From c39319e9467068486a82f50b053f6d0148d876a1 Mon Sep 17 00:00:00 2001 From: tawei-amd Date: Fri, 16 Oct 2026 17:55:27 -0700 Subject: [PATCH 1/6] fix(audio): decode without FFmpeg; Whisper uses AudioReader AudioReader and Whisper's own decoder used FFmpeg (avformat/avcodec + swresample). Replace both with permissively licensed decoders and a resampler that reproduces swresample's default filter, so audio reaching the models stays numerically the same: - audio_decoder.cpp: format sniffing plus - WAV/RF64/W64/AIFF, FLAC (native and Ogg) and MP3 via dr_libs (public domain / MIT-0), vendored at commit dfe83776, which carries the post-release hardening fixes (e.g. dr_mp3 Xing heap over-read); - Ogg Vorbis via libvorbis/vorbisfile + libogg (BSD-3-Clause); - Ogg Opus and WebM/Matroska Opus via libopus (BSD-3-Clause) with a small Ogg demuxer and a bounds-checked EBML/Matroska parser (unknown- size elements, all lacing modes, DiscardPadding). Ogg Opus length follows RFC 7845 (start offset + pre-skip + final granule). - AAC (.m4a/.mp4/ADTS) is rejected with a clear message listing the supported formats. - audio_resampler.cpp: Kaiser-windowed sinc with swresample's defaults (beta 9, cutoff 0.97, filter_size 32, even tap count, exact rational phases up to 1024 else interpolated, mirrored edges as in invert_initial_buffer/resample_flush). - AudioReader keeps its API and downmix modes; Whisper's two _load_audio overloads (~500 lines of FFmpeg code) now call AudioReader at 16 kHz with MEAN downmix, matching the previous swr result (s16 output made swr average stereo), but without the 16-bit quantization. Build: src/cmake/FlmAudioCodecs.cmake builds libopus 1.6.1, libogg 1.3.6 and libvorbis 1.3.7 statically via ExternalProject (pinned URLs and SHA-256, short dirs for Windows MAX_PATH) and compiles audio_decoder.cpp with -fwrapv. The src/test harnesses link it (CMake) or the system libraries (Makefiles) and no longer reference FFmpeg at all; phi4_core's FFmpeg stub headers are gone since no production header includes FFmpeg. THIRD_PARTY_NOTICES.txt lists the new libraries. FFmpeg itself is now unused by the code; dropping it from the build and packages follows separately. Verified against FFmpeg 8 (PyAV) and libsndfile on 22 files: - lossless (WAV 8/16/24-bit/float, 6ch, RF64, W64, AIFF, FLAC): bit-exact; - Ogg Vorbis, Ogg Opus (two muxers), WebM Opus mono/stereo: 133-139 dB SNR and identical lengths; MP3 ~90 dB (different decoder), real test_audio.mp3 max abs diff 4.6e-5; - resampler vs swresample on 11 rate pairs incl. edges: 132-142 dB SNR; - AAC (M4A, ADTS) rejected. 6,000 mutated/truncated inputs under ASan+UBSan: no crash, leak or UB (an earlier stb_vorbis-based version leaked and hit UB, hence libvorbis). Co-Authored-By: Claude --- THIRD_PARTY_NOTICES.txt | 194 + src/CMakeLists.txt | 4 +- src/cmake/FlmAudioCodecs.cmake | 95 + src/common/audio/audio_decoder.cpp | 767 + src/common/audio/audio_reader.cpp | 334 +- src/common/audio/audio_resampler.cpp | 145 + src/common/whisper/modeling_whisper_audio.cpp | 529 +- src/include/audio/audio_decoder.hpp | 28 + src/include/audio/audio_reader.hpp | 4 +- src/include/audio/audio_resampler.hpp | 19 + src/include/dr_libs/LICENSE | 47 + src/include/dr_libs/dr_flac.h | 12760 ++++++++++++++++ src/include/dr_libs/dr_mp3.h | 5430 +++++++ src/include/dr_libs/dr_wav.h | 9209 +++++++++++ src/include/whisper/modeling_whisper.hpp | 8 - src/test/CMakeLists.txt | 13 + src/test/gemma4_12b/CMakeLists.txt | 3 - src/test/gemma4_12b/Makefile | 4 +- src/test/gemma4_core/test_gemma4_frontend.cpp | 4 +- src/test/gemma4e_flash/CMakeLists.txt | 4 - src/test/gemma4e_flash/Makefile | 4 +- src/test/gemma4e_npu/CMakeLists.txt | 4 - src/test/gemma4e_npu/Makefile | 4 +- src/test/minicpm_v_4_7_npu/Makefile | 4 +- src/test/phi4_core/CMakeLists.txt | 16 - src/test/qwen3_5_omni_npu/CMakeLists.txt | 3 - src/test/qwen3_5_omni_npu/Makefile | 4 +- src/test/qwen3_npu/CMakeLists.txt | 4 - src/test/whisper_npu/CMakeLists.txt | 5 - src/test/whisper_npu/Makefile | 5 +- 30 files changed, 28797 insertions(+), 857 deletions(-) create mode 100644 src/cmake/FlmAudioCodecs.cmake create mode 100644 src/common/audio/audio_decoder.cpp create mode 100644 src/common/audio/audio_resampler.cpp create mode 100644 src/include/audio/audio_decoder.hpp create mode 100644 src/include/audio/audio_resampler.hpp create mode 100644 src/include/dr_libs/LICENSE create mode 100644 src/include/dr_libs/dr_flac.h create mode 100644 src/include/dr_libs/dr_mp3.h create mode 100644 src/include/dr_libs/dr_wav.h diff --git a/THIRD_PARTY_NOTICES.txt b/THIRD_PARTY_NOTICES.txt index c37f58cc5..f67913cc3 100644 --- a/THIRD_PARTY_NOTICES.txt +++ b/THIRD_PARTY_NOTICES.txt @@ -38,6 +38,10 @@ Components 23. Microsoft Visual C++ Runtime 24. libjpeg-turbo 25. LodePNG + 26. Opus (libopus) + 27. libogg + 28. libvorbis + 29. dr_libs (dr_wav, dr_flac, dr_mp3) ============================================================================== 1. FFmpeg @@ -884,6 +888,196 @@ freely, subject to the following restrictions: 3. This notice may not be removed or altered from any source distribution. +============================================================================== +26. Opus (libopus) +============================================================================== + +License: BSD 3-Clause License +Homepage: https://opus-codec.org +Used in: Compiled into flm (Opus audio decoding), libopus 1.6.1 built as a static + library. + +Copyright 2001-2023 Xiph.Org, Skype Limited, Octasic, + Jean-Marc Valin, Timothy B. Terriberry, + CSIRO, Gregory Maxwell, Mark Borgerding, + Erik de Castro Lopo, Mozilla, Amazon + +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: + +- Redistributions of source code must retain the above copyright +notice, this list of conditions and the following disclaimer. + +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. + +- Neither the name of Internet Society, IETF or IETF Trust, nor the +names of specific contributors, may be used to endorse or promote +products derived from this software without specific prior written +permission. + +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS +``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT +LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR +A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE COPYRIGHT OWNER +OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, SPECIAL, +EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO, +PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR +PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF +LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING +NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS +SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + +Opus is subject to the royalty-free patent licenses which are +specified at: + +Xiph.Org Foundation: +https://datatracker.ietf.org/ipr/1524/ + +Microsoft Corporation: +https://datatracker.ietf.org/ipr/1914/ + +Broadcom Corporation: +https://datatracker.ietf.org/ipr/1526/ + +============================================================================== +27. libogg +============================================================================== + +License: BSD 3-Clause License +Homepage: https://xiph.org/ogg/ +Used in: Compiled into flm (Ogg container for Vorbis decoding), libogg 1.3.6 built + as a static library. + +Copyright (c) 2002, Xiph.org Foundation + +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: + +- Redistributions of source code must retain the above copyright +notice, this list of conditions and the following disclaimer. + +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. + +- Neither the name of the Xiph.org Foundation nor the names of its +contributors may be used to endorse or promote products derived from +this software without specific prior written permission. + +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS +``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT +LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR +A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION +OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, +SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT +LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, +DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY +THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT +(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE +OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + +============================================================================== +28. libvorbis +============================================================================== + +License: BSD 3-Clause License +Homepage: https://xiph.org/vorbis/ +Used in: Compiled into flm (Ogg Vorbis audio decoding), libvorbis/libvorbisfile + 1.3.7 built as static libraries. + +Copyright (c) 2002-2020 Xiph.org Foundation + +Redistribution and use in source and binary forms, with or without +modification, are permitted provided that the following conditions +are met: + +- Redistributions of source code must retain the above copyright +notice, this list of conditions and the following disclaimer. + +- Redistributions in binary form must reproduce the above copyright +notice, this list of conditions and the following disclaimer in the +documentation and/or other materials provided with the distribution. + +- Neither the name of the Xiph.org Foundation nor the names of its +contributors may be used to endorse or promote products derived from +this software without specific prior written permission. + +THIS SOFTWARE IS PROVIDED BY THE COPYRIGHT HOLDERS AND CONTRIBUTORS +``AS IS'' AND ANY EXPRESS OR IMPLIED WARRANTIES, INCLUDING, BUT NOT +LIMITED TO, THE IMPLIED WARRANTIES OF MERCHANTABILITY AND FITNESS FOR +A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO EVENT SHALL THE FOUNDATION +OR CONTRIBUTORS BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL, +SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT +LIMITED TO, PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, +DATA, OR PROFITS; OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY +THEORY OF LIABILITY, WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT +(INCLUDING NEGLIGENCE OR OTHERWISE) ARISING IN ANY WAY OUT OF THE USE +OF THIS SOFTWARE, EVEN IF ADVISED OF THE POSSIBILITY OF SUCH DAMAGE. + +============================================================================== +29. dr_libs (dr_wav, dr_flac, dr_mp3) +============================================================================== + +License: Public Domain (Unlicense) or MIT No Attribution (MIT-0), at the + licensee's choice +Homepage: https://github.com/mackron/dr_libs +Used in: Compiled into flm (WAV/AIFF, FLAC and MP3 audio decoding), vendored in + src/include/dr_libs from commit dfe83776 (2026-08-31), which includes + hardening fixes made after the dr_wav 0.14.5, dr_flac 0.13.3 and + dr_mp3 0.7.3 releases. + +This software is available as a choice of the following licenses. Choose +whichever you prefer. + +=============================================================================== +ALTERNATIVE 1 - Public Domain (www.unlicense.org) +=============================================================================== +This is free and unencumbered software released into the public domain. + +Anyone is free to copy, modify, publish, use, compile, sell, or distribute this +software, either in source code form or as a compiled binary, for any purpose, +commercial or non-commercial, and by any means. + +In jurisdictions that recognize copyright laws, the author or authors of this +software dedicate any and all copyright interest in the software to the public +domain. We make this dedication for the benefit of the public at large and to +the detriment of our heirs and successors. We intend this dedication to be an +overt act of relinquishment in perpetuity of all present and future rights to +this software under copyright law. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN +ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION +WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + +For more information, please refer to + +=============================================================================== +ALTERNATIVE 2 - MIT No Attribution +=============================================================================== +Copyright 2020 David Reid + +Permission is hereby granted, free of charge, to any person obtaining a copy of +this software and associated documentation files (the "Software"), to deal in +the Software without restriction, including without limitation the rights to +use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies +of the Software, and to permit persons to whom the Software is furnished to do +so. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +SOFTWARE. + ============================================================================== Full license texts ============================================================================== diff --git a/src/CMakeLists.txt b/src/CMakeLists.txt index bb75b7f11..1ea4e97a8 100644 --- a/src/CMakeLists.txt +++ b/src/CMakeLists.txt @@ -294,6 +294,8 @@ add_subdirectory(${CMAKE_SOURCE_DIR}/../third_party/tokenizers-cpp # JPEG/PNG decoding for ImageReader: static libjpeg-turbo + vendored lodepng. include(${CMAKE_SOURCE_DIR}/cmake/FlmImageCodecs.cmake) +# Audio decoding for AudioReader: static libopus/libogg/libvorbis + vendored dr_libs. +include(${CMAKE_SOURCE_DIR}/cmake/FlmAudioCodecs.cmake) # ——————————————————————————————————————————————— # Gather your sources @@ -749,7 +751,7 @@ endif() target_link_libraries(flm PRIVATE tokenizers_cpp ) -target_link_libraries(flm PRIVATE flm_image_codecs) +target_link_libraries(flm PRIVATE flm_image_codecs flm_audio_codecs) if(WIN32) target_link_libraries(flm PRIVATE diff --git a/src/cmake/FlmAudioCodecs.cmake b/src/cmake/FlmAudioCodecs.cmake new file mode 100644 index 000000000..e9650d078 --- /dev/null +++ b/src/cmake/FlmAudioCodecs.cmake @@ -0,0 +1,95 @@ +# Audio codecs for AudioReader (src/common/audio/audio_decoder.cpp). +# +# - libopus 1.6.1, libogg 1.3.6 and libvorbis 1.3.7 (all BSD-3-Clause), built +# as static libraries through ExternalProject from pinned release tarballs. +# libvorbis finds libogg through the shared install prefix. +# - dr_wav/dr_flac/dr_mp3 (public domain or MIT-0), vendored header-only in +# include/dr_libs and compiled into audio_decoder.cpp. +# +# Defines the INTERFACE target flm_audio_codecs. Link it from anything that +# compiles audio_decoder.cpp. +include_guard(GLOBAL) +include(ExternalProject) + +set(_flm_audio_prefix ${CMAKE_BINARY_DIR}/audio-codecs) +if(MSVC) + set(_flm_lib_prefix "") + set(_flm_lib_suffix ".lib") +else() + set(_flm_lib_prefix "lib") + set(_flm_lib_suffix ".a") +endif() +foreach(_lib opus ogg vorbis vorbisfile) + set(_flm_${_lib}_lib ${_flm_audio_prefix}/lib/${_flm_lib_prefix}${_lib}${_flm_lib_suffix}) +endforeach() + +set(_flm_audio_args + -DCMAKE_INSTALL_PREFIX= + -DCMAKE_INSTALL_LIBDIR=lib + -DCMAKE_PREFIX_PATH=${_flm_audio_prefix} + -DCMAKE_BUILD_TYPE=Release + -DCMAKE_POSITION_INDEPENDENT_CODE=ON + -DBUILD_SHARED_LIBS=OFF + -DBUILD_TESTING=OFF) +if(CMAKE_C_COMPILER) + list(APPEND _flm_audio_args -DCMAKE_C_COMPILER=${CMAKE_C_COMPILER}) +endif() + +# Short working directories keep MSBuild under Windows' 260-char MAX_PATH on +# CI runners with deep work directories (see FlmImageCodecs.cmake). +function(_flm_audio_project name url sha256 byproduct) + set(_work ${CMAKE_BINARY_DIR}/ac/${name}) + ExternalProject_Add(flm_${name} + URL ${url} + URL_HASH SHA256=${sha256} + DOWNLOAD_EXTRACT_TIMESTAMP TRUE + EXCLUDE_FROM_ALL TRUE + PREFIX ${_work} + DOWNLOAD_DIR ${_work}/dl + SOURCE_DIR ${_work}/s + BINARY_DIR ${_work}/b + STAMP_DIR ${_work}/st + TMP_DIR ${_work}/t + INSTALL_DIR ${_flm_audio_prefix} + CMAKE_ARGS ${_flm_audio_args} ${ARGN} + BUILD_BYPRODUCTS ${byproduct}) +endfunction() + +_flm_audio_project(opus + https://downloads.xiph.org/releases/opus/opus-1.6.1.tar.gz + 6ffcb593207be92584df15b32466ed64bbec99109f007c82205f0194572411a1 + ${_flm_opus_lib} + -DOPUS_BUILD_PROGRAMS=OFF + -DOPUS_INSTALL_PKG_CONFIG_MODULE=OFF + -DOPUS_INSTALL_CMAKE_CONFIG_MODULE=OFF) +_flm_audio_project(ogg + https://downloads.xiph.org/releases/ogg/libogg-1.3.6.tar.gz + 83e6704730683d004d20e21b8f7f55dcb3383cdf84c0daedf30bde175f774638 + ${_flm_ogg_lib} + -DINSTALL_DOCS=OFF) +_flm_audio_project(vorbis + https://downloads.xiph.org/releases/vorbis/libvorbis-1.3.7.tar.gz + 0e982409a9c3fc82ee06e08205b1355e5c6aa4c36bca58146ef399621b0ce5ab + "${_flm_vorbis_lib};${_flm_vorbisfile_lib}") +ExternalProject_Add_StepDependencies(flm_vorbis configure flm_ogg) + +# The include directory must exist at configure time for the imported targets. +file(MAKE_DIRECTORY ${_flm_audio_prefix}/include) +add_library(flm_audio_codecs INTERFACE) +foreach(_lib vorbisfile vorbis ogg opus) + add_library(flm_${_lib}_imported STATIC IMPORTED GLOBAL) + set_target_properties(flm_${_lib}_imported PROPERTIES + IMPORTED_LOCATION ${_flm_${_lib}_lib} + INTERFACE_INCLUDE_DIRECTORIES ${_flm_audio_prefix}/include) + target_link_libraries(flm_audio_codecs INTERFACE flm_${_lib}_imported) +endforeach() +add_dependencies(flm_vorbisfile_imported flm_vorbis) +add_dependencies(flm_vorbis_imported flm_vorbis) +add_dependencies(flm_ogg_imported flm_ogg) +add_dependencies(flm_opus_imported flm_opus) + +# The decoders do wrapping integer arithmetic on corrupt input (e.g. FLAC LPC +# prediction); make signed overflow defined there instead of undefined. +get_filename_component(_flm_audio_decoder_src ${CMAKE_CURRENT_LIST_DIR}/../common/audio/audio_decoder.cpp ABSOLUTE) +set_source_files_properties(${_flm_audio_decoder_src} PROPERTIES + COMPILE_OPTIONS "$<$>:-fwrapv>") diff --git a/src/common/audio/audio_decoder.cpp b/src/common/audio/audio_decoder.cpp new file mode 100644 index 000000000..453427066 --- /dev/null +++ b/src/common/audio/audio_decoder.cpp @@ -0,0 +1,767 @@ +/// \file audio_decoder.cpp +/// \brief Container/codec decoding for AudioReader, built on permissively +/// licensed decoders: dr_wav/dr_flac/dr_mp3 (public domain or MIT-0), +/// libvorbis/libogg and libopus (BSD-3-Clause). +/// \author FastFlowLM Team + +#include "audio/audio_decoder.hpp" + +#include +#include +#include +#include + +#define DR_WAV_IMPLEMENTATION +#define DR_WAV_NO_STDIO +#include "dr_libs/dr_wav.h" +#define DR_FLAC_IMPLEMENTATION +#define DR_FLAC_NO_STDIO +#include "dr_libs/dr_flac.h" +#define DR_MP3_IMPLEMENTATION +#define DR_MP3_NO_STDIO +#include "dr_libs/dr_mp3.h" + +#include +#include + +namespace audio { +namespace { + +constexpr size_t kChunkFrames = 65536; +constexpr int kMaxChannels = 64; + +const char* const kUnsupportedHint = + "supported formats are WAV, FLAC, MP3, Ogg (Vorbis/Opus/FLAC) and WebM/Matroska (Opus)"; + +uint16_t le16(const uint8_t* p) { return static_cast(p[0] | (p[1] << 8)); } +uint32_t le32(const uint8_t* p) { + return static_cast(p[0]) | (static_cast(p[1]) << 8) | + (static_cast(p[2]) << 16) | (static_cast(p[3]) << 24); +} +uint64_t le64(const uint8_t* p) { return le32(p) | (static_cast(le32(p + 4)) << 32); } + +bool has_bytes(const uint8_t* data, size_t size, size_t offset, const char* magic, size_t len) { + return offset + len <= size && std::memcmp(data + offset, magic, len) == 0; +} + +bool valid_layout(int sample_rate, int channels) { + return sample_rate > 0 && channels > 0 && channels <= kMaxChannels; +} + +// ——————————————————————————————————————————————— +// Format detection +// ——————————————————————————————————————————————— +enum class Format { Wav, Flac, Mp3, Ogg, Matroska, Mp4, Adts, Unknown }; + +// Size of a leading ID3v2 tag (0 if none). +size_t id3v2_size(const uint8_t* data, size_t size) { + if (size < 10 || !has_bytes(data, size, 0, "ID3", 3)) { + return 0; + } + const size_t body = (static_cast(data[6] & 0x7F) << 21) | (static_cast(data[7] & 0x7F) << 14) | + (static_cast(data[8] & 0x7F) << 7) | static_cast(data[9] & 0x7F); + const size_t footer = (data[5] & 0x10) ? 10 : 0; + return std::min(size, 10 + body + footer); +} + +Format sniff(const uint8_t* data, size_t size, size_t& offset) { + offset = 0; + static const uint8_t kW64Riff[16] = {'r', 'i', 'f', 'f', 0x2E, 0x91, 0xCF, 0x11, + 0xA5, 0xD6, 0x28, 0xDB, 0x04, 0xC1, 0x00, 0x00}; + if ((has_bytes(data, size, 0, "RIFF", 4) || has_bytes(data, size, 0, "RF64", 4) || + has_bytes(data, size, 0, "BW64", 4)) && has_bytes(data, size, 8, "WAVE", 4)) { + return Format::Wav; + } + if (has_bytes(data, size, 0, "FORM", 4) && + (has_bytes(data, size, 8, "AIFF", 4) || has_bytes(data, size, 8, "AIFC", 4))) { + return Format::Wav; + } + if (size >= 16 && std::memcmp(data, kW64Riff, 16) == 0) { + return Format::Wav; + } + if (has_bytes(data, size, 0, "OggS", 4)) { + return Format::Ogg; + } + if (has_bytes(data, size, 0, "\x1A\x45\xDF\xA3", 4)) { + return Format::Matroska; + } + if (has_bytes(data, size, 4, "ftyp", 4)) { + return Format::Mp4; + } + offset = id3v2_size(data, size); + if (has_bytes(data, size, offset, "fLaC", 4)) { + return Format::Flac; + } + if (offset + 2 <= size && data[offset] == 0xFF && (data[offset + 1] & 0xE0) == 0xE0) { + // MPEG audio frame sync; layer bits 00 mean ADTS (AAC). + return ((data[offset + 1] >> 1) & 0x03) == 0 ? Format::Adts : Format::Mp3; + } + if (offset > 0) { + offset = 0; + return Format::Mp3; // ID3-tagged; let dr_mp3 find the first frame. + } + return Format::Unknown; +} + +// ——————————————————————————————————————————————— +// dr_libs / libvorbis +// ——————————————————————————————————————————————— +bool decode_wav(const uint8_t* data, size_t size, DecodedAudio& out, std::string& error) { + drwav wav; + if (!drwav_init_memory(&wav, data, size, nullptr)) { + error = "invalid or unsupported WAV/AIFF file"; + return false; + } + out.sample_rate = static_cast(wav.sampleRate); + out.channels = static_cast(wav.channels); + if (!valid_layout(out.sample_rate, out.channels)) { + drwav_uninit(&wav); + error = "WAV file has an invalid sample rate or channel count"; + return false; + } + // dr_wav maps 8-bit PCM as u8/255*2-1; FFmpeg (and libsndfile) use + // (u8-128)/128, so convert those samples here to keep the previous values. + const bool u8_pcm = wav.translatedFormatTag == DR_WAVE_FORMAT_PCM && wav.bitsPerSample == 8; + std::vector chunk(kChunkFrames * out.channels); + std::vector raw(u8_pcm ? kChunkFrames * out.channels : 0); + for (;;) { + const drwav_uint64 n = u8_pcm ? drwav_read_pcm_frames(&wav, kChunkFrames, raw.data()) + : drwav_read_pcm_frames_f32(&wav, kChunkFrames, chunk.data()); + if (n == 0) { + break; + } + if (u8_pcm) { + for (size_t i = 0; i < n * out.channels; ++i) { + chunk[i] = (static_cast(raw[i]) - 128.0f) / 128.0f; + } + } + out.samples.insert(out.samples.end(), chunk.begin(), chunk.begin() + n * out.channels); + } + drwav_uninit(&wav); + return true; +} + +bool decode_flac(const uint8_t* data, size_t size, DecodedAudio& out, std::string& error) { + drflac* flac = drflac_open_memory(data, size, nullptr); + if (!flac) { + error = "invalid FLAC file"; + return false; + } + out.sample_rate = static_cast(flac->sampleRate); + out.channels = static_cast(flac->channels); + if (!valid_layout(out.sample_rate, out.channels)) { + drflac_close(flac); + error = "FLAC file has an invalid sample rate or channel count"; + return false; + } + std::vector chunk(kChunkFrames * out.channels); + for (;;) { + const drflac_uint64 n = drflac_read_pcm_frames_f32(flac, kChunkFrames, chunk.data()); + if (n == 0) { + break; + } + out.samples.insert(out.samples.end(), chunk.begin(), chunk.begin() + n * out.channels); + } + drflac_close(flac); + return true; +} + +bool decode_mp3(const uint8_t* data, size_t size, DecodedAudio& out, std::string& error) { + drmp3 mp3; + if (!drmp3_init_memory(&mp3, data, size, nullptr)) { + error = "invalid MP3 file"; + return false; + } + out.sample_rate = static_cast(mp3.sampleRate); + out.channels = static_cast(mp3.channels); + if (!valid_layout(out.sample_rate, out.channels)) { + drmp3_uninit(&mp3); + error = "MP3 file has an invalid sample rate or channel count"; + return false; + } + std::vector chunk(kChunkFrames * out.channels); + for (;;) { + const drmp3_uint64 n = drmp3_read_pcm_frames_f32(&mp3, kChunkFrames, chunk.data()); + if (n == 0) { + break; + } + out.samples.insert(out.samples.end(), chunk.begin(), chunk.begin() + n * out.channels); + } + drmp3_uninit(&mp3); + return true; +} + +// vorbisfile reads through callbacks; serve them from the in-memory file. +struct MemoryFile { + const uint8_t* data; + size_t size; + size_t pos; +}; + +size_t memory_read(void* ptr, size_t size, size_t nmemb, void* source) { + auto* f = static_cast(source); + if (size == 0) { + return 0; + } + const size_t n = std::min(nmemb, (f->size - f->pos) / size); + std::memcpy(ptr, f->data + f->pos, n * size); + f->pos += n * size; + return n; +} + +int memory_seek(void* source, ogg_int64_t offset, int whence) { + auto* f = static_cast(source); + int64_t base = 0; + if (whence == SEEK_CUR) { + base = static_cast(f->pos); + } else if (whence == SEEK_END) { + base = static_cast(f->size); + } else if (whence != SEEK_SET) { + return -1; + } + const int64_t target = base + offset; + if (target < 0 || target > static_cast(f->size)) { + return -1; + } + f->pos = static_cast(target); + return 0; +} + +long memory_tell(void* source) { return static_cast(static_cast(source)->pos); } + +bool decode_vorbis(const uint8_t* data, size_t size, DecodedAudio& out, std::string& error) { + MemoryFile file{data, size, 0}; + const ov_callbacks callbacks{memory_read, memory_seek, nullptr, memory_tell}; + OggVorbis_File vf; + if (ov_open_callbacks(&file, &vf, nullptr, 0, callbacks) < 0) { + error = "invalid Ogg Vorbis file"; + return false; + } + const vorbis_info* info = ov_info(&vf, -1); + out.sample_rate = info ? static_cast(info->rate) : 0; + out.channels = info ? info->channels : 0; + if (!valid_layout(out.sample_rate, out.channels)) { + ov_clear(&vf); + error = "Ogg Vorbis file has an invalid sample rate or channel count"; + return false; + } + int first_section = -1; + for (;;) { + float** pcm = nullptr; + int section = 0; + const long n = ov_read_float(&vf, &pcm, 4096, §ion); + if (n == OV_HOLE) { + continue; // damaged data; keep going, as FFmpeg does + } + if (n <= 0) { + break; + } + if (first_section < 0) { + first_section = section; + } else if (section != first_section) { + break; // chained stream: decode the first link only + } + const size_t base = out.samples.size(); + out.samples.resize(base + static_cast(n) * out.channels); + for (long i = 0; i < n; ++i) { + for (int c = 0; c < out.channels; ++c) { + out.samples[base + static_cast(i) * out.channels + c] = pcm[c][i]; + } + } + } + ov_clear(&vf); + return true; +} + +// ——————————————————————————————————————————————— +// Opus (RFC 7845 header; packets from Ogg or Matroska) +// ——————————————————————————————————————————————— +struct OpusPacket { + std::vector data; + int64_t discard_end = 0; // samples (48 kHz) to drop from the end of this packet's output +}; + +bool decode_opus(const std::vector& head, const std::vector& packets, + int64_t total_samples, DecodedAudio& out, std::string& error) { + if (head.size() < 19 || std::memcmp(head.data(), "OpusHead", 8) != 0 || (head[8] >> 4) != 0) { + error = "invalid Opus header"; + return false; + } + const int channels = head[9]; + const int pre_skip = le16(&head[10]); + const int16_t gain_q8 = static_cast(le16(&head[16])); + const int family = head[18]; + + int streams = 1; + int coupled = channels > 1 ? 1 : 0; + unsigned char mapping[255] = {0, 1}; + if (family == 0) { + if (channels < 1 || channels > 2) { + error = "invalid Opus channel count"; + return false; + } + } else if (family == 1 || family == 255) { + if (channels < 1 || head.size() < 21u + channels) { + error = "invalid Opus channel mapping"; + return false; + } + streams = head[19]; + coupled = head[20]; + std::memcpy(mapping, &head[21], channels); + } else { + error = "unsupported Opus channel mapping family " + std::to_string(family); + return false; + } + + int err = 0; + std::unique_ptr dec( + opus_multistream_decoder_create(48000, channels, streams, coupled, mapping, &err), + opus_multistream_decoder_destroy); + if (!dec || err != OPUS_OK) { + error = std::string("could not create Opus decoder: ") + opus_strerror(err); + return false; + } + + constexpr int kMaxFrame = 5760; // 120 ms at 48 kHz + std::vector pcm(static_cast(kMaxFrame) * channels); + for (const OpusPacket& p : packets) { + if (p.data.empty() || p.data.size() > static_cast(INT32_MAX)) { + continue; + } + int n = opus_multistream_decode_float(dec.get(), p.data.data(), static_cast(p.data.size()), + pcm.data(), kMaxFrame, 0); + if (n < 0) { + continue; // skip a corrupt packet, as FFmpeg does + } + n -= static_cast(std::min(std::max(p.discard_end, 0), n)); + out.samples.insert(out.samples.end(), pcm.begin(), pcm.begin() + static_cast(n) * channels); + } + + const size_t skip = std::min(out.samples.size(), static_cast(pre_skip) * channels); + out.samples.erase(out.samples.begin(), out.samples.begin() + skip); + if (total_samples >= 0 && static_cast(total_samples) * channels < out.samples.size()) { + out.samples.resize(static_cast(total_samples) * channels); + } + if (gain_q8 != 0) { + const float gain = static_cast(std::pow(10.0, gain_q8 / (20.0 * 256.0))); + for (float& s : out.samples) { + s *= gain; + } + } + out.sample_rate = 48000; + out.channels = channels; + return true; +} + +// ——————————————————————————————————————————————— +// Ogg (RFC 3533): packets of the first logical stream +// ——————————————————————————————————————————————— +struct OggStream { + std::vector> packets; + std::vector packet_page; // page on which each packet completes + std::vector page_granule; // granule position of each page of the stream + int64_t last_granule = -1; +}; + +bool read_ogg_stream(const uint8_t* data, size_t size, OggStream& stream) { + int64_t serial = -1; + std::vector partial; + size_t pos = 0; + while (pos + 27 <= size) { + if (std::memcmp(data + pos, "OggS", 4) != 0 || data[pos + 4] != 0) { + break; + } + const uint8_t header_type = data[pos + 5]; + const int64_t granule = static_cast(le64(data + pos + 6)); + const uint32_t page_serial = le32(data + pos + 14); + const size_t segments = data[pos + 26]; + const size_t body = pos + 27 + segments; + if (body > size) { + break; + } + size_t body_len = 0; + for (size_t i = 0; i < segments; ++i) { + body_len += data[pos + 27 + i]; + } + if (body + body_len > size) { + break; + } + if (serial < 0 && (header_type & 0x02)) { + serial = page_serial; + } + if (static_cast(page_serial) == serial) { + const size_t page = stream.page_granule.size(); + stream.page_granule.push_back(granule); + size_t off = body; + for (size_t i = 0; i < segments; ++i) { + const size_t len = data[pos + 27 + i]; + partial.insert(partial.end(), data + off, data + off + len); + off += len; + if (len < 255) { + stream.packets.push_back(std::move(partial)); + stream.packet_page.push_back(page); + partial.clear(); + } + } + if (granule != -1) { + stream.last_granule = granule; + } + } + pos = body + body_len; + } + return !stream.packets.empty(); +} + +bool decode_ogg(const uint8_t* data, size_t size, DecodedAudio& out, std::string& error) { + OggStream stream; + if (!read_ogg_stream(data, size, stream)) { + error = "invalid Ogg file"; + return false; + } + const std::vector& first = stream.packets.front(); + if (first.size() >= 7 && std::memcmp(first.data(), "\x01vorbis", 7) == 0) { + return decode_vorbis(data, size, out, error); + } + if (first.size() >= 5 && std::memcmp(first.data(), "\x7F""FLAC", 5) == 0) { + return decode_flac(data, size, out, error); + } + if (first.size() >= 8 && std::memcmp(first.data(), "OpusHead", 8) == 0) { + // Packet 0 is OpusHead, packet 1 OpusTags; audio follows. Playback + // length (RFC 7845, 4.3-4.5): last granule minus the stream's start + // position minus pre-skip, where the start position is the first audio + // page's granule minus the samples of the packets completing on it. + const int pre_skip = first.size() >= 12 ? le16(&first[10]) : 0; + int64_t start = 0; + if (stream.packets.size() > 2) { + const size_t first_page = stream.packet_page[2]; + int64_t first_samples = 0; + for (size_t i = 2; i < stream.packets.size() && stream.packet_page[i] == first_page; ++i) { + const std::vector& pkt = stream.packets[i]; + const int n = pkt.empty() ? 0 : opus_packet_get_nb_samples(pkt.data(), static_cast(pkt.size()), 48000); + first_samples += std::max(n, 0); + } + const int64_t granule = stream.page_granule[first_page]; + start = granule >= 0 ? std::max(granule - first_samples, 0) : 0; + } + const int64_t total = stream.last_granule >= 0 ? stream.last_granule - start - pre_skip : -1; + + std::vector packets; + for (size_t i = 2; i < stream.packets.size(); ++i) { + packets.push_back({std::move(stream.packets[i]), 0}); + } + return decode_opus(first, packets, total >= 0 ? total : -1, out, error); + } + error = std::string("unsupported codec in Ogg file; ") + kUnsupportedHint; + return false; +} + +// ——————————————————————————————————————————————— +// Matroska / WebM: Opus audio track only +// ——————————————————————————————————————————————— +namespace ebml { +constexpr uint32_t kSegment = 0x18538067, kTracks = 0x1654AE6B, kTrackEntry = 0xAE, kTrackNumber = 0xD7, + kTrackType = 0x83, kCodecId = 0x86, kCodecPrivate = 0x63A2, kCluster = 0x1F43B675, + kSimpleBlock = 0xA3, kBlockGroup = 0xA0, kBlock = 0xA1, kDiscardPadding = 0x75A2; + +// Element ID: 1-4 bytes, length marker kept. +bool read_id(const uint8_t* d, size_t n, size_t& pos, uint32_t& id) { + if (pos >= n || d[pos] == 0) { + return false; + } + int len = 1; + while (len <= 4 && !(d[pos] & (0x80 >> (len - 1)))) { + ++len; + } + if (len > 4 || pos + len > n) { + return false; + } + id = 0; + for (int i = 0; i < len; ++i) { + id = (id << 8) | d[pos + i]; + } + pos += len; + return true; +} + +// Variable-size integer: 1-8 bytes, length marker removed. `unknown` is set +// when all value bits are 1 (unknown element size). +bool read_vint(const uint8_t* d, size_t n, size_t& pos, uint64_t& value, bool* unknown = nullptr, int* length = nullptr) { + if (pos >= n || d[pos] == 0) { + return false; + } + int len = 1; + while (!(d[pos] & (0x80 >> (len - 1)))) { + ++len; + } + if (pos + len > n) { + return false; + } + value = d[pos] & (0xFF >> len); + bool all_ones = value == (0xFFu >> len); + for (int i = 1; i < len; ++i) { + value = (value << 8) | d[pos + i]; + all_ones = all_ones && d[pos + i] == 0xFF; + } + if (unknown) { + *unknown = all_ones; + } + if (length) { + *length = len; + } + pos += len; + return true; +} + +uint64_t read_uint(const uint8_t* d, size_t len) { + uint64_t v = 0; + for (size_t i = 0; i < len && i < 8; ++i) { + v = (v << 8) | d[i]; + } + return v; +} + +int64_t read_sint(const uint8_t* d, size_t len) { + if (len == 0 || len > 8) { + return 0; + } + uint64_t v = read_uint(d, len); + if (d[0] & 0x80) { + v |= len == 8 ? 0 : (~uint64_t{0} << (8 * len)); + } + return static_cast(v); +} + +// Split a (Simple)Block payload into frames for `track` (RFC 9559, 10.2). +bool block_frames(const uint8_t* d, size_t n, uint64_t track, std::vector>& frames) { + size_t pos = 0; + uint64_t number = 0; + if (!read_vint(d, n, pos, number) || pos + 3 > n) { + return false; + } + if (number != track) { + return true; + } + const uint8_t flags = d[pos + 2]; + pos += 3; + const int lacing = (flags >> 1) & 0x03; + if (lacing == 0) { + frames.emplace_back(d + pos, d + n); + return true; + } + if (pos >= n) { + return false; + } + const size_t count = static_cast(d[pos++]) + 1; + std::vector sizes; + if (lacing == 1) { // Xiph + for (size_t i = 0; i + 1 < count; ++i) { + uint64_t s = 0; + uint8_t b = 0; + do { + if (pos >= n) { + return false; + } + b = d[pos++]; + s += b; + } while (b == 255); + sizes.push_back(s); + } + } else if (lacing == 3) { // EBML + uint64_t s = 0; + if (count > 1) { + if (!read_vint(d, n, pos, s)) { + return false; + } + sizes.push_back(s); + } + for (size_t i = 1; i + 1 < count; ++i) { + uint64_t raw = 0; + int len = 0; + if (!read_vint(d, n, pos, raw, nullptr, &len)) { + return false; + } + const int64_t delta = static_cast(raw) - ((int64_t{1} << (7 * len - 1)) - 1); + const int64_t next = static_cast(s) + delta; + if (next < 0) { + return false; + } + s = static_cast(next); + sizes.push_back(s); + } + } else { // fixed + const size_t each = (n - pos) / count; + if (each * count != n - pos) { + return false; + } + sizes.assign(count - 1, each); + } + for (uint64_t s : sizes) { + if (s > n - pos) { + return false; + } + frames.emplace_back(d + pos, d + pos + s); + pos += static_cast(s); + } + frames.emplace_back(d + pos, d + n); + return true; +} +} // namespace ebml + +bool decode_matroska(const uint8_t* data, size_t size, DecodedAudio& out, std::string& error) { + struct Track { + uint64_t number = 0; + uint64_t type = 0; + std::string codec; + std::vector codec_private; + }; + std::vector tracks; + int audio_index = -1; // index into tracks once the first block selects it + std::vector packets; + size_t group_first_packet = 0; + int64_t pending_discard_ns = 0; + + // Elements are visited in file order. Master elements we need children + // from are entered instead of skipped, which also copes with the + // unknown-size Segment/Cluster elements live recorders (MediaRecorder) + // write. + size_t pos = 0; + while (pos < size) { + uint32_t id = 0; + uint64_t len = 0; + bool unknown = false; + if (!ebml::read_id(data, size, pos, id) || !ebml::read_vint(data, size, pos, len, &unknown)) { + break; + } + if (id == ebml::kSegment || id == ebml::kTracks || id == ebml::kCluster) { + continue; + } + if (id == ebml::kTrackEntry) { + if (audio_index < 0) { + tracks.emplace_back(); + } + continue; + } + if (id == ebml::kBlockGroup) { + group_first_packet = packets.size(); + pending_discard_ns = 0; + continue; + } + if (unknown) { + break; // only master elements may have an unknown size + } + if (len > size - pos) { + len = size - pos; // truncated file: use what is there + } + const uint8_t* body = data + pos; + const size_t blen = static_cast(len); + pos += blen; + + if (!tracks.empty() && audio_index < 0) { + Track& t = tracks.back(); + if (id == ebml::kTrackNumber) { + t.number = ebml::read_uint(body, blen); + } else if (id == ebml::kTrackType) { + t.type = ebml::read_uint(body, blen); + } else if (id == ebml::kCodecId) { + // Codec IDs are short ASCII strings; keep untrusted bytes out of + // error messages. + t.codec.clear(); + for (size_t i = 0; i < blen && t.codec.size() < 32; ++i) { + const char ch = static_cast(body[i]); + if (ch == '\0') { + break; + } + t.codec.push_back(ch >= 0x20 && ch < 0x7F ? ch : '?'); + } + } else if (id == ebml::kCodecPrivate) { + t.codec_private.assign(body, body + blen); + } + } + if (id == ebml::kSimpleBlock || id == ebml::kBlock) { + if (audio_index < 0) { + for (size_t i = 0; i < tracks.size(); ++i) { + if (tracks[i].type == 2) { + audio_index = static_cast(i); + break; + } + } + if (audio_index < 0) { + error = "no audio track in Matroska/WebM file"; + return false; + } + if (tracks[audio_index].codec != "A_OPUS") { + error = "unsupported audio codec " + tracks[audio_index].codec + " in Matroska/WebM file; " + + kUnsupportedHint; + return false; + } + } + std::vector> frames; + if (!ebml::block_frames(body, blen, tracks[audio_index].number, frames)) { + continue; + } + for (auto& f : frames) { + packets.push_back({std::move(f), 0}); + } + if (id == ebml::kBlock && pending_discard_ns > 0 && packets.size() > group_first_packet) { + packets.back().discard_end = pending_discard_ns * 48000 / 1000000000; + pending_discard_ns = 0; + } + } else if (id == ebml::kDiscardPadding) { + const int64_t ns = ebml::read_sint(body, blen); + if (packets.size() > group_first_packet) { + packets.back().discard_end = ns * 48000 / 1000000000; + } else { + pending_discard_ns = ns; + } + } + } + + if (audio_index < 0) { + error = "no audio data in Matroska/WebM file"; + return false; + } + return decode_opus(tracks[audio_index].codec_private, packets, -1, out, error); +} + +} // namespace + +bool decode_audio_file(const uint8_t* data, size_t size, DecodedAudio& out, std::string& error) { + out = DecodedAudio{}; + if (!data || size == 0) { + error = "empty audio data"; + return false; + } + size_t offset = 0; + bool ok = false; + switch (sniff(data, size, offset)) { + case Format::Wav: + ok = decode_wav(data, size, out, error); + break; + case Format::Flac: + ok = decode_flac(data + offset, size - offset, out, error); + break; + case Format::Mp3: + ok = decode_mp3(data, size, out, error); + break; + case Format::Ogg: + ok = decode_ogg(data, size, out, error); + break; + case Format::Matroska: + ok = decode_matroska(data, size, out, error); + break; + case Format::Mp4: + case Format::Adts: + error = std::string("AAC/M4A/MP4 audio is not supported; ") + kUnsupportedHint; + return false; + case Format::Unknown: + error = std::string("unrecognized audio format; ") + kUnsupportedHint; + return false; + } + if (ok && out.samples.empty()) { + error = "audio file contains no samples"; + return false; + } + return ok; +} + +} // namespace audio diff --git a/src/common/audio/audio_reader.cpp b/src/common/audio/audio_reader.cpp index 4de965cfd..024137588 100644 --- a/src/common/audio/audio_reader.cpp +++ b/src/common/audio/audio_reader.cpp @@ -1,78 +1,19 @@ /// \file audio_reader.cpp -/// \brief AudioReader implementation – decode and resample audio via FFmpeg +/// \brief AudioReader implementation – decode (audio_decoder) and resample (audio_resampler) /// \author FastFlowLM Team #include "audio/audio_reader.hpp" +#include "audio/audio_decoder.hpp" +#include "audio/audio_resampler.hpp" #include #include -#include -#include +#include #include -extern "C" { -#include -#include -#include -#include -#include -#include -#include -} - -// --------------------------------------------------------------------------- -// helpers -// --------------------------------------------------------------------------- -namespace { - -static std::once_flag g_ffmpeg_init_flag; - -static void init_ffmpeg_once() { - std::call_once(g_ffmpeg_init_flag, []() { - av_log_set_level(AV_LOG_ERROR); - }); -} - -// Custom AVIO context for reading from memory -struct MemoryIOContext { - const uint8_t* data; - size_t size; - size_t pos; -}; - -static int mem_read_packet(void* opaque, uint8_t* buf, int buf_size) { - auto* ctx = static_cast(opaque); - size_t remaining = ctx->size - ctx->pos; - if (remaining == 0) return AVERROR_EOF; - size_t to_read = static_cast(buf_size); - if (to_read > remaining) to_read = remaining; - std::memcpy(buf, ctx->data + ctx->pos, to_read); - ctx->pos += to_read; - return static_cast(to_read); -} - -static int64_t mem_seek(void* opaque, int64_t offset, int whence) { - auto* ctx = static_cast(opaque); - int64_t new_pos = -1; - switch (whence) { - case SEEK_SET: new_pos = offset; break; - case SEEK_CUR: new_pos = static_cast(ctx->pos) + offset; break; - case SEEK_END: new_pos = static_cast(ctx->size) + offset; break; - case AVSEEK_SIZE: return static_cast(ctx->size); - default: return AVERROR(EINVAL); - } - if (new_pos < 0 || new_pos > static_cast(ctx->size)) return AVERROR(EINVAL); - ctx->pos = static_cast(new_pos); - return new_pos; -} - -} // anonymous namespace - // --------------------------------------------------------------------------- // AudioReader // --------------------------------------------------------------------------- -AudioReader::AudioReader() { - init_ffmpeg_once(); -} +AudioReader::AudioReader() = default; AudioReader::~AudioReader() = default; @@ -82,22 +23,25 @@ bool AudioReader::load_audio(const std::string& filename, int target_sample_rate, MonoDownmixMode downmix) { - init_ffmpeg_once(); - if (!std::filesystem::exists(filename)) { std::cerr << "Error: audio file not found: " << filename << std::endl; return false; } - AVFormatContext* format_ctx = nullptr; - if (avformat_open_input(&format_ctx, filename.c_str(), nullptr, nullptr) < 0) { + std::ifstream file(filename, std::ios::binary | std::ios::ate); + const std::streamsize file_size = file.is_open() ? static_cast(file.tellg()) : -1; + if (file_size <= 0) { std::cerr << "Error: could not open audio file: " << filename << std::endl; return false; } + file.seekg(0, std::ios::beg); + std::vector data(static_cast(file_size)); + if (!file.read(reinterpret_cast(data.data()), file_size)) { + std::cerr << "Error: could not read audio file: " << filename << std::endl; + return false; + } - bool ok = decode_audio(format_ctx, out_audio, target_sample_rate, downmix); - avformat_close_input(&format_ctx); - return ok; + return decode_audio(data.data(), data.size(), out_audio, target_sample_rate, downmix); } // --------------------------------------------------------------------------- @@ -106,245 +50,61 @@ bool AudioReader::load_audio_from_memory(const uint8_t* data, size_t size, int target_sample_rate, MonoDownmixMode downmix) { - init_ffmpeg_once(); - if (!data || size == 0) { std::cerr << "Error: empty audio buffer" << std::endl; return false; } - - constexpr int kAVIOBufSize = 32768; - auto* avio_buf = static_cast(av_malloc(kAVIOBufSize)); - if (!avio_buf) { - std::cerr << "Error: could not allocate AVIO buffer" << std::endl; - return false; - } - - MemoryIOContext mem_ctx{data, size, 0}; - AVIOContext* avio_ctx = avio_alloc_context(avio_buf, kAVIOBufSize, 0, - &mem_ctx, mem_read_packet, nullptr, mem_seek); - if (!avio_ctx) { - av_free(avio_buf); - std::cerr << "Error: could not create AVIO context" << std::endl; - return false; - } - - AVFormatContext* format_ctx = avformat_alloc_context(); - if (!format_ctx) { - avio_context_free(&avio_ctx); - std::cerr << "Error: could not allocate AVFormatContext" << std::endl; - return false; - } - format_ctx->pb = avio_ctx; - - if (avformat_open_input(&format_ctx, nullptr, nullptr, nullptr) < 0) { - // avformat_open_input frees format_ctx on failure - avio_context_free(&avio_ctx); - std::cerr << "Error: could not open audio from memory buffer" << std::endl; - return false; - } - - bool ok = decode_audio(format_ctx, out_audio, target_sample_rate, downmix); - avformat_close_input(&format_ctx); - avio_context_free(&avio_ctx); - return ok; + return decode_audio(data, size, out_audio, target_sample_rate, downmix); } // --------------------------------------------------------------------------- -// Core decode + resample logic (shared by file and memory paths) +// Core decode + downmix + resample logic (shared by file and memory paths) // --------------------------------------------------------------------------- -bool AudioReader::decode_audio(AVFormatContext* format_ctx, +bool AudioReader::decode_audio(const uint8_t* data, size_t size, audio_data_t& out_audio, int target_sample_rate, MonoDownmixMode downmix) { - if (avformat_find_stream_info(format_ctx, nullptr) < 0) { - std::cerr << "Error: could not find stream information" << std::endl; - return false; - } - - // Find the first audio stream - int audio_stream_idx = -1; - for (unsigned i = 0; i < format_ctx->nb_streams; ++i) { - if (format_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { - audio_stream_idx = static_cast(i); - break; - } - } - if (audio_stream_idx < 0) { - std::cerr << "Error: no audio stream found" << std::endl; - return false; - } - - AVStream* audio_stream = format_ctx->streams[audio_stream_idx]; - AVCodecParameters* codec_params = audio_stream->codecpar; - - // Stash original metadata before decoding - const int original_sample_rate = codec_params->sample_rate; - const int original_channels = codec_params->ch_layout.nb_channels; - - // Open decoder - const AVCodec* codec = avcodec_find_decoder(codec_params->codec_id); - if (!codec) { - std::cerr << "Error: unsupported audio codec" << std::endl; - return false; - } - - AVCodecContext* codec_ctx = avcodec_alloc_context3(codec); - if (!codec_ctx) { - std::cerr << "Error: could not allocate codec context" << std::endl; - return false; - } - - if (avcodec_parameters_to_context(codec_ctx, codec_params) < 0) { - avcodec_free_context(&codec_ctx); - std::cerr << "Error: could not copy codec parameters" << std::endl; + if (target_sample_rate <= 0) { + std::cerr << "Error: invalid target sample rate " << target_sample_rate << std::endl; return false; } - if (avcodec_open2(codec_ctx, codec, nullptr) < 0) { - avcodec_free_context(&codec_ctx); - std::cerr << "Error: could not open audio codec" << std::endl; + audio::DecodedAudio decoded; + std::string error; + if (!audio::decode_audio_file(data, size, decoded, error)) { + std::cerr << "Error: could not decode audio: " << error << std::endl; return false; } - // Setup resampler: any input → mono float32 at target_sample_rate - SwrContext* swr_ctx = swr_alloc(); - if (!swr_ctx) { - avcodec_free_context(&codec_ctx); - std::cerr << "Error: could not allocate resampler" << std::endl; - return false; - } - - av_opt_set_chlayout(swr_ctx, "in_chlayout", &codec_ctx->ch_layout, 0); - av_opt_set_int(swr_ctx, "in_sample_rate", codec_ctx->sample_rate, 0); - av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", codec_ctx->sample_fmt, 0); + const int original_sample_rate = decoded.sample_rate; + const int original_channels = decoded.channels; + std::vector samples = std::move(decoded.samples); - // Determine output channel layout + // Downmix to mono before resampling (cheaper, and the order libswresample + // used): every channel weighted by the same coefficient. const bool do_downmix = (downmix != MonoDownmixMode::NONE) && (original_channels > 1); - AVChannelLayout out_chlayout; - if (do_downmix) { - out_chlayout = AV_CHANNEL_LAYOUT_MONO; - } else { - out_chlayout = codec_ctx->ch_layout; - } - const int out_channels = out_chlayout.nb_channels; - - av_opt_set_chlayout(swr_ctx, "out_chlayout", &out_chlayout, 0); - av_opt_set_int(swr_ctx, "out_sample_rate", target_sample_rate, 0); - av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", AV_SAMPLE_FMT_FLT, 0); - - // Override FFmpeg's default downmix matrix with custom coefficients - // via swr_set_matrix(). Must be called before swr_init(). + int out_channels = original_channels; if (do_downmix) { - // matrix is out_channels x in_channels (row-major), stride = in_channels - std::vector matrix(original_channels); - double coeff; - switch (downmix) { - case MonoDownmixMode::MEAN: - // Simple average: 1/N per channel (matches librosa's to_mono) - coeff = 1.0 / original_channels; - break; - case MonoDownmixMode::RMS: - // Energy-preserving: 1/sqrt(N) per channel (FFmpeg default) - coeff = 1.0 / std::sqrt(static_cast(original_channels)); - break; - default: - coeff = 1.0 / original_channels; - break; - } - for (int c = 0; c < original_channels; ++c) { - matrix[c] = coeff; - } - swr_set_matrix(swr_ctx, matrix.data(), original_channels); - } - - if (swr_init(swr_ctx) < 0) { - swr_free(&swr_ctx); - avcodec_free_context(&codec_ctx); - std::cerr << "Error: could not initialize resampler" << std::endl; - return false; - } - - AVPacket* packet = av_packet_alloc(); - AVFrame* frame = av_frame_alloc(); - if (!packet || !frame) { - av_packet_free(&packet); - av_frame_free(&frame); - swr_free(&swr_ctx); - avcodec_free_context(&codec_ctx); - std::cerr << "Error: could not allocate packet/frame" << std::endl; - return false; - } - - std::vector samples; - - // Lambda: resample one decoded frame and append to samples - auto resample_frame = [&](AVFrame* f) { - const int out_count = av_rescale_rnd( - swr_get_delay(swr_ctx, codec_ctx->sample_rate) + f->nb_samples, - target_sample_rate, - codec_ctx->sample_rate, - AV_ROUND_UP); - - // Allocate temporary output buffer - uint8_t* out_buf = nullptr; - int out_linesize = 0; - if (av_samples_alloc(&out_buf, &out_linesize, out_channels, out_count, AV_SAMPLE_FMT_FLT, 0) < 0) { - return; - } - - int converted = swr_convert(swr_ctx, - &out_buf, out_count, - const_cast(f->data), f->nb_samples); - if (converted > 0) { - const float* flt = reinterpret_cast(out_buf); - samples.insert(samples.end(), flt, flt + converted * out_channels); - } - av_freep(&out_buf); - }; - - // Read & decode packets - while (av_read_frame(format_ctx, packet) >= 0) { - if (packet->stream_index == audio_stream_idx) { - if (avcodec_send_packet(codec_ctx, packet) >= 0) { - while (avcodec_receive_frame(codec_ctx, frame) >= 0) { - resample_frame(frame); - } - } - } - av_packet_unref(packet); - } - - // Flush decoder - avcodec_send_packet(codec_ctx, nullptr); - while (avcodec_receive_frame(codec_ctx, frame) >= 0) { - resample_frame(frame); - } - - // Flush resampler - { - int tail = av_rescale_rnd(swr_get_delay(swr_ctx, codec_ctx->sample_rate), - target_sample_rate, codec_ctx->sample_rate, AV_ROUND_UP); - if (tail > 0) { - uint8_t* out_buf = nullptr; - int out_linesize = 0; - if (av_samples_alloc(&out_buf, &out_linesize, out_channels, tail, AV_SAMPLE_FMT_FLT, 0) >= 0) { - int converted = swr_convert(swr_ctx, &out_buf, tail, nullptr, 0); - if (converted > 0) { - const float* flt = reinterpret_cast(out_buf); - samples.insert(samples.end(), flt, flt + converted * out_channels); - } - av_freep(&out_buf); + const float coeff = downmix == MonoDownmixMode::RMS + // Energy-preserving: 1/sqrt(N) per channel (FFmpeg default) + ? static_cast(1.0 / std::sqrt(static_cast(original_channels))) + // Simple average: 1/N per channel (matches librosa's to_mono) + : 1.0f / static_cast(original_channels); + const size_t frames = samples.size() / original_channels; + std::vector mono(frames); + for (size_t f = 0; f < frames; ++f) { + float sum = 0.0f; + for (int c = 0; c < original_channels; ++c) { + sum += samples[f * original_channels + c]; } + mono[f] = sum * coeff; } + samples = std::move(mono); + out_channels = 1; } - // Cleanup FFmpeg objects - av_packet_free(&packet); - av_frame_free(&frame); - swr_free(&swr_ctx); - avcodec_free_context(&codec_ctx); + samples = audio::resample(samples, out_channels, original_sample_rate, target_sample_rate); // Fill output struct out_audio.samples = std::move(samples); @@ -356,9 +116,7 @@ bool AudioReader::decode_audio(AVFormatContext* format_ctx, out_audio.num_frames = (out_channels > 0) ? out_audio.num_samples / out_channels : 0; - out_audio.duration_seconds = (target_sample_rate > 0) - ? static_cast(out_audio.num_frames) / target_sample_rate - : 0.0; + out_audio.duration_seconds = static_cast(out_audio.num_frames) / target_sample_rate; return out_audio.num_samples > 0; } diff --git a/src/common/audio/audio_resampler.cpp b/src/common/audio/audio_resampler.cpp new file mode 100644 index 000000000..b37eb1b69 --- /dev/null +++ b/src/common/audio/audio_resampler.cpp @@ -0,0 +1,145 @@ +/// \file audio_resampler.cpp +/// \brief Sample-rate conversion matching libswresample's default filter design +/// \author FastFlowLM Team + +#include "audio/audio_resampler.hpp" + +#include +#include +#include +#include + +namespace audio { +namespace { + +// libswresample defaults (libswresample/options.c). +constexpr int kFilterSize = 32; +constexpr int kPhaseShift = 10; +constexpr double kCutoff = 0.97; +constexpr double kKaiserBeta = 9.0; + +// Zeroth-order modified Bessel function of the first kind (power series). +double bessel_i0(double x) { + double sum = 1.0; + double term = 1.0; + const double q = x * x / 4.0; + for (int k = 1; k < 64; ++k) { + term *= q / (static_cast(k) * k); + sum += term; + if (term < sum * 1e-17) { + break; + } + } + return sum; +} + +// Kaiser-windowed sinc filter bank, one row of `taps` coefficients per phase, +// normalized by the DC gain of phase 0 (as build_filter() in +// libswresample/resample.c does). +std::vector build_filter_bank(double factor, int taps, int phases) { + const int center = (taps - 1) / 2; + std::vector bank(static_cast(phases + 1) * taps); + std::vector row(taps); + double norm = 0.0; + for (int ph = 0; ph <= phases; ++ph) { + for (int i = 0; i < taps; ++i) { + const double x = M_PI * ((i - center) - static_cast(ph) / phases) * factor; + double y = x == 0.0 ? 1.0 : std::sin(x) / x; + const double w = 2.0 * x / (factor * taps * M_PI); + y *= bessel_i0(kKaiserBeta * std::sqrt(std::max(1.0 - w * w, 0.0))); + row[i] = y; + if (ph == 0) { + norm += y; + } + } + for (int i = 0; i < taps; ++i) { + bank[static_cast(ph) * taps + i] = static_cast(row[i] / norm); + } + } + return bank; +} + +// Index into the signal extended the way libswresample extends it: mirrored +// about the first sample at the start (x[-k] = x[k], invert_initial_buffer) +// and about the end at the tail (x[n+k] = x[n-1-k], resample_flush). The +// filter never reaches further than its half-length past either edge. +inline int64_t mirror(int64_t i, int64_t n) { + if (i < 0) { + i = -i; + } else if (i >= n) { + i = 2 * n - 1 - i; + } + return std::clamp(i, 0, n - 1); +} + +} // namespace + +std::vector resample(const std::vector& in, int channels, int in_rate, int out_rate) { + if (in_rate == out_rate || in.empty() || channels <= 0 || in_rate <= 0 || out_rate <= 0) { + return in; + } + const int64_t in_frames = static_cast(in.size()) / channels; + + const double factor = std::min(out_rate * kCutoff / in_rate, 1.0); + // libswresample rounds the filter length up to an even number of taps. + int taps = std::max(static_cast(std::ceil(kFilterSize / factor)), 1); + if (taps > 1) { + taps += taps & 1; + } + const int center = (taps - 1) / 2; + + // Output sample n sits at input position n * in_rate / out_rate. With the + // ratio reduced to in_step / out_step, the fractional part takes exactly + // out_step values; use one filter phase each when that fits in the default + // 1024 phases, otherwise 1024 phases with linear interpolation. + const int g = std::gcd(in_rate, out_rate); + const int64_t in_step = in_rate / g; + const int64_t out_step = out_rate / g; + const bool exact = out_step <= (1 << kPhaseShift); + const int phases = exact ? static_cast(out_step) : (1 << kPhaseShift); + const std::vector bank = build_filter_bank(factor, taps, phases); + + const int64_t out_frames = (in_frames * out_step + in_step - 1) / in_step; + std::vector out(static_cast(out_frames) * channels); + + for (int64_t n = 0; n < out_frames; ++n) { + const int64_t pos = n * in_step; + const int64_t idx = pos / out_step; + const int64_t rem = pos % out_step; + const int64_t first = idx - center; + const bool interior = first >= 0 && first + taps <= in_frames; + + for (int c = 0; c < channels; ++c) { + auto sample = [&](int i) -> float { + const int64_t k = interior ? first + i : mirror(first + i, in_frames); + return in[static_cast(k) * channels + c]; + }; + float v; + if (exact) { + const float* h = &bank[static_cast(rem) * taps]; + v = 0.0f; + for (int i = 0; i < taps; ++i) { + v += sample(i) * h[i]; + } + } else { + const double fpos = static_cast(rem) * phases / out_step; + const int ph = static_cast(fpos); + const float frac = static_cast(fpos - ph); + const float* h1 = &bank[static_cast(ph) * taps]; + const float* h2 = h1 + taps; + float v1 = 0.0f; + float v2 = 0.0f; + for (int i = 0; i < taps; ++i) { + const float s = sample(i); + v1 += s * h1[i]; + v2 += s * h2[i]; + } + v = v1 + (v2 - v1) * frac; + } + out[static_cast(n) * channels + c] = v; + } + } + return out; +} + +} // namespace audio diff --git a/src/common/whisper/modeling_whisper_audio.cpp b/src/common/whisper/modeling_whisper_audio.cpp index fd54b8650..b484c2003 100644 --- a/src/common/whisper/modeling_whisper_audio.cpp +++ b/src/common/whisper/modeling_whisper_audio.cpp @@ -22,524 +22,33 @@ #include "utils/debug_utils.hpp" #include "buffer.hpp" #include "tensor_2d.hpp" +#include "audio/audio_reader.hpp" #include // For AVX intrinsics -extern "C" { -#include -#include -#include -#include +// Decoding, downmix and resampling go through AudioReader. Whisper expects +// 16 kHz mono; stereo is averaged (0.5 L + 0.5 R), which is what the previous +// FFmpeg path produced, but samples now stay float instead of passing through +// 16-bit PCM. +static std::vector load_whisper_audio(AudioReader& reader, const uint8_t* data, size_t size, + const std::string& what) { + constexpr int TARGET_SAMPLE_RATE = 16000; + audio_data_t audio; + const bool ok = data ? reader.load_audio_from_memory(data, size, audio, TARGET_SAMPLE_RATE, MonoDownmixMode::MEAN) + : reader.load_audio(what, audio, TARGET_SAMPLE_RATE, MonoDownmixMode::MEAN); + if (!ok) { + throw std::runtime_error("Could not decode audio " + what); + } + return std::move(audio.samples); } std::vector Whisper::_load_audio(std::string& filename) { - AVFormatContext* format_ctx = nullptr; - AVCodecContext* codec_ctx = nullptr; - SwrContext* swr_ctx = nullptr; - std::vector result_audio_data; - - const int TARGET_SAMPLE_RATE = 16000; - - // Suppress FFmpeg warnings and info messages - av_log_set_level(AV_LOG_ERROR); - - try { - // Open input file - if (avformat_open_input(&format_ctx, filename.c_str(), nullptr, nullptr) < 0) { - throw std::runtime_error("Could not open audio file: " + filename); - } - - // Retrieve stream information - if (avformat_find_stream_info(format_ctx, nullptr) < 0) { - throw std::runtime_error("Could not find stream information"); - } - - // Find the audio stream - int audio_stream_index = -1; - for (unsigned int i = 0; i < format_ctx->nb_streams; i++) { - if (format_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { - audio_stream_index = i; - break; - } - } - - if (audio_stream_index == -1) { - throw std::runtime_error("Could not find audio stream"); - } - - AVStream* audio_stream = format_ctx->streams[audio_stream_index]; - AVCodecParameters* codec_params = audio_stream->codecpar; - - // Find decoder - const AVCodec* codec = avcodec_find_decoder(codec_params->codec_id); - if (!codec) { - throw std::runtime_error("Unsupported codec"); - } - - // Allocate codec context - codec_ctx = avcodec_alloc_context3(codec); - if (!codec_ctx) { - throw std::runtime_error("Could not allocate codec context"); - } - - // Copy codec parameters to codec context - if (avcodec_parameters_to_context(codec_ctx, codec_params) < 0) { - throw std::runtime_error("Could not copy codec parameters to codec context"); - } - - // Open codec - if (avcodec_open2(codec_ctx, codec, nullptr) < 0) { - throw std::runtime_error("Could not open codec"); - } - - // Get input sample rate and channel layout - int input_sample_rate = codec_ctx->sample_rate; - AVChannelLayout input_ch_layout = codec_ctx->ch_layout; - AVSampleFormat input_sample_fmt = codec_ctx->sample_fmt; - - // Always setup resampler to convert to mono 16kHz s16le (like FFmpeg command) - swr_ctx = swr_alloc(); - if (!swr_ctx) { - throw std::runtime_error("Could not allocate resampler context"); - } - - // Setup output channel layout (mono) - AVChannelLayout out_ch_layout = AV_CHANNEL_LAYOUT_MONO; - - // Set resampler options to match FFmpeg command: -f s16le -ac 1 -ar 16000 - av_opt_set_chlayout(swr_ctx, "in_chlayout", &input_ch_layout, 0); - av_opt_set_int(swr_ctx, "in_sample_rate", input_sample_rate, 0); - av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", input_sample_fmt, 0); - - av_opt_set_chlayout(swr_ctx, "out_chlayout", &out_ch_layout, 0); - av_opt_set_int(swr_ctx, "out_sample_rate", TARGET_SAMPLE_RATE, 0); - av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0); // s16le format - - // Initialize resampler - if (swr_init(swr_ctx) < 0) { - throw std::runtime_error("Could not initialize resampler"); - } - - // Read frames and decode - AVPacket* packet = av_packet_alloc(); - AVFrame* frame = av_frame_alloc(); - - if (!packet || !frame) { - av_packet_free(&packet); - av_frame_free(&frame); - throw std::runtime_error("Could not allocate packet or frame"); - } - - while (av_read_frame(format_ctx, packet) >= 0) { - if (packet->stream_index == audio_stream_index) { - // Send packet to decoder - if (avcodec_send_packet(codec_ctx, packet) >= 0) { - // Receive decoded frames - while (avcodec_receive_frame(codec_ctx, frame) >= 0) { - // Calculate output samples - int out_samples = av_rescale_rnd( - swr_get_delay(swr_ctx, input_sample_rate) + frame->nb_samples, - TARGET_SAMPLE_RATE, - input_sample_rate, - AV_ROUND_UP - ); - - // Allocate output buffer for s16le format - uint8_t* output_buffer = nullptr; - int out_linesize; - av_samples_alloc(&output_buffer, &out_linesize, 1, out_samples, - AV_SAMPLE_FMT_S16, 0); - - // Resample to s16le format - int converted_samples = swr_convert( - swr_ctx, - &output_buffer, - out_samples, - (const uint8_t**)frame->data, - frame->nb_samples - ); - - if (converted_samples > 0) { - // Convert s16le to float32 (normalize to [-1.0, 1.0]) - int16_t* s16_data = reinterpret_cast(output_buffer); - for (int i = 0; i < converted_samples; i++) { - result_audio_data.push_back(s16_data[i] / 32768.0f); - } - } - - av_freep(&output_buffer); - } - } - } - av_packet_unref(packet); - } - - // Flush decoder - avcodec_send_packet(codec_ctx, nullptr); - while (avcodec_receive_frame(codec_ctx, frame) >= 0) { - int out_samples = av_rescale_rnd( - swr_get_delay(swr_ctx, input_sample_rate) + frame->nb_samples, - TARGET_SAMPLE_RATE, - input_sample_rate, - AV_ROUND_UP - ); - - uint8_t* output_buffer = nullptr; - int out_linesize; - av_samples_alloc(&output_buffer, &out_linesize, 1, out_samples, - AV_SAMPLE_FMT_S16, 0); - - int converted_samples = swr_convert( - swr_ctx, - &output_buffer, - out_samples, - (const uint8_t**)frame->data, - frame->nb_samples - ); - - if (converted_samples > 0) { - // Convert s16le to float32 (normalize to [-1.0, 1.0]) - int16_t* s16_data = reinterpret_cast(output_buffer); - for (int i = 0; i < converted_samples; i++) { - result_audio_data.push_back(s16_data[i] / 32768.0f); - } - } - - av_freep(&output_buffer); - } - - // Flush resampler - if (swr_ctx) { - int out_samples = av_rescale_rnd(swr_get_delay(swr_ctx, input_sample_rate), - TARGET_SAMPLE_RATE, input_sample_rate, AV_ROUND_UP); - if (out_samples > 0) { - uint8_t* output_buffer = nullptr; - int out_linesize; - av_samples_alloc(&output_buffer, &out_linesize, 1, out_samples, - AV_SAMPLE_FMT_S16, 0); - - int converted_samples = swr_convert(swr_ctx, &output_buffer, out_samples, - nullptr, 0); - - if (converted_samples > 0) { - // Convert s16le to float32 (normalize to [-1.0, 1.0]) - int16_t* s16_data = reinterpret_cast(output_buffer); - for (int i = 0; i < converted_samples; i++) { - result_audio_data.push_back(s16_data[i] / 32768.0f); - } - } - - av_freep(&output_buffer); - } - } - - // Cleanup - av_packet_free(&packet); - av_frame_free(&frame); - - if (swr_ctx) { - swr_free(&swr_ctx); - } - if (codec_ctx) { - avcodec_free_context(&codec_ctx); - } - if (format_ctx) { - avformat_close_input(&format_ctx); - } - - return result_audio_data; - - } catch (...) { - // Cleanup on error - if (swr_ctx) { - swr_free(&swr_ctx); - } - if (codec_ctx) { - avcodec_free_context(&codec_ctx); - } - if (format_ctx) { - avformat_close_input(&format_ctx); - } - throw; - } + AudioReader reader; + return load_whisper_audio(reader, nullptr, 0, filename); } - - std::vector Whisper::_load_audio(std::vector& audio_data) { - AVFormatContext* format_ctx = nullptr; - AVCodecContext* codec_ctx = nullptr; - SwrContext* swr_ctx = nullptr; - std::vector result_audio_data; - - const int TARGET_SAMPLE_RATE = 16000; - - // Suppress FFmpeg warnings and info messages - av_log_set_level(AV_LOG_ERROR); - - try { - // Allocate buffer using FFmpeg's allocator so it can be safely freed - uint8_t* io_buffer = (uint8_t*)av_malloc(audio_data.size()); - if (!io_buffer) { - throw std::runtime_error("Could not allocate IO buffer for memory"); - } - std::memcpy(io_buffer, audio_data.data(), audio_data.size()); - - // Create a custom IO context for memory buffer - AVIOContext* raw_io_ctx = avio_alloc_context( - io_buffer, // buffer (now FFmpeg-managed) - audio_data.size(), // buffer size - 0, // write flag (0 = read-only) - nullptr, // opaque (not needed) - nullptr, // read_packet (not needed for read-only) - nullptr, // write_packet (not needed for read-only) - nullptr // seek (not needed for read-only) - ); - - if (!raw_io_ctx) { - av_free(io_buffer); - throw std::runtime_error("Could not allocate IO context for memory buffer"); - } - - // Create shared_ptr with custom deleter for FFmpeg-allocated memory - std::shared_ptr io_ctx(raw_io_ctx, [](AVIOContext* ctx) { - if (ctx) { - // Free the buffer first, then the context - av_free(ctx->buffer); - avio_context_free(&ctx); - } - }); - - // Allocate format context - format_ctx = avformat_alloc_context(); - if (!format_ctx) { - throw std::runtime_error("Could not allocate format context"); - } - - // Set the custom IO context - format_ctx->pb = io_ctx.get(); - - // Open input from memory buffer - if (avformat_open_input(&format_ctx, nullptr, nullptr, nullptr) < 0) { - throw std::runtime_error("Could not open audio data from memory buffer"); - } - - // Retrieve stream information - if (avformat_find_stream_info(format_ctx, nullptr) < 0) { - throw std::runtime_error("Could not find stream information"); - } - - // Find the audio stream - int audio_stream_index = -1; - for (unsigned int i = 0; i < format_ctx->nb_streams; i++) { - if (format_ctx->streams[i]->codecpar->codec_type == AVMEDIA_TYPE_AUDIO) { - audio_stream_index = i; - break; - } - } - - if (audio_stream_index == -1) { - throw std::runtime_error("Could not find audio stream"); - } - - AVStream* audio_stream = format_ctx->streams[audio_stream_index]; - AVCodecParameters* codec_params = audio_stream->codecpar; - - // Find decoder - const AVCodec* codec = avcodec_find_decoder(codec_params->codec_id); - if (!codec) { - throw std::runtime_error("Unsupported codec"); - } - - // Allocate codec context - codec_ctx = avcodec_alloc_context3(codec); - if (!codec_ctx) { - throw std::runtime_error("Could not allocate codec context"); - } - - // Copy codec parameters to codec context - if (avcodec_parameters_to_context(codec_ctx, codec_params) < 0) { - throw std::runtime_error("Could not copy codec parameters to codec context"); - } - - // Open codec - if (avcodec_open2(codec_ctx, codec, nullptr) < 0) { - throw std::runtime_error("Could not open codec"); - } - - // Get input sample rate and channel layout - int input_sample_rate = codec_ctx->sample_rate; - AVChannelLayout input_ch_layout = codec_ctx->ch_layout; - AVSampleFormat input_sample_fmt = codec_ctx->sample_fmt; - - // Always setup resampler to convert to mono 16kHz s16le (like FFmpeg command) - swr_ctx = swr_alloc(); - if (!swr_ctx) { - throw std::runtime_error("Could not allocate resampler context"); - } - - // Setup output channel layout (mono) - AVChannelLayout out_ch_layout = AV_CHANNEL_LAYOUT_MONO; - - // Set resampler options to match FFmpeg command: -f s16le -ac 1 -ar 16000 - av_opt_set_chlayout(swr_ctx, "in_chlayout", &input_ch_layout, 0); - av_opt_set_int(swr_ctx, "in_sample_rate", input_sample_rate, 0); - av_opt_set_sample_fmt(swr_ctx, "in_sample_fmt", input_sample_fmt, 0); - - av_opt_set_chlayout(swr_ctx, "out_chlayout", &out_ch_layout, 0); - av_opt_set_int(swr_ctx, "out_sample_rate", TARGET_SAMPLE_RATE, 0); - av_opt_set_sample_fmt(swr_ctx, "out_sample_fmt", AV_SAMPLE_FMT_S16, 0); // s16le format - - // Initialize resampler - if (swr_init(swr_ctx) < 0) { - throw std::runtime_error("Could not initialize resampler"); - } - - // Read frames and decode - AVPacket* packet = av_packet_alloc(); - AVFrame* frame = av_frame_alloc(); - - if (!packet || !frame) { - av_packet_free(&packet); - av_frame_free(&frame); - throw std::runtime_error("Could not allocate packet or frame"); - } - - while (av_read_frame(format_ctx, packet) >= 0) { - if (packet->stream_index == audio_stream_index) { - // Send packet to decoder - if (avcodec_send_packet(codec_ctx, packet) >= 0) { - // Receive decoded frames - while (avcodec_receive_frame(codec_ctx, frame) >= 0) { - // Calculate output samples - int out_samples = av_rescale_rnd( - swr_get_delay(swr_ctx, input_sample_rate) + frame->nb_samples, - TARGET_SAMPLE_RATE, - input_sample_rate, - AV_ROUND_UP - ); - - // Allocate output buffer for s16le format - uint8_t* output_buffer = nullptr; - int out_linesize; - av_samples_alloc(&output_buffer, &out_linesize, 1, out_samples, - AV_SAMPLE_FMT_S16, 0); - - // Resample to s16le format - int converted_samples = swr_convert( - swr_ctx, - &output_buffer, - out_samples, - (const uint8_t**)frame->data, - frame->nb_samples - ); - - if (converted_samples > 0) { - // Convert s16le to float32 (normalize to [-1.0, 1.0]) - int16_t* s16_data = reinterpret_cast(output_buffer); - for (int i = 0; i < converted_samples; i++) { - result_audio_data.push_back(s16_data[i] / 32768.0f); - } - } - - av_freep(&output_buffer); - } - } - } - av_packet_unref(packet); - } - - // Flush decoder - avcodec_send_packet(codec_ctx, nullptr); - while (avcodec_receive_frame(codec_ctx, frame) >= 0) { - int out_samples = av_rescale_rnd( - swr_get_delay(swr_ctx, input_sample_rate) + frame->nb_samples, - TARGET_SAMPLE_RATE, - input_sample_rate, - AV_ROUND_UP - ); - - uint8_t* output_buffer = nullptr; - int out_linesize; - av_samples_alloc(&output_buffer, &out_linesize, 1, out_samples, - AV_SAMPLE_FMT_S16, 0); - - int converted_samples = swr_convert( - swr_ctx, - &output_buffer, - out_samples, - (const uint8_t**)frame->data, - frame->nb_samples - ); - - if (converted_samples > 0) { - // Convert s16le to float32 (normalize to [-1.0, 1.0]) - int16_t* s16_data = reinterpret_cast(output_buffer); - for (int i = 0; i < converted_samples; i++) { - result_audio_data.push_back(s16_data[i] / 32768.0f); - } - } - - av_freep(&output_buffer); - } - - // Flush resampler - if (swr_ctx) { - int out_samples = av_rescale_rnd(swr_get_delay(swr_ctx, input_sample_rate), - TARGET_SAMPLE_RATE, input_sample_rate, AV_ROUND_UP); - if (out_samples > 0) { - uint8_t* output_buffer = nullptr; - int out_linesize; - av_samples_alloc(&output_buffer, &out_linesize, 1, out_samples, - AV_SAMPLE_FMT_S16, 0); - - int converted_samples = swr_convert(swr_ctx, &output_buffer, out_samples, - nullptr, 0); - - if (converted_samples > 0) { - // Convert s16le to float32 (normalize to [-1.0, 1.0]) - int16_t* s16_data = reinterpret_cast(output_buffer); - for (int i = 0; i < converted_samples; i++) { - result_audio_data.push_back(s16_data[i] / 32768.0f); - } - } - - av_freep(&output_buffer); - } - } - - // Cleanup - av_packet_free(&packet); - av_frame_free(&frame); - - if (swr_ctx) { - swr_free(&swr_ctx); - } - if (codec_ctx) { - avcodec_free_context(&codec_ctx); - } - if (format_ctx) { - avformat_close_input(&format_ctx); - } - // Note: io_ctx will be automatically freed when format_ctx is closed - // The shared_ptr will handle the cleanup when it goes out of scope - - return result_audio_data; - - } catch (...) { - // Cleanup on error - if (swr_ctx) { - swr_free(&swr_ctx); - } - if (codec_ctx) { - avcodec_free_context(&codec_ctx); - } - if (format_ctx) { - avformat_close_input(&format_ctx); - } - // Note: io_ctx will be automatically freed when format_ctx is closed - // The shared_ptr will handle the cleanup when it goes out of scope - throw; - } + AudioReader reader; + return load_whisper_audio(reader, audio_data.data(), audio_data.size(), "data"); } static float hann_window[400] = { diff --git a/src/include/audio/audio_decoder.hpp b/src/include/audio/audio_decoder.hpp new file mode 100644 index 000000000..a8ea7acd6 --- /dev/null +++ b/src/include/audio/audio_decoder.hpp @@ -0,0 +1,28 @@ +/// \file audio_decoder.hpp +/// \brief Container/codec decoding for AudioReader (no FFmpeg) +/// \author FastFlowLM Team +#pragma once + +#include +#include +#include +#include + +namespace audio { + +/// Decoded PCM at the source's native sample rate and channel count. +struct DecodedAudio { + /// Interleaved float32 samples, nominally in [-1.0, 1.0]. + std::vector samples; + int sample_rate = 0; + int channels = 0; +}; + +/// Decode a complete audio file held in memory. +/// +/// Supported: WAV (RIFF/RF64/W64/AIFF), FLAC (native and in Ogg), MP3, +/// Ogg Vorbis, Ogg Opus and WebM/Matroska Opus. Anything else, notably AAC +/// (.m4a/.mp4/ADTS), fails with a descriptive message in \p error. +bool decode_audio_file(const uint8_t* data, size_t size, DecodedAudio& out, std::string& error); + +} // namespace audio diff --git a/src/include/audio/audio_reader.hpp b/src/include/audio/audio_reader.hpp index 051f9d611..8d54be3dd 100644 --- a/src/include/audio/audio_reader.hpp +++ b/src/include/audio/audio_reader.hpp @@ -55,7 +55,7 @@ class AudioReader { ~AudioReader(); /// Load audio from a file, decode, resample to target_sample_rate, and convert to mono float32. - /// Supports any format FFmpeg can demux (mp3, wav, flac, ogg, etc.). + /// Supports WAV, FLAC, MP3, Ogg (Vorbis/Opus/FLAC) and WebM/Matroska (Opus); see audio_decoder.hpp. /// @param filename path to the audio file /// @param out_audio output struct filled with samples and metadata /// @param target_sample_rate desired output sample rate (default 16000) @@ -75,7 +75,7 @@ class AudioReader { bool clip_audio_length(audio_data_t& audio, double max_duration_second); private: - bool decode_audio(struct AVFormatContext* format_ctx, audio_data_t& out_audio, int target_sample_rate, MonoDownmixMode downmix); + bool decode_audio(const uint8_t* data, size_t size, audio_data_t& out_audio, int target_sample_rate, MonoDownmixMode downmix); }; diff --git a/src/include/audio/audio_resampler.hpp b/src/include/audio/audio_resampler.hpp new file mode 100644 index 000000000..38d33d8ba --- /dev/null +++ b/src/include/audio/audio_resampler.hpp @@ -0,0 +1,19 @@ +/// \file audio_resampler.hpp +/// \brief Sample-rate conversion for AudioReader (no FFmpeg) +/// \author FastFlowLM Team +#pragma once + +#include + +namespace audio { + +/// Resample interleaved float32 PCM from \p in_rate to \p out_rate. +/// +/// Uses the filter design libswresample (FFmpeg) applies by default, so output +/// stays close to what AudioReader produced before: Kaiser-windowed sinc +/// (beta 9), cutoff 0.97 of the lower Nyquist, filter_size 32, exact rational +/// phases when they fit in 1024 (otherwise 1024 phases with linear +/// interpolation), and mirrored signal edges. +std::vector resample(const std::vector& in, int channels, int in_rate, int out_rate); + +} // namespace audio diff --git a/src/include/dr_libs/LICENSE b/src/include/dr_libs/LICENSE new file mode 100644 index 000000000..16dfbdfda --- /dev/null +++ b/src/include/dr_libs/LICENSE @@ -0,0 +1,47 @@ +This software is available as a choice of the following licenses. Choose +whichever you prefer. + +=============================================================================== +ALTERNATIVE 1 - Public Domain (www.unlicense.org) +=============================================================================== +This is free and unencumbered software released into the public domain. + +Anyone is free to copy, modify, publish, use, compile, sell, or distribute this +software, either in source code form or as a compiled binary, for any purpose, +commercial or non-commercial, and by any means. + +In jurisdictions that recognize copyright laws, the author or authors of this +software dedicate any and all copyright interest in the software to the public +domain. We make this dedication for the benefit of the public at large and to +the detriment of our heirs and successors. We intend this dedication to be an +overt act of relinquishment in perpetuity of all present and future rights to +this software under copyright law. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN +ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION +WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + +For more information, please refer to + +=============================================================================== +ALTERNATIVE 2 - MIT No Attribution +=============================================================================== +Copyright 2020 David Reid + +Permission is hereby granted, free of charge, to any person obtaining a copy of +this software and associated documentation files (the "Software"), to deal in +the Software without restriction, including without limitation the rights to +use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies +of the Software, and to permit persons to whom the Software is furnished to do +so. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +SOFTWARE. diff --git a/src/include/dr_libs/dr_flac.h b/src/include/dr_libs/dr_flac.h new file mode 100644 index 000000000..051f37c66 --- /dev/null +++ b/src/include/dr_libs/dr_flac.h @@ -0,0 +1,12760 @@ +/* +FLAC audio decoder. Choice of public domain or MIT-0. See license statements at the end of this file. +dr_flac - v0.13.4 - TBD + +David Reid - mackron@gmail.com + +GitHub: https://github.com/mackron/dr_libs +*/ + +/* +Introduction +============ +dr_flac is a single file library. To use it, do something like the following in one .c file. + + ```c + #define DR_FLAC_IMPLEMENTATION + #include "dr_flac.h" + ``` + +You can then #include this file in other parts of the program as you would with any other header file. To decode audio data, do something like the following: + + ```c + drflac* pFlac = drflac_open_file("MySong.flac", NULL); + if (pFlac == NULL) { + // Failed to open FLAC file + } + + drflac_int32* pSamples = malloc(pFlac->totalPCMFrameCount * pFlac->channels * sizeof(drflac_int32)); + drflac_uint64 numberOfInterleavedSamplesActuallyRead = drflac_read_pcm_frames_s32(pFlac, pFlac->totalPCMFrameCount, pSamples); + ``` + +The drflac object represents the decoder. It is a transparent type so all the information you need, such as the number of channels and the bits per sample, +should be directly accessible - just make sure you don't change their values. Samples are always output as interleaved signed 32-bit PCM. In the example above +a native FLAC stream was opened, however dr_flac has seamless support for Ogg encapsulated FLAC streams as well. + +You do not need to decode the entire stream in one go - you just specify how many samples you'd like at any given time and the decoder will give you as many +samples as it can, up to the amount requested. Later on when you need the next batch of samples, just call it again. Example: + + ```c + while (drflac_read_pcm_frames_s32(pFlac, chunkSizeInPCMFrames, pChunkSamples) > 0) { + do_something(); + } + ``` + +You can seek to a specific PCM frame with `drflac_seek_to_pcm_frame()`. + +If you just want to quickly decode an entire FLAC file in one go you can do something like this: + + ```c + unsigned int channels; + unsigned int sampleRate; + drflac_uint64 totalPCMFrameCount; + drflac_int32* pSampleData = drflac_open_file_and_read_pcm_frames_s32("MySong.flac", &channels, &sampleRate, &totalPCMFrameCount, NULL); + if (pSampleData == NULL) { + // Failed to open and decode FLAC file. + } + + ... + + drflac_free(pSampleData, NULL); + ``` + +You can read samples as signed 16-bit integer and 32-bit floating-point PCM with the *_s16() and *_f32() family of APIs respectively, but note that these +should be considered lossy. + + +If you need access to metadata (album art, etc.), use `drflac_open_with_metadata()`, `drflac_open_file_with_metdata()` or `drflac_open_memory_with_metadata()`. +The rationale for keeping these APIs separate is that they're slightly slower than the normal versions and also just a little bit harder to use. dr_flac +reports metadata to the application through the use of a callback, and every metadata block is reported before `drflac_open_with_metdata()` returns. + +The main opening APIs (`drflac_open()`, etc.) will fail if the header is not present. The presents a problem in certain scenarios such as broadcast style +streams or internet radio where the header may not be present because the user has started playback mid-stream. To handle this, use the relaxed APIs: + + `drflac_open_relaxed()` + `drflac_open_with_metadata_relaxed()` + +It is not recommended to use these APIs for file based streams because a missing header would usually indicate a corrupt or perverse file. In addition, these +APIs can take a long time to initialize because they may need to spend a lot of time finding the first frame. + + + +Build Options +============= +#define these options before including this file. + +#define DR_FLAC_NO_STDIO + Disable `drflac_open_file()` and family. + +#define DR_FLAC_NO_OGG + Disables support for Ogg/FLAC streams. + +#define DR_FLAC_BUFFER_SIZE + Defines the size of the internal buffer to store data from onRead(). This buffer is used to reduce the number of calls back to the client for more data. + Larger values means more memory, but better performance. My tests show diminishing returns after about 4KB (which is the default). Consider reducing this if + you have a very efficient implementation of onRead(), or increase it if it's very inefficient. Must be a multiple of 8. + +#define DR_FLAC_NO_CRC + Disables CRC checks. This will offer a performance boost when CRC is unnecessary. This will disable binary search seeking. When seeking, the seek table will + be used if available. Otherwise the seek will be performed using brute force. + +#define DR_FLAC_NO_SIMD + Disables SIMD optimizations (SSE on x86/x64 architectures, NEON on ARM architectures). Use this if you are having compatibility issues with your compiler. + +#define DR_FLAC_NO_WCHAR + Disables all functions ending with `_w`. Use this if your compiler does not provide wchar.h. Not required if DR_FLAC_NO_STDIO is also defined. + + + +Notes +===== +- dr_flac does not support changing the sample rate nor channel count mid stream. +- dr_flac is not thread-safe, but its APIs can be called from any thread so long as you do your own synchronization. +- When using Ogg encapsulation, a corrupted metadata block will result in `drflac_open_with_metadata()` and `drflac_open()` returning inconsistent samples due + to differences in corrupted stream recorvery logic between the two APIs. +*/ + +#ifndef dr_flac_h +#define dr_flac_h + +#ifdef __cplusplus +extern "C" { +#endif + +#define DRFLAC_STRINGIFY(x) #x +#define DRFLAC_XSTRINGIFY(x) DRFLAC_STRINGIFY(x) + +#define DRFLAC_VERSION_MAJOR 0 +#define DRFLAC_VERSION_MINOR 13 +#define DRFLAC_VERSION_REVISION 4 +#define DRFLAC_VERSION_STRING DRFLAC_XSTRINGIFY(DRFLAC_VERSION_MAJOR) "." DRFLAC_XSTRINGIFY(DRFLAC_VERSION_MINOR) "." DRFLAC_XSTRINGIFY(DRFLAC_VERSION_REVISION) + +#include /* For size_t. */ + +/* Sized Types */ +typedef signed char drflac_int8; +typedef unsigned char drflac_uint8; +typedef signed short drflac_int16; +typedef unsigned short drflac_uint16; +typedef signed int drflac_int32; +typedef unsigned int drflac_uint32; +#if defined(_MSC_VER) && !defined(__clang__) + typedef signed __int64 drflac_int64; + typedef unsigned __int64 drflac_uint64; +#else + #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic push + #pragma GCC diagnostic ignored "-Wlong-long" + #if defined(__clang__) + #pragma GCC diagnostic ignored "-Wc++11-long-long" + #endif + #endif + typedef signed long long drflac_int64; + typedef unsigned long long drflac_uint64; + #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic pop + #endif +#endif +#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined(_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(__powerpc64__) + typedef drflac_uint64 drflac_uintptr; +#else + typedef drflac_uint32 drflac_uintptr; +#endif +typedef drflac_uint8 drflac_bool8; +typedef drflac_uint32 drflac_bool32; +#define DRFLAC_TRUE 1 +#define DRFLAC_FALSE 0 +/* End Sized Types */ + +/* Decorations */ +#if !defined(DRFLAC_API) + #if defined(DRFLAC_DLL) + #if defined(_WIN32) + #define DRFLAC_DLL_IMPORT __declspec(dllimport) + #define DRFLAC_DLL_EXPORT __declspec(dllexport) + #define DRFLAC_DLL_PRIVATE static + #else + #if defined(__GNUC__) && __GNUC__ >= 4 + #define DRFLAC_DLL_IMPORT __attribute__((visibility("default"))) + #define DRFLAC_DLL_EXPORT __attribute__((visibility("default"))) + #define DRFLAC_DLL_PRIVATE __attribute__((visibility("hidden"))) + #else + #define DRFLAC_DLL_IMPORT + #define DRFLAC_DLL_EXPORT + #define DRFLAC_DLL_PRIVATE static + #endif + #endif + + #if defined(DR_FLAC_IMPLEMENTATION) || defined(DRFLAC_IMPLEMENTATION) + #define DRFLAC_API DRFLAC_DLL_EXPORT + #else + #define DRFLAC_API DRFLAC_DLL_IMPORT + #endif + #define DRFLAC_PRIVATE DRFLAC_DLL_PRIVATE + #else + #define DRFLAC_API extern + #define DRFLAC_PRIVATE static + #endif +#endif +/* End Decorations */ + +#if defined(_MSC_VER) && _MSC_VER >= 1700 /* Visual Studio 2012 */ + #define DRFLAC_DEPRECATED __declspec(deprecated) +#elif (defined(__GNUC__) && __GNUC__ >= 4) /* GCC 4 */ + #define DRFLAC_DEPRECATED __attribute__((deprecated)) +#elif defined(__has_feature) /* Clang */ + #if __has_feature(attribute_deprecated) + #define DRFLAC_DEPRECATED __attribute__((deprecated)) + #else + #define DRFLAC_DEPRECATED + #endif +#else + #define DRFLAC_DEPRECATED +#endif + +DRFLAC_API void drflac_version(drflac_uint32* pMajor, drflac_uint32* pMinor, drflac_uint32* pRevision); +DRFLAC_API const char* drflac_version_string(void); + +/* Allocation Callbacks */ +typedef struct +{ + void* pUserData; + void* (* onMalloc)(size_t sz, void* pUserData); + void* (* onRealloc)(void* p, size_t sz, void* pUserData); + void (* onFree)(void* p, void* pUserData); +} drflac_allocation_callbacks; +/* End Allocation Callbacks */ + +/* +As data is read from the client it is placed into an internal buffer for fast access. This controls the size of that buffer. Larger values means more speed, +but also more memory. In my testing there is diminishing returns after about 4KB, but you can fiddle with this to suit your own needs. Must be a multiple of 8. +*/ +#ifndef DR_FLAC_BUFFER_SIZE +#define DR_FLAC_BUFFER_SIZE 4096 +#endif + + +/* Architecture Detection */ +#if defined(_WIN64) || defined(_LP64) || defined(__LP64__) +#define DRFLAC_64BIT +#endif + +#if defined(__x86_64__) || (defined(_M_X64) && !defined(_M_ARM64EC)) + #define DRFLAC_X64 +#elif defined(__i386) || defined(_M_IX86) + #define DRFLAC_X86 +#elif defined(__arm__) || defined(_M_ARM) || defined(__arm64) || defined(__arm64__) || defined(__aarch64__) || defined(_M_ARM64) || defined(_M_ARM64EC) + #define DRFLAC_ARM +#endif +/* End Architecture Detection */ + + +#ifdef DRFLAC_64BIT +typedef drflac_uint64 drflac_cache_t; +#else +typedef drflac_uint32 drflac_cache_t; +#endif + +/* The various metadata block types. */ +#define DRFLAC_METADATA_BLOCK_TYPE_STREAMINFO 0 +#define DRFLAC_METADATA_BLOCK_TYPE_PADDING 1 +#define DRFLAC_METADATA_BLOCK_TYPE_APPLICATION 2 +#define DRFLAC_METADATA_BLOCK_TYPE_SEEKTABLE 3 +#define DRFLAC_METADATA_BLOCK_TYPE_VORBIS_COMMENT 4 +#define DRFLAC_METADATA_BLOCK_TYPE_CUESHEET 5 +#define DRFLAC_METADATA_BLOCK_TYPE_PICTURE 6 +#define DRFLAC_METADATA_BLOCK_TYPE_INVALID 127 + +/* The various picture types specified in the PICTURE block. */ +#define DRFLAC_PICTURE_TYPE_OTHER 0 +#define DRFLAC_PICTURE_TYPE_FILE_ICON 1 +#define DRFLAC_PICTURE_TYPE_OTHER_FILE_ICON 2 +#define DRFLAC_PICTURE_TYPE_COVER_FRONT 3 +#define DRFLAC_PICTURE_TYPE_COVER_BACK 4 +#define DRFLAC_PICTURE_TYPE_LEAFLET_PAGE 5 +#define DRFLAC_PICTURE_TYPE_MEDIA 6 +#define DRFLAC_PICTURE_TYPE_LEAD_ARTIST 7 +#define DRFLAC_PICTURE_TYPE_ARTIST 8 +#define DRFLAC_PICTURE_TYPE_CONDUCTOR 9 +#define DRFLAC_PICTURE_TYPE_BAND 10 +#define DRFLAC_PICTURE_TYPE_COMPOSER 11 +#define DRFLAC_PICTURE_TYPE_LYRICIST 12 +#define DRFLAC_PICTURE_TYPE_RECORDING_LOCATION 13 +#define DRFLAC_PICTURE_TYPE_DURING_RECORDING 14 +#define DRFLAC_PICTURE_TYPE_DURING_PERFORMANCE 15 +#define DRFLAC_PICTURE_TYPE_SCREEN_CAPTURE 16 +#define DRFLAC_PICTURE_TYPE_BRIGHT_COLORED_FISH 17 +#define DRFLAC_PICTURE_TYPE_ILLUSTRATION 18 +#define DRFLAC_PICTURE_TYPE_BAND_LOGOTYPE 19 +#define DRFLAC_PICTURE_TYPE_PUBLISHER_LOGOTYPE 20 + +typedef enum +{ + drflac_container_native, + drflac_container_ogg, + drflac_container_unknown +} drflac_container; + +typedef enum +{ + DRFLAC_SEEK_SET, + DRFLAC_SEEK_CUR, + DRFLAC_SEEK_END +} drflac_seek_origin; + +/* The order of members in this structure is important because we map this directly to the raw data within the SEEKTABLE metadata block. */ +typedef struct +{ + drflac_uint64 firstPCMFrame; + drflac_uint64 flacFrameOffset; /* The offset from the first byte of the header of the first frame. */ + drflac_uint16 pcmFrameCount; +} drflac_seekpoint; + +typedef struct +{ + drflac_uint16 minBlockSizeInPCMFrames; + drflac_uint16 maxBlockSizeInPCMFrames; + drflac_uint32 minFrameSizeInPCMFrames; + drflac_uint32 maxFrameSizeInPCMFrames; + drflac_uint32 sampleRate; + drflac_uint8 channels; + drflac_uint8 bitsPerSample; + drflac_uint64 totalPCMFrameCount; + drflac_uint8 md5[16]; +} drflac_streaminfo; + +typedef struct +{ + /* + The metadata type. Use this to know how to interpret the data below. Will be set to one of the + DRFLAC_METADATA_BLOCK_TYPE_* tokens. + */ + drflac_uint32 type; + + /* The size in bytes of the block and the buffer pointed to by pRawData if it's non-NULL. */ + drflac_uint32 rawDataSize; + + /* The offset in the stream of the raw data. */ + drflac_uint64 rawDataOffset; + + /* + A pointer to the raw data. This points to a temporary buffer so don't hold on to it. It's best to + not modify the contents of this buffer. Use the structures below for more meaningful and structured + information about the metadata. It's possible for this to be null. + */ + const void* pRawData; + + union + { + drflac_streaminfo streaminfo; + + struct + { + int unused; + } padding; + + struct + { + drflac_uint32 id; + const void* pData; + drflac_uint32 dataSize; + } application; + + struct + { + drflac_uint32 seekpointCount; + const drflac_seekpoint* pSeekpoints; + } seektable; + + struct + { + drflac_uint32 vendorLength; + const char* vendor; + drflac_uint32 commentCount; + const void* pComments; + } vorbis_comment; + + struct + { + char catalog[128]; + drflac_uint64 leadInSampleCount; + drflac_bool32 isCD; + drflac_uint8 trackCount; + const void* pTrackData; + } cuesheet; + + struct + { + drflac_uint32 type; + drflac_uint32 mimeLength; + const char* mime; + drflac_uint32 descriptionLength; + const char* description; + drflac_uint32 width; + drflac_uint32 height; + drflac_uint32 colorDepth; + drflac_uint32 indexColorCount; + drflac_uint32 pictureDataSize; + drflac_uint64 pictureDataOffset; /* Offset from the start of the stream. */ + const drflac_uint8* pPictureData; + } picture; + } data; +} drflac_metadata; + + +/* +Callback for when data needs to be read from the client. + + +Parameters +---------- +pUserData (in) + The user data that was passed to drflac_open() and family. + +pBufferOut (out) + The output buffer. + +bytesToRead (in) + The number of bytes to read. + + +Return Value +------------ +The number of bytes actually read. + + +Remarks +------- +A return value of less than bytesToRead indicates the end of the stream. Do _not_ return from this callback until either the entire bytesToRead is filled or +you have reached the end of the stream. +*/ +typedef size_t (* drflac_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead); + +/* +Callback for when data needs to be seeked. + + +Parameters +---------- +pUserData (in) + The user data that was passed to drflac_open() and family. + +offset (in) + The number of bytes to move, relative to the origin. Will never be negative. + +origin (in) + The origin of the seek - the current position, the start of the stream, or the end of the stream. + + +Return Value +------------ +Whether or not the seek was successful. + + +Remarks +------- +Seeking relative to the start and the current position must always be supported. If seeking from the end of the stream is not supported, return DRFLAC_FALSE. + +When seeking to a PCM frame using drflac_seek_to_pcm_frame(), dr_flac may call this with an offset beyond the end of the FLAC stream. This needs to be detected +and handled by returning DRFLAC_FALSE. +*/ +typedef drflac_bool32 (* drflac_seek_proc)(void* pUserData, int offset, drflac_seek_origin origin); + +/* +Callback for when the current position in the stream needs to be retrieved. + + +Parameters +---------- +pUserData (in) + The user data that was passed to drflac_open() and family. + +pCursor (out) + A pointer to a variable to receive the current position in the stream. + + +Return Value +------------ +Whether or not the operation was successful. +*/ +typedef drflac_bool32 (* drflac_tell_proc)(void* pUserData, drflac_int64* pCursor); + +/* +Callback for when a metadata block is read. + + +Parameters +---------- +pUserData (in) + The user data that was passed to drflac_open() and family. + +pMetadata (in) + A pointer to a structure containing the data of the metadata block. + + +Remarks +------- +Use pMetadata->type to determine which metadata block is being handled and how to read the data. This +will be set to one of the DRFLAC_METADATA_BLOCK_TYPE_* tokens. +*/ +typedef void (* drflac_meta_proc)(void* pUserData, drflac_metadata* pMetadata); + + +/* Structure for internal use. Only used for decoders opened with drflac_open_memory. */ +typedef struct +{ + const drflac_uint8* data; + size_t dataSize; + size_t currentReadPos; +} drflac__memory_stream; + +/* Structure for internal use. Used for bit streaming. */ +typedef struct +{ + /* The function to call when more data needs to be read. */ + drflac_read_proc onRead; + + /* The function to call when the current read position needs to be moved. */ + drflac_seek_proc onSeek; + + /* The function to call when the current read position needs to be retrieved. */ + drflac_tell_proc onTell; + + /* The user data to pass around to onRead and onSeek. */ + void* pUserData; + + + /* + The number of unaligned bytes in the L2 cache. This will always be 0 until the end of the stream is hit. At the end of the + stream there will be a number of bytes that don't cleanly fit in an L1 cache line, so we use this variable to know whether + or not the bistreamer needs to run on a slower path to read those last bytes. This will never be more than sizeof(drflac_cache_t). + */ + size_t unalignedByteCount; + + /* The content of the unaligned bytes. */ + drflac_cache_t unalignedCache; + + /* The index of the next valid cache line in the "L2" cache. */ + drflac_uint32 nextL2Line; + + /* The number of bits that have been consumed by the cache. This is used to determine how many valid bits are remaining. */ + drflac_uint32 consumedBits; + + /* + The cached data which was most recently read from the client. There are two levels of cache. Data flows as such: + Client -> L2 -> L1. The L2 -> L1 movement is aligned and runs on a fast path in just a few instructions. + */ + drflac_cache_t cacheL2[DR_FLAC_BUFFER_SIZE/sizeof(drflac_cache_t)]; + drflac_cache_t cache; + + /* + CRC-16. This is updated whenever bits are read from the bit stream. Manually set this to 0 to reset the CRC. For FLAC, this + is reset to 0 at the beginning of each frame. + */ + drflac_uint16 crc16; + drflac_cache_t crc16Cache; /* A cache for optimizing CRC calculations. This is filled when when the L1 cache is reloaded. */ + drflac_uint32 crc16CacheIgnoredBytes; /* The number of bytes to ignore when updating the CRC-16 from the CRC-16 cache. */ +} drflac_bs; + +typedef struct +{ + /* The type of the subframe: SUBFRAME_CONSTANT, SUBFRAME_VERBATIM, SUBFRAME_FIXED or SUBFRAME_LPC. */ + drflac_uint8 subframeType; + + /* The number of wasted bits per sample as specified by the sub-frame header. */ + drflac_uint8 wastedBitsPerSample; + + /* The order to use for the prediction stage for SUBFRAME_FIXED and SUBFRAME_LPC. */ + drflac_uint8 lpcOrder; + + /* A pointer to the buffer containing the decoded samples in the subframe. This pointer is an offset from drflac::pExtraData. */ + drflac_int32* pSamplesS32; +} drflac_subframe; + +typedef struct +{ + /* + If the stream uses variable block sizes, this will be set to the index of the first PCM frame. If fixed block sizes are used, this will + always be set to 0. This is 64-bit because the decoded PCM frame number will be 36 bits. + */ + drflac_uint64 pcmFrameNumber; + + /* + If the stream uses fixed block sizes, this will be set to the frame number. If variable block sizes are used, this will always be 0. This + is 32-bit because in fixed block sizes, the maximum frame number will be 31 bits. + */ + drflac_uint32 flacFrameNumber; + + /* The sample rate of this frame. */ + drflac_uint32 sampleRate; + + /* The number of PCM frames in each sub-frame within this frame. */ + drflac_uint16 blockSizeInPCMFrames; + + /* + The channel assignment of this frame. This is not always set to the channel count. If interchannel decorrelation is being used this + will be set to DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE, DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE or DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE. + */ + drflac_uint8 channelAssignment; + + /* The number of bits per sample within this frame. */ + drflac_uint8 bitsPerSample; + + /* The frame's CRC. */ + drflac_uint8 crc8; +} drflac_frame_header; + +typedef struct +{ + /* The header. */ + drflac_frame_header header; + + /* + The number of PCM frames left to be read in this FLAC frame. This is initially set to the block size. As PCM frames are read, + this will be decremented. When it reaches 0, the decoder will see this frame as fully consumed and load the next frame. + */ + drflac_uint32 pcmFramesRemaining; + + /* The list of sub-frames within the frame. There is one sub-frame for each channel, and there's a maximum of 8 channels. */ + drflac_subframe subframes[8]; +} drflac_frame; + +typedef struct +{ + /* The function to call when a metadata block is read. */ + drflac_meta_proc onMeta; + + /* The user data posted to the metadata callback function. */ + void* pUserDataMD; + + /* Memory allocation callbacks. */ + drflac_allocation_callbacks allocationCallbacks; + + + /* The sample rate. Will be set to something like 44100. */ + drflac_uint32 sampleRate; + + /* + The number of channels. This will be set to 1 for monaural streams, 2 for stereo, etc. Maximum 8. This is set based on the + value specified in the STREAMINFO block. + */ + drflac_uint8 channels; + + /* The bits per sample. Will be set to something like 16, 24, etc. */ + drflac_uint8 bitsPerSample; + + /* The maximum block size, in samples. This number represents the number of samples in each channel (not combined). */ + drflac_uint16 maxBlockSizeInPCMFrames; + + /* + The total number of PCM Frames making up the stream. Can be 0 in which case it's still a valid stream, but just means + the total PCM frame count is unknown. Likely the case with streams like internet radio. + */ + drflac_uint64 totalPCMFrameCount; + + + /* The container type. This is set based on whether or not the decoder was opened from a native or Ogg stream. */ + drflac_container container; + + /* The number of seekpoints in the seektable. */ + drflac_uint32 seekpointCount; + + + /* Information about the frame the decoder is currently sitting on. */ + drflac_frame currentFLACFrame; + + + /* The index of the PCM frame the decoder is currently sitting on. This is only used for seeking. */ + drflac_uint64 currentPCMFrame; + + /* The position of the first FLAC frame in the stream. This is only ever used for seeking. */ + drflac_uint64 firstFLACFramePosInBytes; + + + /* A hack to avoid a malloc() when opening a decoder with drflac_open_memory(). */ + drflac__memory_stream memoryStream; + + + /* A pointer to the decoded sample data. This is an offset of pExtraData. */ + drflac_int32* pDecodedSamples; + + /* A pointer to the seek table. This is an offset of pExtraData, or NULL if there is no seek table. */ + drflac_seekpoint* pSeekpoints; + + /* Internal use only. Only used with Ogg containers. Points to a drflac_oggbs object. This is an offset of pExtraData. */ + void* _oggbs; + + /* Internal use only. Used for profiling and testing different seeking modes. */ + drflac_bool32 _noSeekTableSeek : 1; + drflac_bool32 _noBinarySearchSeek : 1; + drflac_bool32 _noBruteForceSeek : 1; + + /* The bit streamer. The raw FLAC data is fed through this object. */ + drflac_bs bs; + + /* Variable length extra data. We attach this to the end of the object so we can avoid unnecessary mallocs. */ + drflac_uint8 pExtraData[1]; +} drflac; + + +/* +Opens a FLAC decoder. + + +Parameters +---------- +onRead (in) + The function to call when data needs to be read from the client. + +onSeek (in) + The function to call when the read position of the client data needs to move. + +onTell (in) + The function to call when the read position of the client needs to be queried. + +pUserData (in, optional) + A pointer to application defined data that will be passed to onRead and onSeek. + +pAllocationCallbacks (in, optional) + A pointer to application defined callbacks for managing memory allocations. + + +Return Value +------------ +Returns a pointer to an object representing the decoder. + + +Remarks +------- +Close the decoder with `drflac_close()`. + +`pAllocationCallbacks` can be NULL in which case it will use `DRFLAC_MALLOC`, `DRFLAC_REALLOC` and `DRFLAC_FREE`. + +This function will automatically detect whether or not you are attempting to open a native or Ogg encapsulated FLAC, both of which should work seamlessly +without any manual intervention. Ogg encapsulation also works with multiplexed streams which basically means it can play FLAC encoded audio tracks in videos. + +This is the lowest level function for opening a FLAC stream. You can also use `drflac_open_file()` and `drflac_open_memory()` to open the stream from a file or +from a block of memory respectively. + +The STREAMINFO block must be present for this to succeed. Use `drflac_open_relaxed()` to open a FLAC stream where the header may not be present. + +Use `drflac_open_with_metadata()` if you need access to metadata. + + +Seek Also +--------- +drflac_open_file() +drflac_open_memory() +drflac_open_with_metadata() +drflac_close() +*/ +DRFLAC_API drflac* drflac_open(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_tell_proc onTell, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* +Opens a FLAC stream with relaxed validation of the header block. + + +Parameters +---------- +onRead (in) + The function to call when data needs to be read from the client. + +onSeek (in) + The function to call when the read position of the client data needs to move. + +onTell (in) + The function to call when the read position of the client needs to be queried. + +container (in) + Whether or not the FLAC stream is encapsulated using standard FLAC encapsulation or Ogg encapsulation. + +pUserData (in, optional) + A pointer to application defined data that will be passed to onRead and onSeek. + +pAllocationCallbacks (in, optional) + A pointer to application defined callbacks for managing memory allocations. + + +Return Value +------------ +A pointer to an object representing the decoder. + + +Remarks +------- +The same as drflac_open(), except attempts to open the stream even when a header block is not present. + +Because the header is not necessarily available, the caller must explicitly define the container (Native or Ogg). Do not set this to `drflac_container_unknown` +as that is for internal use only. + +Opening in relaxed mode will continue reading data from onRead until it finds a valid frame. If a frame is never found it will continue forever. To abort, +force your `onRead` callback to return 0, which dr_flac will use as an indicator that the end of the stream was found. + +Use `drflac_open_with_metadata_relaxed()` if you need access to metadata. +*/ +DRFLAC_API drflac* drflac_open_relaxed(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_tell_proc onTell, drflac_container container, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* +Opens a FLAC decoder and notifies the caller of the metadata chunks (album art, etc.). + + +Parameters +---------- +onRead (in) + The function to call when data needs to be read from the client. + +onSeek (in) + The function to call when the read position of the client data needs to move. + +onTell (in) + The function to call when the read position of the client needs to be queried. + +onMeta (in) + The function to call for every metadata block. + +pUserData (in, optional) + A pointer to application defined data that will be passed to onRead, onSeek and onMeta. + +pAllocationCallbacks (in, optional) + A pointer to application defined callbacks for managing memory allocations. + + +Return Value +------------ +A pointer to an object representing the decoder. + + +Remarks +------- +Close the decoder with `drflac_close()`. + +`pAllocationCallbacks` can be NULL in which case it will use `DRFLAC_MALLOC`, `DRFLAC_REALLOC` and `DRFLAC_FREE`. + +This is slower than `drflac_open()`, so avoid this one if you don't need metadata. Internally, this will allocate and free memory on the heap for every +metadata block except for STREAMINFO and PADDING blocks. + +The caller is notified of the metadata via the `onMeta` callback. All metadata blocks will be handled before the function returns. This callback takes a +pointer to a `drflac_metadata` object which is a union containing the data of all relevant metadata blocks. Use the `type` member to discriminate against +the different metadata types. + +The STREAMINFO block must be present for this to succeed. Use `drflac_open_with_metadata_relaxed()` to open a FLAC stream where the header may not be present. + +Note that this will behave inconsistently with `drflac_open()` if the stream is an Ogg encapsulated stream and a metadata block is corrupted. This is due to +the way the Ogg stream recovers from corrupted pages. When `drflac_open_with_metadata()` is being used, the open routine will try to read the contents of the +metadata block, whereas `drflac_open()` will simply seek past it (for the sake of efficiency). This inconsistency can result in different samples being +returned depending on whether or not the stream is being opened with metadata. + + +Seek Also +--------- +drflac_open_file_with_metadata() +drflac_open_memory_with_metadata() +drflac_open() +drflac_close() +*/ +DRFLAC_API drflac* drflac_open_with_metadata(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_tell_proc onTell, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* +The same as drflac_open_with_metadata(), except attempts to open the stream even when a header block is not present. + +See Also +-------- +drflac_open_with_metadata() +drflac_open_relaxed() +*/ +DRFLAC_API drflac* drflac_open_with_metadata_relaxed(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_tell_proc onTell, drflac_meta_proc onMeta, drflac_container container, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* +Closes the given FLAC decoder. + + +Parameters +---------- +pFlac (in) + The decoder to close. + + +Remarks +------- +This will destroy the decoder object. + + +See Also +-------- +drflac_open() +drflac_open_with_metadata() +drflac_open_file() +drflac_open_file_w() +drflac_open_file_with_metadata() +drflac_open_file_with_metadata_w() +drflac_open_memory() +drflac_open_memory_with_metadata() +*/ +DRFLAC_API void drflac_close(drflac* pFlac); + + +/* +Reads sample data from the given FLAC decoder, output as interleaved signed 32-bit PCM. + + +Parameters +---------- +pFlac (in) + The decoder. + +framesToRead (in) + The number of PCM frames to read. + +pBufferOut (out, optional) + A pointer to the buffer that will receive the decoded samples. + + +Return Value +------------ +Returns the number of PCM frames actually read. If the return value is less than `framesToRead` it has reached the end. + + +Remarks +------- +pBufferOut can be null, in which case the call will act as a seek, and the return value will be the number of frames seeked. +*/ +DRFLAC_API drflac_uint64 drflac_read_pcm_frames_s32(drflac* pFlac, drflac_uint64 framesToRead, drflac_int32* pBufferOut); + + +/* +Reads sample data from the given FLAC decoder, output as interleaved signed 16-bit PCM. + + +Parameters +---------- +pFlac (in) + The decoder. + +framesToRead (in) + The number of PCM frames to read. + +pBufferOut (out, optional) + A pointer to the buffer that will receive the decoded samples. + + +Return Value +------------ +Returns the number of PCM frames actually read. If the return value is less than `framesToRead` it has reached the end. + + +Remarks +------- +pBufferOut can be null, in which case the call will act as a seek, and the return value will be the number of frames seeked. + +Note that this is lossy for streams where the bits per sample is larger than 16. +*/ +DRFLAC_API drflac_uint64 drflac_read_pcm_frames_s16(drflac* pFlac, drflac_uint64 framesToRead, drflac_int16* pBufferOut); + +/* +Reads sample data from the given FLAC decoder, output as interleaved 32-bit floating point PCM. + + +Parameters +---------- +pFlac (in) + The decoder. + +framesToRead (in) + The number of PCM frames to read. + +pBufferOut (out, optional) + A pointer to the buffer that will receive the decoded samples. + + +Return Value +------------ +Returns the number of PCM frames actually read. If the return value is less than `framesToRead` it has reached the end. + + +Remarks +------- +pBufferOut can be null, in which case the call will act as a seek, and the return value will be the number of frames seeked. + +Note that this should be considered lossy due to the nature of floating point numbers not being able to exactly represent every possible number. +*/ +DRFLAC_API drflac_uint64 drflac_read_pcm_frames_f32(drflac* pFlac, drflac_uint64 framesToRead, float* pBufferOut); + +/* +Seeks to the PCM frame at the given index. + + +Parameters +---------- +pFlac (in) + The decoder. + +pcmFrameIndex (in) + The index of the PCM frame to seek to. See notes below. + + +Return Value +------------- +`DRFLAC_TRUE` if successful; `DRFLAC_FALSE` otherwise. +*/ +DRFLAC_API drflac_bool32 drflac_seek_to_pcm_frame(drflac* pFlac, drflac_uint64 pcmFrameIndex); + + + +#ifndef DR_FLAC_NO_STDIO +/* +Opens a FLAC decoder from the file at the given path. + + +Parameters +---------- +pFileName (in) + The path of the file to open, either absolute or relative to the current directory. + +pAllocationCallbacks (in, optional) + A pointer to application defined callbacks for managing memory allocations. + + +Return Value +------------ +A pointer to an object representing the decoder. + + +Remarks +------- +Close the decoder with drflac_close(). + + +Remarks +------- +This will hold a handle to the file until the decoder is closed with drflac_close(). Some platforms will restrict the number of files a process can have open +at any given time, so keep this mind if you have many decoders open at the same time. + + +See Also +-------- +drflac_open_file_with_metadata() +drflac_open() +drflac_close() +*/ +DRFLAC_API drflac* drflac_open_file(const char* pFileName, const drflac_allocation_callbacks* pAllocationCallbacks); +DRFLAC_API drflac* drflac_open_file_w(const wchar_t* pFileName, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* +Opens a FLAC decoder from the file at the given path and notifies the caller of the metadata chunks (album art, etc.) + + +Parameters +---------- +pFileName (in) + The path of the file to open, either absolute or relative to the current directory. + +pAllocationCallbacks (in, optional) + A pointer to application defined callbacks for managing memory allocations. + +onMeta (in) + The callback to fire for each metadata block. + +pUserData (in) + A pointer to the user data to pass to the metadata callback. + +pAllocationCallbacks (in) + A pointer to application defined callbacks for managing memory allocations. + + +Remarks +------- +Look at the documentation for drflac_open_with_metadata() for more information on how metadata is handled. + + +See Also +-------- +drflac_open_with_metadata() +drflac_open() +drflac_close() +*/ +DRFLAC_API drflac* drflac_open_file_with_metadata(const char* pFileName, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks); +DRFLAC_API drflac* drflac_open_file_with_metadata_w(const wchar_t* pFileName, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks); +#endif + +/* +Opens a FLAC decoder from a pre-allocated block of memory + + +Parameters +---------- +pData (in) + A pointer to the raw encoded FLAC data. + +dataSize (in) + The size in bytes of `data`. + +pAllocationCallbacks (in) + A pointer to application defined callbacks for managing memory allocations. + + +Return Value +------------ +A pointer to an object representing the decoder. + + +Remarks +------- +This does not create a copy of the data. It is up to the application to ensure the buffer remains valid for the lifetime of the decoder. + + +See Also +-------- +drflac_open() +drflac_close() +*/ +DRFLAC_API drflac* drflac_open_memory(const void* pData, size_t dataSize, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* +Opens a FLAC decoder from a pre-allocated block of memory and notifies the caller of the metadata chunks (album art, etc.) + + +Parameters +---------- +pData (in) + A pointer to the raw encoded FLAC data. + +dataSize (in) + The size in bytes of `data`. + +onMeta (in) + The callback to fire for each metadata block. + +pUserData (in) + A pointer to the user data to pass to the metadata callback. + +pAllocationCallbacks (in) + A pointer to application defined callbacks for managing memory allocations. + + +Remarks +------- +Look at the documentation for drflac_open_with_metadata() for more information on how metadata is handled. + + +See Also +------- +drflac_open_with_metadata() +drflac_open() +drflac_close() +*/ +DRFLAC_API drflac* drflac_open_memory_with_metadata(const void* pData, size_t dataSize, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks); + + + +/* High Level APIs */ + +/* +Opens a FLAC stream from the given callbacks and fully decodes it in a single operation. The return value is a +pointer to the sample data as interleaved signed 32-bit PCM. The returned data must be freed with drflac_free(). + +You can pass in custom memory allocation callbacks via the pAllocationCallbacks parameter. This can be NULL in which +case it will use DRFLAC_MALLOC, DRFLAC_REALLOC and DRFLAC_FREE. + +Sometimes a FLAC file won't keep track of the total sample count. In this situation the function will continuously +read samples into a dynamically sized buffer on the heap until no samples are left. + +Do not call this function on a broadcast type of stream (like internet radio streams and whatnot). +*/ +DRFLAC_API drflac_int32* drflac_open_and_read_pcm_frames_s32(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_tell_proc onTell, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* Same as drflac_open_and_read_pcm_frames_s32(), except returns signed 16-bit integer samples. */ +DRFLAC_API drflac_int16* drflac_open_and_read_pcm_frames_s16(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_tell_proc onTell, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* Same as drflac_open_and_read_pcm_frames_s32(), except returns 32-bit floating-point samples. */ +DRFLAC_API float* drflac_open_and_read_pcm_frames_f32(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_tell_proc onTell, void* pUserData, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); + +#ifndef DR_FLAC_NO_STDIO +/* Same as drflac_open_and_read_pcm_frames_s32() except opens the decoder from a file. */ +DRFLAC_API drflac_int32* drflac_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* Same as drflac_open_file_and_read_pcm_frames_s32(), except returns signed 16-bit integer samples. */ +DRFLAC_API drflac_int16* drflac_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* Same as drflac_open_file_and_read_pcm_frames_s32(), except returns 32-bit floating-point samples. */ +DRFLAC_API float* drflac_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); +#endif + +/* Same as drflac_open_and_read_pcm_frames_s32() except opens the decoder from a block of memory. */ +DRFLAC_API drflac_int32* drflac_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* Same as drflac_open_memory_and_read_pcm_frames_s32(), except returns signed 16-bit integer samples. */ +DRFLAC_API drflac_int16* drflac_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* Same as drflac_open_memory_and_read_pcm_frames_s32(), except returns 32-bit floating-point samples. */ +DRFLAC_API float* drflac_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks); + +/* +Frees memory that was allocated internally by dr_flac. + +Set pAllocationCallbacks to the same object that was passed to drflac_open_*_and_read_pcm_frames_*(). If you originally passed in NULL, pass in NULL for this. +*/ +DRFLAC_API void drflac_free(void* p, const drflac_allocation_callbacks* pAllocationCallbacks); + + +/* Structure representing an iterator for vorbis comments in a VORBIS_COMMENT metadata block. */ +typedef struct +{ + drflac_uint32 countRemaining; + const char* pRunningData; +} drflac_vorbis_comment_iterator; + +/* +Initializes a vorbis comment iterator. This can be used for iterating over the vorbis comments in a VORBIS_COMMENT +metadata block. +*/ +DRFLAC_API void drflac_init_vorbis_comment_iterator(drflac_vorbis_comment_iterator* pIter, drflac_uint32 commentCount, const void* pComments); + +/* +Goes to the next vorbis comment in the given iterator. If null is returned it means there are no more comments. The +returned string is NOT null terminated. +*/ +DRFLAC_API const char* drflac_next_vorbis_comment(drflac_vorbis_comment_iterator* pIter, drflac_uint32* pCommentLengthOut); + + +/* Structure representing an iterator for cuesheet tracks in a CUESHEET metadata block. */ +typedef struct +{ + drflac_uint32 countRemaining; + const char* pRunningData; +} drflac_cuesheet_track_iterator; + +/* The order of members here is important because we map this directly to the raw data within the CUESHEET metadata block. */ +typedef struct +{ + drflac_uint64 offset; + drflac_uint8 index; + drflac_uint8 reserved[3]; +} drflac_cuesheet_track_index; + +typedef struct +{ + drflac_uint64 offset; + drflac_uint8 trackNumber; + char ISRC[12]; + drflac_bool8 isAudio; + drflac_bool8 preEmphasis; + drflac_uint8 indexCount; + const drflac_cuesheet_track_index* pIndexPoints; +} drflac_cuesheet_track; + +/* +Initializes a cuesheet track iterator. This can be used for iterating over the cuesheet tracks in a CUESHEET metadata +block. +*/ +DRFLAC_API void drflac_init_cuesheet_track_iterator(drflac_cuesheet_track_iterator* pIter, drflac_uint32 trackCount, const void* pTrackData); + +/* Goes to the next cuesheet track in the given iterator. If DRFLAC_FALSE is returned it means there are no more comments. */ +DRFLAC_API drflac_bool32 drflac_next_cuesheet_track(drflac_cuesheet_track_iterator* pIter, drflac_cuesheet_track* pCuesheetTrack); + + +#ifdef __cplusplus +} +#endif +#endif /* dr_flac_h */ + + +/************************************************************************************************************************************************************ + ************************************************************************************************************************************************************ + + IMPLEMENTATION + + ************************************************************************************************************************************************************ + ************************************************************************************************************************************************************/ +#if defined(DR_FLAC_IMPLEMENTATION) || defined(DRFLAC_IMPLEMENTATION) +#ifndef dr_flac_c +#define dr_flac_c + +/* Disable some annoying warnings. */ +#if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic push + #if __GNUC__ >= 7 + #pragma GCC diagnostic ignored "-Wimplicit-fallthrough" + #endif +#endif + +#ifdef __linux__ + #ifndef _BSD_SOURCE + #define _BSD_SOURCE + #endif + #ifndef _DEFAULT_SOURCE + #define _DEFAULT_SOURCE + #endif + #ifndef __USE_BSD + #define __USE_BSD + #endif + #include +#endif + +#include +#include + +/* Inline */ +#ifdef _MSC_VER + #define DRFLAC_INLINE __forceinline +#elif defined(__GNUC__) + /* + I've had a bug report where GCC is emitting warnings about functions possibly not being inlineable. This warning happens when + the __attribute__((always_inline)) attribute is defined without an "inline" statement. I think therefore there must be some + case where "__inline__" is not always defined, thus the compiler emitting these warnings. When using -std=c89 or -ansi on the + command line, we cannot use the "inline" keyword and instead need to use "__inline__". In an attempt to work around this issue + I am using "__inline__" only when we're compiling in strict ANSI mode. + */ + #if defined(__STRICT_ANSI__) + #define DRFLAC_GNUC_INLINE_HINT __inline__ + #else + #define DRFLAC_GNUC_INLINE_HINT inline + #endif + + #if (__GNUC__ > 3 || (__GNUC__ == 3 && __GNUC_MINOR__ >= 2)) || defined(__clang__) + #define DRFLAC_INLINE DRFLAC_GNUC_INLINE_HINT __attribute__((always_inline)) + #else + #define DRFLAC_INLINE DRFLAC_GNUC_INLINE_HINT + #endif +#elif defined(__WATCOMC__) + #define DRFLAC_INLINE __inline +#else + #define DRFLAC_INLINE +#endif +/* End Inline */ + +/* +Intrinsics Support + +There's a bug in GCC 4.2.x which results in an incorrect compilation error when using _mm_slli_epi32() where it complains with + + "error: shift must be an immediate" + +Unfortuantely dr_flac depends on this for a few things so we're just going to disable SSE on GCC 4.2 and below. +*/ +#if !defined(DR_FLAC_NO_SIMD) + #if defined(DRFLAC_X64) || defined(DRFLAC_X86) + #if defined(_MSC_VER) && !defined(__clang__) + /* MSVC. */ + #if _MSC_VER >= 1400 && !defined(DRFLAC_NO_SSE2) /* 2005 */ + #define DRFLAC_SUPPORT_SSE2 + #endif + #if _MSC_VER >= 1600 && !defined(DRFLAC_NO_SSE41) /* 2010 */ + #define DRFLAC_SUPPORT_SSE41 + #endif + #elif defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 3))) + /* Assume GNUC-style. */ + #if defined(__SSE2__) && !defined(DRFLAC_NO_SSE2) + #define DRFLAC_SUPPORT_SSE2 + #endif + #if defined(__SSE4_1__) && !defined(DRFLAC_NO_SSE41) + #define DRFLAC_SUPPORT_SSE41 + #endif + #endif + + /* If at this point we still haven't determined compiler support for the intrinsics just fall back to __has_include. */ + #if !defined(__GNUC__) && !defined(__clang__) && defined(__has_include) + #if !defined(DRFLAC_SUPPORT_SSE2) && !defined(DRFLAC_NO_SSE2) && __has_include() + #define DRFLAC_SUPPORT_SSE2 + #endif + #if !defined(DRFLAC_SUPPORT_SSE41) && !defined(DRFLAC_NO_SSE41) && __has_include() + #define DRFLAC_SUPPORT_SSE41 + #endif + #endif + + #if defined(DRFLAC_SUPPORT_SSE41) + #include + #elif defined(DRFLAC_SUPPORT_SSE2) + #include + #endif + #endif + + #if defined(DRFLAC_ARM) + #if !defined(DRFLAC_NO_NEON) && (defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64)) + #define DRFLAC_SUPPORT_NEON + #include + #endif + #endif +#endif + +/* Compile-time CPU feature support. */ +#if !defined(DR_FLAC_NO_SIMD) && (defined(DRFLAC_X86) || defined(DRFLAC_X64)) + #if defined(_MSC_VER) && !defined(__clang__) + #if _MSC_VER >= 1400 + #include + static void drflac__cpuid(int info[4], int fid) + { + __cpuid(info, fid); + } + #else + #define DRFLAC_NO_CPUID + #endif + #else + #if defined(__GNUC__) || defined(__clang__) + static void drflac__cpuid(int info[4], int fid) + { + /* + It looks like the -fPIC option uses the ebx register which GCC complains about. We can work around this by just using a different register, the + specific register of which I'm letting the compiler decide on. The "k" prefix is used to specify a 32-bit register. The {...} syntax is for + supporting different assembly dialects. + + What's basically happening is that we're saving and restoring the ebx register manually. + */ + #if defined(DRFLAC_X86) && defined(__PIC__) + __asm__ __volatile__ ( + "xchg{l} {%%}ebx, %k1;" + "cpuid;" + "xchg{l} {%%}ebx, %k1;" + : "=a"(info[0]), "=&r"(info[1]), "=c"(info[2]), "=d"(info[3]) : "a"(fid), "c"(0) + ); + #else + __asm__ __volatile__ ( + "cpuid" : "=a"(info[0]), "=b"(info[1]), "=c"(info[2]), "=d"(info[3]) : "a"(fid), "c"(0) + ); + #endif + } + #else + #define DRFLAC_NO_CPUID + #endif + #endif +#else + #define DRFLAC_NO_CPUID +#endif + +static DRFLAC_INLINE drflac_bool32 drflac_has_sse2(void) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + #if (defined(DRFLAC_X64) || defined(DRFLAC_X86)) && !defined(DRFLAC_NO_SSE2) + #if defined(DRFLAC_X64) + return DRFLAC_TRUE; /* 64-bit targets always support SSE2. */ + #elif (defined(_M_IX86_FP) && _M_IX86_FP == 2) || defined(__SSE2__) + return DRFLAC_TRUE; /* If the compiler is allowed to freely generate SSE2 code we can assume support. */ + #else + #if defined(DRFLAC_NO_CPUID) + return DRFLAC_FALSE; + #else + int info[4]; + drflac__cpuid(info, 1); + return (info[3] & (1 << 26)) != 0; + #endif + #endif + #else + return DRFLAC_FALSE; /* SSE2 is only supported on x86 and x64 architectures. */ + #endif +#else + return DRFLAC_FALSE; /* No compiler support. */ +#endif +} + +static DRFLAC_INLINE drflac_bool32 drflac_has_sse41(void) +{ +#if defined(DRFLAC_SUPPORT_SSE41) + #if (defined(DRFLAC_X64) || defined(DRFLAC_X86)) && !defined(DRFLAC_NO_SSE41) + #if defined(__SSE4_1__) || defined(__AVX__) + return DRFLAC_TRUE; /* If the compiler is allowed to freely generate SSE41 code we can assume support. */ + #else + #if defined(DRFLAC_NO_CPUID) + return DRFLAC_FALSE; + #else + int info[4]; + drflac__cpuid(info, 1); + return (info[2] & (1 << 19)) != 0; + #endif + #endif + #else + return DRFLAC_FALSE; /* SSE41 is only supported on x86 and x64 architectures. */ + #endif +#else + return DRFLAC_FALSE; /* No compiler support. */ +#endif +} + + +#if defined(_MSC_VER) && _MSC_VER >= 1500 && (defined(DRFLAC_X86) || defined(DRFLAC_X64)) && !defined(__clang__) + #define DRFLAC_HAS_LZCNT_INTRINSIC +#elif (defined(__GNUC__) && ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 7))) + #define DRFLAC_HAS_LZCNT_INTRINSIC +#elif defined(__clang__) + #if defined(__has_builtin) + #if __has_builtin(__builtin_clzll) || __has_builtin(__builtin_clzl) + #define DRFLAC_HAS_LZCNT_INTRINSIC + #endif + #endif +#endif + +#if defined(_MSC_VER) && _MSC_VER >= 1400 && !defined(__clang__) + #define DRFLAC_HAS_BYTESWAP16_INTRINSIC + #define DRFLAC_HAS_BYTESWAP32_INTRINSIC + #define DRFLAC_HAS_BYTESWAP64_INTRINSIC +#elif defined(__clang__) + #if defined(__has_builtin) + #if __has_builtin(__builtin_bswap16) + #define DRFLAC_HAS_BYTESWAP16_INTRINSIC + #endif + #if __has_builtin(__builtin_bswap32) + #define DRFLAC_HAS_BYTESWAP32_INTRINSIC + #endif + #if __has_builtin(__builtin_bswap64) + #define DRFLAC_HAS_BYTESWAP64_INTRINSIC + #endif + #endif +#elif defined(__GNUC__) + #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 3)) + #define DRFLAC_HAS_BYTESWAP32_INTRINSIC + #define DRFLAC_HAS_BYTESWAP64_INTRINSIC + #endif + #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8)) + #define DRFLAC_HAS_BYTESWAP16_INTRINSIC + #endif +#elif defined(__WATCOMC__) && defined(__386__) + #define DRFLAC_HAS_BYTESWAP16_INTRINSIC + #define DRFLAC_HAS_BYTESWAP32_INTRINSIC + #define DRFLAC_HAS_BYTESWAP64_INTRINSIC + extern __inline drflac_uint16 _watcom_bswap16(drflac_uint16); + extern __inline drflac_uint32 _watcom_bswap32(drflac_uint32); + extern __inline drflac_uint64 _watcom_bswap64(drflac_uint64); +#pragma aux _watcom_bswap16 = \ + "xchg al, ah" \ + parm [ax] \ + value [ax] \ + modify nomemory; +#pragma aux _watcom_bswap32 = \ + "bswap eax" \ + parm [eax] \ + value [eax] \ + modify nomemory; +#pragma aux _watcom_bswap64 = \ + "bswap eax" \ + "bswap edx" \ + "xchg eax,edx" \ + parm [eax edx] \ + value [eax edx] \ + modify nomemory; +#endif + + +/* Standard library stuff. */ +#ifndef DRFLAC_ASSERT +#include +#define DRFLAC_ASSERT(expression) assert(expression) +#endif +#ifndef DRFLAC_MALLOC +#define DRFLAC_MALLOC(sz) malloc((sz)) +#endif +#ifndef DRFLAC_REALLOC +#define DRFLAC_REALLOC(p, sz) realloc((p), (sz)) +#endif +#ifndef DRFLAC_FREE +#define DRFLAC_FREE(p) free((p)) +#endif +#ifndef DRFLAC_COPY_MEMORY +#define DRFLAC_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz)) +#endif +#ifndef DRFLAC_ZERO_MEMORY +#define DRFLAC_ZERO_MEMORY(p, sz) memset((p), 0, (sz)) +#endif +#ifndef DRFLAC_ZERO_OBJECT +#define DRFLAC_ZERO_OBJECT(p) DRFLAC_ZERO_MEMORY((p), sizeof(*(p))) +#endif + +#define DRFLAC_MIN(a, b) (((a) < (b)) ? (a) : (b)) + +#define DRFLAC_MAX_SIMD_VECTOR_SIZE 64 /* 64 for AVX-512 in the future. */ + +/* Result Codes */ +typedef drflac_int32 drflac_result; +#define DRFLAC_SUCCESS 0 +#define DRFLAC_ERROR -1 /* A generic error. */ +#define DRFLAC_INVALID_ARGS -2 +#define DRFLAC_INVALID_OPERATION -3 +#define DRFLAC_OUT_OF_MEMORY -4 +#define DRFLAC_OUT_OF_RANGE -5 +#define DRFLAC_ACCESS_DENIED -6 +#define DRFLAC_DOES_NOT_EXIST -7 +#define DRFLAC_ALREADY_EXISTS -8 +#define DRFLAC_TOO_MANY_OPEN_FILES -9 +#define DRFLAC_INVALID_FILE -10 +#define DRFLAC_TOO_BIG -11 +#define DRFLAC_PATH_TOO_LONG -12 +#define DRFLAC_NAME_TOO_LONG -13 +#define DRFLAC_NOT_DIRECTORY -14 +#define DRFLAC_IS_DIRECTORY -15 +#define DRFLAC_DIRECTORY_NOT_EMPTY -16 +#define DRFLAC_END_OF_FILE -17 +#define DRFLAC_NO_SPACE -18 +#define DRFLAC_BUSY -19 +#define DRFLAC_IO_ERROR -20 +#define DRFLAC_INTERRUPT -21 +#define DRFLAC_UNAVAILABLE -22 +#define DRFLAC_ALREADY_IN_USE -23 +#define DRFLAC_BAD_ADDRESS -24 +#define DRFLAC_BAD_SEEK -25 +#define DRFLAC_BAD_PIPE -26 +#define DRFLAC_DEADLOCK -27 +#define DRFLAC_TOO_MANY_LINKS -28 +#define DRFLAC_NOT_IMPLEMENTED -29 +#define DRFLAC_NO_MESSAGE -30 +#define DRFLAC_BAD_MESSAGE -31 +#define DRFLAC_NO_DATA_AVAILABLE -32 +#define DRFLAC_INVALID_DATA -33 +#define DRFLAC_TIMEOUT -34 +#define DRFLAC_NO_NETWORK -35 +#define DRFLAC_NOT_UNIQUE -36 +#define DRFLAC_NOT_SOCKET -37 +#define DRFLAC_NO_ADDRESS -38 +#define DRFLAC_BAD_PROTOCOL -39 +#define DRFLAC_PROTOCOL_UNAVAILABLE -40 +#define DRFLAC_PROTOCOL_NOT_SUPPORTED -41 +#define DRFLAC_PROTOCOL_FAMILY_NOT_SUPPORTED -42 +#define DRFLAC_ADDRESS_FAMILY_NOT_SUPPORTED -43 +#define DRFLAC_SOCKET_NOT_SUPPORTED -44 +#define DRFLAC_CONNECTION_RESET -45 +#define DRFLAC_ALREADY_CONNECTED -46 +#define DRFLAC_NOT_CONNECTED -47 +#define DRFLAC_CONNECTION_REFUSED -48 +#define DRFLAC_NO_HOST -49 +#define DRFLAC_IN_PROGRESS -50 +#define DRFLAC_CANCELLED -51 +#define DRFLAC_MEMORY_ALREADY_MAPPED -52 +#define DRFLAC_AT_END -53 + +#define DRFLAC_CRC_MISMATCH -100 +/* End Result Codes */ + + +#define DRFLAC_SUBFRAME_CONSTANT 0 +#define DRFLAC_SUBFRAME_VERBATIM 1 +#define DRFLAC_SUBFRAME_FIXED 8 +#define DRFLAC_SUBFRAME_LPC 32 +#define DRFLAC_SUBFRAME_RESERVED 255 + +#define DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE 0 +#define DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE2 1 + +#define DRFLAC_CHANNEL_ASSIGNMENT_INDEPENDENT 0 +#define DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE 8 +#define DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE 9 +#define DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE 10 + +#define DRFLAC_SEEKPOINT_SIZE_IN_BYTES 18 +#define DRFLAC_CUESHEET_TRACK_SIZE_IN_BYTES 36 +#define DRFLAC_CUESHEET_TRACK_INDEX_SIZE_IN_BYTES 12 + +#define drflac_align(x, a) ((((x) + (a) - 1) / (a)) * (a)) + + +DRFLAC_API void drflac_version(drflac_uint32* pMajor, drflac_uint32* pMinor, drflac_uint32* pRevision) +{ + if (pMajor) { + *pMajor = DRFLAC_VERSION_MAJOR; + } + + if (pMinor) { + *pMinor = DRFLAC_VERSION_MINOR; + } + + if (pRevision) { + *pRevision = DRFLAC_VERSION_REVISION; + } +} + +DRFLAC_API const char* drflac_version_string(void) +{ + return DRFLAC_VERSION_STRING; +} + + +/* CPU caps. */ +#if defined(__has_feature) + #if __has_feature(thread_sanitizer) + #define DRFLAC_NO_THREAD_SANITIZE __attribute__((no_sanitize("thread"))) + #else + #define DRFLAC_NO_THREAD_SANITIZE + #endif +#else + #define DRFLAC_NO_THREAD_SANITIZE +#endif + +#if defined(DRFLAC_HAS_LZCNT_INTRINSIC) +static drflac_bool32 drflac__gIsLZCNTSupported = DRFLAC_FALSE; +#endif + +#ifndef DRFLAC_NO_CPUID +static drflac_bool32 drflac__gIsSSE2Supported = DRFLAC_FALSE; +static drflac_bool32 drflac__gIsSSE41Supported = DRFLAC_FALSE; + +/* +I've had a bug report that Clang's ThreadSanitizer presents a warning in this function. Having reviewed this, this does +actually make sense. However, since CPU caps should never differ for a running process, I don't think the trade off of +complicating internal API's by passing around CPU caps versus just disabling the warnings is worthwhile. I'm therefore +just going to disable these warnings. This is disabled via the DRFLAC_NO_THREAD_SANITIZE attribute. +*/ +DRFLAC_NO_THREAD_SANITIZE static void drflac__init_cpu_caps(void) +{ + static drflac_bool32 isCPUCapsInitialized = DRFLAC_FALSE; + + if (!isCPUCapsInitialized) { + /* LZCNT */ +#if defined(DRFLAC_HAS_LZCNT_INTRINSIC) + int info[4] = {0}; + drflac__cpuid(info, 0x80000001); + drflac__gIsLZCNTSupported = (info[2] & (1 << 5)) != 0; +#endif + + /* SSE2 */ + drflac__gIsSSE2Supported = drflac_has_sse2(); + + /* SSE4.1 */ + drflac__gIsSSE41Supported = drflac_has_sse41(); + + /* Initialized. */ + isCPUCapsInitialized = DRFLAC_TRUE; + } +} +#else +static drflac_bool32 drflac__gIsNEONSupported = DRFLAC_FALSE; + +static DRFLAC_INLINE drflac_bool32 drflac__has_neon(void) +{ +#if defined(DRFLAC_SUPPORT_NEON) + #if defined(DRFLAC_ARM) && !defined(DRFLAC_NO_NEON) + #if (defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64)) + return DRFLAC_TRUE; /* If the compiler is allowed to freely generate NEON code we can assume support. */ + #else + /* TODO: Runtime check. */ + return DRFLAC_FALSE; + #endif + #else + return DRFLAC_FALSE; /* NEON is only supported on ARM architectures. */ + #endif +#else + return DRFLAC_FALSE; /* No compiler support. */ +#endif +} + +DRFLAC_NO_THREAD_SANITIZE static void drflac__init_cpu_caps(void) +{ + drflac__gIsNEONSupported = drflac__has_neon(); + +#if defined(DRFLAC_HAS_LZCNT_INTRINSIC) && defined(DRFLAC_ARM) && (defined(__ARM_ARCH) && __ARM_ARCH >= 5) + drflac__gIsLZCNTSupported = DRFLAC_TRUE; +#endif +} +#endif + + +/* Endian Management */ +static DRFLAC_INLINE drflac_bool32 drflac__is_little_endian(void) +{ +#if defined(DRFLAC_X86) || defined(DRFLAC_X64) + return DRFLAC_TRUE; +#elif defined(__BYTE_ORDER) && defined(__LITTLE_ENDIAN) && __BYTE_ORDER == __LITTLE_ENDIAN + return DRFLAC_TRUE; +#else + int n = 1; + return (*(char*)&n) == 1; +#endif +} + +static DRFLAC_INLINE drflac_uint16 drflac__swap_endian_uint16(drflac_uint16 n) +{ +#ifdef DRFLAC_HAS_BYTESWAP16_INTRINSIC + #if defined(_MSC_VER) && !defined(__clang__) + return _byteswap_ushort(n); + #elif defined(__GNUC__) || defined(__clang__) + return __builtin_bswap16(n); + #elif defined(__WATCOMC__) && defined(__386__) + return _watcom_bswap16(n); + #else + #error "This compiler does not support the byte swap intrinsic." + #endif +#else + return ((n & 0xFF00) >> 8) | + ((n & 0x00FF) << 8); +#endif +} + +static DRFLAC_INLINE drflac_uint32 drflac__swap_endian_uint32(drflac_uint32 n) +{ +#ifdef DRFLAC_HAS_BYTESWAP32_INTRINSIC + #if defined(_MSC_VER) && !defined(__clang__) + return _byteswap_ulong(n); + #elif defined(__GNUC__) || defined(__clang__) + #if defined(DRFLAC_ARM) && (defined(__ARM_ARCH) && __ARM_ARCH >= 6) && !defined(__ARM_ARCH_6M__) && !defined(DRFLAC_64BIT) /* <-- 64-bit inline assembly has not been tested, so disabling for now. */ + /* Inline assembly optimized implementation for ARM. In my testing, GCC does not generate optimized code with __builtin_bswap32(). */ + drflac_uint32 r; + __asm__ __volatile__ ( + #if defined(DRFLAC_64BIT) + "rev %w[out], %w[in]" : [out]"=r"(r) : [in]"r"(n) /* <-- This is untested. If someone in the community could test this, that would be appreciated! */ + #else + "rev %[out], %[in]" : [out]"=r"(r) : [in]"r"(n) + #endif + ); + return r; + #else + return __builtin_bswap32(n); + #endif + #elif defined(__WATCOMC__) && defined(__386__) + return _watcom_bswap32(n); + #else + #error "This compiler does not support the byte swap intrinsic." + #endif +#else + return ((n & 0xFF000000) >> 24) | + ((n & 0x00FF0000) >> 8) | + ((n & 0x0000FF00) << 8) | + ((n & 0x000000FF) << 24); +#endif +} + +static DRFLAC_INLINE drflac_uint64 drflac__swap_endian_uint64(drflac_uint64 n) +{ +#ifdef DRFLAC_HAS_BYTESWAP64_INTRINSIC + #if defined(_MSC_VER) && !defined(__clang__) + return _byteswap_uint64(n); + #elif defined(__GNUC__) || defined(__clang__) + return __builtin_bswap64(n); + #elif defined(__WATCOMC__) && defined(__386__) + return _watcom_bswap64(n); + #else + #error "This compiler does not support the byte swap intrinsic." + #endif +#else + /* Weird "<< 32" bitshift is required for C89 because it doesn't support 64-bit constants. Should be optimized out by a good compiler. */ + return ((n & ((drflac_uint64)0xFF000000 << 32)) >> 56) | + ((n & ((drflac_uint64)0x00FF0000 << 32)) >> 40) | + ((n & ((drflac_uint64)0x0000FF00 << 32)) >> 24) | + ((n & ((drflac_uint64)0x000000FF << 32)) >> 8) | + ((n & ((drflac_uint64)0xFF000000 )) << 8) | + ((n & ((drflac_uint64)0x00FF0000 )) << 24) | + ((n & ((drflac_uint64)0x0000FF00 )) << 40) | + ((n & ((drflac_uint64)0x000000FF )) << 56); +#endif +} + + +static DRFLAC_INLINE drflac_uint16 drflac__be2host_16(drflac_uint16 n) +{ + if (drflac__is_little_endian()) { + return drflac__swap_endian_uint16(n); + } + + return n; +} + +static DRFLAC_INLINE drflac_uint32 drflac__be2host_32(drflac_uint32 n) +{ + if (drflac__is_little_endian()) { + return drflac__swap_endian_uint32(n); + } + + return n; +} + +static DRFLAC_INLINE drflac_uint32 drflac__be2host_32_ptr_unaligned(const void* pData) +{ + const drflac_uint8* pNum = (drflac_uint8*)pData; + return *(pNum) << 24 | *(pNum+1) << 16 | *(pNum+2) << 8 | *(pNum+3); +} + +static DRFLAC_INLINE drflac_uint64 drflac__be2host_64(drflac_uint64 n) +{ + if (drflac__is_little_endian()) { + return drflac__swap_endian_uint64(n); + } + + return n; +} + + +static DRFLAC_INLINE drflac_uint32 drflac__le2host_32(drflac_uint32 n) +{ + if (!drflac__is_little_endian()) { + return drflac__swap_endian_uint32(n); + } + + return n; +} + +static DRFLAC_INLINE drflac_uint32 drflac__le2host_32_ptr_unaligned(const void* pData) +{ + const drflac_uint8* pNum = (drflac_uint8*)pData; + return *pNum | *(pNum+1) << 8 | *(pNum+2) << 16 | *(pNum+3) << 24; +} + + +static DRFLAC_INLINE drflac_uint32 drflac__unsynchsafe_32(drflac_uint32 n) +{ + drflac_uint32 result = 0; + result |= (n & 0x7F000000) >> 3; + result |= (n & 0x007F0000) >> 2; + result |= (n & 0x00007F00) >> 1; + result |= (n & 0x0000007F) >> 0; + + return result; +} + + + +/* The CRC code below is based on this document: http://zlib.net/crc_v3.txt */ +static drflac_uint8 drflac__crc8_table[] = { + 0x00, 0x07, 0x0E, 0x09, 0x1C, 0x1B, 0x12, 0x15, 0x38, 0x3F, 0x36, 0x31, 0x24, 0x23, 0x2A, 0x2D, + 0x70, 0x77, 0x7E, 0x79, 0x6C, 0x6B, 0x62, 0x65, 0x48, 0x4F, 0x46, 0x41, 0x54, 0x53, 0x5A, 0x5D, + 0xE0, 0xE7, 0xEE, 0xE9, 0xFC, 0xFB, 0xF2, 0xF5, 0xD8, 0xDF, 0xD6, 0xD1, 0xC4, 0xC3, 0xCA, 0xCD, + 0x90, 0x97, 0x9E, 0x99, 0x8C, 0x8B, 0x82, 0x85, 0xA8, 0xAF, 0xA6, 0xA1, 0xB4, 0xB3, 0xBA, 0xBD, + 0xC7, 0xC0, 0xC9, 0xCE, 0xDB, 0xDC, 0xD5, 0xD2, 0xFF, 0xF8, 0xF1, 0xF6, 0xE3, 0xE4, 0xED, 0xEA, + 0xB7, 0xB0, 0xB9, 0xBE, 0xAB, 0xAC, 0xA5, 0xA2, 0x8F, 0x88, 0x81, 0x86, 0x93, 0x94, 0x9D, 0x9A, + 0x27, 0x20, 0x29, 0x2E, 0x3B, 0x3C, 0x35, 0x32, 0x1F, 0x18, 0x11, 0x16, 0x03, 0x04, 0x0D, 0x0A, + 0x57, 0x50, 0x59, 0x5E, 0x4B, 0x4C, 0x45, 0x42, 0x6F, 0x68, 0x61, 0x66, 0x73, 0x74, 0x7D, 0x7A, + 0x89, 0x8E, 0x87, 0x80, 0x95, 0x92, 0x9B, 0x9C, 0xB1, 0xB6, 0xBF, 0xB8, 0xAD, 0xAA, 0xA3, 0xA4, + 0xF9, 0xFE, 0xF7, 0xF0, 0xE5, 0xE2, 0xEB, 0xEC, 0xC1, 0xC6, 0xCF, 0xC8, 0xDD, 0xDA, 0xD3, 0xD4, + 0x69, 0x6E, 0x67, 0x60, 0x75, 0x72, 0x7B, 0x7C, 0x51, 0x56, 0x5F, 0x58, 0x4D, 0x4A, 0x43, 0x44, + 0x19, 0x1E, 0x17, 0x10, 0x05, 0x02, 0x0B, 0x0C, 0x21, 0x26, 0x2F, 0x28, 0x3D, 0x3A, 0x33, 0x34, + 0x4E, 0x49, 0x40, 0x47, 0x52, 0x55, 0x5C, 0x5B, 0x76, 0x71, 0x78, 0x7F, 0x6A, 0x6D, 0x64, 0x63, + 0x3E, 0x39, 0x30, 0x37, 0x22, 0x25, 0x2C, 0x2B, 0x06, 0x01, 0x08, 0x0F, 0x1A, 0x1D, 0x14, 0x13, + 0xAE, 0xA9, 0xA0, 0xA7, 0xB2, 0xB5, 0xBC, 0xBB, 0x96, 0x91, 0x98, 0x9F, 0x8A, 0x8D, 0x84, 0x83, + 0xDE, 0xD9, 0xD0, 0xD7, 0xC2, 0xC5, 0xCC, 0xCB, 0xE6, 0xE1, 0xE8, 0xEF, 0xFA, 0xFD, 0xF4, 0xF3 +}; + +static drflac_uint16 drflac__crc16_table[] = { + 0x0000, 0x8005, 0x800F, 0x000A, 0x801B, 0x001E, 0x0014, 0x8011, + 0x8033, 0x0036, 0x003C, 0x8039, 0x0028, 0x802D, 0x8027, 0x0022, + 0x8063, 0x0066, 0x006C, 0x8069, 0x0078, 0x807D, 0x8077, 0x0072, + 0x0050, 0x8055, 0x805F, 0x005A, 0x804B, 0x004E, 0x0044, 0x8041, + 0x80C3, 0x00C6, 0x00CC, 0x80C9, 0x00D8, 0x80DD, 0x80D7, 0x00D2, + 0x00F0, 0x80F5, 0x80FF, 0x00FA, 0x80EB, 0x00EE, 0x00E4, 0x80E1, + 0x00A0, 0x80A5, 0x80AF, 0x00AA, 0x80BB, 0x00BE, 0x00B4, 0x80B1, + 0x8093, 0x0096, 0x009C, 0x8099, 0x0088, 0x808D, 0x8087, 0x0082, + 0x8183, 0x0186, 0x018C, 0x8189, 0x0198, 0x819D, 0x8197, 0x0192, + 0x01B0, 0x81B5, 0x81BF, 0x01BA, 0x81AB, 0x01AE, 0x01A4, 0x81A1, + 0x01E0, 0x81E5, 0x81EF, 0x01EA, 0x81FB, 0x01FE, 0x01F4, 0x81F1, + 0x81D3, 0x01D6, 0x01DC, 0x81D9, 0x01C8, 0x81CD, 0x81C7, 0x01C2, + 0x0140, 0x8145, 0x814F, 0x014A, 0x815B, 0x015E, 0x0154, 0x8151, + 0x8173, 0x0176, 0x017C, 0x8179, 0x0168, 0x816D, 0x8167, 0x0162, + 0x8123, 0x0126, 0x012C, 0x8129, 0x0138, 0x813D, 0x8137, 0x0132, + 0x0110, 0x8115, 0x811F, 0x011A, 0x810B, 0x010E, 0x0104, 0x8101, + 0x8303, 0x0306, 0x030C, 0x8309, 0x0318, 0x831D, 0x8317, 0x0312, + 0x0330, 0x8335, 0x833F, 0x033A, 0x832B, 0x032E, 0x0324, 0x8321, + 0x0360, 0x8365, 0x836F, 0x036A, 0x837B, 0x037E, 0x0374, 0x8371, + 0x8353, 0x0356, 0x035C, 0x8359, 0x0348, 0x834D, 0x8347, 0x0342, + 0x03C0, 0x83C5, 0x83CF, 0x03CA, 0x83DB, 0x03DE, 0x03D4, 0x83D1, + 0x83F3, 0x03F6, 0x03FC, 0x83F9, 0x03E8, 0x83ED, 0x83E7, 0x03E2, + 0x83A3, 0x03A6, 0x03AC, 0x83A9, 0x03B8, 0x83BD, 0x83B7, 0x03B2, + 0x0390, 0x8395, 0x839F, 0x039A, 0x838B, 0x038E, 0x0384, 0x8381, + 0x0280, 0x8285, 0x828F, 0x028A, 0x829B, 0x029E, 0x0294, 0x8291, + 0x82B3, 0x02B6, 0x02BC, 0x82B9, 0x02A8, 0x82AD, 0x82A7, 0x02A2, + 0x82E3, 0x02E6, 0x02EC, 0x82E9, 0x02F8, 0x82FD, 0x82F7, 0x02F2, + 0x02D0, 0x82D5, 0x82DF, 0x02DA, 0x82CB, 0x02CE, 0x02C4, 0x82C1, + 0x8243, 0x0246, 0x024C, 0x8249, 0x0258, 0x825D, 0x8257, 0x0252, + 0x0270, 0x8275, 0x827F, 0x027A, 0x826B, 0x026E, 0x0264, 0x8261, + 0x0220, 0x8225, 0x822F, 0x022A, 0x823B, 0x023E, 0x0234, 0x8231, + 0x8213, 0x0216, 0x021C, 0x8219, 0x0208, 0x820D, 0x8207, 0x0202 +}; + +static DRFLAC_INLINE drflac_uint8 drflac_crc8_byte(drflac_uint8 crc, drflac_uint8 data) +{ + return drflac__crc8_table[crc ^ data]; +} + +static DRFLAC_INLINE drflac_uint8 drflac_crc8(drflac_uint8 crc, drflac_uint32 data, drflac_uint32 count) +{ +#ifdef DR_FLAC_NO_CRC + (void)crc; + (void)data; + (void)count; + return 0; +#else +#if 0 + /* REFERENCE (use of this implementation requires an explicit flush by doing "drflac_crc8(crc, 0, 8);") */ + drflac_uint8 p = 0x07; + for (int i = count-1; i >= 0; --i) { + drflac_uint8 bit = (data & (1 << i)) >> i; + if (crc & 0x80) { + crc = ((crc << 1) | bit) ^ p; + } else { + crc = ((crc << 1) | bit); + } + } + return crc; +#else + drflac_uint32 wholeBytes; + drflac_uint32 leftoverBits; + drflac_uint64 leftoverDataMask; + + static drflac_uint64 leftoverDataMaskTable[8] = { + 0x00, 0x01, 0x03, 0x07, 0x0F, 0x1F, 0x3F, 0x7F + }; + + DRFLAC_ASSERT(count <= 32); + + wholeBytes = count >> 3; + leftoverBits = count - (wholeBytes*8); + leftoverDataMask = leftoverDataMaskTable[leftoverBits]; + + switch (wholeBytes) { + case 4: crc = drflac_crc8_byte(crc, (drflac_uint8)((data & (0xFF000000UL << leftoverBits)) >> (24 + leftoverBits))); + case 3: crc = drflac_crc8_byte(crc, (drflac_uint8)((data & (0x00FF0000UL << leftoverBits)) >> (16 + leftoverBits))); + case 2: crc = drflac_crc8_byte(crc, (drflac_uint8)((data & (0x0000FF00UL << leftoverBits)) >> ( 8 + leftoverBits))); + case 1: crc = drflac_crc8_byte(crc, (drflac_uint8)((data & (0x000000FFUL << leftoverBits)) >> ( 0 + leftoverBits))); + case 0: if (leftoverBits > 0) crc = (drflac_uint8)((crc << leftoverBits) ^ drflac__crc8_table[(crc >> (8 - leftoverBits)) ^ (data & leftoverDataMask)]); + } + return crc; +#endif +#endif +} + +static DRFLAC_INLINE drflac_uint16 drflac_crc16_byte(drflac_uint16 crc, drflac_uint8 data) +{ + return (crc << 8) ^ drflac__crc16_table[(drflac_uint8)(crc >> 8) ^ data]; +} + +static DRFLAC_INLINE drflac_uint16 drflac_crc16_cache(drflac_uint16 crc, drflac_cache_t data) +{ +#ifdef DRFLAC_64BIT + crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 56) & 0xFF)); + crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 48) & 0xFF)); + crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 40) & 0xFF)); + crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 32) & 0xFF)); +#endif + crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 24) & 0xFF)); + crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 16) & 0xFF)); + crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 8) & 0xFF)); + crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 0) & 0xFF)); + + return crc; +} + +static DRFLAC_INLINE drflac_uint16 drflac_crc16_bytes(drflac_uint16 crc, drflac_cache_t data, drflac_uint32 byteCount) +{ + switch (byteCount) + { +#ifdef DRFLAC_64BIT + case 8: crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 56) & 0xFF)); + case 7: crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 48) & 0xFF)); + case 6: crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 40) & 0xFF)); + case 5: crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 32) & 0xFF)); +#endif + case 4: crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 24) & 0xFF)); + case 3: crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 16) & 0xFF)); + case 2: crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 8) & 0xFF)); + case 1: crc = drflac_crc16_byte(crc, (drflac_uint8)((data >> 0) & 0xFF)); + } + + return crc; +} + +#if 0 +static DRFLAC_INLINE drflac_uint16 drflac_crc16__32bit(drflac_uint16 crc, drflac_uint32 data, drflac_uint32 count) +{ +#ifdef DR_FLAC_NO_CRC + (void)crc; + (void)data; + (void)count; + return 0; +#else +#if 0 + /* REFERENCE (use of this implementation requires an explicit flush by doing "drflac_crc16(crc, 0, 16);") */ + drflac_uint16 p = 0x8005; + for (int i = count-1; i >= 0; --i) { + drflac_uint16 bit = (data & (1ULL << i)) >> i; + if (r & 0x8000) { + r = ((r << 1) | bit) ^ p; + } else { + r = ((r << 1) | bit); + } + } + + return crc; +#else + drflac_uint32 wholeBytes; + drflac_uint32 leftoverBits; + drflac_uint64 leftoverDataMask; + + static drflac_uint64 leftoverDataMaskTable[8] = { + 0x00, 0x01, 0x03, 0x07, 0x0F, 0x1F, 0x3F, 0x7F + }; + + DRFLAC_ASSERT(count <= 64); + + wholeBytes = count >> 3; + leftoverBits = count & 7; + leftoverDataMask = leftoverDataMaskTable[leftoverBits]; + + switch (wholeBytes) { + default: + case 4: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (0xFF000000UL << leftoverBits)) >> (24 + leftoverBits))); + case 3: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (0x00FF0000UL << leftoverBits)) >> (16 + leftoverBits))); + case 2: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (0x0000FF00UL << leftoverBits)) >> ( 8 + leftoverBits))); + case 1: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (0x000000FFUL << leftoverBits)) >> ( 0 + leftoverBits))); + case 0: if (leftoverBits > 0) crc = (crc << leftoverBits) ^ drflac__crc16_table[(crc >> (16 - leftoverBits)) ^ (data & leftoverDataMask)]; + } + return crc; +#endif +#endif +} + +static DRFLAC_INLINE drflac_uint16 drflac_crc16__64bit(drflac_uint16 crc, drflac_uint64 data, drflac_uint32 count) +{ +#ifdef DR_FLAC_NO_CRC + (void)crc; + (void)data; + (void)count; + return 0; +#else + drflac_uint32 wholeBytes; + drflac_uint32 leftoverBits; + drflac_uint64 leftoverDataMask; + + static drflac_uint64 leftoverDataMaskTable[8] = { + 0x00, 0x01, 0x03, 0x07, 0x0F, 0x1F, 0x3F, 0x7F + }; + + DRFLAC_ASSERT(count <= 64); + + wholeBytes = count >> 3; + leftoverBits = count & 7; + leftoverDataMask = leftoverDataMaskTable[leftoverBits]; + + switch (wholeBytes) { + default: + case 8: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (((drflac_uint64)0xFF000000 << 32) << leftoverBits)) >> (56 + leftoverBits))); /* Weird "<< 32" bitshift is required for C89 because it doesn't support 64-bit constants. Should be optimized out by a good compiler. */ + case 7: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (((drflac_uint64)0x00FF0000 << 32) << leftoverBits)) >> (48 + leftoverBits))); + case 6: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (((drflac_uint64)0x0000FF00 << 32) << leftoverBits)) >> (40 + leftoverBits))); + case 5: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (((drflac_uint64)0x000000FF << 32) << leftoverBits)) >> (32 + leftoverBits))); + case 4: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (((drflac_uint64)0xFF000000 ) << leftoverBits)) >> (24 + leftoverBits))); + case 3: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (((drflac_uint64)0x00FF0000 ) << leftoverBits)) >> (16 + leftoverBits))); + case 2: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (((drflac_uint64)0x0000FF00 ) << leftoverBits)) >> ( 8 + leftoverBits))); + case 1: crc = drflac_crc16_byte(crc, (drflac_uint8)((data & (((drflac_uint64)0x000000FF ) << leftoverBits)) >> ( 0 + leftoverBits))); + case 0: if (leftoverBits > 0) crc = (crc << leftoverBits) ^ drflac__crc16_table[(crc >> (16 - leftoverBits)) ^ (data & leftoverDataMask)]; + } + return crc; +#endif +} + + +static DRFLAC_INLINE drflac_uint16 drflac_crc16(drflac_uint16 crc, drflac_cache_t data, drflac_uint32 count) +{ +#ifdef DRFLAC_64BIT + return drflac_crc16__64bit(crc, data, count); +#else + return drflac_crc16__32bit(crc, data, count); +#endif +} +#endif + + +#ifdef DRFLAC_64BIT +#define drflac__be2host__cache_line drflac__be2host_64 +#else +#define drflac__be2host__cache_line drflac__be2host_32 +#endif + +/* +BIT READING ATTEMPT #2 + +This uses a 32- or 64-bit bit-shifted cache - as bits are read, the cache is shifted such that the first valid bit is sitting +on the most significant bit. It uses the notion of an L1 and L2 cache (borrowed from CPU architecture), where the L1 cache +is a 32- or 64-bit unsigned integer (depending on whether or not a 32- or 64-bit build is being compiled) and the L2 is an +array of "cache lines", with each cache line being the same size as the L1. The L2 is a buffer of about 4KB and is where data +from onRead() is read into. +*/ +#define DRFLAC_CACHE_L1_SIZE_BYTES(bs) (sizeof((bs)->cache)) +#define DRFLAC_CACHE_L1_SIZE_BITS(bs) (sizeof((bs)->cache)*8) +#define DRFLAC_CACHE_L1_BITS_REMAINING(bs) (DRFLAC_CACHE_L1_SIZE_BITS(bs) - (bs)->consumedBits) +#define DRFLAC_CACHE_L1_SELECTION_MASK(_bitCount) (~((~(drflac_cache_t)0) >> (_bitCount))) +#define DRFLAC_CACHE_L1_SELECTION_SHIFT(bs, _bitCount) (DRFLAC_CACHE_L1_SIZE_BITS(bs) - (_bitCount)) +#define DRFLAC_CACHE_L1_SELECT(bs, _bitCount) (((bs)->cache) & DRFLAC_CACHE_L1_SELECTION_MASK(_bitCount)) +#define DRFLAC_CACHE_L1_SELECT_AND_SHIFT(bs, _bitCount) (DRFLAC_CACHE_L1_SELECT((bs), (_bitCount)) >> DRFLAC_CACHE_L1_SELECTION_SHIFT((bs), (_bitCount))) +#define DRFLAC_CACHE_L1_SELECT_AND_SHIFT_SAFE(bs, _bitCount)(DRFLAC_CACHE_L1_SELECT((bs), (_bitCount)) >> (DRFLAC_CACHE_L1_SELECTION_SHIFT((bs), (_bitCount)) & (DRFLAC_CACHE_L1_SIZE_BITS(bs)-1))) +#define DRFLAC_CACHE_L2_SIZE_BYTES(bs) (sizeof((bs)->cacheL2)) +#define DRFLAC_CACHE_L2_LINE_COUNT(bs) (DRFLAC_CACHE_L2_SIZE_BYTES(bs) / sizeof((bs)->cacheL2[0])) +#define DRFLAC_CACHE_L2_LINES_REMAINING(bs) (DRFLAC_CACHE_L2_LINE_COUNT(bs) - (bs)->nextL2Line) + + +#ifndef DR_FLAC_NO_CRC +static DRFLAC_INLINE void drflac__reset_crc16(drflac_bs* bs) +{ + bs->crc16 = 0; + bs->crc16CacheIgnoredBytes = bs->consumedBits >> 3; +} + +static DRFLAC_INLINE void drflac__update_crc16(drflac_bs* bs) +{ + if (bs->crc16CacheIgnoredBytes == 0) { + bs->crc16 = drflac_crc16_cache(bs->crc16, bs->crc16Cache); + } else { + bs->crc16 = drflac_crc16_bytes(bs->crc16, bs->crc16Cache, DRFLAC_CACHE_L1_SIZE_BYTES(bs) - bs->crc16CacheIgnoredBytes); + bs->crc16CacheIgnoredBytes = 0; + } +} + +static DRFLAC_INLINE drflac_uint16 drflac__flush_crc16(drflac_bs* bs) +{ + /* We should never be flushing in a situation where we are not aligned on a byte boundary. */ + DRFLAC_ASSERT((DRFLAC_CACHE_L1_BITS_REMAINING(bs) & 7) == 0); + + /* + The bits that were read from the L1 cache need to be accumulated. The number of bytes needing to be accumulated is determined + by the number of bits that have been consumed. + */ + if (DRFLAC_CACHE_L1_BITS_REMAINING(bs) == 0) { + drflac__update_crc16(bs); + } else { + /* We only accumulate the consumed bits. */ + bs->crc16 = drflac_crc16_bytes(bs->crc16, bs->crc16Cache >> DRFLAC_CACHE_L1_BITS_REMAINING(bs), (bs->consumedBits >> 3) - bs->crc16CacheIgnoredBytes); + + /* + The bits that we just accumulated should never be accumulated again. We need to keep track of how many bytes were accumulated + so we can handle that later. + */ + bs->crc16CacheIgnoredBytes = bs->consumedBits >> 3; + } + + return bs->crc16; +} +#endif + +static DRFLAC_INLINE drflac_bool32 drflac__reload_l1_cache_from_l2(drflac_bs* bs) +{ + size_t bytesRead; + size_t alignedL1LineCount; + + /* Fast path. Try loading straight from L2. */ + if (bs->nextL2Line < DRFLAC_CACHE_L2_LINE_COUNT(bs)) { + bs->cache = bs->cacheL2[bs->nextL2Line++]; + return DRFLAC_TRUE; + } + + /* + If we get here it means we've run out of data in the L2 cache. We'll need to fetch more from the client, if there's + any left. + */ + if (bs->unalignedByteCount > 0) { + return DRFLAC_FALSE; /* If we have any unaligned bytes it means there's no more aligned bytes left in the client. */ + } + + bytesRead = bs->onRead(bs->pUserData, bs->cacheL2, DRFLAC_CACHE_L2_SIZE_BYTES(bs)); + + bs->nextL2Line = 0; + if (bytesRead == DRFLAC_CACHE_L2_SIZE_BYTES(bs)) { + bs->cache = bs->cacheL2[bs->nextL2Line++]; + return DRFLAC_TRUE; + } + + + /* + If we get here it means we were unable to retrieve enough data to fill the entire L2 cache. It probably + means we've just reached the end of the file. We need to move the valid data down to the end of the buffer + and adjust the index of the next line accordingly. Also keep in mind that the L2 cache must be aligned to + the size of the L1 so we'll need to seek backwards by any misaligned bytes. + */ + alignedL1LineCount = bytesRead / DRFLAC_CACHE_L1_SIZE_BYTES(bs); + + /* We need to keep track of any unaligned bytes for later use. */ + bs->unalignedByteCount = bytesRead - (alignedL1LineCount * DRFLAC_CACHE_L1_SIZE_BYTES(bs)); + if (bs->unalignedByteCount > 0) { + bs->unalignedCache = bs->cacheL2[alignedL1LineCount]; + } + + if (alignedL1LineCount > 0) { + size_t offset = DRFLAC_CACHE_L2_LINE_COUNT(bs) - alignedL1LineCount; + size_t i; + for (i = alignedL1LineCount; i > 0; --i) { + bs->cacheL2[i-1 + offset] = bs->cacheL2[i-1]; + } + + bs->nextL2Line = (drflac_uint32)offset; + bs->cache = bs->cacheL2[bs->nextL2Line++]; + return DRFLAC_TRUE; + } else { + /* If we get into this branch it means we weren't able to load any L1-aligned data. */ + bs->nextL2Line = DRFLAC_CACHE_L2_LINE_COUNT(bs); + return DRFLAC_FALSE; + } +} + +static drflac_bool32 drflac__reload_cache(drflac_bs* bs) +{ + size_t bytesRead; + +#ifndef DR_FLAC_NO_CRC + drflac__update_crc16(bs); +#endif + + /* Fast path. Try just moving the next value in the L2 cache to the L1 cache. */ + if (drflac__reload_l1_cache_from_l2(bs)) { + bs->cache = drflac__be2host__cache_line(bs->cache); + bs->consumedBits = 0; +#ifndef DR_FLAC_NO_CRC + bs->crc16Cache = bs->cache; +#endif + return DRFLAC_TRUE; + } + + /* Slow path. */ + + /* + If we get here it means we have failed to load the L1 cache from the L2. Likely we've just reached the end of the stream and the last + few bytes did not meet the alignment requirements for the L2 cache. In this case we need to fall back to a slower path and read the + data from the unaligned cache. + */ + bytesRead = bs->unalignedByteCount; + if (bytesRead == 0) { + bs->consumedBits = DRFLAC_CACHE_L1_SIZE_BITS(bs); /* <-- The stream has been exhausted, so marked the bits as consumed. */ + return DRFLAC_FALSE; + } + + DRFLAC_ASSERT(bytesRead < DRFLAC_CACHE_L1_SIZE_BYTES(bs)); + bs->consumedBits = (drflac_uint32)(DRFLAC_CACHE_L1_SIZE_BYTES(bs) - bytesRead) * 8; + + bs->cache = drflac__be2host__cache_line(bs->unalignedCache); + bs->cache &= DRFLAC_CACHE_L1_SELECTION_MASK(DRFLAC_CACHE_L1_BITS_REMAINING(bs)); /* <-- Make sure the consumed bits are always set to zero. Other parts of the library depend on this property. */ + bs->unalignedByteCount = 0; /* <-- At this point the unaligned bytes have been moved into the cache and we thus have no more unaligned bytes. */ + +#ifndef DR_FLAC_NO_CRC + bs->crc16Cache = bs->cache >> bs->consumedBits; + bs->crc16CacheIgnoredBytes = bs->consumedBits >> 3; +#endif + return DRFLAC_TRUE; +} + +static void drflac__reset_cache(drflac_bs* bs) +{ + bs->nextL2Line = DRFLAC_CACHE_L2_LINE_COUNT(bs); /* <-- This clears the L2 cache. */ + bs->consumedBits = DRFLAC_CACHE_L1_SIZE_BITS(bs); /* <-- This clears the L1 cache. */ + bs->cache = 0; + bs->unalignedByteCount = 0; /* <-- This clears the trailing unaligned bytes. */ + bs->unalignedCache = 0; + +#ifndef DR_FLAC_NO_CRC + bs->crc16Cache = 0; + bs->crc16CacheIgnoredBytes = 0; +#endif +} + + +static DRFLAC_INLINE drflac_bool32 drflac__read_uint32(drflac_bs* bs, unsigned int bitCount, drflac_uint32* pResultOut) +{ + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pResultOut != NULL); + DRFLAC_ASSERT(bitCount > 0); + DRFLAC_ASSERT(bitCount <= 32); + + if (bs->consumedBits == DRFLAC_CACHE_L1_SIZE_BITS(bs)) { + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + } + + if (bitCount <= DRFLAC_CACHE_L1_BITS_REMAINING(bs)) { + /* + If we want to load all 32-bits from a 32-bit cache we need to do it slightly differently because we can't do + a 32-bit shift on a 32-bit integer. This will never be the case on 64-bit caches, so we can have a slightly + more optimal solution for this. + */ +#ifdef DRFLAC_64BIT + *pResultOut = (drflac_uint32)DRFLAC_CACHE_L1_SELECT_AND_SHIFT(bs, bitCount); + bs->consumedBits += bitCount; + bs->cache <<= bitCount; +#else + if (bitCount < DRFLAC_CACHE_L1_SIZE_BITS(bs)) { + *pResultOut = (drflac_uint32)DRFLAC_CACHE_L1_SELECT_AND_SHIFT(bs, bitCount); + bs->consumedBits += bitCount; + bs->cache <<= bitCount; + } else { + /* Cannot shift by 32-bits, so need to do it differently. */ + *pResultOut = (drflac_uint32)bs->cache; + bs->consumedBits = DRFLAC_CACHE_L1_SIZE_BITS(bs); + bs->cache = 0; + } +#endif + + return DRFLAC_TRUE; + } else { + /* It straddles the cached data. It will never cover more than the next chunk. We just read the number in two parts and combine them. */ + drflac_uint32 bitCountHi = DRFLAC_CACHE_L1_BITS_REMAINING(bs); + drflac_uint32 bitCountLo = bitCount - bitCountHi; + drflac_uint32 resultHi; + + DRFLAC_ASSERT(bitCountHi > 0); + DRFLAC_ASSERT(bitCountHi < 32); + resultHi = (drflac_uint32)DRFLAC_CACHE_L1_SELECT_AND_SHIFT(bs, bitCountHi); + + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + if (bitCountLo > DRFLAC_CACHE_L1_BITS_REMAINING(bs)) { + /* This happens when we get to end of stream */ + return DRFLAC_FALSE; + } + + *pResultOut = (resultHi << bitCountLo) | (drflac_uint32)DRFLAC_CACHE_L1_SELECT_AND_SHIFT(bs, bitCountLo); + bs->consumedBits += bitCountLo; + bs->cache <<= bitCountLo; + return DRFLAC_TRUE; + } +} + +static drflac_bool32 drflac__read_int32(drflac_bs* bs, unsigned int bitCount, drflac_int32* pResult) +{ + drflac_uint32 result; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pResult != NULL); + DRFLAC_ASSERT(bitCount > 0); + DRFLAC_ASSERT(bitCount <= 32); + + if (!drflac__read_uint32(bs, bitCount, &result)) { + return DRFLAC_FALSE; + } + + /* Do not attempt to shift by 32 as it's undefined. */ + if (bitCount < 32) { + drflac_uint32 signbit; + signbit = ((result >> (bitCount-1)) & 0x01); + result |= (~signbit + 1) << bitCount; + } + + *pResult = (drflac_int32)result; + return DRFLAC_TRUE; +} + +#ifdef DRFLAC_64BIT +static drflac_bool32 drflac__read_uint64(drflac_bs* bs, unsigned int bitCount, drflac_uint64* pResultOut) +{ + drflac_uint32 resultHi; + drflac_uint32 resultLo; + + DRFLAC_ASSERT(bitCount <= 64); + DRFLAC_ASSERT(bitCount > 32); + + if (!drflac__read_uint32(bs, bitCount - 32, &resultHi)) { + return DRFLAC_FALSE; + } + + if (!drflac__read_uint32(bs, 32, &resultLo)) { + return DRFLAC_FALSE; + } + + *pResultOut = (((drflac_uint64)resultHi) << 32) | ((drflac_uint64)resultLo); + return DRFLAC_TRUE; +} +#endif + +/* Function below is unused, but leaving it here in case I need to quickly add it again. */ +#if 0 +static drflac_bool32 drflac__read_int64(drflac_bs* bs, unsigned int bitCount, drflac_int64* pResultOut) +{ + drflac_uint64 result; + drflac_uint64 signbit; + + DRFLAC_ASSERT(bitCount <= 64); + + if (!drflac__read_uint64(bs, bitCount, &result)) { + return DRFLAC_FALSE; + } + + signbit = ((result >> (bitCount-1)) & 0x01); + result |= (~signbit + 1) << bitCount; + + *pResultOut = (drflac_int64)result; + return DRFLAC_TRUE; +} +#endif + +static drflac_bool32 drflac__read_uint16(drflac_bs* bs, unsigned int bitCount, drflac_uint16* pResult) +{ + drflac_uint32 result; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pResult != NULL); + DRFLAC_ASSERT(bitCount > 0); + DRFLAC_ASSERT(bitCount <= 16); + + if (!drflac__read_uint32(bs, bitCount, &result)) { + return DRFLAC_FALSE; + } + + *pResult = (drflac_uint16)result; + return DRFLAC_TRUE; +} + +#if 0 +static drflac_bool32 drflac__read_int16(drflac_bs* bs, unsigned int bitCount, drflac_int16* pResult) +{ + drflac_int32 result; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pResult != NULL); + DRFLAC_ASSERT(bitCount > 0); + DRFLAC_ASSERT(bitCount <= 16); + + if (!drflac__read_int32(bs, bitCount, &result)) { + return DRFLAC_FALSE; + } + + *pResult = (drflac_int16)result; + return DRFLAC_TRUE; +} +#endif + +static drflac_bool32 drflac__read_uint8(drflac_bs* bs, unsigned int bitCount, drflac_uint8* pResult) +{ + drflac_uint32 result; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pResult != NULL); + DRFLAC_ASSERT(bitCount > 0); + DRFLAC_ASSERT(bitCount <= 8); + + if (!drflac__read_uint32(bs, bitCount, &result)) { + return DRFLAC_FALSE; + } + + *pResult = (drflac_uint8)result; + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__read_int8(drflac_bs* bs, unsigned int bitCount, drflac_int8* pResult) +{ + drflac_int32 result; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pResult != NULL); + DRFLAC_ASSERT(bitCount > 0); + DRFLAC_ASSERT(bitCount <= 8); + + if (!drflac__read_int32(bs, bitCount, &result)) { + return DRFLAC_FALSE; + } + + *pResult = (drflac_int8)result; + return DRFLAC_TRUE; +} + + +static drflac_bool32 drflac__seek_bits(drflac_bs* bs, size_t bitsToSeek) +{ + if (bitsToSeek <= DRFLAC_CACHE_L1_BITS_REMAINING(bs)) { + bs->consumedBits += (drflac_uint32)bitsToSeek; + bs->cache <<= bitsToSeek; + return DRFLAC_TRUE; + } else { + /* It straddles the cached data. This function isn't called too frequently so I'm favouring simplicity here. */ + bitsToSeek -= DRFLAC_CACHE_L1_BITS_REMAINING(bs); + bs->consumedBits += DRFLAC_CACHE_L1_BITS_REMAINING(bs); + bs->cache = 0; + + /* Simple case. Seek in groups of the same number as bits that fit within a cache line. */ +#ifdef DRFLAC_64BIT + while (bitsToSeek >= DRFLAC_CACHE_L1_SIZE_BITS(bs)) { + drflac_uint64 bin; + if (!drflac__read_uint64(bs, DRFLAC_CACHE_L1_SIZE_BITS(bs), &bin)) { + return DRFLAC_FALSE; + } + bitsToSeek -= DRFLAC_CACHE_L1_SIZE_BITS(bs); + } +#else + while (bitsToSeek >= DRFLAC_CACHE_L1_SIZE_BITS(bs)) { + drflac_uint32 bin; + if (!drflac__read_uint32(bs, DRFLAC_CACHE_L1_SIZE_BITS(bs), &bin)) { + return DRFLAC_FALSE; + } + bitsToSeek -= DRFLAC_CACHE_L1_SIZE_BITS(bs); + } +#endif + + /* Whole leftover bytes. */ + while (bitsToSeek >= 8) { + drflac_uint8 bin; + if (!drflac__read_uint8(bs, 8, &bin)) { + return DRFLAC_FALSE; + } + bitsToSeek -= 8; + } + + /* Leftover bits. */ + if (bitsToSeek > 0) { + drflac_uint8 bin; + if (!drflac__read_uint8(bs, (drflac_uint32)bitsToSeek, &bin)) { + return DRFLAC_FALSE; + } + bitsToSeek = 0; /* <-- Necessary for the assert below. */ + } + + DRFLAC_ASSERT(bitsToSeek == 0); + return DRFLAC_TRUE; + } +} + + +/* This function moves the bit streamer to the first bit after the sync code (bit 15 of the of the frame header). It will also update the CRC-16. */ +static drflac_bool32 drflac__find_and_seek_to_next_sync_code(drflac_bs* bs) +{ + DRFLAC_ASSERT(bs != NULL); + + /* + The sync code is always aligned to 8 bits. This is convenient for us because it means we can do byte-aligned movements. The first + thing to do is align to the next byte. + */ + if (!drflac__seek_bits(bs, DRFLAC_CACHE_L1_BITS_REMAINING(bs) & 7)) { + return DRFLAC_FALSE; + } + + for (;;) { + drflac_uint8 hi; + +#ifndef DR_FLAC_NO_CRC + drflac__reset_crc16(bs); +#endif + + if (!drflac__read_uint8(bs, 8, &hi)) { + return DRFLAC_FALSE; + } + + if (hi == 0xFF) { + drflac_uint8 lo; + if (!drflac__read_uint8(bs, 6, &lo)) { + return DRFLAC_FALSE; + } + + if (lo == 0x3E) { + return DRFLAC_TRUE; + } else { + if (!drflac__seek_bits(bs, DRFLAC_CACHE_L1_BITS_REMAINING(bs) & 7)) { + return DRFLAC_FALSE; + } + } + } + } + + /* Should never get here. */ + /*return DRFLAC_FALSE;*/ +} + + +#if defined(DRFLAC_HAS_LZCNT_INTRINSIC) +#define DRFLAC_IMPLEMENT_CLZ_LZCNT +#endif +#if defined(_MSC_VER) && _MSC_VER >= 1400 && (defined(DRFLAC_X64) || defined(DRFLAC_X86)) && !defined(__clang__) +#define DRFLAC_IMPLEMENT_CLZ_MSVC +#endif +#if defined(__WATCOMC__) && defined(__386__) +#define DRFLAC_IMPLEMENT_CLZ_WATCOM +#endif +#ifdef __MRC__ +#include +#define DRFLAC_IMPLEMENT_CLZ_MRC +#endif + +static DRFLAC_INLINE drflac_uint32 drflac__clz_software(drflac_cache_t x) +{ + drflac_uint32 n; + static drflac_uint32 clz_table_4[] = { + 0, + 4, + 3, 3, + 2, 2, 2, 2, + 1, 1, 1, 1, 1, 1, 1, 1 + }; + + if (x == 0) { + return sizeof(x)*8; + } + + n = clz_table_4[x >> (sizeof(x)*8 - 4)]; + if (n == 0) { +#ifdef DRFLAC_64BIT + if ((x & ((drflac_uint64)0xFFFFFFFF << 32)) == 0) { n = 32; x <<= 32; } + if ((x & ((drflac_uint64)0xFFFF0000 << 32)) == 0) { n += 16; x <<= 16; } + if ((x & ((drflac_uint64)0xFF000000 << 32)) == 0) { n += 8; x <<= 8; } + if ((x & ((drflac_uint64)0xF0000000 << 32)) == 0) { n += 4; x <<= 4; } +#else + if ((x & 0xFFFF0000) == 0) { n = 16; x <<= 16; } + if ((x & 0xFF000000) == 0) { n += 8; x <<= 8; } + if ((x & 0xF0000000) == 0) { n += 4; x <<= 4; } +#endif + n += clz_table_4[x >> (sizeof(x)*8 - 4)]; + } + + return n - 1; +} + +#ifdef DRFLAC_IMPLEMENT_CLZ_LZCNT +static DRFLAC_INLINE drflac_bool32 drflac__is_lzcnt_supported(void) +{ + /* Fast compile time check for ARM. */ +#if defined(DRFLAC_HAS_LZCNT_INTRINSIC) && defined(DRFLAC_ARM) && (defined(__ARM_ARCH) && __ARM_ARCH >= 5) + return DRFLAC_TRUE; +#elif defined(__MRC__) + return DRFLAC_TRUE; +#else + /* If the compiler itself does not support the intrinsic then we'll need to return false. */ + #ifdef DRFLAC_HAS_LZCNT_INTRINSIC + return drflac__gIsLZCNTSupported; + #else + return DRFLAC_FALSE; + #endif +#endif +} + +static DRFLAC_INLINE drflac_uint32 drflac__clz_lzcnt(drflac_cache_t x) +{ + /* + It's critical for competitive decoding performance that this function be highly optimal. With MSVC we can use the __lzcnt64() and __lzcnt() intrinsics + to achieve good performance, however on GCC and Clang it's a little bit more annoying. The __builtin_clzl() and __builtin_clzll() intrinsics leave + it undefined as to the return value when `x` is 0. We need this to be well defined as returning 32 or 64, depending on whether or not it's a 32- or + 64-bit build. To work around this we would need to add a conditional to check for the x = 0 case, but this creates unnecessary inefficiency. To work + around this problem I have written some inline assembly to emit the LZCNT (x86) or CLZ (ARM) instruction directly which removes the need to include + the conditional. This has worked well in the past, but for some reason Clang's MSVC compatible driver, clang-cl, does not seem to be handling this + in the same way as the normal Clang driver. It seems that `clang-cl` is just outputting the wrong results sometimes, maybe due to some register + getting clobbered? + + I'm not sure if this is a bug with dr_flac's inlined assembly (most likely), a bug in `clang-cl` or just a misunderstanding on my part with inline + assembly rules for `clang-cl`. If somebody can identify an error in dr_flac's inlined assembly I'm happy to get that fixed. + + Fortunately there is an easy workaround for this. Clang implements MSVC-specific intrinsics for compatibility. It also defines _MSC_VER for extra + compatibility. We can therefore just check for _MSC_VER and use the MSVC intrinsic which, fortunately for us, Clang supports. It would still be nice + to know how to fix the inlined assembly for correctness sake, however. + */ + +#if defined(_MSC_VER) /*&& !defined(__clang__)*/ /* <-- Intentionally wanting Clang to use the MSVC __lzcnt64/__lzcnt intrinsics due to above ^. */ + #ifdef DRFLAC_64BIT + return (drflac_uint32)__lzcnt64(x); + #else + return (drflac_uint32)__lzcnt(x); + #endif +#else + #if defined(__GNUC__) || defined(__clang__) + #if defined(DRFLAC_X64) + { + /* + A note on lzcnt. + + We check for the presence of the lzcnt instruction at runtime before calling this function, but we still generate this code. I have had + a report where the assembler does not recognize the lzcnt instruction. To work around this we are going to use `rep; bsr` instead which + has an identical byte encoding as lzcnt, and should hopefully improve compatibility with older assemblers. + */ + drflac_uint64 r; + __asm__ __volatile__ ( + "rep; bsr{q %1, %0| %0, %1}" : "=r"(r) : "r"(x) : "cc" + /*"lzcnt{ %1, %0| %0, %1}" : "=r"(r) : "r"(x) : "cc"*/ + ); + + return (drflac_uint32)r; + } + #elif defined(DRFLAC_X86) + { + drflac_uint32 r; + __asm__ __volatile__ ( + "rep; bsr{l %1, %0| %0, %1}" : "=r"(r) : "r"(x) : "cc" + /*"lzcnt{l %1, %0| %0, %1}" : "=r"(r) : "r"(x) : "cc"*/ + ); + + return r; + } + #elif defined(DRFLAC_ARM) && (defined(__ARM_ARCH) && __ARM_ARCH >= 5) && !defined(__ARM_ARCH_6M__) && !(defined(__thumb__) && !defined(__thumb2__)) && !defined(DRFLAC_64BIT) /* <-- I haven't tested 64-bit inline assembly, so only enabling this for the 32-bit build for now. */ + { + unsigned int r; + __asm__ __volatile__ ( + #if defined(DRFLAC_64BIT) + "clz %w[out], %w[in]" : [out]"=r"(r) : [in]"r"(x) /* <-- This is untested. If someone in the community could test this, that would be appreciated! */ + #else + "clz %[out], %[in]" : [out]"=r"(r) : [in]"r"(x) + #endif + ); + + return r; + } + #else + if (x == 0) { + return sizeof(x)*8; + } + #ifdef DRFLAC_64BIT + return (drflac_uint32)__builtin_clzll((drflac_uint64)x); + #else + return (drflac_uint32)__builtin_clzl((drflac_uint32)x); + #endif + #endif + #else + /* Unsupported compiler. */ + #error "This compiler does not support the lzcnt intrinsic." + #endif +#endif +} +#endif + +#ifdef DRFLAC_IMPLEMENT_CLZ_MSVC +#include /* For BitScanReverse(). */ + +static DRFLAC_INLINE drflac_uint32 drflac__clz_msvc(drflac_cache_t x) +{ + drflac_uint32 n; + + if (x == 0) { + return sizeof(x)*8; + } + +#ifdef DRFLAC_64BIT + _BitScanReverse64((unsigned long*)&n, x); +#else + _BitScanReverse((unsigned long*)&n, x); +#endif + return sizeof(x)*8 - n - 1; +} +#endif + +#ifdef DRFLAC_IMPLEMENT_CLZ_WATCOM +static __inline drflac_uint32 drflac__clz_watcom (drflac_uint32); +#ifdef DRFLAC_IMPLEMENT_CLZ_WATCOM_LZCNT +/* Use the LZCNT instruction (only available on some processors since the 2010s). */ +#pragma aux drflac__clz_watcom_lzcnt = \ + "db 0F3h, 0Fh, 0BDh, 0C0h" /* lzcnt eax, eax */ \ + parm [eax] \ + value [eax] \ + modify nomemory; +#else +/* Use the 386+-compatible implementation. */ +#pragma aux drflac__clz_watcom = \ + "bsr eax, eax" \ + "xor eax, 31" \ + parm [eax] nomemory \ + value [eax] \ + modify exact [eax] nomemory; +#endif +#endif + +static DRFLAC_INLINE drflac_uint32 drflac__clz(drflac_cache_t x) +{ +#ifdef DRFLAC_IMPLEMENT_CLZ_LZCNT + if (drflac__is_lzcnt_supported()) { + return drflac__clz_lzcnt(x); + } else +#endif + { +#ifdef DRFLAC_IMPLEMENT_CLZ_MSVC + return drflac__clz_msvc(x); +#elif defined(DRFLAC_IMPLEMENT_CLZ_WATCOM_LZCNT) + return drflac__clz_watcom_lzcnt(x); +#elif defined(DRFLAC_IMPLEMENT_CLZ_WATCOM) + return (x == 0) ? sizeof(x)*8 : drflac__clz_watcom(x); +#elif defined(__MRC__) + return __cntlzw(x); +#else + return drflac__clz_software(x); +#endif + } +} + + +static DRFLAC_INLINE drflac_bool32 drflac__seek_past_next_set_bit(drflac_bs* bs, unsigned int* pOffsetOut) +{ + drflac_uint32 zeroCounter = 0; + drflac_uint32 setBitOffsetPlus1; + + while (bs->cache == 0) { + zeroCounter += (drflac_uint32)DRFLAC_CACHE_L1_BITS_REMAINING(bs); + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + } + + if (bs->cache == 1) { + /* Not catching this would lead to undefined behaviour: a shift of a 32-bit number by 32 or more is undefined */ + *pOffsetOut = zeroCounter + (drflac_uint32)DRFLAC_CACHE_L1_BITS_REMAINING(bs) - 1; + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + + return DRFLAC_TRUE; + } + + setBitOffsetPlus1 = drflac__clz(bs->cache); + setBitOffsetPlus1 += 1; + + if (setBitOffsetPlus1 > DRFLAC_CACHE_L1_BITS_REMAINING(bs)) { + /* This happens when we get to end of stream */ + return DRFLAC_FALSE; + } + + bs->consumedBits += setBitOffsetPlus1; + bs->cache <<= setBitOffsetPlus1; + + *pOffsetOut = zeroCounter + setBitOffsetPlus1 - 1; + return DRFLAC_TRUE; +} + + + +static drflac_bool32 drflac__seek_to_byte(drflac_bs* bs, drflac_uint64 offsetFromStart) +{ + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(offsetFromStart > 0); + + /* + Seeking from the start is not quite as trivial as it sounds because the onSeek callback takes a signed 32-bit integer (which + is intentional because it simplifies the implementation of the onSeek callbacks), however offsetFromStart is unsigned 64-bit. + To resolve we just need to do an initial seek from the start, and then a series of offset seeks to make up the remainder. + */ + if (offsetFromStart > 0x7FFFFFFF) { + drflac_uint64 bytesRemaining = offsetFromStart; + if (!bs->onSeek(bs->pUserData, 0x7FFFFFFF, DRFLAC_SEEK_SET)) { + return DRFLAC_FALSE; + } + bytesRemaining -= 0x7FFFFFFF; + + while (bytesRemaining > 0x7FFFFFFF) { + if (!bs->onSeek(bs->pUserData, 0x7FFFFFFF, DRFLAC_SEEK_CUR)) { + return DRFLAC_FALSE; + } + bytesRemaining -= 0x7FFFFFFF; + } + + if (bytesRemaining > 0) { + if (!bs->onSeek(bs->pUserData, (int)bytesRemaining, DRFLAC_SEEK_CUR)) { + return DRFLAC_FALSE; + } + } + } else { + if (!bs->onSeek(bs->pUserData, (int)offsetFromStart, DRFLAC_SEEK_SET)) { + return DRFLAC_FALSE; + } + } + + /* The cache should be reset to force a reload of fresh data from the client. */ + drflac__reset_cache(bs); + return DRFLAC_TRUE; +} + + +static drflac_result drflac__read_utf8_coded_number(drflac_bs* bs, drflac_uint64* pNumberOut, drflac_uint8* pCRCOut) +{ + drflac_uint8 crc; + drflac_uint64 result; + drflac_uint8 utf8[7] = {0}; + int byteCount; + int i; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pNumberOut != NULL); + DRFLAC_ASSERT(pCRCOut != NULL); + + crc = *pCRCOut; + + if (!drflac__read_uint8(bs, 8, utf8)) { + *pNumberOut = 0; + return DRFLAC_AT_END; + } + crc = drflac_crc8(crc, utf8[0], 8); + + if ((utf8[0] & 0x80) == 0) { + *pNumberOut = utf8[0]; + *pCRCOut = crc; + return DRFLAC_SUCCESS; + } + + /*byteCount = 1;*/ + if ((utf8[0] & 0xE0) == 0xC0) { + byteCount = 2; + } else if ((utf8[0] & 0xF0) == 0xE0) { + byteCount = 3; + } else if ((utf8[0] & 0xF8) == 0xF0) { + byteCount = 4; + } else if ((utf8[0] & 0xFC) == 0xF8) { + byteCount = 5; + } else if ((utf8[0] & 0xFE) == 0xFC) { + byteCount = 6; + } else if ((utf8[0] & 0xFF) == 0xFE) { + byteCount = 7; + } else { + *pNumberOut = 0; + return DRFLAC_CRC_MISMATCH; /* Bad UTF-8 encoding. */ + } + + /* Read extra bytes. */ + DRFLAC_ASSERT(byteCount > 1); + + result = (drflac_uint64)(utf8[0] & (0xFF >> (byteCount + 1))); + for (i = 1; i < byteCount; ++i) { + if (!drflac__read_uint8(bs, 8, utf8 + i)) { + *pNumberOut = 0; + return DRFLAC_AT_END; + } + crc = drflac_crc8(crc, utf8[i], 8); + + result = (result << 6) | (utf8[i] & 0x3F); + } + + *pNumberOut = result; + *pCRCOut = crc; + return DRFLAC_SUCCESS; +} + + +static DRFLAC_INLINE drflac_uint32 drflac__ilog2_u32(drflac_uint32 x) +{ +#if 1 /* Needs optimizing. */ + drflac_uint32 result = 0; + while (x > 0) { + result += 1; + x >>= 1; + } + + return result; +#endif +} + +static DRFLAC_INLINE drflac_bool32 drflac__use_64_bit_prediction(drflac_uint32 bitsPerSample, drflac_uint32 order, drflac_uint32 precision) +{ + /* https://web.archive.org/web/20220205005724/https://github.com/ietf-wg-cellar/flac-specification/blob/37a49aa48ba4ba12e8757badfc59c0df35435fec/rfc_backmatter.md */ + return bitsPerSample + precision + drflac__ilog2_u32(order) > 32; +} + + +/* +The next two functions are responsible for calculating the prediction. + +When the bits per sample is >16 we need to use 64-bit integer arithmetic because otherwise we'll run out of precision. It's +safe to assume this will be slower on 32-bit platforms so we use a more optimal solution when the bits per sample is <=16. +*/ +#if defined(__clang__) +__attribute__((no_sanitize("signed-integer-overflow"))) +#endif +static DRFLAC_INLINE drflac_int32 drflac__calculate_prediction_32(drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pDecodedSamples) +{ + drflac_int32 prediction = 0; + + DRFLAC_ASSERT(order <= 32); + + /* 32-bit version. */ + + /* VC++ optimizes this to a single jmp. I've not yet verified this for other compilers. */ + switch (order) + { + case 32: prediction += coefficients[31] * pDecodedSamples[-32]; + case 31: prediction += coefficients[30] * pDecodedSamples[-31]; + case 30: prediction += coefficients[29] * pDecodedSamples[-30]; + case 29: prediction += coefficients[28] * pDecodedSamples[-29]; + case 28: prediction += coefficients[27] * pDecodedSamples[-28]; + case 27: prediction += coefficients[26] * pDecodedSamples[-27]; + case 26: prediction += coefficients[25] * pDecodedSamples[-26]; + case 25: prediction += coefficients[24] * pDecodedSamples[-25]; + case 24: prediction += coefficients[23] * pDecodedSamples[-24]; + case 23: prediction += coefficients[22] * pDecodedSamples[-23]; + case 22: prediction += coefficients[21] * pDecodedSamples[-22]; + case 21: prediction += coefficients[20] * pDecodedSamples[-21]; + case 20: prediction += coefficients[19] * pDecodedSamples[-20]; + case 19: prediction += coefficients[18] * pDecodedSamples[-19]; + case 18: prediction += coefficients[17] * pDecodedSamples[-18]; + case 17: prediction += coefficients[16] * pDecodedSamples[-17]; + case 16: prediction += coefficients[15] * pDecodedSamples[-16]; + case 15: prediction += coefficients[14] * pDecodedSamples[-15]; + case 14: prediction += coefficients[13] * pDecodedSamples[-14]; + case 13: prediction += coefficients[12] * pDecodedSamples[-13]; + case 12: prediction += coefficients[11] * pDecodedSamples[-12]; + case 11: prediction += coefficients[10] * pDecodedSamples[-11]; + case 10: prediction += coefficients[ 9] * pDecodedSamples[-10]; + case 9: prediction += coefficients[ 8] * pDecodedSamples[- 9]; + case 8: prediction += coefficients[ 7] * pDecodedSamples[- 8]; + case 7: prediction += coefficients[ 6] * pDecodedSamples[- 7]; + case 6: prediction += coefficients[ 5] * pDecodedSamples[- 6]; + case 5: prediction += coefficients[ 4] * pDecodedSamples[- 5]; + case 4: prediction += coefficients[ 3] * pDecodedSamples[- 4]; + case 3: prediction += coefficients[ 2] * pDecodedSamples[- 3]; + case 2: prediction += coefficients[ 1] * pDecodedSamples[- 2]; + case 1: prediction += coefficients[ 0] * pDecodedSamples[- 1]; + } + + return (drflac_int32)(prediction >> shift); +} + +static DRFLAC_INLINE drflac_int32 drflac__calculate_prediction_64(drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pDecodedSamples) +{ + drflac_int64 prediction; + + DRFLAC_ASSERT(order <= 32); + + /* 64-bit version. */ + + /* This method is faster on the 32-bit build when compiling with VC++. See note below. */ +#ifndef DRFLAC_64BIT + if (order == 8) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + prediction += coefficients[4] * (drflac_int64)pDecodedSamples[-5]; + prediction += coefficients[5] * (drflac_int64)pDecodedSamples[-6]; + prediction += coefficients[6] * (drflac_int64)pDecodedSamples[-7]; + prediction += coefficients[7] * (drflac_int64)pDecodedSamples[-8]; + } + else if (order == 7) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + prediction += coefficients[4] * (drflac_int64)pDecodedSamples[-5]; + prediction += coefficients[5] * (drflac_int64)pDecodedSamples[-6]; + prediction += coefficients[6] * (drflac_int64)pDecodedSamples[-7]; + } + else if (order == 3) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + } + else if (order == 6) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + prediction += coefficients[4] * (drflac_int64)pDecodedSamples[-5]; + prediction += coefficients[5] * (drflac_int64)pDecodedSamples[-6]; + } + else if (order == 5) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + prediction += coefficients[4] * (drflac_int64)pDecodedSamples[-5]; + } + else if (order == 4) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + } + else if (order == 12) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + prediction += coefficients[4] * (drflac_int64)pDecodedSamples[-5]; + prediction += coefficients[5] * (drflac_int64)pDecodedSamples[-6]; + prediction += coefficients[6] * (drflac_int64)pDecodedSamples[-7]; + prediction += coefficients[7] * (drflac_int64)pDecodedSamples[-8]; + prediction += coefficients[8] * (drflac_int64)pDecodedSamples[-9]; + prediction += coefficients[9] * (drflac_int64)pDecodedSamples[-10]; + prediction += coefficients[10] * (drflac_int64)pDecodedSamples[-11]; + prediction += coefficients[11] * (drflac_int64)pDecodedSamples[-12]; + } + else if (order == 2) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + } + else if (order == 1) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + } + else if (order == 10) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + prediction += coefficients[4] * (drflac_int64)pDecodedSamples[-5]; + prediction += coefficients[5] * (drflac_int64)pDecodedSamples[-6]; + prediction += coefficients[6] * (drflac_int64)pDecodedSamples[-7]; + prediction += coefficients[7] * (drflac_int64)pDecodedSamples[-8]; + prediction += coefficients[8] * (drflac_int64)pDecodedSamples[-9]; + prediction += coefficients[9] * (drflac_int64)pDecodedSamples[-10]; + } + else if (order == 9) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + prediction += coefficients[4] * (drflac_int64)pDecodedSamples[-5]; + prediction += coefficients[5] * (drflac_int64)pDecodedSamples[-6]; + prediction += coefficients[6] * (drflac_int64)pDecodedSamples[-7]; + prediction += coefficients[7] * (drflac_int64)pDecodedSamples[-8]; + prediction += coefficients[8] * (drflac_int64)pDecodedSamples[-9]; + } + else if (order == 11) + { + prediction = coefficients[0] * (drflac_int64)pDecodedSamples[-1]; + prediction += coefficients[1] * (drflac_int64)pDecodedSamples[-2]; + prediction += coefficients[2] * (drflac_int64)pDecodedSamples[-3]; + prediction += coefficients[3] * (drflac_int64)pDecodedSamples[-4]; + prediction += coefficients[4] * (drflac_int64)pDecodedSamples[-5]; + prediction += coefficients[5] * (drflac_int64)pDecodedSamples[-6]; + prediction += coefficients[6] * (drflac_int64)pDecodedSamples[-7]; + prediction += coefficients[7] * (drflac_int64)pDecodedSamples[-8]; + prediction += coefficients[8] * (drflac_int64)pDecodedSamples[-9]; + prediction += coefficients[9] * (drflac_int64)pDecodedSamples[-10]; + prediction += coefficients[10] * (drflac_int64)pDecodedSamples[-11]; + } + else + { + int j; + + prediction = 0; + for (j = 0; j < (int)order; ++j) { + prediction += coefficients[j] * (drflac_int64)pDecodedSamples[-j-1]; + } + } +#endif + + /* + VC++ optimizes this to a single jmp instruction, but only the 64-bit build. The 32-bit build generates less efficient code for some + reason. The ugly version above is faster so we'll just switch between the two depending on the target platform. + */ +#ifdef DRFLAC_64BIT + prediction = 0; + switch (order) + { + case 32: prediction += coefficients[31] * (drflac_int64)pDecodedSamples[-32]; + case 31: prediction += coefficients[30] * (drflac_int64)pDecodedSamples[-31]; + case 30: prediction += coefficients[29] * (drflac_int64)pDecodedSamples[-30]; + case 29: prediction += coefficients[28] * (drflac_int64)pDecodedSamples[-29]; + case 28: prediction += coefficients[27] * (drflac_int64)pDecodedSamples[-28]; + case 27: prediction += coefficients[26] * (drflac_int64)pDecodedSamples[-27]; + case 26: prediction += coefficients[25] * (drflac_int64)pDecodedSamples[-26]; + case 25: prediction += coefficients[24] * (drflac_int64)pDecodedSamples[-25]; + case 24: prediction += coefficients[23] * (drflac_int64)pDecodedSamples[-24]; + case 23: prediction += coefficients[22] * (drflac_int64)pDecodedSamples[-23]; + case 22: prediction += coefficients[21] * (drflac_int64)pDecodedSamples[-22]; + case 21: prediction += coefficients[20] * (drflac_int64)pDecodedSamples[-21]; + case 20: prediction += coefficients[19] * (drflac_int64)pDecodedSamples[-20]; + case 19: prediction += coefficients[18] * (drflac_int64)pDecodedSamples[-19]; + case 18: prediction += coefficients[17] * (drflac_int64)pDecodedSamples[-18]; + case 17: prediction += coefficients[16] * (drflac_int64)pDecodedSamples[-17]; + case 16: prediction += coefficients[15] * (drflac_int64)pDecodedSamples[-16]; + case 15: prediction += coefficients[14] * (drflac_int64)pDecodedSamples[-15]; + case 14: prediction += coefficients[13] * (drflac_int64)pDecodedSamples[-14]; + case 13: prediction += coefficients[12] * (drflac_int64)pDecodedSamples[-13]; + case 12: prediction += coefficients[11] * (drflac_int64)pDecodedSamples[-12]; + case 11: prediction += coefficients[10] * (drflac_int64)pDecodedSamples[-11]; + case 10: prediction += coefficients[ 9] * (drflac_int64)pDecodedSamples[-10]; + case 9: prediction += coefficients[ 8] * (drflac_int64)pDecodedSamples[- 9]; + case 8: prediction += coefficients[ 7] * (drflac_int64)pDecodedSamples[- 8]; + case 7: prediction += coefficients[ 6] * (drflac_int64)pDecodedSamples[- 7]; + case 6: prediction += coefficients[ 5] * (drflac_int64)pDecodedSamples[- 6]; + case 5: prediction += coefficients[ 4] * (drflac_int64)pDecodedSamples[- 5]; + case 4: prediction += coefficients[ 3] * (drflac_int64)pDecodedSamples[- 4]; + case 3: prediction += coefficients[ 2] * (drflac_int64)pDecodedSamples[- 3]; + case 2: prediction += coefficients[ 1] * (drflac_int64)pDecodedSamples[- 2]; + case 1: prediction += coefficients[ 0] * (drflac_int64)pDecodedSamples[- 1]; + } +#endif + + return (drflac_int32)(prediction >> shift); +} + + +#if 0 +/* +Reference implementation for reading and decoding samples with residual. This is intentionally left unoptimized for the +sake of readability and should only be used as a reference. +*/ +static drflac_bool32 drflac__decode_samples_with_residual__rice__reference(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + drflac_uint32 i; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pSamplesOut != NULL); + + for (i = 0; i < count; ++i) { + drflac_uint32 zeroCounter = 0; + for (;;) { + drflac_uint8 bit; + if (!drflac__read_uint8(bs, 1, &bit)) { + return DRFLAC_FALSE; + } + + if (bit == 0) { + zeroCounter += 1; + } else { + break; + } + } + + drflac_uint32 decodedRice; + if (riceParam > 0) { + if (!drflac__read_uint32(bs, riceParam, &decodedRice)) { + return DRFLAC_FALSE; + } + } else { + decodedRice = 0; + } + + decodedRice |= (zeroCounter << riceParam); + if ((decodedRice & 0x01)) { + decodedRice = ~(decodedRice >> 1); + } else { + decodedRice = (decodedRice >> 1); + } + + + if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) { + pSamplesOut[i] = decodedRice + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + i); + } else { + pSamplesOut[i] = decodedRice + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + i); + } + } + + return DRFLAC_TRUE; +} +#endif + +#if 0 +static drflac_bool32 drflac__read_rice_parts__reference(drflac_bs* bs, drflac_uint8 riceParam, drflac_uint32* pZeroCounterOut, drflac_uint32* pRiceParamPartOut) +{ + drflac_uint32 zeroCounter = 0; + drflac_uint32 decodedRice; + + for (;;) { + drflac_uint8 bit; + if (!drflac__read_uint8(bs, 1, &bit)) { + return DRFLAC_FALSE; + } + + if (bit == 0) { + zeroCounter += 1; + } else { + break; + } + } + + if (riceParam > 0) { + if (!drflac__read_uint32(bs, riceParam, &decodedRice)) { + return DRFLAC_FALSE; + } + } else { + decodedRice = 0; + } + + *pZeroCounterOut = zeroCounter; + *pRiceParamPartOut = decodedRice; + return DRFLAC_TRUE; +} +#endif + +#if 0 +static DRFLAC_INLINE drflac_bool32 drflac__read_rice_parts(drflac_bs* bs, drflac_uint8 riceParam, drflac_uint32* pZeroCounterOut, drflac_uint32* pRiceParamPartOut) +{ + drflac_cache_t riceParamMask; + drflac_uint32 zeroCounter; + drflac_uint32 setBitOffsetPlus1; + drflac_uint32 riceParamPart; + drflac_uint32 riceLength; + + DRFLAC_ASSERT(riceParam > 0); /* <-- riceParam should never be 0. drflac__read_rice_parts__param_equals_zero() should be used instead for this case. */ + + riceParamMask = DRFLAC_CACHE_L1_SELECTION_MASK(riceParam); + + zeroCounter = 0; + while (bs->cache == 0) { + zeroCounter += (drflac_uint32)DRFLAC_CACHE_L1_BITS_REMAINING(bs); + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + } + + setBitOffsetPlus1 = drflac__clz(bs->cache); + zeroCounter += setBitOffsetPlus1; + setBitOffsetPlus1 += 1; + + riceLength = setBitOffsetPlus1 + riceParam; + if (riceLength < DRFLAC_CACHE_L1_BITS_REMAINING(bs)) { + riceParamPart = (drflac_uint32)((bs->cache & (riceParamMask >> setBitOffsetPlus1)) >> DRFLAC_CACHE_L1_SELECTION_SHIFT(bs, riceLength)); + + bs->consumedBits += riceLength; + bs->cache <<= riceLength; + } else { + drflac_uint32 bitCountLo; + drflac_cache_t resultHi; + + bs->consumedBits += riceLength; + bs->cache <<= setBitOffsetPlus1 & (DRFLAC_CACHE_L1_SIZE_BITS(bs)-1); /* <-- Equivalent to "if (setBitOffsetPlus1 < DRFLAC_CACHE_L1_SIZE_BITS(bs)) { bs->cache <<= setBitOffsetPlus1; }" */ + + /* It straddles the cached data. It will never cover more than the next chunk. We just read the number in two parts and combine them. */ + bitCountLo = bs->consumedBits - DRFLAC_CACHE_L1_SIZE_BITS(bs); + resultHi = DRFLAC_CACHE_L1_SELECT_AND_SHIFT(bs, riceParam); /* <-- Use DRFLAC_CACHE_L1_SELECT_AND_SHIFT_SAFE() if ever this function allows riceParam=0. */ + + if (bs->nextL2Line < DRFLAC_CACHE_L2_LINE_COUNT(bs)) { +#ifndef DR_FLAC_NO_CRC + drflac__update_crc16(bs); +#endif + bs->cache = drflac__be2host__cache_line(bs->cacheL2[bs->nextL2Line++]); + bs->consumedBits = 0; +#ifndef DR_FLAC_NO_CRC + bs->crc16Cache = bs->cache; +#endif + } else { + /* Slow path. We need to fetch more data from the client. */ + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + if (bitCountLo > DRFLAC_CACHE_L1_BITS_REMAINING(bs)) { + /* This happens when we get to end of stream */ + return DRFLAC_FALSE; + } + } + + riceParamPart = (drflac_uint32)(resultHi | DRFLAC_CACHE_L1_SELECT_AND_SHIFT_SAFE(bs, bitCountLo)); + + bs->consumedBits += bitCountLo; + bs->cache <<= bitCountLo; + } + + pZeroCounterOut[0] = zeroCounter; + pRiceParamPartOut[0] = riceParamPart; + + return DRFLAC_TRUE; +} +#endif + +static DRFLAC_INLINE drflac_bool32 drflac__read_rice_parts_x1(drflac_bs* bs, drflac_uint8 riceParam, drflac_uint32* pZeroCounterOut, drflac_uint32* pRiceParamPartOut) +{ + drflac_uint32 riceParamPlus1 = riceParam + 1; + /*drflac_cache_t riceParamPlus1Mask = DRFLAC_CACHE_L1_SELECTION_MASK(riceParamPlus1);*/ + drflac_uint32 riceParamPlus1Shift = DRFLAC_CACHE_L1_SELECTION_SHIFT(bs, riceParamPlus1); + drflac_uint32 riceParamPlus1MaxConsumedBits = DRFLAC_CACHE_L1_SIZE_BITS(bs) - riceParamPlus1; + + /* + The idea here is to use local variables for the cache in an attempt to encourage the compiler to store them in registers. I have + no idea how this will work in practice... + */ + drflac_cache_t bs_cache = bs->cache; + drflac_uint32 bs_consumedBits = bs->consumedBits; + + /* The first thing to do is find the first unset bit. Most likely a bit will be set in the current cache line. */ + drflac_uint32 lzcount = drflac__clz(bs_cache); + if (lzcount < sizeof(bs_cache)*8) { + pZeroCounterOut[0] = lzcount; + + /* + It is most likely that the riceParam part (which comes after the zero counter) is also on this cache line. When extracting + this, we include the set bit from the unary coded part because it simplifies cache management. This bit will be handled + outside of this function at a higher level. + */ + extract_rice_param_part: + bs_cache <<= lzcount; + bs_consumedBits += lzcount; + + if (bs_consumedBits <= riceParamPlus1MaxConsumedBits) { + /* Getting here means the rice parameter part is wholly contained within the current cache line. */ + pRiceParamPartOut[0] = (drflac_uint32)(bs_cache >> riceParamPlus1Shift); + bs_cache <<= riceParamPlus1; + bs_consumedBits += riceParamPlus1; + } else { + drflac_uint32 riceParamPartHi; + drflac_uint32 riceParamPartLo; + drflac_uint32 riceParamPartLoBitCount; + + /* + Getting here means the rice parameter part straddles the cache line. We need to read from the tail of the current cache + line, reload the cache, and then combine it with the head of the next cache line. + */ + + /* Grab the high part of the rice parameter part. */ + riceParamPartHi = (drflac_uint32)(bs_cache >> riceParamPlus1Shift); + + /* Before reloading the cache we need to grab the size in bits of the low part. */ + riceParamPartLoBitCount = bs_consumedBits - riceParamPlus1MaxConsumedBits; + DRFLAC_ASSERT(riceParamPartLoBitCount > 0 && riceParamPartLoBitCount < 32); + + /* Now reload the cache. */ + if (bs->nextL2Line < DRFLAC_CACHE_L2_LINE_COUNT(bs)) { + #ifndef DR_FLAC_NO_CRC + drflac__update_crc16(bs); + #endif + bs_cache = drflac__be2host__cache_line(bs->cacheL2[bs->nextL2Line++]); + bs_consumedBits = riceParamPartLoBitCount; + #ifndef DR_FLAC_NO_CRC + bs->crc16Cache = bs_cache; + #endif + } else { + /* Slow path. We need to fetch more data from the client. */ + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + if (riceParamPartLoBitCount > DRFLAC_CACHE_L1_BITS_REMAINING(bs)) { + /* This happens when we get to end of stream */ + return DRFLAC_FALSE; + } + + bs_cache = bs->cache; + bs_consumedBits = bs->consumedBits + riceParamPartLoBitCount; + } + + /* We should now have enough information to construct the rice parameter part. */ + riceParamPartLo = (drflac_uint32)(bs_cache >> (DRFLAC_CACHE_L1_SELECTION_SHIFT(bs, riceParamPartLoBitCount))); + pRiceParamPartOut[0] = riceParamPartHi | riceParamPartLo; + + bs_cache <<= riceParamPartLoBitCount; + } + } else { + /* + Getting here means there are no bits set on the cache line. This is a less optimal case because we just wasted a call + to drflac__clz() and we need to reload the cache. + */ + drflac_uint32 zeroCounter = (drflac_uint32)(DRFLAC_CACHE_L1_SIZE_BITS(bs) - bs_consumedBits); + for (;;) { + if (bs->nextL2Line < DRFLAC_CACHE_L2_LINE_COUNT(bs)) { + #ifndef DR_FLAC_NO_CRC + drflac__update_crc16(bs); + #endif + bs_cache = drflac__be2host__cache_line(bs->cacheL2[bs->nextL2Line++]); + bs_consumedBits = 0; + #ifndef DR_FLAC_NO_CRC + bs->crc16Cache = bs_cache; + #endif + } else { + /* Slow path. We need to fetch more data from the client. */ + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + + bs_cache = bs->cache; + bs_consumedBits = bs->consumedBits; + } + + lzcount = drflac__clz(bs_cache); + zeroCounter += lzcount; + + if (lzcount < sizeof(bs_cache)*8) { + break; + } + } + + pZeroCounterOut[0] = zeroCounter; + goto extract_rice_param_part; + } + + /* Make sure the cache is restored at the end of it all. */ + bs->cache = bs_cache; + bs->consumedBits = bs_consumedBits; + + return DRFLAC_TRUE; +} + +static DRFLAC_INLINE drflac_bool32 drflac__seek_rice_parts(drflac_bs* bs, drflac_uint8 riceParam) +{ + drflac_uint32 riceParamPlus1 = riceParam + 1; + drflac_uint32 riceParamPlus1MaxConsumedBits = DRFLAC_CACHE_L1_SIZE_BITS(bs) - riceParamPlus1; + + /* + The idea here is to use local variables for the cache in an attempt to encourage the compiler to store them in registers. I have + no idea how this will work in practice... + */ + drflac_cache_t bs_cache = bs->cache; + drflac_uint32 bs_consumedBits = bs->consumedBits; + + /* The first thing to do is find the first unset bit. Most likely a bit will be set in the current cache line. */ + drflac_uint32 lzcount = drflac__clz(bs_cache); + if (lzcount < sizeof(bs_cache)*8) { + /* + It is most likely that the riceParam part (which comes after the zero counter) is also on this cache line. When extracting + this, we include the set bit from the unary coded part because it simplifies cache management. This bit will be handled + outside of this function at a higher level. + */ + extract_rice_param_part: + bs_cache <<= lzcount; + bs_consumedBits += lzcount; + + if (bs_consumedBits <= riceParamPlus1MaxConsumedBits) { + /* Getting here means the rice parameter part is wholly contained within the current cache line. */ + bs_cache <<= riceParamPlus1; + bs_consumedBits += riceParamPlus1; + } else { + /* + Getting here means the rice parameter part straddles the cache line. We need to read from the tail of the current cache + line, reload the cache, and then combine it with the head of the next cache line. + */ + + /* Before reloading the cache we need to grab the size in bits of the low part. */ + drflac_uint32 riceParamPartLoBitCount = bs_consumedBits - riceParamPlus1MaxConsumedBits; + DRFLAC_ASSERT(riceParamPartLoBitCount > 0 && riceParamPartLoBitCount < 32); + + /* Now reload the cache. */ + if (bs->nextL2Line < DRFLAC_CACHE_L2_LINE_COUNT(bs)) { + #ifndef DR_FLAC_NO_CRC + drflac__update_crc16(bs); + #endif + bs_cache = drflac__be2host__cache_line(bs->cacheL2[bs->nextL2Line++]); + bs_consumedBits = riceParamPartLoBitCount; + #ifndef DR_FLAC_NO_CRC + bs->crc16Cache = bs_cache; + #endif + } else { + /* Slow path. We need to fetch more data from the client. */ + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + + if (riceParamPartLoBitCount > DRFLAC_CACHE_L1_BITS_REMAINING(bs)) { + /* This happens when we get to end of stream */ + return DRFLAC_FALSE; + } + + bs_cache = bs->cache; + bs_consumedBits = bs->consumedBits + riceParamPartLoBitCount; + } + + bs_cache <<= riceParamPartLoBitCount; + } + } else { + /* + Getting here means there are no bits set on the cache line. This is a less optimal case because we just wasted a call + to drflac__clz() and we need to reload the cache. + */ + for (;;) { + if (bs->nextL2Line < DRFLAC_CACHE_L2_LINE_COUNT(bs)) { + #ifndef DR_FLAC_NO_CRC + drflac__update_crc16(bs); + #endif + bs_cache = drflac__be2host__cache_line(bs->cacheL2[bs->nextL2Line++]); + bs_consumedBits = 0; + #ifndef DR_FLAC_NO_CRC + bs->crc16Cache = bs_cache; + #endif + } else { + /* Slow path. We need to fetch more data from the client. */ + if (!drflac__reload_cache(bs)) { + return DRFLAC_FALSE; + } + + bs_cache = bs->cache; + bs_consumedBits = bs->consumedBits; + } + + lzcount = drflac__clz(bs_cache); + if (lzcount < sizeof(bs_cache)*8) { + break; + } + } + + goto extract_rice_param_part; + } + + /* Make sure the cache is restored at the end of it all. */ + bs->cache = bs_cache; + bs->consumedBits = bs_consumedBits; + + return DRFLAC_TRUE; +} + + +static drflac_bool32 drflac__decode_samples_with_residual__rice__scalar_zeroorder(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + drflac_uint32 t[2] = {0x00000000, 0xFFFFFFFF}; + drflac_uint32 zeroCountPart0; + drflac_uint32 riceParamPart0; + drflac_uint32 riceParamMask; + drflac_uint32 i; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pSamplesOut != NULL); + + (void)bitsPerSample; + (void)order; + (void)shift; + (void)coefficients; + + riceParamMask = (drflac_uint32)~((~0UL) << riceParam); + + i = 0; + while (i < count) { + /* Rice extraction. */ + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart0, &riceParamPart0)) { + return DRFLAC_FALSE; + } + + /* Rice reconstruction. */ + riceParamPart0 &= riceParamMask; + riceParamPart0 |= (zeroCountPart0 << riceParam); + riceParamPart0 = (riceParamPart0 >> 1) ^ t[riceParamPart0 & 0x01]; + + pSamplesOut[i] = riceParamPart0; + + i += 1; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_samples_with_residual__rice__scalar(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + drflac_uint32 t[2] = {0x00000000, 0xFFFFFFFF}; + drflac_uint32 zeroCountPart0 = 0; + drflac_uint32 zeroCountPart1 = 0; + drflac_uint32 zeroCountPart2 = 0; + drflac_uint32 zeroCountPart3 = 0; + drflac_uint32 riceParamPart0 = 0; + drflac_uint32 riceParamPart1 = 0; + drflac_uint32 riceParamPart2 = 0; + drflac_uint32 riceParamPart3 = 0; + drflac_uint32 riceParamMask; + const drflac_int32* pSamplesOutEnd; + drflac_uint32 i; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pSamplesOut != NULL); + + if (lpcOrder == 0) { + return drflac__decode_samples_with_residual__rice__scalar_zeroorder(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, coefficients, pSamplesOut); + } + + riceParamMask = (drflac_uint32)~((~0UL) << riceParam); + pSamplesOutEnd = pSamplesOut + (count & ~3); + + if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) { + while (pSamplesOut < pSamplesOutEnd) { + /* + Rice extraction. It's faster to do this one at a time against local variables than it is to use the x4 version + against an array. Not sure why, but perhaps it's making more efficient use of registers? + */ + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart0, &riceParamPart0) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart1, &riceParamPart1) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart2, &riceParamPart2) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart3, &riceParamPart3)) { + return DRFLAC_FALSE; + } + + riceParamPart0 &= riceParamMask; + riceParamPart1 &= riceParamMask; + riceParamPart2 &= riceParamMask; + riceParamPart3 &= riceParamMask; + + riceParamPart0 |= (zeroCountPart0 << riceParam); + riceParamPart1 |= (zeroCountPart1 << riceParam); + riceParamPart2 |= (zeroCountPart2 << riceParam); + riceParamPart3 |= (zeroCountPart3 << riceParam); + + riceParamPart0 = (riceParamPart0 >> 1) ^ t[riceParamPart0 & 0x01]; + riceParamPart1 = (riceParamPart1 >> 1) ^ t[riceParamPart1 & 0x01]; + riceParamPart2 = (riceParamPart2 >> 1) ^ t[riceParamPart2 & 0x01]; + riceParamPart3 = (riceParamPart3 >> 1) ^ t[riceParamPart3 & 0x01]; + + pSamplesOut[0] = riceParamPart0 + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + 0); + pSamplesOut[1] = riceParamPart1 + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + 1); + pSamplesOut[2] = riceParamPart2 + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + 2); + pSamplesOut[3] = riceParamPart3 + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + 3); + + pSamplesOut += 4; + } + } else { + while (pSamplesOut < pSamplesOutEnd) { + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart0, &riceParamPart0) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart1, &riceParamPart1) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart2, &riceParamPart2) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart3, &riceParamPart3)) { + return DRFLAC_FALSE; + } + + riceParamPart0 &= riceParamMask; + riceParamPart1 &= riceParamMask; + riceParamPart2 &= riceParamMask; + riceParamPart3 &= riceParamMask; + + riceParamPart0 |= (zeroCountPart0 << riceParam); + riceParamPart1 |= (zeroCountPart1 << riceParam); + riceParamPart2 |= (zeroCountPart2 << riceParam); + riceParamPart3 |= (zeroCountPart3 << riceParam); + + riceParamPart0 = (riceParamPart0 >> 1) ^ t[riceParamPart0 & 0x01]; + riceParamPart1 = (riceParamPart1 >> 1) ^ t[riceParamPart1 & 0x01]; + riceParamPart2 = (riceParamPart2 >> 1) ^ t[riceParamPart2 & 0x01]; + riceParamPart3 = (riceParamPart3 >> 1) ^ t[riceParamPart3 & 0x01]; + + pSamplesOut[0] = riceParamPart0 + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + 0); + pSamplesOut[1] = riceParamPart1 + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + 1); + pSamplesOut[2] = riceParamPart2 + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + 2); + pSamplesOut[3] = riceParamPart3 + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + 3); + + pSamplesOut += 4; + } + } + + i = (count & ~3); + while (i < count) { + /* Rice extraction. */ + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountPart0, &riceParamPart0)) { + return DRFLAC_FALSE; + } + + /* Rice reconstruction. */ + riceParamPart0 &= riceParamMask; + riceParamPart0 |= (zeroCountPart0 << riceParam); + riceParamPart0 = (riceParamPart0 >> 1) ^ t[riceParamPart0 & 0x01]; + /*riceParamPart0 = (riceParamPart0 >> 1) ^ (~(riceParamPart0 & 0x01) + 1);*/ + + /* Sample reconstruction. */ + if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) { + pSamplesOut[0] = riceParamPart0 + drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + 0); + } else { + pSamplesOut[0] = riceParamPart0 + drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + 0); + } + + i += 1; + pSamplesOut += 1; + } + + return DRFLAC_TRUE; +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE __m128i drflac__mm_packs_interleaved_epi32(__m128i a, __m128i b) +{ + __m128i r; + + /* Pack. */ + r = _mm_packs_epi32(a, b); + + /* a3a2 a1a0 b3b2 b1b0 -> a3a2 b3b2 a1a0 b1b0 */ + r = _mm_shuffle_epi32(r, _MM_SHUFFLE(3, 1, 2, 0)); + + /* a3a2 b3b2 a1a0 b1b0 -> a3b3 a2b2 a1b1 a0b0 */ + r = _mm_shufflehi_epi16(r, _MM_SHUFFLE(3, 1, 2, 0)); + r = _mm_shufflelo_epi16(r, _MM_SHUFFLE(3, 1, 2, 0)); + + return r; +} +#endif + +#if defined(DRFLAC_SUPPORT_SSE41) +static DRFLAC_INLINE __m128i drflac__mm_not_si128(__m128i a) +{ + return _mm_xor_si128(a, _mm_cmpeq_epi32(_mm_setzero_si128(), _mm_setzero_si128())); +} + +static DRFLAC_INLINE __m128i drflac__mm_hadd_epi32(__m128i x) +{ + __m128i x64 = _mm_add_epi32(x, _mm_shuffle_epi32(x, _MM_SHUFFLE(1, 0, 3, 2))); + __m128i x32 = _mm_shufflelo_epi16(x64, _MM_SHUFFLE(1, 0, 3, 2)); + return _mm_add_epi32(x64, x32); +} + +static DRFLAC_INLINE __m128i drflac__mm_hadd_epi64(__m128i x) +{ + return _mm_add_epi64(x, _mm_shuffle_epi32(x, _MM_SHUFFLE(1, 0, 3, 2))); +} + +static DRFLAC_INLINE __m128i drflac__mm_srai_epi64(__m128i x, int count) +{ + /* + To simplify this we are assuming count < 32. This restriction allows us to work on a low side and a high side. The low side + is shifted with zero bits, whereas the right side is shifted with sign bits. + */ + __m128i lo = _mm_srli_epi64(x, count); + __m128i hi = _mm_srai_epi32(x, count); + + hi = _mm_and_si128(hi, _mm_set_epi32(0xFFFFFFFF, 0, 0xFFFFFFFF, 0)); /* The high part needs to have the low part cleared. */ + + return _mm_or_si128(lo, hi); +} + +static drflac_bool32 drflac__decode_samples_with_residual__rice__sse41_32(drflac_bs* bs, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + int i; + drflac_uint32 riceParamMask; + drflac_int32* pDecodedSamples = pSamplesOut; + drflac_int32* pDecodedSamplesEnd = pSamplesOut + (count & ~3); + drflac_uint32 zeroCountParts0 = 0; + drflac_uint32 zeroCountParts1 = 0; + drflac_uint32 zeroCountParts2 = 0; + drflac_uint32 zeroCountParts3 = 0; + drflac_uint32 riceParamParts0 = 0; + drflac_uint32 riceParamParts1 = 0; + drflac_uint32 riceParamParts2 = 0; + drflac_uint32 riceParamParts3 = 0; + __m128i coefficients128_0; + __m128i coefficients128_4; + __m128i coefficients128_8; + __m128i samples128_0; + __m128i samples128_4; + __m128i samples128_8; + __m128i riceParamMask128; + + const drflac_uint32 t[2] = {0x00000000, 0xFFFFFFFF}; + + riceParamMask = (drflac_uint32)~((~0UL) << riceParam); + riceParamMask128 = _mm_set1_epi32(riceParamMask); + + /* Pre-load. */ + coefficients128_0 = _mm_setzero_si128(); + coefficients128_4 = _mm_setzero_si128(); + coefficients128_8 = _mm_setzero_si128(); + + samples128_0 = _mm_setzero_si128(); + samples128_4 = _mm_setzero_si128(); + samples128_8 = _mm_setzero_si128(); + + /* + Pre-loading the coefficients and prior samples is annoying because we need to ensure we don't try reading more than + what's available in the input buffers. It would be convenient to use a fall-through switch to do this, but this results + in strict aliasing warnings with GCC. To work around this I'm just doing something hacky. This feels a bit convoluted + so I think there's opportunity for this to be simplified. + */ +#if 1 + { + int runningOrder = order; + + /* 0 - 3. */ + if (runningOrder >= 4) { + coefficients128_0 = _mm_loadu_si128((const __m128i*)(coefficients + 0)); + samples128_0 = _mm_loadu_si128((const __m128i*)(pSamplesOut - 4)); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: coefficients128_0 = _mm_set_epi32(0, coefficients[2], coefficients[1], coefficients[0]); samples128_0 = _mm_set_epi32(pSamplesOut[-1], pSamplesOut[-2], pSamplesOut[-3], 0); break; + case 2: coefficients128_0 = _mm_set_epi32(0, 0, coefficients[1], coefficients[0]); samples128_0 = _mm_set_epi32(pSamplesOut[-1], pSamplesOut[-2], 0, 0); break; + case 1: coefficients128_0 = _mm_set_epi32(0, 0, 0, coefficients[0]); samples128_0 = _mm_set_epi32(pSamplesOut[-1], 0, 0, 0); break; + } + runningOrder = 0; + } + + /* 4 - 7 */ + if (runningOrder >= 4) { + coefficients128_4 = _mm_loadu_si128((const __m128i*)(coefficients + 4)); + samples128_4 = _mm_loadu_si128((const __m128i*)(pSamplesOut - 8)); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: coefficients128_4 = _mm_set_epi32(0, coefficients[6], coefficients[5], coefficients[4]); samples128_4 = _mm_set_epi32(pSamplesOut[-5], pSamplesOut[-6], pSamplesOut[-7], 0); break; + case 2: coefficients128_4 = _mm_set_epi32(0, 0, coefficients[5], coefficients[4]); samples128_4 = _mm_set_epi32(pSamplesOut[-5], pSamplesOut[-6], 0, 0); break; + case 1: coefficients128_4 = _mm_set_epi32(0, 0, 0, coefficients[4]); samples128_4 = _mm_set_epi32(pSamplesOut[-5], 0, 0, 0); break; + } + runningOrder = 0; + } + + /* 8 - 11 */ + if (runningOrder == 4) { + coefficients128_8 = _mm_loadu_si128((const __m128i*)(coefficients + 8)); + samples128_8 = _mm_loadu_si128((const __m128i*)(pSamplesOut - 12)); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: coefficients128_8 = _mm_set_epi32(0, coefficients[10], coefficients[9], coefficients[8]); samples128_8 = _mm_set_epi32(pSamplesOut[-9], pSamplesOut[-10], pSamplesOut[-11], 0); break; + case 2: coefficients128_8 = _mm_set_epi32(0, 0, coefficients[9], coefficients[8]); samples128_8 = _mm_set_epi32(pSamplesOut[-9], pSamplesOut[-10], 0, 0); break; + case 1: coefficients128_8 = _mm_set_epi32(0, 0, 0, coefficients[8]); samples128_8 = _mm_set_epi32(pSamplesOut[-9], 0, 0, 0); break; + } + runningOrder = 0; + } + + /* Coefficients need to be shuffled for our streaming algorithm below to work. Samples are already in the correct order from the loading routine above. */ + coefficients128_0 = _mm_shuffle_epi32(coefficients128_0, _MM_SHUFFLE(0, 1, 2, 3)); + coefficients128_4 = _mm_shuffle_epi32(coefficients128_4, _MM_SHUFFLE(0, 1, 2, 3)); + coefficients128_8 = _mm_shuffle_epi32(coefficients128_8, _MM_SHUFFLE(0, 1, 2, 3)); + } +#else + /* This causes strict-aliasing warnings with GCC. */ + switch (order) + { + case 12: ((drflac_int32*)&coefficients128_8)[0] = coefficients[11]; ((drflac_int32*)&samples128_8)[0] = pDecodedSamples[-12]; + case 11: ((drflac_int32*)&coefficients128_8)[1] = coefficients[10]; ((drflac_int32*)&samples128_8)[1] = pDecodedSamples[-11]; + case 10: ((drflac_int32*)&coefficients128_8)[2] = coefficients[ 9]; ((drflac_int32*)&samples128_8)[2] = pDecodedSamples[-10]; + case 9: ((drflac_int32*)&coefficients128_8)[3] = coefficients[ 8]; ((drflac_int32*)&samples128_8)[3] = pDecodedSamples[- 9]; + case 8: ((drflac_int32*)&coefficients128_4)[0] = coefficients[ 7]; ((drflac_int32*)&samples128_4)[0] = pDecodedSamples[- 8]; + case 7: ((drflac_int32*)&coefficients128_4)[1] = coefficients[ 6]; ((drflac_int32*)&samples128_4)[1] = pDecodedSamples[- 7]; + case 6: ((drflac_int32*)&coefficients128_4)[2] = coefficients[ 5]; ((drflac_int32*)&samples128_4)[2] = pDecodedSamples[- 6]; + case 5: ((drflac_int32*)&coefficients128_4)[3] = coefficients[ 4]; ((drflac_int32*)&samples128_4)[3] = pDecodedSamples[- 5]; + case 4: ((drflac_int32*)&coefficients128_0)[0] = coefficients[ 3]; ((drflac_int32*)&samples128_0)[0] = pDecodedSamples[- 4]; + case 3: ((drflac_int32*)&coefficients128_0)[1] = coefficients[ 2]; ((drflac_int32*)&samples128_0)[1] = pDecodedSamples[- 3]; + case 2: ((drflac_int32*)&coefficients128_0)[2] = coefficients[ 1]; ((drflac_int32*)&samples128_0)[2] = pDecodedSamples[- 2]; + case 1: ((drflac_int32*)&coefficients128_0)[3] = coefficients[ 0]; ((drflac_int32*)&samples128_0)[3] = pDecodedSamples[- 1]; + } +#endif + + /* For this version we are doing one sample at a time. */ + while (pDecodedSamples < pDecodedSamplesEnd) { + __m128i prediction128; + __m128i zeroCountPart128; + __m128i riceParamPart128; + + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts0, &riceParamParts0) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts1, &riceParamParts1) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts2, &riceParamParts2) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts3, &riceParamParts3)) { + return DRFLAC_FALSE; + } + + zeroCountPart128 = _mm_set_epi32(zeroCountParts3, zeroCountParts2, zeroCountParts1, zeroCountParts0); + riceParamPart128 = _mm_set_epi32(riceParamParts3, riceParamParts2, riceParamParts1, riceParamParts0); + + riceParamPart128 = _mm_and_si128(riceParamPart128, riceParamMask128); + riceParamPart128 = _mm_or_si128(riceParamPart128, _mm_slli_epi32(zeroCountPart128, riceParam)); + riceParamPart128 = _mm_xor_si128(_mm_srli_epi32(riceParamPart128, 1), _mm_add_epi32(drflac__mm_not_si128(_mm_and_si128(riceParamPart128, _mm_set1_epi32(0x01))), _mm_set1_epi32(0x01))); /* <-- SSE2 compatible */ + /*riceParamPart128 = _mm_xor_si128(_mm_srli_epi32(riceParamPart128, 1), _mm_mullo_epi32(_mm_and_si128(riceParamPart128, _mm_set1_epi32(0x01)), _mm_set1_epi32(0xFFFFFFFF)));*/ /* <-- Only supported from SSE4.1 and is slower in my testing... */ + + if (order <= 4) { + for (i = 0; i < 4; i += 1) { + prediction128 = _mm_mullo_epi32(coefficients128_0, samples128_0); + + /* Horizontal add and shift. */ + prediction128 = drflac__mm_hadd_epi32(prediction128); + prediction128 = _mm_srai_epi32(prediction128, shift); + prediction128 = _mm_add_epi32(riceParamPart128, prediction128); + + samples128_0 = _mm_alignr_epi8(prediction128, samples128_0, 4); + riceParamPart128 = _mm_alignr_epi8(_mm_setzero_si128(), riceParamPart128, 4); + } + } else if (order <= 8) { + for (i = 0; i < 4; i += 1) { + prediction128 = _mm_mullo_epi32(coefficients128_4, samples128_4); + prediction128 = _mm_add_epi32(prediction128, _mm_mullo_epi32(coefficients128_0, samples128_0)); + + /* Horizontal add and shift. */ + prediction128 = drflac__mm_hadd_epi32(prediction128); + prediction128 = _mm_srai_epi32(prediction128, shift); + prediction128 = _mm_add_epi32(riceParamPart128, prediction128); + + samples128_4 = _mm_alignr_epi8(samples128_0, samples128_4, 4); + samples128_0 = _mm_alignr_epi8(prediction128, samples128_0, 4); + riceParamPart128 = _mm_alignr_epi8(_mm_setzero_si128(), riceParamPart128, 4); + } + } else { + for (i = 0; i < 4; i += 1) { + prediction128 = _mm_mullo_epi32(coefficients128_8, samples128_8); + prediction128 = _mm_add_epi32(prediction128, _mm_mullo_epi32(coefficients128_4, samples128_4)); + prediction128 = _mm_add_epi32(prediction128, _mm_mullo_epi32(coefficients128_0, samples128_0)); + + /* Horizontal add and shift. */ + prediction128 = drflac__mm_hadd_epi32(prediction128); + prediction128 = _mm_srai_epi32(prediction128, shift); + prediction128 = _mm_add_epi32(riceParamPart128, prediction128); + + samples128_8 = _mm_alignr_epi8(samples128_4, samples128_8, 4); + samples128_4 = _mm_alignr_epi8(samples128_0, samples128_4, 4); + samples128_0 = _mm_alignr_epi8(prediction128, samples128_0, 4); + riceParamPart128 = _mm_alignr_epi8(_mm_setzero_si128(), riceParamPart128, 4); + } + } + + /* We store samples in groups of 4. */ + _mm_storeu_si128((__m128i*)pDecodedSamples, samples128_0); + pDecodedSamples += 4; + } + + /* Make sure we process the last few samples. */ + i = (count & ~3); + while (i < (int)count) { + /* Rice extraction. */ + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts0, &riceParamParts0)) { + return DRFLAC_FALSE; + } + + /* Rice reconstruction. */ + riceParamParts0 &= riceParamMask; + riceParamParts0 |= (zeroCountParts0 << riceParam); + riceParamParts0 = (riceParamParts0 >> 1) ^ t[riceParamParts0 & 0x01]; + + /* Sample reconstruction. */ + pDecodedSamples[0] = riceParamParts0 + drflac__calculate_prediction_32(order, shift, coefficients, pDecodedSamples); + + i += 1; + pDecodedSamples += 1; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_samples_with_residual__rice__sse41_64(drflac_bs* bs, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + int i; + drflac_uint32 riceParamMask; + drflac_int32* pDecodedSamples = pSamplesOut; + drflac_int32* pDecodedSamplesEnd = pSamplesOut + (count & ~3); + drflac_uint32 zeroCountParts0 = 0; + drflac_uint32 zeroCountParts1 = 0; + drflac_uint32 zeroCountParts2 = 0; + drflac_uint32 zeroCountParts3 = 0; + drflac_uint32 riceParamParts0 = 0; + drflac_uint32 riceParamParts1 = 0; + drflac_uint32 riceParamParts2 = 0; + drflac_uint32 riceParamParts3 = 0; + __m128i coefficients128_0; + __m128i coefficients128_4; + __m128i coefficients128_8; + __m128i samples128_0; + __m128i samples128_4; + __m128i samples128_8; + __m128i prediction128; + __m128i riceParamMask128; + + const drflac_uint32 t[2] = {0x00000000, 0xFFFFFFFF}; + + DRFLAC_ASSERT(order <= 12); + + riceParamMask = (drflac_uint32)~((~0UL) << riceParam); + riceParamMask128 = _mm_set1_epi32(riceParamMask); + + prediction128 = _mm_setzero_si128(); + + /* Pre-load. */ + coefficients128_0 = _mm_setzero_si128(); + coefficients128_4 = _mm_setzero_si128(); + coefficients128_8 = _mm_setzero_si128(); + + samples128_0 = _mm_setzero_si128(); + samples128_4 = _mm_setzero_si128(); + samples128_8 = _mm_setzero_si128(); + +#if 1 + { + int runningOrder = order; + + /* 0 - 3. */ + if (runningOrder >= 4) { + coefficients128_0 = _mm_loadu_si128((const __m128i*)(coefficients + 0)); + samples128_0 = _mm_loadu_si128((const __m128i*)(pSamplesOut - 4)); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: coefficients128_0 = _mm_set_epi32(0, coefficients[2], coefficients[1], coefficients[0]); samples128_0 = _mm_set_epi32(pSamplesOut[-1], pSamplesOut[-2], pSamplesOut[-3], 0); break; + case 2: coefficients128_0 = _mm_set_epi32(0, 0, coefficients[1], coefficients[0]); samples128_0 = _mm_set_epi32(pSamplesOut[-1], pSamplesOut[-2], 0, 0); break; + case 1: coefficients128_0 = _mm_set_epi32(0, 0, 0, coefficients[0]); samples128_0 = _mm_set_epi32(pSamplesOut[-1], 0, 0, 0); break; + } + runningOrder = 0; + } + + /* 4 - 7 */ + if (runningOrder >= 4) { + coefficients128_4 = _mm_loadu_si128((const __m128i*)(coefficients + 4)); + samples128_4 = _mm_loadu_si128((const __m128i*)(pSamplesOut - 8)); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: coefficients128_4 = _mm_set_epi32(0, coefficients[6], coefficients[5], coefficients[4]); samples128_4 = _mm_set_epi32(pSamplesOut[-5], pSamplesOut[-6], pSamplesOut[-7], 0); break; + case 2: coefficients128_4 = _mm_set_epi32(0, 0, coefficients[5], coefficients[4]); samples128_4 = _mm_set_epi32(pSamplesOut[-5], pSamplesOut[-6], 0, 0); break; + case 1: coefficients128_4 = _mm_set_epi32(0, 0, 0, coefficients[4]); samples128_4 = _mm_set_epi32(pSamplesOut[-5], 0, 0, 0); break; + } + runningOrder = 0; + } + + /* 8 - 11 */ + if (runningOrder == 4) { + coefficients128_8 = _mm_loadu_si128((const __m128i*)(coefficients + 8)); + samples128_8 = _mm_loadu_si128((const __m128i*)(pSamplesOut - 12)); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: coefficients128_8 = _mm_set_epi32(0, coefficients[10], coefficients[9], coefficients[8]); samples128_8 = _mm_set_epi32(pSamplesOut[-9], pSamplesOut[-10], pSamplesOut[-11], 0); break; + case 2: coefficients128_8 = _mm_set_epi32(0, 0, coefficients[9], coefficients[8]); samples128_8 = _mm_set_epi32(pSamplesOut[-9], pSamplesOut[-10], 0, 0); break; + case 1: coefficients128_8 = _mm_set_epi32(0, 0, 0, coefficients[8]); samples128_8 = _mm_set_epi32(pSamplesOut[-9], 0, 0, 0); break; + } + runningOrder = 0; + } + + /* Coefficients need to be shuffled for our streaming algorithm below to work. Samples are already in the correct order from the loading routine above. */ + coefficients128_0 = _mm_shuffle_epi32(coefficients128_0, _MM_SHUFFLE(0, 1, 2, 3)); + coefficients128_4 = _mm_shuffle_epi32(coefficients128_4, _MM_SHUFFLE(0, 1, 2, 3)); + coefficients128_8 = _mm_shuffle_epi32(coefficients128_8, _MM_SHUFFLE(0, 1, 2, 3)); + } +#else + switch (order) + { + case 12: ((drflac_int32*)&coefficients128_8)[0] = coefficients[11]; ((drflac_int32*)&samples128_8)[0] = pDecodedSamples[-12]; + case 11: ((drflac_int32*)&coefficients128_8)[1] = coefficients[10]; ((drflac_int32*)&samples128_8)[1] = pDecodedSamples[-11]; + case 10: ((drflac_int32*)&coefficients128_8)[2] = coefficients[ 9]; ((drflac_int32*)&samples128_8)[2] = pDecodedSamples[-10]; + case 9: ((drflac_int32*)&coefficients128_8)[3] = coefficients[ 8]; ((drflac_int32*)&samples128_8)[3] = pDecodedSamples[- 9]; + case 8: ((drflac_int32*)&coefficients128_4)[0] = coefficients[ 7]; ((drflac_int32*)&samples128_4)[0] = pDecodedSamples[- 8]; + case 7: ((drflac_int32*)&coefficients128_4)[1] = coefficients[ 6]; ((drflac_int32*)&samples128_4)[1] = pDecodedSamples[- 7]; + case 6: ((drflac_int32*)&coefficients128_4)[2] = coefficients[ 5]; ((drflac_int32*)&samples128_4)[2] = pDecodedSamples[- 6]; + case 5: ((drflac_int32*)&coefficients128_4)[3] = coefficients[ 4]; ((drflac_int32*)&samples128_4)[3] = pDecodedSamples[- 5]; + case 4: ((drflac_int32*)&coefficients128_0)[0] = coefficients[ 3]; ((drflac_int32*)&samples128_0)[0] = pDecodedSamples[- 4]; + case 3: ((drflac_int32*)&coefficients128_0)[1] = coefficients[ 2]; ((drflac_int32*)&samples128_0)[1] = pDecodedSamples[- 3]; + case 2: ((drflac_int32*)&coefficients128_0)[2] = coefficients[ 1]; ((drflac_int32*)&samples128_0)[2] = pDecodedSamples[- 2]; + case 1: ((drflac_int32*)&coefficients128_0)[3] = coefficients[ 0]; ((drflac_int32*)&samples128_0)[3] = pDecodedSamples[- 1]; + } +#endif + + /* For this version we are doing one sample at a time. */ + while (pDecodedSamples < pDecodedSamplesEnd) { + __m128i zeroCountPart128; + __m128i riceParamPart128; + + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts0, &riceParamParts0) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts1, &riceParamParts1) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts2, &riceParamParts2) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts3, &riceParamParts3)) { + return DRFLAC_FALSE; + } + + zeroCountPart128 = _mm_set_epi32(zeroCountParts3, zeroCountParts2, zeroCountParts1, zeroCountParts0); + riceParamPart128 = _mm_set_epi32(riceParamParts3, riceParamParts2, riceParamParts1, riceParamParts0); + + riceParamPart128 = _mm_and_si128(riceParamPart128, riceParamMask128); + riceParamPart128 = _mm_or_si128(riceParamPart128, _mm_slli_epi32(zeroCountPart128, riceParam)); + riceParamPart128 = _mm_xor_si128(_mm_srli_epi32(riceParamPart128, 1), _mm_add_epi32(drflac__mm_not_si128(_mm_and_si128(riceParamPart128, _mm_set1_epi32(1))), _mm_set1_epi32(1))); + + for (i = 0; i < 4; i += 1) { + prediction128 = _mm_xor_si128(prediction128, prediction128); /* Reset to 0. */ + + switch (order) + { + case 12: + case 11: prediction128 = _mm_add_epi64(prediction128, _mm_mul_epi32(_mm_shuffle_epi32(coefficients128_8, _MM_SHUFFLE(1, 1, 0, 0)), _mm_shuffle_epi32(samples128_8, _MM_SHUFFLE(1, 1, 0, 0)))); + case 10: + case 9: prediction128 = _mm_add_epi64(prediction128, _mm_mul_epi32(_mm_shuffle_epi32(coefficients128_8, _MM_SHUFFLE(3, 3, 2, 2)), _mm_shuffle_epi32(samples128_8, _MM_SHUFFLE(3, 3, 2, 2)))); + case 8: + case 7: prediction128 = _mm_add_epi64(prediction128, _mm_mul_epi32(_mm_shuffle_epi32(coefficients128_4, _MM_SHUFFLE(1, 1, 0, 0)), _mm_shuffle_epi32(samples128_4, _MM_SHUFFLE(1, 1, 0, 0)))); + case 6: + case 5: prediction128 = _mm_add_epi64(prediction128, _mm_mul_epi32(_mm_shuffle_epi32(coefficients128_4, _MM_SHUFFLE(3, 3, 2, 2)), _mm_shuffle_epi32(samples128_4, _MM_SHUFFLE(3, 3, 2, 2)))); + case 4: + case 3: prediction128 = _mm_add_epi64(prediction128, _mm_mul_epi32(_mm_shuffle_epi32(coefficients128_0, _MM_SHUFFLE(1, 1, 0, 0)), _mm_shuffle_epi32(samples128_0, _MM_SHUFFLE(1, 1, 0, 0)))); + case 2: + case 1: prediction128 = _mm_add_epi64(prediction128, _mm_mul_epi32(_mm_shuffle_epi32(coefficients128_0, _MM_SHUFFLE(3, 3, 2, 2)), _mm_shuffle_epi32(samples128_0, _MM_SHUFFLE(3, 3, 2, 2)))); + } + + /* Horizontal add and shift. */ + prediction128 = drflac__mm_hadd_epi64(prediction128); + prediction128 = drflac__mm_srai_epi64(prediction128, shift); + prediction128 = _mm_add_epi32(riceParamPart128, prediction128); + + /* Our value should be sitting in prediction128[0]. We need to combine this with our SSE samples. */ + samples128_8 = _mm_alignr_epi8(samples128_4, samples128_8, 4); + samples128_4 = _mm_alignr_epi8(samples128_0, samples128_4, 4); + samples128_0 = _mm_alignr_epi8(prediction128, samples128_0, 4); + + /* Slide our rice parameter down so that the value in position 0 contains the next one to process. */ + riceParamPart128 = _mm_alignr_epi8(_mm_setzero_si128(), riceParamPart128, 4); + } + + /* We store samples in groups of 4. */ + _mm_storeu_si128((__m128i*)pDecodedSamples, samples128_0); + pDecodedSamples += 4; + } + + /* Make sure we process the last few samples. */ + i = (count & ~3); + while (i < (int)count) { + /* Rice extraction. */ + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts0, &riceParamParts0)) { + return DRFLAC_FALSE; + } + + /* Rice reconstruction. */ + riceParamParts0 &= riceParamMask; + riceParamParts0 |= (zeroCountParts0 << riceParam); + riceParamParts0 = (riceParamParts0 >> 1) ^ t[riceParamParts0 & 0x01]; + + /* Sample reconstruction. */ + pDecodedSamples[0] = riceParamParts0 + drflac__calculate_prediction_64(order, shift, coefficients, pDecodedSamples); + + i += 1; + pDecodedSamples += 1; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_samples_with_residual__rice__sse41(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pSamplesOut != NULL); + + /* In my testing the order is rarely > 12, so in this case I'm going to simplify the SSE implementation by only handling order <= 12. */ + if (lpcOrder > 0 && lpcOrder <= 12) { + if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) { + return drflac__decode_samples_with_residual__rice__sse41_64(bs, count, riceParam, lpcOrder, lpcShift, coefficients, pSamplesOut); + } else { + return drflac__decode_samples_with_residual__rice__sse41_32(bs, count, riceParam, lpcOrder, lpcShift, coefficients, pSamplesOut); + } + } else { + return drflac__decode_samples_with_residual__rice__scalar(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut); + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac__vst2q_s32(drflac_int32* p, int32x4x2_t x) +{ + vst1q_s32(p+0, x.val[0]); + vst1q_s32(p+4, x.val[1]); +} + +static DRFLAC_INLINE void drflac__vst2q_u32(drflac_uint32* p, uint32x4x2_t x) +{ + vst1q_u32(p+0, x.val[0]); + vst1q_u32(p+4, x.val[1]); +} + +static DRFLAC_INLINE void drflac__vst2q_f32(float* p, float32x4x2_t x) +{ + vst1q_f32(p+0, x.val[0]); + vst1q_f32(p+4, x.val[1]); +} + +static DRFLAC_INLINE void drflac__vst2q_s16(drflac_int16* p, int16x4x2_t x) +{ + vst1q_s16(p, vcombine_s16(x.val[0], x.val[1])); +} + +static DRFLAC_INLINE void drflac__vst2q_u16(drflac_uint16* p, uint16x4x2_t x) +{ + vst1q_u16(p, vcombine_u16(x.val[0], x.val[1])); +} + +static DRFLAC_INLINE int32x4_t drflac__vdupq_n_s32x4(drflac_int32 x3, drflac_int32 x2, drflac_int32 x1, drflac_int32 x0) +{ + drflac_int32 x[4]; + x[3] = x3; + x[2] = x2; + x[1] = x1; + x[0] = x0; + return vld1q_s32(x); +} + +static DRFLAC_INLINE int32x4_t drflac__valignrq_s32_1(int32x4_t a, int32x4_t b) +{ + /* Equivalent to SSE's _mm_alignr_epi8(a, b, 4) */ + + /* Reference */ + /*return drflac__vdupq_n_s32x4( + vgetq_lane_s32(a, 0), + vgetq_lane_s32(b, 3), + vgetq_lane_s32(b, 2), + vgetq_lane_s32(b, 1) + );*/ + + return vextq_s32(b, a, 1); +} + +static DRFLAC_INLINE uint32x4_t drflac__valignrq_u32_1(uint32x4_t a, uint32x4_t b) +{ + /* Equivalent to SSE's _mm_alignr_epi8(a, b, 4) */ + + /* Reference */ + /*return drflac__vdupq_n_s32x4( + vgetq_lane_s32(a, 0), + vgetq_lane_s32(b, 3), + vgetq_lane_s32(b, 2), + vgetq_lane_s32(b, 1) + );*/ + + return vextq_u32(b, a, 1); +} + +static DRFLAC_INLINE int32x2_t drflac__vhaddq_s32(int32x4_t x) +{ + /* The sum must end up in position 0. */ + + /* Reference */ + /*return vdupq_n_s32( + vgetq_lane_s32(x, 3) + + vgetq_lane_s32(x, 2) + + vgetq_lane_s32(x, 1) + + vgetq_lane_s32(x, 0) + );*/ + + int32x2_t r = vadd_s32(vget_high_s32(x), vget_low_s32(x)); + return vpadd_s32(r, r); +} + +static DRFLAC_INLINE int64x1_t drflac__vhaddq_s64(int64x2_t x) +{ + return vadd_s64(vget_high_s64(x), vget_low_s64(x)); +} + +static DRFLAC_INLINE int32x4_t drflac__vrevq_s32(int32x4_t x) +{ + /* Reference */ + /*return drflac__vdupq_n_s32x4( + vgetq_lane_s32(x, 0), + vgetq_lane_s32(x, 1), + vgetq_lane_s32(x, 2), + vgetq_lane_s32(x, 3) + );*/ + + return vrev64q_s32(vcombine_s32(vget_high_s32(x), vget_low_s32(x))); +} + +static DRFLAC_INLINE int32x4_t drflac__vnotq_s32(int32x4_t x) +{ + return veorq_s32(x, vdupq_n_s32(0xFFFFFFFF)); +} + +static DRFLAC_INLINE uint32x4_t drflac__vnotq_u32(uint32x4_t x) +{ + return veorq_u32(x, vdupq_n_u32(0xFFFFFFFF)); +} + +static drflac_bool32 drflac__decode_samples_with_residual__rice__neon_32(drflac_bs* bs, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + int i; + drflac_uint32 riceParamMask; + drflac_int32* pDecodedSamples = pSamplesOut; + drflac_int32* pDecodedSamplesEnd = pSamplesOut + (count & ~3); + drflac_uint32 zeroCountParts[4]; + drflac_uint32 riceParamParts[4]; + int32x4_t coefficients128_0; + int32x4_t coefficients128_4; + int32x4_t coefficients128_8; + int32x4_t samples128_0; + int32x4_t samples128_4; + int32x4_t samples128_8; + uint32x4_t riceParamMask128; + int32x4_t riceParam128; + int32x2_t shift64; + uint32x4_t one128; + + const drflac_uint32 t[2] = {0x00000000, 0xFFFFFFFF}; + + riceParamMask = (drflac_uint32)~((~0UL) << riceParam); + riceParamMask128 = vdupq_n_u32(riceParamMask); + + riceParam128 = vdupq_n_s32(riceParam); + shift64 = vdup_n_s32(-shift); /* Negate the shift because we'll be doing a variable shift using vshlq_s32(). */ + one128 = vdupq_n_u32(1); + + /* + Pre-loading the coefficients and prior samples is annoying because we need to ensure we don't try reading more than + what's available in the input buffers. It would be conenient to use a fall-through switch to do this, but this results + in strict aliasing warnings with GCC. To work around this I'm just doing something hacky. This feels a bit convoluted + so I think there's opportunity for this to be simplified. + */ + { + int runningOrder = order; + drflac_int32 tempC[4] = {0, 0, 0, 0}; + drflac_int32 tempS[4] = {0, 0, 0, 0}; + + /* 0 - 3. */ + if (runningOrder >= 4) { + coefficients128_0 = vld1q_s32(coefficients + 0); + samples128_0 = vld1q_s32(pSamplesOut - 4); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: tempC[2] = coefficients[2]; tempS[1] = pSamplesOut[-3]; /* fallthrough */ + case 2: tempC[1] = coefficients[1]; tempS[2] = pSamplesOut[-2]; /* fallthrough */ + case 1: tempC[0] = coefficients[0]; tempS[3] = pSamplesOut[-1]; /* fallthrough */ + } + + coefficients128_0 = vld1q_s32(tempC); + samples128_0 = vld1q_s32(tempS); + runningOrder = 0; + } + + /* 4 - 7 */ + if (runningOrder >= 4) { + coefficients128_4 = vld1q_s32(coefficients + 4); + samples128_4 = vld1q_s32(pSamplesOut - 8); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: tempC[2] = coefficients[6]; tempS[1] = pSamplesOut[-7]; /* fallthrough */ + case 2: tempC[1] = coefficients[5]; tempS[2] = pSamplesOut[-6]; /* fallthrough */ + case 1: tempC[0] = coefficients[4]; tempS[3] = pSamplesOut[-5]; /* fallthrough */ + } + + coefficients128_4 = vld1q_s32(tempC); + samples128_4 = vld1q_s32(tempS); + runningOrder = 0; + } + + /* 8 - 11 */ + if (runningOrder == 4) { + coefficients128_8 = vld1q_s32(coefficients + 8); + samples128_8 = vld1q_s32(pSamplesOut - 12); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: tempC[2] = coefficients[10]; tempS[1] = pSamplesOut[-11]; /* fallthrough */ + case 2: tempC[1] = coefficients[ 9]; tempS[2] = pSamplesOut[-10]; /* fallthrough */ + case 1: tempC[0] = coefficients[ 8]; tempS[3] = pSamplesOut[- 9]; /* fallthrough */ + } + + coefficients128_8 = vld1q_s32(tempC); + samples128_8 = vld1q_s32(tempS); + runningOrder = 0; + } + + /* Coefficients need to be shuffled for our streaming algorithm below to work. Samples are already in the correct order from the loading routine above. */ + coefficients128_0 = drflac__vrevq_s32(coefficients128_0); + coefficients128_4 = drflac__vrevq_s32(coefficients128_4); + coefficients128_8 = drflac__vrevq_s32(coefficients128_8); + } + + /* For this version we are doing one sample at a time. */ + while (pDecodedSamples < pDecodedSamplesEnd) { + int32x4_t prediction128; + int32x2_t prediction64; + uint32x4_t zeroCountPart128; + uint32x4_t riceParamPart128; + + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[0], &riceParamParts[0]) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[1], &riceParamParts[1]) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[2], &riceParamParts[2]) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[3], &riceParamParts[3])) { + return DRFLAC_FALSE; + } + + zeroCountPart128 = vld1q_u32(zeroCountParts); + riceParamPart128 = vld1q_u32(riceParamParts); + + riceParamPart128 = vandq_u32(riceParamPart128, riceParamMask128); + riceParamPart128 = vorrq_u32(riceParamPart128, vshlq_u32(zeroCountPart128, riceParam128)); + riceParamPart128 = veorq_u32(vshrq_n_u32(riceParamPart128, 1), vaddq_u32(drflac__vnotq_u32(vandq_u32(riceParamPart128, one128)), one128)); + + if (order <= 4) { + for (i = 0; i < 4; i += 1) { + prediction128 = vmulq_s32(coefficients128_0, samples128_0); + + /* Horizontal add and shift. */ + prediction64 = drflac__vhaddq_s32(prediction128); + prediction64 = vshl_s32(prediction64, shift64); + prediction64 = vadd_s32(prediction64, vget_low_s32(vreinterpretq_s32_u32(riceParamPart128))); + + samples128_0 = drflac__valignrq_s32_1(vcombine_s32(prediction64, vdup_n_s32(0)), samples128_0); + riceParamPart128 = drflac__valignrq_u32_1(vdupq_n_u32(0), riceParamPart128); + } + } else if (order <= 8) { + for (i = 0; i < 4; i += 1) { + prediction128 = vmulq_s32(coefficients128_4, samples128_4); + prediction128 = vmlaq_s32(prediction128, coefficients128_0, samples128_0); + + /* Horizontal add and shift. */ + prediction64 = drflac__vhaddq_s32(prediction128); + prediction64 = vshl_s32(prediction64, shift64); + prediction64 = vadd_s32(prediction64, vget_low_s32(vreinterpretq_s32_u32(riceParamPart128))); + + samples128_4 = drflac__valignrq_s32_1(samples128_0, samples128_4); + samples128_0 = drflac__valignrq_s32_1(vcombine_s32(prediction64, vdup_n_s32(0)), samples128_0); + riceParamPart128 = drflac__valignrq_u32_1(vdupq_n_u32(0), riceParamPart128); + } + } else { + for (i = 0; i < 4; i += 1) { + prediction128 = vmulq_s32(coefficients128_8, samples128_8); + prediction128 = vmlaq_s32(prediction128, coefficients128_4, samples128_4); + prediction128 = vmlaq_s32(prediction128, coefficients128_0, samples128_0); + + /* Horizontal add and shift. */ + prediction64 = drflac__vhaddq_s32(prediction128); + prediction64 = vshl_s32(prediction64, shift64); + prediction64 = vadd_s32(prediction64, vget_low_s32(vreinterpretq_s32_u32(riceParamPart128))); + + samples128_8 = drflac__valignrq_s32_1(samples128_4, samples128_8); + samples128_4 = drflac__valignrq_s32_1(samples128_0, samples128_4); + samples128_0 = drflac__valignrq_s32_1(vcombine_s32(prediction64, vdup_n_s32(0)), samples128_0); + riceParamPart128 = drflac__valignrq_u32_1(vdupq_n_u32(0), riceParamPart128); + } + } + + /* We store samples in groups of 4. */ + vst1q_s32(pDecodedSamples, samples128_0); + pDecodedSamples += 4; + } + + /* Make sure we process the last few samples. */ + i = (count & ~3); + while (i < (int)count) { + /* Rice extraction. */ + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[0], &riceParamParts[0])) { + return DRFLAC_FALSE; + } + + /* Rice reconstruction. */ + riceParamParts[0] &= riceParamMask; + riceParamParts[0] |= (zeroCountParts[0] << riceParam); + riceParamParts[0] = (riceParamParts[0] >> 1) ^ t[riceParamParts[0] & 0x01]; + + /* Sample reconstruction. */ + pDecodedSamples[0] = riceParamParts[0] + drflac__calculate_prediction_32(order, shift, coefficients, pDecodedSamples); + + i += 1; + pDecodedSamples += 1; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_samples_with_residual__rice__neon_64(drflac_bs* bs, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 order, drflac_int32 shift, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + int i; + drflac_uint32 riceParamMask; + drflac_int32* pDecodedSamples = pSamplesOut; + drflac_int32* pDecodedSamplesEnd = pSamplesOut + (count & ~3); + drflac_uint32 zeroCountParts[4]; + drflac_uint32 riceParamParts[4]; + int32x4_t coefficients128_0; + int32x4_t coefficients128_4; + int32x4_t coefficients128_8; + int32x4_t samples128_0; + int32x4_t samples128_4; + int32x4_t samples128_8; + uint32x4_t riceParamMask128; + int32x4_t riceParam128; + int64x1_t shift64; + uint32x4_t one128; + int64x2_t prediction128 = { 0 }; + uint32x4_t zeroCountPart128; + uint32x4_t riceParamPart128; + + const drflac_uint32 t[2] = {0x00000000, 0xFFFFFFFF}; + + riceParamMask = (drflac_uint32)~((~0UL) << riceParam); + riceParamMask128 = vdupq_n_u32(riceParamMask); + + riceParam128 = vdupq_n_s32(riceParam); + shift64 = vdup_n_s64(-shift); /* Negate the shift because we'll be doing a variable shift using vshlq_s32(). */ + one128 = vdupq_n_u32(1); + + /* + Pre-loading the coefficients and prior samples is annoying because we need to ensure we don't try reading more than + what's available in the input buffers. It would be convenient to use a fall-through switch to do this, but this results + in strict aliasing warnings with GCC. To work around this I'm just doing something hacky. This feels a bit convoluted + so I think there's opportunity for this to be simplified. + */ + { + int runningOrder = order; + drflac_int32 tempC[4] = {0, 0, 0, 0}; + drflac_int32 tempS[4] = {0, 0, 0, 0}; + + /* 0 - 3. */ + if (runningOrder >= 4) { + coefficients128_0 = vld1q_s32(coefficients + 0); + samples128_0 = vld1q_s32(pSamplesOut - 4); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: tempC[2] = coefficients[2]; tempS[1] = pSamplesOut[-3]; /* fallthrough */ + case 2: tempC[1] = coefficients[1]; tempS[2] = pSamplesOut[-2]; /* fallthrough */ + case 1: tempC[0] = coefficients[0]; tempS[3] = pSamplesOut[-1]; /* fallthrough */ + } + + coefficients128_0 = vld1q_s32(tempC); + samples128_0 = vld1q_s32(tempS); + runningOrder = 0; + } + + /* 4 - 7 */ + if (runningOrder >= 4) { + coefficients128_4 = vld1q_s32(coefficients + 4); + samples128_4 = vld1q_s32(pSamplesOut - 8); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: tempC[2] = coefficients[6]; tempS[1] = pSamplesOut[-7]; /* fallthrough */ + case 2: tempC[1] = coefficients[5]; tempS[2] = pSamplesOut[-6]; /* fallthrough */ + case 1: tempC[0] = coefficients[4]; tempS[3] = pSamplesOut[-5]; /* fallthrough */ + } + + coefficients128_4 = vld1q_s32(tempC); + samples128_4 = vld1q_s32(tempS); + runningOrder = 0; + } + + /* 8 - 11 */ + if (runningOrder == 4) { + coefficients128_8 = vld1q_s32(coefficients + 8); + samples128_8 = vld1q_s32(pSamplesOut - 12); + runningOrder -= 4; + } else { + switch (runningOrder) { + case 3: tempC[2] = coefficients[10]; tempS[1] = pSamplesOut[-11]; /* fallthrough */ + case 2: tempC[1] = coefficients[ 9]; tempS[2] = pSamplesOut[-10]; /* fallthrough */ + case 1: tempC[0] = coefficients[ 8]; tempS[3] = pSamplesOut[- 9]; /* fallthrough */ + } + + coefficients128_8 = vld1q_s32(tempC); + samples128_8 = vld1q_s32(tempS); + runningOrder = 0; + } + + /* Coefficients need to be shuffled for our streaming algorithm below to work. Samples are already in the correct order from the loading routine above. */ + coefficients128_0 = drflac__vrevq_s32(coefficients128_0); + coefficients128_4 = drflac__vrevq_s32(coefficients128_4); + coefficients128_8 = drflac__vrevq_s32(coefficients128_8); + } + + /* For this version we are doing one sample at a time. */ + while (pDecodedSamples < pDecodedSamplesEnd) { + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[0], &riceParamParts[0]) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[1], &riceParamParts[1]) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[2], &riceParamParts[2]) || + !drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[3], &riceParamParts[3])) { + return DRFLAC_FALSE; + } + + zeroCountPart128 = vld1q_u32(zeroCountParts); + riceParamPart128 = vld1q_u32(riceParamParts); + + riceParamPart128 = vandq_u32(riceParamPart128, riceParamMask128); + riceParamPart128 = vorrq_u32(riceParamPart128, vshlq_u32(zeroCountPart128, riceParam128)); + riceParamPart128 = veorq_u32(vshrq_n_u32(riceParamPart128, 1), vaddq_u32(drflac__vnotq_u32(vandq_u32(riceParamPart128, one128)), one128)); + + for (i = 0; i < 4; i += 1) { + int64x1_t prediction64; + + prediction128 = veorq_s64(prediction128, prediction128); /* Reset to 0. */ + switch (order) + { + case 12: + case 11: prediction128 = vaddq_s64(prediction128, vmull_s32(vget_low_s32(coefficients128_8), vget_low_s32(samples128_8))); + case 10: + case 9: prediction128 = vaddq_s64(prediction128, vmull_s32(vget_high_s32(coefficients128_8), vget_high_s32(samples128_8))); + case 8: + case 7: prediction128 = vaddq_s64(prediction128, vmull_s32(vget_low_s32(coefficients128_4), vget_low_s32(samples128_4))); + case 6: + case 5: prediction128 = vaddq_s64(prediction128, vmull_s32(vget_high_s32(coefficients128_4), vget_high_s32(samples128_4))); + case 4: + case 3: prediction128 = vaddq_s64(prediction128, vmull_s32(vget_low_s32(coefficients128_0), vget_low_s32(samples128_0))); + case 2: + case 1: prediction128 = vaddq_s64(prediction128, vmull_s32(vget_high_s32(coefficients128_0), vget_high_s32(samples128_0))); + } + + /* Horizontal add and shift. */ + prediction64 = drflac__vhaddq_s64(prediction128); + prediction64 = vshl_s64(prediction64, shift64); + prediction64 = vadd_s64(prediction64, vdup_n_s64(vgetq_lane_u32(riceParamPart128, 0))); + + /* Our value should be sitting in prediction64[0]. We need to combine this with our SSE samples. */ + samples128_8 = drflac__valignrq_s32_1(samples128_4, samples128_8); + samples128_4 = drflac__valignrq_s32_1(samples128_0, samples128_4); + samples128_0 = drflac__valignrq_s32_1(vcombine_s32(vreinterpret_s32_s64(prediction64), vdup_n_s32(0)), samples128_0); + + /* Slide our rice parameter down so that the value in position 0 contains the next one to process. */ + riceParamPart128 = drflac__valignrq_u32_1(vdupq_n_u32(0), riceParamPart128); + } + + /* We store samples in groups of 4. */ + vst1q_s32(pDecodedSamples, samples128_0); + pDecodedSamples += 4; + } + + /* Make sure we process the last few samples. */ + i = (count & ~3); + while (i < (int)count) { + /* Rice extraction. */ + if (!drflac__read_rice_parts_x1(bs, riceParam, &zeroCountParts[0], &riceParamParts[0])) { + return DRFLAC_FALSE; + } + + /* Rice reconstruction. */ + riceParamParts[0] &= riceParamMask; + riceParamParts[0] |= (zeroCountParts[0] << riceParam); + riceParamParts[0] = (riceParamParts[0] >> 1) ^ t[riceParamParts[0] & 0x01]; + + /* Sample reconstruction. */ + pDecodedSamples[0] = riceParamParts[0] + drflac__calculate_prediction_64(order, shift, coefficients, pDecodedSamples); + + i += 1; + pDecodedSamples += 1; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_samples_with_residual__rice__neon(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(pSamplesOut != NULL); + + /* In my testing the order is rarely > 12, so in this case I'm going to simplify the NEON implementation by only handling order <= 12. */ + if (lpcOrder > 0 && lpcOrder <= 12) { + if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) { + return drflac__decode_samples_with_residual__rice__neon_64(bs, count, riceParam, lpcOrder, lpcShift, coefficients, pSamplesOut); + } else { + return drflac__decode_samples_with_residual__rice__neon_32(bs, count, riceParam, lpcOrder, lpcShift, coefficients, pSamplesOut); + } + } else { + return drflac__decode_samples_with_residual__rice__scalar(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut); + } +} +#endif + +static drflac_bool32 drflac__decode_samples_with_residual__rice(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 riceParam, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ +#if defined(DRFLAC_SUPPORT_SSE41) + if (drflac__gIsSSE41Supported) { + return drflac__decode_samples_with_residual__rice__sse41(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported) { + return drflac__decode_samples_with_residual__rice__neon(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut); + } else +#endif + { + /* Scalar fallback. */ + #if 0 + return drflac__decode_samples_with_residual__rice__reference(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut); + #else + return drflac__decode_samples_with_residual__rice__scalar(bs, bitsPerSample, count, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pSamplesOut); + #endif + } +} + +/* Reads and seeks past a string of residual values as Rice codes. The decoder should be sitting on the first bit of the Rice codes. */ +static drflac_bool32 drflac__read_and_seek_residual__rice(drflac_bs* bs, drflac_uint32 count, drflac_uint8 riceParam) +{ + drflac_uint32 i; + + DRFLAC_ASSERT(bs != NULL); + + for (i = 0; i < count; ++i) { + if (!drflac__seek_rice_parts(bs, riceParam)) { + return DRFLAC_FALSE; + } + } + + return DRFLAC_TRUE; +} + +#if defined(__clang__) +__attribute__((no_sanitize("signed-integer-overflow"))) +#endif +static drflac_bool32 drflac__decode_samples_with_residual__unencoded(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 count, drflac_uint8 unencodedBitsPerSample, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pSamplesOut) +{ + drflac_uint32 i; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(unencodedBitsPerSample <= 31); /* <-- unencodedBitsPerSample is a 5 bit number, so cannot exceed 31. */ + DRFLAC_ASSERT(pSamplesOut != NULL); + + for (i = 0; i < count; ++i) { + if (unencodedBitsPerSample > 0) { + if (!drflac__read_int32(bs, unencodedBitsPerSample, pSamplesOut + i)) { + return DRFLAC_FALSE; + } + } else { + pSamplesOut[i] = 0; + } + + if (drflac__use_64_bit_prediction(bitsPerSample, lpcOrder, lpcPrecision)) { + pSamplesOut[i] += drflac__calculate_prediction_64(lpcOrder, lpcShift, coefficients, pSamplesOut + i); + } else { + pSamplesOut[i] += drflac__calculate_prediction_32(lpcOrder, lpcShift, coefficients, pSamplesOut + i); + } + } + + return DRFLAC_TRUE; +} + + +/* +Reads and decodes the residual for the sub-frame the decoder is currently sitting on. This function should be called +when the decoder is sitting at the very start of the RESIDUAL block. The first residuals will be ignored. The + and parameters are used to determine how many residual values need to be decoded. +*/ +static drflac_bool32 drflac__decode_samples_with_residual(drflac_bs* bs, drflac_uint32 bitsPerSample, drflac_uint32 blockSize, drflac_uint32 lpcOrder, drflac_int32 lpcShift, drflac_uint32 lpcPrecision, const drflac_int32* coefficients, drflac_int32* pDecodedSamples) +{ + drflac_uint8 residualMethod; + drflac_uint8 partitionOrder; + drflac_uint32 samplesInPartition; + drflac_uint32 partitionsRemaining; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(blockSize != 0); + DRFLAC_ASSERT(pDecodedSamples != NULL); /* <-- Should we allow NULL, in which case we just seek past the residual rather than do a full decode? */ + + if (!drflac__read_uint8(bs, 2, &residualMethod)) { + return DRFLAC_FALSE; + } + + if (residualMethod != DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE && residualMethod != DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE2) { + return DRFLAC_FALSE; /* Unknown or unsupported residual coding method. */ + } + + /* Ignore the first values. */ + pDecodedSamples += lpcOrder; + + if (!drflac__read_uint8(bs, 4, &partitionOrder)) { + return DRFLAC_FALSE; + } + + /* + From the FLAC spec: + The Rice partition order in a Rice-coded residual section must be less than or equal to 8. + */ + if (partitionOrder > 8) { + return DRFLAC_FALSE; + } + + /* Validation check. */ + if ((blockSize / (1 << partitionOrder)) < lpcOrder) { + return DRFLAC_FALSE; + } + + samplesInPartition = (blockSize / (1 << partitionOrder)) - lpcOrder; + partitionsRemaining = (1 << partitionOrder); + for (;;) { + drflac_uint8 riceParam = 0; + if (residualMethod == DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE) { + if (!drflac__read_uint8(bs, 4, &riceParam)) { + return DRFLAC_FALSE; + } + if (riceParam == 15) { + riceParam = 0xFF; + } + } else if (residualMethod == DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE2) { + if (!drflac__read_uint8(bs, 5, &riceParam)) { + return DRFLAC_FALSE; + } + if (riceParam == 31) { + riceParam = 0xFF; + } + } + + if (riceParam != 0xFF) { + if (!drflac__decode_samples_with_residual__rice(bs, bitsPerSample, samplesInPartition, riceParam, lpcOrder, lpcShift, lpcPrecision, coefficients, pDecodedSamples)) { + return DRFLAC_FALSE; + } + } else { + drflac_uint8 unencodedBitsPerSample = 0; + if (!drflac__read_uint8(bs, 5, &unencodedBitsPerSample)) { + return DRFLAC_FALSE; + } + + if (!drflac__decode_samples_with_residual__unencoded(bs, bitsPerSample, samplesInPartition, unencodedBitsPerSample, lpcOrder, lpcShift, lpcPrecision, coefficients, pDecodedSamples)) { + return DRFLAC_FALSE; + } + } + + pDecodedSamples += samplesInPartition; + + if (partitionsRemaining == 1) { + break; + } + + partitionsRemaining -= 1; + + if (partitionOrder != 0) { + samplesInPartition = blockSize / (1 << partitionOrder); + } + } + + return DRFLAC_TRUE; +} + +/* +Reads and seeks past the residual for the sub-frame the decoder is currently sitting on. This function should be called +when the decoder is sitting at the very start of the RESIDUAL block. The first residuals will be set to 0. The + and parameters are used to determine how many residual values need to be decoded. +*/ +static drflac_bool32 drflac__read_and_seek_residual(drflac_bs* bs, drflac_uint32 blockSize, drflac_uint32 order) +{ + drflac_uint8 residualMethod; + drflac_uint8 partitionOrder; + drflac_uint32 samplesInPartition; + drflac_uint32 partitionsRemaining; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(blockSize != 0); + + if (!drflac__read_uint8(bs, 2, &residualMethod)) { + return DRFLAC_FALSE; + } + + if (residualMethod != DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE && residualMethod != DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE2) { + return DRFLAC_FALSE; /* Unknown or unsupported residual coding method. */ + } + + if (!drflac__read_uint8(bs, 4, &partitionOrder)) { + return DRFLAC_FALSE; + } + + /* + From the FLAC spec: + The Rice partition order in a Rice-coded residual section must be less than or equal to 8. + */ + if (partitionOrder > 8) { + return DRFLAC_FALSE; + } + + /* Validation check. */ + if ((blockSize / (1 << partitionOrder)) <= order) { + return DRFLAC_FALSE; + } + + samplesInPartition = (blockSize / (1 << partitionOrder)) - order; + partitionsRemaining = (1 << partitionOrder); + for (;;) + { + drflac_uint8 riceParam = 0; + if (residualMethod == DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE) { + if (!drflac__read_uint8(bs, 4, &riceParam)) { + return DRFLAC_FALSE; + } + if (riceParam == 15) { + riceParam = 0xFF; + } + } else if (residualMethod == DRFLAC_RESIDUAL_CODING_METHOD_PARTITIONED_RICE2) { + if (!drflac__read_uint8(bs, 5, &riceParam)) { + return DRFLAC_FALSE; + } + if (riceParam == 31) { + riceParam = 0xFF; + } + } + + if (riceParam != 0xFF) { + if (!drflac__read_and_seek_residual__rice(bs, samplesInPartition, riceParam)) { + return DRFLAC_FALSE; + } + } else { + drflac_uint8 unencodedBitsPerSample = 0; + if (!drflac__read_uint8(bs, 5, &unencodedBitsPerSample)) { + return DRFLAC_FALSE; + } + + if (!drflac__seek_bits(bs, unencodedBitsPerSample * samplesInPartition)) { + return DRFLAC_FALSE; + } + } + + + if (partitionsRemaining == 1) { + break; + } + + partitionsRemaining -= 1; + samplesInPartition = blockSize / (1 << partitionOrder); + } + + return DRFLAC_TRUE; +} + + +static drflac_bool32 drflac__decode_samples__constant(drflac_bs* bs, drflac_uint32 blockSize, drflac_uint32 subframeBitsPerSample, drflac_int32* pDecodedSamples) +{ + drflac_uint32 i; + + /* Only a single sample needs to be decoded here. */ + drflac_int32 sample; + if (!drflac__read_int32(bs, subframeBitsPerSample, &sample)) { + return DRFLAC_FALSE; + } + + /* + We don't really need to expand this, but it does simplify the process of reading samples. If this becomes a performance issue (unlikely) + we'll want to look at a more efficient way. + */ + for (i = 0; i < blockSize; ++i) { + pDecodedSamples[i] = sample; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_samples__verbatim(drflac_bs* bs, drflac_uint32 blockSize, drflac_uint32 subframeBitsPerSample, drflac_int32* pDecodedSamples) +{ + drflac_uint32 i; + + for (i = 0; i < blockSize; ++i) { + drflac_int32 sample; + if (!drflac__read_int32(bs, subframeBitsPerSample, &sample)) { + return DRFLAC_FALSE; + } + + pDecodedSamples[i] = sample; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_samples__fixed(drflac_bs* bs, drflac_uint32 blockSize, drflac_uint32 subframeBitsPerSample, drflac_uint8 lpcOrder, drflac_int32* pDecodedSamples) +{ + drflac_uint32 i; + + static drflac_int32 lpcCoefficientsTable[5][4] = { + {0, 0, 0, 0}, + {1, 0, 0, 0}, + {2, -1, 0, 0}, + {3, -3, 1, 0}, + {4, -6, 4, -1} + }; + + /* Warm up samples and coefficients. */ + for (i = 0; i < lpcOrder; ++i) { + drflac_int32 sample; + if (!drflac__read_int32(bs, subframeBitsPerSample, &sample)) { + return DRFLAC_FALSE; + } + + pDecodedSamples[i] = sample; + } + + if (!drflac__decode_samples_with_residual(bs, subframeBitsPerSample, blockSize, lpcOrder, 0, 4, lpcCoefficientsTable[lpcOrder], pDecodedSamples)) { + return DRFLAC_FALSE; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_samples__lpc(drflac_bs* bs, drflac_uint32 blockSize, drflac_uint32 bitsPerSample, drflac_uint8 lpcOrder, drflac_int32* pDecodedSamples) +{ + drflac_uint8 i; + drflac_uint8 lpcPrecision; + drflac_int8 lpcShift; + drflac_int32 coefficients[32]; + + /* Warm up samples. */ + for (i = 0; i < lpcOrder; ++i) { + drflac_int32 sample; + if (!drflac__read_int32(bs, bitsPerSample, &sample)) { + return DRFLAC_FALSE; + } + + pDecodedSamples[i] = sample; + } + + if (!drflac__read_uint8(bs, 4, &lpcPrecision)) { + return DRFLAC_FALSE; + } + if (lpcPrecision == 15) { + return DRFLAC_FALSE; /* Invalid. */ + } + lpcPrecision += 1; + + if (!drflac__read_int8(bs, 5, &lpcShift)) { + return DRFLAC_FALSE; + } + + /* + From the FLAC specification: + + Quantized linear predictor coefficient shift needed in bits (NOTE: this number is signed two's-complement) + + Emphasis on the "signed two's-complement". In practice there does not seem to be any encoders nor decoders supporting negative shifts. For now dr_flac is + not going to support negative shifts as I don't have any reference files. However, when a reference file comes through I will consider adding support. + */ + if (lpcShift < 0) { + return DRFLAC_FALSE; + } + + DRFLAC_ZERO_MEMORY(coefficients, sizeof(coefficients)); + for (i = 0; i < lpcOrder; ++i) { + if (!drflac__read_int32(bs, lpcPrecision, coefficients + i)) { + return DRFLAC_FALSE; + } + } + + if (!drflac__decode_samples_with_residual(bs, bitsPerSample, blockSize, lpcOrder, lpcShift, lpcPrecision, coefficients, pDecodedSamples)) { + return DRFLAC_FALSE; + } + + return DRFLAC_TRUE; +} + + +static drflac_bool32 drflac__read_next_flac_frame_header(drflac_bs* bs, drflac_uint8 streaminfoBitsPerSample, drflac_frame_header* header) +{ + const drflac_uint32 sampleRateTable[12] = {0, 88200, 176400, 192000, 8000, 16000, 22050, 24000, 32000, 44100, 48000, 96000}; + const drflac_uint8 bitsPerSampleTable[8] = {0, 8, 12, (drflac_uint8)-1, 16, 20, 24, (drflac_uint8)-1}; /* -1 = reserved. */ + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(header != NULL); + + /* Keep looping until we find a valid sync code. */ + for (;;) { + drflac_uint8 crc8 = 0xCE; /* 0xCE = drflac_crc8(0, 0x3FFE, 14); */ + drflac_uint8 reserved = 0; + drflac_uint8 blockingStrategy = 0; + drflac_uint8 blockSize = 0; + drflac_uint8 sampleRate = 0; + drflac_uint8 channelAssignment = 0; + drflac_uint8 bitsPerSample = 0; + drflac_bool32 isVariableBlockSize; + + if (!drflac__find_and_seek_to_next_sync_code(bs)) { + return DRFLAC_FALSE; + } + + if (!drflac__read_uint8(bs, 1, &reserved)) { + return DRFLAC_FALSE; + } + if (reserved == 1) { + continue; + } + crc8 = drflac_crc8(crc8, reserved, 1); + + if (!drflac__read_uint8(bs, 1, &blockingStrategy)) { + return DRFLAC_FALSE; + } + crc8 = drflac_crc8(crc8, blockingStrategy, 1); + + if (!drflac__read_uint8(bs, 4, &blockSize)) { + return DRFLAC_FALSE; + } + if (blockSize == 0) { + continue; + } + crc8 = drflac_crc8(crc8, blockSize, 4); + + if (!drflac__read_uint8(bs, 4, &sampleRate)) { + return DRFLAC_FALSE; + } + crc8 = drflac_crc8(crc8, sampleRate, 4); + + if (!drflac__read_uint8(bs, 4, &channelAssignment)) { + return DRFLAC_FALSE; + } + if (channelAssignment > 10) { + continue; + } + crc8 = drflac_crc8(crc8, channelAssignment, 4); + + if (!drflac__read_uint8(bs, 3, &bitsPerSample)) { + return DRFLAC_FALSE; + } + if (bitsPerSample == 3 || bitsPerSample == 7) { + continue; + } + crc8 = drflac_crc8(crc8, bitsPerSample, 3); + + + if (!drflac__read_uint8(bs, 1, &reserved)) { + return DRFLAC_FALSE; + } + if (reserved == 1) { + continue; + } + crc8 = drflac_crc8(crc8, reserved, 1); + + + isVariableBlockSize = blockingStrategy == 1; + if (isVariableBlockSize) { + drflac_uint64 pcmFrameNumber; + drflac_result result = drflac__read_utf8_coded_number(bs, &pcmFrameNumber, &crc8); + if (result != DRFLAC_SUCCESS) { + if (result == DRFLAC_AT_END) { + return DRFLAC_FALSE; + } else { + continue; + } + } + header->flacFrameNumber = 0; + header->pcmFrameNumber = pcmFrameNumber; + } else { + drflac_uint64 flacFrameNumber = 0; + drflac_result result = drflac__read_utf8_coded_number(bs, &flacFrameNumber, &crc8); + if (result != DRFLAC_SUCCESS) { + if (result == DRFLAC_AT_END) { + return DRFLAC_FALSE; + } else { + continue; + } + } + header->flacFrameNumber = (drflac_uint32)flacFrameNumber; /* <-- Safe cast. */ + header->pcmFrameNumber = 0; + } + + + DRFLAC_ASSERT(blockSize > 0); + if (blockSize == 1) { + header->blockSizeInPCMFrames = 192; + } else if (blockSize <= 5) { + DRFLAC_ASSERT(blockSize >= 2); + header->blockSizeInPCMFrames = 576 * (1 << (blockSize - 2)); + } else if (blockSize == 6) { + if (!drflac__read_uint16(bs, 8, &header->blockSizeInPCMFrames)) { + return DRFLAC_FALSE; + } + crc8 = drflac_crc8(crc8, header->blockSizeInPCMFrames, 8); + header->blockSizeInPCMFrames += 1; + } else if (blockSize == 7) { + if (!drflac__read_uint16(bs, 16, &header->blockSizeInPCMFrames)) { + return DRFLAC_FALSE; + } + crc8 = drflac_crc8(crc8, header->blockSizeInPCMFrames, 16); + if (header->blockSizeInPCMFrames == 0xFFFF) { + return DRFLAC_FALSE; /* Frame is too big. This is the size of the frame minus 1. The STREAMINFO block defines the max block size which is 16-bits. Adding one will make it 17 bits and therefore too big. */ + } + header->blockSizeInPCMFrames += 1; + } else { + DRFLAC_ASSERT(blockSize >= 8); + header->blockSizeInPCMFrames = 256 * (1 << (blockSize - 8)); + } + + + if (sampleRate <= 11) { + header->sampleRate = sampleRateTable[sampleRate]; + } else if (sampleRate == 12) { + if (!drflac__read_uint32(bs, 8, &header->sampleRate)) { + return DRFLAC_FALSE; + } + crc8 = drflac_crc8(crc8, header->sampleRate, 8); + header->sampleRate *= 1000; + } else if (sampleRate == 13) { + if (!drflac__read_uint32(bs, 16, &header->sampleRate)) { + return DRFLAC_FALSE; + } + crc8 = drflac_crc8(crc8, header->sampleRate, 16); + } else if (sampleRate == 14) { + if (!drflac__read_uint32(bs, 16, &header->sampleRate)) { + return DRFLAC_FALSE; + } + crc8 = drflac_crc8(crc8, header->sampleRate, 16); + header->sampleRate *= 10; + } else { + continue; /* Invalid. Assume an invalid block. */ + } + + + header->channelAssignment = channelAssignment; + + header->bitsPerSample = bitsPerSampleTable[bitsPerSample]; + if (header->bitsPerSample == 0) { + header->bitsPerSample = streaminfoBitsPerSample; + } + + if (header->bitsPerSample != streaminfoBitsPerSample) { + /* If this subframe has a different bitsPerSample then streaminfo or the first frame, reject it */ + return DRFLAC_FALSE; + } + + if (!drflac__read_uint8(bs, 8, &header->crc8)) { + return DRFLAC_FALSE; + } + +#ifndef DR_FLAC_NO_CRC + if (header->crc8 != crc8) { + continue; /* CRC mismatch. Loop back to the top and find the next sync code. */ + } +#endif + return DRFLAC_TRUE; + } +} + +static drflac_bool32 drflac__read_subframe_header(drflac_bs* bs, drflac_subframe* pSubframe) +{ + drflac_uint8 header; + int type; + + if (!drflac__read_uint8(bs, 8, &header)) { + return DRFLAC_FALSE; + } + + /* First bit should always be 0. */ + if ((header & 0x80) != 0) { + return DRFLAC_FALSE; + } + + /* + Default to 0 for the LPC order. It's important that we always set this to 0 for non LPC + and FIXED subframes because we'll be using it in a generic validation check later. + */ + pSubframe->lpcOrder = 0; + + type = (header & 0x7E) >> 1; + if (type == 0) { + pSubframe->subframeType = DRFLAC_SUBFRAME_CONSTANT; + } else if (type == 1) { + pSubframe->subframeType = DRFLAC_SUBFRAME_VERBATIM; + } else { + if ((type & 0x20) != 0) { + pSubframe->subframeType = DRFLAC_SUBFRAME_LPC; + pSubframe->lpcOrder = (drflac_uint8)(type & 0x1F) + 1; + } else if ((type & 0x08) != 0) { + pSubframe->subframeType = DRFLAC_SUBFRAME_FIXED; + pSubframe->lpcOrder = (drflac_uint8)(type & 0x07); + if (pSubframe->lpcOrder > 4) { + pSubframe->subframeType = DRFLAC_SUBFRAME_RESERVED; + pSubframe->lpcOrder = 0; + } + } else { + pSubframe->subframeType = DRFLAC_SUBFRAME_RESERVED; + } + } + + if (pSubframe->subframeType == DRFLAC_SUBFRAME_RESERVED) { + return DRFLAC_FALSE; + } + + /* Wasted bits per sample. */ + pSubframe->wastedBitsPerSample = 0; + if ((header & 0x01) == 1) { + unsigned int wastedBitsPerSample; + if (!drflac__seek_past_next_set_bit(bs, &wastedBitsPerSample)) { + return DRFLAC_FALSE; + } + pSubframe->wastedBitsPerSample = (drflac_uint8)wastedBitsPerSample + 1; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_subframe(drflac_bs* bs, drflac_frame* frame, int subframeIndex, drflac_int32* pDecodedSamplesOut) +{ + drflac_subframe* pSubframe; + drflac_uint32 subframeBitsPerSample; + drflac_bool32 decodeResult; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(frame != NULL); + + pSubframe = frame->subframes + subframeIndex; + if (!drflac__read_subframe_header(bs, pSubframe)) { + return DRFLAC_FALSE; + } + + /* Side channels require an extra bit per sample. Took a while to figure that one out... */ + subframeBitsPerSample = frame->header.bitsPerSample; + if ((frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE || frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE) && subframeIndex == 1) { + subframeBitsPerSample += 1; + } else if (frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE && subframeIndex == 0) { + subframeBitsPerSample += 1; + } + + if (subframeBitsPerSample > 32) { + /* libFLAC and ffmpeg reject 33-bit subframes as well */ + return DRFLAC_FALSE; + } + + /* Need to handle wasted bits per sample. */ + if (pSubframe->wastedBitsPerSample >= subframeBitsPerSample) { + return DRFLAC_FALSE; + } + subframeBitsPerSample -= pSubframe->wastedBitsPerSample; + + pSubframe->pSamplesS32 = pDecodedSamplesOut; + + /* + pDecodedSamplesOut will be pointing to a buffer that was allocated with enough memory to store + maxBlockSizeInPCMFrames samples (as specified in the FLAC header). We need to guard against an + overflow here. At a higher level we are checking maxBlockSizeInPCMFrames from the header, but + here we need to do an additional check to ensure this frame's block size fully encompasses any + warmup samples which is determined by the LPC order. For non LPC and FIXED subframes, the LPC + order will be have been set to 0 in drflac__read_subframe_header(). + */ + if (frame->header.blockSizeInPCMFrames < pSubframe->lpcOrder) { + return DRFLAC_FALSE; + } + + switch (pSubframe->subframeType) + { + case DRFLAC_SUBFRAME_CONSTANT: + { + decodeResult = drflac__decode_samples__constant(bs, frame->header.blockSizeInPCMFrames, subframeBitsPerSample, pSubframe->pSamplesS32); + } break; + + case DRFLAC_SUBFRAME_VERBATIM: + { + decodeResult = drflac__decode_samples__verbatim(bs, frame->header.blockSizeInPCMFrames, subframeBitsPerSample, pSubframe->pSamplesS32); + } break; + + case DRFLAC_SUBFRAME_FIXED: + { + decodeResult = drflac__decode_samples__fixed(bs, frame->header.blockSizeInPCMFrames, subframeBitsPerSample, pSubframe->lpcOrder, pSubframe->pSamplesS32); + } break; + + case DRFLAC_SUBFRAME_LPC: + { + decodeResult = drflac__decode_samples__lpc(bs, frame->header.blockSizeInPCMFrames, subframeBitsPerSample, pSubframe->lpcOrder, pSubframe->pSamplesS32); + } break; + + default: decodeResult = DRFLAC_FALSE; + } + + return decodeResult; +} + +static drflac_bool32 drflac__seek_subframe(drflac_bs* bs, drflac_frame* frame, int subframeIndex) +{ + drflac_subframe* pSubframe; + drflac_uint32 subframeBitsPerSample; + + DRFLAC_ASSERT(bs != NULL); + DRFLAC_ASSERT(frame != NULL); + + pSubframe = frame->subframes + subframeIndex; + if (!drflac__read_subframe_header(bs, pSubframe)) { + return DRFLAC_FALSE; + } + + /* Side channels require an extra bit per sample. Took a while to figure that one out... */ + subframeBitsPerSample = frame->header.bitsPerSample; + if ((frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE || frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE) && subframeIndex == 1) { + subframeBitsPerSample += 1; + } else if (frame->header.channelAssignment == DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE && subframeIndex == 0) { + subframeBitsPerSample += 1; + } + + /* Need to handle wasted bits per sample. */ + if (pSubframe->wastedBitsPerSample >= subframeBitsPerSample) { + return DRFLAC_FALSE; + } + subframeBitsPerSample -= pSubframe->wastedBitsPerSample; + + pSubframe->pSamplesS32 = NULL; + + switch (pSubframe->subframeType) + { + case DRFLAC_SUBFRAME_CONSTANT: + { + if (!drflac__seek_bits(bs, subframeBitsPerSample)) { + return DRFLAC_FALSE; + } + } break; + + case DRFLAC_SUBFRAME_VERBATIM: + { + unsigned int bitsToSeek = frame->header.blockSizeInPCMFrames * subframeBitsPerSample; + if (!drflac__seek_bits(bs, bitsToSeek)) { + return DRFLAC_FALSE; + } + } break; + + case DRFLAC_SUBFRAME_FIXED: + { + unsigned int bitsToSeek = pSubframe->lpcOrder * subframeBitsPerSample; + if (!drflac__seek_bits(bs, bitsToSeek)) { + return DRFLAC_FALSE; + } + + if (!drflac__read_and_seek_residual(bs, frame->header.blockSizeInPCMFrames, pSubframe->lpcOrder)) { + return DRFLAC_FALSE; + } + } break; + + case DRFLAC_SUBFRAME_LPC: + { + drflac_uint8 lpcPrecision; + + unsigned int bitsToSeek = pSubframe->lpcOrder * subframeBitsPerSample; + if (!drflac__seek_bits(bs, bitsToSeek)) { + return DRFLAC_FALSE; + } + + if (!drflac__read_uint8(bs, 4, &lpcPrecision)) { + return DRFLAC_FALSE; + } + if (lpcPrecision == 15) { + return DRFLAC_FALSE; /* Invalid. */ + } + lpcPrecision += 1; + + + bitsToSeek = (pSubframe->lpcOrder * lpcPrecision) + 5; /* +5 for shift. */ + if (!drflac__seek_bits(bs, bitsToSeek)) { + return DRFLAC_FALSE; + } + + if (!drflac__read_and_seek_residual(bs, frame->header.blockSizeInPCMFrames, pSubframe->lpcOrder)) { + return DRFLAC_FALSE; + } + } break; + + default: return DRFLAC_FALSE; + } + + return DRFLAC_TRUE; +} + + +static DRFLAC_INLINE drflac_uint8 drflac__get_channel_count_from_channel_assignment(drflac_int8 channelAssignment) +{ + drflac_uint8 lookup[] = {1, 2, 3, 4, 5, 6, 7, 8, 2, 2, 2}; + + DRFLAC_ASSERT(channelAssignment <= 10); + return lookup[channelAssignment]; +} + +static drflac_result drflac__decode_flac_frame(drflac* pFlac) +{ + int channelCount; + int i; + drflac_uint8 paddingSizeInBits; + drflac_uint16 desiredCRC16; +#ifndef DR_FLAC_NO_CRC + drflac_uint16 actualCRC16; +#endif + + /* This function should be called while the stream is sitting on the first byte after the frame header. */ + pFlac->currentFLACFrame.pcmFramesRemaining = 0; + DRFLAC_ZERO_MEMORY(pFlac->currentFLACFrame.subframes, sizeof(pFlac->currentFLACFrame.subframes)); + + /* The frame block size must never be larger than the maximum block size defined by the FLAC stream. */ + if (pFlac->currentFLACFrame.header.blockSizeInPCMFrames > pFlac->maxBlockSizeInPCMFrames) { + return DRFLAC_ERROR; + } + + /* The number of channels in the frame must match the channel count from the STREAMINFO block. */ + channelCount = drflac__get_channel_count_from_channel_assignment(pFlac->currentFLACFrame.header.channelAssignment); + if (channelCount != (int)pFlac->channels) { + return DRFLAC_ERROR; + } + + for (i = 0; i < channelCount; ++i) { + if (!drflac__decode_subframe(&pFlac->bs, &pFlac->currentFLACFrame, i, pFlac->pDecodedSamples + (pFlac->currentFLACFrame.header.blockSizeInPCMFrames * i))) { + return DRFLAC_ERROR; + } + } + + paddingSizeInBits = (drflac_uint8)(DRFLAC_CACHE_L1_BITS_REMAINING(&pFlac->bs) & 7); + if (paddingSizeInBits > 0) { + drflac_uint8 padding = 0; + if (!drflac__read_uint8(&pFlac->bs, paddingSizeInBits, &padding)) { + return DRFLAC_AT_END; + } + } + +#ifndef DR_FLAC_NO_CRC + actualCRC16 = drflac__flush_crc16(&pFlac->bs); +#endif + if (!drflac__read_uint16(&pFlac->bs, 16, &desiredCRC16)) { + return DRFLAC_AT_END; + } + +#ifndef DR_FLAC_NO_CRC + if (actualCRC16 != desiredCRC16) { + return DRFLAC_CRC_MISMATCH; /* CRC mismatch. */ + } +#endif + + pFlac->currentFLACFrame.pcmFramesRemaining = pFlac->currentFLACFrame.header.blockSizeInPCMFrames; + + return DRFLAC_SUCCESS; +} + +static drflac_result drflac__seek_flac_frame(drflac* pFlac) +{ + drflac_result result; + int channelCount; + int i; + drflac_uint16 desiredCRC16; +#ifndef DR_FLAC_NO_CRC + drflac_uint16 actualCRC16; +#endif + + pFlac->currentFLACFrame.pcmFramesRemaining = 0; + + channelCount = drflac__get_channel_count_from_channel_assignment(pFlac->currentFLACFrame.header.channelAssignment); + for (i = 0; i < channelCount; ++i) { + if (!drflac__seek_subframe(&pFlac->bs, &pFlac->currentFLACFrame, i)) { + result = DRFLAC_ERROR; + goto error; + } + } + + /* Padding. */ + if (!drflac__seek_bits(&pFlac->bs, DRFLAC_CACHE_L1_BITS_REMAINING(&pFlac->bs) & 7)) { + result = DRFLAC_ERROR; + goto error; + } + + /* CRC. */ +#ifndef DR_FLAC_NO_CRC + actualCRC16 = drflac__flush_crc16(&pFlac->bs); +#endif + if (!drflac__read_uint16(&pFlac->bs, 16, &desiredCRC16)) { + result = DRFLAC_AT_END; + goto error; + } + +#ifndef DR_FLAC_NO_CRC + if (actualCRC16 != desiredCRC16) { + result = DRFLAC_CRC_MISMATCH; /* CRC mismatch. */ + goto error; + } +#endif + + return DRFLAC_SUCCESS; + +error: + DRFLAC_ZERO_MEMORY(pFlac->currentFLACFrame.subframes, sizeof(pFlac->currentFLACFrame.subframes)); + return result; +} + +static drflac_bool32 drflac__read_and_decode_next_flac_frame(drflac* pFlac) +{ + DRFLAC_ASSERT(pFlac != NULL); + + for (;;) { + drflac_result result; + + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + return DRFLAC_FALSE; + } + + result = drflac__decode_flac_frame(pFlac); + if (result != DRFLAC_SUCCESS) { + if (result == DRFLAC_CRC_MISMATCH) { + continue; /* CRC mismatch. Skip to the next frame. */ + } else { + return DRFLAC_FALSE; + } + } + + return DRFLAC_TRUE; + } +} + +static void drflac__get_pcm_frame_range_of_current_flac_frame(drflac* pFlac, drflac_uint64* pFirstPCMFrame, drflac_uint64* pLastPCMFrame) +{ + drflac_uint64 firstPCMFrame; + drflac_uint64 lastPCMFrame; + + DRFLAC_ASSERT(pFlac != NULL); + + firstPCMFrame = pFlac->currentFLACFrame.header.pcmFrameNumber; + if (firstPCMFrame == 0) { + firstPCMFrame = ((drflac_uint64)pFlac->currentFLACFrame.header.flacFrameNumber) * pFlac->maxBlockSizeInPCMFrames; + } + + lastPCMFrame = firstPCMFrame + pFlac->currentFLACFrame.header.blockSizeInPCMFrames; + if (lastPCMFrame > 0) { + lastPCMFrame -= 1; /* Needs to be zero based. */ + } + + if (pFirstPCMFrame) { + *pFirstPCMFrame = firstPCMFrame; + } + if (pLastPCMFrame) { + *pLastPCMFrame = lastPCMFrame; + } +} + +static drflac_bool32 drflac__seek_to_first_frame(drflac* pFlac) +{ + drflac_bool32 result; + + DRFLAC_ASSERT(pFlac != NULL); + + result = drflac__seek_to_byte(&pFlac->bs, pFlac->firstFLACFramePosInBytes); + + DRFLAC_ZERO_MEMORY(&pFlac->currentFLACFrame, sizeof(pFlac->currentFLACFrame)); + pFlac->currentPCMFrame = 0; + + return result; +} + +static DRFLAC_INLINE drflac_result drflac__seek_to_next_flac_frame(drflac* pFlac) +{ + /* This function should only ever be called while the decoder is sitting on the first byte past the FRAME_HEADER section. */ + DRFLAC_ASSERT(pFlac != NULL); + return drflac__seek_flac_frame(pFlac); +} + + +static drflac_uint64 drflac__seek_forward_by_pcm_frames(drflac* pFlac, drflac_uint64 pcmFramesToSeek) +{ + drflac_uint64 pcmFramesRead = 0; + while (pcmFramesToSeek > 0) { + if (pFlac->currentFLACFrame.pcmFramesRemaining == 0) { + if (!drflac__read_and_decode_next_flac_frame(pFlac)) { + break; /* Couldn't read the next frame, so just break from the loop and return. */ + } + } else { + if (pFlac->currentFLACFrame.pcmFramesRemaining > pcmFramesToSeek) { + pcmFramesRead += pcmFramesToSeek; + pFlac->currentFLACFrame.pcmFramesRemaining -= (drflac_uint32)pcmFramesToSeek; /* <-- Safe cast. Will always be < currentFrame.pcmFramesRemaining < 65536. */ + pcmFramesToSeek = 0; + } else { + pcmFramesRead += pFlac->currentFLACFrame.pcmFramesRemaining; + pcmFramesToSeek -= pFlac->currentFLACFrame.pcmFramesRemaining; + pFlac->currentFLACFrame.pcmFramesRemaining = 0; + } + } + } + + pFlac->currentPCMFrame += pcmFramesRead; + return pcmFramesRead; +} + + +static drflac_bool32 drflac__seek_to_pcm_frame__brute_force(drflac* pFlac, drflac_uint64 pcmFrameIndex) +{ + drflac_bool32 isMidFrame = DRFLAC_FALSE; + drflac_uint64 runningPCMFrameCount; + + DRFLAC_ASSERT(pFlac != NULL); + + /* If we are seeking forward we start from the current position. Otherwise we need to start all the way from the start of the file. */ + if (pcmFrameIndex >= pFlac->currentPCMFrame) { + /* Seeking forward. Need to seek from the current position. */ + runningPCMFrameCount = pFlac->currentPCMFrame; + + /* The frame header for the first frame may not yet have been read. We need to do that if necessary. */ + if (pFlac->currentPCMFrame == 0 && pFlac->currentFLACFrame.pcmFramesRemaining == 0) { + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + return DRFLAC_FALSE; + } + } else { + isMidFrame = DRFLAC_TRUE; + } + } else { + /* Seeking backwards. Need to seek from the start of the file. */ + runningPCMFrameCount = 0; + + /* Move back to the start. */ + if (!drflac__seek_to_first_frame(pFlac)) { + return DRFLAC_FALSE; + } + + /* Decode the first frame in preparation for sample-exact seeking below. */ + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + return DRFLAC_FALSE; + } + } + + /* + We need to as quickly as possible find the frame that contains the target sample. To do this, we iterate over each frame and inspect its + header. If based on the header we can determine that the frame contains the sample, we do a full decode of that frame. + */ + for (;;) { + drflac_uint64 pcmFrameCountInThisFLACFrame; + drflac_uint64 firstPCMFrameInFLACFrame = 0; + drflac_uint64 lastPCMFrameInFLACFrame = 0; + + drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &firstPCMFrameInFLACFrame, &lastPCMFrameInFLACFrame); + + pcmFrameCountInThisFLACFrame = (lastPCMFrameInFLACFrame - firstPCMFrameInFLACFrame) + 1; + if (pcmFrameIndex < (runningPCMFrameCount + pcmFrameCountInThisFLACFrame)) { + /* + The sample should be in this frame. We need to fully decode it, however if it's an invalid frame (a CRC mismatch), we need to pretend + it never existed and keep iterating. + */ + drflac_uint64 pcmFramesToDecode = pcmFrameIndex - runningPCMFrameCount; + + if (!isMidFrame) { + drflac_result result = drflac__decode_flac_frame(pFlac); + if (result == DRFLAC_SUCCESS) { + /* The frame is valid. We just need to skip over some samples to ensure it's sample-exact. */ + return drflac__seek_forward_by_pcm_frames(pFlac, pcmFramesToDecode) == pcmFramesToDecode; /* <-- If this fails, something bad has happened (it should never fail). */ + } else { + if (result == DRFLAC_CRC_MISMATCH) { + goto next_iteration; /* CRC mismatch. Pretend this frame never existed. */ + } else { + return DRFLAC_FALSE; + } + } + } else { + /* We started seeking mid-frame which means we need to skip the frame decoding part. */ + return drflac__seek_forward_by_pcm_frames(pFlac, pcmFramesToDecode) == pcmFramesToDecode; + } + } else { + /* + It's not in this frame. We need to seek past the frame, but check if there was a CRC mismatch. If so, we pretend this + frame never existed and leave the running sample count untouched. + */ + if (!isMidFrame) { + drflac_result result = drflac__seek_to_next_flac_frame(pFlac); + if (result == DRFLAC_SUCCESS) { + runningPCMFrameCount += pcmFrameCountInThisFLACFrame; + } else { + if (result == DRFLAC_CRC_MISMATCH) { + goto next_iteration; /* CRC mismatch. Pretend this frame never existed. */ + } else { + return DRFLAC_FALSE; + } + } + } else { + /* + We started seeking mid-frame which means we need to seek by reading to the end of the frame instead of with + drflac__seek_to_next_flac_frame() which only works if the decoder is sitting on the byte just after the frame header. + */ + runningPCMFrameCount += pFlac->currentFLACFrame.pcmFramesRemaining; + pFlac->currentFLACFrame.pcmFramesRemaining = 0; + isMidFrame = DRFLAC_FALSE; + } + + /* If we are seeking to the end of the file and we've just hit it, we're done. */ + if (pcmFrameIndex == pFlac->totalPCMFrameCount && runningPCMFrameCount == pFlac->totalPCMFrameCount) { + return DRFLAC_TRUE; + } + } + + next_iteration: + /* Grab the next frame in preparation for the next iteration. */ + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + return DRFLAC_FALSE; + } + } +} + + +#if !defined(DR_FLAC_NO_CRC) +/* +We use an average compression ratio to determine our approximate start location. FLAC files are generally about 50%-70% the size of their +uncompressed counterparts so we'll use this as a basis. I'm going to split the middle and use a factor of 0.6 to determine the starting +location. +*/ +#define DRFLAC_BINARY_SEARCH_APPROX_COMPRESSION_RATIO 0.6f + +static drflac_bool32 drflac__seek_to_approximate_flac_frame_to_byte(drflac* pFlac, drflac_uint64 targetByte, drflac_uint64 rangeLo, drflac_uint64 rangeHi, drflac_uint64* pLastSuccessfulSeekOffset) +{ + DRFLAC_ASSERT(pFlac != NULL); + DRFLAC_ASSERT(pLastSuccessfulSeekOffset != NULL); + DRFLAC_ASSERT(targetByte >= rangeLo); + DRFLAC_ASSERT(targetByte <= rangeHi); + + *pLastSuccessfulSeekOffset = pFlac->firstFLACFramePosInBytes; + + for (;;) { + /* After rangeLo == rangeHi == targetByte fails, we need to break out. */ + drflac_uint64 lastTargetByte = targetByte; + + /* When seeking to a byte, failure probably means we've attempted to seek beyond the end of the stream. To counter this we just halve it each attempt. */ + if (!drflac__seek_to_byte(&pFlac->bs, targetByte)) { + /* If we couldn't even seek to the first byte in the stream we have a problem. Just abandon the whole thing. */ + if (targetByte == 0) { + drflac__seek_to_first_frame(pFlac); /* Try to recover. */ + return DRFLAC_FALSE; + } + + /* Halve the byte location and continue. */ + targetByte = rangeLo + ((rangeHi - rangeLo)/2); + rangeHi = targetByte; + } else { + /* Getting here should mean that we have seeked to an appropriate byte. */ + + /* Clear the details of the FLAC frame so we don't misreport data. */ + DRFLAC_ZERO_MEMORY(&pFlac->currentFLACFrame, sizeof(pFlac->currentFLACFrame)); + + /* + Now seek to the next FLAC frame. We need to decode the entire frame (not just the header) because it's possible for the header to incorrectly pass the + CRC check and return bad data. We need to decode the entire frame to be more certain. Although this seems unlikely, this has happened to me in testing + so it needs to stay this way for now. + */ +#if 1 + if (!drflac__read_and_decode_next_flac_frame(pFlac)) { + /* Halve the byte location and continue. */ + targetByte = rangeLo + ((rangeHi - rangeLo)/2); + rangeHi = targetByte; + } else { + break; + } +#else + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + /* Halve the byte location and continue. */ + targetByte = rangeLo + ((rangeHi - rangeLo)/2); + rangeHi = targetByte; + } else { + break; + } +#endif + } + + /* We already tried this byte and there are no more to try, break out. */ + if(targetByte == lastTargetByte) { + return DRFLAC_FALSE; + } + } + + /* The current PCM frame needs to be updated based on the frame we just seeked to. */ + drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &pFlac->currentPCMFrame, NULL); + + DRFLAC_ASSERT(targetByte <= rangeHi); + + *pLastSuccessfulSeekOffset = targetByte; + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__decode_flac_frame_and_seek_forward_by_pcm_frames(drflac* pFlac, drflac_uint64 offset) +{ + /* This section of code would be used if we were only decoding the FLAC frame header when calling drflac__seek_to_approximate_flac_frame_to_byte(). */ +#if 0 + if (drflac__decode_flac_frame(pFlac) != DRFLAC_SUCCESS) { + /* We failed to decode this frame which may be due to it being corrupt. We'll just use the next valid FLAC frame. */ + if (drflac__read_and_decode_next_flac_frame(pFlac) == DRFLAC_FALSE) { + return DRFLAC_FALSE; + } + } +#endif + + return drflac__seek_forward_by_pcm_frames(pFlac, offset) == offset; +} + + +static drflac_bool32 drflac__seek_to_pcm_frame__binary_search_internal(drflac* pFlac, drflac_uint64 pcmFrameIndex, drflac_uint64 byteRangeLo, drflac_uint64 byteRangeHi) +{ + /* This assumes pFlac->currentPCMFrame is sitting on byteRangeLo upon entry. */ + + drflac_uint64 targetByte; + drflac_uint64 pcmRangeLo = pFlac->totalPCMFrameCount; + drflac_uint64 pcmRangeHi = 0; + drflac_uint64 lastSuccessfulSeekOffset = (drflac_uint64)-1; + drflac_uint64 closestSeekOffsetBeforeTargetPCMFrame = byteRangeLo; + drflac_uint32 seekForwardThreshold = (pFlac->maxBlockSizeInPCMFrames != 0) ? pFlac->maxBlockSizeInPCMFrames*2 : 4096; + + targetByte = byteRangeLo + (drflac_uint64)(((drflac_int64)((pcmFrameIndex - pFlac->currentPCMFrame) * pFlac->channels * pFlac->bitsPerSample)/8.0f) * DRFLAC_BINARY_SEARCH_APPROX_COMPRESSION_RATIO); + if (targetByte > byteRangeHi) { + targetByte = byteRangeHi; + } + + for (;;) { + /* + If only two adjacent byte offsets remain, binary search cannot narrow the range any further. Seek to the closest frame before the target and decode + forward from there. + */ + if ((byteRangeHi - byteRangeLo) == 1) { + if (!drflac__seek_to_approximate_flac_frame_to_byte(pFlac, closestSeekOffsetBeforeTargetPCMFrame, closestSeekOffsetBeforeTargetPCMFrame, byteRangeHi, &lastSuccessfulSeekOffset)) { + break; + } + + if (pFlac->currentPCMFrame <= pcmFrameIndex && drflac__decode_flac_frame_and_seek_forward_by_pcm_frames(pFlac, pcmFrameIndex - pFlac->currentPCMFrame)) { + return DRFLAC_TRUE; + } + + break; + } + + if (drflac__seek_to_approximate_flac_frame_to_byte(pFlac, targetByte, byteRangeLo, byteRangeHi, &lastSuccessfulSeekOffset)) { + /* We found a FLAC frame. We need to check if it contains the sample we're looking for. */ + drflac_uint64 newPCMRangeLo; + drflac_uint64 newPCMRangeHi; + drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &newPCMRangeLo, &newPCMRangeHi); + + /* If we selected the same frame, it means we should be pretty close. Just decode the rest. */ + if (pcmRangeLo == newPCMRangeLo) { + if (!drflac__seek_to_approximate_flac_frame_to_byte(pFlac, closestSeekOffsetBeforeTargetPCMFrame, closestSeekOffsetBeforeTargetPCMFrame, byteRangeHi, &lastSuccessfulSeekOffset)) { + break; /* Failed to seek to closest frame. */ + } + + if (drflac__decode_flac_frame_and_seek_forward_by_pcm_frames(pFlac, pcmFrameIndex - pFlac->currentPCMFrame)) { + return DRFLAC_TRUE; + } else { + break; /* Failed to seek forward. */ + } + } + + pcmRangeLo = newPCMRangeLo; + pcmRangeHi = newPCMRangeHi; + + if (pcmRangeLo <= pcmFrameIndex && pcmRangeHi >= pcmFrameIndex) { + /* The target PCM frame is in this FLAC frame. */ + if (drflac__decode_flac_frame_and_seek_forward_by_pcm_frames(pFlac, pcmFrameIndex - pFlac->currentPCMFrame) ) { + return DRFLAC_TRUE; + } else { + break; /* Failed to seek to FLAC frame. */ + } + } else { + if (pcmRangeLo > pcmFrameIndex) { + /* We seeked too far forward. We need to move our target byte backward and try again. */ + byteRangeHi = lastSuccessfulSeekOffset; + if (byteRangeLo > byteRangeHi) { + byteRangeLo = byteRangeHi; + } + + targetByte = byteRangeLo + ((byteRangeHi - byteRangeLo) / 2); + if (targetByte < byteRangeLo) { + targetByte = byteRangeLo; + } + } else /*if (pcmRangeHi < pcmFrameIndex)*/ { + /* We didn't seek far enough. We need to move our target byte forward and try again. */ + + /* If we're close enough we can just seek forward. */ + if ((pcmFrameIndex - pcmRangeLo) < seekForwardThreshold) { + if (drflac__decode_flac_frame_and_seek_forward_by_pcm_frames(pFlac, pcmFrameIndex - pFlac->currentPCMFrame)) { + return DRFLAC_TRUE; + } else { + break; /* Failed to seek to FLAC frame. */ + } + } else { + const double approxCompressionRatio = (drflac_int64)(lastSuccessfulSeekOffset - pFlac->firstFLACFramePosInBytes) / ((drflac_int64)(pcmRangeLo * pFlac->channels * pFlac->bitsPerSample)/8.0); + + byteRangeLo = lastSuccessfulSeekOffset; + if (byteRangeHi < byteRangeLo) { + byteRangeHi = byteRangeLo; + } + + targetByte = lastSuccessfulSeekOffset + (drflac_uint64)(((drflac_int64)((pcmFrameIndex-pcmRangeLo) * pFlac->channels * pFlac->bitsPerSample)/8.0) * approxCompressionRatio); + if (targetByte > byteRangeHi) { + targetByte = byteRangeHi; + } + + if (closestSeekOffsetBeforeTargetPCMFrame < lastSuccessfulSeekOffset) { + closestSeekOffsetBeforeTargetPCMFrame = lastSuccessfulSeekOffset; + } + } + } + } + } else { + /* Getting here is really bad. We just recover as best we can, but moving to the first frame in the stream, and then abort. */ + break; + } + } + + drflac__seek_to_first_frame(pFlac); /* <-- Try to recover. */ + return DRFLAC_FALSE; +} + +static drflac_bool32 drflac__seek_to_pcm_frame__binary_search(drflac* pFlac, drflac_uint64 pcmFrameIndex) +{ + drflac_uint64 byteRangeLo; + drflac_uint64 byteRangeHi; + drflac_uint32 seekForwardThreshold = (pFlac->maxBlockSizeInPCMFrames != 0) ? pFlac->maxBlockSizeInPCMFrames*2 : 4096; + + /* Our algorithm currently assumes the FLAC stream is currently sitting at the start. */ + if (drflac__seek_to_first_frame(pFlac) == DRFLAC_FALSE) { + return DRFLAC_FALSE; + } + + /* If we're close enough to the start, just move to the start and seek forward. */ + if (pcmFrameIndex < seekForwardThreshold) { + return drflac__seek_forward_by_pcm_frames(pFlac, pcmFrameIndex) == pcmFrameIndex; + } + + /* + Our starting byte range is the byte position of the first FLAC frame and the approximate end of the file as if it were completely uncompressed. This ensures + the entire file is included, even though most of the time it'll exceed the end of the actual stream. This is OK as the frame searching logic will handle it. + */ + byteRangeLo = pFlac->firstFLACFramePosInBytes; + byteRangeHi = pFlac->firstFLACFramePosInBytes + (drflac_uint64)((drflac_int64)(pFlac->totalPCMFrameCount * pFlac->channels * pFlac->bitsPerSample)/8.0f); + + return drflac__seek_to_pcm_frame__binary_search_internal(pFlac, pcmFrameIndex, byteRangeLo, byteRangeHi); +} +#endif /* !DR_FLAC_NO_CRC */ + +static drflac_bool32 drflac__seek_to_pcm_frame__seek_table(drflac* pFlac, drflac_uint64 pcmFrameIndex) +{ + drflac_uint32 iClosestSeekpoint = 0; + drflac_bool32 isMidFrame = DRFLAC_FALSE; + drflac_uint64 runningPCMFrameCount; + drflac_uint32 iSeekpoint; + + + DRFLAC_ASSERT(pFlac != NULL); + + if (pFlac->pSeekpoints == NULL || pFlac->seekpointCount == 0) { + return DRFLAC_FALSE; + } + + /* Do not use the seektable if pcmFramIndex is not coverd by it. */ + if (pFlac->pSeekpoints[0].firstPCMFrame > pcmFrameIndex) { + return DRFLAC_FALSE; + } + + for (iSeekpoint = 0; iSeekpoint < pFlac->seekpointCount; ++iSeekpoint) { + if (pFlac->pSeekpoints[iSeekpoint].firstPCMFrame >= pcmFrameIndex) { + break; + } + + iClosestSeekpoint = iSeekpoint; + } + + /* There's been cases where the seek table contains only zeros. We need to do some basic validation on the closest seekpoint. */ + if (pFlac->pSeekpoints[iClosestSeekpoint].pcmFrameCount == 0 || pFlac->pSeekpoints[iClosestSeekpoint].pcmFrameCount > pFlac->maxBlockSizeInPCMFrames) { + return DRFLAC_FALSE; + } + if (pFlac->pSeekpoints[iClosestSeekpoint].firstPCMFrame > pFlac->totalPCMFrameCount && pFlac->totalPCMFrameCount > 0) { + return DRFLAC_FALSE; + } + +#if !defined(DR_FLAC_NO_CRC) + /* At this point we should know the closest seek point. We can use a binary search for this. We need to know the total sample count for this. */ + if (pFlac->totalPCMFrameCount > 0) { + drflac_uint64 byteRangeLo; + drflac_uint64 byteRangeHi; + + byteRangeHi = pFlac->firstFLACFramePosInBytes + (drflac_uint64)((drflac_int64)(pFlac->totalPCMFrameCount * pFlac->channels * pFlac->bitsPerSample)/8.0f); + byteRangeLo = pFlac->firstFLACFramePosInBytes + pFlac->pSeekpoints[iClosestSeekpoint].flacFrameOffset; + + /* + If our closest seek point is not the last one, we only need to search between it and the next one. The section below calculates an appropriate starting + value for byteRangeHi which will clamp it appropriately. + + Note that the next seekpoint must have an offset greater than the closest seekpoint because otherwise our binary search algorithm will break down. There + have been cases where a seektable consists of seek points where every byte offset is set to 0 which causes problems. If this happens we need to abort. + */ + if (iClosestSeekpoint < pFlac->seekpointCount-1) { + drflac_uint32 iNextSeekpoint = iClosestSeekpoint + 1; + + /* Basic validation on the seekpoints to ensure they're usable. */ + if (pFlac->pSeekpoints[iClosestSeekpoint].flacFrameOffset >= pFlac->pSeekpoints[iNextSeekpoint].flacFrameOffset || pFlac->pSeekpoints[iNextSeekpoint].pcmFrameCount == 0) { + return DRFLAC_FALSE; /* The next seekpoint doesn't look right. The seek table cannot be trusted from here. Abort. */ + } + + if (pFlac->pSeekpoints[iNextSeekpoint].firstPCMFrame != (((drflac_uint64)0xFFFFFFFF << 32) | 0xFFFFFFFF)) { /* Make sure it's not a placeholder seekpoint. */ + byteRangeHi = pFlac->firstFLACFramePosInBytes + pFlac->pSeekpoints[iNextSeekpoint].flacFrameOffset - 1; /* byteRangeHi must be zero based. */ + } + } + + if (drflac__seek_to_byte(&pFlac->bs, pFlac->firstFLACFramePosInBytes + pFlac->pSeekpoints[iClosestSeekpoint].flacFrameOffset)) { + if (drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &pFlac->currentPCMFrame, NULL); + + if (drflac__seek_to_pcm_frame__binary_search_internal(pFlac, pcmFrameIndex, byteRangeLo, byteRangeHi)) { + return DRFLAC_TRUE; + } + } + } + } +#endif /* !DR_FLAC_NO_CRC */ + + /* Getting here means we need to use a slower algorithm because the binary search method failed or cannot be used. */ + + /* + If we are seeking forward and the closest seekpoint is _before_ the current sample, we just seek forward from where we are. Otherwise we start seeking + from the seekpoint's first sample. + */ + if (pcmFrameIndex >= pFlac->currentPCMFrame && pFlac->pSeekpoints[iClosestSeekpoint].firstPCMFrame <= pFlac->currentPCMFrame) { + /* Optimized case. Just seek forward from where we are. */ + runningPCMFrameCount = pFlac->currentPCMFrame; + + /* The frame header for the first frame may not yet have been read. We need to do that if necessary. */ + if (pFlac->currentPCMFrame == 0 && pFlac->currentFLACFrame.pcmFramesRemaining == 0) { + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + return DRFLAC_FALSE; + } + } else { + isMidFrame = DRFLAC_TRUE; + } + } else { + /* Slower case. Seek to the start of the seekpoint and then seek forward from there. */ + runningPCMFrameCount = pFlac->pSeekpoints[iClosestSeekpoint].firstPCMFrame; + + if (!drflac__seek_to_byte(&pFlac->bs, pFlac->firstFLACFramePosInBytes + pFlac->pSeekpoints[iClosestSeekpoint].flacFrameOffset)) { + return DRFLAC_FALSE; + } + + /* Grab the frame the seekpoint is sitting on in preparation for the sample-exact seeking below. */ + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + return DRFLAC_FALSE; + } + } + + for (;;) { + drflac_uint64 pcmFrameCountInThisFLACFrame; + drflac_uint64 firstPCMFrameInFLACFrame = 0; + drflac_uint64 lastPCMFrameInFLACFrame = 0; + + drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &firstPCMFrameInFLACFrame, &lastPCMFrameInFLACFrame); + + pcmFrameCountInThisFLACFrame = (lastPCMFrameInFLACFrame - firstPCMFrameInFLACFrame) + 1; + if (pcmFrameIndex < (runningPCMFrameCount + pcmFrameCountInThisFLACFrame)) { + /* + The sample should be in this frame. We need to fully decode it, but if it's an invalid frame (a CRC mismatch) we need to pretend + it never existed and keep iterating. + */ + drflac_uint64 pcmFramesToDecode = pcmFrameIndex - runningPCMFrameCount; + + if (!isMidFrame) { + drflac_result result = drflac__decode_flac_frame(pFlac); + if (result == DRFLAC_SUCCESS) { + /* The frame is valid. We just need to skip over some samples to ensure it's sample-exact. */ + return drflac__seek_forward_by_pcm_frames(pFlac, pcmFramesToDecode) == pcmFramesToDecode; /* <-- If this fails, something bad has happened (it should never fail). */ + } else { + if (result == DRFLAC_CRC_MISMATCH) { + goto next_iteration; /* CRC mismatch. Pretend this frame never existed. */ + } else { + return DRFLAC_FALSE; + } + } + } else { + /* We started seeking mid-frame which means we need to skip the frame decoding part. */ + return drflac__seek_forward_by_pcm_frames(pFlac, pcmFramesToDecode) == pcmFramesToDecode; + } + } else { + /* + It's not in this frame. We need to seek past the frame, but check if there was a CRC mismatch. If so, we pretend this + frame never existed and leave the running sample count untouched. + */ + if (!isMidFrame) { + drflac_result result = drflac__seek_to_next_flac_frame(pFlac); + if (result == DRFLAC_SUCCESS) { + runningPCMFrameCount += pcmFrameCountInThisFLACFrame; + } else { + if (result == DRFLAC_CRC_MISMATCH) { + goto next_iteration; /* CRC mismatch. Pretend this frame never existed. */ + } else { + return DRFLAC_FALSE; + } + } + } else { + /* + We started seeking mid-frame which means we need to seek by reading to the end of the frame instead of with + drflac__seek_to_next_flac_frame() which only works if the decoder is sitting on the byte just after the frame header. + */ + runningPCMFrameCount += pFlac->currentFLACFrame.pcmFramesRemaining; + pFlac->currentFLACFrame.pcmFramesRemaining = 0; + isMidFrame = DRFLAC_FALSE; + } + + /* If we are seeking to the end of the file and we've just hit it, we're done. */ + if (pcmFrameIndex == pFlac->totalPCMFrameCount && runningPCMFrameCount == pFlac->totalPCMFrameCount) { + return DRFLAC_TRUE; + } + } + + next_iteration: + /* Grab the next frame in preparation for the next iteration. */ + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + return DRFLAC_FALSE; + } + } +} + + +#ifndef DR_FLAC_NO_OGG +typedef struct +{ + drflac_uint8 capturePattern[4]; /* Should be "OggS" */ + drflac_uint8 structureVersion; /* Always 0. */ + drflac_uint8 headerType; + drflac_uint64 granulePosition; + drflac_uint32 serialNumber; + drflac_uint32 sequenceNumber; + drflac_uint32 checksum; + drflac_uint8 segmentCount; + drflac_uint8 segmentTable[255]; +} drflac_ogg_page_header; +#endif + +typedef struct +{ + drflac_read_proc onRead; + drflac_seek_proc onSeek; + drflac_tell_proc onTell; + drflac_meta_proc onMeta; + drflac_container container; + void* pUserData; + void* pUserDataMD; + drflac_uint32 sampleRate; + drflac_uint8 channels; + drflac_uint8 bitsPerSample; + drflac_uint64 totalPCMFrameCount; + drflac_uint16 maxBlockSizeInPCMFrames; + drflac_uint64 runningFilePos; + drflac_bool32 hasStreamInfoBlock; + drflac_bool32 hasMetadataBlocks; + drflac_bs bs; /* <-- A bit streamer is required for loading data during initialization. */ + drflac_frame_header firstFrameHeader; /* <-- The header of the first frame that was read during relaxed initalization. Only set if there is no STREAMINFO block. */ + +#ifndef DR_FLAC_NO_OGG + drflac_uint32 oggSerial; + drflac_uint64 oggFirstBytePos; + drflac_ogg_page_header oggBosHeader; +#endif +} drflac_init_info; + +static DRFLAC_INLINE void drflac__decode_block_header(drflac_uint32 blockHeader, drflac_uint8* isLastBlock, drflac_uint8* blockType, drflac_uint32* blockSize) +{ + blockHeader = drflac__be2host_32(blockHeader); + *isLastBlock = (drflac_uint8)((blockHeader & 0x80000000UL) >> 31); + *blockType = (drflac_uint8)((blockHeader & 0x7F000000UL) >> 24); + *blockSize = (blockHeader & 0x00FFFFFFUL); +} + +static DRFLAC_INLINE drflac_bool32 drflac__read_and_decode_block_header(drflac_read_proc onRead, void* pUserData, drflac_uint8* isLastBlock, drflac_uint8* blockType, drflac_uint32* blockSize) +{ + drflac_uint32 blockHeader; + + *blockSize = 0; + if (onRead(pUserData, &blockHeader, 4) != 4) { + return DRFLAC_FALSE; + } + + drflac__decode_block_header(blockHeader, isLastBlock, blockType, blockSize); + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__read_streaminfo(drflac_read_proc onRead, void* pUserData, drflac_streaminfo* pStreamInfo) +{ + drflac_uint32 blockSizes; + drflac_uint64 frameSizes = 0; + drflac_uint64 importantProps; + drflac_uint8 md5[16]; + + /* min/max block size. */ + if (onRead(pUserData, &blockSizes, 4) != 4) { + return DRFLAC_FALSE; + } + + /* min/max frame size. */ + if (onRead(pUserData, &frameSizes, 6) != 6) { + return DRFLAC_FALSE; + } + + /* Sample rate, channels, bits per sample and total sample count. */ + if (onRead(pUserData, &importantProps, 8) != 8) { + return DRFLAC_FALSE; + } + + /* MD5 */ + if (onRead(pUserData, md5, sizeof(md5)) != sizeof(md5)) { + return DRFLAC_FALSE; + } + + blockSizes = drflac__be2host_32(blockSizes); + frameSizes = drflac__be2host_64(frameSizes); + importantProps = drflac__be2host_64(importantProps); + + pStreamInfo->minBlockSizeInPCMFrames = (drflac_uint16)((blockSizes & 0xFFFF0000) >> 16); + pStreamInfo->maxBlockSizeInPCMFrames = (drflac_uint16) (blockSizes & 0x0000FFFF); + pStreamInfo->minFrameSizeInPCMFrames = (drflac_uint32)((frameSizes & (((drflac_uint64)0x00FFFFFF << 16) << 24)) >> 40); + pStreamInfo->maxFrameSizeInPCMFrames = (drflac_uint32)((frameSizes & (((drflac_uint64)0x00FFFFFF << 16) << 0)) >> 16); + pStreamInfo->sampleRate = (drflac_uint32)((importantProps & (((drflac_uint64)0x000FFFFF << 16) << 28)) >> 44); + pStreamInfo->channels = (drflac_uint8 )((importantProps & (((drflac_uint64)0x0000000E << 16) << 24)) >> 41) + 1; + pStreamInfo->bitsPerSample = (drflac_uint8 )((importantProps & (((drflac_uint64)0x0000001F << 16) << 20)) >> 36) + 1; + pStreamInfo->totalPCMFrameCount = ((importantProps & ((((drflac_uint64)0x0000000F << 16) << 16) | 0xFFFFFFFF))); + DRFLAC_COPY_MEMORY(pStreamInfo->md5, md5, sizeof(md5)); + + return DRFLAC_TRUE; +} + + +static void* drflac__malloc_default(size_t sz, void* pUserData) +{ + (void)pUserData; + return DRFLAC_MALLOC(sz); +} + +static void* drflac__realloc_default(void* p, size_t sz, void* pUserData) +{ + (void)pUserData; + return DRFLAC_REALLOC(p, sz); +} + +static void drflac__free_default(void* p, void* pUserData) +{ + (void)pUserData; + DRFLAC_FREE(p); +} + + +static void* drflac__malloc_from_callbacks(size_t sz, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks == NULL) { + return NULL; + } + + if (pAllocationCallbacks->onMalloc != NULL) { + return pAllocationCallbacks->onMalloc(sz, pAllocationCallbacks->pUserData); + } + + /* Try using realloc(). */ + if (pAllocationCallbacks->onRealloc != NULL) { + return pAllocationCallbacks->onRealloc(NULL, sz, pAllocationCallbacks->pUserData); + } + + return NULL; +} + +static void* drflac__realloc_from_callbacks(void* p, size_t szNew, size_t szOld, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks == NULL) { + return NULL; + } + + if (pAllocationCallbacks->onRealloc != NULL) { + return pAllocationCallbacks->onRealloc(p, szNew, pAllocationCallbacks->pUserData); + } + + /* Try emulating realloc() in terms of malloc()/free(). */ + if (pAllocationCallbacks->onMalloc != NULL && pAllocationCallbacks->onFree != NULL) { + void* p2; + + p2 = pAllocationCallbacks->onMalloc(szNew, pAllocationCallbacks->pUserData); + if (p2 == NULL) { + return NULL; + } + + if (p != NULL) { + DRFLAC_COPY_MEMORY(p2, p, DRFLAC_MIN(szNew, szOld)); + pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData); + } + + return p2; + } + + return NULL; +} + +static void drflac__free_from_callbacks(void* p, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + if (p == NULL || pAllocationCallbacks == NULL) { + return; + } + + if (pAllocationCallbacks->onFree != NULL) { + pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData); + } +} + + +static drflac_bool32 drflac__read_and_decode_metadata(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_tell_proc onTell, drflac_meta_proc onMeta, void* pUserData, void* pUserDataMD, drflac_uint64* pFirstFramePos, drflac_uint64* pSeektablePos, drflac_uint32* pSeekpointCount, drflac_allocation_callbacks* pAllocationCallbacks) +{ + /* + We want to keep track of the byte position in the stream of the seektable. At the time of calling this function we know that + we'll be sitting on byte 42. + */ + drflac_uint64 runningFilePos = 42; + drflac_uint64 seektablePos = 0; + drflac_uint32 seektableSize = 0; + drflac_int64 fileSize = 0; + drflac_bool32 hasKnownFileSize = DRFLAC_FALSE; + + /* We'll be doing some memory allocations here against untrusted data. We'll do a basic validation check that they don't exceed the size of the file. */ + if (onTell != NULL && onSeek != NULL) { + if (onSeek(pUserData, 0, DRFLAC_SEEK_END)) { + if (onTell(pUserData, &fileSize)) { + hasKnownFileSize = DRFLAC_TRUE; + } + + onSeek(pUserData, (int)runningFilePos, DRFLAC_SEEK_SET); /* Safe cast because runningFilePos should always be 42 at this point. */ + } + } + + for (;;) { + drflac_metadata metadata; + drflac_uint8 isLastBlock = 0; + drflac_uint8 blockType = 0; + drflac_uint32 blockSize; + if (drflac__read_and_decode_block_header(onRead, pUserData, &isLastBlock, &blockType, &blockSize) == DRFLAC_FALSE) { + return DRFLAC_FALSE; + } + + if (hasKnownFileSize && (blockSize > ((drflac_uint64)fileSize - runningFilePos))) { + return DRFLAC_FALSE; /* Block size exceeds the size of the file. */ + } + + runningFilePos += 4; + + metadata.type = blockType; + metadata.rawDataSize = 0; + metadata.rawDataOffset = runningFilePos; + metadata.pRawData = NULL; + + switch (blockType) + { + case DRFLAC_METADATA_BLOCK_TYPE_APPLICATION: + { + if (blockSize < 4) { + return DRFLAC_FALSE; + } + + if (onMeta) { + void* pRawData = drflac__malloc_from_callbacks(blockSize, pAllocationCallbacks); + if (pRawData == NULL) { + return DRFLAC_FALSE; + } + + if (onRead(pUserData, pRawData, blockSize) != blockSize) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + metadata.pRawData = pRawData; + metadata.rawDataSize = blockSize; + metadata.data.application.id = drflac__be2host_32(*(drflac_uint32*)pRawData); + metadata.data.application.pData = (const void*)((drflac_uint8*)pRawData + sizeof(drflac_uint32)); + metadata.data.application.dataSize = blockSize - sizeof(drflac_uint32); + onMeta(pUserDataMD, &metadata); + + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + } + } break; + + case DRFLAC_METADATA_BLOCK_TYPE_SEEKTABLE: + { + seektablePos = runningFilePos; + seektableSize = blockSize; + + if (onMeta) { + drflac_uint32 seekpointCount; + drflac_uint32 iSeekpoint; + void* pRawData; + size_t rawDataSize; + + seekpointCount = blockSize/DRFLAC_SEEKPOINT_SIZE_IN_BYTES; + rawDataSize = seekpointCount * sizeof(drflac_seekpoint); + + pRawData = drflac__malloc_from_callbacks(rawDataSize, pAllocationCallbacks); + if (pRawData == NULL) { + return DRFLAC_FALSE; + } + + /* We need to read seekpoint by seekpoint and do some processing. */ + for (iSeekpoint = 0; iSeekpoint < seekpointCount; ++iSeekpoint) { + drflac_seekpoint* pSeekpoint = (drflac_seekpoint*)pRawData + iSeekpoint; + + if (onRead(pUserData, pSeekpoint, DRFLAC_SEEKPOINT_SIZE_IN_BYTES) != DRFLAC_SEEKPOINT_SIZE_IN_BYTES) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + /* Endian swap. */ + pSeekpoint->firstPCMFrame = drflac__be2host_64(pSeekpoint->firstPCMFrame); + pSeekpoint->flacFrameOffset = drflac__be2host_64(pSeekpoint->flacFrameOffset); + pSeekpoint->pcmFrameCount = drflac__be2host_16(pSeekpoint->pcmFrameCount); + } + + metadata.pRawData = pRawData; + metadata.rawDataSize = rawDataSize; + metadata.data.seektable.seekpointCount = seekpointCount; + metadata.data.seektable.pSeekpoints = (const drflac_seekpoint*)pRawData; + + onMeta(pUserDataMD, &metadata); + + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + } + } break; + + case DRFLAC_METADATA_BLOCK_TYPE_VORBIS_COMMENT: + { + if (blockSize < 8) { + return DRFLAC_FALSE; + } + + if (onMeta) { + void* pRawData; + const char* pRunningData; + const char* pRunningDataEnd; + drflac_uint32 i; + + pRawData = drflac__malloc_from_callbacks(blockSize, pAllocationCallbacks); + if (pRawData == NULL) { + return DRFLAC_FALSE; + } + + if (onRead(pUserData, pRawData, blockSize) != blockSize) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + metadata.pRawData = pRawData; + metadata.rawDataSize = blockSize; + + pRunningData = (const char*)pRawData; + pRunningDataEnd = (const char*)pRawData + blockSize; + + metadata.data.vorbis_comment.vendorLength = drflac__le2host_32_ptr_unaligned(pRunningData); pRunningData += 4; + + /* Need space for the rest of the block */ + if ((pRunningDataEnd - pRunningData) - 4 < (drflac_int64)metadata.data.vorbis_comment.vendorLength) { /* <-- Note the order of operations to avoid overflow to a valid value */ + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + metadata.data.vorbis_comment.vendor = pRunningData; pRunningData += metadata.data.vorbis_comment.vendorLength; + metadata.data.vorbis_comment.commentCount = drflac__le2host_32_ptr_unaligned(pRunningData); pRunningData += 4; + + /* Need space for 'commentCount' comments after the block, which at minimum is a drflac_uint32 per comment */ + if ((pRunningDataEnd - pRunningData) / sizeof(drflac_uint32) < metadata.data.vorbis_comment.commentCount) { /* <-- Note the order of operations to avoid overflow to a valid value */ + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + metadata.data.vorbis_comment.pComments = pRunningData; + + /* Check that the comments section is valid before passing it to the callback */ + for (i = 0; i < metadata.data.vorbis_comment.commentCount; ++i) { + drflac_uint32 commentLength; + + if (pRunningDataEnd - pRunningData < 4) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + commentLength = drflac__le2host_32_ptr_unaligned(pRunningData); pRunningData += 4; + if (pRunningDataEnd - pRunningData < (drflac_int64)commentLength) { /* <-- Note the order of operations to avoid overflow to a valid value */ + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + pRunningData += commentLength; + } + + onMeta(pUserDataMD, &metadata); + + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + } + } break; + + case DRFLAC_METADATA_BLOCK_TYPE_CUESHEET: + { + if (blockSize < 396) { + return DRFLAC_FALSE; + } + + if (onMeta) { + void* pRawData; + const char* pRunningData; + const char* pRunningDataEnd; + size_t bufferSize; + drflac_uint8 iTrack; + drflac_uint8 iIndex; + void* pTrackData; + + /* + This needs to be loaded in two passes. The first pass is used to calculate the size of the memory allocation + we need for storing the necessary data. The second pass will fill that buffer with usable data. + */ + pRawData = drflac__malloc_from_callbacks(blockSize, pAllocationCallbacks); + if (pRawData == NULL) { + return DRFLAC_FALSE; + } + + if (onRead(pUserData, pRawData, blockSize) != blockSize) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + metadata.pRawData = pRawData; + metadata.rawDataSize = blockSize; + + pRunningData = (const char*)pRawData; + pRunningDataEnd = (const char*)pRawData + blockSize; + + DRFLAC_COPY_MEMORY(metadata.data.cuesheet.catalog, pRunningData, 128); pRunningData += 128; + metadata.data.cuesheet.leadInSampleCount = drflac__be2host_64(*(const drflac_uint64*)pRunningData); pRunningData += 8; + metadata.data.cuesheet.isCD = (pRunningData[0] & 0x80) != 0; pRunningData += 259; + metadata.data.cuesheet.trackCount = pRunningData[0]; pRunningData += 1; + metadata.data.cuesheet.pTrackData = NULL; /* Will be filled later. */ + + /* Pass 1: Calculate the size of the buffer for the track data. */ + { + const char* pRunningDataSaved = pRunningData; /* Will be restored at the end in preparation for the second pass. */ + + bufferSize = metadata.data.cuesheet.trackCount * DRFLAC_CUESHEET_TRACK_SIZE_IN_BYTES; + + for (iTrack = 0; iTrack < metadata.data.cuesheet.trackCount; ++iTrack) { + drflac_uint8 indexCount; + drflac_uint32 indexPointSize; + + if (pRunningDataEnd - pRunningData < DRFLAC_CUESHEET_TRACK_SIZE_IN_BYTES) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + /* Skip to the index point count */ + pRunningData += 35; + + indexCount = pRunningData[0]; + pRunningData += 1; + + bufferSize += indexCount * sizeof(drflac_cuesheet_track_index); + + /* Quick validation check. */ + indexPointSize = indexCount * DRFLAC_CUESHEET_TRACK_INDEX_SIZE_IN_BYTES; + if (pRunningDataEnd - pRunningData < (drflac_int64)indexPointSize) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + pRunningData += indexPointSize; + } + + pRunningData = pRunningDataSaved; + } + + /* Pass 2: Allocate a buffer and fill the data. Validation was done in the step above so can be skipped. */ + { + char* pRunningTrackData; + + pTrackData = drflac__malloc_from_callbacks(bufferSize, pAllocationCallbacks); + if (pTrackData == NULL) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + + pRunningTrackData = (char*)pTrackData; + + for (iTrack = 0; iTrack < metadata.data.cuesheet.trackCount; ++iTrack) { + drflac_uint8 indexCount; + + DRFLAC_COPY_MEMORY(pRunningTrackData, pRunningData, DRFLAC_CUESHEET_TRACK_SIZE_IN_BYTES); + pRunningData += DRFLAC_CUESHEET_TRACK_SIZE_IN_BYTES-1; /* Skip forward, but not beyond the last byte in the CUESHEET_TRACK block which is the index count. */ + pRunningTrackData += DRFLAC_CUESHEET_TRACK_SIZE_IN_BYTES-1; + + /* Grab the index count for the next part. */ + indexCount = pRunningData[0]; + pRunningData += 1; + pRunningTrackData += 1; + + /* Extract each track index. */ + for (iIndex = 0; iIndex < indexCount; ++iIndex) { + drflac_cuesheet_track_index* pTrackIndex = (drflac_cuesheet_track_index*)pRunningTrackData; + + DRFLAC_COPY_MEMORY(pRunningTrackData, pRunningData, DRFLAC_CUESHEET_TRACK_INDEX_SIZE_IN_BYTES); + pRunningData += DRFLAC_CUESHEET_TRACK_INDEX_SIZE_IN_BYTES; + pRunningTrackData += sizeof(drflac_cuesheet_track_index); + + pTrackIndex->offset = drflac__be2host_64(pTrackIndex->offset); + } + } + + metadata.data.cuesheet.pTrackData = pTrackData; + } + + /* The original data is no longer needed. */ + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + pRawData = NULL; + + onMeta(pUserDataMD, &metadata); + + drflac__free_from_callbacks(pTrackData, pAllocationCallbacks); + pTrackData = NULL; + } + } break; + + case DRFLAC_METADATA_BLOCK_TYPE_PICTURE: + { + if (blockSize < 32) { + return DRFLAC_FALSE; + } + + if (onMeta) { + drflac_bool32 result = DRFLAC_TRUE; + drflac_uint32 blockSizeRemaining = blockSize; + char* pMime = NULL; + char* pDescription = NULL; + void* pPictureData = NULL; + + if (blockSizeRemaining < 4 || onRead(pUserData, &metadata.data.picture.type, 4) != 4) { + result = DRFLAC_FALSE; + goto done_flac; + } + blockSizeRemaining -= 4; + metadata.data.picture.type = drflac__be2host_32(metadata.data.picture.type); + + + if (blockSizeRemaining < 4 || onRead(pUserData, &metadata.data.picture.mimeLength, 4) != 4) { + result = DRFLAC_FALSE; + goto done_flac; + } + blockSizeRemaining -= 4; + metadata.data.picture.mimeLength = drflac__be2host_32(metadata.data.picture.mimeLength); + + if (blockSizeRemaining < metadata.data.picture.mimeLength) { + result = DRFLAC_FALSE; + goto done_flac; + } + + pMime = (char*)drflac__malloc_from_callbacks(metadata.data.picture.mimeLength + 1, pAllocationCallbacks); /* +1 for null terminator. */ + if (pMime == NULL) { + result = DRFLAC_FALSE; + goto done_flac; + } + + if (onRead(pUserData, pMime, metadata.data.picture.mimeLength) != metadata.data.picture.mimeLength) { + result = DRFLAC_FALSE; + goto done_flac; + } + blockSizeRemaining -= metadata.data.picture.mimeLength; + pMime[metadata.data.picture.mimeLength] = '\0'; /* Null terminate for safety. */ + metadata.data.picture.mime = (const char*)pMime; + + + if (blockSizeRemaining < 4 || onRead(pUserData, &metadata.data.picture.descriptionLength, 4) != 4) { + result = DRFLAC_FALSE; + goto done_flac; + } + blockSizeRemaining -= 4; + metadata.data.picture.descriptionLength = drflac__be2host_32(metadata.data.picture.descriptionLength); + + if (blockSizeRemaining < metadata.data.picture.descriptionLength) { + result = DRFLAC_FALSE; + goto done_flac; + } + + pDescription = (char*)drflac__malloc_from_callbacks(metadata.data.picture.descriptionLength + 1, pAllocationCallbacks); /* +1 for null terminator. */ + if (pDescription == NULL) { + result = DRFLAC_FALSE; + goto done_flac; + } + + if (onRead(pUserData, pDescription, metadata.data.picture.descriptionLength) != metadata.data.picture.descriptionLength) { + result = DRFLAC_FALSE; + goto done_flac; + } + blockSizeRemaining -= metadata.data.picture.descriptionLength; + pDescription[metadata.data.picture.descriptionLength] = '\0'; /* Null terminate for safety. */ + metadata.data.picture.description = (const char*)pDescription; + + + if (blockSizeRemaining < 4 || onRead(pUserData, &metadata.data.picture.width, 4) != 4) { + result = DRFLAC_FALSE; + goto done_flac; + } + blockSizeRemaining -= 4; + metadata.data.picture.width = drflac__be2host_32(metadata.data.picture.width); + + if (blockSizeRemaining < 4 || onRead(pUserData, &metadata.data.picture.height, 4) != 4) { + result = DRFLAC_FALSE; + goto done_flac; + } + blockSizeRemaining -= 4; + metadata.data.picture.height = drflac__be2host_32(metadata.data.picture.height); + + if (blockSizeRemaining < 4 || onRead(pUserData, &metadata.data.picture.colorDepth, 4) != 4) { + result = DRFLAC_FALSE; + goto done_flac; + } + blockSizeRemaining -= 4; + metadata.data.picture.colorDepth = drflac__be2host_32(metadata.data.picture.colorDepth); + + if (blockSizeRemaining < 4 || onRead(pUserData, &metadata.data.picture.indexColorCount, 4) != 4) { + result = DRFLAC_FALSE; + goto done_flac; + } + blockSizeRemaining -= 4; + metadata.data.picture.indexColorCount = drflac__be2host_32(metadata.data.picture.indexColorCount); + + + /* Picture data. */ + if (blockSizeRemaining < 4 || onRead(pUserData, &metadata.data.picture.pictureDataSize, 4) != 4) { + result = DRFLAC_FALSE; + goto done_flac; + } + blockSizeRemaining -= 4; + metadata.data.picture.pictureDataSize = drflac__be2host_32(metadata.data.picture.pictureDataSize); + + if (blockSizeRemaining < metadata.data.picture.pictureDataSize) { + result = DRFLAC_FALSE; + goto done_flac; + } + + /* For the actual image data we want to store the offset to the start of the stream. */ + metadata.data.picture.pictureDataOffset = runningFilePos + (blockSize - blockSizeRemaining); + + /* + For the allocation of image data, we can allow memory allocation to fail, in which case we just leave + the pointer as null. If it fails, we need to fall back to seeking past the image data. + */ + #ifndef DR_FLAC_NO_PICTURE_METADATA_MALLOC + pPictureData = drflac__malloc_from_callbacks(metadata.data.picture.pictureDataSize, pAllocationCallbacks); + if (pPictureData != NULL) { + if (onRead(pUserData, pPictureData, metadata.data.picture.pictureDataSize) != metadata.data.picture.pictureDataSize) { + result = DRFLAC_FALSE; + goto done_flac; + } + } else + #endif + { + /* Allocation failed. We need to seek past the picture data. */ + if (!onSeek(pUserData, metadata.data.picture.pictureDataSize, DRFLAC_SEEK_CUR)) { + result = DRFLAC_FALSE; + goto done_flac; + } + } + + blockSizeRemaining -= metadata.data.picture.pictureDataSize; + (void)blockSizeRemaining; + + metadata.data.picture.pPictureData = (const drflac_uint8*)pPictureData; + + + /* Only fire the callback if we actually have a way to read the image data. We must have either a valid offset, or a valid data pointer. */ + if (metadata.data.picture.pictureDataOffset != 0 || metadata.data.picture.pPictureData != NULL) { + onMeta(pUserDataMD, &metadata); + } else { + /* Don't have a valid offset or data pointer, so just pretend we don't have a picture metadata. */ + } + + done_flac: + drflac__free_from_callbacks(pMime, pAllocationCallbacks); + drflac__free_from_callbacks(pDescription, pAllocationCallbacks); + drflac__free_from_callbacks(pPictureData, pAllocationCallbacks); + + if (result != DRFLAC_TRUE) { + return DRFLAC_FALSE; + } + } + } break; + + case DRFLAC_METADATA_BLOCK_TYPE_PADDING: + { + if (onMeta) { + metadata.data.padding.unused = 0; + + /* Padding doesn't have anything meaningful in it, so just skip over it, but make sure the caller is aware of it by firing the callback. */ + if (!onSeek(pUserData, blockSize, DRFLAC_SEEK_CUR)) { + isLastBlock = DRFLAC_TRUE; /* An error occurred while seeking. Attempt to recover by treating this as the last block which will in turn terminate the loop. */ + } else { + onMeta(pUserDataMD, &metadata); + } + } + } break; + + case DRFLAC_METADATA_BLOCK_TYPE_INVALID: + { + /* Invalid chunk. Just skip over this one. */ + if (onMeta) { + if (!onSeek(pUserData, blockSize, DRFLAC_SEEK_CUR)) { + isLastBlock = DRFLAC_TRUE; /* An error occurred while seeking. Attempt to recover by treating this as the last block which will in turn terminate the loop. */ + } + } + } break; + + default: + { + /* + It's an unknown chunk, but not necessarily invalid. There's a chance more metadata blocks might be defined later on, so we + can at the very least report the chunk to the application and let it look at the raw data. + */ + if (onMeta) { + void* pRawData = drflac__malloc_from_callbacks(blockSize, pAllocationCallbacks); + if (pRawData != NULL) { + if (onRead(pUserData, pRawData, blockSize) != blockSize) { + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + return DRFLAC_FALSE; + } + } else { + /* Allocation failed. We need to seek past the block. */ + if (!onSeek(pUserData, blockSize, DRFLAC_SEEK_CUR)) { + return DRFLAC_FALSE; + } + } + + metadata.pRawData = pRawData; + metadata.rawDataSize = blockSize; + onMeta(pUserDataMD, &metadata); + + drflac__free_from_callbacks(pRawData, pAllocationCallbacks); + } + } break; + } + + /* If we're not handling metadata, just skip over the block. If we are, it will have been handled earlier in the switch statement above. */ + if (onMeta == NULL && blockSize > 0) { + if (!onSeek(pUserData, blockSize, DRFLAC_SEEK_CUR)) { + isLastBlock = DRFLAC_TRUE; + } + } + + runningFilePos += blockSize; + if (isLastBlock) { + break; + } + } + + *pSeektablePos = seektablePos; + *pSeekpointCount = seektableSize / DRFLAC_SEEKPOINT_SIZE_IN_BYTES; + *pFirstFramePos = runningFilePos; + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__init_private__native(drflac_init_info* pInit, drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, void* pUserData, void* pUserDataMD, drflac_bool32 relaxed) +{ + /* Pre Condition: The bit stream should be sitting just past the 4-byte id header. */ + + drflac_uint8 isLastBlock; + drflac_uint8 blockType; + drflac_uint32 blockSize; + + (void)onSeek; + + pInit->container = drflac_container_native; + + /* The first metadata block should be the STREAMINFO block. */ + if (!drflac__read_and_decode_block_header(onRead, pUserData, &isLastBlock, &blockType, &blockSize)) { + return DRFLAC_FALSE; + } + + if (blockType != DRFLAC_METADATA_BLOCK_TYPE_STREAMINFO || blockSize != 34) { + if (!relaxed) { + /* We're opening in strict mode and the first block is not the STREAMINFO block. Error. */ + return DRFLAC_FALSE; + } else { + /* + Relaxed mode. To open from here we need to just find the first frame and set the sample rate, etc. to whatever is defined + for that frame. + */ + pInit->hasStreamInfoBlock = DRFLAC_FALSE; + pInit->hasMetadataBlocks = DRFLAC_FALSE; + + if (!drflac__read_next_flac_frame_header(&pInit->bs, 0, &pInit->firstFrameHeader)) { + return DRFLAC_FALSE; /* Couldn't find a frame. */ + } + + if (pInit->firstFrameHeader.bitsPerSample == 0) { + return DRFLAC_FALSE; /* Failed to initialize because the first frame depends on the STREAMINFO block, which does not exist. */ + } + + pInit->sampleRate = pInit->firstFrameHeader.sampleRate; + pInit->channels = drflac__get_channel_count_from_channel_assignment(pInit->firstFrameHeader.channelAssignment); + pInit->bitsPerSample = pInit->firstFrameHeader.bitsPerSample; + pInit->maxBlockSizeInPCMFrames = 65535; /* <-- See notes here: https://xiph.org/flac/format.html#metadata_block_streaminfo */ + return DRFLAC_TRUE; + } + } else { + drflac_streaminfo streaminfo; + if (!drflac__read_streaminfo(onRead, pUserData, &streaminfo)) { + return DRFLAC_FALSE; + } + + pInit->hasStreamInfoBlock = DRFLAC_TRUE; + pInit->sampleRate = streaminfo.sampleRate; + pInit->channels = streaminfo.channels; + pInit->bitsPerSample = streaminfo.bitsPerSample; + pInit->totalPCMFrameCount = streaminfo.totalPCMFrameCount; + pInit->maxBlockSizeInPCMFrames = streaminfo.maxBlockSizeInPCMFrames; /* Don't care about the min block size - only the max (used for determining the size of the memory allocation). */ + pInit->hasMetadataBlocks = !isLastBlock; + + if (onMeta) { + drflac_metadata metadata; + metadata.type = DRFLAC_METADATA_BLOCK_TYPE_STREAMINFO; + metadata.pRawData = NULL; + metadata.rawDataSize = 0; + metadata.data.streaminfo = streaminfo; + onMeta(pUserDataMD, &metadata); + } + + return DRFLAC_TRUE; + } +} + +#ifndef DR_FLAC_NO_OGG +#define DRFLAC_OGG_MAX_PAGE_SIZE 65307 +#define DRFLAC_OGG_CAPTURE_PATTERN_CRC32 1605413199 /* CRC-32 of "OggS". */ + +typedef enum +{ + drflac_ogg_recover_on_crc_mismatch, + drflac_ogg_fail_on_crc_mismatch +} drflac_ogg_crc_mismatch_recovery; + +#ifndef DR_FLAC_NO_CRC +static drflac_uint32 drflac__crc32_table[] = { + 0x00000000L, 0x04C11DB7L, 0x09823B6EL, 0x0D4326D9L, + 0x130476DCL, 0x17C56B6BL, 0x1A864DB2L, 0x1E475005L, + 0x2608EDB8L, 0x22C9F00FL, 0x2F8AD6D6L, 0x2B4BCB61L, + 0x350C9B64L, 0x31CD86D3L, 0x3C8EA00AL, 0x384FBDBDL, + 0x4C11DB70L, 0x48D0C6C7L, 0x4593E01EL, 0x4152FDA9L, + 0x5F15ADACL, 0x5BD4B01BL, 0x569796C2L, 0x52568B75L, + 0x6A1936C8L, 0x6ED82B7FL, 0x639B0DA6L, 0x675A1011L, + 0x791D4014L, 0x7DDC5DA3L, 0x709F7B7AL, 0x745E66CDL, + 0x9823B6E0L, 0x9CE2AB57L, 0x91A18D8EL, 0x95609039L, + 0x8B27C03CL, 0x8FE6DD8BL, 0x82A5FB52L, 0x8664E6E5L, + 0xBE2B5B58L, 0xBAEA46EFL, 0xB7A96036L, 0xB3687D81L, + 0xAD2F2D84L, 0xA9EE3033L, 0xA4AD16EAL, 0xA06C0B5DL, + 0xD4326D90L, 0xD0F37027L, 0xDDB056FEL, 0xD9714B49L, + 0xC7361B4CL, 0xC3F706FBL, 0xCEB42022L, 0xCA753D95L, + 0xF23A8028L, 0xF6FB9D9FL, 0xFBB8BB46L, 0xFF79A6F1L, + 0xE13EF6F4L, 0xE5FFEB43L, 0xE8BCCD9AL, 0xEC7DD02DL, + 0x34867077L, 0x30476DC0L, 0x3D044B19L, 0x39C556AEL, + 0x278206ABL, 0x23431B1CL, 0x2E003DC5L, 0x2AC12072L, + 0x128E9DCFL, 0x164F8078L, 0x1B0CA6A1L, 0x1FCDBB16L, + 0x018AEB13L, 0x054BF6A4L, 0x0808D07DL, 0x0CC9CDCAL, + 0x7897AB07L, 0x7C56B6B0L, 0x71159069L, 0x75D48DDEL, + 0x6B93DDDBL, 0x6F52C06CL, 0x6211E6B5L, 0x66D0FB02L, + 0x5E9F46BFL, 0x5A5E5B08L, 0x571D7DD1L, 0x53DC6066L, + 0x4D9B3063L, 0x495A2DD4L, 0x44190B0DL, 0x40D816BAL, + 0xACA5C697L, 0xA864DB20L, 0xA527FDF9L, 0xA1E6E04EL, + 0xBFA1B04BL, 0xBB60ADFCL, 0xB6238B25L, 0xB2E29692L, + 0x8AAD2B2FL, 0x8E6C3698L, 0x832F1041L, 0x87EE0DF6L, + 0x99A95DF3L, 0x9D684044L, 0x902B669DL, 0x94EA7B2AL, + 0xE0B41DE7L, 0xE4750050L, 0xE9362689L, 0xEDF73B3EL, + 0xF3B06B3BL, 0xF771768CL, 0xFA325055L, 0xFEF34DE2L, + 0xC6BCF05FL, 0xC27DEDE8L, 0xCF3ECB31L, 0xCBFFD686L, + 0xD5B88683L, 0xD1799B34L, 0xDC3ABDEDL, 0xD8FBA05AL, + 0x690CE0EEL, 0x6DCDFD59L, 0x608EDB80L, 0x644FC637L, + 0x7A089632L, 0x7EC98B85L, 0x738AAD5CL, 0x774BB0EBL, + 0x4F040D56L, 0x4BC510E1L, 0x46863638L, 0x42472B8FL, + 0x5C007B8AL, 0x58C1663DL, 0x558240E4L, 0x51435D53L, + 0x251D3B9EL, 0x21DC2629L, 0x2C9F00F0L, 0x285E1D47L, + 0x36194D42L, 0x32D850F5L, 0x3F9B762CL, 0x3B5A6B9BL, + 0x0315D626L, 0x07D4CB91L, 0x0A97ED48L, 0x0E56F0FFL, + 0x1011A0FAL, 0x14D0BD4DL, 0x19939B94L, 0x1D528623L, + 0xF12F560EL, 0xF5EE4BB9L, 0xF8AD6D60L, 0xFC6C70D7L, + 0xE22B20D2L, 0xE6EA3D65L, 0xEBA91BBCL, 0xEF68060BL, + 0xD727BBB6L, 0xD3E6A601L, 0xDEA580D8L, 0xDA649D6FL, + 0xC423CD6AL, 0xC0E2D0DDL, 0xCDA1F604L, 0xC960EBB3L, + 0xBD3E8D7EL, 0xB9FF90C9L, 0xB4BCB610L, 0xB07DABA7L, + 0xAE3AFBA2L, 0xAAFBE615L, 0xA7B8C0CCL, 0xA379DD7BL, + 0x9B3660C6L, 0x9FF77D71L, 0x92B45BA8L, 0x9675461FL, + 0x8832161AL, 0x8CF30BADL, 0x81B02D74L, 0x857130C3L, + 0x5D8A9099L, 0x594B8D2EL, 0x5408ABF7L, 0x50C9B640L, + 0x4E8EE645L, 0x4A4FFBF2L, 0x470CDD2BL, 0x43CDC09CL, + 0x7B827D21L, 0x7F436096L, 0x7200464FL, 0x76C15BF8L, + 0x68860BFDL, 0x6C47164AL, 0x61043093L, 0x65C52D24L, + 0x119B4BE9L, 0x155A565EL, 0x18197087L, 0x1CD86D30L, + 0x029F3D35L, 0x065E2082L, 0x0B1D065BL, 0x0FDC1BECL, + 0x3793A651L, 0x3352BBE6L, 0x3E119D3FL, 0x3AD08088L, + 0x2497D08DL, 0x2056CD3AL, 0x2D15EBE3L, 0x29D4F654L, + 0xC5A92679L, 0xC1683BCEL, 0xCC2B1D17L, 0xC8EA00A0L, + 0xD6AD50A5L, 0xD26C4D12L, 0xDF2F6BCBL, 0xDBEE767CL, + 0xE3A1CBC1L, 0xE760D676L, 0xEA23F0AFL, 0xEEE2ED18L, + 0xF0A5BD1DL, 0xF464A0AAL, 0xF9278673L, 0xFDE69BC4L, + 0x89B8FD09L, 0x8D79E0BEL, 0x803AC667L, 0x84FBDBD0L, + 0x9ABC8BD5L, 0x9E7D9662L, 0x933EB0BBL, 0x97FFAD0CL, + 0xAFB010B1L, 0xAB710D06L, 0xA6322BDFL, 0xA2F33668L, + 0xBCB4666DL, 0xB8757BDAL, 0xB5365D03L, 0xB1F740B4L +}; +#endif + +static DRFLAC_INLINE drflac_uint32 drflac_crc32_byte(drflac_uint32 crc32, drflac_uint8 data) +{ +#ifndef DR_FLAC_NO_CRC + return (crc32 << 8) ^ drflac__crc32_table[(drflac_uint8)((crc32 >> 24) & 0xFF) ^ data]; +#else + (void)data; + return crc32; +#endif +} + +#if 0 +static DRFLAC_INLINE drflac_uint32 drflac_crc32_uint32(drflac_uint32 crc32, drflac_uint32 data) +{ + crc32 = drflac_crc32_byte(crc32, (drflac_uint8)((data >> 24) & 0xFF)); + crc32 = drflac_crc32_byte(crc32, (drflac_uint8)((data >> 16) & 0xFF)); + crc32 = drflac_crc32_byte(crc32, (drflac_uint8)((data >> 8) & 0xFF)); + crc32 = drflac_crc32_byte(crc32, (drflac_uint8)((data >> 0) & 0xFF)); + return crc32; +} + +static DRFLAC_INLINE drflac_uint32 drflac_crc32_uint64(drflac_uint32 crc32, drflac_uint64 data) +{ + crc32 = drflac_crc32_uint32(crc32, (drflac_uint32)((data >> 32) & 0xFFFFFFFF)); + crc32 = drflac_crc32_uint32(crc32, (drflac_uint32)((data >> 0) & 0xFFFFFFFF)); + return crc32; +} +#endif + +static DRFLAC_INLINE drflac_uint32 drflac_crc32_buffer(drflac_uint32 crc32, drflac_uint8* pData, drflac_uint32 dataSize) +{ + /* This can be optimized. */ + drflac_uint32 i; + for (i = 0; i < dataSize; ++i) { + crc32 = drflac_crc32_byte(crc32, pData[i]); + } + return crc32; +} + + +static DRFLAC_INLINE drflac_bool32 drflac_ogg__is_capture_pattern(drflac_uint8 pattern[4]) +{ + return pattern[0] == 'O' && pattern[1] == 'g' && pattern[2] == 'g' && pattern[3] == 'S'; +} + +static DRFLAC_INLINE drflac_uint32 drflac_ogg__get_page_header_size(drflac_ogg_page_header* pHeader) +{ + return 27 + pHeader->segmentCount; +} + +static DRFLAC_INLINE drflac_uint32 drflac_ogg__get_page_body_size(drflac_ogg_page_header* pHeader) +{ + drflac_uint32 pageBodySize = 0; + int i; + + for (i = 0; i < pHeader->segmentCount; ++i) { + pageBodySize += pHeader->segmentTable[i]; + } + + return pageBodySize; +} + +static drflac_result drflac_ogg__read_page_header_after_capture_pattern(drflac_read_proc onRead, void* pUserData, drflac_ogg_page_header* pHeader, drflac_uint32* pBytesRead, drflac_uint32* pCRC32) +{ + drflac_uint8 data[23]; + drflac_uint32 i; + + DRFLAC_ASSERT(*pCRC32 == DRFLAC_OGG_CAPTURE_PATTERN_CRC32); + + if (onRead(pUserData, data, 23) != 23) { + return DRFLAC_AT_END; + } + *pBytesRead += 23; + + /* + It's not actually used, but set the capture pattern to 'OggS' for completeness. Not doing this will cause static analysers to complain about + us trying to access uninitialized data. We could alternatively just comment out this member of the drflac_ogg_page_header structure, but I + like to have it map to the structure of the underlying data. + */ + pHeader->capturePattern[0] = 'O'; + pHeader->capturePattern[1] = 'g'; + pHeader->capturePattern[2] = 'g'; + pHeader->capturePattern[3] = 'S'; + + pHeader->structureVersion = data[0]; + pHeader->headerType = data[1]; + DRFLAC_COPY_MEMORY(&pHeader->granulePosition, &data[ 2], 8); + DRFLAC_COPY_MEMORY(&pHeader->serialNumber, &data[10], 4); + DRFLAC_COPY_MEMORY(&pHeader->sequenceNumber, &data[14], 4); + DRFLAC_COPY_MEMORY(&pHeader->checksum, &data[18], 4); + pHeader->segmentCount = data[22]; + + /* Calculate the CRC. Note that for the calculation the checksum part of the page needs to be set to 0. */ + data[18] = 0; + data[19] = 0; + data[20] = 0; + data[21] = 0; + + for (i = 0; i < 23; ++i) { + *pCRC32 = drflac_crc32_byte(*pCRC32, data[i]); + } + + + if (onRead(pUserData, pHeader->segmentTable, pHeader->segmentCount) != pHeader->segmentCount) { + return DRFLAC_AT_END; + } + *pBytesRead += pHeader->segmentCount; + + for (i = 0; i < pHeader->segmentCount; ++i) { + *pCRC32 = drflac_crc32_byte(*pCRC32, pHeader->segmentTable[i]); + } + + return DRFLAC_SUCCESS; +} + +static drflac_result drflac_ogg__read_page_header(drflac_read_proc onRead, void* pUserData, drflac_ogg_page_header* pHeader, drflac_uint32* pBytesRead, drflac_uint32* pCRC32) +{ + drflac_uint8 id[4]; + + *pBytesRead = 0; + + if (onRead(pUserData, id, 4) != 4) { + return DRFLAC_AT_END; + } + *pBytesRead += 4; + + /* We need to read byte-by-byte until we find the OggS capture pattern. */ + for (;;) { + if (drflac_ogg__is_capture_pattern(id)) { + drflac_result result; + + *pCRC32 = DRFLAC_OGG_CAPTURE_PATTERN_CRC32; + + result = drflac_ogg__read_page_header_after_capture_pattern(onRead, pUserData, pHeader, pBytesRead, pCRC32); + if (result == DRFLAC_SUCCESS) { + return DRFLAC_SUCCESS; + } else { + if (result == DRFLAC_CRC_MISMATCH) { + continue; + } else { + return result; + } + } + } else { + /* The first 4 bytes did not equal the capture pattern. Read the next byte and try again. */ + id[0] = id[1]; + id[1] = id[2]; + id[2] = id[3]; + if (onRead(pUserData, &id[3], 1) != 1) { + return DRFLAC_AT_END; + } + *pBytesRead += 1; + } + } +} + + +/* +The main part of the Ogg encapsulation is the conversion from the physical Ogg bitstream to the native FLAC bitstream. It works +in three general stages: Ogg Physical Bitstream -> Ogg/FLAC Logical Bitstream -> FLAC Native Bitstream. dr_flac is designed +in such a way that the core sections assume everything is delivered in native format. Therefore, for each encapsulation type +dr_flac is supporting there needs to be a layer sitting on top of the onRead and onSeek callbacks that ensures the bits read from +the physical Ogg bitstream are converted and delivered in native FLAC format. +*/ +typedef struct +{ + drflac_read_proc onRead; /* The original onRead callback from drflac_open() and family. */ + drflac_seek_proc onSeek; /* The original onSeek callback from drflac_open() and family. */ + drflac_tell_proc onTell; /* The original onTell callback from drflac_open() and family. */ + void* pUserData; /* The user data passed on onRead and onSeek. This is the user data that was passed on drflac_open() and family. */ + drflac_uint64 currentBytePos; /* The position of the byte we are sitting on in the physical byte stream. Used for efficient seeking. */ + drflac_uint64 firstBytePos; /* The position of the first byte in the physical bitstream. Points to the start of the "OggS" identifier of the FLAC bos page. */ + drflac_uint32 serialNumber; /* The serial number of the FLAC audio pages. This is determined by the initial header page that was read during initialization. */ + drflac_ogg_page_header bosPageHeader; /* Used for seeking. */ + drflac_ogg_page_header currentPageHeader; + drflac_uint32 bytesRemainingInPage; + drflac_uint32 pageDataSize; + drflac_uint8 pageData[DRFLAC_OGG_MAX_PAGE_SIZE]; +} drflac_oggbs; /* oggbs = Ogg Bitstream */ + +static size_t drflac_oggbs__read_physical(drflac_oggbs* oggbs, void* bufferOut, size_t bytesToRead) +{ + size_t bytesActuallyRead = oggbs->onRead(oggbs->pUserData, bufferOut, bytesToRead); + oggbs->currentBytePos += bytesActuallyRead; + + return bytesActuallyRead; +} + +static drflac_bool32 drflac_oggbs__seek_physical(drflac_oggbs* oggbs, drflac_uint64 offset, drflac_seek_origin origin) +{ + if (origin == DRFLAC_SEEK_SET) { + if (offset <= 0x7FFFFFFF) { + if (!oggbs->onSeek(oggbs->pUserData, (int)offset, DRFLAC_SEEK_SET)) { + return DRFLAC_FALSE; + } + oggbs->currentBytePos = offset; + + return DRFLAC_TRUE; + } else { + if (!oggbs->onSeek(oggbs->pUserData, 0x7FFFFFFF, DRFLAC_SEEK_SET)) { + return DRFLAC_FALSE; + } + oggbs->currentBytePos = offset; + + return drflac_oggbs__seek_physical(oggbs, offset - 0x7FFFFFFF, DRFLAC_SEEK_CUR); + } + } else { + while (offset > 0x7FFFFFFF) { + if (!oggbs->onSeek(oggbs->pUserData, 0x7FFFFFFF, DRFLAC_SEEK_CUR)) { + return DRFLAC_FALSE; + } + oggbs->currentBytePos += 0x7FFFFFFF; + offset -= 0x7FFFFFFF; + } + + if (!oggbs->onSeek(oggbs->pUserData, (int)offset, DRFLAC_SEEK_CUR)) { /* <-- Safe cast thanks to the loop above. */ + return DRFLAC_FALSE; + } + oggbs->currentBytePos += offset; + + return DRFLAC_TRUE; + } +} + +static drflac_bool32 drflac_oggbs__goto_next_page(drflac_oggbs* oggbs, drflac_ogg_crc_mismatch_recovery recoveryMethod) +{ + drflac_ogg_page_header header; + for (;;) { + drflac_uint32 crc32 = 0; + drflac_uint32 bytesRead; + drflac_uint32 pageBodySize; +#ifndef DR_FLAC_NO_CRC + drflac_uint32 actualCRC32; +#endif + + if (drflac_ogg__read_page_header(oggbs->onRead, oggbs->pUserData, &header, &bytesRead, &crc32) != DRFLAC_SUCCESS) { + return DRFLAC_FALSE; + } + oggbs->currentBytePos += bytesRead; + + pageBodySize = drflac_ogg__get_page_body_size(&header); + if (pageBodySize > DRFLAC_OGG_MAX_PAGE_SIZE) { + continue; /* Invalid page size. Assume it's corrupted and just move to the next page. */ + } + + if (header.serialNumber != oggbs->serialNumber) { + /* It's not a FLAC page. Skip it. */ + if (pageBodySize > 0 && !drflac_oggbs__seek_physical(oggbs, pageBodySize, DRFLAC_SEEK_CUR)) { + return DRFLAC_FALSE; + } + continue; + } + + + /* We need to read the entire page and then do a CRC check on it. If there's a CRC mismatch we need to skip this page. */ + if (drflac_oggbs__read_physical(oggbs, oggbs->pageData, pageBodySize) != pageBodySize) { + return DRFLAC_FALSE; + } + oggbs->pageDataSize = pageBodySize; + +#ifndef DR_FLAC_NO_CRC + actualCRC32 = drflac_crc32_buffer(crc32, oggbs->pageData, oggbs->pageDataSize); + if (actualCRC32 != header.checksum) { + if (recoveryMethod == drflac_ogg_recover_on_crc_mismatch) { + continue; /* CRC mismatch. Skip this page. */ + } else { + /* + Even though we are failing on a CRC mismatch, we still want our stream to be in a good state. Therefore we + go to the next valid page to ensure we're in a good state, but return false to let the caller know that the + seek did not fully complete. + */ + drflac_oggbs__goto_next_page(oggbs, drflac_ogg_recover_on_crc_mismatch); + return DRFLAC_FALSE; + } + } +#else + (void)recoveryMethod; /* <-- Silence a warning. */ +#endif + + oggbs->currentPageHeader = header; + oggbs->bytesRemainingInPage = pageBodySize; + return DRFLAC_TRUE; + } +} + +/* Function below is unused at the moment, but I might be re-adding it later. */ +#if 0 +static drflac_uint8 drflac_oggbs__get_current_segment_index(drflac_oggbs* oggbs, drflac_uint8* pBytesRemainingInSeg) +{ + drflac_uint32 bytesConsumedInPage = drflac_ogg__get_page_body_size(&oggbs->currentPageHeader) - oggbs->bytesRemainingInPage; + drflac_uint8 iSeg = 0; + drflac_uint32 iByte = 0; + while (iByte < bytesConsumedInPage) { + drflac_uint8 segmentSize = oggbs->currentPageHeader.segmentTable[iSeg]; + if (iByte + segmentSize > bytesConsumedInPage) { + break; + } else { + iSeg += 1; + iByte += segmentSize; + } + } + + *pBytesRemainingInSeg = oggbs->currentPageHeader.segmentTable[iSeg] - (drflac_uint8)(bytesConsumedInPage - iByte); + return iSeg; +} + +static drflac_bool32 drflac_oggbs__seek_to_next_packet(drflac_oggbs* oggbs) +{ + /* The current packet ends when we get to the segment with a lacing value of < 255 which is not at the end of a page. */ + for (;;) { + drflac_bool32 atEndOfPage = DRFLAC_FALSE; + + drflac_uint8 bytesRemainingInSeg; + drflac_uint8 iFirstSeg = drflac_oggbs__get_current_segment_index(oggbs, &bytesRemainingInSeg); + + drflac_uint32 bytesToEndOfPacketOrPage = bytesRemainingInSeg; + for (drflac_uint8 iSeg = iFirstSeg; iSeg < oggbs->currentPageHeader.segmentCount; ++iSeg) { + drflac_uint8 segmentSize = oggbs->currentPageHeader.segmentTable[iSeg]; + if (segmentSize < 255) { + if (iSeg == oggbs->currentPageHeader.segmentCount-1) { + atEndOfPage = DRFLAC_TRUE; + } + + break; + } + + bytesToEndOfPacketOrPage += segmentSize; + } + + /* + At this point we will have found either the packet or the end of the page. If were at the end of the page we'll + want to load the next page and keep searching for the end of the packet. + */ + drflac_oggbs__seek_physical(oggbs, bytesToEndOfPacketOrPage, DRFLAC_SEEK_CUR); + oggbs->bytesRemainingInPage -= bytesToEndOfPacketOrPage; + + if (atEndOfPage) { + /* + We're potentially at the next packet, but we need to check the next page first to be sure because the packet may + straddle pages. + */ + if (!drflac_oggbs__goto_next_page(oggbs)) { + return DRFLAC_FALSE; + } + + /* If it's a fresh packet it most likely means we're at the next packet. */ + if ((oggbs->currentPageHeader.headerType & 0x01) == 0) { + return DRFLAC_TRUE; + } + } else { + /* We're at the next packet. */ + return DRFLAC_TRUE; + } + } +} + +static drflac_bool32 drflac_oggbs__seek_to_next_frame(drflac_oggbs* oggbs) +{ + /* The bitstream should be sitting on the first byte just after the header of the frame. */ + + /* What we're actually doing here is seeking to the start of the next packet. */ + return drflac_oggbs__seek_to_next_packet(oggbs); +} +#endif + +static size_t drflac__on_read_ogg(void* pUserData, void* bufferOut, size_t bytesToRead) +{ + drflac_oggbs* oggbs = (drflac_oggbs*)pUserData; + drflac_uint8* pRunningBufferOut = (drflac_uint8*)bufferOut; + size_t bytesRead = 0; + + DRFLAC_ASSERT(oggbs != NULL); + DRFLAC_ASSERT(pRunningBufferOut != NULL); + + /* Reading is done page-by-page. If we've run out of bytes in the page we need to move to the next one. */ + while (bytesRead < bytesToRead) { + size_t bytesRemainingToRead = bytesToRead - bytesRead; + + if (oggbs->bytesRemainingInPage >= bytesRemainingToRead) { + DRFLAC_COPY_MEMORY(pRunningBufferOut, oggbs->pageData + (oggbs->pageDataSize - oggbs->bytesRemainingInPage), bytesRemainingToRead); + bytesRead += bytesRemainingToRead; + oggbs->bytesRemainingInPage -= (drflac_uint32)bytesRemainingToRead; + break; + } + + /* If we get here it means some of the requested data is contained in the next pages. */ + if (oggbs->bytesRemainingInPage > 0) { + DRFLAC_COPY_MEMORY(pRunningBufferOut, oggbs->pageData + (oggbs->pageDataSize - oggbs->bytesRemainingInPage), oggbs->bytesRemainingInPage); + bytesRead += oggbs->bytesRemainingInPage; + pRunningBufferOut += oggbs->bytesRemainingInPage; + oggbs->bytesRemainingInPage = 0; + } + + DRFLAC_ASSERT(bytesRemainingToRead > 0); + if (!drflac_oggbs__goto_next_page(oggbs, drflac_ogg_recover_on_crc_mismatch)) { + break; /* Failed to go to the next page. Might have simply hit the end of the stream. */ + } + } + + return bytesRead; +} + +static drflac_bool32 drflac__on_seek_ogg(void* pUserData, int offset, drflac_seek_origin origin) +{ + drflac_oggbs* oggbs = (drflac_oggbs*)pUserData; + int bytesSeeked = 0; + + DRFLAC_ASSERT(oggbs != NULL); + DRFLAC_ASSERT(offset >= 0); /* <-- Never seek backwards. */ + + /* Seeking is always forward which makes things a lot simpler. */ + if (origin == DRFLAC_SEEK_SET) { + if (!drflac_oggbs__seek_physical(oggbs, (int)oggbs->firstBytePos, DRFLAC_SEEK_SET)) { + return DRFLAC_FALSE; + } + + if (!drflac_oggbs__goto_next_page(oggbs, drflac_ogg_fail_on_crc_mismatch)) { + return DRFLAC_FALSE; + } + + return drflac__on_seek_ogg(pUserData, offset, DRFLAC_SEEK_CUR); + } else if (origin == DRFLAC_SEEK_CUR) { + while (bytesSeeked < offset) { + int bytesRemainingToSeek = offset - bytesSeeked; + DRFLAC_ASSERT(bytesRemainingToSeek >= 0); + + if (oggbs->bytesRemainingInPage >= (size_t)bytesRemainingToSeek) { + bytesSeeked += bytesRemainingToSeek; + (void)bytesSeeked; /* <-- Silence a dead store warning emitted by Clang Static Analyzer. */ + oggbs->bytesRemainingInPage -= bytesRemainingToSeek; + break; + } + + /* If we get here it means some of the requested data is contained in the next pages. */ + if (oggbs->bytesRemainingInPage > 0) { + bytesSeeked += (int)oggbs->bytesRemainingInPage; + oggbs->bytesRemainingInPage = 0; + } + + DRFLAC_ASSERT(bytesRemainingToSeek > 0); + if (!drflac_oggbs__goto_next_page(oggbs, drflac_ogg_fail_on_crc_mismatch)) { + /* Failed to go to the next page. We either hit the end of the stream or had a CRC mismatch. */ + return DRFLAC_FALSE; + } + } + } else if (origin == DRFLAC_SEEK_END) { + /* Seeking to the end is not supported. */ + return DRFLAC_FALSE; + } + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__on_tell_ogg(void* pUserData, drflac_int64* pCursor) +{ + /* + Not implemented for Ogg containers because we don't currently track the byte position of the logical bitstream. To support this, we'll need + to track the position in drflac__on_read_ogg and drflac__on_seek_ogg. + */ + (void)pUserData; + (void)pCursor; + return DRFLAC_FALSE; +} + + +static drflac_bool32 drflac_ogg__seek_to_pcm_frame(drflac* pFlac, drflac_uint64 pcmFrameIndex) +{ + drflac_oggbs* oggbs = (drflac_oggbs*)pFlac->_oggbs; + drflac_uint64 originalBytePos; + drflac_uint64 runningGranulePosition; + drflac_uint64 runningFrameBytePos; + drflac_uint64 runningPCMFrameCount; + + DRFLAC_ASSERT(oggbs != NULL); + + originalBytePos = oggbs->currentBytePos; /* For recovery. Points to the OggS identifier. */ + + /* First seek to the first frame. */ + if (!drflac__seek_to_byte(&pFlac->bs, pFlac->firstFLACFramePosInBytes)) { + return DRFLAC_FALSE; + } + oggbs->bytesRemainingInPage = 0; + + runningGranulePosition = 0; + for (;;) { + if (!drflac_oggbs__goto_next_page(oggbs, drflac_ogg_recover_on_crc_mismatch)) { + drflac_oggbs__seek_physical(oggbs, originalBytePos, DRFLAC_SEEK_SET); + return DRFLAC_FALSE; /* Never did find that sample... */ + } + + runningFrameBytePos = oggbs->currentBytePos - drflac_ogg__get_page_header_size(&oggbs->currentPageHeader) - oggbs->pageDataSize; + if (oggbs->currentPageHeader.granulePosition >= pcmFrameIndex) { + break; /* The sample is somewhere in the previous page. */ + } + + /* + At this point we know the sample is not in the previous page. It could possibly be in this page. For simplicity we + disregard any pages that do not begin a fresh packet. + */ + if ((oggbs->currentPageHeader.headerType & 0x01) == 0) { /* <-- Is it a fresh page? */ + if (oggbs->currentPageHeader.segmentTable[0] >= 2) { + drflac_uint8 firstBytesInPage[2]; + firstBytesInPage[0] = oggbs->pageData[0]; + firstBytesInPage[1] = oggbs->pageData[1]; + + if ((firstBytesInPage[0] == 0xFF) && (firstBytesInPage[1] & 0xFC) == 0xF8) { /* <-- Does the page begin with a frame's sync code? */ + runningGranulePosition = oggbs->currentPageHeader.granulePosition; + } + + continue; + } + } + } + + /* + We found the page that that is closest to the sample, so now we need to find it. The first thing to do is seek to the + start of that page. In the loop above we checked that it was a fresh page which means this page is also the start of + a new frame. This property means that after we've seeked to the page we can immediately start looping over frames until + we find the one containing the target sample. + */ + if (!drflac_oggbs__seek_physical(oggbs, runningFrameBytePos, DRFLAC_SEEK_SET)) { + return DRFLAC_FALSE; + } + if (!drflac_oggbs__goto_next_page(oggbs, drflac_ogg_recover_on_crc_mismatch)) { + return DRFLAC_FALSE; + } + + /* + At this point we'll be sitting on the first byte of the frame header of the first frame in the page. We just keep + looping over these frames until we find the one containing the sample we're after. + */ + runningPCMFrameCount = runningGranulePosition; + for (;;) { + /* + There are two ways to find the sample and seek past irrelevant frames: + 1) Use the native FLAC decoder. + 2) Use Ogg's framing system. + + Both of these options have their own pros and cons. Using the native FLAC decoder is slower because it needs to + do a full decode of the frame. Using Ogg's framing system is faster, but more complicated and involves some code + duplication for the decoding of frame headers. + + Another thing to consider is that using the Ogg framing system will perform direct seeking of the physical Ogg + bitstream. This is important to consider because it means we cannot read data from the drflac_bs object using the + standard drflac__*() APIs because that will read in extra data for its own internal caching which in turn breaks + the positioning of the read pointer of the physical Ogg bitstream. Therefore, anything that would normally be read + using the native FLAC decoding APIs, such as drflac__read_next_flac_frame_header(), need to be re-implemented so as to + avoid the use of the drflac_bs object. + + Considering these issues, I have decided to use the slower native FLAC decoding method for the following reasons: + 1) Seeking is already partially accelerated using Ogg's paging system in the code block above. + 2) Seeking in an Ogg encapsulated FLAC stream is probably quite uncommon. + 3) Simplicity. + */ + drflac_uint64 firstPCMFrameInFLACFrame = 0; + drflac_uint64 lastPCMFrameInFLACFrame = 0; + drflac_uint64 pcmFrameCountInThisFrame; + + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + return DRFLAC_FALSE; + } + + drflac__get_pcm_frame_range_of_current_flac_frame(pFlac, &firstPCMFrameInFLACFrame, &lastPCMFrameInFLACFrame); + + pcmFrameCountInThisFrame = (lastPCMFrameInFLACFrame - firstPCMFrameInFLACFrame) + 1; + + /* If we are seeking to the end of the file and we've just hit it, we're done. */ + if (pcmFrameIndex == pFlac->totalPCMFrameCount && (runningPCMFrameCount + pcmFrameCountInThisFrame) == pFlac->totalPCMFrameCount) { + drflac_result result = drflac__decode_flac_frame(pFlac); + if (result == DRFLAC_SUCCESS) { + pFlac->currentPCMFrame = pcmFrameIndex; + pFlac->currentFLACFrame.pcmFramesRemaining = 0; + return DRFLAC_TRUE; + } else { + return DRFLAC_FALSE; + } + } + + if (pcmFrameIndex < (runningPCMFrameCount + pcmFrameCountInThisFrame)) { + /* + The sample should be in this FLAC frame. We need to fully decode it, however if it's an invalid frame (a CRC mismatch), we need to pretend + it never existed and keep iterating. + */ + drflac_result result = drflac__decode_flac_frame(pFlac); + if (result == DRFLAC_SUCCESS) { + /* The frame is valid. We just need to skip over some samples to ensure it's sample-exact. */ + drflac_uint64 pcmFramesToDecode = (size_t)(pcmFrameIndex - runningPCMFrameCount); /* <-- Safe cast because the maximum number of samples in a frame is 65535. */ + if (pcmFramesToDecode == 0) { + return DRFLAC_TRUE; + } + + pFlac->currentPCMFrame = runningPCMFrameCount; + + return drflac__seek_forward_by_pcm_frames(pFlac, pcmFramesToDecode) == pcmFramesToDecode; /* <-- If this fails, something bad has happened (it should never fail). */ + } else { + if (result == DRFLAC_CRC_MISMATCH) { + continue; /* CRC mismatch. Pretend this frame never existed. */ + } else { + return DRFLAC_FALSE; + } + } + } else { + /* + It's not in this frame. We need to seek past the frame, but check if there was a CRC mismatch. If so, we pretend this + frame never existed and leave the running sample count untouched. + */ + drflac_result result = drflac__seek_to_next_flac_frame(pFlac); + if (result == DRFLAC_SUCCESS) { + runningPCMFrameCount += pcmFrameCountInThisFrame; + } else { + if (result == DRFLAC_CRC_MISMATCH) { + continue; /* CRC mismatch. Pretend this frame never existed. */ + } else { + return DRFLAC_FALSE; + } + } + } + } +} + + + +static drflac_bool32 drflac__init_private__ogg(drflac_init_info* pInit, drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_meta_proc onMeta, void* pUserData, void* pUserDataMD, drflac_bool32 relaxed) +{ + drflac_ogg_page_header header; + drflac_uint32 crc32 = DRFLAC_OGG_CAPTURE_PATTERN_CRC32; + drflac_uint32 bytesRead = 0; + + /* Pre Condition: The bit stream should be sitting just past the 4-byte OggS capture pattern. */ + (void)relaxed; + + pInit->container = drflac_container_ogg; + pInit->oggFirstBytePos = 0; + + /* + We'll get here if the first 4 bytes of the stream were the OggS capture pattern, however it doesn't necessarily mean the + stream includes FLAC encoded audio. To check for this we need to scan the beginning-of-stream page markers and check if + any match the FLAC specification. Important to keep in mind that the stream may be multiplexed. + */ + if (drflac_ogg__read_page_header_after_capture_pattern(onRead, pUserData, &header, &bytesRead, &crc32) != DRFLAC_SUCCESS) { + return DRFLAC_FALSE; + } + pInit->runningFilePos += bytesRead; + + for (;;) { + int pageBodySize; + + /* Break if we're past the beginning of stream page. */ + if ((header.headerType & 0x02) == 0) { + return DRFLAC_FALSE; + } + + /* Check if it's a FLAC header. */ + pageBodySize = drflac_ogg__get_page_body_size(&header); + if (pageBodySize == 51) { /* 51 = the lacing value of the FLAC header packet. */ + /* It could be a FLAC page... */ + drflac_uint32 bytesRemainingInPage = pageBodySize; + drflac_uint8 packetType; + + if (onRead(pUserData, &packetType, 1) != 1) { + return DRFLAC_FALSE; + } + + bytesRemainingInPage -= 1; + if (packetType == 0x7F) { + /* Increasingly more likely to be a FLAC page... */ + drflac_uint8 sig[4]; + if (onRead(pUserData, sig, 4) != 4) { + return DRFLAC_FALSE; + } + + bytesRemainingInPage -= 4; + if (sig[0] == 'F' && sig[1] == 'L' && sig[2] == 'A' && sig[3] == 'C') { + /* Almost certainly a FLAC page... */ + drflac_uint8 mappingVersion[2]; + if (onRead(pUserData, mappingVersion, 2) != 2) { + return DRFLAC_FALSE; + } + + if (mappingVersion[0] != 1) { + return DRFLAC_FALSE; /* Only supporting version 1.x of the Ogg mapping. */ + } + + /* + The next 2 bytes are the non-audio packets, not including this one. We don't care about this because we're going to + be handling it in a generic way based on the serial number and packet types. + */ + if (!onSeek(pUserData, 2, DRFLAC_SEEK_CUR)) { + return DRFLAC_FALSE; + } + + /* Expecting the native FLAC signature "fLaC". */ + if (onRead(pUserData, sig, 4) != 4) { + return DRFLAC_FALSE; + } + + if (sig[0] == 'f' && sig[1] == 'L' && sig[2] == 'a' && sig[3] == 'C') { + /* The remaining data in the page should be the STREAMINFO block. */ + drflac_streaminfo streaminfo; + drflac_uint8 isLastBlock; + drflac_uint8 blockType; + drflac_uint32 blockSize; + if (!drflac__read_and_decode_block_header(onRead, pUserData, &isLastBlock, &blockType, &blockSize)) { + return DRFLAC_FALSE; + } + + if (blockType != DRFLAC_METADATA_BLOCK_TYPE_STREAMINFO || blockSize != 34) { + return DRFLAC_FALSE; /* Invalid block type. First block must be the STREAMINFO block. */ + } + + if (drflac__read_streaminfo(onRead, pUserData, &streaminfo)) { + /* Success! */ + pInit->hasStreamInfoBlock = DRFLAC_TRUE; + pInit->sampleRate = streaminfo.sampleRate; + pInit->channels = streaminfo.channels; + pInit->bitsPerSample = streaminfo.bitsPerSample; + pInit->totalPCMFrameCount = streaminfo.totalPCMFrameCount; + pInit->maxBlockSizeInPCMFrames = streaminfo.maxBlockSizeInPCMFrames; + pInit->hasMetadataBlocks = !isLastBlock; + + if (onMeta) { + drflac_metadata metadata; + metadata.type = DRFLAC_METADATA_BLOCK_TYPE_STREAMINFO; + metadata.pRawData = NULL; + metadata.rawDataSize = 0; + metadata.data.streaminfo = streaminfo; + onMeta(pUserDataMD, &metadata); + } + + pInit->runningFilePos += pageBodySize; + pInit->oggFirstBytePos = pInit->runningFilePos - 79; /* Subtracting 79 will place us right on top of the "OggS" identifier of the FLAC bos page. */ + pInit->oggSerial = header.serialNumber; + pInit->oggBosHeader = header; + break; + } else { + /* Failed to read STREAMINFO block. Aww, so close... */ + return DRFLAC_FALSE; + } + } else { + /* Invalid file. */ + return DRFLAC_FALSE; + } + } else { + /* Not a FLAC header. Skip it. */ + if (!onSeek(pUserData, bytesRemainingInPage, DRFLAC_SEEK_CUR)) { + return DRFLAC_FALSE; + } + } + } else { + /* Not a FLAC header. Seek past the entire page and move on to the next. */ + if (!onSeek(pUserData, bytesRemainingInPage, DRFLAC_SEEK_CUR)) { + return DRFLAC_FALSE; + } + } + } else { + if (!onSeek(pUserData, pageBodySize, DRFLAC_SEEK_CUR)) { + return DRFLAC_FALSE; + } + } + + pInit->runningFilePos += pageBodySize; + + + /* Read the header of the next page. */ + if (drflac_ogg__read_page_header(onRead, pUserData, &header, &bytesRead, &crc32) != DRFLAC_SUCCESS) { + return DRFLAC_FALSE; + } + pInit->runningFilePos += bytesRead; + } + + /* + If we get here it means we found a FLAC audio stream. We should be sitting on the first byte of the header of the next page. The next + packets in the FLAC logical stream contain the metadata. The only thing left to do in the initialization phase for Ogg is to create the + Ogg bistream object. + */ + pInit->hasMetadataBlocks = DRFLAC_TRUE; /* <-- Always have at least VORBIS_COMMENT metadata block. */ + return DRFLAC_TRUE; +} +#endif + +static drflac_bool32 drflac__init_private(drflac_init_info* pInit, drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_tell_proc onTell, drflac_meta_proc onMeta, drflac_container container, void* pUserData, void* pUserDataMD) +{ + drflac_bool32 relaxed; + drflac_uint8 id[4]; + + if (pInit == NULL || onRead == NULL || onSeek == NULL) { /* <-- onTell is optional. */ + return DRFLAC_FALSE; + } + + DRFLAC_ZERO_MEMORY(pInit, sizeof(*pInit)); + pInit->onRead = onRead; + pInit->onSeek = onSeek; + pInit->onTell = onTell; + pInit->onMeta = onMeta; + pInit->container = container; + pInit->pUserData = pUserData; + pInit->pUserDataMD = pUserDataMD; + + pInit->bs.onRead = onRead; + pInit->bs.onSeek = onSeek; + pInit->bs.onTell = onTell; + pInit->bs.pUserData = pUserData; + drflac__reset_cache(&pInit->bs); + + + /* If the container is explicitly defined then we can try opening in relaxed mode. */ + relaxed = container != drflac_container_unknown; + + /* Skip over any ID3 tags. */ + for (;;) { + if (onRead(pUserData, id, 4) != 4) { + return DRFLAC_FALSE; /* Ran out of data. */ + } + pInit->runningFilePos += 4; + + if (id[0] == 'I' && id[1] == 'D' && id[2] == '3') { + drflac_uint8 header[6]; + drflac_uint8 flags; + drflac_uint32 headerSize; + + if (onRead(pUserData, header, 6) != 6) { + return DRFLAC_FALSE; /* Ran out of data. */ + } + pInit->runningFilePos += 6; + + flags = header[1]; + + DRFLAC_COPY_MEMORY(&headerSize, header+2, 4); + headerSize = drflac__unsynchsafe_32(drflac__be2host_32(headerSize)); + if (flags & 0x10) { + headerSize += 10; + } + + if (!onSeek(pUserData, headerSize, DRFLAC_SEEK_CUR)) { + return DRFLAC_FALSE; /* Failed to seek past the tag. */ + } + pInit->runningFilePos += headerSize; + } else { + break; + } + } + + if (id[0] == 'f' && id[1] == 'L' && id[2] == 'a' && id[3] == 'C') { + return drflac__init_private__native(pInit, onRead, onSeek, onMeta, pUserData, pUserDataMD, relaxed); + } +#ifndef DR_FLAC_NO_OGG + if (id[0] == 'O' && id[1] == 'g' && id[2] == 'g' && id[3] == 'S') { + return drflac__init_private__ogg(pInit, onRead, onSeek, onMeta, pUserData, pUserDataMD, relaxed); + } +#endif + + /* If we get here it means we likely don't have a header. Try opening in relaxed mode, if applicable. */ + if (relaxed) { + if (container == drflac_container_native) { + return drflac__init_private__native(pInit, onRead, onSeek, onMeta, pUserData, pUserDataMD, relaxed); + } +#ifndef DR_FLAC_NO_OGG + if (container == drflac_container_ogg) { + return drflac__init_private__ogg(pInit, onRead, onSeek, onMeta, pUserData, pUserDataMD, relaxed); + } +#endif + } + + /* Unsupported container. */ + return DRFLAC_FALSE; +} + +static void drflac__init_from_info(drflac* pFlac, const drflac_init_info* pInit) +{ + DRFLAC_ASSERT(pFlac != NULL); + DRFLAC_ASSERT(pInit != NULL); + + DRFLAC_ZERO_MEMORY(pFlac, sizeof(*pFlac)); + pFlac->bs = pInit->bs; + pFlac->onMeta = pInit->onMeta; + pFlac->pUserDataMD = pInit->pUserDataMD; + pFlac->maxBlockSizeInPCMFrames = pInit->maxBlockSizeInPCMFrames; + pFlac->sampleRate = pInit->sampleRate; + pFlac->channels = (drflac_uint8)pInit->channels; + pFlac->bitsPerSample = (drflac_uint8)pInit->bitsPerSample; + pFlac->totalPCMFrameCount = pInit->totalPCMFrameCount; + pFlac->container = pInit->container; +} + + +static drflac* drflac_open_with_metadata_private(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_tell_proc onTell, drflac_meta_proc onMeta, drflac_container container, void* pUserData, void* pUserDataMD, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac_init_info init; + drflac_uint32 allocationSize; + drflac_uint32 wholeSIMDVectorCountPerChannel; + drflac_uint32 decodedSamplesAllocationSize; +#ifndef DR_FLAC_NO_OGG + drflac_oggbs* pOggbs = NULL; +#endif + drflac_uint64 firstFramePos; + drflac_uint64 seektablePos; + drflac_uint32 seekpointCount; + drflac_allocation_callbacks allocationCallbacks; + drflac* pFlac; + + /* CPU support first. */ + drflac__init_cpu_caps(); + + if (!drflac__init_private(&init, onRead, onSeek, onTell, onMeta, container, pUserData, pUserDataMD)) { + return NULL; + } + + if (pAllocationCallbacks != NULL) { + allocationCallbacks = *pAllocationCallbacks; + if (allocationCallbacks.onFree == NULL || (allocationCallbacks.onMalloc == NULL && allocationCallbacks.onRealloc == NULL)) { + return NULL; /* Invalid allocation callbacks. */ + } + } else { + allocationCallbacks.pUserData = NULL; + allocationCallbacks.onMalloc = drflac__malloc_default; + allocationCallbacks.onRealloc = drflac__realloc_default; + allocationCallbacks.onFree = drflac__free_default; + } + + + /* + The size of the allocation for the drflac object needs to be large enough to fit the following: + 1) The main members of the drflac structure + 2) A block of memory large enough to store the decoded samples of the largest frame in the stream + 3) If the container is Ogg, a drflac_oggbs object + + The complicated part of the allocation is making sure there's enough room the decoded samples, taking into consideration + the different SIMD instruction sets. + */ + allocationSize = sizeof(drflac); + + /* + The allocation size for decoded frames depends on the number of 32-bit integers that fit inside the largest SIMD vector + we are supporting. + */ + if ((init.maxBlockSizeInPCMFrames % (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32))) == 0) { + wholeSIMDVectorCountPerChannel = (init.maxBlockSizeInPCMFrames / (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32))); + } else { + wholeSIMDVectorCountPerChannel = (init.maxBlockSizeInPCMFrames / (DRFLAC_MAX_SIMD_VECTOR_SIZE / sizeof(drflac_int32))) + 1; + } + + decodedSamplesAllocationSize = wholeSIMDVectorCountPerChannel * DRFLAC_MAX_SIMD_VECTOR_SIZE * init.channels; + + allocationSize += decodedSamplesAllocationSize; + allocationSize += DRFLAC_MAX_SIMD_VECTOR_SIZE; /* Allocate extra bytes to ensure we have enough for alignment. */ + +#ifndef DR_FLAC_NO_OGG + /* There's additional data required for Ogg streams. */ + if (init.container == drflac_container_ogg) { + allocationSize += sizeof(drflac_oggbs); + + pOggbs = (drflac_oggbs*)drflac__malloc_from_callbacks(sizeof(*pOggbs), &allocationCallbacks); + if (pOggbs == NULL) { + return NULL; /*DRFLAC_OUT_OF_MEMORY;*/ + } + + DRFLAC_ZERO_MEMORY(pOggbs, sizeof(*pOggbs)); + pOggbs->onRead = onRead; + pOggbs->onSeek = onSeek; + pOggbs->onTell = onTell; + pOggbs->pUserData = pUserData; + pOggbs->currentBytePos = init.oggFirstBytePos; + pOggbs->firstBytePos = init.oggFirstBytePos; + pOggbs->serialNumber = init.oggSerial; + pOggbs->bosPageHeader = init.oggBosHeader; + pOggbs->bytesRemainingInPage = 0; + } +#endif + + /* + This part is a bit awkward. We need to load the seektable so that it can be referenced in-memory, but I want the drflac object to + consist of only a single heap allocation. To this, the size of the seek table needs to be known, which we determine when reading + and decoding the metadata. + */ + firstFramePos = 42; /* <-- We know we are at byte 42 at this point. */ + seektablePos = 0; + seekpointCount = 0; + if (init.hasMetadataBlocks) { + drflac_read_proc onReadOverride = onRead; + drflac_seek_proc onSeekOverride = onSeek; + drflac_tell_proc onTellOverride = onTell; + void* pUserDataOverride = pUserData; + +#ifndef DR_FLAC_NO_OGG + if (init.container == drflac_container_ogg) { + onReadOverride = drflac__on_read_ogg; + onSeekOverride = drflac__on_seek_ogg; + onTellOverride = drflac__on_tell_ogg; + pUserDataOverride = (void*)pOggbs; + } +#endif + + if (!drflac__read_and_decode_metadata(onReadOverride, onSeekOverride, onTellOverride, onMeta, pUserDataOverride, pUserDataMD, &firstFramePos, &seektablePos, &seekpointCount, &allocationCallbacks)) { + #ifndef DR_FLAC_NO_OGG + drflac__free_from_callbacks(pOggbs, &allocationCallbacks); + #endif + return NULL; + } + + if ((0xFFFFFFFF - (seekpointCount * sizeof(drflac_seekpoint))) < allocationSize) { + #ifndef DR_FLAC_NO_OGG + drflac__free_from_callbacks(pOggbs, &allocationCallbacks); + #endif + return NULL; + } + + allocationSize += seekpointCount * sizeof(drflac_seekpoint); + } + + pFlac = (drflac*)drflac__malloc_from_callbacks((size_t)allocationSize, &allocationCallbacks); + if (pFlac == NULL) { + #ifndef DR_FLAC_NO_OGG + drflac__free_from_callbacks(pOggbs, &allocationCallbacks); + #endif + return NULL; + } + + drflac__init_from_info(pFlac, &init); + pFlac->allocationCallbacks = allocationCallbacks; + pFlac->pDecodedSamples = (drflac_int32*)drflac_align((size_t)pFlac->pExtraData, DRFLAC_MAX_SIMD_VECTOR_SIZE); + +#ifndef DR_FLAC_NO_OGG + if (init.container == drflac_container_ogg) { + drflac_oggbs* pInternalOggbs = (drflac_oggbs*)((drflac_uint8*)pFlac->pDecodedSamples + decodedSamplesAllocationSize + (seekpointCount * sizeof(drflac_seekpoint))); + DRFLAC_COPY_MEMORY(pInternalOggbs, pOggbs, sizeof(*pOggbs)); + + /* At this point the pOggbs object has been handed over to pInternalOggbs and can be freed. */ + drflac__free_from_callbacks(pOggbs, &allocationCallbacks); + pOggbs = NULL; + + /* The Ogg bistream needs to be layered on top of the original bitstream. */ + pFlac->bs.onRead = drflac__on_read_ogg; + pFlac->bs.onSeek = drflac__on_seek_ogg; + pFlac->bs.onTell = drflac__on_tell_ogg; + pFlac->bs.pUserData = (void*)pInternalOggbs; + pFlac->_oggbs = (void*)pInternalOggbs; + } +#endif + + pFlac->firstFLACFramePosInBytes = firstFramePos; + + /* NOTE: Seektables are not currently compatible with Ogg encapsulation (Ogg has its own accelerated seeking system). I may change this later, so I'm leaving this here for now. */ +#ifndef DR_FLAC_NO_OGG + if (init.container == drflac_container_ogg) + { + pFlac->pSeekpoints = NULL; + pFlac->seekpointCount = 0; + } + else +#endif + { + /* If we have a seektable we need to load it now, making sure we move back to where we were previously. */ + if (seektablePos != 0) { + pFlac->seekpointCount = seekpointCount; + pFlac->pSeekpoints = (drflac_seekpoint*)((drflac_uint8*)pFlac->pDecodedSamples + decodedSamplesAllocationSize); + + DRFLAC_ASSERT(pFlac->bs.onSeek != NULL); + DRFLAC_ASSERT(pFlac->bs.onRead != NULL); + + /* Seek to the seektable, then just read directly into our seektable buffer. */ + if (pFlac->bs.onSeek(pFlac->bs.pUserData, (int)seektablePos, DRFLAC_SEEK_SET)) { + drflac_uint32 iSeekpoint; + + for (iSeekpoint = 0; iSeekpoint < seekpointCount; iSeekpoint += 1) { + if (pFlac->bs.onRead(pFlac->bs.pUserData, pFlac->pSeekpoints + iSeekpoint, DRFLAC_SEEKPOINT_SIZE_IN_BYTES) == DRFLAC_SEEKPOINT_SIZE_IN_BYTES) { + /* Endian swap. */ + pFlac->pSeekpoints[iSeekpoint].firstPCMFrame = drflac__be2host_64(pFlac->pSeekpoints[iSeekpoint].firstPCMFrame); + pFlac->pSeekpoints[iSeekpoint].flacFrameOffset = drflac__be2host_64(pFlac->pSeekpoints[iSeekpoint].flacFrameOffset); + pFlac->pSeekpoints[iSeekpoint].pcmFrameCount = drflac__be2host_16(pFlac->pSeekpoints[iSeekpoint].pcmFrameCount); + } else { + /* Failed to read the seektable. Pretend we don't have one. */ + pFlac->pSeekpoints = NULL; + pFlac->seekpointCount = 0; + break; + } + } + + /* We need to seek back to where we were. If this fails it's a critical error. */ + if (!pFlac->bs.onSeek(pFlac->bs.pUserData, (int)pFlac->firstFLACFramePosInBytes, DRFLAC_SEEK_SET)) { + drflac__free_from_callbacks(pFlac, &allocationCallbacks); + return NULL; + } + } else { + /* Failed to seek to the seektable. Ominous sign, but for now we can just pretend we don't have one. */ + pFlac->pSeekpoints = NULL; + pFlac->seekpointCount = 0; + } + } + } + + + /* + If we get here, but don't have a STREAMINFO block, it means we've opened the stream in relaxed mode and need to decode + the first frame. + */ + if (!init.hasStreamInfoBlock) { + pFlac->currentFLACFrame.header = init.firstFrameHeader; + for (;;) { + drflac_result result = drflac__decode_flac_frame(pFlac); + if (result == DRFLAC_SUCCESS) { + break; + } else { + if (result == DRFLAC_CRC_MISMATCH) { + if (!drflac__read_next_flac_frame_header(&pFlac->bs, pFlac->bitsPerSample, &pFlac->currentFLACFrame.header)) { + drflac__free_from_callbacks(pFlac, &allocationCallbacks); + return NULL; + } + continue; + } else { + drflac__free_from_callbacks(pFlac, &allocationCallbacks); + return NULL; + } + } + } + } + + return pFlac; +} + + + +#ifndef DR_FLAC_NO_STDIO +#include +#ifndef DR_FLAC_NO_WCHAR +#include /* For wcslen(), wcsrtombs() */ +#endif + +/* Errno */ +/* drflac_result_from_errno() is only used for fopen() and wfopen() so putting it inside DR_WAV_NO_STDIO for now. If something else needs this later we can move it out. */ +#include +static drflac_result drflac_result_from_errno(int e) +{ + switch (e) + { + case 0: return DRFLAC_SUCCESS; + #ifdef EPERM + case EPERM: return DRFLAC_INVALID_OPERATION; + #endif + #ifdef ENOENT + case ENOENT: return DRFLAC_DOES_NOT_EXIST; + #endif + #ifdef ESRCH + case ESRCH: return DRFLAC_DOES_NOT_EXIST; + #endif + #ifdef EINTR + case EINTR: return DRFLAC_INTERRUPT; + #endif + #ifdef EIO + case EIO: return DRFLAC_IO_ERROR; + #endif + #ifdef ENXIO + case ENXIO: return DRFLAC_DOES_NOT_EXIST; + #endif + #ifdef E2BIG + case E2BIG: return DRFLAC_INVALID_ARGS; + #endif + #ifdef ENOEXEC + case ENOEXEC: return DRFLAC_INVALID_FILE; + #endif + #ifdef EBADF + case EBADF: return DRFLAC_INVALID_FILE; + #endif + #ifdef ECHILD + case ECHILD: return DRFLAC_ERROR; + #endif + #ifdef EAGAIN + case EAGAIN: return DRFLAC_UNAVAILABLE; + #endif + #ifdef ENOMEM + case ENOMEM: return DRFLAC_OUT_OF_MEMORY; + #endif + #ifdef EACCES + case EACCES: return DRFLAC_ACCESS_DENIED; + #endif + #ifdef EFAULT + case EFAULT: return DRFLAC_BAD_ADDRESS; + #endif + #ifdef ENOTBLK + case ENOTBLK: return DRFLAC_ERROR; + #endif + #ifdef EBUSY + case EBUSY: return DRFLAC_BUSY; + #endif + #ifdef EEXIST + case EEXIST: return DRFLAC_ALREADY_EXISTS; + #endif + #ifdef EXDEV + case EXDEV: return DRFLAC_ERROR; + #endif + #ifdef ENODEV + case ENODEV: return DRFLAC_DOES_NOT_EXIST; + #endif + #ifdef ENOTDIR + case ENOTDIR: return DRFLAC_NOT_DIRECTORY; + #endif + #ifdef EISDIR + case EISDIR: return DRFLAC_IS_DIRECTORY; + #endif + #ifdef EINVAL + case EINVAL: return DRFLAC_INVALID_ARGS; + #endif + #ifdef ENFILE + case ENFILE: return DRFLAC_TOO_MANY_OPEN_FILES; + #endif + #ifdef EMFILE + case EMFILE: return DRFLAC_TOO_MANY_OPEN_FILES; + #endif + #ifdef ENOTTY + case ENOTTY: return DRFLAC_INVALID_OPERATION; + #endif + #ifdef ETXTBSY + case ETXTBSY: return DRFLAC_BUSY; + #endif + #ifdef EFBIG + case EFBIG: return DRFLAC_TOO_BIG; + #endif + #ifdef ENOSPC + case ENOSPC: return DRFLAC_NO_SPACE; + #endif + #ifdef ESPIPE + case ESPIPE: return DRFLAC_BAD_SEEK; + #endif + #ifdef EROFS + case EROFS: return DRFLAC_ACCESS_DENIED; + #endif + #ifdef EMLINK + case EMLINK: return DRFLAC_TOO_MANY_LINKS; + #endif + #ifdef EPIPE + case EPIPE: return DRFLAC_BAD_PIPE; + #endif + #ifdef EDOM + case EDOM: return DRFLAC_OUT_OF_RANGE; + #endif + #ifdef ERANGE + case ERANGE: return DRFLAC_OUT_OF_RANGE; + #endif + #ifdef EDEADLK + case EDEADLK: return DRFLAC_DEADLOCK; + #endif + #ifdef ENAMETOOLONG + case ENAMETOOLONG: return DRFLAC_PATH_TOO_LONG; + #endif + #ifdef ENOLCK + case ENOLCK: return DRFLAC_ERROR; + #endif + #ifdef ENOSYS + case ENOSYS: return DRFLAC_NOT_IMPLEMENTED; + #endif + #if defined(ENOTEMPTY) && ENOTEMPTY != EEXIST /* In AIX, ENOTEMPTY and EEXIST use the same value. */ + case ENOTEMPTY: return DRFLAC_DIRECTORY_NOT_EMPTY; + #endif + #ifdef ELOOP + case ELOOP: return DRFLAC_TOO_MANY_LINKS; + #endif + #ifdef ENOMSG + case ENOMSG: return DRFLAC_NO_MESSAGE; + #endif + #ifdef EIDRM + case EIDRM: return DRFLAC_ERROR; + #endif + #ifdef ECHRNG + case ECHRNG: return DRFLAC_ERROR; + #endif + #ifdef EL2NSYNC + case EL2NSYNC: return DRFLAC_ERROR; + #endif + #ifdef EL3HLT + case EL3HLT: return DRFLAC_ERROR; + #endif + #ifdef EL3RST + case EL3RST: return DRFLAC_ERROR; + #endif + #ifdef ELNRNG + case ELNRNG: return DRFLAC_OUT_OF_RANGE; + #endif + #ifdef EUNATCH + case EUNATCH: return DRFLAC_ERROR; + #endif + #ifdef ENOCSI + case ENOCSI: return DRFLAC_ERROR; + #endif + #ifdef EL2HLT + case EL2HLT: return DRFLAC_ERROR; + #endif + #ifdef EBADE + case EBADE: return DRFLAC_ERROR; + #endif + #ifdef EBADR + case EBADR: return DRFLAC_ERROR; + #endif + #ifdef EXFULL + case EXFULL: return DRFLAC_ERROR; + #endif + #ifdef ENOANO + case ENOANO: return DRFLAC_ERROR; + #endif + #ifdef EBADRQC + case EBADRQC: return DRFLAC_ERROR; + #endif + #ifdef EBADSLT + case EBADSLT: return DRFLAC_ERROR; + #endif + #ifdef EBFONT + case EBFONT: return DRFLAC_INVALID_FILE; + #endif + #ifdef ENOSTR + case ENOSTR: return DRFLAC_ERROR; + #endif + #ifdef ENODATA + case ENODATA: return DRFLAC_NO_DATA_AVAILABLE; + #endif + #ifdef ETIME + case ETIME: return DRFLAC_TIMEOUT; + #endif + #ifdef ENOSR + case ENOSR: return DRFLAC_NO_DATA_AVAILABLE; + #endif + #ifdef ENONET + case ENONET: return DRFLAC_NO_NETWORK; + #endif + #ifdef ENOPKG + case ENOPKG: return DRFLAC_ERROR; + #endif + #ifdef EREMOTE + case EREMOTE: return DRFLAC_ERROR; + #endif + #ifdef ENOLINK + case ENOLINK: return DRFLAC_ERROR; + #endif + #ifdef EADV + case EADV: return DRFLAC_ERROR; + #endif + #ifdef ESRMNT + case ESRMNT: return DRFLAC_ERROR; + #endif + #ifdef ECOMM + case ECOMM: return DRFLAC_ERROR; + #endif + #ifdef EPROTO + case EPROTO: return DRFLAC_ERROR; + #endif + #ifdef EMULTIHOP + case EMULTIHOP: return DRFLAC_ERROR; + #endif + #ifdef EDOTDOT + case EDOTDOT: return DRFLAC_ERROR; + #endif + #ifdef EBADMSG + case EBADMSG: return DRFLAC_BAD_MESSAGE; + #endif + #ifdef EOVERFLOW + case EOVERFLOW: return DRFLAC_TOO_BIG; + #endif + #ifdef ENOTUNIQ + case ENOTUNIQ: return DRFLAC_NOT_UNIQUE; + #endif + #ifdef EBADFD + case EBADFD: return DRFLAC_ERROR; + #endif + #ifdef EREMCHG + case EREMCHG: return DRFLAC_ERROR; + #endif + #ifdef ELIBACC + case ELIBACC: return DRFLAC_ACCESS_DENIED; + #endif + #ifdef ELIBBAD + case ELIBBAD: return DRFLAC_INVALID_FILE; + #endif + #ifdef ELIBSCN + case ELIBSCN: return DRFLAC_INVALID_FILE; + #endif + #ifdef ELIBMAX + case ELIBMAX: return DRFLAC_ERROR; + #endif + #ifdef ELIBEXEC + case ELIBEXEC: return DRFLAC_ERROR; + #endif + #ifdef EILSEQ + case EILSEQ: return DRFLAC_INVALID_DATA; + #endif + #ifdef ERESTART + case ERESTART: return DRFLAC_ERROR; + #endif + #ifdef ESTRPIPE + case ESTRPIPE: return DRFLAC_ERROR; + #endif + #ifdef EUSERS + case EUSERS: return DRFLAC_ERROR; + #endif + #ifdef ENOTSOCK + case ENOTSOCK: return DRFLAC_NOT_SOCKET; + #endif + #ifdef EDESTADDRREQ + case EDESTADDRREQ: return DRFLAC_NO_ADDRESS; + #endif + #ifdef EMSGSIZE + case EMSGSIZE: return DRFLAC_TOO_BIG; + #endif + #ifdef EPROTOTYPE + case EPROTOTYPE: return DRFLAC_BAD_PROTOCOL; + #endif + #ifdef ENOPROTOOPT + case ENOPROTOOPT: return DRFLAC_PROTOCOL_UNAVAILABLE; + #endif + #ifdef EPROTONOSUPPORT + case EPROTONOSUPPORT: return DRFLAC_PROTOCOL_NOT_SUPPORTED; + #endif + #ifdef ESOCKTNOSUPPORT + case ESOCKTNOSUPPORT: return DRFLAC_SOCKET_NOT_SUPPORTED; + #endif + #ifdef EOPNOTSUPP + case EOPNOTSUPP: return DRFLAC_INVALID_OPERATION; + #endif + #ifdef EPFNOSUPPORT + case EPFNOSUPPORT: return DRFLAC_PROTOCOL_FAMILY_NOT_SUPPORTED; + #endif + #ifdef EAFNOSUPPORT + case EAFNOSUPPORT: return DRFLAC_ADDRESS_FAMILY_NOT_SUPPORTED; + #endif + #ifdef EADDRINUSE + case EADDRINUSE: return DRFLAC_ALREADY_IN_USE; + #endif + #ifdef EADDRNOTAVAIL + case EADDRNOTAVAIL: return DRFLAC_ERROR; + #endif + #ifdef ENETDOWN + case ENETDOWN: return DRFLAC_NO_NETWORK; + #endif + #ifdef ENETUNREACH + case ENETUNREACH: return DRFLAC_NO_NETWORK; + #endif + #ifdef ENETRESET + case ENETRESET: return DRFLAC_NO_NETWORK; + #endif + #ifdef ECONNABORTED + case ECONNABORTED: return DRFLAC_NO_NETWORK; + #endif + #ifdef ECONNRESET + case ECONNRESET: return DRFLAC_CONNECTION_RESET; + #endif + #ifdef ENOBUFS + case ENOBUFS: return DRFLAC_NO_SPACE; + #endif + #ifdef EISCONN + case EISCONN: return DRFLAC_ALREADY_CONNECTED; + #endif + #ifdef ENOTCONN + case ENOTCONN: return DRFLAC_NOT_CONNECTED; + #endif + #ifdef ESHUTDOWN + case ESHUTDOWN: return DRFLAC_ERROR; + #endif + #ifdef ETOOMANYREFS + case ETOOMANYREFS: return DRFLAC_ERROR; + #endif + #ifdef ETIMEDOUT + case ETIMEDOUT: return DRFLAC_TIMEOUT; + #endif + #ifdef ECONNREFUSED + case ECONNREFUSED: return DRFLAC_CONNECTION_REFUSED; + #endif + #ifdef EHOSTDOWN + case EHOSTDOWN: return DRFLAC_NO_HOST; + #endif + #ifdef EHOSTUNREACH + case EHOSTUNREACH: return DRFLAC_NO_HOST; + #endif + #ifdef EALREADY + case EALREADY: return DRFLAC_IN_PROGRESS; + #endif + #ifdef EINPROGRESS + case EINPROGRESS: return DRFLAC_IN_PROGRESS; + #endif + #ifdef ESTALE + case ESTALE: return DRFLAC_INVALID_FILE; + #endif + #ifdef EUCLEAN + case EUCLEAN: return DRFLAC_ERROR; + #endif + #ifdef ENOTNAM + case ENOTNAM: return DRFLAC_ERROR; + #endif + #ifdef ENAVAIL + case ENAVAIL: return DRFLAC_ERROR; + #endif + #ifdef EISNAM + case EISNAM: return DRFLAC_ERROR; + #endif + #ifdef EREMOTEIO + case EREMOTEIO: return DRFLAC_IO_ERROR; + #endif + #ifdef EDQUOT + case EDQUOT: return DRFLAC_NO_SPACE; + #endif + #ifdef ENOMEDIUM + case ENOMEDIUM: return DRFLAC_DOES_NOT_EXIST; + #endif + #ifdef EMEDIUMTYPE + case EMEDIUMTYPE: return DRFLAC_ERROR; + #endif + #ifdef ECANCELED + case ECANCELED: return DRFLAC_CANCELLED; + #endif + #ifdef ENOKEY + case ENOKEY: return DRFLAC_ERROR; + #endif + #ifdef EKEYEXPIRED + case EKEYEXPIRED: return DRFLAC_ERROR; + #endif + #ifdef EKEYREVOKED + case EKEYREVOKED: return DRFLAC_ERROR; + #endif + #ifdef EKEYREJECTED + case EKEYREJECTED: return DRFLAC_ERROR; + #endif + #ifdef EOWNERDEAD + case EOWNERDEAD: return DRFLAC_ERROR; + #endif + #ifdef ENOTRECOVERABLE + case ENOTRECOVERABLE: return DRFLAC_ERROR; + #endif + #ifdef ERFKILL + case ERFKILL: return DRFLAC_ERROR; + #endif + #ifdef EHWPOISON + case EHWPOISON: return DRFLAC_ERROR; + #endif + default: return DRFLAC_ERROR; + } +} +/* End Errno */ + +/* fopen */ +static drflac_result drflac_fopen(FILE** ppFile, const char* pFilePath, const char* pOpenMode) +{ +#if defined(_MSC_VER) && _MSC_VER >= 1400 + errno_t err; +#endif + + if (ppFile != NULL) { + *ppFile = NULL; /* Safety. */ + } + + if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) { + return DRFLAC_INVALID_ARGS; + } + +#if defined(_MSC_VER) && _MSC_VER >= 1400 + err = fopen_s(ppFile, pFilePath, pOpenMode); + if (err != 0) { + return drflac_result_from_errno(err); + } +#else +#if defined(_WIN32) || defined(__APPLE__) + *ppFile = fopen(pFilePath, pOpenMode); +#else + #if defined(_FILE_OFFSET_BITS) && _FILE_OFFSET_BITS == 64 && defined(_LARGEFILE64_SOURCE) + *ppFile = fopen64(pFilePath, pOpenMode); + #else + *ppFile = fopen(pFilePath, pOpenMode); + #endif +#endif + if (*ppFile == NULL) { + drflac_result result = drflac_result_from_errno(errno); + if (result == DRFLAC_SUCCESS) { + result = DRFLAC_ERROR; /* Just a safety check to make sure we never ever return success when pFile == NULL. */ + } + + return result; + } +#endif + + return DRFLAC_SUCCESS; +} + +/* +_wfopen() isn't always available in all compilation environments. + + * Windows only. + * MSVC seems to support it universally as far back as VC6 from what I can tell (haven't checked further back). + * MinGW-64 (both 32- and 64-bit) seems to support it. + * MinGW wraps it in !defined(__STRICT_ANSI__). + * OpenWatcom wraps it in !defined(_NO_EXT_KEYS). + +This can be reviewed as compatibility issues arise. The preference is to use _wfopen_s() and _wfopen() as opposed to the wcsrtombs() +fallback, so if you notice your compiler not detecting this properly I'm happy to look at adding support. +*/ +#if defined(_WIN32) + #if defined(_MSC_VER) || defined(__MINGW64__) || (!defined(__STRICT_ANSI__) && !defined(_NO_EXT_KEYS)) + #define DRFLAC_HAS_WFOPEN + #endif +#endif + +#ifndef DR_FLAC_NO_WCHAR +static drflac_result drflac_wfopen(FILE** ppFile, const wchar_t* pFilePath, const wchar_t* pOpenMode, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + if (ppFile != NULL) { + *ppFile = NULL; /* Safety. */ + } + + if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) { + return DRFLAC_INVALID_ARGS; + } + +#if defined(DRFLAC_HAS_WFOPEN) + { + /* Use _wfopen() on Windows. */ + #if defined(_MSC_VER) && _MSC_VER >= 1400 + errno_t err = _wfopen_s(ppFile, pFilePath, pOpenMode); + if (err != 0) { + return drflac_result_from_errno(err); + } + #else + *ppFile = _wfopen(pFilePath, pOpenMode); + if (*ppFile == NULL) { + return drflac_result_from_errno(errno); + } + #endif + (void)pAllocationCallbacks; + } +#else + /* + Use fopen() on anything other than Windows. Requires a conversion. This is annoying because + fopen() is locale specific. The only real way I can think of to do this is with wcsrtombs(). Note + that wcstombs() is apparently not thread-safe because it uses a static global mbstate_t object for + maintaining state. I've checked this with -std=c89 and it works, but if somebody get's a compiler + error I'll look into improving compatibility. + */ + + /* + Some compilers don't support wchar_t or wcsrtombs() which we're using below. In this case we just + need to abort with an error. If you encounter a compiler lacking such support, add it to this list + and submit a bug report and it'll be added to the library upstream. + */ + #if defined(__DJGPP__) + { + /* Nothing to do here. This will fall through to the error check below. */ + } + #else + { + mbstate_t mbs; + size_t lenMB; + const wchar_t* pFilePathTemp = pFilePath; + char* pFilePathMB = NULL; + char pOpenModeMB[32] = {0}; + + /* Get the length first. */ + DRFLAC_ZERO_OBJECT(&mbs); + lenMB = wcsrtombs(NULL, &pFilePathTemp, 0, &mbs); + if (lenMB == (size_t)-1) { + return drflac_result_from_errno(errno); + } + + pFilePathMB = (char*)drflac__malloc_from_callbacks(lenMB + 1, pAllocationCallbacks); + if (pFilePathMB == NULL) { + return DRFLAC_OUT_OF_MEMORY; + } + + pFilePathTemp = pFilePath; + DRFLAC_ZERO_OBJECT(&mbs); + wcsrtombs(pFilePathMB, &pFilePathTemp, lenMB + 1, &mbs); + + /* The open mode should always consist of ASCII characters so we should be able to do a trivial conversion. */ + { + size_t i = 0; + for (;;) { + if (pOpenMode[i] == 0) { + pOpenModeMB[i] = '\0'; + break; + } + + pOpenModeMB[i] = (char)pOpenMode[i]; + i += 1; + } + } + + *ppFile = fopen(pFilePathMB, pOpenModeMB); + + drflac__free_from_callbacks(pFilePathMB, pAllocationCallbacks); + } + #endif + + if (*ppFile == NULL) { + return DRFLAC_ERROR; + } +#endif + + return DRFLAC_SUCCESS; +} +#endif +/* End fopen */ + +static size_t drflac__on_read_stdio(void* pUserData, void* bufferOut, size_t bytesToRead) +{ + return fread(bufferOut, 1, bytesToRead, (FILE*)pUserData); +} + +static drflac_bool32 drflac__on_seek_stdio(void* pUserData, int offset, drflac_seek_origin origin) +{ + int whence = SEEK_SET; + if (origin == DRFLAC_SEEK_CUR) { + whence = SEEK_CUR; + } else if (origin == DRFLAC_SEEK_END) { + whence = SEEK_END; + } + + return fseek((FILE*)pUserData, offset, whence) == 0; +} + +static drflac_bool32 drflac__on_tell_stdio(void* pUserData, drflac_int64* pCursor) +{ + FILE* pFileStdio = (FILE*)pUserData; + drflac_int64 result; + + /* These were all validated at a higher level. */ + DRFLAC_ASSERT(pFileStdio != NULL); + DRFLAC_ASSERT(pCursor != NULL); + +#if defined(_WIN32) && !defined(NXDK) + #if defined(_MSC_VER) && _MSC_VER > 1200 + result = _ftelli64(pFileStdio); + #else + result = ftell(pFileStdio); + #endif +#else + result = ftell(pFileStdio); +#endif + + *pCursor = result; + + return DRFLAC_TRUE; +} + + + +DRFLAC_API drflac* drflac_open_file(const char* pFileName, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + FILE* pFile; + + if (drflac_fopen(&pFile, pFileName, "rb") != DRFLAC_SUCCESS) { + return NULL; + } + + pFlac = drflac_open(drflac__on_read_stdio, drflac__on_seek_stdio, drflac__on_tell_stdio, (void*)pFile, pAllocationCallbacks); + if (pFlac == NULL) { + fclose(pFile); + return NULL; + } + + return pFlac; +} + +#ifndef DR_FLAC_NO_WCHAR +DRFLAC_API drflac* drflac_open_file_w(const wchar_t* pFileName, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + FILE* pFile; + + if (drflac_wfopen(&pFile, pFileName, L"rb", pAllocationCallbacks) != DRFLAC_SUCCESS) { + return NULL; + } + + pFlac = drflac_open(drflac__on_read_stdio, drflac__on_seek_stdio, drflac__on_tell_stdio, (void*)pFile, pAllocationCallbacks); + if (pFlac == NULL) { + fclose(pFile); + return NULL; + } + + return pFlac; +} +#endif + +DRFLAC_API drflac* drflac_open_file_with_metadata(const char* pFileName, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + FILE* pFile; + + if (drflac_fopen(&pFile, pFileName, "rb") != DRFLAC_SUCCESS) { + return NULL; + } + + pFlac = drflac_open_with_metadata_private(drflac__on_read_stdio, drflac__on_seek_stdio, drflac__on_tell_stdio, onMeta, drflac_container_unknown, (void*)pFile, pUserData, pAllocationCallbacks); + if (pFlac == NULL) { + fclose(pFile); + return pFlac; + } + + return pFlac; +} + +#ifndef DR_FLAC_NO_WCHAR +DRFLAC_API drflac* drflac_open_file_with_metadata_w(const wchar_t* pFileName, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + FILE* pFile; + + if (drflac_wfopen(&pFile, pFileName, L"rb", pAllocationCallbacks) != DRFLAC_SUCCESS) { + return NULL; + } + + pFlac = drflac_open_with_metadata_private(drflac__on_read_stdio, drflac__on_seek_stdio, drflac__on_tell_stdio, onMeta, drflac_container_unknown, (void*)pFile, pUserData, pAllocationCallbacks); + if (pFlac == NULL) { + fclose(pFile); + return pFlac; + } + + return pFlac; +} +#endif +#endif /* DR_FLAC_NO_STDIO */ + +static size_t drflac__on_read_memory(void* pUserData, void* bufferOut, size_t bytesToRead) +{ + drflac__memory_stream* memoryStream = (drflac__memory_stream*)pUserData; + size_t bytesRemaining; + + DRFLAC_ASSERT(memoryStream != NULL); + DRFLAC_ASSERT(memoryStream->dataSize >= memoryStream->currentReadPos); + + bytesRemaining = memoryStream->dataSize - memoryStream->currentReadPos; + if (bytesToRead > bytesRemaining) { + bytesToRead = bytesRemaining; + } + + if (bytesToRead > 0) { + DRFLAC_COPY_MEMORY(bufferOut, memoryStream->data + memoryStream->currentReadPos, bytesToRead); + memoryStream->currentReadPos += bytesToRead; + } + + return bytesToRead; +} + +static drflac_bool32 drflac__on_seek_memory(void* pUserData, int offset, drflac_seek_origin origin) +{ + drflac__memory_stream* memoryStream = (drflac__memory_stream*)pUserData; + drflac_int64 newCursor; + + DRFLAC_ASSERT(memoryStream != NULL); + + if (origin == DRFLAC_SEEK_SET) { + newCursor = 0; + } else if (origin == DRFLAC_SEEK_CUR) { + newCursor = (drflac_int64)memoryStream->currentReadPos; + } else if (origin == DRFLAC_SEEK_END) { + newCursor = (drflac_int64)memoryStream->dataSize; + } else { + DRFLAC_ASSERT(!"Invalid seek origin"); + return DRFLAC_FALSE; + } + + newCursor += offset; + + if (newCursor < 0) { + return DRFLAC_FALSE; /* Trying to seek prior to the start of the buffer. */ + } + if ((size_t)newCursor > memoryStream->dataSize) { + return DRFLAC_FALSE; /* Trying to seek beyond the end of the buffer. */ + } + + memoryStream->currentReadPos = (size_t)newCursor; + + return DRFLAC_TRUE; +} + +static drflac_bool32 drflac__on_tell_memory(void* pUserData, drflac_int64* pCursor) +{ + drflac__memory_stream* memoryStream = (drflac__memory_stream*)pUserData; + + DRFLAC_ASSERT(memoryStream != NULL); + DRFLAC_ASSERT(pCursor != NULL); + + *pCursor = (drflac_int64)memoryStream->currentReadPos; + return DRFLAC_TRUE; +} + +DRFLAC_API drflac* drflac_open_memory(const void* pData, size_t dataSize, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac__memory_stream memoryStream; + drflac* pFlac; + + memoryStream.data = (const drflac_uint8*)pData; + memoryStream.dataSize = dataSize; + memoryStream.currentReadPos = 0; + pFlac = drflac_open(drflac__on_read_memory, drflac__on_seek_memory, drflac__on_tell_memory, &memoryStream, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + pFlac->memoryStream = memoryStream; + + /* This is an awful hack... */ +#ifndef DR_FLAC_NO_OGG + if (pFlac->container == drflac_container_ogg) + { + drflac_oggbs* oggbs = (drflac_oggbs*)pFlac->_oggbs; + oggbs->pUserData = &pFlac->memoryStream; + } + else +#endif + { + pFlac->bs.pUserData = &pFlac->memoryStream; + } + + return pFlac; +} + +DRFLAC_API drflac* drflac_open_memory_with_metadata(const void* pData, size_t dataSize, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac__memory_stream memoryStream; + drflac* pFlac; + + memoryStream.data = (const drflac_uint8*)pData; + memoryStream.dataSize = dataSize; + memoryStream.currentReadPos = 0; + pFlac = drflac_open_with_metadata_private(drflac__on_read_memory, drflac__on_seek_memory, drflac__on_tell_memory, onMeta, drflac_container_unknown, &memoryStream, pUserData, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + pFlac->memoryStream = memoryStream; + + /* This is an awful hack... */ +#ifndef DR_FLAC_NO_OGG + if (pFlac->container == drflac_container_ogg) + { + drflac_oggbs* oggbs = (drflac_oggbs*)pFlac->_oggbs; + oggbs->pUserData = &pFlac->memoryStream; + } + else +#endif + { + pFlac->bs.pUserData = &pFlac->memoryStream; + } + + return pFlac; +} + + + +DRFLAC_API drflac* drflac_open(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_tell_proc onTell, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + return drflac_open_with_metadata_private(onRead, onSeek, onTell, NULL, drflac_container_unknown, pUserData, pUserData, pAllocationCallbacks); +} +DRFLAC_API drflac* drflac_open_relaxed(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_tell_proc onTell, drflac_container container, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + return drflac_open_with_metadata_private(onRead, onSeek, onTell, NULL, container, pUserData, pUserData, pAllocationCallbacks); +} + +DRFLAC_API drflac* drflac_open_with_metadata(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_tell_proc onTell, drflac_meta_proc onMeta, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + return drflac_open_with_metadata_private(onRead, onSeek, onTell, onMeta, drflac_container_unknown, pUserData, pUserData, pAllocationCallbacks); +} +DRFLAC_API drflac* drflac_open_with_metadata_relaxed(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_tell_proc onTell, drflac_meta_proc onMeta, drflac_container container, void* pUserData, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + return drflac_open_with_metadata_private(onRead, onSeek, onTell, onMeta, container, pUserData, pUserData, pAllocationCallbacks); +} + +DRFLAC_API void drflac_close(drflac* pFlac) +{ + if (pFlac == NULL) { + return; + } + +#ifndef DR_FLAC_NO_STDIO + /* + If we opened the file with drflac_open_file() we will want to close the file handle. We can know whether or not drflac_open_file() + was used by looking at the callbacks. + */ + if (pFlac->bs.onRead == drflac__on_read_stdio) { + fclose((FILE*)pFlac->bs.pUserData); + } + +#ifndef DR_FLAC_NO_OGG + /* Need to clean up Ogg streams a bit differently due to the way the bit streaming is chained. */ + if (pFlac->container == drflac_container_ogg) { + drflac_oggbs* oggbs = (drflac_oggbs*)pFlac->_oggbs; + DRFLAC_ASSERT(pFlac->bs.onRead == drflac__on_read_ogg); + + if (oggbs->onRead == drflac__on_read_stdio) { + fclose((FILE*)oggbs->pUserData); + } + } +#endif +#endif + + drflac__free_from_callbacks(pFlac, &pFlac->allocationCallbacks); +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_left_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + for (i = 0; i < frameCount; ++i) { + drflac_uint32 left = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + drflac_uint32 side = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + drflac_uint32 right = left - side; + + pOutputSamples[i*2+0] = (drflac_int32)left; + pOutputSamples[i*2+1] = (drflac_int32)right; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_left_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 left0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 left1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 left2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 left3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << shift1; + + drflac_uint32 right0 = left0 - side0; + drflac_uint32 right1 = left1 - side1; + drflac_uint32 right2 = left2 - side2; + drflac_uint32 right3 = left3 - side3; + + pOutputSamples[i*8+0] = (drflac_int32)left0; + pOutputSamples[i*8+1] = (drflac_int32)right0; + pOutputSamples[i*8+2] = (drflac_int32)left1; + pOutputSamples[i*8+3] = (drflac_int32)right1; + pOutputSamples[i*8+4] = (drflac_int32)left2; + pOutputSamples[i*8+5] = (drflac_int32)right2; + pOutputSamples[i*8+6] = (drflac_int32)left3; + pOutputSamples[i*8+7] = (drflac_int32)right3; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + pOutputSamples[i*2+0] = (drflac_int32)left; + pOutputSamples[i*2+1] = (drflac_int32)right; + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_left_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + for (i = 0; i < frameCount4; ++i) { + __m128i left = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + __m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + __m128i right = _mm_sub_epi32(left, side); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 0), _mm_unpacklo_epi32(left, right)); + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 4), _mm_unpackhi_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + pOutputSamples[i*2+0] = (drflac_int32)left; + pOutputSamples[i*2+1] = (drflac_int32)right; + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_left_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + int32x4_t shift0_4; + int32x4_t shift1_4; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + shift0_4 = vdupq_n_s32(shift0); + shift1_4 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + uint32x4_t left; + uint32x4_t side; + uint32x4_t right; + + left = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4); + side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4); + right = vsubq_u32(left, side); + + drflac__vst2q_u32((drflac_uint32*)pOutputSamples + i*8, vzipq_u32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + pOutputSamples[i*2+0] = (drflac_int32)left; + pOutputSamples[i*2+1] = (drflac_int32)right; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_left_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s32__decode_left_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s32__decode_left_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_s32__decode_left_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_s32__decode_left_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_right_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + for (i = 0; i < frameCount; ++i) { + drflac_uint32 side = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + drflac_uint32 right = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + drflac_uint32 left = right + side; + + pOutputSamples[i*2+0] = (drflac_int32)left; + pOutputSamples[i*2+1] = (drflac_int32)right; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_right_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 side0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 side1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 side2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 side3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 right0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 right1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 right2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 right3 = pInputSamples1U32[i*4+3] << shift1; + + drflac_uint32 left0 = right0 + side0; + drflac_uint32 left1 = right1 + side1; + drflac_uint32 left2 = right2 + side2; + drflac_uint32 left3 = right3 + side3; + + pOutputSamples[i*8+0] = (drflac_int32)left0; + pOutputSamples[i*8+1] = (drflac_int32)right0; + pOutputSamples[i*8+2] = (drflac_int32)left1; + pOutputSamples[i*8+3] = (drflac_int32)right1; + pOutputSamples[i*8+4] = (drflac_int32)left2; + pOutputSamples[i*8+5] = (drflac_int32)right2; + pOutputSamples[i*8+6] = (drflac_int32)left3; + pOutputSamples[i*8+7] = (drflac_int32)right3; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + pOutputSamples[i*2+0] = (drflac_int32)left; + pOutputSamples[i*2+1] = (drflac_int32)right; + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_right_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + for (i = 0; i < frameCount4; ++i) { + __m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + __m128i right = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + __m128i left = _mm_add_epi32(right, side); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 0), _mm_unpacklo_epi32(left, right)); + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 4), _mm_unpackhi_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + pOutputSamples[i*2+0] = (drflac_int32)left; + pOutputSamples[i*2+1] = (drflac_int32)right; + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_right_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + int32x4_t shift0_4; + int32x4_t shift1_4; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + shift0_4 = vdupq_n_s32(shift0); + shift1_4 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + uint32x4_t side; + uint32x4_t right; + uint32x4_t left; + + side = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4); + right = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4); + left = vaddq_u32(right, side); + + drflac__vst2q_u32((drflac_uint32*)pOutputSamples + i*8, vzipq_u32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + pOutputSamples[i*2+0] = (drflac_int32)left; + pOutputSamples[i*2+1] = (drflac_int32)right; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_right_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s32__decode_right_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s32__decode_right_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_s32__decode_right_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_s32__decode_right_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_mid_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + for (drflac_uint64 i = 0; i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid + side) >> 1) << unusedBitsPerSample); + pOutputSamples[i*2+1] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid - side) >> 1) << unusedBitsPerSample); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_mid_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_int32 shift = unusedBitsPerSample; + + if (shift > 0) { + shift -= 1; + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 temp0L; + drflac_uint32 temp1L; + drflac_uint32 temp2L; + drflac_uint32 temp3L; + drflac_uint32 temp0R; + drflac_uint32 temp1R; + drflac_uint32 temp2R; + drflac_uint32 temp3R; + + drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid0 = (mid0 << 1) | (side0 & 0x01); + mid1 = (mid1 << 1) | (side1 & 0x01); + mid2 = (mid2 << 1) | (side2 & 0x01); + mid3 = (mid3 << 1) | (side3 & 0x01); + + temp0L = (mid0 + side0) << shift; + temp1L = (mid1 + side1) << shift; + temp2L = (mid2 + side2) << shift; + temp3L = (mid3 + side3) << shift; + + temp0R = (mid0 - side0) << shift; + temp1R = (mid1 - side1) << shift; + temp2R = (mid2 - side2) << shift; + temp3R = (mid3 - side3) << shift; + + pOutputSamples[i*8+0] = (drflac_int32)temp0L; + pOutputSamples[i*8+1] = (drflac_int32)temp0R; + pOutputSamples[i*8+2] = (drflac_int32)temp1L; + pOutputSamples[i*8+3] = (drflac_int32)temp1R; + pOutputSamples[i*8+4] = (drflac_int32)temp2L; + pOutputSamples[i*8+5] = (drflac_int32)temp2R; + pOutputSamples[i*8+6] = (drflac_int32)temp3L; + pOutputSamples[i*8+7] = (drflac_int32)temp3R; + } + } else { + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 temp0L; + drflac_uint32 temp1L; + drflac_uint32 temp2L; + drflac_uint32 temp3L; + drflac_uint32 temp0R; + drflac_uint32 temp1R; + drflac_uint32 temp2R; + drflac_uint32 temp3R; + + drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid0 = (mid0 << 1) | (side0 & 0x01); + mid1 = (mid1 << 1) | (side1 & 0x01); + mid2 = (mid2 << 1) | (side2 & 0x01); + mid3 = (mid3 << 1) | (side3 & 0x01); + + temp0L = (drflac_uint32)((drflac_int32)(mid0 + side0) >> 1); + temp1L = (drflac_uint32)((drflac_int32)(mid1 + side1) >> 1); + temp2L = (drflac_uint32)((drflac_int32)(mid2 + side2) >> 1); + temp3L = (drflac_uint32)((drflac_int32)(mid3 + side3) >> 1); + + temp0R = (drflac_uint32)((drflac_int32)(mid0 - side0) >> 1); + temp1R = (drflac_uint32)((drflac_int32)(mid1 - side1) >> 1); + temp2R = (drflac_uint32)((drflac_int32)(mid2 - side2) >> 1); + temp3R = (drflac_uint32)((drflac_int32)(mid3 - side3) >> 1); + + pOutputSamples[i*8+0] = (drflac_int32)temp0L; + pOutputSamples[i*8+1] = (drflac_int32)temp0R; + pOutputSamples[i*8+2] = (drflac_int32)temp1L; + pOutputSamples[i*8+3] = (drflac_int32)temp1R; + pOutputSamples[i*8+4] = (drflac_int32)temp2L; + pOutputSamples[i*8+5] = (drflac_int32)temp2R; + pOutputSamples[i*8+6] = (drflac_int32)temp3L; + pOutputSamples[i*8+7] = (drflac_int32)temp3R; + } + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid + side) >> 1) << unusedBitsPerSample); + pOutputSamples[i*2+1] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid - side) >> 1) << unusedBitsPerSample); + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_mid_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_int32 shift = unusedBitsPerSample; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + if (shift == 0) { + for (i = 0; i < frameCount4; ++i) { + __m128i mid; + __m128i side; + __m128i left; + __m128i right; + + mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + + mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01))); + + left = _mm_srai_epi32(_mm_add_epi32(mid, side), 1); + right = _mm_srai_epi32(_mm_sub_epi32(mid, side), 1); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 0), _mm_unpacklo_epi32(left, right)); + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 4), _mm_unpackhi_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)(mid + side) >> 1; + pOutputSamples[i*2+1] = (drflac_int32)(mid - side) >> 1; + } + } else { + shift -= 1; + for (i = 0; i < frameCount4; ++i) { + __m128i mid; + __m128i side; + __m128i left; + __m128i right; + + mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + + mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01))); + + left = _mm_slli_epi32(_mm_add_epi32(mid, side), shift); + right = _mm_slli_epi32(_mm_sub_epi32(mid, side), shift); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 0), _mm_unpacklo_epi32(left, right)); + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 4), _mm_unpackhi_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)((mid + side) << shift); + pOutputSamples[i*2+1] = (drflac_int32)((mid - side) << shift); + } + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_mid_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_int32 shift = unusedBitsPerSample; + int32x4_t wbpsShift0_4; /* wbps = Wasted Bits Per Sample */ + int32x4_t wbpsShift1_4; /* wbps = Wasted Bits Per Sample */ + uint32x4_t one4; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + wbpsShift0_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + wbpsShift1_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + one4 = vdupq_n_u32(1); + + if (shift == 0) { + for (i = 0; i < frameCount4; ++i) { + uint32x4_t mid; + uint32x4_t side; + int32x4_t left; + int32x4_t right; + + mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbpsShift0_4); + side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbpsShift1_4); + + mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, one4)); + + left = vshrq_n_s32(vreinterpretq_s32_u32(vaddq_u32(mid, side)), 1); + right = vshrq_n_s32(vreinterpretq_s32_u32(vsubq_u32(mid, side)), 1); + + drflac__vst2q_s32(pOutputSamples + i*8, vzipq_s32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)(mid + side) >> 1; + pOutputSamples[i*2+1] = (drflac_int32)(mid - side) >> 1; + } + } else { + int32x4_t shift4; + + shift -= 1; + shift4 = vdupq_n_s32(shift); + + for (i = 0; i < frameCount4; ++i) { + uint32x4_t mid; + uint32x4_t side; + int32x4_t left; + int32x4_t right; + + mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbpsShift0_4); + side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbpsShift1_4); + + mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, one4)); + + left = vreinterpretq_s32_u32(vshlq_u32(vaddq_u32(mid, side), shift4)); + right = vreinterpretq_s32_u32(vshlq_u32(vsubq_u32(mid, side), shift4)); + + drflac__vst2q_s32(pOutputSamples + i*8, vzipq_s32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)((mid + side) << shift); + pOutputSamples[i*2+1] = (drflac_int32)((mid - side) << shift); + } + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_mid_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s32__decode_mid_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s32__decode_mid_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_s32__decode_mid_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_s32__decode_mid_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_independent_stereo__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + for (drflac_uint64 i = 0; i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int32)((drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample)); + pOutputSamples[i*2+1] = (drflac_int32)((drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample)); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_independent_stereo__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 tempL0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 tempL1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 tempL2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 tempL3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 tempR0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 tempR1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 tempR2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 tempR3 = pInputSamples1U32[i*4+3] << shift1; + + pOutputSamples[i*8+0] = (drflac_int32)tempL0; + pOutputSamples[i*8+1] = (drflac_int32)tempR0; + pOutputSamples[i*8+2] = (drflac_int32)tempL1; + pOutputSamples[i*8+3] = (drflac_int32)tempR1; + pOutputSamples[i*8+4] = (drflac_int32)tempL2; + pOutputSamples[i*8+5] = (drflac_int32)tempR2; + pOutputSamples[i*8+6] = (drflac_int32)tempL3; + pOutputSamples[i*8+7] = (drflac_int32)tempR3; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0); + pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1); + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_independent_stereo__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + for (i = 0; i < frameCount4; ++i) { + __m128i left = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + __m128i right = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 0), _mm_unpacklo_epi32(left, right)); + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8 + 4), _mm_unpackhi_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0); + pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1); + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_independent_stereo__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + int32x4_t shift4_0 = vdupq_n_s32(shift0); + int32x4_t shift4_1 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + int32x4_t left; + int32x4_t right; + + left = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift4_0)); + right = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift4_1)); + + drflac__vst2q_s32(pOutputSamples + i*8, vzipq_s32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0); + pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s32__decode_independent_stereo(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int32* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s32__decode_independent_stereo__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s32__decode_independent_stereo__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_s32__decode_independent_stereo__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_s32__decode_independent_stereo__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +static drflac_bool32 drflac__is_current_flac_frame_valid(drflac* pFlac) +{ + drflac_uint32 iChannel; + + if (pFlac->currentFLACFrame.header.blockSizeInPCMFrames > pFlac->maxBlockSizeInPCMFrames || pFlac->currentFLACFrame.pcmFramesRemaining > pFlac->currentFLACFrame.header.blockSizeInPCMFrames) { + return DRFLAC_FALSE; + } + + for (iChannel = 0; iChannel < pFlac->channels; iChannel += 1) { + if (pFlac->currentFLACFrame.subframes[iChannel].pSamplesS32 == NULL) { + return DRFLAC_FALSE; + } + } + + return DRFLAC_TRUE; +} + +DRFLAC_API drflac_uint64 drflac_read_pcm_frames_s32(drflac* pFlac, drflac_uint64 framesToRead, drflac_int32* pBufferOut) +{ + drflac_uint64 framesRead; + drflac_uint32 unusedBitsPerSample; + + if (pFlac == NULL || framesToRead == 0) { + return 0; + } + + if (pBufferOut == NULL) { + return drflac__seek_forward_by_pcm_frames(pFlac, framesToRead); + } + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 32); + unusedBitsPerSample = 32 - pFlac->bitsPerSample; + + framesRead = 0; + while (framesToRead > 0) { + /* If we've run out of samples in this frame, go to the next. */ + if (pFlac->currentFLACFrame.pcmFramesRemaining == 0) { + if (!drflac__read_and_decode_next_flac_frame(pFlac)) { + break; /* Couldn't read the next frame, so just break from the loop and return. */ + } + } else { + unsigned int channelCount = drflac__get_channel_count_from_channel_assignment(pFlac->currentFLACFrame.header.channelAssignment); + drflac_uint64 iFirstPCMFrame = pFlac->currentFLACFrame.header.blockSizeInPCMFrames - pFlac->currentFLACFrame.pcmFramesRemaining; + drflac_uint64 frameCountThisIteration = framesToRead; + + if (frameCountThisIteration > pFlac->currentFLACFrame.pcmFramesRemaining) { + frameCountThisIteration = pFlac->currentFLACFrame.pcmFramesRemaining; + } + + if (channelCount == 2) { + const drflac_int32* pDecodedSamples0 = pFlac->currentFLACFrame.subframes[0].pSamplesS32 + iFirstPCMFrame; + const drflac_int32* pDecodedSamples1 = pFlac->currentFLACFrame.subframes[1].pSamplesS32 + iFirstPCMFrame; + + switch (pFlac->currentFLACFrame.header.channelAssignment) + { + case DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE: + { + drflac_read_pcm_frames_s32__decode_left_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE: + { + drflac_read_pcm_frames_s32__decode_right_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE: + { + drflac_read_pcm_frames_s32__decode_mid_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_INDEPENDENT: + default: + { + drflac_read_pcm_frames_s32__decode_independent_stereo(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + } + } else { + /* Generic interleaving. */ + drflac_uint64 i; + for (i = 0; i < frameCountThisIteration; ++i) { + unsigned int j; + for (j = 0; j < channelCount; ++j) { + pBufferOut[(i*channelCount)+j] = (drflac_int32)((drflac_uint32)(pFlac->currentFLACFrame.subframes[j].pSamplesS32[iFirstPCMFrame + i]) << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[j].wastedBitsPerSample)); + } + } + } + + framesRead += frameCountThisIteration; + pBufferOut += frameCountThisIteration * channelCount; + framesToRead -= frameCountThisIteration; + pFlac->currentPCMFrame += frameCountThisIteration; + pFlac->currentFLACFrame.pcmFramesRemaining -= (drflac_uint32)frameCountThisIteration; + } + } + + return framesRead; +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_left_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + for (i = 0; i < frameCount; ++i) { + drflac_uint32 left = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + drflac_uint32 side = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + drflac_uint32 right = left - side; + + left >>= 16; + right >>= 16; + + pOutputSamples[i*2+0] = (drflac_int16)left; + pOutputSamples[i*2+1] = (drflac_int16)right; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_left_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 left0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 left1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 left2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 left3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << shift1; + + drflac_uint32 right0 = left0 - side0; + drflac_uint32 right1 = left1 - side1; + drflac_uint32 right2 = left2 - side2; + drflac_uint32 right3 = left3 - side3; + + left0 >>= 16; + left1 >>= 16; + left2 >>= 16; + left3 >>= 16; + + right0 >>= 16; + right1 >>= 16; + right2 >>= 16; + right3 >>= 16; + + pOutputSamples[i*8+0] = (drflac_int16)left0; + pOutputSamples[i*8+1] = (drflac_int16)right0; + pOutputSamples[i*8+2] = (drflac_int16)left1; + pOutputSamples[i*8+3] = (drflac_int16)right1; + pOutputSamples[i*8+4] = (drflac_int16)left2; + pOutputSamples[i*8+5] = (drflac_int16)right2; + pOutputSamples[i*8+6] = (drflac_int16)left3; + pOutputSamples[i*8+7] = (drflac_int16)right3; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + left >>= 16; + right >>= 16; + + pOutputSamples[i*2+0] = (drflac_int16)left; + pOutputSamples[i*2+1] = (drflac_int16)right; + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_left_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + for (i = 0; i < frameCount4; ++i) { + __m128i left = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + __m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + __m128i right = _mm_sub_epi32(left, side); + + left = _mm_srai_epi32(left, 16); + right = _mm_srai_epi32(right, 16); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8), drflac__mm_packs_interleaved_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + left >>= 16; + right >>= 16; + + pOutputSamples[i*2+0] = (drflac_int16)left; + pOutputSamples[i*2+1] = (drflac_int16)right; + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_left_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + int32x4_t shift0_4; + int32x4_t shift1_4; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + shift0_4 = vdupq_n_s32(shift0); + shift1_4 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + uint32x4_t left; + uint32x4_t side; + uint32x4_t right; + + left = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4); + side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4); + right = vsubq_u32(left, side); + + left = vshrq_n_u32(left, 16); + right = vshrq_n_u32(right, 16); + + drflac__vst2q_u16((drflac_uint16*)pOutputSamples + i*8, vzip_u16(vmovn_u32(left), vmovn_u32(right))); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + left >>= 16; + right >>= 16; + + pOutputSamples[i*2+0] = (drflac_int16)left; + pOutputSamples[i*2+1] = (drflac_int16)right; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_left_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s16__decode_left_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s16__decode_left_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_s16__decode_left_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_s16__decode_left_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_right_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + for (i = 0; i < frameCount; ++i) { + drflac_uint32 side = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + drflac_uint32 right = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + drflac_uint32 left = right + side; + + left >>= 16; + right >>= 16; + + pOutputSamples[i*2+0] = (drflac_int16)left; + pOutputSamples[i*2+1] = (drflac_int16)right; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_right_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 side0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 side1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 side2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 side3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 right0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 right1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 right2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 right3 = pInputSamples1U32[i*4+3] << shift1; + + drflac_uint32 left0 = right0 + side0; + drflac_uint32 left1 = right1 + side1; + drflac_uint32 left2 = right2 + side2; + drflac_uint32 left3 = right3 + side3; + + left0 >>= 16; + left1 >>= 16; + left2 >>= 16; + left3 >>= 16; + + right0 >>= 16; + right1 >>= 16; + right2 >>= 16; + right3 >>= 16; + + pOutputSamples[i*8+0] = (drflac_int16)left0; + pOutputSamples[i*8+1] = (drflac_int16)right0; + pOutputSamples[i*8+2] = (drflac_int16)left1; + pOutputSamples[i*8+3] = (drflac_int16)right1; + pOutputSamples[i*8+4] = (drflac_int16)left2; + pOutputSamples[i*8+5] = (drflac_int16)right2; + pOutputSamples[i*8+6] = (drflac_int16)left3; + pOutputSamples[i*8+7] = (drflac_int16)right3; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + left >>= 16; + right >>= 16; + + pOutputSamples[i*2+0] = (drflac_int16)left; + pOutputSamples[i*2+1] = (drflac_int16)right; + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_right_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + for (i = 0; i < frameCount4; ++i) { + __m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + __m128i right = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + __m128i left = _mm_add_epi32(right, side); + + left = _mm_srai_epi32(left, 16); + right = _mm_srai_epi32(right, 16); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8), drflac__mm_packs_interleaved_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + left >>= 16; + right >>= 16; + + pOutputSamples[i*2+0] = (drflac_int16)left; + pOutputSamples[i*2+1] = (drflac_int16)right; + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_right_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + int32x4_t shift0_4; + int32x4_t shift1_4; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + shift0_4 = vdupq_n_s32(shift0); + shift1_4 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + uint32x4_t side; + uint32x4_t right; + uint32x4_t left; + + side = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4); + right = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4); + left = vaddq_u32(right, side); + + left = vshrq_n_u32(left, 16); + right = vshrq_n_u32(right, 16); + + drflac__vst2q_u16((drflac_uint16*)pOutputSamples + i*8, vzip_u16(vmovn_u32(left), vmovn_u32(right))); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + left >>= 16; + right >>= 16; + + pOutputSamples[i*2+0] = (drflac_int16)left; + pOutputSamples[i*2+1] = (drflac_int16)right; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_right_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s16__decode_right_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s16__decode_right_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_s16__decode_right_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_s16__decode_right_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_mid_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + for (drflac_uint64 i = 0; i < frameCount; ++i) { + drflac_uint32 mid = (drflac_uint32)pInputSamples0[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = (drflac_uint32)pInputSamples1[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int16)(((drflac_uint32)((drflac_int32)(mid + side) >> 1) << unusedBitsPerSample) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)(((drflac_uint32)((drflac_int32)(mid - side) >> 1) << unusedBitsPerSample) >> 16); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_mid_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift = unusedBitsPerSample; + + if (shift > 0) { + shift -= 1; + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 temp0L; + drflac_uint32 temp1L; + drflac_uint32 temp2L; + drflac_uint32 temp3L; + drflac_uint32 temp0R; + drflac_uint32 temp1R; + drflac_uint32 temp2R; + drflac_uint32 temp3R; + + drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid0 = (mid0 << 1) | (side0 & 0x01); + mid1 = (mid1 << 1) | (side1 & 0x01); + mid2 = (mid2 << 1) | (side2 & 0x01); + mid3 = (mid3 << 1) | (side3 & 0x01); + + temp0L = (mid0 + side0) << shift; + temp1L = (mid1 + side1) << shift; + temp2L = (mid2 + side2) << shift; + temp3L = (mid3 + side3) << shift; + + temp0R = (mid0 - side0) << shift; + temp1R = (mid1 - side1) << shift; + temp2R = (mid2 - side2) << shift; + temp3R = (mid3 - side3) << shift; + + temp0L >>= 16; + temp1L >>= 16; + temp2L >>= 16; + temp3L >>= 16; + + temp0R >>= 16; + temp1R >>= 16; + temp2R >>= 16; + temp3R >>= 16; + + pOutputSamples[i*8+0] = (drflac_int16)temp0L; + pOutputSamples[i*8+1] = (drflac_int16)temp0R; + pOutputSamples[i*8+2] = (drflac_int16)temp1L; + pOutputSamples[i*8+3] = (drflac_int16)temp1R; + pOutputSamples[i*8+4] = (drflac_int16)temp2L; + pOutputSamples[i*8+5] = (drflac_int16)temp2R; + pOutputSamples[i*8+6] = (drflac_int16)temp3L; + pOutputSamples[i*8+7] = (drflac_int16)temp3R; + } + } else { + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 temp0L; + drflac_uint32 temp1L; + drflac_uint32 temp2L; + drflac_uint32 temp3L; + drflac_uint32 temp0R; + drflac_uint32 temp1R; + drflac_uint32 temp2R; + drflac_uint32 temp3R; + + drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid0 = (mid0 << 1) | (side0 & 0x01); + mid1 = (mid1 << 1) | (side1 & 0x01); + mid2 = (mid2 << 1) | (side2 & 0x01); + mid3 = (mid3 << 1) | (side3 & 0x01); + + temp0L = ((drflac_int32)(mid0 + side0) >> 1); + temp1L = ((drflac_int32)(mid1 + side1) >> 1); + temp2L = ((drflac_int32)(mid2 + side2) >> 1); + temp3L = ((drflac_int32)(mid3 + side3) >> 1); + + temp0R = ((drflac_int32)(mid0 - side0) >> 1); + temp1R = ((drflac_int32)(mid1 - side1) >> 1); + temp2R = ((drflac_int32)(mid2 - side2) >> 1); + temp3R = ((drflac_int32)(mid3 - side3) >> 1); + + temp0L >>= 16; + temp1L >>= 16; + temp2L >>= 16; + temp3L >>= 16; + + temp0R >>= 16; + temp1R >>= 16; + temp2R >>= 16; + temp3R >>= 16; + + pOutputSamples[i*8+0] = (drflac_int16)temp0L; + pOutputSamples[i*8+1] = (drflac_int16)temp0R; + pOutputSamples[i*8+2] = (drflac_int16)temp1L; + pOutputSamples[i*8+3] = (drflac_int16)temp1R; + pOutputSamples[i*8+4] = (drflac_int16)temp2L; + pOutputSamples[i*8+5] = (drflac_int16)temp2R; + pOutputSamples[i*8+6] = (drflac_int16)temp3L; + pOutputSamples[i*8+7] = (drflac_int16)temp3R; + } + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int16)(((drflac_uint32)((drflac_int32)(mid + side) >> 1) << unusedBitsPerSample) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)(((drflac_uint32)((drflac_int32)(mid - side) >> 1) << unusedBitsPerSample) >> 16); + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_mid_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift = unusedBitsPerSample; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + if (shift == 0) { + for (i = 0; i < frameCount4; ++i) { + __m128i mid; + __m128i side; + __m128i left; + __m128i right; + + mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + + mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01))); + + left = _mm_srai_epi32(_mm_add_epi32(mid, side), 1); + right = _mm_srai_epi32(_mm_sub_epi32(mid, side), 1); + + left = _mm_srai_epi32(left, 16); + right = _mm_srai_epi32(right, 16); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8), drflac__mm_packs_interleaved_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int16)(((drflac_int32)(mid + side) >> 1) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)(((drflac_int32)(mid - side) >> 1) >> 16); + } + } else { + shift -= 1; + for (i = 0; i < frameCount4; ++i) { + __m128i mid; + __m128i side; + __m128i left; + __m128i right; + + mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + + mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01))); + + left = _mm_slli_epi32(_mm_add_epi32(mid, side), shift); + right = _mm_slli_epi32(_mm_sub_epi32(mid, side), shift); + + left = _mm_srai_epi32(left, 16); + right = _mm_srai_epi32(right, 16); + + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8), drflac__mm_packs_interleaved_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int16)(((mid + side) << shift) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)(((mid - side) << shift) >> 16); + } + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_mid_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift = unusedBitsPerSample; + int32x4_t wbpsShift0_4; /* wbps = Wasted Bits Per Sample */ + int32x4_t wbpsShift1_4; /* wbps = Wasted Bits Per Sample */ + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + wbpsShift0_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + wbpsShift1_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + + if (shift == 0) { + for (i = 0; i < frameCount4; ++i) { + uint32x4_t mid; + uint32x4_t side; + int32x4_t left; + int32x4_t right; + + mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbpsShift0_4); + side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbpsShift1_4); + + mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, vdupq_n_u32(1))); + + left = vshrq_n_s32(vreinterpretq_s32_u32(vaddq_u32(mid, side)), 1); + right = vshrq_n_s32(vreinterpretq_s32_u32(vsubq_u32(mid, side)), 1); + + left = vshrq_n_s32(left, 16); + right = vshrq_n_s32(right, 16); + + drflac__vst2q_s16(pOutputSamples + i*8, vzip_s16(vmovn_s32(left), vmovn_s32(right))); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int16)(((drflac_int32)(mid + side) >> 1) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)(((drflac_int32)(mid - side) >> 1) >> 16); + } + } else { + int32x4_t shift4; + + shift -= 1; + shift4 = vdupq_n_s32(shift); + + for (i = 0; i < frameCount4; ++i) { + uint32x4_t mid; + uint32x4_t side; + int32x4_t left; + int32x4_t right; + + mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbpsShift0_4); + side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbpsShift1_4); + + mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, vdupq_n_u32(1))); + + left = vreinterpretq_s32_u32(vshlq_u32(vaddq_u32(mid, side), shift4)); + right = vreinterpretq_s32_u32(vshlq_u32(vsubq_u32(mid, side), shift4)); + + left = vshrq_n_s32(left, 16); + right = vshrq_n_s32(right, 16); + + drflac__vst2q_s16(pOutputSamples + i*8, vzip_s16(vmovn_s32(left), vmovn_s32(right))); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int16)(((mid + side) << shift) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)(((mid - side) << shift) >> 16); + } + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_mid_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s16__decode_mid_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s16__decode_mid_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_s16__decode_mid_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_s16__decode_mid_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_independent_stereo__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + for (drflac_uint64 i = 0; i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int16)((drflac_int32)((drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample)) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)((drflac_int32)((drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample)) >> 16); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_independent_stereo__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 tempL0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 tempL1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 tempL2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 tempL3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 tempR0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 tempR1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 tempR2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 tempR3 = pInputSamples1U32[i*4+3] << shift1; + + tempL0 >>= 16; + tempL1 >>= 16; + tempL2 >>= 16; + tempL3 >>= 16; + + tempR0 >>= 16; + tempR1 >>= 16; + tempR2 >>= 16; + tempR3 >>= 16; + + pOutputSamples[i*8+0] = (drflac_int16)tempL0; + pOutputSamples[i*8+1] = (drflac_int16)tempR0; + pOutputSamples[i*8+2] = (drflac_int16)tempL1; + pOutputSamples[i*8+3] = (drflac_int16)tempR1; + pOutputSamples[i*8+4] = (drflac_int16)tempL2; + pOutputSamples[i*8+5] = (drflac_int16)tempR2; + pOutputSamples[i*8+6] = (drflac_int16)tempL3; + pOutputSamples[i*8+7] = (drflac_int16)tempR3; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int16)((pInputSamples0U32[i] << shift0) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)((pInputSamples1U32[i] << shift1) >> 16); + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_independent_stereo__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + for (i = 0; i < frameCount4; ++i) { + __m128i left = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + __m128i right = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + + left = _mm_srai_epi32(left, 16); + right = _mm_srai_epi32(right, 16); + + /* At this point we have results. We can now pack and interleave these into a single __m128i object and then store the in the output buffer. */ + _mm_storeu_si128((__m128i*)(pOutputSamples + i*8), drflac__mm_packs_interleaved_epi32(left, right)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int16)((pInputSamples0U32[i] << shift0) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)((pInputSamples1U32[i] << shift1) >> 16); + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_independent_stereo__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + int32x4_t shift0_4 = vdupq_n_s32(shift0); + int32x4_t shift1_4 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + int32x4_t left; + int32x4_t right; + + left = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4)); + right = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4)); + + left = vshrq_n_s32(left, 16); + right = vshrq_n_s32(right, 16); + + drflac__vst2q_s16(pOutputSamples + i*8, vzip_s16(vmovn_s32(left), vmovn_s32(right))); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int16)((pInputSamples0U32[i] << shift0) >> 16); + pOutputSamples[i*2+1] = (drflac_int16)((pInputSamples1U32[i] << shift1) >> 16); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_s16__decode_independent_stereo(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, drflac_int16* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s16__decode_independent_stereo__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_s16__decode_independent_stereo__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_s16__decode_independent_stereo__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_s16__decode_independent_stereo__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + +DRFLAC_API drflac_uint64 drflac_read_pcm_frames_s16(drflac* pFlac, drflac_uint64 framesToRead, drflac_int16* pBufferOut) +{ + drflac_uint64 framesRead; + drflac_uint32 unusedBitsPerSample; + + if (pFlac == NULL || framesToRead == 0) { + return 0; + } + + if (pBufferOut == NULL) { + return drflac__seek_forward_by_pcm_frames(pFlac, framesToRead); + } + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 32); + unusedBitsPerSample = 32 - pFlac->bitsPerSample; + + framesRead = 0; + while (framesToRead > 0) { + /* If we've run out of samples in this frame, go to the next. */ + if (pFlac->currentFLACFrame.pcmFramesRemaining == 0) { + if (!drflac__read_and_decode_next_flac_frame(pFlac)) { + break; /* Couldn't read the next frame, so just break from the loop and return. */ + } + } else { + unsigned int channelCount = drflac__get_channel_count_from_channel_assignment(pFlac->currentFLACFrame.header.channelAssignment); + drflac_uint64 iFirstPCMFrame = pFlac->currentFLACFrame.header.blockSizeInPCMFrames - pFlac->currentFLACFrame.pcmFramesRemaining; + drflac_uint64 frameCountThisIteration = framesToRead; + + if (frameCountThisIteration > pFlac->currentFLACFrame.pcmFramesRemaining) { + frameCountThisIteration = pFlac->currentFLACFrame.pcmFramesRemaining; + } + + if (channelCount == 2) { + const drflac_int32* pDecodedSamples0 = pFlac->currentFLACFrame.subframes[0].pSamplesS32 + iFirstPCMFrame; + const drflac_int32* pDecodedSamples1 = pFlac->currentFLACFrame.subframes[1].pSamplesS32 + iFirstPCMFrame; + + switch (pFlac->currentFLACFrame.header.channelAssignment) + { + case DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE: + { + drflac_read_pcm_frames_s16__decode_left_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE: + { + drflac_read_pcm_frames_s16__decode_right_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE: + { + drflac_read_pcm_frames_s16__decode_mid_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_INDEPENDENT: + default: + { + drflac_read_pcm_frames_s16__decode_independent_stereo(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + } + } else { + /* Generic interleaving. */ + drflac_uint64 i; + for (i = 0; i < frameCountThisIteration; ++i) { + unsigned int j; + for (j = 0; j < channelCount; ++j) { + drflac_int32 sampleS32 = (drflac_int32)((drflac_uint32)(pFlac->currentFLACFrame.subframes[j].pSamplesS32[iFirstPCMFrame + i]) << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[j].wastedBitsPerSample)); + pBufferOut[(i*channelCount)+j] = (drflac_int16)(sampleS32 >> 16); + } + } + } + + framesRead += frameCountThisIteration; + pBufferOut += frameCountThisIteration * channelCount; + framesToRead -= frameCountThisIteration; + pFlac->currentPCMFrame += frameCountThisIteration; + pFlac->currentFLACFrame.pcmFramesRemaining -= (drflac_uint32)frameCountThisIteration; + } + } + + return framesRead; +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_left_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + for (i = 0; i < frameCount; ++i) { + drflac_uint32 left = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + drflac_uint32 side = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + drflac_uint32 right = left - side; + + pOutputSamples[i*2+0] = (float)((drflac_int32)left / 2147483648.0); + pOutputSamples[i*2+1] = (float)((drflac_int32)right / 2147483648.0); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_left_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + float factor = 1 / 2147483648.0; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 left0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 left1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 left2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 left3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << shift1; + + drflac_uint32 right0 = left0 - side0; + drflac_uint32 right1 = left1 - side1; + drflac_uint32 right2 = left2 - side2; + drflac_uint32 right3 = left3 - side3; + + pOutputSamples[i*8+0] = (drflac_int32)left0 * factor; + pOutputSamples[i*8+1] = (drflac_int32)right0 * factor; + pOutputSamples[i*8+2] = (drflac_int32)left1 * factor; + pOutputSamples[i*8+3] = (drflac_int32)right1 * factor; + pOutputSamples[i*8+4] = (drflac_int32)left2 * factor; + pOutputSamples[i*8+5] = (drflac_int32)right2 * factor; + pOutputSamples[i*8+6] = (drflac_int32)left3 * factor; + pOutputSamples[i*8+7] = (drflac_int32)right3 * factor; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + pOutputSamples[i*2+0] = (drflac_int32)left * factor; + pOutputSamples[i*2+1] = (drflac_int32)right * factor; + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_left_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8; + drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8; + __m128 factor; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + factor = _mm_set1_ps(1.0f / 8388608.0f); + + for (i = 0; i < frameCount4; ++i) { + __m128i left = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + __m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + __m128i right = _mm_sub_epi32(left, side); + __m128 leftf = _mm_mul_ps(_mm_cvtepi32_ps(left), factor); + __m128 rightf = _mm_mul_ps(_mm_cvtepi32_ps(right), factor); + + _mm_storeu_ps(pOutputSamples + i*8 + 0, _mm_unpacklo_ps(leftf, rightf)); + _mm_storeu_ps(pOutputSamples + i*8 + 4, _mm_unpackhi_ps(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + pOutputSamples[i*2+0] = (drflac_int32)left / 8388608.0f; + pOutputSamples[i*2+1] = (drflac_int32)right / 8388608.0f; + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_left_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8; + drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8; + float32x4_t factor4; + int32x4_t shift0_4; + int32x4_t shift1_4; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + factor4 = vdupq_n_f32(1.0f / 8388608.0f); + shift0_4 = vdupq_n_s32(shift0); + shift1_4 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + uint32x4_t left; + uint32x4_t side; + uint32x4_t right; + float32x4_t leftf; + float32x4_t rightf; + + left = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4); + side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4); + right = vsubq_u32(left, side); + leftf = vmulq_f32(vcvtq_f32_s32(vreinterpretq_s32_u32(left)), factor4); + rightf = vmulq_f32(vcvtq_f32_s32(vreinterpretq_s32_u32(right)), factor4); + + drflac__vst2q_f32(pOutputSamples + i*8, vzipq_f32(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 left = pInputSamples0U32[i] << shift0; + drflac_uint32 side = pInputSamples1U32[i] << shift1; + drflac_uint32 right = left - side; + + pOutputSamples[i*2+0] = (drflac_int32)left / 8388608.0f; + pOutputSamples[i*2+1] = (drflac_int32)right / 8388608.0f; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_left_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_f32__decode_left_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_f32__decode_left_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_f32__decode_left_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_f32__decode_left_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_right_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + for (i = 0; i < frameCount; ++i) { + drflac_uint32 side = (drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + drflac_uint32 right = (drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + drflac_uint32 left = right + side; + + pOutputSamples[i*2+0] = (float)((drflac_int32)left / 2147483648.0); + pOutputSamples[i*2+1] = (float)((drflac_int32)right / 2147483648.0); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_right_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + float factor = 1 / 2147483648.0; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 side0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 side1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 side2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 side3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 right0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 right1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 right2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 right3 = pInputSamples1U32[i*4+3] << shift1; + + drflac_uint32 left0 = right0 + side0; + drflac_uint32 left1 = right1 + side1; + drflac_uint32 left2 = right2 + side2; + drflac_uint32 left3 = right3 + side3; + + pOutputSamples[i*8+0] = (drflac_int32)left0 * factor; + pOutputSamples[i*8+1] = (drflac_int32)right0 * factor; + pOutputSamples[i*8+2] = (drflac_int32)left1 * factor; + pOutputSamples[i*8+3] = (drflac_int32)right1 * factor; + pOutputSamples[i*8+4] = (drflac_int32)left2 * factor; + pOutputSamples[i*8+5] = (drflac_int32)right2 * factor; + pOutputSamples[i*8+6] = (drflac_int32)left3 * factor; + pOutputSamples[i*8+7] = (drflac_int32)right3 * factor; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + pOutputSamples[i*2+0] = (drflac_int32)left * factor; + pOutputSamples[i*2+1] = (drflac_int32)right * factor; + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_right_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8; + drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8; + __m128 factor; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + factor = _mm_set1_ps(1.0f / 8388608.0f); + + for (i = 0; i < frameCount4; ++i) { + __m128i side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + __m128i right = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + __m128i left = _mm_add_epi32(right, side); + __m128 leftf = _mm_mul_ps(_mm_cvtepi32_ps(left), factor); + __m128 rightf = _mm_mul_ps(_mm_cvtepi32_ps(right), factor); + + _mm_storeu_ps(pOutputSamples + i*8 + 0, _mm_unpacklo_ps(leftf, rightf)); + _mm_storeu_ps(pOutputSamples + i*8 + 4, _mm_unpackhi_ps(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + pOutputSamples[i*2+0] = (drflac_int32)left / 8388608.0f; + pOutputSamples[i*2+1] = (drflac_int32)right / 8388608.0f; + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_right_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8; + drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8; + float32x4_t factor4; + int32x4_t shift0_4; + int32x4_t shift1_4; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + factor4 = vdupq_n_f32(1.0f / 8388608.0f); + shift0_4 = vdupq_n_s32(shift0); + shift1_4 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + uint32x4_t side; + uint32x4_t right; + uint32x4_t left; + float32x4_t leftf; + float32x4_t rightf; + + side = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4); + right = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4); + left = vaddq_u32(right, side); + leftf = vmulq_f32(vcvtq_f32_s32(vreinterpretq_s32_u32(left)), factor4); + rightf = vmulq_f32(vcvtq_f32_s32(vreinterpretq_s32_u32(right)), factor4); + + drflac__vst2q_f32(pOutputSamples + i*8, vzipq_f32(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 side = pInputSamples0U32[i] << shift0; + drflac_uint32 right = pInputSamples1U32[i] << shift1; + drflac_uint32 left = right + side; + + pOutputSamples[i*2+0] = (drflac_int32)left / 8388608.0f; + pOutputSamples[i*2+1] = (drflac_int32)right / 8388608.0f; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_right_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_f32__decode_right_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_f32__decode_right_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_f32__decode_right_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_f32__decode_right_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_mid_side__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + for (drflac_uint64 i = 0; i < frameCount; ++i) { + drflac_uint32 mid = (drflac_uint32)pInputSamples0[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = (drflac_uint32)pInputSamples1[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (float)((((drflac_int32)(mid + side) >> 1) << (unusedBitsPerSample)) / 2147483648.0); + pOutputSamples[i*2+1] = (float)((((drflac_int32)(mid - side) >> 1) << (unusedBitsPerSample)) / 2147483648.0); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_mid_side__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift = unusedBitsPerSample; + float factor = 1 / 2147483648.0; + + if (shift > 0) { + shift -= 1; + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 temp0L; + drflac_uint32 temp1L; + drflac_uint32 temp2L; + drflac_uint32 temp3L; + drflac_uint32 temp0R; + drflac_uint32 temp1R; + drflac_uint32 temp2R; + drflac_uint32 temp3R; + + drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid0 = (mid0 << 1) | (side0 & 0x01); + mid1 = (mid1 << 1) | (side1 & 0x01); + mid2 = (mid2 << 1) | (side2 & 0x01); + mid3 = (mid3 << 1) | (side3 & 0x01); + + temp0L = (mid0 + side0) << shift; + temp1L = (mid1 + side1) << shift; + temp2L = (mid2 + side2) << shift; + temp3L = (mid3 + side3) << shift; + + temp0R = (mid0 - side0) << shift; + temp1R = (mid1 - side1) << shift; + temp2R = (mid2 - side2) << shift; + temp3R = (mid3 - side3) << shift; + + pOutputSamples[i*8+0] = (drflac_int32)temp0L * factor; + pOutputSamples[i*8+1] = (drflac_int32)temp0R * factor; + pOutputSamples[i*8+2] = (drflac_int32)temp1L * factor; + pOutputSamples[i*8+3] = (drflac_int32)temp1R * factor; + pOutputSamples[i*8+4] = (drflac_int32)temp2L * factor; + pOutputSamples[i*8+5] = (drflac_int32)temp2R * factor; + pOutputSamples[i*8+6] = (drflac_int32)temp3L * factor; + pOutputSamples[i*8+7] = (drflac_int32)temp3R * factor; + } + } else { + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 temp0L; + drflac_uint32 temp1L; + drflac_uint32 temp2L; + drflac_uint32 temp3L; + drflac_uint32 temp0R; + drflac_uint32 temp1R; + drflac_uint32 temp2R; + drflac_uint32 temp3R; + + drflac_uint32 mid0 = pInputSamples0U32[i*4+0] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid1 = pInputSamples0U32[i*4+1] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid2 = pInputSamples0U32[i*4+2] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 mid3 = pInputSamples0U32[i*4+3] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + + drflac_uint32 side0 = pInputSamples1U32[i*4+0] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side1 = pInputSamples1U32[i*4+1] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side2 = pInputSamples1U32[i*4+2] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + drflac_uint32 side3 = pInputSamples1U32[i*4+3] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid0 = (mid0 << 1) | (side0 & 0x01); + mid1 = (mid1 << 1) | (side1 & 0x01); + mid2 = (mid2 << 1) | (side2 & 0x01); + mid3 = (mid3 << 1) | (side3 & 0x01); + + temp0L = (drflac_uint32)((drflac_int32)(mid0 + side0) >> 1); + temp1L = (drflac_uint32)((drflac_int32)(mid1 + side1) >> 1); + temp2L = (drflac_uint32)((drflac_int32)(mid2 + side2) >> 1); + temp3L = (drflac_uint32)((drflac_int32)(mid3 + side3) >> 1); + + temp0R = (drflac_uint32)((drflac_int32)(mid0 - side0) >> 1); + temp1R = (drflac_uint32)((drflac_int32)(mid1 - side1) >> 1); + temp2R = (drflac_uint32)((drflac_int32)(mid2 - side2) >> 1); + temp3R = (drflac_uint32)((drflac_int32)(mid3 - side3) >> 1); + + pOutputSamples[i*8+0] = (drflac_int32)temp0L * factor; + pOutputSamples[i*8+1] = (drflac_int32)temp0R * factor; + pOutputSamples[i*8+2] = (drflac_int32)temp1L * factor; + pOutputSamples[i*8+3] = (drflac_int32)temp1R * factor; + pOutputSamples[i*8+4] = (drflac_int32)temp2L * factor; + pOutputSamples[i*8+5] = (drflac_int32)temp2R * factor; + pOutputSamples[i*8+6] = (drflac_int32)temp3L * factor; + pOutputSamples[i*8+7] = (drflac_int32)temp3R * factor; + } + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid + side) >> 1) << unusedBitsPerSample) * factor; + pOutputSamples[i*2+1] = (drflac_int32)((drflac_uint32)((drflac_int32)(mid - side) >> 1) << unusedBitsPerSample) * factor; + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_mid_side__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift = unusedBitsPerSample - 8; + float factor; + __m128 factor128; + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + factor = 1.0f / 8388608.0f; + factor128 = _mm_set1_ps(factor); + + if (shift == 0) { + for (i = 0; i < frameCount4; ++i) { + __m128i mid; + __m128i side; + __m128i tempL; + __m128i tempR; + __m128 leftf; + __m128 rightf; + + mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + + mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01))); + + tempL = _mm_srai_epi32(_mm_add_epi32(mid, side), 1); + tempR = _mm_srai_epi32(_mm_sub_epi32(mid, side), 1); + + leftf = _mm_mul_ps(_mm_cvtepi32_ps(tempL), factor128); + rightf = _mm_mul_ps(_mm_cvtepi32_ps(tempR), factor128); + + _mm_storeu_ps(pOutputSamples + i*8 + 0, _mm_unpacklo_ps(leftf, rightf)); + _mm_storeu_ps(pOutputSamples + i*8 + 4, _mm_unpackhi_ps(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = ((drflac_int32)(mid + side) >> 1) * factor; + pOutputSamples[i*2+1] = ((drflac_int32)(mid - side) >> 1) * factor; + } + } else { + shift -= 1; + for (i = 0; i < frameCount4; ++i) { + __m128i mid; + __m128i side; + __m128i tempL; + __m128i tempR; + __m128 leftf; + __m128 rightf; + + mid = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + side = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + + mid = _mm_or_si128(_mm_slli_epi32(mid, 1), _mm_and_si128(side, _mm_set1_epi32(0x01))); + + tempL = _mm_slli_epi32(_mm_add_epi32(mid, side), shift); + tempR = _mm_slli_epi32(_mm_sub_epi32(mid, side), shift); + + leftf = _mm_mul_ps(_mm_cvtepi32_ps(tempL), factor128); + rightf = _mm_mul_ps(_mm_cvtepi32_ps(tempR), factor128); + + _mm_storeu_ps(pOutputSamples + i*8 + 0, _mm_unpacklo_ps(leftf, rightf)); + _mm_storeu_ps(pOutputSamples + i*8 + 4, _mm_unpackhi_ps(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)((mid + side) << shift) * factor; + pOutputSamples[i*2+1] = (drflac_int32)((mid - side) << shift) * factor; + } + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_mid_side__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift = unusedBitsPerSample - 8; + float factor; + float32x4_t factor4; + int32x4_t shift4; + int32x4_t wbps0_4; /* Wasted Bits Per Sample */ + int32x4_t wbps1_4; /* Wasted Bits Per Sample */ + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 24); + + factor = 1.0f / 8388608.0f; + factor4 = vdupq_n_f32(factor); + wbps0_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample); + wbps1_4 = vdupq_n_s32(pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample); + + if (shift == 0) { + for (i = 0; i < frameCount4; ++i) { + int32x4_t lefti; + int32x4_t righti; + float32x4_t leftf; + float32x4_t rightf; + + uint32x4_t mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbps0_4); + uint32x4_t side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbps1_4); + + mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, vdupq_n_u32(1))); + + lefti = vshrq_n_s32(vreinterpretq_s32_u32(vaddq_u32(mid, side)), 1); + righti = vshrq_n_s32(vreinterpretq_s32_u32(vsubq_u32(mid, side)), 1); + + leftf = vmulq_f32(vcvtq_f32_s32(lefti), factor4); + rightf = vmulq_f32(vcvtq_f32_s32(righti), factor4); + + drflac__vst2q_f32(pOutputSamples + i*8, vzipq_f32(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = ((drflac_int32)(mid + side) >> 1) * factor; + pOutputSamples[i*2+1] = ((drflac_int32)(mid - side) >> 1) * factor; + } + } else { + shift -= 1; + shift4 = vdupq_n_s32(shift); + for (i = 0; i < frameCount4; ++i) { + uint32x4_t mid; + uint32x4_t side; + int32x4_t lefti; + int32x4_t righti; + float32x4_t leftf; + float32x4_t rightf; + + mid = vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), wbps0_4); + side = vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), wbps1_4); + + mid = vorrq_u32(vshlq_n_u32(mid, 1), vandq_u32(side, vdupq_n_u32(1))); + + lefti = vreinterpretq_s32_u32(vshlq_u32(vaddq_u32(mid, side), shift4)); + righti = vreinterpretq_s32_u32(vshlq_u32(vsubq_u32(mid, side), shift4)); + + leftf = vmulq_f32(vcvtq_f32_s32(lefti), factor4); + rightf = vmulq_f32(vcvtq_f32_s32(righti), factor4); + + drflac__vst2q_f32(pOutputSamples + i*8, vzipq_f32(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + drflac_uint32 mid = pInputSamples0U32[i] << pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 side = pInputSamples1U32[i] << pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + + mid = (mid << 1) | (side & 0x01); + + pOutputSamples[i*2+0] = (drflac_int32)((mid + side) << shift) * factor; + pOutputSamples[i*2+1] = (drflac_int32)((mid - side) << shift) * factor; + } + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_mid_side(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_f32__decode_mid_side__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_f32__decode_mid_side__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_f32__decode_mid_side__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_f32__decode_mid_side__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + +#if 0 +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_independent_stereo__reference(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + for (drflac_uint64 i = 0; i < frameCount; ++i) { + pOutputSamples[i*2+0] = (float)((drflac_int32)((drflac_uint32)pInputSamples0[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample)) / 2147483648.0); + pOutputSamples[i*2+1] = (float)((drflac_int32)((drflac_uint32)pInputSamples1[i] << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample)) / 2147483648.0); + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_independent_stereo__scalar(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample; + drflac_uint32 shift1 = unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample; + float factor = 1 / 2147483648.0; + + for (i = 0; i < frameCount4; ++i) { + drflac_uint32 tempL0 = pInputSamples0U32[i*4+0] << shift0; + drflac_uint32 tempL1 = pInputSamples0U32[i*4+1] << shift0; + drflac_uint32 tempL2 = pInputSamples0U32[i*4+2] << shift0; + drflac_uint32 tempL3 = pInputSamples0U32[i*4+3] << shift0; + + drflac_uint32 tempR0 = pInputSamples1U32[i*4+0] << shift1; + drflac_uint32 tempR1 = pInputSamples1U32[i*4+1] << shift1; + drflac_uint32 tempR2 = pInputSamples1U32[i*4+2] << shift1; + drflac_uint32 tempR3 = pInputSamples1U32[i*4+3] << shift1; + + pOutputSamples[i*8+0] = (drflac_int32)tempL0 * factor; + pOutputSamples[i*8+1] = (drflac_int32)tempR0 * factor; + pOutputSamples[i*8+2] = (drflac_int32)tempL1 * factor; + pOutputSamples[i*8+3] = (drflac_int32)tempR1 * factor; + pOutputSamples[i*8+4] = (drflac_int32)tempL2 * factor; + pOutputSamples[i*8+5] = (drflac_int32)tempR2 * factor; + pOutputSamples[i*8+6] = (drflac_int32)tempL3 * factor; + pOutputSamples[i*8+7] = (drflac_int32)tempR3 * factor; + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0) * factor; + pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1) * factor; + } +} + +#if defined(DRFLAC_SUPPORT_SSE2) +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_independent_stereo__sse2(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8; + drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8; + + float factor = 1.0f / 8388608.0f; + __m128 factor128 = _mm_set1_ps(factor); + + for (i = 0; i < frameCount4; ++i) { + __m128i lefti; + __m128i righti; + __m128 leftf; + __m128 rightf; + + lefti = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples0 + i), shift0); + righti = _mm_slli_epi32(_mm_loadu_si128((const __m128i*)pInputSamples1 + i), shift1); + + leftf = _mm_mul_ps(_mm_cvtepi32_ps(lefti), factor128); + rightf = _mm_mul_ps(_mm_cvtepi32_ps(righti), factor128); + + _mm_storeu_ps(pOutputSamples + i*8 + 0, _mm_unpacklo_ps(leftf, rightf)); + _mm_storeu_ps(pOutputSamples + i*8 + 4, _mm_unpackhi_ps(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0) * factor; + pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1) * factor; + } +} +#endif + +#if defined(DRFLAC_SUPPORT_NEON) +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_independent_stereo__neon(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ + drflac_uint64 i; + drflac_uint64 frameCount4 = frameCount >> 2; + const drflac_uint32* pInputSamples0U32 = (const drflac_uint32*)pInputSamples0; + const drflac_uint32* pInputSamples1U32 = (const drflac_uint32*)pInputSamples1; + drflac_uint32 shift0 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[0].wastedBitsPerSample) - 8; + drflac_uint32 shift1 = (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[1].wastedBitsPerSample) - 8; + + float factor = 1.0f / 8388608.0f; + float32x4_t factor4 = vdupq_n_f32(factor); + int32x4_t shift0_4 = vdupq_n_s32(shift0); + int32x4_t shift1_4 = vdupq_n_s32(shift1); + + for (i = 0; i < frameCount4; ++i) { + int32x4_t lefti; + int32x4_t righti; + float32x4_t leftf; + float32x4_t rightf; + + lefti = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples0U32 + i*4), shift0_4)); + righti = vreinterpretq_s32_u32(vshlq_u32(vld1q_u32(pInputSamples1U32 + i*4), shift1_4)); + + leftf = vmulq_f32(vcvtq_f32_s32(lefti), factor4); + rightf = vmulq_f32(vcvtq_f32_s32(righti), factor4); + + drflac__vst2q_f32(pOutputSamples + i*8, vzipq_f32(leftf, rightf)); + } + + for (i = (frameCount4 << 2); i < frameCount; ++i) { + pOutputSamples[i*2+0] = (drflac_int32)(pInputSamples0U32[i] << shift0) * factor; + pOutputSamples[i*2+1] = (drflac_int32)(pInputSamples1U32[i] << shift1) * factor; + } +} +#endif + +static DRFLAC_INLINE void drflac_read_pcm_frames_f32__decode_independent_stereo(drflac* pFlac, drflac_uint64 frameCount, drflac_uint32 unusedBitsPerSample, const drflac_int32* pInputSamples0, const drflac_int32* pInputSamples1, float* pOutputSamples) +{ +#if defined(DRFLAC_SUPPORT_SSE2) + if (drflac__gIsSSE2Supported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_f32__decode_independent_stereo__sse2(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#elif defined(DRFLAC_SUPPORT_NEON) + if (drflac__gIsNEONSupported && pFlac->bitsPerSample <= 24) { + drflac_read_pcm_frames_f32__decode_independent_stereo__neon(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); + } else +#endif + { + /* Scalar fallback. */ +#if 0 + drflac_read_pcm_frames_f32__decode_independent_stereo__reference(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#else + drflac_read_pcm_frames_f32__decode_independent_stereo__scalar(pFlac, frameCount, unusedBitsPerSample, pInputSamples0, pInputSamples1, pOutputSamples); +#endif + } +} + +DRFLAC_API drflac_uint64 drflac_read_pcm_frames_f32(drflac* pFlac, drflac_uint64 framesToRead, float* pBufferOut) +{ + drflac_uint64 framesRead; + drflac_uint32 unusedBitsPerSample; + + if (pFlac == NULL || framesToRead == 0) { + return 0; + } + + if (pBufferOut == NULL) { + return drflac__seek_forward_by_pcm_frames(pFlac, framesToRead); + } + + DRFLAC_ASSERT(pFlac->bitsPerSample <= 32); + unusedBitsPerSample = 32 - pFlac->bitsPerSample; + + framesRead = 0; + while (framesToRead > 0) { + /* If we've run out of samples in this frame, go to the next. */ + if (pFlac->currentFLACFrame.pcmFramesRemaining == 0) { + if (!drflac__read_and_decode_next_flac_frame(pFlac)) { + break; /* Couldn't read the next frame, so just break from the loop and return. */ + } + } else { + unsigned int channelCount = drflac__get_channel_count_from_channel_assignment(pFlac->currentFLACFrame.header.channelAssignment); + drflac_uint64 iFirstPCMFrame = pFlac->currentFLACFrame.header.blockSizeInPCMFrames - pFlac->currentFLACFrame.pcmFramesRemaining; + drflac_uint64 frameCountThisIteration = framesToRead; + + if (frameCountThisIteration > pFlac->currentFLACFrame.pcmFramesRemaining) { + frameCountThisIteration = pFlac->currentFLACFrame.pcmFramesRemaining; + } + + if (channelCount == 2) { + const drflac_int32* pDecodedSamples0 = pFlac->currentFLACFrame.subframes[0].pSamplesS32 + iFirstPCMFrame; + const drflac_int32* pDecodedSamples1 = pFlac->currentFLACFrame.subframes[1].pSamplesS32 + iFirstPCMFrame; + + switch (pFlac->currentFLACFrame.header.channelAssignment) + { + case DRFLAC_CHANNEL_ASSIGNMENT_LEFT_SIDE: + { + drflac_read_pcm_frames_f32__decode_left_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_RIGHT_SIDE: + { + drflac_read_pcm_frames_f32__decode_right_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_MID_SIDE: + { + drflac_read_pcm_frames_f32__decode_mid_side(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + + case DRFLAC_CHANNEL_ASSIGNMENT_INDEPENDENT: + default: + { + drflac_read_pcm_frames_f32__decode_independent_stereo(pFlac, frameCountThisIteration, unusedBitsPerSample, pDecodedSamples0, pDecodedSamples1, pBufferOut); + } break; + } + } else { + /* Generic interleaving. */ + drflac_uint64 i; + for (i = 0; i < frameCountThisIteration; ++i) { + unsigned int j; + for (j = 0; j < channelCount; ++j) { + drflac_int32 sampleS32 = (drflac_int32)((drflac_uint32)(pFlac->currentFLACFrame.subframes[j].pSamplesS32[iFirstPCMFrame + i]) << (unusedBitsPerSample + pFlac->currentFLACFrame.subframes[j].wastedBitsPerSample)); + pBufferOut[(i*channelCount)+j] = (float)(sampleS32 / 2147483648.0); + } + } + } + + framesRead += frameCountThisIteration; + pBufferOut += frameCountThisIteration * channelCount; + framesToRead -= frameCountThisIteration; + pFlac->currentPCMFrame += frameCountThisIteration; + pFlac->currentFLACFrame.pcmFramesRemaining -= (unsigned int)frameCountThisIteration; + } + } + + return framesRead; +} + + +DRFLAC_API drflac_bool32 drflac_seek_to_pcm_frame(drflac* pFlac, drflac_uint64 pcmFrameIndex) +{ + if (pFlac == NULL) { + return DRFLAC_FALSE; + } + + /* Don't do anything if we're already on the seek point. */ + if (pFlac->currentPCMFrame == pcmFrameIndex) { + return DRFLAC_TRUE; + } + + /* + If we don't know where the first frame begins then we can't seek. This will happen when the STREAMINFO block was not present + when the decoder was opened. + */ + if (pFlac->firstFLACFramePosInBytes == 0) { + return DRFLAC_FALSE; + } + + if (pcmFrameIndex == 0) { + pFlac->currentPCMFrame = 0; + return drflac__seek_to_first_frame(pFlac); + } else { + drflac_bool32 wasSuccessful = DRFLAC_FALSE; + drflac_uint64 originalPCMFrame = pFlac->currentPCMFrame; + + /* Clamp the sample to the end. */ + if (pcmFrameIndex > pFlac->totalPCMFrameCount) { + pcmFrameIndex = pFlac->totalPCMFrameCount; + } + + /* If the target sample and the current sample are in the same frame we just move the position forward. */ + if (drflac__is_current_flac_frame_valid(pFlac)) { + if (pcmFrameIndex > pFlac->currentPCMFrame) { + /* Forward. */ + drflac_uint32 offset = (drflac_uint32)(pcmFrameIndex - pFlac->currentPCMFrame); + if (pFlac->currentFLACFrame.pcmFramesRemaining > offset) { + pFlac->currentFLACFrame.pcmFramesRemaining -= offset; + pFlac->currentPCMFrame = pcmFrameIndex; + return DRFLAC_TRUE; + } + } else { + /* Backward. */ + drflac_uint32 offsetAbs = (drflac_uint32)(pFlac->currentPCMFrame - pcmFrameIndex); + drflac_uint32 currentFLACFramePCMFrameCount = pFlac->currentFLACFrame.header.blockSizeInPCMFrames; + drflac_uint32 currentFLACFramePCMFramesConsumed = currentFLACFramePCMFrameCount - pFlac->currentFLACFrame.pcmFramesRemaining; + if (currentFLACFramePCMFramesConsumed > offsetAbs) { + pFlac->currentFLACFrame.pcmFramesRemaining += offsetAbs; + pFlac->currentPCMFrame = pcmFrameIndex; + return DRFLAC_TRUE; + } + } + } + + /* + Different techniques depending on encapsulation. Using the native FLAC seektable with Ogg encapsulation is a bit awkward so + we'll instead use Ogg's natural seeking facility. + */ +#ifndef DR_FLAC_NO_OGG + if (pFlac->container == drflac_container_ogg) + { + wasSuccessful = drflac_ogg__seek_to_pcm_frame(pFlac, pcmFrameIndex); + } + else +#endif + { + /* First try seeking via the seek table. If this fails, fall back to a brute force seek which is much slower. */ + if (/*!wasSuccessful && */!pFlac->_noSeekTableSeek) { + wasSuccessful = drflac__seek_to_pcm_frame__seek_table(pFlac, pcmFrameIndex); + } + +#if !defined(DR_FLAC_NO_CRC) + /* Fall back to binary search if seek table seeking fails. This requires the length of the stream to be known. */ + if (!wasSuccessful && !pFlac->_noBinarySearchSeek && pFlac->totalPCMFrameCount > 0) { + wasSuccessful = drflac__seek_to_pcm_frame__binary_search(pFlac, pcmFrameIndex); + } +#endif + + /* Fall back to brute force if all else fails. */ + if (!wasSuccessful && !pFlac->_noBruteForceSeek) { + wasSuccessful = drflac__seek_to_pcm_frame__brute_force(pFlac, pcmFrameIndex); + } + } + + if (wasSuccessful) { + pFlac->currentPCMFrame = pcmFrameIndex; + } else { + /* Seek failed. Try putting the decoder back to it's original state. */ + if (drflac_seek_to_pcm_frame(pFlac, originalPCMFrame) == DRFLAC_FALSE) { + /* Failed to seek back to the original PCM frame. Fall back to 0. */ + drflac_seek_to_pcm_frame(pFlac, 0); + } + } + + return wasSuccessful; + } +} + + + +/* High Level APIs */ + +/* SIZE_MAX */ +#if defined(SIZE_MAX) + #define DRFLAC_SIZE_MAX SIZE_MAX +#else + #if defined(DRFLAC_64BIT) + #define DRFLAC_SIZE_MAX ((drflac_uint64)0xFFFFFFFFFFFFFFFF) + #else + #define DRFLAC_SIZE_MAX 0xFFFFFFFF + #endif +#endif +/* End SIZE_MAX */ + + +/* Using a macro as the definition of the drflac__full_decode_and_close_*() API family. Sue me. */ +#define DRFLAC_DEFINE_FULL_READ_AND_CLOSE(extension, type) \ +static type* drflac__full_read_and_close_ ## extension (drflac* pFlac, unsigned int* channelsOut, unsigned int* sampleRateOut, drflac_uint64* totalPCMFrameCountOut)\ +{ \ + type* pSampleData = NULL; \ + drflac_uint64 totalPCMFrameCount; \ + type buffer[4096]; \ + drflac_uint64 pcmFramesRead; \ + size_t sampleDataBufferSize = sizeof(buffer); \ + \ + DRFLAC_ASSERT(pFlac != NULL); \ + \ + totalPCMFrameCount = 0; \ + \ + pSampleData = (type*)drflac__malloc_from_callbacks(sampleDataBufferSize, &pFlac->allocationCallbacks); \ + if (pSampleData == NULL) { \ + goto on_error; \ + } \ + \ + while ((pcmFramesRead = (drflac_uint64)drflac_read_pcm_frames_##extension(pFlac, sizeof(buffer)/sizeof(buffer[0])/pFlac->channels, buffer)) > 0) { \ + if (((totalPCMFrameCount + pcmFramesRead) * pFlac->channels * sizeof(type)) > sampleDataBufferSize) { \ + type* pNewSampleData; \ + size_t newSampleDataBufferSize; \ + \ + newSampleDataBufferSize = sampleDataBufferSize * 2; \ + pNewSampleData = (type*)drflac__realloc_from_callbacks(pSampleData, newSampleDataBufferSize, sampleDataBufferSize, &pFlac->allocationCallbacks); \ + if (pNewSampleData == NULL) { \ + drflac__free_from_callbacks(pSampleData, &pFlac->allocationCallbacks); \ + goto on_error; \ + } \ + \ + sampleDataBufferSize = newSampleDataBufferSize; \ + pSampleData = pNewSampleData; \ + } \ + \ + DRFLAC_COPY_MEMORY(pSampleData + (totalPCMFrameCount*pFlac->channels), buffer, (size_t)(pcmFramesRead*pFlac->channels*sizeof(type))); \ + totalPCMFrameCount += pcmFramesRead; \ + } \ + \ + /* At this point everything should be decoded, but we just want to fill the unused part buffer with silence - need to \ + protect those ears from random noise! */ \ + DRFLAC_ZERO_MEMORY(pSampleData + (totalPCMFrameCount*pFlac->channels), (size_t)(sampleDataBufferSize - totalPCMFrameCount*pFlac->channels*sizeof(type))); \ + \ + if (sampleRateOut) *sampleRateOut = pFlac->sampleRate; \ + if (channelsOut) *channelsOut = pFlac->channels; \ + if (totalPCMFrameCountOut) *totalPCMFrameCountOut = totalPCMFrameCount; \ + \ + drflac_close(pFlac); \ + return pSampleData; \ + \ +on_error: \ + drflac_close(pFlac); \ + return NULL; \ +} + +DRFLAC_DEFINE_FULL_READ_AND_CLOSE(s32, drflac_int32) +DRFLAC_DEFINE_FULL_READ_AND_CLOSE(s16, drflac_int16) +DRFLAC_DEFINE_FULL_READ_AND_CLOSE(f32, float) + +DRFLAC_API drflac_int32* drflac_open_and_read_pcm_frames_s32(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_tell_proc onTell, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drflac_uint64* totalPCMFrameCountOut, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalPCMFrameCountOut) { + *totalPCMFrameCountOut = 0; + } + + pFlac = drflac_open(onRead, onSeek, onTell, pUserData, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_s32(pFlac, channelsOut, sampleRateOut, totalPCMFrameCountOut); +} + +DRFLAC_API drflac_int16* drflac_open_and_read_pcm_frames_s16(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_tell_proc onTell, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drflac_uint64* totalPCMFrameCountOut, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalPCMFrameCountOut) { + *totalPCMFrameCountOut = 0; + } + + pFlac = drflac_open(onRead, onSeek, onTell, pUserData, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_s16(pFlac, channelsOut, sampleRateOut, totalPCMFrameCountOut); +} + +DRFLAC_API float* drflac_open_and_read_pcm_frames_f32(drflac_read_proc onRead, drflac_seek_proc onSeek, drflac_tell_proc onTell, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drflac_uint64* totalPCMFrameCountOut, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalPCMFrameCountOut) { + *totalPCMFrameCountOut = 0; + } + + pFlac = drflac_open(onRead, onSeek, onTell, pUserData, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_f32(pFlac, channelsOut, sampleRateOut, totalPCMFrameCountOut); +} + +#ifndef DR_FLAC_NO_STDIO +DRFLAC_API drflac_int32* drflac_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalPCMFrameCount) { + *totalPCMFrameCount = 0; + } + + pFlac = drflac_open_file(filename, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_s32(pFlac, channels, sampleRate, totalPCMFrameCount); +} + +DRFLAC_API drflac_int16* drflac_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalPCMFrameCount) { + *totalPCMFrameCount = 0; + } + + pFlac = drflac_open_file(filename, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_s16(pFlac, channels, sampleRate, totalPCMFrameCount); +} + +DRFLAC_API float* drflac_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalPCMFrameCount) { + *totalPCMFrameCount = 0; + } + + pFlac = drflac_open_file(filename, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_f32(pFlac, channels, sampleRate, totalPCMFrameCount); +} +#endif + +DRFLAC_API drflac_int32* drflac_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalPCMFrameCount) { + *totalPCMFrameCount = 0; + } + + pFlac = drflac_open_memory(data, dataSize, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_s32(pFlac, channels, sampleRate, totalPCMFrameCount); +} + +DRFLAC_API drflac_int16* drflac_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalPCMFrameCount) { + *totalPCMFrameCount = 0; + } + + pFlac = drflac_open_memory(data, dataSize, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_s16(pFlac, channels, sampleRate, totalPCMFrameCount); +} + +DRFLAC_API float* drflac_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channels, unsigned int* sampleRate, drflac_uint64* totalPCMFrameCount, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + drflac* pFlac; + + if (sampleRate) { + *sampleRate = 0; + } + if (channels) { + *channels = 0; + } + if (totalPCMFrameCount) { + *totalPCMFrameCount = 0; + } + + pFlac = drflac_open_memory(data, dataSize, pAllocationCallbacks); + if (pFlac == NULL) { + return NULL; + } + + return drflac__full_read_and_close_f32(pFlac, channels, sampleRate, totalPCMFrameCount); +} + + +DRFLAC_API void drflac_free(void* p, const drflac_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks != NULL) { + drflac__free_from_callbacks(p, pAllocationCallbacks); + } else { + drflac__free_default(p, NULL); + } +} + + + + +DRFLAC_API void drflac_init_vorbis_comment_iterator(drflac_vorbis_comment_iterator* pIter, drflac_uint32 commentCount, const void* pComments) +{ + if (pIter == NULL) { + return; + } + + pIter->countRemaining = commentCount; + pIter->pRunningData = (const char*)pComments; +} + +DRFLAC_API const char* drflac_next_vorbis_comment(drflac_vorbis_comment_iterator* pIter, drflac_uint32* pCommentLengthOut) +{ + drflac_int32 length; + const char* pComment; + + /* Safety. */ + if (pCommentLengthOut) { + *pCommentLengthOut = 0; + } + + if (pIter == NULL || pIter->countRemaining == 0 || pIter->pRunningData == NULL) { + return NULL; + } + + length = drflac__le2host_32_ptr_unaligned(pIter->pRunningData); + pIter->pRunningData += 4; + + pComment = pIter->pRunningData; + pIter->pRunningData += length; + pIter->countRemaining -= 1; + + if (pCommentLengthOut) { + *pCommentLengthOut = length; + } + + return pComment; +} + + + + +DRFLAC_API void drflac_init_cuesheet_track_iterator(drflac_cuesheet_track_iterator* pIter, drflac_uint32 trackCount, const void* pTrackData) +{ + if (pIter == NULL) { + return; + } + + pIter->countRemaining = trackCount; + pIter->pRunningData = (const char*)pTrackData; +} + +DRFLAC_API drflac_bool32 drflac_next_cuesheet_track(drflac_cuesheet_track_iterator* pIter, drflac_cuesheet_track* pCuesheetTrack) +{ + drflac_cuesheet_track cuesheetTrack; + const char* pRunningData; + drflac_uint64 offsetHi; + drflac_uint64 offsetLo; + + if (pIter == NULL || pIter->countRemaining == 0 || pIter->pRunningData == NULL) { + return DRFLAC_FALSE; + } + + pRunningData = pIter->pRunningData; + + offsetHi = drflac__be2host_32(*(const drflac_uint32*)pRunningData); pRunningData += 4; + offsetLo = drflac__be2host_32(*(const drflac_uint32*)pRunningData); pRunningData += 4; + cuesheetTrack.offset = offsetLo | (offsetHi << 32); + cuesheetTrack.trackNumber = pRunningData[0]; pRunningData += 1; + DRFLAC_COPY_MEMORY(cuesheetTrack.ISRC, pRunningData, sizeof(cuesheetTrack.ISRC)); pRunningData += 12; + cuesheetTrack.isAudio = (pRunningData[0] & 0x80) != 0; + cuesheetTrack.preEmphasis = (pRunningData[0] & 0x40) != 0; pRunningData += 14; + cuesheetTrack.indexCount = pRunningData[0]; pRunningData += 1; + cuesheetTrack.pIndexPoints = (const drflac_cuesheet_track_index*)pRunningData; pRunningData += cuesheetTrack.indexCount * sizeof(drflac_cuesheet_track_index); + + pIter->pRunningData = pRunningData; + pIter->countRemaining -= 1; + + if (pCuesheetTrack) { + *pCuesheetTrack = cuesheetTrack; + } + + return DRFLAC_TRUE; +} + +#if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic pop +#endif +#endif /* dr_flac_c */ +#endif /* DR_FLAC_IMPLEMENTATION */ + + +/* +REVISION HISTORY +================ +v0.13.4 - TBD + - Add a bounds check when allocating memory during metadata processing. + - Fix a possible overflow error when parsing picture metadata. + - Fix an error with seek point parsing. + - Fix a possible deadlock when seeking. + - Fix an error where the decoder can be put into a bad state when seeking fails which then results in a crash when reading and seeking. + +v0.13.3 - 2026-01-17 + - Fix a compiler compatibility issue with some inlined assembly. + - Fix a compilation warning. + +v0.13.2 - 2025-12-02 + - Improve robustness of the parsing of picture metadata to improve support for memory constrained embedded devices. + - Fix a warning about an assigned by unused variable. + - Improvements to drflac_open_and_read_pcm_frames_*() and family to avoid excessively large memory allocations from malformed files. + +v0.13.1 - 2025-09-10 + - Fix an error with the NXDK build. + +v0.13.0 - 2025-07-23 + - API CHANGE: Seek origin enums have been renamed to match the naming convention used by other dr_libs libraries: + - drflac_seek_origin_start -> DRFLAC_SEEK_SET + - drflac_seek_origin_current -> DRFLAC_SEEK_CUR + - DRFLAC_SEEK_END (new) + - API CHANGE: A new seek origin has been added to allow seeking from the end of the file. If you implement your own `onSeek` callback, you should now detect and handle `DRFLAC_SEEK_END`. If seeking to the end is not supported, return `DRFLAC_FALSE`. If you only use `*_open_file()` or `*_open_memory()`, you need not change anything. + - API CHANGE: An `onTell` callback has been added to the following functions: + - drflac_open() + - drflac_open_relaxed() + - drflac_open_with_metadata() + - drflac_open_with_metadata_relaxed() + - drflac_open_and_read_pcm_frames_s32() + - drflac_open_and_read_pcm_frames_s16() + - drflac_open_and_read_pcm_frames_f32() + - Fix compilation for AIX OS. + +v0.12.43 - 2024-12-17 + - Fix a possible buffer overflow during decoding. + - Improve detection of ARM64EC + +v0.12.42 - 2023-11-02 + - Fix build for ARMv6-M. + - Fix a compilation warning with GCC. + +v0.12.41 - 2023-06-17 + - Fix an incorrect date in revision history. No functional change. + +v0.12.40 - 2023-05-22 + - Minor code restructure. No functional change. + +v0.12.39 - 2022-09-17 + - Fix compilation with DJGPP. + - Fix compilation error with Visual Studio 2019 and the ARM build. + - Fix an error with SSE 4.1 detection. + - Add support for disabling wchar_t with DR_WAV_NO_WCHAR. + - Improve compatibility with compilers which lack support for explicit struct packing. + - Improve compatibility with low-end and embedded hardware by reducing the amount of stack + allocation when loading an Ogg encapsulated file. + +v0.12.38 - 2022-04-10 + - Fix compilation error on older versions of GCC. + +v0.12.37 - 2022-02-12 + - Improve ARM detection. + +v0.12.36 - 2022-02-07 + - Fix a compilation error with the ARM build. + +v0.12.35 - 2022-02-06 + - Fix a bug due to underestimating the amount of precision required for the prediction stage. + - Fix some bugs found from fuzz testing. + +v0.12.34 - 2022-01-07 + - Fix some misalignment bugs when reading metadata. + +v0.12.33 - 2021-12-22 + - Fix a bug with seeking when the seek table does not start at PCM frame 0. + +v0.12.32 - 2021-12-11 + - Fix a warning with Clang. + +v0.12.31 - 2021-08-16 + - Silence some warnings. + +v0.12.30 - 2021-07-31 + - Fix platform detection for ARM64. + +v0.12.29 - 2021-04-02 + - Fix a bug where the running PCM frame index is set to an invalid value when over-seeking. + - Fix a decoding error due to an incorrect validation check. + +v0.12.28 - 2021-02-21 + - Fix a warning due to referencing _MSC_VER when it is undefined. + +v0.12.27 - 2021-01-31 + - Fix a static analysis warning. + +v0.12.26 - 2021-01-17 + - Fix a compilation warning due to _BSD_SOURCE being deprecated. + +v0.12.25 - 2020-12-26 + - Update documentation. + +v0.12.24 - 2020-11-29 + - Fix ARM64/NEON detection when compiling with MSVC. + +v0.12.23 - 2020-11-21 + - Fix compilation with OpenWatcom. + +v0.12.22 - 2020-11-01 + - Fix an error with the previous release. + +v0.12.21 - 2020-11-01 + - Fix a possible deadlock when seeking. + - Improve compiler support for older versions of GCC. + +v0.12.20 - 2020-09-08 + - Fix a compilation error on older compilers. + +v0.12.19 - 2020-08-30 + - Fix a bug due to an undefined 32-bit shift. + +v0.12.18 - 2020-08-14 + - Fix a crash when compiling with clang-cl. + +v0.12.17 - 2020-08-02 + - Simplify sized types. + +v0.12.16 - 2020-07-25 + - Fix a compilation warning. + +v0.12.15 - 2020-07-06 + - Check for negative LPC shifts and return an error. + +v0.12.14 - 2020-06-23 + - Add include guard for the implementation section. + +v0.12.13 - 2020-05-16 + - Add compile-time and run-time version querying. + - DRFLAC_VERSION_MINOR + - DRFLAC_VERSION_MAJOR + - DRFLAC_VERSION_REVISION + - DRFLAC_VERSION_STRING + - drflac_version() + - drflac_version_string() + +v0.12.12 - 2020-04-30 + - Fix compilation errors with VC6. + +v0.12.11 - 2020-04-19 + - Fix some pedantic warnings. + - Fix some undefined behaviour warnings. + +v0.12.10 - 2020-04-10 + - Fix some bugs when trying to seek with an invalid seek table. + +v0.12.9 - 2020-04-05 + - Fix warnings. + +v0.12.8 - 2020-04-04 + - Add drflac_open_file_w() and drflac_open_file_with_metadata_w(). + - Fix some static analysis warnings. + - Minor documentation updates. + +v0.12.7 - 2020-03-14 + - Fix compilation errors with VC6. + +v0.12.6 - 2020-03-07 + - Fix compilation error with Visual Studio .NET 2003. + +v0.12.5 - 2020-01-30 + - Silence some static analysis warnings. + +v0.12.4 - 2020-01-29 + - Silence some static analysis warnings. + +v0.12.3 - 2019-12-02 + - Fix some warnings when compiling with GCC and the -Og flag. + - Fix a crash in out-of-memory situations. + - Fix potential integer overflow bug. + - Fix some static analysis warnings. + - Fix a possible crash when using custom memory allocators without a custom realloc() implementation. + - Fix a bug with binary search seeking where the bits per sample is not a multiple of 8. + +v0.12.2 - 2019-10-07 + - Internal code clean up. + +v0.12.1 - 2019-09-29 + - Fix some Clang Static Analyzer warnings. + - Fix an unused variable warning. + +v0.12.0 - 2019-09-23 + - API CHANGE: Add support for user defined memory allocation routines. This system allows the program to specify their own memory allocation + routines with a user data pointer for client-specific contextual data. This adds an extra parameter to the end of the following APIs: + - drflac_open() + - drflac_open_relaxed() + - drflac_open_with_metadata() + - drflac_open_with_metadata_relaxed() + - drflac_open_file() + - drflac_open_file_with_metadata() + - drflac_open_memory() + - drflac_open_memory_with_metadata() + - drflac_open_and_read_pcm_frames_s32() + - drflac_open_and_read_pcm_frames_s16() + - drflac_open_and_read_pcm_frames_f32() + - drflac_open_file_and_read_pcm_frames_s32() + - drflac_open_file_and_read_pcm_frames_s16() + - drflac_open_file_and_read_pcm_frames_f32() + - drflac_open_memory_and_read_pcm_frames_s32() + - drflac_open_memory_and_read_pcm_frames_s16() + - drflac_open_memory_and_read_pcm_frames_f32() + Set this extra parameter to NULL to use defaults which is the same as the previous behaviour. Setting this NULL will use + DRFLAC_MALLOC, DRFLAC_REALLOC and DRFLAC_FREE. + - Remove deprecated APIs: + - drflac_read_s32() + - drflac_read_s16() + - drflac_read_f32() + - drflac_seek_to_sample() + - drflac_open_and_decode_s32() + - drflac_open_and_decode_s16() + - drflac_open_and_decode_f32() + - drflac_open_and_decode_file_s32() + - drflac_open_and_decode_file_s16() + - drflac_open_and_decode_file_f32() + - drflac_open_and_decode_memory_s32() + - drflac_open_and_decode_memory_s16() + - drflac_open_and_decode_memory_f32() + - Remove drflac.totalSampleCount which is now replaced with drflac.totalPCMFrameCount. You can emulate drflac.totalSampleCount + by doing pFlac->totalPCMFrameCount*pFlac->channels. + - Rename drflac.currentFrame to drflac.currentFLACFrame to remove ambiguity with PCM frames. + - Fix errors when seeking to the end of a stream. + - Optimizations to seeking. + - SSE improvements and optimizations. + - ARM NEON optimizations. + - Optimizations to drflac_read_pcm_frames_s16(). + - Optimizations to drflac_read_pcm_frames_s32(). + +v0.11.10 - 2019-06-26 + - Fix a compiler error. + +v0.11.9 - 2019-06-16 + - Silence some ThreadSanitizer warnings. + +v0.11.8 - 2019-05-21 + - Fix warnings. + +v0.11.7 - 2019-05-06 + - C89 fixes. + +v0.11.6 - 2019-05-05 + - Add support for C89. + - Fix a compiler warning when CRC is disabled. + - Change license to choice of public domain or MIT-0. + +v0.11.5 - 2019-04-19 + - Fix a compiler error with GCC. + +v0.11.4 - 2019-04-17 + - Fix some warnings with GCC when compiling with -std=c99. + +v0.11.3 - 2019-04-07 + - Silence warnings with GCC. + +v0.11.2 - 2019-03-10 + - Fix a warning. + +v0.11.1 - 2019-02-17 + - Fix a potential bug with seeking. + +v0.11.0 - 2018-12-16 + - API CHANGE: Deprecated drflac_read_s32(), drflac_read_s16() and drflac_read_f32() and replaced them with + drflac_read_pcm_frames_s32(), drflac_read_pcm_frames_s16() and drflac_read_pcm_frames_f32(). The new APIs take + and return PCM frame counts instead of sample counts. To upgrade you will need to change the input count by + dividing it by the channel count, and then do the same with the return value. + - API_CHANGE: Deprecated drflac_seek_to_sample() and replaced with drflac_seek_to_pcm_frame(). Same rules as + the changes to drflac_read_*() apply. + - API CHANGE: Deprecated drflac_open_and_decode_*() and replaced with drflac_open_*_and_read_*(). Same rules as + the changes to drflac_read_*() apply. + - Optimizations. + +v0.10.0 - 2018-09-11 + - Remove the DR_FLAC_NO_WIN32_IO option and the Win32 file IO functionality. If you need to use Win32 file IO you + need to do it yourself via the callback API. + - Fix the clang build. + - Fix undefined behavior. + - Fix errors with CUESHEET metdata blocks. + - Add an API for iterating over each cuesheet track in the CUESHEET metadata block. This works the same way as the + Vorbis comment API. + - Other miscellaneous bug fixes, mostly relating to invalid FLAC streams. + - Minor optimizations. + +v0.9.11 - 2018-08-29 + - Fix a bug with sample reconstruction. + +v0.9.10 - 2018-08-07 + - Improve 64-bit detection. + +v0.9.9 - 2018-08-05 + - Fix C++ build on older versions of GCC. + +v0.9.8 - 2018-07-24 + - Fix compilation errors. + +v0.9.7 - 2018-07-05 + - Fix a warning. + +v0.9.6 - 2018-06-29 + - Fix some typos. + +v0.9.5 - 2018-06-23 + - Fix some warnings. + +v0.9.4 - 2018-06-14 + - Optimizations to seeking. + - Clean up. + +v0.9.3 - 2018-05-22 + - Bug fix. + +v0.9.2 - 2018-05-12 + - Fix a compilation error due to a missing break statement. + +v0.9.1 - 2018-04-29 + - Fix compilation error with Clang. + +v0.9 - 2018-04-24 + - Fix Clang build. + - Start using major.minor.revision versioning. + +v0.8g - 2018-04-19 + - Fix build on non-x86/x64 architectures. + +v0.8f - 2018-02-02 + - Stop pretending to support changing rate/channels mid stream. + +v0.8e - 2018-02-01 + - Fix a crash when the block size of a frame is larger than the maximum block size defined by the FLAC stream. + - Fix a crash the the Rice partition order is invalid. + +v0.8d - 2017-09-22 + - Add support for decoding streams with ID3 tags. ID3 tags are just skipped. + +v0.8c - 2017-09-07 + - Fix warning on non-x86/x64 architectures. + +v0.8b - 2017-08-19 + - Fix build on non-x86/x64 architectures. + +v0.8a - 2017-08-13 + - A small optimization for the Clang build. + +v0.8 - 2017-08-12 + - API CHANGE: Rename dr_* types to drflac_*. + - Optimizations. This brings dr_flac back to about the same class of efficiency as the reference implementation. + - Add support for custom implementations of malloc(), realloc(), etc. + - Add CRC checking to Ogg encapsulated streams. + - Fix VC++ 6 build. This is only for the C++ compiler. The C compiler is not currently supported. + - Bug fixes. + +v0.7 - 2017-07-23 + - Add support for opening a stream without a header block. To do this, use drflac_open_relaxed() / drflac_open_with_metadata_relaxed(). + +v0.6 - 2017-07-22 + - Add support for recovering from invalid frames. With this change, dr_flac will simply skip over invalid frames as if they + never existed. Frames are checked against their sync code, the CRC-8 of the frame header and the CRC-16 of the whole frame. + +v0.5 - 2017-07-16 + - Fix typos. + - Change drflac_bool* types to unsigned. + - Add CRC checking. This makes dr_flac slower, but can be disabled with #define DR_FLAC_NO_CRC. + +v0.4f - 2017-03-10 + - Fix a couple of bugs with the bitstreaming code. + +v0.4e - 2017-02-17 + - Fix some warnings. + +v0.4d - 2016-12-26 + - Add support for 32-bit floating-point PCM decoding. + - Use drflac_int* and drflac_uint* sized types to improve compiler support. + - Minor improvements to documentation. + +v0.4c - 2016-12-26 + - Add support for signed 16-bit integer PCM decoding. + +v0.4b - 2016-10-23 + - A minor change to drflac_bool8 and drflac_bool32 types. + +v0.4a - 2016-10-11 + - Rename drBool32 to drflac_bool32 for styling consistency. + +v0.4 - 2016-09-29 + - API/ABI CHANGE: Use fixed size 32-bit booleans instead of the built-in bool type. + - API CHANGE: Rename drflac_open_and_decode*() to drflac_open_and_decode*_s32(). + - API CHANGE: Swap the order of "channels" and "sampleRate" parameters in drflac_open_and_decode*(). Rationale for this is to + keep it consistent with drflac_audio. + +v0.3f - 2016-09-21 + - Fix a warning with GCC. + +v0.3e - 2016-09-18 + - Fixed a bug where GCC 4.3+ was not getting properly identified. + - Fixed a few typos. + - Changed date formats to ISO 8601 (YYYY-MM-DD). + +v0.3d - 2016-06-11 + - Minor clean up. + +v0.3c - 2016-05-28 + - Fixed compilation error. + +v0.3b - 2016-05-16 + - Fixed Linux/GCC build. + - Updated documentation. + +v0.3a - 2016-05-15 + - Minor fixes to documentation. + +v0.3 - 2016-05-11 + - Optimizations. Now at about parity with the reference implementation on 32-bit builds. + - Lots of clean up. + +v0.2b - 2016-05-10 + - Bug fixes. + +v0.2a - 2016-05-10 + - Made drflac_open_and_decode() more robust. + - Removed an unused debugging variable + +v0.2 - 2016-05-09 + - Added support for Ogg encapsulation. + - API CHANGE. Have the onSeek callback take a third argument which specifies whether or not the seek + should be relative to the start or the current position. Also changes the seeking rules such that + seeking offsets will never be negative. + - Have drflac_open_and_decode() fail gracefully if the stream has an unknown total sample count. + +v0.1b - 2016-05-07 + - Properly close the file handle in drflac_open_file() and family when the decoder fails to initialize. + - Removed a stale comment. + +v0.1a - 2016-05-05 + - Minor formatting changes. + - Fixed a warning on the GCC build. + +v0.1 - 2016-05-03 + - Initial versioned release. +*/ + +/* +This software is available as a choice of the following licenses. Choose +whichever you prefer. + +=============================================================================== +ALTERNATIVE 1 - Public Domain (www.unlicense.org) +=============================================================================== +This is free and unencumbered software released into the public domain. + +Anyone is free to copy, modify, publish, use, compile, sell, or distribute this +software, either in source code form or as a compiled binary, for any purpose, +commercial or non-commercial, and by any means. + +In jurisdictions that recognize copyright laws, the author or authors of this +software dedicate any and all copyright interest in the software to the public +domain. We make this dedication for the benefit of the public at large and to +the detriment of our heirs and successors. We intend this dedication to be an +overt act of relinquishment in perpetuity of all present and future rights to +this software under copyright law. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN +ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION +WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + +For more information, please refer to + +=============================================================================== +ALTERNATIVE 2 - MIT No Attribution +=============================================================================== +Copyright 2023 David Reid + +Permission is hereby granted, free of charge, to any person obtaining a copy of +this software and associated documentation files (the "Software"), to deal in +the Software without restriction, including without limitation the rights to +use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies +of the Software, and to permit persons to whom the Software is furnished to do +so. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +SOFTWARE. +*/ diff --git a/src/include/dr_libs/dr_mp3.h b/src/include/dr_libs/dr_mp3.h new file mode 100644 index 000000000..a631ae40f --- /dev/null +++ b/src/include/dr_libs/dr_mp3.h @@ -0,0 +1,5430 @@ +/* +MP3 audio decoder. Choice of public domain or MIT-0. See license statements at the end of this file. +dr_mp3 - v0.7.4 - TBD + +David Reid - mackron@gmail.com + +GitHub: https://github.com/mackron/dr_libs + +Based on minimp3 (https://github.com/lieff/minimp3) which is where the real work was done. See the bottom of this file for differences between minimp3 and dr_mp3. +*/ + +/* +Introduction +============= +dr_mp3 is a single file library. To use it, do something like the following in one .c file. + + ```c + #define DR_MP3_IMPLEMENTATION + #include "dr_mp3.h" + ``` + +You can then #include this file in other parts of the program as you would with any other header file. To decode audio data, do something like the following: + + ```c + drmp3 mp3; + if (!drmp3_init_file(&mp3, "MySong.mp3", NULL)) { + // Failed to open file + } + + ... + + drmp3_uint64 framesRead = drmp3_read_pcm_frames_f32(pMP3, framesToRead, pFrames); + ``` + +The drmp3 object is transparent so you can get access to the channel count and sample rate like so: + + ``` + drmp3_uint32 channels = mp3.channels; + drmp3_uint32 sampleRate = mp3.sampleRate; + ``` + +The example above initializes a decoder from a file, but you can also initialize it from a block of memory and read and seek callbacks with +`drmp3_init_memory()` and `drmp3_init()` respectively. + +You do not need to do any annoying memory management when reading PCM frames - this is all managed internally. You can request any number of PCM frames in each +call to `drmp3_read_pcm_frames_f32()` and it will return as many PCM frames as it can, up to the requested amount. + +You can also decode an entire file in one go with `drmp3_open_and_read_pcm_frames_f32()`, `drmp3_open_memory_and_read_pcm_frames_f32()` and +`drmp3_open_file_and_read_pcm_frames_f32()`. + + +Build Options +============= +#define these options before including this file. + +#define DR_MP3_NO_STDIO + Disable drmp3_init_file(), etc. + +#define DR_MP3_NO_SIMD + Disable SIMD optimizations. +*/ + +#ifndef dr_mp3_h +#define dr_mp3_h + +#ifdef __cplusplus +extern "C" { +#endif + +#define DRMP3_STRINGIFY(x) #x +#define DRMP3_XSTRINGIFY(x) DRMP3_STRINGIFY(x) + +#define DRMP3_VERSION_MAJOR 0 +#define DRMP3_VERSION_MINOR 7 +#define DRMP3_VERSION_REVISION 4 +#define DRMP3_VERSION_STRING DRMP3_XSTRINGIFY(DRMP3_VERSION_MAJOR) "." DRMP3_XSTRINGIFY(DRMP3_VERSION_MINOR) "." DRMP3_XSTRINGIFY(DRMP3_VERSION_REVISION) + +#include /* For size_t. */ + +/* Sized Types */ +typedef signed char drmp3_int8; +typedef unsigned char drmp3_uint8; +typedef signed short drmp3_int16; +typedef unsigned short drmp3_uint16; +typedef signed int drmp3_int32; +typedef unsigned int drmp3_uint32; +#if defined(_MSC_VER) && !defined(__clang__) + typedef signed __int64 drmp3_int64; + typedef unsigned __int64 drmp3_uint64; +#else + #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic push + #pragma GCC diagnostic ignored "-Wlong-long" + #if defined(__clang__) + #pragma GCC diagnostic ignored "-Wc++11-long-long" + #endif + #endif + typedef signed long long drmp3_int64; + typedef unsigned long long drmp3_uint64; + #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic pop + #endif +#endif +#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined (_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(_M_ARM64EC) || defined(__powerpc64__) + typedef drmp3_uint64 drmp3_uintptr; +#else + typedef drmp3_uint32 drmp3_uintptr; +#endif +typedef drmp3_uint8 drmp3_bool8; +typedef drmp3_uint32 drmp3_bool32; +#define DRMP3_TRUE 1 +#define DRMP3_FALSE 0 + +/* Weird shifting syntax is for VC6 compatibility. */ +#define DRMP3_UINT64_MAX (((drmp3_uint64)0xFFFFFFFF << 32) | (drmp3_uint64)0xFFFFFFFF) +/* End Sized Types */ + +/* Decorations */ +#if !defined(DRMP3_API) + #if defined(DRMP3_DLL) + #if defined(_WIN32) + #define DRMP3_DLL_IMPORT __declspec(dllimport) + #define DRMP3_DLL_EXPORT __declspec(dllexport) + #define DRMP3_DLL_PRIVATE static + #else + #if defined(__GNUC__) && __GNUC__ >= 4 + #define DRMP3_DLL_IMPORT __attribute__((visibility("default"))) + #define DRMP3_DLL_EXPORT __attribute__((visibility("default"))) + #define DRMP3_DLL_PRIVATE __attribute__((visibility("hidden"))) + #else + #define DRMP3_DLL_IMPORT + #define DRMP3_DLL_EXPORT + #define DRMP3_DLL_PRIVATE static + #endif + #endif + + #if defined(DR_MP3_IMPLEMENTATION) + #define DRMP3_API DRMP3_DLL_EXPORT + #else + #define DRMP3_API DRMP3_DLL_IMPORT + #endif + #define DRMP3_PRIVATE DRMP3_DLL_PRIVATE + #else + #define DRMP3_API extern + #define DRMP3_PRIVATE static + #endif +#endif +/* End Decorations */ + +/* Result Codes */ +typedef drmp3_int32 drmp3_result; +#define DRMP3_SUCCESS 0 +#define DRMP3_ERROR -1 /* A generic error. */ +#define DRMP3_INVALID_ARGS -2 +#define DRMP3_INVALID_OPERATION -3 +#define DRMP3_OUT_OF_MEMORY -4 +#define DRMP3_OUT_OF_RANGE -5 +#define DRMP3_ACCESS_DENIED -6 +#define DRMP3_DOES_NOT_EXIST -7 +#define DRMP3_ALREADY_EXISTS -8 +#define DRMP3_TOO_MANY_OPEN_FILES -9 +#define DRMP3_INVALID_FILE -10 +#define DRMP3_TOO_BIG -11 +#define DRMP3_PATH_TOO_LONG -12 +#define DRMP3_NAME_TOO_LONG -13 +#define DRMP3_NOT_DIRECTORY -14 +#define DRMP3_IS_DIRECTORY -15 +#define DRMP3_DIRECTORY_NOT_EMPTY -16 +#define DRMP3_END_OF_FILE -17 +#define DRMP3_NO_SPACE -18 +#define DRMP3_BUSY -19 +#define DRMP3_IO_ERROR -20 +#define DRMP3_INTERRUPT -21 +#define DRMP3_UNAVAILABLE -22 +#define DRMP3_ALREADY_IN_USE -23 +#define DRMP3_BAD_ADDRESS -24 +#define DRMP3_BAD_SEEK -25 +#define DRMP3_BAD_PIPE -26 +#define DRMP3_DEADLOCK -27 +#define DRMP3_TOO_MANY_LINKS -28 +#define DRMP3_NOT_IMPLEMENTED -29 +#define DRMP3_NO_MESSAGE -30 +#define DRMP3_BAD_MESSAGE -31 +#define DRMP3_NO_DATA_AVAILABLE -32 +#define DRMP3_INVALID_DATA -33 +#define DRMP3_TIMEOUT -34 +#define DRMP3_NO_NETWORK -35 +#define DRMP3_NOT_UNIQUE -36 +#define DRMP3_NOT_SOCKET -37 +#define DRMP3_NO_ADDRESS -38 +#define DRMP3_BAD_PROTOCOL -39 +#define DRMP3_PROTOCOL_UNAVAILABLE -40 +#define DRMP3_PROTOCOL_NOT_SUPPORTED -41 +#define DRMP3_PROTOCOL_FAMILY_NOT_SUPPORTED -42 +#define DRMP3_ADDRESS_FAMILY_NOT_SUPPORTED -43 +#define DRMP3_SOCKET_NOT_SUPPORTED -44 +#define DRMP3_CONNECTION_RESET -45 +#define DRMP3_ALREADY_CONNECTED -46 +#define DRMP3_NOT_CONNECTED -47 +#define DRMP3_CONNECTION_REFUSED -48 +#define DRMP3_NO_HOST -49 +#define DRMP3_IN_PROGRESS -50 +#define DRMP3_CANCELLED -51 +#define DRMP3_MEMORY_ALREADY_MAPPED -52 +#define DRMP3_AT_END -53 +/* End Result Codes */ + +#define DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME 1152 +#define DRMP3_MAX_SAMPLES_PER_FRAME (DRMP3_MAX_PCM_FRAMES_PER_MP3_FRAME*2) + +/* Inline */ +#ifdef _MSC_VER + #define DRMP3_INLINE __forceinline +#elif defined(__GNUC__) + /* + I've had a bug report where GCC is emitting warnings about functions possibly not being inlineable. This warning happens when + the __attribute__((always_inline)) attribute is defined without an "inline" statement. I think therefore there must be some + case where "__inline__" is not always defined, thus the compiler emitting these warnings. When using -std=c89 or -ansi on the + command line, we cannot use the "inline" keyword and instead need to use "__inline__". In an attempt to work around this issue + I am using "__inline__" only when we're compiling in strict ANSI mode. + */ + #if defined(__STRICT_ANSI__) + #define DRMP3_GNUC_INLINE_HINT __inline__ + #else + #define DRMP3_GNUC_INLINE_HINT inline + #endif + + #if (__GNUC__ > 3 || (__GNUC__ == 3 && __GNUC_MINOR__ >= 2)) || defined(__clang__) + #define DRMP3_INLINE DRMP3_GNUC_INLINE_HINT __attribute__((always_inline)) + #else + #define DRMP3_INLINE DRMP3_GNUC_INLINE_HINT + #endif +#elif defined(__WATCOMC__) + #define DRMP3_INLINE __inline +#else + #define DRMP3_INLINE +#endif +/* End Inline */ + + +DRMP3_API void drmp3_version(drmp3_uint32* pMajor, drmp3_uint32* pMinor, drmp3_uint32* pRevision); +DRMP3_API const char* drmp3_version_string(void); + + +/* Allocation Callbacks */ +typedef struct +{ + void* pUserData; + void* (* onMalloc)(size_t sz, void* pUserData); + void* (* onRealloc)(void* p, size_t sz, void* pUserData); + void (* onFree)(void* p, void* pUserData); +} drmp3_allocation_callbacks; +/* End Allocation Callbacks */ + + +/* +Low Level Push API +================== +*/ +#define DRMP3_MAX_BITRESERVOIR_BYTES 511 +#define DRMP3_MAX_FREE_FORMAT_FRAME_SIZE 2304 /* more than ISO spec's */ +#define DRMP3_MAX_L3_FRAME_PAYLOAD_BYTES DRMP3_MAX_FREE_FORMAT_FRAME_SIZE /* MUST be >= 320000/8/32000*1152 = 1440 */ + +typedef struct +{ + int frame_bytes, channels, sample_rate, layer, bitrate_kbps; +} drmp3dec_frame_info; + +typedef struct +{ + const drmp3_uint8 *buf; + int pos, limit; +} drmp3_bs; + +typedef struct +{ + const drmp3_uint8 *sfbtab; + drmp3_uint16 part_23_length, big_values, scalefac_compress; + drmp3_uint8 global_gain, block_type, mixed_block_flag, n_long_sfb, n_short_sfb; + drmp3_uint8 table_select[3], region_count[3], subblock_gain[3]; + drmp3_uint8 preflag, scalefac_scale, count1_table, scfsi; +} drmp3_L3_gr_info; + +typedef struct +{ + drmp3_bs bs; + drmp3_uint8 maindata[DRMP3_MAX_BITRESERVOIR_BYTES + DRMP3_MAX_L3_FRAME_PAYLOAD_BYTES]; + drmp3_L3_gr_info gr_info[4]; + float grbuf[2][576], scf[40], syn[18 + 15][2*32]; + drmp3_uint8 ist_pos[2][39]; +} drmp3dec_scratch; + +typedef struct +{ + float mdct_overlap[2][9*32], qmf_state[15*2*32]; + int reserv, free_format_bytes; + drmp3_uint8 header[4], reserv_buf[511]; + drmp3dec_scratch scratch; +} drmp3dec; + +/* Initializes a low level decoder. */ +DRMP3_API void drmp3dec_init(drmp3dec *dec); + +/* Reads a frame from a low level decoder. */ +DRMP3_API int drmp3dec_decode_frame(drmp3dec *dec, const drmp3_uint8 *mp3, int mp3_bytes, void *pcm, drmp3dec_frame_info *info); + +/* Helper for converting between f32 and s16. */ +DRMP3_API void drmp3dec_f32_to_s16(const float *in, drmp3_int16 *out, size_t num_samples); + + + +/* +Main API (Pull API) +=================== +*/ +typedef enum +{ + DRMP3_SEEK_SET, + DRMP3_SEEK_CUR, + DRMP3_SEEK_END +} drmp3_seek_origin; + +typedef struct +{ + drmp3_uint64 seekPosInBytes; /* Points to the first byte of an MP3 frame. */ + drmp3_uint64 pcmFrameIndex; /* The index of the PCM frame this seek point targets. */ + drmp3_uint16 mp3FramesToDiscard; /* The number of whole MP3 frames to be discarded before pcmFramesToDiscard. */ + drmp3_uint16 pcmFramesToDiscard; /* The number of leading samples to read and discard. These are discarded after mp3FramesToDiscard. */ +} drmp3_seek_point; + +typedef enum +{ + DRMP3_METADATA_TYPE_ID3V1, + DRMP3_METADATA_TYPE_ID3V2, + DRMP3_METADATA_TYPE_APE, + DRMP3_METADATA_TYPE_XING, + DRMP3_METADATA_TYPE_VBRI +} drmp3_metadata_type; + +typedef struct +{ + drmp3_metadata_type type; + const void* pRawData; /* A pointer to the raw data. */ + size_t rawDataSize; +} drmp3_metadata; + + +/* +Callback for when data is read. Return value is the number of bytes actually read. + +pUserData [in] The user data that was passed to drmp3_init(), and family. +pBufferOut [out] The output buffer. +bytesToRead [in] The number of bytes to read. + +Returns the number of bytes actually read. + +A return value of less than bytesToRead indicates the end of the stream. Do _not_ return from this callback until +either the entire bytesToRead is filled or you have reached the end of the stream. +*/ +typedef size_t (* drmp3_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead); + +/* +Callback for when data needs to be seeked. + +pUserData [in] The user data that was passed to drmp3_init(), and family. +offset [in] The number of bytes to move, relative to the origin. Can be negative. +origin [in] The origin of the seek. + +Returns whether or not the seek was successful. +*/ +typedef drmp3_bool32 (* drmp3_seek_proc)(void* pUserData, int offset, drmp3_seek_origin origin); + +/* +Callback for retrieving the current cursor position. + +pUserData [in] The user data that was passed to drmp3_init(), and family. +pCursor [out] The cursor position in bytes from the start of the stream. + +Returns whether or not the cursor position was successfully retrieved. +*/ +typedef drmp3_bool32 (* drmp3_tell_proc)(void* pUserData, drmp3_int64* pCursor); + + +/* +Callback for when metadata is read. + +Only the raw data is provided. The client is responsible for parsing the contents of the data themsevles. +*/ +typedef void (* drmp3_meta_proc)(void* pUserData, const drmp3_metadata* pMetadata); + + +typedef struct +{ + drmp3_uint32 channels; + drmp3_uint32 sampleRate; +} drmp3_config; + +typedef struct +{ + drmp3dec decoder; + drmp3_uint32 channels; + drmp3_uint32 sampleRate; + drmp3_read_proc onRead; + drmp3_seek_proc onSeek; + drmp3_meta_proc onMeta; + void* pUserData; + void* pUserDataMeta; + drmp3_allocation_callbacks allocationCallbacks; + drmp3_uint32 mp3FrameChannels; /* The number of channels in the currently loaded MP3 frame. Internal use only. */ + drmp3_uint32 mp3FrameSampleRate; /* The sample rate of the currently loaded MP3 frame. Internal use only. */ + drmp3_uint32 pcmFramesConsumedInMP3Frame; + drmp3_uint32 pcmFramesRemainingInMP3Frame; + drmp3_uint8 pcmFrames[sizeof(float)*DRMP3_MAX_SAMPLES_PER_FRAME]; /* <-- Multipled by sizeof(float) to ensure there's enough room for DR_MP3_FLOAT_OUTPUT. */ + drmp3_uint64 currentPCMFrame; /* The current PCM frame, globally. */ + drmp3_uint64 streamCursor; /* The current byte the decoder is sitting on in the raw stream. */ + drmp3_uint64 streamLength; /* The length of the stream in bytes. dr_mp3 will not read beyond this. If a ID3v1 or APE tag is present, this will be set to the first byte of the tag. */ + drmp3_uint64 streamStartOffset; /* The offset of the start of the MP3 data. This is used for skipping ID3v2 and VBR tags. */ + drmp3_seek_point* pSeekPoints; /* NULL by default. Set with drmp3_bind_seek_table(). Memory is owned by the client. dr_mp3 will never attempt to free this pointer. */ + drmp3_uint32 seekPointCount; /* The number of items in pSeekPoints. When set to 0 assumes to no seek table. Defaults to zero. */ + drmp3_uint32 delayInPCMFrames; + drmp3_uint32 paddingInPCMFrames; + drmp3_uint64 totalPCMFrameCount; /* Set to DRMP3_UINT64_MAX if the length is unknown. Includes delay and padding. */ + drmp3_bool32 isVBR; + drmp3_bool32 isCBR; + size_t dataSize; + size_t dataCapacity; + size_t dataConsumed; + drmp3_uint8* pData; + drmp3_bool32 atEnd; + struct + { + const drmp3_uint8* pData; + size_t dataSize; + size_t currentReadPos; + } memory; /* Only used for decoders that were opened against a block of memory. */ +} drmp3; + +/* +Initializes an MP3 decoder. + +onRead [in] The function to call when data needs to be read from the client. +onSeek [in] The function to call when the read position of the client data needs to move. +onTell [in] The function to call when the read position of the client data needs to be retrieved. +pUserData [in, optional] A pointer to application defined data that will be passed to onRead and onSeek. + +Returns true if successful; false otherwise. + +Close the loader with drmp3_uninit(). + +See also: drmp3_init_file(), drmp3_init_memory(), drmp3_uninit() +*/ +DRMP3_API drmp3_bool32 drmp3_init(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, drmp3_tell_proc onTell, drmp3_meta_proc onMeta, void* pUserData, const drmp3_allocation_callbacks* pAllocationCallbacks); + +/* +Initializes an MP3 decoder from a block of memory. + +This does not create a copy of the data. It is up to the application to ensure the buffer remains valid for +the lifetime of the drmp3 object. + +The buffer should contain the contents of the entire MP3 file. +*/ +DRMP3_API drmp3_bool32 drmp3_init_memory_with_metadata(drmp3* pMP3, const void* pData, size_t dataSize, drmp3_meta_proc onMeta, void* pUserDataMeta, const drmp3_allocation_callbacks* pAllocationCallbacks); +DRMP3_API drmp3_bool32 drmp3_init_memory(drmp3* pMP3, const void* pData, size_t dataSize, const drmp3_allocation_callbacks* pAllocationCallbacks); + +#ifndef DR_MP3_NO_STDIO +/* +Initializes an MP3 decoder from a file. + +This holds the internal FILE object until drmp3_uninit() is called. Keep this in mind if you're caching drmp3 +objects because the operating system may restrict the number of file handles an application can have open at +any given time. +*/ +DRMP3_API drmp3_bool32 drmp3_init_file_with_metadata(drmp3* pMP3, const char* pFilePath, drmp3_meta_proc onMeta, void* pUserDataMeta, const drmp3_allocation_callbacks* pAllocationCallbacks); +DRMP3_API drmp3_bool32 drmp3_init_file_with_metadata_w(drmp3* pMP3, const wchar_t* pFilePath, drmp3_meta_proc onMeta, void* pUserDataMeta, const drmp3_allocation_callbacks* pAllocationCallbacks); + +DRMP3_API drmp3_bool32 drmp3_init_file(drmp3* pMP3, const char* pFilePath, const drmp3_allocation_callbacks* pAllocationCallbacks); +DRMP3_API drmp3_bool32 drmp3_init_file_w(drmp3* pMP3, const wchar_t* pFilePath, const drmp3_allocation_callbacks* pAllocationCallbacks); +#endif + +/* +Uninitializes an MP3 decoder. +*/ +DRMP3_API void drmp3_uninit(drmp3* pMP3); + +/* +Reads PCM frames as interleaved 32-bit IEEE floating point PCM. + +Note that framesToRead specifies the number of PCM frames to read, _not_ the number of MP3 frames. +*/ +DRMP3_API drmp3_uint64 drmp3_read_pcm_frames_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBufferOut); + +/* +Reads PCM frames as interleaved signed 16-bit integer PCM. + +Note that framesToRead specifies the number of PCM frames to read, _not_ the number of MP3 frames. +*/ +DRMP3_API drmp3_uint64 drmp3_read_pcm_frames_s16(drmp3* pMP3, drmp3_uint64 framesToRead, drmp3_int16* pBufferOut); + +/* +Seeks to a specific frame. + +Note that this is _not_ an MP3 frame, but rather a PCM frame. +*/ +DRMP3_API drmp3_bool32 drmp3_seek_to_pcm_frame(drmp3* pMP3, drmp3_uint64 frameIndex); + +/* +Calculates the total number of PCM frames in the MP3 stream. Cannot be used for infinite streams such as internet +radio. Runs in linear time. Returns 0 on error. +*/ +DRMP3_API drmp3_uint64 drmp3_get_pcm_frame_count(drmp3* pMP3); + +/* +Calculates the total number of MP3 frames in the MP3 stream. Cannot be used for infinite streams such as internet +radio. Runs in linear time. Returns 0 on error. +*/ +DRMP3_API drmp3_uint64 drmp3_get_mp3_frame_count(drmp3* pMP3); + +/* +Calculates the total number of MP3 and PCM frames in the MP3 stream. Cannot be used for infinite streams such as internet +radio. Runs in linear time. Returns 0 on error. + +This is equivalent to calling drmp3_get_mp3_frame_count() and drmp3_get_pcm_frame_count() except that it's more efficient. +*/ +DRMP3_API drmp3_bool32 drmp3_get_mp3_and_pcm_frame_count(drmp3* pMP3, drmp3_uint64* pMP3FrameCount, drmp3_uint64* pPCMFrameCount); + +/* +Calculates the seekpoints based on PCM frames. This is slow. + +pSeekpoint count is a pointer to a uint32 containing the seekpoint count. On input it contains the desired count. +On output it contains the actual count. The reason for this design is that the client may request too many +seekpoints, in which case dr_mp3 will return a corrected count. + +Note that seektable seeking is not quite sample exact when the MP3 stream contains inconsistent sample rates. +*/ +DRMP3_API drmp3_bool32 drmp3_calculate_seek_points(drmp3* pMP3, drmp3_uint32* pSeekPointCount, drmp3_seek_point* pSeekPoints); + +/* +Binds a seek table to the decoder. + +This does _not_ make a copy of pSeekPoints - it only references it. It is up to the application to ensure this +remains valid while it is bound to the decoder. + +Use drmp3_calculate_seek_points() to calculate the seek points. +*/ +DRMP3_API drmp3_bool32 drmp3_bind_seek_table(drmp3* pMP3, drmp3_uint32 seekPointCount, drmp3_seek_point* pSeekPoints); + + +/* +Opens an decodes an entire MP3 stream as a single operation. + +On output pConfig will receive the channel count and sample rate of the stream. + +Free the returned pointer with drmp3_free(). +*/ +DRMP3_API float* drmp3_open_and_read_pcm_frames_f32(drmp3_read_proc onRead, drmp3_seek_proc onSeek, drmp3_tell_proc onTell, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks); +DRMP3_API drmp3_int16* drmp3_open_and_read_pcm_frames_s16(drmp3_read_proc onRead, drmp3_seek_proc onSeek, drmp3_tell_proc onTell, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks); + +DRMP3_API float* drmp3_open_memory_and_read_pcm_frames_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks); +DRMP3_API drmp3_int16* drmp3_open_memory_and_read_pcm_frames_s16(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks); + +#ifndef DR_MP3_NO_STDIO +DRMP3_API float* drmp3_open_file_and_read_pcm_frames_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks); +DRMP3_API drmp3_int16* drmp3_open_file_and_read_pcm_frames_s16(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks); +#endif + +/* +Allocates a block of memory on the heap. +*/ +DRMP3_API void* drmp3_malloc(size_t sz, const drmp3_allocation_callbacks* pAllocationCallbacks); + +/* +Frees any memory that was allocated by a public drmp3 API. +*/ +DRMP3_API void drmp3_free(void* p, const drmp3_allocation_callbacks* pAllocationCallbacks); + +#ifdef __cplusplus +} +#endif +#endif /* dr_mp3_h */ + + +/************************************************************************************************************************************************************ + ************************************************************************************************************************************************************ + + IMPLEMENTATION + + ************************************************************************************************************************************************************ + ************************************************************************************************************************************************************/ +#if defined(DR_MP3_IMPLEMENTATION) +#ifndef dr_mp3_c +#define dr_mp3_c + +#include +#include +#include /* For INT_MAX */ + +DRMP3_API void drmp3_version(drmp3_uint32* pMajor, drmp3_uint32* pMinor, drmp3_uint32* pRevision) +{ + if (pMajor) { + *pMajor = DRMP3_VERSION_MAJOR; + } + + if (pMinor) { + *pMinor = DRMP3_VERSION_MINOR; + } + + if (pRevision) { + *pRevision = DRMP3_VERSION_REVISION; + } +} + +DRMP3_API const char* drmp3_version_string(void) +{ + return DRMP3_VERSION_STRING; +} + +/* Disable SIMD when compiling with TCC for now. */ +#if defined(__TINYC__) +#define DR_MP3_NO_SIMD +#endif + +#define DRMP3_OFFSET_PTR(p, offset) ((void*)((drmp3_uint8*)(p) + (offset))) + +#ifndef DRMP3_MAX_FRAME_SYNC_MATCHES +#define DRMP3_MAX_FRAME_SYNC_MATCHES 10 +#endif + +#define DRMP3_SHORT_BLOCK_TYPE 2 +#define DRMP3_STOP_BLOCK_TYPE 3 +#define DRMP3_MODE_MONO 3 +#define DRMP3_MODE_JOINT_STEREO 1 +#define DRMP3_HDR_SIZE 4 +#define DRMP3_HDR_IS_MONO(h) (((h[3]) & 0xC0) == 0xC0) +#define DRMP3_HDR_IS_MS_STEREO(h) (((h[3]) & 0xE0) == 0x60) +#define DRMP3_HDR_IS_FREE_FORMAT(h) (((h[2]) & 0xF0) == 0) +#define DRMP3_HDR_IS_CRC(h) (!((h[1]) & 1)) +#define DRMP3_HDR_TEST_PADDING(h) ((h[2]) & 0x2) +#define DRMP3_HDR_TEST_MPEG1(h) ((h[1]) & 0x8) +#define DRMP3_HDR_TEST_NOT_MPEG25(h) ((h[1]) & 0x10) +#define DRMP3_HDR_TEST_I_STEREO(h) ((h[3]) & 0x10) +#define DRMP3_HDR_TEST_MS_STEREO(h) ((h[3]) & 0x20) +#define DRMP3_HDR_GET_STEREO_MODE(h) (((h[3]) >> 6) & 3) +#define DRMP3_HDR_GET_STEREO_MODE_EXT(h) (((h[3]) >> 4) & 3) +#define DRMP3_HDR_GET_LAYER(h) (((h[1]) >> 1) & 3) +#define DRMP3_HDR_GET_BITRATE(h) ((h[2]) >> 4) +#define DRMP3_HDR_GET_SAMPLE_RATE(h) (((h[2]) >> 2) & 3) +#define DRMP3_HDR_GET_MY_SAMPLE_RATE(h) (DRMP3_HDR_GET_SAMPLE_RATE(h) + (((h[1] >> 3) & 1) + ((h[1] >> 4) & 1))*3) +#define DRMP3_HDR_IS_FRAME_576(h) ((h[1] & 14) == 2) +#define DRMP3_HDR_IS_LAYER_1(h) ((h[1] & 6) == 6) + +#define DRMP3_BITS_DEQUANTIZER_OUT -1 +#define DRMP3_MAX_SCF (255 + DRMP3_BITS_DEQUANTIZER_OUT*4 - 210) +#define DRMP3_MAX_SCFI ((DRMP3_MAX_SCF + 3) & ~3) + +#define DRMP3_MIN(a, b) ((a) > (b) ? (b) : (a)) +#define DRMP3_MAX(a, b) ((a) < (b) ? (b) : (a)) + +#if !defined(DR_MP3_NO_SIMD) + +#if !defined(DR_MP3_ONLY_SIMD) && ((defined(_MSC_VER) && _MSC_VER >= 1400) && defined(_M_X64)) || ((defined(__i386) || defined(_M_IX86) || defined(__i386__) || defined(__x86_64__)) && ((defined(_M_IX86_FP) && _M_IX86_FP == 2) || defined(__SSE2__))) +#define DR_MP3_ONLY_SIMD +#endif +#if !defined(DR_MP3_ONLY_SIMD) && (defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64) || defined(_M_ARM64EC)) +#define DR_MP3_ONLY_SIMD +#endif + +#if ((defined(_MSC_VER) && _MSC_VER >= 1400) && defined(_M_X64)) || ((defined(__i386) || defined(_M_IX86) || defined(__i386__) || defined(__x86_64__)) && ((defined(_M_IX86_FP) && _M_IX86_FP == 2) || defined(__SSE2__))) +#if defined(_MSC_VER) +#include +#endif +#include +#define DRMP3_HAVE_SSE 1 +#define DRMP3_HAVE_SIMD 1 +#define DRMP3_VSTORE _mm_storeu_ps +#define DRMP3_VLD _mm_loadu_ps +#define DRMP3_VSET _mm_set1_ps +#define DRMP3_VADD _mm_add_ps +#define DRMP3_VSUB _mm_sub_ps +#define DRMP3_VMUL _mm_mul_ps +#define DRMP3_VMAC(a, x, y) _mm_add_ps(a, _mm_mul_ps(x, y)) +#define DRMP3_VMSB(a, x, y) _mm_sub_ps(a, _mm_mul_ps(x, y)) +#define DRMP3_VMUL_S(x, s) _mm_mul_ps(x, _mm_set1_ps(s)) +#define DRMP3_VREV(x) _mm_shuffle_ps(x, x, _MM_SHUFFLE(0, 1, 2, 3)) +typedef __m128 drmp3_f4; +#if (defined(_MSC_VER) || defined(DR_MP3_ONLY_SIMD)) && !defined(__clang__) +#define drmp3_cpuid __cpuid +#else +static __inline__ __attribute__((always_inline)) void drmp3_cpuid(int CPUInfo[], const int InfoType) +{ +#if defined(__PIC__) + __asm__ __volatile__( +#if defined(__x86_64__) + "push %%rbx\n" + "cpuid\n" + "xchgl %%ebx, %1\n" + "pop %%rbx\n" +#else + "xchgl %%ebx, %1\n" + "cpuid\n" + "xchgl %%ebx, %1\n" +#endif + : "=a" (CPUInfo[0]), "=r" (CPUInfo[1]), "=c" (CPUInfo[2]), "=d" (CPUInfo[3]) + : "a" (InfoType)); +#else + __asm__ __volatile__( + "cpuid" + : "=a" (CPUInfo[0]), "=b" (CPUInfo[1]), "=c" (CPUInfo[2]), "=d" (CPUInfo[3]) + : "a" (InfoType)); +#endif +} +#endif +static int drmp3_have_simd(void) +{ +#ifdef DR_MP3_ONLY_SIMD + return 1; +#else + static int g_have_simd; + int CPUInfo[4]; +#ifdef MINIMP3_TEST + static int g_counter; + if (g_counter++ > 100) + return 0; +#endif + if (g_have_simd) + goto end; + drmp3_cpuid(CPUInfo, 0); + if (CPUInfo[0] > 0) + { + drmp3_cpuid(CPUInfo, 1); + g_have_simd = (CPUInfo[3] & (1 << 26)) + 1; /* SSE2 */ + return g_have_simd - 1; + } + +end: + return g_have_simd - 1; +#endif +} +#elif defined(__ARM_NEON) || defined(__aarch64__) || defined(_M_ARM64) || defined(_M_ARM64EC) +#include +#define DRMP3_HAVE_SSE 0 +#define DRMP3_HAVE_SIMD 1 +#define DRMP3_VSTORE vst1q_f32 +#define DRMP3_VLD vld1q_f32 +#define DRMP3_VSET vmovq_n_f32 +#define DRMP3_VADD vaddq_f32 +#define DRMP3_VSUB vsubq_f32 +#define DRMP3_VMUL vmulq_f32 +#define DRMP3_VMAC(a, x, y) vmlaq_f32(a, x, y) +#define DRMP3_VMSB(a, x, y) vmlsq_f32(a, x, y) +#define DRMP3_VMUL_S(x, s) vmulq_f32(x, vmovq_n_f32(s)) +#define DRMP3_VREV(x) vcombine_f32(vget_high_f32(vrev64q_f32(x)), vget_low_f32(vrev64q_f32(x))) +typedef float32x4_t drmp3_f4; +static int drmp3_have_simd(void) +{ /* TODO: detect neon for !DR_MP3_ONLY_SIMD */ + return 1; +} +#else +#define DRMP3_HAVE_SSE 0 +#define DRMP3_HAVE_SIMD 0 +#ifdef DR_MP3_ONLY_SIMD +#error DR_MP3_ONLY_SIMD used, but SSE/NEON not enabled +#endif +#endif + +#else + +#define DRMP3_HAVE_SIMD 0 + +#endif + +#if defined(__ARM_ARCH) && (__ARM_ARCH >= 6) && !defined(__aarch64__) && !defined(_M_ARM64) && !defined(_M_ARM64EC) && !defined(__ARM_ARCH_6M__) +#define DRMP3_HAVE_ARMV6 1 +static __inline__ __attribute__((always_inline)) drmp3_int32 drmp3_clip_int16_arm(drmp3_int32 a) +{ + drmp3_int32 x = 0; + __asm__ ("ssat %0, #16, %1" : "=r"(x) : "r"(a)); + return x; +} +#else +#define DRMP3_HAVE_ARMV6 0 +#endif + + +/* Standard library stuff. */ +#ifndef DRMP3_ASSERT +#include +#define DRMP3_ASSERT(expression) assert(expression) +#endif +#ifndef DRMP3_COPY_MEMORY +#define DRMP3_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz)) +#endif +#ifndef DRMP3_MOVE_MEMORY +#define DRMP3_MOVE_MEMORY(dst, src, sz) memmove((dst), (src), (sz)) +#endif +#ifndef DRMP3_ZERO_MEMORY +#define DRMP3_ZERO_MEMORY(p, sz) memset((p), 0, (sz)) +#endif +#define DRMP3_ZERO_OBJECT(p) DRMP3_ZERO_MEMORY((p), sizeof(*(p))) +#ifndef DRMP3_MALLOC +#define DRMP3_MALLOC(sz) malloc((sz)) +#endif +#ifndef DRMP3_REALLOC +#define DRMP3_REALLOC(p, sz) realloc((p), (sz)) +#endif +#ifndef DRMP3_FREE +#define DRMP3_FREE(p) free((p)) +#endif + + + +typedef struct +{ + float scf[3*64]; + drmp3_uint8 total_bands, stereo_bands, bitalloc[64], scfcod[64]; +} drmp3_L12_scale_info; + +typedef struct +{ + drmp3_uint8 tab_offset, code_tab_width, band_count; +} drmp3_L12_subband_alloc; + +static void drmp3_bs_init(drmp3_bs *bs, const drmp3_uint8 *data, int bytes) +{ + bs->buf = data; + bs->pos = 0; + bs->limit = bytes*8; +} + +static drmp3_uint32 drmp3_bs_get_bits(drmp3_bs *bs, int n) +{ + drmp3_uint32 next, cache = 0, s = bs->pos & 7; + int shl = n + s; + const drmp3_uint8 *p = bs->buf + (bs->pos >> 3); + if ((bs->pos += n) > bs->limit) + return 0; + next = *p++ & (255 >> s); + while ((shl -= 8) > 0) + { + cache |= next << shl; + next = *p++; + } + return cache | (next >> -shl); +} + +static int drmp3_hdr_valid(const drmp3_uint8 *h) +{ + return h[0] == 0xff && + ((h[1] & 0xF0) == 0xf0 || (h[1] & 0xFE) == 0xe2) && + (DRMP3_HDR_GET_LAYER(h) != 0) && + (DRMP3_HDR_GET_BITRATE(h) != 15) && + (DRMP3_HDR_GET_SAMPLE_RATE(h) != 3); +} + +static int drmp3_hdr_compare(const drmp3_uint8 *h1, const drmp3_uint8 *h2) +{ + return drmp3_hdr_valid(h2) && + ((h1[1] ^ h2[1]) & 0xFE) == 0 && + ((h1[2] ^ h2[2]) & 0x0C) == 0 && + !(DRMP3_HDR_IS_FREE_FORMAT(h1) ^ DRMP3_HDR_IS_FREE_FORMAT(h2)); +} + +static unsigned drmp3_hdr_bitrate_kbps(const drmp3_uint8 *h) +{ + static const drmp3_uint8 halfrate[2][3][15] = { + { { 0,4,8,12,16,20,24,28,32,40,48,56,64,72,80 }, { 0,4,8,12,16,20,24,28,32,40,48,56,64,72,80 }, { 0,16,24,28,32,40,48,56,64,72,80,88,96,112,128 } }, + { { 0,16,20,24,28,32,40,48,56,64,80,96,112,128,160 }, { 0,16,24,28,32,40,48,56,64,80,96,112,128,160,192 }, { 0,16,32,48,64,80,96,112,128,144,160,176,192,208,224 } }, + }; + return 2*halfrate[!!DRMP3_HDR_TEST_MPEG1(h)][DRMP3_HDR_GET_LAYER(h) - 1][DRMP3_HDR_GET_BITRATE(h)]; +} + +static unsigned drmp3_hdr_sample_rate_hz(const drmp3_uint8 *h) +{ + static const unsigned g_hz[3] = { 44100, 48000, 32000 }; + return g_hz[DRMP3_HDR_GET_SAMPLE_RATE(h)] >> (int)!DRMP3_HDR_TEST_MPEG1(h) >> (int)!DRMP3_HDR_TEST_NOT_MPEG25(h); +} + +static unsigned drmp3_hdr_frame_samples(const drmp3_uint8 *h) +{ + return DRMP3_HDR_IS_LAYER_1(h) ? 384 : (1152 >> (int)DRMP3_HDR_IS_FRAME_576(h)); +} + +static int drmp3_hdr_frame_bytes(const drmp3_uint8 *h, int free_format_size) +{ + int frame_bytes = drmp3_hdr_frame_samples(h)*drmp3_hdr_bitrate_kbps(h)*125/drmp3_hdr_sample_rate_hz(h); + if (DRMP3_HDR_IS_LAYER_1(h)) + { + frame_bytes &= ~3; /* slot align */ + } + return frame_bytes ? frame_bytes : free_format_size; +} + +static int drmp3_hdr_padding(const drmp3_uint8 *h) +{ + return DRMP3_HDR_TEST_PADDING(h) ? (DRMP3_HDR_IS_LAYER_1(h) ? 4 : 1) : 0; +} + +#ifndef DR_MP3_ONLY_MP3 +static const drmp3_L12_subband_alloc *drmp3_L12_subband_alloc_table(const drmp3_uint8 *hdr, drmp3_L12_scale_info *sci) +{ + const drmp3_L12_subband_alloc *alloc; + int mode = DRMP3_HDR_GET_STEREO_MODE(hdr); + int nbands, stereo_bands = (mode == DRMP3_MODE_MONO) ? 0 : (mode == DRMP3_MODE_JOINT_STEREO) ? (DRMP3_HDR_GET_STEREO_MODE_EXT(hdr) << 2) + 4 : 32; + + if (DRMP3_HDR_IS_LAYER_1(hdr)) + { + static const drmp3_L12_subband_alloc g_alloc_L1[] = { { 76, 4, 32 } }; + alloc = g_alloc_L1; + nbands = 32; + } else if (!DRMP3_HDR_TEST_MPEG1(hdr)) + { + static const drmp3_L12_subband_alloc g_alloc_L2M2[] = { { 60, 4, 4 }, { 44, 3, 7 }, { 44, 2, 19 } }; + alloc = g_alloc_L2M2; + nbands = 30; + } else + { + static const drmp3_L12_subband_alloc g_alloc_L2M1[] = { { 0, 4, 3 }, { 16, 4, 8 }, { 32, 3, 12 }, { 40, 2, 7 } }; + int sample_rate_idx = DRMP3_HDR_GET_SAMPLE_RATE(hdr); + unsigned kbps = drmp3_hdr_bitrate_kbps(hdr) >> (int)(mode != DRMP3_MODE_MONO); + if (!kbps) /* free-format */ + { + kbps = 192; + } + + alloc = g_alloc_L2M1; + nbands = 27; + if (kbps < 56) + { + static const drmp3_L12_subband_alloc g_alloc_L2M1_lowrate[] = { { 44, 4, 2 }, { 44, 3, 10 } }; + alloc = g_alloc_L2M1_lowrate; + nbands = sample_rate_idx == 2 ? 12 : 8; + } else if (kbps >= 96 && sample_rate_idx != 1) + { + nbands = 30; + } + } + + sci->total_bands = (drmp3_uint8)nbands; + sci->stereo_bands = (drmp3_uint8)DRMP3_MIN(stereo_bands, nbands); + + return alloc; +} + +static void drmp3_L12_read_scalefactors(drmp3_bs *bs, drmp3_uint8 *pba, drmp3_uint8 *scfcod, int bands, float *scf) +{ + static const float g_deq_L12[18*3] = { +#define DRMP3_DQ(x) 9.53674316e-07f/x, 7.56931807e-07f/x, 6.00777173e-07f/x + DRMP3_DQ(3),DRMP3_DQ(7),DRMP3_DQ(15),DRMP3_DQ(31),DRMP3_DQ(63),DRMP3_DQ(127),DRMP3_DQ(255),DRMP3_DQ(511),DRMP3_DQ(1023),DRMP3_DQ(2047),DRMP3_DQ(4095),DRMP3_DQ(8191),DRMP3_DQ(16383),DRMP3_DQ(32767),DRMP3_DQ(65535),DRMP3_DQ(3),DRMP3_DQ(5),DRMP3_DQ(9) + }; + int i, m; + for (i = 0; i < bands; i++) + { + float s = 0; + int ba = *pba++; + int mask = ba ? 4 + ((19 >> scfcod[i]) & 3) : 0; + for (m = 4; m; m >>= 1) + { + if (mask & m) + { + int b = drmp3_bs_get_bits(bs, 6); + s = g_deq_L12[ba*3 - 6 + b % 3]*(int)(1 << 21 >> b/3); + } + *scf++ = s; + } + } +} + +static void drmp3_L12_read_scale_info(const drmp3_uint8 *hdr, drmp3_bs *bs, drmp3_L12_scale_info *sci) +{ + static const drmp3_uint8 g_bitalloc_code_tab[] = { + 0,17, 3, 4, 5,6,7, 8,9,10,11,12,13,14,15,16, + 0,17,18, 3,19,4,5, 6,7, 8, 9,10,11,12,13,16, + 0,17,18, 3,19,4,5,16, + 0,17,18,16, + 0,17,18,19, 4,5,6, 7,8, 9,10,11,12,13,14,15, + 0,17,18, 3,19,4,5, 6,7, 8, 9,10,11,12,13,14, + 0, 2, 3, 4, 5,6,7, 8,9,10,11,12,13,14,15,16 + }; + const drmp3_L12_subband_alloc *subband_alloc = drmp3_L12_subband_alloc_table(hdr, sci); + + int i, k = 0, ba_bits = 0; + const drmp3_uint8 *ba_code_tab = g_bitalloc_code_tab; + + for (i = 0; i < sci->total_bands; i++) + { + drmp3_uint8 ba; + if (i == k) + { + k += subband_alloc->band_count; + ba_bits = subband_alloc->code_tab_width; + ba_code_tab = g_bitalloc_code_tab + subband_alloc->tab_offset; + subband_alloc++; + } + ba = ba_code_tab[drmp3_bs_get_bits(bs, ba_bits)]; + sci->bitalloc[2*i] = ba; + if (i < sci->stereo_bands) + { + ba = ba_code_tab[drmp3_bs_get_bits(bs, ba_bits)]; + } + sci->bitalloc[2*i + 1] = sci->stereo_bands ? ba : 0; + } + + for (i = 0; i < 2*sci->total_bands; i++) + { + sci->scfcod[i] = (drmp3_uint8)(sci->bitalloc[i] ? DRMP3_HDR_IS_LAYER_1(hdr) ? 2 : drmp3_bs_get_bits(bs, 2) : 6); + } + + drmp3_L12_read_scalefactors(bs, sci->bitalloc, sci->scfcod, sci->total_bands*2, sci->scf); + + for (i = sci->stereo_bands; i < sci->total_bands; i++) + { + sci->bitalloc[2*i + 1] = 0; + } +} + +static int drmp3_L12_dequantize_granule(float *grbuf, drmp3_bs *bs, drmp3_L12_scale_info *sci, int group_size) +{ + int i, j, k, choff = 576; + for (j = 0; j < 4; j++) + { + float *dst = grbuf + group_size*j; + for (i = 0; i < 2*sci->total_bands; i++) + { + int ba = sci->bitalloc[i]; + if (ba != 0) + { + if (ba < 17) + { + int half = (1 << (ba - 1)) - 1; + for (k = 0; k < group_size; k++) + { + dst[k] = (float)((int)drmp3_bs_get_bits(bs, ba) - half); + } + } else + { + unsigned mod = (2 << (ba - 17)) + 1; /* 3, 5, 9 */ + unsigned code = drmp3_bs_get_bits(bs, mod + 2 - (mod >> 3)); /* 5, 7, 10 */ + for (k = 0; k < group_size; k++, code /= mod) + { + dst[k] = (float)((int)(code % mod - mod/2)); + } + } + } + dst += choff; + choff = 18 - choff; + } + } + return group_size*4; +} + +static void drmp3_L12_apply_scf_384(drmp3_L12_scale_info *sci, const float *scf, float *dst) +{ + int i, k; + DRMP3_COPY_MEMORY(dst + 576 + sci->stereo_bands*18, dst + sci->stereo_bands*18, (sci->total_bands - sci->stereo_bands)*18*sizeof(float)); + for (i = 0; i < sci->total_bands; i++, dst += 18, scf += 6) + { + for (k = 0; k < 12; k++) + { + dst[k + 0] *= scf[0]; + dst[k + 576] *= scf[3]; + } + } +} +#endif + +static int drmp3_L3_read_side_info(drmp3_bs *bs, drmp3_L3_gr_info *gr, const drmp3_uint8 *hdr) +{ + static const drmp3_uint8 g_scf_long[8][23] = { + { 6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54,0 }, + { 12,12,12,12,12,12,16,20,24,28,32,40,48,56,64,76,90,2,2,2,2,2,0 }, + { 6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54,0 }, + { 6,6,6,6,6,6,8,10,12,14,16,18,22,26,32,38,46,54,62,70,76,36,0 }, + { 6,6,6,6,6,6,8,10,12,14,16,20,24,28,32,38,46,52,60,68,58,54,0 }, + { 4,4,4,4,4,4,6,6,8,8,10,12,16,20,24,28,34,42,50,54,76,158,0 }, + { 4,4,4,4,4,4,6,6,6,8,10,12,16,18,22,28,34,40,46,54,54,192,0 }, + { 4,4,4,4,4,4,6,6,8,10,12,16,20,24,30,38,46,56,68,84,102,26,0 } + }; + static const drmp3_uint8 g_scf_short[8][40] = { + { 4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 }, + { 8,8,8,8,8,8,8,8,8,12,12,12,16,16,16,20,20,20,24,24,24,28,28,28,36,36,36,2,2,2,2,2,2,2,2,2,26,26,26,0 }, + { 4,4,4,4,4,4,4,4,4,6,6,6,6,6,6,8,8,8,10,10,10,14,14,14,18,18,18,26,26,26,32,32,32,42,42,42,18,18,18,0 }, + { 4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,32,32,32,44,44,44,12,12,12,0 }, + { 4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 }, + { 4,4,4,4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,22,22,22,30,30,30,56,56,56,0 }, + { 4,4,4,4,4,4,4,4,4,4,4,4,6,6,6,6,6,6,10,10,10,12,12,12,14,14,14,16,16,16,20,20,20,26,26,26,66,66,66,0 }, + { 4,4,4,4,4,4,4,4,4,4,4,4,6,6,6,8,8,8,12,12,12,16,16,16,20,20,20,26,26,26,34,34,34,42,42,42,12,12,12,0 } + }; + static const drmp3_uint8 g_scf_mixed[8][40] = { + { 6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 }, + { 12,12,12,4,4,4,8,8,8,12,12,12,16,16,16,20,20,20,24,24,24,28,28,28,36,36,36,2,2,2,2,2,2,2,2,2,26,26,26,0 }, + { 6,6,6,6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,14,14,14,18,18,18,26,26,26,32,32,32,42,42,42,18,18,18,0 }, + { 6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,32,32,32,44,44,44,12,12,12,0 }, + { 6,6,6,6,6,6,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,24,24,24,30,30,30,40,40,40,18,18,18,0 }, + { 4,4,4,4,4,4,6,6,4,4,4,6,6,6,8,8,8,10,10,10,12,12,12,14,14,14,18,18,18,22,22,22,30,30,30,56,56,56,0 }, + { 4,4,4,4,4,4,6,6,4,4,4,6,6,6,6,6,6,10,10,10,12,12,12,14,14,14,16,16,16,20,20,20,26,26,26,66,66,66,0 }, + { 4,4,4,4,4,4,6,6,4,4,4,6,6,6,8,8,8,12,12,12,16,16,16,20,20,20,26,26,26,34,34,34,42,42,42,12,12,12,0 } + }; + + unsigned tables, scfsi = 0; + int main_data_begin, part_23_sum = 0; + int gr_count = DRMP3_HDR_IS_MONO(hdr) ? 1 : 2; + int sr_idx = DRMP3_HDR_GET_MY_SAMPLE_RATE(hdr); sr_idx -= (sr_idx != 0); + + if (DRMP3_HDR_TEST_MPEG1(hdr)) + { + gr_count *= 2; + main_data_begin = drmp3_bs_get_bits(bs, 9); + scfsi = drmp3_bs_get_bits(bs, 7 + gr_count); + } else + { + main_data_begin = drmp3_bs_get_bits(bs, 8 + gr_count) >> gr_count; + } + + do + { + if (DRMP3_HDR_IS_MONO(hdr)) + { + scfsi <<= 4; + } + gr->part_23_length = (drmp3_uint16)drmp3_bs_get_bits(bs, 12); + part_23_sum += gr->part_23_length; + gr->big_values = (drmp3_uint16)drmp3_bs_get_bits(bs, 9); + if (gr->big_values > 288) + { + return -1; + } + gr->global_gain = (drmp3_uint8)drmp3_bs_get_bits(bs, 8); + gr->scalefac_compress = (drmp3_uint16)drmp3_bs_get_bits(bs, DRMP3_HDR_TEST_MPEG1(hdr) ? 4 : 9); + gr->sfbtab = g_scf_long[sr_idx]; + gr->n_long_sfb = 22; + gr->n_short_sfb = 0; + if (drmp3_bs_get_bits(bs, 1)) + { + gr->block_type = (drmp3_uint8)drmp3_bs_get_bits(bs, 2); + if (!gr->block_type) + { + return -1; + } + gr->mixed_block_flag = (drmp3_uint8)drmp3_bs_get_bits(bs, 1); + gr->region_count[0] = 7; + gr->region_count[1] = 255; + if (gr->block_type == DRMP3_SHORT_BLOCK_TYPE) + { + scfsi &= 0x0F0F; + if (!gr->mixed_block_flag) + { + gr->region_count[0] = 8; + gr->sfbtab = g_scf_short[sr_idx]; + gr->n_long_sfb = 0; + gr->n_short_sfb = 39; + } else + { + gr->sfbtab = g_scf_mixed[sr_idx]; + gr->n_long_sfb = DRMP3_HDR_TEST_MPEG1(hdr) ? 8 : 6; + gr->n_short_sfb = 30; + } + } + tables = drmp3_bs_get_bits(bs, 10); + tables <<= 5; + gr->subblock_gain[0] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3); + gr->subblock_gain[1] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3); + gr->subblock_gain[2] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3); + } else + { + gr->block_type = 0; + gr->mixed_block_flag = 0; + tables = drmp3_bs_get_bits(bs, 15); + gr->region_count[0] = (drmp3_uint8)drmp3_bs_get_bits(bs, 4); + gr->region_count[1] = (drmp3_uint8)drmp3_bs_get_bits(bs, 3); + gr->region_count[2] = 255; + } + gr->table_select[0] = (drmp3_uint8)(tables >> 10); + gr->table_select[1] = (drmp3_uint8)((tables >> 5) & 31); + gr->table_select[2] = (drmp3_uint8)((tables) & 31); + gr->preflag = (drmp3_uint8)(DRMP3_HDR_TEST_MPEG1(hdr) ? drmp3_bs_get_bits(bs, 1) : (gr->scalefac_compress >= 500)); + gr->scalefac_scale = (drmp3_uint8)drmp3_bs_get_bits(bs, 1); + gr->count1_table = (drmp3_uint8)drmp3_bs_get_bits(bs, 1); + gr->scfsi = (drmp3_uint8)((scfsi >> 12) & 15); + scfsi <<= 4; + gr++; + } while(--gr_count); + + if (part_23_sum + bs->pos > bs->limit + main_data_begin*8) + { + return -1; + } + + return main_data_begin; +} + +static void drmp3_L3_read_scalefactors(drmp3_uint8 *scf, drmp3_uint8 *ist_pos, const drmp3_uint8 *scf_size, const drmp3_uint8 *scf_count, drmp3_bs *bitbuf, int scfsi) +{ + int i, k; + for (i = 0; i < 4 && scf_count[i]; i++, scfsi *= 2) + { + int cnt = scf_count[i]; + if (scfsi & 8) + { + DRMP3_COPY_MEMORY(scf, ist_pos, cnt); + } else + { + int bits = scf_size[i]; + if (!bits) + { + DRMP3_ZERO_MEMORY(scf, cnt); + DRMP3_ZERO_MEMORY(ist_pos, cnt); + } else + { + int max_scf = (scfsi < 0) ? (1 << bits) - 1 : -1; + for (k = 0; k < cnt; k++) + { + int s = drmp3_bs_get_bits(bitbuf, bits); + ist_pos[k] = (drmp3_uint8)(s == max_scf ? -1 : s); + scf[k] = (drmp3_uint8)s; + } + } + } + ist_pos += cnt; + scf += cnt; + } + scf[0] = scf[1] = scf[2] = 0; +} + +static float drmp3_L3_ldexp_q2(float y, int exp_q2) +{ + static const float g_expfrac[4] = { 9.31322575e-10f,7.83145814e-10f,6.58544508e-10f,5.53767716e-10f }; + int e; + do + { + e = DRMP3_MIN(30*4, exp_q2); + y *= g_expfrac[e & 3]*(1 << 30 >> (e >> 2)); + } while ((exp_q2 -= e) > 0); + return y; +} + +/* +I've had reports of GCC 14 throwing an incorrect -Wstringop-overflow warning here. This is an attempt +to silence this warning. +*/ +#if (defined(__GNUC__) && (__GNUC__ >= 13)) && !defined(__clang__) + #pragma GCC diagnostic push + #pragma GCC diagnostic ignored "-Wstringop-overflow" +#endif +static void drmp3_L3_decode_scalefactors(const drmp3_uint8 *hdr, drmp3_uint8 *ist_pos, drmp3_bs *bs, const drmp3_L3_gr_info *gr, float *scf, int ch) +{ + static const drmp3_uint8 g_scf_partitions[3][28] = { + { 6,5,5, 5,6,5,5,5,6,5, 7,3,11,10,0,0, 7, 7, 7,0, 6, 6,6,3, 8, 8,5,0 }, + { 8,9,6,12,6,9,9,9,6,9,12,6,15,18,0,0, 6,15,12,0, 6,12,9,6, 6,18,9,0 }, + { 9,9,6,12,9,9,9,9,9,9,12,6,18,18,0,0,12,12,12,0,12, 9,9,6,15,12,9,0 } + }; + const drmp3_uint8 *scf_partition = g_scf_partitions[!!gr->n_short_sfb + !gr->n_long_sfb]; + drmp3_uint8 scf_size[4], iscf[40]; + int i, scf_shift = gr->scalefac_scale + 1, gain_exp, scfsi = gr->scfsi; + float gain; + + if (DRMP3_HDR_TEST_MPEG1(hdr)) + { + static const drmp3_uint8 g_scfc_decode[16] = { 0,1,2,3, 12,5,6,7, 9,10,11,13, 14,15,18,19 }; + int part = g_scfc_decode[gr->scalefac_compress]; + scf_size[1] = scf_size[0] = (drmp3_uint8)(part >> 2); + scf_size[3] = scf_size[2] = (drmp3_uint8)(part & 3); + } else + { + static const drmp3_uint8 g_mod[6*4] = { 5,5,4,4,5,5,4,1,4,3,1,1,5,6,6,1,4,4,4,1,4,3,1,1 }; + int k, modprod, sfc, ist = DRMP3_HDR_TEST_I_STEREO(hdr) && ch; + sfc = gr->scalefac_compress >> ist; + for (k = ist*3*4; sfc >= 0; sfc -= modprod, k += 4) + { + for (modprod = 1, i = 3; i >= 0; i--) + { + scf_size[i] = (drmp3_uint8)(sfc / modprod % g_mod[k + i]); + modprod *= g_mod[k + i]; + } + } + scf_partition += k; + scfsi = -16; + } + drmp3_L3_read_scalefactors(iscf, ist_pos, scf_size, scf_partition, bs, scfsi); + + if (gr->n_short_sfb) + { + int sh = 3 - scf_shift; + for (i = 0; i < gr->n_short_sfb; i += 3) + { + iscf[gr->n_long_sfb + i + 0] = (drmp3_uint8)(iscf[gr->n_long_sfb + i + 0] + (gr->subblock_gain[0] << sh)); + iscf[gr->n_long_sfb + i + 1] = (drmp3_uint8)(iscf[gr->n_long_sfb + i + 1] + (gr->subblock_gain[1] << sh)); + iscf[gr->n_long_sfb + i + 2] = (drmp3_uint8)(iscf[gr->n_long_sfb + i + 2] + (gr->subblock_gain[2] << sh)); + } + } else if (gr->preflag) + { + static const drmp3_uint8 g_preamp[10] = { 1,1,1,1,2,2,3,3,3,2 }; + for (i = 0; i < 10; i++) + { + iscf[11 + i] = (drmp3_uint8)(iscf[11 + i] + g_preamp[i]); + } + } + + gain_exp = gr->global_gain + DRMP3_BITS_DEQUANTIZER_OUT*4 - 210 - (DRMP3_HDR_IS_MS_STEREO(hdr) ? 2 : 0); + gain = drmp3_L3_ldexp_q2(1 << (DRMP3_MAX_SCFI/4), DRMP3_MAX_SCFI - gain_exp); + for (i = 0; i < (int)(gr->n_long_sfb + gr->n_short_sfb); i++) + { + scf[i] = drmp3_L3_ldexp_q2(gain, iscf[i] << scf_shift); + } +} +#if (defined(__GNUC__) && (__GNUC__ >= 13)) && !defined(__clang__) + #pragma GCC diagnostic pop +#endif + +static const float g_drmp3_pow43[129 + 16] = { + 0,-1,-2.519842f,-4.326749f,-6.349604f,-8.549880f,-10.902724f,-13.390518f,-16.000000f,-18.720754f,-21.544347f,-24.463781f,-27.473142f,-30.567351f,-33.741992f,-36.993181f, + 0,1,2.519842f,4.326749f,6.349604f,8.549880f,10.902724f,13.390518f,16.000000f,18.720754f,21.544347f,24.463781f,27.473142f,30.567351f,33.741992f,36.993181f,40.317474f,43.711787f,47.173345f,50.699631f,54.288352f,57.937408f,61.644865f,65.408941f,69.227979f,73.100443f,77.024898f,81.000000f,85.024491f,89.097188f,93.216975f,97.382800f,101.593667f,105.848633f,110.146801f,114.487321f,118.869381f,123.292209f,127.755065f,132.257246f,136.798076f,141.376907f,145.993119f,150.646117f,155.335327f,160.060199f,164.820202f,169.614826f,174.443577f,179.305980f,184.201575f,189.129918f,194.090580f,199.083145f,204.107210f,209.162385f,214.248292f,219.364564f,224.510845f,229.686789f,234.892058f,240.126328f,245.389280f,250.680604f,256.000000f,261.347174f,266.721841f,272.123723f,277.552547f,283.008049f,288.489971f,293.998060f,299.532071f,305.091761f,310.676898f,316.287249f,321.922592f,327.582707f,333.267377f,338.976394f,344.709550f,350.466646f,356.247482f,362.051866f,367.879608f,373.730522f,379.604427f,385.501143f,391.420496f,397.362314f,403.326427f,409.312672f,415.320884f,421.350905f,427.402579f,433.475750f,439.570269f,445.685987f,451.822757f,457.980436f,464.158883f,470.357960f,476.577530f,482.817459f,489.077615f,495.357868f,501.658090f,507.978156f,514.317941f,520.677324f,527.056184f,533.454404f,539.871867f,546.308458f,552.764065f,559.238575f,565.731879f,572.243870f,578.774440f,585.323483f,591.890898f,598.476581f,605.080431f,611.702349f,618.342238f,625.000000f,631.675540f,638.368763f,645.079578f +}; + +static float drmp3_L3_pow_43(int x) +{ + float frac; + int sign, mult = 256; + + if (x < 129) + { + return g_drmp3_pow43[16 + x]; + } + + if (x < 1024) + { + mult = 16; + x <<= 3; + } + + sign = 2*x & 64; + frac = (float)((x & 63) - sign) / ((x & ~63) + sign); + return g_drmp3_pow43[16 + ((x + sign) >> 6)]*(1.f + frac*((4.f/3) + frac*(2.f/9)))*mult; +} + +static void drmp3_L3_huffman(float *dst, drmp3_bs *bs, const drmp3_L3_gr_info *gr_info, const float *scf, int layer3gr_limit) +{ + static const drmp3_int16 tabs[] = { 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, 0, + 785,785,785,785,784,784,784,784,513,513,513,513,513,513,513,513,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256, + -255,1313,1298,1282,785,785,785,785,784,784,784,784,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,290,288, + -255,1313,1298,1282,769,769,769,769,529,529,529,529,529,529,529,529,528,528,528,528,528,528,528,528,512,512,512,512,512,512,512,512,290,288, + -253,-318,-351,-367,785,785,785,785,784,784,784,784,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,819,818,547,547,275,275,275,275,561,560,515,546,289,274,288,258, + -254,-287,1329,1299,1314,1312,1057,1057,1042,1042,1026,1026,784,784,784,784,529,529,529,529,529,529,529,529,769,769,769,769,768,768,768,768,563,560,306,306,291,259, + -252,-413,-477,-542,1298,-575,1041,1041,784,784,784,784,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,-383,-399,1107,1092,1106,1061,849,849,789,789,1104,1091,773,773,1076,1075,341,340,325,309,834,804,577,577,532,532,516,516,832,818,803,816,561,561,531,531,515,546,289,289,288,258, + -252,-429,-493,-559,1057,1057,1042,1042,529,529,529,529,529,529,529,529,784,784,784,784,769,769,769,769,512,512,512,512,512,512,512,512,-382,1077,-415,1106,1061,1104,849,849,789,789,1091,1076,1029,1075,834,834,597,581,340,340,339,324,804,833,532,532,832,772,818,803,817,787,816,771,290,290,290,290,288,258, + -253,-349,-414,-447,-463,1329,1299,-479,1314,1312,1057,1057,1042,1042,1026,1026,785,785,785,785,784,784,784,784,769,769,769,769,768,768,768,768,-319,851,821,-335,836,850,805,849,341,340,325,336,533,533,579,579,564,564,773,832,578,548,563,516,321,276,306,291,304,259, + -251,-572,-733,-830,-863,-879,1041,1041,784,784,784,784,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,-511,-527,-543,1396,1351,1381,1366,1395,1335,1380,-559,1334,1138,1138,1063,1063,1350,1392,1031,1031,1062,1062,1364,1363,1120,1120,1333,1348,881,881,881,881,375,374,359,373,343,358,341,325,791,791,1123,1122,-703,1105,1045,-719,865,865,790,790,774,774,1104,1029,338,293,323,308,-799,-815,833,788,772,818,803,816,322,292,307,320,561,531,515,546,289,274,288,258, + -251,-525,-605,-685,-765,-831,-846,1298,1057,1057,1312,1282,785,785,785,785,784,784,784,784,769,769,769,769,512,512,512,512,512,512,512,512,1399,1398,1383,1367,1382,1396,1351,-511,1381,1366,1139,1139,1079,1079,1124,1124,1364,1349,1363,1333,882,882,882,882,807,807,807,807,1094,1094,1136,1136,373,341,535,535,881,775,867,822,774,-591,324,338,-671,849,550,550,866,864,609,609,293,336,534,534,789,835,773,-751,834,804,308,307,833,788,832,772,562,562,547,547,305,275,560,515,290,290, + -252,-397,-477,-557,-622,-653,-719,-735,-750,1329,1299,1314,1057,1057,1042,1042,1312,1282,1024,1024,785,785,785,785,784,784,784,784,769,769,769,769,-383,1127,1141,1111,1126,1140,1095,1110,869,869,883,883,1079,1109,882,882,375,374,807,868,838,881,791,-463,867,822,368,263,852,837,836,-543,610,610,550,550,352,336,534,534,865,774,851,821,850,805,593,533,579,564,773,832,578,578,548,548,577,577,307,276,306,291,516,560,259,259, + -250,-2107,-2507,-2764,-2909,-2974,-3007,-3023,1041,1041,1040,1040,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,-767,-1052,-1213,-1277,-1358,-1405,-1469,-1535,-1550,-1582,-1614,-1647,-1662,-1694,-1726,-1759,-1774,-1807,-1822,-1854,-1886,1565,-1919,-1935,-1951,-1967,1731,1730,1580,1717,-1983,1729,1564,-1999,1548,-2015,-2031,1715,1595,-2047,1714,-2063,1610,-2079,1609,-2095,1323,1323,1457,1457,1307,1307,1712,1547,1641,1700,1699,1594,1685,1625,1442,1442,1322,1322,-780,-973,-910,1279,1278,1277,1262,1276,1261,1275,1215,1260,1229,-959,974,974,989,989,-943,735,478,478,495,463,506,414,-1039,1003,958,1017,927,942,987,957,431,476,1272,1167,1228,-1183,1256,-1199,895,895,941,941,1242,1227,1212,1135,1014,1014,490,489,503,487,910,1013,985,925,863,894,970,955,1012,847,-1343,831,755,755,984,909,428,366,754,559,-1391,752,486,457,924,997,698,698,983,893,740,740,908,877,739,739,667,667,953,938,497,287,271,271,683,606,590,712,726,574,302,302,738,736,481,286,526,725,605,711,636,724,696,651,589,681,666,710,364,467,573,695,466,466,301,465,379,379,709,604,665,679,316,316,634,633,436,436,464,269,424,394,452,332,438,363,347,408,393,448,331,422,362,407,392,421,346,406,391,376,375,359,1441,1306,-2367,1290,-2383,1337,-2399,-2415,1426,1321,-2431,1411,1336,-2447,-2463,-2479,1169,1169,1049,1049,1424,1289,1412,1352,1319,-2495,1154,1154,1064,1064,1153,1153,416,390,360,404,403,389,344,374,373,343,358,372,327,357,342,311,356,326,1395,1394,1137,1137,1047,1047,1365,1392,1287,1379,1334,1364,1349,1378,1318,1363,792,792,792,792,1152,1152,1032,1032,1121,1121,1046,1046,1120,1120,1030,1030,-2895,1106,1061,1104,849,849,789,789,1091,1076,1029,1090,1060,1075,833,833,309,324,532,532,832,772,818,803,561,561,531,560,515,546,289,274,288,258, + -250,-1179,-1579,-1836,-1996,-2124,-2253,-2333,-2413,-2477,-2542,-2574,-2607,-2622,-2655,1314,1313,1298,1312,1282,785,785,785,785,1040,1040,1025,1025,768,768,768,768,-766,-798,-830,-862,-895,-911,-927,-943,-959,-975,-991,-1007,-1023,-1039,-1055,-1070,1724,1647,-1103,-1119,1631,1767,1662,1738,1708,1723,-1135,1780,1615,1779,1599,1677,1646,1778,1583,-1151,1777,1567,1737,1692,1765,1722,1707,1630,1751,1661,1764,1614,1736,1676,1763,1750,1645,1598,1721,1691,1762,1706,1582,1761,1566,-1167,1749,1629,767,766,751,765,494,494,735,764,719,749,734,763,447,447,748,718,477,506,431,491,446,476,461,505,415,430,475,445,504,399,460,489,414,503,383,474,429,459,502,502,746,752,488,398,501,473,413,472,486,271,480,270,-1439,-1455,1357,-1471,-1487,-1503,1341,1325,-1519,1489,1463,1403,1309,-1535,1372,1448,1418,1476,1356,1462,1387,-1551,1475,1340,1447,1402,1386,-1567,1068,1068,1474,1461,455,380,468,440,395,425,410,454,364,467,466,464,453,269,409,448,268,432,1371,1473,1432,1417,1308,1460,1355,1446,1459,1431,1083,1083,1401,1416,1458,1445,1067,1067,1370,1457,1051,1051,1291,1430,1385,1444,1354,1415,1400,1443,1082,1082,1173,1113,1186,1066,1185,1050,-1967,1158,1128,1172,1097,1171,1081,-1983,1157,1112,416,266,375,400,1170,1142,1127,1065,793,793,1169,1033,1156,1096,1141,1111,1155,1080,1126,1140,898,898,808,808,897,897,792,792,1095,1152,1032,1125,1110,1139,1079,1124,882,807,838,881,853,791,-2319,867,368,263,822,852,837,866,806,865,-2399,851,352,262,534,534,821,836,594,594,549,549,593,593,533,533,848,773,579,579,564,578,548,563,276,276,577,576,306,291,516,560,305,305,275,259, + -251,-892,-2058,-2620,-2828,-2957,-3023,-3039,1041,1041,1040,1040,769,769,769,769,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,256,-511,-527,-543,-559,1530,-575,-591,1528,1527,1407,1526,1391,1023,1023,1023,1023,1525,1375,1268,1268,1103,1103,1087,1087,1039,1039,1523,-604,815,815,815,815,510,495,509,479,508,463,507,447,431,505,415,399,-734,-782,1262,-815,1259,1244,-831,1258,1228,-847,-863,1196,-879,1253,987,987,748,-767,493,493,462,477,414,414,686,669,478,446,461,445,474,429,487,458,412,471,1266,1264,1009,1009,799,799,-1019,-1276,-1452,-1581,-1677,-1757,-1821,-1886,-1933,-1997,1257,1257,1483,1468,1512,1422,1497,1406,1467,1496,1421,1510,1134,1134,1225,1225,1466,1451,1374,1405,1252,1252,1358,1480,1164,1164,1251,1251,1238,1238,1389,1465,-1407,1054,1101,-1423,1207,-1439,830,830,1248,1038,1237,1117,1223,1148,1236,1208,411,426,395,410,379,269,1193,1222,1132,1235,1221,1116,976,976,1192,1162,1177,1220,1131,1191,963,963,-1647,961,780,-1663,558,558,994,993,437,408,393,407,829,978,813,797,947,-1743,721,721,377,392,844,950,828,890,706,706,812,859,796,960,948,843,934,874,571,571,-1919,690,555,689,421,346,539,539,944,779,918,873,932,842,903,888,570,570,931,917,674,674,-2575,1562,-2591,1609,-2607,1654,1322,1322,1441,1441,1696,1546,1683,1593,1669,1624,1426,1426,1321,1321,1639,1680,1425,1425,1305,1305,1545,1668,1608,1623,1667,1592,1638,1666,1320,1320,1652,1607,1409,1409,1304,1304,1288,1288,1664,1637,1395,1395,1335,1335,1622,1636,1394,1394,1319,1319,1606,1621,1392,1392,1137,1137,1137,1137,345,390,360,375,404,373,1047,-2751,-2767,-2783,1062,1121,1046,-2799,1077,-2815,1106,1061,789,789,1105,1104,263,355,310,340,325,354,352,262,339,324,1091,1076,1029,1090,1060,1075,833,833,788,788,1088,1028,818,818,803,803,561,561,531,531,816,771,546,546,289,274,288,258, + -253,-317,-381,-446,-478,-509,1279,1279,-811,-1179,-1451,-1756,-1900,-2028,-2189,-2253,-2333,-2414,-2445,-2511,-2526,1313,1298,-2559,1041,1041,1040,1040,1025,1025,1024,1024,1022,1007,1021,991,1020,975,1019,959,687,687,1018,1017,671,671,655,655,1016,1015,639,639,758,758,623,623,757,607,756,591,755,575,754,559,543,543,1009,783,-575,-621,-685,-749,496,-590,750,749,734,748,974,989,1003,958,988,973,1002,942,987,957,972,1001,926,986,941,971,956,1000,910,985,925,999,894,970,-1071,-1087,-1102,1390,-1135,1436,1509,1451,1374,-1151,1405,1358,1480,1420,-1167,1507,1494,1389,1342,1465,1435,1450,1326,1505,1310,1493,1373,1479,1404,1492,1464,1419,428,443,472,397,736,526,464,464,486,457,442,471,484,482,1357,1449,1434,1478,1388,1491,1341,1490,1325,1489,1463,1403,1309,1477,1372,1448,1418,1433,1476,1356,1462,1387,-1439,1475,1340,1447,1402,1474,1324,1461,1371,1473,269,448,1432,1417,1308,1460,-1711,1459,-1727,1441,1099,1099,1446,1386,1431,1401,-1743,1289,1083,1083,1160,1160,1458,1445,1067,1067,1370,1457,1307,1430,1129,1129,1098,1098,268,432,267,416,266,400,-1887,1144,1187,1082,1173,1113,1186,1066,1050,1158,1128,1143,1172,1097,1171,1081,420,391,1157,1112,1170,1142,1127,1065,1169,1049,1156,1096,1141,1111,1155,1080,1126,1154,1064,1153,1140,1095,1048,-2159,1125,1110,1137,-2175,823,823,1139,1138,807,807,384,264,368,263,868,838,853,791,867,822,852,837,866,806,865,790,-2319,851,821,836,352,262,850,805,849,-2399,533,533,835,820,336,261,578,548,563,577,532,532,832,772,562,562,547,547,305,275,560,515,290,290,288,258 }; + static const drmp3_uint8 tab32[] = { 130,162,193,209,44,28,76,140,9,9,9,9,9,9,9,9,190,254,222,238,126,94,157,157,109,61,173,205}; + static const drmp3_uint8 tab33[] = { 252,236,220,204,188,172,156,140,124,108,92,76,60,44,28,12 }; + static const drmp3_int16 tabindex[2*16] = { 0,32,64,98,0,132,180,218,292,364,426,538,648,746,0,1126,1460,1460,1460,1460,1460,1460,1460,1460,1842,1842,1842,1842,1842,1842,1842,1842 }; + static const drmp3_uint8 g_linbits[] = { 0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,0,1,2,3,4,6,8,10,13,4,5,6,7,8,9,11,13 }; + +#define DRMP3_PEEK_BITS(n) (bs_cache >> (32 - (n))) +#define DRMP3_FLUSH_BITS(n) { bs_cache <<= (n); bs_sh += (n); } +#define DRMP3_CHECK_BITS while (bs_sh >= 0) { bs_cache |= (drmp3_uint32)*bs_next_ptr++ << bs_sh; bs_sh -= 8; } +#define DRMP3_BSPOS ((bs_next_ptr - bs->buf)*8 - 24 + bs_sh) + + float one = 0.0f; + int ireg = 0, big_val_cnt = gr_info->big_values; + const drmp3_uint8 *sfb = gr_info->sfbtab; + const drmp3_uint8 *bs_next_ptr = bs->buf + bs->pos/8; + drmp3_uint32 bs_cache = (((bs_next_ptr[0]*256u + bs_next_ptr[1])*256u + bs_next_ptr[2])*256u + bs_next_ptr[3]) << (bs->pos & 7); + int pairs_to_decode, np, bs_sh = (bs->pos & 7) - 8; + bs_next_ptr += 4; + + while (big_val_cnt > 0) + { + int tab_num = gr_info->table_select[ireg]; + int sfb_cnt = gr_info->region_count[ireg++]; + const drmp3_int16 *codebook = tabs + tabindex[tab_num]; + int linbits = g_linbits[tab_num]; + if (linbits) + { + do + { + np = *sfb++ / 2; + pairs_to_decode = DRMP3_MIN(big_val_cnt, np); + one = *scf++; + do + { + int j, w = 5; + int leaf = codebook[DRMP3_PEEK_BITS(w)]; + while (leaf < 0) + { + DRMP3_FLUSH_BITS(w); + w = leaf & 7; + leaf = codebook[DRMP3_PEEK_BITS(w) - (leaf >> 3)]; + } + DRMP3_FLUSH_BITS(leaf >> 8); + + for (j = 0; j < 2; j++, dst++, leaf >>= 4) + { + int lsb = leaf & 0x0F; + if (lsb == 15) + { + lsb += DRMP3_PEEK_BITS(linbits); + DRMP3_FLUSH_BITS(linbits); + DRMP3_CHECK_BITS; + *dst = one*drmp3_L3_pow_43(lsb)*((drmp3_int32)bs_cache < 0 ? -1: 1); + } else + { + *dst = g_drmp3_pow43[16 + lsb - 16*(bs_cache >> 31)]*one; + } + DRMP3_FLUSH_BITS(lsb ? 1 : 0); + } + DRMP3_CHECK_BITS; + } while (--pairs_to_decode); + } while ((big_val_cnt -= np) > 0 && --sfb_cnt >= 0); + } else + { + do + { + np = *sfb++ / 2; + pairs_to_decode = DRMP3_MIN(big_val_cnt, np); + one = *scf++; + do + { + int j, w = 5; + int leaf = codebook[DRMP3_PEEK_BITS(w)]; + while (leaf < 0) + { + DRMP3_FLUSH_BITS(w); + w = leaf & 7; + leaf = codebook[DRMP3_PEEK_BITS(w) - (leaf >> 3)]; + } + DRMP3_FLUSH_BITS(leaf >> 8); + + for (j = 0; j < 2; j++, dst++, leaf >>= 4) + { + int lsb = leaf & 0x0F; + *dst = g_drmp3_pow43[16 + lsb - 16*(bs_cache >> 31)]*one; + DRMP3_FLUSH_BITS(lsb ? 1 : 0); + } + DRMP3_CHECK_BITS; + } while (--pairs_to_decode); + } while ((big_val_cnt -= np) > 0 && --sfb_cnt >= 0); + } + } + + for (np = 1 - big_val_cnt;; dst += 4) + { + const drmp3_uint8 *codebook_count1 = (gr_info->count1_table) ? tab33 : tab32; + int leaf = codebook_count1[DRMP3_PEEK_BITS(4)]; + if (!(leaf & 8)) + { + leaf = codebook_count1[(leaf >> 3) + (bs_cache << 4 >> (32 - (leaf & 3)))]; + } + DRMP3_FLUSH_BITS(leaf & 7); + if (DRMP3_BSPOS > layer3gr_limit) + { + break; + } +#define DRMP3_RELOAD_SCALEFACTOR if (!--np) { np = *sfb++/2; if (!np) break; one = *scf++; } +#define DRMP3_DEQ_COUNT1(s) if (leaf & (128 >> s)) { dst[s] = ((drmp3_int32)bs_cache < 0) ? -one : one; DRMP3_FLUSH_BITS(1) } + DRMP3_RELOAD_SCALEFACTOR; + DRMP3_DEQ_COUNT1(0); + DRMP3_DEQ_COUNT1(1); + DRMP3_RELOAD_SCALEFACTOR; + DRMP3_DEQ_COUNT1(2); + DRMP3_DEQ_COUNT1(3); + DRMP3_CHECK_BITS; + } + + bs->pos = layer3gr_limit; +} + +static void drmp3_L3_midside_stereo(float *left, int n) +{ + int i = 0; + float *right = left + 576; +#if DRMP3_HAVE_SIMD + if (drmp3_have_simd()) + { + for (; i < n - 3; i += 4) + { + drmp3_f4 vl = DRMP3_VLD(left + i); + drmp3_f4 vr = DRMP3_VLD(right + i); + DRMP3_VSTORE(left + i, DRMP3_VADD(vl, vr)); + DRMP3_VSTORE(right + i, DRMP3_VSUB(vl, vr)); + } +#ifdef __GNUC__ + /* Workaround for spurious -Waggressive-loop-optimizations warning from gcc. + * For more info see: https://github.com/lieff/minimp3/issues/88 + */ + if (__builtin_constant_p(n % 4 == 0) && n % 4 == 0) + return; +#endif + } +#endif + for (; i < n; i++) + { + float a = left[i]; + float b = right[i]; + left[i] = a + b; + right[i] = a - b; + } +} + +static void drmp3_L3_intensity_stereo_band(float *left, int n, float kl, float kr) +{ + int i; + for (i = 0; i < n; i++) + { + left[i + 576] = left[i]*kr; + left[i] = left[i]*kl; + } +} + +static void drmp3_L3_stereo_top_band(const float *right, const drmp3_uint8 *sfb, int nbands, int max_band[3]) +{ + int i, k; + + max_band[0] = max_band[1] = max_band[2] = -1; + + for (i = 0; i < nbands; i++) + { + for (k = 0; k < sfb[i]; k += 2) + { + if (right[k] != 0 || right[k + 1] != 0) + { + max_band[i % 3] = i; + break; + } + } + right += sfb[i]; + } +} + +static void drmp3_L3_stereo_process(float *left, const drmp3_uint8 *ist_pos, const drmp3_uint8 *sfb, const drmp3_uint8 *hdr, int max_band[3], int mpeg2_sh) +{ + static const float g_pan[7*2] = { 0,1,0.21132487f,0.78867513f,0.36602540f,0.63397460f,0.5f,0.5f,0.63397460f,0.36602540f,0.78867513f,0.21132487f,1,0 }; + unsigned i, max_pos = DRMP3_HDR_TEST_MPEG1(hdr) ? 7 : 64; + + for (i = 0; sfb[i]; i++) + { + unsigned ipos = ist_pos[i]; + if ((int)i > max_band[i % 3] && ipos < max_pos) + { + float kl, kr, s = DRMP3_HDR_TEST_MS_STEREO(hdr) ? 1.41421356f : 1; + if (DRMP3_HDR_TEST_MPEG1(hdr)) + { + kl = g_pan[2*ipos]; + kr = g_pan[2*ipos + 1]; + } else + { + kl = 1; + kr = drmp3_L3_ldexp_q2(1, (ipos + 1) >> 1 << mpeg2_sh); + if (ipos & 1) + { + kl = kr; + kr = 1; + } + } + drmp3_L3_intensity_stereo_band(left, sfb[i], kl*s, kr*s); + } else if (DRMP3_HDR_TEST_MS_STEREO(hdr)) + { + drmp3_L3_midside_stereo(left, sfb[i]); + } + left += sfb[i]; + } +} + +static void drmp3_L3_intensity_stereo(float *left, drmp3_uint8 *ist_pos, const drmp3_L3_gr_info *gr, const drmp3_uint8 *hdr) +{ + int max_band[3], n_sfb = gr->n_long_sfb + gr->n_short_sfb; + int i, max_blocks = gr->n_short_sfb ? 3 : 1; + + drmp3_L3_stereo_top_band(left + 576, gr->sfbtab, n_sfb, max_band); + if (gr->n_long_sfb) + { + max_band[0] = max_band[1] = max_band[2] = DRMP3_MAX(DRMP3_MAX(max_band[0], max_band[1]), max_band[2]); + } + for (i = 0; i < max_blocks; i++) + { + int default_pos = DRMP3_HDR_TEST_MPEG1(hdr) ? 3 : 0; + int itop = n_sfb - max_blocks + i; + int prev = itop - max_blocks; + ist_pos[itop] = (drmp3_uint8)(max_band[i] >= prev ? default_pos : ist_pos[prev]); + } + drmp3_L3_stereo_process(left, ist_pos, gr->sfbtab, hdr, max_band, gr[1].scalefac_compress & 1); +} + +static void drmp3_L3_reorder(float *grbuf, float *scratch, const drmp3_uint8 *sfb) +{ + int i, len; + float *src = grbuf, *dst = scratch; + + for (;0 != (len = *sfb); sfb += 3, src += 2*len) + { + for (i = 0; i < len; i++, src++) + { + *dst++ = src[0*len]; + *dst++ = src[1*len]; + *dst++ = src[2*len]; + } + } + DRMP3_COPY_MEMORY(grbuf, scratch, (dst - scratch)*sizeof(float)); +} + +static void drmp3_L3_antialias(float *grbuf, int nbands) +{ + static const float g_aa[2][8] = { + {0.85749293f,0.88174200f,0.94962865f,0.98331459f,0.99551782f,0.99916056f,0.99989920f,0.99999316f}, + {0.51449576f,0.47173197f,0.31337745f,0.18191320f,0.09457419f,0.04096558f,0.01419856f,0.00369997f} + }; + + for (; nbands > 0; nbands--, grbuf += 18) + { + int i = 0; +#if DRMP3_HAVE_SIMD + if (drmp3_have_simd()) for (; i < 8; i += 4) + { + drmp3_f4 vu = DRMP3_VLD(grbuf + 18 + i); + drmp3_f4 vd = DRMP3_VLD(grbuf + 14 - i); + drmp3_f4 vc0 = DRMP3_VLD(g_aa[0] + i); + drmp3_f4 vc1 = DRMP3_VLD(g_aa[1] + i); + vd = DRMP3_VREV(vd); + DRMP3_VSTORE(grbuf + 18 + i, DRMP3_VSUB(DRMP3_VMUL(vu, vc0), DRMP3_VMUL(vd, vc1))); + vd = DRMP3_VADD(DRMP3_VMUL(vu, vc1), DRMP3_VMUL(vd, vc0)); + DRMP3_VSTORE(grbuf + 14 - i, DRMP3_VREV(vd)); + } +#endif +#ifndef DR_MP3_ONLY_SIMD + for(; i < 8; i++) + { + float u = grbuf[18 + i]; + float d = grbuf[17 - i]; + grbuf[18 + i] = u*g_aa[0][i] - d*g_aa[1][i]; + grbuf[17 - i] = u*g_aa[1][i] + d*g_aa[0][i]; + } +#endif + } +} + +static void drmp3_L3_dct3_9(float *y) +{ + float s0, s1, s2, s3, s4, s5, s6, s7, s8, t0, t2, t4; + + s0 = y[0]; s2 = y[2]; s4 = y[4]; s6 = y[6]; s8 = y[8]; + t0 = s0 + s6*0.5f; + s0 -= s6; + t4 = (s4 + s2)*0.93969262f; + t2 = (s8 + s2)*0.76604444f; + s6 = (s4 - s8)*0.17364818f; + s4 += s8 - s2; + + s2 = s0 - s4*0.5f; + y[4] = s4 + s0; + s8 = t0 - t2 + s6; + s0 = t0 - t4 + t2; + s4 = t0 + t4 - s6; + + s1 = y[1]; s3 = y[3]; s5 = y[5]; s7 = y[7]; + + s3 *= 0.86602540f; + t0 = (s5 + s1)*0.98480775f; + t4 = (s5 - s7)*0.34202014f; + t2 = (s1 + s7)*0.64278761f; + s1 = (s1 - s5 - s7)*0.86602540f; + + s5 = t0 - s3 - t2; + s7 = t4 - s3 - t0; + s3 = t4 + s3 - t2; + + y[0] = s4 - s7; + y[1] = s2 + s1; + y[2] = s0 - s3; + y[3] = s8 + s5; + y[5] = s8 - s5; + y[6] = s0 + s3; + y[7] = s2 - s1; + y[8] = s4 + s7; +} + +static void drmp3_L3_imdct36(float *grbuf, float *overlap, const float *window, int nbands) +{ + int i, j; + static const float g_twid9[18] = { + 0.73727734f,0.79335334f,0.84339145f,0.88701083f,0.92387953f,0.95371695f,0.97629601f,0.99144486f,0.99904822f,0.67559021f,0.60876143f,0.53729961f,0.46174861f,0.38268343f,0.30070580f,0.21643961f,0.13052619f,0.04361938f + }; + + for (j = 0; j < nbands; j++, grbuf += 18, overlap += 9) + { + float co[9], si[9]; + co[0] = -grbuf[0]; + si[0] = grbuf[17]; + for (i = 0; i < 4; i++) + { + si[8 - 2*i] = grbuf[4*i + 1] - grbuf[4*i + 2]; + co[1 + 2*i] = grbuf[4*i + 1] + grbuf[4*i + 2]; + si[7 - 2*i] = grbuf[4*i + 4] - grbuf[4*i + 3]; + co[2 + 2*i] = -(grbuf[4*i + 3] + grbuf[4*i + 4]); + } + drmp3_L3_dct3_9(co); + drmp3_L3_dct3_9(si); + + si[1] = -si[1]; + si[3] = -si[3]; + si[5] = -si[5]; + si[7] = -si[7]; + + i = 0; + +#if DRMP3_HAVE_SIMD + if (drmp3_have_simd()) for (; i < 8; i += 4) + { + drmp3_f4 vovl = DRMP3_VLD(overlap + i); + drmp3_f4 vc = DRMP3_VLD(co + i); + drmp3_f4 vs = DRMP3_VLD(si + i); + drmp3_f4 vr0 = DRMP3_VLD(g_twid9 + i); + drmp3_f4 vr1 = DRMP3_VLD(g_twid9 + 9 + i); + drmp3_f4 vw0 = DRMP3_VLD(window + i); + drmp3_f4 vw1 = DRMP3_VLD(window + 9 + i); + drmp3_f4 vsum = DRMP3_VADD(DRMP3_VMUL(vc, vr1), DRMP3_VMUL(vs, vr0)); + DRMP3_VSTORE(overlap + i, DRMP3_VSUB(DRMP3_VMUL(vc, vr0), DRMP3_VMUL(vs, vr1))); + DRMP3_VSTORE(grbuf + i, DRMP3_VSUB(DRMP3_VMUL(vovl, vw0), DRMP3_VMUL(vsum, vw1))); + vsum = DRMP3_VADD(DRMP3_VMUL(vovl, vw1), DRMP3_VMUL(vsum, vw0)); + DRMP3_VSTORE(grbuf + 14 - i, DRMP3_VREV(vsum)); + } +#endif + for (; i < 9; i++) + { + float ovl = overlap[i]; + float sum = co[i]*g_twid9[9 + i] + si[i]*g_twid9[0 + i]; + overlap[i] = co[i]*g_twid9[0 + i] - si[i]*g_twid9[9 + i]; + grbuf[i] = ovl*window[0 + i] - sum*window[9 + i]; + grbuf[17 - i] = ovl*window[9 + i] + sum*window[0 + i]; + } + } +} + +static void drmp3_L3_idct3(float x0, float x1, float x2, float *dst) +{ + float m1 = x1*0.86602540f; + float a1 = x0 - x2*0.5f; + dst[1] = x0 + x2; + dst[0] = a1 + m1; + dst[2] = a1 - m1; +} + +static void drmp3_L3_imdct12(float *x, float *dst, float *overlap) +{ + static const float g_twid3[6] = { 0.79335334f,0.92387953f,0.99144486f, 0.60876143f,0.38268343f,0.13052619f }; + float co[3], si[3]; + int i; + + drmp3_L3_idct3(-x[0], x[6] + x[3], x[12] + x[9], co); + drmp3_L3_idct3(x[15], x[12] - x[9], x[6] - x[3], si); + si[1] = -si[1]; + + for (i = 0; i < 3; i++) + { + float ovl = overlap[i]; + float sum = co[i]*g_twid3[3 + i] + si[i]*g_twid3[0 + i]; + overlap[i] = co[i]*g_twid3[0 + i] - si[i]*g_twid3[3 + i]; + dst[i] = ovl*g_twid3[2 - i] - sum*g_twid3[5 - i]; + dst[5 - i] = ovl*g_twid3[5 - i] + sum*g_twid3[2 - i]; + } +} + +static void drmp3_L3_imdct_short(float *grbuf, float *overlap, int nbands) +{ + for (;nbands > 0; nbands--, overlap += 9, grbuf += 18) + { + float tmp[18]; + DRMP3_COPY_MEMORY(tmp, grbuf, sizeof(tmp)); + DRMP3_COPY_MEMORY(grbuf, overlap, 6*sizeof(float)); + drmp3_L3_imdct12(tmp, grbuf + 6, overlap + 6); + drmp3_L3_imdct12(tmp + 1, grbuf + 12, overlap + 6); + drmp3_L3_imdct12(tmp + 2, overlap, overlap + 6); + } +} + +static void drmp3_L3_change_sign(float *grbuf) +{ + int b, i; + for (b = 0, grbuf += 18; b < 32; b += 2, grbuf += 36) + for (i = 1; i < 18; i += 2) + grbuf[i] = -grbuf[i]; +} + +static void drmp3_L3_imdct_gr(float *grbuf, float *overlap, unsigned block_type, unsigned n_long_bands) +{ + static const float g_mdct_window[2][18] = { + { 0.99904822f,0.99144486f,0.97629601f,0.95371695f,0.92387953f,0.88701083f,0.84339145f,0.79335334f,0.73727734f,0.04361938f,0.13052619f,0.21643961f,0.30070580f,0.38268343f,0.46174861f,0.53729961f,0.60876143f,0.67559021f }, + { 1,1,1,1,1,1,0.99144486f,0.92387953f,0.79335334f,0,0,0,0,0,0,0.13052619f,0.38268343f,0.60876143f } + }; + if (n_long_bands) + { + drmp3_L3_imdct36(grbuf, overlap, g_mdct_window[0], n_long_bands); + grbuf += 18*n_long_bands; + overlap += 9*n_long_bands; + } + if (block_type == DRMP3_SHORT_BLOCK_TYPE) + drmp3_L3_imdct_short(grbuf, overlap, 32 - n_long_bands); + else + drmp3_L3_imdct36(grbuf, overlap, g_mdct_window[block_type == DRMP3_STOP_BLOCK_TYPE], 32 - n_long_bands); +} + +static void drmp3_L3_save_reservoir(drmp3dec *h, drmp3dec_scratch *s) +{ + int pos = (s->bs.pos + 7)/8u; + int remains = s->bs.limit/8u - pos; + if (remains > DRMP3_MAX_BITRESERVOIR_BYTES) + { + pos += remains - DRMP3_MAX_BITRESERVOIR_BYTES; + remains = DRMP3_MAX_BITRESERVOIR_BYTES; + } + if (remains > 0) + { + DRMP3_MOVE_MEMORY(h->reserv_buf, s->maindata + pos, remains); + } + h->reserv = remains; +} + +static int drmp3_L3_restore_reservoir(drmp3dec *h, drmp3_bs *bs, drmp3dec_scratch *s, int main_data_begin) +{ + int frame_bytes = (bs->limit - bs->pos)/8; + int bytes_have = DRMP3_MIN(h->reserv, main_data_begin); + DRMP3_COPY_MEMORY(s->maindata, h->reserv_buf + DRMP3_MAX(0, h->reserv - main_data_begin), DRMP3_MIN(h->reserv, main_data_begin)); + DRMP3_COPY_MEMORY(s->maindata + bytes_have, bs->buf + bs->pos/8, frame_bytes); + drmp3_bs_init(&s->bs, s->maindata, bytes_have + frame_bytes); + return h->reserv >= main_data_begin; +} + +static void drmp3_L3_decode(drmp3dec *h, drmp3dec_scratch *s, drmp3_L3_gr_info *gr_info, int nch) +{ + int ch; + + for (ch = 0; ch < nch; ch++) + { + int layer3gr_limit = s->bs.pos + gr_info[ch].part_23_length; + drmp3_L3_decode_scalefactors(h->header, s->ist_pos[ch], &s->bs, gr_info + ch, s->scf, ch); + drmp3_L3_huffman(s->grbuf[ch], &s->bs, gr_info + ch, s->scf, layer3gr_limit); + } + + if (DRMP3_HDR_TEST_I_STEREO(h->header)) + { + drmp3_L3_intensity_stereo(s->grbuf[0], s->ist_pos[1], gr_info, h->header); + } else if (DRMP3_HDR_IS_MS_STEREO(h->header)) + { + drmp3_L3_midside_stereo(s->grbuf[0], 576); + } + + for (ch = 0; ch < nch; ch++, gr_info++) + { + int aa_bands = 31; + int n_long_bands = (gr_info->mixed_block_flag ? 2 : 0) << (int)(DRMP3_HDR_GET_MY_SAMPLE_RATE(h->header) == 2); + + if (gr_info->n_short_sfb) + { + aa_bands = n_long_bands - 1; + drmp3_L3_reorder(s->grbuf[ch] + n_long_bands*18, s->syn[0], gr_info->sfbtab + gr_info->n_long_sfb); + } + + drmp3_L3_antialias(s->grbuf[ch], aa_bands); + drmp3_L3_imdct_gr(s->grbuf[ch], h->mdct_overlap[ch], gr_info->block_type, n_long_bands); + drmp3_L3_change_sign(s->grbuf[ch]); + } +} + +static void drmp3d_DCT_II(float *grbuf, int n) +{ + static const float g_sec[24] = { + 10.19000816f,0.50060302f,0.50241929f,3.40760851f,0.50547093f,0.52249861f,2.05778098f,0.51544732f,0.56694406f,1.48416460f,0.53104258f,0.64682180f,1.16943991f,0.55310392f,0.78815460f,0.97256821f,0.58293498f,1.06067765f,0.83934963f,0.62250412f,1.72244716f,0.74453628f,0.67480832f,5.10114861f + }; + int i, k = 0; +#if DRMP3_HAVE_SIMD + if (drmp3_have_simd()) for (; k < n; k += 4) + { + drmp3_f4 t[4][8], *x; + float *y = grbuf + k; + + for (x = t[0], i = 0; i < 8; i++, x++) + { + drmp3_f4 x0 = DRMP3_VLD(&y[i*18]); + drmp3_f4 x1 = DRMP3_VLD(&y[(15 - i)*18]); + drmp3_f4 x2 = DRMP3_VLD(&y[(16 + i)*18]); + drmp3_f4 x3 = DRMP3_VLD(&y[(31 - i)*18]); + drmp3_f4 t0 = DRMP3_VADD(x0, x3); + drmp3_f4 t1 = DRMP3_VADD(x1, x2); + drmp3_f4 t2 = DRMP3_VMUL_S(DRMP3_VSUB(x1, x2), g_sec[3*i + 0]); + drmp3_f4 t3 = DRMP3_VMUL_S(DRMP3_VSUB(x0, x3), g_sec[3*i + 1]); + x[0] = DRMP3_VADD(t0, t1); + x[8] = DRMP3_VMUL_S(DRMP3_VSUB(t0, t1), g_sec[3*i + 2]); + x[16] = DRMP3_VADD(t3, t2); + x[24] = DRMP3_VMUL_S(DRMP3_VSUB(t3, t2), g_sec[3*i + 2]); + } + for (x = t[0], i = 0; i < 4; i++, x += 8) + { + drmp3_f4 x0 = x[0], x1 = x[1], x2 = x[2], x3 = x[3], x4 = x[4], x5 = x[5], x6 = x[6], x7 = x[7], xt; + xt = DRMP3_VSUB(x0, x7); x0 = DRMP3_VADD(x0, x7); + x7 = DRMP3_VSUB(x1, x6); x1 = DRMP3_VADD(x1, x6); + x6 = DRMP3_VSUB(x2, x5); x2 = DRMP3_VADD(x2, x5); + x5 = DRMP3_VSUB(x3, x4); x3 = DRMP3_VADD(x3, x4); + x4 = DRMP3_VSUB(x0, x3); x0 = DRMP3_VADD(x0, x3); + x3 = DRMP3_VSUB(x1, x2); x1 = DRMP3_VADD(x1, x2); + x[0] = DRMP3_VADD(x0, x1); + x[4] = DRMP3_VMUL_S(DRMP3_VSUB(x0, x1), 0.70710677f); + x5 = DRMP3_VADD(x5, x6); + x6 = DRMP3_VMUL_S(DRMP3_VADD(x6, x7), 0.70710677f); + x7 = DRMP3_VADD(x7, xt); + x3 = DRMP3_VMUL_S(DRMP3_VADD(x3, x4), 0.70710677f); + x5 = DRMP3_VSUB(x5, DRMP3_VMUL_S(x7, 0.198912367f)); /* rotate by PI/8 */ + x7 = DRMP3_VADD(x7, DRMP3_VMUL_S(x5, 0.382683432f)); + x5 = DRMP3_VSUB(x5, DRMP3_VMUL_S(x7, 0.198912367f)); + x0 = DRMP3_VSUB(xt, x6); xt = DRMP3_VADD(xt, x6); + x[1] = DRMP3_VMUL_S(DRMP3_VADD(xt, x7), 0.50979561f); + x[2] = DRMP3_VMUL_S(DRMP3_VADD(x4, x3), 0.54119611f); + x[3] = DRMP3_VMUL_S(DRMP3_VSUB(x0, x5), 0.60134488f); + x[5] = DRMP3_VMUL_S(DRMP3_VADD(x0, x5), 0.89997619f); + x[6] = DRMP3_VMUL_S(DRMP3_VSUB(x4, x3), 1.30656302f); + x[7] = DRMP3_VMUL_S(DRMP3_VSUB(xt, x7), 2.56291556f); + } + + if (k > n - 3) + { +#if DRMP3_HAVE_SSE +#define DRMP3_VSAVE2(i, v) _mm_storel_pi((__m64 *)(void*)&y[i*18], v) +#else +#define DRMP3_VSAVE2(i, v) vst1_f32((float32_t *)&y[(i)*18], vget_low_f32(v)) +#endif + for (i = 0; i < 7; i++, y += 4*18) + { + drmp3_f4 s = DRMP3_VADD(t[3][i], t[3][i + 1]); + DRMP3_VSAVE2(0, t[0][i]); + DRMP3_VSAVE2(1, DRMP3_VADD(t[2][i], s)); + DRMP3_VSAVE2(2, DRMP3_VADD(t[1][i], t[1][i + 1])); + DRMP3_VSAVE2(3, DRMP3_VADD(t[2][1 + i], s)); + } + DRMP3_VSAVE2(0, t[0][7]); + DRMP3_VSAVE2(1, DRMP3_VADD(t[2][7], t[3][7])); + DRMP3_VSAVE2(2, t[1][7]); + DRMP3_VSAVE2(3, t[3][7]); + } else + { +#define DRMP3_VSAVE4(i, v) DRMP3_VSTORE(&y[(i)*18], v) + for (i = 0; i < 7; i++, y += 4*18) + { + drmp3_f4 s = DRMP3_VADD(t[3][i], t[3][i + 1]); + DRMP3_VSAVE4(0, t[0][i]); + DRMP3_VSAVE4(1, DRMP3_VADD(t[2][i], s)); + DRMP3_VSAVE4(2, DRMP3_VADD(t[1][i], t[1][i + 1])); + DRMP3_VSAVE4(3, DRMP3_VADD(t[2][1 + i], s)); + } + DRMP3_VSAVE4(0, t[0][7]); + DRMP3_VSAVE4(1, DRMP3_VADD(t[2][7], t[3][7])); + DRMP3_VSAVE4(2, t[1][7]); + DRMP3_VSAVE4(3, t[3][7]); + } + } else +#endif +#ifdef DR_MP3_ONLY_SIMD + {} /* for HAVE_SIMD=1, MINIMP3_ONLY_SIMD=1 case we do not need non-intrinsic "else" branch */ +#else + for (; k < n; k++) + { + float t[4][8], *x, *y = grbuf + k; + + for (x = t[0], i = 0; i < 8; i++, x++) + { + float x0 = y[i*18]; + float x1 = y[(15 - i)*18]; + float x2 = y[(16 + i)*18]; + float x3 = y[(31 - i)*18]; + float t0 = x0 + x3; + float t1 = x1 + x2; + float t2 = (x1 - x2)*g_sec[3*i + 0]; + float t3 = (x0 - x3)*g_sec[3*i + 1]; + x[0] = t0 + t1; + x[8] = (t0 - t1)*g_sec[3*i + 2]; + x[16] = t3 + t2; + x[24] = (t3 - t2)*g_sec[3*i + 2]; + } + for (x = t[0], i = 0; i < 4; i++, x += 8) + { + float x0 = x[0], x1 = x[1], x2 = x[2], x3 = x[3], x4 = x[4], x5 = x[5], x6 = x[6], x7 = x[7], xt; + xt = x0 - x7; x0 += x7; + x7 = x1 - x6; x1 += x6; + x6 = x2 - x5; x2 += x5; + x5 = x3 - x4; x3 += x4; + x4 = x0 - x3; x0 += x3; + x3 = x1 - x2; x1 += x2; + x[0] = x0 + x1; + x[4] = (x0 - x1)*0.70710677f; + x5 = x5 + x6; + x6 = (x6 + x7)*0.70710677f; + x7 = x7 + xt; + x3 = (x3 + x4)*0.70710677f; + x5 -= x7*0.198912367f; /* rotate by PI/8 */ + x7 += x5*0.382683432f; + x5 -= x7*0.198912367f; + x0 = xt - x6; xt += x6; + x[1] = (xt + x7)*0.50979561f; + x[2] = (x4 + x3)*0.54119611f; + x[3] = (x0 - x5)*0.60134488f; + x[5] = (x0 + x5)*0.89997619f; + x[6] = (x4 - x3)*1.30656302f; + x[7] = (xt - x7)*2.56291556f; + + } + for (i = 0; i < 7; i++, y += 4*18) + { + y[0*18] = t[0][i]; + y[1*18] = t[2][i] + t[3][i] + t[3][i + 1]; + y[2*18] = t[1][i] + t[1][i + 1]; + y[3*18] = t[2][i + 1] + t[3][i] + t[3][i + 1]; + } + y[0*18] = t[0][7]; + y[1*18] = t[2][7] + t[3][7]; + y[2*18] = t[1][7]; + y[3*18] = t[3][7]; + } +#endif +} + +#ifndef DR_MP3_FLOAT_OUTPUT +typedef drmp3_int16 drmp3d_sample_t; + +static drmp3_int16 drmp3d_scale_pcm(float sample) +{ + drmp3_int16 s; +#if DRMP3_HAVE_ARMV6 + drmp3_int32 s32 = (drmp3_int32)(sample + .5f); + s32 -= (s32 < 0); + s = (drmp3_int16)drmp3_clip_int16_arm(s32); +#else + if (sample >= 32766.5f) return (drmp3_int16) 32767; + if (sample <= -32767.5f) return (drmp3_int16)-32768; + s = (drmp3_int16)(sample + .5f); + s -= (s < 0); /* away from zero, to be compliant */ +#endif + return s; +} +#else +typedef float drmp3d_sample_t; + +static float drmp3d_scale_pcm(float sample) +{ + return sample*(1.f/32768.f); +} +#endif + +static void drmp3d_synth_pair(drmp3d_sample_t *pcm, int nch, const float *z) +{ + float a; + a = (z[14*64] - z[ 0]) * 29; + a += (z[ 1*64] + z[13*64]) * 213; + a += (z[12*64] - z[ 2*64]) * 459; + a += (z[ 3*64] + z[11*64]) * 2037; + a += (z[10*64] - z[ 4*64]) * 5153; + a += (z[ 5*64] + z[ 9*64]) * 6574; + a += (z[ 8*64] - z[ 6*64]) * 37489; + a += z[ 7*64] * 75038; + pcm[0] = drmp3d_scale_pcm(a); + + z += 2; + a = z[14*64] * 104; + a += z[12*64] * 1567; + a += z[10*64] * 9727; + a += z[ 8*64] * 64019; + a += z[ 6*64] * -9975; + a += z[ 4*64] * -45; + a += z[ 2*64] * 146; + a += z[ 0*64] * -5; + pcm[16*nch] = drmp3d_scale_pcm(a); +} + +static void drmp3d_synth(float *xl, drmp3d_sample_t *dstl, int nch, float *lins) +{ + int i; + float *xr = xl + 576*(nch - 1); + drmp3d_sample_t *dstr = dstl + (nch - 1); + + static const float g_win[] = { + -1,26,-31,208,218,401,-519,2063,2000,4788,-5517,7134,5959,35640,-39336,74992, + -1,24,-35,202,222,347,-581,2080,1952,4425,-5879,7640,5288,33791,-41176,74856, + -1,21,-38,196,225,294,-645,2087,1893,4063,-6237,8092,4561,31947,-43006,74630, + -1,19,-41,190,227,244,-711,2085,1822,3705,-6589,8492,3776,30112,-44821,74313, + -1,17,-45,183,228,197,-779,2075,1739,3351,-6935,8840,2935,28289,-46617,73908, + -1,16,-49,176,228,153,-848,2057,1644,3004,-7271,9139,2037,26482,-48390,73415, + -2,14,-53,169,227,111,-919,2032,1535,2663,-7597,9389,1082,24694,-50137,72835, + -2,13,-58,161,224,72,-991,2001,1414,2330,-7910,9592,70,22929,-51853,72169, + -2,11,-63,154,221,36,-1064,1962,1280,2006,-8209,9750,-998,21189,-53534,71420, + -2,10,-68,147,215,2,-1137,1919,1131,1692,-8491,9863,-2122,19478,-55178,70590, + -3,9,-73,139,208,-29,-1210,1870,970,1388,-8755,9935,-3300,17799,-56778,69679, + -3,8,-79,132,200,-57,-1283,1817,794,1095,-8998,9966,-4533,16155,-58333,68692, + -4,7,-85,125,189,-83,-1356,1759,605,814,-9219,9959,-5818,14548,-59838,67629, + -4,7,-91,117,177,-106,-1428,1698,402,545,-9416,9916,-7154,12980,-61289,66494, + -5,6,-97,111,163,-127,-1498,1634,185,288,-9585,9838,-8540,11455,-62684,65290 + }; + float *zlin = lins + 15*64; + const float *w = g_win; + + zlin[4*15] = xl[18*16]; + zlin[4*15 + 1] = xr[18*16]; + zlin[4*15 + 2] = xl[0]; + zlin[4*15 + 3] = xr[0]; + + zlin[4*31] = xl[1 + 18*16]; + zlin[4*31 + 1] = xr[1 + 18*16]; + zlin[4*31 + 2] = xl[1]; + zlin[4*31 + 3] = xr[1]; + + drmp3d_synth_pair(dstr, nch, lins + 4*15 + 1); + drmp3d_synth_pair(dstr + 32*nch, nch, lins + 4*15 + 64 + 1); + drmp3d_synth_pair(dstl, nch, lins + 4*15); + drmp3d_synth_pair(dstl + 32*nch, nch, lins + 4*15 + 64); + +#if DRMP3_HAVE_SIMD + if (drmp3_have_simd()) for (i = 14; i >= 0; i--) + { +#define DRMP3_VLOAD(k) drmp3_f4 w0 = DRMP3_VSET(*w++); drmp3_f4 w1 = DRMP3_VSET(*w++); drmp3_f4 vz = DRMP3_VLD(&zlin[4*i - 64*k]); drmp3_f4 vy = DRMP3_VLD(&zlin[4*i - 64*(15 - k)]); +#define DRMP3_V0(k) { DRMP3_VLOAD(k) b = DRMP3_VADD(DRMP3_VMUL(vz, w1), DRMP3_VMUL(vy, w0)) ; a = DRMP3_VSUB(DRMP3_VMUL(vz, w0), DRMP3_VMUL(vy, w1)); } +#define DRMP3_V1(k) { DRMP3_VLOAD(k) b = DRMP3_VADD(b, DRMP3_VADD(DRMP3_VMUL(vz, w1), DRMP3_VMUL(vy, w0))); a = DRMP3_VADD(a, DRMP3_VSUB(DRMP3_VMUL(vz, w0), DRMP3_VMUL(vy, w1))); } +#define DRMP3_V2(k) { DRMP3_VLOAD(k) b = DRMP3_VADD(b, DRMP3_VADD(DRMP3_VMUL(vz, w1), DRMP3_VMUL(vy, w0))); a = DRMP3_VADD(a, DRMP3_VSUB(DRMP3_VMUL(vy, w1), DRMP3_VMUL(vz, w0))); } + drmp3_f4 a, b; + zlin[4*i] = xl[18*(31 - i)]; + zlin[4*i + 1] = xr[18*(31 - i)]; + zlin[4*i + 2] = xl[1 + 18*(31 - i)]; + zlin[4*i + 3] = xr[1 + 18*(31 - i)]; + zlin[4*i + 64] = xl[1 + 18*(1 + i)]; + zlin[4*i + 64 + 1] = xr[1 + 18*(1 + i)]; + zlin[4*i - 64 + 2] = xl[18*(1 + i)]; + zlin[4*i - 64 + 3] = xr[18*(1 + i)]; + + DRMP3_V0(0) DRMP3_V2(1) DRMP3_V1(2) DRMP3_V2(3) DRMP3_V1(4) DRMP3_V2(5) DRMP3_V1(6) DRMP3_V2(7) + + { +#ifndef DR_MP3_FLOAT_OUTPUT +#if DRMP3_HAVE_SSE + static const drmp3_f4 g_max = { 32767.0f, 32767.0f, 32767.0f, 32767.0f }; + static const drmp3_f4 g_min = { -32768.0f, -32768.0f, -32768.0f, -32768.0f }; + __m128i pcm8 = _mm_packs_epi32(_mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(a, g_max), g_min)), + _mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(b, g_max), g_min))); + dstr[(15 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 1); + dstr[(17 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 5); + dstl[(15 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 0); + dstl[(17 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 4); + dstr[(47 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 3); + dstr[(49 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 7); + dstl[(47 - i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 2); + dstl[(49 + i)*nch] = (drmp3_int16)_mm_extract_epi16(pcm8, 6); +#else + int16x4_t pcma, pcmb; + a = DRMP3_VADD(a, DRMP3_VSET(0.5f)); + b = DRMP3_VADD(b, DRMP3_VSET(0.5f)); + pcma = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(a), vreinterpretq_s32_u32(vcltq_f32(a, DRMP3_VSET(0))))); + pcmb = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(b), vreinterpretq_s32_u32(vcltq_f32(b, DRMP3_VSET(0))))); + vst1_lane_s16(dstr + (15 - i)*nch, pcma, 1); + vst1_lane_s16(dstr + (17 + i)*nch, pcmb, 1); + vst1_lane_s16(dstl + (15 - i)*nch, pcma, 0); + vst1_lane_s16(dstl + (17 + i)*nch, pcmb, 0); + vst1_lane_s16(dstr + (47 - i)*nch, pcma, 3); + vst1_lane_s16(dstr + (49 + i)*nch, pcmb, 3); + vst1_lane_s16(dstl + (47 - i)*nch, pcma, 2); + vst1_lane_s16(dstl + (49 + i)*nch, pcmb, 2); +#endif +#else + #if DRMP3_HAVE_SSE + static const drmp3_f4 g_scale = { 1.0f/32768.0f, 1.0f/32768.0f, 1.0f/32768.0f, 1.0f/32768.0f }; + #else + const drmp3_f4 g_scale = vdupq_n_f32(1.0f/32768.0f); + #endif + a = DRMP3_VMUL(a, g_scale); + b = DRMP3_VMUL(b, g_scale); +#if DRMP3_HAVE_SSE + _mm_store_ss(dstr + (15 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(1, 1, 1, 1))); + _mm_store_ss(dstr + (17 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(1, 1, 1, 1))); + _mm_store_ss(dstl + (15 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(0, 0, 0, 0))); + _mm_store_ss(dstl + (17 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(0, 0, 0, 0))); + _mm_store_ss(dstr + (47 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(3, 3, 3, 3))); + _mm_store_ss(dstr + (49 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(3, 3, 3, 3))); + _mm_store_ss(dstl + (47 - i)*nch, _mm_shuffle_ps(a, a, _MM_SHUFFLE(2, 2, 2, 2))); + _mm_store_ss(dstl + (49 + i)*nch, _mm_shuffle_ps(b, b, _MM_SHUFFLE(2, 2, 2, 2))); +#else + vst1q_lane_f32(dstr + (15 - i)*nch, a, 1); + vst1q_lane_f32(dstr + (17 + i)*nch, b, 1); + vst1q_lane_f32(dstl + (15 - i)*nch, a, 0); + vst1q_lane_f32(dstl + (17 + i)*nch, b, 0); + vst1q_lane_f32(dstr + (47 - i)*nch, a, 3); + vst1q_lane_f32(dstr + (49 + i)*nch, b, 3); + vst1q_lane_f32(dstl + (47 - i)*nch, a, 2); + vst1q_lane_f32(dstl + (49 + i)*nch, b, 2); +#endif +#endif /* DR_MP3_FLOAT_OUTPUT */ + } + } else +#endif +#ifdef DR_MP3_ONLY_SIMD + {} /* for HAVE_SIMD=1, MINIMP3_ONLY_SIMD=1 case we do not need non-intrinsic "else" branch */ +#else + for (i = 14; i >= 0; i--) + { +#define DRMP3_LOAD(k) float w0 = *w++; float w1 = *w++; float *vz = &zlin[4*i - k*64]; float *vy = &zlin[4*i - (15 - k)*64]; +#define DRMP3_S0(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] = vz[j]*w1 + vy[j]*w0, a[j] = vz[j]*w0 - vy[j]*w1; } +#define DRMP3_S1(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] += vz[j]*w1 + vy[j]*w0, a[j] += vz[j]*w0 - vy[j]*w1; } +#define DRMP3_S2(k) { int j; DRMP3_LOAD(k); for (j = 0; j < 4; j++) b[j] += vz[j]*w1 + vy[j]*w0, a[j] += vy[j]*w1 - vz[j]*w0; } + float a[4], b[4]; + + zlin[4*i] = xl[18*(31 - i)]; + zlin[4*i + 1] = xr[18*(31 - i)]; + zlin[4*i + 2] = xl[1 + 18*(31 - i)]; + zlin[4*i + 3] = xr[1 + 18*(31 - i)]; + zlin[4*(i + 16)] = xl[1 + 18*(1 + i)]; + zlin[4*(i + 16) + 1] = xr[1 + 18*(1 + i)]; + zlin[4*(i - 16) + 2] = xl[18*(1 + i)]; + zlin[4*(i - 16) + 3] = xr[18*(1 + i)]; + + DRMP3_S0(0) DRMP3_S2(1) DRMP3_S1(2) DRMP3_S2(3) DRMP3_S1(4) DRMP3_S2(5) DRMP3_S1(6) DRMP3_S2(7) + + dstr[(15 - i)*nch] = drmp3d_scale_pcm(a[1]); + dstr[(17 + i)*nch] = drmp3d_scale_pcm(b[1]); + dstl[(15 - i)*nch] = drmp3d_scale_pcm(a[0]); + dstl[(17 + i)*nch] = drmp3d_scale_pcm(b[0]); + dstr[(47 - i)*nch] = drmp3d_scale_pcm(a[3]); + dstr[(49 + i)*nch] = drmp3d_scale_pcm(b[3]); + dstl[(47 - i)*nch] = drmp3d_scale_pcm(a[2]); + dstl[(49 + i)*nch] = drmp3d_scale_pcm(b[2]); + } +#endif +} + +static void drmp3d_synth_granule(float *qmf_state, float *grbuf, int nbands, int nch, drmp3d_sample_t *pcm, float *lins) +{ + int i; + for (i = 0; i < nch; i++) + { + drmp3d_DCT_II(grbuf + 576*i, nbands); + } + + DRMP3_COPY_MEMORY(lins, qmf_state, sizeof(float)*15*64); + + for (i = 0; i < nbands; i += 2) + { + drmp3d_synth(grbuf + i, pcm + 32*nch*i, nch, lins + i*64); + } +#ifndef DR_MP3_NONSTANDARD_BUT_LOGICAL + if (nch == 1) + { + for (i = 0; i < 15*64; i += 2) + { + qmf_state[i] = lins[nbands*64 + i]; + } + } else +#endif + { + DRMP3_COPY_MEMORY(qmf_state, lins + nbands*64, sizeof(float)*15*64); + } +} + +static int drmp3d_match_frame(const drmp3_uint8 *hdr, int mp3_bytes, int frame_bytes) +{ + int i, nmatch; + for (i = 0, nmatch = 0; nmatch < DRMP3_MAX_FRAME_SYNC_MATCHES; nmatch++) + { + i += drmp3_hdr_frame_bytes(hdr + i, frame_bytes) + drmp3_hdr_padding(hdr + i); + if (i + DRMP3_HDR_SIZE > mp3_bytes) + return nmatch > 0; + if (!drmp3_hdr_compare(hdr, hdr + i)) + return 0; + } + return 1; +} + +static int drmp3d_find_frame(const drmp3_uint8 *mp3, int mp3_bytes, int *free_format_bytes, int *ptr_frame_bytes) +{ + int i, k; + for (i = 0; i < mp3_bytes - DRMP3_HDR_SIZE; i++, mp3++) + { + if (drmp3_hdr_valid(mp3)) + { + int frame_bytes = drmp3_hdr_frame_bytes(mp3, *free_format_bytes); + int frame_and_padding = frame_bytes + drmp3_hdr_padding(mp3); + + for (k = DRMP3_HDR_SIZE; !frame_bytes && k < DRMP3_MAX_FREE_FORMAT_FRAME_SIZE && i + 2*k < mp3_bytes - DRMP3_HDR_SIZE; k++) + { + if (drmp3_hdr_compare(mp3, mp3 + k)) + { + int fb = k - drmp3_hdr_padding(mp3); + int nextfb = fb + drmp3_hdr_padding(mp3 + k); + if (i + k + nextfb + DRMP3_HDR_SIZE > mp3_bytes || !drmp3_hdr_compare(mp3, mp3 + k + nextfb)) + continue; + frame_and_padding = k; + frame_bytes = fb; + *free_format_bytes = fb; + } + } + + if ((frame_bytes && i + frame_and_padding <= mp3_bytes && + drmp3d_match_frame(mp3, mp3_bytes - i, frame_bytes)) || + (!i && frame_and_padding == mp3_bytes)) + { + *ptr_frame_bytes = frame_and_padding; + return i; + } + *free_format_bytes = 0; + } + } + *ptr_frame_bytes = 0; + return mp3_bytes; +} + +DRMP3_API void drmp3dec_init(drmp3dec *dec) +{ + dec->header[0] = 0; +} + +DRMP3_API int drmp3dec_decode_frame(drmp3dec *dec, const drmp3_uint8 *mp3, int mp3_bytes, void *pcm, drmp3dec_frame_info *info) +{ + int i = 0, igr, frame_size = 0, success = 1; + const drmp3_uint8 *hdr; + drmp3_bs bs_frame[1]; + + if (mp3_bytes > 4 && dec->header[0] == 0xff && drmp3_hdr_compare(dec->header, mp3)) + { + frame_size = drmp3_hdr_frame_bytes(mp3, dec->free_format_bytes) + drmp3_hdr_padding(mp3); + if (frame_size != mp3_bytes && (frame_size + DRMP3_HDR_SIZE > mp3_bytes || !drmp3_hdr_compare(mp3, mp3 + frame_size))) + { + frame_size = 0; + } + } + if (!frame_size) + { + DRMP3_ZERO_MEMORY(dec, sizeof(drmp3dec)); + i = drmp3d_find_frame(mp3, mp3_bytes, &dec->free_format_bytes, &frame_size); + if (!frame_size || i + frame_size > mp3_bytes) + { + info->frame_bytes = i; + return 0; + } + } + + hdr = mp3 + i; + DRMP3_COPY_MEMORY(dec->header, hdr, DRMP3_HDR_SIZE); + info->frame_bytes = i + frame_size; + info->channels = DRMP3_HDR_IS_MONO(hdr) ? 1 : 2; + info->sample_rate = drmp3_hdr_sample_rate_hz(hdr); + info->layer = 4 - DRMP3_HDR_GET_LAYER(hdr); + info->bitrate_kbps = drmp3_hdr_bitrate_kbps(hdr); + + drmp3_bs_init(bs_frame, hdr + DRMP3_HDR_SIZE, frame_size - DRMP3_HDR_SIZE); + if (DRMP3_HDR_IS_CRC(hdr)) + { + drmp3_bs_get_bits(bs_frame, 16); + } + + if (info->layer == 3) + { + int main_data_begin = drmp3_L3_read_side_info(bs_frame, dec->scratch.gr_info, hdr); + if (main_data_begin < 0 || bs_frame->pos > bs_frame->limit) + { + drmp3dec_init(dec); + return 0; + } + success = drmp3_L3_restore_reservoir(dec, bs_frame, &dec->scratch, main_data_begin); + if (success && pcm != NULL) + { + for (igr = 0; igr < (DRMP3_HDR_TEST_MPEG1(hdr) ? 2 : 1); igr++, pcm = DRMP3_OFFSET_PTR(pcm, sizeof(drmp3d_sample_t)*576*info->channels)) + { + DRMP3_ZERO_MEMORY(dec->scratch.grbuf[0], 576*2*sizeof(float)); + drmp3_L3_decode(dec, &dec->scratch, dec->scratch.gr_info + igr*info->channels, info->channels); + drmp3d_synth_granule(dec->qmf_state, dec->scratch.grbuf[0], 18, info->channels, (drmp3d_sample_t*)pcm, dec->scratch.syn[0]); + } + } + drmp3_L3_save_reservoir(dec, &dec->scratch); + } else + { +#ifdef DR_MP3_ONLY_MP3 + return 0; +#else + drmp3_L12_scale_info sci[1]; + + if (pcm == NULL) { + return drmp3_hdr_frame_samples(hdr); + } + + drmp3_L12_read_scale_info(hdr, bs_frame, sci); + + DRMP3_ZERO_MEMORY(dec->scratch.grbuf[0], 576*2*sizeof(float)); + for (i = 0, igr = 0; igr < 3; igr++) + { + if (12 == (i += drmp3_L12_dequantize_granule(dec->scratch.grbuf[0] + i, bs_frame, sci, info->layer | 1))) + { + i = 0; + drmp3_L12_apply_scf_384(sci, sci->scf + igr, dec->scratch.grbuf[0]); + drmp3d_synth_granule(dec->qmf_state, dec->scratch.grbuf[0], 12, info->channels, (drmp3d_sample_t*)pcm, dec->scratch.syn[0]); + DRMP3_ZERO_MEMORY(dec->scratch.grbuf[0], 576*2*sizeof(float)); + pcm = DRMP3_OFFSET_PTR(pcm, sizeof(drmp3d_sample_t)*384*info->channels); + } + if (bs_frame->pos > bs_frame->limit) + { + drmp3dec_init(dec); + return 0; + } + } +#endif + } + + return success*drmp3_hdr_frame_samples(dec->header); +} + +DRMP3_API void drmp3dec_f32_to_s16(const float *in, drmp3_int16 *out, size_t num_samples) +{ + size_t i = 0; +#if DRMP3_HAVE_SIMD + size_t aligned_count = num_samples & ~7; + for(; i < aligned_count; i+=8) + { + drmp3_f4 scale = DRMP3_VSET(32768.0f); + drmp3_f4 a = DRMP3_VMUL(DRMP3_VLD(&in[i ]), scale); + drmp3_f4 b = DRMP3_VMUL(DRMP3_VLD(&in[i+4]), scale); +#if DRMP3_HAVE_SSE + drmp3_f4 s16max = DRMP3_VSET( 32767.0f); + drmp3_f4 s16min = DRMP3_VSET(-32768.0f); + __m128i pcm8 = _mm_packs_epi32(_mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(a, s16max), s16min)), + _mm_cvtps_epi32(_mm_max_ps(_mm_min_ps(b, s16max), s16min))); + out[i ] = (drmp3_int16)_mm_extract_epi16(pcm8, 0); + out[i+1] = (drmp3_int16)_mm_extract_epi16(pcm8, 1); + out[i+2] = (drmp3_int16)_mm_extract_epi16(pcm8, 2); + out[i+3] = (drmp3_int16)_mm_extract_epi16(pcm8, 3); + out[i+4] = (drmp3_int16)_mm_extract_epi16(pcm8, 4); + out[i+5] = (drmp3_int16)_mm_extract_epi16(pcm8, 5); + out[i+6] = (drmp3_int16)_mm_extract_epi16(pcm8, 6); + out[i+7] = (drmp3_int16)_mm_extract_epi16(pcm8, 7); +#else + int16x4_t pcma, pcmb; + a = DRMP3_VADD(a, DRMP3_VSET(0.5f)); + b = DRMP3_VADD(b, DRMP3_VSET(0.5f)); + pcma = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(a), vreinterpretq_s32_u32(vcltq_f32(a, DRMP3_VSET(0))))); + pcmb = vqmovn_s32(vqaddq_s32(vcvtq_s32_f32(b), vreinterpretq_s32_u32(vcltq_f32(b, DRMP3_VSET(0))))); + vst1_lane_s16(out+i , pcma, 0); + vst1_lane_s16(out+i+1, pcma, 1); + vst1_lane_s16(out+i+2, pcma, 2); + vst1_lane_s16(out+i+3, pcma, 3); + vst1_lane_s16(out+i+4, pcmb, 0); + vst1_lane_s16(out+i+5, pcmb, 1); + vst1_lane_s16(out+i+6, pcmb, 2); + vst1_lane_s16(out+i+7, pcmb, 3); +#endif + } +#endif + for(; i < num_samples; i++) + { + float sample = in[i] * 32768.0f; + if (sample >= 32766.5f) + out[i] = (drmp3_int16) 32767; + else if (sample <= -32767.5f) + out[i] = (drmp3_int16)-32768; + else + { + short s = (drmp3_int16)(sample + .5f); + s -= (s < 0); /* away from zero, to be compliant */ + out[i] = s; + } + } +} + + + +/************************************************************************************************************************************************************ + + Main Public API + + ************************************************************************************************************************************************************/ +/* SIZE_MAX */ +#if defined(SIZE_MAX) + #define DRMP3_SIZE_MAX SIZE_MAX +#else + #if defined(_WIN64) || defined(_LP64) || defined(__LP64__) + #define DRMP3_SIZE_MAX ((drmp3_uint64)0xFFFFFFFFFFFFFFFF) + #else + #define DRMP3_SIZE_MAX 0xFFFFFFFF + #endif +#endif +/* End SIZE_MAX */ + +/* Options. */ +#ifndef DRMP3_SEEK_LEADING_MP3_FRAMES +#define DRMP3_SEEK_LEADING_MP3_FRAMES 2 +#endif + +#define DRMP3_MIN_DATA_CHUNK_SIZE 16384 + +/* The size in bytes of each chunk of data to read from the MP3 stream. minimp3 recommends at least 16K, but in an attempt to reduce data movement I'm making this slightly larger. */ +#ifndef DRMP3_DATA_CHUNK_SIZE +#define DRMP3_DATA_CHUNK_SIZE (DRMP3_MIN_DATA_CHUNK_SIZE*4) +#endif + + +#define DRMP3_COUNTOF(x) (sizeof(x) / sizeof(x[0])) +#define DRMP3_CLAMP(x, lo, hi) (DRMP3_MAX(lo, DRMP3_MIN(x, hi))) + +#ifndef DRMP3_PI_D +#define DRMP3_PI_D 3.14159265358979323846264 +#endif + +#define DRMP3_DEFAULT_RESAMPLER_LPF_ORDER 2 + +static DRMP3_INLINE float drmp3_mix_f32(float x, float y, float a) +{ + return x*(1-a) + y*a; +} +static DRMP3_INLINE float drmp3_mix_f32_fast(float x, float y, float a) +{ + float r0 = (y - x); + float r1 = r0*a; + return x + r1; + /*return x + (y - x)*a;*/ +} + + +/* +Greatest common factor using Euclid's algorithm iteratively. +*/ +static DRMP3_INLINE drmp3_uint32 drmp3_gcf_u32(drmp3_uint32 a, drmp3_uint32 b) +{ + for (;;) { + if (b == 0) { + break; + } else { + drmp3_uint32 t = a; + a = b; + b = t % a; + } + } + + return a; +} + + +static void* drmp3__malloc_default(size_t sz, void* pUserData) +{ + (void)pUserData; + return DRMP3_MALLOC(sz); +} + +static void* drmp3__realloc_default(void* p, size_t sz, void* pUserData) +{ + (void)pUserData; + return DRMP3_REALLOC(p, sz); +} + +static void drmp3__free_default(void* p, void* pUserData) +{ + (void)pUserData; + DRMP3_FREE(p); +} + + +static void* drmp3__malloc_from_callbacks(size_t sz, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks == NULL) { + return NULL; + } + + if (pAllocationCallbacks->onMalloc != NULL) { + return pAllocationCallbacks->onMalloc(sz, pAllocationCallbacks->pUserData); + } + + /* Try using realloc(). */ + if (pAllocationCallbacks->onRealloc != NULL) { + return pAllocationCallbacks->onRealloc(NULL, sz, pAllocationCallbacks->pUserData); + } + + return NULL; +} + +static void* drmp3__realloc_from_callbacks(void* p, size_t szNew, size_t szOld, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks == NULL) { + return NULL; + } + + if (pAllocationCallbacks->onRealloc != NULL) { + return pAllocationCallbacks->onRealloc(p, szNew, pAllocationCallbacks->pUserData); + } + + /* Try emulating realloc() in terms of malloc()/free(). */ + if (pAllocationCallbacks->onMalloc != NULL && pAllocationCallbacks->onFree != NULL) { + void* p2; + + p2 = pAllocationCallbacks->onMalloc(szNew, pAllocationCallbacks->pUserData); + if (p2 == NULL) { + return NULL; + } + + if (p != NULL) { + DRMP3_COPY_MEMORY(p2, p, szOld); + pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData); + } + + return p2; + } + + return NULL; +} + +static void drmp3__free_from_callbacks(void* p, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + if (p == NULL || pAllocationCallbacks == NULL) { + return; + } + + if (pAllocationCallbacks->onFree != NULL) { + pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData); + } +} + + +static drmp3_allocation_callbacks drmp3_copy_allocation_callbacks_or_defaults(const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks != NULL) { + /* Copy. */ + return *pAllocationCallbacks; + } else { + /* Defaults. */ + drmp3_allocation_callbacks allocationCallbacks; + allocationCallbacks.pUserData = NULL; + allocationCallbacks.onMalloc = drmp3__malloc_default; + allocationCallbacks.onRealloc = drmp3__realloc_default; + allocationCallbacks.onFree = drmp3__free_default; + return allocationCallbacks; + } +} + + + +static size_t drmp3__on_read(drmp3* pMP3, void* pBufferOut, size_t bytesToRead) +{ + size_t bytesRead; + + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pMP3->onRead != NULL); + + /* + Don't try reading 0 bytes from the callback. This can happen when the stream is clamped against + ID3v1 or APE tags at the end of the stream. + */ + if (bytesToRead == 0) { + return 0; + } + + bytesRead = pMP3->onRead(pMP3->pUserData, pBufferOut, bytesToRead); + pMP3->streamCursor += bytesRead; + + return bytesRead; +} + +static size_t drmp3__on_read_clamped(drmp3* pMP3, void* pBufferOut, size_t bytesToRead) +{ + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pMP3->onRead != NULL); + + if (pMP3->streamLength == DRMP3_UINT64_MAX) { + return drmp3__on_read(pMP3, pBufferOut, bytesToRead); + } else { + drmp3_uint64 bytesRemaining; + + bytesRemaining = (pMP3->streamLength - pMP3->streamCursor); + if (bytesToRead > bytesRemaining) { + bytesToRead = (size_t)bytesRemaining; + } + + return drmp3__on_read(pMP3, pBufferOut, bytesToRead); + } +} + +static drmp3_bool32 drmp3__on_seek(drmp3* pMP3, int offset, drmp3_seek_origin origin) +{ + DRMP3_ASSERT(offset >= 0); + DRMP3_ASSERT(origin == DRMP3_SEEK_SET || origin == DRMP3_SEEK_CUR); + + if (!pMP3->onSeek(pMP3->pUserData, offset, origin)) { + return DRMP3_FALSE; + } + + if (origin == DRMP3_SEEK_SET) { + pMP3->streamCursor = (drmp3_uint64)offset; + } else{ + pMP3->streamCursor += offset; + } + + return DRMP3_TRUE; +} + +static drmp3_bool32 drmp3__on_seek_64(drmp3* pMP3, drmp3_uint64 offset, drmp3_seek_origin origin) +{ + if (offset <= 0x7FFFFFFF) { + return drmp3__on_seek(pMP3, (int)offset, origin); + } + + /* Getting here "offset" is too large for a 32-bit integer. We just keep seeking forward until we hit the offset. */ + if (!drmp3__on_seek(pMP3, 0x7FFFFFFF, DRMP3_SEEK_SET)) { + return DRMP3_FALSE; + } + + offset -= 0x7FFFFFFF; + while (offset > 0) { + if (offset <= 0x7FFFFFFF) { + if (!drmp3__on_seek(pMP3, (int)offset, DRMP3_SEEK_CUR)) { + return DRMP3_FALSE; + } + offset = 0; + } else { + if (!drmp3__on_seek(pMP3, 0x7FFFFFFF, DRMP3_SEEK_CUR)) { + return DRMP3_FALSE; + } + offset -= 0x7FFFFFFF; + } + } + + return DRMP3_TRUE; +} + +static void drmp3__on_meta(drmp3* pMP3, drmp3_metadata_type type, const void* pRawData, size_t rawDataSize) +{ + if (pMP3->onMeta) { + drmp3_metadata metadata; + + DRMP3_ZERO_OBJECT(&metadata); + metadata.type = type; + metadata.pRawData = pRawData; + metadata.rawDataSize = rawDataSize; + + pMP3->onMeta(pMP3->pUserDataMeta, &metadata); + } +} + + +static drmp3_uint32 drmp3_decode_next_frame_ex__callbacks(drmp3* pMP3, drmp3d_sample_t* pPCMFrames, drmp3dec_frame_info* pMP3FrameInfo, const drmp3_uint8** ppMP3FrameData) +{ + drmp3_uint32 pcmFramesRead = 0; + + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pMP3->onRead != NULL); + + if (pMP3->atEnd) { + return 0; + } + + for (;;) { + drmp3dec_frame_info info; + + /* minimp3 recommends doing data submission in chunks of at least 16K. If we don't have at least 16K bytes available, get more. */ + if (pMP3->dataSize < DRMP3_MIN_DATA_CHUNK_SIZE) { + size_t bytesRead; + + /* First we need to move the data down. */ + if (pMP3->pData != NULL) { + DRMP3_MOVE_MEMORY(pMP3->pData, pMP3->pData + pMP3->dataConsumed, pMP3->dataSize); + } + + pMP3->dataConsumed = 0; + + if (pMP3->dataCapacity < DRMP3_DATA_CHUNK_SIZE) { + drmp3_uint8* pNewData; + size_t newDataCap; + + newDataCap = DRMP3_DATA_CHUNK_SIZE; + + pNewData = (drmp3_uint8*)drmp3__realloc_from_callbacks(pMP3->pData, newDataCap, pMP3->dataCapacity, &pMP3->allocationCallbacks); + if (pNewData == NULL) { + return 0; /* Out of memory. */ + } + + pMP3->pData = pNewData; + pMP3->dataCapacity = newDataCap; + } + + bytesRead = drmp3__on_read_clamped(pMP3, pMP3->pData + pMP3->dataSize, (pMP3->dataCapacity - pMP3->dataSize)); + if (bytesRead == 0) { + if (pMP3->dataSize == 0) { + pMP3->atEnd = DRMP3_TRUE; + return 0; /* No data. */ + } + } + + pMP3->dataSize += bytesRead; + } + + if (pMP3->dataSize > INT_MAX) { + pMP3->atEnd = DRMP3_TRUE; + return 0; /* File too big. */ + } + + DRMP3_ASSERT(pMP3->pData != NULL); + DRMP3_ASSERT(pMP3->dataCapacity > 0); + + /* Do a runtime check here to try silencing a false-positive from clang-analyzer. */ + if (pMP3->pData == NULL) { + return 0; + } + + pcmFramesRead = drmp3dec_decode_frame(&pMP3->decoder, pMP3->pData + pMP3->dataConsumed, (int)pMP3->dataSize, pPCMFrames, &info); /* <-- Safe size_t -> int conversion thanks to the check above. */ + + /* Consume the data. */ + pMP3->dataConsumed += (size_t)info.frame_bytes; + pMP3->dataSize -= (size_t)info.frame_bytes; + + /* pcmFramesRead will be equal to 0 if decoding failed. If it is zero and info.frame_bytes > 0 then we have successfully decoded the frame. */ + if (pcmFramesRead > 0) { + pcmFramesRead = drmp3_hdr_frame_samples(pMP3->decoder.header); + pMP3->pcmFramesConsumedInMP3Frame = 0; + pMP3->pcmFramesRemainingInMP3Frame = pcmFramesRead; + pMP3->mp3FrameChannels = info.channels; + pMP3->mp3FrameSampleRate = info.sample_rate; + + if (pMP3FrameInfo != NULL) { + *pMP3FrameInfo = info; + } + + if (ppMP3FrameData != NULL) { + *ppMP3FrameData = pMP3->pData + pMP3->dataConsumed - (size_t)info.frame_bytes; + } + + break; + } else if (info.frame_bytes == 0) { + /* Need more data. minimp3 recommends doing data submission in 16K chunks. */ + size_t bytesRead; + + /* First we need to move the data down. */ + DRMP3_MOVE_MEMORY(pMP3->pData, pMP3->pData + pMP3->dataConsumed, pMP3->dataSize); + pMP3->dataConsumed = 0; + + if (pMP3->dataCapacity == pMP3->dataSize) { + /* No room. Expand. */ + drmp3_uint8* pNewData; + size_t newDataCap; + + newDataCap = pMP3->dataCapacity + DRMP3_DATA_CHUNK_SIZE; + + pNewData = (drmp3_uint8*)drmp3__realloc_from_callbacks(pMP3->pData, newDataCap, pMP3->dataCapacity, &pMP3->allocationCallbacks); + if (pNewData == NULL) { + return 0; /* Out of memory. */ + } + + pMP3->pData = pNewData; + pMP3->dataCapacity = newDataCap; + } + + /* Fill in a chunk. */ + bytesRead = drmp3__on_read_clamped(pMP3, pMP3->pData + pMP3->dataSize, (pMP3->dataCapacity - pMP3->dataSize)); + if (bytesRead == 0) { + pMP3->atEnd = DRMP3_TRUE; + return 0; /* Error reading more data. */ + } + + pMP3->dataSize += bytesRead; + } + }; + + return pcmFramesRead; +} + +static drmp3_uint32 drmp3_decode_next_frame_ex__memory(drmp3* pMP3, drmp3d_sample_t* pPCMFrames, drmp3dec_frame_info* pMP3FrameInfo, const drmp3_uint8** ppMP3FrameData) +{ + drmp3_uint32 pcmFramesRead = 0; + drmp3dec_frame_info info; + + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pMP3->memory.pData != NULL); + + if (pMP3->atEnd) { + return 0; + } + + for (;;) { + pcmFramesRead = drmp3dec_decode_frame(&pMP3->decoder, pMP3->memory.pData + pMP3->memory.currentReadPos, (int)(pMP3->memory.dataSize - pMP3->memory.currentReadPos), pPCMFrames, &info); + if (pcmFramesRead > 0) { + pcmFramesRead = drmp3_hdr_frame_samples(pMP3->decoder.header); + pMP3->pcmFramesConsumedInMP3Frame = 0; + pMP3->pcmFramesRemainingInMP3Frame = pcmFramesRead; + pMP3->mp3FrameChannels = info.channels; + pMP3->mp3FrameSampleRate = info.sample_rate; + + if (pMP3FrameInfo != NULL) { + *pMP3FrameInfo = info; + } + + if (ppMP3FrameData != NULL) { + *ppMP3FrameData = pMP3->memory.pData + pMP3->memory.currentReadPos; + } + + break; + } else if (info.frame_bytes > 0) { + /* No frames were read, but it looks like we skipped past one. Read the next MP3 frame. */ + pMP3->memory.currentReadPos += (size_t)info.frame_bytes; + pMP3->streamCursor += (size_t)info.frame_bytes; + } else { + /* Nothing at all was read. Abort. */ + break; + } + } + + /* Consume the data. */ + pMP3->memory.currentReadPos += (size_t)info.frame_bytes; + pMP3->streamCursor += (size_t)info.frame_bytes; + + return pcmFramesRead; +} + +static drmp3_uint32 drmp3_decode_next_frame_ex(drmp3* pMP3, drmp3d_sample_t* pPCMFrames, drmp3dec_frame_info* pMP3FrameInfo, const drmp3_uint8** ppMP3FrameData) +{ + if (pMP3->memory.pData != NULL && pMP3->memory.dataSize > 0) { + return drmp3_decode_next_frame_ex__memory(pMP3, pPCMFrames, pMP3FrameInfo, ppMP3FrameData); + } else { + return drmp3_decode_next_frame_ex__callbacks(pMP3, pPCMFrames, pMP3FrameInfo, ppMP3FrameData); + } +} + +static drmp3_uint32 drmp3_decode_next_frame(drmp3* pMP3) +{ + DRMP3_ASSERT(pMP3 != NULL); + return drmp3_decode_next_frame_ex(pMP3, (drmp3d_sample_t*)pMP3->pcmFrames, NULL, NULL); +} + +#if 0 +static drmp3_uint32 drmp3_seek_next_frame(drmp3* pMP3) +{ + drmp3_uint32 pcmFrameCount; + + DRMP3_ASSERT(pMP3 != NULL); + + pcmFrameCount = drmp3_decode_next_frame_ex(pMP3, NULL, NULL, NULL); + if (pcmFrameCount == 0) { + return 0; + } + + /* We have essentially just skipped past the frame, so just set the remaining samples to 0. */ + pMP3->currentPCMFrame += pcmFrameCount; + pMP3->pcmFramesConsumedInMP3Frame = pcmFrameCount; + pMP3->pcmFramesRemainingInMP3Frame = 0; + + return pcmFrameCount; +} +#endif + +static drmp3_bool32 drmp3_init_internal(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, drmp3_tell_proc onTell, drmp3_meta_proc onMeta, void* pUserData, void* pUserDataMeta, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3dec_frame_info firstFrameInfo; + const drmp3_uint8* pFirstFrameData; + drmp3_uint32 firstFramePCMFrameCount; + drmp3_uint32 detectedMP3FrameCount = 0xFFFFFFFF; + + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(onRead != NULL); + + /* This function assumes the output object has already been reset to 0. Do not do that here, otherwise things will break. */ + drmp3dec_init(&pMP3->decoder); + + pMP3->onRead = onRead; + pMP3->onSeek = onSeek; + pMP3->onMeta = onMeta; + pMP3->pUserData = pUserData; + pMP3->pUserDataMeta = pUserDataMeta; + pMP3->allocationCallbacks = drmp3_copy_allocation_callbacks_or_defaults(pAllocationCallbacks); + + if (pMP3->allocationCallbacks.onFree == NULL || (pMP3->allocationCallbacks.onMalloc == NULL && pMP3->allocationCallbacks.onRealloc == NULL)) { + return DRMP3_FALSE; /* Invalid allocation callbacks. */ + } + + pMP3->streamCursor = 0; + pMP3->streamLength = DRMP3_UINT64_MAX; + pMP3->streamStartOffset = 0; + pMP3->delayInPCMFrames = 0; + pMP3->paddingInPCMFrames = 0; + pMP3->totalPCMFrameCount = DRMP3_UINT64_MAX; + + /* We'll first check for any ID3v1 or APE tags. */ + #if 1 + if (onSeek != NULL && onTell != NULL) { + if (onSeek(pUserData, 0, DRMP3_SEEK_END)) { + drmp3_int64 streamLen; + int streamEndOffset = 0; + + /* First get the length of the stream. We need this so we can ensure the stream is big enough to store the tags. */ + if (onTell(pUserData, &streamLen)) { + /* ID3v1 */ + if (streamLen > 128) { + char id3[3]; + if (onSeek(pUserData, streamEndOffset - 128, DRMP3_SEEK_END)) { + if (onRead(pUserData, id3, 3) == 3 && id3[0] == 'T' && id3[1] == 'A' && id3[2] == 'G') { + /* We have an ID3v1 tag. */ + streamEndOffset -= 128; + streamLen -= 128; + + /* Fire a metadata callback for the TAG data. */ + if (onMeta != NULL) { + drmp3_uint8 tag[128]; + tag[0] = 'T'; tag[1] = 'A'; tag[2] = 'G'; + + if (onRead(pUserData, tag + 3, 125) == 125) { + drmp3__on_meta(pMP3, DRMP3_METADATA_TYPE_ID3V1, tag, 128); + } + } + } else { + /* No ID3v1 tag. */ + } + } else { + /* Failed to seek to the ID3v1 tag. */ + } + } else { + /* Stream too short. No ID3v1 tag. */ + } + + /* APE */ + if (streamLen > 32) { + char ape[32]; /* The footer. */ + if (onSeek(pUserData, streamEndOffset - 32, DRMP3_SEEK_END)) { + if (onRead(pUserData, ape, 32) == 32 && ape[0] == 'A' && ape[1] == 'P' && ape[2] == 'E' && ape[3] == 'T' && ape[4] == 'A' && ape[5] == 'G' && ape[6] == 'E' && ape[7] == 'X') { + /* We have an APE tag. */ + drmp3_uint32 tagSize = + ((drmp3_uint32)ape[24] << 0) | + ((drmp3_uint32)ape[25] << 8) | + ((drmp3_uint32)ape[26] << 16) | + ((drmp3_uint32)ape[27] << 24); + + if (32 + tagSize < streamLen) { + streamEndOffset -= 32 + tagSize; + streamLen -= 32 + tagSize; + + /* Fire a metadata callback for the APE data. Must include both the main content and footer. */ + if (onMeta != NULL) { + /* We first need to seek to the start of the APE tag. */ + if (onSeek(pUserData, streamEndOffset, DRMP3_SEEK_END)) { + size_t apeTagSize = (size_t)tagSize + 32; + drmp3_uint8* pTagData = (drmp3_uint8*)drmp3_malloc(apeTagSize, pAllocationCallbacks); + if (pTagData != NULL) { + if (onRead(pUserData, pTagData, apeTagSize) == apeTagSize) { + drmp3__on_meta(pMP3, DRMP3_METADATA_TYPE_APE, pTagData, apeTagSize); + } + + drmp3_free(pTagData, pAllocationCallbacks); + } + } + } + } else { + /* The tag size is larger than the stream. Invalid APE tag. */ + } + } + } + } else { + /* Stream too short. No APE tag. */ + } + + /* Seek back to the start. */ + if (!onSeek(pUserData, 0, DRMP3_SEEK_SET)) { + return DRMP3_FALSE; /* Failed to seek back to the start. */ + } + + pMP3->streamLength = (drmp3_uint64)streamLen; + + if (pMP3->memory.pData != NULL) { + pMP3->memory.dataSize = (size_t)pMP3->streamLength; + } + } else { + /* Failed to get the length of the stream. ID3v1 and APE tags cannot be skipped. */ + if (!onSeek(pUserData, 0, DRMP3_SEEK_SET)) { + return DRMP3_FALSE; /* Failed to seek back to the start. */ + } + } + } else { + /* Failed to seek to the end. Cannot skip ID3v1 or APE tags. */ + } + } else { + /* No onSeek or onTell callback. Cannot skip ID3v1 or APE tags. */ + } + #endif + + + /* ID3v2 tags */ + #if 1 + { + char header[10]; + if (onRead(pUserData, header, 10) == 10) { + if (header[0] == 'I' && header[1] == 'D' && header[2] == '3') { + drmp3_uint32 tagSize = + (((drmp3_uint32)header[6] & 0x7F) << 21) | + (((drmp3_uint32)header[7] & 0x7F) << 14) | + (((drmp3_uint32)header[8] & 0x7F) << 7) | + (((drmp3_uint32)header[9] & 0x7F) << 0); + + /* Account for the footer. */ + if (header[5] & 0x10) { + tagSize += 10; + } + + /* Read the tag content and fire a metadata callback. */ + if (onMeta != NULL) { + size_t tagSizeWithHeader = 10 + tagSize; + drmp3_uint8* pTagData = (drmp3_uint8*)drmp3_malloc(tagSizeWithHeader, pAllocationCallbacks); + if (pTagData != NULL) { + DRMP3_COPY_MEMORY(pTagData, header, 10); + + if (onRead(pUserData, pTagData + 10, tagSize) == tagSize) { + drmp3__on_meta(pMP3, DRMP3_METADATA_TYPE_ID3V2, pTagData, tagSizeWithHeader); + } + + drmp3_free(pTagData, pAllocationCallbacks); + } + } else { + /* Don't have a metadata callback, so just skip the tag. */ + if (onSeek != NULL) { + if (!onSeek(pUserData, tagSize, DRMP3_SEEK_CUR)) { + return DRMP3_FALSE; /* Failed to seek past the ID3v2 tag. */ + } + } else { + /* Don't have a seek callback. Read and discard. */ + char discard[1024]; + + while (tagSize > 0) { + size_t bytesToRead = tagSize; + if (bytesToRead > sizeof(discard)) { + bytesToRead = sizeof(discard); + } + + if (onRead(pUserData, discard, bytesToRead) != bytesToRead) { + return DRMP3_FALSE; /* Failed to read data. */ + } + + tagSize -= (drmp3_uint32)bytesToRead; + } + } + } + + pMP3->streamStartOffset += 10 + tagSize; /* +10 for the header. */ + pMP3->streamCursor = pMP3->streamStartOffset; + } else { + /* Not an ID3v2 tag. Seek back to the start. */ + if (onSeek != NULL) { + if (!onSeek(pUserData, 0, DRMP3_SEEK_SET)) { + return DRMP3_FALSE; /* Failed to seek back to the start. */ + } + } else { + /* Don't have a seek callback to move backwards. We'll just fall through and let the decoding process re-sync. The ideal solution here would be to read into the cache. */ + + /* + TODO: Copy the header into the cache. Will need to allocate space. See drmp3_decode_next_frame_ex__callbacks. There is not need + to handle the memory case because that will always have a seek implementation and will never hit this code path. + */ + } + } + } else { + /* Failed to read the header. We can return false here. If we couldn't read 10 bytes there's no way we'll have a valid MP3 stream. */ + return DRMP3_FALSE; + } + } + #endif + + /* + Decode the first frame to confirm that it is indeed a valid MP3 stream. Note that it's possible the first frame + is actually a Xing/LAME/VBRI header. If this is the case we need to skip over it. + */ + firstFramePCMFrameCount = drmp3_decode_next_frame_ex(pMP3, (drmp3d_sample_t*)pMP3->pcmFrames, &firstFrameInfo, &pFirstFrameData); + if (firstFramePCMFrameCount > 0) { + DRMP3_ASSERT(pFirstFrameData != NULL); + + /* + It might be a header. If so, we need to clear out the cached PCM frames in order to trigger a reload of fresh + data when decoding starts. We can assume all validation has already been performed to check if this is a valid + MP3 frame and that there is more than 0 bytes making up the frame. + + We're going to be basing this parsing code off the minimp3_ex implementation. + */ + #if 1 + DRMP3_ASSERT(firstFrameInfo.frame_bytes > 0); + { + drmp3_bs bs; + drmp3_L3_gr_info grInfo[4]; + + drmp3_bs_init(&bs, pFirstFrameData + DRMP3_HDR_SIZE, firstFrameInfo.frame_bytes - DRMP3_HDR_SIZE); + + if (DRMP3_HDR_IS_CRC(pFirstFrameData)) { + drmp3_bs_get_bits(&bs, 16); /* CRC. */ + } + + if (drmp3_L3_read_side_info(&bs, grInfo, pFirstFrameData) >= 0) { + drmp3_bool32 isXing = DRMP3_FALSE; + drmp3_bool32 isInfo = DRMP3_FALSE; + const drmp3_uint8* pTagData; + const drmp3_uint8* pTagDataBeg; + const void* pDataBufferEnd = NULL; + size_t frameBytes; + + pTagDataBeg = pFirstFrameData + DRMP3_HDR_SIZE + (bs.pos/8); + pTagData = pTagDataBeg; + + /* + We need to determine how many bytes are actually available in pTagData. Unfortunately this is different depending on + whether or not it's being decoded from memory or callbacks. + */ + if (pMP3->memory.pData != NULL && pMP3->memory.dataSize > 0) { + pDataBufferEnd = pMP3->memory.pData + pMP3->memory.dataSize; + } else { + pDataBufferEnd = pMP3->pData + pMP3->dataCapacity; + } + + frameBytes = DRMP3_MIN((size_t)firstFrameInfo.frame_bytes, (size_t)((drmp3_uint8*)pDataBufferEnd - pFirstFrameData)); + + if (frameBytes < (size_t)(pTagData - pFirstFrameData) + 8) { + goto done_xing_info; /* Frame too small for a Xing/Info tag. */ + } + + /* Check for both "Xing" and "Info" identifiers. */ + isXing = (pTagData[0] == 'X' && pTagData[1] == 'i' && pTagData[2] == 'n' && pTagData[3] == 'g'); + isInfo = (pTagData[0] == 'I' && pTagData[1] == 'n' && pTagData[2] == 'f' && pTagData[3] == 'o'); + + if (isXing || isInfo) { + drmp3_uint32 bytes = 0; + drmp3_uint32 flags = pTagData[7]; + + pTagData += 8; /* Skip past the ID and flags. */ + + if (flags & 0x01) { /* FRAMES flag. */ + if (frameBytes < (size_t)(pTagData - pFirstFrameData) + 4) { + goto done_xing_info; /* Invalid Xing/Info tag. */ + } + + detectedMP3FrameCount = (drmp3_uint32)pTagData[0] << 24 | (drmp3_uint32)pTagData[1] << 16 | (drmp3_uint32)pTagData[2] << 8 | (drmp3_uint32)pTagData[3]; + pTagData += 4; + } + + if (flags & 0x02) { /* BYTES flag. */ + if (frameBytes < (size_t)(pTagData - pFirstFrameData) + 4) { + goto done_xing_info; /* Invalid Xing/Info tag. */ + } + + bytes = (drmp3_uint32)pTagData[0] << 24 | (drmp3_uint32)pTagData[1] << 16 | (drmp3_uint32)pTagData[2] << 8 | (drmp3_uint32)pTagData[3]; + (void)bytes; /* <-- Just to silence a warning about `bytes` being assigned but unused. Want to leave this here in case I want to make use of it later. */ + pTagData += 4; + } + + if (flags & 0x04) { /* TOC flag. */ + if (frameBytes < (size_t)(pTagData - pFirstFrameData) + 100) { + goto done_xing_info; /* Invalid Xing/Info tag. */ + } + + /* TODO: Extract and bind seek points. */ + pTagData += 100; + } + + if (flags & 0x08) { /* SCALE flag. */ + if (frameBytes < (size_t)(pTagData - pFirstFrameData) + 4) { + goto done_xing_info; /* Invalid Xing/Info tag. */ + } + + pTagData += 4; + } + + /* At this point we're done with the Xing/Info header. Now we can look at the LAME data. */ + if (frameBytes < (size_t)(pTagData - pFirstFrameData) + 1) { + goto done_xing_info; /* Not enough data left to check for a LAME header. */ + } + + if (pTagData[0]) { + int delayInPCMFrames; + int paddingInPCMFrames; + + if (frameBytes < (size_t)(pTagData - pFirstFrameData) + 36) { + goto done_xing_info; /* Invalid Xing/Info tag. */ + } + + pTagData += 21; + + delayInPCMFrames = (( (drmp3_uint32)pTagData[0] << 4) | ((drmp3_uint32)pTagData[1] >> 4)) + (528 + 1); + paddingInPCMFrames = ((((drmp3_uint32)pTagData[1] & 0xF) << 8) | ((drmp3_uint32)pTagData[2] )) - (528 + 1); + if (paddingInPCMFrames < 0) { + paddingInPCMFrames = 0; /* Padding cannot be negative. Probably a malformed file. Ignore. */ + } + + pMP3->delayInPCMFrames = (drmp3_uint32)delayInPCMFrames; + pMP3->paddingInPCMFrames = (drmp3_uint32)paddingInPCMFrames; + } + + /* + My understanding is that if the "Xing" header is present we can consider this to be a VBR stream and if the "Info" header is + present it's a CBR stream. If this is not the case let me know! I'm just tracking this for the time being in case I want to + look at doing some CBR optimizations later on, such as faster seeking. + */ + if (isXing) { + pMP3->isVBR = DRMP3_TRUE; + } else if (isInfo) { + pMP3->isCBR = DRMP3_TRUE; + } + + /* Post the raw data of the tag to the metadata callback. */ + if (onMeta != NULL) { + drmp3_metadata_type metadataType = isXing ? DRMP3_METADATA_TYPE_XING : DRMP3_METADATA_TYPE_VBRI; + size_t tagDataSize; + + tagDataSize = (size_t)firstFrameInfo.frame_bytes; + tagDataSize -= (size_t)(pTagDataBeg - pFirstFrameData); + + drmp3__on_meta(pMP3, metadataType, pTagDataBeg, tagDataSize); + } + + /* Since this was identified as a tag, we don't want to treat it as audio. We need to clear out the PCM cache. */ + pMP3->pcmFramesRemainingInMP3Frame = 0; + + /* The start offset needs to be moved to the end of this frame so it's not included in any audio processing after seeking. */ + pMP3->streamStartOffset += (drmp3_uint32)(firstFrameInfo.frame_bytes); + pMP3->streamCursor = pMP3->streamStartOffset; + + /* + The internal decoder needs to be reset to clear out any state. If we don't reset this state, it's possible for + there to be inconsistencies in the number of samples read when reading to the end of the stream depending on + whether or not the caller seeks to the start of the stream. + */ + drmp3dec_init(&pMP3->decoder); + } + + done_xing_info:; + } else { + /* Failed to read the side info. */ + } + } + #endif + } else { + /* Not a valid MP3 stream. */ + drmp3__free_from_callbacks(pMP3->pData, &pMP3->allocationCallbacks); /* The call above may have allocated memory. Need to make sure it's freed before aborting. */ + return DRMP3_FALSE; + } + + if (detectedMP3FrameCount != 0xFFFFFFFF) { + pMP3->totalPCMFrameCount = (drmp3_uint64)detectedMP3FrameCount * firstFramePCMFrameCount; + } + + pMP3->channels = pMP3->mp3FrameChannels; + pMP3->sampleRate = pMP3->mp3FrameSampleRate; + + return DRMP3_TRUE; +} + +DRMP3_API drmp3_bool32 drmp3_init(drmp3* pMP3, drmp3_read_proc onRead, drmp3_seek_proc onSeek, drmp3_tell_proc onTell, drmp3_meta_proc onMeta, void* pUserData, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + if (pMP3 == NULL || onRead == NULL) { + return DRMP3_FALSE; + } + + DRMP3_ZERO_OBJECT(pMP3); + return drmp3_init_internal(pMP3, onRead, onSeek, onTell, onMeta, pUserData, pUserData, pAllocationCallbacks); +} + + +static size_t drmp3__on_read_memory(void* pUserData, void* pBufferOut, size_t bytesToRead) +{ + drmp3* pMP3 = (drmp3*)pUserData; + size_t bytesRemaining; + + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pMP3->memory.dataSize >= pMP3->memory.currentReadPos); + + bytesRemaining = pMP3->memory.dataSize - pMP3->memory.currentReadPos; + if (bytesToRead > bytesRemaining) { + bytesToRead = bytesRemaining; + } + + if (bytesToRead > 0) { + DRMP3_COPY_MEMORY(pBufferOut, pMP3->memory.pData + pMP3->memory.currentReadPos, bytesToRead); + pMP3->memory.currentReadPos += bytesToRead; + } + + return bytesToRead; +} + +static drmp3_bool32 drmp3__on_seek_memory(void* pUserData, int byteOffset, drmp3_seek_origin origin) +{ + drmp3* pMP3 = (drmp3*)pUserData; + drmp3_int64 newCursor; + + DRMP3_ASSERT(pMP3 != NULL); + + if (origin == DRMP3_SEEK_SET) { + newCursor = 0; + } else if (origin == DRMP3_SEEK_CUR) { + newCursor = (drmp3_int64)pMP3->memory.currentReadPos; + } else if (origin == DRMP3_SEEK_END) { + newCursor = (drmp3_int64)pMP3->memory.dataSize; + } else { + DRMP3_ASSERT(!"Invalid seek origin"); + return DRMP3_FALSE; + } + + newCursor += byteOffset; + + if (newCursor < 0) { + return DRMP3_FALSE; /* Trying to seek prior to the start of the buffer. */ + } + if ((size_t)newCursor > pMP3->memory.dataSize) { + return DRMP3_FALSE; /* Trying to seek beyond the end of the buffer. */ + } + + pMP3->memory.currentReadPos = (size_t)newCursor; + + return DRMP3_TRUE; +} + +static drmp3_bool32 drmp3__on_tell_memory(void* pUserData, drmp3_int64* pCursor) +{ + drmp3* pMP3 = (drmp3*)pUserData; + + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pCursor != NULL); + + *pCursor = (drmp3_int64)pMP3->memory.currentReadPos; + return DRMP3_TRUE; +} + +DRMP3_API drmp3_bool32 drmp3_init_memory_with_metadata(drmp3* pMP3, const void* pData, size_t dataSize, drmp3_meta_proc onMeta, void* pUserDataMeta, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3_bool32 result; + + if (pMP3 == NULL) { + return DRMP3_FALSE; + } + + DRMP3_ZERO_OBJECT(pMP3); + + if (pData == NULL || dataSize == 0) { + return DRMP3_FALSE; + } + + pMP3->memory.pData = (const drmp3_uint8*)pData; + pMP3->memory.dataSize = dataSize; + pMP3->memory.currentReadPos = 0; + + result = drmp3_init_internal(pMP3, drmp3__on_read_memory, drmp3__on_seek_memory, drmp3__on_tell_memory, onMeta, pMP3, pUserDataMeta, pAllocationCallbacks); + if (result == DRMP3_FALSE) { + return DRMP3_FALSE; + } + + /* Adjust the length of the memory stream to account for ID3v1 and APE tags. */ + if (pMP3->streamLength <= (drmp3_uint64)DRMP3_SIZE_MAX) { + pMP3->memory.dataSize = (size_t)pMP3->streamLength; /* Safe cast. */ + } + + if (pMP3->streamStartOffset > (drmp3_uint64)DRMP3_SIZE_MAX) { + return DRMP3_FALSE; /* Tags too big. */ + } + + return DRMP3_TRUE; +} + +DRMP3_API drmp3_bool32 drmp3_init_memory(drmp3* pMP3, const void* pData, size_t dataSize, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + return drmp3_init_memory_with_metadata(pMP3, pData, dataSize, NULL, NULL, pAllocationCallbacks); +} + + +#ifndef DR_MP3_NO_STDIO +#include +#include /* For wcslen(), wcsrtombs() */ + +/* Errno */ +/* drmp3_result_from_errno() is only used inside DR_MP3_NO_STDIO for now. Move this out if it's ever used elsewhere. */ +#include +static drmp3_result drmp3_result_from_errno(int e) +{ + switch (e) + { + case 0: return DRMP3_SUCCESS; + #ifdef EPERM + case EPERM: return DRMP3_INVALID_OPERATION; + #endif + #ifdef ENOENT + case ENOENT: return DRMP3_DOES_NOT_EXIST; + #endif + #ifdef ESRCH + case ESRCH: return DRMP3_DOES_NOT_EXIST; + #endif + #ifdef EINTR + case EINTR: return DRMP3_INTERRUPT; + #endif + #ifdef EIO + case EIO: return DRMP3_IO_ERROR; + #endif + #ifdef ENXIO + case ENXIO: return DRMP3_DOES_NOT_EXIST; + #endif + #ifdef E2BIG + case E2BIG: return DRMP3_INVALID_ARGS; + #endif + #ifdef ENOEXEC + case ENOEXEC: return DRMP3_INVALID_FILE; + #endif + #ifdef EBADF + case EBADF: return DRMP3_INVALID_FILE; + #endif + #ifdef ECHILD + case ECHILD: return DRMP3_ERROR; + #endif + #ifdef EAGAIN + case EAGAIN: return DRMP3_UNAVAILABLE; + #endif + #ifdef ENOMEM + case ENOMEM: return DRMP3_OUT_OF_MEMORY; + #endif + #ifdef EACCES + case EACCES: return DRMP3_ACCESS_DENIED; + #endif + #ifdef EFAULT + case EFAULT: return DRMP3_BAD_ADDRESS; + #endif + #ifdef ENOTBLK + case ENOTBLK: return DRMP3_ERROR; + #endif + #ifdef EBUSY + case EBUSY: return DRMP3_BUSY; + #endif + #ifdef EEXIST + case EEXIST: return DRMP3_ALREADY_EXISTS; + #endif + #ifdef EXDEV + case EXDEV: return DRMP3_ERROR; + #endif + #ifdef ENODEV + case ENODEV: return DRMP3_DOES_NOT_EXIST; + #endif + #ifdef ENOTDIR + case ENOTDIR: return DRMP3_NOT_DIRECTORY; + #endif + #ifdef EISDIR + case EISDIR: return DRMP3_IS_DIRECTORY; + #endif + #ifdef EINVAL + case EINVAL: return DRMP3_INVALID_ARGS; + #endif + #ifdef ENFILE + case ENFILE: return DRMP3_TOO_MANY_OPEN_FILES; + #endif + #ifdef EMFILE + case EMFILE: return DRMP3_TOO_MANY_OPEN_FILES; + #endif + #ifdef ENOTTY + case ENOTTY: return DRMP3_INVALID_OPERATION; + #endif + #ifdef ETXTBSY + case ETXTBSY: return DRMP3_BUSY; + #endif + #ifdef EFBIG + case EFBIG: return DRMP3_TOO_BIG; + #endif + #ifdef ENOSPC + case ENOSPC: return DRMP3_NO_SPACE; + #endif + #ifdef ESPIPE + case ESPIPE: return DRMP3_BAD_SEEK; + #endif + #ifdef EROFS + case EROFS: return DRMP3_ACCESS_DENIED; + #endif + #ifdef EMLINK + case EMLINK: return DRMP3_TOO_MANY_LINKS; + #endif + #ifdef EPIPE + case EPIPE: return DRMP3_BAD_PIPE; + #endif + #ifdef EDOM + case EDOM: return DRMP3_OUT_OF_RANGE; + #endif + #ifdef ERANGE + case ERANGE: return DRMP3_OUT_OF_RANGE; + #endif + #ifdef EDEADLK + case EDEADLK: return DRMP3_DEADLOCK; + #endif + #ifdef ENAMETOOLONG + case ENAMETOOLONG: return DRMP3_PATH_TOO_LONG; + #endif + #ifdef ENOLCK + case ENOLCK: return DRMP3_ERROR; + #endif + #ifdef ENOSYS + case ENOSYS: return DRMP3_NOT_IMPLEMENTED; + #endif + #if defined(ENOTEMPTY) && ENOTEMPTY != EEXIST /* In AIX, ENOTEMPTY and EEXIST use the same value. */ + case ENOTEMPTY: return DRMP3_DIRECTORY_NOT_EMPTY; + #endif + #ifdef ELOOP + case ELOOP: return DRMP3_TOO_MANY_LINKS; + #endif + #ifdef ENOMSG + case ENOMSG: return DRMP3_NO_MESSAGE; + #endif + #ifdef EIDRM + case EIDRM: return DRMP3_ERROR; + #endif + #ifdef ECHRNG + case ECHRNG: return DRMP3_ERROR; + #endif + #ifdef EL2NSYNC + case EL2NSYNC: return DRMP3_ERROR; + #endif + #ifdef EL3HLT + case EL3HLT: return DRMP3_ERROR; + #endif + #ifdef EL3RST + case EL3RST: return DRMP3_ERROR; + #endif + #ifdef ELNRNG + case ELNRNG: return DRMP3_OUT_OF_RANGE; + #endif + #ifdef EUNATCH + case EUNATCH: return DRMP3_ERROR; + #endif + #ifdef ENOCSI + case ENOCSI: return DRMP3_ERROR; + #endif + #ifdef EL2HLT + case EL2HLT: return DRMP3_ERROR; + #endif + #ifdef EBADE + case EBADE: return DRMP3_ERROR; + #endif + #ifdef EBADR + case EBADR: return DRMP3_ERROR; + #endif + #ifdef EXFULL + case EXFULL: return DRMP3_ERROR; + #endif + #ifdef ENOANO + case ENOANO: return DRMP3_ERROR; + #endif + #ifdef EBADRQC + case EBADRQC: return DRMP3_ERROR; + #endif + #ifdef EBADSLT + case EBADSLT: return DRMP3_ERROR; + #endif + #ifdef EBFONT + case EBFONT: return DRMP3_INVALID_FILE; + #endif + #ifdef ENOSTR + case ENOSTR: return DRMP3_ERROR; + #endif + #ifdef ENODATA + case ENODATA: return DRMP3_NO_DATA_AVAILABLE; + #endif + #ifdef ETIME + case ETIME: return DRMP3_TIMEOUT; + #endif + #ifdef ENOSR + case ENOSR: return DRMP3_NO_DATA_AVAILABLE; + #endif + #ifdef ENONET + case ENONET: return DRMP3_NO_NETWORK; + #endif + #ifdef ENOPKG + case ENOPKG: return DRMP3_ERROR; + #endif + #ifdef EREMOTE + case EREMOTE: return DRMP3_ERROR; + #endif + #ifdef ENOLINK + case ENOLINK: return DRMP3_ERROR; + #endif + #ifdef EADV + case EADV: return DRMP3_ERROR; + #endif + #ifdef ESRMNT + case ESRMNT: return DRMP3_ERROR; + #endif + #ifdef ECOMM + case ECOMM: return DRMP3_ERROR; + #endif + #ifdef EPROTO + case EPROTO: return DRMP3_ERROR; + #endif + #ifdef EMULTIHOP + case EMULTIHOP: return DRMP3_ERROR; + #endif + #ifdef EDOTDOT + case EDOTDOT: return DRMP3_ERROR; + #endif + #ifdef EBADMSG + case EBADMSG: return DRMP3_BAD_MESSAGE; + #endif + #ifdef EOVERFLOW + case EOVERFLOW: return DRMP3_TOO_BIG; + #endif + #ifdef ENOTUNIQ + case ENOTUNIQ: return DRMP3_NOT_UNIQUE; + #endif + #ifdef EBADFD + case EBADFD: return DRMP3_ERROR; + #endif + #ifdef EREMCHG + case EREMCHG: return DRMP3_ERROR; + #endif + #ifdef ELIBACC + case ELIBACC: return DRMP3_ACCESS_DENIED; + #endif + #ifdef ELIBBAD + case ELIBBAD: return DRMP3_INVALID_FILE; + #endif + #ifdef ELIBSCN + case ELIBSCN: return DRMP3_INVALID_FILE; + #endif + #ifdef ELIBMAX + case ELIBMAX: return DRMP3_ERROR; + #endif + #ifdef ELIBEXEC + case ELIBEXEC: return DRMP3_ERROR; + #endif + #ifdef EILSEQ + case EILSEQ: return DRMP3_INVALID_DATA; + #endif + #ifdef ERESTART + case ERESTART: return DRMP3_ERROR; + #endif + #ifdef ESTRPIPE + case ESTRPIPE: return DRMP3_ERROR; + #endif + #ifdef EUSERS + case EUSERS: return DRMP3_ERROR; + #endif + #ifdef ENOTSOCK + case ENOTSOCK: return DRMP3_NOT_SOCKET; + #endif + #ifdef EDESTADDRREQ + case EDESTADDRREQ: return DRMP3_NO_ADDRESS; + #endif + #ifdef EMSGSIZE + case EMSGSIZE: return DRMP3_TOO_BIG; + #endif + #ifdef EPROTOTYPE + case EPROTOTYPE: return DRMP3_BAD_PROTOCOL; + #endif + #ifdef ENOPROTOOPT + case ENOPROTOOPT: return DRMP3_PROTOCOL_UNAVAILABLE; + #endif + #ifdef EPROTONOSUPPORT + case EPROTONOSUPPORT: return DRMP3_PROTOCOL_NOT_SUPPORTED; + #endif + #ifdef ESOCKTNOSUPPORT + case ESOCKTNOSUPPORT: return DRMP3_SOCKET_NOT_SUPPORTED; + #endif + #ifdef EOPNOTSUPP + case EOPNOTSUPP: return DRMP3_INVALID_OPERATION; + #endif + #ifdef EPFNOSUPPORT + case EPFNOSUPPORT: return DRMP3_PROTOCOL_FAMILY_NOT_SUPPORTED; + #endif + #ifdef EAFNOSUPPORT + case EAFNOSUPPORT: return DRMP3_ADDRESS_FAMILY_NOT_SUPPORTED; + #endif + #ifdef EADDRINUSE + case EADDRINUSE: return DRMP3_ALREADY_IN_USE; + #endif + #ifdef EADDRNOTAVAIL + case EADDRNOTAVAIL: return DRMP3_ERROR; + #endif + #ifdef ENETDOWN + case ENETDOWN: return DRMP3_NO_NETWORK; + #endif + #ifdef ENETUNREACH + case ENETUNREACH: return DRMP3_NO_NETWORK; + #endif + #ifdef ENETRESET + case ENETRESET: return DRMP3_NO_NETWORK; + #endif + #ifdef ECONNABORTED + case ECONNABORTED: return DRMP3_NO_NETWORK; + #endif + #ifdef ECONNRESET + case ECONNRESET: return DRMP3_CONNECTION_RESET; + #endif + #ifdef ENOBUFS + case ENOBUFS: return DRMP3_NO_SPACE; + #endif + #ifdef EISCONN + case EISCONN: return DRMP3_ALREADY_CONNECTED; + #endif + #ifdef ENOTCONN + case ENOTCONN: return DRMP3_NOT_CONNECTED; + #endif + #ifdef ESHUTDOWN + case ESHUTDOWN: return DRMP3_ERROR; + #endif + #ifdef ETOOMANYREFS + case ETOOMANYREFS: return DRMP3_ERROR; + #endif + #ifdef ETIMEDOUT + case ETIMEDOUT: return DRMP3_TIMEOUT; + #endif + #ifdef ECONNREFUSED + case ECONNREFUSED: return DRMP3_CONNECTION_REFUSED; + #endif + #ifdef EHOSTDOWN + case EHOSTDOWN: return DRMP3_NO_HOST; + #endif + #ifdef EHOSTUNREACH + case EHOSTUNREACH: return DRMP3_NO_HOST; + #endif + #ifdef EALREADY + case EALREADY: return DRMP3_IN_PROGRESS; + #endif + #ifdef EINPROGRESS + case EINPROGRESS: return DRMP3_IN_PROGRESS; + #endif + #ifdef ESTALE + case ESTALE: return DRMP3_INVALID_FILE; + #endif + #ifdef EUCLEAN + case EUCLEAN: return DRMP3_ERROR; + #endif + #ifdef ENOTNAM + case ENOTNAM: return DRMP3_ERROR; + #endif + #ifdef ENAVAIL + case ENAVAIL: return DRMP3_ERROR; + #endif + #ifdef EISNAM + case EISNAM: return DRMP3_ERROR; + #endif + #ifdef EREMOTEIO + case EREMOTEIO: return DRMP3_IO_ERROR; + #endif + #ifdef EDQUOT + case EDQUOT: return DRMP3_NO_SPACE; + #endif + #ifdef ENOMEDIUM + case ENOMEDIUM: return DRMP3_DOES_NOT_EXIST; + #endif + #ifdef EMEDIUMTYPE + case EMEDIUMTYPE: return DRMP3_ERROR; + #endif + #ifdef ECANCELED + case ECANCELED: return DRMP3_CANCELLED; + #endif + #ifdef ENOKEY + case ENOKEY: return DRMP3_ERROR; + #endif + #ifdef EKEYEXPIRED + case EKEYEXPIRED: return DRMP3_ERROR; + #endif + #ifdef EKEYREVOKED + case EKEYREVOKED: return DRMP3_ERROR; + #endif + #ifdef EKEYREJECTED + case EKEYREJECTED: return DRMP3_ERROR; + #endif + #ifdef EOWNERDEAD + case EOWNERDEAD: return DRMP3_ERROR; + #endif + #ifdef ENOTRECOVERABLE + case ENOTRECOVERABLE: return DRMP3_ERROR; + #endif + #ifdef ERFKILL + case ERFKILL: return DRMP3_ERROR; + #endif + #ifdef EHWPOISON + case EHWPOISON: return DRMP3_ERROR; + #endif + default: return DRMP3_ERROR; + } +} +/* End Errno */ + +/* fopen */ +static drmp3_result drmp3_fopen(FILE** ppFile, const char* pFilePath, const char* pOpenMode) +{ +#if defined(_MSC_VER) && _MSC_VER >= 1400 + errno_t err; +#endif + + if (ppFile != NULL) { + *ppFile = NULL; /* Safety. */ + } + + if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) { + return DRMP3_INVALID_ARGS; + } + +#if defined(_MSC_VER) && _MSC_VER >= 1400 + err = fopen_s(ppFile, pFilePath, pOpenMode); + if (err != 0) { + return drmp3_result_from_errno(err); + } +#else +#if defined(_WIN32) || defined(__APPLE__) + *ppFile = fopen(pFilePath, pOpenMode); +#else + #if defined(_FILE_OFFSET_BITS) && _FILE_OFFSET_BITS == 64 && defined(_LARGEFILE64_SOURCE) + *ppFile = fopen64(pFilePath, pOpenMode); + #else + *ppFile = fopen(pFilePath, pOpenMode); + #endif +#endif + if (*ppFile == NULL) { + drmp3_result result = drmp3_result_from_errno(errno); + if (result == DRMP3_SUCCESS) { + result = DRMP3_ERROR; /* Just a safety check to make sure we never ever return success when pFile == NULL. */ + } + + return result; + } +#endif + + return DRMP3_SUCCESS; +} + +/* +_wfopen() isn't always available in all compilation environments. + + * Windows only. + * MSVC seems to support it universally as far back as VC6 from what I can tell (haven't checked further back). + * MinGW-64 (both 32- and 64-bit) seems to support it. + * MinGW wraps it in !defined(__STRICT_ANSI__). + * OpenWatcom wraps it in !defined(_NO_EXT_KEYS). + +This can be reviewed as compatibility issues arise. The preference is to use _wfopen_s() and _wfopen() as opposed to the wcsrtombs() +fallback, so if you notice your compiler not detecting this properly I'm happy to look at adding support. +*/ +#if defined(_WIN32) + #if defined(_MSC_VER) || defined(__MINGW64__) || (!defined(__STRICT_ANSI__) && !defined(_NO_EXT_KEYS)) + #define DRMP3_HAS_WFOPEN + #endif +#endif + +static drmp3_result drmp3_wfopen(FILE** ppFile, const wchar_t* pFilePath, const wchar_t* pOpenMode, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + if (ppFile != NULL) { + *ppFile = NULL; /* Safety. */ + } + + if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) { + return DRMP3_INVALID_ARGS; + } + +#if defined(DRMP3_HAS_WFOPEN) + { + /* Use _wfopen() on Windows. */ + #if defined(_MSC_VER) && _MSC_VER >= 1400 + errno_t err = _wfopen_s(ppFile, pFilePath, pOpenMode); + if (err != 0) { + return drmp3_result_from_errno(err); + } + #else + *ppFile = _wfopen(pFilePath, pOpenMode); + if (*ppFile == NULL) { + return drmp3_result_from_errno(errno); + } + #endif + (void)pAllocationCallbacks; + } +#else + /* + Use fopen() on anything other than Windows. Requires a conversion. This is annoying because + fopen() is locale specific. The only real way I can think of to do this is with wcsrtombs(). Note + that wcstombs() is apparently not thread-safe because it uses a static global mbstate_t object for + maintaining state. I've checked this with -std=c89 and it works, but if somebody get's a compiler + error I'll look into improving compatibility. + */ + + /* + Some compilers don't support wchar_t or wcsrtombs() which we're using below. In this case we just + need to abort with an error. If you encounter a compiler lacking such support, add it to this list + and submit a bug report and it'll be added to the library upstream. + */ + #if defined(__DJGPP__) + { + /* Nothing to do here. This will fall through to the error check below. */ + } + #else + { + mbstate_t mbs; + size_t lenMB; + const wchar_t* pFilePathTemp = pFilePath; + char* pFilePathMB = NULL; + char pOpenModeMB[32] = {0}; + + /* Get the length first. */ + DRMP3_ZERO_OBJECT(&mbs); + lenMB = wcsrtombs(NULL, &pFilePathTemp, 0, &mbs); + if (lenMB == (size_t)-1) { + return drmp3_result_from_errno(errno); + } + + pFilePathMB = (char*)drmp3__malloc_from_callbacks(lenMB + 1, pAllocationCallbacks); + if (pFilePathMB == NULL) { + return DRMP3_OUT_OF_MEMORY; + } + + pFilePathTemp = pFilePath; + DRMP3_ZERO_OBJECT(&mbs); + wcsrtombs(pFilePathMB, &pFilePathTemp, lenMB + 1, &mbs); + + /* The open mode should always consist of ASCII characters so we should be able to do a trivial conversion. */ + { + size_t i = 0; + for (;;) { + if (pOpenMode[i] == 0) { + pOpenModeMB[i] = '\0'; + break; + } + + pOpenModeMB[i] = (char)pOpenMode[i]; + i += 1; + } + } + + *ppFile = fopen(pFilePathMB, pOpenModeMB); + + drmp3__free_from_callbacks(pFilePathMB, pAllocationCallbacks); + } + #endif + + if (*ppFile == NULL) { + return DRMP3_ERROR; + } +#endif + + return DRMP3_SUCCESS; +} +/* End fopen */ + + +static size_t drmp3__on_read_stdio(void* pUserData, void* pBufferOut, size_t bytesToRead) +{ + return fread(pBufferOut, 1, bytesToRead, (FILE*)pUserData); +} + +static drmp3_bool32 drmp3__on_seek_stdio(void* pUserData, int offset, drmp3_seek_origin origin) +{ + int whence = SEEK_SET; + if (origin == DRMP3_SEEK_CUR) { + whence = SEEK_CUR; + } else if (origin == DRMP3_SEEK_END) { + whence = SEEK_END; + } + + return fseek((FILE*)pUserData, offset, whence) == 0; +} + +static drmp3_bool32 drmp3__on_tell_stdio(void* pUserData, drmp3_int64* pCursor) +{ + FILE* pFileStdio = (FILE*)pUserData; + drmp3_int64 result; + + /* These were all validated at a higher level. */ + DRMP3_ASSERT(pFileStdio != NULL); + DRMP3_ASSERT(pCursor != NULL); + +#if defined(_WIN32) && !defined(NXDK) + #if defined(_MSC_VER) && _MSC_VER > 1200 + result = _ftelli64(pFileStdio); + #else + result = ftell(pFileStdio); + #endif +#else + result = ftell(pFileStdio); +#endif + + *pCursor = result; + + return DRMP3_TRUE; +} + +DRMP3_API drmp3_bool32 drmp3_init_file_with_metadata(drmp3* pMP3, const char* pFilePath, drmp3_meta_proc onMeta, void* pUserDataMeta, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3_bool32 result; + FILE* pFile; + + if (pMP3 == NULL) { + return DRMP3_FALSE; + } + + DRMP3_ZERO_OBJECT(pMP3); + + if (drmp3_fopen(&pFile, pFilePath, "rb") != DRMP3_SUCCESS) { + return DRMP3_FALSE; + } + + result = drmp3_init_internal(pMP3, drmp3__on_read_stdio, drmp3__on_seek_stdio, drmp3__on_tell_stdio, onMeta, (void*)pFile, pUserDataMeta, pAllocationCallbacks); + if (result != DRMP3_TRUE) { + fclose(pFile); + return result; + } + + return DRMP3_TRUE; +} + +DRMP3_API drmp3_bool32 drmp3_init_file_with_metadata_w(drmp3* pMP3, const wchar_t* pFilePath, drmp3_meta_proc onMeta, void* pUserDataMeta, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3_bool32 result; + FILE* pFile; + + if (pMP3 == NULL) { + return DRMP3_FALSE; + } + + DRMP3_ZERO_OBJECT(pMP3); + + if (drmp3_wfopen(&pFile, pFilePath, L"rb", pAllocationCallbacks) != DRMP3_SUCCESS) { + return DRMP3_FALSE; + } + + result = drmp3_init_internal(pMP3, drmp3__on_read_stdio, drmp3__on_seek_stdio, drmp3__on_tell_stdio, onMeta, (void*)pFile, pUserDataMeta, pAllocationCallbacks); + if (result != DRMP3_TRUE) { + fclose(pFile); + return result; + } + + return DRMP3_TRUE; +} + +DRMP3_API drmp3_bool32 drmp3_init_file(drmp3* pMP3, const char* pFilePath, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + return drmp3_init_file_with_metadata(pMP3, pFilePath, NULL, NULL, pAllocationCallbacks); +} + +DRMP3_API drmp3_bool32 drmp3_init_file_w(drmp3* pMP3, const wchar_t* pFilePath, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + return drmp3_init_file_with_metadata_w(pMP3, pFilePath, NULL, NULL, pAllocationCallbacks); +} +#endif + +DRMP3_API void drmp3_uninit(drmp3* pMP3) +{ + if (pMP3 == NULL) { + return; + } + +#ifndef DR_MP3_NO_STDIO + if (pMP3->onRead == drmp3__on_read_stdio) { + FILE* pFile = (FILE*)pMP3->pUserData; + if (pFile != NULL) { + fclose(pFile); + pMP3->pUserData = NULL; /* Make sure the file handle is cleared to NULL to we don't attempt to close it a second time. */ + } + } +#endif + + drmp3__free_from_callbacks(pMP3->pData, &pMP3->allocationCallbacks); +} + +#if defined(DR_MP3_FLOAT_OUTPUT) +static void drmp3_f32_to_s16(drmp3_int16* dst, const float* src, drmp3_uint64 sampleCount) +{ + drmp3_uint64 i; + drmp3_uint64 i4; + drmp3_uint64 sampleCount4; + + /* Unrolled. */ + i = 0; + sampleCount4 = sampleCount >> 2; + for (i4 = 0; i4 < sampleCount4; i4 += 1) { + float x0 = src[i+0]; + float x1 = src[i+1]; + float x2 = src[i+2]; + float x3 = src[i+3]; + + x0 = ((x0 < -1) ? -1 : ((x0 > 1) ? 1 : x0)); + x1 = ((x1 < -1) ? -1 : ((x1 > 1) ? 1 : x1)); + x2 = ((x2 < -1) ? -1 : ((x2 > 1) ? 1 : x2)); + x3 = ((x3 < -1) ? -1 : ((x3 > 1) ? 1 : x3)); + + x0 = x0 * 32767.0f; + x1 = x1 * 32767.0f; + x2 = x2 * 32767.0f; + x3 = x3 * 32767.0f; + + dst[i+0] = (drmp3_int16)x0; + dst[i+1] = (drmp3_int16)x1; + dst[i+2] = (drmp3_int16)x2; + dst[i+3] = (drmp3_int16)x3; + + i += 4; + } + + /* Leftover. */ + for (; i < sampleCount; i += 1) { + float x = src[i]; + x = ((x < -1) ? -1 : ((x > 1) ? 1 : x)); /* clip */ + x = x * 32767.0f; /* -1..1 to -32767..32767 */ + + dst[i] = (drmp3_int16)x; + } +} +#endif + +#if !defined(DR_MP3_FLOAT_OUTPUT) +static void drmp3_s16_to_f32(float* dst, const drmp3_int16* src, drmp3_uint64 sampleCount) +{ + drmp3_uint64 i; + for (i = 0; i < sampleCount; i += 1) { + float x = (float)src[i]; + x = x * 0.000030517578125f; /* -32768..32767 to -1..0.999969482421875 */ + dst[i] = x; + } +} +#endif + + +static drmp3_uint64 drmp3_read_pcm_frames_raw(drmp3* pMP3, drmp3_uint64 framesToRead, void* pBufferOut) +{ + drmp3_uint64 totalFramesRead = 0; + + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pMP3->onRead != NULL); + + while (framesToRead > 0) { + drmp3_uint32 framesToConsume; + + /* Skip frames if necessary. */ + if (pMP3->currentPCMFrame < pMP3->delayInPCMFrames) { + drmp3_uint32 framesToSkip = (drmp3_uint32)DRMP3_MIN(pMP3->pcmFramesRemainingInMP3Frame, pMP3->delayInPCMFrames - pMP3->currentPCMFrame); + + pMP3->currentPCMFrame += framesToSkip; + pMP3->pcmFramesConsumedInMP3Frame += framesToSkip; + pMP3->pcmFramesRemainingInMP3Frame -= framesToSkip; + } + + framesToConsume = (drmp3_uint32)DRMP3_MIN(pMP3->pcmFramesRemainingInMP3Frame, framesToRead); + + /* Clamp the number of frames to read to the padding. */ + if (pMP3->totalPCMFrameCount != DRMP3_UINT64_MAX && pMP3->totalPCMFrameCount > pMP3->paddingInPCMFrames) { + if (pMP3->currentPCMFrame < (pMP3->totalPCMFrameCount - pMP3->paddingInPCMFrames)) { + drmp3_uint64 framesRemainigToPadding = (pMP3->totalPCMFrameCount - pMP3->paddingInPCMFrames) - pMP3->currentPCMFrame; + if (framesToConsume > framesRemainigToPadding) { + framesToConsume = (drmp3_uint32)framesRemainigToPadding; + } + } else { + /* We're into the padding. Abort. */ + break; + } + } + + if (pBufferOut != NULL) { + #if defined(DR_MP3_FLOAT_OUTPUT) + { + /* f32 */ + float* pFramesOutF32 = (float*)DRMP3_OFFSET_PTR(pBufferOut, sizeof(float) * totalFramesRead * pMP3->channels); + float* pFramesInF32 = (float*)DRMP3_OFFSET_PTR(&pMP3->pcmFrames[0], sizeof(float) * pMP3->pcmFramesConsumedInMP3Frame * pMP3->mp3FrameChannels); + DRMP3_COPY_MEMORY(pFramesOutF32, pFramesInF32, sizeof(float) * framesToConsume * pMP3->channels); + } + #else + { + /* s16 */ + drmp3_int16* pFramesOutS16 = (drmp3_int16*)DRMP3_OFFSET_PTR(pBufferOut, sizeof(drmp3_int16) * totalFramesRead * pMP3->channels); + drmp3_int16* pFramesInS16 = (drmp3_int16*)DRMP3_OFFSET_PTR(&pMP3->pcmFrames[0], sizeof(drmp3_int16) * pMP3->pcmFramesConsumedInMP3Frame * pMP3->mp3FrameChannels); + DRMP3_COPY_MEMORY(pFramesOutS16, pFramesInS16, sizeof(drmp3_int16) * framesToConsume * pMP3->channels); + } + #endif + } + + pMP3->currentPCMFrame += framesToConsume; + pMP3->pcmFramesConsumedInMP3Frame += framesToConsume; + pMP3->pcmFramesRemainingInMP3Frame -= framesToConsume; + totalFramesRead += framesToConsume; + framesToRead -= framesToConsume; + + if (framesToRead == 0) { + break; + } + + /* If the cursor is already at the padding we need to abort. */ + if (pMP3->totalPCMFrameCount != DRMP3_UINT64_MAX && pMP3->totalPCMFrameCount > pMP3->paddingInPCMFrames && pMP3->currentPCMFrame >= (pMP3->totalPCMFrameCount - pMP3->paddingInPCMFrames)) { + break; + } + + DRMP3_ASSERT(pMP3->pcmFramesRemainingInMP3Frame == 0); + + /* At this point we have exhausted our in-memory buffer so we need to re-fill. */ + if (drmp3_decode_next_frame(pMP3) == 0) { + break; + } + } + + return totalFramesRead; +} + + +DRMP3_API drmp3_uint64 drmp3_read_pcm_frames_f32(drmp3* pMP3, drmp3_uint64 framesToRead, float* pBufferOut) +{ + if (pMP3 == NULL || pMP3->onRead == NULL) { + return 0; + } + +#if defined(DR_MP3_FLOAT_OUTPUT) + /* Fast path. No conversion required. */ + return drmp3_read_pcm_frames_raw(pMP3, framesToRead, pBufferOut); +#else + /* Slow path. Convert from s16 to f32. */ + { + drmp3_int16 pTempS16[1152*2]; /* MP3 frames have a maximum per-channel sample count of 1152. Times 2 to account for stereo. */ + drmp3_uint64 totalPCMFramesRead = 0; + + while (totalPCMFramesRead < framesToRead) { + drmp3_uint64 framesJustRead; + drmp3_uint64 framesRemaining = framesToRead - totalPCMFramesRead; + drmp3_uint64 framesToReadNow = DRMP3_COUNTOF(pTempS16) / pMP3->channels; + if (framesToReadNow > framesRemaining) { + framesToReadNow = framesRemaining; + } + + framesJustRead = drmp3_read_pcm_frames_raw(pMP3, framesToReadNow, pTempS16); + if (framesJustRead == 0) { + break; + } + + drmp3_s16_to_f32((float*)DRMP3_OFFSET_PTR(pBufferOut, sizeof(float) * totalPCMFramesRead * pMP3->channels), pTempS16, framesJustRead * pMP3->channels); + totalPCMFramesRead += framesJustRead; + } + + return totalPCMFramesRead; + } +#endif +} + +DRMP3_API drmp3_uint64 drmp3_read_pcm_frames_s16(drmp3* pMP3, drmp3_uint64 framesToRead, drmp3_int16* pBufferOut) +{ + if (pMP3 == NULL || pMP3->onRead == NULL) { + return 0; + } + +#if !defined(DR_MP3_FLOAT_OUTPUT) + /* Fast path. No conversion required. */ + return drmp3_read_pcm_frames_raw(pMP3, framesToRead, pBufferOut); +#else + /* Slow path. Convert from f32 to s16. */ + { + float pTempF32[1152*2]; /* MP3 frames have a maximum per-channel sample count of 1152. Times 2 to account for stereo. */ + drmp3_uint64 totalPCMFramesRead = 0; + + while (totalPCMFramesRead < framesToRead) { + drmp3_uint64 framesJustRead; + drmp3_uint64 framesRemaining = framesToRead - totalPCMFramesRead; + drmp3_uint64 framesToReadNow = DRMP3_COUNTOF(pTempF32) / pMP3->channels; + if (framesToReadNow > framesRemaining) { + framesToReadNow = framesRemaining; + } + + framesJustRead = drmp3_read_pcm_frames_raw(pMP3, framesToReadNow, pTempF32); + if (framesJustRead == 0) { + break; + } + + drmp3_f32_to_s16((drmp3_int16*)DRMP3_OFFSET_PTR(pBufferOut, sizeof(drmp3_int16) * totalPCMFramesRead * pMP3->channels), pTempF32, framesJustRead * pMP3->channels); + totalPCMFramesRead += framesJustRead; + } + + return totalPCMFramesRead; + } +#endif +} + +static void drmp3_reset(drmp3* pMP3) +{ + DRMP3_ASSERT(pMP3 != NULL); + + pMP3->pcmFramesConsumedInMP3Frame = 0; + pMP3->pcmFramesRemainingInMP3Frame = 0; + pMP3->currentPCMFrame = 0; + pMP3->dataSize = 0; + pMP3->atEnd = DRMP3_FALSE; + drmp3dec_init(&pMP3->decoder); +} + +static drmp3_bool32 drmp3_seek_to_start_of_stream(drmp3* pMP3) +{ + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pMP3->onSeek != NULL); + + /* Seek to the start of the stream to begin with. */ + if (!drmp3__on_seek_64(pMP3, pMP3->streamStartOffset, DRMP3_SEEK_SET)) { + return DRMP3_FALSE; + } + + /* Clear any cached data. */ + drmp3_reset(pMP3); + return DRMP3_TRUE; +} + + +static drmp3_bool32 drmp3_seek_forward_by_pcm_frames__brute_force(drmp3* pMP3, drmp3_uint64 frameOffset) +{ + drmp3_uint64 framesRead; + + /* + Just using a dumb read-and-discard for now. What would be nice is to parse only the header of the MP3 frame, and then skip over leading + frames without spending the time doing a full decode. I cannot see an easy way to do this in minimp3, however, so it may involve some + kind of manual processing. + */ +#if defined(DR_MP3_FLOAT_OUTPUT) + framesRead = drmp3_read_pcm_frames_f32(pMP3, frameOffset, NULL); +#else + framesRead = drmp3_read_pcm_frames_s16(pMP3, frameOffset, NULL); +#endif + if (framesRead != frameOffset) { + return DRMP3_FALSE; + } + + return DRMP3_TRUE; +} + +static drmp3_bool32 drmp3_seek_to_pcm_frame__brute_force(drmp3* pMP3, drmp3_uint64 frameIndex) +{ + DRMP3_ASSERT(pMP3 != NULL); + + if (frameIndex == pMP3->currentPCMFrame) { + return DRMP3_TRUE; + } + + /* + If we're moving foward we just read from where we're at. Otherwise we need to move back to the start of + the stream and read from the beginning. + */ + if (frameIndex < pMP3->currentPCMFrame) { + /* Moving backward. Move to the start of the stream and then move forward. */ + if (!drmp3_seek_to_start_of_stream(pMP3)) { + return DRMP3_FALSE; + } + } + + DRMP3_ASSERT(frameIndex >= pMP3->currentPCMFrame); + return drmp3_seek_forward_by_pcm_frames__brute_force(pMP3, (frameIndex - pMP3->currentPCMFrame)); +} + +static drmp3_bool32 drmp3_find_closest_seek_point(drmp3* pMP3, drmp3_uint64 frameIndex, drmp3_uint32* pSeekPointIndex) +{ + drmp3_uint32 iSeekPoint; + + DRMP3_ASSERT(pSeekPointIndex != NULL); + + *pSeekPointIndex = 0; + + if (frameIndex < pMP3->pSeekPoints[0].pcmFrameIndex) { + return DRMP3_FALSE; + } + + /* Linear search for simplicity to begin with while I'm getting this thing working. Once it's all working change this to a binary search. */ + for (iSeekPoint = 0; iSeekPoint < pMP3->seekPointCount; ++iSeekPoint) { + if (pMP3->pSeekPoints[iSeekPoint].pcmFrameIndex > frameIndex) { + break; /* Found it. */ + } + + *pSeekPointIndex = iSeekPoint; + } + + return DRMP3_TRUE; +} + +static drmp3_bool32 drmp3_seek_to_pcm_frame__seek_table(drmp3* pMP3, drmp3_uint64 frameIndex) +{ + drmp3_seek_point seekPoint; + drmp3_uint32 priorSeekPointIndex; + drmp3_uint16 iMP3Frame; + drmp3_uint64 leftoverFrames; + + DRMP3_ASSERT(pMP3 != NULL); + DRMP3_ASSERT(pMP3->pSeekPoints != NULL); + DRMP3_ASSERT(pMP3->seekPointCount > 0); + + /* If there is no prior seekpoint it means the target PCM frame comes before the first seek point. Just assume a seekpoint at the start of the file in this case. */ + if (drmp3_find_closest_seek_point(pMP3, frameIndex, &priorSeekPointIndex)) { + seekPoint = pMP3->pSeekPoints[priorSeekPointIndex]; + } else { + seekPoint.seekPosInBytes = 0; + seekPoint.pcmFrameIndex = 0; + seekPoint.mp3FramesToDiscard = 0; + seekPoint.pcmFramesToDiscard = 0; + } + + /* First thing to do is seek to the first byte of the relevant MP3 frame. */ + if (!drmp3__on_seek_64(pMP3, seekPoint.seekPosInBytes, DRMP3_SEEK_SET)) { + return DRMP3_FALSE; /* Failed to seek. */ + } + + /* Clear any cached data. */ + drmp3_reset(pMP3); + + /* Whole MP3 frames need to be discarded first. */ + for (iMP3Frame = 0; iMP3Frame < seekPoint.mp3FramesToDiscard; ++iMP3Frame) { + drmp3_uint32 pcmFramesRead; + drmp3d_sample_t* pPCMFrames; + + /* Pass in non-null for the last frame because we want to ensure the sample rate converter is preloaded correctly. */ + pPCMFrames = NULL; + if (iMP3Frame == seekPoint.mp3FramesToDiscard-1) { + pPCMFrames = (drmp3d_sample_t*)pMP3->pcmFrames; + } + + /* We first need to decode the next frame. */ + pcmFramesRead = drmp3_decode_next_frame_ex(pMP3, pPCMFrames, NULL, NULL); + if (pcmFramesRead == 0) { + return DRMP3_FALSE; + } + } + + /* We seeked to an MP3 frame in the raw stream so we need to make sure the current PCM frame is set correctly. */ + pMP3->currentPCMFrame = seekPoint.pcmFrameIndex - seekPoint.pcmFramesToDiscard; + + /* + Now at this point we can follow the same process as the brute force technique where we just skip over unnecessary MP3 frames and then + read-and-discard at least 2 whole MP3 frames. + */ + leftoverFrames = frameIndex - pMP3->currentPCMFrame; + return drmp3_seek_forward_by_pcm_frames__brute_force(pMP3, leftoverFrames); +} + +DRMP3_API drmp3_bool32 drmp3_seek_to_pcm_frame(drmp3* pMP3, drmp3_uint64 frameIndex) +{ + if (pMP3 == NULL || pMP3->onSeek == NULL) { + return DRMP3_FALSE; + } + + if (frameIndex == 0) { + return drmp3_seek_to_start_of_stream(pMP3); + } + + /* Use the seek table if we have one. */ + if (pMP3->pSeekPoints != NULL && pMP3->seekPointCount > 0) { + return drmp3_seek_to_pcm_frame__seek_table(pMP3, frameIndex); + } else { + return drmp3_seek_to_pcm_frame__brute_force(pMP3, frameIndex); + } +} + +DRMP3_API drmp3_bool32 drmp3_get_mp3_and_pcm_frame_count(drmp3* pMP3, drmp3_uint64* pMP3FrameCount, drmp3_uint64* pPCMFrameCount) +{ + drmp3_uint64 currentPCMFrame; + drmp3_uint64 totalPCMFrameCount; + drmp3_uint64 totalMP3FrameCount; + + if (pMP3 == NULL) { + return DRMP3_FALSE; + } + + /* + The way this works is we move back to the start of the stream, iterate over each MP3 frame and calculate the frame count based + on our output sample rate, the seek back to the PCM frame we were sitting on before calling this function. + */ + + /* The stream must support seeking for this to work. */ + if (pMP3->onSeek == NULL) { + return DRMP3_FALSE; + } + + /* We'll need to seek back to where we were, so grab the PCM frame we're currently sitting on so we can restore later. */ + currentPCMFrame = pMP3->currentPCMFrame; + + if (!drmp3_seek_to_start_of_stream(pMP3)) { + return DRMP3_FALSE; + } + + totalPCMFrameCount = 0; + totalMP3FrameCount = 0; + + for (;;) { + drmp3_uint32 pcmFramesInCurrentMP3Frame; + + pcmFramesInCurrentMP3Frame = drmp3_decode_next_frame_ex(pMP3, NULL, NULL, NULL); + if (pcmFramesInCurrentMP3Frame == 0) { + break; + } + + totalPCMFrameCount += pcmFramesInCurrentMP3Frame; + totalMP3FrameCount += 1; + } + + /* Finally, we need to seek back to where we were. */ + if (!drmp3_seek_to_start_of_stream(pMP3)) { + return DRMP3_FALSE; + } + + if (!drmp3_seek_to_pcm_frame(pMP3, currentPCMFrame)) { + return DRMP3_FALSE; + } + + if (pMP3FrameCount != NULL) { + *pMP3FrameCount = totalMP3FrameCount; + } + if (pPCMFrameCount != NULL) { + *pPCMFrameCount = totalPCMFrameCount; + } + + return DRMP3_TRUE; +} + +DRMP3_API drmp3_uint64 drmp3_get_pcm_frame_count(drmp3* pMP3) +{ + drmp3_uint64 totalPCMFrameCount; + + if (pMP3 == NULL) { + return 0; + } + + if (pMP3->totalPCMFrameCount != DRMP3_UINT64_MAX) { + totalPCMFrameCount = pMP3->totalPCMFrameCount; + + if (totalPCMFrameCount >= pMP3->delayInPCMFrames) { + totalPCMFrameCount -= pMP3->delayInPCMFrames; + } else { + /* The delay is greater than the frame count reported by the Xing/Info tag. Assume it's invalid and ignore. */ + } + + if (totalPCMFrameCount >= pMP3->paddingInPCMFrames) { + totalPCMFrameCount -= pMP3->paddingInPCMFrames; + } else { + /* The padding is greater than the frame count reported by the Xing/Info tag. Assume it's invalid and ignore. */ + } + + return totalPCMFrameCount; + } else { + /* Unknown frame count. Need to calculate it. */ + if (!drmp3_get_mp3_and_pcm_frame_count(pMP3, NULL, &totalPCMFrameCount)) { + return 0; + } + + return totalPCMFrameCount; + } +} + +DRMP3_API drmp3_uint64 drmp3_get_mp3_frame_count(drmp3* pMP3) +{ + drmp3_uint64 totalMP3FrameCount; + if (!drmp3_get_mp3_and_pcm_frame_count(pMP3, &totalMP3FrameCount, NULL)) { + return 0; + } + + return totalMP3FrameCount; +} + +static void drmp3__accumulate_running_pcm_frame_count(drmp3* pMP3, drmp3_uint32 pcmFrameCountIn, drmp3_uint64* pRunningPCMFrameCount, float* pRunningPCMFrameCountFractionalPart) +{ + float srcRatio; + float pcmFrameCountOutF; + drmp3_uint32 pcmFrameCountOut; + + srcRatio = (float)pMP3->mp3FrameSampleRate / (float)pMP3->sampleRate; + DRMP3_ASSERT(srcRatio > 0); + + pcmFrameCountOutF = *pRunningPCMFrameCountFractionalPart + (pcmFrameCountIn / srcRatio); + pcmFrameCountOut = (drmp3_uint32)pcmFrameCountOutF; + *pRunningPCMFrameCountFractionalPart = pcmFrameCountOutF - pcmFrameCountOut; + *pRunningPCMFrameCount += pcmFrameCountOut; +} + +typedef struct +{ + drmp3_uint64 bytePos; + drmp3_uint64 pcmFrameIndex; /* <-- After sample rate conversion. */ +} drmp3__seeking_mp3_frame_info; + +DRMP3_API drmp3_bool32 drmp3_calculate_seek_points(drmp3* pMP3, drmp3_uint32* pSeekPointCount, drmp3_seek_point* pSeekPoints) +{ + drmp3_uint32 seekPointCount; + drmp3_uint64 currentPCMFrame; + drmp3_uint64 totalMP3FrameCount; + drmp3_uint64 totalPCMFrameCount; + + if (pMP3 == NULL || pSeekPointCount == NULL || pSeekPoints == NULL) { + return DRMP3_FALSE; /* Invalid args. */ + } + + seekPointCount = *pSeekPointCount; + if (seekPointCount == 0) { + return DRMP3_FALSE; /* The client has requested no seek points. Consider this to be invalid arguments since the client has probably not intended this. */ + } + + /* We'll need to seek back to the current sample after calculating the seekpoints so we need to go ahead and grab the current location at the top. */ + currentPCMFrame = pMP3->currentPCMFrame; + + /* We never do more than the total number of MP3 frames and we limit it to 32-bits. */ + if (!drmp3_get_mp3_and_pcm_frame_count(pMP3, &totalMP3FrameCount, &totalPCMFrameCount)) { + return DRMP3_FALSE; + } + + /* If there's less than DRMP3_SEEK_LEADING_MP3_FRAMES+1 frames we just report 1 seek point which will be the very start of the stream. */ + if (totalMP3FrameCount < DRMP3_SEEK_LEADING_MP3_FRAMES+1) { + seekPointCount = 1; + pSeekPoints[0].seekPosInBytes = 0; + pSeekPoints[0].pcmFrameIndex = 0; + pSeekPoints[0].mp3FramesToDiscard = 0; + pSeekPoints[0].pcmFramesToDiscard = 0; + } else { + drmp3_uint64 pcmFramesBetweenSeekPoints; + drmp3__seeking_mp3_frame_info mp3FrameInfo[DRMP3_SEEK_LEADING_MP3_FRAMES+1]; + drmp3_uint64 runningPCMFrameCount = 0; + float runningPCMFrameCountFractionalPart = 0; + drmp3_uint64 nextTargetPCMFrame; + drmp3_uint32 iMP3Frame; + drmp3_uint32 iSeekPoint; + + if (seekPointCount > totalMP3FrameCount-1) { + seekPointCount = (drmp3_uint32)totalMP3FrameCount-1; + } + + pcmFramesBetweenSeekPoints = totalPCMFrameCount / (seekPointCount+1); + + /* + Here is where we actually calculate the seek points. We need to start by moving the start of the stream. We then enumerate over each + MP3 frame. + */ + if (!drmp3_seek_to_start_of_stream(pMP3)) { + return DRMP3_FALSE; + } + + /* + We need to cache the byte positions of the previous MP3 frames. As a new MP3 frame is iterated, we cycle the byte positions in this + array. The value in the first item in this array is the byte position that will be reported in the next seek point. + */ + + /* We need to initialize the array of MP3 byte positions for the leading MP3 frames. */ + for (iMP3Frame = 0; iMP3Frame < DRMP3_SEEK_LEADING_MP3_FRAMES+1; ++iMP3Frame) { + drmp3_uint32 pcmFramesInCurrentMP3FrameIn; + + /* The byte position of the next frame will be the stream's cursor position, minus whatever is sitting in the buffer. */ + DRMP3_ASSERT(pMP3->streamCursor >= pMP3->dataSize); + mp3FrameInfo[iMP3Frame].bytePos = pMP3->streamCursor - pMP3->dataSize; + mp3FrameInfo[iMP3Frame].pcmFrameIndex = runningPCMFrameCount; + + /* We need to get information about this frame so we can know how many samples it contained. */ + pcmFramesInCurrentMP3FrameIn = drmp3_decode_next_frame_ex(pMP3, NULL, NULL, NULL); + if (pcmFramesInCurrentMP3FrameIn == 0) { + return DRMP3_FALSE; /* This should never happen. */ + } + + drmp3__accumulate_running_pcm_frame_count(pMP3, pcmFramesInCurrentMP3FrameIn, &runningPCMFrameCount, &runningPCMFrameCountFractionalPart); + } + + /* + At this point we will have extracted the byte positions of the leading MP3 frames. We can now start iterating over each seek point and + calculate them. + */ + nextTargetPCMFrame = 0; + for (iSeekPoint = 0; iSeekPoint < seekPointCount; ++iSeekPoint) { + nextTargetPCMFrame += pcmFramesBetweenSeekPoints; + + for (;;) { + if (nextTargetPCMFrame < runningPCMFrameCount) { + /* The next seek point is in the current MP3 frame. */ + pSeekPoints[iSeekPoint].seekPosInBytes = mp3FrameInfo[0].bytePos; + pSeekPoints[iSeekPoint].pcmFrameIndex = nextTargetPCMFrame; + pSeekPoints[iSeekPoint].mp3FramesToDiscard = DRMP3_SEEK_LEADING_MP3_FRAMES; + pSeekPoints[iSeekPoint].pcmFramesToDiscard = (drmp3_uint16)(nextTargetPCMFrame - mp3FrameInfo[DRMP3_SEEK_LEADING_MP3_FRAMES-1].pcmFrameIndex); + break; + } else { + size_t i; + drmp3_uint32 pcmFramesInCurrentMP3FrameIn; + + /* + The next seek point is not in the current MP3 frame, so continue on to the next one. The first thing to do is cycle the cached + MP3 frame info. + */ + for (i = 0; i < DRMP3_COUNTOF(mp3FrameInfo)-1; ++i) { + mp3FrameInfo[i] = mp3FrameInfo[i+1]; + } + + /* Cache previous MP3 frame info. */ + mp3FrameInfo[DRMP3_COUNTOF(mp3FrameInfo)-1].bytePos = pMP3->streamCursor - pMP3->dataSize; + mp3FrameInfo[DRMP3_COUNTOF(mp3FrameInfo)-1].pcmFrameIndex = runningPCMFrameCount; + + /* + Go to the next MP3 frame. This shouldn't ever fail, but just in case it does we just set the seek point and break. If it happens, it + should only ever do it for the last seek point. + */ + pcmFramesInCurrentMP3FrameIn = drmp3_decode_next_frame_ex(pMP3, NULL, NULL, NULL); + if (pcmFramesInCurrentMP3FrameIn == 0) { + pSeekPoints[iSeekPoint].seekPosInBytes = mp3FrameInfo[0].bytePos; + pSeekPoints[iSeekPoint].pcmFrameIndex = nextTargetPCMFrame; + pSeekPoints[iSeekPoint].mp3FramesToDiscard = DRMP3_SEEK_LEADING_MP3_FRAMES; + pSeekPoints[iSeekPoint].pcmFramesToDiscard = (drmp3_uint16)(nextTargetPCMFrame - mp3FrameInfo[DRMP3_SEEK_LEADING_MP3_FRAMES-1].pcmFrameIndex); + break; + } + + drmp3__accumulate_running_pcm_frame_count(pMP3, pcmFramesInCurrentMP3FrameIn, &runningPCMFrameCount, &runningPCMFrameCountFractionalPart); + } + } + } + + /* Finally, we need to seek back to where we were. */ + if (!drmp3_seek_to_start_of_stream(pMP3)) { + return DRMP3_FALSE; + } + if (!drmp3_seek_to_pcm_frame(pMP3, currentPCMFrame)) { + return DRMP3_FALSE; + } + } + + *pSeekPointCount = seekPointCount; + return DRMP3_TRUE; +} + +DRMP3_API drmp3_bool32 drmp3_bind_seek_table(drmp3* pMP3, drmp3_uint32 seekPointCount, drmp3_seek_point* pSeekPoints) +{ + if (pMP3 == NULL) { + return DRMP3_FALSE; + } + + if (seekPointCount == 0 || pSeekPoints == NULL) { + /* Unbinding. */ + pMP3->seekPointCount = 0; + pMP3->pSeekPoints = NULL; + } else { + /* Binding. */ + pMP3->seekPointCount = seekPointCount; + pMP3->pSeekPoints = pSeekPoints; + } + + return DRMP3_TRUE; +} + + +static float* drmp3__full_read_and_close_f32(drmp3* pMP3, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) +{ + drmp3_uint64 totalFramesRead = 0; + drmp3_uint64 framesCapacity = 0; + float* pFrames = NULL; + float temp[1152*2]; /* MP3 frames have a maximum per-channel sample count of 1152. Times 2 to account for stereo. */ + + DRMP3_ASSERT(pMP3 != NULL); + + for (;;) { + drmp3_uint64 framesToReadRightNow = DRMP3_COUNTOF(temp) / pMP3->channels; + drmp3_uint64 framesJustRead = drmp3_read_pcm_frames_f32(pMP3, framesToReadRightNow, temp); + if (framesJustRead == 0) { + break; + } + + /* Reallocate the output buffer if there's not enough room. */ + if (framesCapacity < totalFramesRead + framesJustRead) { + drmp3_uint64 oldFramesBufferSize; + drmp3_uint64 newFramesBufferSize; + drmp3_uint64 newFramesCap; + float* pNewFrames; + + newFramesCap = framesCapacity * 2; + if (newFramesCap < totalFramesRead + framesJustRead) { + newFramesCap = totalFramesRead + framesJustRead; + } + + oldFramesBufferSize = framesCapacity * pMP3->channels * sizeof(float); + newFramesBufferSize = newFramesCap * pMP3->channels * sizeof(float); + if (newFramesBufferSize > (drmp3_uint64)DRMP3_SIZE_MAX) { + break; + } + + pNewFrames = (float*)drmp3__realloc_from_callbacks(pFrames, (size_t)newFramesBufferSize, (size_t)oldFramesBufferSize, &pMP3->allocationCallbacks); + if (pNewFrames == NULL) { + drmp3__free_from_callbacks(pFrames, &pMP3->allocationCallbacks); + pFrames = NULL; + totalFramesRead = 0; + break; + } + + pFrames = pNewFrames; + framesCapacity = newFramesCap; + } + + DRMP3_COPY_MEMORY(pFrames + totalFramesRead*pMP3->channels, temp, (size_t)(framesJustRead*pMP3->channels*sizeof(float))); + totalFramesRead += framesJustRead; + + /* If the number of frames we asked for is less that what we actually read it means we've reached the end. */ + if (framesJustRead != framesToReadRightNow) { + break; + } + } + + if (pConfig != NULL) { + pConfig->channels = pMP3->channels; + pConfig->sampleRate = pMP3->sampleRate; + } + + drmp3_uninit(pMP3); + + if (pTotalFrameCount) { + *pTotalFrameCount = totalFramesRead; + } + + return pFrames; +} + +static drmp3_int16* drmp3__full_read_and_close_s16(drmp3* pMP3, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount) +{ + drmp3_uint64 totalFramesRead = 0; + drmp3_uint64 framesCapacity = 0; + drmp3_int16* pFrames = NULL; + drmp3_int16 temp[1152*2]; /* MP3 frames have a maximum per-channel sample count of 1152. Times 2 to account for stereo. */ + + DRMP3_ASSERT(pMP3 != NULL); + + for (;;) { + drmp3_uint64 framesToReadRightNow = DRMP3_COUNTOF(temp) / pMP3->channels; + drmp3_uint64 framesJustRead = drmp3_read_pcm_frames_s16(pMP3, framesToReadRightNow, temp); + if (framesJustRead == 0) { + break; + } + + /* Reallocate the output buffer if there's not enough room. */ + if (framesCapacity < totalFramesRead + framesJustRead) { + drmp3_uint64 newFramesBufferSize; + drmp3_uint64 oldFramesBufferSize; + drmp3_uint64 newFramesCap; + drmp3_int16* pNewFrames; + + newFramesCap = framesCapacity * 2; + if (newFramesCap < totalFramesRead + framesJustRead) { + newFramesCap = totalFramesRead + framesJustRead; + } + + oldFramesBufferSize = framesCapacity * pMP3->channels * sizeof(drmp3_int16); + newFramesBufferSize = newFramesCap * pMP3->channels * sizeof(drmp3_int16); + if (newFramesBufferSize > (drmp3_uint64)DRMP3_SIZE_MAX) { + break; + } + + pNewFrames = (drmp3_int16*)drmp3__realloc_from_callbacks(pFrames, (size_t)newFramesBufferSize, (size_t)oldFramesBufferSize, &pMP3->allocationCallbacks); + if (pNewFrames == NULL) { + drmp3__free_from_callbacks(pFrames, &pMP3->allocationCallbacks); + pFrames = NULL; + totalFramesRead = 0; + break; + } + + pFrames = pNewFrames; + framesCapacity = newFramesCap; + } + + DRMP3_COPY_MEMORY(pFrames + totalFramesRead*pMP3->channels, temp, (size_t)(framesJustRead*pMP3->channels*sizeof(drmp3_int16))); + totalFramesRead += framesJustRead; + + /* If the number of frames we asked for is less that what we actually read it means we've reached the end. */ + if (framesJustRead != framesToReadRightNow) { + break; + } + } + + if (pConfig != NULL) { + pConfig->channels = pMP3->channels; + pConfig->sampleRate = pMP3->sampleRate; + } + + drmp3_uninit(pMP3); + + if (pTotalFrameCount) { + *pTotalFrameCount = totalFramesRead; + } + + return pFrames; +} + + +DRMP3_API float* drmp3_open_and_read_pcm_frames_f32(drmp3_read_proc onRead, drmp3_seek_proc onSeek, drmp3_tell_proc onTell, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3 mp3; + if (!drmp3_init(&mp3, onRead, onSeek, onTell, NULL, pUserData, pAllocationCallbacks)) { + return NULL; + } + + return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount); +} + +DRMP3_API drmp3_int16* drmp3_open_and_read_pcm_frames_s16(drmp3_read_proc onRead, drmp3_seek_proc onSeek, drmp3_tell_proc onTell, void* pUserData, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3 mp3; + if (!drmp3_init(&mp3, onRead, onSeek, onTell, NULL, pUserData, pAllocationCallbacks)) { + return NULL; + } + + return drmp3__full_read_and_close_s16(&mp3, pConfig, pTotalFrameCount); +} + + +DRMP3_API float* drmp3_open_memory_and_read_pcm_frames_f32(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3 mp3; + if (!drmp3_init_memory(&mp3, pData, dataSize, pAllocationCallbacks)) { + return NULL; + } + + return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount); +} + +DRMP3_API drmp3_int16* drmp3_open_memory_and_read_pcm_frames_s16(const void* pData, size_t dataSize, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3 mp3; + if (!drmp3_init_memory(&mp3, pData, dataSize, pAllocationCallbacks)) { + return NULL; + } + + return drmp3__full_read_and_close_s16(&mp3, pConfig, pTotalFrameCount); +} + + +#ifndef DR_MP3_NO_STDIO +DRMP3_API float* drmp3_open_file_and_read_pcm_frames_f32(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3 mp3; + if (!drmp3_init_file(&mp3, filePath, pAllocationCallbacks)) { + return NULL; + } + + return drmp3__full_read_and_close_f32(&mp3, pConfig, pTotalFrameCount); +} + +DRMP3_API drmp3_int16* drmp3_open_file_and_read_pcm_frames_s16(const char* filePath, drmp3_config* pConfig, drmp3_uint64* pTotalFrameCount, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + drmp3 mp3; + if (!drmp3_init_file(&mp3, filePath, pAllocationCallbacks)) { + return NULL; + } + + return drmp3__full_read_and_close_s16(&mp3, pConfig, pTotalFrameCount); +} +#endif + +DRMP3_API void* drmp3_malloc(size_t sz, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks != NULL) { + return drmp3__malloc_from_callbacks(sz, pAllocationCallbacks); + } else { + return drmp3__malloc_default(sz, NULL); + } +} + +DRMP3_API void drmp3_free(void* p, const drmp3_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks != NULL) { + drmp3__free_from_callbacks(p, pAllocationCallbacks); + } else { + drmp3__free_default(p, NULL); + } +} + +#endif /* dr_mp3_c */ +#endif /*DR_MP3_IMPLEMENTATION*/ + +/* +DIFFERENCES BETWEEN minimp3 AND dr_mp3 +====================================== +- First, keep in mind that minimp3 (https://github.com/lieff/minimp3) is where all the real work was done. All of the + code relating to the actual decoding remains mostly unmodified, apart from some namespacing changes. +- dr_mp3 adds a pulling style API which allows you to deliver raw data via callbacks. So, rather than pushing data + to the decoder, the decoder _pulls_ data from your callbacks. +- In addition to callbacks, a decoder can be initialized from a block of memory and a file. +- The dr_mp3 pull API reads PCM frames rather than whole MP3 frames. +- dr_mp3 adds convenience APIs for opening and decoding entire files in one go. +- dr_mp3 is fully namespaced, including the implementation section, which is more suitable when compiling projects + as a single translation unit (aka unity builds). At the time of writing this, a unity build is not possible when + using minimp3 in conjunction with stb_vorbis. dr_mp3 addresses this. +*/ + +/* +REVISION HISTORY +================ +v0.7.4 - TBD + - Fix an overflow error with "Xing" and "Info" tag parsing. + - Add some validation checks for "Xing" and "Info" tag parsing. + - Reduce size of some stack allocations. + - Improvements to SIMD detection. + +v0.7.3 - 2026-01-17 + - Fix an error in drmp3_open_and_read_pcm_frames_s16() and family when memory allocation fails. + - Fix some compilation warnings. + +v0.7.2 - 2025-12-02 + - Reduce stack space to improve robustness on embedded systems. + - Fix a compilation error with MSVC Clang toolset relating to cpuid. + - Fix an error with APE tag parsing. + +v0.7.1 - 2025-09-10 + - Silence a warning with GCC. + - Fix an error with the NXDK build. + - Fix a decoding inconsistency when seeking. Prior to this change, reading to the end of the stream immediately after initializing will result in a different number of samples read than if the stream is seeked to the start and read to the end. + +v0.7.0 - 2025-07-23 + - The old `DRMP3_IMPLEMENTATION` has been removed. Use `DR_MP3_IMPLEMENTATION` instead. The reason for this change is that in the future everything will eventually be using the underscored naming convention in the future, so `drmp3` will become `dr_mp3`. + - API CHANGE: Seek origins have been renamed to match the naming convention used by dr_wav and my other libraries. + - drmp3_seek_origin_start -> DRMP3_SEEK_SET + - drmp3_seek_origin_current -> DRMP3_SEEK_CUR + - DRMP3_SEEK_END (new) + - API CHANGE: Add DRMP3_SEEK_END as a seek origin for the seek callback. This is required for detection of ID3v1 and APE tags. + - API CHANGE: Add onTell callback to `drmp3_init()`. This is needed in order to track the location of ID3v1 and APE tags. + - API CHANGE: Add onMeta callback to `drmp3_init()`. This is used for reporting tag data back to the caller. Currently this only reports the raw tag data which means applications need to parse the data themselves. + - API CHANGE: Rename `drmp3dec_frame_info.hz` to `drmp3dec_frame_info.sample_rate`. + - Add detection of ID3v2, ID3v1, APE and Xing/VBRI tags. This should fix errors with some files where the decoder was reading tags as audio data. + - Delay and padding samples from LAME tags are now handled. + - Fix compilation for AIX OS. + +v0.6.40 - 2024-12-17 + - Improve detection of ARM64EC + +v0.6.39 - 2024-02-27 + - Fix a Wdouble-promotion warning. + +v0.6.38 - 2023-11-02 + - Fix build for ARMv6-M. + +v0.6.37 - 2023-07-07 + - Silence a static analysis warning. + +v0.6.36 - 2023-06-17 + - Fix an incorrect date in revision history. No functional change. + +v0.6.35 - 2023-05-22 + - Minor code restructure. No functional change. + +v0.6.34 - 2022-09-17 + - Fix compilation with DJGPP. + - Fix compilation when compiling with x86 with no SSE2. + - Remove an unnecessary variable from the drmp3 structure. + +v0.6.33 - 2022-04-10 + - Fix compilation error with the MSVC ARM64 build. + - Fix compilation error on older versions of GCC. + - Remove some unused functions. + +v0.6.32 - 2021-12-11 + - Fix a warning with Clang. + +v0.6.31 - 2021-08-22 + - Fix a bug when loading from memory. + +v0.6.30 - 2021-08-16 + - Silence some warnings. + - Replace memory operations with DRMP3_* macros. + +v0.6.29 - 2021-08-08 + - Bring up to date with minimp3. + +v0.6.28 - 2021-07-31 + - Fix platform detection for ARM64. + - Fix a compilation error with C89. + +v0.6.27 - 2021-02-21 + - Fix a warning due to referencing _MSC_VER when it is undefined. + +v0.6.26 - 2021-01-31 + - Bring up to date with minimp3. + +v0.6.25 - 2020-12-26 + - Remove DRMP3_DEFAULT_CHANNELS and DRMP3_DEFAULT_SAMPLE_RATE which are leftovers from some removed APIs. + +v0.6.24 - 2020-12-07 + - Fix a typo in version date for 0.6.23. + +v0.6.23 - 2020-12-03 + - Fix an error where a file can be closed twice when initialization of the decoder fails. + +v0.6.22 - 2020-12-02 + - Fix an error where it's possible for a file handle to be left open when initialization of the decoder fails. + +v0.6.21 - 2020-11-28 + - Bring up to date with minimp3. + +v0.6.20 - 2020-11-21 + - Fix compilation with OpenWatcom. + +v0.6.19 - 2020-11-13 + - Minor code clean up. + +v0.6.18 - 2020-11-01 + - Improve compiler support for older versions of GCC. + +v0.6.17 - 2020-09-28 + - Bring up to date with minimp3. + +v0.6.16 - 2020-08-02 + - Simplify sized types. + +v0.6.15 - 2020-07-25 + - Fix a compilation warning. + +v0.6.14 - 2020-07-23 + - Fix undefined behaviour with memmove(). + +v0.6.13 - 2020-07-06 + - Fix a bug when converting from s16 to f32 in drmp3_read_pcm_frames_f32(). + +v0.6.12 - 2020-06-23 + - Add include guard for the implementation section. + +v0.6.11 - 2020-05-26 + - Fix use of uninitialized variable error. + +v0.6.10 - 2020-05-16 + - Add compile-time and run-time version querying. + - DRMP3_VERSION_MINOR + - DRMP3_VERSION_MAJOR + - DRMP3_VERSION_REVISION + - DRMP3_VERSION_STRING + - drmp3_version() + - drmp3_version_string() + +v0.6.9 - 2020-04-30 + - Change the `pcm` parameter of drmp3dec_decode_frame() to a `const drmp3_uint8*` for consistency with internal APIs. + +v0.6.8 - 2020-04-26 + - Optimizations to decoding when initializing from memory. + +v0.6.7 - 2020-04-25 + - Fix a compilation error with DR_MP3_NO_STDIO + - Optimization to decoding by reducing some data movement. + +v0.6.6 - 2020-04-23 + - Fix a minor bug with the running PCM frame counter. + +v0.6.5 - 2020-04-19 + - Fix compilation error on ARM builds. + +v0.6.4 - 2020-04-19 + - Bring up to date with changes to minimp3. + +v0.6.3 - 2020-04-13 + - Fix some pedantic warnings. + +v0.6.2 - 2020-04-10 + - Fix a crash in drmp3_open_*_and_read_pcm_frames_*() if the output config object is NULL. + +v0.6.1 - 2020-04-05 + - Fix warnings. + +v0.6.0 - 2020-04-04 + - API CHANGE: Remove the pConfig parameter from the following APIs: + - drmp3_init() + - drmp3_init_memory() + - drmp3_init_file() + - Add drmp3_init_file_w() for opening a file from a wchar_t encoded path. + +v0.5.6 - 2020-02-12 + - Bring up to date with minimp3. + +v0.5.5 - 2020-01-29 + - Fix a memory allocation bug in high level s16 decoding APIs. + +v0.5.4 - 2019-12-02 + - Fix a possible null pointer dereference when using custom memory allocators for realloc(). + +v0.5.3 - 2019-11-14 + - Fix typos in documentation. + +v0.5.2 - 2019-11-02 + - Bring up to date with minimp3. + +v0.5.1 - 2019-10-08 + - Fix a warning with GCC. + +v0.5.0 - 2019-10-07 + - API CHANGE: Add support for user defined memory allocation routines. This system allows the program to specify their own memory allocation + routines with a user data pointer for client-specific contextual data. This adds an extra parameter to the end of the following APIs: + - drmp3_init() + - drmp3_init_file() + - drmp3_init_memory() + - drmp3_open_and_read_pcm_frames_f32() + - drmp3_open_and_read_pcm_frames_s16() + - drmp3_open_memory_and_read_pcm_frames_f32() + - drmp3_open_memory_and_read_pcm_frames_s16() + - drmp3_open_file_and_read_pcm_frames_f32() + - drmp3_open_file_and_read_pcm_frames_s16() + - API CHANGE: Renamed the following APIs: + - drmp3_open_and_read_f32() -> drmp3_open_and_read_pcm_frames_f32() + - drmp3_open_and_read_s16() -> drmp3_open_and_read_pcm_frames_s16() + - drmp3_open_memory_and_read_f32() -> drmp3_open_memory_and_read_pcm_frames_f32() + - drmp3_open_memory_and_read_s16() -> drmp3_open_memory_and_read_pcm_frames_s16() + - drmp3_open_file_and_read_f32() -> drmp3_open_file_and_read_pcm_frames_f32() + - drmp3_open_file_and_read_s16() -> drmp3_open_file_and_read_pcm_frames_s16() + +v0.4.7 - 2019-07-28 + - Fix a compiler error. + +v0.4.6 - 2019-06-14 + - Fix a compiler error. + +v0.4.5 - 2019-06-06 + - Bring up to date with minimp3. + +v0.4.4 - 2019-05-06 + - Fixes to the VC6 build. + +v0.4.3 - 2019-05-05 + - Use the channel count and/or sample rate of the first MP3 frame instead of DRMP3_DEFAULT_CHANNELS and + DRMP3_DEFAULT_SAMPLE_RATE when they are set to 0. To use the old behaviour, just set the relevant property to + DRMP3_DEFAULT_CHANNELS or DRMP3_DEFAULT_SAMPLE_RATE. + - Add s16 reading APIs + - drmp3_read_pcm_frames_s16 + - drmp3_open_memory_and_read_pcm_frames_s16 + - drmp3_open_and_read_pcm_frames_s16 + - drmp3_open_file_and_read_pcm_frames_s16 + - Add drmp3_get_mp3_and_pcm_frame_count() to the public header section. + - Add support for C89. + - Change license to choice of public domain or MIT-0. + +v0.4.2 - 2019-02-21 + - Fix a warning. + +v0.4.1 - 2018-12-30 + - Fix a warning. + +v0.4.0 - 2018-12-16 + - API CHANGE: Rename some APIs: + - drmp3_read_f32 -> to drmp3_read_pcm_frames_f32 + - drmp3_seek_to_frame -> drmp3_seek_to_pcm_frame + - drmp3_open_and_decode_f32 -> drmp3_open_and_read_pcm_frames_f32 + - drmp3_open_and_decode_memory_f32 -> drmp3_open_memory_and_read_pcm_frames_f32 + - drmp3_open_and_decode_file_f32 -> drmp3_open_file_and_read_pcm_frames_f32 + - Add drmp3_get_pcm_frame_count(). + - Add drmp3_get_mp3_frame_count(). + - Improve seeking performance. + +v0.3.2 - 2018-09-11 + - Fix a couple of memory leaks. + - Bring up to date with minimp3. + +v0.3.1 - 2018-08-25 + - Fix C++ build. + +v0.3.0 - 2018-08-25 + - Bring up to date with minimp3. This has a minor API change: the "pcm" parameter of drmp3dec_decode_frame() has + been changed from short* to void* because it can now output both s16 and f32 samples, depending on whether or + not the DR_MP3_FLOAT_OUTPUT option is set. + +v0.2.11 - 2018-08-08 + - Fix a bug where the last part of a file is not read. + +v0.2.10 - 2018-08-07 + - Improve 64-bit detection. + +v0.2.9 - 2018-08-05 + - Fix C++ build on older versions of GCC. + - Bring up to date with minimp3. + +v0.2.8 - 2018-08-02 + - Fix compilation errors with older versions of GCC. + +v0.2.7 - 2018-07-13 + - Bring up to date with minimp3. + +v0.2.6 - 2018-07-12 + - Bring up to date with minimp3. + +v0.2.5 - 2018-06-22 + - Bring up to date with minimp3. + +v0.2.4 - 2018-05-12 + - Bring up to date with minimp3. + +v0.2.3 - 2018-04-29 + - Fix TCC build. + +v0.2.2 - 2018-04-28 + - Fix bug when opening a decoder from memory. + +v0.2.1 - 2018-04-27 + - Efficiency improvements when the decoder reaches the end of the stream. + +v0.2 - 2018-04-21 + - Bring up to date with minimp3. + - Start using major.minor.revision versioning. + +v0.1d - 2018-03-30 + - Bring up to date with minimp3. + +v0.1c - 2018-03-11 + - Fix C++ build error. + +v0.1b - 2018-03-07 + - Bring up to date with minimp3. + +v0.1a - 2018-02-28 + - Fix compilation error on GCC/Clang. + - Fix some warnings. + +v0.1 - 2018-02-xx + - Initial versioned release. +*/ + +/* +This software is available as a choice of the following licenses. Choose +whichever you prefer. + +=============================================================================== +ALTERNATIVE 1 - Public Domain (www.unlicense.org) +=============================================================================== +This is free and unencumbered software released into the public domain. + +Anyone is free to copy, modify, publish, use, compile, sell, or distribute this +software, either in source code form or as a compiled binary, for any purpose, +commercial or non-commercial, and by any means. + +In jurisdictions that recognize copyright laws, the author or authors of this +software dedicate any and all copyright interest in the software to the public +domain. We make this dedication for the benefit of the public at large and to +the detriment of our heirs and successors. We intend this dedication to be an +overt act of relinquishment in perpetuity of all present and future rights to +this software under copyright law. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN +ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION +WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + +For more information, please refer to + +=============================================================================== +ALTERNATIVE 2 - MIT No Attribution +=============================================================================== +Copyright 2023 David Reid + +Permission is hereby granted, free of charge, to any person obtaining a copy of +this software and associated documentation files (the "Software"), to deal in +the Software without restriction, including without limitation the rights to +use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies +of the Software, and to permit persons to whom the Software is furnished to do +so. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +SOFTWARE. +*/ + +/* + https://github.com/lieff/minimp3 + To the extent possible under law, the author(s) have dedicated all copyright and related and neighboring rights to this software to the public domain worldwide. + This software is distributed without any warranty. + See . +*/ diff --git a/src/include/dr_libs/dr_wav.h b/src/include/dr_libs/dr_wav.h new file mode 100644 index 000000000..bfe842682 --- /dev/null +++ b/src/include/dr_libs/dr_wav.h @@ -0,0 +1,9209 @@ +/* +WAV audio loader and writer. Choice of public domain or MIT-0. See license statements at the end of this file. +dr_wav - v0.14.6 - TBD + +David Reid - mackron@gmail.com + +GitHub: https://github.com/mackron/dr_libs +*/ + +/* +Introduction +============ +This is a single file library. To use it, do something like the following in one .c file. + + ```c + #define DR_WAV_IMPLEMENTATION + #include "dr_wav.h" + ``` + +You can then #include this file in other parts of the program as you would with any other header file. Do something like the following to read audio data: + + ```c + drwav wav; + if (!drwav_init_file(&wav, "my_song.wav", NULL)) { + // Error opening WAV file. + } + + drwav_int32* pDecodedInterleavedPCMFrames = malloc(wav.totalPCMFrameCount * wav.channels * sizeof(drwav_int32)); + size_t numberOfSamplesActuallyDecoded = drwav_read_pcm_frames_s32(&wav, wav.totalPCMFrameCount, pDecodedInterleavedPCMFrames); + + ... + + drwav_uninit(&wav); + ``` + +If you just want to quickly open and read the audio data in a single operation you can do something like this: + + ```c + unsigned int channels; + unsigned int sampleRate; + drwav_uint64 totalPCMFrameCount; + float* pSampleData = drwav_open_file_and_read_pcm_frames_f32("my_song.wav", &channels, &sampleRate, &totalPCMFrameCount, NULL); + if (pSampleData == NULL) { + // Error opening and reading WAV file. + } + + ... + + drwav_free(pSampleData, NULL); + ``` + +The examples above use versions of the API that convert the audio data to a consistent format (32-bit signed PCM, in this case), but you can still output the +audio data in its internal format (see notes below for supported formats): + + ```c + size_t framesRead = drwav_read_pcm_frames(&wav, wav.totalPCMFrameCount, pDecodedInterleavedPCMFrames); + ``` + +You can also read the raw bytes of audio data, which could be useful if dr_wav does not have native support for a particular data format: + + ```c + size_t bytesRead = drwav_read_raw(&wav, bytesToRead, pRawDataBuffer); + ``` + +dr_wav can also be used to output WAV files. This does not currently support compressed formats. To use this, look at `drwav_init_write()`, +`drwav_init_file_write()`, etc. Use `drwav_write_pcm_frames()` to write samples, or `drwav_write_raw()` to write raw data in the "data" chunk. + + ```c + drwav_data_format format; + format.container = drwav_container_riff; // <-- drwav_container_riff = normal WAV files, drwav_container_w64 = Sony Wave64. + format.format = DR_WAVE_FORMAT_PCM; // <-- Any of the DR_WAVE_FORMAT_* codes. + format.channels = 2; + format.sampleRate = 44100; + format.bitsPerSample = 16; + drwav_init_file_write(&wav, "data/recording.wav", &format, NULL); + + ... + + drwav_uint64 framesWritten = drwav_write_pcm_frames(pWav, frameCount, pSamples); + ``` + +Note that writing to AIFF or RIFX is not supported. + +dr_wav has support for decoding from a number of different encapsulation formats. See below for details. + + +Build Options +============= +#define these options before including this file. + +#define DR_WAV_NO_CONVERSION_API + Disables conversion APIs such as `drwav_read_pcm_frames_f32()` and `drwav_s16_to_f32()`. + +#define DR_WAV_NO_STDIO + Disables APIs that initialize a decoder from a file such as `drwav_init_file()`, `drwav_init_file_write()`, etc. + +#define DR_WAV_NO_WCHAR + Disables all functions ending with `_w`. Use this if your compiler does not provide wchar.h. Not required if DR_WAV_NO_STDIO is also defined. + + +Supported Encapsulations +======================== +- RIFF (Regular WAV) +- RIFX (Big-Endian) +- AIFF (Does not currently support ADPCM) +- RF64 +- W64 + +Note that AIFF and RIFX do not support write mode, nor do they support reading of metadata. + + +Supported Encodings +=================== +- Unsigned 8-bit PCM +- Signed 12-bit PCM +- Signed 16-bit PCM +- Signed 24-bit PCM +- Signed 32-bit PCM +- IEEE 32-bit floating point +- IEEE 64-bit floating point +- A-law and u-law +- Microsoft ADPCM +- IMA ADPCM (DVI, format code 0x11) + +8-bit PCM encodings are always assumed to be unsigned. Signed 8-bit encoding can only be read with `drwav_read_raw()`. + +Note that ADPCM is not currently supported with AIFF. Contributions welcome. + + +Notes +===== +- Samples are always interleaved. +- The default read function does not do any data conversion. Use `drwav_read_pcm_frames_f32()`, `drwav_read_pcm_frames_s32()` and `drwav_read_pcm_frames_s16()` + to read and convert audio data to 32-bit floating point, signed 32-bit integer and signed 16-bit integer samples respectively. +- dr_wav will try to read the WAV file as best it can, even if it's not strictly conformant to the WAV format. +*/ + +#ifndef dr_wav_h +#define dr_wav_h + +#ifdef __cplusplus +extern "C" { +#endif + +#define DRWAV_STRINGIFY(x) #x +#define DRWAV_XSTRINGIFY(x) DRWAV_STRINGIFY(x) + +#define DRWAV_VERSION_MAJOR 0 +#define DRWAV_VERSION_MINOR 14 +#define DRWAV_VERSION_REVISION 6 +#define DRWAV_VERSION_STRING DRWAV_XSTRINGIFY(DRWAV_VERSION_MAJOR) "." DRWAV_XSTRINGIFY(DRWAV_VERSION_MINOR) "." DRWAV_XSTRINGIFY(DRWAV_VERSION_REVISION) + +#include /* For size_t. */ + +/* Sized Types */ +typedef signed char drwav_int8; +typedef unsigned char drwav_uint8; +typedef signed short drwav_int16; +typedef unsigned short drwav_uint16; +typedef signed int drwav_int32; +typedef unsigned int drwav_uint32; +#if defined(_MSC_VER) && !defined(__clang__) + typedef signed __int64 drwav_int64; + typedef unsigned __int64 drwav_uint64; +#else + #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic push + #pragma GCC diagnostic ignored "-Wlong-long" + #if defined(__clang__) + #pragma GCC diagnostic ignored "-Wc++11-long-long" + #endif + #endif + typedef signed long long drwav_int64; + typedef unsigned long long drwav_uint64; + #if defined(__clang__) || (defined(__GNUC__) && (__GNUC__ > 4 || (__GNUC__ == 4 && __GNUC_MINOR__ >= 6))) + #pragma GCC diagnostic pop + #endif +#endif +#if defined(__LP64__) || defined(_WIN64) || (defined(__x86_64__) && !defined(__ILP32__)) || defined(_M_X64) || defined(__ia64) || defined (_M_IA64) || defined(__aarch64__) || defined(_M_ARM64) || defined(_M_ARM64EC) || defined(__powerpc64__) + typedef drwav_uint64 drwav_uintptr; +#else + typedef drwav_uint32 drwav_uintptr; +#endif +typedef drwav_uint8 drwav_bool8; +typedef drwav_uint32 drwav_bool32; +#define DRWAV_TRUE 1 +#define DRWAV_FALSE 0 +/* End Sized Types */ + +/* Decorations */ +#if !defined(DRWAV_API) + #if defined(DRWAV_DLL) + #if defined(_WIN32) + #define DRWAV_DLL_IMPORT __declspec(dllimport) + #define DRWAV_DLL_EXPORT __declspec(dllexport) + #define DRWAV_DLL_PRIVATE static + #else + #if defined(__GNUC__) && __GNUC__ >= 4 + #define DRWAV_DLL_IMPORT __attribute__((visibility("default"))) + #define DRWAV_DLL_EXPORT __attribute__((visibility("default"))) + #define DRWAV_DLL_PRIVATE __attribute__((visibility("hidden"))) + #else + #define DRWAV_DLL_IMPORT + #define DRWAV_DLL_EXPORT + #define DRWAV_DLL_PRIVATE static + #endif + #endif + + #if defined(DR_WAV_IMPLEMENTATION) || defined(DRWAV_IMPLEMENTATION) + #define DRWAV_API DRWAV_DLL_EXPORT + #else + #define DRWAV_API DRWAV_DLL_IMPORT + #endif + #define DRWAV_PRIVATE DRWAV_DLL_PRIVATE + #else + #define DRWAV_API extern + #define DRWAV_PRIVATE static + #endif +#endif +/* End Decorations */ + +/* Result Codes */ +typedef drwav_int32 drwav_result; +#define DRWAV_SUCCESS 0 +#define DRWAV_ERROR -1 /* A generic error. */ +#define DRWAV_INVALID_ARGS -2 +#define DRWAV_INVALID_OPERATION -3 +#define DRWAV_OUT_OF_MEMORY -4 +#define DRWAV_OUT_OF_RANGE -5 +#define DRWAV_ACCESS_DENIED -6 +#define DRWAV_DOES_NOT_EXIST -7 +#define DRWAV_ALREADY_EXISTS -8 +#define DRWAV_TOO_MANY_OPEN_FILES -9 +#define DRWAV_INVALID_FILE -10 +#define DRWAV_TOO_BIG -11 +#define DRWAV_PATH_TOO_LONG -12 +#define DRWAV_NAME_TOO_LONG -13 +#define DRWAV_NOT_DIRECTORY -14 +#define DRWAV_IS_DIRECTORY -15 +#define DRWAV_DIRECTORY_NOT_EMPTY -16 +#define DRWAV_END_OF_FILE -17 +#define DRWAV_NO_SPACE -18 +#define DRWAV_BUSY -19 +#define DRWAV_IO_ERROR -20 +#define DRWAV_INTERRUPT -21 +#define DRWAV_UNAVAILABLE -22 +#define DRWAV_ALREADY_IN_USE -23 +#define DRWAV_BAD_ADDRESS -24 +#define DRWAV_BAD_SEEK -25 +#define DRWAV_BAD_PIPE -26 +#define DRWAV_DEADLOCK -27 +#define DRWAV_TOO_MANY_LINKS -28 +#define DRWAV_NOT_IMPLEMENTED -29 +#define DRWAV_NO_MESSAGE -30 +#define DRWAV_BAD_MESSAGE -31 +#define DRWAV_NO_DATA_AVAILABLE -32 +#define DRWAV_INVALID_DATA -33 +#define DRWAV_TIMEOUT -34 +#define DRWAV_NO_NETWORK -35 +#define DRWAV_NOT_UNIQUE -36 +#define DRWAV_NOT_SOCKET -37 +#define DRWAV_NO_ADDRESS -38 +#define DRWAV_BAD_PROTOCOL -39 +#define DRWAV_PROTOCOL_UNAVAILABLE -40 +#define DRWAV_PROTOCOL_NOT_SUPPORTED -41 +#define DRWAV_PROTOCOL_FAMILY_NOT_SUPPORTED -42 +#define DRWAV_ADDRESS_FAMILY_NOT_SUPPORTED -43 +#define DRWAV_SOCKET_NOT_SUPPORTED -44 +#define DRWAV_CONNECTION_RESET -45 +#define DRWAV_ALREADY_CONNECTED -46 +#define DRWAV_NOT_CONNECTED -47 +#define DRWAV_CONNECTION_REFUSED -48 +#define DRWAV_NO_HOST -49 +#define DRWAV_IN_PROGRESS -50 +#define DRWAV_CANCELLED -51 +#define DRWAV_MEMORY_ALREADY_MAPPED -52 +#define DRWAV_AT_END -53 +/* End Result Codes */ + +/* Common data formats. */ +#define DR_WAVE_FORMAT_PCM 0x1 +#define DR_WAVE_FORMAT_ADPCM 0x2 +#define DR_WAVE_FORMAT_IEEE_FLOAT 0x3 +#define DR_WAVE_FORMAT_ALAW 0x6 +#define DR_WAVE_FORMAT_MULAW 0x7 +#define DR_WAVE_FORMAT_DTS 0x8 +#define DR_WAVE_FORMAT_DVI_ADPCM 0x11 +#define DR_WAVE_FORMAT_EXTENSIBLE 0xFFFE + +/* Flags to pass into drwav_init_ex(), etc. */ +#define DRWAV_SEQUENTIAL 0x00000001 +#define DRWAV_WITH_METADATA 0x00000002 + +DRWAV_API void drwav_version(drwav_uint32* pMajor, drwav_uint32* pMinor, drwav_uint32* pRevision); +DRWAV_API const char* drwav_version_string(void); + +/* Allocation Callbacks */ +typedef struct +{ + void* pUserData; + void* (* onMalloc)(size_t sz, void* pUserData); + void* (* onRealloc)(void* p, size_t sz, void* pUserData); + void (* onFree)(void* p, void* pUserData); +} drwav_allocation_callbacks; +/* End Allocation Callbacks */ + +typedef enum +{ + DRWAV_SEEK_SET, + DRWAV_SEEK_CUR, + DRWAV_SEEK_END +} drwav_seek_origin; + +typedef enum +{ + drwav_container_riff, + drwav_container_rifx, + drwav_container_w64, + drwav_container_rf64, + drwav_container_aiff +} drwav_container; + +typedef struct +{ + union + { + drwav_uint8 fourcc[4]; + drwav_uint8 guid[16]; + } id; + + /* The size in bytes of the chunk. */ + drwav_uint64 sizeInBytes; + + /* + RIFF = 2 byte alignment. + W64 = 8 byte alignment. + */ + unsigned int paddingSize; +} drwav_chunk_header; + +typedef struct +{ + /* + The format tag exactly as specified in the wave file's "fmt" chunk. This can be used by applications + that require support for data formats not natively supported by dr_wav. + */ + drwav_uint16 formatTag; + + /* The number of channels making up the audio data. When this is set to 1 it is mono, 2 is stereo, etc. */ + drwav_uint16 channels; + + /* The sample rate. Usually set to something like 44100. */ + drwav_uint32 sampleRate; + + /* Average bytes per second. You probably don't need this, but it's left here for informational purposes. */ + drwav_uint32 avgBytesPerSec; + + /* Block align. This is equal to the number of channels * bytes per sample. */ + drwav_uint16 blockAlign; + + /* Bits per sample. */ + drwav_uint16 bitsPerSample; + + /* The size of the extended data. Only used internally for validation, but left here for informational purposes. */ + drwav_uint16 extendedSize; + + /* + The number of valid bits per sample. When is equal to WAVE_FORMAT_EXTENSIBLE, + is always rounded up to the nearest multiple of 8. This variable contains information about exactly how + many bits are valid per sample. Mainly used for informational purposes. + */ + drwav_uint16 validBitsPerSample; + + /* The channel mask. Not used at the moment. */ + drwav_uint32 channelMask; + + /* The sub-format, exactly as specified by the wave file. */ + drwav_uint8 subFormat[16]; +} drwav_fmt; + +DRWAV_API drwav_uint16 drwav_fmt_get_format(const drwav_fmt* pFMT); + + +/* +Callback for when data is read. Return value is the number of bytes actually read. + +pUserData [in] The user data that was passed to drwav_init() and family. +pBufferOut [out] The output buffer. +bytesToRead [in] The number of bytes to read. + +Returns the number of bytes actually read. + +A return value of less than bytesToRead indicates the end of the stream. Do _not_ return from this callback until +either the entire bytesToRead is filled or you have reached the end of the stream. +*/ +typedef size_t (* drwav_read_proc)(void* pUserData, void* pBufferOut, size_t bytesToRead); + +/* +Callback for when data is written. Returns value is the number of bytes actually written. + +pUserData [in] The user data that was passed to drwav_init_write() and family. +pData [out] A pointer to the data to write. +bytesToWrite [in] The number of bytes to write. + +Returns the number of bytes actually written. + +If the return value differs from bytesToWrite, it indicates an error. +*/ +typedef size_t (* drwav_write_proc)(void* pUserData, const void* pData, size_t bytesToWrite); + +/* +Callback for when data needs to be seeked. + +pUserData [in] The user data that was passed to drwav_init() and family. +offset [in] The number of bytes to move, relative to the origin. Will never be negative. +origin [in] The origin of the seek - the current position or the start of the stream. + +Returns whether or not the seek was successful. + +Whether or not it is relative to the beginning or current position is determined by the "origin" parameter which will be either DRWAV_SEEK_SET or +DRWAV_SEEK_CUR. +*/ +typedef drwav_bool32 (* drwav_seek_proc)(void* pUserData, int offset, drwav_seek_origin origin); + +/* +Callback for when the current position in the stream needs to be retrieved. + +pUserData [in] The user data that was passed to drwav_init() and family. +pCursor [out] A pointer to a variable to receive the current position in the stream. + +Returns whether or not the operation was successful. +*/ +typedef drwav_bool32 (* drwav_tell_proc)(void* pUserData, drwav_int64* pCursor); + +/* +Callback for when drwav_init_ex() finds a chunk. + +pChunkUserData [in] The user data that was passed to the pChunkUserData parameter of drwav_init_ex() and family. +onRead [in] A pointer to the function to call when reading. +onSeek [in] A pointer to the function to call when seeking. +pReadSeekUserData [in] The user data that was passed to the pReadSeekUserData parameter of drwav_init_ex() and family. +pChunkHeader [in] A pointer to an object containing basic header information about the chunk. Use this to identify the chunk. +container [in] Whether or not the WAV file is a RIFF or Wave64 container. If you're unsure of the difference, assume RIFF. +pFMT [in] A pointer to the object containing the contents of the "fmt" chunk. + +Returns the number of bytes read + seeked. + +To read data from the chunk, call onRead(), passing in pReadSeekUserData as the first parameter. Do the same for seeking with onSeek(). The return value must +be the total number of bytes you have read _plus_ seeked. + +Use the `container` argument to discriminate the fields in `pChunkHeader->id`. If the container is `drwav_container_riff` or `drwav_container_rf64` you should +use `id.fourcc`, otherwise you should use `id.guid`. + +The `pFMT` parameter can be used to determine the data format of the wave file. Use `drwav_fmt_get_format()` to get the sample format, which will be one of the +`DR_WAVE_FORMAT_*` identifiers. + +The read pointer will be sitting on the first byte after the chunk's header. You must not attempt to read beyond the boundary of the chunk. +*/ +typedef drwav_uint64 (* drwav_chunk_proc)(void* pChunkUserData, drwav_read_proc onRead, drwav_seek_proc onSeek, void* pReadSeekUserData, const drwav_chunk_header* pChunkHeader, drwav_container container, const drwav_fmt* pFMT); + + +/* Structure for internal use. Only used for loaders opened with drwav_init_memory(). */ +typedef struct +{ + const drwav_uint8* data; + size_t dataSize; + size_t currentReadPos; +} drwav__memory_stream; + +/* Structure for internal use. Only used for writers opened with drwav_init_memory_write(). */ +typedef struct +{ + void** ppData; + size_t* pDataSize; + size_t dataSize; + size_t dataCapacity; + size_t currentWritePos; +} drwav__memory_stream_write; + +typedef struct +{ + drwav_container container; /* RIFF, W64. */ + drwav_uint32 format; /* DR_WAVE_FORMAT_* */ + drwav_uint32 channels; + drwav_uint32 sampleRate; + drwav_uint32 bitsPerSample; +} drwav_data_format; + +typedef enum +{ + drwav_metadata_type_none = 0, + + /* + Unknown simply means a chunk that drwav does not handle specifically. You can still ask to + receive these chunks as metadata objects. It is then up to you to interpret the chunk's data. + You can also write unknown metadata to a wav file. Be careful writing unknown chunks if you + have also edited the audio data. The unknown chunks could represent offsets/sizes that no + longer correctly correspond to the audio data. + */ + drwav_metadata_type_unknown = 1 << 0, + + /* Only 1 of each of these metadata items are allowed in a wav file. */ + drwav_metadata_type_smpl = 1 << 1, + drwav_metadata_type_inst = 1 << 2, + drwav_metadata_type_cue = 1 << 3, + drwav_metadata_type_acid = 1 << 4, + drwav_metadata_type_bext = 1 << 5, + + /* + Wav files often have a LIST chunk. This is a chunk that contains a set of subchunks. For this + higher-level metadata API, we don't make a distinction between a regular chunk and a LIST + subchunk. Instead, they are all just 'metadata' items. + + There can be multiple of these metadata items in a wav file. + */ + drwav_metadata_type_list_label = 1 << 6, + drwav_metadata_type_list_note = 1 << 7, + drwav_metadata_type_list_labelled_cue_region = 1 << 8, + + drwav_metadata_type_list_info_software = 1 << 9, + drwav_metadata_type_list_info_copyright = 1 << 10, + drwav_metadata_type_list_info_title = 1 << 11, + drwav_metadata_type_list_info_artist = 1 << 12, + drwav_metadata_type_list_info_comment = 1 << 13, + drwav_metadata_type_list_info_date = 1 << 14, + drwav_metadata_type_list_info_genre = 1 << 15, + drwav_metadata_type_list_info_album = 1 << 16, + drwav_metadata_type_list_info_tracknumber = 1 << 17, + drwav_metadata_type_list_info_location = 1 << 18, + drwav_metadata_type_list_info_organization = 1 << 19, + drwav_metadata_type_list_info_keywords = 1 << 20, + drwav_metadata_type_list_info_medium = 1 << 21, + drwav_metadata_type_list_info_description = 1 << 22, + + /* Other type constants for convenience. */ + drwav_metadata_type_list_all_info_strings = drwav_metadata_type_list_info_software + | drwav_metadata_type_list_info_copyright + | drwav_metadata_type_list_info_title + | drwav_metadata_type_list_info_artist + | drwav_metadata_type_list_info_comment + | drwav_metadata_type_list_info_date + | drwav_metadata_type_list_info_genre + | drwav_metadata_type_list_info_album + | drwav_metadata_type_list_info_tracknumber + | drwav_metadata_type_list_info_location + | drwav_metadata_type_list_info_organization + | drwav_metadata_type_list_info_keywords + | drwav_metadata_type_list_info_medium + | drwav_metadata_type_list_info_description, + + drwav_metadata_type_list_all_adtl = drwav_metadata_type_list_label + | drwav_metadata_type_list_note + | drwav_metadata_type_list_labelled_cue_region, + + drwav_metadata_type_all = -2, /*0xFFFFFFFF & ~drwav_metadata_type_unknown,*/ + drwav_metadata_type_all_including_unknown = -1 /*0xFFFFFFFF,*/ +} drwav_metadata_type; + +/* +Sampler Metadata + +The sampler chunk contains information about how a sound should be played in the context of a whole +audio production, and when used in a sampler. See https://en.wikipedia.org/wiki/Sample-based_synthesis. +*/ +typedef enum +{ + drwav_smpl_loop_type_forward = 0, + drwav_smpl_loop_type_pingpong = 1, + drwav_smpl_loop_type_backward = 2 +} drwav_smpl_loop_type; + +typedef struct +{ + /* The ID of the associated cue point, see drwav_cue and drwav_cue_point. As with all cue point IDs, this can correspond to a label chunk to give this loop a name, see drwav_list_label_or_note. */ + drwav_uint32 cuePointId; + + /* See drwav_smpl_loop_type. */ + drwav_uint32 type; + + /* The offset of the first sample to be played in the loop. */ + drwav_uint32 firstSampleOffset; + + /* The offset into the audio data of the last sample to be played in the loop. */ + drwav_uint32 lastSampleOffset; + + /* A value to represent that playback should occur at a point between samples. This value ranges from 0 to UINT32_MAX. Where a value of 0 means no fraction, and a value of (UINT32_MAX / 2) would mean half a sample. */ + drwav_uint32 sampleFraction; + + /* Number of times to play the loop. 0 means loop infinitely. */ + drwav_uint32 playCount; +} drwav_smpl_loop; + +typedef struct +{ + /* IDs for a particular MIDI manufacturer. 0 if not used. */ + drwav_uint32 manufacturerId; + drwav_uint32 productId; + + /* The period of 1 sample in nanoseconds. */ + drwav_uint32 samplePeriodNanoseconds; + + /* The MIDI root note of this file. 0 to 127. */ + drwav_uint32 midiUnityNote; + + /* The fraction of a semitone up from the given MIDI note. This is a value from 0 to UINT32_MAX, where 0 means no change and (UINT32_MAX / 2) is half a semitone (AKA 50 cents). */ + drwav_uint32 midiPitchFraction; + + /* Data relating to SMPTE standards which are used for syncing audio and video. 0 if not used. */ + drwav_uint32 smpteFormat; + drwav_uint32 smpteOffset; + + /* drwav_smpl_loop loops. */ + drwav_uint32 sampleLoopCount; + + /* Optional sampler-specific data. */ + drwav_uint32 samplerSpecificDataSizeInBytes; + + drwav_smpl_loop* pLoops; + drwav_uint8* pSamplerSpecificData; +} drwav_smpl; + +/* +Instrument Metadata + +The inst metadata contains data about how a sound should be played as part of an instrument. This +commonly read by samplers. See https://en.wikipedia.org/wiki/Sample-based_synthesis. +*/ +typedef struct +{ + drwav_int8 midiUnityNote; /* The root note of the audio as a MIDI note number. 0 to 127. */ + drwav_int8 fineTuneCents; /* -50 to +50 */ + drwav_int8 gainDecibels; /* -64 to +64 */ + drwav_int8 lowNote; /* 0 to 127 */ + drwav_int8 highNote; /* 0 to 127 */ + drwav_int8 lowVelocity; /* 1 to 127 */ + drwav_int8 highVelocity; /* 1 to 127 */ +} drwav_inst; + +/* +Cue Metadata + +Cue points are markers at specific points in the audio. They often come with an associated piece of +drwav_list_label_or_note metadata which contains the text for the marker. +*/ +typedef struct +{ + /* Unique identification value. */ + drwav_uint32 id; + + /* Set to 0. This is only relevant if there is a 'playlist' chunk - which is not supported by dr_wav. */ + drwav_uint32 playOrderPosition; + + /* Should always be "data". This represents the fourcc value of the chunk that this cue point corresponds to. dr_wav only supports a single data chunk so this should always be "data". */ + drwav_uint8 dataChunkId[4]; + + /* Set to 0. This is only relevant if there is a wave list chunk. dr_wav, like lots of readers/writers, do not support this. */ + drwav_uint32 chunkStart; + + /* Set to 0 for uncompressed formats. Else the last byte in compressed wave data where decompression can begin to find the value of the corresponding sample value. */ + drwav_uint32 blockStart; + + /* For uncompressed formats this is the offset of the cue point into the audio data. For compressed formats this is relative to the block specified with blockStart. */ + drwav_uint32 sampleOffset; +} drwav_cue_point; + +typedef struct +{ + drwav_uint32 cuePointCount; + drwav_cue_point *pCuePoints; +} drwav_cue; + +/* +Acid Metadata + +This chunk contains some information about the time signature and the tempo of the audio. +*/ +typedef enum +{ + drwav_acid_flag_one_shot = 1, /* If this is not set, then it is a loop instead of a one-shot. */ + drwav_acid_flag_root_note_set = 2, + drwav_acid_flag_stretch = 4, + drwav_acid_flag_disk_based = 8, + drwav_acid_flag_acidizer = 16 /* Not sure what this means. */ +} drwav_acid_flag; + +typedef struct +{ + /* A bit-field, see drwav_acid_flag. */ + drwav_uint32 flags; + + /* Valid if flags contains drwav_acid_flag_root_note_set. It represents the MIDI root note the file - a value from 0 to 127. */ + drwav_uint16 midiUnityNote; + + /* Reserved values that should probably be ignored. reserved1 seems to often be 128 and reserved2 is 0. */ + drwav_uint16 reserved1; + float reserved2; + + /* Number of beats. */ + drwav_uint32 numBeats; + + /* The time signature of the audio. */ + drwav_uint16 meterDenominator; + drwav_uint16 meterNumerator; + + /* Beats per minute of the track. Setting a value of 0 suggests that there is no tempo. */ + float tempo; +} drwav_acid; + +/* +Cue Label or Note metadata + +These are 2 different types of metadata, but they have the exact same format. Labels tend to be the +more common and represent a short name for a cue point. Notes might be used to represent a longer +comment. +*/ +typedef struct +{ + /* The ID of a cue point that this label or note corresponds to. */ + drwav_uint32 cuePointId; + + /* Size of the string not including any null terminator. */ + drwav_uint32 stringLength; + + /* The string. The *init_with_metadata functions null terminate this for convenience. */ + char* pString; +} drwav_list_label_or_note; + +/* +BEXT metadata, also known as Broadcast Wave Format (BWF) + +This metadata adds some extra description to an audio file. You must check the version field to +determine if the UMID or the loudness fields are valid. +*/ +typedef struct +{ + /* + These top 3 fields, and the umid field are actually defined in the standard as a statically + sized buffers. In order to reduce the size of this struct (and therefore the union in the + metadata struct), we instead store these as pointers. + */ + char* pDescription; /* Can be NULL or a null-terminated string, must be <= 256 characters. */ + char* pOriginatorName; /* Can be NULL or a null-terminated string, must be <= 32 characters. */ + char* pOriginatorReference; /* Can be NULL or a null-terminated string, must be <= 32 characters. */ + char pOriginationDate[10]; /* ASCII "yyyy:mm:dd". */ + char pOriginationTime[8]; /* ASCII "hh:mm:ss". */ + drwav_uint64 timeReference; /* First sample count since midnight. */ + drwav_uint16 version; /* Version of the BWF, check this to see if the fields below are valid. */ + + /* + Unrestricted ASCII characters containing a collection of strings terminated by CR/LF. Each + string shall contain a description of a coding process applied to the audio data. + */ + char* pCodingHistory; + drwav_uint32 codingHistorySize; + + /* Fields below this point are only valid if the version is 1 or above. */ + drwav_uint8* pUMID; /* Exactly 64 bytes of SMPTE UMID */ + + /* Fields below this point are only valid if the version is 2 or above. */ + drwav_uint16 loudnessValue; /* Integrated Loudness Value of the file in LUFS (multiplied by 100). */ + drwav_uint16 loudnessRange; /* Loudness Range of the file in LU (multiplied by 100). */ + drwav_uint16 maxTruePeakLevel; /* Maximum True Peak Level of the file expressed as dBTP (multiplied by 100). */ + drwav_uint16 maxMomentaryLoudness; /* Highest value of the Momentary Loudness Level of the file in LUFS (multiplied by 100). */ + drwav_uint16 maxShortTermLoudness; /* Highest value of the Short-Term Loudness Level of the file in LUFS (multiplied by 100). */ +} drwav_bext; + +/* +Info Text Metadata + +There a many different types of information text that can be saved in this format. This is where +things like the album name, the artists, the year it was produced, etc are saved. See +drwav_metadata_type for the full list of types that dr_wav supports. +*/ +typedef struct +{ + /* Size of the string not including any null terminator. */ + drwav_uint32 stringLength; + + /* The string. The *init_with_metadata functions null terminate this for convenience. */ + char* pString; +} drwav_list_info_text; + +/* +Labelled Cue Region Metadata + +The labelled cue region metadata is used to associate some region of audio with text. The region +starts at a cue point, and extends for the given number of samples. +*/ +typedef struct +{ + /* The ID of a cue point that this object corresponds to. */ + drwav_uint32 cuePointId; + + /* The number of samples from the cue point forwards that should be considered this region */ + drwav_uint32 sampleLength; + + /* Four characters used to say what the purpose of this region is. */ + drwav_uint8 purposeId[4]; + + /* Unsure of the exact meanings of these. It appears to be acceptable to set them all to 0. */ + drwav_uint16 country; + drwav_uint16 language; + drwav_uint16 dialect; + drwav_uint16 codePage; + + /* Size of the string not including any null terminator. */ + drwav_uint32 stringLength; + + /* The string. The *init_with_metadata functions null terminate this for convenience. */ + char* pString; +} drwav_list_labelled_cue_region; + +/* +Unknown Metadata + +This chunk just represents a type of chunk that dr_wav does not understand. + +Unknown metadata has a location attached to it. This is because wav files can have a LIST chunk +that contains subchunks. These LIST chunks can be one of two types. An adtl list, or an INFO +list. This enum is used to specify the location of a chunk that dr_wav currently doesn't support. +*/ +typedef enum +{ + drwav_metadata_location_invalid, + drwav_metadata_location_top_level, + drwav_metadata_location_inside_info_list, + drwav_metadata_location_inside_adtl_list +} drwav_metadata_location; + +typedef struct +{ + drwav_uint8 id[4]; + drwav_metadata_location chunkLocation; + drwav_uint32 dataSizeInBytes; + drwav_uint8* pData; +} drwav_unknown_metadata; + +/* +Metadata is saved as a union of all the supported types. +*/ +typedef struct +{ + /* Determines which item in the union is valid. */ + drwav_metadata_type type; + + union + { + drwav_cue cue; + drwav_smpl smpl; + drwav_acid acid; + drwav_inst inst; + drwav_bext bext; + drwav_list_label_or_note labelOrNote; /* List label or list note. */ + drwav_list_labelled_cue_region labelledCueRegion; + drwav_list_info_text infoText; /* Any of the list info types. */ + drwav_unknown_metadata unknown; + } data; +} drwav_metadata; + +typedef struct +{ + /* A pointer to the function to call when more data is needed. */ + drwav_read_proc onRead; + + /* A pointer to the function to call when data needs to be written. Only used when the drwav object is opened in write mode. */ + drwav_write_proc onWrite; + + /* A pointer to the function to call when the wav file needs to be seeked. */ + drwav_seek_proc onSeek; + + /* A pointer to the function to call when the position of the stream needs to be retrieved. */ + drwav_tell_proc onTell; + + /* The user data to pass to callbacks. */ + void* pUserData; + + /* Allocation callbacks. */ + drwav_allocation_callbacks allocationCallbacks; + + + /* Whether or not the WAV file is formatted as a standard RIFF file or W64. */ + drwav_container container; + + + /* Structure containing format information exactly as specified by the wav file. */ + drwav_fmt fmt; + + /* The sample rate. Will be set to something like 44100. */ + drwav_uint32 sampleRate; + + /* The number of channels. This will be set to 1 for monaural streams, 2 for stereo, etc. */ + drwav_uint16 channels; + + /* The bits per sample. Will be set to something like 16, 24, etc. */ + drwav_uint16 bitsPerSample; + + /* Equal to fmt.formatTag, or the value specified by fmt.subFormat if fmt.formatTag is equal to 65534 (WAVE_FORMAT_EXTENSIBLE). */ + drwav_uint16 translatedFormatTag; + + /* The total number of PCM frames making up the audio data. */ + drwav_uint64 totalPCMFrameCount; + + + /* The size in bytes of the data chunk. */ + drwav_uint64 dataChunkDataSize; + + /* The position in the stream of the first data byte of the data chunk. This is used for seeking. */ + drwav_uint64 dataChunkDataPos; + + /* The number of bytes remaining in the data chunk. */ + drwav_uint64 bytesRemaining; + + /* The current read position in PCM frames. */ + drwav_uint64 readCursorInPCMFrames; + + + /* + Only used in sequential write mode. Keeps track of the desired size of the "data" chunk at the point of initialization time. Always + set to 0 for non-sequential writes and when the drwav object is opened in read mode. Used for validation. + */ + drwav_uint64 dataChunkDataSizeTargetWrite; + + /* Keeps track of whether or not the wav writer was initialized in sequential mode. */ + drwav_bool32 isSequentialWrite; + + + /* A array of metadata. This is valid after the *init_with_metadata call returns. It will be valid until drwav_uninit() is called. You can take ownership of this data with drwav_take_ownership_of_metadata(). */ + drwav_metadata* pMetadata; + drwav_uint32 metadataCount; + + + /* A hack to avoid a DRWAV_MALLOC() when opening a decoder with drwav_init_memory(). */ + drwav__memory_stream memoryStream; + drwav__memory_stream_write memoryStreamWrite; + + + /* Microsoft ADPCM specific data. */ + struct + { + drwav_uint32 bytesRemainingInBlock; + drwav_uint16 predictor[2]; + drwav_int32 delta[2]; + drwav_int32 cachedFrames[4]; /* Samples are stored in this cache during decoding. */ + drwav_uint32 cachedFrameCount; + drwav_int32 prevFrames[2][2]; /* The previous 2 samples for each channel (2 channels at most). */ + } msadpcm; + + /* IMA ADPCM specific data. */ + struct + { + drwav_uint32 bytesRemainingInBlock; + drwav_int32 predictor[2]; + drwav_int32 stepIndex[2]; + drwav_int32 cachedFrames[16]; /* Samples are stored in this cache during decoding. */ + drwav_uint32 cachedFrameCount; + } ima; + + /* AIFF specific data. */ + struct + { + drwav_bool8 isLE; /* Will be set to true if the audio data is little-endian encoded. */ + drwav_bool8 isUnsigned; /* Only used for 8-bit samples. When set to true, will be treated as unsigned. */ + } aiff; +} drwav; + + +/* +Initializes a pre-allocated drwav object for reading. + +pWav [out] A pointer to the drwav object being initialized. +onRead [in] The function to call when data needs to be read from the client. +onSeek [in] The function to call when the read position of the client data needs to move. +onChunk [in, optional] The function to call when a chunk is enumerated at initialized time. +pUserData, pReadSeekUserData [in, optional] A pointer to application defined data that will be passed to onRead and onSeek. +pChunkUserData [in, optional] A pointer to application defined data that will be passed to onChunk. +flags [in, optional] A set of flags for controlling how things are loaded. + +Returns true if successful; false otherwise. + +Close the loader with drwav_uninit(). + +This is the lowest level function for initializing a WAV file. You can also use drwav_init_file() and drwav_init_memory() +to open the stream from a file or from a block of memory respectively. + +Possible values for flags: + DRWAV_SEQUENTIAL: Never perform a backwards seek while loading. This disables the chunk callback and will cause this function + to return as soon as the data chunk is found. Any chunks after the data chunk will be ignored. + +drwav_init() is equivalent to "drwav_init_ex(pWav, onRead, onSeek, NULL, pUserData, NULL, 0);". + +The onChunk callback is not called for the WAVE or FMT chunks. The contents of the FMT chunk can be read from pWav->fmt +after the function returns. + +See also: drwav_init_file(), drwav_init_memory(), drwav_uninit() +*/ +DRWAV_API drwav_bool32 drwav_init(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_ex(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, drwav_chunk_proc onChunk, void* pReadSeekTellUserData, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_with_metadata(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); + +/* +Initializes a pre-allocated drwav object for writing. + +onWrite [in] The function to call when data needs to be written. +onSeek [in] The function to call when the write position needs to move. +pUserData [in, optional] A pointer to application defined data that will be passed to onWrite and onSeek. +metadata, numMetadata [in, optional] An array of metadata objects that should be written to the file. The array is not edited. You are responsible for this metadata memory and it must maintain valid until drwav_uninit() is called. + +Returns true if successful; false otherwise. + +Close the writer with drwav_uninit(). + +This is the lowest level function for initializing a WAV file. You can also use drwav_init_file_write() and drwav_init_memory_write() +to open the stream from a file or from a block of memory respectively. + +If the total sample count is known, you can use drwav_init_write_sequential(). This avoids the need for dr_wav to perform +a post-processing step for storing the total sample count and the size of the data chunk which requires a backwards seek. + +See also: drwav_init_file_write(), drwav_init_memory_write(), drwav_uninit() +*/ +DRWAV_API drwav_bool32 drwav_init_write(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_write_sequential(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_write_sequential_pcm_frames(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_write_with_metadata(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks, drwav_metadata* pMetadata, drwav_uint32 metadataCount); + +/* +Utility function to determine the target size of the entire data to be written (including all headers and chunks). + +Returns the target size in bytes. + +The metadata argument can be NULL meaning no metadata exists. + +Useful if the application needs to know the size to allocate. + +Only writing to the RIFF chunk and one data chunk is currently supported. + +See also: drwav_init_write(), drwav_init_file_write(), drwav_init_memory_write() +*/ +DRWAV_API drwav_uint64 drwav_target_write_size_bytes(const drwav_data_format* pFormat, drwav_uint64 totalFrameCount, drwav_metadata* pMetadata, drwav_uint32 metadataCount); + +/* +Take ownership of the metadata objects that were allocated via one of the init_with_metadata() function calls. The init_with_metdata functions perform a single heap allocation for this metadata. + +Useful if you want the data to persist beyond the lifetime of the drwav object. + +You must free the data returned from this function using drwav_free(). +*/ +DRWAV_API drwav_metadata* drwav_take_ownership_of_metadata(drwav* pWav); + +/* +Uninitializes the given drwav object. + +Use this only for objects initialized with drwav_init*() functions (drwav_init(), drwav_init_ex(), drwav_init_write(), drwav_init_write_sequential()). +*/ +DRWAV_API drwav_result drwav_uninit(drwav* pWav); + + +/* +Reads raw audio data. + +This is the lowest level function for reading audio data. It simply reads the given number of +bytes of the raw internal sample data. + +Consider using drwav_read_pcm_frames_s16(), drwav_read_pcm_frames_s32() or drwav_read_pcm_frames_f32() for +reading sample data in a consistent format. + +pBufferOut can be NULL in which case a seek will be performed. + +Returns the number of bytes actually read. +*/ +DRWAV_API size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOut); + +/* +Reads up to the specified number of PCM frames from the WAV file. + +The output data will be in the file's internal format, converted to native-endian byte order. Use +drwav_read_pcm_frames_s16/f32/s32() to read data in a specific format. + +If the return value is less than it means the end of the file has been reached or +you have requested more PCM frames than can possibly fit in the output buffer. + +This function will only work when sample data is of a fixed size and uncompressed. If you are +using a compressed format consider using drwav_read_raw() or drwav_read_pcm_frames_s16/s32/f32(). + +pBufferOut can be NULL in which case a seek will be performed. +*/ +DRWAV_API drwav_uint64 drwav_read_pcm_frames(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_le(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_be(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut); + +/* +Seeks to the given PCM frame. + +Returns true if successful; false otherwise. +*/ +DRWAV_API drwav_bool32 drwav_seek_to_pcm_frame(drwav* pWav, drwav_uint64 targetFrameIndex); + +/* +Retrieves the current read position in pcm frames. +*/ +DRWAV_API drwav_result drwav_get_cursor_in_pcm_frames(drwav* pWav, drwav_uint64* pCursor); + +/* +Retrieves the length of the file. +*/ +DRWAV_API drwav_result drwav_get_length_in_pcm_frames(drwav* pWav, drwav_uint64* pLength); + + +/* +Writes raw audio data. + +Returns the number of bytes actually written. If this differs from bytesToWrite, it indicates an error. +*/ +DRWAV_API size_t drwav_write_raw(drwav* pWav, size_t bytesToWrite, const void* pData); + +/* +Writes PCM frames. + +Returns the number of PCM frames written. + +Input samples need to be in native-endian byte order. On big-endian architectures the input data will be converted to +little-endian. Use drwav_write_raw() to write raw audio data without performing any conversion. +*/ +DRWAV_API drwav_uint64 drwav_write_pcm_frames(drwav* pWav, drwav_uint64 framesToWrite, const void* pData); +DRWAV_API drwav_uint64 drwav_write_pcm_frames_le(drwav* pWav, drwav_uint64 framesToWrite, const void* pData); +DRWAV_API drwav_uint64 drwav_write_pcm_frames_be(drwav* pWav, drwav_uint64 framesToWrite, const void* pData); + +/* Conversion Utilities */ +#ifndef DR_WAV_NO_CONVERSION_API + +/* +Reads a chunk of audio data and converts it to signed 16-bit PCM samples. + +pBufferOut can be NULL in which case a seek will be performed. + +Returns the number of PCM frames actually read. + +If the return value is less than it means the end of the file has been reached. +*/ +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16le(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16be(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut); + +/* Low-level function for converting unsigned 8-bit PCM samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_u8_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 24-bit PCM samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_s24_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 32-bit PCM samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_s32_to_s16(drwav_int16* pOut, const drwav_int32* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 32-bit floating point samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_f32_to_s16(drwav_int16* pOut, const float* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 64-bit floating point samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_f64_to_s16(drwav_int16* pOut, const double* pIn, size_t sampleCount); + +/* Low-level function for converting A-law samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_alaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting u-law samples to signed 16-bit PCM samples. */ +DRWAV_API void drwav_mulaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount); + + +/* +Reads a chunk of audio data and converts it to IEEE 32-bit floating point samples. + +pBufferOut can be NULL in which case a seek will be performed. + +Returns the number of PCM frames actually read. + +If the return value is less than it means the end of the file has been reached. +*/ +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32le(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32be(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut); + +/* Low-level function for converting unsigned 8-bit PCM samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_u8_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 16-bit PCM samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_s16_to_f32(float* pOut, const drwav_int16* pIn, size_t sampleCount); + +/* Low-level function for converting signed 24-bit PCM samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_s24_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 32-bit PCM samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_s32_to_f32(float* pOut, const drwav_int32* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 64-bit floating point samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_f64_to_f32(float* pOut, const double* pIn, size_t sampleCount); + +/* Low-level function for converting A-law samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_alaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting u-law samples to IEEE 32-bit floating point samples. */ +DRWAV_API void drwav_mulaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount); + + +/* +Reads a chunk of audio data and converts it to signed 32-bit PCM samples. + +pBufferOut can be NULL in which case a seek will be performed. + +Returns the number of PCM frames actually read. + +If the return value is less than it means the end of the file has been reached. +*/ +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32le(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut); +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32be(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut); + +/* Low-level function for converting unsigned 8-bit PCM samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_u8_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting signed 16-bit PCM samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_s16_to_s32(drwav_int32* pOut, const drwav_int16* pIn, size_t sampleCount); + +/* Low-level function for converting signed 24-bit PCM samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_s24_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 32-bit floating point samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_f32_to_s32(drwav_int32* pOut, const float* pIn, size_t sampleCount); + +/* Low-level function for converting IEEE 64-bit floating point samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_f64_to_s32(drwav_int32* pOut, const double* pIn, size_t sampleCount); + +/* Low-level function for converting A-law samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_alaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +/* Low-level function for converting u-law samples to signed 32-bit PCM samples. */ +DRWAV_API void drwav_mulaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount); + +#endif /* DR_WAV_NO_CONVERSION_API */ + + +/* High-Level Convenience Helpers */ + +#ifndef DR_WAV_NO_STDIO +/* +Helper for initializing a wave file for reading using stdio. + +This holds the internal FILE object until drwav_uninit() is called. Keep this in mind if you're caching drwav +objects because the operating system may restrict the number of file handles an application can have open at +any given time. +*/ +DRWAV_API drwav_bool32 drwav_init_file(drwav* pWav, const char* filename, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_ex(drwav* pWav, const char* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_w(drwav* pWav, const wchar_t* filename, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_ex_w(drwav* pWav, const wchar_t* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_with_metadata(drwav* pWav, const char* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_with_metadata_w(drwav* pWav, const wchar_t* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); + + +/* +Helper for initializing a wave file for writing using stdio. + +This holds the internal FILE object until drwav_uninit() is called. Keep this in mind if you're caching drwav +objects because the operating system may restrict the number of file handles an application can have open at +any given time. +*/ +DRWAV_API drwav_bool32 drwav_init_file_write(drwav* pWav, const char* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_write_sequential(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_write_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks); +#endif /* DR_WAV_NO_STDIO */ + +/* +Helper for initializing a loader from a pre-allocated memory buffer. + +This does not create a copy of the data. It is up to the application to ensure the buffer remains valid for +the lifetime of the drwav object. + +The buffer should contain the contents of the entire wave file, not just the sample data. +*/ +DRWAV_API drwav_bool32 drwav_init_memory(drwav* pWav, const void* data, size_t dataSize, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_memory_ex(drwav* pWav, const void* data, size_t dataSize, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_memory_with_metadata(drwav* pWav, const void* data, size_t dataSize, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks); + +/* +Helper for initializing a writer which outputs data to a memory buffer. + +dr_wav will manage the memory allocations, however it is up to the caller to free the data with drwav_free(). + +The buffer will remain allocated even after drwav_uninit() is called. The buffer should not be considered valid +until after drwav_uninit() has been called. +*/ +DRWAV_API drwav_bool32 drwav_init_memory_write(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_memory_write_sequential(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_bool32 drwav_init_memory_write_sequential_pcm_frames(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks); + + +#ifndef DR_WAV_NO_CONVERSION_API +/* +Opens and reads an entire wav file in a single operation. + +The return value is a heap-allocated buffer containing the audio data. Use drwav_free() to free the buffer. +*/ +DRWAV_API drwav_int16* drwav_open_and_read_pcm_frames_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API float* drwav_open_and_read_pcm_frames_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_int32* drwav_open_and_read_pcm_frames_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +#ifndef DR_WAV_NO_STDIO +/* +Opens and decodes an entire wav file in a single operation. + +The return value is a heap-allocated buffer containing the audio data. Use drwav_free() to free the buffer. +*/ +DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +#endif +/* +Opens and decodes an entire wav file from a block of memory in a single operation. + +The return value is a heap-allocated buffer containing the audio data. Use drwav_free() to free the buffer. +*/ +DRWAV_API drwav_int16* drwav_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API float* drwav_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +DRWAV_API drwav_int32* drwav_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks); +#endif + +/* Frees data that was allocated internally by dr_wav. */ +DRWAV_API void drwav_free(void* p, const drwav_allocation_callbacks* pAllocationCallbacks); + +/* Converts bytes from a wav stream to a sized type of native endian. */ +DRWAV_API drwav_uint16 drwav_bytes_to_u16(const drwav_uint8* data); +DRWAV_API drwav_int16 drwav_bytes_to_s16(const drwav_uint8* data); +DRWAV_API drwav_uint32 drwav_bytes_to_u32(const drwav_uint8* data); +DRWAV_API drwav_int32 drwav_bytes_to_s32(const drwav_uint8* data); +DRWAV_API drwav_uint64 drwav_bytes_to_u64(const drwav_uint8* data); +DRWAV_API drwav_int64 drwav_bytes_to_s64(const drwav_uint8* data); +DRWAV_API float drwav_bytes_to_f32(const drwav_uint8* data); + +/* Compares a GUID for the purpose of checking the type of a Wave64 chunk. */ +DRWAV_API drwav_bool32 drwav_guid_equal(const drwav_uint8 a[16], const drwav_uint8 b[16]); + +/* Compares a four-character-code for the purpose of checking the type of a RIFF chunk. */ +DRWAV_API drwav_bool32 drwav_fourcc_equal(const drwav_uint8* a, const char* b); + +#ifdef __cplusplus +} +#endif +#endif /* dr_wav_h */ + + +/************************************************************************************************************************************************************ + ************************************************************************************************************************************************************ + + IMPLEMENTATION + + ************************************************************************************************************************************************************ + ************************************************************************************************************************************************************/ +#if defined(DR_WAV_IMPLEMENTATION) || defined(DRWAV_IMPLEMENTATION) +#ifndef dr_wav_c +#define dr_wav_c + +#ifdef __MRC__ +/* MrC currently doesn't compile dr_wav correctly with any optimizations enabled. */ +#pragma options opt off +#endif + +#include +#include +#include /* For INT_MAX */ + +#ifndef DR_WAV_NO_STDIO +#include +#ifndef DR_WAV_NO_WCHAR +#include +#endif +#endif + +/* Standard library stuff. */ +#ifndef DRWAV_ASSERT +#include +#define DRWAV_ASSERT(expression) assert(expression) +#endif +#ifndef DRWAV_MALLOC +#define DRWAV_MALLOC(sz) malloc((sz)) +#endif +#ifndef DRWAV_REALLOC +#define DRWAV_REALLOC(p, sz) realloc((p), (sz)) +#endif +#ifndef DRWAV_FREE +#define DRWAV_FREE(p) free((p)) +#endif +#ifndef DRWAV_COPY_MEMORY +#define DRWAV_COPY_MEMORY(dst, src, sz) memcpy((dst), (src), (sz)) +#endif +#ifndef DRWAV_ZERO_MEMORY +#define DRWAV_ZERO_MEMORY(p, sz) memset((p), 0, (sz)) +#endif +#ifndef DRWAV_ZERO_OBJECT +#define DRWAV_ZERO_OBJECT(p) DRWAV_ZERO_MEMORY((p), sizeof(*p)) +#endif + +#define drwav_countof(x) (sizeof(x) / sizeof(x[0])) +#define drwav_align(x, a) ((((x) + (a) - 1) / (a)) * (a)) +#define drwav_min(a, b) (((a) < (b)) ? (a) : (b)) +#define drwav_max(a, b) (((a) > (b)) ? (a) : (b)) +#define drwav_clamp(x, lo, hi) (drwav_max((lo), drwav_min((hi), (x)))) +#define drwav_offset_ptr(p, offset) (((drwav_uint8*)(p)) + (offset)) + +#define DRWAV_MAX_SIMD_VECTOR_SIZE 32 + +/* Architecture Detection */ +#if defined(__x86_64__) || (defined(_M_X64) && !defined(_M_ARM64EC)) + #define DRWAV_X64 +#elif defined(__i386) || defined(_M_IX86) + #define DRWAV_X86 +#elif defined(__arm__) || defined(_M_ARM) + #define DRWAV_ARM +#endif +/* End Architecture Detection */ + +/* Inline */ +#ifdef _MSC_VER + #define DRWAV_INLINE __forceinline +#elif defined(__GNUC__) + /* + I've had a bug report where GCC is emitting warnings about functions possibly not being inlineable. This warning happens when + the __attribute__((always_inline)) attribute is defined without an "inline" statement. I think therefore there must be some + case where "__inline__" is not always defined, thus the compiler emitting these warnings. When using -std=c89 or -ansi on the + command line, we cannot use the "inline" keyword and instead need to use "__inline__". In an attempt to work around this issue + I am using "__inline__" only when we're compiling in strict ANSI mode. + */ + #if defined(__STRICT_ANSI__) + #define DRWAV_GNUC_INLINE_HINT __inline__ + #else + #define DRWAV_GNUC_INLINE_HINT inline + #endif + + #if (__GNUC__ > 3 || (__GNUC__ == 3 && __GNUC_MINOR__ >= 2)) || defined(__clang__) + #define DRWAV_INLINE DRWAV_GNUC_INLINE_HINT __attribute__((always_inline)) + #else + #define DRWAV_INLINE DRWAV_GNUC_INLINE_HINT + #endif +#elif defined(__WATCOMC__) + #define DRWAV_INLINE __inline +#else + #define DRWAV_INLINE +#endif +/* End Inline */ + +/* SIZE_MAX */ +#if defined(SIZE_MAX) + #define DRWAV_SIZE_MAX SIZE_MAX +#else + #if defined(_WIN64) || defined(_LP64) || defined(__LP64__) + #define DRWAV_SIZE_MAX ((drwav_uint64)0xFFFFFFFFFFFFFFFF) + #else + #define DRWAV_SIZE_MAX 0xFFFFFFFF + #endif +#endif +/* End SIZE_MAX */ + +/* Weird bit manipulation is for C89 compatibility (no direct support for 64-bit integers). */ +#define DRWAV_INT64_MIN ((drwav_int64) ((drwav_uint64)0x80000000 << 32)) +#define DRWAV_INT64_MAX ((drwav_int64)(((drwav_uint64)0x7FFFFFFF << 32) | 0xFFFFFFFF)) + +#if defined(_MSC_VER) && _MSC_VER >= 1400 + #define DRWAV_HAS_BYTESWAP16_INTRINSIC + #define DRWAV_HAS_BYTESWAP32_INTRINSIC + #define DRWAV_HAS_BYTESWAP64_INTRINSIC +#elif defined(__clang__) + #if defined(__has_builtin) + #if __has_builtin(__builtin_bswap16) + #define DRWAV_HAS_BYTESWAP16_INTRINSIC + #endif + #if __has_builtin(__builtin_bswap32) + #define DRWAV_HAS_BYTESWAP32_INTRINSIC + #endif + #if __has_builtin(__builtin_bswap64) + #define DRWAV_HAS_BYTESWAP64_INTRINSIC + #endif + #endif +#elif defined(__GNUC__) + #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 3)) + #define DRWAV_HAS_BYTESWAP32_INTRINSIC + #define DRWAV_HAS_BYTESWAP64_INTRINSIC + #endif + #if ((__GNUC__ > 4) || (__GNUC__ == 4 && __GNUC_MINOR__ >= 8)) + #define DRWAV_HAS_BYTESWAP16_INTRINSIC + #endif +#endif + +DRWAV_API void drwav_version(drwav_uint32* pMajor, drwav_uint32* pMinor, drwav_uint32* pRevision) +{ + if (pMajor) { + *pMajor = DRWAV_VERSION_MAJOR; + } + + if (pMinor) { + *pMinor = DRWAV_VERSION_MINOR; + } + + if (pRevision) { + *pRevision = DRWAV_VERSION_REVISION; + } +} + +DRWAV_API const char* drwav_version_string(void) +{ + return DRWAV_VERSION_STRING; +} + +/* +These limits are used for basic validation when initializing the decoder. If you exceed these limits, first of all: what on Earth are +you doing?! (Let me know, I'd be curious!) Second, you can adjust these by #define-ing them before the dr_wav implementation. +*/ +#ifndef DRWAV_MAX_SAMPLE_RATE +#define DRWAV_MAX_SAMPLE_RATE 384000 +#endif +#ifndef DRWAV_MAX_CHANNELS +#define DRWAV_MAX_CHANNELS 256 +#endif +#ifndef DRWAV_MAX_BITS_PER_SAMPLE +#define DRWAV_MAX_BITS_PER_SAMPLE 64 +#endif + +static const drwav_uint8 drwavGUID_W64_RIFF[16] = {0x72,0x69,0x66,0x66, 0x2E,0x91, 0xCF,0x11, 0xA5,0xD6, 0x28,0xDB,0x04,0xC1,0x00,0x00}; /* 66666972-912E-11CF-A5D6-28DB04C10000 */ +static const drwav_uint8 drwavGUID_W64_WAVE[16] = {0x77,0x61,0x76,0x65, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 65766177-ACF3-11D3-8CD1-00C04F8EDB8A */ +/*static const drwav_uint8 drwavGUID_W64_JUNK[16] = {0x6A,0x75,0x6E,0x6B, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A};*/ /* 6B6E756A-ACF3-11D3-8CD1-00C04F8EDB8A */ +static const drwav_uint8 drwavGUID_W64_FMT [16] = {0x66,0x6D,0x74,0x20, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 20746D66-ACF3-11D3-8CD1-00C04F8EDB8A */ +static const drwav_uint8 drwavGUID_W64_FACT[16] = {0x66,0x61,0x63,0x74, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 74636166-ACF3-11D3-8CD1-00C04F8EDB8A */ +static const drwav_uint8 drwavGUID_W64_DATA[16] = {0x64,0x61,0x74,0x61, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A}; /* 61746164-ACF3-11D3-8CD1-00C04F8EDB8A */ +/*static const drwav_uint8 drwavGUID_W64_SMPL[16] = {0x73,0x6D,0x70,0x6C, 0xF3,0xAC, 0xD3,0x11, 0x8C,0xD1, 0x00,0xC0,0x4F,0x8E,0xDB,0x8A};*/ /* 6C706D73-ACF3-11D3-8CD1-00C04F8EDB8A */ + + +static DRWAV_INLINE int drwav__is_little_endian(void) +{ +#if defined(DRWAV_X86) || defined(DRWAV_X64) + return DRWAV_TRUE; +#elif defined(__BYTE_ORDER) && defined(__LITTLE_ENDIAN) && __BYTE_ORDER == __LITTLE_ENDIAN + return DRWAV_TRUE; +#else + int n = 1; + return (*(char*)&n) == 1; +#endif +} + + +static DRWAV_INLINE void drwav_bytes_to_guid(const drwav_uint8* data, drwav_uint8* guid) +{ + int i; + for (i = 0; i < 16; ++i) { + guid[i] = data[i]; + } +} + + +static DRWAV_INLINE drwav_uint16 drwav__bswap16(drwav_uint16 n) +{ +#ifdef DRWAV_HAS_BYTESWAP16_INTRINSIC + #if defined(_MSC_VER) + return _byteswap_ushort(n); + #elif defined(__GNUC__) || defined(__clang__) + return __builtin_bswap16(n); + #else + #error "This compiler does not support the byte swap intrinsic." + #endif +#else + return ((n & 0xFF00) >> 8) | + ((n & 0x00FF) << 8); +#endif +} + +static DRWAV_INLINE drwav_uint32 drwav__bswap32(drwav_uint32 n) +{ +#ifdef DRWAV_HAS_BYTESWAP32_INTRINSIC + #if defined(_MSC_VER) + return _byteswap_ulong(n); + #elif defined(__GNUC__) || defined(__clang__) + #if defined(DRWAV_ARM) && (defined(__ARM_ARCH) && __ARM_ARCH >= 6) && !defined(DRWAV_64BIT) /* <-- 64-bit inline assembly has not been tested, so disabling for now. */ + /* Inline assembly optimized implementation for ARM. In my testing, GCC does not generate optimized code with __builtin_bswap32(). */ + drwav_uint32 r; + __asm__ __volatile__ ( + #if defined(DRWAV_64BIT) + "rev %w[out], %w[in]" : [out]"=r"(r) : [in]"r"(n) /* <-- This is untested. If someone in the community could test this, that would be appreciated! */ + #else + "rev %[out], %[in]" : [out]"=r"(r) : [in]"r"(n) + #endif + ); + return r; + #else + return __builtin_bswap32(n); + #endif + #else + #error "This compiler does not support the byte swap intrinsic." + #endif +#else + return ((n & 0xFF000000) >> 24) | + ((n & 0x00FF0000) >> 8) | + ((n & 0x0000FF00) << 8) | + ((n & 0x000000FF) << 24); +#endif +} + +static DRWAV_INLINE drwav_uint64 drwav__bswap64(drwav_uint64 n) +{ +#ifdef DRWAV_HAS_BYTESWAP64_INTRINSIC + #if defined(_MSC_VER) + return _byteswap_uint64(n); + #elif defined(__GNUC__) || defined(__clang__) + return __builtin_bswap64(n); + #else + #error "This compiler does not support the byte swap intrinsic." + #endif +#else + /* Weird "<< 32" bitshift is required for C89 because it doesn't support 64-bit constants. Should be optimized out by a good compiler. */ + return ((n & ((drwav_uint64)0xFF000000 << 32)) >> 56) | + ((n & ((drwav_uint64)0x00FF0000 << 32)) >> 40) | + ((n & ((drwav_uint64)0x0000FF00 << 32)) >> 24) | + ((n & ((drwav_uint64)0x000000FF << 32)) >> 8) | + ((n & ((drwav_uint64)0xFF000000 )) << 8) | + ((n & ((drwav_uint64)0x00FF0000 )) << 24) | + ((n & ((drwav_uint64)0x0000FF00 )) << 40) | + ((n & ((drwav_uint64)0x000000FF )) << 56); +#endif +} + + +static DRWAV_INLINE drwav_int16 drwav__bswap_s16(drwav_int16 n) +{ + return (drwav_int16)drwav__bswap16((drwav_uint16)n); +} + +static DRWAV_INLINE void drwav__bswap_samples_s16(drwav_int16* pSamples, drwav_uint64 sampleCount) +{ + drwav_uint64 iSample; + for (iSample = 0; iSample < sampleCount; iSample += 1) { + pSamples[iSample] = drwav__bswap_s16(pSamples[iSample]); + } +} + + +static DRWAV_INLINE void drwav__bswap_s24(drwav_uint8* p) +{ + drwav_uint8 t; + t = p[0]; + p[0] = p[2]; + p[2] = t; +} + +static DRWAV_INLINE void drwav__bswap_samples_s24(drwav_uint8* pSamples, drwav_uint64 sampleCount) +{ + drwav_uint64 iSample; + for (iSample = 0; iSample < sampleCount; iSample += 1) { + drwav_uint8* pSample = pSamples + (iSample*3); + drwav__bswap_s24(pSample); + } +} + + +static DRWAV_INLINE drwav_int32 drwav__bswap_s32(drwav_int32 n) +{ + return (drwav_int32)drwav__bswap32((drwav_uint32)n); +} + +static DRWAV_INLINE void drwav__bswap_samples_s32(drwav_int32* pSamples, drwav_uint64 sampleCount) +{ + drwav_uint64 iSample; + for (iSample = 0; iSample < sampleCount; iSample += 1) { + pSamples[iSample] = drwav__bswap_s32(pSamples[iSample]); + } +} + + +static DRWAV_INLINE drwav_int64 drwav__bswap_s64(drwav_int64 n) +{ + return (drwav_int64)drwav__bswap64((drwav_uint64)n); +} + +static DRWAV_INLINE void drwav__bswap_samples_s64(drwav_int64* pSamples, drwav_uint64 sampleCount) +{ + drwav_uint64 iSample; + for (iSample = 0; iSample < sampleCount; iSample += 1) { + pSamples[iSample] = drwav__bswap_s64(pSamples[iSample]); + } +} + + +static DRWAV_INLINE float drwav__bswap_f32(float n) +{ + union { + drwav_uint32 i; + float f; + } x; + x.f = n; + x.i = drwav__bswap32(x.i); + + return x.f; +} + +static DRWAV_INLINE void drwav__bswap_samples_f32(float* pSamples, drwav_uint64 sampleCount) +{ + drwav_uint64 iSample; + for (iSample = 0; iSample < sampleCount; iSample += 1) { + pSamples[iSample] = drwav__bswap_f32(pSamples[iSample]); + } +} + + +static DRWAV_INLINE void drwav__bswap_samples(void* pSamples, drwav_uint64 sampleCount, drwav_uint32 bytesPerSample) +{ + switch (bytesPerSample) + { + case 1: + { + /* No-op. */ + } break; + case 2: + { + drwav__bswap_samples_s16((drwav_int16*)pSamples, sampleCount); + } break; + case 3: + { + drwav__bswap_samples_s24((drwav_uint8*)pSamples, sampleCount); + } break; + case 4: + { + drwav__bswap_samples_s32((drwav_int32*)pSamples, sampleCount); + } break; + case 8: + { + drwav__bswap_samples_s64((drwav_int64*)pSamples, sampleCount); + } break; + default: + { + drwav_uint64 iSample; + + for (iSample = 0; iSample < sampleCount; iSample += 1) { + drwav_uint8* pSample = (drwav_uint8*)pSamples + (iSample * bytesPerSample); + drwav_uint32 iByte; + + for (iByte = 0; iByte < bytesPerSample / 2; iByte += 1) { + drwav_uint8 temp = pSample[iByte]; + pSample[iByte] = pSample[bytesPerSample - iByte - 1]; + pSample[bytesPerSample - iByte - 1] = temp; + } + } + } break; + } +} + + + +DRWAV_PRIVATE DRWAV_INLINE drwav_bool32 drwav_is_container_be(drwav_container container) +{ + if (container == drwav_container_rifx || container == drwav_container_aiff) { + return DRWAV_TRUE; + } else { + return DRWAV_FALSE; + } +} + + +DRWAV_PRIVATE DRWAV_INLINE drwav_uint16 drwav_bytes_to_u16_le(const drwav_uint8* data) +{ + return ((drwav_uint16)data[0] << 0) | ((drwav_uint16)data[1] << 8); +} + +DRWAV_PRIVATE DRWAV_INLINE drwav_uint16 drwav_bytes_to_u16_be(const drwav_uint8* data) +{ + return ((drwav_uint16)data[1] << 0) | ((drwav_uint16)data[0] << 8); +} + +DRWAV_PRIVATE DRWAV_INLINE drwav_uint16 drwav_bytes_to_u16_ex(const drwav_uint8* data, drwav_container container) +{ + if (drwav_is_container_be(container)) { + return drwav_bytes_to_u16_be(data); + } else { + return drwav_bytes_to_u16_le(data); + } +} + + +DRWAV_PRIVATE DRWAV_INLINE drwav_uint32 drwav_bytes_to_u32_le(const drwav_uint8* data) +{ + return ((drwav_uint32)data[0] << 0) | ((drwav_uint32)data[1] << 8) | ((drwav_uint32)data[2] << 16) | ((drwav_uint32)data[3] << 24); +} + +DRWAV_PRIVATE DRWAV_INLINE drwav_uint32 drwav_bytes_to_u32_be(const drwav_uint8* data) +{ + return ((drwav_uint32)data[3] << 0) | ((drwav_uint32)data[2] << 8) | ((drwav_uint32)data[1] << 16) | ((drwav_uint32)data[0] << 24); +} + +DRWAV_PRIVATE DRWAV_INLINE drwav_uint32 drwav_bytes_to_u32_ex(const drwav_uint8* data, drwav_container container) +{ + if (drwav_is_container_be(container)) { + return drwav_bytes_to_u32_be(data); + } else { + return drwav_bytes_to_u32_le(data); + } +} + + + +DRWAV_PRIVATE drwav_int64 drwav_aiff_extented_to_s64(const drwav_uint8* data) +{ + drwav_uint32 exponent = ((drwav_uint32)data[0] << 8) | data[1]; + drwav_uint64 hi = ((drwav_uint64)data[2] << 24) | ((drwav_uint64)data[3] << 16) | ((drwav_uint64)data[4] << 8) | ((drwav_uint64)data[5] << 0); + drwav_uint64 lo = ((drwav_uint64)data[6] << 24) | ((drwav_uint64)data[7] << 16) | ((drwav_uint64)data[8] << 8) | ((drwav_uint64)data[9] << 0); + drwav_uint64 significand = (hi << 32) | lo; + int sign = exponent >> 15; + + /* Remove sign bit. */ + exponent &= 0x7FFF; + + /* Special cases. */ + if (exponent == 0 && significand == 0) { + return 0; + } else if (exponent == 0x7FFF) { + return sign ? DRWAV_INT64_MIN : DRWAV_INT64_MAX; /* Infinite. */ + } + + exponent -= 16383; + + if (exponent > 63) { + return sign ? DRWAV_INT64_MIN : DRWAV_INT64_MAX; /* Too big for a 64-bit integer. */ + } else if (exponent < 1) { + return 0; /* Number is less than 1, so rounds down to 0. */ + } + + significand >>= (63 - exponent); + + if (sign) { + return -(drwav_int64)significand; + } else { + return (drwav_int64)significand; + } +} + + +DRWAV_PRIVATE void* drwav__malloc_default(size_t sz, void* pUserData) +{ + (void)pUserData; + return DRWAV_MALLOC(sz); +} + +DRWAV_PRIVATE void* drwav__realloc_default(void* p, size_t sz, void* pUserData) +{ + (void)pUserData; + return DRWAV_REALLOC(p, sz); +} + +DRWAV_PRIVATE void drwav__free_default(void* p, void* pUserData) +{ + (void)pUserData; + DRWAV_FREE(p); +} + + +DRWAV_PRIVATE void* drwav__malloc_from_callbacks(size_t sz, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks == NULL) { + return NULL; + } + + if (pAllocationCallbacks->onMalloc != NULL) { + return pAllocationCallbacks->onMalloc(sz, pAllocationCallbacks->pUserData); + } + + /* Try using realloc(). */ + if (pAllocationCallbacks->onRealloc != NULL) { + return pAllocationCallbacks->onRealloc(NULL, sz, pAllocationCallbacks->pUserData); + } + + return NULL; +} + +DRWAV_PRIVATE void* drwav__realloc_from_callbacks(void* p, size_t szNew, size_t szOld, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks == NULL) { + return NULL; + } + + if (pAllocationCallbacks->onRealloc != NULL) { + return pAllocationCallbacks->onRealloc(p, szNew, pAllocationCallbacks->pUserData); + } + + /* Try emulating realloc() in terms of malloc()/free(). */ + if (pAllocationCallbacks->onMalloc != NULL && pAllocationCallbacks->onFree != NULL) { + void* p2; + + p2 = pAllocationCallbacks->onMalloc(szNew, pAllocationCallbacks->pUserData); + if (p2 == NULL) { + return NULL; + } + + if (p != NULL) { + DRWAV_COPY_MEMORY(p2, p, szOld); + pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData); + } + + return p2; + } + + return NULL; +} + +DRWAV_PRIVATE void drwav__free_from_callbacks(void* p, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (p == NULL || pAllocationCallbacks == NULL) { + return; + } + + if (pAllocationCallbacks->onFree != NULL) { + pAllocationCallbacks->onFree(p, pAllocationCallbacks->pUserData); + } +} + + +DRWAV_PRIVATE drwav_allocation_callbacks drwav_copy_allocation_callbacks_or_defaults(const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks != NULL) { + /* Copy. */ + return *pAllocationCallbacks; + } else { + /* Defaults. */ + drwav_allocation_callbacks allocationCallbacks; + allocationCallbacks.pUserData = NULL; + allocationCallbacks.onMalloc = drwav__malloc_default; + allocationCallbacks.onRealloc = drwav__realloc_default; + allocationCallbacks.onFree = drwav__free_default; + return allocationCallbacks; + } +} + + +static DRWAV_INLINE drwav_bool32 drwav__is_compressed_format_tag(drwav_uint16 formatTag) +{ + return + formatTag == DR_WAVE_FORMAT_ADPCM || + formatTag == DR_WAVE_FORMAT_DVI_ADPCM; +} + +DRWAV_PRIVATE unsigned int drwav__chunk_padding_size_riff(drwav_uint64 chunkSize) +{ + return (unsigned int)(chunkSize % 2); +} + +DRWAV_PRIVATE unsigned int drwav__chunk_padding_size_w64(drwav_uint64 chunkSize) +{ + return (unsigned int)(chunkSize % 8); +} + +DRWAV_PRIVATE unsigned int drwav_calculate_padding_size(drwav_container container, drwav_uint64 chunkSize) +{ + if (container == drwav_container_riff || container == drwav_container_rf64) { + return drwav__chunk_padding_size_riff(chunkSize); + } else { + return drwav__chunk_padding_size_w64(chunkSize); + } +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__msadpcm(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut); +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__ima(drwav* pWav, drwav_uint64 samplesToRead, drwav_int16* pBufferOut); +DRWAV_PRIVATE drwav_bool32 drwav_init_write__internal(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount); + +DRWAV_PRIVATE drwav_result drwav__read_chunk_header(drwav_read_proc onRead, void* pUserData, drwav_container container, drwav_uint64* pRunningBytesReadOut, drwav_chunk_header* pHeaderOut) +{ + if (container == drwav_container_riff || container == drwav_container_rifx || container == drwav_container_rf64 || container == drwav_container_aiff) { + drwav_uint8 sizeInBytes[4]; + + if (onRead(pUserData, pHeaderOut->id.fourcc, 4) != 4) { + return DRWAV_AT_END; + } + + if (onRead(pUserData, sizeInBytes, 4) != 4) { + return DRWAV_INVALID_FILE; + } + + pHeaderOut->sizeInBytes = drwav_bytes_to_u32_ex(sizeInBytes, container); + pHeaderOut->paddingSize = drwav__chunk_padding_size_riff(pHeaderOut->sizeInBytes); + + *pRunningBytesReadOut += 8; + } else if (container == drwav_container_w64) { + drwav_uint8 sizeInBytes[8]; + + if (onRead(pUserData, pHeaderOut->id.guid, 16) != 16) { + return DRWAV_AT_END; + } + + if (onRead(pUserData, sizeInBytes, 8) != 8) { + return DRWAV_INVALID_FILE; + } + + pHeaderOut->sizeInBytes = drwav_bytes_to_u64(sizeInBytes); + + /* Subtract 24 from the size because with w64 the reported chunk size includes the size of the header itself. */ + if (pHeaderOut->sizeInBytes >= 24) { + pHeaderOut->sizeInBytes -= 24; + } else { + return DRWAV_INVALID_FILE; + } + + pHeaderOut->paddingSize = drwav__chunk_padding_size_w64(pHeaderOut->sizeInBytes); + *pRunningBytesReadOut += 24; + } else { + return DRWAV_INVALID_FILE; + } + + return DRWAV_SUCCESS; +} + +DRWAV_PRIVATE drwav_bool32 drwav__seek_forward(drwav_seek_proc onSeek, drwav_uint64 offset, void* pUserData) +{ + drwav_uint64 bytesRemainingToSeek = offset; + while (bytesRemainingToSeek > 0) { + if (bytesRemainingToSeek > 0x7FFFFFFF) { + if (!onSeek(pUserData, 0x7FFFFFFF, DRWAV_SEEK_CUR)) { + return DRWAV_FALSE; + } + bytesRemainingToSeek -= 0x7FFFFFFF; + } else { + if (!onSeek(pUserData, (int)bytesRemainingToSeek, DRWAV_SEEK_CUR)) { + return DRWAV_FALSE; + } + bytesRemainingToSeek = 0; + } + } + + return DRWAV_TRUE; +} + +DRWAV_PRIVATE drwav_bool32 drwav__seek_from_start(drwav_seek_proc onSeek, drwav_uint64 offset, void* pUserData) +{ + if (offset <= 0x7FFFFFFF) { + return onSeek(pUserData, (int)offset, DRWAV_SEEK_SET); + } + + /* Larger than 32-bit seek. */ + if (!onSeek(pUserData, 0x7FFFFFFF, DRWAV_SEEK_SET)) { + return DRWAV_FALSE; + } + offset -= 0x7FFFFFFF; + + for (;;) { + if (offset <= 0x7FFFFFFF) { + return onSeek(pUserData, (int)offset, DRWAV_SEEK_CUR); + } + + if (!onSeek(pUserData, 0x7FFFFFFF, DRWAV_SEEK_CUR)) { + return DRWAV_FALSE; + } + offset -= 0x7FFFFFFF; + } + + /* Should never get here. */ + /*return DRWAV_TRUE; */ +} + + + +DRWAV_PRIVATE size_t drwav__on_read(drwav_read_proc onRead, void* pUserData, void* pBufferOut, size_t bytesToRead, drwav_uint64* pCursor) +{ + size_t bytesRead; + + DRWAV_ASSERT(onRead != NULL); + DRWAV_ASSERT(pCursor != NULL); + + bytesRead = onRead(pUserData, pBufferOut, bytesToRead); + *pCursor += bytesRead; + return bytesRead; +} + +#if 0 +DRWAV_PRIVATE drwav_bool32 drwav__on_seek(drwav_seek_proc onSeek, void* pUserData, int offset, drwav_seek_origin origin, drwav_uint64* pCursor) +{ + DRWAV_ASSERT(onSeek != NULL); + DRWAV_ASSERT(pCursor != NULL); + + if (!onSeek(pUserData, offset, origin)) { + return DRWAV_FALSE; + } + + if (origin == DRWAV_SEEK_SET) { + *pCursor = offset; + } else { + *pCursor += offset; + } + + return DRWAV_TRUE; +} +#endif + + +#define DRWAV_SMPL_BYTES 36 +#define DRWAV_SMPL_LOOP_BYTES 24 +#define DRWAV_INST_BYTES 7 +#define DRWAV_ACID_BYTES 24 +#define DRWAV_CUE_BYTES 4 +#define DRWAV_BEXT_BYTES 602 +#define DRWAV_BEXT_DESCRIPTION_BYTES 256 +#define DRWAV_BEXT_ORIGINATOR_NAME_BYTES 32 +#define DRWAV_BEXT_ORIGINATOR_REF_BYTES 32 +#define DRWAV_BEXT_RESERVED_BYTES 180 +#define DRWAV_BEXT_UMID_BYTES 64 +#define DRWAV_CUE_POINT_BYTES 24 +#define DRWAV_LIST_LABEL_OR_NOTE_BYTES 4 +#define DRWAV_LIST_LABELLED_TEXT_BYTES 20 + +#define DRWAV_METADATA_ALIGNMENT 8 + +typedef enum +{ + drwav__metadata_parser_stage_count, + drwav__metadata_parser_stage_read +} drwav__metadata_parser_stage; + +typedef struct +{ + drwav_read_proc onRead; + drwav_seek_proc onSeek; + void *pReadSeekUserData; + drwav__metadata_parser_stage stage; + drwav_metadata *pMetadata; + drwav_uint32 metadataCount; + drwav_uint8 *pData; + drwav_uint8 *pDataCursor; + drwav_uint64 metadataCursor; + drwav_uint64 extraCapacity; +} drwav__metadata_parser; + +DRWAV_PRIVATE size_t drwav__metadata_memory_capacity(drwav__metadata_parser* pParser) +{ + drwav_uint64 cap = sizeof(drwav_metadata) * (drwav_uint64)pParser->metadataCount + pParser->extraCapacity; + if (cap > DRWAV_SIZE_MAX) { + return 0; /* Too big. */ + } + + return (size_t)cap; /* Safe cast thanks to the check above. */ +} + +DRWAV_PRIVATE drwav_uint8* drwav__metadata_get_memory(drwav__metadata_parser* pParser, size_t size, size_t align) +{ + drwav_uint8* pResult; + + if (align) { + drwav_uintptr modulo = (drwav_uintptr)pParser->pDataCursor % align; + if (modulo != 0) { + pParser->pDataCursor += align - modulo; + } + } + + pResult = pParser->pDataCursor; + + /* + Getting to the point where this function is called means there should always be memory + available. Out of memory checks should have been done at an earlier stage. + */ + DRWAV_ASSERT((pResult + size) <= (pParser->pData + drwav__metadata_memory_capacity(pParser))); + + pParser->pDataCursor += size; + return pResult; +} + +DRWAV_PRIVATE void drwav__metadata_request_extra_memory_for_stage_2(drwav__metadata_parser* pParser, size_t bytes, size_t align) +{ + size_t extra = bytes + (align ? (align - 1) : 0); + pParser->extraCapacity += extra; +} + +DRWAV_PRIVATE drwav_result drwav__metadata_alloc(drwav__metadata_parser* pParser, drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pParser->extraCapacity != 0 || pParser->metadataCount != 0) { + pAllocationCallbacks->onFree(pParser->pData, pAllocationCallbacks->pUserData); + + pParser->pData = (drwav_uint8*)pAllocationCallbacks->onMalloc(drwav__metadata_memory_capacity(pParser), pAllocationCallbacks->pUserData); + pParser->pDataCursor = pParser->pData; + + if (pParser->pData == NULL) { + return DRWAV_OUT_OF_MEMORY; + } + + /* + We don't need to worry about specifying an alignment here because malloc always returns something + of suitable alignment. This also means pParser->pMetadata is all that we need to store in order + for us to free when we are done. + */ + pParser->pMetadata = (drwav_metadata*)drwav__metadata_get_memory(pParser, sizeof(drwav_metadata) * pParser->metadataCount, 1); + pParser->metadataCursor = 0; + } + + return DRWAV_SUCCESS; +} + +DRWAV_PRIVATE size_t drwav__metadata_parser_read(drwav__metadata_parser* pParser, void* pBufferOut, size_t bytesToRead, drwav_uint64* pCursor) +{ + if (pCursor != NULL) { + return drwav__on_read(pParser->onRead, pParser->pReadSeekUserData, pBufferOut, bytesToRead, pCursor); + } else { + return pParser->onRead(pParser->pReadSeekUserData, pBufferOut, bytesToRead); + } +} + +DRWAV_PRIVATE drwav_bool32 drwav__metadata_validate_smpl_chunk(const drwav_chunk_header* pChunkHeader, drwav_uint32 loopCount, drwav_uint32 samplerSpecificDataSizeInBytes, drwav_uint64* pTrailingDataSizeInBytes) +{ + drwav_uint64 remainingDataSizeInBytes; + + DRWAV_ASSERT(pChunkHeader != NULL); + + if (pChunkHeader->sizeInBytes < DRWAV_SMPL_BYTES) { + return DRWAV_FALSE; + } + + remainingDataSizeInBytes = pChunkHeader->sizeInBytes - DRWAV_SMPL_BYTES; + if ((drwav_uint64)loopCount > remainingDataSizeInBytes / DRWAV_SMPL_LOOP_BYTES) { + return DRWAV_FALSE; + } + + remainingDataSizeInBytes -= (drwav_uint64)loopCount * DRWAV_SMPL_LOOP_BYTES; + if ((drwav_uint64)samplerSpecificDataSizeInBytes > remainingDataSizeInBytes) { + return DRWAV_FALSE; + } + + if (pTrailingDataSizeInBytes != NULL) { + *pTrailingDataSizeInBytes = remainingDataSizeInBytes - samplerSpecificDataSizeInBytes; + } + + return DRWAV_TRUE; +} + +DRWAV_PRIVATE drwav_uint64 drwav__read_smpl_to_metadata_obj(drwav__metadata_parser* pParser, const drwav_chunk_header* pChunkHeader, drwav_metadata* pMetadata) +{ + drwav_uint8 smplHeaderData[DRWAV_SMPL_BYTES]; + drwav_uint64 totalBytesRead = 0; + size_t bytesJustRead; + + if (pMetadata == NULL) { + return 0; + } + + bytesJustRead = drwav__metadata_parser_read(pParser, smplHeaderData, sizeof(smplHeaderData), &totalBytesRead); + + DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read); + DRWAV_ASSERT(pChunkHeader != NULL); + + if (pMetadata != NULL && bytesJustRead == sizeof(smplHeaderData)) { + drwav_uint32 iSampleLoop; + drwav_uint32 loopCount; + drwav_uint32 samplerSpecificDataSizeInBytes; + drwav_uint64 trailingDataSizeInBytes; + + /* + When we calculated the amount of memory required for the "smpl" chunk we excluded the chunk entirely + if its loop or sampler-specific data exceeded the chunk size. When this happens, the second stage will + still hit this path but the `pMetadata` will either point at the end of the allocation or at another + chunk. We need to repeat the validation before dereferencing the pMetadata object. + */ + loopCount = drwav_bytes_to_u32(smplHeaderData + 28); + samplerSpecificDataSizeInBytes = drwav_bytes_to_u32(smplHeaderData + 32); + if (!drwav__metadata_validate_smpl_chunk(pChunkHeader, loopCount, samplerSpecificDataSizeInBytes, &trailingDataSizeInBytes)) { + return totalBytesRead; + } + + pMetadata->type = drwav_metadata_type_smpl; + pMetadata->data.smpl.manufacturerId = drwav_bytes_to_u32(smplHeaderData + 0); + pMetadata->data.smpl.productId = drwav_bytes_to_u32(smplHeaderData + 4); + pMetadata->data.smpl.samplePeriodNanoseconds = drwav_bytes_to_u32(smplHeaderData + 8); + pMetadata->data.smpl.midiUnityNote = drwav_bytes_to_u32(smplHeaderData + 12); + pMetadata->data.smpl.midiPitchFraction = drwav_bytes_to_u32(smplHeaderData + 16); + pMetadata->data.smpl.smpteFormat = drwav_bytes_to_u32(smplHeaderData + 20); + pMetadata->data.smpl.smpteOffset = drwav_bytes_to_u32(smplHeaderData + 24); + pMetadata->data.smpl.sampleLoopCount = loopCount; + pMetadata->data.smpl.samplerSpecificDataSizeInBytes = samplerSpecificDataSizeInBytes; + + pMetadata->data.smpl.pLoops = (drwav_smpl_loop*)drwav__metadata_get_memory(pParser, sizeof(drwav_smpl_loop) * pMetadata->data.smpl.sampleLoopCount, DRWAV_METADATA_ALIGNMENT); + + for (iSampleLoop = 0; iSampleLoop < pMetadata->data.smpl.sampleLoopCount; ++iSampleLoop) { + drwav_uint8 smplLoopData[DRWAV_SMPL_LOOP_BYTES]; + bytesJustRead = drwav__metadata_parser_read(pParser, smplLoopData, sizeof(smplLoopData), &totalBytesRead); + + if (bytesJustRead == sizeof(smplLoopData)) { + pMetadata->data.smpl.pLoops[iSampleLoop].cuePointId = drwav_bytes_to_u32(smplLoopData + 0); + pMetadata->data.smpl.pLoops[iSampleLoop].type = drwav_bytes_to_u32(smplLoopData + 4); + pMetadata->data.smpl.pLoops[iSampleLoop].firstSampleOffset = drwav_bytes_to_u32(smplLoopData + 8); + pMetadata->data.smpl.pLoops[iSampleLoop].lastSampleOffset = drwav_bytes_to_u32(smplLoopData + 12); + pMetadata->data.smpl.pLoops[iSampleLoop].sampleFraction = drwav_bytes_to_u32(smplLoopData + 16); + pMetadata->data.smpl.pLoops[iSampleLoop].playCount = drwav_bytes_to_u32(smplLoopData + 20); + } else { + return totalBytesRead; + } + } + + if (pMetadata->data.smpl.samplerSpecificDataSizeInBytes > 0) { + pMetadata->data.smpl.pSamplerSpecificData = drwav__metadata_get_memory(pParser, pMetadata->data.smpl.samplerSpecificDataSizeInBytes, 1); + DRWAV_ASSERT(pMetadata->data.smpl.pSamplerSpecificData != NULL); + + bytesJustRead = drwav__metadata_parser_read(pParser, pMetadata->data.smpl.pSamplerSpecificData, pMetadata->data.smpl.samplerSpecificDataSizeInBytes, &totalBytesRead); + if (bytesJustRead != pMetadata->data.smpl.samplerSpecificDataSizeInBytes) { + return totalBytesRead; + } + } + + if (trailingDataSizeInBytes > 0) { + if (!drwav__seek_forward(pParser->onSeek, trailingDataSizeInBytes, pParser->pReadSeekUserData)) { + return totalBytesRead; + } + totalBytesRead += trailingDataSizeInBytes; + } + } + + return totalBytesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav__read_cue_to_metadata_obj(drwav__metadata_parser* pParser, const drwav_chunk_header* pChunkHeader, drwav_metadata* pMetadata) +{ + drwav_uint8 cueHeaderSectionData[DRWAV_CUE_BYTES]; + drwav_uint64 totalBytesRead = 0; + size_t bytesJustRead; + + if (pMetadata == NULL) { + return 0; + } + + bytesJustRead = drwav__metadata_parser_read(pParser, cueHeaderSectionData, sizeof(cueHeaderSectionData), &totalBytesRead); + + DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read); + + if (bytesJustRead == sizeof(cueHeaderSectionData)) { + pMetadata->type = drwav_metadata_type_cue; + pMetadata->data.cue.cuePointCount = drwav_bytes_to_u32(cueHeaderSectionData); + + /* + We need to validate the cue point count against the size of the chunk so we don't read + beyond the chunk. + */ + if (pMetadata->data.cue.cuePointCount == (pChunkHeader->sizeInBytes - DRWAV_CUE_BYTES) / DRWAV_CUE_POINT_BYTES) { + pMetadata->data.cue.pCuePoints = (drwav_cue_point*)drwav__metadata_get_memory(pParser, sizeof(drwav_cue_point) * pMetadata->data.cue.cuePointCount, DRWAV_METADATA_ALIGNMENT); + DRWAV_ASSERT(pMetadata->data.cue.pCuePoints != NULL); + + if (pMetadata->data.cue.cuePointCount > 0) { + drwav_uint32 iCuePoint; + + for (iCuePoint = 0; iCuePoint < pMetadata->data.cue.cuePointCount; ++iCuePoint) { + drwav_uint8 cuePointData[DRWAV_CUE_POINT_BYTES]; + bytesJustRead = drwav__metadata_parser_read(pParser, cuePointData, sizeof(cuePointData), &totalBytesRead); + + if (bytesJustRead == sizeof(cuePointData)) { + pMetadata->data.cue.pCuePoints[iCuePoint].id = drwav_bytes_to_u32(cuePointData + 0); + pMetadata->data.cue.pCuePoints[iCuePoint].playOrderPosition = drwav_bytes_to_u32(cuePointData + 4); + pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[0] = cuePointData[8]; + pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[1] = cuePointData[9]; + pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[2] = cuePointData[10]; + pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId[3] = cuePointData[11]; + pMetadata->data.cue.pCuePoints[iCuePoint].chunkStart = drwav_bytes_to_u32(cuePointData + 12); + pMetadata->data.cue.pCuePoints[iCuePoint].blockStart = drwav_bytes_to_u32(cuePointData + 16); + pMetadata->data.cue.pCuePoints[iCuePoint].sampleOffset = drwav_bytes_to_u32(cuePointData + 20); + } else { + break; + } + } + } + } + } + + return totalBytesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav__read_inst_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata) +{ + drwav_uint8 instData[DRWAV_INST_BYTES]; + drwav_uint64 bytesRead; + + if (pMetadata == NULL) { + return 0; + } + + bytesRead = drwav__metadata_parser_read(pParser, instData, sizeof(instData), NULL); + + DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read); + + if (bytesRead == sizeof(instData)) { + pMetadata->type = drwav_metadata_type_inst; + pMetadata->data.inst.midiUnityNote = (drwav_int8)instData[0]; + pMetadata->data.inst.fineTuneCents = (drwav_int8)instData[1]; + pMetadata->data.inst.gainDecibels = (drwav_int8)instData[2]; + pMetadata->data.inst.lowNote = (drwav_int8)instData[3]; + pMetadata->data.inst.highNote = (drwav_int8)instData[4]; + pMetadata->data.inst.lowVelocity = (drwav_int8)instData[5]; + pMetadata->data.inst.highVelocity = (drwav_int8)instData[6]; + } + + return bytesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav__read_acid_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata) +{ + drwav_uint8 acidData[DRWAV_ACID_BYTES]; + drwav_uint64 bytesRead; + + if (pMetadata == NULL) { + return 0; + } + + bytesRead = drwav__metadata_parser_read(pParser, acidData, sizeof(acidData), NULL); + + DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read); + + if (bytesRead == sizeof(acidData)) { + pMetadata->type = drwav_metadata_type_acid; + pMetadata->data.acid.flags = drwav_bytes_to_u32(acidData + 0); + pMetadata->data.acid.midiUnityNote = drwav_bytes_to_u16(acidData + 4); + pMetadata->data.acid.reserved1 = drwav_bytes_to_u16(acidData + 6); + pMetadata->data.acid.reserved2 = drwav_bytes_to_f32(acidData + 8); + pMetadata->data.acid.numBeats = drwav_bytes_to_u32(acidData + 12); + pMetadata->data.acid.meterDenominator = drwav_bytes_to_u16(acidData + 16); + pMetadata->data.acid.meterNumerator = drwav_bytes_to_u16(acidData + 18); + pMetadata->data.acid.tempo = drwav_bytes_to_f32(acidData + 20); + } + + return bytesRead; +} + +DRWAV_PRIVATE size_t drwav__strlen(const char* str) +{ + size_t result = 0; + + while (*str++) { + result += 1; + } + + return result; +} + +DRWAV_PRIVATE size_t drwav__strlen_clamped(const char* str, size_t maxToRead) +{ + size_t result = 0; + + while (*str++ && result < maxToRead) { + result += 1; + } + + return result; +} + +DRWAV_PRIVATE char* drwav__metadata_copy_string(drwav__metadata_parser* pParser, const char* str, size_t maxToRead) +{ + size_t len = drwav__strlen_clamped(str, maxToRead); + + if (len) { + char* result = (char*)drwav__metadata_get_memory(pParser, len + 1, 1); + DRWAV_ASSERT(result != NULL); + + DRWAV_COPY_MEMORY(result, str, len); + result[len] = '\0'; + + return result; + } else { + return NULL; + } +} + +typedef struct +{ + const void* pBuffer; + size_t sizeInBytes; + size_t cursor; +} drwav_buffer_reader; + +DRWAV_PRIVATE drwav_result drwav_buffer_reader_init(const void* pBuffer, size_t sizeInBytes, drwav_buffer_reader* pReader) +{ + DRWAV_ASSERT(pBuffer != NULL); + DRWAV_ASSERT(pReader != NULL); + + DRWAV_ZERO_OBJECT(pReader); + + pReader->pBuffer = pBuffer; + pReader->sizeInBytes = sizeInBytes; + pReader->cursor = 0; + + return DRWAV_SUCCESS; +} + +DRWAV_PRIVATE const void* drwav_buffer_reader_ptr(const drwav_buffer_reader* pReader) +{ + DRWAV_ASSERT(pReader != NULL); + + return drwav_offset_ptr(pReader->pBuffer, pReader->cursor); +} + +DRWAV_PRIVATE drwav_result drwav_buffer_reader_seek(drwav_buffer_reader* pReader, size_t bytesToSeek) +{ + DRWAV_ASSERT(pReader != NULL); + + if (pReader->cursor + bytesToSeek > pReader->sizeInBytes) { + return DRWAV_BAD_SEEK; /* Seeking too far forward. */ + } + + pReader->cursor += bytesToSeek; + + return DRWAV_SUCCESS; +} + +DRWAV_PRIVATE drwav_result drwav_buffer_reader_read(drwav_buffer_reader* pReader, void* pDst, size_t bytesToRead, size_t* pBytesRead) +{ + drwav_result result = DRWAV_SUCCESS; + size_t bytesRemaining; + + DRWAV_ASSERT(pReader != NULL); + + if (pBytesRead != NULL) { + *pBytesRead = 0; + } + + bytesRemaining = (pReader->sizeInBytes - pReader->cursor); + if (bytesToRead > bytesRemaining) { + bytesToRead = bytesRemaining; + } + + if (pDst == NULL) { + /* Seek. */ + result = drwav_buffer_reader_seek(pReader, bytesToRead); + } else { + /* Read. */ + DRWAV_COPY_MEMORY(pDst, drwav_buffer_reader_ptr(pReader), bytesToRead); + pReader->cursor += bytesToRead; + } + + DRWAV_ASSERT(pReader->cursor <= pReader->sizeInBytes); + + if (result == DRWAV_SUCCESS) { + if (pBytesRead != NULL) { + *pBytesRead = bytesToRead; + } + } + + return DRWAV_SUCCESS; +} + +DRWAV_PRIVATE drwav_result drwav_buffer_reader_read_u16(drwav_buffer_reader* pReader, drwav_uint16* pDst) +{ + drwav_result result; + size_t bytesRead; + drwav_uint8 data[2]; + + DRWAV_ASSERT(pReader != NULL); + DRWAV_ASSERT(pDst != NULL); + + *pDst = 0; /* Safety. */ + + result = drwav_buffer_reader_read(pReader, data, sizeof(*pDst), &bytesRead); + if (result != DRWAV_SUCCESS || bytesRead != sizeof(*pDst)) { + return result; + } + + *pDst = drwav_bytes_to_u16(data); + + return DRWAV_SUCCESS; +} + +DRWAV_PRIVATE drwav_result drwav_buffer_reader_read_u32(drwav_buffer_reader* pReader, drwav_uint32* pDst) +{ + drwav_result result; + size_t bytesRead; + drwav_uint8 data[4]; + + DRWAV_ASSERT(pReader != NULL); + DRWAV_ASSERT(pDst != NULL); + + *pDst = 0; /* Safety. */ + + result = drwav_buffer_reader_read(pReader, data, sizeof(*pDst), &bytesRead); + if (result != DRWAV_SUCCESS || bytesRead != sizeof(*pDst)) { + return result; + } + + *pDst = drwav_bytes_to_u32(data); + + return DRWAV_SUCCESS; +} + + + +DRWAV_PRIVATE drwav_uint64 drwav__read_bext_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata, drwav_uint64 chunkSize) +{ + drwav_uint8 bextData[DRWAV_BEXT_BYTES]; + size_t bytesRead = drwav__metadata_parser_read(pParser, bextData, sizeof(bextData), NULL); + + DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read); + + if (bytesRead == sizeof(bextData)) { + drwav_buffer_reader reader; + drwav_uint32 timeReferenceLow; + drwav_uint32 timeReferenceHigh; + size_t extraBytes; + + pMetadata->type = drwav_metadata_type_bext; + + if (drwav_buffer_reader_init(bextData, bytesRead, &reader) == DRWAV_SUCCESS) { + pMetadata->data.bext.pDescription = drwav__metadata_copy_string(pParser, (const char*)drwav_buffer_reader_ptr(&reader), DRWAV_BEXT_DESCRIPTION_BYTES); + drwav_buffer_reader_seek(&reader, DRWAV_BEXT_DESCRIPTION_BYTES); + + pMetadata->data.bext.pOriginatorName = drwav__metadata_copy_string(pParser, (const char*)drwav_buffer_reader_ptr(&reader), DRWAV_BEXT_ORIGINATOR_NAME_BYTES); + drwav_buffer_reader_seek(&reader, DRWAV_BEXT_ORIGINATOR_NAME_BYTES); + + pMetadata->data.bext.pOriginatorReference = drwav__metadata_copy_string(pParser, (const char*)drwav_buffer_reader_ptr(&reader), DRWAV_BEXT_ORIGINATOR_REF_BYTES); + drwav_buffer_reader_seek(&reader, DRWAV_BEXT_ORIGINATOR_REF_BYTES); + + drwav_buffer_reader_read(&reader, pMetadata->data.bext.pOriginationDate, sizeof(pMetadata->data.bext.pOriginationDate), NULL); + drwav_buffer_reader_read(&reader, pMetadata->data.bext.pOriginationTime, sizeof(pMetadata->data.bext.pOriginationTime), NULL); + + drwav_buffer_reader_read_u32(&reader, &timeReferenceLow); + drwav_buffer_reader_read_u32(&reader, &timeReferenceHigh); + pMetadata->data.bext.timeReference = ((drwav_uint64)timeReferenceHigh << 32) + timeReferenceLow; + + drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.version); + + pMetadata->data.bext.pUMID = drwav__metadata_get_memory(pParser, DRWAV_BEXT_UMID_BYTES, 1); + drwav_buffer_reader_read(&reader, pMetadata->data.bext.pUMID, DRWAV_BEXT_UMID_BYTES, NULL); + + drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.loudnessValue); + drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.loudnessRange); + drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.maxTruePeakLevel); + drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.maxMomentaryLoudness); + drwav_buffer_reader_read_u16(&reader, &pMetadata->data.bext.maxShortTermLoudness); + + DRWAV_ASSERT((drwav_offset_ptr(drwav_buffer_reader_ptr(&reader), DRWAV_BEXT_RESERVED_BYTES)) == (bextData + DRWAV_BEXT_BYTES)); + + extraBytes = (size_t)(chunkSize - DRWAV_BEXT_BYTES); + if (extraBytes > 0) { + pMetadata->data.bext.pCodingHistory = (char*)drwav__metadata_get_memory(pParser, extraBytes + 1, 1); + DRWAV_ASSERT(pMetadata->data.bext.pCodingHistory != NULL); + + pMetadata->data.bext.codingHistorySize = (drwav_uint32)drwav__metadata_parser_read(pParser, pMetadata->data.bext.pCodingHistory, extraBytes, NULL); + pMetadata->data.bext.pCodingHistory[pMetadata->data.bext.codingHistorySize] = '\0'; /* <-- Explicit null terminator in case of a badly formed file. */ + + bytesRead += pMetadata->data.bext.codingHistorySize; + } else { + pMetadata->data.bext.pCodingHistory = NULL; + pMetadata->data.bext.codingHistorySize = 0; + } + } + } + + return bytesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav__read_list_label_or_note_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata, drwav_uint64 chunkSize, drwav_metadata_type type) +{ + drwav_uint8 cueIDBuffer[DRWAV_LIST_LABEL_OR_NOTE_BYTES]; + drwav_uint64 totalBytesRead = 0; + size_t bytesJustRead = drwav__metadata_parser_read(pParser, cueIDBuffer, sizeof(cueIDBuffer), &totalBytesRead); + + DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read); + + if (bytesJustRead == sizeof(cueIDBuffer)) { + drwav_uint32 sizeIncludingNullTerminator; + + pMetadata->type = type; + pMetadata->data.labelOrNote.cuePointId = drwav_bytes_to_u32(cueIDBuffer); + + sizeIncludingNullTerminator = (drwav_uint32)chunkSize - DRWAV_LIST_LABEL_OR_NOTE_BYTES; + if (sizeIncludingNullTerminator > 0) { + pMetadata->data.labelOrNote.stringLength = sizeIncludingNullTerminator - 1; + pMetadata->data.labelOrNote.pString = (char*)drwav__metadata_get_memory(pParser, sizeIncludingNullTerminator, 1); + DRWAV_ASSERT(pMetadata->data.labelOrNote.pString != NULL); + + drwav__metadata_parser_read(pParser, pMetadata->data.labelOrNote.pString, sizeIncludingNullTerminator, &totalBytesRead); + } else { + pMetadata->data.labelOrNote.stringLength = 0; + pMetadata->data.labelOrNote.pString = NULL; + } + } + + return totalBytesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav__read_list_labelled_cue_region_to_metadata_obj(drwav__metadata_parser* pParser, drwav_metadata* pMetadata, drwav_uint64 chunkSize) +{ + drwav_uint8 buffer[DRWAV_LIST_LABELLED_TEXT_BYTES]; + drwav_uint64 totalBytesRead = 0; + size_t bytesJustRead = drwav__metadata_parser_read(pParser, buffer, sizeof(buffer), &totalBytesRead); + + DRWAV_ASSERT(pParser->stage == drwav__metadata_parser_stage_read); + + if (bytesJustRead == sizeof(buffer)) { + drwav_uint32 sizeIncludingNullTerminator; + + pMetadata->type = drwav_metadata_type_list_labelled_cue_region; + pMetadata->data.labelledCueRegion.cuePointId = drwav_bytes_to_u32(buffer + 0); + pMetadata->data.labelledCueRegion.sampleLength = drwav_bytes_to_u32(buffer + 4); + pMetadata->data.labelledCueRegion.purposeId[0] = buffer[8]; + pMetadata->data.labelledCueRegion.purposeId[1] = buffer[9]; + pMetadata->data.labelledCueRegion.purposeId[2] = buffer[10]; + pMetadata->data.labelledCueRegion.purposeId[3] = buffer[11]; + pMetadata->data.labelledCueRegion.country = drwav_bytes_to_u16(buffer + 12); + pMetadata->data.labelledCueRegion.language = drwav_bytes_to_u16(buffer + 14); + pMetadata->data.labelledCueRegion.dialect = drwav_bytes_to_u16(buffer + 16); + pMetadata->data.labelledCueRegion.codePage = drwav_bytes_to_u16(buffer + 18); + + sizeIncludingNullTerminator = (drwav_uint32)chunkSize - DRWAV_LIST_LABELLED_TEXT_BYTES; + if (sizeIncludingNullTerminator > 0) { + pMetadata->data.labelledCueRegion.stringLength = sizeIncludingNullTerminator - 1; + pMetadata->data.labelledCueRegion.pString = (char*)drwav__metadata_get_memory(pParser, sizeIncludingNullTerminator, 1); + DRWAV_ASSERT(pMetadata->data.labelledCueRegion.pString != NULL); + + drwav__metadata_parser_read(pParser, pMetadata->data.labelledCueRegion.pString, sizeIncludingNullTerminator, &totalBytesRead); + } else { + pMetadata->data.labelledCueRegion.stringLength = 0; + pMetadata->data.labelledCueRegion.pString = NULL; + } + } + + return totalBytesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav__metadata_process_info_text_chunk(drwav__metadata_parser* pParser, drwav_uint64 chunkSize, drwav_metadata_type type) +{ + drwav_uint64 bytesRead = 0; + drwav_uint32 stringSizeWithNullTerminator = (drwav_uint32)chunkSize; + + if (pParser->stage == drwav__metadata_parser_stage_count) { + pParser->metadataCount += 1; + drwav__metadata_request_extra_memory_for_stage_2(pParser, stringSizeWithNullTerminator, 1); + } else { + drwav_metadata* pMetadata = &pParser->pMetadata[pParser->metadataCursor]; + pMetadata->type = type; + if (stringSizeWithNullTerminator > 0) { + pMetadata->data.infoText.stringLength = stringSizeWithNullTerminator - 1; + pMetadata->data.infoText.pString = (char*)drwav__metadata_get_memory(pParser, stringSizeWithNullTerminator, 1); + DRWAV_ASSERT(pMetadata->data.infoText.pString != NULL); + + bytesRead = drwav__metadata_parser_read(pParser, pMetadata->data.infoText.pString, (size_t)stringSizeWithNullTerminator, NULL); + if (bytesRead == chunkSize) { + pParser->metadataCursor += 1; + } else { + /* Failed to parse. */ + } + } else { + pMetadata->data.infoText.stringLength = 0; + pMetadata->data.infoText.pString = NULL; + pParser->metadataCursor += 1; + } + } + + return bytesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav__metadata_process_unknown_chunk(drwav__metadata_parser* pParser, const drwav_uint8* pChunkId, drwav_uint64 chunkSize, drwav_metadata_location location) +{ + drwav_uint64 bytesRead = 0; + + if (location == drwav_metadata_location_invalid) { + return 0; + } + + if (drwav_fourcc_equal(pChunkId, "data") || drwav_fourcc_equal(pChunkId, "fmt ") || drwav_fourcc_equal(pChunkId, "fact")) { + return 0; + } + + if (pParser->stage == drwav__metadata_parser_stage_count) { + pParser->metadataCount += 1; + drwav__metadata_request_extra_memory_for_stage_2(pParser, (size_t)chunkSize, 1); + } else { + drwav_metadata* pMetadata = &pParser->pMetadata[pParser->metadataCursor]; + pMetadata->type = drwav_metadata_type_unknown; + pMetadata->data.unknown.chunkLocation = location; + pMetadata->data.unknown.id[0] = pChunkId[0]; + pMetadata->data.unknown.id[1] = pChunkId[1]; + pMetadata->data.unknown.id[2] = pChunkId[2]; + pMetadata->data.unknown.id[3] = pChunkId[3]; + pMetadata->data.unknown.dataSizeInBytes = (drwav_uint32)chunkSize; + pMetadata->data.unknown.pData = (drwav_uint8 *)drwav__metadata_get_memory(pParser, (size_t)chunkSize, 1); + DRWAV_ASSERT(pMetadata->data.unknown.pData != NULL); + + bytesRead = drwav__metadata_parser_read(pParser, pMetadata->data.unknown.pData, pMetadata->data.unknown.dataSizeInBytes, NULL); + if (bytesRead == pMetadata->data.unknown.dataSizeInBytes) { + pParser->metadataCursor += 1; + } else { + /* Failed to read. */ + } + } + + return bytesRead; +} + +DRWAV_PRIVATE drwav_bool32 drwav__chunk_matches(drwav_metadata_type allowedMetadataTypes, const drwav_uint8* pChunkID, drwav_metadata_type type, const char* pID) +{ + return (allowedMetadataTypes & type) && drwav_fourcc_equal(pChunkID, pID); +} + +DRWAV_PRIVATE drwav_uint64 drwav__metadata_process_chunk(drwav__metadata_parser* pParser, const drwav_chunk_header* pChunkHeader, drwav_metadata_type allowedMetadataTypes) +{ + const drwav_uint8 *pChunkID = pChunkHeader->id.fourcc; + drwav_uint64 bytesRead = 0; + + if (drwav__chunk_matches(allowedMetadataTypes, pChunkID, drwav_metadata_type_smpl, "smpl")) { + if (pChunkHeader->sizeInBytes >= DRWAV_SMPL_BYTES) { + if (pParser->stage == drwav__metadata_parser_stage_count) { + drwav_uint8 buffer[4]; + size_t bytesJustRead; + + if (!pParser->onSeek(pParser->pReadSeekUserData, 28, DRWAV_SEEK_CUR)) { + return bytesRead; + } + bytesRead += 28; + + bytesJustRead = drwav__metadata_parser_read(pParser, buffer, sizeof(buffer), &bytesRead); + if (bytesJustRead == sizeof(buffer)) { + drwav_uint32 loopCount = drwav_bytes_to_u32(buffer); + + bytesJustRead = drwav__metadata_parser_read(pParser, buffer, sizeof(buffer), &bytesRead); + if (bytesJustRead == sizeof(buffer)) { + drwav_uint32 samplerSpecificDataSizeInBytes = drwav_bytes_to_u32(buffer); + + if (drwav__metadata_validate_smpl_chunk(pChunkHeader, loopCount, samplerSpecificDataSizeInBytes, NULL)) { + pParser->metadataCount += 1; + drwav__metadata_request_extra_memory_for_stage_2(pParser, sizeof(drwav_smpl_loop) * loopCount, DRWAV_METADATA_ALIGNMENT); + drwav__metadata_request_extra_memory_for_stage_2(pParser, samplerSpecificDataSizeInBytes, 1); + } else { + /* Incorrectly formed chunk. Loop or sampler-specific data exceeds the size of the chunk. */ + } + } + } + } else { + bytesRead = drwav__read_smpl_to_metadata_obj(pParser, pChunkHeader, &pParser->pMetadata[pParser->metadataCursor]); + if (bytesRead == pChunkHeader->sizeInBytes) { + pParser->metadataCursor += 1; + } else { + /* Failed to parse. */ + } + } + } else { + /* Incorrectly formed chunk. */ + } + } else if (drwav__chunk_matches(allowedMetadataTypes, pChunkID, drwav_metadata_type_inst, "inst")) { + if (pChunkHeader->sizeInBytes == DRWAV_INST_BYTES) { + if (pParser->stage == drwav__metadata_parser_stage_count) { + pParser->metadataCount += 1; + } else { + bytesRead = drwav__read_inst_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor]); + if (bytesRead == pChunkHeader->sizeInBytes) { + pParser->metadataCursor += 1; + } else { + /* Failed to parse. */ + } + } + } else { + /* Incorrectly formed chunk. */ + } + } else if (drwav__chunk_matches(allowedMetadataTypes, pChunkID, drwav_metadata_type_acid, "acid")) { + if (pChunkHeader->sizeInBytes == DRWAV_ACID_BYTES) { + if (pParser->stage == drwav__metadata_parser_stage_count) { + pParser->metadataCount += 1; + } else { + bytesRead = drwav__read_acid_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor]); + if (bytesRead == pChunkHeader->sizeInBytes) { + pParser->metadataCursor += 1; + } else { + /* Failed to parse. */ + } + } + } else { + /* Incorrectly formed chunk. */ + } + } else if (drwav__chunk_matches(allowedMetadataTypes, pChunkID, drwav_metadata_type_cue, "cue ")) { + if (pChunkHeader->sizeInBytes >= DRWAV_CUE_BYTES) { + if (pParser->stage == drwav__metadata_parser_stage_count) { + size_t cueCount; + + pParser->metadataCount += 1; + cueCount = (size_t)(pChunkHeader->sizeInBytes - DRWAV_CUE_BYTES) / DRWAV_CUE_POINT_BYTES; + drwav__metadata_request_extra_memory_for_stage_2(pParser, sizeof(drwav_cue_point) * cueCount, DRWAV_METADATA_ALIGNMENT); + } else { + bytesRead = drwav__read_cue_to_metadata_obj(pParser, pChunkHeader, &pParser->pMetadata[pParser->metadataCursor]); + if (bytesRead == pChunkHeader->sizeInBytes) { + pParser->metadataCursor += 1; + } else { + /* Failed to parse. */ + } + } + } else { + /* Incorrectly formed chunk. */ + } + } else if (drwav__chunk_matches(allowedMetadataTypes, pChunkID, drwav_metadata_type_bext, "bext")) { + if (pChunkHeader->sizeInBytes >= DRWAV_BEXT_BYTES) { + if (pParser->stage == drwav__metadata_parser_stage_count) { + /* The description field is the largest one in a bext chunk, so that is the max size of this temporary buffer. */ + char buffer[DRWAV_BEXT_DESCRIPTION_BYTES + 1]; + size_t allocSizeNeeded = DRWAV_BEXT_UMID_BYTES; /* We know we will need SMPTE umid size. */ + size_t bytesJustRead; + + buffer[DRWAV_BEXT_DESCRIPTION_BYTES] = '\0'; + bytesJustRead = drwav__metadata_parser_read(pParser, buffer, DRWAV_BEXT_DESCRIPTION_BYTES, &bytesRead); + if (bytesJustRead != DRWAV_BEXT_DESCRIPTION_BYTES) { + return bytesRead; + } + allocSizeNeeded += drwav__strlen(buffer) + 1; + + buffer[DRWAV_BEXT_ORIGINATOR_NAME_BYTES] = '\0'; + bytesJustRead = drwav__metadata_parser_read(pParser, buffer, DRWAV_BEXT_ORIGINATOR_NAME_BYTES, &bytesRead); + if (bytesJustRead != DRWAV_BEXT_ORIGINATOR_NAME_BYTES) { + return bytesRead; + } + allocSizeNeeded += drwav__strlen(buffer) + 1; + + buffer[DRWAV_BEXT_ORIGINATOR_REF_BYTES] = '\0'; + bytesJustRead = drwav__metadata_parser_read(pParser, buffer, DRWAV_BEXT_ORIGINATOR_REF_BYTES, &bytesRead); + if (bytesJustRead != DRWAV_BEXT_ORIGINATOR_REF_BYTES) { + return bytesRead; + } + allocSizeNeeded += drwav__strlen(buffer) + 1; + + /* Coding history. */ + allocSizeNeeded += (size_t)pChunkHeader->sizeInBytes - DRWAV_BEXT_BYTES + 1; + + drwav__metadata_request_extra_memory_for_stage_2(pParser, allocSizeNeeded, 1); + + pParser->metadataCount += 1; + } else { + bytesRead = drwav__read_bext_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor], pChunkHeader->sizeInBytes); + if (bytesRead == pChunkHeader->sizeInBytes) { + pParser->metadataCursor += 1; + } else { + /* Failed to parse. */ + } + } + } else { + /* Incorrectly formed chunk. */ + } + } else if (drwav_fourcc_equal(pChunkID, "LIST") || drwav_fourcc_equal(pChunkID, "list")) { + drwav_metadata_location listType = drwav_metadata_location_invalid; + while (bytesRead < pChunkHeader->sizeInBytes) { + drwav_uint8 subchunkId[4]; + drwav_uint8 subchunkSizeBuffer[4]; + drwav_uint64 subchunkDataSize; + drwav_uint64 subchunkBytesRead = 0; + drwav_uint64 bytesJustRead = drwav__metadata_parser_read(pParser, subchunkId, sizeof(subchunkId), &bytesRead); + if (bytesJustRead != sizeof(subchunkId)) { + break; + } + + /* + The first thing in a list chunk should be "adtl" or "INFO". + + - adtl means this list is a Associated Data List Chunk and will contain labels, notes + or labelled cue regions. + - INFO means this list is an Info List Chunk containing info text chunks such as IPRD + which would specifies the album of this wav file. + + No data follows the adtl or INFO id so we just make note of what type this list is and + continue. + */ + if (drwav_fourcc_equal(subchunkId, "adtl")) { + listType = drwav_metadata_location_inside_adtl_list; + continue; + } else if (drwav_fourcc_equal(subchunkId, "INFO")) { + listType = drwav_metadata_location_inside_info_list; + continue; + } + + bytesJustRead = drwav__metadata_parser_read(pParser, subchunkSizeBuffer, sizeof(subchunkSizeBuffer), &bytesRead); + if (bytesJustRead != sizeof(subchunkSizeBuffer)) { + break; + } + subchunkDataSize = drwav_bytes_to_u32(subchunkSizeBuffer); + + if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_label, "labl") || drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_note, "note")) { + if (subchunkDataSize >= DRWAV_LIST_LABEL_OR_NOTE_BYTES) { + drwav_uint64 stringSizeWithNullTerm = subchunkDataSize - DRWAV_LIST_LABEL_OR_NOTE_BYTES; + if (pParser->stage == drwav__metadata_parser_stage_count) { + pParser->metadataCount += 1; + drwav__metadata_request_extra_memory_for_stage_2(pParser, (size_t)stringSizeWithNullTerm, 1); + } else { + subchunkBytesRead = drwav__read_list_label_or_note_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor], subchunkDataSize, drwav_fourcc_equal(subchunkId, "labl") ? drwav_metadata_type_list_label : drwav_metadata_type_list_note); + if (subchunkBytesRead == subchunkDataSize) { + pParser->metadataCursor += 1; + } else { + /* Failed to parse. */ + } + } + } else { + /* Incorrectly formed chunk. */ + } + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_labelled_cue_region, "ltxt")) { + if (subchunkDataSize >= DRWAV_LIST_LABELLED_TEXT_BYTES) { + drwav_uint64 stringSizeWithNullTerminator = subchunkDataSize - DRWAV_LIST_LABELLED_TEXT_BYTES; + if (pParser->stage == drwav__metadata_parser_stage_count) { + pParser->metadataCount += 1; + drwav__metadata_request_extra_memory_for_stage_2(pParser, (size_t)stringSizeWithNullTerminator, 1); + } else { + subchunkBytesRead = drwav__read_list_labelled_cue_region_to_metadata_obj(pParser, &pParser->pMetadata[pParser->metadataCursor], subchunkDataSize); + if (subchunkBytesRead == subchunkDataSize) { + pParser->metadataCursor += 1; + } else { + /* Failed to parse. */ + } + } + } else { + /* Incorrectly formed chunk. */ + } + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_software, "ISFT")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_software); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_copyright, "ICOP")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_copyright); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_title, "INAM")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_title); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_artist, "IART")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_artist); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_comment, "ICMT")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_comment); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_date, "ICRD")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_date); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_genre, "IGNR")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_genre); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_album, "IPRD")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_album); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_tracknumber, "ITRK")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_tracknumber); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_location, "IARL")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_location); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_organization, "ICMS")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_organization); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_keywords, "IKEY")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_keywords); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_medium, "IMED")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_medium); + } else if (drwav__chunk_matches(allowedMetadataTypes, subchunkId, drwav_metadata_type_list_info_description, "ISBJ")) { + subchunkBytesRead = drwav__metadata_process_info_text_chunk(pParser, subchunkDataSize, drwav_metadata_type_list_info_description); + } else if ((allowedMetadataTypes & drwav_metadata_type_unknown) != 0) { + subchunkBytesRead = drwav__metadata_process_unknown_chunk(pParser, subchunkId, subchunkDataSize, listType); + } + + bytesRead += subchunkBytesRead; + DRWAV_ASSERT(subchunkBytesRead <= subchunkDataSize); + + if (subchunkBytesRead < subchunkDataSize) { + drwav_uint64 bytesToSeek = subchunkDataSize - subchunkBytesRead; + + if (!pParser->onSeek(pParser->pReadSeekUserData, (int)bytesToSeek, DRWAV_SEEK_CUR)) { + break; + } + bytesRead += bytesToSeek; + } + + if ((subchunkDataSize % 2) == 1) { + if (!pParser->onSeek(pParser->pReadSeekUserData, 1, DRWAV_SEEK_CUR)) { + break; + } + bytesRead += 1; + } + } + } else if ((allowedMetadataTypes & drwav_metadata_type_unknown) != 0) { + bytesRead = drwav__metadata_process_unknown_chunk(pParser, pChunkID, pChunkHeader->sizeInBytes, drwav_metadata_location_top_level); + } + + return bytesRead; +} + + +DRWAV_PRIVATE drwav_uint32 drwav_get_bytes_per_pcm_frame(drwav* pWav) +{ + drwav_uint32 bytesPerFrame; + + /* + The bytes per frame is a bit ambiguous. It can be either be based on the bits per sample, or the block align. The way I'm doing it here + is that if the bits per sample is a multiple of 8, use floor(bitsPerSample*channels/8), otherwise fall back to the block align. + */ + if ((pWav->bitsPerSample & 0x7) == 0) { + /* Bits per sample is a multiple of 8. */ + bytesPerFrame = (pWav->bitsPerSample * pWav->fmt.channels) >> 3; + } else { + bytesPerFrame = pWav->fmt.blockAlign; + } + + /* Validation for known formats. a-law and mu-law should be 1 byte per channel. If it's not, it's not decodable. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW || pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) { + if (bytesPerFrame != pWav->fmt.channels) { + return 0; /* Invalid file. */ + } + } + + return bytesPerFrame; +} + +DRWAV_API drwav_uint16 drwav_fmt_get_format(const drwav_fmt* pFMT) +{ + if (pFMT == NULL) { + return 0; + } + + if (pFMT->formatTag != DR_WAVE_FORMAT_EXTENSIBLE) { + return pFMT->formatTag; + } else { + return drwav_bytes_to_u16(pFMT->subFormat); /* Only the first two bytes are required. */ + } +} + +DRWAV_PRIVATE drwav_bool32 drwav_preinit(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pReadSeekTellUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pWav == NULL || onRead == NULL || onSeek == NULL) { /* <-- onTell is optional. */ + return DRWAV_FALSE; + } + + DRWAV_ZERO_MEMORY(pWav, sizeof(*pWav)); + pWav->onRead = onRead; + pWav->onSeek = onSeek; + pWav->onTell = onTell; + pWav->pUserData = pReadSeekTellUserData; + pWav->allocationCallbacks = drwav_copy_allocation_callbacks_or_defaults(pAllocationCallbacks); + + if (pWav->allocationCallbacks.onFree == NULL || (pWav->allocationCallbacks.onMalloc == NULL && pWav->allocationCallbacks.onRealloc == NULL)) { + return DRWAV_FALSE; /* Invalid allocation callbacks. */ + } + + return DRWAV_TRUE; +} + +DRWAV_PRIVATE drwav_bool32 drwav_init__internal(drwav* pWav, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags) +{ + /* This function assumes drwav_preinit() has been called beforehand. */ + drwav_result result; + drwav_uint64 cursor; /* <-- Keeps track of the byte position so we can seek to specific locations. */ + drwav_bool32 sequential; + drwav_uint8 riff[4]; + drwav_fmt fmt; + unsigned short translatedFormatTag; + drwav_uint64 dataChunkSize = 0; /* <-- Important! Don't explicitly set this to 0 anywhere else. Calculation of the size of the data chunk is performed in different paths depending on the container. */ + drwav_uint64 sampleCountFromFactChunk = 0; /* Same as dataChunkSize - make sure this is the only place this is initialized to 0. */ + drwav_uint64 metadataStartPos; + drwav__metadata_parser metadataParser; + drwav_bool8 isProcessingMetadata = DRWAV_FALSE; + drwav_bool8 foundChunk_fmt = DRWAV_FALSE; + drwav_bool8 foundChunk_data = DRWAV_FALSE; + drwav_bool8 isAIFCFormType = DRWAV_FALSE; /* Only used with AIFF. */ + drwav_uint64 aiffFrameCount = 0; + drwav_int64 fileSize; + drwav_bool32 hasKnownFileSize = DRWAV_FALSE; + + cursor = 0; + sequential = (flags & DRWAV_SEQUENTIAL) != 0; + + /* + Whether or not we are processing metadata controls how we load. We can load more efficiently when + metadata is not being processed. Seqential mode cannot support metadata because it involves seeking + backwards. + */ + isProcessingMetadata = !sequential && ((flags & DRWAV_WITH_METADATA) != 0); + + /* + When processing metadata we'll be doing some memory allocations here against untrusted data. We'll do + a basic validation check that they don't exceed the size of the file. + */ + if (isProcessingMetadata && pWav->onTell != NULL && pWav->onSeek != NULL) { + if (pWav->onSeek(pWav->pUserData, 0, DRWAV_SEEK_END)) { + if (pWav->onTell(pWav->pUserData, &fileSize)) { + hasKnownFileSize = DRWAV_TRUE; + } + + pWav->onSeek(pWav->pUserData, 0, DRWAV_SEEK_SET); + } + } + + DRWAV_ZERO_OBJECT(&fmt); + + /* The first 4 bytes should be the RIFF identifier. */ + if (drwav__on_read(pWav->onRead, pWav->pUserData, riff, sizeof(riff), &cursor) != sizeof(riff)) { + return DRWAV_FALSE; + } + + /* + The first 4 bytes can be used to identify the container. For RIFF files it will start with "RIFF" and for + w64 it will start with "riff". + */ + if (drwav_fourcc_equal(riff, "RIFF")) { + pWav->container = drwav_container_riff; + } else if (drwav_fourcc_equal(riff, "RIFX")) { + pWav->container = drwav_container_rifx; + } else if (drwav_fourcc_equal(riff, "riff")) { + int i; + drwav_uint8 riff2[12]; + + pWav->container = drwav_container_w64; + + /* Check the rest of the GUID for validity. */ + if (drwav__on_read(pWav->onRead, pWav->pUserData, riff2, sizeof(riff2), &cursor) != sizeof(riff2)) { + return DRWAV_FALSE; + } + + for (i = 0; i < 12; ++i) { + if (riff2[i] != drwavGUID_W64_RIFF[i+4]) { + return DRWAV_FALSE; + } + } + } else if (drwav_fourcc_equal(riff, "RF64")) { + pWav->container = drwav_container_rf64; + } else if (drwav_fourcc_equal(riff, "FORM")) { + pWav->container = drwav_container_aiff; + } else { + return DRWAV_FALSE; /* Unknown or unsupported container. */ + } + + + if (pWav->container == drwav_container_riff || pWav->container == drwav_container_rifx || pWav->container == drwav_container_rf64) { + drwav_uint8 chunkSizeBytes[4]; + drwav_uint8 wave[4]; + + if (drwav__on_read(pWav->onRead, pWav->pUserData, chunkSizeBytes, sizeof(chunkSizeBytes), &cursor) != sizeof(chunkSizeBytes)) { + return DRWAV_FALSE; + } + + if (pWav->container == drwav_container_riff || pWav->container == drwav_container_rifx) { + if (drwav_bytes_to_u32_ex(chunkSizeBytes, pWav->container) < 36) { + /* + I've had a report of a WAV file failing to load when the size of the WAVE chunk is not encoded + and is instead just set to 0. I'm going to relax the validation here to allow these files to + load. Considering the chunk size isn't actually used this should be safe. With this change my + test suite still passes. + */ + /*return DRWAV_FALSE;*/ /* Chunk size should always be at least 36 bytes. */ + } + } else if (pWav->container == drwav_container_rf64) { + if (drwav_bytes_to_u32_le(chunkSizeBytes) != 0xFFFFFFFF) { + return DRWAV_FALSE; /* Chunk size should always be set to -1/0xFFFFFFFF for RF64. The actual size is retrieved later. */ + } + } else { + return DRWAV_FALSE; /* Should never hit this. */ + } + + if (drwav__on_read(pWav->onRead, pWav->pUserData, wave, sizeof(wave), &cursor) != sizeof(wave)) { + return DRWAV_FALSE; + } + + if (!drwav_fourcc_equal(wave, "WAVE")) { + return DRWAV_FALSE; /* Expecting "WAVE". */ + } + } else if (pWav->container == drwav_container_w64) { + drwav_uint8 chunkSizeBytes[8]; + drwav_uint8 wave[16]; + + if (drwav__on_read(pWav->onRead, pWav->pUserData, chunkSizeBytes, sizeof(chunkSizeBytes), &cursor) != sizeof(chunkSizeBytes)) { + return DRWAV_FALSE; + } + + if (drwav_bytes_to_u64(chunkSizeBytes) < 80) { + return DRWAV_FALSE; + } + + if (drwav__on_read(pWav->onRead, pWav->pUserData, wave, sizeof(wave), &cursor) != sizeof(wave)) { + return DRWAV_FALSE; + } + + if (!drwav_guid_equal(wave, drwavGUID_W64_WAVE)) { + return DRWAV_FALSE; + } + } else if (pWav->container == drwav_container_aiff) { + drwav_uint8 chunkSizeBytes[4]; + drwav_uint8 aiff[4]; + + if (drwav__on_read(pWav->onRead, pWav->pUserData, chunkSizeBytes, sizeof(chunkSizeBytes), &cursor) != sizeof(chunkSizeBytes)) { + return DRWAV_FALSE; + } + + if (drwav_bytes_to_u32_be(chunkSizeBytes) < 18) { + return DRWAV_FALSE; + } + + if (drwav__on_read(pWav->onRead, pWav->pUserData, aiff, sizeof(aiff), &cursor) != sizeof(aiff)) { + return DRWAV_FALSE; + } + + if (drwav_fourcc_equal(aiff, "AIFF")) { + isAIFCFormType = DRWAV_FALSE; + } else if (drwav_fourcc_equal(aiff, "AIFC")) { + isAIFCFormType = DRWAV_TRUE; + } else { + return DRWAV_FALSE; /* Expecting "AIFF" or "AIFC". */ + } + } else { + return DRWAV_FALSE; + } + + + /* For RF64, the "ds64" chunk must come next, before the "fmt " chunk. */ + if (pWav->container == drwav_container_rf64) { + drwav_uint8 sizeBytes[8]; + drwav_uint64 bytesRemainingInChunk; + drwav_chunk_header header; + result = drwav__read_chunk_header(pWav->onRead, pWav->pUserData, pWav->container, &cursor, &header); + if (result != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + if (!drwav_fourcc_equal(header.id.fourcc, "ds64")) { + return DRWAV_FALSE; /* Expecting "ds64". */ + } + + bytesRemainingInChunk = header.sizeInBytes + header.paddingSize; + + /* We don't care about the size of the RIFF chunk - skip it. */ + if (!drwav__seek_forward(pWav->onSeek, 8, pWav->pUserData)) { + return DRWAV_FALSE; + } + bytesRemainingInChunk -= 8; + cursor += 8; + + + /* Next 8 bytes is the size of the "data" chunk. */ + if (drwav__on_read(pWav->onRead, pWav->pUserData, sizeBytes, sizeof(sizeBytes), &cursor) != sizeof(sizeBytes)) { + return DRWAV_FALSE; + } + bytesRemainingInChunk -= 8; + dataChunkSize = drwav_bytes_to_u64(sizeBytes); + + + /* Next 8 bytes is the same count which we would usually derived from the FACT chunk if it was available. */ + if (drwav__on_read(pWav->onRead, pWav->pUserData, sizeBytes, sizeof(sizeBytes), &cursor) != sizeof(sizeBytes)) { + return DRWAV_FALSE; + } + bytesRemainingInChunk -= 8; + sampleCountFromFactChunk = drwav_bytes_to_u64(sizeBytes); + + + /* Skip over everything else. */ + if (!drwav__seek_forward(pWav->onSeek, bytesRemainingInChunk, pWav->pUserData)) { + return DRWAV_FALSE; + } + cursor += bytesRemainingInChunk; + } + + + metadataStartPos = cursor; + + /* Don't allow processing of metadata with untested containers. */ + if (pWav->container != drwav_container_riff && pWav->container != drwav_container_rf64) { + isProcessingMetadata = DRWAV_FALSE; + } + + DRWAV_ZERO_MEMORY(&metadataParser, sizeof(metadataParser)); + if (isProcessingMetadata) { + metadataParser.onRead = pWav->onRead; + metadataParser.onSeek = pWav->onSeek; + metadataParser.pReadSeekUserData = pWav->pUserData; + metadataParser.stage = drwav__metadata_parser_stage_count; + } + + + /* + From here on out, chunks might be in any order. In order to robustly handle metadata we'll need + to loop through every chunk and handle them as we find them. In sequential mode we need to get + out of the loop as soon as we find the data chunk because we won't be able to seek back. + */ + for (;;) { /* For each chunk... */ + drwav_chunk_header header; + drwav_uint64 chunkSize; + + result = drwav__read_chunk_header(pWav->onRead, pWav->pUserData, pWav->container, &cursor, &header); + if (result != DRWAV_SUCCESS) { + break; + } + + chunkSize = header.sizeInBytes; + + + /* + Always tell the caller about this chunk. We cannot do this in sequential mode because the + callback is allowed to read from the file, in which case we'll need to rewind. + */ + if (!sequential && onChunk != NULL) { + drwav_uint64 callbackBytesRead = onChunk(pChunkUserData, pWav->onRead, pWav->onSeek, pWav->pUserData, &header, pWav->container, &fmt); + + /* + dr_wav may need to read the contents of the chunk, so we now need to seek back to the position before + we called the callback. + */ + if (callbackBytesRead > 0) { + if (drwav__seek_from_start(pWav->onSeek, cursor, pWav->pUserData) == DRWAV_FALSE) { + return DRWAV_FALSE; + } + } + } + + + /* Explicitly handle known chunks first. */ + + /* "fmt " */ + if (((pWav->container == drwav_container_riff || pWav->container == drwav_container_rifx || pWav->container == drwav_container_rf64) && drwav_fourcc_equal(header.id.fourcc, "fmt ")) || + ((pWav->container == drwav_container_w64) && drwav_guid_equal(header.id.guid, drwavGUID_W64_FMT))) { + drwav_uint8 fmtData[16]; + + if (header.sizeInBytes < sizeof(fmtData)) { + return DRWAV_FALSE; /* Invalid fmt chunk. */ + } + + foundChunk_fmt = DRWAV_TRUE; + + if (pWav->onRead(pWav->pUserData, fmtData, sizeof(fmtData)) != sizeof(fmtData)) { + return DRWAV_FALSE; + } + cursor += sizeof(fmtData); + + fmt.formatTag = drwav_bytes_to_u16_ex(fmtData + 0, pWav->container); + fmt.channels = drwav_bytes_to_u16_ex(fmtData + 2, pWav->container); + fmt.sampleRate = drwav_bytes_to_u32_ex(fmtData + 4, pWav->container); + fmt.avgBytesPerSec = drwav_bytes_to_u32_ex(fmtData + 8, pWav->container); + fmt.blockAlign = drwav_bytes_to_u16_ex(fmtData + 12, pWav->container); + fmt.bitsPerSample = drwav_bytes_to_u16_ex(fmtData + 14, pWav->container); + + fmt.extendedSize = 0; + fmt.validBitsPerSample = 0; + fmt.channelMask = 0; + DRWAV_ZERO_MEMORY(fmt.subFormat, sizeof(fmt.subFormat)); + + if (header.sizeInBytes > 16) { + drwav_uint8 fmt_cbSize[2]; + int bytesReadSoFar = 0; + drwav_uint64 leftoverBytes; + + if (pWav->onRead(pWav->pUserData, fmt_cbSize, sizeof(fmt_cbSize)) != sizeof(fmt_cbSize)) { + return DRWAV_FALSE; /* Expecting more data. */ + } + cursor += sizeof(fmt_cbSize); + + bytesReadSoFar = 18; + + fmt.extendedSize = drwav_bytes_to_u16_ex(fmt_cbSize, pWav->container); + if (fmt.extendedSize > 0) { + /* Simple validation. */ + if (fmt.formatTag == DR_WAVE_FORMAT_EXTENSIBLE) { + if (fmt.extendedSize != 22) { + return DRWAV_FALSE; + } + } + + if (fmt.formatTag == DR_WAVE_FORMAT_EXTENSIBLE) { + drwav_uint8 fmtext[22]; + + if (pWav->onRead(pWav->pUserData, fmtext, fmt.extendedSize) != fmt.extendedSize) { + return DRWAV_FALSE; /* Expecting more data. */ + } + + fmt.validBitsPerSample = drwav_bytes_to_u16_ex(fmtext + 0, pWav->container); + fmt.channelMask = drwav_bytes_to_u32_ex(fmtext + 2, pWav->container); + drwav_bytes_to_guid(fmtext + 6, fmt.subFormat); + } else { + if (pWav->onSeek(pWav->pUserData, fmt.extendedSize, DRWAV_SEEK_CUR) == DRWAV_FALSE) { + return DRWAV_FALSE; + } + } + cursor += fmt.extendedSize; + + bytesReadSoFar += fmt.extendedSize; + } + + /* Seek past any leftover bytes. For w64 the leftover will be defined based on the chunk size. */ + leftoverBytes = header.sizeInBytes - bytesReadSoFar; + if (leftoverBytes > 0x7FFFFFFF) { + return DRWAV_FALSE; + } + + if (pWav->onSeek(pWav->pUserData, (int)leftoverBytes, DRWAV_SEEK_CUR) == DRWAV_FALSE) { + return DRWAV_FALSE; + } + + cursor += leftoverBytes; + } + + if (header.paddingSize > 0) { + if (drwav__seek_forward(pWav->onSeek, header.paddingSize, pWav->pUserData) == DRWAV_FALSE) { + break; + } + cursor += header.paddingSize; + } + + /* Go to the next chunk. Don't include this chunk in metadata. */ + continue; + } + + /* "data" */ + if (((pWav->container == drwav_container_riff || pWav->container == drwav_container_rifx || pWav->container == drwav_container_rf64) && drwav_fourcc_equal(header.id.fourcc, "data")) || + ((pWav->container == drwav_container_w64) && drwav_guid_equal(header.id.guid, drwavGUID_W64_DATA))) { + foundChunk_data = DRWAV_TRUE; + + pWav->dataChunkDataPos = cursor; + + if (pWav->container != drwav_container_rf64) { /* The data chunk size for RF64 will always be set to 0xFFFFFFFF here. It was set to it's true value earlier. */ + dataChunkSize = chunkSize; + } + + /* If we're running in sequential mode, or we're not reading metadata, we have enough now that we can get out of the loop. */ + if (sequential || !isProcessingMetadata) { + break; /* No need to keep reading beyond the data chunk. */ + } else { + chunkSize += header.paddingSize; /* <-- Make sure we seek past the padding. */ + if (drwav__seek_forward(pWav->onSeek, chunkSize, pWav->pUserData) == DRWAV_FALSE) { + break; + } + cursor += chunkSize; + + continue; /* There may be some more metadata to read. */ + } + } + + /* "fact". This is optional. Can use this to get the sample count which is useful for compressed formats. For RF64 we retrieved the sample count from the ds64 chunk earlier. */ + if (((pWav->container == drwav_container_riff || pWav->container == drwav_container_rifx || pWav->container == drwav_container_rf64) && drwav_fourcc_equal(header.id.fourcc, "fact")) || + ((pWav->container == drwav_container_w64) && drwav_guid_equal(header.id.guid, drwavGUID_W64_FACT))) { + if (pWav->container == drwav_container_riff || pWav->container == drwav_container_rifx) { + drwav_uint8 sampleCount[4]; + + if (chunkSize < 4) { + return DRWAV_FALSE; + } + + if (drwav__on_read(pWav->onRead, pWav->pUserData, &sampleCount, 4, &cursor) != 4) { + return DRWAV_FALSE; + } + + chunkSize -= 4; + + /* + The sample count in the "fact" chunk is either unreliable, or I'm not understanding it properly. For now I am only enabling this + for Microsoft ADPCM formats. + */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + sampleCountFromFactChunk = drwav_bytes_to_u32_ex(sampleCount, pWav->container); + } else { + sampleCountFromFactChunk = 0; + } + } else if (pWav->container == drwav_container_w64) { + if (chunkSize < 8) { + return DRWAV_FALSE; + } + + if (drwav__on_read(pWav->onRead, pWav->pUserData, &sampleCountFromFactChunk, 8, &cursor) != 8) { + return DRWAV_FALSE; + } + + chunkSize -= 8; + } else if (pWav->container == drwav_container_rf64) { + /* We retrieved the sample count from the ds64 chunk earlier so no need to do that here. */ + } + + /* Seek to the next chunk in preparation for the next iteration. */ + chunkSize += header.paddingSize; /* <-- Make sure we seek past the padding. */ + if (drwav__seek_forward(pWav->onSeek, chunkSize, pWav->pUserData) == DRWAV_FALSE) { + break; + } + cursor += chunkSize; + + continue; + } + + + /* "COMM". AIFF/AIFC only. */ + if (pWav->container == drwav_container_aiff && drwav_fourcc_equal(header.id.fourcc, "COMM")) { + drwav_uint8 commData[24]; + drwav_uint32 commDataBytesToRead; + drwav_uint16 channels; + drwav_uint32 frameCount; + drwav_uint16 sampleSizeInBits; + drwav_int64 sampleRate; + drwav_uint16 compressionFormat; + + foundChunk_fmt = DRWAV_TRUE; + + if (isAIFCFormType) { + commDataBytesToRead = 24; + if (header.sizeInBytes < commDataBytesToRead) { + return DRWAV_FALSE; /* Invalid COMM chunk. */ + } + } else { + commDataBytesToRead = 18; + if (header.sizeInBytes != commDataBytesToRead) { + return DRWAV_FALSE; /* INVALID COMM chunk. */ + } + } + + if (drwav__on_read(pWav->onRead, pWav->pUserData, commData, commDataBytesToRead, &cursor) != commDataBytesToRead) { + return DRWAV_FALSE; + } + + + channels = drwav_bytes_to_u16_ex (commData + 0, pWav->container); + frameCount = drwav_bytes_to_u32_ex (commData + 2, pWav->container); + sampleSizeInBits = drwav_bytes_to_u16_ex (commData + 6, pWav->container); + sampleRate = drwav_aiff_extented_to_s64(commData + 8); + + if (sampleRate < 0 || sampleRate > 0xFFFFFFFF) { + return DRWAV_FALSE; /* Invalid sample rate. */ + } + + if (isAIFCFormType) { + const drwav_uint8* type = commData + 18; + + if (drwav_fourcc_equal(type, "NONE")) { + compressionFormat = DR_WAVE_FORMAT_PCM; /* PCM, big-endian. */ + } else if (drwav_fourcc_equal(type, "raw ")) { + compressionFormat = DR_WAVE_FORMAT_PCM; + + /* In my testing, it looks like when the "raw " compression type is used, 8-bit samples should be considered unsigned. */ + if (sampleSizeInBits == 8) { + pWav->aiff.isUnsigned = DRWAV_TRUE; + } + } else if (drwav_fourcc_equal(type, "sowt")) { + compressionFormat = DR_WAVE_FORMAT_PCM; /* PCM, little-endian. */ + pWav->aiff.isLE = DRWAV_TRUE; + } else if (drwav_fourcc_equal(type, "fl32") || drwav_fourcc_equal(type, "fl64") || drwav_fourcc_equal(type, "FL32") || drwav_fourcc_equal(type, "FL64")) { + compressionFormat = DR_WAVE_FORMAT_IEEE_FLOAT; + } else if (drwav_fourcc_equal(type, "alaw") || drwav_fourcc_equal(type, "ALAW")) { + compressionFormat = DR_WAVE_FORMAT_ALAW; + } else if (drwav_fourcc_equal(type, "ulaw") || drwav_fourcc_equal(type, "ULAW")) { + compressionFormat = DR_WAVE_FORMAT_MULAW; + } else if (drwav_fourcc_equal(type, "ima4")) { + compressionFormat = DR_WAVE_FORMAT_DVI_ADPCM; + sampleSizeInBits = 4; + + /* + I haven't been able to figure out how to get correct decoding for IMA ADPCM. Until this is figured out + we'll need to abort when we encounter such an encoding. Advice welcome! + */ + (void)compressionFormat; + (void)sampleSizeInBits; + + return DRWAV_FALSE; + } else { + return DRWAV_FALSE; /* Unknown or unsupported compression format. Need to abort. */ + } + } else { + compressionFormat = DR_WAVE_FORMAT_PCM; /* It's a standard AIFF form which is always compressed. */ + } + + /* With AIFF we want to use the explicitly defined frame count rather than deriving it from the size of the chunk. */ + aiffFrameCount = frameCount; + + /* We should now have enough information to fill out our fmt structure. */ + fmt.formatTag = compressionFormat; + fmt.channels = channels; + fmt.sampleRate = (drwav_uint32)sampleRate; + fmt.bitsPerSample = (sampleSizeInBits + 7) & ~7; /* In AIFF, samples are padded to 8-bit boundaries. We need to round up our bits per sample here. */ + fmt.blockAlign = (drwav_uint16)((drwav_uint32)fmt.channels * fmt.bitsPerSample / 8); + fmt.avgBytesPerSec = fmt.blockAlign * fmt.sampleRate; + + /* + Weird one. I've seen some alaw and ulaw encoded files that for some reason set the bits per sample to 16 when + it should be 8. To get this working I need to explicitly check for this and change it. + */ + if (compressionFormat == DR_WAVE_FORMAT_ALAW || compressionFormat == DR_WAVE_FORMAT_MULAW) { + if (fmt.bitsPerSample > 8) { + fmt.bitsPerSample = 8; + fmt.blockAlign = fmt.channels; + } + } + + /* If the form type is AIFC there will be some additional data in the chunk. We need to seek past it. */ + if (isAIFCFormType) { + if (drwav__seek_forward(pWav->onSeek, (chunkSize - commDataBytesToRead), pWav->pUserData) == DRWAV_FALSE) { + return DRWAV_FALSE; + } + cursor += (chunkSize - commDataBytesToRead); + } + + /* Don't fall through or else we'll end up treating this chunk as metadata which is incorrect. */ + continue; + } + + + /* "SSND". AIFF/AIFC only. This is the AIFF equivalent of the "data" chunk. */ + if (pWav->container == drwav_container_aiff && drwav_fourcc_equal(header.id.fourcc, "SSND")) { + drwav_uint8 offsetAndBlockSizeData[8]; + drwav_uint32 offset; + + foundChunk_data = DRWAV_TRUE; + + if (drwav__on_read(pWav->onRead, pWav->pUserData, offsetAndBlockSizeData, sizeof(offsetAndBlockSizeData), &cursor) != sizeof(offsetAndBlockSizeData)) { + return DRWAV_FALSE; + } + + /* The position of the audio data starts at an offset. */ + offset = drwav_bytes_to_u32_ex(offsetAndBlockSizeData + 0, pWav->container); + pWav->dataChunkDataPos = cursor + offset; + + /* The data chunk size needs to be reduced by the offset or else seeking will break. */ + dataChunkSize = chunkSize; + if (dataChunkSize > offset) { + dataChunkSize -= offset; + } else { + dataChunkSize = 0; + } + + if (sequential) { + if (foundChunk_fmt) { /* <-- Name is misleading, but will be set to true if the COMM chunk has been parsed. */ + /* + Getting here means we're opening in sequential mode and we've found the SSND (data) and COMM (fmt) chunks. We need + to get out of the loop here or else we'll end up going past the data chunk and will have no way of getting back to + it since we're not allowed to seek backwards. + + One subtle detail here is that there is an offset with the SSND chunk. We need to make sure we seek past this offset + so we're left sitting on the first byte of actual audio data. + */ + if (drwav__seek_forward(pWav->onSeek, offset, pWav->pUserData) == DRWAV_FALSE) { + return DRWAV_FALSE; + } + cursor += offset; + + break; + } else { + /* + Getting here means the COMM chunk was not found. In sequential mode, if we haven't yet found the COMM chunk + we'll need to abort because we can't be doing a backwards seek back to the SSND chunk in order to read the + data. For this reason, this configuration of AIFF files are not supported with sequential mode. + */ + return DRWAV_FALSE; + } + } else { + chunkSize += header.paddingSize; /* <-- Make sure we seek past the padding. */ + chunkSize -= sizeof(offsetAndBlockSizeData); /* <-- This was read earlier. */ + + if (drwav__seek_forward(pWav->onSeek, chunkSize, pWav->pUserData) == DRWAV_FALSE) { + break; + } + cursor += chunkSize; + + continue; /* There may be some more metadata to read. */ + } + } + + + /* Getting here means it's not a chunk that we care about internally, but might need to be handled as metadata by the caller. */ + if (isProcessingMetadata) { + if (hasKnownFileSize && header.sizeInBytes > (drwav_uint64)fileSize) { + return DRWAV_FALSE; + } + + drwav__metadata_process_chunk(&metadataParser, &header, drwav_metadata_type_all_including_unknown); + + /* Go back to the start of the chunk so we can normalize the position of the cursor. */ + if (drwav__seek_from_start(pWav->onSeek, cursor, pWav->pUserData) == DRWAV_FALSE) { + break; /* Failed to seek. Can't reliable read the remaining chunks. Get out. */ + } + } + + + /* Make sure we skip past the content of this chunk before we go to the next one. */ + chunkSize += header.paddingSize; /* <-- Make sure we seek past the padding. */ + if (drwav__seek_forward(pWav->onSeek, chunkSize, pWav->pUserData) == DRWAV_FALSE) { + break; + } + cursor += chunkSize; + } + + /* There's some mandatory chunks that must exist. If they were not found in the iteration above we must abort. */ + if (!foundChunk_fmt || !foundChunk_data) { + return DRWAV_FALSE; + } + + /* Basic validation. */ + if ((fmt.sampleRate == 0 || fmt.sampleRate > DRWAV_MAX_SAMPLE_RATE ) || + (fmt.channels == 0 || fmt.channels > DRWAV_MAX_CHANNELS ) || + (fmt.bitsPerSample == 0 || fmt.bitsPerSample > DRWAV_MAX_BITS_PER_SAMPLE) || + fmt.blockAlign == 0) { + return DRWAV_FALSE; /* Probably an invalid WAV file. */ + } + + /* Translate the internal format. */ + translatedFormatTag = fmt.formatTag; + if (translatedFormatTag == DR_WAVE_FORMAT_EXTENSIBLE) { + translatedFormatTag = drwav_bytes_to_u16_ex(fmt.subFormat + 0, pWav->container); + } + + /* We may have moved passed the data chunk. If so we need to move back. If running in sequential mode we can assume we are already sitting on the data chunk. */ + if (!sequential) { + if (!drwav__seek_from_start(pWav->onSeek, pWav->dataChunkDataPos, pWav->pUserData)) { + return DRWAV_FALSE; + } + cursor = pWav->dataChunkDataPos; + } + + + /* + At this point we should have done the initial parsing of each of our chunks, but we now need to + do a second pass to extract the actual contents of the metadata (the first pass just calculated + the length of the memory allocation). + + We only do this if we've actually got metadata to parse. + */ + if (isProcessingMetadata && metadataParser.metadataCount > 0) { + if (drwav__seek_from_start(pWav->onSeek, metadataStartPos, pWav->pUserData) == DRWAV_FALSE) { + return DRWAV_FALSE; + } + + result = drwav__metadata_alloc(&metadataParser, &pWav->allocationCallbacks); + if (result != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + metadataParser.stage = drwav__metadata_parser_stage_read; + + for (;;) { + drwav_chunk_header header; + drwav_uint64 metadataBytesRead; + + result = drwav__read_chunk_header(pWav->onRead, pWav->pUserData, pWav->container, &cursor, &header); + if (result != DRWAV_SUCCESS) { + break; + } + + metadataBytesRead = drwav__metadata_process_chunk(&metadataParser, &header, drwav_metadata_type_all_including_unknown); + + if (metadataParser.metadataCursor == metadataParser.metadataCount) { + break; + } + + /* Move to the end of the chunk so we can keep iterating. */ + if (drwav__seek_forward(pWav->onSeek, (header.sizeInBytes + header.paddingSize) - metadataBytesRead, pWav->pUserData) == DRWAV_FALSE) { + drwav_free(metadataParser.pMetadata, &pWav->allocationCallbacks); + return DRWAV_FALSE; + } + } + + /* Getting here means we're finished parsing the metadata. */ + pWav->pMetadata = metadataParser.pMetadata; + pWav->metadataCount = metadataParser.metadataCount; + } + + /* + It's possible for the size reported in the data chunk to be greater than that of the file. We + need to do a validation check here to make sure we don't exceed the file size. To skip this + check, set the onTell callback to NULL. + */ + if (pWav->onTell != NULL && pWav->onSeek != NULL) { + if (pWav->onSeek(pWav->pUserData, 0, DRWAV_SEEK_END) == DRWAV_TRUE) { + drwav_int64 fileSize; + if (pWav->onTell(pWav->pUserData, &fileSize)) { + if (dataChunkSize + pWav->dataChunkDataPos > (drwav_uint64)fileSize) { + dataChunkSize = (drwav_uint64)fileSize - pWav->dataChunkDataPos; + } + } + } else { + /* + Failed to seek to the end of the file. It might not be supported by the backend so in + this case we cannot perform the validation check. + */ + } + } + + /* + I've seen a WAV file in the wild where a RIFF-ecapsulated file has the size of it's "RIFF" and + "data" chunks set to 0xFFFFFFFF when the file is definitely not that big. In this case we're + going to have to calculate the size by reading and discarding bytes, and then seeking back. We + cannot do this in sequential mode. We just assume that the rest of the file is audio data. + */ + if (dataChunkSize == 0xFFFFFFFF && (pWav->container == drwav_container_riff || pWav->container == drwav_container_rifx) && pWav->isSequentialWrite == DRWAV_FALSE) { + dataChunkSize = 0; + + for (;;) { + drwav_uint8 temp[4096]; + size_t bytesRead = pWav->onRead(pWav->pUserData, temp, sizeof(temp)); + dataChunkSize += bytesRead; + + if (bytesRead < sizeof(temp)) { + break; + } + } + } + + /* At this point we want to be sitting on the first byte of the raw audio data. */ + if (drwav__seek_from_start(pWav->onSeek, pWav->dataChunkDataPos, pWav->pUserData) == DRWAV_FALSE) { + drwav_free(pWav->pMetadata, &pWav->allocationCallbacks); + return DRWAV_FALSE; + } + + + pWav->fmt = fmt; + pWav->sampleRate = fmt.sampleRate; + pWav->channels = fmt.channels; + pWav->bitsPerSample = fmt.bitsPerSample; + pWav->translatedFormatTag = translatedFormatTag; + + /* + I've had a report where files would start glitching after seeking. The reason for this is the data + chunk is not a clean multiple of the PCM frame size in bytes. Where this becomes a problem is when + seeking, because the number of bytes remaining in the data chunk is used to calculate the current + byte position. If this byte position is not aligned to the number of bytes in a PCM frame, it will + result in the seek not being cleanly positioned at the start of the PCM frame thereby resulting in + all decoded frames after that being corrupted. + + To address this, we need to round the data chunk size down to the nearest multiple of the frame size. + */ + if (!drwav__is_compressed_format_tag(translatedFormatTag)) { + drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame > 0) { + dataChunkSize -= (dataChunkSize % bytesPerFrame); + } + } + + pWav->bytesRemaining = dataChunkSize; + pWav->dataChunkDataSize = dataChunkSize; + + if (sampleCountFromFactChunk != 0) { + pWav->totalPCMFrameCount = sampleCountFromFactChunk; + } else if (aiffFrameCount != 0) { + pWav->totalPCMFrameCount = aiffFrameCount; + } else { + drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + drwav_free(pWav->pMetadata, &pWav->allocationCallbacks); + return DRWAV_FALSE; /* Invalid file. */ + } + + pWav->totalPCMFrameCount = dataChunkSize / bytesPerFrame; + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + drwav_uint64 totalBlockHeaderSizeInBytes; + drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign; + + /* Make sure any trailing partial block is accounted for. */ + if ((blockCount * fmt.blockAlign) < dataChunkSize) { + blockCount += 1; + } + + /* We decode two samples per byte. There will be blockCount headers in the data chunk. This is enough to know how to calculate the total PCM frame count. */ + totalBlockHeaderSizeInBytes = blockCount * (6*fmt.channels); + if (totalBlockHeaderSizeInBytes >= dataChunkSize) { /* <-- We'll be subtracting totalBlockHeaderSizeInBytes from dataChunkSize next so it must be validated. */ + drwav_free(pWav->pMetadata, &pWav->allocationCallbacks); + return DRWAV_FALSE; /* Invalid file. */ + } + + pWav->totalPCMFrameCount = ((dataChunkSize - totalBlockHeaderSizeInBytes) * 2) / fmt.channels; + } + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + drwav_uint64 totalBlockHeaderSizeInBytes; + drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign; + + /* Make sure any trailing partial block is accounted for. */ + if ((blockCount * fmt.blockAlign) < dataChunkSize) { + blockCount += 1; + } + + /* We decode two samples per byte. There will be blockCount headers in the data chunk. This is enough to know how to calculate the total PCM frame count. */ + totalBlockHeaderSizeInBytes = blockCount * (4*fmt.channels); + if (totalBlockHeaderSizeInBytes >= dataChunkSize) { /* <-- We'll be subtracting totalBlockHeaderSizeInBytes from dataChunkSize next so it must be validated. */ + drwav_free(pWav->pMetadata, &pWav->allocationCallbacks); + return DRWAV_FALSE; /* Invalid file. */ + } + + pWav->totalPCMFrameCount = ((dataChunkSize - totalBlockHeaderSizeInBytes) * 2) / fmt.channels; + + /* The header includes a decoded sample for each channel which acts as the initial predictor sample. */ + pWav->totalPCMFrameCount += blockCount; + } + } + + /* Some formats only support a certain number of channels. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM || pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + if (pWav->channels > 2) { + drwav_free(pWav->pMetadata, &pWav->allocationCallbacks); + return DRWAV_FALSE; + } + } + + /* The number of bytes per frame must be known. If not, it's an invalid file and not decodable. */ + if (drwav_get_bytes_per_pcm_frame(pWav) == 0) { + drwav_free(pWav->pMetadata, &pWav->allocationCallbacks); + return DRWAV_FALSE; + } + +#ifdef DR_WAV_LIBSNDFILE_COMPAT + /* + I use libsndfile as a benchmark for testing, however in the version I'm using (from the Windows installer on the libsndfile website), + it appears the total sample count libsndfile uses for MS-ADPCM is incorrect. It would seem they are computing the total sample count + from the number of blocks, however this results in the inclusion of extra silent samples at the end of the last block. The correct + way to know the total sample count is to inspect the "fact" chunk, which should always be present for compressed formats, and should + always include the sample count. This little block of code below is only used to emulate the libsndfile logic so I can properly run my + correctness tests against libsndfile, and is disabled by default. + */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign; + pWav->totalPCMFrameCount = (((blockCount * (fmt.blockAlign - (6*pWav->channels))) * 2)) / fmt.channels; /* x2 because two samples per byte. */ + } + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + drwav_uint64 blockCount = dataChunkSize / fmt.blockAlign; + pWav->totalPCMFrameCount = (((blockCount * (fmt.blockAlign - (4*pWav->channels))) * 2) + (blockCount * pWav->channels)) / fmt.channels; + } +#endif + + return DRWAV_TRUE; +} + +DRWAV_API drwav_bool32 drwav_init(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_ex(pWav, onRead, onSeek, onTell, NULL, pUserData, NULL, 0, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_ex(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, drwav_chunk_proc onChunk, void* pReadSeekTellUserData, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (!drwav_preinit(pWav, onRead, onSeek, onTell, pReadSeekTellUserData, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + return drwav_init__internal(pWav, onChunk, pChunkUserData, flags); +} + +DRWAV_API drwav_bool32 drwav_init_with_metadata(drwav* pWav, drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (!drwav_preinit(pWav, onRead, onSeek, onTell, pUserData, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + return drwav_init__internal(pWav, NULL, NULL, flags | DRWAV_WITH_METADATA); +} + +DRWAV_API drwav_metadata* drwav_take_ownership_of_metadata(drwav* pWav) +{ + drwav_metadata *result = pWav->pMetadata; + + pWav->pMetadata = NULL; + pWav->metadataCount = 0; + + return result; +} + + +DRWAV_PRIVATE size_t drwav__write(drwav* pWav, const void* pData, size_t dataSize) +{ + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->onWrite != NULL); + + /* Generic write. Assumes no byte reordering required. */ + return pWav->onWrite(pWav->pUserData, pData, dataSize); +} + +DRWAV_PRIVATE size_t drwav__write_byte(drwav* pWav, drwav_uint8 byte) +{ + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->onWrite != NULL); + + return pWav->onWrite(pWav->pUserData, &byte, 1); +} + +DRWAV_PRIVATE size_t drwav__write_u16ne_to_le(drwav* pWav, drwav_uint16 value) +{ + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->onWrite != NULL); + + if (!drwav__is_little_endian()) { + value = drwav__bswap16(value); + } + + return drwav__write(pWav, &value, 2); +} + +DRWAV_PRIVATE size_t drwav__write_u32ne_to_le(drwav* pWav, drwav_uint32 value) +{ + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->onWrite != NULL); + + if (!drwav__is_little_endian()) { + value = drwav__bswap32(value); + } + + return drwav__write(pWav, &value, 4); +} + +DRWAV_PRIVATE size_t drwav__write_u64ne_to_le(drwav* pWav, drwav_uint64 value) +{ + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->onWrite != NULL); + + if (!drwav__is_little_endian()) { + value = drwav__bswap64(value); + } + + return drwav__write(pWav, &value, 8); +} + +DRWAV_PRIVATE size_t drwav__write_f32ne_to_le(drwav* pWav, float value) +{ + union { + drwav_uint32 u32; + float f32; + } u; + + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->onWrite != NULL); + + u.f32 = value; + + if (!drwav__is_little_endian()) { + u.u32 = drwav__bswap32(u.u32); + } + + return drwav__write(pWav, &u.u32, 4); +} + +DRWAV_PRIVATE size_t drwav__write_or_count(drwav* pWav, const void* pData, size_t dataSize) +{ + if (pWav == NULL) { + return dataSize; + } + + return drwav__write(pWav, pData, dataSize); +} + +DRWAV_PRIVATE size_t drwav__write_or_count_byte(drwav* pWav, drwav_uint8 byte) +{ + if (pWav == NULL) { + return 1; + } + + return drwav__write_byte(pWav, byte); +} + +DRWAV_PRIVATE size_t drwav__write_or_count_u16ne_to_le(drwav* pWav, drwav_uint16 value) +{ + if (pWav == NULL) { + return 2; + } + + return drwav__write_u16ne_to_le(pWav, value); +} + +DRWAV_PRIVATE size_t drwav__write_or_count_u32ne_to_le(drwav* pWav, drwav_uint32 value) +{ + if (pWav == NULL) { + return 4; + } + + return drwav__write_u32ne_to_le(pWav, value); +} + +#if 0 /* Unused for now. */ +DRWAV_PRIVATE size_t drwav__write_or_count_u64ne_to_le(drwav* pWav, drwav_uint64 value) +{ + if (pWav == NULL) { + return 8; + } + + return drwav__write_u64ne_to_le(pWav, value); +} +#endif + +DRWAV_PRIVATE size_t drwav__write_or_count_f32ne_to_le(drwav* pWav, float value) +{ + if (pWav == NULL) { + return 4; + } + + return drwav__write_f32ne_to_le(pWav, value); +} + +DRWAV_PRIVATE size_t drwav__write_or_count_string_to_fixed_size_buf(drwav* pWav, char* str, size_t bufFixedSize) +{ + size_t len; + + if (pWav == NULL) { + return bufFixedSize; + } + + len = drwav__strlen_clamped(str, bufFixedSize); + drwav__write_or_count(pWav, str, len); + + if (len < bufFixedSize) { + size_t i; + for (i = 0; i < bufFixedSize - len; ++i) { + drwav__write_byte(pWav, 0); + } + } + + return bufFixedSize; +} + + +/* pWav can be NULL meaning just count the bytes that would be written. */ +DRWAV_PRIVATE size_t drwav__write_or_count_metadata(drwav* pWav, drwav_metadata* pMetadatas, drwav_uint32 metadataCount) +{ + size_t bytesWritten = 0; + drwav_bool32 hasListAdtl = DRWAV_FALSE; + drwav_bool32 hasListInfo = DRWAV_FALSE; + drwav_uint32 iMetadata; + + if (pMetadatas == NULL || metadataCount == 0) { + return 0; + } + + for (iMetadata = 0; iMetadata < metadataCount; ++iMetadata) { + drwav_metadata* pMetadata = &pMetadatas[iMetadata]; + drwav_uint32 chunkSize = 0; + + if ((pMetadata->type & drwav_metadata_type_list_all_info_strings) || (pMetadata->type == drwav_metadata_type_unknown && pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_info_list)) { + hasListInfo = DRWAV_TRUE; + } + + if ((pMetadata->type & drwav_metadata_type_list_all_adtl) || (pMetadata->type == drwav_metadata_type_unknown && pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_adtl_list)) { + hasListAdtl = DRWAV_TRUE; + } + + switch (pMetadata->type) { + case drwav_metadata_type_smpl: + { + drwav_uint32 iLoop; + + chunkSize = DRWAV_SMPL_BYTES + DRWAV_SMPL_LOOP_BYTES * pMetadata->data.smpl.sampleLoopCount + pMetadata->data.smpl.samplerSpecificDataSizeInBytes; + + bytesWritten += drwav__write_or_count(pWav, "smpl", 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize); + + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.manufacturerId); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.productId); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.samplePeriodNanoseconds); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.midiUnityNote); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.midiPitchFraction); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.smpteFormat); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.smpteOffset); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.sampleLoopCount); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.samplerSpecificDataSizeInBytes); + + for (iLoop = 0; iLoop < pMetadata->data.smpl.sampleLoopCount; ++iLoop) { + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].cuePointId); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].type); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].firstSampleOffset); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].lastSampleOffset); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].sampleFraction); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.smpl.pLoops[iLoop].playCount); + } + + if (pMetadata->data.smpl.samplerSpecificDataSizeInBytes > 0) { + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.smpl.pSamplerSpecificData, pMetadata->data.smpl.samplerSpecificDataSizeInBytes); + } + } break; + + case drwav_metadata_type_inst: + { + chunkSize = DRWAV_INST_BYTES; + + bytesWritten += drwav__write_or_count(pWav, "inst", 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize); + bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.midiUnityNote, 1); + bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.fineTuneCents, 1); + bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.gainDecibels, 1); + bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.lowNote, 1); + bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.highNote, 1); + bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.lowVelocity, 1); + bytesWritten += drwav__write_or_count(pWav, &pMetadata->data.inst.highVelocity, 1); + } break; + + case drwav_metadata_type_cue: + { + drwav_uint32 iCuePoint; + + chunkSize = DRWAV_CUE_BYTES + DRWAV_CUE_POINT_BYTES * pMetadata->data.cue.cuePointCount; + + bytesWritten += drwav__write_or_count(pWav, "cue ", 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.cuePointCount); + for (iCuePoint = 0; iCuePoint < pMetadata->data.cue.cuePointCount; ++iCuePoint) { + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].id); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].playOrderPosition); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].dataChunkId, 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].chunkStart); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].blockStart); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.cue.pCuePoints[iCuePoint].sampleOffset); + } + } break; + + case drwav_metadata_type_acid: + { + chunkSize = DRWAV_ACID_BYTES; + + bytesWritten += drwav__write_or_count(pWav, "acid", 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.acid.flags); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.acid.midiUnityNote); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.acid.reserved1); + bytesWritten += drwav__write_or_count_f32ne_to_le(pWav, pMetadata->data.acid.reserved2); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.acid.numBeats); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.acid.meterDenominator); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.acid.meterNumerator); + bytesWritten += drwav__write_or_count_f32ne_to_le(pWav, pMetadata->data.acid.tempo); + } break; + + case drwav_metadata_type_bext: + { + char reservedBuf[DRWAV_BEXT_RESERVED_BYTES]; + drwav_uint32 timeReferenceLow; + drwav_uint32 timeReferenceHigh; + + chunkSize = DRWAV_BEXT_BYTES + pMetadata->data.bext.codingHistorySize; + + bytesWritten += drwav__write_or_count(pWav, "bext", 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize); + + bytesWritten += drwav__write_or_count_string_to_fixed_size_buf(pWav, pMetadata->data.bext.pDescription, DRWAV_BEXT_DESCRIPTION_BYTES); + bytesWritten += drwav__write_or_count_string_to_fixed_size_buf(pWav, pMetadata->data.bext.pOriginatorName, DRWAV_BEXT_ORIGINATOR_NAME_BYTES); + bytesWritten += drwav__write_or_count_string_to_fixed_size_buf(pWav, pMetadata->data.bext.pOriginatorReference, DRWAV_BEXT_ORIGINATOR_REF_BYTES); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.bext.pOriginationDate, sizeof(pMetadata->data.bext.pOriginationDate)); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.bext.pOriginationTime, sizeof(pMetadata->data.bext.pOriginationTime)); + + timeReferenceLow = (drwav_uint32)(pMetadata->data.bext.timeReference & 0xFFFFFFFF); + timeReferenceHigh = (drwav_uint32)(pMetadata->data.bext.timeReference >> 32); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, timeReferenceLow); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, timeReferenceHigh); + + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.version); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.bext.pUMID, DRWAV_BEXT_UMID_BYTES); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.loudnessValue); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.loudnessRange); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.maxTruePeakLevel); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.maxMomentaryLoudness); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.bext.maxShortTermLoudness); + + DRWAV_ZERO_MEMORY(reservedBuf, sizeof(reservedBuf)); + bytesWritten += drwav__write_or_count(pWav, reservedBuf, sizeof(reservedBuf)); + + if (pMetadata->data.bext.codingHistorySize > 0) { + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.bext.pCodingHistory, pMetadata->data.bext.codingHistorySize); + } + } break; + + case drwav_metadata_type_unknown: + { + if (pMetadata->data.unknown.chunkLocation == drwav_metadata_location_top_level) { + chunkSize = pMetadata->data.unknown.dataSizeInBytes; + + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.id, 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.pData, pMetadata->data.unknown.dataSizeInBytes); + } + } break; + + default: break; + } + if ((chunkSize % 2) != 0) { + bytesWritten += drwav__write_or_count_byte(pWav, 0); + } + } + + if (hasListInfo) { + drwav_uint32 chunkSize = 4; /* Start with 4 bytes for "INFO". */ + for (iMetadata = 0; iMetadata < metadataCount; ++iMetadata) { + drwav_metadata* pMetadata = &pMetadatas[iMetadata]; + + if ((pMetadata->type & drwav_metadata_type_list_all_info_strings)) { + chunkSize += 8; /* For id and string size. */ + chunkSize += pMetadata->data.infoText.stringLength + 1; /* Include null terminator. */ + } else if (pMetadata->type == drwav_metadata_type_unknown && pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_info_list) { + chunkSize += 8; /* For id string size. */ + chunkSize += pMetadata->data.unknown.dataSizeInBytes; + } + + if ((chunkSize % 2) != 0) { + chunkSize += 1; + } + } + + bytesWritten += drwav__write_or_count(pWav, "LIST", 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize); + bytesWritten += drwav__write_or_count(pWav, "INFO", 4); + + for (iMetadata = 0; iMetadata < metadataCount; ++iMetadata) { + drwav_metadata* pMetadata = &pMetadatas[iMetadata]; + drwav_uint32 subchunkSize = 0; + + if (pMetadata->type & drwav_metadata_type_list_all_info_strings) { + const char* pID = NULL; + + switch (pMetadata->type) { + case drwav_metadata_type_list_info_software: pID = "ISFT"; break; + case drwav_metadata_type_list_info_copyright: pID = "ICOP"; break; + case drwav_metadata_type_list_info_title: pID = "INAM"; break; + case drwav_metadata_type_list_info_artist: pID = "IART"; break; + case drwav_metadata_type_list_info_comment: pID = "ICMT"; break; + case drwav_metadata_type_list_info_date: pID = "ICRD"; break; + case drwav_metadata_type_list_info_genre: pID = "IGNR"; break; + case drwav_metadata_type_list_info_album: pID = "IPRD"; break; + case drwav_metadata_type_list_info_tracknumber: pID = "ITRK"; break; + case drwav_metadata_type_list_info_location: pID = "IARL"; break; + case drwav_metadata_type_list_info_organization: pID = "ICMS"; break; + case drwav_metadata_type_list_info_keywords: pID = "IKEY"; break; + case drwav_metadata_type_list_info_medium: pID = "IMED"; break; + case drwav_metadata_type_list_info_description: pID = "ISBJ"; break; + default: break; + } + + DRWAV_ASSERT(pID != NULL); + + if (pMetadata->data.infoText.stringLength) { + subchunkSize = pMetadata->data.infoText.stringLength + 1; + bytesWritten += drwav__write_or_count(pWav, pID, 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, subchunkSize); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.infoText.pString, pMetadata->data.infoText.stringLength); + bytesWritten += drwav__write_or_count_byte(pWav, '\0'); + } + } else if (pMetadata->type == drwav_metadata_type_unknown && pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_info_list) { + if (pMetadata->data.unknown.dataSizeInBytes) { + subchunkSize = pMetadata->data.unknown.dataSizeInBytes; + + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.id, 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.unknown.dataSizeInBytes); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.pData, subchunkSize); + } + } + + if ((subchunkSize % 2) != 0) { + bytesWritten += drwav__write_or_count_byte(pWav, 0); + } + } + } + + if (hasListAdtl) { + drwav_uint32 chunkSize = 4; /* start with 4 bytes for "adtl" */ + + for (iMetadata = 0; iMetadata < metadataCount; ++iMetadata) { + drwav_metadata* pMetadata = &pMetadatas[iMetadata]; + + switch (pMetadata->type) + { + case drwav_metadata_type_list_label: + case drwav_metadata_type_list_note: + { + chunkSize += 8; /* for id and chunk size */ + chunkSize += DRWAV_LIST_LABEL_OR_NOTE_BYTES; + + if (pMetadata->data.labelOrNote.stringLength > 0) { + chunkSize += pMetadata->data.labelOrNote.stringLength + 1; + } + } break; + + case drwav_metadata_type_list_labelled_cue_region: + { + chunkSize += 8; /* for id and chunk size */ + chunkSize += DRWAV_LIST_LABELLED_TEXT_BYTES; + + if (pMetadata->data.labelledCueRegion.stringLength > 0) { + chunkSize += pMetadata->data.labelledCueRegion.stringLength + 1; + } + } break; + + case drwav_metadata_type_unknown: + { + if (pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_adtl_list) { + chunkSize += 8; /* for id and chunk size */ + chunkSize += pMetadata->data.unknown.dataSizeInBytes; + } + } break; + + default: break; + } + + if ((chunkSize % 2) != 0) { + chunkSize += 1; + } + } + + bytesWritten += drwav__write_or_count(pWav, "LIST", 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, chunkSize); + bytesWritten += drwav__write_or_count(pWav, "adtl", 4); + + for (iMetadata = 0; iMetadata < metadataCount; ++iMetadata) { + drwav_metadata* pMetadata = &pMetadatas[iMetadata]; + drwav_uint32 subchunkSize = 0; + + switch (pMetadata->type) + { + case drwav_metadata_type_list_label: + case drwav_metadata_type_list_note: + { + if (pMetadata->data.labelOrNote.stringLength > 0) { + const char *pID = NULL; + + if (pMetadata->type == drwav_metadata_type_list_label) { + pID = "labl"; + } + else if (pMetadata->type == drwav_metadata_type_list_note) { + pID = "note"; + } + + DRWAV_ASSERT(pID != NULL); + DRWAV_ASSERT(pMetadata->data.labelOrNote.pString != NULL); + + subchunkSize = DRWAV_LIST_LABEL_OR_NOTE_BYTES; + + bytesWritten += drwav__write_or_count(pWav, pID, 4); + subchunkSize += pMetadata->data.labelOrNote.stringLength + 1; + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, subchunkSize); + + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.labelOrNote.cuePointId); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.labelOrNote.pString, pMetadata->data.labelOrNote.stringLength); + bytesWritten += drwav__write_or_count_byte(pWav, '\0'); + } + } break; + + case drwav_metadata_type_list_labelled_cue_region: + { + subchunkSize = DRWAV_LIST_LABELLED_TEXT_BYTES; + + bytesWritten += drwav__write_or_count(pWav, "ltxt", 4); + if (pMetadata->data.labelledCueRegion.stringLength > 0) { + subchunkSize += pMetadata->data.labelledCueRegion.stringLength + 1; + } + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, subchunkSize); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.labelledCueRegion.cuePointId); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, pMetadata->data.labelledCueRegion.sampleLength); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.labelledCueRegion.purposeId, 4); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.labelledCueRegion.country); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.labelledCueRegion.language); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.labelledCueRegion.dialect); + bytesWritten += drwav__write_or_count_u16ne_to_le(pWav, pMetadata->data.labelledCueRegion.codePage); + + if (pMetadata->data.labelledCueRegion.stringLength > 0) { + DRWAV_ASSERT(pMetadata->data.labelledCueRegion.pString != NULL); + + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.labelledCueRegion.pString, pMetadata->data.labelledCueRegion.stringLength); + bytesWritten += drwav__write_or_count_byte(pWav, '\0'); + } + } break; + + case drwav_metadata_type_unknown: + { + if (pMetadata->data.unknown.chunkLocation == drwav_metadata_location_inside_adtl_list) { + subchunkSize = pMetadata->data.unknown.dataSizeInBytes; + + DRWAV_ASSERT(pMetadata->data.unknown.pData != NULL); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.id, 4); + bytesWritten += drwav__write_or_count_u32ne_to_le(pWav, subchunkSize); + bytesWritten += drwav__write_or_count(pWav, pMetadata->data.unknown.pData, subchunkSize); + } + } break; + + default: break; + } + + if ((subchunkSize % 2) != 0) { + bytesWritten += drwav__write_or_count_byte(pWav, 0); + } + } + } + + DRWAV_ASSERT((bytesWritten % 2) == 0); + + return bytesWritten; +} + +DRWAV_PRIVATE drwav_uint32 drwav__riff_chunk_size_riff(drwav_uint64 dataChunkSize, drwav_metadata* pMetadata, drwav_uint32 metadataCount) +{ + drwav_uint64 chunkSize = 4 + 24 + (drwav_uint64)drwav__write_or_count_metadata(NULL, pMetadata, metadataCount) + 8 + dataChunkSize + drwav__chunk_padding_size_riff(dataChunkSize); /* 4 = "WAVE". 24 = "fmt " chunk. 8 = "data" + u32 data size. */ + if (chunkSize > 0xFFFFFFFFUL) { + chunkSize = 0xFFFFFFFFUL; + } + + return (drwav_uint32)chunkSize; /* Safe cast due to the clamp above. */ +} + +DRWAV_PRIVATE drwav_uint32 drwav__data_chunk_size_riff(drwav_uint64 dataChunkSize) +{ + if (dataChunkSize <= 0xFFFFFFFFUL) { + return (drwav_uint32)dataChunkSize; + } else { + return 0xFFFFFFFFUL; + } +} + +DRWAV_PRIVATE drwav_uint64 drwav__riff_chunk_size_w64(drwav_uint64 dataChunkSize) +{ + drwav_uint64 dataSubchunkPaddingSize = drwav__chunk_padding_size_w64(dataChunkSize); + + return 80 + 24 + dataChunkSize + dataSubchunkPaddingSize; /* +24 because W64 includes the size of the GUID and size fields. */ +} + +DRWAV_PRIVATE drwav_uint64 drwav__data_chunk_size_w64(drwav_uint64 dataChunkSize) +{ + return 24 + dataChunkSize; /* +24 because W64 includes the size of the GUID and size fields. */ +} + +DRWAV_PRIVATE drwav_uint64 drwav__riff_chunk_size_rf64(drwav_uint64 dataChunkSize, drwav_metadata *metadata, drwav_uint32 numMetadata) +{ + drwav_uint64 chunkSize = 4 + 36 + 24 + (drwav_uint64)drwav__write_or_count_metadata(NULL, metadata, numMetadata) + 8 + dataChunkSize + drwav__chunk_padding_size_riff(dataChunkSize); /* 4 = "WAVE". 36 = "ds64" chunk. 24 = "fmt " chunk. 8 = "data" + u32 data size. */ + if (chunkSize > 0xFFFFFFFFUL) { + chunkSize = 0xFFFFFFFFUL; + } + + return chunkSize; +} + +DRWAV_PRIVATE drwav_uint64 drwav__data_chunk_size_rf64(drwav_uint64 dataChunkSize) +{ + return dataChunkSize; +} + + + +DRWAV_PRIVATE drwav_bool32 drwav_preinit_write(drwav* pWav, const drwav_data_format* pFormat, drwav_bool32 isSequential, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pWav == NULL || onWrite == NULL) { + return DRWAV_FALSE; + } + + if (!isSequential && onSeek == NULL) { + return DRWAV_FALSE; /* <-- onSeek is required when in non-sequential mode. */ + } + + /* Not currently supporting compressed formats. Will need to add support for the "fact" chunk before we enable this. */ + if (pFormat->format == DR_WAVE_FORMAT_EXTENSIBLE) { + return DRWAV_FALSE; + } + if (pFormat->format == DR_WAVE_FORMAT_ADPCM || pFormat->format == DR_WAVE_FORMAT_DVI_ADPCM) { + return DRWAV_FALSE; + } + + DRWAV_ZERO_MEMORY(pWav, sizeof(*pWav)); + pWav->onWrite = onWrite; + pWav->onSeek = onSeek; + pWav->pUserData = pUserData; + pWav->allocationCallbacks = drwav_copy_allocation_callbacks_or_defaults(pAllocationCallbacks); + + if (pWav->allocationCallbacks.onFree == NULL || (pWav->allocationCallbacks.onMalloc == NULL && pWav->allocationCallbacks.onRealloc == NULL)) { + return DRWAV_FALSE; /* Invalid allocation callbacks. */ + } + + pWav->fmt.formatTag = (drwav_uint16)pFormat->format; + pWav->fmt.channels = (drwav_uint16)pFormat->channels; + pWav->fmt.sampleRate = pFormat->sampleRate; + pWav->fmt.avgBytesPerSec = (drwav_uint32)((pFormat->bitsPerSample * pFormat->sampleRate * pFormat->channels) / 8); + pWav->fmt.blockAlign = (drwav_uint16)((pFormat->channels * pFormat->bitsPerSample) / 8); + pWav->fmt.bitsPerSample = (drwav_uint16)pFormat->bitsPerSample; + pWav->fmt.extendedSize = 0; + pWav->isSequentialWrite = isSequential; + + return DRWAV_TRUE; +} + + +DRWAV_PRIVATE drwav_bool32 drwav_init_write__internal(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount) +{ + /* The function assumes drwav_preinit_write() was called beforehand. */ + + size_t runningPos = 0; + drwav_uint64 initialDataChunkSize = 0; + drwav_uint64 chunkSizeFMT; + + /* + The initial values for the "RIFF" and "data" chunks depends on whether or not we are initializing in sequential mode or not. In + sequential mode we set this to its final values straight away since they can be calculated from the total sample count. In non- + sequential mode we initialize it all to zero and fill it out in drwav_uninit() using a backwards seek. + */ + if (pWav->isSequentialWrite) { + initialDataChunkSize = (totalSampleCount * pWav->fmt.bitsPerSample) / 8; + + /* + The RIFF container has a limit on the number of samples. drwav is not allowing this. There's no practical limits for Wave64 + so for the sake of simplicity I'm not doing any validation for that. + */ + if (pFormat->container == drwav_container_riff) { + if (initialDataChunkSize > (0xFFFFFFFFUL - 36)) { + return DRWAV_FALSE; /* Not enough room to store every sample. */ + } + } + } + + pWav->dataChunkDataSizeTargetWrite = initialDataChunkSize; + + + /* "RIFF" chunk. */ + if (pFormat->container == drwav_container_riff) { + drwav_uint32 chunkSizeRIFF = 36 + (drwav_uint32)initialDataChunkSize; /* +36 = "WAVE" + [sizeof "fmt " chunk] + [data chunk header] */ + runningPos += drwav__write(pWav, "RIFF", 4); + runningPos += drwav__write_u32ne_to_le(pWav, chunkSizeRIFF); + runningPos += drwav__write(pWav, "WAVE", 4); + } else if (pFormat->container == drwav_container_w64) { + drwav_uint64 chunkSizeRIFF = 80 + 24 + initialDataChunkSize; /* +24 because W64 includes the size of the GUID and size fields. */ + runningPos += drwav__write(pWav, drwavGUID_W64_RIFF, 16); + runningPos += drwav__write_u64ne_to_le(pWav, chunkSizeRIFF); + runningPos += drwav__write(pWav, drwavGUID_W64_WAVE, 16); + } else if (pFormat->container == drwav_container_rf64) { + runningPos += drwav__write(pWav, "RF64", 4); + runningPos += drwav__write_u32ne_to_le(pWav, 0xFFFFFFFF); /* Always 0xFFFFFFFF for RF64. Set to a proper value in the "ds64" chunk. */ + runningPos += drwav__write(pWav, "WAVE", 4); + } else { + return DRWAV_FALSE; /* Container not supported for writing. */ + } + + + /* "ds64" chunk (RF64 only). */ + if (pFormat->container == drwav_container_rf64) { + drwav_uint32 initialds64ChunkSize = 28; /* 28 = [Size of RIFF (8 bytes)] + [Size of DATA (8 bytes)] + [Sample Count (8 bytes)] + [Table Length (4 bytes)]. Table length always set to 0. */ + drwav_uint64 initialRiffChunkSize = 8 + initialds64ChunkSize + initialDataChunkSize; /* +8 for the ds64 header. */ + + runningPos += drwav__write(pWav, "ds64", 4); + runningPos += drwav__write_u32ne_to_le(pWav, initialds64ChunkSize); /* Size of ds64. */ + runningPos += drwav__write_u64ne_to_le(pWav, initialRiffChunkSize); /* Size of RIFF. Set to true value at the end. */ + runningPos += drwav__write_u64ne_to_le(pWav, initialDataChunkSize); /* Size of DATA. Set to true value at the end. */ + runningPos += drwav__write_u64ne_to_le(pWav, totalSampleCount); /* Sample count. */ + runningPos += drwav__write_u32ne_to_le(pWav, 0); /* Table length. Always set to zero in our case since we're not doing any other chunks than "DATA". */ + } + + + /* "fmt " chunk. */ + if (pFormat->container == drwav_container_riff || pFormat->container == drwav_container_rf64) { + chunkSizeFMT = 16; + runningPos += drwav__write(pWav, "fmt ", 4); + runningPos += drwav__write_u32ne_to_le(pWav, (drwav_uint32)chunkSizeFMT); + } else if (pFormat->container == drwav_container_w64) { + chunkSizeFMT = 40; + runningPos += drwav__write(pWav, drwavGUID_W64_FMT, 16); + runningPos += drwav__write_u64ne_to_le(pWav, chunkSizeFMT); + } + + runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.formatTag); + runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.channels); + runningPos += drwav__write_u32ne_to_le(pWav, pWav->fmt.sampleRate); + runningPos += drwav__write_u32ne_to_le(pWav, pWav->fmt.avgBytesPerSec); + runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.blockAlign); + runningPos += drwav__write_u16ne_to_le(pWav, pWav->fmt.bitsPerSample); + + /* TODO: is a 'fact' chunk required for DR_WAVE_FORMAT_IEEE_FLOAT? */ + + if (!pWav->isSequentialWrite && pWav->pMetadata != NULL && pWav->metadataCount > 0 && (pFormat->container == drwav_container_riff || pFormat->container == drwav_container_rf64)) { + runningPos += drwav__write_or_count_metadata(pWav, pWav->pMetadata, pWav->metadataCount); + } + + pWav->dataChunkDataPos = runningPos; + + /* "data" chunk. */ + if (pFormat->container == drwav_container_riff) { + drwav_uint32 chunkSizeDATA = (drwav_uint32)initialDataChunkSize; + runningPos += drwav__write(pWav, "data", 4); + runningPos += drwav__write_u32ne_to_le(pWav, chunkSizeDATA); + } else if (pFormat->container == drwav_container_w64) { + drwav_uint64 chunkSizeDATA = 24 + initialDataChunkSize; /* +24 because W64 includes the size of the GUID and size fields. */ + runningPos += drwav__write(pWav, drwavGUID_W64_DATA, 16); + runningPos += drwav__write_u64ne_to_le(pWav, chunkSizeDATA); + } else if (pFormat->container == drwav_container_rf64) { + runningPos += drwav__write(pWav, "data", 4); + runningPos += drwav__write_u32ne_to_le(pWav, 0xFFFFFFFF); /* Always set to 0xFFFFFFFF for RF64. The true size of the data chunk is specified in the ds64 chunk. */ + } + + /* Set some properties for the client's convenience. */ + pWav->container = pFormat->container; + pWav->channels = (drwav_uint16)pFormat->channels; + pWav->sampleRate = pFormat->sampleRate; + pWav->bitsPerSample = (drwav_uint16)pFormat->bitsPerSample; + pWav->translatedFormatTag = (drwav_uint16)pFormat->format; + pWav->dataChunkDataPos = runningPos; + + return DRWAV_TRUE; +} + + +DRWAV_API drwav_bool32 drwav_init_write(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (!drwav_preinit_write(pWav, pFormat, DRWAV_FALSE, onWrite, onSeek, pUserData, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + return drwav_init_write__internal(pWav, pFormat, 0); /* DRWAV_FALSE = Not Sequential */ +} + +DRWAV_API drwav_bool32 drwav_init_write_sequential(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (!drwav_preinit_write(pWav, pFormat, DRWAV_TRUE, onWrite, NULL, pUserData, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + return drwav_init_write__internal(pWav, pFormat, totalSampleCount); /* DRWAV_TRUE = Sequential */ +} + +DRWAV_API drwav_bool32 drwav_init_write_sequential_pcm_frames(drwav* pWav, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, drwav_write_proc onWrite, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pFormat == NULL) { + return DRWAV_FALSE; + } + + return drwav_init_write_sequential(pWav, pFormat, totalPCMFrameCount*pFormat->channels, onWrite, pUserData, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_write_with_metadata(drwav* pWav, const drwav_data_format* pFormat, drwav_write_proc onWrite, drwav_seek_proc onSeek, void* pUserData, const drwav_allocation_callbacks* pAllocationCallbacks, drwav_metadata* pMetadata, drwav_uint32 metadataCount) +{ + if (!drwav_preinit_write(pWav, pFormat, DRWAV_FALSE, onWrite, onSeek, pUserData, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + pWav->pMetadata = pMetadata; + pWav->metadataCount = metadataCount; + + return drwav_init_write__internal(pWav, pFormat, 0); +} + + +DRWAV_API drwav_uint64 drwav_target_write_size_bytes(const drwav_data_format* pFormat, drwav_uint64 totalFrameCount, drwav_metadata* pMetadata, drwav_uint32 metadataCount) +{ + /* Casting totalFrameCount to drwav_int64 for VC6 compatibility. No issues in practice because nobody is going to exhaust the whole 63 bits. */ + drwav_uint64 targetDataSizeBytes = (drwav_uint64)((drwav_int64)totalFrameCount * pFormat->channels * pFormat->bitsPerSample/8.0); + drwav_uint64 riffChunkSizeBytes; + drwav_uint64 fileSizeBytes = 0; + + if (pFormat->container == drwav_container_riff) { + riffChunkSizeBytes = drwav__riff_chunk_size_riff(targetDataSizeBytes, pMetadata, metadataCount); + fileSizeBytes = (8 + riffChunkSizeBytes); /* +8 because WAV doesn't include the size of the ChunkID and ChunkSize fields. */ + } else if (pFormat->container == drwav_container_w64) { + riffChunkSizeBytes = drwav__riff_chunk_size_w64(targetDataSizeBytes); + fileSizeBytes = riffChunkSizeBytes; + } else if (pFormat->container == drwav_container_rf64) { + riffChunkSizeBytes = drwav__riff_chunk_size_rf64(targetDataSizeBytes, pMetadata, metadataCount); + fileSizeBytes = (8 + riffChunkSizeBytes); /* +8 because WAV doesn't include the size of the ChunkID and ChunkSize fields. */ + } + + return fileSizeBytes; +} + + +#ifndef DR_WAV_NO_STDIO + +/* Errno */ +/* drwav_result_from_errno() is only used for fopen() and wfopen() so putting it inside DR_WAV_NO_STDIO for now. If something else needs this later we can move it out. */ +#include +DRWAV_PRIVATE drwav_result drwav_result_from_errno(int e) +{ + switch (e) + { + case 0: return DRWAV_SUCCESS; + #ifdef EPERM + case EPERM: return DRWAV_INVALID_OPERATION; + #endif + #ifdef ENOENT + case ENOENT: return DRWAV_DOES_NOT_EXIST; + #endif + #ifdef ESRCH + case ESRCH: return DRWAV_DOES_NOT_EXIST; + #endif + #ifdef EINTR + case EINTR: return DRWAV_INTERRUPT; + #endif + #ifdef EIO + case EIO: return DRWAV_IO_ERROR; + #endif + #ifdef ENXIO + case ENXIO: return DRWAV_DOES_NOT_EXIST; + #endif + #ifdef E2BIG + case E2BIG: return DRWAV_INVALID_ARGS; + #endif + #ifdef ENOEXEC + case ENOEXEC: return DRWAV_INVALID_FILE; + #endif + #ifdef EBADF + case EBADF: return DRWAV_INVALID_FILE; + #endif + #ifdef ECHILD + case ECHILD: return DRWAV_ERROR; + #endif + #ifdef EAGAIN + case EAGAIN: return DRWAV_UNAVAILABLE; + #endif + #ifdef ENOMEM + case ENOMEM: return DRWAV_OUT_OF_MEMORY; + #endif + #ifdef EACCES + case EACCES: return DRWAV_ACCESS_DENIED; + #endif + #ifdef EFAULT + case EFAULT: return DRWAV_BAD_ADDRESS; + #endif + #ifdef ENOTBLK + case ENOTBLK: return DRWAV_ERROR; + #endif + #ifdef EBUSY + case EBUSY: return DRWAV_BUSY; + #endif + #ifdef EEXIST + case EEXIST: return DRWAV_ALREADY_EXISTS; + #endif + #ifdef EXDEV + case EXDEV: return DRWAV_ERROR; + #endif + #ifdef ENODEV + case ENODEV: return DRWAV_DOES_NOT_EXIST; + #endif + #ifdef ENOTDIR + case ENOTDIR: return DRWAV_NOT_DIRECTORY; + #endif + #ifdef EISDIR + case EISDIR: return DRWAV_IS_DIRECTORY; + #endif + #ifdef EINVAL + case EINVAL: return DRWAV_INVALID_ARGS; + #endif + #ifdef ENFILE + case ENFILE: return DRWAV_TOO_MANY_OPEN_FILES; + #endif + #ifdef EMFILE + case EMFILE: return DRWAV_TOO_MANY_OPEN_FILES; + #endif + #ifdef ENOTTY + case ENOTTY: return DRWAV_INVALID_OPERATION; + #endif + #ifdef ETXTBSY + case ETXTBSY: return DRWAV_BUSY; + #endif + #ifdef EFBIG + case EFBIG: return DRWAV_TOO_BIG; + #endif + #ifdef ENOSPC + case ENOSPC: return DRWAV_NO_SPACE; + #endif + #ifdef ESPIPE + case ESPIPE: return DRWAV_BAD_SEEK; + #endif + #ifdef EROFS + case EROFS: return DRWAV_ACCESS_DENIED; + #endif + #ifdef EMLINK + case EMLINK: return DRWAV_TOO_MANY_LINKS; + #endif + #ifdef EPIPE + case EPIPE: return DRWAV_BAD_PIPE; + #endif + #ifdef EDOM + case EDOM: return DRWAV_OUT_OF_RANGE; + #endif + #ifdef ERANGE + case ERANGE: return DRWAV_OUT_OF_RANGE; + #endif + #ifdef EDEADLK + case EDEADLK: return DRWAV_DEADLOCK; + #endif + #ifdef ENAMETOOLONG + case ENAMETOOLONG: return DRWAV_PATH_TOO_LONG; + #endif + #ifdef ENOLCK + case ENOLCK: return DRWAV_ERROR; + #endif + #ifdef ENOSYS + case ENOSYS: return DRWAV_NOT_IMPLEMENTED; + #endif + #if defined(ENOTEMPTY) && ENOTEMPTY != EEXIST /* In AIX, ENOTEMPTY and EEXIST use the same value. */ + case ENOTEMPTY: return DRWAV_DIRECTORY_NOT_EMPTY; + #endif + #ifdef ELOOP + case ELOOP: return DRWAV_TOO_MANY_LINKS; + #endif + #ifdef ENOMSG + case ENOMSG: return DRWAV_NO_MESSAGE; + #endif + #ifdef EIDRM + case EIDRM: return DRWAV_ERROR; + #endif + #ifdef ECHRNG + case ECHRNG: return DRWAV_ERROR; + #endif + #ifdef EL2NSYNC + case EL2NSYNC: return DRWAV_ERROR; + #endif + #ifdef EL3HLT + case EL3HLT: return DRWAV_ERROR; + #endif + #ifdef EL3RST + case EL3RST: return DRWAV_ERROR; + #endif + #ifdef ELNRNG + case ELNRNG: return DRWAV_OUT_OF_RANGE; + #endif + #ifdef EUNATCH + case EUNATCH: return DRWAV_ERROR; + #endif + #ifdef ENOCSI + case ENOCSI: return DRWAV_ERROR; + #endif + #ifdef EL2HLT + case EL2HLT: return DRWAV_ERROR; + #endif + #ifdef EBADE + case EBADE: return DRWAV_ERROR; + #endif + #ifdef EBADR + case EBADR: return DRWAV_ERROR; + #endif + #ifdef EXFULL + case EXFULL: return DRWAV_ERROR; + #endif + #ifdef ENOANO + case ENOANO: return DRWAV_ERROR; + #endif + #ifdef EBADRQC + case EBADRQC: return DRWAV_ERROR; + #endif + #ifdef EBADSLT + case EBADSLT: return DRWAV_ERROR; + #endif + #ifdef EBFONT + case EBFONT: return DRWAV_INVALID_FILE; + #endif + #ifdef ENOSTR + case ENOSTR: return DRWAV_ERROR; + #endif + #ifdef ENODATA + case ENODATA: return DRWAV_NO_DATA_AVAILABLE; + #endif + #ifdef ETIME + case ETIME: return DRWAV_TIMEOUT; + #endif + #ifdef ENOSR + case ENOSR: return DRWAV_NO_DATA_AVAILABLE; + #endif + #ifdef ENONET + case ENONET: return DRWAV_NO_NETWORK; + #endif + #ifdef ENOPKG + case ENOPKG: return DRWAV_ERROR; + #endif + #ifdef EREMOTE + case EREMOTE: return DRWAV_ERROR; + #endif + #ifdef ENOLINK + case ENOLINK: return DRWAV_ERROR; + #endif + #ifdef EADV + case EADV: return DRWAV_ERROR; + #endif + #ifdef ESRMNT + case ESRMNT: return DRWAV_ERROR; + #endif + #ifdef ECOMM + case ECOMM: return DRWAV_ERROR; + #endif + #ifdef EPROTO + case EPROTO: return DRWAV_ERROR; + #endif + #ifdef EMULTIHOP + case EMULTIHOP: return DRWAV_ERROR; + #endif + #ifdef EDOTDOT + case EDOTDOT: return DRWAV_ERROR; + #endif + #ifdef EBADMSG + case EBADMSG: return DRWAV_BAD_MESSAGE; + #endif + #ifdef EOVERFLOW + case EOVERFLOW: return DRWAV_TOO_BIG; + #endif + #ifdef ENOTUNIQ + case ENOTUNIQ: return DRWAV_NOT_UNIQUE; + #endif + #ifdef EBADFD + case EBADFD: return DRWAV_ERROR; + #endif + #ifdef EREMCHG + case EREMCHG: return DRWAV_ERROR; + #endif + #ifdef ELIBACC + case ELIBACC: return DRWAV_ACCESS_DENIED; + #endif + #ifdef ELIBBAD + case ELIBBAD: return DRWAV_INVALID_FILE; + #endif + #ifdef ELIBSCN + case ELIBSCN: return DRWAV_INVALID_FILE; + #endif + #ifdef ELIBMAX + case ELIBMAX: return DRWAV_ERROR; + #endif + #ifdef ELIBEXEC + case ELIBEXEC: return DRWAV_ERROR; + #endif + #ifdef EILSEQ + case EILSEQ: return DRWAV_INVALID_DATA; + #endif + #ifdef ERESTART + case ERESTART: return DRWAV_ERROR; + #endif + #ifdef ESTRPIPE + case ESTRPIPE: return DRWAV_ERROR; + #endif + #ifdef EUSERS + case EUSERS: return DRWAV_ERROR; + #endif + #ifdef ENOTSOCK + case ENOTSOCK: return DRWAV_NOT_SOCKET; + #endif + #ifdef EDESTADDRREQ + case EDESTADDRREQ: return DRWAV_NO_ADDRESS; + #endif + #ifdef EMSGSIZE + case EMSGSIZE: return DRWAV_TOO_BIG; + #endif + #ifdef EPROTOTYPE + case EPROTOTYPE: return DRWAV_BAD_PROTOCOL; + #endif + #ifdef ENOPROTOOPT + case ENOPROTOOPT: return DRWAV_PROTOCOL_UNAVAILABLE; + #endif + #ifdef EPROTONOSUPPORT + case EPROTONOSUPPORT: return DRWAV_PROTOCOL_NOT_SUPPORTED; + #endif + #ifdef ESOCKTNOSUPPORT + case ESOCKTNOSUPPORT: return DRWAV_SOCKET_NOT_SUPPORTED; + #endif + #ifdef EOPNOTSUPP + case EOPNOTSUPP: return DRWAV_INVALID_OPERATION; + #endif + #ifdef EPFNOSUPPORT + case EPFNOSUPPORT: return DRWAV_PROTOCOL_FAMILY_NOT_SUPPORTED; + #endif + #ifdef EAFNOSUPPORT + case EAFNOSUPPORT: return DRWAV_ADDRESS_FAMILY_NOT_SUPPORTED; + #endif + #ifdef EADDRINUSE + case EADDRINUSE: return DRWAV_ALREADY_IN_USE; + #endif + #ifdef EADDRNOTAVAIL + case EADDRNOTAVAIL: return DRWAV_ERROR; + #endif + #ifdef ENETDOWN + case ENETDOWN: return DRWAV_NO_NETWORK; + #endif + #ifdef ENETUNREACH + case ENETUNREACH: return DRWAV_NO_NETWORK; + #endif + #ifdef ENETRESET + case ENETRESET: return DRWAV_NO_NETWORK; + #endif + #ifdef ECONNABORTED + case ECONNABORTED: return DRWAV_NO_NETWORK; + #endif + #ifdef ECONNRESET + case ECONNRESET: return DRWAV_CONNECTION_RESET; + #endif + #ifdef ENOBUFS + case ENOBUFS: return DRWAV_NO_SPACE; + #endif + #ifdef EISCONN + case EISCONN: return DRWAV_ALREADY_CONNECTED; + #endif + #ifdef ENOTCONN + case ENOTCONN: return DRWAV_NOT_CONNECTED; + #endif + #ifdef ESHUTDOWN + case ESHUTDOWN: return DRWAV_ERROR; + #endif + #ifdef ETOOMANYREFS + case ETOOMANYREFS: return DRWAV_ERROR; + #endif + #ifdef ETIMEDOUT + case ETIMEDOUT: return DRWAV_TIMEOUT; + #endif + #ifdef ECONNREFUSED + case ECONNREFUSED: return DRWAV_CONNECTION_REFUSED; + #endif + #ifdef EHOSTDOWN + case EHOSTDOWN: return DRWAV_NO_HOST; + #endif + #ifdef EHOSTUNREACH + case EHOSTUNREACH: return DRWAV_NO_HOST; + #endif + #ifdef EALREADY + case EALREADY: return DRWAV_IN_PROGRESS; + #endif + #ifdef EINPROGRESS + case EINPROGRESS: return DRWAV_IN_PROGRESS; + #endif + #ifdef ESTALE + case ESTALE: return DRWAV_INVALID_FILE; + #endif + #ifdef EUCLEAN + case EUCLEAN: return DRWAV_ERROR; + #endif + #ifdef ENOTNAM + case ENOTNAM: return DRWAV_ERROR; + #endif + #ifdef ENAVAIL + case ENAVAIL: return DRWAV_ERROR; + #endif + #ifdef EISNAM + case EISNAM: return DRWAV_ERROR; + #endif + #ifdef EREMOTEIO + case EREMOTEIO: return DRWAV_IO_ERROR; + #endif + #ifdef EDQUOT + case EDQUOT: return DRWAV_NO_SPACE; + #endif + #ifdef ENOMEDIUM + case ENOMEDIUM: return DRWAV_DOES_NOT_EXIST; + #endif + #ifdef EMEDIUMTYPE + case EMEDIUMTYPE: return DRWAV_ERROR; + #endif + #ifdef ECANCELED + case ECANCELED: return DRWAV_CANCELLED; + #endif + #ifdef ENOKEY + case ENOKEY: return DRWAV_ERROR; + #endif + #ifdef EKEYEXPIRED + case EKEYEXPIRED: return DRWAV_ERROR; + #endif + #ifdef EKEYREVOKED + case EKEYREVOKED: return DRWAV_ERROR; + #endif + #ifdef EKEYREJECTED + case EKEYREJECTED: return DRWAV_ERROR; + #endif + #ifdef EOWNERDEAD + case EOWNERDEAD: return DRWAV_ERROR; + #endif + #ifdef ENOTRECOVERABLE + case ENOTRECOVERABLE: return DRWAV_ERROR; + #endif + #ifdef ERFKILL + case ERFKILL: return DRWAV_ERROR; + #endif + #ifdef EHWPOISON + case EHWPOISON: return DRWAV_ERROR; + #endif + default: return DRWAV_ERROR; + } +} +/* End Errno */ + +/* fopen */ +DRWAV_PRIVATE drwav_result drwav_fopen(FILE** ppFile, const char* pFilePath, const char* pOpenMode) +{ +#if defined(_MSC_VER) && _MSC_VER >= 1400 + errno_t err; +#endif + + if (ppFile != NULL) { + *ppFile = NULL; /* Safety. */ + } + + if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) { + return DRWAV_INVALID_ARGS; + } + +#if defined(_MSC_VER) && _MSC_VER >= 1400 + err = fopen_s(ppFile, pFilePath, pOpenMode); + if (err != 0) { + return drwav_result_from_errno(err); + } +#else +#if defined(_WIN32) || defined(__APPLE__) + *ppFile = fopen(pFilePath, pOpenMode); +#else + #if defined(_FILE_OFFSET_BITS) && _FILE_OFFSET_BITS == 64 && defined(_LARGEFILE64_SOURCE) + *ppFile = fopen64(pFilePath, pOpenMode); + #else + *ppFile = fopen(pFilePath, pOpenMode); + #endif +#endif + if (*ppFile == NULL) { + drwav_result result = drwav_result_from_errno(errno); + if (result == DRWAV_SUCCESS) { + result = DRWAV_ERROR; /* Just a safety check to make sure we never ever return success when pFile == NULL. */ + } + + return result; + } +#endif + + return DRWAV_SUCCESS; +} + +/* +_wfopen() isn't always available in all compilation environments. + + * Windows only. + * MSVC seems to support it universally as far back as VC6 from what I can tell (haven't checked further back). + * MinGW-64 (both 32- and 64-bit) seems to support it. + * MinGW wraps it in !defined(__STRICT_ANSI__). + * OpenWatcom wraps it in !defined(_NO_EXT_KEYS). + +This can be reviewed as compatibility issues arise. The preference is to use _wfopen_s() and _wfopen() as opposed to the wcsrtombs() +fallback, so if you notice your compiler not detecting this properly I'm happy to look at adding support. +*/ +#if defined(_WIN32) + #if defined(_MSC_VER) || defined(__MINGW64__) || (!defined(__STRICT_ANSI__) && !defined(_NO_EXT_KEYS)) + #define DRWAV_HAS_WFOPEN + #endif +#endif + +#ifndef DR_WAV_NO_WCHAR +DRWAV_PRIVATE drwav_result drwav_wfopen(FILE** ppFile, const wchar_t* pFilePath, const wchar_t* pOpenMode, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (ppFile != NULL) { + *ppFile = NULL; /* Safety. */ + } + + if (pFilePath == NULL || pOpenMode == NULL || ppFile == NULL) { + return DRWAV_INVALID_ARGS; + } + +#if defined(DRWAV_HAS_WFOPEN) + { + /* Use _wfopen() on Windows. */ + #if defined(_MSC_VER) && _MSC_VER >= 1400 + errno_t err = _wfopen_s(ppFile, pFilePath, pOpenMode); + if (err != 0) { + return drwav_result_from_errno(err); + } + #else + *ppFile = _wfopen(pFilePath, pOpenMode); + if (*ppFile == NULL) { + return drwav_result_from_errno(errno); + } + #endif + (void)pAllocationCallbacks; + } +#else + /* + Use fopen() on anything other than Windows. Requires a conversion. This is annoying because + fopen() is locale specific. The only real way I can think of to do this is with wcsrtombs(). Note + that wcstombs() is apparently not thread-safe because it uses a static global mbstate_t object for + maintaining state. I've checked this with -std=c89 and it works, but if somebody get's a compiler + error I'll look into improving compatibility. + */ + + /* + Some compilers don't support wchar_t or wcsrtombs() which we're using below. In this case we just + need to abort with an error. If you encounter a compiler lacking such support, add it to this list + and submit a bug report and it'll be added to the library upstream. + */ + #if defined(__DJGPP__) + { + /* Nothing to do here. This will fall through to the error check below. */ + } + #else + { + mbstate_t mbs; + size_t lenMB; + const wchar_t* pFilePathTemp = pFilePath; + char* pFilePathMB = NULL; + char pOpenModeMB[32] = {0}; + + /* Get the length first. */ + DRWAV_ZERO_OBJECT(&mbs); + lenMB = wcsrtombs(NULL, &pFilePathTemp, 0, &mbs); + if (lenMB == (size_t)-1) { + return drwav_result_from_errno(errno); + } + + pFilePathMB = (char*)drwav__malloc_from_callbacks(lenMB + 1, pAllocationCallbacks); + if (pFilePathMB == NULL) { + return DRWAV_OUT_OF_MEMORY; + } + + pFilePathTemp = pFilePath; + DRWAV_ZERO_OBJECT(&mbs); + wcsrtombs(pFilePathMB, &pFilePathTemp, lenMB + 1, &mbs); + + /* The open mode should always consist of ASCII characters so we should be able to do a trivial conversion. */ + { + size_t i = 0; + for (;;) { + if (pOpenMode[i] == 0) { + pOpenModeMB[i] = '\0'; + break; + } + + pOpenModeMB[i] = (char)pOpenMode[i]; + i += 1; + } + } + + *ppFile = fopen(pFilePathMB, pOpenModeMB); + + drwav__free_from_callbacks(pFilePathMB, pAllocationCallbacks); + } + #endif + + if (*ppFile == NULL) { + return DRWAV_ERROR; + } +#endif + + return DRWAV_SUCCESS; +} +#endif +/* End fopen */ + + +DRWAV_PRIVATE size_t drwav__on_read_stdio(void* pUserData, void* pBufferOut, size_t bytesToRead) +{ + return fread(pBufferOut, 1, bytesToRead, (FILE*)pUserData); +} + +DRWAV_PRIVATE size_t drwav__on_write_stdio(void* pUserData, const void* pData, size_t bytesToWrite) +{ + return fwrite(pData, 1, bytesToWrite, (FILE*)pUserData); +} + +DRWAV_PRIVATE drwav_bool32 drwav__on_seek_stdio(void* pUserData, int offset, drwav_seek_origin origin) +{ + int whence = SEEK_SET; + if (origin == DRWAV_SEEK_CUR) { + whence = SEEK_CUR; + } else if (origin == DRWAV_SEEK_END) { + whence = SEEK_END; + } + + return fseek((FILE*)pUserData, offset, whence) == 0; +} + +DRWAV_PRIVATE drwav_bool32 drwav__on_tell_stdio(void* pUserData, drwav_int64* pCursor) +{ + FILE* pFileStdio = (FILE*)pUserData; + drwav_int64 result; + + /* These were all validated at a higher level. */ + DRWAV_ASSERT(pFileStdio != NULL); + DRWAV_ASSERT(pCursor != NULL); + +#if defined(_WIN32) && !defined(NXDK) + #if defined(_MSC_VER) && _MSC_VER > 1200 + result = _ftelli64(pFileStdio); + #else + result = ftell(pFileStdio); + #endif +#else + result = ftell(pFileStdio); +#endif + + *pCursor = result; + + return DRWAV_TRUE; +} + +DRWAV_API drwav_bool32 drwav_init_file(drwav* pWav, const char* filename, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_ex(pWav, filename, NULL, NULL, 0, pAllocationCallbacks); +} + + +DRWAV_PRIVATE drwav_bool32 drwav_init_file__internal_FILE(drwav* pWav, FILE* pFile, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav_bool32 result; + + result = drwav_preinit(pWav, drwav__on_read_stdio, drwav__on_seek_stdio, drwav__on_tell_stdio, (void*)pFile, pAllocationCallbacks); + if (result != DRWAV_TRUE) { + fclose(pFile); + return result; + } + + result = drwav_init__internal(pWav, onChunk, pChunkUserData, flags); + if (result != DRWAV_TRUE) { + fclose(pFile); + return result; + } + + return DRWAV_TRUE; +} + +DRWAV_API drwav_bool32 drwav_init_file_ex(drwav* pWav, const char* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + FILE* pFile; + if (drwav_fopen(&pFile, filename, "rb") != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + /* This takes ownership of the FILE* object. */ + return drwav_init_file__internal_FILE(pWav, pFile, onChunk, pChunkUserData, flags, pAllocationCallbacks); +} + +#ifndef DR_WAV_NO_WCHAR +DRWAV_API drwav_bool32 drwav_init_file_w(drwav* pWav, const wchar_t* filename, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_ex_w(pWav, filename, NULL, NULL, 0, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_ex_w(drwav* pWav, const wchar_t* filename, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + FILE* pFile; + if (drwav_wfopen(&pFile, filename, L"rb", pAllocationCallbacks) != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + /* This takes ownership of the FILE* object. */ + return drwav_init_file__internal_FILE(pWav, pFile, onChunk, pChunkUserData, flags, pAllocationCallbacks); +} +#endif + +DRWAV_API drwav_bool32 drwav_init_file_with_metadata(drwav* pWav, const char* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + FILE* pFile; + if (drwav_fopen(&pFile, filename, "rb") != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + /* This takes ownership of the FILE* object. */ + return drwav_init_file__internal_FILE(pWav, pFile, NULL, NULL, flags | DRWAV_WITH_METADATA, pAllocationCallbacks); +} + +#ifndef DR_WAV_NO_WCHAR +DRWAV_API drwav_bool32 drwav_init_file_with_metadata_w(drwav* pWav, const wchar_t* filename, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + FILE* pFile; + if (drwav_wfopen(&pFile, filename, L"rb", pAllocationCallbacks) != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + /* This takes ownership of the FILE* object. */ + return drwav_init_file__internal_FILE(pWav, pFile, NULL, NULL, flags | DRWAV_WITH_METADATA, pAllocationCallbacks); +} +#endif + + +DRWAV_PRIVATE drwav_bool32 drwav_init_file_write__internal_FILE(drwav* pWav, FILE* pFile, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav_bool32 result; + + result = drwav_preinit_write(pWav, pFormat, isSequential, drwav__on_write_stdio, drwav__on_seek_stdio, (void*)pFile, pAllocationCallbacks); + if (result != DRWAV_TRUE) { + fclose(pFile); + return result; + } + + result = drwav_init_write__internal(pWav, pFormat, totalSampleCount); + if (result != DRWAV_TRUE) { + fclose(pFile); + return result; + } + + return DRWAV_TRUE; +} + +DRWAV_PRIVATE drwav_bool32 drwav_init_file_write__internal(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + FILE* pFile; + if (drwav_fopen(&pFile, filename, "wb") != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + /* This takes ownership of the FILE* object. */ + return drwav_init_file_write__internal_FILE(pWav, pFile, pFormat, totalSampleCount, isSequential, pAllocationCallbacks); +} + +#ifndef DR_WAV_NO_WCHAR +DRWAV_PRIVATE drwav_bool32 drwav_init_file_write_w__internal(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + FILE* pFile; + if (drwav_wfopen(&pFile, filename, L"wb", pAllocationCallbacks) != DRWAV_SUCCESS) { + return DRWAV_FALSE; + } + + /* This takes ownership of the FILE* object. */ + return drwav_init_file_write__internal_FILE(pWav, pFile, pFormat, totalSampleCount, isSequential, pAllocationCallbacks); +} +#endif + +DRWAV_API drwav_bool32 drwav_init_file_write(drwav* pWav, const char* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_write__internal(pWav, filename, pFormat, 0, DRWAV_FALSE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_write_sequential(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_write__internal(pWav, filename, pFormat, totalSampleCount, DRWAV_TRUE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames(drwav* pWav, const char* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pFormat == NULL) { + return DRWAV_FALSE; + } + + return drwav_init_file_write_sequential(pWav, filename, pFormat, totalPCMFrameCount*pFormat->channels, pAllocationCallbacks); +} + +#ifndef DR_WAV_NO_WCHAR +DRWAV_API drwav_bool32 drwav_init_file_write_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_write_w__internal(pWav, filename, pFormat, 0, DRWAV_FALSE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_file_write_w__internal(pWav, filename, pFormat, totalSampleCount, DRWAV_TRUE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_file_write_sequential_pcm_frames_w(drwav* pWav, const wchar_t* filename, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pFormat == NULL) { + return DRWAV_FALSE; + } + + return drwav_init_file_write_sequential_w(pWav, filename, pFormat, totalPCMFrameCount*pFormat->channels, pAllocationCallbacks); +} +#endif +#endif /* DR_WAV_NO_STDIO */ + + +DRWAV_PRIVATE size_t drwav__on_read_memory(void* pUserData, void* pBufferOut, size_t bytesToRead) +{ + drwav* pWav = (drwav*)pUserData; + size_t bytesRemaining; + + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->memoryStream.dataSize >= pWav->memoryStream.currentReadPos); + + bytesRemaining = pWav->memoryStream.dataSize - pWav->memoryStream.currentReadPos; + if (bytesToRead > bytesRemaining) { + bytesToRead = bytesRemaining; + } + + if (bytesToRead > 0) { + DRWAV_COPY_MEMORY(pBufferOut, pWav->memoryStream.data + pWav->memoryStream.currentReadPos, bytesToRead); + pWav->memoryStream.currentReadPos += bytesToRead; + } + + return bytesToRead; +} + +DRWAV_PRIVATE drwav_bool32 drwav__on_seek_memory(void* pUserData, int offset, drwav_seek_origin origin) +{ + drwav* pWav = (drwav*)pUserData; + drwav_int64 newCursor; + + DRWAV_ASSERT(pWav != NULL); + + if (origin == DRWAV_SEEK_SET) { + newCursor = 0; + } else if (origin == DRWAV_SEEK_CUR) { + newCursor = (drwav_int64)pWav->memoryStream.currentReadPos; + } else if (origin == DRWAV_SEEK_END) { + newCursor = (drwav_int64)pWav->memoryStream.dataSize; + } else { + DRWAV_ASSERT(!"Invalid seek origin"); + return DRWAV_FALSE; + } + + newCursor += offset; + + if (newCursor < 0) { + return DRWAV_FALSE; /* Trying to seek prior to the start of the buffer. */ + } + if ((size_t)newCursor > pWav->memoryStream.dataSize) { + return DRWAV_FALSE; /* Trying to seek beyond the end of the buffer. */ + } + + pWav->memoryStream.currentReadPos = (size_t)newCursor; + + return DRWAV_TRUE; +} + +DRWAV_PRIVATE size_t drwav__on_write_memory(void* pUserData, const void* pDataIn, size_t bytesToWrite) +{ + drwav* pWav = (drwav*)pUserData; + size_t bytesRemaining; + + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pWav->memoryStreamWrite.dataCapacity >= pWav->memoryStreamWrite.currentWritePos); + + bytesRemaining = pWav->memoryStreamWrite.dataCapacity - pWav->memoryStreamWrite.currentWritePos; + if (bytesRemaining < bytesToWrite) { + /* Need to reallocate. */ + void* pNewData; + size_t newDataCapacity = (pWav->memoryStreamWrite.dataCapacity == 0) ? 256 : pWav->memoryStreamWrite.dataCapacity * 2; + + /* If doubling wasn't enough, just make it the minimum required size to write the data. */ + if ((newDataCapacity - pWav->memoryStreamWrite.currentWritePos) < bytesToWrite) { + newDataCapacity = pWav->memoryStreamWrite.currentWritePos + bytesToWrite; + } + + pNewData = drwav__realloc_from_callbacks(*pWav->memoryStreamWrite.ppData, newDataCapacity, pWav->memoryStreamWrite.dataCapacity, &pWav->allocationCallbacks); + if (pNewData == NULL) { + return 0; + } + + *pWav->memoryStreamWrite.ppData = pNewData; + pWav->memoryStreamWrite.dataCapacity = newDataCapacity; + } + + DRWAV_COPY_MEMORY(((drwav_uint8*)(*pWav->memoryStreamWrite.ppData)) + pWav->memoryStreamWrite.currentWritePos, pDataIn, bytesToWrite); + + pWav->memoryStreamWrite.currentWritePos += bytesToWrite; + if (pWav->memoryStreamWrite.dataSize < pWav->memoryStreamWrite.currentWritePos) { + pWav->memoryStreamWrite.dataSize = pWav->memoryStreamWrite.currentWritePos; + } + + *pWav->memoryStreamWrite.pDataSize = pWav->memoryStreamWrite.dataSize; + + return bytesToWrite; +} + +DRWAV_PRIVATE drwav_bool32 drwav__on_seek_memory_write(void* pUserData, int offset, drwav_seek_origin origin) +{ + drwav* pWav = (drwav*)pUserData; + drwav_int64 newCursor; + + DRWAV_ASSERT(pWav != NULL); + + if (origin == DRWAV_SEEK_SET) { + newCursor = 0; + } else if (origin == DRWAV_SEEK_CUR) { + newCursor = (drwav_int64)pWav->memoryStreamWrite.currentWritePos; + } else if (origin == DRWAV_SEEK_END) { + newCursor = (drwav_int64)pWav->memoryStreamWrite.dataSize; + } else { + DRWAV_ASSERT(!"Invalid seek origin"); + return DRWAV_FALSE; + } + + newCursor += offset; + + if (newCursor < 0) { + return DRWAV_FALSE; /* Trying to seek prior to the start of the buffer. */ + } + if ((size_t)newCursor > pWav->memoryStreamWrite.dataSize) { + return DRWAV_FALSE; /* Trying to seek beyond the end of the buffer. */ + } + + pWav->memoryStreamWrite.currentWritePos = (size_t)newCursor; + + return DRWAV_TRUE; +} + +DRWAV_PRIVATE drwav_bool32 drwav__on_tell_memory(void* pUserData, drwav_int64* pCursor) +{ + drwav* pWav = (drwav*)pUserData; + + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(pCursor != NULL); + + *pCursor = (drwav_int64)pWav->memoryStream.currentReadPos; + return DRWAV_TRUE; +} + +DRWAV_API drwav_bool32 drwav_init_memory(drwav* pWav, const void* data, size_t dataSize, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_memory_ex(pWav, data, dataSize, NULL, NULL, 0, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_memory_ex(drwav* pWav, const void* data, size_t dataSize, drwav_chunk_proc onChunk, void* pChunkUserData, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (data == NULL || dataSize == 0) { + return DRWAV_FALSE; + } + + if (!drwav_preinit(pWav, drwav__on_read_memory, drwav__on_seek_memory, drwav__on_tell_memory, pWav, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + pWav->memoryStream.data = (const drwav_uint8*)data; + pWav->memoryStream.dataSize = dataSize; + pWav->memoryStream.currentReadPos = 0; + + return drwav_init__internal(pWav, onChunk, pChunkUserData, flags); +} + +DRWAV_API drwav_bool32 drwav_init_memory_with_metadata(drwav* pWav, const void* data, size_t dataSize, drwav_uint32 flags, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (data == NULL || dataSize == 0) { + return DRWAV_FALSE; + } + + if (!drwav_preinit(pWav, drwav__on_read_memory, drwav__on_seek_memory, drwav__on_tell_memory, pWav, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + pWav->memoryStream.data = (const drwav_uint8*)data; + pWav->memoryStream.dataSize = dataSize; + pWav->memoryStream.currentReadPos = 0; + + return drwav_init__internal(pWav, NULL, NULL, flags | DRWAV_WITH_METADATA); +} + + +DRWAV_PRIVATE drwav_bool32 drwav_init_memory_write__internal(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, drwav_bool32 isSequential, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (ppData == NULL || pDataSize == NULL) { + return DRWAV_FALSE; + } + + *ppData = NULL; /* Important because we're using realloc()! */ + *pDataSize = 0; + + if (!drwav_preinit_write(pWav, pFormat, isSequential, drwav__on_write_memory, drwav__on_seek_memory_write, pWav, pAllocationCallbacks)) { + return DRWAV_FALSE; + } + + pWav->memoryStreamWrite.ppData = ppData; + pWav->memoryStreamWrite.pDataSize = pDataSize; + pWav->memoryStreamWrite.dataSize = 0; + pWav->memoryStreamWrite.dataCapacity = 0; + pWav->memoryStreamWrite.currentWritePos = 0; + + return drwav_init_write__internal(pWav, pFormat, totalSampleCount); +} + +DRWAV_API drwav_bool32 drwav_init_memory_write(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_memory_write__internal(pWav, ppData, pDataSize, pFormat, 0, DRWAV_FALSE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_memory_write_sequential(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalSampleCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + return drwav_init_memory_write__internal(pWav, ppData, pDataSize, pFormat, totalSampleCount, DRWAV_TRUE, pAllocationCallbacks); +} + +DRWAV_API drwav_bool32 drwav_init_memory_write_sequential_pcm_frames(drwav* pWav, void** ppData, size_t* pDataSize, const drwav_data_format* pFormat, drwav_uint64 totalPCMFrameCount, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pFormat == NULL) { + return DRWAV_FALSE; + } + + return drwav_init_memory_write_sequential(pWav, ppData, pDataSize, pFormat, totalPCMFrameCount*pFormat->channels, pAllocationCallbacks); +} + +DRWAV_PRIVATE drwav_uint32 drwav_write_padding(drwav* pWav) +{ + /* Do not adjust pWav->dataChunkDataSize - this should not include the padding. */ + drwav_uint32 paddingSize = drwav_calculate_padding_size(pWav->container, pWav->dataChunkDataSize); + + if (paddingSize > 0) { + drwav_uint64 paddingData = 0; + drwav__write(pWav, &paddingData, paddingSize); /* Byte order does not matter for this. */ + } + + return paddingSize; +} + +DRWAV_PRIVATE void drwav_write_chunk_sizes(drwav* pWav) +{ + /* + When using sequential mode, these will have been filled in at initialization time. We only need + to do this when using non-sequential mode. + */ + if (pWav->onSeek != NULL && !pWav->isSequentialWrite) { + if (pWav->container == drwav_container_riff) { + /* The "RIFF" chunk size. */ + if (pWav->onSeek(pWav->pUserData, 4, DRWAV_SEEK_SET)) { + drwav_uint32 riffChunkSize = drwav__riff_chunk_size_riff(pWav->dataChunkDataSize, pWav->pMetadata, pWav->metadataCount); + drwav__write_u32ne_to_le(pWav, riffChunkSize); + } + + /* The "data" chunk size. */ + if (pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos - 4, DRWAV_SEEK_SET)) { + drwav_uint32 dataChunkSize = drwav__data_chunk_size_riff(pWav->dataChunkDataSize); + drwav__write_u32ne_to_le(pWav, dataChunkSize); + } + } else if (pWav->container == drwav_container_w64) { + /* The "RIFF" chunk size. */ + if (pWav->onSeek(pWav->pUserData, 16, DRWAV_SEEK_SET)) { + drwav_uint64 riffChunkSize = drwav__riff_chunk_size_w64(pWav->dataChunkDataSize); + drwav__write_u64ne_to_le(pWav, riffChunkSize); + } + + /* The "data" chunk size. */ + if (pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos - 8, DRWAV_SEEK_SET)) { + drwav_uint64 dataChunkSize = drwav__data_chunk_size_w64(pWav->dataChunkDataSize); + drwav__write_u64ne_to_le(pWav, dataChunkSize); + } + } else if (pWav->container == drwav_container_rf64) { + /* We only need to update the ds64 chunk. The "RIFF" and "data" chunks always have their sizes set to 0xFFFFFFFF for RF64. */ + int ds64BodyPos = 12 + 8; + + /* The "RIFF" chunk size. */ + if (pWav->onSeek(pWav->pUserData, ds64BodyPos + 0, DRWAV_SEEK_SET)) { + drwav_uint64 riffChunkSize = drwav__riff_chunk_size_rf64(pWav->dataChunkDataSize, pWav->pMetadata, pWav->metadataCount); + drwav__write_u64ne_to_le(pWav, riffChunkSize); + } + + /* The "data" chunk size. */ + if (pWav->onSeek(pWav->pUserData, ds64BodyPos + 8, DRWAV_SEEK_SET)) { + drwav_uint64 dataChunkSize = drwav__data_chunk_size_rf64(pWav->dataChunkDataSize); + drwav__write_u64ne_to_le(pWav, dataChunkSize); + } + } + } +} + +DRWAV_API drwav_result drwav_uninit(drwav* pWav) +{ + drwav_result result = DRWAV_SUCCESS; + + if (pWav == NULL) { + return DRWAV_INVALID_ARGS; + } + + if (pWav->onWrite != NULL) { + if (pWav->isSequentialWrite) { + /* + Padding will not have been written in `drwav_write_*()` in sequential mode so we'll want to + do it explicitly here. + */ + drwav_write_padding(pWav); + + /* Validation for sequential mode. */ + if (pWav->dataChunkDataSize != pWav->dataChunkDataSizeTargetWrite) { + result = DRWAV_INVALID_FILE; + } + } + } else { + drwav_free(pWav->pMetadata, &pWav->allocationCallbacks); + } + +#ifndef DR_WAV_NO_STDIO + /* + If we opened the file with drwav_open_file() we will want to close the file handle. We can know whether or not drwav_open_file() + was used by looking at the onRead and onSeek callbacks. + */ + if (pWav->onRead == drwav__on_read_stdio || pWav->onWrite == drwav__on_write_stdio) { + fclose((FILE*)pWav->pUserData); + } +#endif + + return result; +} + + + +DRWAV_API size_t drwav_read_raw(drwav* pWav, size_t bytesToRead, void* pBufferOut) +{ + size_t bytesRead; + drwav_uint32 bytesPerFrame; + + if (pWav == NULL || bytesToRead == 0) { + return 0; /* Invalid args. */ + } + + if (bytesToRead > pWav->bytesRemaining) { + bytesToRead = (size_t)pWav->bytesRemaining; + } + + if (bytesToRead == 0) { + return 0; /* At end. */ + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; /* Could not determine the bytes per frame. */ + } + + if (pBufferOut != NULL) { + bytesRead = pWav->onRead(pWav->pUserData, pBufferOut, bytesToRead); + } else { + /* We need to seek. If we fail, we need to read-and-discard to make sure we get a good byte count. */ + bytesRead = 0; + while (bytesRead < bytesToRead) { + size_t bytesToSeek = (bytesToRead - bytesRead); + if (bytesToSeek > 0x7FFFFFFF) { + bytesToSeek = 0x7FFFFFFF; + } + + if (pWav->onSeek(pWav->pUserData, (int)bytesToSeek, DRWAV_SEEK_CUR) == DRWAV_FALSE) { + break; + } + + bytesRead += bytesToSeek; + } + + /* When we get here we may need to read-and-discard some data. */ + while (bytesRead < bytesToRead) { + drwav_uint8 buffer[4096]; + size_t bytesSeeked; + size_t bytesToSeek = (bytesToRead - bytesRead); + if (bytesToSeek > sizeof(buffer)) { + bytesToSeek = sizeof(buffer); + } + + bytesSeeked = pWav->onRead(pWav->pUserData, buffer, bytesToSeek); + bytesRead += bytesSeeked; + + if (bytesSeeked < bytesToSeek) { + break; /* Reached the end. */ + } + } + } + + pWav->readCursorInPCMFrames += bytesRead / bytesPerFrame; + + pWav->bytesRemaining -= bytesRead; + return bytesRead; +} + + + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_le(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut) +{ + drwav_uint32 bytesPerFrame; + drwav_uint64 bytesToRead; /* Intentionally uint64 instead of size_t so we can do a check that we're not reading too much on 32-bit builds. */ + drwav_uint64 framesRemainingInFile; + + if (pWav == NULL || framesToRead == 0) { + return 0; + } + + /* Cannot use this function for compressed formats. */ + if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) { + return 0; + } + + framesRemainingInFile = pWav->totalPCMFrameCount - pWav->readCursorInPCMFrames; + if (framesToRead > framesRemainingInFile) { + framesToRead = framesRemainingInFile; + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + /* Don't try to read more samples than can potentially fit in the output buffer. */ + bytesToRead = framesToRead * bytesPerFrame; + if (bytesToRead > DRWAV_SIZE_MAX) { + bytesToRead = (DRWAV_SIZE_MAX / bytesPerFrame) * bytesPerFrame; /* Round the number of bytes to read to a clean frame boundary. */ + } + + /* + Doing an explicit check here just to make it clear that we don't want to be attempt to read anything if there's no bytes to read. There + *could* be a time where it evaluates to 0 due to overflowing. + */ + if (bytesToRead == 0) { + return 0; + } + + return drwav_read_raw(pWav, (size_t)bytesToRead, pBufferOut) / bytesPerFrame; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_be(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_le(pWav, framesToRead, pBufferOut); + + if (pBufferOut != NULL) { + drwav_uint32 bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; /* Could not get the bytes per frame which means bytes per sample cannot be determined and we don't know how to byte swap. */ + } + + drwav__bswap_samples(pBufferOut, framesRead*pWav->channels, bytesPerFrame/pWav->channels); + } + + return framesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames(drwav* pWav, drwav_uint64 framesToRead, void* pBufferOut) +{ + drwav_uint64 framesRead = 0; + + if (drwav_is_container_be(pWav->container)) { + /* + Special case for AIFF. AIFF is a big-endian encoded format, but it supports a format that is + PCM in little-endian encoding. In this case, we fall through this branch and treate it as + little-endian. + */ + if (pWav->container != drwav_container_aiff || pWav->aiff.isLE == DRWAV_FALSE) { + if (drwav__is_little_endian()) { + framesRead = drwav_read_pcm_frames_be(pWav, framesToRead, pBufferOut); + } else { + framesRead = drwav_read_pcm_frames_le(pWav, framesToRead, pBufferOut); + } + + goto post_process; + } + } + + /* Getting here means the data should be considered little-endian. */ + if (drwav__is_little_endian()) { + framesRead = drwav_read_pcm_frames_le(pWav, framesToRead, pBufferOut); + } else { + framesRead = drwav_read_pcm_frames_be(pWav, framesToRead, pBufferOut); + } + + /* + Here is where we check if we need to do a signed/unsigned conversion for AIFF. The reason we need to do this + is because dr_wav always assumes an 8-bit sample is unsigned, whereas AIFF can have signed 8-bit formats. + */ + post_process: + { + if (pWav->container == drwav_container_aiff && pWav->bitsPerSample == 8 && pWav->aiff.isUnsigned == DRWAV_FALSE) { + if (pBufferOut != NULL) { + drwav_uint64 iSample; + + for (iSample = 0; iSample < framesRead * pWav->channels; iSample += 1) { + ((drwav_uint8*)pBufferOut)[iSample] += 128; + } + } + } + } + + return framesRead; +} + + + +DRWAV_PRIVATE drwav_bool32 drwav_seek_to_first_pcm_frame(drwav* pWav) +{ + if (pWav->onWrite != NULL) { + return DRWAV_FALSE; /* No seeking in write mode. */ + } + + if (!pWav->onSeek(pWav->pUserData, (int)pWav->dataChunkDataPos, DRWAV_SEEK_SET)) { + return DRWAV_FALSE; + } + + if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) { + /* Cached data needs to be cleared for compressed formats. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + DRWAV_ZERO_OBJECT(&pWav->msadpcm); + } else if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + DRWAV_ZERO_OBJECT(&pWav->ima); + } else { + DRWAV_ASSERT(DRWAV_FALSE); /* If this assertion is triggered it means I've implemented a new compressed format but forgot to add a branch for it here. */ + } + } + + pWav->readCursorInPCMFrames = 0; + pWav->bytesRemaining = pWav->dataChunkDataSize; + + return DRWAV_TRUE; +} + +DRWAV_API drwav_bool32 drwav_seek_to_pcm_frame(drwav* pWav, drwav_uint64 targetFrameIndex) +{ + /* Seeking should be compatible with wave files > 2GB. */ + + if (pWav == NULL || pWav->onSeek == NULL) { + return DRWAV_FALSE; + } + + /* No seeking in write mode. */ + if (pWav->onWrite != NULL) { + return DRWAV_FALSE; + } + + /* If there are no samples, just return DRWAV_TRUE without doing anything. */ + if (pWav->totalPCMFrameCount == 0) { + return DRWAV_TRUE; + } + + /* Make sure the sample is clamped. */ + if (targetFrameIndex > pWav->totalPCMFrameCount) { + targetFrameIndex = pWav->totalPCMFrameCount; + } + + /* + For compressed formats we just use a slow generic seek. If we are seeking forward we just seek forward. If we are going backwards we need + to seek back to the start. + */ + if (drwav__is_compressed_format_tag(pWav->translatedFormatTag)) { + /* TODO: This can be optimized. */ + + /* + If we're seeking forward it's simple - just keep reading samples until we hit the sample we're requesting. If we're seeking backwards, + we first need to seek back to the start and then just do the same thing as a forward seek. + */ + if (targetFrameIndex < pWav->readCursorInPCMFrames) { + if (!drwav_seek_to_first_pcm_frame(pWav)) { + return DRWAV_FALSE; + } + } + + if (targetFrameIndex > pWav->readCursorInPCMFrames) { + drwav_uint64 offsetInFrames = targetFrameIndex - pWav->readCursorInPCMFrames; + + drwav_int16 devnull[2048]; + while (offsetInFrames > 0) { + drwav_uint64 framesRead = 0; + drwav_uint64 framesToRead = offsetInFrames; + if (framesToRead > drwav_countof(devnull)/pWav->channels) { + framesToRead = drwav_countof(devnull)/pWav->channels; + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + framesRead = drwav_read_pcm_frames_s16__msadpcm(pWav, framesToRead, devnull); + } else if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + framesRead = drwav_read_pcm_frames_s16__ima(pWav, framesToRead, devnull); + } else { + DRWAV_ASSERT(DRWAV_FALSE); /* If this assertion is triggered it means I've implemented a new compressed format but forgot to add a branch for it here. */ + } + + if (framesRead != framesToRead) { + return DRWAV_FALSE; + } + + offsetInFrames -= framesRead; + } + } + } else { + drwav_uint64 totalSizeInBytes; + drwav_uint64 currentBytePos; + drwav_uint64 targetBytePos; + drwav_uint64 offset; + drwav_uint32 bytesPerFrame; + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return DRWAV_FALSE; /* Not able to calculate offset. */ + } + + totalSizeInBytes = pWav->totalPCMFrameCount * bytesPerFrame; + /*DRWAV_ASSERT(totalSizeInBytes >= pWav->bytesRemaining);*/ + + currentBytePos = totalSizeInBytes - pWav->bytesRemaining; + targetBytePos = targetFrameIndex * bytesPerFrame; + + if (currentBytePos < targetBytePos) { + /* Offset forwards. */ + offset = (targetBytePos - currentBytePos); + } else { + /* Offset backwards. */ + if (!drwav_seek_to_first_pcm_frame(pWav)) { + return DRWAV_FALSE; + } + offset = targetBytePos; + } + + while (offset > 0) { + int offset32 = ((offset > INT_MAX) ? INT_MAX : (int)offset); + if (!pWav->onSeek(pWav->pUserData, offset32, DRWAV_SEEK_CUR)) { + return DRWAV_FALSE; + } + + pWav->readCursorInPCMFrames += offset32 / bytesPerFrame; + pWav->bytesRemaining -= offset32; + offset -= offset32; + } + } + + return DRWAV_TRUE; +} + +DRWAV_API drwav_result drwav_get_cursor_in_pcm_frames(drwav* pWav, drwav_uint64* pCursor) +{ + if (pCursor == NULL) { + return DRWAV_INVALID_ARGS; + } + + *pCursor = 0; /* Safety. */ + + if (pWav == NULL) { + return DRWAV_INVALID_ARGS; + } + + *pCursor = pWav->readCursorInPCMFrames; + + return DRWAV_SUCCESS; +} + +DRWAV_API drwav_result drwav_get_length_in_pcm_frames(drwav* pWav, drwav_uint64* pLength) +{ + if (pLength == NULL) { + return DRWAV_INVALID_ARGS; + } + + *pLength = 0; /* Safety. */ + + if (pWav == NULL) { + return DRWAV_INVALID_ARGS; + } + + *pLength = pWav->totalPCMFrameCount; + + return DRWAV_SUCCESS; +} + + +DRWAV_API size_t drwav_write_raw(drwav* pWav, size_t bytesToWrite, const void* pData) +{ + size_t bytesWritten; + + if (pWav == NULL || bytesToWrite == 0 || pData == NULL) { + return 0; + } + + bytesWritten = pWav->onWrite(pWav->pUserData, pData, bytesToWrite); + pWav->dataChunkDataSize += bytesWritten; + + if (!pWav->isSequentialWrite) { + drwav_uint32 padding; + + /* Padding. */ + padding = drwav_write_padding(pWav); + + /* Chunk sizes. */ + drwav_write_chunk_sizes(pWav); + + /* Now seek back to just before the padding in preparation for the next writes. */ + if (pWav->onSeek != NULL) { + pWav->onSeek(pWav->pUserData, -(int)padding, DRWAV_SEEK_END); /* Safe cast. */ + } + } + + return bytesWritten; +} + +DRWAV_API drwav_uint64 drwav_write_pcm_frames_le(drwav* pWav, drwav_uint64 framesToWrite, const void* pData) +{ + drwav_uint64 bytesToWrite; + drwav_uint64 bytesWritten; + const drwav_uint8* pRunningData; + + if (pWav == NULL || framesToWrite == 0 || pData == NULL) { + return 0; + } + + bytesToWrite = ((framesToWrite * pWav->channels * pWav->bitsPerSample) / 8); + if (bytesToWrite > DRWAV_SIZE_MAX) { + return 0; + } + + bytesWritten = 0; + pRunningData = (const drwav_uint8*)pData; + + while (bytesToWrite > 0) { + size_t bytesJustWritten; + drwav_uint64 bytesToWriteThisIteration; + + bytesToWriteThisIteration = bytesToWrite; + DRWAV_ASSERT(bytesToWriteThisIteration <= DRWAV_SIZE_MAX); /* <-- This is checked above. */ + + bytesJustWritten = drwav_write_raw(pWav, (size_t)bytesToWriteThisIteration, pRunningData); + if (bytesJustWritten == 0) { + break; + } + + bytesToWrite -= bytesJustWritten; + bytesWritten += bytesJustWritten; + pRunningData += bytesJustWritten; + } + + return (bytesWritten * 8) / pWav->bitsPerSample / pWav->channels; +} + +DRWAV_API drwav_uint64 drwav_write_pcm_frames_be(drwav* pWav, drwav_uint64 framesToWrite, const void* pData) +{ + drwav_uint64 bytesToWrite; + drwav_uint64 bytesWritten; + drwav_uint32 bytesPerSample; + const drwav_uint8* pRunningData; + + if (pWav == NULL || framesToWrite == 0 || pData == NULL) { + return 0; + } + + bytesToWrite = ((framesToWrite * pWav->channels * pWav->bitsPerSample) / 8); + if (bytesToWrite > DRWAV_SIZE_MAX) { + return 0; + } + + bytesWritten = 0; + pRunningData = (const drwav_uint8*)pData; + + bytesPerSample = drwav_get_bytes_per_pcm_frame(pWav) / pWav->channels; + if (bytesPerSample == 0) { + return 0; /* Cannot determine bytes per sample, or bytes per sample is less than one byte. */ + } + + while (bytesToWrite > 0) { + drwav_uint8 temp[4096]; + drwav_uint32 sampleCount; + size_t bytesJustWritten; + drwav_uint64 bytesToWriteThisIteration; + + bytesToWriteThisIteration = bytesToWrite; + DRWAV_ASSERT(bytesToWriteThisIteration <= DRWAV_SIZE_MAX); /* <-- This is checked above. */ + + /* + WAV files are always little-endian. We need to byte swap on big-endian architectures. Since our input buffer is read-only we need + to use an intermediary buffer for the conversion. + */ + sampleCount = sizeof(temp)/bytesPerSample; + + if (bytesToWriteThisIteration > ((drwav_uint64)sampleCount)*bytesPerSample) { + bytesToWriteThisIteration = ((drwav_uint64)sampleCount)*bytesPerSample; + } + + DRWAV_COPY_MEMORY(temp, pRunningData, (size_t)bytesToWriteThisIteration); + drwav__bswap_samples(temp, sampleCount, bytesPerSample); + + bytesJustWritten = drwav_write_raw(pWav, (size_t)bytesToWriteThisIteration, temp); + if (bytesJustWritten == 0) { + break; + } + + bytesToWrite -= bytesJustWritten; + bytesWritten += bytesJustWritten; + pRunningData += bytesJustWritten; + } + + return (bytesWritten * 8) / pWav->bitsPerSample / pWav->channels; +} + +DRWAV_API drwav_uint64 drwav_write_pcm_frames(drwav* pWav, drwav_uint64 framesToWrite, const void* pData) +{ + if (drwav__is_little_endian()) { + return drwav_write_pcm_frames_le(pWav, framesToWrite, pData); + } else { + return drwav_write_pcm_frames_be(pWav, framesToWrite, pData); + } +} + + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__msadpcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalFramesRead = 0; + + static const drwav_int32 adaptationTable[] = { + 230, 230, 230, 230, 307, 409, 512, 614, + 768, 614, 512, 409, 307, 230, 230, 230 + }; + static const drwav_int32 coeff1Table[] = { 256, 512, 0, 192, 240, 460, 392 }; + static const drwav_int32 coeff2Table[] = { 0, -256, 0, 64, 0, -208, -232 }; + + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(framesToRead > 0); + + /* TODO: Lots of room for optimization here. */ + + while (pWav->readCursorInPCMFrames < pWav->totalPCMFrameCount) { + DRWAV_ASSERT(framesToRead > 0); /* This loop iteration will never get hit with framesToRead == 0 because it's asserted at the top, and we check for 0 inside the loop just below. */ + + /* If there are no cached frames we need to load a new block. */ + if (pWav->msadpcm.cachedFrameCount == 0 && pWav->msadpcm.bytesRemainingInBlock == 0) { + if (pWav->channels == 1) { + /* Mono. */ + drwav_uint8 header[7]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalFramesRead; + } + pWav->msadpcm.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + pWav->msadpcm.predictor[0] = header[0]; + pWav->msadpcm.delta[0] = drwav_bytes_to_s16(header + 1); + pWav->msadpcm.prevFrames[0][1] = (drwav_int32)drwav_bytes_to_s16(header + 3); + pWav->msadpcm.prevFrames[0][0] = (drwav_int32)drwav_bytes_to_s16(header + 5); + pWav->msadpcm.cachedFrames[2] = pWav->msadpcm.prevFrames[0][0]; + pWav->msadpcm.cachedFrames[3] = pWav->msadpcm.prevFrames[0][1]; + pWav->msadpcm.cachedFrameCount = 2; + + /* The predictor is used as an index into coeff1Table so we'll need to validate to ensure it never overflows. */ + if (pWav->msadpcm.predictor[0] >= drwav_countof(coeff1Table) || pWav->msadpcm.predictor[0] >= drwav_countof(coeff2Table)) { + return totalFramesRead; /* Invalid file. */ + } + } else { + /* Stereo. */ + drwav_uint8 header[14]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalFramesRead; + } + pWav->msadpcm.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + pWav->msadpcm.predictor[0] = header[0]; + pWav->msadpcm.predictor[1] = header[1]; + pWav->msadpcm.delta[0] = drwav_bytes_to_s16(header + 2); + pWav->msadpcm.delta[1] = drwav_bytes_to_s16(header + 4); + pWav->msadpcm.prevFrames[0][1] = (drwav_int32)drwav_bytes_to_s16(header + 6); + pWav->msadpcm.prevFrames[1][1] = (drwav_int32)drwav_bytes_to_s16(header + 8); + pWav->msadpcm.prevFrames[0][0] = (drwav_int32)drwav_bytes_to_s16(header + 10); + pWav->msadpcm.prevFrames[1][0] = (drwav_int32)drwav_bytes_to_s16(header + 12); + + pWav->msadpcm.cachedFrames[0] = pWav->msadpcm.prevFrames[0][0]; + pWav->msadpcm.cachedFrames[1] = pWav->msadpcm.prevFrames[1][0]; + pWav->msadpcm.cachedFrames[2] = pWav->msadpcm.prevFrames[0][1]; + pWav->msadpcm.cachedFrames[3] = pWav->msadpcm.prevFrames[1][1]; + pWav->msadpcm.cachedFrameCount = 2; + + /* The predictor is used as an index into coeff1Table so we'll need to validate to ensure it never overflows. */ + if (pWav->msadpcm.predictor[0] >= drwav_countof(coeff1Table) || pWav->msadpcm.predictor[0] >= drwav_countof(coeff2Table) || + pWav->msadpcm.predictor[1] >= drwav_countof(coeff1Table) || pWav->msadpcm.predictor[1] >= drwav_countof(coeff2Table)) { + return totalFramesRead; /* Invalid file. */ + } + } + } + + /* Output anything that's cached. */ + while (framesToRead > 0 && pWav->msadpcm.cachedFrameCount > 0 && pWav->readCursorInPCMFrames < pWav->totalPCMFrameCount) { + if (pBufferOut != NULL) { + drwav_uint32 iSample = 0; + for (iSample = 0; iSample < pWav->channels; iSample += 1) { + pBufferOut[iSample] = (drwav_int16)pWav->msadpcm.cachedFrames[(drwav_countof(pWav->msadpcm.cachedFrames) - (pWav->msadpcm.cachedFrameCount*pWav->channels)) + iSample]; + } + + pBufferOut += pWav->channels; + } + + framesToRead -= 1; + totalFramesRead += 1; + pWav->readCursorInPCMFrames += 1; + pWav->msadpcm.cachedFrameCount -= 1; + } + + if (framesToRead == 0) { + break; + } + + + /* + If there's nothing left in the cache, just go ahead and load more. If there's nothing left to load in the current block we just continue to the next + loop iteration which will trigger the loading of a new block. + */ + if (pWav->msadpcm.cachedFrameCount == 0) { + if (pWav->msadpcm.bytesRemainingInBlock == 0) { + continue; + } else { + drwav_uint8 nibbles; + drwav_int32 nibble0; + drwav_int32 nibble1; + + if (pWav->onRead(pWav->pUserData, &nibbles, 1) != 1) { + return totalFramesRead; + } + pWav->msadpcm.bytesRemainingInBlock -= 1; + + /* TODO: Optimize away these if statements. */ + nibble0 = ((nibbles & 0xF0) >> 4); if ((nibbles & 0x80)) { nibble0 |= 0xFFFFFFF0UL; } + nibble1 = ((nibbles & 0x0F) >> 0); if ((nibbles & 0x08)) { nibble1 |= 0xFFFFFFF0UL; } + + if (pWav->channels == 1) { + /* Mono. */ + drwav_int32 newSample0; + drwav_int32 newSample1; + + /* The predictor is read from the file and then indexed into a table. Check that it's in bounds. */ + if (pWav->msadpcm.predictor[0] >= drwav_countof(coeff1Table) || pWav->msadpcm.predictor[0] >= drwav_countof(coeff2Table)) { + return totalFramesRead; + } + + newSample0 = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8; + newSample0 += nibble0 * pWav->msadpcm.delta[0]; + newSample0 = drwav_clamp(newSample0, -32768, 32767); + + pWav->msadpcm.delta[0] = (drwav_int32)drwav_clamp(((drwav_int64)adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8, 16, 0x7FFFFFFF); + + pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1]; + pWav->msadpcm.prevFrames[0][1] = newSample0; + + + newSample1 = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8; + newSample1 += nibble1 * pWav->msadpcm.delta[0]; + newSample1 = drwav_clamp(newSample1, -32768, 32767); + + pWav->msadpcm.delta[0] = (drwav_int32)drwav_clamp(((drwav_int64)adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[0]) >> 8, 16, 0x7FFFFFFF); + + pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1]; + pWav->msadpcm.prevFrames[0][1] = newSample1; + + pWav->msadpcm.cachedFrames[2] = newSample0; + pWav->msadpcm.cachedFrames[3] = newSample1; + pWav->msadpcm.cachedFrameCount = 2; + } else { + /* Stereo. */ + drwav_int32 newSample0; + drwav_int32 newSample1; + + /* Left. */ + if (pWav->msadpcm.predictor[0] >= drwav_countof(coeff1Table) || pWav->msadpcm.predictor[0] >= drwav_countof(coeff2Table)) { + return totalFramesRead; /* Out of bounds. Invalid file. */ + } + + newSample0 = ((pWav->msadpcm.prevFrames[0][1] * coeff1Table[pWav->msadpcm.predictor[0]]) + (pWav->msadpcm.prevFrames[0][0] * coeff2Table[pWav->msadpcm.predictor[0]])) >> 8; + newSample0 += nibble0 * pWav->msadpcm.delta[0]; + newSample0 = drwav_clamp(newSample0, -32768, 32767); + + pWav->msadpcm.delta[0] = (drwav_int32)drwav_clamp(((drwav_int64)adaptationTable[((nibbles & 0xF0) >> 4)] * pWav->msadpcm.delta[0]) >> 8, 16, 0x7FFFFFFF); + + pWav->msadpcm.prevFrames[0][0] = pWav->msadpcm.prevFrames[0][1]; + pWav->msadpcm.prevFrames[0][1] = newSample0; + + + /* Right. */ + if (pWav->msadpcm.predictor[1] >= drwav_countof(coeff1Table) || pWav->msadpcm.predictor[1] >= drwav_countof(coeff2Table)) { + return totalFramesRead; /* Out of bounds. Invalid file. */ + } + + newSample1 = ((pWav->msadpcm.prevFrames[1][1] * coeff1Table[pWav->msadpcm.predictor[1]]) + (pWav->msadpcm.prevFrames[1][0] * coeff2Table[pWav->msadpcm.predictor[1]])) >> 8; + newSample1 += nibble1 * pWav->msadpcm.delta[1]; + newSample1 = drwav_clamp(newSample1, -32768, 32767); + + pWav->msadpcm.delta[1] = (drwav_int32)drwav_clamp(((drwav_int64)adaptationTable[((nibbles & 0x0F) >> 0)] * pWav->msadpcm.delta[1]) >> 8, 16, 0x7FFFFFFF); + + pWav->msadpcm.prevFrames[1][0] = pWav->msadpcm.prevFrames[1][1]; + pWav->msadpcm.prevFrames[1][1] = newSample1; + + pWav->msadpcm.cachedFrames[2] = newSample0; + pWav->msadpcm.cachedFrames[3] = newSample1; + pWav->msadpcm.cachedFrameCount = 1; + } + } + } + } + + return totalFramesRead; +} + + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__ima(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalFramesRead = 0; + drwav_uint32 iChannel; + + static const drwav_int32 indexTable[16] = { + -1, -1, -1, -1, 2, 4, 6, 8, + -1, -1, -1, -1, 2, 4, 6, 8 + }; + + static const drwav_int32 stepTable[89] = { + 7, 8, 9, 10, 11, 12, 13, 14, 16, 17, + 19, 21, 23, 25, 28, 31, 34, 37, 41, 45, + 50, 55, 60, 66, 73, 80, 88, 97, 107, 118, + 130, 143, 157, 173, 190, 209, 230, 253, 279, 307, + 337, 371, 408, 449, 494, 544, 598, 658, 724, 796, + 876, 963, 1060, 1166, 1282, 1411, 1552, 1707, 1878, 2066, + 2272, 2499, 2749, 3024, 3327, 3660, 4026, 4428, 4871, 5358, + 5894, 6484, 7132, 7845, 8630, 9493, 10442, 11487, 12635, 13899, + 15289, 16818, 18500, 20350, 22385, 24623, 27086, 29794, 32767 + }; + + DRWAV_ASSERT(pWav != NULL); + DRWAV_ASSERT(framesToRead > 0); + + /* TODO: Lots of room for optimization here. */ + + while (pWav->readCursorInPCMFrames < pWav->totalPCMFrameCount) { + DRWAV_ASSERT(framesToRead > 0); /* This loop iteration will never get hit with framesToRead == 0 because it's asserted at the top, and we check for 0 inside the loop just below. */ + + /* If there are no cached samples we need to load a new block. */ + if (pWav->ima.cachedFrameCount == 0 && pWav->ima.bytesRemainingInBlock == 0) { + if (pWav->channels == 1) { + /* Mono. */ + drwav_uint8 header[4]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalFramesRead; + } + pWav->ima.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + if (header[2] >= drwav_countof(stepTable)) { + pWav->onSeek(pWav->pUserData, pWav->ima.bytesRemainingInBlock, DRWAV_SEEK_CUR); + pWav->ima.bytesRemainingInBlock = 0; + return totalFramesRead; /* Invalid data. */ + } + + pWav->ima.predictor[0] = (drwav_int16)drwav_bytes_to_u16(header + 0); + pWav->ima.stepIndex[0] = drwav_clamp(header[2], 0, (drwav_int32)drwav_countof(stepTable)-1); /* Clamp not necessary because we checked above, but adding here to silence a static analysis warning. */ + pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 1] = pWav->ima.predictor[0]; + pWav->ima.cachedFrameCount = 1; + } else { + /* Stereo. */ + drwav_uint8 header[8]; + if (pWav->onRead(pWav->pUserData, header, sizeof(header)) != sizeof(header)) { + return totalFramesRead; + } + pWav->ima.bytesRemainingInBlock = pWav->fmt.blockAlign - sizeof(header); + + if (header[2] >= drwav_countof(stepTable) || header[6] >= drwav_countof(stepTable)) { + pWav->onSeek(pWav->pUserData, pWav->ima.bytesRemainingInBlock, DRWAV_SEEK_CUR); + pWav->ima.bytesRemainingInBlock = 0; + return totalFramesRead; /* Invalid data. */ + } + + pWav->ima.predictor[0] = drwav_bytes_to_s16(header + 0); + pWav->ima.stepIndex[0] = drwav_clamp(header[2], 0, (drwav_int32)drwav_countof(stepTable)-1); /* Clamp not necessary because we checked above, but adding here to silence a static analysis warning. */ + pWav->ima.predictor[1] = drwav_bytes_to_s16(header + 4); + pWav->ima.stepIndex[1] = drwav_clamp(header[6], 0, (drwav_int32)drwav_countof(stepTable)-1); /* Clamp not necessary because we checked above, but adding here to silence a static analysis warning. */ + + pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 2] = pWav->ima.predictor[0]; + pWav->ima.cachedFrames[drwav_countof(pWav->ima.cachedFrames) - 1] = pWav->ima.predictor[1]; + pWav->ima.cachedFrameCount = 1; + } + } + + /* Output anything that's cached. */ + while (framesToRead > 0 && pWav->ima.cachedFrameCount > 0 && pWav->readCursorInPCMFrames < pWav->totalPCMFrameCount) { + if (pBufferOut != NULL) { + drwav_uint32 iSample; + for (iSample = 0; iSample < pWav->channels; iSample += 1) { + pBufferOut[iSample] = (drwav_int16)pWav->ima.cachedFrames[(drwav_countof(pWav->ima.cachedFrames) - (pWav->ima.cachedFrameCount*pWav->channels)) + iSample]; + } + pBufferOut += pWav->channels; + } + + framesToRead -= 1; + totalFramesRead += 1; + pWav->readCursorInPCMFrames += 1; + pWav->ima.cachedFrameCount -= 1; + } + + if (framesToRead == 0) { + break; + } + + /* + If there's nothing left in the cache, just go ahead and load more. If there's nothing left to load in the current block we just continue to the next + loop iteration which will trigger the loading of a new block. + */ + if (pWav->ima.cachedFrameCount == 0) { + if (pWav->ima.bytesRemainingInBlock == 0) { + continue; + } else { + /* + From what I can tell with stereo streams, it looks like every 4 bytes (8 samples) is for one channel. So it goes 4 bytes for the + left channel, 4 bytes for the right channel. + */ + pWav->ima.cachedFrameCount = 8; + for (iChannel = 0; iChannel < pWav->channels; ++iChannel) { + drwav_uint32 iByte; + drwav_uint8 nibbles[4]; + if (pWav->onRead(pWav->pUserData, &nibbles, 4) != 4) { + pWav->ima.cachedFrameCount = 0; + return totalFramesRead; + } + pWav->ima.bytesRemainingInBlock -= 4; + + for (iByte = 0; iByte < 4; ++iByte) { + drwav_uint8 nibble0 = ((nibbles[iByte] & 0x0F) >> 0); + drwav_uint8 nibble1 = ((nibbles[iByte] & 0xF0) >> 4); + + drwav_int32 step = stepTable[pWav->ima.stepIndex[iChannel]]; + drwav_int32 predictor = pWav->ima.predictor[iChannel]; + + drwav_int32 diff = step >> 3; + if (nibble0 & 1) diff += step >> 2; + if (nibble0 & 2) diff += step >> 1; + if (nibble0 & 4) diff += step; + if (nibble0 & 8) diff = -diff; + + predictor = drwav_clamp(predictor + diff, -32768, 32767); + pWav->ima.predictor[iChannel] = predictor; + pWav->ima.stepIndex[iChannel] = drwav_clamp(pWav->ima.stepIndex[iChannel] + indexTable[nibble0], 0, (drwav_int32)drwav_countof(stepTable)-1); + pWav->ima.cachedFrames[(drwav_countof(pWav->ima.cachedFrames) - (pWav->ima.cachedFrameCount*pWav->channels)) + (iByte*2+0)*pWav->channels + iChannel] = predictor; + + + step = stepTable[pWav->ima.stepIndex[iChannel]]; + predictor = pWav->ima.predictor[iChannel]; + + diff = step >> 3; + if (nibble1 & 1) diff += step >> 2; + if (nibble1 & 2) diff += step >> 1; + if (nibble1 & 4) diff += step; + if (nibble1 & 8) diff = -diff; + + predictor = drwav_clamp(predictor + diff, -32768, 32767); + pWav->ima.predictor[iChannel] = predictor; + pWav->ima.stepIndex[iChannel] = drwav_clamp(pWav->ima.stepIndex[iChannel] + indexTable[nibble1], 0, (drwav_int32)drwav_countof(stepTable)-1); + pWav->ima.cachedFrames[(drwav_countof(pWav->ima.cachedFrames) - (pWav->ima.cachedFrameCount*pWav->channels)) + (iByte*2+1)*pWav->channels + iChannel] = predictor; + } + } + } + } + } + + return totalFramesRead; +} + + +#ifndef DR_WAV_NO_CONVERSION_API +static const unsigned short g_drwavAlawTable[256] = { + 0xEA80, 0xEB80, 0xE880, 0xE980, 0xEE80, 0xEF80, 0xEC80, 0xED80, 0xE280, 0xE380, 0xE080, 0xE180, 0xE680, 0xE780, 0xE480, 0xE580, + 0xF540, 0xF5C0, 0xF440, 0xF4C0, 0xF740, 0xF7C0, 0xF640, 0xF6C0, 0xF140, 0xF1C0, 0xF040, 0xF0C0, 0xF340, 0xF3C0, 0xF240, 0xF2C0, + 0xAA00, 0xAE00, 0xA200, 0xA600, 0xBA00, 0xBE00, 0xB200, 0xB600, 0x8A00, 0x8E00, 0x8200, 0x8600, 0x9A00, 0x9E00, 0x9200, 0x9600, + 0xD500, 0xD700, 0xD100, 0xD300, 0xDD00, 0xDF00, 0xD900, 0xDB00, 0xC500, 0xC700, 0xC100, 0xC300, 0xCD00, 0xCF00, 0xC900, 0xCB00, + 0xFEA8, 0xFEB8, 0xFE88, 0xFE98, 0xFEE8, 0xFEF8, 0xFEC8, 0xFED8, 0xFE28, 0xFE38, 0xFE08, 0xFE18, 0xFE68, 0xFE78, 0xFE48, 0xFE58, + 0xFFA8, 0xFFB8, 0xFF88, 0xFF98, 0xFFE8, 0xFFF8, 0xFFC8, 0xFFD8, 0xFF28, 0xFF38, 0xFF08, 0xFF18, 0xFF68, 0xFF78, 0xFF48, 0xFF58, + 0xFAA0, 0xFAE0, 0xFA20, 0xFA60, 0xFBA0, 0xFBE0, 0xFB20, 0xFB60, 0xF8A0, 0xF8E0, 0xF820, 0xF860, 0xF9A0, 0xF9E0, 0xF920, 0xF960, + 0xFD50, 0xFD70, 0xFD10, 0xFD30, 0xFDD0, 0xFDF0, 0xFD90, 0xFDB0, 0xFC50, 0xFC70, 0xFC10, 0xFC30, 0xFCD0, 0xFCF0, 0xFC90, 0xFCB0, + 0x1580, 0x1480, 0x1780, 0x1680, 0x1180, 0x1080, 0x1380, 0x1280, 0x1D80, 0x1C80, 0x1F80, 0x1E80, 0x1980, 0x1880, 0x1B80, 0x1A80, + 0x0AC0, 0x0A40, 0x0BC0, 0x0B40, 0x08C0, 0x0840, 0x09C0, 0x0940, 0x0EC0, 0x0E40, 0x0FC0, 0x0F40, 0x0CC0, 0x0C40, 0x0DC0, 0x0D40, + 0x5600, 0x5200, 0x5E00, 0x5A00, 0x4600, 0x4200, 0x4E00, 0x4A00, 0x7600, 0x7200, 0x7E00, 0x7A00, 0x6600, 0x6200, 0x6E00, 0x6A00, + 0x2B00, 0x2900, 0x2F00, 0x2D00, 0x2300, 0x2100, 0x2700, 0x2500, 0x3B00, 0x3900, 0x3F00, 0x3D00, 0x3300, 0x3100, 0x3700, 0x3500, + 0x0158, 0x0148, 0x0178, 0x0168, 0x0118, 0x0108, 0x0138, 0x0128, 0x01D8, 0x01C8, 0x01F8, 0x01E8, 0x0198, 0x0188, 0x01B8, 0x01A8, + 0x0058, 0x0048, 0x0078, 0x0068, 0x0018, 0x0008, 0x0038, 0x0028, 0x00D8, 0x00C8, 0x00F8, 0x00E8, 0x0098, 0x0088, 0x00B8, 0x00A8, + 0x0560, 0x0520, 0x05E0, 0x05A0, 0x0460, 0x0420, 0x04E0, 0x04A0, 0x0760, 0x0720, 0x07E0, 0x07A0, 0x0660, 0x0620, 0x06E0, 0x06A0, + 0x02B0, 0x0290, 0x02F0, 0x02D0, 0x0230, 0x0210, 0x0270, 0x0250, 0x03B0, 0x0390, 0x03F0, 0x03D0, 0x0330, 0x0310, 0x0370, 0x0350 +}; + +static const unsigned short g_drwavMulawTable[256] = { + 0x8284, 0x8684, 0x8A84, 0x8E84, 0x9284, 0x9684, 0x9A84, 0x9E84, 0xA284, 0xA684, 0xAA84, 0xAE84, 0xB284, 0xB684, 0xBA84, 0xBE84, + 0xC184, 0xC384, 0xC584, 0xC784, 0xC984, 0xCB84, 0xCD84, 0xCF84, 0xD184, 0xD384, 0xD584, 0xD784, 0xD984, 0xDB84, 0xDD84, 0xDF84, + 0xE104, 0xE204, 0xE304, 0xE404, 0xE504, 0xE604, 0xE704, 0xE804, 0xE904, 0xEA04, 0xEB04, 0xEC04, 0xED04, 0xEE04, 0xEF04, 0xF004, + 0xF0C4, 0xF144, 0xF1C4, 0xF244, 0xF2C4, 0xF344, 0xF3C4, 0xF444, 0xF4C4, 0xF544, 0xF5C4, 0xF644, 0xF6C4, 0xF744, 0xF7C4, 0xF844, + 0xF8A4, 0xF8E4, 0xF924, 0xF964, 0xF9A4, 0xF9E4, 0xFA24, 0xFA64, 0xFAA4, 0xFAE4, 0xFB24, 0xFB64, 0xFBA4, 0xFBE4, 0xFC24, 0xFC64, + 0xFC94, 0xFCB4, 0xFCD4, 0xFCF4, 0xFD14, 0xFD34, 0xFD54, 0xFD74, 0xFD94, 0xFDB4, 0xFDD4, 0xFDF4, 0xFE14, 0xFE34, 0xFE54, 0xFE74, + 0xFE8C, 0xFE9C, 0xFEAC, 0xFEBC, 0xFECC, 0xFEDC, 0xFEEC, 0xFEFC, 0xFF0C, 0xFF1C, 0xFF2C, 0xFF3C, 0xFF4C, 0xFF5C, 0xFF6C, 0xFF7C, + 0xFF88, 0xFF90, 0xFF98, 0xFFA0, 0xFFA8, 0xFFB0, 0xFFB8, 0xFFC0, 0xFFC8, 0xFFD0, 0xFFD8, 0xFFE0, 0xFFE8, 0xFFF0, 0xFFF8, 0x0000, + 0x7D7C, 0x797C, 0x757C, 0x717C, 0x6D7C, 0x697C, 0x657C, 0x617C, 0x5D7C, 0x597C, 0x557C, 0x517C, 0x4D7C, 0x497C, 0x457C, 0x417C, + 0x3E7C, 0x3C7C, 0x3A7C, 0x387C, 0x367C, 0x347C, 0x327C, 0x307C, 0x2E7C, 0x2C7C, 0x2A7C, 0x287C, 0x267C, 0x247C, 0x227C, 0x207C, + 0x1EFC, 0x1DFC, 0x1CFC, 0x1BFC, 0x1AFC, 0x19FC, 0x18FC, 0x17FC, 0x16FC, 0x15FC, 0x14FC, 0x13FC, 0x12FC, 0x11FC, 0x10FC, 0x0FFC, + 0x0F3C, 0x0EBC, 0x0E3C, 0x0DBC, 0x0D3C, 0x0CBC, 0x0C3C, 0x0BBC, 0x0B3C, 0x0ABC, 0x0A3C, 0x09BC, 0x093C, 0x08BC, 0x083C, 0x07BC, + 0x075C, 0x071C, 0x06DC, 0x069C, 0x065C, 0x061C, 0x05DC, 0x059C, 0x055C, 0x051C, 0x04DC, 0x049C, 0x045C, 0x041C, 0x03DC, 0x039C, + 0x036C, 0x034C, 0x032C, 0x030C, 0x02EC, 0x02CC, 0x02AC, 0x028C, 0x026C, 0x024C, 0x022C, 0x020C, 0x01EC, 0x01CC, 0x01AC, 0x018C, + 0x0174, 0x0164, 0x0154, 0x0144, 0x0134, 0x0124, 0x0114, 0x0104, 0x00F4, 0x00E4, 0x00D4, 0x00C4, 0x00B4, 0x00A4, 0x0094, 0x0084, + 0x0078, 0x0070, 0x0068, 0x0060, 0x0058, 0x0050, 0x0048, 0x0040, 0x0038, 0x0030, 0x0028, 0x0020, 0x0018, 0x0010, 0x0008, 0x0000 +}; + +static DRWAV_INLINE drwav_int16 drwav__alaw_to_s16(drwav_uint8 sampleIn) +{ + return (short)g_drwavAlawTable[sampleIn]; +} + +static DRWAV_INLINE drwav_int16 drwav__mulaw_to_s16(drwav_uint8 sampleIn) +{ + return (short)g_drwavMulawTable[sampleIn]; +} + + + +DRWAV_PRIVATE void drwav__pcm_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample) +{ + size_t i; + + /* Special case for 8-bit sample data because it's treated as unsigned. */ + if (bytesPerSample == 1) { + drwav_u8_to_s16(pOut, pIn, totalSampleCount); + return; + } + + + /* Slightly more optimal implementation for common formats. */ + if (bytesPerSample == 2) { + for (i = 0; i < totalSampleCount; ++i) { + *pOut++ = ((const drwav_int16*)pIn)[i]; + } + return; + } + if (bytesPerSample == 3) { + drwav_s24_to_s16(pOut, pIn, totalSampleCount); + return; + } + if (bytesPerSample == 4) { + drwav_s32_to_s16(pOut, (const drwav_int32*)pIn, totalSampleCount); + return; + } + + + /* Anything more than 64 bits per sample is not supported. */ + if (bytesPerSample > 8) { + DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut)); + return; + } + + + /* Generic, slow converter. */ + for (i = 0; i < totalSampleCount; ++i) { + drwav_uint64 sample = 0; + unsigned int shift = (8 - bytesPerSample) * 8; + + unsigned int j; + for (j = 0; j < bytesPerSample; j += 1) { + DRWAV_ASSERT(j < 8); + sample |= (drwav_uint64)(pIn[j]) << shift; + shift += 8; + } + + if (!drwav__is_little_endian()) { + sample = drwav__bswap64(sample); + } + + pIn += j; + *pOut++ = (drwav_int16)((drwav_int64)sample >> 48); + } +} + +DRWAV_PRIVATE void drwav__ieee_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample) +{ + if (bytesPerSample == 4) { + drwav_f32_to_s16(pOut, (const float*)pIn, totalSampleCount); + return; + } else if (bytesPerSample == 8) { + drwav_f64_to_s16(pOut, (const double*)pIn, totalSampleCount); + return; + } else { + /* Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. */ + DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut)); + return; + } +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__pcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + /* Fast path. */ + if ((pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM && pWav->bitsPerSample == 16) || pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav__pcm_to_s16(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__ieee(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav__ieee_to_s16(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); /* Safe cast. */ + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__alaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav_alaw_to_s16(pBufferOut, sampleData, (size_t)samplesRead); + + /* + For some reason libsndfile seems to be returning samples of the opposite sign for a-law, but only + with AIFF files. For WAV files it seems to be the same as dr_wav. This is resulting in dr_wav's + automated tests failing. I'm not sure which is correct, but will assume dr_wav. If we're enforcing + libsndfile compatibility we'll swap the signs here. + */ + #ifdef DR_WAV_LIBSNDFILE_COMPAT + { + if (pWav->container == drwav_container_aiff) { + drwav_uint64 iSample; + for (iSample = 0; iSample < samplesRead; iSample += 1) { + pBufferOut[iSample] = -pBufferOut[iSample]; + } + } + } + #endif + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s16__mulaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav_mulaw_to_s16(pBufferOut, sampleData, (size_t)samplesRead); + + /* + Just like with alaw, for some reason the signs between libsndfile and dr_wav are opposite. We just need to + swap the sign if we're compiling with libsndfile compatiblity so our automated tests don't fail. + */ + #ifdef DR_WAV_LIBSNDFILE_COMPAT + { + if (pWav->container == drwav_container_aiff) { + drwav_uint64 iSample; + for (iSample = 0; iSample < samplesRead; iSample += 1) { + pBufferOut[iSample] = -pBufferOut[iSample]; + } + } + } + #endif + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + if (pWav == NULL || framesToRead == 0) { + return 0; + } + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + /* Don't try to read more samples than can potentially fit in the output buffer. */ + if (framesToRead * pWav->channels * sizeof(drwav_int16) > DRWAV_SIZE_MAX) { + framesToRead = DRWAV_SIZE_MAX / sizeof(drwav_int16) / pWav->channels; + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) { + return drwav_read_pcm_frames_s16__pcm(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) { + return drwav_read_pcm_frames_s16__ieee(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) { + return drwav_read_pcm_frames_s16__alaw(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) { + return drwav_read_pcm_frames_s16__mulaw(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM) { + return drwav_read_pcm_frames_s16__msadpcm(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + return drwav_read_pcm_frames_s16__ima(pWav, framesToRead, pBufferOut); + } + + return 0; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16le(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_FALSE) { + drwav__bswap_samples_s16(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s16be(drwav* pWav, drwav_uint64 framesToRead, drwav_int16* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_TRUE) { + drwav__bswap_samples_s16(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + + +DRWAV_API void drwav_u8_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + int x = pIn[i]; + r = x << 8; + r = r - 32768; + pOut[i] = (short)r; + } +} + +DRWAV_API void drwav_s24_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + int x = ((int)(((unsigned int)(((const drwav_uint8*)pIn)[i*3+0]) << 8) | ((unsigned int)(((const drwav_uint8*)pIn)[i*3+1]) << 16) | ((unsigned int)(((const drwav_uint8*)pIn)[i*3+2])) << 24)) >> 8; + r = x >> 8; + pOut[i] = (short)r; + } +} + +DRWAV_API void drwav_s32_to_s16(drwav_int16* pOut, const drwav_int32* pIn, size_t sampleCount) +{ + int r; + size_t i; + for (i = 0; i < sampleCount; ++i) { + int x = pIn[i]; + r = x >> 16; + pOut[i] = (short)r; + } +} + +DRWAV_API void drwav_f32_to_s16(drwav_int16* pOut, const float* pIn, size_t sampleCount) +{ + size_t i; + for (i = 0; i < sampleCount; ++i) { + float x = pIn[i]; + if (x != x) { + pOut[i] = 0; /* NaN */ + } else if (x <= -1) { + pOut[i] = (-32767 - 1); + } else if (x >= 1) { + pOut[i] = 32767; + } else { + pOut[i] = (drwav_int16)(x * 32768.0f); + } + } +} + +DRWAV_API void drwav_f64_to_s16(drwav_int16* pOut, const double* pIn, size_t sampleCount) +{ + size_t i; + for (i = 0; i < sampleCount; ++i) { + double x = pIn[i]; + if (x != x) { + pOut[i] = 0; /* NaN */ + } else if (x <= -1) { + pOut[i] = (-32767 - 1); + } else if (x >= 1) { + pOut[i] = 32767; + } else { + pOut[i] = (drwav_int16)(x * 32768.0); + } + } +} + +DRWAV_API void drwav_alaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + for (i = 0; i < sampleCount; ++i) { + pOut[i] = drwav__alaw_to_s16(pIn[i]); + } +} + +DRWAV_API void drwav_mulaw_to_s16(drwav_int16* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + for (i = 0; i < sampleCount; ++i) { + pOut[i] = drwav__mulaw_to_s16(pIn[i]); + } +} + + +DRWAV_PRIVATE void drwav__pcm_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount, unsigned int bytesPerSample) +{ + unsigned int i; + + /* Special case for 8-bit sample data because it's treated as unsigned. */ + if (bytesPerSample == 1) { + drwav_u8_to_f32(pOut, pIn, sampleCount); + return; + } + + /* Slightly more optimal implementation for common formats. */ + if (bytesPerSample == 2) { + drwav_s16_to_f32(pOut, (const drwav_int16*)pIn, sampleCount); + return; + } + if (bytesPerSample == 3) { + drwav_s24_to_f32(pOut, pIn, sampleCount); + return; + } + if (bytesPerSample == 4) { + drwav_s32_to_f32(pOut, (const drwav_int32*)pIn, sampleCount); + return; + } + + + /* Anything more than 64 bits per sample is not supported. */ + if (bytesPerSample > 8) { + DRWAV_ZERO_MEMORY(pOut, sampleCount * sizeof(*pOut)); + return; + } + + + /* Generic, slow converter. */ + for (i = 0; i < sampleCount; ++i) { + drwav_uint64 sample = 0; + unsigned int shift = (8 - bytesPerSample) * 8; + + unsigned int j; + for (j = 0; j < bytesPerSample; j += 1) { + DRWAV_ASSERT(j < 8); + sample |= (drwav_uint64)(pIn[j]) << shift; + shift += 8; + } + + if (!drwav__is_little_endian()) { + sample = drwav__bswap64(sample); + } + + pIn += j; + *pOut++ = (float)((drwav_int64)sample / 9223372036854775807.0); + } +} + +DRWAV_PRIVATE void drwav__ieee_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount, unsigned int bytesPerSample) +{ + if (bytesPerSample == 4) { + unsigned int i; + for (i = 0; i < sampleCount; ++i) { + *pOut++ = ((const float*)pIn)[i]; + } + return; + } else if (bytesPerSample == 8) { + drwav_f64_to_f32(pOut, (const double*)pIn, sampleCount); + return; + } else { + /* Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. */ + DRWAV_ZERO_MEMORY(pOut, sampleCount * sizeof(*pOut)); + return; + } +} + + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__pcm(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav__pcm_to_f32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__msadpcm_ima(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + /* + We're just going to borrow the implementation from the drwav_read_s16() since ADPCM is a little bit more complicated than other formats and I don't + want to duplicate that code. + */ + drwav_uint64 totalFramesRead; + drwav_int16 samples16[2048]; + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels); + drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToReadThisIteration, samples16); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + drwav_s16_to_f32(pBufferOut, samples16, (size_t)(framesRead*pWav->channels)); /* <-- Safe cast because we're clamping to 2048. */ + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__ieee(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + /* Fast path. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT && pWav->bitsPerSample == 32) { + return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav__ieee_to_f32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__alaw(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav_alaw_to_f32(pBufferOut, sampleData, (size_t)samplesRead); + + #ifdef DR_WAV_LIBSNDFILE_COMPAT + { + if (pWav->container == drwav_container_aiff) { + drwav_uint64 iSample; + for (iSample = 0; iSample < samplesRead; iSample += 1) { + pBufferOut[iSample] = -pBufferOut[iSample]; + } + } + } + #endif + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_f32__mulaw(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav_mulaw_to_f32(pBufferOut, sampleData, (size_t)samplesRead); + + #ifdef DR_WAV_LIBSNDFILE_COMPAT + { + if (pWav->container == drwav_container_aiff) { + drwav_uint64 iSample; + for (iSample = 0; iSample < samplesRead; iSample += 1) { + pBufferOut[iSample] = -pBufferOut[iSample]; + } + } + } + #endif + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + if (pWav == NULL || framesToRead == 0) { + return 0; + } + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + /* Don't try to read more samples than can potentially fit in the output buffer. */ + if (framesToRead * pWav->channels * sizeof(float) > DRWAV_SIZE_MAX) { + framesToRead = DRWAV_SIZE_MAX / sizeof(float) / pWav->channels; + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) { + return drwav_read_pcm_frames_f32__pcm(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM || pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + return drwav_read_pcm_frames_f32__msadpcm_ima(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) { + return drwav_read_pcm_frames_f32__ieee(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) { + return drwav_read_pcm_frames_f32__alaw(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) { + return drwav_read_pcm_frames_f32__mulaw(pWav, framesToRead, pBufferOut); + } + + return 0; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32le(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_f32(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_FALSE) { + drwav__bswap_samples_f32(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_f32be(drwav* pWav, drwav_uint64 framesToRead, float* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_f32(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_TRUE) { + drwav__bswap_samples_f32(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + + +DRWAV_API void drwav_u8_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + +#ifdef DR_WAV_LIBSNDFILE_COMPAT + /* + It appears libsndfile uses slightly different logic for the u8 -> f32 conversion to dr_wav, which in my opinion is incorrect. It appears + libsndfile performs the conversion something like "f32 = (u8 / 256) * 2 - 1", however I think it should be "f32 = (u8 / 255) * 2 - 1" (note + the divisor of 256 vs 255). I use libsndfile as a benchmark for testing, so I'm therefore leaving this block here just for my automated + correctness testing. This is disabled by default. + */ + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (pIn[i] / 256.0f) * 2 - 1; + } +#else + for (i = 0; i < sampleCount; ++i) { + float x = pIn[i]; + x = x * 0.00784313725490196078f; /* 0..255 to 0..2 */ + x = x - 1; /* 0..2 to -1..1 */ + + *pOut++ = x; + } +#endif +} + +DRWAV_API void drwav_s16_to_f32(float* pOut, const drwav_int16* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = pIn[i] * 0.000030517578125f; + } +} + +DRWAV_API void drwav_s24_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + double x; + drwav_uint32 a = ((drwav_uint32)(pIn[i*3+0]) << 8); + drwav_uint32 b = ((drwav_uint32)(pIn[i*3+1]) << 16); + drwav_uint32 c = ((drwav_uint32)(pIn[i*3+2]) << 24); + + x = (double)((drwav_int32)(a | b | c) >> 8); + *pOut++ = (float)(x * 0.00000011920928955078125); + } +} + +DRWAV_API void drwav_s32_to_f32(float* pOut, const drwav_int32* pIn, size_t sampleCount) +{ + size_t i; + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (float)(pIn[i] / 2147483648.0); + } +} + +DRWAV_API void drwav_f64_to_f32(float* pOut, const double* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (float)pIn[i]; + } +} + +DRWAV_API void drwav_alaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = drwav__alaw_to_s16(pIn[i]) / 32768.0f; + } +} + +DRWAV_API void drwav_mulaw_to_f32(float* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = drwav__mulaw_to_s16(pIn[i]) / 32768.0f; + } +} + + + +DRWAV_PRIVATE void drwav__pcm_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample) +{ + unsigned int i; + + /* Special case for 8-bit sample data because it's treated as unsigned. */ + if (bytesPerSample == 1) { + drwav_u8_to_s32(pOut, pIn, totalSampleCount); + return; + } + + /* Slightly more optimal implementation for common formats. */ + if (bytesPerSample == 2) { + drwav_s16_to_s32(pOut, (const drwav_int16*)pIn, totalSampleCount); + return; + } + if (bytesPerSample == 3) { + drwav_s24_to_s32(pOut, pIn, totalSampleCount); + return; + } + if (bytesPerSample == 4) { + for (i = 0; i < totalSampleCount; ++i) { + *pOut++ = ((const drwav_int32*)pIn)[i]; + } + return; + } + + + /* Anything more than 64 bits per sample is not supported. */ + if (bytesPerSample > 8) { + DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut)); + return; + } + + + /* Generic, slow converter. */ + for (i = 0; i < totalSampleCount; ++i) { + drwav_uint64 sample = 0; + unsigned int shift = (8 - bytesPerSample) * 8; + + unsigned int j; + for (j = 0; j < bytesPerSample; j += 1) { + DRWAV_ASSERT(j < 8); + sample |= (drwav_uint64)(pIn[j]) << shift; + shift += 8; + } + + if (!drwav__is_little_endian()) { + sample = drwav__bswap64(sample); + } + + pIn += j; + *pOut++ = (drwav_int32)((drwav_int64)sample >> 32); + } +} + +DRWAV_PRIVATE void drwav__ieee_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t totalSampleCount, unsigned int bytesPerSample) +{ + if (bytesPerSample == 4) { + drwav_f32_to_s32(pOut, (const float*)pIn, totalSampleCount); + return; + } else if (bytesPerSample == 8) { + drwav_f64_to_s32(pOut, (const double*)pIn, totalSampleCount); + return; + } else { + /* Only supporting 32- and 64-bit float. Output silence in all other cases. Contributions welcome for 16-bit float. */ + DRWAV_ZERO_MEMORY(pOut, totalSampleCount * sizeof(*pOut)); + return; + } +} + + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__pcm(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + /* Fast path. */ + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM && pWav->bitsPerSample == 32) { + return drwav_read_pcm_frames(pWav, framesToRead, pBufferOut); + } + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav__pcm_to_s32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__msadpcm_ima(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + /* + We're just going to borrow the implementation from the drwav_read_s16() since ADPCM is a little bit more complicated than other formats and I don't + want to duplicate that code. + */ + drwav_uint64 totalFramesRead = 0; + drwav_int16 samples16[2048]; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, drwav_countof(samples16)/pWav->channels); + drwav_uint64 framesRead = drwav_read_pcm_frames_s16(pWav, framesToReadThisIteration, samples16); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + drwav_s16_to_s32(pBufferOut, samples16, (size_t)(framesRead*pWav->channels)); /* <-- Safe cast because we're clamping to 2048. */ + + pBufferOut += framesRead*pWav->channels; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__ieee(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav__ieee_to_s32(pBufferOut, sampleData, (size_t)samplesRead, bytesPerSample); + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__alaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav_alaw_to_s32(pBufferOut, sampleData, (size_t)samplesRead); + + #ifdef DR_WAV_LIBSNDFILE_COMPAT + { + if (pWav->container == drwav_container_aiff) { + drwav_uint64 iSample; + for (iSample = 0; iSample < samplesRead; iSample += 1) { + pBufferOut[iSample] = -pBufferOut[iSample]; + } + } + } + #endif + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_PRIVATE drwav_uint64 drwav_read_pcm_frames_s32__mulaw(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 totalFramesRead; + drwav_uint8 sampleData[4096] = {0}; + drwav_uint32 bytesPerFrame; + drwav_uint32 bytesPerSample; + drwav_uint64 samplesRead; + + bytesPerFrame = drwav_get_bytes_per_pcm_frame(pWav); + if (bytesPerFrame == 0) { + return 0; + } + + bytesPerSample = bytesPerFrame / pWav->channels; + if (bytesPerSample == 0 || (bytesPerFrame % pWav->channels) != 0) { + return 0; /* Only byte-aligned formats are supported. */ + } + + totalFramesRead = 0; + + while (framesToRead > 0) { + drwav_uint64 framesToReadThisIteration = drwav_min(framesToRead, sizeof(sampleData)/bytesPerFrame); + drwav_uint64 framesRead = drwav_read_pcm_frames(pWav, framesToReadThisIteration, sampleData); + if (framesRead == 0) { + break; + } + + DRWAV_ASSERT(framesRead <= framesToReadThisIteration); /* If this fails it means there's a bug in drwav_read_pcm_frames(). */ + + /* Validation to ensure we don't read too much from out intermediary buffer. This is to protect from invalid files. */ + samplesRead = framesRead * pWav->channels; + if ((samplesRead * bytesPerSample) > sizeof(sampleData)) { + DRWAV_ASSERT(DRWAV_FALSE); /* This should never happen with a valid file. */ + break; + } + + drwav_mulaw_to_s32(pBufferOut, sampleData, (size_t)samplesRead); + + #ifdef DR_WAV_LIBSNDFILE_COMPAT + { + if (pWav->container == drwav_container_aiff) { + drwav_uint64 iSample; + for (iSample = 0; iSample < samplesRead; iSample += 1) { + pBufferOut[iSample] = -pBufferOut[iSample]; + } + } + } + #endif + + pBufferOut += samplesRead; + framesToRead -= framesRead; + totalFramesRead += framesRead; + } + + return totalFramesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + if (pWav == NULL || framesToRead == 0) { + return 0; + } + + if (pBufferOut == NULL) { + return drwav_read_pcm_frames(pWav, framesToRead, NULL); + } + + /* Don't try to read more samples than can potentially fit in the output buffer. */ + if (framesToRead * pWav->channels * sizeof(drwav_int32) > DRWAV_SIZE_MAX) { + framesToRead = DRWAV_SIZE_MAX / sizeof(drwav_int32) / pWav->channels; + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_PCM) { + return drwav_read_pcm_frames_s32__pcm(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ADPCM || pWav->translatedFormatTag == DR_WAVE_FORMAT_DVI_ADPCM) { + return drwav_read_pcm_frames_s32__msadpcm_ima(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_IEEE_FLOAT) { + return drwav_read_pcm_frames_s32__ieee(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_ALAW) { + return drwav_read_pcm_frames_s32__alaw(pWav, framesToRead, pBufferOut); + } + + if (pWav->translatedFormatTag == DR_WAVE_FORMAT_MULAW) { + return drwav_read_pcm_frames_s32__mulaw(pWav, framesToRead, pBufferOut); + } + + return 0; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32le(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_s32(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_FALSE) { + drwav__bswap_samples_s32(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + +DRWAV_API drwav_uint64 drwav_read_pcm_frames_s32be(drwav* pWav, drwav_uint64 framesToRead, drwav_int32* pBufferOut) +{ + drwav_uint64 framesRead = drwav_read_pcm_frames_s32(pWav, framesToRead, pBufferOut); + if (pBufferOut != NULL && drwav__is_little_endian() == DRWAV_TRUE) { + drwav__bswap_samples_s32(pBufferOut, framesRead*pWav->channels); + } + + return framesRead; +} + + +DRWAV_API void drwav_u8_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = ((int)pIn[i] - 128) * 16777216; + } +} + +DRWAV_API void drwav_s16_to_s32(drwav_int32* pOut, const drwav_int16* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (drwav_int32)pIn[i] * 65536; + } +} + +DRWAV_API void drwav_s24_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + unsigned int s0 = pIn[i*3 + 0]; + unsigned int s1 = pIn[i*3 + 1]; + unsigned int s2 = pIn[i*3 + 2]; + + drwav_int32 sample32 = (drwav_int32)((s0 << 8) | (s1 << 16) | (s2 << 24)); + *pOut++ = sample32; + } +} + +DRWAV_API void drwav_f32_to_s32(drwav_int32* pOut, const float* pIn, size_t sampleCount) +{ + size_t i; + for (i = 0; i < sampleCount; ++i) { + float x = pIn[i]; + if (x != x) { + pOut[i] = 0; /* NaN */ + } else if (x <= -1) { + pOut[i] = (-2147483647 - 1); + } else if (x >= 1) { + pOut[i] = 2147483647; + } else { + pOut[i] = (drwav_int32)(x * 2147483648.0f); + } + } +} + +DRWAV_API void drwav_f64_to_s32(drwav_int32* pOut, const double* pIn, size_t sampleCount) +{ + size_t i; + for (i = 0; i < sampleCount; ++i) { + double x = pIn[i]; + if (x != x) { + pOut[i] = 0; /* NaN */ + } else if (x <= -1) { + pOut[i] = (-2147483647 - 1); + } else if (x >= 1) { + pOut[i] = 2147483647; + } else { + pOut[i] = (drwav_int32)(x * 2147483648.0); + } + } +} + +DRWAV_API void drwav_alaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i = 0; i < sampleCount; ++i) { + *pOut++ = (drwav_int32)drwav__alaw_to_s16(pIn[i]) * 65536; + } +} + +DRWAV_API void drwav_mulaw_to_s32(drwav_int32* pOut, const drwav_uint8* pIn, size_t sampleCount) +{ + size_t i; + + if (pOut == NULL || pIn == NULL) { + return; + } + + for (i= 0; i < sampleCount; ++i) { + *pOut++ = (drwav_int32)drwav__mulaw_to_s16(pIn[i]) * 65536; + } +} + + + +DRWAV_PRIVATE drwav_int16* drwav__read_pcm_frames_and_close_s16(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount) +{ + drwav_uint64 sampleDataSize; + drwav_int16* pSampleData; + drwav_uint64 framesRead; + + DRWAV_ASSERT(pWav != NULL); + + /* Check for overflow before multiplication. */ + if (pWav->channels == 0 || pWav->totalPCMFrameCount > DRWAV_SIZE_MAX / pWav->channels / sizeof(drwav_int16)) { + drwav_uninit(pWav); + return NULL; /* Overflow or invalid channels. */ + } + + sampleDataSize = pWav->totalPCMFrameCount * pWav->channels * sizeof(drwav_int16); + if (sampleDataSize > DRWAV_SIZE_MAX) { + drwav_uninit(pWav); + return NULL; /* File's too big. */ + } + + pSampleData = (drwav_int16*)drwav__malloc_from_callbacks((size_t)sampleDataSize, &pWav->allocationCallbacks); /* <-- Safe cast due to the check above. */ + if (pSampleData == NULL) { + drwav_uninit(pWav); + return NULL; /* Failed to allocate memory. */ + } + + framesRead = drwav_read_pcm_frames_s16(pWav, (size_t)pWav->totalPCMFrameCount, pSampleData); + if (framesRead != pWav->totalPCMFrameCount) { + drwav__free_from_callbacks(pSampleData, &pWav->allocationCallbacks); + drwav_uninit(pWav); + return NULL; /* There was an error reading the samples. */ + } + + drwav_uninit(pWav); + + if (sampleRate) { + *sampleRate = pWav->sampleRate; + } + if (channels) { + *channels = pWav->channels; + } + if (totalFrameCount) { + *totalFrameCount = pWav->totalPCMFrameCount; + } + + return pSampleData; +} + +DRWAV_PRIVATE float* drwav__read_pcm_frames_and_close_f32(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount) +{ + drwav_uint64 sampleDataSize; + float* pSampleData; + drwav_uint64 framesRead; + + DRWAV_ASSERT(pWav != NULL); + + /* Check for overflow before multiplication. */ + if (pWav->channels == 0 || pWav->totalPCMFrameCount > DRWAV_SIZE_MAX / pWav->channels / sizeof(float)) { + drwav_uninit(pWav); + return NULL; /* Overflow or invalid channels. */ + } + + sampleDataSize = pWav->totalPCMFrameCount * pWav->channels * sizeof(float); + if (sampleDataSize > DRWAV_SIZE_MAX) { + drwav_uninit(pWav); + return NULL; /* File's too big. */ + } + + pSampleData = (float*)drwav__malloc_from_callbacks((size_t)sampleDataSize, &pWav->allocationCallbacks); /* <-- Safe cast due to the check above. */ + if (pSampleData == NULL) { + drwav_uninit(pWav); + return NULL; /* Failed to allocate memory. */ + } + + framesRead = drwav_read_pcm_frames_f32(pWav, (size_t)pWav->totalPCMFrameCount, pSampleData); + if (framesRead != pWav->totalPCMFrameCount) { + drwav__free_from_callbacks(pSampleData, &pWav->allocationCallbacks); + drwav_uninit(pWav); + return NULL; /* There was an error reading the samples. */ + } + + drwav_uninit(pWav); + + if (sampleRate) { + *sampleRate = pWav->sampleRate; + } + if (channels) { + *channels = pWav->channels; + } + if (totalFrameCount) { + *totalFrameCount = pWav->totalPCMFrameCount; + } + + return pSampleData; +} + +DRWAV_PRIVATE drwav_int32* drwav__read_pcm_frames_and_close_s32(drwav* pWav, unsigned int* channels, unsigned int* sampleRate, drwav_uint64* totalFrameCount) +{ + drwav_uint64 sampleDataSize; + drwav_int32* pSampleData; + drwav_uint64 framesRead; + + DRWAV_ASSERT(pWav != NULL); + + /* Check for overflow before multiplication. */ + if (pWav->channels == 0 || pWav->totalPCMFrameCount > DRWAV_SIZE_MAX / pWav->channels / sizeof(drwav_int32)) { + drwav_uninit(pWav); + return NULL; /* Overflow or invalid channels. */ + } + + sampleDataSize = pWav->totalPCMFrameCount * pWav->channels * sizeof(drwav_int32); + if (sampleDataSize > DRWAV_SIZE_MAX) { + drwav_uninit(pWav); + return NULL; /* File's too big. */ + } + + pSampleData = (drwav_int32*)drwav__malloc_from_callbacks((size_t)sampleDataSize, &pWav->allocationCallbacks); /* <-- Safe cast due to the check above. */ + if (pSampleData == NULL) { + drwav_uninit(pWav); + return NULL; /* Failed to allocate memory. */ + } + + framesRead = drwav_read_pcm_frames_s32(pWav, (size_t)pWav->totalPCMFrameCount, pSampleData); + if (framesRead != pWav->totalPCMFrameCount) { + drwav__free_from_callbacks(pSampleData, &pWav->allocationCallbacks); + drwav_uninit(pWav); + return NULL; /* There was an error reading the samples. */ + } + + drwav_uninit(pWav); + + if (sampleRate) { + *sampleRate = pWav->sampleRate; + } + if (channels) { + *channels = pWav->channels; + } + if (totalFrameCount) { + *totalFrameCount = pWav->totalPCMFrameCount; + } + + return pSampleData; +} + + + +DRWAV_API drwav_int16* drwav_open_and_read_pcm_frames_s16(drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init(&wav, onRead, onSeek, onTell, pUserData, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API float* drwav_open_and_read_pcm_frames_f32(drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init(&wav, onRead, onSeek, onTell, pUserData, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API drwav_int32* drwav_open_and_read_pcm_frames_s32(drwav_read_proc onRead, drwav_seek_proc onSeek, drwav_tell_proc onTell, void* pUserData, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init(&wav, onRead, onSeek, onTell, pUserData, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +#ifndef DR_WAV_NO_STDIO +DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32(const char* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + + +#ifndef DR_WAV_NO_WCHAR +DRWAV_API drwav_int16* drwav_open_file_and_read_pcm_frames_s16_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (channelsOut) { + *channelsOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file_w(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API float* drwav_open_file_and_read_pcm_frames_f32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (channelsOut) { + *channelsOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file_w(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API drwav_int32* drwav_open_file_and_read_pcm_frames_s32_w(const wchar_t* filename, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (channelsOut) { + *channelsOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_file_w(&wav, filename, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} +#endif /* DR_WAV_NO_WCHAR */ +#endif /* DR_WAV_NO_STDIO */ + +DRWAV_API drwav_int16* drwav_open_memory_and_read_pcm_frames_s16(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_memory(&wav, data, dataSize, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s16(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API float* drwav_open_memory_and_read_pcm_frames_f32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_memory(&wav, data, dataSize, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_f32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} + +DRWAV_API drwav_int32* drwav_open_memory_and_read_pcm_frames_s32(const void* data, size_t dataSize, unsigned int* channelsOut, unsigned int* sampleRateOut, drwav_uint64* totalFrameCountOut, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + drwav wav; + + if (channelsOut) { + *channelsOut = 0; + } + if (sampleRateOut) { + *sampleRateOut = 0; + } + if (totalFrameCountOut) { + *totalFrameCountOut = 0; + } + + if (!drwav_init_memory(&wav, data, dataSize, pAllocationCallbacks)) { + return NULL; + } + + return drwav__read_pcm_frames_and_close_s32(&wav, channelsOut, sampleRateOut, totalFrameCountOut); +} +#endif /* DR_WAV_NO_CONVERSION_API */ + + +DRWAV_API void drwav_free(void* p, const drwav_allocation_callbacks* pAllocationCallbacks) +{ + if (pAllocationCallbacks != NULL) { + drwav__free_from_callbacks(p, pAllocationCallbacks); + } else { + drwav__free_default(p, NULL); + } +} + +DRWAV_API drwav_uint16 drwav_bytes_to_u16(const drwav_uint8* data) +{ + return ((drwav_uint16)data[0] << 0) | ((drwav_uint16)data[1] << 8); +} + +DRWAV_API drwav_int16 drwav_bytes_to_s16(const drwav_uint8* data) +{ + return (drwav_int16)drwav_bytes_to_u16(data); +} + +DRWAV_API drwav_uint32 drwav_bytes_to_u32(const drwav_uint8* data) +{ + return drwav_bytes_to_u32_le(data); +} + +DRWAV_API float drwav_bytes_to_f32(const drwav_uint8* data) +{ + union { + drwav_uint32 u32; + float f32; + } value; + + value.u32 = drwav_bytes_to_u32(data); + return value.f32; +} + +DRWAV_API drwav_int32 drwav_bytes_to_s32(const drwav_uint8* data) +{ + return (drwav_int32)drwav_bytes_to_u32(data); +} + +DRWAV_API drwav_uint64 drwav_bytes_to_u64(const drwav_uint8* data) +{ + return + ((drwav_uint64)data[0] << 0) | ((drwav_uint64)data[1] << 8) | ((drwav_uint64)data[2] << 16) | ((drwav_uint64)data[3] << 24) | + ((drwav_uint64)data[4] << 32) | ((drwav_uint64)data[5] << 40) | ((drwav_uint64)data[6] << 48) | ((drwav_uint64)data[7] << 56); +} + +DRWAV_API drwav_int64 drwav_bytes_to_s64(const drwav_uint8* data) +{ + return (drwav_int64)drwav_bytes_to_u64(data); +} + + +DRWAV_API drwav_bool32 drwav_guid_equal(const drwav_uint8 a[16], const drwav_uint8 b[16]) +{ + int i; + for (i = 0; i < 16; i += 1) { + if (a[i] != b[i]) { + return DRWAV_FALSE; + } + } + + return DRWAV_TRUE; +} + +DRWAV_API drwav_bool32 drwav_fourcc_equal(const drwav_uint8* a, const char* b) +{ + return + a[0] == b[0] && + a[1] == b[1] && + a[2] == b[2] && + a[3] == b[3]; +} + +#ifdef __MRC__ +/* Undo the pragma at the beginning of this file. */ +#pragma options opt reset +#endif + +#endif /* dr_wav_c */ +#endif /* DR_WAV_IMPLEMENTATION */ + +/* +REVISION HISTORY +================ +v0.14.6 - TBD + - Encoders will now write out header information each write so that a valid file is still produced when an explicit `drwav_uninit()` is not called. + - Fix an error when loading files with a malformed "bext" chunk. + - Fix an error when loading files with a malformed "fmt" chunk. + - Fix an error when loading files with a malformed "fact" chunk. + - Fix an error when loading files with a malformed "smpl" chunk. + - Fix an underflow error with badly formed ADPCM encoded files. + - Fix an underflow error with badly formed W64 files. + - Fix an error when converting from >32 bit samples to s16/f32/s32 on big-endian architectures. + - Fix an error with conversion from u8, 16, alaw and mulaw to s32. + - Fix an error with AIFF files with an unusual bit depth. + - Fix some NaN conversion errors when converting from floating point to s16 and s32. + - Add some bound checking when processing metadata chunks. + +v0.14.5 - 2026-03-03 + - Fix a crash when loading files with a malformed "smpl" chunk. + - Fix a signed overflow bug with the MS-ADPCM decoder. + +v0.14.4 - 2026-01-17 + - Fix some compilation warnings. + +v0.14.3 - 2025-12-14 + - Fix a possible out-of-bounds read when reading from MS-ADPCM encoded files. + - Fix a possible integer overflow error. + +v0.14.2 - 2025-12-02 + - Fix a compilation warning. + +v0.14.1 - 2025-09-10 + - Fix an error with the NXDK build. + +v0.14.0 - 2025-07-23 + - API CHANGE: Seek origin enums have been renamed to the following: + - drwav_seek_origin_start -> DRWAV_SEEK_SET + - drwav_seek_origin_current -> DRWAV_SEEK_CUR + - DRWAV_SEEK_END (new) + - API CHANGE: A new seek origin has been added to allow seeking from the end of the file. If you implement your own `onSeek` callback, you must now handle `DRWAV_SEEK_END`. If you only use `*_init_file()` or `*_init_memory()`, you need not change anything. + - API CHANGE: An `onTell` callback has been added to the following functions: + - drwav_init() + - drwav_init_ex() + - drwav_init_with_metadata() + - drwav_open_and_read_pcm_frames_s16() + - drwav_open_and_read_pcm_frames_f32() + - drwav_open_and_read_pcm_frames_s32() + - API CHANGE: The `firstSampleByteOffset`, `lastSampleByteOffset` and `sampleByteOffset` members of `drwav_cue_point` have been renamed to `firstSampleOffset`, `lastSampleOffset` and `sampleOffset`, respectively. + - Fix a static analysis warning. + - Fix compilation for AIX OS. + +v0.13.17 - 2024-12-17 + - Fix a possible crash when reading from MS-ADPCM encoded files. + - Improve detection of ARM64EC + +v0.13.16 - 2024-02-27 + - Fix a Wdouble-promotion warning. + +v0.13.15 - 2024-01-23 + - Relax some unnecessary validation that prevented some files from loading. + +v0.13.14 - 2023-12-02 + - Fix a warning about an unused variable. + +v0.13.13 - 2023-11-02 + - Fix a warning when compiling with Clang. + +v0.13.12 - 2023-08-07 + - Fix a possible crash in drwav_read_pcm_frames(). + +v0.13.11 - 2023-07-07 + - AIFF compatibility improvements. + +v0.13.10 - 2023-05-29 + - Fix a bug where drwav_init_with_metadata() does not decode any frames after initializtion. + +v0.13.9 - 2023-05-22 + - Add support for AIFF decoding (writing and metadata not supported). + - Add support for RIFX decoding (writing and metadata not supported). + - Fix a bug where metadata is not processed if it's located before the "fmt " chunk. + - Add a workaround for a type of malformed WAV file where the size of the "RIFF" and "data" chunks + are incorrectly set to 0xFFFFFFFF. + +v0.13.8 - 2023-03-25 + - Fix a possible null pointer dereference. + - Fix a crash when loading files with badly formed metadata. + +v0.13.7 - 2022-09-17 + - Fix compilation with DJGPP. + - Add support for disabling wchar_t with DR_WAV_NO_WCHAR. + +v0.13.6 - 2022-04-10 + - Fix compilation error on older versions of GCC. + - Remove some dependencies on the standard library. + +v0.13.5 - 2022-01-26 + - Fix an error when seeking to the end of the file. + +v0.13.4 - 2021-12-08 + - Fix some static analysis warnings. + +v0.13.3 - 2021-11-24 + - Fix an incorrect assertion when trying to endian swap 1-byte sample formats. This is now a no-op + rather than a failed assertion. + - Fix a bug with parsing of the bext chunk. + - Fix some static analysis warnings. + +v0.13.2 - 2021-10-02 + - Fix a possible buffer overflow when reading from compressed formats. + +v0.13.1 - 2021-07-31 + - Fix platform detection for ARM64. + +v0.13.0 - 2021-07-01 + - Improve support for reading and writing metadata. Use the `_with_metadata()` APIs to initialize + a WAV decoder and store the metadata within the `drwav` object. Use the `pMetadata` and + `metadataCount` members of the `drwav` object to read the data. The old way of handling metadata + via a callback is still usable and valid. + - API CHANGE: drwav_target_write_size_bytes() now takes extra parameters for calculating the + required write size when writing metadata. + - Add drwav_get_cursor_in_pcm_frames() + - Add drwav_get_length_in_pcm_frames() + - Fix a bug where drwav_read_raw() can call the read callback with a byte count of zero. + +v0.12.20 - 2021-06-11 + - Fix some undefined behavior. + +v0.12.19 - 2021-02-21 + - Fix a warning due to referencing _MSC_VER when it is undefined. + - Minor improvements to the management of some internal state concerning the data chunk cursor. + +v0.12.18 - 2021-01-31 + - Clean up some static analysis warnings. + +v0.12.17 - 2021-01-17 + - Minor fix to sample code in documentation. + - Correctly qualify a private API as private rather than public. + - Code cleanup. + +v0.12.16 - 2020-12-02 + - Fix a bug when trying to read more bytes than can fit in a size_t. + +v0.12.15 - 2020-11-21 + - Fix compilation with OpenWatcom. + +v0.12.14 - 2020-11-13 + - Minor code clean up. + +v0.12.13 - 2020-11-01 + - Improve compiler support for older versions of GCC. + +v0.12.12 - 2020-09-28 + - Add support for RF64. + - Fix a bug in writing mode where the size of the RIFF chunk incorrectly includes the header section. + +v0.12.11 - 2020-09-08 + - Fix a compilation error on older compilers. + +v0.12.10 - 2020-08-24 + - Fix a bug when seeking with ADPCM formats. + +v0.12.9 - 2020-08-02 + - Simplify sized types. + +v0.12.8 - 2020-07-25 + - Fix a compilation warning. + +v0.12.7 - 2020-07-15 + - Fix some bugs on big-endian architectures. + - Fix an error in s24 to f32 conversion. + +v0.12.6 - 2020-06-23 + - Change drwav_read_*() to allow NULL to be passed in as the output buffer which is equivalent to a forward seek. + - Fix a buffer overflow when trying to decode invalid IMA-ADPCM files. + - Add include guard for the implementation section. + +v0.12.5 - 2020-05-27 + - Minor documentation fix. + +v0.12.4 - 2020-05-16 + - Replace assert() with DRWAV_ASSERT(). + - Add compile-time and run-time version querying. + - DRWAV_VERSION_MINOR + - DRWAV_VERSION_MAJOR + - DRWAV_VERSION_REVISION + - DRWAV_VERSION_STRING + - drwav_version() + - drwav_version_string() + +v0.12.3 - 2020-04-30 + - Fix compilation errors with VC6. + +v0.12.2 - 2020-04-21 + - Fix a bug where drwav_init_file() does not close the file handle after attempting to load an erroneous file. + +v0.12.1 - 2020-04-13 + - Fix some pedantic warnings. + +v0.12.0 - 2020-04-04 + - API CHANGE: Add container and format parameters to the chunk callback. + - Minor documentation updates. + +v0.11.5 - 2020-03-07 + - Fix compilation error with Visual Studio .NET 2003. + +v0.11.4 - 2020-01-29 + - Fix some static analysis warnings. + - Fix a bug when reading f32 samples from an A-law encoded stream. + +v0.11.3 - 2020-01-12 + - Minor changes to some f32 format conversion routines. + - Minor bug fix for ADPCM conversion when end of file is reached. + +v0.11.2 - 2019-12-02 + - Fix a possible crash when using custom memory allocators without a custom realloc() implementation. + - Fix an integer overflow bug. + - Fix a null pointer dereference bug. + - Add limits to sample rate, channels and bits per sample to tighten up some validation. + +v0.11.1 - 2019-10-07 + - Internal code clean up. + +v0.11.0 - 2019-10-06 + - API CHANGE: Add support for user defined memory allocation routines. This system allows the program to specify their own memory allocation + routines with a user data pointer for client-specific contextual data. This adds an extra parameter to the end of the following APIs: + - drwav_init() + - drwav_init_ex() + - drwav_init_file() + - drwav_init_file_ex() + - drwav_init_file_w() + - drwav_init_file_w_ex() + - drwav_init_memory() + - drwav_init_memory_ex() + - drwav_init_write() + - drwav_init_write_sequential() + - drwav_init_write_sequential_pcm_frames() + - drwav_init_file_write() + - drwav_init_file_write_sequential() + - drwav_init_file_write_sequential_pcm_frames() + - drwav_init_file_write_w() + - drwav_init_file_write_sequential_w() + - drwav_init_file_write_sequential_pcm_frames_w() + - drwav_init_memory_write() + - drwav_init_memory_write_sequential() + - drwav_init_memory_write_sequential_pcm_frames() + - drwav_open_and_read_pcm_frames_s16() + - drwav_open_and_read_pcm_frames_f32() + - drwav_open_and_read_pcm_frames_s32() + - drwav_open_file_and_read_pcm_frames_s16() + - drwav_open_file_and_read_pcm_frames_f32() + - drwav_open_file_and_read_pcm_frames_s32() + - drwav_open_file_and_read_pcm_frames_s16_w() + - drwav_open_file_and_read_pcm_frames_f32_w() + - drwav_open_file_and_read_pcm_frames_s32_w() + - drwav_open_memory_and_read_pcm_frames_s16() + - drwav_open_memory_and_read_pcm_frames_f32() + - drwav_open_memory_and_read_pcm_frames_s32() + Set this extra parameter to NULL to use defaults which is the same as the previous behaviour. Setting this NULL will use + DRWAV_MALLOC, DRWAV_REALLOC and DRWAV_FREE. + - Add support for reading and writing PCM frames in an explicit endianness. New APIs: + - drwav_read_pcm_frames_le() + - drwav_read_pcm_frames_be() + - drwav_read_pcm_frames_s16le() + - drwav_read_pcm_frames_s16be() + - drwav_read_pcm_frames_f32le() + - drwav_read_pcm_frames_f32be() + - drwav_read_pcm_frames_s32le() + - drwav_read_pcm_frames_s32be() + - drwav_write_pcm_frames_le() + - drwav_write_pcm_frames_be() + - Remove deprecated APIs. + - API CHANGE: The following APIs now return native-endian data. Previously they returned little-endian data. + - drwav_read_pcm_frames() + - drwav_read_pcm_frames_s16() + - drwav_read_pcm_frames_s32() + - drwav_read_pcm_frames_f32() + - drwav_open_and_read_pcm_frames_s16() + - drwav_open_and_read_pcm_frames_s32() + - drwav_open_and_read_pcm_frames_f32() + - drwav_open_file_and_read_pcm_frames_s16() + - drwav_open_file_and_read_pcm_frames_s32() + - drwav_open_file_and_read_pcm_frames_f32() + - drwav_open_file_and_read_pcm_frames_s16_w() + - drwav_open_file_and_read_pcm_frames_s32_w() + - drwav_open_file_and_read_pcm_frames_f32_w() + - drwav_open_memory_and_read_pcm_frames_s16() + - drwav_open_memory_and_read_pcm_frames_s32() + - drwav_open_memory_and_read_pcm_frames_f32() + +v0.10.1 - 2019-08-31 + - Correctly handle partial trailing ADPCM blocks. + +v0.10.0 - 2019-08-04 + - Remove deprecated APIs. + - Add wchar_t variants for file loading APIs: + drwav_init_file_w() + drwav_init_file_ex_w() + drwav_init_file_write_w() + drwav_init_file_write_sequential_w() + - Add drwav_target_write_size_bytes() which calculates the total size in bytes of a WAV file given a format and sample count. + - Add APIs for specifying the PCM frame count instead of the sample count when opening in sequential write mode: + drwav_init_write_sequential_pcm_frames() + drwav_init_file_write_sequential_pcm_frames() + drwav_init_file_write_sequential_pcm_frames_w() + drwav_init_memory_write_sequential_pcm_frames() + - Deprecate drwav_open*() and drwav_close(): + drwav_open() + drwav_open_ex() + drwav_open_write() + drwav_open_write_sequential() + drwav_open_file() + drwav_open_file_ex() + drwav_open_file_write() + drwav_open_file_write_sequential() + drwav_open_memory() + drwav_open_memory_ex() + drwav_open_memory_write() + drwav_open_memory_write_sequential() + drwav_close() + - Minor documentation updates. + +v0.9.2 - 2019-05-21 + - Fix warnings. + +v0.9.1 - 2019-05-05 + - Add support for C89. + - Change license to choice of public domain or MIT-0. + +v0.9.0 - 2018-12-16 + - API CHANGE: Add new reading APIs for reading by PCM frames instead of samples. Old APIs have been deprecated and + will be removed in v0.10.0. Deprecated APIs and their replacements: + drwav_read() -> drwav_read_pcm_frames() + drwav_read_s16() -> drwav_read_pcm_frames_s16() + drwav_read_f32() -> drwav_read_pcm_frames_f32() + drwav_read_s32() -> drwav_read_pcm_frames_s32() + drwav_seek_to_sample() -> drwav_seek_to_pcm_frame() + drwav_write() -> drwav_write_pcm_frames() + drwav_open_and_read_s16() -> drwav_open_and_read_pcm_frames_s16() + drwav_open_and_read_f32() -> drwav_open_and_read_pcm_frames_f32() + drwav_open_and_read_s32() -> drwav_open_and_read_pcm_frames_s32() + drwav_open_file_and_read_s16() -> drwav_open_file_and_read_pcm_frames_s16() + drwav_open_file_and_read_f32() -> drwav_open_file_and_read_pcm_frames_f32() + drwav_open_file_and_read_s32() -> drwav_open_file_and_read_pcm_frames_s32() + drwav_open_memory_and_read_s16() -> drwav_open_memory_and_read_pcm_frames_s16() + drwav_open_memory_and_read_f32() -> drwav_open_memory_and_read_pcm_frames_f32() + drwav_open_memory_and_read_s32() -> drwav_open_memory_and_read_pcm_frames_s32() + drwav::totalSampleCount -> drwav::totalPCMFrameCount + - API CHANGE: Rename drwav_open_and_read_file_*() to drwav_open_file_and_read_*(). + - API CHANGE: Rename drwav_open_and_read_memory_*() to drwav_open_memory_and_read_*(). + - Add built-in support for smpl chunks. + - Add support for firing a callback for each chunk in the file at initialization time. + - This is enabled through the drwav_init_ex(), etc. family of APIs. + - Handle invalid FMT chunks more robustly. + +v0.8.5 - 2018-09-11 + - Const correctness. + - Fix a potential stack overflow. + +v0.8.4 - 2018-08-07 + - Improve 64-bit detection. + +v0.8.3 - 2018-08-05 + - Fix C++ build on older versions of GCC. + +v0.8.2 - 2018-08-02 + - Fix some big-endian bugs. + +v0.8.1 - 2018-06-29 + - Add support for sequential writing APIs. + - Disable seeking in write mode. + - Fix bugs with Wave64. + - Fix typos. + +v0.8 - 2018-04-27 + - Bug fix. + - Start using major.minor.revision versioning. + +v0.7f - 2018-02-05 + - Restrict ADPCM formats to a maximum of 2 channels. + +v0.7e - 2018-02-02 + - Fix a crash. + +v0.7d - 2018-02-01 + - Fix a crash. + +v0.7c - 2018-02-01 + - Set drwav.bytesPerSample to 0 for all compressed formats. + - Fix a crash when reading 16-bit floating point WAV files. In this case dr_wav will output silence for + all format conversion reading APIs (*_s16, *_s32, *_f32 APIs). + - Fix some divide-by-zero errors. + +v0.7b - 2018-01-22 + - Fix errors with seeking of compressed formats. + - Fix compilation error when DR_WAV_NO_CONVERSION_API + +v0.7a - 2017-11-17 + - Fix some GCC warnings. + +v0.7 - 2017-11-04 + - Add writing APIs. + +v0.6 - 2017-08-16 + - API CHANGE: Rename dr_* types to drwav_*. + - Add support for custom implementations of malloc(), realloc(), etc. + - Add support for Microsoft ADPCM. + - Add support for IMA ADPCM (DVI, format code 0x11). + - Optimizations to drwav_read_s16(). + - Bug fixes. + +v0.5g - 2017-07-16 + - Change underlying type for booleans to unsigned. + +v0.5f - 2017-04-04 + - Fix a minor bug with drwav_open_and_read_s16() and family. + +v0.5e - 2016-12-29 + - Added support for reading samples as signed 16-bit integers. Use the _s16() family of APIs for this. + - Minor fixes to documentation. + +v0.5d - 2016-12-28 + - Use drwav_int* and drwav_uint* sized types to improve compiler support. + +v0.5c - 2016-11-11 + - Properly handle JUNK chunks that come before the FMT chunk. + +v0.5b - 2016-10-23 + - A minor change to drwav_bool8 and drwav_bool32 types. + +v0.5a - 2016-10-11 + - Fixed a bug with drwav_open_and_read() and family due to incorrect argument ordering. + - Improve A-law and mu-law efficiency. + +v0.5 - 2016-09-29 + - API CHANGE. Swap the order of "channels" and "sampleRate" parameters in drwav_open_and_read*(). Rationale for this is to + keep it consistent with dr_audio and dr_flac. + +v0.4b - 2016-09-18 + - Fixed a typo in documentation. + +v0.4a - 2016-09-18 + - Fixed a typo. + - Change date format to ISO 8601 (YYYY-MM-DD) + +v0.4 - 2016-07-13 + - API CHANGE. Make onSeek consistent with dr_flac. + - API CHANGE. Rename drwav_seek() to drwav_seek_to_sample() for clarity and consistency with dr_flac. + - Added support for Sony Wave64. + +v0.3a - 2016-05-28 + - API CHANGE. Return drwav_bool32 instead of int in onSeek callback. + - Fixed a memory leak. + +v0.3 - 2016-05-22 + - Lots of API changes for consistency. + +v0.2a - 2016-05-16 + - Fixed Linux/GCC build. + +v0.2 - 2016-05-11 + - Added support for reading data as signed 32-bit PCM for consistency with dr_flac. + +v0.1a - 2016-05-07 + - Fixed a bug in drwav_open_file() where the file handle would not be closed if the loader failed to initialize. + +v0.1 - 2016-05-04 + - Initial versioned release. +*/ + +/* +This software is available as a choice of the following licenses. Choose +whichever you prefer. + +=============================================================================== +ALTERNATIVE 1 - Public Domain (www.unlicense.org) +=============================================================================== +This is free and unencumbered software released into the public domain. + +Anyone is free to copy, modify, publish, use, compile, sell, or distribute this +software, either in source code form or as a compiled binary, for any purpose, +commercial or non-commercial, and by any means. + +In jurisdictions that recognize copyright laws, the author or authors of this +software dedicate any and all copyright interest in the software to the public +domain. We make this dedication for the benefit of the public at large and to +the detriment of our heirs and successors. We intend this dedication to be an +overt act of relinquishment in perpetuity of all present and future rights to +this software under copyright law. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER LIABILITY, WHETHER IN AN +ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, OUT OF OR IN CONNECTION +WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE SOFTWARE. + +For more information, please refer to + +=============================================================================== +ALTERNATIVE 2 - MIT No Attribution +=============================================================================== +Copyright 2023 David Reid + +Permission is hereby granted, free of charge, to any person obtaining a copy of +this software and associated documentation files (the "Software"), to deal in +the Software without restriction, including without limitation the rights to +use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies +of the Software, and to permit persons to whom the Software is furnished to do +so. + +THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR +IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, +FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL THE +AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER +LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING FROM, +OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS IN THE +SOFTWARE. +*/ diff --git a/src/include/whisper/modeling_whisper.hpp b/src/include/whisper/modeling_whisper.hpp index 00b031cfa..8b5dd57e1 100644 --- a/src/include/whisper/modeling_whisper.hpp +++ b/src/include/whisper/modeling_whisper.hpp @@ -15,14 +15,6 @@ #include "utils/profiler.hpp" #include "lm_config.hpp" #include "tensor_utils/q4_npu_eXpress.hpp" -// FFmpeg includes for image processing only -extern "C" { -#include -#include -#include -#include -#include -} #include #include "typedef.hpp" diff --git a/src/test/CMakeLists.txt b/src/test/CMakeLists.txt index cac196b84..0c0b631b1 100644 --- a/src/test/CMakeLists.txt +++ b/src/test/CMakeLists.txt @@ -112,6 +112,19 @@ function(add_npu_test TARGET_NAME OUTPUT_SUBDIR) target_link_libraries(${TARGET_NAME} PRIVATE flm_image_codecs) endif() + # AudioReader (also used by Whisper) decodes with libopus/libvorbis/dr_libs + # (see src/cmake/FlmAudioCodecs.cmake). + if(NPU_TEST_SOURCES MATCHES "audio/audio_reader\\.cpp|whisper/modeling_whisper_audio\\.cpp") + include(${CMAKE_CURRENT_FUNCTION_LIST_DIR}/../cmake/FlmAudioCodecs.cmake) + target_sources(${TARGET_NAME} PRIVATE + ${CMAKE_CURRENT_FUNCTION_LIST_DIR}/../common/audio/audio_decoder.cpp + ${CMAKE_CURRENT_FUNCTION_LIST_DIR}/../common/audio/audio_resampler.cpp) + if(NOT NPU_TEST_SOURCES MATCHES "audio/audio_reader\\.cpp") + target_sources(${TARGET_NAME} PRIVATE ${CMAKE_CURRENT_FUNCTION_LIST_DIR}/../common/audio/audio_reader.cpp) + endif() + target_link_libraries(${TARGET_NAME} PRIVATE flm_audio_codecs) + endif() + target_link_libraries(${TARGET_NAME} PRIVATE q4_npu_eXpress lm_head diff --git a/src/test/gemma4_12b/CMakeLists.txt b/src/test/gemma4_12b/CMakeLists.txt index f4e21e309..876b8d215 100644 --- a/src/test/gemma4_12b/CMakeLists.txt +++ b/src/test/gemma4_12b/CMakeLists.txt @@ -39,9 +39,6 @@ endif() target_link_libraries(test_gemma4_12b PUBLIC gemma4_12b_npu xrt_coreutil - avformat - avcodec - avutil zlib ) diff --git a/src/test/gemma4_12b/Makefile b/src/test/gemma4_12b/Makefile index 80e608074..6e7bf9bf4 100644 --- a/src/test/gemma4_12b/Makefile +++ b/src/test/gemma4_12b/Makefile @@ -28,6 +28,8 @@ SOURCES += ../../common/image_process_utils/imageprocAVX512.cpp SOURCES += ../../common/image/image_reader.cpp SOURCES += ../../include/lodepng/lodepng.cpp SOURCES += ../../common/audio/audio_reader.cpp +SOURCES += ../../common/audio/audio_decoder.cpp +SOURCES += ../../common/audio/audio_resampler.cpp SOURCES += ../../common/tokenizer/tokenizer.cpp SOURCES += ../../common/modules/sampler.cpp @@ -37,7 +39,7 @@ HEADERS += ../../include/AutoModel/modeling_gemma4_12b.hpp ifeq ($(WSL), 0) LDFLAGS += -lgemma4_12b_npu -LDFLAGS += -lavformat -lavcodec -lavutil -lswresample -ljpeg +LDFLAGS += -ljpeg -lvorbisfile -lvorbis -logg -lopus CPP_SOURCES := $(filter %.cpp,$(SOURCES)) OBJECTS := $(addprefix $(BUILD_DIR)/,$(notdir $(CPP_SOURCES:.cpp=.o))) diff --git a/src/test/gemma4_core/test_gemma4_frontend.cpp b/src/test/gemma4_core/test_gemma4_frontend.cpp index bbb953e5b..402b34d47 100644 --- a/src/test/gemma4_core/test_gemma4_frontend.cpp +++ b/src/test/gemma4_core/test_gemma4_frontend.cpp @@ -225,14 +225,14 @@ lm_uniform_input_t TextInput(const std::string& prompt) { // - SafeTensors, ImageMemoryPool/ImageReader, AudioReader: value members // Gemma4e (or the headers it drags in) constructs and destroys as part of // an ordinary object lifetime. Their real implementations open weight -// files or decode through FFmpeg; a text-only test never calls into them, +// files or decode media (libjpeg-turbo/libopus/...); a text-only test never calls into them, // so only construction/destruction needs to link. // - Gemma4e's own load_image/load_image_base64/preprocess_image/load_audio/ // load_audio_base64/clip_audio_length/extract_spectrogram: implemented in // modeling_gemma4e_image.cpp and modeling_gemma4e_audio.cpp, which this // target deliberately does not compile (task A3 is scoped to the // dynamic_cast helper in modeling_gemma4e.cpp only, and pulling those two -/// files in would pull in FFmpeg). insert() still *calls* these on the +/// files in would pull in the image/audio codec libraries). insert() still *calls* these on the // image/audio branches, so the symbols must exist; a text-only run never // takes those branches, so the bodies are never exercised. // --------------------------------------------------------------------------- diff --git a/src/test/gemma4e_flash/CMakeLists.txt b/src/test/gemma4e_flash/CMakeLists.txt index 4f23d0b4f..d7d1eade6 100644 --- a/src/test/gemma4e_flash/CMakeLists.txt +++ b/src/test/gemma4e_flash/CMakeLists.txt @@ -52,10 +52,6 @@ target_link_libraries(test_gemma4e_flash PUBLIC # modeling_gemma4e.cpp serves both wrappers, so both engines must be linked. gemma4e_npu xrt_coreutil - avformat - avcodec - avutil - swresample zlib ) diff --git a/src/test/gemma4e_flash/Makefile b/src/test/gemma4e_flash/Makefile index 09e6139df..a50a14e8d 100644 --- a/src/test/gemma4e_flash/Makefile +++ b/src/test/gemma4e_flash/Makefile @@ -28,6 +28,8 @@ SOURCES += ../../common/audio_process_utils/audioprocAVX512.cpp SOURCES += ../../common/image/image_reader.cpp SOURCES += ../../include/lodepng/lodepng.cpp SOURCES += ../../common/audio/audio_reader.cpp +SOURCES += ../../common/audio/audio_decoder.cpp +SOURCES += ../../common/audio/audio_resampler.cpp SOURCES += ../../common/tokenizer/tokenizer.cpp SOURCES += ../../common/modules/sampler.cpp @@ -50,7 +52,7 @@ ifeq ($(WSL), 0) LDFLAGS += -lgemma4e_flash # modeling_gemma4e.cpp serves both wrappers, so both engines must be linked. LDFLAGS += -lgemma4e_npu -LDFLAGS += -lavformat -lavcodec -lavutil -lswresample -ljpeg +LDFLAGS += -ljpeg -lvorbisfile -lvorbis -logg -lopus CPP_SOURCES := $(filter %.cpp,$(SOURCES)) OBJECTS := $(addprefix $(BUILD_DIR)/,$(notdir $(CPP_SOURCES:.cpp=.o))) diff --git a/src/test/gemma4e_npu/CMakeLists.txt b/src/test/gemma4e_npu/CMakeLists.txt index 9a2b9d52d..6f360fbf6 100644 --- a/src/test/gemma4e_npu/CMakeLists.txt +++ b/src/test/gemma4e_npu/CMakeLists.txt @@ -52,10 +52,6 @@ target_link_libraries(test_gemma4e_npu PUBLIC # modeling_gemma4e.cpp serves both wrappers, so both engines must be linked. gemma4e_flash xrt_coreutil - avformat - avcodec - avutil - swresample zlib ) diff --git a/src/test/gemma4e_npu/Makefile b/src/test/gemma4e_npu/Makefile index 6c5fbf7d0..c76c651a4 100644 --- a/src/test/gemma4e_npu/Makefile +++ b/src/test/gemma4e_npu/Makefile @@ -28,6 +28,8 @@ SOURCES += ../../common/audio_process_utils/audioprocAVX512.cpp SOURCES += ../../common/image/image_reader.cpp SOURCES += ../../include/lodepng/lodepng.cpp SOURCES += ../../common/audio/audio_reader.cpp +SOURCES += ../../common/audio/audio_decoder.cpp +SOURCES += ../../common/audio/audio_resampler.cpp SOURCES += ../../common/tokenizer/tokenizer.cpp SOURCES += ../../common/modules/sampler.cpp @@ -48,7 +50,7 @@ ifeq ($(WSL), 0) LDFLAGS += -lgemma4e_npu # modeling_gemma4e.cpp serves both wrappers, so both engines must be linked. LDFLAGS += -lgemma4e_flash -LDFLAGS += -lavformat -lavcodec -lavutil -lswresample -ljpeg +LDFLAGS += -ljpeg -lvorbisfile -lvorbis -logg -lopus CPP_SOURCES := $(filter %.cpp,$(SOURCES)) OBJECTS := $(addprefix $(BUILD_DIR)/,$(notdir $(CPP_SOURCES:.cpp=.o))) diff --git a/src/test/minicpm_v_4_7_npu/Makefile b/src/test/minicpm_v_4_7_npu/Makefile index f0479d312..9d8c61e61 100644 --- a/src/test/minicpm_v_4_7_npu/Makefile +++ b/src/test/minicpm_v_4_7_npu/Makefile @@ -18,7 +18,7 @@ SOURCES += ../../common/AutoModel/automodel.cpp SOURCES += ../../common/AutoModel/model_backend.cpp SOURCES += ../../common/AutoModel/modeling_minicpm_v.cpp # Image support. modeling_minicpm_v_image.cpp holds the slicing, the PIL-exact -# resize and image_placeholder(); image_reader.cpp decodes the file via ffmpeg; +# resize and image_placeholder(); image_reader.cpp decodes the file (libjpeg-turbo/lodepng); # imageprocAVX512.cpp supplies rescale_and_normalize_avx512(), and pulls in # imageproc.cpp for resize_bicubic_* at link time. # Same file list as the other VLM harnesses. MiniCPM does NOT call the shared @@ -42,7 +42,7 @@ HEADERS += ../../include/AutoModel/modeling_minicpm_v.hpp ifeq ($(WSL), 0) LDFLAGS += -lminicpm_v_4_7_npu -# image_reader.cpp decodes through ffmpeg, same as every other VLM harness +# image_reader.cpp decodes through libjpeg-turbo (+ vendored lodepng), same as every other VLM harness LDFLAGS += -ljpeg CPP_SOURCES := $(filter %.cpp,$(SOURCES)) diff --git a/src/test/phi4_core/CMakeLists.txt b/src/test/phi4_core/CMakeLists.txt index fe6113295..2188e5def 100644 --- a/src/test/phi4_core/CMakeLists.txt +++ b/src/test/phi4_core/CMakeLists.txt @@ -166,21 +166,6 @@ target_link_directories(test_phi4_frontend_off PRIVATE "${XRT_INCLUDE_DIR}/../li target_link_libraries(test_phi4_frontend_off PRIVATE xrt_coreutil) # Compile the actual production frontend translation units in both feature modes. -# Empty declaration-only FFmpeg headers isolate this compile check from an -# unrelated optional SDK that is absent on the standalone test host. -set(FRONTEND_STUB_INCLUDE_DIR "${CMAKE_CURRENT_BINARY_DIR}/frontend-compile-stubs") -foreach(STUB_HEADER IN ITEMS - libavcodec/avcodec.h - libavformat/avformat.h - libswscale/swscale.h - libavutil/imgutils.h - libavutil/frame.h - libavutil/pixfmt.h) - get_filename_component(STUB_PARENT - "${FRONTEND_STUB_INCLUDE_DIR}/${STUB_HEADER}" DIRECTORY) - file(MAKE_DIRECTORY "${STUB_PARENT}") - file(WRITE "${FRONTEND_STUB_INCLUDE_DIR}/${STUB_HEADER}" "#pragma once\n") -endforeach() set(FLM_PRODUCTION_FRONTEND_SOURCES ${PHI4_FRONTEND_SOURCES} @@ -196,7 +181,6 @@ function(add_frontend_compile_guard TARGET_NAME ENABLE_CORELIB) "${FLM_SOURCE_DIR}/server" "${FLM_SOURCE_DIR}/pull" "${FLM_SOURCE_DIR}/../third_party/tokenizers-cpp/include" - "${FRONTEND_STUB_INCLUDE_DIR}" "${BOOST_INCLUDE_DIR}" "${XRT_INCLUDE_DIR}") target_compile_definitions(${TARGET_NAME} PRIVATE diff --git a/src/test/qwen3_5_omni_npu/CMakeLists.txt b/src/test/qwen3_5_omni_npu/CMakeLists.txt index 2a632891b..3d03e0643 100644 --- a/src/test/qwen3_5_omni_npu/CMakeLists.txt +++ b/src/test/qwen3_5_omni_npu/CMakeLists.txt @@ -50,9 +50,6 @@ endif() target_link_libraries(test_qwen3_5_omni_npu PUBLIC qwen3_5_omni_npu xrt_coreutil - avformat - avcodec - avutil zlib ) diff --git a/src/test/qwen3_5_omni_npu/Makefile b/src/test/qwen3_5_omni_npu/Makefile index 2adf349c9..eadb47639 100644 --- a/src/test/qwen3_5_omni_npu/Makefile +++ b/src/test/qwen3_5_omni_npu/Makefile @@ -30,6 +30,8 @@ SOURCES += ../../common/audio_process_utils/audioprocAVX512.cpp SOURCES += ../../common/image/image_reader.cpp SOURCES += ../../include/lodepng/lodepng.cpp SOURCES += ../../common/audio/audio_reader.cpp +SOURCES += ../../common/audio/audio_decoder.cpp +SOURCES += ../../common/audio/audio_resampler.cpp SOURCES += ../../common/tokenizer/tokenizer.cpp SOURCES += ../../common/modules/sampler.cpp @@ -47,7 +49,7 @@ TYPE ?= 0 ifeq ($(WSL), 0) LDFLAGS += -lqwen3_5_omni_npu -LDFLAGS += -lavformat -lavcodec -lavutil -lswresample -ljpeg +LDFLAGS += -ljpeg -lvorbisfile -lvorbis -logg -lopus # Map each source to an object file in BUILD_DIR (basenames are unique across # the source set). VPATH lets the pattern rule locate sources in their dirs. diff --git a/src/test/qwen3_npu/CMakeLists.txt b/src/test/qwen3_npu/CMakeLists.txt index c2d7d3269..c0da5819f 100644 --- a/src/test/qwen3_npu/CMakeLists.txt +++ b/src/test/qwen3_npu/CMakeLists.txt @@ -17,10 +17,6 @@ add_npu_test( target_link_libraries(test_qwen3_npu PUBLIC qwen3_npu xrt_coreutil - avformat - avcodec - avutil - swscale zlib ) diff --git a/src/test/whisper_npu/CMakeLists.txt b/src/test/whisper_npu/CMakeLists.txt index 2328057ea..e402754db 100644 --- a/src/test/whisper_npu/CMakeLists.txt +++ b/src/test/whisper_npu/CMakeLists.txt @@ -29,11 +29,6 @@ target_link_libraries(test_whisper_npu PUBLIC whisper_npu xrt_coreutil aiebu_static - avformat - avcodec - avutil - swscale - swresample zlib libboost_program_options-vc143-mt-x64-1_88 ) diff --git a/src/test/whisper_npu/Makefile b/src/test/whisper_npu/Makefile index 13bbc35ab..69020d289 100644 --- a/src/test/whisper_npu/Makefile +++ b/src/test/whisper_npu/Makefile @@ -21,6 +21,9 @@ SOURCES += ../../common/AutoModel/automodel.cpp SOURCES += ../../common/modules/sampler.cpp SOURCES += ../../common/tokenizer/tokenizer.cpp SOURCES += ../../common/whisper/modeling_whisper_audio.cpp +SOURCES += ../../common/audio/audio_reader.cpp +SOURCES += ../../common/audio/audio_decoder.cpp +SOURCES += ../../common/audio/audio_resampler.cpp SOURCES += ../../common/whisper/modeling_whisper.cpp HEADERS += ../../include/llama/whisper_npu.hpp @@ -35,7 +38,7 @@ ifeq ($(WSL), 0) # Use g++-13 directly without CMake LDFLAGS += -lwhisper_npu -LDFLAGS += -lavformat -lavcodec -lavutil -lswscale -lswresample +LDFLAGS += -lvorbisfile -lvorbis -logg -lopus CPP_SOURCES := $(filter %.cpp,$(SOURCES)) From 6fd56face0eea070d36932b9da578de5470fa969 Mon Sep 17 00:00:00 2001 From: tawei-amd Date: Fri, 16 Oct 2026 18:00:55 -0700 Subject: [PATCH 2/6] build(audio): let libvorbis 1.3.7 configure under CMake 4 libvorbis 1.3.7 declares cmake_minimum_required(VERSION 2.8.12), which CMake 4 (CI runners and build containers) rejects: 'Compatibility with CMake < 3.5 has been removed'. Pass CMAKE_POLICY_VERSION_MINIMUM=3.5 to that ExternalProject; older CMake ignores it. Verified with CMake 3.28 and 4.4. Co-Authored-By: Claude --- src/cmake/FlmAudioCodecs.cmake | 6 +++++- 1 file changed, 5 insertions(+), 1 deletion(-) diff --git a/src/cmake/FlmAudioCodecs.cmake b/src/cmake/FlmAudioCodecs.cmake index e9650d078..f8b6a1d60 100644 --- a/src/cmake/FlmAudioCodecs.cmake +++ b/src/cmake/FlmAudioCodecs.cmake @@ -67,10 +67,14 @@ _flm_audio_project(ogg 83e6704730683d004d20e21b8f7f55dcb3383cdf84c0daedf30bde175f774638 ${_flm_ogg_lib} -DINSTALL_DOCS=OFF) +# libvorbis 1.3.7 declares cmake_minimum_required(VERSION 2.8.12), which +# CMake 4 rejects; let it configure under 3.5 policies (older CMake ignores +# this variable). _flm_audio_project(vorbis https://downloads.xiph.org/releases/vorbis/libvorbis-1.3.7.tar.gz 0e982409a9c3fc82ee06e08205b1355e5c6aa4c36bca58146ef399621b0ce5ab - "${_flm_vorbis_lib};${_flm_vorbisfile_lib}") + "${_flm_vorbis_lib};${_flm_vorbisfile_lib}" + -DCMAKE_POLICY_VERSION_MINIMUM=3.5) ExternalProject_Add_StepDependencies(flm_vorbis configure flm_ogg) # The include directory must exist at configure time for the imported targets. From 5c3f092ef40b6b37960ef2dae523e4ab23192e69 Mon Sep 17 00:00:00 2001 From: tawei-amd Date: Fri, 16 Oct 2026 18:07:21 -0700 Subject: [PATCH 3/6] build(audio): add download mirrors for the Xiph tarballs A self-hosted Windows runner had its connection to downloads.xiph.org reset while fetching opus-1.6.1.tar.gz. Give each ExternalProject a mirror list (GitHub release assets where they exist, downloads.xiph.org, and the OSUOSL mirror); CMake tries them in order and the pinned SHA-256 still guarantees the same tarball. All mirrors verified hash-identical. Co-Authored-By: Claude --- src/cmake/FlmAudioCodecs.cmake | 13 +++++++------ 1 file changed, 7 insertions(+), 6 deletions(-) diff --git a/src/cmake/FlmAudioCodecs.cmake b/src/cmake/FlmAudioCodecs.cmake index f8b6a1d60..cd74dd7e9 100644 --- a/src/cmake/FlmAudioCodecs.cmake +++ b/src/cmake/FlmAudioCodecs.cmake @@ -36,11 +36,12 @@ if(CMAKE_C_COMPILER) endif() # Short working directories keep MSBuild under Windows' 260-char MAX_PATH on -# CI runners with deep work directories (see FlmImageCodecs.cmake). -function(_flm_audio_project name url sha256 byproduct) +# CI runners with deep work directories (see FlmImageCodecs.cmake). `urls` is +# a list of mirrors tried in order; the SHA-256 pins the exact tarball. +function(_flm_audio_project name urls sha256 byproduct) set(_work ${CMAKE_BINARY_DIR}/ac/${name}) ExternalProject_Add(flm_${name} - URL ${url} + URL ${urls} URL_HASH SHA256=${sha256} DOWNLOAD_EXTRACT_TIMESTAMP TRUE EXCLUDE_FROM_ALL TRUE @@ -56,14 +57,14 @@ function(_flm_audio_project name url sha256 byproduct) endfunction() _flm_audio_project(opus - https://downloads.xiph.org/releases/opus/opus-1.6.1.tar.gz + "https://downloads.xiph.org/releases/opus/opus-1.6.1.tar.gz;https://ftp.osuosl.org/pub/xiph/releases/opus/opus-1.6.1.tar.gz" 6ffcb593207be92584df15b32466ed64bbec99109f007c82205f0194572411a1 ${_flm_opus_lib} -DOPUS_BUILD_PROGRAMS=OFF -DOPUS_INSTALL_PKG_CONFIG_MODULE=OFF -DOPUS_INSTALL_CMAKE_CONFIG_MODULE=OFF) _flm_audio_project(ogg - https://downloads.xiph.org/releases/ogg/libogg-1.3.6.tar.gz + "https://github.com/xiph/ogg/releases/download/v1.3.6/libogg-1.3.6.tar.gz;https://downloads.xiph.org/releases/ogg/libogg-1.3.6.tar.gz;https://ftp.osuosl.org/pub/xiph/releases/ogg/libogg-1.3.6.tar.gz" 83e6704730683d004d20e21b8f7f55dcb3383cdf84c0daedf30bde175f774638 ${_flm_ogg_lib} -DINSTALL_DOCS=OFF) @@ -71,7 +72,7 @@ _flm_audio_project(ogg # CMake 4 rejects; let it configure under 3.5 policies (older CMake ignores # this variable). _flm_audio_project(vorbis - https://downloads.xiph.org/releases/vorbis/libvorbis-1.3.7.tar.gz + "https://github.com/xiph/vorbis/releases/download/v1.3.7/libvorbis-1.3.7.tar.gz;https://downloads.xiph.org/releases/vorbis/libvorbis-1.3.7.tar.gz;https://ftp.osuosl.org/pub/xiph/releases/vorbis/libvorbis-1.3.7.tar.gz" 0e982409a9c3fc82ee06e08205b1355e5c6aa4c36bca58146ef399621b0ce5ab "${_flm_vorbis_lib};${_flm_vorbisfile_lib}" -DCMAKE_POLICY_VERSION_MINIMUM=3.5) From 581d395c12ee5cec4b1bb8ddb6d59be3eb7df44a Mon Sep 17 00:00:00 2001 From: tawei-amd Date: Fri, 16 Oct 2026 18:18:51 -0700 Subject: [PATCH 4/6] fix(audio): build with MSVC (no M_PI) M_PI is not standard C++; MSVC only defines it with _USE_MATH_DEFINES, so audio_resampler.cpp failed to compile on Windows. Use a local constant. Also include explicitly for SEEK_* in audio_decoder.cpp instead of relying on transitive includes. Co-Authored-By: Claude --- src/common/audio/audio_decoder.cpp | 1 + src/common/audio/audio_resampler.cpp | 6 ++++-- 2 files changed, 5 insertions(+), 2 deletions(-) diff --git a/src/common/audio/audio_decoder.cpp b/src/common/audio/audio_decoder.cpp index 453427066..a0b62cdd5 100644 --- a/src/common/audio/audio_decoder.cpp +++ b/src/common/audio/audio_decoder.cpp @@ -8,6 +8,7 @@ #include #include +#include #include #include diff --git a/src/common/audio/audio_resampler.cpp b/src/common/audio/audio_resampler.cpp index b37eb1b69..d3df74226 100644 --- a/src/common/audio/audio_resampler.cpp +++ b/src/common/audio/audio_resampler.cpp @@ -17,6 +17,8 @@ constexpr int kFilterSize = 32; constexpr int kPhaseShift = 10; constexpr double kCutoff = 0.97; constexpr double kKaiserBeta = 9.0; +// M_PI is not standard (MSVC needs _USE_MATH_DEFINES). +constexpr double kPi = 3.14159265358979323846; // Zeroth-order modified Bessel function of the first kind (power series). double bessel_i0(double x) { @@ -43,9 +45,9 @@ std::vector build_filter_bank(double factor, int taps, int phases) { double norm = 0.0; for (int ph = 0; ph <= phases; ++ph) { for (int i = 0; i < taps; ++i) { - const double x = M_PI * ((i - center) - static_cast(ph) / phases) * factor; + const double x = kPi * ((i - center) - static_cast(ph) / phases) * factor; double y = x == 0.0 ? 1.0 : std::sin(x) / x; - const double w = 2.0 * x / (factor * taps * M_PI); + const double w = 2.0 * x / (factor * taps * kPi); y *= bessel_i0(kKaiserBeta * std::sqrt(std::max(1.0 - w * w, 0.0))); row[i] = y; if (ph == 0) { From 9a0f1fc77c78015d885bfae35974ba1310d660e9 Mon Sep 17 00:00:00 2001 From: tawei-amd Date: Fri, 16 Oct 2026 23:39:23 -0700 Subject: [PATCH 5/6] fix(audio): decode MP3 in float; reject implausible sample rates The ASR CI on this PR showed gemma4 chat WER worse on every platform (5.05% -> 8-9%). All eval clips are 16 kHz mono MP3, and dr_mp3 by default synthesizes 16-bit PCM and converts it to float, which on these quiet clips is only ~35-45 dB from FFmpeg/mpg123 (they agree with each other at 108-121 dB). Define DR_MP3_FLOAT_OUTPUT: our decode now matches mpg123 at >=127.8 dB and main's FFmpeg input at >=108 dB on all 84 eval clips (identical lengths), and synthetic/real MP3s at 118-124 dB. Fuzzing also found that a WAV header claiming a 1 Hz sample rate makes the 16 kHz resample of a 0.5 MB upload ~8.5 GB. Accept only 4-768 kHz source rates, which caps resampling expansion at 4x. 6,000 mutated inputs under ASan+UBSan are clean, with no slow inputs. Co-Authored-By: Claude --- src/common/audio/audio_decoder.cpp | 12 +++++++++++- 1 file changed, 11 insertions(+), 1 deletion(-) diff --git a/src/common/audio/audio_decoder.cpp b/src/common/audio/audio_decoder.cpp index a0b62cdd5..56023d32e 100644 --- a/src/common/audio/audio_decoder.cpp +++ b/src/common/audio/audio_decoder.cpp @@ -20,6 +20,9 @@ #include "dr_libs/dr_flac.h" #define DR_MP3_IMPLEMENTATION #define DR_MP3_NO_STDIO +// Decode in float. Without this dr_mp3 synthesizes 16-bit PCM and converts it, +// which quantizes quiet recordings audibly (~35 dB SNR on 16 kHz speech). +#define DR_MP3_FLOAT_OUTPUT #include "dr_libs/dr_mp3.h" #include @@ -45,8 +48,15 @@ bool has_bytes(const uint8_t* data, size_t size, size_t offset, const char* magi return offset + len <= size && std::memcmp(data + offset, magic, len) == 0; } +// Real audio sits between telephony (8 kHz) and 384 kHz. Rejecting rates +// outside a generous range also bounds what resampling can amplify: a header +// claiming 1 Hz would otherwise turn a 0.5 MB upload into gigabytes at 16 kHz. +constexpr int kMinSampleRate = 4000; +constexpr int kMaxSampleRate = 768000; + bool valid_layout(int sample_rate, int channels) { - return sample_rate > 0 && channels > 0 && channels <= kMaxChannels; + return sample_rate >= kMinSampleRate && sample_rate <= kMaxSampleRate && + channels > 0 && channels <= kMaxChannels; } // ——————————————————————————————————————————————— From f42e2664cb6c1950b41c04cacf5991f4bc268bd8 Mon Sep 17 00:00:00 2001 From: tawei-amd Date: Sat, 17 Oct 2026 05:47:52 -0700 Subject: [PATCH 6/6] fix(whisper): keep the 16-bit input quantization of the FFmpeg path The previous Whisper decoder asked swresample for s16 output and divided by 32768, so Whisper has always seen 16-bit-quantized audio. Feeding it unquantized floats moved the ASR CI's pooled WER by about +2 points on 3 of 4 platforms (and the opposite way when the input happened to be quantized), i.e. the eval is sensitive to that rounding and this PR is not the place to change it. Quantize exactly as swresample did (lrint(x * 32768), clip to int16, / 32768). On all 84 ASR eval clips the result is identical to main's Whisper input for 99.95% of samples; the rest differ by 1 LSB (MP3 decoder rounding), with identical lengths. Co-Authored-By: Claude --- src/common/whisper/modeling_whisper_audio.cpp | 11 ++++++++--- 1 file changed, 8 insertions(+), 3 deletions(-) diff --git a/src/common/whisper/modeling_whisper_audio.cpp b/src/common/whisper/modeling_whisper_audio.cpp index b484c2003..099f7197c 100644 --- a/src/common/whisper/modeling_whisper_audio.cpp +++ b/src/common/whisper/modeling_whisper_audio.cpp @@ -26,9 +26,10 @@ #include // For AVX intrinsics // Decoding, downmix and resampling go through AudioReader. Whisper expects -// 16 kHz mono; stereo is averaged (0.5 L + 0.5 R), which is what the previous -// FFmpeg path produced, but samples now stay float instead of passing through -// 16-bit PCM. +// 16 kHz mono; stereo is averaged (0.5 L + 0.5 R). The samples then pass +// through 16-bit PCM exactly as the previous FFmpeg path did (swresample's +// float -> s16 conversion, then s16 / 32768), so Whisper sees the same input +// it always has. static std::vector load_whisper_audio(AudioReader& reader, const uint8_t* data, size_t size, const std::string& what) { constexpr int TARGET_SAMPLE_RATE = 16000; @@ -38,6 +39,10 @@ static std::vector load_whisper_audio(AudioReader& reader, const uint8_t* if (!ok) { throw std::runtime_error("Could not decode audio " + what); } + for (float& s : audio.samples) { + const long q = std::lrint(s * 32768.0f); + s = static_cast(std::clamp(q, -32768L, 32767L)) / 32768.0f; + } return std::move(audio.samples); }