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Copy pathrtsp_client_multitrack.py
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executable file
·446 lines (392 loc) · 16.3 KB
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"""Multi-track RTSP client that works with rtsp_server_multitrack.py.
Discovers an RTSP server via mDNS, connects, and receives frames.
Demonstrates the new generic frame callback and depacketizer APIs.
Usage:
# Connect to any discovered RTSP server (auto-detect codec from SDP)
python rtsp_client_multitrack.py
# Specify the server path
python rtsp_client_multitrack.py --path /stream
"""
import sys
import time
import queue
import argparse
import socket
import cv2
import numpy as np
from zeroconf import ServiceBrowser, ServiceListener, Zeroconf
import espp
try:
import sounddevice as sd
except ImportError:
sd = None
class MyListener(ServiceListener):
def __init__(self, service_name=None):
self.service_ip = None
self.service_port = None
self.found_service = False
self.service_name = service_name
def update_service(self, zc, type_, name):
pass
def remove_service(self, zc, type_, name):
print(f"Service {name} removed")
def add_service(self, zc, type_, name):
if self.service_name and name != f"{self.service_name}._rtsp._tcp.local.":
return
info = zc.get_service_info(type_, name)
print(f"Service {name} added, service info: {info}")
self.service_ip = socket.inet_ntoa(info.addresses[0])
self.service_port = info.port
self.found_service = True
frame_queue = queue.Queue()
audio_frame_count = 0
audio_valid_frame_count = 0
audio_invalid_frame_count = 0
video_frame_count = 0
decoded_video_frame_count = 0
display_video_frames = True
expected_audio_bytes = 320
min_audio_peak = 0
audio_player = None
audio_sample_rate = 8000
audio_channels = 1
class AudioPlayer:
def __init__(self, sample_rate=8000, channels=1, bytes_per_sample=2, block_samples=160,
device=None):
self.sample_rate = sample_rate
self.channels = channels
self.bytes_per_sample = bytes_per_sample
self.block_samples = block_samples
self.device = device
self.frame_queue = queue.Queue(maxsize=128)
self.pending_bytes = bytearray()
self.stream = None
self.started = False
self.underflow_count = 0
self.dropped_frames = 0
def start(self):
if sd is None:
print("Audio playback unavailable: install sounddevice")
return False
try:
self.stream = sd.RawOutputStream(
samplerate=self.sample_rate,
channels=self.channels,
dtype='int16',
blocksize=self.block_samples,
device=self.device,
callback=self._callback,
)
self.stream.start()
self.started = True
print(f"Audio playback enabled at {self.sample_rate} Hz")
return True
except Exception as exc:
print(f"Audio playback unavailable: {exc}")
self.stream = None
return False
def stop(self):
if self.stream is not None:
try:
self.stream.stop()
finally:
self.stream.close()
self.stream = None
self.started = False
def enqueue(self, frame_bytes):
if not self.started:
return
try:
self.frame_queue.put_nowait(frame_bytes)
except queue.Full:
try:
self.frame_queue.get_nowait()
except queue.Empty:
# Another consumer/drain path already emptied the queue.
pass
self.dropped_frames += 1
try:
self.frame_queue.put_nowait(frame_bytes)
except queue.Full:
self.dropped_frames += 1
def _callback(self, outdata, frames, time_info, status):
del frames, time_info
if status.output_underflow:
self.underflow_count += 1
bytes_needed = len(outdata)
while len(self.pending_bytes) < bytes_needed:
try:
self.pending_bytes.extend(self.frame_queue.get_nowait())
except queue.Empty:
break
available = min(len(self.pending_bytes), bytes_needed)
if available:
outdata[:available] = self.pending_bytes[:available]
del self.pending_bytes[:available]
if available < bytes_needed:
outdata[available:bytes_needed] = b'\x00' * (bytes_needed - available)
self.underflow_count += 1
def as_frame_bytes(data):
if isinstance(data, bytes):
return data
if isinstance(data, bytearray):
return bytes(data)
if isinstance(data, memoryview):
return data.tobytes()
return bytes(data)
def validate_audio_frame(frame_bytes):
global min_audio_peak, expected_audio_bytes
if len(frame_bytes) != expected_audio_bytes:
return False, f"unexpected audio frame size: {len(frame_bytes)}"
if len(frame_bytes) % 2 != 0:
return False, f"audio frame size is not 16-bit aligned: {len(frame_bytes)}"
samples = np.frombuffer(frame_bytes, dtype='<i2')
if samples.size == 0:
return False, "audio frame contained no samples"
peak = int(np.max(np.abs(samples)))
if min_audio_peak > 0 and peak < min_audio_peak:
return False, f"audio peak too small: {peak}"
return True, f"{samples.size} samples, peak {peak}"
def error_message(ec):
if ec is None or not bool(ec):
return ""
return str(ec)
def on_receive_jpeg_frame(frame):
"""Legacy JPEG frame callback."""
global video_frame_count, decoded_video_frame_count
buf = as_frame_bytes(frame.get_data())
video_frame_count += 1
decoded = cv2.imdecode(np.frombuffer(buf, dtype=np.uint8), cv2.IMREAD_COLOR)
if decoded is not None:
decoded_video_frame_count += 1
if display_video_frames:
frame_queue.put_nowait(decoded)
def on_receive_generic_frame(track_id, data):
"""Generic frame callback for any track/codec."""
global audio_frame_count, audio_valid_frame_count, audio_invalid_frame_count
global video_frame_count, decoded_video_frame_count
global audio_player
frame_bytes = as_frame_bytes(data)
if track_id == 0:
video_frame_count += 1
if frame_bytes.startswith(b'\xff\xd8'):
decoded = cv2.imdecode(np.frombuffer(frame_bytes, dtype=np.uint8), cv2.IMREAD_COLOR)
if decoded is not None:
decoded_video_frame_count += 1
if display_video_frames:
frame_queue.put_nowait(decoded)
else:
print(f"[Track {track_id}] JPEG frame: {len(frame_bytes)} bytes (decode failed)")
elif track_id == 1:
audio_frame_count += 1
valid, detail = validate_audio_frame(frame_bytes)
if valid:
audio_valid_frame_count += 1
if audio_player is not None:
audio_player.enqueue(frame_bytes)
else:
audio_invalid_frame_count += 1
print(f"[Track {track_id}] Invalid audio frame: {detail}")
if audio_frame_count % 50 == 0:
print(f"[Track {track_id}] Audio frames received: {audio_frame_count} ({detail})")
else:
print(f"[Track {track_id}] Frame: {len(frame_bytes)} bytes")
def main(argv=None):
global audio_frame_count, audio_valid_frame_count, audio_invalid_frame_count
global video_frame_count, decoded_video_frame_count
global display_video_frames, expected_audio_bytes, min_audio_peak, audio_player
global audio_sample_rate, audio_channels
parser = argparse.ArgumentParser(description='Multi-track RTSP Client')
parser.add_argument('--path', type=str, default='/stream',
help='RTSP stream path (default: /stream)')
parser.add_argument('--service-name', type=str, default=None,
help='Only connect to this mDNS service name')
parser.add_argument('--discovery-timeout', type=float, default=10.0,
help='Seconds to wait for mDNS discovery before failing')
parser.add_argument('--duration', type=float, default=0.0,
help='Auto-stop after this many seconds (default: run until quit)')
parser.add_argument('--headless', action='store_true',
help='Do not create an OpenCV window')
parser.add_argument('--expect-audio', action='store_true',
help='Fail if valid audio frames are not received')
parser.add_argument('--require-decoded-video', action='store_true',
help='Fail if no JPEG video frames are successfully decoded')
parser.add_argument('--min-video-frames', type=int, default=1,
help='Minimum number of video frames expected before success')
parser.add_argument('--min-audio-frames', type=int, default=1,
help='Minimum number of valid audio frames expected before success')
parser.add_argument('--expected-audio-bytes', type=int, default=None,
help='Expected bytes per audio frame (default: derive from SDP or fall back to 320)')
parser.add_argument('--min-audio-peak', type=int, default=0,
help='Minimum absolute sample peak required to treat audio as valid')
parser.add_argument('--play-audio', dest='play_audio', action='store_true',
help='Play the audio track in real time')
parser.add_argument('--no-audio-playback', dest='play_audio', action='store_false',
help='Disable real-time audio playback')
parser.add_argument('--audio-device', type=int, default=None,
help='Optional sounddevice output device index')
parser.add_argument('--legacy', action='store_true',
help='Use legacy JPEG-only mode (on_jpeg_frame callback)')
parser.set_defaults(play_audio=None)
args = parser.parse_args(argv)
display_video_frames = not args.headless
if args.play_audio is None:
args.play_audio = not args.headless
expected_audio_bytes = args.expected_audio_bytes
min_audio_peak = args.min_audio_peak
audio_frame_count = 0
audio_valid_frame_count = 0
audio_invalid_frame_count = 0
video_frame_count = 0
decoded_video_frame_count = 0
audio_player = None
audio_sample_rate = 8000
audio_channels = 1
# Discover RTSP server via mDNS
print("Discovering RTSP service via mDNS...")
zeroconf = Zeroconf()
listener = MyListener(service_name=args.service_name)
browser = ServiceBrowser(zeroconf, "_rtsp._tcp.local.", listener)
discovery_start = time.monotonic()
while not listener.found_service:
if args.discovery_timeout > 0 and (time.monotonic() - discovery_start) >= args.discovery_timeout:
print(f"Error: timed out after {args.discovery_timeout:.1f}s waiting for RTSP service discovery")
browser.cancel()
zeroconf.close()
if audio_player is not None:
audio_player.stop()
return 1
time.sleep(0.1)
server_address = listener.service_ip
server_port = listener.service_port
rtsp_uri = f"rtsp://{server_address}:{server_port}{args.path}"
print(f"Found service at: {rtsp_uri}\n")
# Create RTSP client with appropriate callback mode
if args.legacy:
print("Using legacy JPEG-only mode (on_jpeg_frame)")
client_config = espp.RtspClient.Config(
server_address=server_address,
rtsp_port=server_port,
path=args.path,
on_jpeg_frame=on_receive_jpeg_frame,
log_level=espp.Logger.Verbosity.info
)
else:
print("Using generic multi-track mode (on_frame)")
client_config = espp.RtspClient.Config(
server_address=server_address,
rtsp_port=server_port,
path=args.path,
on_frame=on_receive_generic_frame,
log_level=espp.Logger.Verbosity.info
)
rtsp_client = espp.RtspClient(client_config)
ec = espp.Error()
# Connect and setup
print("Connecting...")
rtsp_client.connect(ec)
connect_error = error_message(ec)
if connect_error:
print(f"Error connecting: {connect_error}")
return 1
print("Describing...")
rtsp_client.describe(ec)
describe_error = error_message(ec)
if describe_error:
print(f"Error describing: {describe_error}")
return 1
tracks = rtsp_client.tracks()
audio_track = next((track for track in tracks if track.media_type.lower() == 'audio'), None)
if audio_track is not None:
audio_sample_rate = audio_track.clock_rate or audio_sample_rate
audio_channels = audio_track.channels or audio_channels
if args.expected_audio_bytes is None:
expected_audio_bytes = max(2 * audio_channels,
(audio_sample_rate * audio_channels * 2) // 50)
print(f"Discovered audio track: PT={audio_track.payload_type}, "
f"{audio_track.encoding_name}/{audio_sample_rate}/{audio_channels}, "
f"expecting {expected_audio_bytes} bytes per frame")
elif expected_audio_bytes is None:
expected_audio_bytes = 320
if args.play_audio and audio_track is not None:
audio_player = AudioPlayer(
sample_rate=audio_sample_rate,
channels=audio_channels,
block_samples=max(1, expected_audio_bytes // max(1, 2 * audio_channels)),
device=args.audio_device,
)
if not audio_player.start():
audio_player = None
print("Setting up...")
rtsp_client.setup(ec)
setup_error = error_message(ec)
if setup_error:
print(f"Error setting up: {setup_error}")
return 1
print("Playing...")
rtsp_client.play(ec)
play_error = error_message(ec)
if play_error:
print(f"Error playing: {play_error}")
return 1
window_name = 'RTSP Multi-Track Client'
if not args.headless:
cv2.namedWindow(window_name, cv2.WINDOW_NORMAL)
cv2.setWindowProperty(window_name, cv2.WND_PROP_TOPMOST, 1)
start_time = time.time()
print("Receiving frames... Press 'q' to quit.\n")
try:
while True:
try:
frame = frame_queue.get_nowait()
if not args.headless:
cv2.imshow(window_name, frame)
except queue.Empty:
# No decoded frame is ready yet from the non-blocking queue.
pass
if args.duration > 0 and (time.time() - start_time) >= args.duration:
break
if not args.headless:
key = cv2.waitKey(1) & 0xFF
if key == ord('q'):
break
else:
time.sleep(0.01)
except KeyboardInterrupt:
# Ctrl+C should fall through to the normal teardown path below.
pass
elapsed = time.time() - start_time
fps = video_frame_count / elapsed if elapsed > 0 else 0
print(f"\nVideo: {video_frame_count} frames in {elapsed:.1f}s ({fps:.1f} FPS)")
print(f"Decoded video frames: {decoded_video_frame_count}")
print(f"Audio: {audio_frame_count} frames received, {audio_valid_frame_count} valid, "
f"{audio_invalid_frame_count} invalid")
rtsp_client.teardown(ec)
browser.cancel()
zeroconf.close()
if audio_player is not None:
audio_player.stop()
print(f"Audio playback underflows: {audio_player.underflow_count}, "
f"dropped frames: {audio_player.dropped_frames}")
failures = []
if video_frame_count < args.min_video_frames:
failures.append(
f"expected at least {args.min_video_frames} video frames, got {video_frame_count}"
)
if args.require_decoded_video and decoded_video_frame_count == 0:
failures.append("expected at least one decoded JPEG video frame")
if args.expect_audio and audio_valid_frame_count < args.min_audio_frames:
failures.append(
f"expected at least {args.min_audio_frames} valid audio frames, "
f"got {audio_valid_frame_count}"
)
if audio_invalid_frame_count > 0:
failures.append(f"received {audio_invalid_frame_count} invalid audio frames")
if failures:
for failure in failures:
print(f"[FAIL] {failure}")
return 1
print("[PASS] Multitrack RTSP client validation succeeded")
return 0
if __name__ == "__main__":
sys.exit(main())