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fix webrtc audio problem

This commit is contained in:
lipku 2024-04-27 00:07:37 +08:00
parent f2d81f88d3
commit 027e15201a
2 changed files with 19 additions and 12 deletions
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13
nerfreal.py
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@ -167,14 +167,21 @@ class NeRFReal:
#print(f'[INFO] get_audio_out shape ',frame.shape) #print(f'[INFO] get_audio_out shape ',frame.shape)
self.streamer.stream_frame_audio(frame) self.streamer.stream_frame_audio(frame)
else: else:
frame1 = self.asr.get_audio_out() for _ in range(2):
frame2 = self.asr.get_audio_out() frame = self.asr.get_audio_out()
frame = np.concatenate((frame1,frame2))
frame = (frame * 32767).astype(np.int16) frame = (frame * 32767).astype(np.int16)
new_frame = AudioFrame(format='s16', layout='mono', samples=frame.shape[0]) new_frame = AudioFrame(format='s16', layout='mono', samples=frame.shape[0])
new_frame.planes[0].update(frame.tobytes()) new_frame.planes[0].update(frame.tobytes())
new_frame.sample_rate=16000 new_frame.sample_rate=16000
asyncio.run_coroutine_threadsafe(audio_track._queue.put(new_frame), loop) asyncio.run_coroutine_threadsafe(audio_track._queue.put(new_frame), loop)
# frame1 = self.asr.get_audio_out()
# frame2 = self.asr.get_audio_out()
# frame = np.concatenate((frame1,frame2))
# frame = (frame * 32767).astype(np.int16)
# new_frame = AudioFrame(format='s16', layout='mono', samples=frame.shape[0])
# new_frame.planes[0].update(frame.tobytes())
# new_frame.sample_rate=16000
# asyncio.run_coroutine_threadsafe(audio_track._queue.put(new_frame), loop)
# frame = (frame * 32767).astype(np.int16).tobytes() # frame = (frame * 32767).astype(np.int16).tobytes()
# self.fifo_audio.write(frame) # self.fifo_audio.write(frame)
else: else:

8
webrtc.py
View File

@ -46,8 +46,8 @@ class PlayerStreamTrack(MediaStreamTrack):
if self.kind == 'video': if self.kind == 'video':
if hasattr(self, "_timestamp"): if hasattr(self, "_timestamp"):
self._timestamp = (time.time()-self._start) * VIDEO_CLOCK_RATE # self._timestamp = (time.time()-self._start) * VIDEO_CLOCK_RATE
# self._timestamp += int(VIDEO_PTIME * VIDEO_CLOCK_RATE) self._timestamp += int(VIDEO_PTIME * VIDEO_CLOCK_RATE)
# wait = self._start + (self._timestamp / VIDEO_CLOCK_RATE) - time.time() # wait = self._start + (self._timestamp / VIDEO_CLOCK_RATE) - time.time()
# if wait>0: # if wait>0:
# await asyncio.sleep(wait) # await asyncio.sleep(wait)
@ -57,8 +57,8 @@ class PlayerStreamTrack(MediaStreamTrack):
return self._timestamp, VIDEO_TIME_BASE return self._timestamp, VIDEO_TIME_BASE
else: #audio else: #audio
if hasattr(self, "_timestamp"): if hasattr(self, "_timestamp"):
self._timestamp = (time.time()-self._start) * SAMPLE_RATE # self._timestamp = (time.time()-self._start) * SAMPLE_RATE
# self._timestamp += int(AUDIO_PTIME * SAMPLE_RATE) self._timestamp += int(AUDIO_PTIME * SAMPLE_RATE)
# wait = self._start + (self._timestamp / SAMPLE_RATE) - time.time() # wait = self._start + (self._timestamp / SAMPLE_RATE) - time.time()
# if wait>0: # if wait>0:
# await asyncio.sleep(wait) # await asyncio.sleep(wait)