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fix webrtc audio

This commit is contained in:
lipku 2024-04-20 18:40:34 +08:00
parent a3a86bf299
commit 6d4952c1bf
1 changed files with 13 additions and 10 deletions
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23
nerfreal.py
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@ -161,17 +161,20 @@ class NeRFReal:
new_frame = VideoFrame.from_ndarray(image_fullbody, format="rgb24")
asyncio.run_coroutine_threadsafe(video_track._queue.put(new_frame), loop)
#self.pipe.stdin.write(image.tostring())
for _ in range(2):
frame = self.asr.get_audio_out()
#print(f'[INFO] get_audio_out shape ',frame.shape)
if self.opt.transport=='rtmp':
if self.opt.transport=='rtmp':
for _ in range(2):
frame = self.asr.get_audio_out()
#print(f'[INFO] get_audio_out shape ',frame.shape)
self.streamer.stream_frame_audio(frame)
else:
frame = (frame * 32767).astype(np.int16)
new_frame = AudioFrame(format='s16', layout='mono', samples=320)
new_frame.planes[0].update(frame.tobytes())
new_frame.sample_rate=16000
asyncio.run_coroutine_threadsafe(audio_track._queue.put(new_frame), loop)
else:
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()
# self.fifo_audio.write(frame)
else: