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| from typing import List | |
| import av | |
| import asyncio | |
| from collections import deque | |
| import threading | |
| import numpy as np | |
| import ray | |
| from webrtc_av_queue_actor import WebRtcAVQueueActor | |
| import pydub | |
| import torch | |
| class StreamlitAVQueue: | |
| def __init__(self, audio_bit_rate=16000): | |
| self._output_channels = 2 | |
| self._audio_bit_rate = audio_bit_rate | |
| self.queue_actor = WebRtcAVQueueActor.options( | |
| name="WebRtcAVQueueActor", | |
| get_if_exists=True, | |
| ).remote() | |
| async def queued_video_frames_callback( | |
| self, | |
| frames: List[av.AudioFrame], | |
| ) -> av.AudioFrame: | |
| try: | |
| for frame in frames: | |
| shared_tensor = torch.from_numpy(frame.to_ndarray()) | |
| shared_tensor_ref = ray.put(shared_tensor) | |
| await self.queue_actor.enqueue_in_video_frame.remote(shared_tensor_ref) | |
| except Exception as e: | |
| print (e) | |
| return frames | |
| async def queued_audio_frames_callback( | |
| self, | |
| frames: List[av.AudioFrame], | |
| ) -> av.AudioFrame: | |
| try: | |
| sound_chunk = pydub.AudioSegment.empty() | |
| if len(frames) > 0: | |
| for frame in frames: | |
| sound = pydub.AudioSegment( | |
| data=frame.to_ndarray().tobytes(), | |
| sample_width=frame.format.bytes, | |
| frame_rate=frame.sample_rate, | |
| channels=len(frame.layout.channels), | |
| ) | |
| sound = sound.set_channels(1) | |
| sound = sound.set_frame_rate(self._audio_bit_rate) | |
| sound_chunk += sound | |
| shared_buffer = np.array(sound_chunk.get_array_of_samples()) | |
| shared_buffer_ref = ray.put(shared_buffer) | |
| await self.queue_actor.enqueue_in_audio_frame.remote(shared_buffer_ref) | |
| except Exception as e: | |
| print (e) | |
| # return empty frames to avoid echo | |
| new_frames = [] | |
| try: | |
| for frame in frames: | |
| required_samples = frame.samples | |
| # print (f"frame: {frame.format.name}, {frame.layout.name}, {frame.sample_rate}, {frame.samples}") | |
| assert frame.format.bytes == 2 | |
| assert frame.format.name == 's16' | |
| frame_as_bytes = await self.queue_actor.get_out_audio_frame.remote() | |
| if frame_as_bytes: | |
| # print(f"frame_as_bytes: {len(frame_as_bytes)}") | |
| assert len(frame_as_bytes) == frame.samples * frame.format.bytes | |
| samples = np.frombuffer(frame_as_bytes, dtype=np.int16) | |
| else: | |
| samples = np.zeros((required_samples * 2 * 1), dtype=np.int16) | |
| if self._output_channels == 2: | |
| samples = np.vstack((samples, samples)).reshape((-1,), order='F') | |
| samples = samples.reshape(1, -1) | |
| layout = 'stereo' if self._output_channels == 2 else 'mono' | |
| new_frame = av.AudioFrame.from_ndarray(samples, format='s16', layout=layout) | |
| new_frame.sample_rate = frame.sample_rate | |
| new_frames.append(new_frame) | |
| except Exception as e: | |
| print (e) | |
| return new_frames | |
| async def get_in_audio_frames_async(self) -> List[av.AudioFrame]: | |
| shared_buffers = await self.queue_actor.get_in_audio_frames.remote() | |
| return shared_buffers | |
| async def get_video_frames_async(self) -> List[av.AudioFrame]: | |
| shared_tensors = await self.queue_actor.get_in_video_frames.remote() | |
| return shared_tensors | |
| def get_out_audio_queue(self): | |
| return self.queue_actor.get_out_audio_queue.remote() | |
| # def get_out_audio_frame(self): | |
| # return self.queue_actor.get_out_audio_frame.remote() |