mirror of
https://github.com/QuentinFuxa/WhisperLiveKit.git
synced 2026-03-07 22:33:36 +00:00
Refactor AudioProcessor methods for improved async handling and WebSocket integration
This commit is contained in:
31
audio.py
31
audio.py
@@ -54,7 +54,7 @@ class AudioProcessor:
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/ 32768.0)
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return pcm_array
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async def start_ffmpeg_decoder(self):
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def start_ffmpeg_decoder(self):
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"""
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Start an FFmpeg process in async streaming mode that reads WebM from stdin
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and outputs raw s16le PCM on stdout. Returns the process object.
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@@ -79,7 +79,7 @@ class AudioProcessor:
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await asyncio.get_event_loop().run_in_executor(None, self.ffmpeg_process.wait)
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except Exception as e:
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logger.warning(f"Error killing FFmpeg process: {e}")
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self.ffmpeg_process = await self.start_ffmpeg_decoder()
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self.ffmpeg_process = self.start_ffmpeg_decoder()
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self.pcm_buffer = bytearray()
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async def ffmpeg_stdout_reader(self):
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@@ -198,10 +198,9 @@ class AudioProcessor:
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finally:
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self.diarization_queue.task_done()
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async def results_formatter(self, websocket):
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async def results_formatter(self):
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while True:
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try:
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# Get the current state
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state = await self.shared_state.get_current_state()
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tokens = state["tokens"]
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buffer_transcription = state["buffer_transcription"]
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@@ -217,7 +216,6 @@ class AudioProcessor:
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sleep(0.5)
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state = await self.shared_state.get_current_state()
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tokens = state["tokens"]
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# Process tokens to create response
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previous_speaker = -1
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lines = []
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last_end_diarized = 0
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@@ -273,22 +271,21 @@ class AudioProcessor:
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"beg": format_time(0),
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"end": format_time(tokens[-1].end) if tokens else format_time(0),
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"diff": 0
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}],
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}],
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"buffer_transcription": buffer_transcription,
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"buffer_diarization": buffer_diarization,
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"remaining_time_transcription": remaining_time_transcription,
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"remaining_time_diarization": remaining_time_diarization
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}
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response_content = ' '.join([str(line['speaker']) + ' ' + line["text"] for line in lines]) + ' | ' + buffer_transcription + ' | ' + buffer_diarization
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if response_content != self.shared_state.last_response_content:
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if lines or buffer_transcription or buffer_diarization:
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await websocket.send_json(response)
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yield response
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self.shared_state.last_response_content = response_content
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# Add a small delay to avoid overwhelming the client
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#small delay to avoid overwhelming the client
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await asyncio.sleep(0.1)
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except Exception as e:
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@@ -296,18 +293,22 @@ class AudioProcessor:
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logger.warning(f"Traceback: {traceback.format_exc()}")
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await asyncio.sleep(0.5) # Back off on error
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async def create_tasks(self, websocket, diarization):
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async def create_tasks(self, diarization=None):
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if diarization:
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self.diarization = diarization
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tasks = []
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if self.args.transcription and self.online:
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tasks.append(asyncio.create_task(self.transcription_processor()))
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if self.args.diarization and diarization:
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tasks.append(asyncio.create_task(self.diarization_processor(diarization)))
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formatter_task = asyncio.create_task(self.results_formatter(websocket))
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tasks.append(formatter_task)
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if self.args.diarization and self.diarization:
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tasks.append(asyncio.create_task(self.diarization_processor(self.diarization)))
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stdout_reader_task = asyncio.create_task(self.ffmpeg_stdout_reader())
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tasks.append(stdout_reader_task)
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self.tasks = tasks
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self.diarization = diarization
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return self.results_formatter()
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async def cleanup(self):
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for task in self.tasks:
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@@ -5,6 +5,7 @@ from fastapi.responses import HTMLResponse
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from fastapi.middleware.cors import CORSMiddleware
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from whisper_streaming_custom.whisper_online import backend_factory, warmup_asr
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import asyncio
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import logging
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from parse_args import parse_args
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from audio import AudioProcessor
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@@ -51,6 +52,16 @@ with open("web/live_transcription.html", "r", encoding="utf-8") as f:
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async def get():
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return HTMLResponse(html)
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async def handle_websocket_results(websocket, results_generator):
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"""Consumes results from the audio processor and sends them via WebSocket."""
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try:
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async for response in results_generator:
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await websocket.send_json(response)
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except Exception as e:
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logger.warning(f"Error in WebSocket results handler: {e}")
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@app.websocket("/asr")
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async def websocket_endpoint(websocket: WebSocket):
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audio_processor = AudioProcessor(args, asr, tokenizer)
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@@ -58,14 +69,17 @@ async def websocket_endpoint(websocket: WebSocket):
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await websocket.accept()
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logger.info("WebSocket connection opened.")
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await audio_processor.create_tasks(websocket, diarization)
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results_generator = await audio_processor.create_tasks(diarization)
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websocket_task = asyncio.create_task(handle_websocket_results(websocket, results_generator))
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try:
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while True:
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message = await websocket.receive_bytes()
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audio_processor.process_audio(message)
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await audio_processor.process_audio(message)
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except WebSocketDisconnect:
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logger.warning("WebSocket disconnected.")
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finally:
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websocket_task.cancel()
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audio_processor.cleanup()
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logger.info("WebSocket endpoint cleaned up.")
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