add diarization (beta). Disabled by default
Browse files
src/diarization/diarization_online.py
ADDED
@@ -0,0 +1,86 @@
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+
from diart import SpeakerDiarization
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from diart.inference import StreamingInference
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from diart.sources import AudioSource
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from rx.subject import Subject
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import threading
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import numpy as np
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import asyncio
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class WebSocketAudioSource(AudioSource):
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"""
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Simple custom AudioSource that blocks in read()
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until close() is called.
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push_audio() is used to inject new PCM chunks.
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"""
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def __init__(self, uri: str = "websocket", sample_rate: int = 16000):
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super().__init__(uri, sample_rate)
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self._close_event = threading.Event()
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self._closed = False
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def read(self):
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self._close_event.wait()
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def close(self):
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if not self._closed:
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self._closed = True
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self.stream.on_completed()
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self._close_event.set()
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def push_audio(self, chunk: np.ndarray):
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chunk = np.expand_dims(chunk, axis=0)
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if not self._closed:
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self.stream.on_next(chunk)
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def create_pipeline(SAMPLE_RATE):
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diar_pipeline = SpeakerDiarization()
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ws_source = WebSocketAudioSource(uri="websocket_source", sample_rate=SAMPLE_RATE)
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inference = StreamingInference(
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pipeline=diar_pipeline,
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source=ws_source,
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do_plot=False,
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show_progress=False,
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)
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return inference, ws_source
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def init_diart(SAMPLE_RATE):
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inference, ws_source = create_pipeline(SAMPLE_RATE)
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def diar_hook(result):
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"""
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Hook called each time Diart processes a chunk.
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result is (annotation, audio).
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We store the label of the last segment in 'current_speaker'.
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"""
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global l_speakers
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l_speakers = []
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annotation, audio = result
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for speaker in annotation._labels:
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segment = annotation._labels[speaker].__str__()
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asyncio.create_task(
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l_speakers_queue.put({"speaker": speaker, "segment": segment})
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)
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l_speakers_queue = asyncio.Queue()
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inference.attach_hooks(diar_hook)
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# Launch Diart in a background thread
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loop = asyncio.get_event_loop()
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diar_future = loop.run_in_executor(None, inference)
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return inference, l_speakers_queue, ws_source
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class DiartDiarization():
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def __init__(self, SAMPLE_RATE):
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self.inference, self.l_speakers_queue, self.ws_source = init_diart(SAMPLE_RATE)
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async def get_speakers(self, pcm_array):
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self.ws_source.push_audio(pcm_array)
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speakers = []
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while not self.l_speakers_queue.empty():
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speakers.append(await self.l_speakers_queue.get())
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return speakers
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def close(self):
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self.ws_source.close()
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whisper_fastapi_online_server.py
CHANGED
@@ -10,7 +10,6 @@ from fastapi.responses import HTMLResponse
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from fastapi.middleware.cors import CORSMiddleware
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from whisper_online import backend_factory, online_factory, add_shared_args
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-
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app = FastAPI()
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app.add_middleware(
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CORSMiddleware,
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@@ -37,11 +36,24 @@ parser.add_argument(
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dest="warmup_file",
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help="The path to a speech audio wav file to warm up Whisper so that the very first chunk processing is fast. It can be e.g. https://github.com/ggerganov/whisper.cpp/raw/master/samples/jfk.wav .",
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)
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add_shared_args(parser)
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args = parser.parse_args()
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asr, tokenizer = backend_factory(args)
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# Load demo HTML for the root endpoint
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with open("src/web/live_transcription.html", "r", encoding="utf-8") as f:
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html = f.read()
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@@ -89,6 +101,9 @@ async def websocket_endpoint(websocket: WebSocket):
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online = online_factory(args, asr, tokenizer)
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print("Online loaded.")
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# Continuously read decoded PCM from ffmpeg stdout in a background task
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async def ffmpeg_stdout_reader():
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nonlocal pcm_buffer
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@@ -136,9 +151,13 @@ async def websocket_endpoint(websocket: WebSocket):
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buffer in full_transcription
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): # With VAC, the buffer is not updated until the next chunk is processed
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buffer = ""
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-
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-
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except Exception as e:
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print(f"Exception in ffmpeg_stdout_reader: {e}")
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break
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@@ -174,6 +193,11 @@ async def websocket_endpoint(websocket: WebSocket):
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ffmpeg_process.wait()
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del online
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from fastapi.middleware.cors import CORSMiddleware
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from whisper_online import backend_factory, online_factory, add_shared_args
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app = FastAPI()
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app.add_middleware(
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CORSMiddleware,
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dest="warmup_file",
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help="The path to a speech audio wav file to warm up Whisper so that the very first chunk processing is fast. It can be e.g. https://github.com/ggerganov/whisper.cpp/raw/master/samples/jfk.wav .",
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)
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parser.add_argument(
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"--diarization",
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type=bool,
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default=False,
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help="Whether to enable speaker diarization.",
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)
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add_shared_args(parser)
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args = parser.parse_args()
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asr, tokenizer = backend_factory(args)
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if args.diarization:
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from src.diarization.diarization_online import DiartDiarization
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# Load demo HTML for the root endpoint
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with open("src/web/live_transcription.html", "r", encoding="utf-8") as f:
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html = f.read()
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online = online_factory(args, asr, tokenizer)
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print("Online loaded.")
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if args.diarization:
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diarization = DiartDiarization(SAMPLE_RATE)
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# Continuously read decoded PCM from ffmpeg stdout in a background task
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async def ffmpeg_stdout_reader():
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nonlocal pcm_buffer
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buffer in full_transcription
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): # With VAC, the buffer is not updated until the next chunk is processed
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buffer = ""
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response = {"transcription": transcription, "buffer": buffer}
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if args.diarization:
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speakers = await diarization.get_speakers(pcm_array)
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response["speakers"] = speakers
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await websocket.send_json(response)
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except Exception as e:
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print(f"Exception in ffmpeg_stdout_reader: {e}")
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break
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ffmpeg_process.wait()
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del online
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if args.diarization:
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# Stop Diart
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diarization.close()
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