Update app.py
Browse files
app.py
CHANGED
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@@ -1,427 +1,427 @@
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import os
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import io
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import requests
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import argparse
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import asyncio
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import numpy as np
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import ffmpeg
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from time import time
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from fastapi import FastAPI, WebSocket, WebSocketDisconnect
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from fastapi.responses import HTMLResponse
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from fastapi.middleware.cors import CORSMiddleware
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from src.whisper_streaming.whisper_online import backend_factory, online_factory, add_shared_args
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import logging
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import logging.config
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from transformers import pipeline
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from huggingface_hub import login
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HUGGING_FACE_TOKEN = os.environ['HUGGING_FACE_TOKEN']
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login(HUGGING_FACE_TOKEN)
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# os.environ['HF_HOME'] = './.cache'
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MODEL_NAME = 'Helsinki-NLP/opus-tatoeba-en-ja'
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TRANSLATOR = pipeline('translation', model=MODEL_NAME, device='cuda')
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TRANSLATOR('Warming up!')
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def translator_wrapper(source_text, translation_target_lang, mode):
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if mode == 'deepl':
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params = {
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'auth_key' : os.environ['DEEPL_API_KEY'],
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'text' : source_text,
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'source_lang' : 'EN', # 翻訳対象の言語
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"target_lang": 'JA', # 翻訳後の言語
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}
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# リクエストを投げる
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try:
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request = requests.post("https://api-free.deepl.com/v2/translate", data=params, timeout=5) # URIは有償版, 無償版で異なるため要注意
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result = request.json()['translations'][0]['text']
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except requests.exceptions.Timeout:
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result = "(timed out)"
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return result
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elif mode == 'marianmt':
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return TRANSLATOR(source_text)[0]['translation_text']
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elif mode == 'google':
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import requests
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# https://www.eyoucms.com/news/ziliao/other/29445.html
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language_type = ""
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url = "https://translation.googleapis.com/language/translate/v2"
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data = {
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'key':"AIzaSyCX0-Wdxl_rgvcZzklNjnqJ1W9YiKjcHUs", # 認証の設定:APIキー
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'source': language_type,
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'target': translation_target_lang,
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'q': source_text,
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'format': "text"
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}
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#headers = {'X-HTTP-Method-Override': 'GET'}
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#response = requests.post(url, data=data, headers=headers)
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response = requests.post(url, data)
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# print(response.json())
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print(response)
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res = response.json()
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print(res["data"]["translations"][0]["translatedText"])
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result = res["data"]["translations"][0]["translatedText"]
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print(result)
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return result
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def setup_logging():
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logging_config = {
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'version': 1,
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'disable_existing_loggers': False,
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'formatters': {
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'standard': {
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'format': '%(asctime)s %(levelname)s [%(name)s]: %(message)s',
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},
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},
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'handlers': {
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'console': {
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'level': 'INFO',
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'class': 'logging.StreamHandler',
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'formatter': 'standard',
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},
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},
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'root': {
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'handlers': ['console'],
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'level': 'DEBUG',
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},
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'loggers': {
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'uvicorn': {
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'handlers': ['console'],
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'level': 'INFO',
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'propagate': False,
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},
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'uvicorn.error': {
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'level': 'INFO',
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},
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'uvicorn.access': {
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'level': 'INFO',
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},
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'src.whisper_streaming.online_asr': { # Add your specific module here
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'handlers': ['console'],
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'level': 'DEBUG',
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'propagate': False,
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},
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'src.whisper_streaming.whisper_streaming': { # Add your specific module here
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'handlers': ['console'],
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'level': 'DEBUG',
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'propagate': False,
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},
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},
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}
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logging.config.dictConfig(logging_config)
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setup_logging()
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logger = logging.getLogger(__name__)
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app = FastAPI()
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app.add_middleware(
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CORSMiddleware,
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allow_origins=["*"],
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allow_credentials=True,
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allow_methods=["*"],
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allow_headers=["*"],
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)
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parser = argparse.ArgumentParser(description="Whisper FastAPI Online Server")
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parser.add_argument(
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"--host",
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type=str,
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default="localhost",
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help="The host address to bind the server to.",
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)
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parser.add_argument(
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"--port", type=int, default=8000, help="The port number to bind the server to."
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)
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parser.add_argument(
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"--warmup-file",
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type=str,
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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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parser.add_argument(
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"--generate-audio",
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type=bool,
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default=False,
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help="Whether to generate translation audio.",
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)
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add_shared_args(parser)
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args = parser.parse_args()
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# args.model = 'medium'
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if args.lan == 'ja':
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translation_target_lang = 'en'
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elif args.lan == 'en':
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translation_target_lang = 'ja'
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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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@app.get("/")
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async def get():
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return HTMLResponse(html)
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SAMPLE_RATE = 16000
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CHANNELS = 1
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SAMPLES_PER_SEC = int(SAMPLE_RATE * args.min_chunk_size)
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BYTES_PER_SAMPLE = 2 # s16le = 2 bytes per sample
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BYTES_PER_SEC = SAMPLES_PER_SEC * BYTES_PER_SAMPLE
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print('SAMPLE_RATE', SAMPLE_RATE)
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print('CHANNELS', CHANNELS)
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print('SAMPLES_PER_SEC', SAMPLES_PER_SEC)
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print('BYTES_PER_SAMPLE', BYTES_PER_SAMPLE)
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print('BYTES_PER_SEC', BYTES_PER_SEC)
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def generate_audio(japanese_text, speed=1.0):
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api_url = "https://j6im8slpwcevr7g0.us-east-1.aws.endpoints.huggingface.cloud"
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headers = {
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"Accept" : "application/json",
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"Authorization": f"Bearer {HUGGING_FACE_TOKEN}",
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"Content-Type": "application/json"
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}
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payload = {
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"inputs": japanese_text,
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"speed": speed,
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}
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response = requests.post(api_url, headers=headers, json=payload).json()
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if 'error' in response:
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print(response)
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return ''
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return response
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async def start_ffmpeg_decoder():
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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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"""
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process = (
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ffmpeg
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.input("pipe:0", format="webm")
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.output(
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"pipe:1",
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format="s16le",
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acodec="pcm_s16le",
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ac=CHANNELS,
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ar=str(SAMPLE_RATE),
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# fflags='nobuffer',
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)
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.global_args('-loglevel', 'quiet')
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.run_async(pipe_stdin=True, pipe_stdout=True, pipe_stderr=False, quiet=True)
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)
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return process
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import queue
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import threading
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@app.websocket("/asr")
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async def websocket_endpoint(websocket: WebSocket):
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await websocket.accept()
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print("WebSocket connection opened.")
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ffmpeg_process = await start_ffmpeg_decoder()
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pcm_buffer = bytearray()
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print("Loading online.")
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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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loop = asyncio.get_event_loop()
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full_transcription = ""
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beg = time()
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chunk_history = [] # Will store dicts: {beg, end, text, speaker}
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buffers = [{'speaker': '0', 'text': '', 'translation': None, 'audio_url': None}]
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buffer_line = ''
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# Create a queue to hold the chunks
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chunk_queue = queue.Queue()
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# Function to read from ffmpeg stdout in a separate thread
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def read_ffmpeg_stdout():
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while True:
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try:
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chunk = ffmpeg_process.stdout.read(BYTES_PER_SEC)
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if not chunk:
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break
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chunk_queue.put(chunk)
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except Exception as e:
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print(f"Exception in read_ffmpeg_stdout: {e}")
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break
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# Start the thread
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threading.Thread(target=read_ffmpeg_stdout, daemon=True).start()
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while True:
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try:
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# Get the chunk from the queue
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chunk = await loop.run_in_executor(None, chunk_queue.get)
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if not chunk:
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print("FFmpeg stdout closed.")
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break
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pcm_buffer.extend(chunk)
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print('len(pcm_buffer): ', len(pcm_buffer))
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print('BYTES_PER_SEC: ', BYTES_PER_SEC)
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if len(pcm_buffer) >= BYTES_PER_SEC:
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# Convert int16 -> float32
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pcm_array = (np.frombuffer(pcm_buffer, dtype=np.int16).astype(np.float32) / 32768.0)
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pcm_buffer = bytearray() # Initialize the PCM buffer
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online.insert_audio_chunk(pcm_array)
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beg_trans, end_trans, trans = online.process_iter()
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if trans:
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chunk_history.append({
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"beg": beg_trans,
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"end": end_trans,
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"text": trans,
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"speaker": "0"
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})
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full_transcription += trans
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# ----------------
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# Process buffer
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# ----------------
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if args.vac:
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# We need to access the underlying online object to get the buffer
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buffer_text = online.online.concatenate_tsw(online.online.transcript_buffer.buffer)[2]
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else:
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buffer_text = online.concatenate_tsw(online.transcript_buffer.buffer)[2]
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if buffer_text in full_transcription: # With VAC, the buffer is not updated until the next chunk is processed
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buffer_text = ""
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buffer_line += buffer_text
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punctuations = (',', '.', '?', '!', 'and', 'or', 'but', 'however')
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if not any(punctuation in buffer_line for punctuation in punctuations):
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continue
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last_punctuation_index = max((buffer_line.rfind(p) + len(p) + 1) for p in punctuations if p in buffer_line)
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extracted_text = buffer_line[:last_punctuation_index]
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buffer_line = buffer_line[last_punctuation_index:]
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buffer = {'speaker': '0', 'text': extracted_text, 'translation': None}
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translation = translator_wrapper(buffer['text'], translation_target_lang, mode='google')
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buffer['translation'] = translation
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buffer['text'] += ('|' + translation)
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buffer['audio_url'] = generate_audio(translation, speed=
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buffers.append(buffer)
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# ----------------
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# Process lines
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# ----------------
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'''
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print('Process lines')
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lines = [{"speaker": "0", "text": ""}]
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if args.diarization:
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await diarization.diarize(pcm_array)
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# diarization.assign_speakers_to_chunks(chunk_history)
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chunk_history = diarization.assign_speakers_to_chunks(chunk_history)
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for ch in chunk_history:
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if args.diarization and ch["speaker"] and ch["speaker"][-1] != lines[-1]["speaker"]:
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lines.append({"speaker": ch["speaker"], "text": ch['text']})
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else:
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lines.append({"speaker": ch["speaker"], "text": ch['text']})
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for i, line in enumerate(lines):
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if line['text'].strip() == '':
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continue
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# translation = translator(line['text'])[0]['translation_text']
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# translation = translation.replace(' ', '')
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# lines[i]['text'] = line['text'] + translation
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lines[i]['text'] = line['text']
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'''
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print('Before making response')
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response = {'line': buffer, 'buffer': ''}
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print(response)
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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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print("Exiting ffmpeg_stdout_reader...")
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stdout_reader_task = asyncio.create_task(ffmpeg_stdout_reader())
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try:
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while True:
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# Receive incoming WebM audio chunks from the client
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| 391 |
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message = await websocket.receive_bytes()
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# Pass them to ffmpeg via stdin
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ffmpeg_process.stdin.write(message)
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ffmpeg_process.stdin.flush()
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except WebSocketDisconnect:
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print("WebSocket connection closed.")
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| 398 |
-
except Exception as e:
|
| 399 |
-
print(f"Error in websocket loop: {e}")
|
| 400 |
-
finally:
|
| 401 |
-
# Clean up ffmpeg and the reader task
|
| 402 |
-
try:
|
| 403 |
-
ffmpeg_process.stdin.close()
|
| 404 |
-
except:
|
| 405 |
-
pass
|
| 406 |
-
stdout_reader_task.cancel()
|
| 407 |
-
|
| 408 |
-
try:
|
| 409 |
-
ffmpeg_process.stdout.close()
|
| 410 |
-
except:
|
| 411 |
-
pass
|
| 412 |
-
|
| 413 |
-
ffmpeg_process.wait()
|
| 414 |
-
del online
|
| 415 |
-
|
| 416 |
-
if args.diarization:
|
| 417 |
-
# Stop Diart
|
| 418 |
-
diarization.close()
|
| 419 |
-
|
| 420 |
-
|
| 421 |
-
if __name__ == "__main__":
|
| 422 |
-
import uvicorn
|
| 423 |
-
|
| 424 |
-
uvicorn.run(
|
| 425 |
-
"app:app", host=args.host, port=args.port, reload=True,
|
| 426 |
-
log_level="info"
|
| 427 |
-
)
|
|
|
|
| 1 |
+
import os
|
| 2 |
+
import io
|
| 3 |
+
import requests
|
| 4 |
+
import argparse
|
| 5 |
+
import asyncio
|
| 6 |
+
import numpy as np
|
| 7 |
+
import ffmpeg
|
| 8 |
+
from time import time
|
| 9 |
+
|
| 10 |
+
from fastapi import FastAPI, WebSocket, WebSocketDisconnect
|
| 11 |
+
from fastapi.responses import HTMLResponse
|
| 12 |
+
from fastapi.middleware.cors import CORSMiddleware
|
| 13 |
+
|
| 14 |
+
from src.whisper_streaming.whisper_online import backend_factory, online_factory, add_shared_args
|
| 15 |
+
|
| 16 |
+
import logging
|
| 17 |
+
import logging.config
|
| 18 |
+
from transformers import pipeline
|
| 19 |
+
from huggingface_hub import login
|
| 20 |
+
|
| 21 |
+
HUGGING_FACE_TOKEN = os.environ['HUGGING_FACE_TOKEN']
|
| 22 |
+
login(HUGGING_FACE_TOKEN)
|
| 23 |
+
|
| 24 |
+
# os.environ['HF_HOME'] = './.cache'
|
| 25 |
+
|
| 26 |
+
MODEL_NAME = 'Helsinki-NLP/opus-tatoeba-en-ja'
|
| 27 |
+
TRANSLATOR = pipeline('translation', model=MODEL_NAME, device='cuda')
|
| 28 |
+
TRANSLATOR('Warming up!')
|
| 29 |
+
|
| 30 |
+
def translator_wrapper(source_text, translation_target_lang, mode):
|
| 31 |
+
if mode == 'deepl':
|
| 32 |
+
params = {
|
| 33 |
+
'auth_key' : os.environ['DEEPL_API_KEY'],
|
| 34 |
+
'text' : source_text,
|
| 35 |
+
'source_lang' : 'EN', # 翻訳対象の言語
|
| 36 |
+
"target_lang": 'JA', # 翻訳後の言語
|
| 37 |
+
}
|
| 38 |
+
|
| 39 |
+
# リクエストを投げる
|
| 40 |
+
try:
|
| 41 |
+
request = requests.post("https://api-free.deepl.com/v2/translate", data=params, timeout=5) # URIは有償版, 無償版で異なるため要注意
|
| 42 |
+
result = request.json()['translations'][0]['text']
|
| 43 |
+
except requests.exceptions.Timeout:
|
| 44 |
+
result = "(timed out)"
|
| 45 |
+
return result
|
| 46 |
+
|
| 47 |
+
elif mode == 'marianmt':
|
| 48 |
+
return TRANSLATOR(source_text)[0]['translation_text']
|
| 49 |
+
|
| 50 |
+
elif mode == 'google':
|
| 51 |
+
import requests
|
| 52 |
+
|
| 53 |
+
# https://www.eyoucms.com/news/ziliao/other/29445.html
|
| 54 |
+
language_type = ""
|
| 55 |
+
url = "https://translation.googleapis.com/language/translate/v2"
|
| 56 |
+
data = {
|
| 57 |
+
'key':"AIzaSyCX0-Wdxl_rgvcZzklNjnqJ1W9YiKjcHUs", # 認証の設定:APIキー
|
| 58 |
+
'source': language_type,
|
| 59 |
+
'target': translation_target_lang,
|
| 60 |
+
'q': source_text,
|
| 61 |
+
'format': "text"
|
| 62 |
+
}
|
| 63 |
+
#headers = {'X-HTTP-Method-Override': 'GET'}
|
| 64 |
+
#response = requests.post(url, data=data, headers=headers)
|
| 65 |
+
response = requests.post(url, data)
|
| 66 |
+
# print(response.json())
|
| 67 |
+
print(response)
|
| 68 |
+
res = response.json()
|
| 69 |
+
print(res["data"]["translations"][0]["translatedText"])
|
| 70 |
+
result = res["data"]["translations"][0]["translatedText"]
|
| 71 |
+
print(result)
|
| 72 |
+
return result
|
| 73 |
+
|
| 74 |
+
|
| 75 |
+
def setup_logging():
|
| 76 |
+
logging_config = {
|
| 77 |
+
'version': 1,
|
| 78 |
+
'disable_existing_loggers': False,
|
| 79 |
+
'formatters': {
|
| 80 |
+
'standard': {
|
| 81 |
+
'format': '%(asctime)s %(levelname)s [%(name)s]: %(message)s',
|
| 82 |
+
},
|
| 83 |
+
},
|
| 84 |
+
'handlers': {
|
| 85 |
+
'console': {
|
| 86 |
+
'level': 'INFO',
|
| 87 |
+
'class': 'logging.StreamHandler',
|
| 88 |
+
'formatter': 'standard',
|
| 89 |
+
},
|
| 90 |
+
},
|
| 91 |
+
'root': {
|
| 92 |
+
'handlers': ['console'],
|
| 93 |
+
'level': 'DEBUG',
|
| 94 |
+
},
|
| 95 |
+
'loggers': {
|
| 96 |
+
'uvicorn': {
|
| 97 |
+
'handlers': ['console'],
|
| 98 |
+
'level': 'INFO',
|
| 99 |
+
'propagate': False,
|
| 100 |
+
},
|
| 101 |
+
'uvicorn.error': {
|
| 102 |
+
'level': 'INFO',
|
| 103 |
+
},
|
| 104 |
+
'uvicorn.access': {
|
| 105 |
+
'level': 'INFO',
|
| 106 |
+
},
|
| 107 |
+
'src.whisper_streaming.online_asr': { # Add your specific module here
|
| 108 |
+
'handlers': ['console'],
|
| 109 |
+
'level': 'DEBUG',
|
| 110 |
+
'propagate': False,
|
| 111 |
+
},
|
| 112 |
+
'src.whisper_streaming.whisper_streaming': { # Add your specific module here
|
| 113 |
+
'handlers': ['console'],
|
| 114 |
+
'level': 'DEBUG',
|
| 115 |
+
'propagate': False,
|
| 116 |
+
},
|
| 117 |
+
},
|
| 118 |
+
}
|
| 119 |
+
|
| 120 |
+
logging.config.dictConfig(logging_config)
|
| 121 |
+
|
| 122 |
+
setup_logging()
|
| 123 |
+
logger = logging.getLogger(__name__)
|
| 124 |
+
|
| 125 |
+
app = FastAPI()
|
| 126 |
+
app.add_middleware(
|
| 127 |
+
CORSMiddleware,
|
| 128 |
+
allow_origins=["*"],
|
| 129 |
+
allow_credentials=True,
|
| 130 |
+
allow_methods=["*"],
|
| 131 |
+
allow_headers=["*"],
|
| 132 |
+
)
|
| 133 |
+
|
| 134 |
+
parser = argparse.ArgumentParser(description="Whisper FastAPI Online Server")
|
| 135 |
+
parser.add_argument(
|
| 136 |
+
"--host",
|
| 137 |
+
type=str,
|
| 138 |
+
default="localhost",
|
| 139 |
+
help="The host address to bind the server to.",
|
| 140 |
+
)
|
| 141 |
+
parser.add_argument(
|
| 142 |
+
"--port", type=int, default=8000, help="The port number to bind the server to."
|
| 143 |
+
)
|
| 144 |
+
parser.add_argument(
|
| 145 |
+
"--warmup-file",
|
| 146 |
+
type=str,
|
| 147 |
+
dest="warmup_file",
|
| 148 |
+
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 .",
|
| 149 |
+
)
|
| 150 |
+
parser.add_argument(
|
| 151 |
+
"--diarization",
|
| 152 |
+
type=bool,
|
| 153 |
+
default=False,
|
| 154 |
+
help="Whether to enable speaker diarization.",
|
| 155 |
+
)
|
| 156 |
+
parser.add_argument(
|
| 157 |
+
"--generate-audio",
|
| 158 |
+
type=bool,
|
| 159 |
+
default=False,
|
| 160 |
+
help="Whether to generate translation audio.",
|
| 161 |
+
)
|
| 162 |
+
|
| 163 |
+
|
| 164 |
+
add_shared_args(parser)
|
| 165 |
+
args = parser.parse_args()
|
| 166 |
+
# args.model = 'medium'
|
| 167 |
+
|
| 168 |
+
if args.lan == 'ja':
|
| 169 |
+
translation_target_lang = 'en'
|
| 170 |
+
elif args.lan == 'en':
|
| 171 |
+
translation_target_lang = 'ja'
|
| 172 |
+
|
| 173 |
+
asr, tokenizer = backend_factory(args)
|
| 174 |
+
|
| 175 |
+
if args.diarization:
|
| 176 |
+
from src.diarization.diarization_online import DiartDiarization
|
| 177 |
+
|
| 178 |
+
|
| 179 |
+
# Load demo HTML for the root endpoint
|
| 180 |
+
with open("src/web/live_transcription.html", "r", encoding="utf-8") as f:
|
| 181 |
+
html = f.read()
|
| 182 |
+
|
| 183 |
+
|
| 184 |
+
@app.get("/")
|
| 185 |
+
async def get():
|
| 186 |
+
return HTMLResponse(html)
|
| 187 |
+
|
| 188 |
+
|
| 189 |
+
SAMPLE_RATE = 16000
|
| 190 |
+
CHANNELS = 1
|
| 191 |
+
SAMPLES_PER_SEC = int(SAMPLE_RATE * args.min_chunk_size)
|
| 192 |
+
BYTES_PER_SAMPLE = 2 # s16le = 2 bytes per sample
|
| 193 |
+
BYTES_PER_SEC = SAMPLES_PER_SEC * BYTES_PER_SAMPLE
|
| 194 |
+
print('SAMPLE_RATE', SAMPLE_RATE)
|
| 195 |
+
print('CHANNELS', CHANNELS)
|
| 196 |
+
print('SAMPLES_PER_SEC', SAMPLES_PER_SEC)
|
| 197 |
+
print('BYTES_PER_SAMPLE', BYTES_PER_SAMPLE)
|
| 198 |
+
print('BYTES_PER_SEC', BYTES_PER_SEC)
|
| 199 |
+
|
| 200 |
+
|
| 201 |
+
def generate_audio(japanese_text, speed=1.0):
|
| 202 |
+
api_url = "https://j6im8slpwcevr7g0.us-east-1.aws.endpoints.huggingface.cloud"
|
| 203 |
+
headers = {
|
| 204 |
+
"Accept" : "application/json",
|
| 205 |
+
"Authorization": f"Bearer {HUGGING_FACE_TOKEN}",
|
| 206 |
+
"Content-Type": "application/json"
|
| 207 |
+
}
|
| 208 |
+
|
| 209 |
+
payload = {
|
| 210 |
+
"inputs": japanese_text,
|
| 211 |
+
"speed": speed,
|
| 212 |
+
}
|
| 213 |
+
|
| 214 |
+
response = requests.post(api_url, headers=headers, json=payload).json()
|
| 215 |
+
if 'error' in response:
|
| 216 |
+
print(response)
|
| 217 |
+
return ''
|
| 218 |
+
return response
|
| 219 |
+
|
| 220 |
+
|
| 221 |
+
async def start_ffmpeg_decoder():
|
| 222 |
+
"""
|
| 223 |
+
Start an FFmpeg process in async streaming mode that reads WebM from stdin
|
| 224 |
+
and outputs raw s16le PCM on stdout. Returns the process object.
|
| 225 |
+
"""
|
| 226 |
+
process = (
|
| 227 |
+
ffmpeg
|
| 228 |
+
.input("pipe:0", format="webm")
|
| 229 |
+
.output(
|
| 230 |
+
"pipe:1",
|
| 231 |
+
format="s16le",
|
| 232 |
+
acodec="pcm_s16le",
|
| 233 |
+
ac=CHANNELS,
|
| 234 |
+
ar=str(SAMPLE_RATE),
|
| 235 |
+
# fflags='nobuffer',
|
| 236 |
+
)
|
| 237 |
+
.global_args('-loglevel', 'quiet')
|
| 238 |
+
.run_async(pipe_stdin=True, pipe_stdout=True, pipe_stderr=False, quiet=True)
|
| 239 |
+
)
|
| 240 |
+
return process
|
| 241 |
+
|
| 242 |
+
import queue
|
| 243 |
+
import threading
|
| 244 |
+
|
| 245 |
+
@app.websocket("/asr")
|
| 246 |
+
async def websocket_endpoint(websocket: WebSocket):
|
| 247 |
+
await websocket.accept()
|
| 248 |
+
print("WebSocket connection opened.")
|
| 249 |
+
|
| 250 |
+
ffmpeg_process = await start_ffmpeg_decoder()
|
| 251 |
+
pcm_buffer = bytearray()
|
| 252 |
+
print("Loading online.")
|
| 253 |
+
online = online_factory(args, asr, tokenizer)
|
| 254 |
+
print("Online loaded.")
|
| 255 |
+
|
| 256 |
+
if args.diarization:
|
| 257 |
+
diarization = DiartDiarization(SAMPLE_RATE)
|
| 258 |
+
|
| 259 |
+
# Continuously read decoded PCM from ffmpeg stdout in a background task
|
| 260 |
+
async def ffmpeg_stdout_reader():
|
| 261 |
+
nonlocal pcm_buffer
|
| 262 |
+
loop = asyncio.get_event_loop()
|
| 263 |
+
full_transcription = ""
|
| 264 |
+
beg = time()
|
| 265 |
+
|
| 266 |
+
chunk_history = [] # Will store dicts: {beg, end, text, speaker}
|
| 267 |
+
|
| 268 |
+
buffers = [{'speaker': '0', 'text': '', 'translation': None, 'audio_url': None}]
|
| 269 |
+
buffer_line = ''
|
| 270 |
+
|
| 271 |
+
# Create a queue to hold the chunks
|
| 272 |
+
chunk_queue = queue.Queue()
|
| 273 |
+
|
| 274 |
+
# Function to read from ffmpeg stdout in a separate thread
|
| 275 |
+
def read_ffmpeg_stdout():
|
| 276 |
+
while True:
|
| 277 |
+
try:
|
| 278 |
+
chunk = ffmpeg_process.stdout.read(BYTES_PER_SEC)
|
| 279 |
+
if not chunk:
|
| 280 |
+
break
|
| 281 |
+
chunk_queue.put(chunk)
|
| 282 |
+
except Exception as e:
|
| 283 |
+
print(f"Exception in read_ffmpeg_stdout: {e}")
|
| 284 |
+
break
|
| 285 |
+
|
| 286 |
+
# Start the thread
|
| 287 |
+
threading.Thread(target=read_ffmpeg_stdout, daemon=True).start()
|
| 288 |
+
|
| 289 |
+
while True:
|
| 290 |
+
try:
|
| 291 |
+
# Get the chunk from the queue
|
| 292 |
+
chunk = await loop.run_in_executor(None, chunk_queue.get)
|
| 293 |
+
if not chunk:
|
| 294 |
+
print("FFmpeg stdout closed.")
|
| 295 |
+
break
|
| 296 |
+
|
| 297 |
+
pcm_buffer.extend(chunk)
|
| 298 |
+
print('len(pcm_buffer): ', len(pcm_buffer))
|
| 299 |
+
print('BYTES_PER_SEC: ', BYTES_PER_SEC)
|
| 300 |
+
|
| 301 |
+
if len(pcm_buffer) >= BYTES_PER_SEC:
|
| 302 |
+
# Convert int16 -> float32
|
| 303 |
+
pcm_array = (np.frombuffer(pcm_buffer, dtype=np.int16).astype(np.float32) / 32768.0)
|
| 304 |
+
pcm_buffer = bytearray() # Initialize the PCM buffer
|
| 305 |
+
online.insert_audio_chunk(pcm_array)
|
| 306 |
+
beg_trans, end_trans, trans = online.process_iter()
|
| 307 |
+
|
| 308 |
+
if trans:
|
| 309 |
+
chunk_history.append({
|
| 310 |
+
"beg": beg_trans,
|
| 311 |
+
"end": end_trans,
|
| 312 |
+
"text": trans,
|
| 313 |
+
"speaker": "0"
|
| 314 |
+
})
|
| 315 |
+
full_transcription += trans
|
| 316 |
+
|
| 317 |
+
# ----------------
|
| 318 |
+
# Process buffer
|
| 319 |
+
# ----------------
|
| 320 |
+
if args.vac:
|
| 321 |
+
# We need to access the underlying online object to get the buffer
|
| 322 |
+
buffer_text = online.online.concatenate_tsw(online.online.transcript_buffer.buffer)[2]
|
| 323 |
+
else:
|
| 324 |
+
buffer_text = online.concatenate_tsw(online.transcript_buffer.buffer)[2]
|
| 325 |
+
|
| 326 |
+
if buffer_text in full_transcription: # With VAC, the buffer is not updated until the next chunk is processed
|
| 327 |
+
buffer_text = ""
|
| 328 |
+
|
| 329 |
+
buffer_line += buffer_text
|
| 330 |
+
|
| 331 |
+
punctuations = (',', '.', '?', '!', 'and', 'or', 'but', 'however')
|
| 332 |
+
if not any(punctuation in buffer_line for punctuation in punctuations):
|
| 333 |
+
continue
|
| 334 |
+
|
| 335 |
+
last_punctuation_index = max((buffer_line.rfind(p) + len(p) + 1) for p in punctuations if p in buffer_line)
|
| 336 |
+
extracted_text = buffer_line[:last_punctuation_index]
|
| 337 |
+
buffer_line = buffer_line[last_punctuation_index:]
|
| 338 |
+
buffer = {'speaker': '0', 'text': extracted_text, 'translation': None}
|
| 339 |
+
|
| 340 |
+
translation = translator_wrapper(buffer['text'], translation_target_lang, mode='google')
|
| 341 |
+
|
| 342 |
+
buffer['translation'] = translation
|
| 343 |
+
buffer['text'] += ('|' + translation)
|
| 344 |
+
buffer['audio_url'] = generate_audio(translation, speed=1.5) if args.generate_audio else ''
|
| 345 |
+
buffers.append(buffer)
|
| 346 |
+
|
| 347 |
+
# ----------------
|
| 348 |
+
# Process lines
|
| 349 |
+
# ----------------
|
| 350 |
+
'''
|
| 351 |
+
print('Process lines')
|
| 352 |
+
lines = [{"speaker": "0", "text": ""}]
|
| 353 |
+
|
| 354 |
+
if args.diarization:
|
| 355 |
+
await diarization.diarize(pcm_array)
|
| 356 |
+
# diarization.assign_speakers_to_chunks(chunk_history)
|
| 357 |
+
chunk_history = diarization.assign_speakers_to_chunks(chunk_history)
|
| 358 |
+
|
| 359 |
+
for ch in chunk_history:
|
| 360 |
+
if args.diarization and ch["speaker"] and ch["speaker"][-1] != lines[-1]["speaker"]:
|
| 361 |
+
lines.append({"speaker": ch["speaker"], "text": ch['text']})
|
| 362 |
+
|
| 363 |
+
else:
|
| 364 |
+
lines.append({"speaker": ch["speaker"], "text": ch['text']})
|
| 365 |
+
|
| 366 |
+
for i, line in enumerate(lines):
|
| 367 |
+
if line['text'].strip() == '':
|
| 368 |
+
continue
|
| 369 |
+
# translation = translator(line['text'])[0]['translation_text']
|
| 370 |
+
# translation = translation.replace(' ', '')
|
| 371 |
+
# lines[i]['text'] = line['text'] + translation
|
| 372 |
+
lines[i]['text'] = line['text']
|
| 373 |
+
'''
|
| 374 |
+
|
| 375 |
+
print('Before making response')
|
| 376 |
+
response = {'line': buffer, 'buffer': ''}
|
| 377 |
+
print(response)
|
| 378 |
+
await websocket.send_json(response)
|
| 379 |
+
|
| 380 |
+
except Exception as e:
|
| 381 |
+
print(f"Exception in ffmpeg_stdout_reader: {e}")
|
| 382 |
+
break
|
| 383 |
+
|
| 384 |
+
print("Exiting ffmpeg_stdout_reader...")
|
| 385 |
+
|
| 386 |
+
stdout_reader_task = asyncio.create_task(ffmpeg_stdout_reader())
|
| 387 |
+
|
| 388 |
+
try:
|
| 389 |
+
while True:
|
| 390 |
+
# Receive incoming WebM audio chunks from the client
|
| 391 |
+
message = await websocket.receive_bytes()
|
| 392 |
+
# Pass them to ffmpeg via stdin
|
| 393 |
+
ffmpeg_process.stdin.write(message)
|
| 394 |
+
ffmpeg_process.stdin.flush()
|
| 395 |
+
|
| 396 |
+
except WebSocketDisconnect:
|
| 397 |
+
print("WebSocket connection closed.")
|
| 398 |
+
except Exception as e:
|
| 399 |
+
print(f"Error in websocket loop: {e}")
|
| 400 |
+
finally:
|
| 401 |
+
# Clean up ffmpeg and the reader task
|
| 402 |
+
try:
|
| 403 |
+
ffmpeg_process.stdin.close()
|
| 404 |
+
except:
|
| 405 |
+
pass
|
| 406 |
+
stdout_reader_task.cancel()
|
| 407 |
+
|
| 408 |
+
try:
|
| 409 |
+
ffmpeg_process.stdout.close()
|
| 410 |
+
except:
|
| 411 |
+
pass
|
| 412 |
+
|
| 413 |
+
ffmpeg_process.wait()
|
| 414 |
+
del online
|
| 415 |
+
|
| 416 |
+
if args.diarization:
|
| 417 |
+
# Stop Diart
|
| 418 |
+
diarization.close()
|
| 419 |
+
|
| 420 |
+
|
| 421 |
+
if __name__ == "__main__":
|
| 422 |
+
import uvicorn
|
| 423 |
+
|
| 424 |
+
uvicorn.run(
|
| 425 |
+
"app:app", host=args.host, port=args.port, reload=True,
|
| 426 |
+
log_level="info"
|
| 427 |
+
)
|