realtime-stt-translation / whisper_fastapi_online_server.py
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import io
import requests
import argparse
import asyncio
import numpy as np
import ffmpeg
from time import time
from fastapi import FastAPI, WebSocket, WebSocketDisconnect
from fastapi.responses import HTMLResponse
from fastapi.middleware.cors import CORSMiddleware
from src.whisper_streaming.whisper_online import backend_factory, online_factory, add_shared_args
import logging
import logging.config
from transformers import pipeline
MODEL_NAME = 'Helsinki-NLP/opus-tatoeba-en-ja'
TRANSLATOR = pipeline('translation', model=MODEL_NAME, device='cuda')
TRANSLATOR('Warming up!')
API_KEY = '3c2b8b0f-4fa9-4eb7-b67d-7cae25546051:fx' # 自身の API キーを指定
SOURCE_LANG = 'EN'
TARGET_LANG = 'JA'
def translator_wrapper(source_text, mode='deepl'):
if mode == 'deepl':
params = {
'auth_key' : API_KEY,
'text' : source_text,
'source_lang' : SOURCE_LANG, # 翻訳対象の言語
"target_lang": TARGET_LANG # 翻訳後の言語
}
# リクエストを投げる
try:
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
result = request.json()['translations'][0]['text']
except requests.exceptions.Timeout:
result = "(timed out)"
return result
elif mode == 'marianmt':
return TRANSLATOR(source_text)[0]['translation_text']
elif mode == 'google':
import requests
# https://www.eyoucms.com/news/ziliao/other/29445.html
language_type = ""
target = 'ja-jp'
url = "https://translation.googleapis.com/language/translate/v2"
data = {
'key':"AIzaSyCX0-Wdxl_rgvcZzklNjnqJ1W9YiKjcHUs", # 認証の設定:APIキー
'source': language_type,
'target': target,
'q': source_text,
'format': "text"
}
#headers = {'X-HTTP-Method-Override': 'GET'}
#response = requests.post(url, data=data, headers=headers)
response = requests.post(url, data)
# print(response.json())
print(response)
res = response.json()
print(res["data"]["translations"][0]["translatedText"])
result = res["data"]["translations"][0]["translatedText"]
print(result)
return result
def setup_logging():
logging_config = {
'version': 1,
'disable_existing_loggers': False,
'formatters': {
'standard': {
'format': '%(asctime)s %(levelname)s [%(name)s]: %(message)s',
},
},
'handlers': {
'console': {
'level': 'INFO',
'class': 'logging.StreamHandler',
'formatter': 'standard',
},
},
'root': {
'handlers': ['console'],
'level': 'DEBUG',
},
'loggers': {
'uvicorn': {
'handlers': ['console'],
'level': 'INFO',
'propagate': False,
},
'uvicorn.error': {
'level': 'INFO',
},
'uvicorn.access': {
'level': 'INFO',
},
'src.whisper_streaming.online_asr': { # Add your specific module here
'handlers': ['console'],
'level': 'DEBUG',
'propagate': False,
},
'src.whisper_streaming.whisper_streaming': { # Add your specific module here
'handlers': ['console'],
'level': 'DEBUG',
'propagate': False,
},
},
}
logging.config.dictConfig(logging_config)
setup_logging()
logger = logging.getLogger(__name__)
app = FastAPI()
app.add_middleware(
CORSMiddleware,
allow_origins=["*"],
allow_credentials=True,
allow_methods=["*"],
allow_headers=["*"],
)
parser = argparse.ArgumentParser(description="Whisper FastAPI Online Server")
parser.add_argument(
"--host",
type=str,
default="localhost",
help="The host address to bind the server to.",
)
parser.add_argument(
"--port", type=int, default=8000, help="The port number to bind the server to."
)
parser.add_argument(
"--warmup-file",
type=str,
dest="warmup_file",
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 .",
)
parser.add_argument(
"--diarization",
type=bool,
default=False,
help="Whether to enable speaker diarization.",
)
add_shared_args(parser)
args = parser.parse_args()
# args.model = 'medium'
asr, tokenizer = backend_factory(args)
if args.diarization:
from src.diarization.diarization_online import DiartDiarization
# Load demo HTML for the root endpoint
with open("src/web/live_transcription.html", "r", encoding="utf-8") as f:
html = f.read()
@app.get("/")
async def get():
return HTMLResponse(html)
SAMPLE_RATE = 16000
CHANNELS = 1
SAMPLES_PER_SEC = SAMPLE_RATE * int(args.min_chunk_size)
BYTES_PER_SAMPLE = 2 # s16le = 2 bytes per sample
BYTES_PER_SEC = SAMPLES_PER_SEC * BYTES_PER_SAMPLE
async def start_ffmpeg_decoder():
"""
Start an FFmpeg process in async streaming mode that reads WebM from stdin
and outputs raw s16le PCM on stdout. Returns the process object.
"""
process = (
ffmpeg.input("pipe:0", format="webm")
.output(
"pipe:1",
format="s16le",
acodec="pcm_s16le",
ac=CHANNELS,
ar=str(SAMPLE_RATE),
)
.run_async(pipe_stdin=True, pipe_stdout=True, pipe_stderr=True)
)
return process
@app.websocket("/asr")
async def websocket_endpoint(websocket: WebSocket):
await websocket.accept()
print("WebSocket connection opened.")
ffmpeg_process = await start_ffmpeg_decoder()
pcm_buffer = bytearray()
print("Loading online.")
online = online_factory(args, asr, tokenizer)
print("Online loaded.")
if args.diarization:
diarization = DiartDiarization(SAMPLE_RATE)
# Continuously read decoded PCM from ffmpeg stdout in a background task
async def ffmpeg_stdout_reader():
nonlocal pcm_buffer
loop = asyncio.get_event_loop()
full_transcription = ""
beg = time()
chunk_history = [] # Will store dicts: {beg, end, text, speaker}
buffers = [{'speaker': '0', 'text': '', 'translation': None}]
buffer_line = ''
while True:
print('in while')
try:
print('try in while')
elapsed_time = int(time() - beg)
beg = time()
print('before await loop.run_in_executor()')
chunk = await loop.run_in_executor(None, ffmpeg_process.stdout.read, 32000 * elapsed_time)
print('before if not chunk')
if not chunk: # The first chunk will be almost empty, FFmpeg is still starting up
chunk = await loop.run_in_executor(None, ffmpeg_process.stdout.read, 4096)
if not chunk: # FFmpeg might have closed
print("FFmpeg stdout closed.")
break
pcm_buffer.extend(chunk)
print('before if len(pcm_buffer)')
if len(pcm_buffer) >= BYTES_PER_SEC:
print('in if len(pcm_buffer)')
# Convert int16 -> float32
pcm_array = (np.frombuffer(pcm_buffer, dtype=np.int16).astype(np.float32) / 32768.0)
pcm_buffer = bytearray() # Initialize the PCM buffer
online.insert_audio_chunk(pcm_array)
beg_trans, end_trans, trans = online.process_iter()
if trans:
chunk_history.append({
"beg": beg_trans,
"end": end_trans,
"text": trans,
"speaker": "0"
})
full_transcription += trans
# ----------------
# Process buffer
# ----------------
if args.vac:
# We need to access the underlying online object to get the buffer
buffer = online.online.concatenate_tsw(online.online.transcript_buffer.buffer)[2]
else:
buffer = online.concatenate_tsw(online.transcript_buffer.buffer)[2]
if buffer in full_transcription: # With VAC, the buffer is not updated until the next chunk is processed
buffer = ""
buffer_line += buffer
punctuations = (',', '.', '?', '!', 'and', 'or', 'but', 'however')
if any(punctuation in buffer_line for punctuation in punctuations):
last_punctuation_index = max((buffer_line.rfind(p) + len(p) + 1) for p in punctuations if p in buffer_line)
extracted_text = buffer_line[:last_punctuation_index]
buffer_line = buffer_line[last_punctuation_index:]
buffers.append({'speaker': '0', 'text': extracted_text, 'translation': None})
# Translation loop
print('buffers for loop')
for i, buffer in enumerate(buffers):
print(i, buffer)
if buffer['translation'] is not None:
continue
if buffer['text'] == '':
continue
transcription = buffer['text']
buffers[i]['translation'] = translator_wrapper(transcription, mode='google')
buffers[i]['text'] += ('|' + buffers[i]['translation'])
# ----------------
# Process lines
# ----------------
print('Process lines')
lines = [{"speaker": "0", "text": ""}]
if args.diarization:
await diarization.diarize(pcm_array)
# diarization.assign_speakers_to_chunks(chunk_history)
chunk_history = diarization.assign_speakers_to_chunks(chunk_history)
for ch in chunk_history:
if args.diarization and ch["speaker"] and ch["speaker"][-1] != lines[-1]["speaker"]:
lines.append({"speaker": ch["speaker"], "text": ch['text']})
else:
lines.append({"speaker": ch["speaker"], "text": ch['text']})
for i, line in enumerate(lines):
if line['text'].strip() == '':
continue
# translation = translator(line['text'])[0]['translation_text']
# translation = translation.replace(' ', '')
# lines[i]['text'] = line['text'] + translation
lines[i]['text'] = line['text']
# translation = translator(buffer)[0]['translation_text']
# translation = translation.replace(' ', '')
# buffer += translation
print('Before making response')
response = {"lines": buffers, "buffer": ''}
await websocket.send_json(response)
except Exception as e:
print(f"Exception in ffmpeg_stdout_reader: {e}")
break
print("Exiting ffmpeg_stdout_reader...")
stdout_reader_task = asyncio.create_task(ffmpeg_stdout_reader())
try:
while True:
# Receive incoming WebM audio chunks from the client
message = await websocket.receive_bytes()
# Pass them to ffmpeg via stdin
ffmpeg_process.stdin.write(message)
ffmpeg_process.stdin.flush()
except WebSocketDisconnect:
print("WebSocket connection closed.")
except Exception as e:
print(f"Error in websocket loop: {e}")
finally:
# Clean up ffmpeg and the reader task
try:
ffmpeg_process.stdin.close()
except:
pass
stdout_reader_task.cancel()
try:
ffmpeg_process.stdout.close()
except:
pass
ffmpeg_process.wait()
del online
if args.diarization:
# Stop Diart
diarization.close()
if __name__ == "__main__":
import uvicorn
uvicorn.run(
"whisper_fastapi_online_server:app", host=args.host, port=args.port, reload=True,
log_level="info"
)