add parameter to disable transcription (only diarization), add time in output
Browse files- whisper_fastapi_online_server.py +54 -30
whisper_fastapi_online_server.py
CHANGED
|
@@ -3,7 +3,7 @@ import argparse
|
|
| 3 |
import asyncio
|
| 4 |
import numpy as np
|
| 5 |
import ffmpeg
|
| 6 |
-
from time import time
|
| 7 |
from contextlib import asynccontextmanager
|
| 8 |
|
| 9 |
from fastapi import FastAPI, WebSocket, WebSocketDisconnect
|
|
@@ -12,9 +12,12 @@ from fastapi.middleware.cors import CORSMiddleware
|
|
| 12 |
|
| 13 |
from src.whisper_streaming.whisper_online import backend_factory, online_factory, add_shared_args
|
| 14 |
|
| 15 |
-
import subprocess
|
| 16 |
import math
|
| 17 |
import logging
|
|
|
|
|
|
|
|
|
|
|
|
|
| 18 |
|
| 19 |
|
| 20 |
logging.basicConfig(level=logging.INFO, format="%(asctime)s - %(levelname)s - %(message)s")
|
|
@@ -48,6 +51,12 @@ parser.add_argument(
|
|
| 48 |
help="Whether to enable speaker diarization.",
|
| 49 |
)
|
| 50 |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| 51 |
|
| 52 |
add_shared_args(parser)
|
| 53 |
args = parser.parse_args()
|
|
@@ -68,7 +77,10 @@ if args.diarization:
|
|
| 68 |
@asynccontextmanager
|
| 69 |
async def lifespan(app: FastAPI):
|
| 70 |
global asr, tokenizer
|
| 71 |
-
|
|
|
|
|
|
|
|
|
|
| 72 |
yield
|
| 73 |
|
| 74 |
app = FastAPI(lifespan=lifespan)
|
|
@@ -117,7 +129,7 @@ async def websocket_endpoint(websocket: WebSocket):
|
|
| 117 |
|
| 118 |
ffmpeg_process = None
|
| 119 |
pcm_buffer = bytearray()
|
| 120 |
-
online = online_factory(args, asr, tokenizer)
|
| 121 |
diarization = DiartDiarization(SAMPLE_RATE) if args.diarization else None
|
| 122 |
|
| 123 |
async def restart_ffmpeg():
|
|
@@ -130,7 +142,7 @@ async def websocket_endpoint(websocket: WebSocket):
|
|
| 130 |
logger.warning(f"Error killing FFmpeg process: {e}")
|
| 131 |
ffmpeg_process = await start_ffmpeg_decoder()
|
| 132 |
pcm_buffer = bytearray()
|
| 133 |
-
online = online_factory(args, asr, tokenizer)
|
| 134 |
if args.diarization:
|
| 135 |
diarization = DiartDiarization(SAMPLE_RATE)
|
| 136 |
logger.info("FFmpeg process started.")
|
|
@@ -142,7 +154,7 @@ async def websocket_endpoint(websocket: WebSocket):
|
|
| 142 |
loop = asyncio.get_event_loop()
|
| 143 |
full_transcription = ""
|
| 144 |
beg = time()
|
| 145 |
-
|
| 146 |
chunk_history = [] # Will store dicts: {beg, end, text, speaker}
|
| 147 |
|
| 148 |
while True:
|
|
@@ -184,45 +196,57 @@ async def websocket_endpoint(websocket: WebSocket):
|
|
| 184 |
/ 32768.0
|
| 185 |
)
|
| 186 |
pcm_buffer = pcm_buffer[MAX_BYTES_PER_SEC:]
|
| 187 |
-
logger.info(f"{len(online.audio_buffer) / online.SAMPLING_RATE} seconds of audio will be processed by the model.")
|
| 188 |
-
online.insert_audio_chunk(pcm_array)
|
| 189 |
-
transcription = online.process_iter()
|
| 190 |
|
| 191 |
-
if transcription:
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| 192 |
chunk_history.append({
|
| 193 |
-
|
| 194 |
-
|
| 195 |
-
|
| 196 |
-
"speaker": "0"
|
| 197 |
})
|
|
|
|
|
|
|
| 198 |
|
| 199 |
-
full_transcription += transcription.text if transcription else ""
|
| 200 |
-
buffer = online.get_buffer()
|
| 201 |
-
|
| 202 |
-
if buffer in full_transcription: # With VAC, the buffer is not updated until the next chunk is processed
|
| 203 |
-
buffer = ""
|
| 204 |
-
|
| 205 |
-
lines = [
|
| 206 |
-
{
|
| 207 |
-
"speaker": "0",
|
| 208 |
-
"text": "",
|
| 209 |
-
}
|
| 210 |
-
]
|
| 211 |
-
|
| 212 |
if args.diarization:
|
| 213 |
await diarization.diarize(pcm_array)
|
| 214 |
diarization.assign_speakers_to_chunks(chunk_history)
|
| 215 |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| 216 |
for ch in chunk_history:
|
| 217 |
-
if args.diarization and ch["speaker"] and ch["speaker"]
|
|
|
|
| 218 |
lines.append(
|
| 219 |
{
|
| 220 |
-
"speaker":
|
| 221 |
-
"text": ch['text']
|
|
|
|
|
|
|
| 222 |
}
|
| 223 |
)
|
|
|
|
| 224 |
else:
|
| 225 |
lines[-1]["text"] += ch['text']
|
|
|
|
| 226 |
|
| 227 |
response = {"lines": lines, "buffer": buffer}
|
| 228 |
await websocket.send_json(response)
|
|
|
|
| 3 |
import asyncio
|
| 4 |
import numpy as np
|
| 5 |
import ffmpeg
|
| 6 |
+
from time import time, sleep
|
| 7 |
from contextlib import asynccontextmanager
|
| 8 |
|
| 9 |
from fastapi import FastAPI, WebSocket, WebSocketDisconnect
|
|
|
|
| 12 |
|
| 13 |
from src.whisper_streaming.whisper_online import backend_factory, online_factory, add_shared_args
|
| 14 |
|
|
|
|
| 15 |
import math
|
| 16 |
import logging
|
| 17 |
+
from datetime import timedelta
|
| 18 |
+
|
| 19 |
+
def format_time(seconds):
|
| 20 |
+
return str(timedelta(seconds=int(seconds)))
|
| 21 |
|
| 22 |
|
| 23 |
logging.basicConfig(level=logging.INFO, format="%(asctime)s - %(levelname)s - %(message)s")
|
|
|
|
| 51 |
help="Whether to enable speaker diarization.",
|
| 52 |
)
|
| 53 |
|
| 54 |
+
parser.add_argument(
|
| 55 |
+
"--transcription",
|
| 56 |
+
type=bool,
|
| 57 |
+
default=True,
|
| 58 |
+
help="To disable to only see live diarization results.",
|
| 59 |
+
)
|
| 60 |
|
| 61 |
add_shared_args(parser)
|
| 62 |
args = parser.parse_args()
|
|
|
|
| 77 |
@asynccontextmanager
|
| 78 |
async def lifespan(app: FastAPI):
|
| 79 |
global asr, tokenizer
|
| 80 |
+
if args.transcription:
|
| 81 |
+
asr, tokenizer = backend_factory(args)
|
| 82 |
+
else:
|
| 83 |
+
asr, tokenizer = None, None
|
| 84 |
yield
|
| 85 |
|
| 86 |
app = FastAPI(lifespan=lifespan)
|
|
|
|
| 129 |
|
| 130 |
ffmpeg_process = None
|
| 131 |
pcm_buffer = bytearray()
|
| 132 |
+
online = online_factory(args, asr, tokenizer) if args.transcription else None
|
| 133 |
diarization = DiartDiarization(SAMPLE_RATE) if args.diarization else None
|
| 134 |
|
| 135 |
async def restart_ffmpeg():
|
|
|
|
| 142 |
logger.warning(f"Error killing FFmpeg process: {e}")
|
| 143 |
ffmpeg_process = await start_ffmpeg_decoder()
|
| 144 |
pcm_buffer = bytearray()
|
| 145 |
+
online = online_factory(args, asr, tokenizer) if args.transcription else None
|
| 146 |
if args.diarization:
|
| 147 |
diarization = DiartDiarization(SAMPLE_RATE)
|
| 148 |
logger.info("FFmpeg process started.")
|
|
|
|
| 154 |
loop = asyncio.get_event_loop()
|
| 155 |
full_transcription = ""
|
| 156 |
beg = time()
|
| 157 |
+
beg_loop = time()
|
| 158 |
chunk_history = [] # Will store dicts: {beg, end, text, speaker}
|
| 159 |
|
| 160 |
while True:
|
|
|
|
| 196 |
/ 32768.0
|
| 197 |
)
|
| 198 |
pcm_buffer = pcm_buffer[MAX_BYTES_PER_SEC:]
|
|
|
|
|
|
|
|
|
|
| 199 |
|
| 200 |
+
if args.transcription:
|
| 201 |
+
logger.info(f"{len(online.audio_buffer) / online.SAMPLING_RATE} seconds of audio will be processed by the model.")
|
| 202 |
+
online.insert_audio_chunk(pcm_array)
|
| 203 |
+
transcription = online.process_iter()
|
| 204 |
+
if transcription.start:
|
| 205 |
+
chunk_history.append({
|
| 206 |
+
"beg": transcription.start,
|
| 207 |
+
"end": transcription.end,
|
| 208 |
+
"text": transcription.text,
|
| 209 |
+
})
|
| 210 |
+
full_transcription += transcription.text if transcription else ""
|
| 211 |
+
buffer = online.get_buffer()
|
| 212 |
+
if buffer in full_transcription: # With VAC, the buffer is not updated until the next chunk is processed
|
| 213 |
+
buffer = ""
|
| 214 |
+
else:
|
| 215 |
chunk_history.append({
|
| 216 |
+
"beg": time() - beg_loop,
|
| 217 |
+
"end": time() - beg_loop + 0.1,
|
| 218 |
+
"text": '',
|
|
|
|
| 219 |
})
|
| 220 |
+
sleep(0.1)
|
| 221 |
+
buffer = ''
|
| 222 |
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
|
| 223 |
if args.diarization:
|
| 224 |
await diarization.diarize(pcm_array)
|
| 225 |
diarization.assign_speakers_to_chunks(chunk_history)
|
| 226 |
|
| 227 |
+
|
| 228 |
+
current_speaker = -1
|
| 229 |
+
lines = [{
|
| 230 |
+
"beg": 0,
|
| 231 |
+
"end": 0,
|
| 232 |
+
"speaker": current_speaker,
|
| 233 |
+
"text": ""
|
| 234 |
+
}]
|
| 235 |
for ch in chunk_history:
|
| 236 |
+
if args.diarization and ch["speaker"] and ch["speaker"] != current_speaker:
|
| 237 |
+
new_speaker = ch["speaker"]
|
| 238 |
lines.append(
|
| 239 |
{
|
| 240 |
+
"speaker": new_speaker,
|
| 241 |
+
"text": ch['text'],
|
| 242 |
+
"beg": format_time(ch['beg']),
|
| 243 |
+
"end": format_time(ch['end']),
|
| 244 |
}
|
| 245 |
)
|
| 246 |
+
current_speaker = new_speaker
|
| 247 |
else:
|
| 248 |
lines[-1]["text"] += ch['text']
|
| 249 |
+
lines[-1]["end"] = format_time(ch['end'])
|
| 250 |
|
| 251 |
response = {"lines": lines, "buffer": buffer}
|
| 252 |
await websocket.send_json(response)
|