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import io
import argparse
import asyncio
import numpy as np
import ffmpeg
from time import time, sleep
from contextlib import asynccontextmanager
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
from src.whisper_streaming.timed_objects import ASRToken
import math
import logging
from datetime import timedelta
import traceback
def format_time(seconds):
return str(timedelta(seconds=int(seconds)))
logging.basicConfig(level=logging.INFO, format="%(asctime)s - %(levelname)s - %(message)s")
logging.getLogger().setLevel(logging.WARNING)
logger = logging.getLogger(__name__)
logger.setLevel(logging.DEBUG)
##### LOAD ARGS #####
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.",
)
parser.add_argument(
"--transcription",
type=bool,
default=True,
help="To disable to only see live diarization results.",
)
add_shared_args(parser)
args = parser.parse_args()
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
MAX_BYTES_PER_SEC = 32000 * 5 # 5 seconds of audio at 32 kHz
class SharedState:
def __init__(self):
self.tokens = []
self.buffer_transcription = ""
self.buffer_diarization = ""
self.full_transcription = ""
self.end_buffer = 0
self.end_attributed_speaker = 0
self.lock = asyncio.Lock()
self.beg_loop = time()
self.sep = " " # Default separator
self.last_response_content = "" # To track changes in response
async def update_transcription(self, new_tokens, buffer, end_buffer, full_transcription, sep):
async with self.lock:
self.tokens.extend(new_tokens)
self.buffer_transcription = buffer
self.end_buffer = end_buffer
self.full_transcription = full_transcription
self.sep = sep
async def update_diarization(self, end_attributed_speaker, buffer_diarization=""):
async with self.lock:
self.end_attributed_speaker = end_attributed_speaker
if buffer_diarization:
self.buffer_diarization = buffer_diarization
async def add_dummy_token(self):
async with self.lock:
current_time = time() - self.beg_loop
dummy_token = ASRToken(
start=current_time,
end=current_time + 0.5,
text="",
speaker=-1
)
self.tokens.append(dummy_token)
async def get_current_state(self):
async with self.lock:
current_time = time()
remaining_time_transcription = 0
remaining_time_diarization = 0
# Calculate remaining time for transcription buffer
if self.end_buffer > 0:
remaining_time_transcription = max(0, round(current_time - self.beg_loop - self.end_buffer, 2))
# Calculate remaining time for diarization
if self.end_attributed_speaker > 0:
remaining_time_diarization = max(0, round(max(self.end_buffer, self.tokens[-1].end if self.tokens else 0) - self.end_attributed_speaker, 2))
return {
"tokens": self.tokens.copy(),
"buffer_transcription": self.buffer_transcription,
"buffer_diarization": self.buffer_diarization,
"end_buffer": self.end_buffer,
"end_attributed_speaker": self.end_attributed_speaker,
"sep": self.sep,
"remaining_time_transcription": remaining_time_transcription,
"remaining_time_diarization": remaining_time_diarization
}
async def reset(self):
"""Reset the state."""
async with self.lock:
self.tokens = []
self.buffer_transcription = ""
self.buffer_diarization = ""
self.end_buffer = 0
self.end_attributed_speaker = 0
self.full_transcription = ""
self.beg_loop = time()
self.last_response_content = ""
##### LOAD APP #####
@asynccontextmanager
async def lifespan(app: FastAPI):
global asr, tokenizer, diarization
if args.transcription:
asr, tokenizer = backend_factory(args)
else:
asr, tokenizer = None, None
if args.diarization:
from src.diarization.diarization_online import DiartDiarization
diarization = DiartDiarization(SAMPLE_RATE)
else :
diarization = None
yield
app = FastAPI(lifespan=lifespan)
app.add_middleware(
CORSMiddleware,
allow_origins=["*"],
allow_credentials=True,
allow_methods=["*"],
allow_headers=["*"],
)
# Load demo HTML for the root endpoint
with open("src/web/live_transcription.html", "r", encoding="utf-8") as f:
html = f.read()
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
async def transcription_processor(shared_state, pcm_queue, online):
full_transcription = ""
sep = online.asr.sep
while True:
try:
pcm_array = await pcm_queue.get()
logger.info(f"{len(online.audio_buffer) / online.SAMPLING_RATE} seconds of audio will be processed by the model.")
# Process transcription
online.insert_audio_chunk(pcm_array)
new_tokens = online.process_iter()
if new_tokens:
full_transcription += sep.join([t.text for t in new_tokens])
_buffer = online.get_buffer()
buffer = _buffer.text
end_buffer = _buffer.end if _buffer.end else (new_tokens[-1].end if new_tokens else 0)
if buffer in full_transcription:
buffer = ""
await shared_state.update_transcription(
new_tokens, buffer, end_buffer, full_transcription, sep)
except Exception as e:
logger.warning(f"Exception in transcription_processor: {e}")
logger.warning(f"Traceback: {traceback.format_exc()}")
finally:
pcm_queue.task_done()
async def diarization_processor(shared_state, pcm_queue, diarization_obj):
buffer_diarization = ""
while True:
try:
pcm_array = await pcm_queue.get()
# Process diarization
await diarization_obj.diarize(pcm_array)
# Get current state
state = await shared_state.get_current_state()
tokens = state["tokens"]
end_attributed_speaker = state["end_attributed_speaker"]
# Update speaker information
new_end_attributed_speaker = diarization_obj.assign_speakers_to_tokens(
end_attributed_speaker, tokens)
await shared_state.update_diarization(new_end_attributed_speaker, buffer_diarization)
except Exception as e:
logger.warning(f"Exception in diarization_processor: {e}")
logger.warning(f"Traceback: {traceback.format_exc()}")
finally:
pcm_queue.task_done()
async def results_formatter(shared_state, websocket):
while True:
try:
# Get the current state
state = await shared_state.get_current_state()
tokens = state["tokens"]
buffer_transcription = state["buffer_transcription"]
buffer_diarization = state["buffer_diarization"]
end_attributed_speaker = state["end_attributed_speaker"]
remaining_time_transcription = state["remaining_time_transcription"]
remaining_time_diarization = state["remaining_time_diarization"]
sep = state["sep"]
# If diarization is enabled but no transcription, add dummy tokens periodically
if not tokens and not args.transcription and args.diarization:
await shared_state.add_dummy_token()
# Re-fetch tokens after adding dummy
state = await shared_state.get_current_state()
tokens = state["tokens"]
# Process tokens to create response
previous_speaker = -10
lines = [
]
last_end_diarized = 0
undiarized_text = []
for token in tokens:
speaker = token.speaker
# Handle diarization differently if diarization is enabled
if args.diarization:
# If token is not yet processed by diarization
if (speaker == -1 or speaker == 0) and token.end >= end_attributed_speaker:
# Add this token's text to undiarized buffer instead of creating a new line
undiarized_text.append(token.text)
continue
# If speaker isn't assigned yet but should be (based on timestamp)
elif (speaker == -1 or speaker == 0) and token.end < end_attributed_speaker:
speaker = previous_speaker
# Track last diarized token end time
if speaker not in [-1, 0]:
last_end_diarized = max(token.end, last_end_diarized)
if speaker != previous_speaker or not lines:
lines.append(
{
"speaker": speaker,
"text": token.text,
"beg": format_time(token.start),
"end": format_time(token.end),
"diff": round(token.end - last_end_diarized, 2)
}
)
previous_speaker = speaker
elif token.text: # Only append if text isn't empty
lines[-1]["text"] += sep + token.text
lines[-1]["end"] = format_time(token.end)
lines[-1]["diff"] = round(token.end - last_end_diarized, 2)
# Update buffer_diarization with undiarized text
if undiarized_text:
combined_buffer_diarization = sep.join(undiarized_text)
if buffer_transcription:
combined_buffer_diarization += sep
await shared_state.update_diarization(end_attributed_speaker, combined_buffer_diarization)
buffer_diarization = combined_buffer_diarization
# Prepare response object
if lines:
response = {
"lines": lines,
"buffer_transcription": buffer_transcription,
"buffer_diarization": buffer_diarization,
"remaining_time_transcription": remaining_time_transcription,
"remaining_time_diarization": remaining_time_diarization
}
else:
response = {
"lines": [{
"speaker": 1,
"text": "",
"beg": format_time(0),
"end": format_time(token.end) if token else format_time(0),
"diff": 0
}],
"buffer_transcription": buffer_transcription,
"buffer_diarization": buffer_diarization,
"remaining_time_transcription": remaining_time_transcription,
"remaining_time_diarization": remaining_time_diarization
}
response_content = ' '.join([str(line['speaker']) + ' ' + line["text"] for line in lines]) + ' | ' + buffer_transcription + ' | ' + buffer_diarization
if response_content != shared_state.last_response_content:
# Only send if there's actual content to send
if lines or buffer_transcription or buffer_diarization:
await websocket.send_json(response)
shared_state.last_response_content = response_content
# Add a small delay to avoid overwhelming the client
await asyncio.sleep(0.1)
except Exception as e:
logger.warning(f"Exception in results_formatter: {e}")
logger.warning(f"Traceback: {traceback.format_exc()}")
await asyncio.sleep(0.5) # Back off on error
##### ENDPOINTS #####
@app.get("/")
async def get():
return HTMLResponse(html)
@app.websocket("/asr")
async def websocket_endpoint(websocket: WebSocket):
await websocket.accept()
logger.info("WebSocket connection opened.")
ffmpeg_process = None
pcm_buffer = bytearray()
shared_state = SharedState()
transcription_queue = asyncio.Queue() if args.transcription else None
diarization_queue = asyncio.Queue() if args.diarization else None
online = None
async def restart_ffmpeg():
nonlocal ffmpeg_process, online, pcm_buffer
if ffmpeg_process:
try:
ffmpeg_process.kill()
await asyncio.get_event_loop().run_in_executor(None, ffmpeg_process.wait)
except Exception as e:
logger.warning(f"Error killing FFmpeg process: {e}")
ffmpeg_process = await start_ffmpeg_decoder()
pcm_buffer = bytearray()
if args.transcription:
online = online_factory(args, asr, tokenizer)
await shared_state.reset()
logger.info("FFmpeg process started.")
await restart_ffmpeg()
tasks = []
if args.transcription and online:
tasks.append(asyncio.create_task(
transcription_processor(shared_state, transcription_queue, online)))
if args.diarization and diarization:
tasks.append(asyncio.create_task(
diarization_processor(shared_state, diarization_queue, diarization)))
formatter_task = asyncio.create_task(results_formatter(shared_state, websocket))
tasks.append(formatter_task)
async def ffmpeg_stdout_reader():
nonlocal ffmpeg_process, pcm_buffer
loop = asyncio.get_event_loop()
beg = time()
while True:
try:
elapsed_time = math.floor((time() - beg) * 10) / 10 # Round to 0.1 sec
ffmpeg_buffer_from_duration = max(int(32000 * elapsed_time), 4096)
beg = time()
# Read chunk with timeout
try:
chunk = await asyncio.wait_for(
loop.run_in_executor(
None, ffmpeg_process.stdout.read, ffmpeg_buffer_from_duration
),
timeout=5.0
)
except asyncio.TimeoutError:
logger.warning("FFmpeg read timeout. Restarting...")
await restart_ffmpeg()
beg = time()
continue # Skip processing and read from new process
if not chunk:
logger.info("FFmpeg stdout closed.")
break
pcm_buffer.extend(chunk)
if len(pcm_buffer) >= BYTES_PER_SEC:
if len(pcm_buffer) > MAX_BYTES_PER_SEC:
logger.warning(
f"""Audio buffer is too large: {len(pcm_buffer) / BYTES_PER_SEC:.2f} seconds.
The model probably struggles to keep up. Consider using a smaller model.
""")
# Convert int16 -> float32
pcm_array = (
np.frombuffer(pcm_buffer[:MAX_BYTES_PER_SEC], dtype=np.int16).astype(np.float32)
/ 32768.0
)
pcm_buffer = pcm_buffer[MAX_BYTES_PER_SEC:]
if args.transcription and transcription_queue:
await transcription_queue.put(pcm_array.copy())
if args.diarization and diarization_queue:
await diarization_queue.put(pcm_array.copy())
if not args.transcription and not args.diarization:
await asyncio.sleep(0.1)
except Exception as e:
logger.warning(f"Exception in ffmpeg_stdout_reader: {e}")
logger.warning(f"Traceback: {traceback.format_exc()}")
break
logger.info("Exiting ffmpeg_stdout_reader...")
stdout_reader_task = asyncio.create_task(ffmpeg_stdout_reader())
tasks.append(stdout_reader_task)
try:
while True:
# Receive incoming WebM audio chunks from the client
message = await websocket.receive_bytes()
try:
ffmpeg_process.stdin.write(message)
ffmpeg_process.stdin.flush()
except (BrokenPipeError, AttributeError) as e:
logger.warning(f"Error writing to FFmpeg: {e}. Restarting...")
await restart_ffmpeg()
ffmpeg_process.stdin.write(message)
ffmpeg_process.stdin.flush()
except WebSocketDisconnect:
logger.warning("WebSocket disconnected.")
finally:
for task in tasks:
task.cancel()
try:
await asyncio.gather(*tasks, return_exceptions=True)
ffmpeg_process.stdin.close()
ffmpeg_process.wait()
except Exception as e:
logger.warning(f"Error during cleanup: {e}")
if args.diarization and diarization:
diarization.close()
logger.info("WebSocket endpoint cleaned up.")
if __name__ == "__main__":
import uvicorn
uvicorn.run(
"whisper_fastapi_online_server:app", host=args.host, port=args.port, reload=True,
log_level="info"
) |