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Update app.py
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app.py
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from fastapi import FastAPI, UploadFile, File, WebSocket
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from fastapi.responses import HTMLResponse, JSONResponse
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from fastapi.middleware.cors import CORSMiddleware
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from pywhispercpp.model import Model
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import uvicorn
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import tempfile
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import os
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from time import time
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app = FastAPI(title="pyWhisperCPP
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# Allow CORS (useful if you host frontend separately, but fine on Spaces too)
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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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# Load Whisper.cpp model ONCE
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# OPTIONS: 'tiny.en', 'base.en', etc.
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model = Model("base.en")
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# ---------- Simple HTML frontend ----------
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@app.get("/", response_class=HTMLResponse)
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async def index():
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# Serve the index.html file
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with open("index.html", "r", encoding="utf-8") as f:
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return HTMLResponse(f.read())
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# ---------- Normal file upload transcription ----------
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@app.post("/transcribe")
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async def transcribe(file: UploadFile = File(...)):
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# Save uploaded audio temporarily
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with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as temp:
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temp.write(await file.read())
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temp.flush()
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audio_path = temp.name
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elapsed = round(time() - start, 3)
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return {
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"text": text,
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"processing_time_seconds": elapsed
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}
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finally:
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os.remove(audio_path)
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#
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periodically transcribes the buffered audio with Whisper.cpp,
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and sends back partial text.
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"""
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await websocket.accept()
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buffer = b""
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MIN_CHUNK_SIZE = 40_000 # bytes before running a transcription (tune this)
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try:
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while True:
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message = await websocket.receive()
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# Handle text messages (control)
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if "text" in message and message["text"] is not None:
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text_msg = message["text"]
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if text_msg == "__END__":
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# Finish stream
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break
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# Ignore other text controls for now
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continue
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# Handle binary audio data
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chunk = message.get("bytes")
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if not chunk:
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continue
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buffer += chunk
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# When enough audio collected, transcribe
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if len(buffer) >= MIN_CHUNK_SIZE:
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with tempfile.NamedTemporaryFile(delete=False, suffix=".webm") as temp:
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temp.write(buffer)
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temp.flush()
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audio_path = temp.name
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try:
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segments = model.transcribe(audio_path)
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text = " ".join(seg.text for seg in segments).strip()
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finally:
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os.remove(audio_path)
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# Send partial transcript to client
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if text:
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await websocket.send_text(text)
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# Clear buffer (or keep tail if you want overlap)
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buffer = b""
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# End-of-stream message
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await websocket.send_text("[stream ended]")
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except WebSocketDisconnect:
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# Client disconnected
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pass
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finally:
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await websocket.close()
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@app.websocket("/ws/transcribe_pcm")
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async def websocket_transcription_pcm(websocket: WebSocket):
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await websocket.accept()
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buffer = b""
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SAMPLE_RATE = 16000
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MIN_PCM_SIZE = SAMPLE_RATE * 2 * 3 # 3 seconds buffer
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try:
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while True:
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chunk = await websocket.receive_bytes()
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# If end control message (optional)
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if chunk == b"__END__":
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break
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if text:
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await websocket.send_text(text)
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pass
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finally:
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await websocket.close()
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if __name__ == "__main__":
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# For local testing. On Spaces, you don't usually run uvicorn manually.
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uvicorn.run(app, host="0.0.0.0", port=7860)
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from fastapi import FastAPI, UploadFile, File, WebSocket
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from pywhispercpp.model import Model
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import uvicorn, tempfile, os
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from time import time
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app = FastAPI(title="pyWhisperCPP API")
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model = Model("base.en")
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@app.get("/")
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def root():
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return {"status": "Whisper.cpp API is running!"}
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@app.post("/transcribe")
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async def transcribe(file: UploadFile = File(...)):
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with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as temp:
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temp.write(await file.read())
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temp.flush()
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audio_path = temp.name
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start = time()
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segments = model.transcribe(audio_path)
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text = " ".join(seg.text for seg in segments])
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os.remove(audio_path)
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return {"text": text}
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# ================================
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# 🔥 Real-time streaming endpoint
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# ================================
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@app.websocket("/ws/live")
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async def websocket_live(websocket: WebSocket):
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await websocket.accept()
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buffer = b""
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while True:
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data = await websocket.receive_bytes()
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if data == b"__END__":
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break
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buffer += data
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if len(buffer) > 32000: # ~1 second PCM16 16kHz
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with tempfile.NamedTemporaryFile(delete=False, suffix=".wav") as temp:
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temp.write(buffer)
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temp.flush()
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audio_path = temp.name
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segments = model.transcribe(audio_path)
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text = " ".join(seg.text for seg in segments])
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await websocket.send_text(text)
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buffer = b""
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os.remove(audio_path)
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await websocket.send_text("[END]")
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await websocket.close()
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if __name__ == "__main__":
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uvicorn.run(app, host="0.0.0.0", port=7860)
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