yjyoon / whisper_streaming_Deprecated star
Quentin Fuxa 2024-12-16
Merge pull request #1 from QuentinFuxa/fast-api-web-interface-with-buffer 
add fastapi server with live webm to pcm conversion and web page show…
@fc9dd80cfe43eb7065a6410c46b6621d02a4c866
6e80c9d
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README.md
--- README.md
+++ README.md
@@ -208,6 +208,51 @@
 
 
 - nc is netcat with server's host and port
 
 
+## Live Transcription Web Interface
 
+
 
+This repository also includes a **FastAPI server** and an **HTML/JavaScript client** for quick testing of live speech transcription in the browser. The client uses native WebSockets and the `MediaRecorder` API to capture microphone audio in **WebM** format and send it to the server—**no additional front-end framework** is required.
 
+
 
+![Demo Screenshot](src/demo.png)
 
+
 
+### How to Launch the Server
 
+
 
+1. **Install Dependencies**:
 
+
 
+    ```bash
 
+    pip install -r requirements.txt
 
+    ```
 
+
 
+2. **Run the FastAPI Server**:
 
+
 
+    ```bash
 
+    python whisper_fastapi_online_server.py --host 0.0.0.0 --port 8000
 
+    ```
 
+
 
+    - `--host` and `--port` let you specify the server’s IP/port.  
+
 
+3. **Open the Provided HTML**:
 
+
 
+    - By default, the server root endpoint `/` serves a simple `live_transcription.html` page.  
+    - Open your browser at `http://localhost:8000` (or replace `localhost` and `8000` with whatever you specified).  
+    - The page uses vanilla JavaScript and the WebSocket API to capture your microphone and stream audio to the server in real time.
 
+
 
+### How the Live Interface Works
 
+
 
+- Once you **allow microphone access**, the page records small chunks of audio using the **MediaRecorder** API in **webm/opus** format.  
+- These chunks are sent over a **WebSocket** to the FastAPI endpoint at `/ws`.  
+- The Python server decodes `.webm` chunks on the fly using **FFmpeg** and streams them into **Whisper** for transcription.  
+- **Partial transcription** appears as soon as enough audio is processed. The “unvalidated” text is shown in **lighter or grey color** (i.e., an ‘aperçu’) to indicate it’s still buffered partial output. Once Whisper finalizes that segment, it’s displayed in normal text.  
+- You can watch the transcription update in near real time, ideal for demos, prototyping, or quick debugging.
 
+
 
+### Deploying to a Remote Server
 
+
 
+If you want to **deploy** this setup:
 
+
 
+1. **Host the FastAPI app** behind a production-grade HTTP server (like **Uvicorn + Nginx** or Docker).  
+2. The **HTML/JS page** can be served by the same FastAPI app or a separate static host.  
+3. Users open the page in **Chrome/Firefox** (any modern browser that supports MediaRecorder + WebSocket).  
+
 
+No additional front-end libraries or frameworks are required. The WebSocket logic in `live_transcription.html` is minimal enough to adapt for your own custom UI or embed in other pages.
 
 
 ## Background
 
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src/demo.png (Binary) (added)
+++ src/demo.png
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src/live_transcription.html (added)
+++ src/live_transcription.html
@@ -0,0 +1,111 @@
+<!DOCTYPE html>
+<html lang="en">
+<head>
+    <meta charset="UTF-8">
+    <meta name="viewport" content="width=device-width, initial-scale=1.0">
+    <title>Audio Transcription</title>
+    <style>
+        body {
+            font-family: 'Inter', sans-serif;
+            text-align: center;
+            margin: 20px;
+        }
+        #recordButton {
+            width: 80px;
+            height: 80px;
+            font-size: 36px;
+            border: none;
+            border-radius: 50%;
+            background-color: white;
+            cursor: pointer;
+            box-shadow: 0 0px 10px rgba(0, 0, 0, 0.2);
+            transition: background-color 0.3s ease, transform 0.2s ease;
+        }
+        #recordButton.recording {
+            background-color: #ff4d4d;
+            color: white;
+        }
+        #recordButton:active {
+            transform: scale(0.95);
+        }
+        #transcriptions {
+            margin-top: 20px;
+            font-size: 18px;
+            text-align: left;
+        }
+        .transcription {
+            display: inline;
+            color: black;
+        }
+        .buffer {
+            display: inline;
+            color: rgb(197, 197, 197);
+        }
+    </style>
+</head>
+<body>
+    <p id="status">Click to start transcription</p>
+    <button id="recordButton">🎙�</button>
+    <div id="transcriptions"></div>
+
+    <script>
+        let isRecording = false, websocket, recorder;
+
+        const statusText = document.getElementById("status");
+        const recordButton = document.getElementById("recordButton");
+        const transcriptionsDiv = document.getElementById("transcriptions");
+
+        let fullTranscription = ""; // Store confirmed transcription
+
+        function setupWebSocket() {
+            websocket = new WebSocket("ws://localhost:8000/ws");
+            websocket.onmessage = (event) => {
+                const data = JSON.parse(event.data);
+                const { transcription, buffer } = data;
+
+                // Update confirmed transcription
+                fullTranscription += transcription;
+
+                // Update the transcription display
+                transcriptionsDiv.innerHTML = `
+                    <span class="transcription">${fullTranscription}</span>
+                    <span class="buffer">${buffer}</span>
+                `;
+            };
+        }
+
+        async function startRecording() {
+            const stream = await navigator.mediaDevices.getUserMedia({ audio: true });
+            recorder = new MediaRecorder(stream, { mimeType: "audio/webm" });
+            recorder.ondataavailable = (e) => websocket?.send(e.data);
+            recorder.start(3000);
+            isRecording = true;
+            updateUI();
+        }
+
+        function stopRecording() {
+            recorder?.stop();
+            recorder = null;
+            isRecording = false;
+            websocket?.close();
+            websocket = null;
+            updateUI();
+        }
+
+        async function toggleRecording() {
+            if (isRecording) stopRecording();
+            else {
+                setupWebSocket();
+                await startRecording();
+            }
+        }
+
+        function updateUI() {
+            recordButton.classList.toggle("recording", isRecording);
+            statusText.textContent = isRecording ? "Recording..." : "Click to start transcription";
+        }
+
+        recordButton.addEventListener("click", toggleRecording);
+    </script>
+</body>
+</html>(파일 끝에 줄바꿈 문자 없음)
 
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whisper_fastapi_online_server.py (added)
+++ whisper_fastapi_online_server.py
@@ -0,0 +1,140 @@
+import io
+import argparse
+import asyncio
+import numpy as np
+import ffmpeg
+
+from fastapi import FastAPI, WebSocket, WebSocketDisconnect
+from fastapi.responses import HTMLResponse
+from fastapi.middleware.cors import CORSMiddleware
+
+from whisper_online import asr_factory, add_shared_args
+
+app = FastAPI()
+app.add_middleware(
+    CORSMiddleware,
+    allow_origins=["*"],
+    allow_credentials=True,
+    allow_methods=["*"],
+    allow_headers=["*"],
+)
+
+
+# Argument parsing
+parser = argparse.ArgumentParser()
+parser.add_argument("--host", type=str, default='localhost')
+parser.add_argument("--port", type=int, default=8000)
+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 .")
+add_shared_args(parser)
+args = parser.parse_args()
+
+# Initialize Whisper
+asr, online = asr_factory(args)
+
+# Load demo HTML for the root endpoint
+with open("live_transcription.html", "r") as f:
+    html = f.read()
+
+@app.get("/")
+async def get():
+    return HTMLResponse(html)
+
+# Streaming constants
+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("/ws")
+async def websocket_endpoint(websocket: WebSocket):
+    await websocket.accept()
+    print("WebSocket connection opened.")
+
+    ffmpeg_process = await start_ffmpeg_decoder()
+    pcm_buffer = bytearray()
+
+    # Continuously read decoded PCM from ffmpeg stdout in a background task
+    async def ffmpeg_stdout_reader():
+        nonlocal pcm_buffer
+        loop = asyncio.get_event_loop()
+        while True:
+            try:
+                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)
+
+                # Process in 3-second batches
+                while len(pcm_buffer) >= BYTES_PER_SEC:
+                    three_sec_chunk = pcm_buffer[:BYTES_PER_SEC]
+                    del pcm_buffer[:BYTES_PER_SEC]
+
+                    # Convert int16 -> float32
+                    pcm_array = np.frombuffer(three_sec_chunk, dtype=np.int16).astype(np.float32) / 32768.0
+
+                    # Send PCM data to Whisper
+                    online.insert_audio_chunk(pcm_array)
+                    transcription = online.process_iter()
+                    buffer = online.to_flush(online.transcript_buffer.buffer)
+
+                    # Return partial transcription results to the client
+                    await websocket.send_json({
+                        "transcription": transcription[2],
+                        "buffer": buffer[2]
+                    })
+            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()
+        
+        
+if __name__ == "__main__":
+    import uvicorn
+    uvicorn.run("whisper_fastapi_online_server:app", host=args.host, port=args.port, reload=True)(파일 끝에 줄바꿈 문자 없음)
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