feat: streaming api
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README_BW.md
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165
README_BW.md
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# Qwen3-ASR
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https://github.com/QwenLM/Qwen3-ASR
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## 📦 Version History
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| Version | Date | Summary |
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|---------|------------|---------------------------------|
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| 0.0.1 | 2026-04-22 | Initial version |
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### 🔄 Version Details
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#### 🆕 0.0.1 – *2026-04-22*
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- ✅ **Core Features**
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- Initial Qwen3-ASR integration
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---
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# Start
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```bash
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docker pull harbor.bwgdi.com/library/qwen3asr:0.0.1
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# Run with custom model path
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# -e ASR_MODEL_PATH: Model name or local path inside container
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docker run -d --restart always -p 8000:8000 --gpus all \
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-e ASR_MODEL_PATH="Qwen/Qwen3-ASR-1.7B" \
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--mount type=bind,source=/path/to/your/models,target=/models \
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harbor.bwgdi.com/library/qwen3asr:0.0.3
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```
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# Usage
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## Non-streaming (HTTP POST)
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Transcribe an entire audio file.
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```bash
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curl -X POST http://localhost:8000/asr/transcribe \
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-F "file=@audio.wav" \
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-F "language=Chinese"
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```
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## Streaming (WebSocket)
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Real-time incremental transcription.
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- **URL**: `ws://localhost:8000/asr/stream`
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- **Protocol**:
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- Client sends `bytes`: float32 PCM 16kHz audio chunks.
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- Client sends `text`: `{"command": "finish"}` to stop.
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- Server sends `text`: `{"session_id": ..., "language": ..., "text": ..., "is_final": bool}`
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Example using Python `websockets`:
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```python
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# coding=utf-8
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import argparse
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import asyncio
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import io
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import json
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import logging
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import urllib.request
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import numpy as np
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import soundfile as sf
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import websockets
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# Configure logging
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logging.basicConfig(level=logging.INFO, format='%(asctime)s - %(levelname)s - %(message)s')
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logger = logging.getLogger(__name__)
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async def stream_audio_to_api(uri: str, audio_path: str, chunk_size_ms: int = 500):
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"""
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Load audio and stream it in chunks to the ASR WebSocket API.
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"""
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logger.info(f"Loading audio from {audio_path}...")
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# Load audio data
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if audio_path.startswith("http"):
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# Download from URL
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req = urllib.request.Request(audio_path, headers={"User-Agent": "Mozilla/5.0"})
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with urllib.request.urlopen(req, timeout=30) as resp:
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audio_bytes = resp.read()
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f = io.BytesIO(audio_bytes)
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else:
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# Load local file
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f = audio_path
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# Read audio as Float32
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wav, sr = sf.read(f, dtype="float32", always_2d=False)
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# Simple resample to 16k if needed (for better accuracy)
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if sr != 16000:
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logger.warning(f"Audio sample rate is {sr}, resampling to 16000...")
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dur = wav.shape[0] / float(sr)
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n16 = int(round(dur * 16000))
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x_old = np.linspace(0.0, dur, num=wav.shape[0], endpoint=False)
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x_new = np.linspace(0.0, dur, num=n16, endpoint=False)
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wav = np.interp(x_new, x_old, wav).astype(np.float32)
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sr = 16000
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# Calculate samples per chunk
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chunk_samples = int(sr * chunk_size_ms / 1000)
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logger.info(f"Connecting to WebSocket at {uri}...")
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try:
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async with websockets.connect(uri) as websocket:
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logger.info("Connected. Streaming audio...")
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pos = 0
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call_id = 0
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while pos < len(wav):
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chunk = wav[pos : pos + chunk_samples]
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pos += len(chunk)
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call_id += 1
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# Send binary Float32 data
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await websocket.send(chunk.tobytes())
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# Wait for immediate response (intermediate result)
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try:
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response = await asyncio.wait_for(websocket.recv(), timeout=2.0)
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result = json.loads(response)
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if "error" in result:
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logger.error(f"API Error: {result['error']}")
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return
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lang = result.get("language", "unknown")
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text = result.get("text", "")
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print(f"[Chunk {call_id:03d}] Lang: {lang:7s} | Text: {text}")
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except asyncio.TimeoutError:
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logger.warning(f"Timeout waiting for response on chunk {call_id}")
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# Optional: simulate real-time performance
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# await asyncio.sleep(chunk_size_ms / 1000)
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# Send finish command
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logger.info("Finished streaming audio. Sending 'finish' command...")
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await websocket.send(json.dumps({"command": "finish"}))
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# Wait for final response
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try:
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final_response = await asyncio.wait_for(websocket.recv(), timeout=5.0)
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final_result = json.loads(final_response)
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print("\n" + "="*50)
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print("FINAL RESULT:")
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print(f"Language: {final_result.get('language')}")
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print(f"Text: {final_result.get('text')}")
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print("="*50)
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except asyncio.TimeoutError:
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logger.error("Timeout waiting for final response")
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except Exception as e:
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logger.error(f"WebSocket Error: {e}")
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def main():
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parser = argparse.ArgumentParser(description="Qwen3-ASR Streaming API Client Test")
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parser.add_argument("--url", default="ws://localhost:8000/asr/stream", help="WebSocket API URI")
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parser.add_argument("--audio", default="https://qianwen-res.oss-cn-beijing.aliyuncs.com/Qwen3-ASR-Repo/asr_en.wav",
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help="Path or URL to audio file")
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parser.add_argument("--chunk-ms", type=int, default=1000, help="Chunk size in milliseconds")
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args = parser.parse_args()
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asyncio.run(stream_audio_to_api(args.url, args.audio, args.chunk_ms))
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if __name__ == "__main__":
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main()
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```
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