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from fastapi import FastAPI, HTTPException |
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from fastapi.middleware.cors import CORSMiddleware |
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from pydantic import BaseModel |
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import librosa |
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import torch |
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import base64 |
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import io |
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import logging |
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import numpy as np |
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from transformers import AutoModel, AutoTokenizer |
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logging.basicConfig(level=logging.INFO) |
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logger = logging.getLogger(__name__) |
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app = FastAPI() |
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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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class AudioRequest(BaseModel): |
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audio_data: str |
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sample_rate: int |
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class AudioResponse(BaseModel): |
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audio_data: str |
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text: str = "" |
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INITIALIZATION_STATUS = { |
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"model_loaded": False, |
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"error": None |
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} |
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class Model: |
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def __init__(self): |
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self.model = model = AutoModel.from_pretrained( |
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'./models/checkpoint', |
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trust_remote_code=True, |
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torch_dtype=torch.bfloat16, |
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attn_implementation='sdpa' |
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) |
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model = model.eval().cuda() |
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self.tokenizer = AutoTokenizer.from_pretrained( |
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'./models/checkpoint', |
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trust_remote_code=True |
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) |
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model.init_tts() |
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model.tts.float() |
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self.model_in_sr = 16000 |
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self.model_out_sr = 24000 |
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self.ref_audio, _ = librosa.load('./ref_audios/female_example.wav', sr=self.model_in_sr, mono=True) |
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self.sys_prompt = model.get_sys_prompt(ref_audio=self.ref_audio, mode='audio_assistant', language='en') |
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audio_data = librosa.load('./ref_audios/male_example.wav', sr=self.model_in_sr, mono=True)[0] |
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_ = self.inference(audio_data, self.model_in_sr) |
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def inference(self, audio_np, input_audio_sr): |
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if input_audio_sr != self.model_in_sr: |
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audio_np = librosa.resample(audio_np, orig_sr=input_audio_sr, target_sr=self.model_in_sr) |
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user_question = {'role': 'user', 'content': [audio_np]} |
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msgs = [self.sys_prompt, user_question] |
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res = self.model.chat( |
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msgs=msgs, |
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tokenizer=self.tokenizer, |
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sampling=True, |
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max_new_tokens=128, |
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use_tts_template=True, |
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generate_audio=True, |
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temperature=0.3, |
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) |
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audio = res["audio_wav"].cpu().numpy() |
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if self.model_out_sr != input_audio_sr: |
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audio = librosa.resample(audio, orig_sr=self.model_out_sr, target_sr=input_audio_sr) |
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return audio, res["text"] |
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def initialize_model(): |
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"""Initialize the MiniCPM model""" |
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global model, INITIALIZATION_STATUS |
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try: |
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logger.info("Initializing model...") |
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model = Model() |
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INITIALIZATION_STATUS["model_loaded"] = True |
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logger.info("MiniCPM model initialized successfully") |
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return True |
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except Exception as e: |
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INITIALIZATION_STATUS["error"] = str(e) |
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logger.error(f"Failed to initialize model: {e}") |
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return False |
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@app.on_event("startup") |
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async def startup_event(): |
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"""Initialize model on startup""" |
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initialize_model() |
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@app.get("/api/v1/health") |
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def health_check(): |
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"""Health check endpoint""" |
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status = { |
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"status": "healthy" if INITIALIZATION_STATUS["model_loaded"] else "initializing", |
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"model_loaded": INITIALIZATION_STATUS["model_loaded"], |
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"error": INITIALIZATION_STATUS["error"] |
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} |
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return status |
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@app.post("/api/v1/inference") |
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async def inference(request: AudioRequest) -> AudioResponse: |
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"""Run inference with MiniCPM model""" |
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if not INITIALIZATION_STATUS["model_loaded"]: |
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raise HTTPException( |
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status_code=503, |
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detail=f"Model not ready. Status: {INITIALIZATION_STATUS}" |
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) |
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try: |
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audio_bytes = base64.b64decode(request.audio_data) |
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audio_np = np.load(io.BytesIO(audio_bytes)).flatten() |
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import time |
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start = time.time() |
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print(f"starting inference with audio length {audio_np.shape}") |
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audio_response, text_response = model.inference(audio_np, request.sample_rate) |
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print(f"inference took {time.time() - start} seconds") |
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buffer = io.BytesIO() |
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np.save(buffer, audio_response) |
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audio_b64 = base64.b64encode(buffer.getvalue()).decode() |
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return AudioResponse( |
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audio_data=audio_b64, |
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text=text_response |
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) |
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except Exception as e: |
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logger.error(f"Inference failed: {str(e)}") |
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raise HTTPException( |
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status_code=500, |
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detail=str(e) |
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) |
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if __name__ == "__main__": |
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import uvicorn |
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uvicorn.run(app, host="0.0.0.0", port=8000) |
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