import json import os.path import tempfile import sys import re import uuid import requests import librosa import numpy as np import torch import uvicorn import torchaudio import base64 import io from argparse import ArgumentParser from fastapi import FastAPI, HTTPException from fastapi.middleware.cors import CORSMiddleware from pydantic import BaseModel from transformers import WhisperFeatureExtractor, AutoTokenizer from speech_tokenizer.modeling_whisper import WhisperVQEncoder sys.path.insert(0, "./cosyvoice") sys.path.insert(0, "./third_party/Matcha-TTS") from speech_tokenizer.utils import extract_speech_token from flow_inference import AudioDecoder # Initialize FastAPI app = FastAPI() app.add_middleware( CORSMiddleware, allow_origins=["*"], allow_credentials=True, allow_methods=["*"], allow_headers=["*"], ) class AudioRequest(BaseModel): audio_data: str # Base64 encoded audio sample_rate: int class AudioResponse(BaseModel): audio_data: str # Base64 encoded response audio text_transcript: str # Global initialization DEVICE = "cuda" audio_decoder = None whisper_model = None feature_extractor = None glm_tokenizer = None def initialize_models(): global audio_decoder, feature_extractor, whisper_model, glm_tokenizer # Initialize Whisper and GLM components glm_tokenizer = AutoTokenizer.from_pretrained("THUDM/glm-4-voice-9b", trust_remote_code=True) # Initialize Whisper model whisper_model = WhisperVQEncoder.from_pretrained("THUDM/glm-4-voice-tokenizer").eval().to(DEVICE) feature_extractor = WhisperFeatureExtractor.from_pretrained("THUDM/glm-4-voice-tokenizer") # Initialize AudioDecoder audio_decoder = AudioDecoder( config_path="./glm-4-voice-decoder/config.yaml", flow_ckpt_path="./glm-4-voice-decoder/flow.pt", hift_ckpt_path="./glm-4-voice-decoder/hift.pt", device=DEVICE ) @app.on_event("startup") async def startup_event(): try: initialize_models() except Exception as e: raise HTTPException(status_code=500, detail=f"Model initialization failed: {str(e)}") def process_audio(audio_bytes: bytes, target_sr: int = 16000): # Convert bytes to numpy array audio_np = np.frombuffer(audio_bytes, dtype=np.int16) # Resample if necessary if target_sr != 16000: # Whisper's default sample rate audio_np = librosa.resample(audio_np, orig_sr=target_sr, target_sr=16000) return audio_np @app.post("/api/voice_chat") async def voice_chat(request: AudioRequest): try: # Decode and process audio audio_bytes = base64.b64decode(request.audio_data) audio_np = process_audio(audio_bytes, request.sample_rate) # Extract tokens with tempfile.TemporaryDirectory() as tmp_dir: tmp_path = os.path.join(tmp_dir, "audio.wav") torchaudio.save(tmp_path, torch.from_numpy(audio_np).unsqueeze(0), request.sample_rate) audio_tokens = extract_speech_token(whisper_model, feature_extractor, [tmp_path])[0] if not audio_tokens: raise HTTPException(400, "No speech detected") # Generate response response = requests.post( "http://localhost:10000/generate_stream", json={ "prompt": f"<|system|>Respond<|user|>{' '.join(f'<|audio_{x}|>' for x in audio_tokens)}<|assistant|>", "temperature": 0.7, "top_p": 0.9, "max_new_tokens": 256 }, stream=True ) # Process response text_tokens = [] audio_tokens = [] audio_offset = glm_tokenizer.convert_tokens_to_ids('<|audio_0|>') for chunk in response.iter_lines(): token_id = json.loads(chunk)["token_id"] if token_id >= audio_offset: audio_tokens.append(token_id - audio_offset) else: text_tokens.append(token_id) # Generate audio tts_token = torch.tensor(audio_tokens, device=DEVICE).unsqueeze(0) tts_speech, _ = audio_decoder.token2wav(tts_token) # Prepare response buffer = io.BytesIO() torchaudio.save(buffer, tts_speech.cpu(), 22050, format="wav") return AudioResponse( audio_data=base64.b64encode(buffer.getvalue()).decode(), text_transcript=glm_tokenizer.decode(text_tokens, skip_special_tokens=True) ) except Exception as e: raise HTTPException(500, str(e)) if __name__ == "__main__": import uvicorn uvicorn.run(app, host="0.0.0.0", port=8000)