""" ALI_KEY = "sk-a47a3fb5f4a94f66bbaf713779101c75" class QwenTTS: def __init__(self, key,format="mp3",sample_rate=44100, model_name="cosyvoice-v1/longxiaochun"): import dashscope print("---begin--init dialog_service QwenTTS--") # cyx self.model_name = model_name dashscope.api_key = key self.synthesizer = None self.callback = None self.is_cosyvoice = False self.voice = "" self.format = format self.sample_rate = sample_rate if '/' in model_name: parts = model_name.split('/', 1) # 返回分离后的两个字符串parts[0], parts[1] if parts[0] == 'cosyvoice-v1': self.is_cosyvoice = True self.voice = parts[1] def init_streaming_call(self, audio_call_back): from dashscope.api_entities.dashscope_response import SpeechSynthesisResponse # cyx 2025 01 19 测试cosyvoice 使用tts_v2 版本 from dashscope.audio.tts_v2 import ResultCallback, SpeechSynthesizer, AudioFormat # , SpeechSynthesisResult from dashscope.audio.tts import SpeechSynthesisResult from collections import deque print(f"--QwenTTS--tts_stream begin-- {self.is_cosyvoice} {self.voice}") # cyx class Callback_v2(ResultCallback): def __init__(self) -> None: self.dque = deque() def _run(self): while True: if not self.dque: time.sleep(0) continue val = self.dque.popleft() if val: yield val else: break def on_open(self): logging.info("Qwen tts open") pass def on_complete(self): self.dque.append(None) def on_error(self, response: SpeechSynthesisResponse): print("Qwen tts error", str(response)) raise RuntimeError(str(response)) def on_close(self): # print("---Qwen call back close") # cyx logging.info("Qwen tts close") pass # canceled for test 语音大模型CosyVoice #def on_event(self, result: SpeechSynthesisResult): # if result.get_audio_frame() is not None: # self.dque.append(result.get_audio_frame()) # def on_event(self, message): # print(f"recv speech synthsis message {message}") pass # 以下适合语音大模型CosyVoice def on_data(self, data: bytes) -> None: if len(data) > 0: #self.dque.append(data) if audio_call_back: audio_call_back(data) # -------------------------- # text = self.normalize_text(text) try: self.callback = Callback_v2() format =self.get_audio_format(self.format,self.sample_rate) self.synthesizer = SpeechSynthesizer( model='cosyvoice-v1', # voice="longyuan", #"longfei", voice=self.voice, callback=self.callback, format=format ) #self.synthesizer.call(text) except Exception as e: print(f"---dale---20 error {e}") # cyx # ----------------------------------- # print(f"---Qwen return data {num_tokens_from_string(text)}") # yield num_tokens_from_string(text) except Exception as e: raise RuntimeError(f"**ERROR**: {e}") def get_audio_format(self, format: str, sample_rate: int): # 动态获取音频格式 from dashscope.audio.tts_v2 import AudioFormat logging.info(f"QwenTTS--get_audio_format-- {format} {sample_rate}") format_map = { (8000, 'mp3'): AudioFormat.MP3_8000HZ_MONO_128KBPS, (8000, 'pcm'): AudioFormat.PCM_8000HZ_MONO_16BIT, (8000, 'wav'): AudioFormat.WAV_8000HZ_MONO_16BIT, (16000, 'pcm'): AudioFormat.PCM_16000HZ_MONO_16BIT, (22050, 'mp3'): AudioFormat.MP3_22050HZ_MONO_256KBPS, (22050, 'pcm'): AudioFormat.PCM_22050HZ_MONO_16BIT, (22050, 'wav'): AudioFormat.WAV_22050HZ_MONO_16BIT, (44100, 'mp3'): AudioFormat.MP3_44100HZ_MONO_256KBPS, (44100, 'pcm'): AudioFormat.PCM_44100HZ_MONO_16BIT, (44100, 'wav'): AudioFormat.WAV_44100HZ_MONO_16BIT, (48800, 'mp3'): AudioFormat.MP3_48000HZ_MONO_256KBPS, (48800, 'pcm'): AudioFormat.PCM_48000HZ_MONO_16BIT, (48800, 'wav'):AudioFormat.WAV_48000HZ_MONO_16BIT } return format_map.get((sample_rate, format), AudioFormat.MP3_16000HZ_MONO_128KBPS) def streaming_call(self,text): if self.synthesizer: self.synthesizer.streaming_call(text) def end_streaming_call(self): if self.synthesizer: self.synthesizer.streaming_complete() class StreamSessionManager1: def __init__(self): self.sessions = {} # {session_id: {'tts_model': obj, 'buffer': queue, 'task_queue': Queue}} self.lock = threading.Lock() self.executor = ThreadPoolExecutor(max_workers=30) # 固定大小线程池 self.gc_interval = 300 # 5分钟清理一次 5 x 60 300秒 self.gc_tts = 10 # 10s 大模型开始输出文本有可能需要比较久，2025年5 24 从3s->10s self.inited = False self.tts_model = None def create_session(self, tts_model,sample_rate =8000, stream_format='mp3'): session_id = str(uuid.uuid4()) with self.lock: self.sessions[session_id] = { 'tts_model': tts_model, 'buffer': queue.Queue(maxsize=300), # 线程安全队列 'task_queue': queue.Queue(), 'active': True, 'last_active': time.time(), 'audio_chunk_count':0, 'finished': threading.Event(), # 添加事件对象 'sample_rate':sample_rate, 'stream_format':stream_format, "tts_chunk_data_valid":False } # 启动任务处理线程 threading.Thread(target=self._process_tasks, args=(session_id,), daemon=True).start() return session_id def append_text(self, session_id, text): with self.lock: session = self.sessions.get(session_id) if not session: return # 将文本放入任务队列（非阻塞） #logging.info(f"StreamSessionManager append_text {text}") try: session['task_queue'].put(text, block=False) except queue.Full: logging.warning(f"Session {session_id} task queue full") def _process_tasks(self, session_id): #任务处理线程（每个会话独立） session = self.sessions.get(session_id) def audio_call_back(chunk): logging.info(f"audio_call_back {len(chunk)}") if session['stream_format'] == 'wav': if first_chunk: chunk_len = len(chunk) if chunk_len > 2048: session['buffer'].put(audio_fade_in(chunk, 1024)) else: session['buffer'].put(audio_fade_in(chunk, chunk_len)) first_chunk = False else: session['buffer'].put(chunk) else: session['buffer'].put(chunk) session['last_active'] = time.time() session['audio_chunk_count'] = session['audio_chunk_count'] + 1 if session['tts_chunk_data_valid'] is False: session['tts_chunk_data_valid'] = True # 20250510 增加，表示连接TTS后台已经返回，可以通知前端了 while True: if not session or not session['active']: break if not self.inited: self.inited = True self.tts_model = QwenTTS(ALI_KEY, session['stream_format'], session['sample_rate']) self.tts_model.init_streaming_call(audio_call_back) try: #logging.info(f"StreamSessionManager _process_tasks {session['task_queue'].qsize()}") # 合并多个文本块（最多等待50ms） texts = [] while len(texts) < 5: # 最大合并5个文本块 try: text = session['task_queue'].get(timeout=0.1) #logging.info(f"StreamSessionManager _process_tasks --0 {len(texts)}") texts.append(text) except queue.Empty: break if texts: session['last_active'] = time.time() # 如果有处理文本，重置活跃时间 # 提交到线程池处理 future=self.executor.submit( self._generate_audio, session_id, ' '.join(texts) # 合并文本减少请求次数 ) future.result() # 等待转换任务执行完毕 session['last_active'] = time.time() # 会话超时检查 if time.time() - session['last_active'] > self.gc_interval: self.close_session(session_id) break if time.time() - session['last_active'] > self.gc_tts: session['finished'].set() if self.tts_model: self.tts_model.end_streaming_call() break except Exception as e: logging.error(f"Task processing error: {str(e)}") def _generate_audio(self, session_id, text): #实际生成音频（线程池执行） session = self.sessions.get(session_id) if not session: return # logging.info(f"_generate_audio:{text}") first_chunk = True try: self.tts_model.streaming_call(text) except Exception as e: session['buffer'].put(f"ERROR:{str(e)}") logging.info(f"--streaming_call--error {str(e)}") def close_session(self, session_id): with self.lock: if session_id in self.sessions: # 标记会话为不活跃 self.sessions[session_id]['active'] = False # 延迟2秒后清理资源 threading.Timer(1, self._clean_session, args=[session_id]).start() def _clean_session(self, session_id): with self.lock: if session_id in self.sessions: del self.sessions[session_id] def get_session(self, session_id): return self.sessions.get(session_id) """