import traceback from config.logger import setup_logging import json import asyncio import time from core.utils.util import remove_punctuation_and_length, get_string_no_punctuation_or_emoji TAG = __name__ logger = setup_logging() async def isLLMWantToFinish(last_text): _, last_text_without_punctuation = remove_punctuation_and_length(last_text) if "再见" in last_text_without_punctuation or "拜拜" in last_text_without_punctuation: return True return False async def sendAudioMessageStream(conn, audios_queue, text, text_index=0): # 发送句子开始消息 if text_index == conn.tts_first_text_index: logger.bind(tag=TAG).info(f"发送第一段语音: {text}") await send_tts_message(conn, "sentence_start", text) # 初始化流控参数 frame_duration = 60 # 毫秒 start_time = time.time() # 使用高精度计时器 play_position = 0 # 已播放的时长(毫秒) time_out_stop = False while True: try: start_get_queue = time.time() # 尝试获取数据,如果没有数据,则等待一小段时间再试 audio_data_chunke = None try: audio_data_chunke = audios_queue.get(timeout=5) # 设置超时为1秒 except Exception as e: # 如果超时,继续等待 logger.bind(tag=TAG).error(f"获取队列超时~{e}") audio_opus_datas = audio_data_chunke.get('data') if audio_data_chunke else None duration = audio_data_chunke.get('duration') if audio_data_chunke else 0 if audio_data_chunke: start_time = time.time() # 检查是否超过 5 秒没有数据 if time.time() - start_time > 15: logger.bind(tag=TAG).error("超过15秒没有数据,退出。") break if audio_data_chunke and audio_data_chunke.get("end", True): break if audio_opus_datas: queue_duration = time.time() - start_get_queue last_duration = conn.tts_duration - queue_duration if last_duration <= 0: last_duration = 0 conn.tts_duration = duration + last_duration for opus_packet in audio_opus_datas: await conn.websocket.send(opus_packet) start_time = time.time() # 更新获取数据的时间 except Exception as e: logger.bind(tag=TAG).error(f"发生错误: {e}") traceback.print_exc() # 打印错误堆栈 await send_tts_message(conn, "sentence_end", text) print(f'{text_index}-{conn.tts_last_text_index}') # 发送结束消息(如果是最后一个文本) if conn.llm_finish_task and text_index == conn.tts_last_text_index: if conn.tts_duration and conn.tts_duration > 0: await asyncio.sleep(conn.tts_duration) await send_tts_message(conn, 'stop', None) if await isLLMWantToFinish(text): await conn.close() async def sendAudioMessage(conn, audios, text, text_index=0): # 发送句子开始消息 if text_index == conn.tts_first_text_index: logger.bind(tag=TAG).info(f"发送第一段语音: {text}") await send_tts_message(conn, "sentence_start", text) # 初始化流控参数 frame_duration = 60 # 毫秒 start_time = time.perf_counter() # 使用高精度计时器 play_position = 0 # 已播放的时长(毫秒) for opus_packet in audios: if conn.client_abort: return # 计算当前包的预期发送时间 expected_time = start_time + (play_position / 1000) current_time = time.perf_counter() # 等待直到预期时间 delay = expected_time - current_time if delay > 0: await asyncio.sleep(delay) # 发送音频包 await conn.websocket.send(opus_packet) play_position += frame_duration # 更新播放位置 await send_tts_message(conn, "sentence_end", text) # 发送结束消息(如果是最后一个文本) if conn.llm_finish_task and text_index == conn.tts_last_text_index: await send_tts_message(conn, 'stop', None) if await isLLMWantToFinish(text): await conn.close() async def send_tts_message(conn, state, text=None): """发送 TTS 状态消息""" message = { "type": "tts", "state": state, "session_id": conn.session_id } if text is not None: message["text"] = text await conn.websocket.send(json.dumps(message)) if state == "stop": conn.clearSpeakStatus() async def send_stt_message(conn, text): """发送 STT 状态消息""" stt_text = get_string_no_punctuation_or_emoji(text) await conn.websocket.send(json.dumps({ "type": "stt", "text": stt_text, "session_id": conn.session_id} )) await conn.websocket.send( json.dumps({ "type": "llm", "text": "😊", "emotion": "happy", "session_id": conn.session_id} )) await send_tts_message(conn, "start")