更新持续流式聊天接口和语音电话接口，使得其可以正常处理完所有数据再退出
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@ -297,95 +297,64 @@ async def streaming_chat_temporary_handler(ws: WebSocket, db, redis):
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#---------------------------------------------------------------------------------------------------
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# 持续流式聊天
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async def streaming_chat_lasting_handler(ws, db, redis):
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print("Websocket连接成功")
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while True:
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#------------------------------------------持续流式聊天----------------------------------------------
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#处理用户输入
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async def scl_user_input_handler(ws,user_input_q,llm_input_q,future_session_id,future_response_type,input_finished_event):
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logger.debug("用户输入处理函数启动")
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is_future_done = False
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try:
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print("等待接受")
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data = await asyncio.wait_for(ws.receive_text(), timeout=60)
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data_json = json.loads(data)
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if data_json["is_close"]:
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close_message = {"type": "close", "code": 200, "msg": ""}
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await ws.send_text(json.dumps(close_message, ensure_ascii=False))
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print("连接关闭")
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await asyncio.sleep(0.5)
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await ws.close()
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return;
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except asyncio.TimeoutError:
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print("连接超时")
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await ws.close()
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return;
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current_message = "" # 用于存储用户消息
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response_type = RESPONSE_TEXT # 用于获取返回类型
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session_id = ""
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if Config.STRAM_CHAT.ASR == "LOCAL":
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try:
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while True:
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if data_json["text"]: # 若文字输入不为空，则表示该输入为文字输入
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if data_json["meta_info"]["voice_synthesize"]:
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response_type = RESPONSE_AUDIO # 查看voice_synthesize判断返回类型
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session_id = data_json["meta_info"]["session_id"]
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current_message = data_json['text']
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while not input_finished_event.is_set():
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scl_data_json = json.loads(await ws.receive_text())
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if scl_data_json['is_close']:
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input_finished_event.set()
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break
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if not data_json['meta_info']['is_end']: # 还在发
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asr_result = asr.streaming_recognize(data_json["audio"])
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current_message += ''.join(asr_result['text'])
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else: # 发完了
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asr_result = asr.streaming_recognize(data_json["audio"], is_end=True)
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session_id = data_json["meta_info"]["session_id"]
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current_message += ''.join(asr_result['text'])
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if data_json["meta_info"]["voice_synthesize"]:
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response_type = RESPONSE_AUDIO # 查看voice_synthesize判断返回类型
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break
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data_json = json.loads(await ws.receive_text())
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except Exception as e:
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error_info = f"接收用户消息错误: {str(e)}"
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error_message = {"type": "error", "code": "500", "msg": error_info}
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logger.error(error_info)
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await ws.send_text(json.dumps(error_message, ensure_ascii=False))
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await ws.close()
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return
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elif Config.STRAM_CHAT.ASR == "REMOTE":
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error_info = f"远程ASR服务暂未开通"
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error_message = {"type": "error", "code": "500", "msg": error_info}
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logger.error(error_info)
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await ws.send_text(json.dumps(error_message, ensure_ascii=False))
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await ws.close()
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return
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print(f"接收到用户消息: {current_message}")
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# 查询Session记录
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session_content_str = ""
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if redis.exists(session_id):
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session_content_str = redis.get(session_id)
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if not is_future_done:
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future_session_id.set_result(scl_data_json['meta_info']['session_id'])
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if scl_data_json['meta_info']['voice_synthesize']:
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future_response_type.set_result(RESPONSE_AUDIO)
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else:
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session_db = db.query(Session).filter(Session.id == session_id).first()
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if not session_db:
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error_info = f"未找到session记录: {str(e)}"
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error_message = {"type": "error", "code": 500, "msg": error_info}
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logger.error(error_info)
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await ws.send_text(json.dumps(error_message, ensure_ascii=False))
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await ws.close()
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return
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session_content_str = session_db.content
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future_response_type.set_result(RESPONSE_TEXT)
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is_future_done = True
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if scl_data_json['text']:
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await llm_input_q.put(scl_data_json['text'])
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if scl_data_json['meta_info']['is_end']:
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user_input_frame = {"audio": scl_data_json['audio'], "is_end": True}
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await user_input_q.put(user_input_frame)
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user_input_frame = {"audio": scl_data_json['audio'], "is_end": False}
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await user_input_q.put(user_input_frame)
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except KeyError as ke:
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if 'state' in scl_data_json and 'method' in scl_data_json:
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logger.debug("收到心跳包")
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session_content = json.loads(session_content_str)
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llm_info = json.loads(session_content["llm_info"])
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tts_info = json.loads(session_content["tts_info"])
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user_info = json.loads(session_content["user_info"])
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#语音识别
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async def scl_asr_handler(user_input_q,llm_input_q,input_finished_event,asr_finished_event):
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logger.debug("语音识别函数启动")
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current_message = ""
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while not (input_finished_event.is_set() and user_input_q.empty()):
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aduio_frame = await user_input_q.get()
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if aduio_frame['is_end']:
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asr_result = asr.streaming_recognize(aduio_frame['audio'], is_end=True)
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current_message += ''.join(asr_result['text'])
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await llm_input_q.put(current_message)
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logger.debug(f"接收到用户消息: {current_message}")
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else:
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asr_result = asr.streaming_recognize(aduio_frame['audio'])
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current_message += ''.join(asr_result['text'])
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asr_finished_event.set()
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#大模型调用
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async def scl_llm_handler(session_id,llm_info,db,redis,llm_input_q,llm_response_q,asr_finished_event,llm_finished_event):
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logger.debug("llm调用函数启动")
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while not (asr_finished_event.is_set() and llm_input_q.empty()):
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session_content = get_session_content(session_id,redis,db)
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messages = json.loads(session_content["messages"])
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current_message = await llm_input_q.get()
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messages.append({'role': 'user', "content": current_message})
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token_count = session_content["token"]
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try:
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payload = json.dumps({
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"model": llm_info["model"],
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"stream": True,
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"messages": messages,
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"tool_choice": "auto",
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"max_tokens": 10000,
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"temperature": llm_info["temperature"],
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"top_p": llm_info["top_p"]
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@ -395,112 +364,121 @@ async def streaming_chat_lasting_handler(ws, db, redis):
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'Content-Type': 'application/json'
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}
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response = requests.request("POST", Config.MINIMAX_LLM.URL, headers=headers, data=payload, stream=True)
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except Exception as e:
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error_info = f"发送信息给大模型时发生错误: {str(e)}"
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error_message = {"type": "error", "code": 500, "msg": error_info}
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logger.error(error_info)
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await ws.send_text(json.dumps(error_message, ensure_ascii=False))
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await ws.close()
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return
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def split_string_with_punctuation(text):
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punctuations = "！？。"
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result = []
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current_sentence = ""
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for char in text:
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current_sentence += char
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if char in punctuations:
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result.append(current_sentence)
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current_sentence = ""
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# 判断最后一个字符是否为标点符号
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if current_sentence and current_sentence[-1] not in punctuations:
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# 如果最后一段不以标点符号结尾，则加入拆分数组
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result.append(current_sentence)
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return result
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llm_response = ""
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response_buf = ""
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try:
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if Config.STRAM_CHAT.TTS == "LOCAL":
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if response.status_code == 200:
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for chunk in response.iter_lines():
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if chunk:
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if response_type == RESPONSE_AUDIO:
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chunk_data = parseChunkDelta(chunk)
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llm_response += chunk_data
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response_buf += chunk_data
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split_buf = split_string_with_punctuation(response_buf)
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response_buf = ""
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if len(split_buf) != 0:
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for sentence in split_buf:
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sr, audio = tts.synthesize(sentence, tts_info["language"], tts_info["speaker_id"], tts_info["noise_scale"], tts_info["noise_scale_w"], tts_info["length_scale"], return_bytes=True)
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text_response = {"type": "text", "code": 200, "msg": sentence}
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await ws.send_text(json.dumps(text_response, ensure_ascii=False)) # 返回文本数据
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await ws.send_bytes(audio) # 返回音频二进制流数据
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if response_type == RESPONSE_TEXT:
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chunk_data = parseChunkDelta(chunk)
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llm_response += chunk_data
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text_response = {"type": "text", "code": 200, "msg": chunk_data}
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await ws.send_text(json.dumps(text_response, ensure_ascii=False)) # 返回文本数据
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llm_frame = {"message": chunk_data, "is_end": False}
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await llm_response_q.put(llm_frame)
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llm_frame = {"message": "", "is_end": True}
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await llm_response_q.put(llm_frame)
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llm_finished_event.set()
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elif Config.STRAM_CHAT.TTS == "REMOTE":
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error_info = f"暂不支持远程音频合成"
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error_message = {"type": "error", "code": 500, "msg": error_info}
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logger.error(error_info)
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await ws.send_text(json.dumps(error_message, ensure_ascii=False))
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await ws.close()
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return
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end_response = {"type": "end", "code": 200, "msg": ""}
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await ws.send_text(json.dumps(end_response, ensure_ascii=False)) # 单次返回结束
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print(f"llm消息: {llm_response}")
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except Exception as e:
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error_info = f"音频合成与向前端返回时错误: {str(e)}"
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error_message = {"type": "error", "code": 500, "msg": error_info}
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logger.error(error_info)
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await ws.send_text(json.dumps(error_message, ensure_ascii=False))
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await ws.close()
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return
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try:
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#大模型返回断句
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async def scl_llm_response_handler(session_id,redis,db,llm_response_q,split_result_q,llm_finished_event,split_finished_event):
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logger.debug("llm返回处理函数启动")
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llm_response = ""
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current_sentence = ""
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is_first = True
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while not (llm_finished_event.is_set() and llm_response_q.empty()):
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llm_frame = await llm_response_q.get()
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llm_response += llm_frame['message']
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sentences,current_sentence,is_first = split_string_with_punctuation(current_sentence,llm_frame['message'],is_first)
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for sentence in sentences:
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sentence_frame = {"message": sentence, "is_end": False}
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await split_result_q.put(sentence_frame)
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if llm_frame['is_end']:
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sentence_frame = {"message": "", "is_end": True}
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await split_result_q.put(sentence_frame)
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is_first = True
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session_content = get_session_content(session_id,redis,db)
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messages = json.loads(session_content["messages"])
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messages.append({'role': 'assistant', "content": llm_response})
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token_count += len(llm_response)
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session_content["messages"] = json.dumps(messages, ensure_ascii=False)
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session_content["token"] = token_count
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redis.set(session_id, json.dumps(session_content, ensure_ascii=False))
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except Exception as e:
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error_info = f"更新session时错误: {str(e)}"
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error_message = {"type": "error", "code": 500, "msg": error_info}
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logger.error(error_info)
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await ws.send_text(json.dumps(error_message, ensure_ascii=False))
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session_content["messages"] = json.dumps(messages,ensure_ascii=False) #更新对话
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redis.set(session_id,json.dumps(session_content,ensure_ascii=False)) #更新session
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logger.debug(f"llm返回结果: {llm_response}")
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llm_response = ""
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current_sentence = ""
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split_finished_event.set()
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#文本返回及语音合成
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async def scl_response_handler(ws,tts_info,response_type,split_result_q,split_finished_event,chat_finish_event):
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logger.debug("返回处理函数启动")
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while not (split_finished_event.is_set() and split_result_q.empty()):
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sentence_frame = await split_result_q.get()
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sentence = sentence_frame['message']
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if sentence_frame['is_end']:
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await ws.send_text(json.dumps({"type": "end", "code": 200, "msg": ""}, ensure_ascii=False))
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continue
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if response_type == RESPONSE_TEXT:
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response_message = {"type": "text", "code":200, "msg": sentence}
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await ws.send_text(json.dumps(response_message, ensure_ascii=False))
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elif response_type == RESPONSE_AUDIO:
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sr,audio = tts.synthesize(sentence, tts_info["speaker_id"], tts_info["language"], tts_info["noise_scale"], tts_info["noise_scale_w"], tts_info["length_scale"],return_bytes=True)
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response_message = {"type": "text", "code":200, "msg": sentence}
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await ws.send_bytes(audio)
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await ws.send_text(json.dumps(response_message, ensure_ascii=False))
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logger.debug(f"websocket返回: {sentence}")
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chat_finish_event.set()
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async def streaming_chat_lasting_handler(ws,db,redis):
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logger.debug("streaming chat lasting websocket 连接建立")
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user_input_q = asyncio.Queue() # 用于存储用户输入
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llm_input_q = asyncio.Queue() # 用于存储llm输入
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llm_response_q = asyncio.Queue() # 用于存储llm输出
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split_result_q = asyncio.Queue() # 用于存储llm返回后断句输出
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input_finished_event = asyncio.Event()
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asr_finished_event = asyncio.Event()
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llm_finished_event = asyncio.Event()
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split_finished_event = asyncio.Event()
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chat_finish_event = asyncio.Event()
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future_session_id = asyncio.Future()
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future_response_type = asyncio.Future()
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asyncio.create_task(scl_user_input_handler(ws,user_input_q,llm_input_q,future_session_id,future_response_type,input_finished_event))
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asyncio.create_task(scl_asr_handler(user_input_q,llm_input_q,input_finished_event,asr_finished_event))
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session_id = await future_session_id #获取session_id
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response_type = await future_response_type #获取返回类型
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tts_info = json.loads(get_session_content(session_id,redis,db)["tts_info"])
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llm_info = json.loads(get_session_content(session_id,redis,db)["llm_info"])
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asyncio.create_task(scl_llm_handler(session_id,llm_info,db,redis,llm_input_q,llm_response_q,asr_finished_event,llm_finished_event))
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asyncio.create_task(scl_llm_response_handler(session_id,redis,db,llm_response_q,split_result_q,llm_finished_event,split_finished_event))
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asyncio.create_task(scl_response_handler(ws,tts_info,response_type,split_result_q,split_finished_event,chat_finish_event))
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while not chat_finish_event.is_set():
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await asyncio.sleep(3)
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await ws.send_text(json.dumps({"type": "close", "code": 200, "msg": ""}, ensure_ascii=False))
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await ws.close()
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return
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print("处理完毕")
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logger.debug("streaming chat lasting websocket 连接断开")
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#---------------------------------------------------------------------------------------------------
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#--------------------------------语音通话接口--------------------------------------
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#音频数据生产函数
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async def voice_call_audio_producer(ws,audio_queue,future,shutdown_event):
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async def voice_call_audio_producer(ws,audio_queue,future,input_finished_event):
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logger.debug("音频数据生产函数启动")
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is_future_done = False
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while not shutdown_event.is_set():
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while not input_finished_event.is_set():
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voice_call_data_json = json.loads(await ws.receive_text())
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if not is_future_done: #在第一次循环中读取session_id
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future.set_result(voice_call_data_json['meta_info']['session_id'])
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is_future_done = True
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if voice_call_data_json["is_close"]:
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shutdown_event.set()
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input_finished_event.set()
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break
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else:
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await audio_queue.put(voice_call_data_json["audio"]) #将音频数据存入audio_q
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#音频数据消费函数
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async def voice_call_audio_consumer(audio_q,asr_result_q,shutdown_event):
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async def voice_call_audio_consumer(audio_q,asr_result_q,input_finished_event,asr_finished_event):
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logger.debug("音频数据消费者函数启动")
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vad = VAD()
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current_message = ""
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vad_count = 0
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while not (shutdown_event.is_set() and audio_q.empty()):
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while not (input_finished_event.is_set() and audio_q.empty()):
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audio_data = await audio_q.get()
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if vad.is_speech(audio_data):
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if vad_count > 0:
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@ -516,11 +494,12 @@ async def voice_call_audio_consumer(audio_q,asr_result_q,shutdown_event):
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await asr_result_q.put(current_message)
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current_message = ""
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vad_count = 0
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asr_finished_event.set()
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#asr结果消费以及llm返回生产函数
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async def voice_call_llm_handler(session_id,llm_info,redis,db,asr_result_q,llm_response_q,shutdown_event):
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async def voice_call_llm_handler(session_id,llm_info,redis,db,asr_result_q,llm_response_q,asr_finished_event,llm_finished_event):
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logger.debug("asr结果消费以及llm返回生产函数启动")
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while not (shutdown_event.is_set() and asr_result_q.empty()):
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while not (asr_finished_event.is_set() and asr_result_q.empty()):
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session_content = get_session_content(session_id,redis,db)
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messages = json.loads(session_content["messages"])
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current_message = await asr_result_q.get()
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|
@ -547,14 +526,15 @@ async def voice_call_llm_handler(session_id,llm_info,redis,db,asr_result_q,llm_r
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|||
await llm_response_q.put(llm_frame)
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llm_frame = {'message':"",'is_end':True}
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await llm_response_q.put(llm_frame)
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llm_finished_event.set()
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|
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#llm结果返回函数
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async def voice_call_llm_response_consumer(session_id,redis,db,llm_response_q,split_result_q,shutdown_event):
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||||
async def voice_call_llm_response_consumer(session_id,redis,db,llm_response_q,split_result_q,llm_finished_event,split_finished_event):
|
||||
logger.debug("llm结果返回函数启动")
|
||||
llm_response = ""
|
||||
current_sentence = ""
|
||||
is_first = True
|
||||
while not (shutdown_event.is_set() and llm_response_q.empty()):
|
||||
while not (llm_finished_event.is_set() and llm_response_q.empty()):
|
||||
llm_frame = await llm_response_q.get()
|
||||
llm_response += llm_frame['message']
|
||||
sentences,current_sentence,is_first = split_string_with_punctuation(current_sentence,llm_frame['message'],is_first)
|
||||
|
|
@ -570,11 +550,12 @@ async def voice_call_llm_response_consumer(session_id,redis,db,llm_response_q,sp
|
|||
logger.debug(f"llm返回结果: {llm_response}")
|
||||
llm_response = ""
|
||||
current_sentence = ""
|
||||
split_finished_event.set()
|
||||
|
||||
#语音合成及返回函数
|
||||
async def voice_call_tts_handler(ws,tts_info,split_result_q,shutdown_event):
|
||||
async def voice_call_tts_handler(ws,tts_info,split_result_q,split_finished_event,voice_call_end_event):
|
||||
logger.debug("语音合成及返回函数启动")
|
||||
while not (shutdown_event.is_set() and split_result_q.empty()):
|
||||
while not (split_finished_event.is_set() and split_result_q.empty()):
|
||||
sentence = await split_result_q.get()
|
||||
sr,audio = tts.synthesize(sentence, tts_info["language"], tts_info["speaker_id"], tts_info["noise_scale"], tts_info["noise_scale_w"], tts_info["length_scale"], return_bytes=True)
|
||||
text_response = {"type": "text", "code": 200, "msg": sentence}
|
||||
|
|
@ -583,6 +564,7 @@ async def voice_call_tts_handler(ws,tts_info,split_result_q,shutdown_event):
|
|||
logger.debug(f"websocket返回:{sentence}")
|
||||
asyncio.sleep(0.5)
|
||||
await ws.close()
|
||||
voice_call_end_event.set()
|
||||
|
||||
|
||||
async def voice_call_handler(ws, db, redis):
|
||||
|
|
@ -592,22 +574,26 @@ async def voice_call_handler(ws, db, redis):
|
|||
llm_response_q = asyncio.Queue()
|
||||
split_result_q = asyncio.Queue()
|
||||
|
||||
shutdown_event = asyncio.Event()
|
||||
input_finished_event = asyncio.Event()
|
||||
asr_finished_event = asyncio.Event()
|
||||
llm_finished_event = asyncio.Event()
|
||||
split_finished_event = asyncio.Event()
|
||||
voice_call_end_event = asyncio.Event()
|
||||
future = asyncio.Future()
|
||||
asyncio.create_task(voice_call_audio_producer(ws,audio_q,future,shutdown_event)) #创建音频数据生产者
|
||||
asyncio.create_task(voice_call_audio_consumer(audio_q,asr_result_q,shutdown_event)) #创建音频数据消费者
|
||||
asyncio.create_task(voice_call_audio_producer(ws,audio_q,future,input_finished_event)) #创建音频数据生产者
|
||||
asyncio.create_task(voice_call_audio_consumer(audio_q,asr_result_q,input_finished_event,asr_finished_event)) #创建音频数据消费者
|
||||
|
||||
#获取session内容
|
||||
session_id = await future #获取session_id
|
||||
tts_info = json.loads(get_session_content(session_id,redis,db)["tts_info"])
|
||||
llm_info = json.loads(get_session_content(session_id,redis,db)["llm_info"])
|
||||
|
||||
asyncio.create_task(voice_call_llm_handler(session_id,llm_info,redis,db,asr_result_q,llm_response_q,shutdown_event)) #创建llm处理者
|
||||
asyncio.create_task(voice_call_llm_response_consumer(session_id,redis,db,llm_response_q,split_result_q,shutdown_event)) #创建llm断句结果
|
||||
asyncio.create_task(voice_call_tts_handler(ws,tts_info,split_result_q,shutdown_event)) #返回tts音频结果
|
||||
asyncio.create_task(voice_call_llm_handler(session_id,llm_info,redis,db,asr_result_q,llm_response_q,asr_finished_event,llm_finished_event)) #创建llm处理者
|
||||
asyncio.create_task(voice_call_llm_response_consumer(session_id,redis,db,llm_response_q,split_result_q,llm_finished_event,split_finished_event)) #创建llm断句结果
|
||||
asyncio.create_task(voice_call_tts_handler(ws,tts_info,split_result_q,split_finished_event,voice_call_end_event)) #返回tts音频结果
|
||||
|
||||
while not shutdown_event.is_set():
|
||||
await asyncio.sleep(5)
|
||||
while not voice_call_end_event.is_set():
|
||||
await asyncio.sleep(3)
|
||||
await ws.close()
|
||||
logger.debug("voice_call websocket 连接断开")
|
||||
#------------------------------------------------------------------------------------------
|
||||
Loading…
Reference in New Issue