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import json
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import asyncio
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import time
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from core.providers.tts.dto.dto import SentenceType
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from core.utils.util import get_string_no_punctuation_or_emoji, analyze_emotion
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from loguru import logger
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TAG = __name__
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emoji_map = {
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"neutral": "😶",
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"happy": "🙂",
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"laughing": "😆",
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"funny": "😂",
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"sad": "😔",
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"angry": "😠",
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"crying": "😭",
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"loving": "😍",
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"embarrassed": "😳",
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"surprised": "😲",
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"shocked": "😱",
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"thinking": "🤔",
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"winking": "😉",
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"cool": "😎",
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"relaxed": "😌",
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"delicious": "🤤",
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"kissy": "😘",
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"confident": "😏",
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"sleepy": "😴",
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"silly": "😜",
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"confused": "🙄",
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}
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async def sendAudioMessage(conn, sentenceType, audios, text):
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# 发送句子开始消息
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conn.logger.bind(tag=TAG).info(f"发送音频消息: {sentenceType}, {text}")
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if text is not None:
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emotion = analyze_emotion(text)
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emoji = emoji_map.get(emotion, "🙂") # 默认使用笑脸
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await conn.websocket.send(
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json.dumps(
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{
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"type": "llm",
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"text": emoji,
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"emotion": emotion,
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"session_id": conn.session_id,
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}
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)
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)
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pre_buffer = False
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if conn.tts.tts_audio_first_sentence and text is not None:
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conn.logger.bind(tag=TAG).info(f"发送第一段语音: {text}")
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conn.tts.tts_audio_first_sentence = False
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pre_buffer = True
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await send_tts_message(conn, "sentence_start", text)
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await sendAudio(conn, audios, pre_buffer)
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await send_tts_message(conn, "sentence_end", text)
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# 发送结束消息（如果是最后一个文本）
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if conn.llm_finish_task and sentenceType == SentenceType.LAST:
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await send_tts_message(conn, "stop", None)
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conn.client_is_speaking = False
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if conn.close_after_chat:
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await conn.close()
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# 播放音频
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async def sendAudio(conn, audios, pre_buffer=True):
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if audios is None or len(audios) == 0:
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return
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# 流控参数优化
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frame_duration = 60 # 帧时长（毫秒），匹配 Opus 编码
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start_time = time.perf_counter()
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play_position = 0
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# 仅当第一句话时执行预缓冲
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if pre_buffer:
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pre_buffer_frames = min(3, len(audios))
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for i in range(pre_buffer_frames):
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await conn.websocket.send(audios[i])
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remaining_audios = audios[pre_buffer_frames:]
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else:
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remaining_audios = audios
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# 播放剩余音频帧
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for opus_packet in remaining_audios:
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if conn.client_abort:
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break
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# 重置没有声音的状态
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conn.last_activity_time = time.time() * 1000
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# 计算预期发送时间
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expected_time = start_time + (play_position / 1000)
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current_time = time.perf_counter()
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delay = expected_time - current_time
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if delay > 0:
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await asyncio.sleep(delay)
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await conn.websocket.send(opus_packet)
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play_position += frame_duration
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async def send_tts_message(conn, state, text=None):
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"""发送 TTS 状态消息"""
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message = {"type": "tts", "state": state, "session_id": conn.session_id}
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if text is not None:
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message["text"] = text
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# TTS播放结束
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if state == "stop":
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# 播放提示音
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tts_notify = conn.config.get("enable_stop_tts_notify", False)
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if tts_notify:
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stop_tts_notify_voice = conn.config.get(
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"stop_tts_notify_voice", "config/assets/tts_notify.mp3"
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)
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audios, _ = conn.tts.audio_to_opus_data(stop_tts_notify_voice)
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await sendAudio(conn, audios)
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# 清除服务端讲话状态
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conn.clearSpeakStatus()
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# 发送消息到客户端
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await conn.websocket.send(json.dumps(message))
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async def send_stt_message(conn, text):
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end_prompt_str = conn.config.get("end_prompt", {}).get("prompt")
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if end_prompt_str and end_prompt_str == text:
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await send_tts_message(conn, "start")
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return
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"""发送 STT 状态消息"""
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# 解析JSON格式，提取实际的用户说话内容
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display_text = text
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try:
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# 尝试解析JSON格式
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if text.strip().startswith('{') and text.strip().endswith('}'):
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parsed_data = json.loads(text)
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if isinstance(parsed_data, dict) and "content" in parsed_data:
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# 如果是包含说话人信息的JSON格式，只显示content部分
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display_text = parsed_data["content"]
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except (json.JSONDecodeError, TypeError):
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# 如果不是JSON格式，直接使用原始文本
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display_text = text
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stt_text = get_string_no_punctuation_or_emoji(display_text)
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await conn.websocket.send(
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json.dumps({"type": "stt", "text": stt_text, "session_id": conn.session_id})
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)
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conn.client_is_speaking = True
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await send_tts_message(conn, "start")
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