xiaozhi-esp32-server/main/xiaozhi-server/core/handle/sendAudioHandle.py

import json
import asyncio
import time
from core.providers.tts.dto.dto import SentenceType
from core.utils import textUtils

TAG = __name__


async def sendAudioMessage(conn, sentenceType, audios, text):
    # 发送句子开始消息
    conn.logger.bind(tag=TAG).info(f"发送音频消息: {sentenceType}, {text}")

    pre_buffer = False
    if conn.tts.tts_audio_first_sentence:
        conn.logger.bind(tag=TAG).info(f"发送第一段语音: {text}")
        conn.tts.tts_audio_first_sentence = False
        pre_buffer = True
        await send_tts_message(conn, "start", None)

    if sentenceType == SentenceType.FIRST:
        await send_tts_message(conn, "sentence_start", text)

    await sendAudio(conn, audios, pre_buffer)

    # 发送结束消息（如果是最后一个文本）
    if conn.llm_finish_task and sentenceType == SentenceType.LAST:
        await send_tts_message(conn, "stop", None)
        conn.client_is_speaking = False
        if conn.close_after_chat:
            await conn.close()


# 播放音频
async def sendAudio(conn, audios, pre_buffer=True):
    if audios is None:
        return
    # 如果audios不是opus数组，则不需要进行遍历，可以直接发送;这里需要进行流控管理，防止发送过快引发客户端溢出
    if isinstance(audios ,bytes):
        await conn.websocket.send(audios)
    else:
        if audios is None or len(audios) == 0:
            return
        # 流控参数优化
        frame_duration = 60  # 帧时长（毫秒），匹配 Opus 编码
        start_time = time.perf_counter()
        play_position = 0
        # 仅当第一句话时执行预缓冲
        if pre_buffer:
            pre_buffer_frames = min(3, len(audios))
            for i in range(pre_buffer_frames):
                await conn.websocket.send(audios[i])
            remaining_audios = audios[pre_buffer_frames:]
        else:
            remaining_audios = audios

        # 播放剩余音频帧
        for opus_packet in remaining_audios:
            if conn.client_abort:
                break

            # 重置没有声音的状态
            conn.last_activity_time = time.time() * 1000

            # 计算预期发送时间
            expected_time = start_time + (play_position / 1000)
            current_time = time.perf_counter()
            delay = expected_time - current_time
            if delay > 0:
                await asyncio.sleep(delay)

            await conn.websocket.send(opus_packet)

            play_position += frame_duration


async def send_tts_message(conn, state, text=None):
    """发送 TTS 状态消息"""
    if text is None and state == "sentence_start":
        return
    message = {"type": "tts", "state": state, "session_id": conn.session_id}
    if text is not None:
        message["text"] = textUtils.check_emoji(text)

    # TTS播放结束
    if state == "stop":
        # 播放提示音
        tts_notify = conn.config.get("enable_stop_tts_notify", False)
        if tts_notify:
            stop_tts_notify_voice = conn.config.get(
                "stop_tts_notify_voice", "config/assets/tts_notify.mp3"
            )
            audios, _ = conn.tts.audio_to_opus_data(stop_tts_notify_voice)
            await sendAudio(conn, audios)
        # 清除服务端讲话状态
        conn.clearSpeakStatus()

    # 发送消息到客户端
    await conn.websocket.send(json.dumps(message))


async def send_stt_message(conn, text):
    end_prompt_str = conn.config.get("end_prompt", {}).get("prompt")
    if end_prompt_str and end_prompt_str == text:
        await send_tts_message(conn, "start")
        return

    """发送 STT 状态消息"""
    
    # 解析JSON格式，提取实际的用户说话内容
    display_text = text
    try:
        # 尝试解析JSON格式
        if text.strip().startswith('{') and text.strip().endswith('}'):
            parsed_data = json.loads(text)
            if isinstance(parsed_data, dict) and "content" in parsed_data:
                # 如果是包含说话人信息的JSON格式，只显示content部分
                display_text = parsed_data["content"]
                # 保存说话人信息到conn对象
                if "speaker" in parsed_data:
                    conn.current_speaker = parsed_data["speaker"]
    except (json.JSONDecodeError, TypeError):
        # 如果不是JSON格式，直接使用原始文本
        display_text = text
    stt_text = textUtils.get_string_no_punctuation_or_emoji(display_text)
    await conn.websocket.send(
        json.dumps({"type": "stt", "text": stt_text, "session_id": conn.session_id})
    )
    conn.client_is_speaking = True
    await send_tts_message(conn, "start")