mirror of
https://github.com/xinnan-tech/xiaozhi-esp32-server.git
synced 2026-07-22 23:23:55 +08:00
145 lines
5.1 KiB
Python
145 lines
5.1 KiB
Python
import traceback
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from config.logger import setup_logging
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import json
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import asyncio
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import time
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from core.utils.util import remove_punctuation_and_length, get_string_no_punctuation_or_emoji
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TAG = __name__
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logger = setup_logging()
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async def isLLMWantToFinish(last_text):
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_, last_text_without_punctuation = remove_punctuation_and_length(last_text)
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if "再见" in last_text_without_punctuation or "拜拜" in last_text_without_punctuation:
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return True
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return False
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async def sendAudioMessageStream(conn, audios_queue, text, text_index=0):
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# 发送句子开始消息
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if text_index == conn.tts_first_text_index:
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logger.bind(tag=TAG).info(f"发送第一段语音: {text}")
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await send_tts_message(conn, "sentence_start", text)
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# 初始化流控参数
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frame_duration = 60 # 毫秒
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start_time = time.time() # 使用高精度计时器
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play_position = 0 # 已播放的时长（毫秒）
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time_out_stop = False
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while True:
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try:
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start_get_queue = time.time()
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# 尝试获取数据，如果没有数据，则等待一小段时间再试
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audio_data_chunke = None
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try:
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audio_data_chunke = audios_queue.get(timeout=5) # 设置超时为1秒
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except Exception as e:
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# 如果超时，继续等待
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logger.bind(tag=TAG).error(f"获取队列超时～{e}")
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audio_opus_datas = audio_data_chunke.get('data') if audio_data_chunke else None
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duration = audio_data_chunke.get('duration') if audio_data_chunke else 0
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if audio_data_chunke:
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start_time = time.time()
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# 检查是否超过 5 秒没有数据
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if time.time() - start_time > 15:
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logger.bind(tag=TAG).error("超过15秒没有数据，退出。")
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break
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if audio_data_chunke and audio_data_chunke.get("end", True):
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break
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if audio_opus_datas:
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queue_duration = time.time() - start_get_queue
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last_duration = conn.tts_duration - queue_duration
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if last_duration <= 0:
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last_duration = 0
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conn.tts_duration = duration + last_duration
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for opus_packet in audio_opus_datas:
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await conn.websocket.send(opus_packet)
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start_time = time.time() # 更新获取数据的时间
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except Exception as e:
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logger.bind(tag=TAG).error(f"发生错误: {e}")
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traceback.print_exc() # 打印错误堆栈
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await send_tts_message(conn, "sentence_end", text)
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print(f'{text_index}-{conn.tts_last_text_index}')
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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 text_index == conn.tts_last_text_index:
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if conn.tts_duration and conn.tts_duration > 0:
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await asyncio.sleep(conn.tts_duration)
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await send_tts_message(conn, 'stop', None)
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if await isLLMWantToFinish(text):
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await conn.close()
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async def sendAudioMessage(conn, audios, text, text_index=0):
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# 发送句子开始消息
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if text_index == conn.tts_first_text_index:
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logger.bind(tag=TAG).info(f"发送第一段语音: {text}")
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await send_tts_message(conn, "sentence_start", text)
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# 初始化流控参数
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frame_duration = 60 # 毫秒
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start_time = time.perf_counter() # 使用高精度计时器
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play_position = 0 # 已播放的时长（毫秒）
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for opus_packet in audios:
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if conn.client_abort:
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return
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# 计算当前包的预期发送时间
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expected_time = start_time + (play_position / 1000)
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current_time = time.perf_counter()
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# 等待直到预期时间
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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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# 发送音频包
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await conn.websocket.send(opus_packet)
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play_position += frame_duration # 更新播放位置
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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 text_index == conn.tts_last_text_index:
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await send_tts_message(conn, 'stop', None)
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if await isLLMWantToFinish(text):
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await conn.close()
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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 = {
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"type": "tts",
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"state": state,
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"session_id": conn.session_id
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}
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if text is not None:
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message["text"] = text
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await conn.websocket.send(json.dumps(message))
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if state == "stop":
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conn.clearSpeakStatus()
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async def send_stt_message(conn, text):
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"""发送 STT 状态消息"""
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stt_text = get_string_no_punctuation_or_emoji(text)
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await conn.websocket.send(json.dumps({
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"type": "stt",
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"text": stt_text,
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"session_id": conn.session_id}
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))
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await conn.websocket.send(
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json.dumps({
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"type": "llm",
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"text": "😊",
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"emotion": "happy",
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"session_id": conn.session_id}
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))
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await send_tts_message(conn, "start")
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