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xiaozhi-esp32-server/core/handle/sendAudioHandle.py
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import traceback
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from config.logger import setup_logging
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import json
import asyncio
import time
from core.utils.util import remove_punctuation_and_length, get_string_no_punctuation_or_emoji
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TAG = __name__
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)
if "再见" in last_text_without_punctuation or "拜拜" in last_text_without_punctuation:
return True
return False
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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):
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# 发送句子开始消息
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
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:
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if conn.client_abort:
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)
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)
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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
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:
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await send_tts_message(conn, 'stop', None)
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if await isLLMWantToFinish(text):
await conn.close()
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async def send_tts_message(conn, state, text=None):
"""发送 TTS 状态消息"""
message = {
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"type": "tts",
"state": state,
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"session_id": conn.session_id
}
if text is not None:
message["text"] = text
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await conn.websocket.send(json.dumps(message))
if state == "stop":
conn.clearSpeakStatus()
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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")