mirror of
https://github.com/xinnan-tech/xiaozhi-esp32-server.git
synced 2026-07-29 13:43:56 +08:00
update: 音频流式优化
This commit is contained in:
@@ -1,32 +1,13 @@
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import time
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import json
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import random
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import asyncio
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from core.utils.dialogue import Message
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from core.utils.util import audio_to_data
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from core.handle.sendAudioHandle import sendAudioMessage, send_stt_message
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from core.utils.util import remove_punctuation_and_length, opus_datas_to_wav_bytes
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from core.providers.tts.dto.dto import ContentType, SentenceType
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from core.providers.tools.device_mcp import (
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MCPClient,
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send_mcp_initialize_message,
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send_mcp_tools_list_request,
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)
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from core.utils.wakeup_word import WakeupWordsConfig
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TAG = __name__
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WAKEUP_CONFIG = {
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"refresh_time": 5,
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"words": ["你好", "你好啊", "嘿,你好", "嗨"],
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}
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# 创建全局的唤醒词配置管理器
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wakeup_words_config = WakeupWordsConfig()
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# 用于防止并发调用wakeupWordsResponse的锁
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_wakeup_response_lock = asyncio.Lock()
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async def handleHelloMessage(conn, msg_json):
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"""处理hello消息"""
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@@ -49,93 +30,3 @@ async def handleHelloMessage(conn, msg_json):
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asyncio.create_task(send_mcp_tools_list_request(conn))
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await conn.websocket.send(json.dumps(conn.welcome_msg))
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async def checkWakeupWords(conn, text):
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enable_wakeup_words_response_cache = conn.config[
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"enable_wakeup_words_response_cache"
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]
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if not enable_wakeup_words_response_cache or not conn.tts:
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return False
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_, filtered_text = remove_punctuation_and_length(text)
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if filtered_text not in conn.config.get("wakeup_words"):
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return False
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conn.just_woken_up = True
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await send_stt_message(conn, text)
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# 获取当前音色
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voice = getattr(conn.tts, "voice", "default")
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if not voice:
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voice = "default"
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# 获取唤醒词回复配置
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response = wakeup_words_config.get_wakeup_response(voice)
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if not response or not response.get("file_path"):
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response = {
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"voice": "default",
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"file_path": "config/assets/wakeup_words.wav",
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"time": 0,
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"text": "哈啰啊,我是小智啦,声音好听的台湾女孩一枚,超开心认识你耶,最近在忙啥,别忘了给我来点有趣的料哦,我超爱听八卦的啦",
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}
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# 播放唤醒词回复
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conn.client_abort = False
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opus_packets = audio_to_data(response.get("file_path"))
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conn.logger.bind(tag=TAG).info(f"播放唤醒词回复: {response.get('text')}")
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await sendAudioMessage(conn, SentenceType.FIRST, opus_packets, response.get("text"))
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await sendAudioMessage(conn, SentenceType.LAST, [], None)
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# 补充对话
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conn.dialogue.put(Message(role="assistant", content=response.get("text")))
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# 检查是否需要更新唤醒词回复
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if time.time() - response.get("time", 0) > WAKEUP_CONFIG["refresh_time"]:
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if not _wakeup_response_lock.locked():
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asyncio.create_task(wakeupWordsResponse(conn))
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return True
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async def wakeupWordsResponse(conn):
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if not conn.tts or not conn.llm or not conn.llm.response_no_stream:
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return
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try:
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# 尝试获取锁,如果获取不到就返回
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if not await _wakeup_response_lock.acquire():
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return
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# 生成唤醒词回复
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wakeup_word = random.choice(WAKEUP_CONFIG["words"])
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question = (
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"此刻用户正在和你说```"
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+ wakeup_word
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+ "```。\n请你根据以上用户的内容进行20-30字回复。要符合系统设置的角色情感和态度,不要像机器人一样说话。\n"
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+ "请勿对这条内容本身进行任何解释和回应,请勿返回表情符号,仅返回对用户的内容的回复。"
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)
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result = conn.llm.response_no_stream(conn.config["prompt"], question)
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if not result or len(result) == 0:
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return
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# 生成TTS音频
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tts_result = await asyncio.to_thread(conn.tts.to_tts, result)
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if not tts_result:
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return
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# 获取当前音色
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voice = getattr(conn.tts, "voice", "default")
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wav_bytes = opus_datas_to_wav_bytes(tts_result, sample_rate=16000)
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file_path = wakeup_words_config.generate_file_path(voice)
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with open(file_path, "wb") as f:
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f.write(wav_bytes)
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# 更新配置
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wakeup_words_config.update_wakeup_response(voice, file_path, result)
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finally:
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# 确保在任何情况下都释放锁
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if _wakeup_response_lock.locked():
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_wakeup_response_lock.release()
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@@ -2,7 +2,6 @@ import json
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import asyncio
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import uuid
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from core.handle.sendAudioHandle import send_stt_message
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from core.handle.helloHandle import checkWakeupWords
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from core.utils.util import remove_punctuation_and_length
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from core.providers.tts.dto.dto import ContentType
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from core.utils.dialogue import Message
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@@ -27,9 +26,6 @@ async def handle_user_intent(conn, text):
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filtered_text = remove_punctuation_and_length(text)[1]
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if await check_direct_exit(conn, filtered_text):
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return True
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# 检查是否是唤醒词
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if await checkWakeupWords(conn, filtered_text):
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return True
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if conn.intent_type == "function_call":
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# 使用支持function calling的聊天方法,不再进行意图分析
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@@ -1,12 +1,12 @@
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import time
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import asyncio
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import json
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from core.handle.sendAudioHandle import send_stt_message
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from core.handle.intentHandler import handle_user_intent
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from core.utils.output_counter import check_device_output_limit
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from core.handle.abortHandle import handleAbortMessage
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import time
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import asyncio
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import json
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from core.handle.sendAudioHandle import SentenceType
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from core.utils.util import audio_to_data
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from core.utils.util import audio_to_data_stream
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TAG = __name__
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@@ -42,7 +42,7 @@ async def startToChat(conn, text):
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# 检查输入是否是JSON格式(包含说话人信息)
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speaker_name = None
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actual_text = text
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try:
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# 尝试解析JSON格式的输入
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if text.strip().startswith('{') and text.strip().endswith('}'):
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@@ -51,13 +51,13 @@ async def startToChat(conn, text):
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speaker_name = data['speaker']
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actual_text = data['content']
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conn.logger.bind(tag=TAG).info(f"解析到说话人信息: {speaker_name}")
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# 直接使用JSON格式的文本,不解析
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actual_text = text
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except (json.JSONDecodeError, KeyError):
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# 如果解析失败,继续使用原始文本
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pass
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# 保存说话人信息到连接对象
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if speaker_name:
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conn.current_speaker = speaker_name
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@@ -121,8 +121,9 @@ async def max_out_size(conn):
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text = "不好意思,我现在有点事情要忙,明天这个时候我们再聊,约好了哦!明天不见不散,拜拜!"
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await send_stt_message(conn, text)
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file_path = "config/assets/max_output_size.wav"
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opus_packets, _ = audio_to_data(file_path)
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conn.tts.tts_audio_queue.put((SentenceType.LAST, opus_packets, text))
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conn.tts.tts_audio_queue.put((SentenceType.FIRST, [], text))
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play_audio_frames(conn, file_path)
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conn.tts.tts_audio_queue.put((SentenceType.LAST, [], None))
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conn.close_after_chat = True
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@@ -140,16 +141,15 @@ async def check_bind_device(conn):
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# 播放提示音
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music_path = "config/assets/bind_code.wav"
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opus_packets = audio_to_data(music_path)
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conn.tts.tts_audio_queue.put((SentenceType.FIRST, opus_packets, text))
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conn.tts.tts_audio_queue.put((SentenceType.FIRST, [], text))
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play_audio_frames(conn, music_path)
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# 逐个播放数字
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for i in range(6): # 确保只播放6位数字
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try:
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digit = conn.bind_code[i]
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num_path = f"config/assets/bind_code/{digit}.wav"
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num_packets = audio_to_data(num_path)
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conn.tts.tts_audio_queue.put((SentenceType.MIDDLE, num_packets, None))
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play_audio_frames(conn, num_path)
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except Exception as e:
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conn.logger.bind(tag=TAG).error(f"播放数字音频失败: {e}")
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continue
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@@ -158,5 +158,18 @@ async def check_bind_device(conn):
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text = f"没有找到该设备的版本信息,请正确配置 OTA地址,然后重新编译固件。"
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await send_stt_message(conn, text)
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music_path = "config/assets/bind_not_found.wav"
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opus_packets = audio_to_data(music_path)
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conn.tts.tts_audio_queue.put((SentenceType.LAST, opus_packets, text))
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conn.tts.tts_audio_queue.put((SentenceType.FIRST, [], text))
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play_audio_frames(conn, music_path)
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conn.tts.tts_audio_queue.put((SentenceType.LAST, [], None))
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def play_audio_frames(conn, file_path):
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"""播放音频文件并处理发送帧数据"""
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def handle_audio_frame(frame_data):
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conn.tts.tts_audio_queue.put((SentenceType.MIDDLE, frame_data, None))
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audio_to_data_stream(
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file_path,
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is_opus=True,
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callback=handle_audio_frame
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)
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@@ -1,6 +1,4 @@
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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 import textUtils
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@@ -8,20 +6,18 @@ TAG = __name__
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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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pre_buffer = False
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if conn.tts.tts_audio_first_sentence:
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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, "start", None)
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if sentenceType == SentenceType.FIRST:
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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 sendAudio(conn, audios)
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# 发送句子开始消息
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if sentenceType is not SentenceType.MIDDLE:
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conn.logger.bind(tag=TAG).info(f"发送音频消息: {sentenceType}, {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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@@ -32,46 +28,12 @@ async def sendAudioMessage(conn, sentenceType, audios, text):
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# 播放音频
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async def sendAudio(conn, audios, pre_buffer=True):
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async def sendAudio(conn, audios):
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if audios is None:
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return
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# 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
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if isinstance(audios ,bytes):
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if isinstance(audios, bytes):
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await conn.websocket.send(audios)
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else:
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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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@@ -100,13 +62,12 @@ async def send_tts_message(conn, state, text=None):
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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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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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@@ -5,7 +5,6 @@ from core.handle.helloHandle import handleHelloMessage
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from core.providers.tools.device_mcp import handle_mcp_message
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from core.utils.util import remove_punctuation_and_length, filter_sensitive_info
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from core.handle.receiveAudioHandle import startToChat, handleAudioMessage
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from core.handle.sendAudioHandle import send_stt_message, send_tts_message
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from core.providers.tools.device_iot import handleIotDescriptors, handleIotStatus
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from core.handle.reportHandle import enqueue_asr_report
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import asyncio
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@@ -51,23 +50,9 @@ async def handleTextMessage(conn, message):
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filtered_len, filtered_text = remove_punctuation_and_length(
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original_text
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)
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# 识别是否是唤醒词
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is_wakeup_words = filtered_text in conn.config.get("wakeup_words")
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# 是否开启唤醒词回复
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enable_greeting = conn.config.get("enable_greeting", True)
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if is_wakeup_words and not enable_greeting:
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# 如果是唤醒词,且关闭了唤醒词回复,就不用回答
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await send_stt_message(conn, original_text)
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await send_tts_message(conn, "stop", None)
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conn.client_is_speaking = False
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elif is_wakeup_words:
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conn.just_woken_up = True
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# 上报纯文字数据(复用ASR上报功能,但不提供音频数据)
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enqueue_asr_report(conn, "嘿,你好呀", [])
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await startToChat(conn, "嘿,你好呀")
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else:
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if not is_wakeup_words:
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# 上报纯文字数据(复用ASR上报功能,但不提供音频数据)
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enqueue_asr_report(conn, original_text, [])
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# 否则需要LLM对文字内容进行答复
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