mirror of
https://github.com/xinnan-tech/xiaozhi-esp32-server.git
synced 2026-07-22 07:03:53 +08:00
update: 恢复唤醒播放机制,待优化和改造(目前只有linkerai拥有正常保存唤醒音频to_tts)
This commit is contained in:
@@ -61,6 +61,10 @@ delete_audio: true
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close_connection_no_voice_time: 120
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# TTS请求超时时间(秒)
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tts_timeout: 10
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# 开启唤醒词加速
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enable_wakeup_words_response_cache: true
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# 开场是否回复唤醒词
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enable_greeting: true
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# 说完话是否开启提示音
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enable_stop_tts_notify: false
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# 说完话是否开启提示音,音效地址
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@@ -1,5 +1,16 @@
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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.providers.tts.dto.dto import SentenceType
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from core.utils.wakeup_word import WakeupWordsConfig
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from core.handle.sendAudioHandle import sendAudioMessage, send_stt_message
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from core.utils.util import (
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audio_to_data_stream,
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remove_punctuation_and_length,
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opus_datas_to_wav_bytes
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)
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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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@@ -8,6 +19,17 @@ from core.providers.tools.device_mcp import (
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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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@@ -30,3 +52,99 @@ 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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# 获取音频数据
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opus_packets = []
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def handle_audio_frame(frame_data):
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opus_packets.append(frame_data)
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audio_to_data_stream(response.get("file_path"), is_opus=True, callback=handle_audio_frame)
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# 播放唤醒词回复
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conn.client_abort = False
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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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@@ -2,6 +2,7 @@ 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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@@ -26,6 +27,10 @@ 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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@@ -1,11 +1,11 @@
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import time
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import json
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from core.handle.sendAudioHandle import send_stt_message
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import asyncio
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from core.utils.util import audio_to_data_stream
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from core.handle.abortHandle import handleAbortMessage
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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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from core.handle.sendAudioHandle import SentenceType
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from core.utils.util import audio_to_data_stream
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from core.handle.sendAudioHandle import send_stt_message, SentenceType
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TAG = __name__
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@@ -13,7 +13,14 @@ TAG = __name__
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async def handleAudioMessage(conn, audio):
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# 当前片段是否有人说话
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have_voice = conn.vad.is_vad(conn, audio)
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# 如果设备刚刚被唤醒,短暂忽略VAD检测
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if have_voice and hasattr(conn, "just_woken_up") and conn.just_woken_up:
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have_voice = False
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# 设置一个短暂延迟后恢复VAD检测
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conn.asr_audio.clear()
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if not hasattr(conn, "vad_resume_task") or conn.vad_resume_task.done():
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conn.vad_resume_task = asyncio.create_task(resume_vad_detection(conn))
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return
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if have_voice:
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if conn.client_is_speaking:
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await handleAbortMessage(conn)
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@@ -22,6 +29,13 @@ async def handleAudioMessage(conn, audio):
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# 接收音频
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await conn.asr.receive_audio(conn, audio, have_voice)
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async def resume_vad_detection(conn):
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# 等待2秒后恢复VAD检测
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await asyncio.sleep(1)
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conn.just_woken_up = False
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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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@@ -146,7 +160,6 @@ async def check_bind_device(conn):
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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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@@ -1,8 +1,8 @@
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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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import asyncio
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from core.utils import textUtils
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from core.providers.tts.dto.dto import SentenceType
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TAG = __name__
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@@ -30,7 +30,7 @@ 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=False):
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async def sendAudio(conn, audios, pre_buffer=False, frame_duration=60):
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"""
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发送单个opus包,支持流控
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Args:
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@@ -50,12 +50,8 @@ async def sendAudio(conn, audios, pre_buffer=False):
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await conn.websocket.send(audios)
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return
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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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# 流控逻辑:确保按60ms的帧时长间隔发送
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frame_duration = 60 # 毫秒
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# 获取或初始化流控状态
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if not hasattr(conn, "audio_flow_control"):
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conn.audio_flow_control = {
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@@ -66,13 +62,10 @@ async def sendAudio(conn, audios, pre_buffer=False):
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flow_control = conn.audio_flow_control
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current_time = time.perf_counter()
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# 计算期望的发送时间
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# 计算预期发送时间
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expected_time = flow_control["start_time"] + (
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flow_control["packet_count"] * frame_duration / 1000
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)
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# 流控延迟
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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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@@ -83,6 +76,38 @@ async def sendAudio(conn, audios, pre_buffer=False):
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# 更新流控状态
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flow_control["packet_count"] += 1
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flow_control["last_send_time"] = time.perf_counter()
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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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# 执行预缓冲
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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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# 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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@@ -1,13 +1,14 @@
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import json
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import time
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import asyncio
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from core.handle.abortHandle import handleAbortMessage
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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.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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from core.providers.tools.device_mcp import handle_mcp_message
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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.utils.util import remove_punctuation_and_length, filter_sensitive_info
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from core.providers.tools.device_iot import handleIotDescriptors, handleIotStatus
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TAG = __name__
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@@ -50,9 +51,23 @@ 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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if not is_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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# 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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@@ -286,8 +286,8 @@ class TTSProviderBase(ABC):
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enqueue_audio = []
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enqueue_text = text
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# 计算音频数据的帧数
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if isinstance(audio_datas, bytes):
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# 收集上报音频数据
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if isinstance(audio_datas, bytes) and enqueue_audio is not None:
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enqueue_audio.append(audio_datas)
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# 发送音频
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@@ -1,8 +1,10 @@
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import os
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import time
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import queue
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import asyncio
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import traceback
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import aiohttp
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import asyncio
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import requests
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import traceback
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from config.logger import setup_logging
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from core.utils.tts import MarkdownCleaner
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from core.providers.tts.base import TTSProviderBase
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@@ -199,3 +201,71 @@ class TTSProvider(TTSProviderBase):
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except Exception as e:
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logger.bind(tag=TAG).error(f"TTS请求异常: {e}")
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self.tts_audio_queue.put((SentenceType.LAST, [], None))
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def to_tts(self, text: str) -> list:
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"""非流式TTS处理,用于测试及保存音频文件的场景
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Args:
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text: 要转换的文本
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Returns:
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list: 返回opus编码后的音频数据列表
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"""
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start_time = time.time()
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text = MarkdownCleaner.clean_markdown(text)
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params = {
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"tts_text": text,
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"spk_id": self.voice,
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"frame_duration": 60,
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"stream": False,
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"target_sr": 16000,
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"audio_format": self.audio_format,
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"instruct_text": "请生成一段自然流畅的语音",
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}
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headers = {
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"Authorization": f"Bearer {self.access_token}",
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"Content-Type": "application/json",
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}
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try:
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with requests.get(
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self.api_url, params=params, headers=headers, timeout=5
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) as response:
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if response.status_code != 200:
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logger.bind(tag=TAG).error(
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f"TTS请求失败: {response.status_code}, {response.text}"
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)
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return []
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|
||||
logger.info(f"TTS请求成功: {text}, 耗时: {time.time() - start_time}秒")
|
||||
|
||||
# 使用opus编码器处理PCM数据
|
||||
opus_datas = []
|
||||
pcm_data = response.content
|
||||
|
||||
# 计算每帧的字节数
|
||||
frame_bytes = int(
|
||||
self.opus_encoder.sample_rate
|
||||
* self.opus_encoder.channels
|
||||
* self.opus_encoder.frame_size_ms
|
||||
/ 1000
|
||||
* 2
|
||||
)
|
||||
|
||||
# 分帧处理PCM数据
|
||||
for i in range(0, len(pcm_data), frame_bytes):
|
||||
frame = pcm_data[i : i + frame_bytes]
|
||||
if len(frame) < frame_bytes:
|
||||
# 最后一帧可能不足,用0填充
|
||||
frame = frame + b"\x00" * (frame_bytes - len(frame))
|
||||
|
||||
self.opus_encoder.encode_pcm_to_opus_stream(
|
||||
frame,
|
||||
end_of_stream=(i + frame_bytes >= len(pcm_data)),
|
||||
callback=lambda opus: opus_datas.append(opus)
|
||||
)
|
||||
|
||||
return opus_datas
|
||||
|
||||
except Exception as e:
|
||||
logger.bind(tag=TAG).error(f"TTS请求异常: {e}")
|
||||
return []
|
||||
@@ -1,16 +1,17 @@
|
||||
import json
|
||||
import socket
|
||||
import subprocess
|
||||
import re
|
||||
import os
|
||||
from io import BytesIO
|
||||
from typing import Callable, Any
|
||||
from core.utils import p3
|
||||
import numpy as np
|
||||
import requests
|
||||
import opuslib_next
|
||||
from pydub import AudioSegment
|
||||
import json
|
||||
import wave
|
||||
import copy
|
||||
import socket
|
||||
import requests
|
||||
import subprocess
|
||||
import numpy as np
|
||||
import opuslib_next
|
||||
from io import BytesIO
|
||||
from core.utils import p3
|
||||
from pydub import AudioSegment
|
||||
from typing import Callable, Any
|
||||
|
||||
TAG = __name__
|
||||
emoji_map = {
|
||||
@@ -274,6 +275,33 @@ def pcm_to_data_stream(raw_data, is_opus=True, callback: Callable[[Any], Any] =
|
||||
callback(frame_data)
|
||||
|
||||
|
||||
def opus_datas_to_wav_bytes(opus_datas, sample_rate=16000, channels=1):
|
||||
"""
|
||||
将opus帧列表解码为wav字节流
|
||||
"""
|
||||
decoder = opuslib_next.Decoder(sample_rate, channels)
|
||||
pcm_datas = []
|
||||
|
||||
frame_duration = 60 # ms
|
||||
frame_size = int(sample_rate * frame_duration / 1000) # 960
|
||||
|
||||
for opus_frame in opus_datas:
|
||||
# 解码为PCM(返回bytes,2字节/采样点)
|
||||
pcm = decoder.decode(opus_frame, frame_size)
|
||||
pcm_datas.append(pcm)
|
||||
|
||||
pcm_bytes = b"".join(pcm_datas)
|
||||
|
||||
# 写入wav字节流
|
||||
wav_buffer = BytesIO()
|
||||
with wave.open(wav_buffer, "wb") as wf:
|
||||
wf.setnchannels(channels)
|
||||
wf.setsampwidth(2) # 16bit
|
||||
wf.setframerate(sample_rate)
|
||||
wf.writeframes(pcm_bytes)
|
||||
return wav_buffer.getvalue()
|
||||
|
||||
|
||||
def check_vad_update(before_config, new_config):
|
||||
if (
|
||||
new_config.get("selected_module") is None
|
||||
|
||||
@@ -0,0 +1,140 @@
|
||||
import os
|
||||
import re
|
||||
import yaml
|
||||
import time
|
||||
import hashlib
|
||||
import portalocker
|
||||
from typing import Dict
|
||||
|
||||
|
||||
class FileLock:
|
||||
def __init__(self, file, timeout=5):
|
||||
self.file = file
|
||||
self.timeout = timeout
|
||||
self.start_time = None
|
||||
|
||||
def __enter__(self):
|
||||
self.start_time = time.time()
|
||||
while True:
|
||||
try:
|
||||
portalocker.lock(self.file, portalocker.LOCK_EX | portalocker.LOCK_NB)
|
||||
return self.file
|
||||
except portalocker.LockException:
|
||||
if time.time() - self.start_time > self.timeout:
|
||||
raise TimeoutError("获取文件锁超时")
|
||||
time.sleep(0.1)
|
||||
|
||||
def __exit__(self, exc_type, exc_val, exc_tb):
|
||||
portalocker.unlock(self.file)
|
||||
|
||||
|
||||
class WakeupWordsConfig:
|
||||
def __init__(self):
|
||||
self.config_file = "data/.wakeup_words.yaml"
|
||||
self.assets_dir = "config/assets/wakeup_words"
|
||||
self._ensure_directories()
|
||||
self._config_cache = None
|
||||
self._last_load_time = 0
|
||||
self._cache_ttl = 1 # 缓存有效期(秒)
|
||||
self._lock_timeout = 5 # 文件锁超时时间(秒)
|
||||
|
||||
def _ensure_directories(self):
|
||||
"""确保必要的目录存在"""
|
||||
os.makedirs(os.path.dirname(self.config_file), exist_ok=True)
|
||||
os.makedirs(self.assets_dir, exist_ok=True)
|
||||
|
||||
def _load_config(self) -> Dict:
|
||||
"""加载配置文件,使用缓存机制"""
|
||||
current_time = time.time()
|
||||
|
||||
# 如果缓存有效,直接返回缓存
|
||||
if (
|
||||
self._config_cache is not None
|
||||
and current_time - self._last_load_time < self._cache_ttl
|
||||
):
|
||||
return self._config_cache
|
||||
|
||||
try:
|
||||
with open(self.config_file, "a+") as f:
|
||||
with FileLock(f, timeout=self._lock_timeout):
|
||||
f.seek(0)
|
||||
content = f.read()
|
||||
config = yaml.safe_load(content) if content else {}
|
||||
self._config_cache = config
|
||||
self._last_load_time = current_time
|
||||
return config
|
||||
except (TimeoutError, IOError) as e:
|
||||
print(f"加载配置文件失败: {e}")
|
||||
return {}
|
||||
except Exception as e:
|
||||
print(f"加载配置文件时发生未知错误: {e}")
|
||||
return {}
|
||||
|
||||
def _save_config(self, config: Dict):
|
||||
"""保存配置到文件,使用文件锁保护"""
|
||||
try:
|
||||
with open(self.config_file, "w") as f:
|
||||
with FileLock(f, timeout=self._lock_timeout):
|
||||
yaml.dump(config, f, allow_unicode=True)
|
||||
self._config_cache = config
|
||||
self._last_load_time = time.time()
|
||||
except (TimeoutError, IOError) as e:
|
||||
print(f"保存配置文件失败: {e}")
|
||||
raise
|
||||
except Exception as e:
|
||||
print(f"保存配置文件时发生未知错误: {e}")
|
||||
raise
|
||||
|
||||
def get_wakeup_response(self, voice: str) -> Dict:
|
||||
voice = hashlib.md5(voice.encode()).hexdigest()
|
||||
"""获取唤醒词回复配置"""
|
||||
config = self._load_config()
|
||||
|
||||
if not config or voice not in config:
|
||||
return None
|
||||
|
||||
# 检查文件大小
|
||||
file_path = config[voice]["file_path"]
|
||||
if not os.path.exists(file_path) or os.stat(file_path).st_size < (15 * 1024):
|
||||
return None
|
||||
|
||||
return config[voice]
|
||||
|
||||
def update_wakeup_response(self, voice: str, file_path: str, text: str):
|
||||
"""更新唤醒词回复配置"""
|
||||
try:
|
||||
# 过滤表情符号
|
||||
filtered_text = re.sub(r'[\U0001F600-\U0001F64F\U0001F900-\U0001F9FF]', '', text)
|
||||
|
||||
config = self._load_config()
|
||||
voice_hash = hashlib.md5(voice.encode()).hexdigest()
|
||||
config[voice_hash] = {
|
||||
"voice": voice,
|
||||
"file_path": file_path,
|
||||
"time": time.time(),
|
||||
"text": filtered_text,
|
||||
}
|
||||
self._save_config(config)
|
||||
except Exception as e:
|
||||
print(f"更新唤醒词回复配置失败: {e}")
|
||||
raise
|
||||
|
||||
def generate_file_path(self, voice: str) -> str:
|
||||
"""生成音频文件路径,使用voice的哈希值作为文件名"""
|
||||
try:
|
||||
# 生成voice的哈希值
|
||||
voice_hash = hashlib.md5(voice.encode()).hexdigest()
|
||||
file_path = os.path.join(self.assets_dir, f"{voice_hash}.wav")
|
||||
|
||||
# 如果文件已存在,先删除
|
||||
if os.path.exists(file_path):
|
||||
try:
|
||||
os.remove(file_path)
|
||||
except Exception as e:
|
||||
print(f"删除已存在的音频文件失败: {e}")
|
||||
raise
|
||||
|
||||
return file_path
|
||||
except Exception as e:
|
||||
print(f"生成音频文件路径失败: {e}")
|
||||
raise
|
||||
Reference in New Issue
Block a user