mirror of
https://github.com/xinnan-tech/xiaozhi-esp32-server.git
synced 2026-07-27 01:23:55 +08:00
Merge branch 'xinnan-tech:main' into dev
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@@ -1,4 +1,3 @@
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from config.logger import setup_logging
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import json
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import uuid
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from core.handle.sendAudioHandle import send_stt_message
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@@ -146,4 +145,3 @@ async def process_intent_result(conn, intent_result, original_text):
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def speak_txt(conn, text):
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conn.tts.tts_one_sentence(conn, ContentType.TEXT, content_detail=text)
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conn.dialogue.put(Message(role="assistant", content=text))
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@@ -1,10 +1,7 @@
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import time
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import copy
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from core.utils.util import remove_punctuation_and_length
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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.reportHandle import enqueue_asr_report
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import time
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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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@@ -13,46 +10,17 @@ TAG = __name__
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async def handleAudioMessage(conn, audio):
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if conn.vad is None:
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conn.logger.bind(tag=TAG).warning("VAD模块未初始化,继续等待")
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return
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if not conn.asr_server_receive:
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conn.logger.bind(tag=TAG).debug(f"前期数据处理中,暂停接收")
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if conn.asr is None:
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conn.logger.bind(tag=TAG).warning("ASR模块未初始化,继续等待")
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return
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if conn.client_listen_mode == "auto" or conn.client_listen_mode == "realtime":
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have_voice = conn.vad.is_vad(conn, audio)
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else:
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have_voice = conn.client_have_voice
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# 如果本次没有声音,本段也没声音,就把声音丢弃了
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if have_voice == False and conn.client_have_voice == False:
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await no_voice_close_connect(conn)
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conn.asr_audio.append(audio)
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conn.asr_audio = conn.asr_audio[
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-10:
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] # 保留最新的10帧音频内容,解决ASR句首丢字问题
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return
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conn.client_no_voice_last_time = 0.0
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conn.asr_audio.append(audio)
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# 如果本段有声音,且已经停止了
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if conn.client_voice_stop:
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conn.client_abort = False
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conn.asr_server_receive = False
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# 音频太短了,无法识别
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if len(conn.asr_audio) < 15:
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conn.asr_server_receive = True
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else:
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raw_text, _ = await conn.asr.speech_to_text(
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conn.asr_audio, conn.session_id
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) # 确保ASR模块返回原始文本
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conn.logger.bind(tag=TAG).info(f"识别文本: {raw_text}")
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text_len, _ = remove_punctuation_and_length(raw_text)
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if text_len > 0:
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# 使用自定义模块进行上报
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await startToChat(conn, raw_text)
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enqueue_asr_report(conn, raw_text, copy.deepcopy(conn.asr_audio))
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else:
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conn.asr_server_receive = True
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conn.asr_audio.clear()
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conn.reset_vad_states()
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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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# 设备长时间空闲检测,用于say goodbye
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await no_voice_close_connect(conn, have_voice)
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# 接收音频
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await conn.asr.receive_audio(audio, have_voice)
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async def startToChat(conn, text):
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@@ -73,7 +41,6 @@ async def startToChat(conn, text):
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if intent_handled:
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# 如果意图已被处理,不再进行聊天
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conn.asr_server_receive = True
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return
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# 意图未被处理,继续常规聊天流程
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@@ -81,7 +48,10 @@ async def startToChat(conn, text):
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conn.executor.submit(conn.chat, text)
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async def no_voice_close_connect(conn):
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async def no_voice_close_connect(conn, have_voice):
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if have_voice:
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conn.client_no_voice_last_time = 0.0
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return
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if conn.client_no_voice_last_time == 0.0:
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conn.client_no_voice_last_time = time.time() * 1000
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else:
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@@ -95,7 +65,6 @@ async def no_voice_close_connect(conn):
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):
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conn.close_after_chat = True
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conn.client_abort = False
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conn.asr_server_receive = False
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end_prompt = conn.config.get("end_prompt", {})
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if end_prompt and end_prompt.get("enable", True) is False:
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conn.logger.bind(tag=TAG).info("结束对话,无需发送结束提示语")
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@@ -8,6 +8,7 @@ TTS上报功能已集成到ConnectionHandler类中。
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具体实现请参考core/connection.py中的相关代码。
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"""
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import time
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import opuslib_next
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@@ -145,4 +146,4 @@ def enqueue_asr_report(conn, text, opus_data):
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f"ASR数据已加入上报队列: {conn.device_id}, 不上报音频"
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)
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except Exception as e:
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conn.logger.bind(tag=TAG).error(f"加入ASR上报队列失败: {text}, {e}")
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conn.logger.bind(tag=TAG).debug(f"加入ASR上报队列失败: {text}, {e}")
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@@ -131,6 +131,11 @@ async def send_tts_message(conn, state, text=None):
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async def send_stt_message(conn, text):
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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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stt_text = get_string_no_punctuation_or_emoji(text)
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await conn.websocket.send(
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@@ -42,12 +42,13 @@ async def handleTextMessage(conn, message):
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if len(conn.asr_audio) > 0:
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await handleAudioMessage(conn, b"")
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elif msg_json["state"] == "detect":
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conn.asr_server_receive = False
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conn.client_have_voice = False
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conn.asr_audio.clear()
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if "text" in msg_json:
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original_text = msg_json["text"] # 保留原始文本
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filtered_len, filtered_text = remove_punctuation_and_length(original_text)
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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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