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https://github.com/xinnan-tech/xiaozhi-esp32-server.git
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add:实时打断对话
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@@ -27,13 +27,13 @@
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## 已实现
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- `xiaozhi-esp32` 通信 WebSocket 协议
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- 支持实时打断对话,试一下在聊天的时候说一句`你好小智`
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- 支持国语、粤语、英语、日语、韩语 5 种语言识别(FunASR(默认))
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- 自由更换 LLM(支持ChatGLM(默认)、Dify、DeepSeek)
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- 自由更换 TTS(支持EdgeTTS(默认)、火山引擎豆包TTS)
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## 正在实现
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- 打断对话
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- 按键手动对话
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- 长时间不聊天进入休眠状态
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- 对话记忆
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@@ -213,7 +213,11 @@ https://espressif.github.io/esp-launchpad/
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建议:大模型和TTS都是依赖接口,如果网络环境不佳,可以考虑换成本地模型。或多尝试切换不同的接口模型。
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## 3、更多问题,可联系我们反馈
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## 3、为啥我的ChatGLMLLM回复有点问题,明明它才是小智,却把我当小智。
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建议:第一步可以先`配置文件`里的调整提示词。第二步,`配置文件`中用的模型是免费的模型:`glm-4-flash`,可以考虑换成收费版本的。
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## 4、更多问题,可联系我们反馈
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+7
-2
@@ -32,13 +32,13 @@ To fully experience this project, follow these steps:
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## Implemented
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- `xiaozhi-esp32` WebSocket communication protocol
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- Support interrupt dialogue in real time, try to say a sentence when chatting `你好小智`
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- Support for 5 languages: Mandarin, Cantonese, English, Japanese, Korean (FunASR - default)
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- Flexible LLM switching (ChatGLM - default, Dify, DeepSeek)
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- Flexible TTS switching (EdgeTTS - default, ByteDance Doubao TTS)
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## In Progress
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- Conversation interruption
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- Manual button-triggered dialogue
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- Sleep mode after inactivity
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- Dialogue memory
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@@ -223,7 +223,12 @@ both are slow, the network environment may need to be optimized.
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Suggestions: Both big models and TTS are dependent interfaces. If the network environment is not good, you can consider
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changing the local model. Or try to switch different interface models.
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## 3、For more questions, contact us to feedback
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## 3、Why is my ChatGLMLLM replying to a bit? Obviously it is Xiaozhi, but treats me as Xiaozhi.
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Suggestion: The first step can be to adjust the prompt words in the configuration file. The second step, the model used
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in the configuration file is the free model: `glm-4-flash`. You might consider switching to the paid version.
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## 4、For more questions, contact us to feedback
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+14
-3
@@ -12,6 +12,7 @@ from collections import deque
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from core.utils.util import is_segment
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from core.utils.dialogue import Message, Dialogue
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from core.handle.textHandle import handleTextMessage
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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.utils.util import get_string_no_punctuation_or_emoji
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from concurrent.futures import ThreadPoolExecutor, TimeoutError
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@@ -28,6 +29,7 @@ class ConnectionHandler:
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self.session_id = None
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self.prompt = None
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self.welcome_msg = None
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self.client_abort = False
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# 线程任务相关
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self.loop = asyncio.get_event_loop()
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@@ -100,6 +102,8 @@ class ConnectionHandler:
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msg_json = json.loads(message)
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if msg_json["type"] == "hello":
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await handleHelloMessage(self, "你好")
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if msg_json["type"] == "abort":
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await handleAbortMessage(self)
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except json.JSONDecodeError:
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await handleTextMessage(self, message)
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@@ -126,6 +130,11 @@ class ConnectionHandler:
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self.llm_finish_task = False
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for content in llm_responses:
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response_message.append(content)
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# 如果中途被打断,就停止生成
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if self.client_abort:
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start = len(response_message)
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break
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end_time = time.time() # 记录结束时间
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self.logger.debug(f"大模型返回时间时间: {end_time - start_time} 秒, 生成token={content}")
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if is_segment(response_message):
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@@ -169,9 +178,11 @@ class ConnectionHandler:
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except Exception as e:
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self.logger.error(f"TTS 任务出错: {e}")
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continue
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asyncio.run_coroutine_threadsafe(
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sendAudioMessage(self, opus_datas, duration, text), self.loop
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)
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if not self.client_abort:
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# 如果没有中途打断就发送语音
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asyncio.run_coroutine_threadsafe(
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sendAudioMessage(self, opus_datas, duration, text), self.loop
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)
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if self.tts.delete_audio_file and os.path.exists(tts_file):
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os.remove(tts_file)
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except Exception as e:
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@@ -0,0 +1,16 @@
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import json
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import logging
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logger = logging.getLogger(__name__)
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async def handleAbortMessage(conn):
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logger.info("Abort message received")
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# 设置成打断状态,会自动打断llm、tts任务
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conn.client_abort = True
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# 打断屏显任务
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conn.stop_all_tasks()
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# 打断客户端说话状态
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await conn.websocket.send(json.dumps({"type": "tts", "state": "stop", "session_id": conn.session_id}))
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conn.clearSpeakStatus()
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logger.info("Abort message received-end")
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@@ -20,6 +20,7 @@ async def handleAudioMessage(conn, audio):
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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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text, file_path = conn.asr.speech_to_text(conn.asr_audio, conn.session_id)
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logger.info(f"识别文本: {text}")
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