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162 Commits
Author SHA1 Message Date
欣南科技andGitHub 2d7d75c290 Merge pull request #1438 from xinnan-tech/vllm
udpate:添加拍照识物教程
2025-06-01 16:29:27 +08:00
hrz 237e88ba89 udpate:添加拍照识物教程 2025-06-01 16:18:00 +08:00
欣南科技andGitHub 48223e4f39 Merge pull request #1435 from xinnan-tech/vllm
update:智控台,完成mcp拍照识图
2025-06-01 13:35:33 +08:00
hrz ce776f210c update:智控台,完成mcp拍照识图 2025-06-01 13:34:32 +08:00
欣南科技andGitHub fb5e25ec16 Merge pull request #1433 from xinnan-tech/vllm
update:单模块部署xiaozhi-server,支持mcp调用摄像头识图
2025-06-01 02:31:13 +08:00
hrz c4f2411fee update:单模块部署xiaozhi-server,支持mcp调用摄像头识图 2025-06-01 02:26:19 +08:00
欣南科技andGitHub 3c3be950e9 Merge pull request #1428 from xinnan-tech/hot-fix
update:修复参数错误
2025-05-31 10:14:50 +08:00
hrz b24ee0b9a6 update:修复参数错误 2025-05-31 10:14:23 +08:00
hrzandGitHub e1d5caa0fd Merge pull request #1426 from bitailab/main
SSEClient支持认证
2025-05-30 23:45:26 +08:00
lihaolong 35ba9b0e6b SSEClient支持认证 2025-05-30 11:08:32 +00:00
欣南科技andGitHub 66de07823d Merge pull request #1424 from xinnan-tech/update-doc
update:更新文档
2025-05-30 18:16:14 +08:00
hrz fd96e71d04 update:更新文档 2025-05-30 18:15:25 +08:00
欣南科技andGitHub 3c85242efc Merge pull request #1421 from xinnan-tech/test-mcp
Test mcp
2025-05-30 17:14:44 +08:00
hrz 3c288e1e59 update:优化客户端MCP服务 2025-05-30 17:13:54 +08:00
hrzandGitHub 4bf2f927b0 Merge pull request #1420 from xinnan-tech/py_test_mcp
Py test mcp
2025-05-30 16:09:06 +08:00
hrzandGitHub ad30030a1f Merge branch 'test-mcp' into py_test_mcp 2025-05-30 16:08:14 +08:00
hrzandGitHub 02d66e8093 Merge pull request #1414 from myifeng/dev
修复 has_tool错误;优化检索复杂度
2025-05-30 16:03:36 +08:00
欣南科技andGitHub 49eab95be9 Merge pull request #1419 from xinnan-tech/fix-server
update:双麦实时打断
2025-05-30 15:48:27 +08:00
hrz b1bfaf5a5c update:双麦实时打断 2025-05-30 15:47:32 +08:00
myifeng 843e605352 handle_speaker_volume_or_screen_brightness 属于IOT功能函数,不作为必要函数自动注册 2025-05-30 14:33:29 +08:00
myifeng 6a7fa6060b 增加MCP协议测试功能 2025-05-30 14:26:52 +08:00
myifeng 43ce1df0ed 修复 has_tool错误;优化检索复杂度 2025-05-30 11:35:12 +08:00
myifengandGitHub a6deb3af8b Merge branch 'xinnan-tech:main' into dev 2025-05-30 10:27:23 +08:00
CGDandGitHub 9c01f5f0b0 Merge pull request #1411 from myifeng/dev
修正错误提交
2025-05-30 09:47:50 +08:00
欣南科技andGitHub 80b89a85ee Merge pull request #1412 from xinnan-tech/update-doc
update:优化文档
2025-05-30 09:39:05 +08:00
hrz 2fd0bb4912 update:优化文档 2025-05-30 09:37:13 +08:00
欣南科技andGitHub c0329618fe Merge pull request #1410 from xinnan-tech/update-doc
update:添加配置说明
2025-05-30 09:33:08 +08:00
hrz c4a240cb04 update:添加配置说明 2025-05-30 09:32:11 +08:00
myifeng 33d5761194 FIX 错误提交 2025-05-30 09:30:15 +08:00
myifeng a58ddd5880 为空判断处理; 2025-05-30 09:20:40 +08:00
CGDandGitHub 3bbb5f9b83 Merge pull request #1396 from myifeng/dev
监听MCP类型消息,等待后续适配
2025-05-30 09:11:57 +08:00
欣南科技andGitHub 3d6bf800f3 Merge pull request #1409 from xinnan-tech/asr_stream
豆包流式ASR
2025-05-30 02:17:36 +08:00
hrz a0b0a0e1e8 update:优化日志打印 2025-05-30 02:16:39 +08:00
hrz 8f48e9ac0c update:调通流式ASR 2025-05-30 02:05:34 +08:00
hrz bb42095ca0 update:兼容豆包流式ASR 2025-05-29 23:56:34 +08:00
myifeng c487f69676 FIX 为空判断 2025-05-29 17:55:01 +08:00
myifengandGitHub 3db46a84b2 Merge branch 'xinnan-tech:main' into dev 2025-05-29 16:13:13 +08:00
myifeng 69520391f2 参数为JSON 2025-05-29 15:41:18 +08:00
myifeng 4b212753fb 对接小智MCP协议控制 2025-05-29 13:42:31 +08:00
hrz d86a2cf9de update:抽离公共方法save_audio_to_file到父类 2025-05-29 10:38:01 +08:00
欣南科技andGitHub 625d079168 Merge pull request #1403 from xinnan-tech/hot-fix
update:修复退出卡壳问题
2025-05-28 21:35:33 +08:00
hrz 2586843654 update:修复退出卡壳问题 2025-05-28 21:34:21 +08:00
myifeng 8ea48f9875 独立处理MCP Message 2025-05-28 19:28:36 +08:00
myifeng d7f3b3caf5 接收并打印MCP数据 2025-05-28 18:32:15 +08:00
欣南科技andGitHub b37d8abf62 Merge pull request #1400 from xinnan-tech/hot-fix
update:优化资源释放判断
2025-05-28 18:23:15 +08:00
hrz cd930edd06 update:优化资源释放判断 2025-05-28 18:22:31 +08:00
欣南科技andGitHub 4956aa6d70 Merge pull request #1399 from xinnan-tech/hot-fix
update:增强TTS流会话关闭和开启逻辑
2025-05-28 18:14:19 +08:00
hrz fd17465d94 update:增强TTS流会话关闭和开启逻辑 2025-05-28 18:07:58 +08:00
欣南科技andGitHub 87a9425241 Merge pull request #1398 from xinnan-tech/huoshan_double_stream
update:优化第一句话
2025-05-28 16:20:47 +08:00
hrz b225e8afd5 update:优化第一句话 2025-05-28 16:19:21 +08:00
hrzandGitHub 297b6e69d5 Merge pull request #1397 from xinnan-tech/huoshan_double_stream
火山双向流式tts优化
2025-05-28 15:52:58 +08:00
王华侨 d7f0e88801 火山双向流式tts优化 2025-05-28 15:47:49 +08:00
myifeng d62b27814c 打印cmp类型消息,后期适配 2025-05-28 14:09:13 +08:00
myifeng 19526878b3 接收hello消息中的features信息 2025-05-28 11:49:18 +08:00
欣南科技andGitHub 14ecad720f update:修复web测试工具Maximum call stack size exceeded 的bug (#1394) 2025-05-28 11:35:32 +08:00
hrzandGitHub 81689c04a5 Merge pull request #1393 from wenbindu/main
修改注释
2025-05-28 11:30:06 +08:00
hrzandGitHub de6b9de341 Update silero.py 2025-05-28 11:29:48 +08:00
hrz 7f864eb84d update:修复web测试工具Maximum call stack size exceeded 的bug 2025-05-28 11:23:39 +08:00
wenbindu 4af0c1e2ce 修改注释 2025-05-28 10:59:59 +08:00
欣南科技andGitHub 6a4ed78812 Merge pull request #1386 from xinnan-tech/py_server_log
update:日志配置动态获取
2025-05-28 01:14:40 +08:00
hrzandGitHub 23b881f364 Merge branch 'main' into py_server_log 2025-05-28 01:13:25 +08:00
hrz 22054cf8bc update:更新日志标识位置 2025-05-28 01:00:48 +08:00
欣南科技andGitHub fc236b1d96 Merge pull request #1383 from xinnan-tech/tts-response
TTS流式框架改造:支持豆包语音双流式+支持传统非流式
2025-05-27 23:34:11 +08:00
hrz 4f81454f93 update:更新sql脚本 2025-05-27 23:30:07 +08:00
hrz b3d6f173f1 update:智控台支持火山双流式配置 2025-05-27 23:29:11 +08:00
hrz 8d2ba39ab8 优化双流式TTS时的声音 2025-05-27 18:51:08 +08:00
hrz 4260e5a1a7 update:优化音频播放 2025-05-27 18:06:44 +08:00
Sakura-RanChen 17585c8294 update:日志配置动态获取 2025-05-27 16:17:13 +08:00
hrzandGitHub d242a9e5c0 Merge branch 'main' into tts-response 2025-05-27 15:36:42 +08:00
欣南科技andGitHub 9ca98f6391 Merge pull request #1351 from xinnan-tech/py_memory_llm
update: 记忆模块使用独立LLM openai增加超参
2025-05-27 15:29:22 +08:00
hrz be7ef08f40 update:智控台添加独立记忆模型配置 2025-05-27 15:28:17 +08:00
hrz 626692df29 update:优化阿里云tts数字参数读取方式 2025-05-27 13:47:07 +08:00
hrz 631787a4f1 update:这几天tts流式改造合并后,self.executor的主要任务将是用来上报聊天记录,因此这里其实可以合并在self.executor 2025-05-27 13:41:53 +08:00
hrzandGitHub 09cf6cdc2d Merge pull request #1372 from tiamohummer/main
增加阿里云TTS private_voice 判断
2025-05-27 13:14:41 +08:00
hrzandGitHub baf174b059 Merge pull request #1223 from GOODDAYDAY/feature/muti_upload
feat: 增加上报线程池
2025-05-27 13:06:55 +08:00
tiamohummerandGitHub 49eb3178a9 Merge branch 'xinnan-tech:main' into main 2025-05-27 09:08:34 +08:00
hrzandGitHub 6a6aceff1d Tts respone update (#1382)
* update:优化

* update:添加编码解码

* update:优化流式tts

* update:优化

* update:优化

* update:旧非流失兼容改造

* update:优化线程

* update:优化

* update:优化公共方法

* update:优化火山双流式tts

* update:优化代码

* update:修改版本号

* update:合并双流式
2025-05-26 22:31:39 +08:00
hrz b8349af3da update:合并双流式 2025-05-26 22:30:45 +08:00
goodyhao 258d783f8e Merge remote-tracking branch 'origin/main' into feature/muti_upload 2025-05-26 20:28:41 +08:00
goodyhao 761fc05331 feat: 增加上报时间 2025-05-26 20:27:22 +08:00
hrz 0c8e943d1b update:修改版本号 2025-05-26 16:18:35 +08:00
hrz 9787ca60da update:优化代码 2025-05-26 16:12:38 +08:00
Sakura-RanChenandGitHub b8e57aeff4 Merge pull request #1377 from xinnan-tech/py_test_page.html_fix
Py test page.html fix
2025-05-26 15:43:12 +08:00
CGD 9a0240ef3e update:优化了页面样式 2025-05-26 15:34:21 +08:00
tiamohummerandGitHub cf5ccafe7e Merge branch 'xinnan-tech:main' into main 2025-05-26 13:42:37 +08:00
hrz ae64233986 update:优化火山双流式tts 2025-05-26 12:48:43 +08:00
hrz 6dda79ee10 update:优化公共方法 2025-05-26 11:57:55 +08:00
CGD 335a855968 Merge branch 'py_test_page.html_fix' of https://github.com/xinnan-tech/xiaozhi-esp32-server into py_test_page.html_fix 2025-05-26 11:26:44 +08:00
CGD 94c92c5f38 fix:修复测试页面音频重复播放问题 2025-05-26 11:23:42 +08:00
CGDandGitHub 35fa1493b3 Merge pull request #1373 from Minamiyama/ENH/test-page-localstorage
refactor(test_page.html): 优化UI布局和输入提示,增加本地存储功能
2025-05-26 11:13:36 +08:00
hrz f78f6fc529 update:优化 2025-05-26 11:04:13 +08:00
hrz 5be65216e2 update:优化线程 2025-05-26 10:44:35 +08:00
hrz 24526ad206 update:旧非流失兼容改造 2025-05-26 02:20:38 +08:00
Minamiyama d7564a65f7 refactor(test_page.html): 优化UI布局和输入提示,增加本地存储功能
- 在OTA和WebSocket服务器地址输入框中添加更详细的占位符提示
- 增加本地存储功能,保存并恢复OTA和WebSocket服务器地址
2025-05-25 15:24:06 +08:00
goodyhao 866d61cfaf Merge branch 'refs/heads/main' into feature/muti_upload
# Conflicts:
#	main/xiaozhi-server/core/connection.py
2025-05-25 13:22:51 +08:00
欣南科技andGitHub 3f3f3fdaa6 Merge pull request #1371 from xinnan-tech/hot-fix
修复测试页面音频重复播放
2025-05-25 09:10:47 +08:00
hrz e0da59096a 修复测试页面音频重复播放 2025-05-25 09:10:11 +08:00
hrz 40632019ac update:优化 2025-05-25 08:56:58 +08:00
hrzandGitHub 3c5563b62f Merge pull request #1369 from xinnan-tech/py_test
fix:修复“小智服务器测试页面播放语音问题”
2025-05-25 08:46:57 +08:00
CGD 562424b74d fix:修复“小智服务器测试页面播放语音问题” 2025-05-25 00:09:36 +08:00
hrz 7a598d5839 update:优化 2025-05-24 23:43:16 +08:00
hrz 574d34bc2c update:优化流式tts 2025-05-24 17:50:03 +08:00
hrz 16a4ccdb12 update:添加编码解码 2025-05-24 14:52:27 +08:00
hrz 920cf4f897 update:优化 2025-05-24 12:11:13 +08:00
欣南科技andGitHub 76ee2c5365 Merge pull request #1366 from xinnan-tech/update-fix
合并chat和chat_with_function_calling
2025-05-24 09:40:28 +08:00
hrz 472106390d update:优化音频播放方法 2025-05-24 09:39:10 +08:00
hrz d97f8b2e9a update:供应器管理更名为字段管理,防止混淆模型管理 2025-05-24 02:17:49 +08:00
hrz 98e2526f8a update:合并chat和chat_with_function_calling 2025-05-24 02:15:06 +08:00
tiamohummer c30c4649a4 增加阿里云TTS private_voice 判断 2025-05-23 17:41:22 +08:00
欣南科技andGitHub ede8676979 Merge pull request #1363 from xinnan-tech/hot-fix
update:修复iotbug
2025-05-23 16:10:21 +08:00
hrz 17fb60b7ae update:修复iotbug 2025-05-23 16:03:15 +08:00
hrz 8119897818 Merge branch 'main' into tts-response
# Conflicts:
#	main/xiaozhi-server/core/handle/intentHandler.py
2025-05-23 09:15:15 +08:00
欣南科技andGitHub 0a9e5d2cea Merge pull request #1357 from xinnan-tech/update-docs
update:更新视频教程
2025-05-22 22:23:21 +08:00
hrz a5f74d7767 update:更新视频教程 2025-05-22 22:22:43 +08:00
Sakura-RanChen 02cb9c35b3 update: 记忆模块使用独立LLM openai增加超参 2025-05-22 17:14:58 +08:00
hrz fdc6dcb26e update:去掉无用代码 2025-05-21 16:25:22 +08:00
hrz 9fd010e91e update:调试 2025-05-21 15:59:24 +08:00
hrz c900498ce8 update:合并main分支 2025-05-21 15:55:40 +08:00
hrz c4c84e44e1 update:合并main分支 2025-05-21 14:52:24 +08:00
hrz 191ac47353 合并main分支 2025-05-21 14:13:52 +08:00
hrz 851365fb58 update:合并非tts代码 2025-05-21 13:18:12 +08:00
hrz ede2bc6a4e update:更新manager-api模块 2025-05-21 11:55:56 +08:00
hrz 38780b6daa update:忽略本地文件 2025-05-21 11:46:26 +08:00
goodyhao 885b7a0b05 feat: 增加上报线程池 2025-05-13 15:54:00 +08:00
CoderMonkeyandGitHub 92e117c10a Merge pull request #737 from HiCoderMonkey/tts-response
Tts response 双向流tts,fix 播放音乐
2025-04-11 11:12:12 +08:00
lizhongxiang 71db92ef61 fix: 播放音乐 2025-04-11 11:08:22 +08:00
lizhongxiang 118bb728a9 fix: 播放咋音问题 2025-04-11 10:25:43 +08:00
hrz d7aac5e10f Merge branch 'main' into tts-response
# Conflicts:
#	main/manager-web/.env.development
2025-04-09 21:46:59 +08:00
CoderMonkeyandGitHub 4359f1d5b3 Merge pull request #719 from HiCoderMonkey/tts-response
Tts response ,fix 播放有咋音
2025-04-09 14:38:02 +08:00
lizhongxiang 8bb888c58e fix: 播放咋音问题 2025-04-09 14:32:32 +08:00
lizhongxiang 4c81585e32 merge oeigin/tts-response 2025-04-07 18:15:09 +08:00
hrz 0e6d1cd677 Merge branch 'main' into tts-response 2025-04-04 02:02:30 +08:00
hrz 98368cbe06 fix:文字回复卡住bug 2025-04-04 02:01:48 +08:00
hrz 0b4a4df1af update:合并最新代码 2025-04-04 00:27:04 +08:00
hrz c8a3d378b7 update:合并main分支 2025-04-03 23:06:59 +08:00
CoderMonkeyandGitHub aa34473560 Merge pull request #640 from HiCoderMonkey/tts-response
fix:删除多余方法
2025-04-02 11:37:23 +08:00
lizhongxiang f479a789db fix:删除多余方法 2025-04-02 11:35:49 +08:00
CoderMonkeyandGitHub d4ce501c75 Merge pull request #639 from HiCoderMonkey/tts-response
fix:播放音乐不会停
2025-04-02 11:26:43 +08:00
lizhongxiang c2643d0b80 fix:merge main 2025-04-02 11:18:26 +08:00
lizhongxiang 078605c188 fix:播放音乐不会停 2025-04-02 11:11:16 +08:00
lizhongxiang cd3e9ed494 fix:播放音乐不会停 2025-04-02 11:07:04 +08:00
CoderMonkeyandGitHub 8a5a9cc0b3 Merge pull request #625 from HiCoderMonkey/tts-response
fix:火山声音复刻
2025-04-01 15:09:12 +08:00
lizhongxiang 347fc4ca44 fix:火上声音复刻 2025-04-01 15:07:16 +08:00
hrz 61db1192e2 update:格式化代码 2025-04-01 14:27:17 +08:00
CoderMonkeyandGitHub d9d973b6aa Merge pull request #612 from HiCoderMonkey/tts-response
fix:aliyun,cozecn,custom,doubao,gpt_sovits_v2,gpt_sovits_v3,minimax,o…
2025-03-31 16:32:08 +08:00
lizhongxiang cc4aed5a8e fix:aliyun,cozecn,custom,doubao,gpt_sovits_v2,gpt_sovits_v3,minimax,openai,siliconflow,ttson,这些tts兼容 2025-03-31 16:31:06 +08:00
CoderMonkeyandGitHub b44d52fb3e Merge pull request #610 from HiCoderMonkey/tts-response
fix:普通聊天调试
2025-03-31 15:35:33 +08:00
lizhongxiang 93265be974 fix:普通聊天调试 2025-03-31 15:33:20 +08:00
CoderMonkeyandGitHub 2028405af6 Merge pull request #606 from HiCoderMonkey/tts-response
tts 双向流改造
2025-03-31 15:21:38 +08:00
lizhongxiang bb9df3f54c add:双向流tts结构改造,火山双向tts语音流式输入输出 2025-03-31 15:15:59 +08:00
lizhongxiang 264487574b add:火山双向tts语音流式输入输出 2025-03-29 14:17:09 +08:00
CoderMonkeyandGitHub 6a00895af8 Merge pull request #443 from HiCoderMonkey/tts-response
Tts 改造
2025-03-20 10:16:47 +08:00
lizhongxiang 1136ca4b24 mergin main 2025-03-20 10:14:49 +08:00
lizhongxiang b153bc11f2 mergin main,速度提升一下 2025-03-20 09:39:25 +08:00
lizhongxiang a067a6ba87 fix: 流式tts,function聊天时候的支持 2025-03-13 17:50:29 +08:00
lizhongxiang c271b88c87 fix: 流式tts,function聊天时候的支持 2025-03-13 17:44:10 +08:00
lizhongxiang 1dfa619243 fix: 流式流畅度问题,英文符号切分,空格切分 2025-03-13 15:30:00 +08:00
lizhongxiang 91bdc05a38 merge main 2025-03-13 14:24:18 +08:00
lizhongxiang 0e8edfe002 merge main,fix gitignore 2025-03-10 11:12:17 +08:00
lizhongxiang c1084923f7 merge main 2025-03-10 10:51:25 +08:00
lizhongxiang b2381fc689 add: 语音流式输出 2025-03-05 17:29:26 +08:00
lizhongxiang fa75f56ffb add: 语音流式输出 2025-03-05 17:26:32 +08:00
99 changed files with 4316 additions and 1450 deletions
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@@ -107,6 +107,7 @@ celerybeat.pid
*.sage.py *.sage.py
# Environments # Environments
.env
.venv .venv
env/ env/
venv/ venv/
@@ -145,24 +146,31 @@ tmp
.history .history
.DS_Store .DS_Store
main/xiaozhi-server/data main/xiaozhi-server/data
main/xiaozhi-server/config/assets/wakeup_words.*
main/manager-web/node_modules main/manager-web/node_modules
.config.yaml .config.yaml
.secrets.yaml .secrets.yaml
.private_config.yaml .private_config.yaml
.env.development
# model files # model files
main/xiaozhi-server/models/SenseVoiceSmall/model.pt main/xiaozhi-server/models/SenseVoiceSmall/model.pt
main/xiaozhi-server/models/sherpa-onnx* main/xiaozhi-server/models/sherpa-onnx*
/main/xiaozhi-server/audio_ref/
/audio_ref/
/asr-models/iic/SenseVoiceSmall/
/main/xiaozhi-server/asr-models/iic/SenseVoiceSmall/
/models/SenseVoiceSmall/model.pt
my_wakeup_words.mp3 my_wakeup_words.mp3
!main/xiaozhi-server/config/assets/bind_code.wav !main/xiaozhi-server/config/assets/bind_code.wav
!main/xiaozhi-server/config/assets/wakeup_words.wav
!main/xiaozhi-server/config/assets/bind_not_found.wav !main/xiaozhi-server/config/assets/bind_not_found.wav
!main/xiaozhi-server/config/assets/bind_code/*.wav !main/xiaozhi-server/config/assets/bind_code/*.wav
!main/xiaozhi-server/config/assets/max_output_size.wav !main/xiaozhi-server/config/assets/max_output_size.wav
main/manager-api/.vscode main/manager-api/.vscode
# Ignore webpack cache directory # Ignore webpack cache directory
main/manager-web/.webpack_cache/ main/manager-web/.webpack_cache/
main/xiaozhi-server/mysql main/xiaozhi-server/mysql
uploadfile uploadfile
*.json
.vscode .vscode
+46 -5
View File
@@ -121,6 +121,33 @@
</a> </a>
</td> </td>
</tr> </tr>
<tr>
<td>
<a href="https://www.bilibili.com/video/BV12J7WzBEaH" target="_blank">
<picture>
<img alt="实时打断" src="docs/images/demo10.png" />
</picture>
</a>
</td>
<td>
<a href="https://www.bilibili.com/video/BV1Co76z7EvK" target="_blank">
<picture>
<img alt="拍照识物品" src="docs/images/demo12.png" />
</picture>
</a>
</td>
<td>
<a href="https://www.bilibili.com/video/BV1TJ7WzzEo6" target="_blank">
<picture>
<img alt="多指令任务" src="docs/images/demo11.png" />
</picture>
</a>
</td>
<td>
</td>
<td>
</td>
</tr>
</table> </table>
--- ---
@@ -141,11 +168,11 @@
本项目提供两种部署方式,请根据您的具体需求选择: 本项目提供两种部署方式,请根据您的具体需求选择:
#### 🚀 部署方式选择 #### 🚀 部署方式选择
| 部署方式 | 特点 | 适用场景 | 部署文档 | 配置要求 | 视频教程 |
|---------|------|---------|---------|---------|---------|
| **最简化安装** | 智能对话、IOT功能,数据存储在配置文件 | 低配置环境,无需数据库 | [Docker版](./docs/Deployment.md#%E6%96%B9%E5%BC%8F%E4%B8%80docker%E5%8F%AA%E8%BF%90%E8%A1%8Cserver) / [源码部署](./docs/Deployment.md#%E6%96%B9%E5%BC%8F%E4%BA%8C%E6%9C%AC%E5%9C%B0%E6%BA%90%E7%A0%81%E5%8F%AA%E8%BF%90%E8%A1%8Cserver)| 如果使用`FunASR`要2核4G,如果全API,要2核2G | - |
| **全模块安装** | 智能对话、IOT、OTA、智控台,数据存储在数据库 | 完整功能体验 |[Docker版](./docs/Deployment_all.md#%E6%96%B9%E5%BC%8F%E4%B8%80docker%E8%BF%90%E8%A1%8C%E5%85%A8%E6%A8%A1%E5%9D%97) / [源码部署](./docs/Deployment_all.md#%E6%96%B9%E5%BC%8F%E4%BA%8C%E6%9C%AC%E5%9C%B0%E6%BA%90%E7%A0%81%E8%BF%90%E8%A1%8C%E5%85%A8%E6%A8%A1%E5%9D%97) | 如果使用`FunASR`要4核8G,如果全API,要2核4G| [本地源码视频教程](https://www.bilibili.com/video/BV1wBJhz4Ewe) |
| 部署方式 | 特点 | 适用场景 | Docker部署文档 | 源码部署文档 |
|---------|------|---------|---------|---------|
| **最简化安装** | 智能对话、IOT功能,数据存储在配置文件 | 低配置环境,无需数据库 | [Docker只运行Server](./docs/Deployment.md#%E6%96%B9%E5%BC%8F%E4%B8%80docker%E5%8F%AA%E8%BF%90%E8%A1%8Cserver) | [本地源码只运行Server](./docs/Deployment.md#%E6%96%B9%E5%BC%8F%E4%BA%8C%E6%9C%AC%E5%9C%B0%E6%BA%90%E7%A0%81%E5%8F%AA%E8%BF%90%E8%A1%8Cserver)|
| **全模块安装** | 智能对话、IOT、OTA、智控台,数据存储在数据库 | 完整功能体验 |[Docker运行全模块](./docs/Deployment_all.md#%E6%96%B9%E5%BC%8F%E4%B8%80docker%E8%BF%90%E8%A1%8C%E5%85%A8%E6%A8%A1%E5%9D%97) | [本地源码运行全模块](./docs/Deployment_all.md#%E6%96%B9%E5%BC%8F%E4%BA%8C%E6%9C%AC%E5%9C%B0%E6%BA%90%E7%A0%81%E8%BF%90%E8%A1%8C%E5%85%A8%E6%A8%A1%E5%9D%97) |
> 💡 提示:以下是按最新代码部署后的测试平台,有需要可烧录测试,并发为6个,每天会清空数据 > 💡 提示:以下是按最新代码部署后的测试平台,有需要可烧录测试,并发为6个,每天会清空数据
@@ -157,9 +184,23 @@ OTA接口地址: https://2662r3426b.vicp.fun/xiaozhi/ota/
Websocket接口地址: wss://2662r3426b.vicp.fun/xiaozhi/v1/ Websocket接口地址: wss://2662r3426b.vicp.fun/xiaozhi/v1/
``` ```
#### 🚩 配置说明和推荐
> [!Note]
> 本项目默认的配置是`入门全免费`设置,如果想效果更优,推荐使用`全流式配置`。
>
> 本项目自`0.5.2`版本,已支持整个生命周期全流式,相比`0.5`版本以前,响应速度提升约`2.5秒`
| 模块名称 | 入门全免费设置 | 全流式配置 |
|---------|---------|------|
| ASR(语音识别) | FunASR(本地) | ✅DoubaoASR(火山流式语音识别) |
| LLM(大模型) | ChatGLMLLM(智谱glm-4-flash) | ✅DoubaoLLM(火山doubao-1-5-pro-32k-250115) |
| TTS(语音合成) | EdgeTTS(微软语音) | ✅HuoshanDoubleStreamTTS(火山双流式语音合成) |
| Intent(意图识别) | function_call(函数调用) | ✅function_call(函数调用) |
| Memory(记忆功能) | mem_local_short(本地短期记忆) | ✅mem_local_short(本地短期记忆) |
--- ---
## 功能清单 ✨ ## 功能清单 ✨
### 已实现 ✅ ### 已实现 ✅
| 功能模块 | 描述 | | 功能模块 | 描述 |
+5 -1
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@@ -108,7 +108,11 @@ VAD:
参考教程[阿里云短信集成指南](./ali-sms-integration.md) 参考教程[阿里云短信集成指南](./ali-sms-integration.md)
### 9、更多问题,可联系我们反馈 💬 ### 9、如何开启视觉模型实现拍照识物 📷
参考教程[视觉模型使用指南](./mcp-vision-integration.md)
### 10、更多问题,可联系我们反馈 💬
可以在[issues](https://github.com/xinnan-tech/xiaozhi-esp32-server/issues)提交您的问题。 可以在[issues](https://github.com/xinnan-tech/xiaozhi-esp32-server/issues)提交您的问题。
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@@ -0,0 +1,171 @@
# 视觉模型使用指南
本教程分为两部分:
- 第一部分:单模块运行xiaozhi-server开启视觉模型
- 第二部分:全模块运行时,如何开启视觉模型
开启视觉模型前,你需要准备三件事:
- 你需要准备一台带摄像头的设备,而且这台设备已经在虾哥仓库里,实现了调用摄像头功能。例如`立创·实战派ESP32-S3开发板`
- 你设备固件的版本升级到1.6.6及以上
- 你已经成功跑通基础对话模块
## 单模块运行xiaozhi-server开启视觉模型
### 第一步确认网络
由于视觉模型会默认启动8003端口。
如果你是docker运行,请确认一下你的`docker-compose.yml`是否放了`8003`端口,如果没有就更新最新的`docker-compose.yml`文件
如果你是源码运行,确认防火墙是否放行`8003`端口
### 第二步选择你的视觉模型
打开你的`data/.config.yaml`文件,设置你的`selected_module.VLLM`设置为某个视觉模型。目前我们已经支持`openai`类型接口的视觉模型。`ChatGLMVLLM`就是其中一款兼容`openai`的模型。
```
selected_module:
VAD: ..
ASR: ..
LLM: ..
VLLM: ChatGLMVLLM
TTS: ..
Memory: ..
Intent: ..
```
假设我们使用`ChatGLMVLLM`作为视觉模型,那我们需要先登录[智谱AI](https://bigmodel.cn/usercenter/proj-mgmt/apikeys)网站,申请密钥。如果你之前已经申请过了密钥,可以复用这个密钥。
在你的配置文件中,增加这个配置,如果已经有了这个配置,就设置好你的api_key。
```
VLLM:
ChatGLMVLLM:
api_key: 你的api_key
```
### 第三步启动xiaozhi-server服务
如果你是源码,就输入命令启动
```
python app.py
```
如果你是docker运行,就重启容器
```
docker restart xiaozhi-esp32-server
```
启动后会输出以下内容的日志。
```
2025-06-01 **** - OTA接口是 http://192.168.4.7:8003/xiaozhi/ota/
2025-06-01 **** - 视觉分析接口是 http://192.168.4.7:8003/mcp/vision/explain
2025-06-01 **** - Websocket地址是 ws://192.168.4.7:8000/xiaozhi/v1/
2025-06-01 **** - =======上面的地址是websocket协议地址,请勿用浏览器访问=======
2025-06-01 **** - 如想测试websocket请用谷歌浏览器打开test目录下的test_page.html
2025-06-01 **** - =============================================================
```
启动后,使用使用浏览器打开日志里`视觉分析接口`连接。看看输出了什么?如果你是linux,没有浏览器,你可以执行这个命令:
```
curl -i 你的视觉分析接口
```
正常来说会这样显示
```
MCP Vision 接口运行正常,视觉解释接口地址是:http://xxxx:8003/mcp/vision/explain
```
请注意,如果你是公网部署,或者docker部署,一定要改一下你的`data/.config.yaml`里这个配置
```
server:
vision_explain: http://你的ip或者域名:端口号/mcp/vision/explain
```
为什么呢?因为视觉解释接口需要下发到设备,如果你的地址是局域网地址,或者是docker内部地址,设备是无法访问的。
假设你的公网地址是`111.111.111.111`,那么`vision_explain`应该这么配
```
server:
vision_explain: http://111.111.111.111:8003/mcp/vision/explain
```
如果你的MCP Vision 接口运行正常,且你也试着用浏览器访问正常打开下发的`视觉解释接口地址`,请继续下一步
### 第四步 设备唤醒开启
对设备说“请打开摄像头,说你你看到了什么”
留意xiaozhi-server的日志输出,看看有没有报错。
## 全模块运行时,如何开启视觉模型
### 第一步 确认网络
由于视觉模型会默认启动8003端口。
如果你是docker运行,请确认一下你的`docker-compose_all.yml`是否映射了`8003`端口,如果没有就更新最新的`docker-compose_all.yml`文件
如果你是源码运行,确认防火墙是否放行`8003`端口
### 第二步 确认你配置文件
打开你的`data/.config.yaml`文件,确认一下你的配置文件的结构,是否和`data/config_from_api.yaml`一样。如果不一样,或缺少某项,请补齐。
### 第三步 配置视觉模型密钥
那我们需要先登录[智谱AI](https://bigmodel.cn/usercenter/proj-mgmt/apikeys)网站,申请密钥。如果你之前已经申请过了密钥,可以复用这个密钥。
登录`智控台`,顶部菜单点击`模型配置`,在左侧栏点击`视觉打语言模型`,找到`VLLM_ChatGLMVLLM`,点击修改按钮,在弹框中,在`API密钥`输入你密钥,点击保存。
保存成功后,去到你需要测试的智能体哪里,点击`配置角色`,在打开的内容里,查看`视觉大语言模型(VLLM)`是否选择了刚才的视觉模型。点击保存。
### 第三步 启动xiaozhi-server模块
如果你是源码,就输入命令启动
```
python app.py
```
如果你是docker运行,就重启容器
```
docker restart xiaozhi-esp32-server
```
启动后会输出以下内容的日志。
```
2025-06-01 **** - 视觉分析接口是 http://192.168.4.7:8003/mcp/vision/explain
2025-06-01 **** - Websocket地址是 ws://192.168.4.7:8000/xiaozhi/v1/
2025-06-01 **** - =======上面的地址是websocket协议地址,请勿用浏览器访问=======
2025-06-01 **** - 如想测试websocket请用谷歌浏览器打开test目录下的test_page.html
2025-06-01 **** - =============================================================
```
启动后,使用使用浏览器打开日志里`视觉分析接口`连接。看看输出了什么?如果你是linux,没有浏览器,你可以执行这个命令:
```
curl -i 你的视觉分析接口
```
正常来说会这样显示
```
MCP Vision 接口运行正常,视觉解释接口地址是:http://xxxx:8003/mcp/vision/explain
```
请注意,如果你是公网部署,或者docker部署,一定要改一下你的`data/.config.yaml`里这个配置
```
server:
vision_explain: http://你的ip或者域名:端口号/mcp/vision/explain
```
为什么呢?因为视觉解释接口需要下发到设备,如果你的地址是局域网地址,或者是docker内部地址,设备是无法访问的。
假设你的公网地址是`111.111.111.111`,那么`vision_explain`应该这么配
```
server:
vision_explain: http://111.111.111.111:8003/mcp/vision/explain
```
如果你的MCP Vision 接口运行正常,且你也试着用浏览器访问正常打开下发的`视觉解释接口地址`,请继续下一步
### 第四步 设备唤醒开启
对设备说“请打开摄像头,说你你看到了什么”
留意xiaozhi-server的日志输出,看看有没有报错。
+1 -1
View File
@@ -258,4 +258,4 @@
</plugin> </plugin>
</plugins> </plugins>
</build> </build>
</project> </project>
@@ -227,7 +227,7 @@ public interface Constant {
/** /**
* 版本号 * 版本号
*/ */
public static final String VERSION = "0.4.4"; public static final String VERSION = "0.5.2";
/** /**
* 无效固件URL * 无效固件URL
@@ -4,7 +4,6 @@ import java.util.List;
import java.util.Objects; import java.util.Objects;
import org.apache.shiro.authz.UnauthorizedException; import org.apache.shiro.authz.UnauthorizedException;
import org.springframework.context.support.DefaultMessageSourceResolvable;
import org.springframework.dao.DuplicateKeyException; import org.springframework.dao.DuplicateKeyException;
import org.springframework.validation.ObjectError; import org.springframework.validation.ObjectError;
import org.springframework.web.bind.MethodArgumentNotValidException; import org.springframework.web.bind.MethodArgumentNotValidException;
@@ -28,4 +28,6 @@ public class AgentChatHistoryReportDTO {
private String content; private String content;
@Schema(description = "base64编码的opus音频数据", example = "") @Schema(description = "base64编码的opus音频数据", example = "")
private String audioBase64; private String audioBase64;
@Schema(description = "上报时间,十位时间戳,空时默认使用当前时间", example = "1745657732")
private Long reportTime;
} }
@@ -27,6 +27,9 @@ public class AgentDTO {
@Schema(description = "大语言模型名称", example = "llm_model_01") @Schema(description = "大语言模型名称", example = "llm_model_01")
private String llmModelName; private String llmModelName;
@Schema(description = "视觉模型名称", example = "vllm_model_01")
private String vllmModelName;
@Schema(description = "记忆模型ID", example = "mem_model_01") @Schema(description = "记忆模型ID", example = "mem_model_01")
private String memModelId; private String memModelId;
@@ -36,6 +36,9 @@ public class AgentEntity {
@Schema(description = "大语言模型标识") @Schema(description = "大语言模型标识")
private String llmModelId; private String llmModelId;
@Schema(description = "VLLM模型标识")
private String vllmModelId;
@Schema(description = "语音合成模型标识") @Schema(description = "语音合成模型标识")
private String ttsModelId; private String ttsModelId;
@@ -49,6 +49,11 @@ public class AgentTemplateEntity implements Serializable {
*/ */
private String llmModelId; private String llmModelId;
/**
* VLLM模型标识
*/
private String vllmModelId;
/** /**
* 语音合成模型标识 * 语音合成模型标识
*/ */
@@ -27,4 +27,4 @@ public interface AgentChatAudioService extends IService<AgentChatAudioEntity> {
* @return 音频数据 * @return 音频数据
*/ */
byte[] getAudio(String audioId); byte[] getAudio(String audioId);
} }
@@ -47,7 +47,8 @@ public class AgentChatHistoryBizServiceImpl implements AgentChatHistoryBizServic
public Boolean report(AgentChatHistoryReportDTO report) { public Boolean report(AgentChatHistoryReportDTO report) {
String macAddress = report.getMacAddress(); String macAddress = report.getMacAddress();
Byte chatType = report.getChatType(); Byte chatType = report.getChatType();
log.info("小智设备聊天上报请求: macAddress={}, type={}", macAddress, chatType); Long reportTimeMillis = null != report.getReportTime() ? report.getReportTime() * 1000 : System.currentTimeMillis();
log.info("小智设备聊天上报请求: macAddress={}, type={} reportTime={}", macAddress, chatType, reportTimeMillis);
// 根据设备MAC地址查询对应的默认智能体,判断是否需要上报 // 根据设备MAC地址查询对应的默认智能体,判断是否需要上报
AgentEntity agentEntity = agentService.getDefaultAgentByMacAddress(macAddress); AgentEntity agentEntity = agentService.getDefaultAgentByMacAddress(macAddress);
@@ -59,10 +60,10 @@ public class AgentChatHistoryBizServiceImpl implements AgentChatHistoryBizServic
String agentId = agentEntity.getId(); String agentId = agentEntity.getId();
if (Objects.equals(chatHistoryConf, Constant.ChatHistoryConfEnum.RECORD_TEXT.getCode())) { if (Objects.equals(chatHistoryConf, Constant.ChatHistoryConfEnum.RECORD_TEXT.getCode())) {
saveChatText(report, agentId, macAddress, null); saveChatText(report, agentId, macAddress, null, reportTimeMillis);
} else if (Objects.equals(chatHistoryConf, Constant.ChatHistoryConfEnum.RECORD_TEXT_AUDIO.getCode())) { } else if (Objects.equals(chatHistoryConf, Constant.ChatHistoryConfEnum.RECORD_TEXT_AUDIO.getCode())) {
String audioId = saveChatAudio(report); String audioId = saveChatAudio(report);
saveChatText(report, agentId, macAddress, audioId); saveChatText(report, agentId, macAddress, audioId, reportTimeMillis);
} }
// 更新设备最后对话时间 // 更新设备最后对话时间
@@ -92,8 +93,7 @@ public class AgentChatHistoryBizServiceImpl implements AgentChatHistoryBizServic
/** /**
* 组装上报数据 * 组装上报数据
*/ */
private void saveChatText(AgentChatHistoryReportDTO report, String agentId, String macAddress, String audioId) { private void saveChatText(AgentChatHistoryReportDTO report, String agentId, String macAddress, String audioId, Long reportTime) {
// 构建聊天记录实体 // 构建聊天记录实体
AgentChatHistoryEntity entity = AgentChatHistoryEntity.builder() AgentChatHistoryEntity entity = AgentChatHistoryEntity.builder()
.macAddress(macAddress) .macAddress(macAddress)
@@ -102,6 +102,8 @@ public class AgentChatHistoryBizServiceImpl implements AgentChatHistoryBizServic
.chatType(report.getChatType()) .chatType(report.getChatType())
.content(report.getContent()) .content(report.getContent())
.audioId(audioId) .audioId(audioId)
.createdAt(new Date(reportTime))
// NOTE(haotian): 2025/5/26 updateAt可以不设置,重点是createAt,而且这样可以看到上报延迟
.build(); .build();
// 保存数据 // 保存数据
@@ -31,4 +31,4 @@ public class AgentChatAudioServiceImpl extends ServiceImpl<AiAgentChatAudioDao,
AgentChatAudioEntity entity = getById(audioId); AgentChatAudioEntity entity = getById(audioId);
return entity != null ? entity.getAudio() : null; return entity != null ? entity.getAudio() : null;
} }
} }
@@ -102,6 +102,9 @@ public class AgentServiceImpl extends BaseServiceImpl<AgentDao, AgentEntity> imp
// 获取 LLM 模型名称 // 获取 LLM 模型名称
dto.setLlmModelName(modelConfigService.getModelNameById(agent.getLlmModelId())); dto.setLlmModelName(modelConfigService.getModelNameById(agent.getLlmModelId()));
// 获取 VLLM 模型名称
dto.setVllmModelName(modelConfigService.getModelNameById(agent.getVllmModelId()));
// 获取记忆模型名称 // 获取记忆模型名称
dto.setMemModelId(agent.getMemModelId()); dto.setMemModelId(agent.getMemModelId());
@@ -72,6 +72,7 @@ public class ConfigServiceImpl implements ConfigService {
null, null,
null, null,
null, null,
null,
result, result,
isCache); isCache);
@@ -140,6 +141,7 @@ public class ConfigServiceImpl implements ConfigService {
agent.getVadModelId(), agent.getVadModelId(),
agent.getAsrModelId(), agent.getAsrModelId(),
agent.getLlmModelId(), agent.getLlmModelId(),
agent.getVllmModelId(),
agent.getTtsModelId(), agent.getTtsModelId(),
agent.getMemModelId(), agent.getMemModelId(),
agent.getIntentModelId(), agent.getIntentModelId(),
@@ -241,6 +243,7 @@ public class ConfigServiceImpl implements ConfigService {
String vadModelId, String vadModelId,
String asrModelId, String asrModelId,
String llmModelId, String llmModelId,
String vllmModelId,
String ttsModelId, String ttsModelId,
String memModelId, String memModelId,
String intentModelId, String intentModelId,
@@ -248,9 +251,10 @@ public class ConfigServiceImpl implements ConfigService {
boolean isCache) { boolean isCache) {
Map<String, String> selectedModule = new HashMap<>(); Map<String, String> selectedModule = new HashMap<>();
String[] modelTypes = { "VAD", "ASR", "TTS", "Memory", "Intent", "LLM" }; String[] modelTypes = { "VAD", "ASR", "TTS", "Memory", "Intent", "LLM", "VLLM" };
String[] modelIds = { vadModelId, asrModelId, ttsModelId, memModelId, intentModelId, llmModelId }; String[] modelIds = { vadModelId, asrModelId, ttsModelId, memModelId, intentModelId, llmModelId, vllmModelId };
String intentLLMModelId = null; String intentLLMModelId = null;
String memLocalShortLLMModelId = null;
for (int i = 0; i < modelIds.length; i++) { for (int i = 0; i < modelIds.length; i++) {
if (modelIds[i] == null) { if (modelIds[i] == null) {
@@ -269,7 +273,7 @@ public class ConfigServiceImpl implements ConfigService {
Map<String, Object> map = (Map<String, Object>) model.getConfigJson(); Map<String, Object> map = (Map<String, Object>) model.getConfigJson();
if ("intent_llm".equals(map.get("type"))) { if ("intent_llm".equals(map.get("type"))) {
intentLLMModelId = (String) map.get("llm"); intentLLMModelId = (String) map.get("llm");
if (intentLLMModelId != null && intentLLMModelId.equals(llmModelId)) { if (StringUtils.isNotBlank(intentLLMModelId) && intentLLMModelId.equals(llmModelId)) {
intentLLMModelId = null; intentLLMModelId = null;
} }
} }
@@ -281,10 +285,31 @@ public class ConfigServiceImpl implements ConfigService {
} }
} }
} }
if ("Memory".equals(modelTypes[i])) {
Map<String, Object> map = (Map<String, Object>) model.getConfigJson();
if ("mem_local_short".equals(map.get("type"))) {
memLocalShortLLMModelId = (String) map.get("llm");
if (StringUtils.isNotBlank(memLocalShortLLMModelId)
&& memLocalShortLLMModelId.equals(llmModelId)) {
memLocalShortLLMModelId = null;
}
}
}
// 如果是LLM类型,且intentLLMModelId不为空,则添加附加模型 // 如果是LLM类型,且intentLLMModelId不为空,则添加附加模型
if ("LLM".equals(modelTypes[i]) && intentLLMModelId != null) { if ("LLM".equals(modelTypes[i])) {
ModelConfigEntity intentLLM = modelConfigService.getModelById(intentLLMModelId, isCache); if (StringUtils.isNotBlank(intentLLMModelId)) {
typeConfig.put(intentLLM.getId(), intentLLM.getConfigJson()); if (!typeConfig.containsKey(intentLLMModelId)) {
ModelConfigEntity intentLLM = modelConfigService.getModelById(intentLLMModelId, isCache);
typeConfig.put(intentLLM.getId(), intentLLM.getConfigJson());
}
}
if (StringUtils.isNotBlank(memLocalShortLLMModelId)) {
if (!typeConfig.containsKey(memLocalShortLLMModelId)) {
ModelConfigEntity memLocalShortLLM = modelConfigService
.getModelById(memLocalShortLLMModelId, isCache);
typeConfig.put(memLocalShortLLM.getId(), memLocalShortLLM.getConfigJson());
}
}
} }
} }
result.put(modelTypes[i], typeConfig); result.put(modelTypes[i], typeConfig);
@@ -48,4 +48,4 @@ spring:
max-active: 8 # 连接池最大连接数(使用负值表示没有限制) max-active: 8 # 连接池最大连接数(使用负值表示没有限制)
max-idle: 8 # 连接池中的最大空闲连接 max-idle: 8 # 连接池中的最大空闲连接
min-idle: 0 # 连接池中的最小空闲连接 min-idle: 0 # 连接池中的最小空闲连接
shutdown-timeout: 100ms # 客户端优雅关闭的等待时间 shutdown-timeout: 100ms # 客户端优雅关闭的等待时间
@@ -0,0 +1,51 @@
-- 本地短期记忆配置可以设置独立的LLM
update `ai_model_provider` set fields = '[{"key":"llm","label":"LLM模型","type":"string"}]' where id = 'SYSTEM_Memory_mem_local_short';
update `ai_model_config` set config_json = '{\"type\": \"mem_local_short\", \"llm\": \"LLM_ChatGLMLLM\"}' where id = 'Memory_mem_local_short';
-- 增加火山双流式TTS供应器和模型配置
delete from `ai_model_provider` where id = 'SYSTEM_TTS_HSDSTTS';
INSERT INTO `ai_model_provider` (`id`, `model_type`, `provider_code`, `name`, `fields`, `sort`, `creator`, `create_date`, `updater`, `update_date`) VALUES
('SYSTEM_TTS_HSDSTTS', 'TTS', 'huoshan_double_stream', '火山双流式语音合成', '[{"key":"ws_url","label":"WebSocket地址","type":"string"},{"key":"appid","label":"应用ID","type":"string"},{"key":"access_token","label":"访问令牌","type":"string"},{"key":"resource_id","label":"资源ID","type":"string"},{"key":"speaker","label":"默认音色","type":"string"}]', 13, 1, NOW(), 1, NOW());
delete from `ai_model_config` where id = 'TTS_HuoshanDoubleStreamTTS';
INSERT INTO `ai_model_config` VALUES ('TTS_HuoshanDoubleStreamTTS', 'TTS', 'HuoshanDoubleStreamTTS', '火山双流式语音合成', 0, 1, '{\"type\": \"huoshan_double_stream\", \"ws_url\": \"wss://openspeech.bytedance.com/api/v3/tts/bidirection\", \"appid\": \"你的火山引擎语音合成服务appid\", \"access_token\": \"你的火山引擎语音合成服务access_token\", \"resource_id\": \"volc.service_type.10029\", \"speaker\": \"zh_female_wanwanxiaohe_moon_bigtts\"}', NULL, NULL, 16, NULL, NULL, NULL, NULL);
-- 火山双流式TT模型配置说明文档
UPDATE `ai_model_config` SET
`doc_link` = 'https://console.volcengine.com/speech/service/10007',
`remark` = '火山引擎语音合成服务配置说明:
1. 访问 https://www.volcengine.com/ 注册并开通火山引擎账号
2. 访问 https://console.volcengine.com/speech/service/10007 开通语音合成大模型,购买音色
3. 在页面底部获取appid和access_token
5. 资源ID固定为:volc.service_type.10029(大模型语音合成及混音)
6. 填入配置文件中' WHERE `id` = 'TTS_HuoshanDoubleStreamTTS';
-- 添加火山双流式TTS音色
delete from `ai_tts_voice` where tts_model_id = 'TTS_HuoshanDoubleStreamTTS';
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0001', 'TTS_HuoshanDoubleStreamTTS', '爽快思思/Skye', 'zh_female_shuangkuaisisi_moon_bigtts', '中文、英文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/Skye.mp3', NULL, 1, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0002', 'TTS_HuoshanDoubleStreamTTS', '温暖阿虎/Alvin', 'zh_male_wennuanahu_moon_bigtts', '中文、英文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/Alvin.mp3', NULL, 2, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0003', 'TTS_HuoshanDoubleStreamTTS', '少年梓辛/Brayan', 'zh_male_shaonianzixin_moon_bigtts', '中文、英文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/Brayan.mp3', NULL, 3, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0004', 'TTS_HuoshanDoubleStreamTTS', '邻家女孩', 'zh_female_linjianvhai_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E9%82%BB%E5%AE%B6%E5%A5%B3%E5%AD%A9.mp3', NULL, 4, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0005', 'TTS_HuoshanDoubleStreamTTS', '渊博小叔', 'zh_male_yuanboxiaoshu_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E6%B8%8A%E5%8D%9A%E5%B0%8F%E5%8F%94.mp3', NULL, 5, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0006', 'TTS_HuoshanDoubleStreamTTS', '阳光青年', 'zh_male_yangguangqingnian_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E9%98%B3%E5%85%89%E9%9D%92%E5%B9%B4.mp3', NULL, 6, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0007', 'TTS_HuoshanDoubleStreamTTS', '京腔侃爷/Harmony', 'zh_male_jingqiangkanye_moon_bigtts', '中文、英文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/Harmony.mp3', NULL, 7, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0008', 'TTS_HuoshanDoubleStreamTTS', '湾湾小何', 'zh_female_wanwanxiaohe_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E6%B9%BE%E6%B9%BE%E5%B0%8F%E4%BD%95.mp3', NULL, 8, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0009', 'TTS_HuoshanDoubleStreamTTS', '湾区大叔', 'zh_female_wanqudashu_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E6%B9%BE%E5%8C%BA%E5%A4%A7%E5%8F%94.mp3', NULL, 9, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0010', 'TTS_HuoshanDoubleStreamTTS', '呆萌川妹', 'zh_female_daimengchuanmei_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E5%91%86%E8%90%8C%E5%B7%9D%E5%A6%B9.mp3', NULL, 10, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0011', 'TTS_HuoshanDoubleStreamTTS', '广州德哥', 'zh_male_guozhoudege_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E5%B9%BF%E5%B7%9E%E5%BE%B7%E5%93%A5.mp3', NULL, 11, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0012', 'TTS_HuoshanDoubleStreamTTS', '北京小爷', 'zh_male_beijingxiaoye_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E5%8C%97%E4%BA%AC%E5%B0%8F%E7%88%B7.mp3', NULL, 12, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0013', 'TTS_HuoshanDoubleStreamTTS', '浩宇小哥', 'zh_male_haoyuxiaoge_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E6%B5%A9%E5%AE%87%E5%B0%8F%E5%93%A5.mp3', NULL, 13, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0014', 'TTS_HuoshanDoubleStreamTTS', '广西远舟', 'zh_male_guangxiyuanzhou_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E5%B9%BF%E8%A5%BF%E8%BF%9C%E8%88%9F.mp3', NULL, 14, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0015', 'TTS_HuoshanDoubleStreamTTS', '妹坨洁儿', 'zh_female_meituojieer_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E5%A6%B9%E5%9D%A8%E6%B4%81%E5%84%BF.mp3', NULL, 15, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0016', 'TTS_HuoshanDoubleStreamTTS', '豫州子轩', 'zh_male_yuzhouzixuan_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E8%B1%AB%E5%B7%9E%E5%AD%90%E8%BD%A9.mp3', NULL, 16, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0017', 'TTS_HuoshanDoubleStreamTTS', '高冷御姐', 'zh_female_gaolengyujie_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E9%AB%98%E5%86%B7%E5%BE%A1%E5%A7%90.mp3', NULL, 17, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0018', 'TTS_HuoshanDoubleStreamTTS', '傲娇霸总', 'zh_male_aojiaobazong_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E5%82%B2%E5%A8%87%E9%9C%B8%E6%80%BB.mp3', NULL, 18, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0019', 'TTS_HuoshanDoubleStreamTTS', '魅力女友', 'zh_female_meilinvyou_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E9%AD%85%E5%8A%9B%E5%A5%B3%E5%8F%8B.mp3', NULL, 19, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0020', 'TTS_HuoshanDoubleStreamTTS', '深夜播客', 'zh_male_shenyeboke_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E6%B7%B1%E5%A4%9C%E6%92%AD%E5%AE%A2.mp3', NULL, 20, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0021', 'TTS_HuoshanDoubleStreamTTS', '柔美女友', 'zh_female_sajiaonvyou_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E6%9F%94%E7%BE%8E%E5%A5%B3%E5%8F%8B.mp3', NULL, 21, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0022', 'TTS_HuoshanDoubleStreamTTS', '撒娇学妹', 'zh_female_yuanqinvyou_moon_bigtts', '中文', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E6%92%92%E5%A8%87%E5%AD%A6%E5%A6%B9.mp3', NULL, 22, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0023', 'TTS_HuoshanDoubleStreamTTS', 'かずね(和音)', 'multi_male_jingqiangkanye_moon_bigtts', '日语、西语', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/Javier.wav', NULL, 23, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0024', 'TTS_HuoshanDoubleStreamTTS', 'はるこ(晴子)', 'multi_female_shuangkuaisisi_moon_bigtts', '日语、西语', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/Esmeralda.mp3', NULL, 24, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0025', 'TTS_HuoshanDoubleStreamTTS', 'あけみ(朱美)', 'multi_female_gaolengyujie_moon_bigtts', '日语', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/%E6%9C%B1%E7%BE%8E.mp3', NULL, 25, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_HuoshanDoubleStreamTTS_0026', 'TTS_HuoshanDoubleStreamTTS', 'ひろし(広志)', 'multi_male_wanqudashu_moon_bigtts', '日语、西语', 'https://lf3-static.bytednsdoc.com/obj/eden-cn/lm_hz_ihsph/ljhwZthlaukjlkulzlp/portal/bigtts/Roberto.wav', NULL, 26, NULL, NULL, NULL, NULL);
@@ -0,0 +1,29 @@
-- VLLM模型供应器
delete from `ai_model_provider` where id = 'SYSTEM_VLLM_openai';
INSERT INTO `ai_model_provider` (`id`, `model_type`, `provider_code`, `name`, `fields`, `sort`, `creator`, `create_date`, `updater`, `update_date`) VALUES
('SYSTEM_VLLM_openai', 'VLLM', 'openai', 'OpenAI接口', '[{"key":"base_url","label":"基础URL","type":"string"},{"key":"model_name","label":"模型名称","type":"string"},{"key":"api_key","label":"API密钥","type":"string"}]', 9, 1, NOW(), 1, NOW());
-- VLLM模型配置
delete from `ai_model_config` where id = 'VLLM_ChatGLMVLLM';
INSERT INTO `ai_model_config` VALUES ('VLLM_ChatGLMVLLM', 'VLLM', 'ChatGLMVLLM', '智谱视觉AI', 1, 1, '{\"type\": \"openai\", \"model_name\": \"glm-4v-flash\", \"base_url\": \"https://open.bigmodel.cn/api/paas/v4/\", \"api_key\": \"你的api_key\"}', NULL, NULL, 1, NULL, NULL, NULL, NULL);
-- 更新文档
UPDATE `ai_model_config` SET
`doc_link` = 'https://bigmodel.cn/usercenter/proj-mgmt/apikeys',
`remark` = '智谱视觉AI配置说明:
1. 访问 https://bigmodel.cn/usercenter/proj-mgmt/apikeys
2. 注册并获取API密钥
3. 填入配置文件中' WHERE `id` = 'VLLM_ChatGLMVLLM';
-- 添加参数
INSERT INTO `sys_params` (id, param_code, param_value, value_type, param_type, remark) VALUES (113, 'server.http_port', '8003', 'number', 1, 'http服务的端口,用于启动视觉分析接口');
INSERT INTO `sys_params` (id, param_code, param_value, value_type, param_type, remark) VALUES (114, 'server.vision_explain', 'null', 'string', 1, '视觉分析接口地址,用于下发到设备,多个用;分隔');
-- 智能体表增加VLLM模型配置
ALTER TABLE `ai_agent`
ADD COLUMN `vllm_model_id` varchar(32) NULL DEFAULT 'VLLM_ChatGLMVLLM' COMMENT '视觉模型标识' AFTER `llm_model_id`;
-- 智能体模版表增加VLLM模型配置
ALTER TABLE `ai_agent_template`
ADD COLUMN `vllm_model_id` varchar(32) NULL DEFAULT 'VLLM_ChatGLMVLLM' COMMENT '视觉模型标识' AFTER `llm_model_id`;
@@ -162,4 +162,18 @@ databaseChangeLog:
changes: changes:
- sqlFile: - sqlFile:
encoding: utf8 encoding: utf8
path: classpath:db/changelog/202505151451.sql path: classpath:db/changelog/202505151451.sql
- changeSet:
id: 202505271414
author: hrz
changes:
- sqlFile:
encoding: utf8
path: classpath:db/changelog/202505271414.sql
- changeSet:
id: 202506010920
author: hrz
changes:
- sqlFile:
encoding: utf8
path: classpath:db/changelog/202506010920.sql
@@ -14,10 +14,10 @@
</div> </div>
</div> </div>
<div class="device-name"> <div class="device-name">
设备型号{{ device.ttsModelName }} 语言模型{{ device.llmModelName }}
</div> </div>
<div class="device-name"> <div class="device-name">
音色模型{{ device.ttsVoiceName }} 音色模型{{ device.ttsModelName }} ({{ device.ttsVoiceName }})
</div> </div>
<div style="display: flex;gap: 10px;align-items: center;"> <div style="display: flex;gap: 10px;align-items: center;">
<div class="settings-btn" @click="handleConfigure"> <div class="settings-btn" @click="handleConfigure">
@@ -51,7 +51,7 @@
字典管理 字典管理
</el-dropdown-item> </el-dropdown-item>
<el-dropdown-item @click.native="goProviderManagement"> <el-dropdown-item @click.native="goProviderManagement">
供应器管理 字段管理
</el-dropdown-item> </el-dropdown-item>
<el-dropdown-item @click.native="goServerSideManagement"> <el-dropdown-item @click.native="goServerSideManagement">
服务端管理 服务端管理
+5 -1
View File
@@ -30,6 +30,9 @@
<el-menu-item index="llm"> <el-menu-item index="llm">
<span class="menu-text">大语言模型</span> <span class="menu-text">大语言模型</span>
</el-menu-item> </el-menu-item>
<el-menu-item index="vllm">
<span class="menu-text">视觉大语言模型</span>
</el-menu-item>
<el-menu-item index="intent"> <el-menu-item index="intent">
<span class="menu-text">意图识别</span> <span class="menu-text">意图识别</span>
</el-menu-item> </el-menu-item>
@@ -173,6 +176,7 @@ export default {
vad: '语言活动检测模型(VAD)', vad: '语言活动检测模型(VAD)',
asr: '语音识别模型(ASR)', asr: '语音识别模型(ASR)',
llm: '大语言模型(LLM', llm: '大语言模型(LLM',
vllm: '视觉大语言模型(VLLM',
intent: '意图识别模型(Intent)', intent: '意图识别模型(Intent)',
tts: '语音合成模型(TTS)', tts: '语音合成模型(TTS)',
memory: '记忆模型(Memory)' memory: '记忆模型(Memory)'
@@ -467,7 +471,7 @@ export default {
.main-wrapper { .main-wrapper {
margin: 5px 22px; margin: 5px 22px;
border-radius: 15px; border-radius: 15px;
min-height: calc(100vh - 24vh); min-height: calc(100vh - 26vh);
height: auto; height: auto;
max-height: 80vh; max-height: 80vh;
box-shadow: 0 2px 12px rgba(0, 0, 0, 0.1); box-shadow: 0 2px 12px rgba(0, 0, 0, 0.1);
@@ -3,7 +3,7 @@
<HeaderBar /> <HeaderBar />
<div class="operation-bar"> <div class="operation-bar">
<h2 class="page-title">供应器管理</h2> <h2 class="page-title">字段管理</h2>
<div class="right-operations"> <div class="right-operations">
<el-dropdown trigger="click" @command="handleSelectModelType" @visible-change="handleDropdownVisibleChange"> <el-dropdown trigger="click" @command="handleSelectModelType" @visible-change="handleDropdownVisibleChange">
<el-button class="category-btn"> <el-button class="category-btn">
+46 -1
View File
@@ -64,7 +64,27 @@
</el-form-item> </el-form-item>
</div> </div>
<div class="form-column"> <div class="form-column">
<el-form-item v-for="(model, index) in models" :key="`model-${index}`" :label="model.label" <div class="model-row">
<el-form-item label="语音活动检测(VAD)" class="model-item">
<div class="model-select-wrapper">
<el-select v-model="form.model.vadModelId" filterable placeholder="请选择" class="form-select"
@change="handleModelChange('VAD', $event)">
<el-option v-for="(item, optionIndex) in modelOptions['VAD']"
:key="`option-vad-${optionIndex}`" :label="item.label" :value="item.value" />
</el-select>
</div>
</el-form-item>
<el-form-item label="语音识别(ASR)" class="model-item">
<div class="model-select-wrapper">
<el-select v-model="form.model.asrModelId" filterable placeholder="请选择" class="form-select"
@change="handleModelChange('ASR', $event)">
<el-option v-for="(item, optionIndex) in modelOptions['ASR']"
:key="`option-asr-${optionIndex}`" :label="item.label" :value="item.value" />
</el-select>
</div>
</el-form-item>
</div>
<el-form-item v-for="(model, index) in models.slice(2)" :key="`model-${index}`" :label="model.label"
class="model-item"> class="model-item">
<div class="model-select-wrapper"> <div class="model-select-wrapper">
<el-select v-model="form.model[model.key]" filterable placeholder="请选择" class="form-select" <el-select v-model="form.model[model.key]" filterable placeholder="请选择" class="form-select"
@@ -148,6 +168,7 @@ export default {
vadModelId: "", vadModelId: "",
asrModelId: "", asrModelId: "",
llmModelId: "", llmModelId: "",
vllmModelId: "",
memModelId: "", memModelId: "",
intentModelId: "", intentModelId: "",
} }
@@ -156,6 +177,7 @@ export default {
{ label: '语音活动检测(VAD)', key: 'vadModelId', type: 'VAD' }, { label: '语音活动检测(VAD)', key: 'vadModelId', type: 'VAD' },
{ label: '语音识别(ASR)', key: 'asrModelId', type: 'ASR' }, { label: '语音识别(ASR)', key: 'asrModelId', type: 'ASR' },
{ label: '大语言模型(LLM)', key: 'llmModelId', type: 'LLM' }, { label: '大语言模型(LLM)', key: 'llmModelId', type: 'LLM' },
{ label: '视觉大语言模型(VLLM)', key: 'vllmModelId', type: 'VLLM' },
{ label: '意图识别(Intent)', key: 'intentModelId', type: 'Intent' }, { label: '意图识别(Intent)', key: 'intentModelId', type: 'Intent' },
{ label: '记忆(Memory)', key: 'memModelId', type: 'Memory' }, { label: '记忆(Memory)', key: 'memModelId', type: 'Memory' },
{ label: '语音合成(TTS)', key: 'ttsModelId', type: 'TTS' }, { label: '语音合成(TTS)', key: 'ttsModelId', type: 'TTS' },
@@ -189,6 +211,7 @@ export default {
asrModelId: this.form.model.asrModelId, asrModelId: this.form.model.asrModelId,
vadModelId: this.form.model.vadModelId, vadModelId: this.form.model.vadModelId,
llmModelId: this.form.model.llmModelId, llmModelId: this.form.model.llmModelId,
vllmModelId: this.form.model.vllmModelId,
ttsModelId: this.form.model.ttsModelId, ttsModelId: this.form.model.ttsModelId,
ttsVoiceId: this.form.ttsVoiceId, ttsVoiceId: this.form.ttsVoiceId,
chatHistoryConf: this.form.chatHistoryConf, chatHistoryConf: this.form.chatHistoryConf,
@@ -236,6 +259,7 @@ export default {
vadModelId: "", vadModelId: "",
asrModelId: "", asrModelId: "",
llmModelId: "", llmModelId: "",
vllmModelId: "",
memModelId: "", memModelId: "",
intentModelId: "", intentModelId: "",
} }
@@ -289,6 +313,7 @@ export default {
vadModelId: templateData.vadModelId || this.form.model.vadModelId, vadModelId: templateData.vadModelId || this.form.model.vadModelId,
asrModelId: templateData.asrModelId || this.form.model.asrModelId, asrModelId: templateData.asrModelId || this.form.model.asrModelId,
llmModelId: templateData.llmModelId || this.form.model.llmModelId, llmModelId: templateData.llmModelId || this.form.model.llmModelId,
vllmModelId: templateData.vllmModelId || this.form.model.vllmModelId,
memModelId: templateData.memModelId || this.form.model.memModelId, memModelId: templateData.memModelId || this.form.model.memModelId,
intentModelId: templateData.intentModelId || this.form.model.intentModelId intentModelId: templateData.intentModelId || this.form.model.intentModelId
} }
@@ -305,6 +330,7 @@ export default {
vadModelId: data.data.vadModelId, vadModelId: data.data.vadModelId,
asrModelId: data.data.asrModelId, asrModelId: data.data.asrModelId,
llmModelId: data.data.llmModelId, llmModelId: data.data.llmModelId,
vllmModelId: data.data.vllmModelId,
memModelId: data.data.memModelId, memModelId: data.data.memModelId,
intentModelId: data.data.intentModelId intentModelId: data.data.intentModelId
} }
@@ -587,6 +613,25 @@ export default {
width: 100%; width: 100%;
} }
.model-row {
display: flex;
gap: 20px;
margin-bottom: 6px;
}
.model-row .model-item {
flex: 1;
margin-bottom: 0;
}
.model-row .el-form-item__label {
font-size: 12px !important;
color: #3d4566 !important;
font-weight: 400;
line-height: 22px;
padding-bottom: 2px;
}
.function-icons { .function-icons {
display: flex; display: flex;
align-items: center; align-items: center;
+19 -11
View File
@@ -1,13 +1,14 @@
import asyncio
import sys import sys
import uuid
import signal import signal
import asyncio
from aioconsole import ainput
from config.settings import load_config from config.settings import load_config
from core.websocket_server import WebSocketServer
from core.ota_server import SimpleOtaServer
from core.utils.util import check_ffmpeg_installed
from config.logger import setup_logging from config.logger import setup_logging
from core.utils.util import get_local_ip from core.utils.util import get_local_ip
from aioconsole import ainput from core.http_server import SimpleHttpServer
from core.websocket_server import WebSocketServer
from core.utils.util import check_ffmpeg_installed
TAG = __name__ TAG = __name__
logger = setup_logging() logger = setup_logging()
@@ -45,25 +46,32 @@ async def main():
check_ffmpeg_installed() check_ffmpeg_installed()
config = load_config() config = load_config()
# 生成随机密钥并添加到配置中
config["server"]["auth_key"] = str(uuid.uuid4().hex)
# 添加 stdin 监控任务 # 添加 stdin 监控任务
stdin_task = asyncio.create_task(monitor_stdin()) stdin_task = asyncio.create_task(monitor_stdin())
# 启动 WebSocket 服务器 # 启动 WebSocket 服务器
ws_server = WebSocketServer(config) ws_server = WebSocketServer(config)
ws_task = asyncio.create_task(ws_server.start()) ws_task = asyncio.create_task(ws_server.start())
ota_task = None # 启动 Simple http 服务器
ota_server = SimpleHttpServer(config)
ota_task = asyncio.create_task(ota_server.start())
read_config_from_api = config.get("read_config_from_api", False) read_config_from_api = config.get("read_config_from_api", False)
port = int(config["server"].get("http_port", 8003))
if not read_config_from_api: if not read_config_from_api:
# 启动 Simple OTA 服务器
ota_server = SimpleOtaServer(config)
ota_task = asyncio.create_task(ota_server.start())
logger.bind(tag=TAG).info( logger.bind(tag=TAG).info(
"OTA接口是\t\thttp://{}:{}/xiaozhi/ota/", "OTA接口是\t\thttp://{}:{}/xiaozhi/ota/",
get_local_ip(), get_local_ip(),
config["server"]["ota_port"], port,
) )
logger.bind(tag=TAG).info(
"视觉分析接口是\thttp://{}:{}/mcp/vision/explain",
get_local_ip(),
port,
)
# 获取WebSocket配置,使用安全的默认值 # 获取WebSocket配置,使用安全的默认值
websocket_port = 8000 websocket_port = 8000
+44 -6
View File
@@ -2,6 +2,7 @@
# 然后你想修改覆盖修改什么配置,就修改【.config.yaml】文件,而不是修改【config.yaml】文件 # 然后你想修改覆盖修改什么配置,就修改【.config.yaml】文件,而不是修改【config.yaml】文件
# 系统会优先读取【data/.config.yaml】文件的配置,如果【.config.yaml】文件里的配置不存在,系统会自动去读取【config.yaml】文件的配置。 # 系统会优先读取【data/.config.yaml】文件的配置,如果【.config.yaml】文件里的配置不存在,系统会自动去读取【config.yaml】文件的配置。
# 这样做,可以最简化配置,保护您的密钥安全。 # 这样做,可以最简化配置,保护您的密钥安全。
# 如果你使用了智控台,那么以下所有配置,都不会生效,请在智控台中修改配置
# ##################################################################################### # #####################################################################################
# #############################以下是服务器基本运行配置#################################### # #############################以下是服务器基本运行配置####################################
@@ -9,13 +10,21 @@ server:
# 服务器监听地址和端口(Server listening address and port) # 服务器监听地址和端口(Server listening address and port)
ip: 0.0.0.0 ip: 0.0.0.0
port: 8000 port: 8000
# OTA接口的端口号 # http服务的端口,用于简单OTA接口(单服务部署),以及视觉分析接口
ota_port: 8002 http_port: 8003
# 这个websocket配置是指ota接口向设备发送的websocket地址 # 这个websocket配置是指ota接口向设备发送的websocket地址
# 如果按默认的写法,ota接口会自动生成websocket地址这个地址你可以直接用浏览器访问ota接口确认一下 # 如果按默认的写法,ota接口会自动生成websocket地址,并输出在启动日志里,这个地址你可以直接用浏览器访问ota接口确认一下
# 当你使用docker部署或使用公网部署(使用ssl、域名)时,不一定准确 # 当你使用docker部署或使用公网部署(使用ssl、域名)时,不一定准确
# 所以如果你使用docker部署或使用公网部署时,请设置正确的websocket地址 # 所以如果你使用docker部署时,将websocket设置成局域网地址
# 如果你使用公网部署时,将vwebsocket设置成公网地址
websocket: ws://你的ip或者域名:端口号/xiaozhi/v1/ websocket: ws://你的ip或者域名:端口号/xiaozhi/v1/
# 视觉分析接口地址
# 向设备发送的视觉分析的接口地址
# 如果按下面默认的写法,系统会自动生成视觉识别地址,并输出在启动日志里,这个地址你可以直接用浏览器访问确认一下
# 当你使用docker部署或使用公网部署(使用ssl、域名)时,不一定准确
# 所以如果你使用docker部署时,将vision_explain设置成局域网地址
# 如果你使用公网部署时,将vision_explain设置成公网地址
vision_explain: http://你的ip或者域名:端口号/mcp/vision/explain
# OTA返回信息时区偏移量 # OTA返回信息时区偏移量
timezone_offset: +8 timezone_offset: +8
# 认证配置 # 认证配置
@@ -85,6 +94,7 @@ module_test:
# 唤醒词,用于识别唤醒词还是讲话内容 # 唤醒词,用于识别唤醒词还是讲话内容
wakeup_words: wakeup_words:
- "你好小智" - "你好小智"
- "嘿你好呀"
- "你好小志" - "你好小志"
- "小爱同学" - "小爱同学"
- "你好小鑫" - "你好小鑫"
@@ -159,6 +169,8 @@ selected_module:
ASR: FunASR ASR: FunASR
# 将根据配置名称对应的type调用实际的LLM适配器 # 将根据配置名称对应的type调用实际的LLM适配器
LLM: ChatGLMLLM LLM: ChatGLMLLM
# 视觉语言大模型
VLLM: ChatGLMVLLM
# TTS将根据配置名称对应的type调用实际的TTS适配器 # TTS将根据配置名称对应的type调用实际的TTS适配器
TTS: EdgeTTS TTS: EdgeTTS
# 记忆模块,默认不开启记忆;如果想使用超长记忆,推荐使用mem0ai;如果注重隐私,请使用本地的mem_local_short # 记忆模块,默认不开启记忆;如果想使用超长记忆,推荐使用mem0ai;如果注重隐私,请使用本地的mem_local_short
@@ -218,8 +230,12 @@ Memory:
# 不想使用记忆功能,可以使用nomem # 不想使用记忆功能,可以使用nomem
type: nomem type: nomem
mem_local_short: mem_local_short:
# 本地记忆功能,通过selected_module的llm总结,数据保存在本地,不会上传到服务器 # 本地记忆功能,通过selected_module的llm总结,数据保存在本地服务器,不会上传到外部服务器
type: mem_local_short type: mem_local_short
# 配备记忆存储独立的思考模型
# 如果这里不填,则会默认使用selected_module.LLM的模型作为意图识别的思考模型
# 如果你的不想使用selected_module.LLM记忆存储,这里最好使用独立的LLM作为意图识别,例如使用免费的ChatGLMLLM
llm: ChatGLMLLM
ASR: ASR:
FunASR: FunASR:
@@ -325,7 +341,7 @@ LLM:
# 定义LLM API类型 # 定义LLM API类型
type: openai type: openai
# 先开通服务,打开以下网址,开通的服务搜索Doubao-1.5-pro,开通它 # 先开通服务,打开以下网址,开通的服务搜索Doubao-1.5-pro,开通它
# 开通地址:https://console.volcengine.com/ark/region:ark+cn-beijing/openManagement?LLM=%7B%7D&OpenTokenDrawer=false # 开通地址:https://console.volcengine.com/ark/region:ark+cn-beijing/openManagement?LLM=%7B%7D&OpenTokenDrawer=false
# 免费额度500000token # 免费额度500000token
# 开通后,进入这里获取密钥:https://console.volcengine.com/ark/region:ark+cn-beijing/apiKey?apikey=%7B%7D # 开通后,进入这里获取密钥:https://console.volcengine.com/ark/region:ark+cn-beijing/apiKey?apikey=%7B%7D
base_url: https://ark.cn-beijing.volces.com/api/v3 base_url: https://ark.cn-beijing.volces.com/api/v3
@@ -427,6 +443,15 @@ LLM:
# Xinference服务地址和模型名称 # Xinference服务地址和模型名称
model_name: qwen2.5:3b-AWQ # 使用的小模型名称,用于意图识别 model_name: qwen2.5:3b-AWQ # 使用的小模型名称,用于意图识别
base_url: http://localhost:9997 # Xinference服务地址 base_url: http://localhost:9997 # Xinference服务地址
# VLLM配置(视觉语言大模型)
VLLM:
ChatGLMVLLM:
type: openai
# glm-4v-flash是智谱免费AI的视觉模型,需要先在智谱AI平台创建API密钥并获取api_key
# 可在这里找到你的api key https://bigmodel.cn/usercenter/proj-mgmt/apikeys
model_name: glm-4v-flash # 智谱AI的视觉模型
url: https://open.bigmodel.cn/api/paas/v4/
api_key: 你的api_key
TTS: TTS:
# 当前支持的type为edge、doubao,可自行适配 # 当前支持的type为edge、doubao,可自行适配
EdgeTTS: EdgeTTS:
@@ -452,6 +477,19 @@ TTS:
speed_ratio: 1.0 speed_ratio: 1.0
volume_ratio: 1.0 volume_ratio: 1.0
pitch_ratio: 1.0 pitch_ratio: 1.0
#火山tts,支持双向流式tts
HuoshanDoubleStreamTTS:
type: huoshan_double_stream
# 访问 https://console.volcengine.com/speech/service/10007 开通语音合成大模型,购买音色
# 在页面底部获取appid和access_token
# 资源ID固定为:volc.service_type.10029(大模型语音合成及混音)
# 如果是机智云,把接口地址换成wss://bytedance.gizwitsapi.com/api/v3/tts/bidirection
# 机智云不需要天填 appid
ws_url: wss://openspeech.bytedance.com/api/v3/tts/bidirection
appid: 你的火山引擎语音合成服务appid
access_token: 你的火山引擎语音合成服务access_token
resource_id: volc.service_type.10029
speaker: zh_female_wanwanxiaohe_moon_bigtts
CosyVoiceSiliconflow: CosyVoiceSiliconflow:
type: siliconflow type: siliconflow
# 硅基流动TTS # 硅基流动TTS
@@ -59,10 +59,14 @@ def get_config_from_api(config):
"url": config["manager-api"].get("url", ""), "url": config["manager-api"].get("url", ""),
"secret": config["manager-api"].get("secret", ""), "secret": config["manager-api"].get("secret", ""),
} }
# server的配置以本地为准
if config.get("server"): if config.get("server"):
config_data["server"] = { config_data["server"] = {
"ip": config["server"].get("ip", ""), "ip": config["server"].get("ip", ""),
"port": config["server"].get("port", ""), "port": config["server"].get("port", ""),
"http_port": config["server"].get("http_port", ""),
"vision_explain": config["server"].get("vision_explain", ""),
"auth_key": config["server"].get("auth_key", ""),
} }
return config_data return config_data
+96 -31
View File
@@ -4,7 +4,8 @@ from loguru import logger
from config.config_loader import load_config from config.config_loader import load_config
from config.settings import check_config_file from config.settings import check_config_file
SERVER_VERSION = "0.4.4" SERVER_VERSION = "0.5.2"
_logger_initialized = False
def get_module_abbreviation(module_name, module_dict): def get_module_abbreviation(module_name, module_dict):
@@ -43,41 +44,105 @@ def setup_logging():
"""从配置文件中读取日志配置,并设置日志输出格式和级别""" """从配置文件中读取日志配置,并设置日志输出格式和级别"""
config = load_config() config = load_config()
log_config = config["log"] log_config = config["log"]
log_format = log_config.get( global _logger_initialized
"log_format",
"<green>{time:YYMMDD HH:mm:ss}</green>[{version}_{selected_module}][<light-blue>{extra[tag]}</light-blue>]-<level>{level}</level>-<light-green>{message}</light-green>",
)
log_format_file = log_config.get(
"log_format_file",
"{time:YYYY-MM-DD HH:mm:ss} - {version_{selected_module}} - {name} - {level} - {extra[tag]} - {message}",
)
selected_module_str = build_module_string(config.get("selected_module", {}))
log_format = log_format.replace("{version}", SERVER_VERSION) # 第一次初始化时配置日志
log_format = log_format.replace("{selected_module}", selected_module_str) if not _logger_initialized:
log_format_file = log_format_file.replace("{version}", SERVER_VERSION) logger.configure(
log_format_file = log_format_file.replace("{selected_module}", selected_module_str) extra={
"selected_module": log_config.get("selected_module", "00000000000000")
}
) # 新增配置
log_format = log_config.get(
"log_format",
"<green>{time:YYMMDD HH:mm:ss}</green>[{version}_{extra[selected_module]}][<light-blue>{extra[tag]}</light-blue>]-<level>{level}</level>-<light-green>{message}</light-green>",
)
log_format_file = log_config.get(
"log_format_file",
"{time:YYYY-MM-DD HH:mm:ss} - {version_{extra[selected_module]}} - {name} - {level} - {extra[tag]} - {message}",
)
selected_module_str = logger._core.extra["selected_module"]
log_level = log_config.get("log_level", "INFO") log_format = log_format.replace("{version}", SERVER_VERSION)
log_dir = log_config.get("log_dir", "tmp") log_format = log_format.replace("{selected_module}", selected_module_str)
log_file = log_config.get("log_file", "server.log") log_format_file = log_format_file.replace("{version}", SERVER_VERSION)
data_dir = log_config.get("data_dir", "data") log_format_file = log_format_file.replace(
"{selected_module}", selected_module_str
)
os.makedirs(log_dir, exist_ok=True) log_level = log_config.get("log_level", "INFO")
os.makedirs(data_dir, exist_ok=True) log_dir = log_config.get("log_dir", "tmp")
log_file = log_config.get("log_file", "server.log")
data_dir = log_config.get("data_dir", "data")
# 配置日志输出 os.makedirs(log_dir, exist_ok=True)
logger.remove() os.makedirs(data_dir, exist_ok=True)
# 输出到控制台 # 配置日志输出
logger.add(sys.stdout, format=log_format, level=log_level, filter=formatter) logger.remove()
# 输出到文件 # 输出到控制台
logger.add( logger.add(sys.stdout, format=log_format, level=log_level, filter=formatter)
os.path.join(log_dir, log_file),
format=log_format_file, # 输出到文件
level=log_level, logger.add(
filter=formatter, os.path.join(log_dir, log_file),
) format=log_format_file,
level=log_level,
filter=formatter,
)
_logger_initialized = True # 标记为已初始化
return logger return logger
def update_module_string(selected_module_str):
"""更新模块字符串并重新配置日志处理器"""
logger.debug(f"更新日志配置组件")
current_module = logger._core.extra["selected_module"]
if current_module == selected_module_str:
return
try:
logger.configure(extra={"selected_module": selected_module_str})
config = load_config()
log_config = config["log"]
log_format = log_config.get(
"log_format",
"<green>{time:YYMMDD HH:mm:ss}</green>[{version}_{extra[selected_module]}][<light-blue>{extra[tag]}</light-blue>]-<level>{level}</level>-<light-green>{message}</light-green>",
)
log_format_file = log_config.get(
"log_format_file",
"{time:YYYY-MM-DD HH:mm:ss} - {version_{extra[selected_module]}} - {name} - {level} - {extra[tag]} - {message}",
)
log_format = log_format.replace("{version}", SERVER_VERSION)
log_format = log_format.replace("{selected_module}", selected_module_str)
log_format_file = log_format_file.replace("{version}", SERVER_VERSION)
log_format_file = log_format_file.replace(
"{selected_module}", selected_module_str
)
logger.remove()
logger.add(
sys.stdout,
format=log_format,
level=log_config.get("log_level", "INFO"),
filter=formatter,
)
logger.add(
os.path.join(
log_config.get("log_dir", "tmp"),
log_config.get("log_file", "server.log"),
),
format=log_format_file,
level=log_config.get("log_level", "INFO"),
filter=formatter,
)
except Exception as e:
logger.error(f"日志配置更新失败: {str(e)}")
raise
@@ -160,7 +160,7 @@ def save_mem_local_short(mac_address: str, short_momery: str) -> Optional[Dict]:
def report( def report(
mac_address: str, session_id: str, chat_type: int, content: str, audio mac_address: str, session_id: str, chat_type: int, content: str, audio, report_time
) -> Optional[Dict]: ) -> Optional[Dict]:
"""带熔断的业务方法示例""" """带熔断的业务方法示例"""
if not content or not ManageApiClient._instance: if not content or not ManageApiClient._instance:
@@ -174,6 +174,7 @@ def report(
"sessionId": session_id, "sessionId": session_id,
"chatType": chat_type, "chatType": chat_type,
"content": content, "content": content,
"reportTime": report_time,
"audioBase64": ( "audioBase64": (
base64.b64encode(audio).decode("utf-8") if audio else None base64.b64encode(audio).decode("utf-8") if audio else None
), ),
+9
View File
@@ -8,6 +8,15 @@
server: server:
ip: 0.0.0.0 ip: 0.0.0.0
port: 8000 port: 8000
# http服务的端口,用于视觉分析接口
http_port: 8003
# 视觉分析接口地址
# 向设备发送的视觉分析的接口地址
# 如果按下面默认的写法,系统会自动生成视觉识别地址,并输出在启动日志里,这个地址你可以直接用浏览器访问确认一下
# 当你使用docker部署或使用公网部署(使用ssl、域名)时,不一定准确
# 所以如果你使用docker部署时,将vision_explain设置成局域网地址
# 如果你使用公网部署时,将vision_explain设置成公网地址
vision_explain: http://你的ip或者域名:端口号/mcp/vision/explain
manager-api: manager-api:
# 你的manager-api的地址,最好使用局域网ip # 你的manager-api的地址,最好使用局域网ip
url: http://127.0.0.1:8002/xiaozhi url: http://127.0.0.1:8002/xiaozhi
@@ -0,0 +1,16 @@
from aiohttp import web
from config.logger import setup_logging
class BaseHandler:
def __init__(self, config: dict):
self.config = config
self.logger = setup_logging()
def _add_cors_headers(self, response):
"""添加CORS头信息"""
response.headers["Access-Control-Allow-Headers"] = (
"client-id, content-type, device-id"
)
response.headers["Access-Control-Allow-Credentials"] = "true"
response.headers["Access-Control-Allow-Origin"] = "*"
@@ -1,18 +1,15 @@
import json import json
import time import time
import asyncio
from aiohttp import web from aiohttp import web
from config.logger import setup_logging from core.utils.util import get_local_ip
from core.connection import ConnectionHandler from core.api.base_handler import BaseHandler
from core.utils.util import get_local_ip, initialize_modules
TAG = __name__ TAG = __name__
class SimpleOtaServer: class OTAHandler(BaseHandler):
def __init__(self, config: dict): def __init__(self, config: dict):
self.config = config super().__init__(config)
self.logger = setup_logging()
def _get_websocket_url(self, local_ip: str, port: int) -> str: def _get_websocket_url(self, local_ip: str, port: int) -> str:
"""获取websocket地址 """获取websocket地址
@@ -25,41 +22,15 @@ class SimpleOtaServer:
str: websocket地址 str: websocket地址
""" """
server_config = self.config["server"] server_config = self.config["server"]
websocket_config = server_config.get("websocket") websocket_config = server_config.get("websocket", "")
if websocket_config and "" not in websocket_config: if "" not in websocket_config:
return websocket_config return websocket_config
else: else:
return f"ws://{local_ip}:{port}/xiaozhi/v1/" return f"ws://{local_ip}:{port}/xiaozhi/v1/"
async def start(self): async def handle_post(self, request):
server_config = self.config["server"] """处理 OTA POST 请求"""
host = server_config.get("ip", "0.0.0.0")
port = int(server_config.get("ota_port"))
if port:
app = web.Application()
# 添加路由
app.add_routes(
[
web.get("/xiaozhi/ota/", self._handle_ota_get_request),
web.post("/xiaozhi/ota/", self._handle_ota_request),
web.options("/xiaozhi/ota/", self._handle_ota_request),
]
)
# 运行服务
runner = web.AppRunner(app)
await runner.setup()
site = web.TCPSite(runner, host, port)
await site.start()
# 保持服务运行
while True:
await asyncio.sleep(3600) # 每隔 1 小时检查一次
async def _handle_ota_request(self, request):
"""处理 /xiaozhi/ota/ 的 POST 请求"""
try: try:
data = await request.text() data = await request.text()
self.logger.bind(tag=TAG).debug(f"OTA请求方法: {request.method}") self.logger.bind(tag=TAG).debug(f"OTA请求方法: {request.method}")
@@ -75,11 +46,9 @@ class SimpleOtaServer:
data_json = json.loads(data) data_json = json.loads(data)
server_config = self.config["server"] server_config = self.config["server"]
host = server_config.get("ip", "0.0.0.0")
port = int(server_config.get("port", 8000)) port = int(server_config.get("port", 8000))
local_ip = get_local_ip() local_ip = get_local_ip()
# OTA基础信息
return_json = { return_json = {
"server_time": { "server_time": {
"timestamp": int(round(time.time() * 1000)), "timestamp": int(round(time.time() * 1000)),
@@ -98,23 +67,17 @@ class SimpleOtaServer:
content_type="application/json", content_type="application/json",
) )
except Exception as e: except Exception as e:
self.logger.bind(tag=TAG).error(f"OTA请求异常: {e}")
return_json = {"success": False, "message": "request error."} return_json = {"success": False, "message": "request error."}
response = web.Response( response = web.Response(
text=json.dumps(return_json, separators=(",", ":")), text=json.dumps(return_json, separators=(",", ":")),
content_type="application/json", content_type="application/json",
) )
finally: finally:
# 添加header,允许跨域访问 self._add_cors_headers(response)
response.headers["Access-Control-Allow-Headers"] = (
"client-id, content-type, device-id"
)
response.headers["Access-Control-Allow-Credentials"] = "true"
response.headers["Access-Control-Allow-Origin"] = "*"
return response return response
async def _handle_ota_get_request(self, request): async def handle_get(self, request):
"""处理 /xiaozhi/ota/ 的 GET 请求""" """处理 OTA GET 请求"""
try: try:
server_config = self.config["server"] server_config = self.config["server"]
local_ip = get_local_ip() local_ip = get_local_ip()
@@ -126,10 +89,5 @@ class SimpleOtaServer:
self.logger.bind(tag=TAG).error(f"OTA GET请求异常: {e}") self.logger.bind(tag=TAG).error(f"OTA GET请求异常: {e}")
response = web.Response(text="OTA接口异常", content_type="text/plain") response = web.Response(text="OTA接口异常", content_type="text/plain")
finally: finally:
# 添加header,允许跨域访问 self._add_cors_headers(response)
response.headers["Access-Control-Allow-Headers"] = (
"client-id, content-type, device-id"
)
response.headers["Access-Control-Allow-Credentials"] = "true"
response.headers["Access-Control-Allow-Origin"] = "*"
return response return response
@@ -0,0 +1,181 @@
import json
import copy
from aiohttp import web
from config.logger import setup_logging
from core.utils.util import get_vision_url, is_valid_image_file
from core.utils.vllm import create_instance
from config.config_loader import get_private_config_from_api
from core.utils.auth import AuthToken
import base64
from typing import Tuple, Optional
TAG = __name__
# 设置最大文件大小为5MB
MAX_FILE_SIZE = 5 * 1024 * 1024
class VisionHandler:
def __init__(self, config: dict):
self.config = config
self.logger = setup_logging()
# 初始化认证工具
self.auth = AuthToken(config["server"]["auth_key"])
def _create_error_response(self, message: str) -> dict:
"""创建统一的错误响应格式"""
return {"success": False, "message": message}
def _verify_auth_token(self, request) -> Tuple[bool, Optional[str]]:
"""验证认证token"""
auth_header = request.headers.get("Authorization", "")
if not auth_header.startswith("Bearer "):
return False, None
token = auth_header[7:] # 移除"Bearer "前缀
return self.auth.verify_token(token)
async def handle_post(self, request):
"""处理 MCP Vision POST 请求"""
try:
# 验证token
is_valid, token_device_id = self._verify_auth_token(request)
if not is_valid:
return web.Response(
text=json.dumps(
self._create_error_response("无效的认证token或token已过期")
),
content_type="application/json",
status=401,
)
# 获取请求头信息
device_id = request.headers.get("Device-Id", "")
client_id = request.headers.get("Client-Id", "")
if device_id != token_device_id:
return web.Response(
text=json.dumps(self._create_error_response("设备ID与token不匹配")),
content_type="application/json",
status=401,
)
# 解析multipart/form-data请求
reader = await request.multipart()
# 读取question字段
question_field = await reader.next()
if question_field is None:
raise ValueError("缺少问题字段")
question = await question_field.text()
self.logger.bind(tag=TAG).debug(f"Question: {question}")
# 读取图片文件
image_field = await reader.next()
if image_field is None:
raise ValueError("缺少图片文件")
# 读取图片数据
image_data = await image_field.read()
if not image_data:
raise ValueError("图片数据为空")
# 检查文件大小
if len(image_data) > MAX_FILE_SIZE:
raise ValueError(
f"图片大小超过限制,最大允许{MAX_FILE_SIZE/1024/1024}MB"
)
# 检查文件格式
if not is_valid_image_file(image_data):
raise ValueError(
"不支持的文件格式,请上传有效的图片文件(支持JPEG、PNG、GIF、BMP、TIFF、WEBP格式)"
)
# 将图片转换为base64编码
image_base64 = base64.b64encode(image_data).decode("utf-8")
# 如果开启了智控台,则从智控台获取模型配置
current_config = copy.deepcopy(self.config)
read_config_from_api = current_config.get("read_config_from_api", False)
if read_config_from_api:
current_config = get_private_config_from_api(
current_config,
device_id,
client_id,
)
select_vllm_module = current_config["selected_module"].get("VLLM")
if not select_vllm_module:
raise ValueError("您还未设置默认的视觉分析模块")
vllm_type = (
select_vllm_module
if "type" not in current_config["VLLM"][select_vllm_module]
else current_config["VLLM"][select_vllm_module]["type"]
)
if not vllm_type:
raise ValueError(f"无法找到VLLM模块对应的供应器{vllm_type}")
vllm = create_instance(
vllm_type, current_config["VLLM"][select_vllm_module]
)
response = vllm.response(question, image_base64)
return_json = {
"success": True,
"result": response,
}
response = web.Response(
text=json.dumps(return_json, separators=(",", ":")),
content_type="application/json",
)
except ValueError as e:
self.logger.bind(tag=TAG).error(f"MCP Vision POST请求异常: {e}")
return_json = self._create_error_response(str(e))
response = web.Response(
text=json.dumps(return_json, separators=(",", ":")),
content_type="application/json",
)
except Exception as e:
self.logger.bind(tag=TAG).error(f"MCP Vision POST请求异常: {e}")
return_json = self._create_error_response("处理请求时发生错误")
response = web.Response(
text=json.dumps(return_json, separators=(",", ":")),
content_type="application/json",
)
finally:
self._add_cors_headers(response)
return response
async def handle_get(self, request):
"""处理 MCP Vision GET 请求"""
try:
vision_explain = get_vision_url(self.config)
if vision_explain and len(vision_explain) > 0 and "null" != vision_explain:
message = (
f"MCP Vision 接口运行正常,视觉解释接口地址是:{vision_explain}"
)
else:
message = "MCP Vision 接口运行不正常,请打开data目录下的.config.yaml文件,找到【server.vision_explain】,设置好地址"
response = web.Response(text=message, content_type="text/plain")
except Exception as e:
self.logger.bind(tag=TAG).error(f"MCP Vision GET请求异常: {e}")
return_json = self._create_error_response("服务器内部错误")
response = web.Response(
text=json.dumps(return_json, separators=(",", ":")),
content_type="application/json",
)
finally:
self._add_cors_headers(response)
return response
def _add_cors_headers(self, response):
"""添加CORS头信息"""
response.headers["Access-Control-Allow-Headers"] = (
"client-id, content-type, device-id"
)
response.headers["Access-Control-Allow-Credentials"] = "true"
response.headers["Access-Control-Allow-Origin"] = "*"
+289 -401
View File
@@ -1,40 +1,45 @@
import os import os
import sys
import copy import copy
import json import json
import subprocess
import sys
import uuid import uuid
import time import time
import queue import queue
import asyncio import asyncio
import traceback
import threading import threading
import traceback
import subprocess
import websockets import websockets
from typing import Dict, Any from core.handle.mcpHandle import call_mcp_tool
from plugins_func.loadplugins import auto_import_modules
from config.logger import setup_logging
from core.utils.dialogue import Message, Dialogue
from core.handle.textHandle import handleTextMessage
from core.utils.util import ( from core.utils.util import (
get_string_no_punctuation_or_emoji,
extract_json_from_string, extract_json_from_string,
initialize_modules,
check_vad_update, check_vad_update,
check_asr_update, check_asr_update,
filter_sensitive_info, filter_sensitive_info,
) )
from concurrent.futures import ThreadPoolExecutor, TimeoutError from typing import Dict, Any
from core.handle.sendAudioHandle import sendAudioMessage from core.mcp.manager import MCPManager
from core.handle.receiveAudioHandle import handleAudioMessage from core.utils.modules_initialize import (
initialize_modules,
initialize_tts,
initialize_asr,
)
from core.handle.reportHandle import report
from core.providers.tts.default import DefaultTTS
from concurrent.futures import ThreadPoolExecutor
from core.utils.dialogue import Message, Dialogue
from core.providers.asr.dto.dto import InterfaceType
from core.handle.textHandle import handleTextMessage
from core.handle.functionHandler import FunctionHandler from core.handle.functionHandler import FunctionHandler
from plugins_func.loadplugins import auto_import_modules
from plugins_func.register import Action, ActionResponse from plugins_func.register import Action, ActionResponse
from core.auth import AuthMiddleware, AuthenticationError from core.auth import AuthMiddleware, AuthenticationError
from core.mcp.manager import MCPManager
from config.config_loader import get_private_config_from_api from config.config_loader import get_private_config_from_api
from core.handle.receiveAudioHandle import handleAudioMessage
from core.providers.tts.dto.dto import ContentType, TTSMessageDTO, SentenceType
from config.logger import setup_logging, build_module_string, update_module_string
from config.manage_api_client import DeviceNotFoundException, DeviceBindException from config.manage_api_client import DeviceNotFoundException, DeviceBindException
from core.utils.output_counter import add_device_output
from core.handle.reportHandle import enqueue_tts_report, report
TAG = __name__ TAG = __name__
@@ -52,7 +57,6 @@ class ConnectionHandler:
_vad, _vad,
_asr, _asr,
_llm, _llm,
_tts,
_memory, _memory,
_intent, _intent,
server=None, server=None,
@@ -80,29 +84,28 @@ class ConnectionHandler:
# 客户端状态相关 # 客户端状态相关
self.client_abort = False self.client_abort = False
self.client_is_speaking = False
self.client_listen_mode = "auto" self.client_listen_mode = "auto"
# 线程任务相关 # 线程任务相关
self.loop = asyncio.get_event_loop() self.loop = asyncio.get_event_loop()
self.stop_event = threading.Event() self.stop_event = threading.Event()
self.tts_queue = queue.Queue() self.executor = ThreadPoolExecutor(max_workers=5)
self.audio_play_queue = queue.Queue()
self.executor = ThreadPoolExecutor(max_workers=10)
# 上报线程 # 添加上报线程
self.report_queue = queue.Queue() self.report_queue = queue.Queue()
self.report_thread = None self.report_thread = None
# TODO(haotian): 2025/5/12 可以通过修改此处,调节asr的上报和tts的上报 # 未来可以通过修改此处,调节asr的上报和tts的上报,目前默认都开启
self.report_asr_enable = self.read_config_from_api self.report_asr_enable = self.read_config_from_api
self.report_tts_enable = self.read_config_from_api self.report_tts_enable = self.read_config_from_api
# 依赖的组件 # 依赖的组件
self.vad = None self.vad = None
self.asr = None self.asr = None
self.tts = None
self._asr = _asr self._asr = _asr
self._vad = _vad self._vad = _vad
self.llm = _llm self.llm = _llm
self.tts = _tts
self.memory = _memory self.memory = _memory
self.intent = _intent self.intent = _intent
@@ -115,15 +118,13 @@ class ConnectionHandler:
# asr相关变量 # asr相关变量
self.asr_audio = [] self.asr_audio = []
self.asr_server_receive = True
# llm相关变量 # llm相关变量
self.llm_finish_task = False self.llm_finish_task = True
self.dialogue = Dialogue() self.dialogue = Dialogue()
# tts相关变量 # tts相关变量
self.tts_first_text_index = -1 self.sentence_id = None
self.tts_last_text_index = -1
# iot相关变量 # iot相关变量
self.iot_descriptors = {} self.iot_descriptors = {}
@@ -146,6 +147,8 @@ class ConnectionHandler:
) # 在原来第一道关闭的基础上加60秒,进行二道关闭 ) # 在原来第一道关闭的基础上加60秒,进行二道关闭
self.audio_format = "opus" self.audio_format = "opus"
# {"mcp":true} 表示启用MCP功能
self.features = None
async def handle_connection(self, ws): async def handle_connection(self, ws):
try: try:
@@ -194,17 +197,6 @@ class ConnectionHandler:
self._initialize_private_config() self._initialize_private_config()
# 异步初始化 # 异步初始化
self.executor.submit(self._initialize_components) self.executor.submit(self._initialize_components)
# tts 消化线程
self.tts_priority_thread = threading.Thread(
target=self._tts_priority_thread, daemon=True
)
self.tts_priority_thread.start()
# 音频播放 消化线程
self.audio_play_priority_thread = threading.Thread(
target=self._audio_play_priority_thread, daemon=True
)
self.audio_play_priority_thread.start()
try: try:
async for message in self.websocket: async for message in self.websocket:
@@ -313,25 +305,43 @@ class ConnectionHandler:
) )
def _initialize_components(self): def _initialize_components(self):
"""初始化组件""" try:
if self.config.get("prompt") is not None: self.selected_module_str = build_module_string(
self.prompt = self.config["prompt"] self.config.get("selected_module", {})
self.change_system_prompt(self.prompt) )
self.logger.bind(tag=TAG).info( update_module_string(self.selected_module_str)
f"初始化组件: prompt成功 {self.prompt[:50]}..." """初始化组件"""
if self.config.get("prompt") is not None:
self.prompt = self.config["prompt"]
self.change_system_prompt(self.prompt)
self.logger.bind(tag=TAG).info(
f"初始化组件: prompt成功 {self.prompt[:50]}..."
)
"""初始化本地组件"""
if self.vad is None:
self.vad = self._vad
if self.asr is None:
self.asr = self._initialize_asr()
# 打开语音识别通道
asyncio.run_coroutine_threadsafe(
self.asr.open_audio_channels(self), self.loop
)
if self.tts is None:
self.tts = self._initialize_tts()
# 打开语音合成通道
asyncio.run_coroutine_threadsafe(
self.tts.open_audio_channels(self), self.loop
) )
"""初始化本地组件""" """加载记忆"""
if self.vad is None: self._initialize_memory()
self.vad = self._vad """加载意图识别"""
if self.asr is None: self._initialize_intent()
self.asr = self._asr """初始化上报线程"""
"""加载记忆""" self._init_report_threads()
self._initialize_memory() except Exception as e:
"""加载意图识别""" self.logger.bind(tag=TAG).error(f"实例化组件失败: {e}")
self._initialize_intent()
"""初始化上报线程"""
self._init_report_threads()
def _init_report_threads(self): def _init_report_threads(self):
"""初始化ASR和TTS上报线程""" """初始化ASR和TTS上报线程"""
@@ -346,6 +356,30 @@ class ConnectionHandler:
self.report_thread.start() self.report_thread.start()
self.logger.bind(tag=TAG).info("TTS上报线程已启动") self.logger.bind(tag=TAG).info("TTS上报线程已启动")
def _initialize_tts(self):
"""初始化TTS"""
tts = None
if not self.need_bind:
tts = initialize_tts(self.config)
if tts is None:
tts = DefaultTTS(self.config, delete_audio_file=True)
return tts
def _initialize_asr(self):
"""初始化ASR"""
if self._asr.interface_type == InterfaceType.LOCAL:
# 如果公共ASR是本地服务,则直接返回
# 因为本地一个实例ASR,可以被多个连接共享
asr = self._asr
else:
# 如果公共ASR是远程服务,则初始化一个新实例
# 因为远程ASR,涉及到websocket连接和接收线程,需要每个连接一个实例
asr = initialize_asr(self.config)
return asr
def _initialize_private_config(self): def _initialize_private_config(self):
"""如果是从配置文件获取,则进行二次实例化""" """如果是从配置文件获取,则进行二次实例化"""
if not self.read_config_from_api: if not self.read_config_from_api:
@@ -444,6 +478,8 @@ class ConnectionHandler:
self.memory = modules["memory"] self.memory = modules["memory"]
def _initialize_memory(self): def _initialize_memory(self):
if self.memory is None:
return
"""初始化记忆模块""" """初始化记忆模块"""
self.memory.init_memory( self.memory.init_memory(
role_id=self.device_id, role_id=self.device_id,
@@ -452,7 +488,40 @@ class ConnectionHandler:
save_to_file=not self.read_config_from_api, save_to_file=not self.read_config_from_api,
) )
# 获取记忆总结配置
memory_config = self.config["Memory"]
memory_type = self.config["Memory"][self.config["selected_module"]["Memory"]][
"type"
]
# 如果使用 nomen,直接返回
if memory_type == "nomem":
return
# 使用 mem_local_short 模式
elif memory_type == "mem_local_short":
memory_llm_name = memory_config[self.config["selected_module"]["Memory"]][
"llm"
]
if memory_llm_name and memory_llm_name in self.config["LLM"]:
# 如果配置了专用LLM,则创建独立的LLM实例
from core.utils import llm as llm_utils
memory_llm_config = self.config["LLM"][memory_llm_name]
memory_llm_type = memory_llm_config.get("type", memory_llm_name)
memory_llm = llm_utils.create_instance(
memory_llm_type, memory_llm_config
)
self.logger.bind(tag=TAG).info(
f"为记忆总结创建了专用LLM: {memory_llm_name}, 类型: {memory_llm_type}"
)
self.memory.set_llm(memory_llm)
else:
# 否则使用主LLM
self.memory.set_llm(self.llm)
self.logger.bind(tag=TAG).info("使用主LLM作为意图识别模型")
def _initialize_intent(self): def _initialize_intent(self):
if self.intent is None:
return
self.intent_type = self.config["Intent"][ self.intent_type = self.config["Intent"][
self.config["selected_module"]["Intent"] self.config["selected_module"]["Intent"]
]["type"] ]["type"]
@@ -506,106 +575,26 @@ class ConnectionHandler:
# 更新系统prompt至上下文 # 更新系统prompt至上下文
self.dialogue.update_system_message(self.prompt) self.dialogue.update_system_message(self.prompt)
def chat(self, query): def chat(self, query, tool_call=False):
self.logger.bind(tag=TAG).info(f"大模型收到用户消息: {query}")
self.dialogue.put(Message(role="user", content=query))
response_message = []
processed_chars = 0 # 跟踪已处理的字符位置
try:
# 使用带记忆的对话
memory_str = None
if self.memory is not None:
future = asyncio.run_coroutine_threadsafe(
self.memory.query_memory(query), self.loop
)
memory_str = future.result()
self.logger.bind(tag=TAG).debug(f"记忆内容: {memory_str}")
llm_responses = self.llm.response(
self.session_id, self.dialogue.get_llm_dialogue_with_memory(memory_str)
)
except Exception as e:
self.logger.bind(tag=TAG).error(f"LLM 处理出错 {query}: {e}")
return None
self.llm_finish_task = False self.llm_finish_task = False
text_index = 0
for content in llm_responses:
response_message.append(content)
if self.client_abort:
break
# 合并当前全部文本并处理未分割部分
full_text = "".join(response_message)
current_text = full_text[processed_chars:] # 从未处理的位置开始
# 查找最后一个有效标点
punctuations = ("", ".", "", "?", "", "!", "", ";", "")
last_punct_pos = -1
number_flag = True
for punct in punctuations:
pos = current_text.rfind(punct)
prev_char = current_text[pos - 1] if pos - 1 >= 0 else ""
# 如果.前面是数字统一判断为小数
if prev_char.isdigit() and punct == ".":
number_flag = False
if pos > last_punct_pos and number_flag:
last_punct_pos = pos
# 找到分割点则处理
if last_punct_pos != -1:
segment_text_raw = current_text[: last_punct_pos + 1]
segment_text = get_string_no_punctuation_or_emoji(segment_text_raw)
if segment_text:
# 强制设置空字符,测试TTS出错返回语音的健壮性
# if text_index % 2 == 0:
# segment_text = " "
text_index += 1
self.recode_first_last_text(segment_text, text_index)
future = self.executor.submit(
self.speak_and_play, segment_text, text_index
)
self.tts_queue.put((future, text_index))
processed_chars += len(segment_text_raw) # 更新已处理字符位置
# 处理最后剩余的文本
full_text = "".join(response_message)
remaining_text = full_text[processed_chars:]
if remaining_text:
segment_text = get_string_no_punctuation_or_emoji(remaining_text)
if segment_text:
text_index += 1
self.recode_first_last_text(segment_text, text_index)
future = self.executor.submit(
self.speak_and_play, segment_text, text_index
)
self.tts_queue.put((future, text_index))
self.llm_finish_task = True
self.dialogue.put(Message(role="assistant", content="".join(response_message)))
self.logger.bind(tag=TAG).debug(
json.dumps(self.dialogue.get_llm_dialogue(), indent=4, ensure_ascii=False)
)
return True
def chat_with_function_calling(self, query, tool_call=False):
self.logger.bind(tag=TAG).debug(f"Chat with function calling start: {query}")
"""Chat with function calling for intent detection using streaming"""
if not tool_call: if not tool_call:
self.dialogue.put(Message(role="user", content=query)) self.dialogue.put(Message(role="user", content=query))
# Define intent functions # Define intent functions
functions = None functions = None
if hasattr(self, "func_handler"): if self.intent_type == "function_call" and hasattr(self, "func_handler"):
functions = self.func_handler.get_functions() functions = self.func_handler.get_functions()
if hasattr(self, "mcp_client"):
mcp_tools = self.mcp_client.get_available_tools()
if mcp_tools is not None and len(mcp_tools) > 0:
if functions is None:
functions = []
functions.extend(mcp_tools)
response_message = [] response_message = []
processed_chars = 0 # 跟踪已处理的字符位置
try: try:
start_time = time.time()
# 使用带记忆的对话 # 使用带记忆的对话
memory_str = None memory_str = None
if self.memory is not None: if self.memory is not None:
@@ -614,94 +603,78 @@ class ConnectionHandler:
) )
memory_str = future.result() memory_str = future.result()
# self.logger.bind(tag=TAG).info(f"对话记录: {self.dialogue.get_llm_dialogue_with_memory(memory_str)}") uuid_str = str(uuid.uuid4()).replace("-", "")
self.sentence_id = uuid_str
# 使用支持functions的streaming接口 if functions is not None:
llm_responses = self.llm.response_with_functions( # 使用支持functions的streaming接口
self.session_id, llm_responses = self.llm.response_with_functions(
self.dialogue.get_llm_dialogue_with_memory(memory_str), self.session_id,
functions=functions, self.dialogue.get_llm_dialogue_with_memory(memory_str),
) functions=functions,
)
else:
llm_responses = self.llm.response(
self.session_id,
self.dialogue.get_llm_dialogue_with_memory(memory_str),
)
except Exception as e: except Exception as e:
self.logger.bind(tag=TAG).error(f"LLM 处理出错 {query}: {e}") self.logger.bind(tag=TAG).error(f"LLM 处理出错 {query}: {e}")
return None return None
self.llm_finish_task = False
text_index = 0
# 处理流式响应 # 处理流式响应
tool_call_flag = False tool_call_flag = False
function_name = None function_name = None
function_id = None function_id = None
function_arguments = "" function_arguments = ""
content_arguments = "" content_arguments = ""
text_index = 0
self.client_abort = False
for response in llm_responses: for response in llm_responses:
content, tools_call = response if self.client_abort:
break
if functions is not None:
content, tools_call = response
if "content" in response:
content = response["content"]
tools_call = None
if content is not None and len(content) > 0:
content_arguments += content
if "content" in response: if not tool_call_flag and content_arguments.startswith("<tool_call>"):
content = response["content"] # print("content_arguments", content_arguments)
tools_call = None tool_call_flag = True
if content is not None and len(content) > 0:
content_arguments += content
if not tool_call_flag and content_arguments.startswith("<tool_call>"):
# print("content_arguments", content_arguments)
tool_call_flag = True
if tools_call is not None:
tool_call_flag = True
if tools_call[0].id is not None:
function_id = tools_call[0].id
if tools_call[0].function.name is not None:
function_name = tools_call[0].function.name
if tools_call[0].function.arguments is not None:
function_arguments += tools_call[0].function.arguments
if tools_call is not None:
tool_call_flag = True
if tools_call[0].id is not None:
function_id = tools_call[0].id
if tools_call[0].function.name is not None:
function_name = tools_call[0].function.name
if tools_call[0].function.arguments is not None:
function_arguments += tools_call[0].function.arguments
else:
content = response
if content is not None and len(content) > 0: if content is not None and len(content) > 0:
if not tool_call_flag: if not tool_call_flag:
response_message.append(content) response_message.append(content)
if text_index == 0:
if self.client_abort: self.tts.tts_text_queue.put(
break TTSMessageDTO(
sentence_id=self.sentence_id,
end_time = time.time() sentence_type=SentenceType.FIRST,
# self.logger.bind(tag=TAG).debug(f"大模型返回时间: {end_time - start_time} 秒, 生成token={content}") content_type=ContentType.ACTION,
# 处理文本分段和TTS逻辑
# 合并当前全部文本并处理未分割部分
full_text = "".join(response_message)
current_text = full_text[processed_chars:] # 从未处理的位置开始
# 查找最后一个有效标点
punctuations = ("", ".", "", "?", "", "!", "", ";", "")
last_punct_pos = -1
number_flag = True
for punct in punctuations:
pos = current_text.rfind(punct)
prev_char = current_text[pos - 1] if pos - 1 >= 0 else ""
# 如果.前面是数字统一判断为小数
if prev_char.isdigit() and punct == ".":
number_flag = False
if pos > last_punct_pos and number_flag:
last_punct_pos = pos
# 找到分割点则处理
if last_punct_pos != -1:
segment_text_raw = current_text[: last_punct_pos + 1]
segment_text = get_string_no_punctuation_or_emoji(
segment_text_raw
)
if segment_text:
text_index += 1
self.recode_first_last_text(segment_text, text_index)
future = self.executor.submit(
self.speak_and_play, segment_text, text_index
) )
self.tts_queue.put((future, text_index)) )
# 更新已处理字符位置 self.tts.tts_text_queue.put(
processed_chars += len(segment_text_raw) TTSMessageDTO(
sentence_id=self.sentence_id,
sentence_type=SentenceType.MIDDLE,
content_type=ContentType.TEXT,
content_detail=content,
)
)
text_index += 1
# 处理function call # 处理function call
if tool_call_flag: if tool_call_flag:
bHasError = False bHasError = False
@@ -736,35 +709,52 @@ class ConnectionHandler:
"arguments": function_arguments, "arguments": function_arguments,
} }
# 处理MCP工具调用 # 处理Server端MCP工具调用
if self.mcp_manager.is_mcp_tool(function_name): if self.mcp_manager.is_mcp_tool(function_name):
result = self._handle_mcp_tool_call(function_call_data) result = self._handle_mcp_tool_call(function_call_data)
elif hasattr(self, "mcp_client") and self.mcp_client.has_tool(
function_name
):
# 如果是小智端MCP工具调用
self.logger.bind(tag=TAG).debug(
f"调用小智端MCP工具: {function_name}, 参数: {function_arguments}"
)
try:
result = asyncio.run_coroutine_threadsafe(
call_mcp_tool(
self, self.mcp_client, function_name, function_arguments
),
self.loop,
).result()
self.logger.bind(tag=TAG).debug(f"MCP工具调用结果: {result}")
result = ActionResponse(
action=Action.REQLLM, result=result, response=""
)
except Exception as e:
self.logger.bind(tag=TAG).error(f"MCP工具调用失败: {e}")
result = ActionResponse(
action=Action.REQLLM, result="MCP工具调用失败", response=""
)
else: else:
# 处理系统函数 # 处理系统函数
result = self.func_handler.handle_llm_function_call( result = self.func_handler.handle_llm_function_call(
self, function_call_data self, function_call_data
) )
self._handle_function_result(result, function_call_data, text_index + 1) self._handle_function_result(result, function_call_data)
# 处理最后剩余的文本
full_text = "".join(response_message)
remaining_text = full_text[processed_chars:]
if remaining_text:
segment_text = get_string_no_punctuation_or_emoji(remaining_text)
if segment_text:
text_index += 1
self.recode_first_last_text(segment_text, text_index)
future = self.executor.submit(
self.speak_and_play, segment_text, text_index
)
self.tts_queue.put((future, text_index))
# 存储对话内容 # 存储对话内容
if len(response_message) > 0: if len(response_message) > 0:
self.dialogue.put( self.dialogue.put(
Message(role="assistant", content="".join(response_message)) Message(role="assistant", content="".join(response_message))
) )
if text_index > 0:
self.tts.tts_text_queue.put(
TTSMessageDTO(
sentence_id=self.sentence_id,
sentence_type=SentenceType.LAST,
content_type=ContentType.ACTION,
)
)
self.llm_finish_task = True self.llm_finish_task = True
self.logger.bind(tag=TAG).debug( self.logger.bind(tag=TAG).debug(
json.dumps(self.dialogue.get_llm_dialogue(), indent=4, ensure_ascii=False) json.dumps(self.dialogue.get_llm_dialogue(), indent=4, ensure_ascii=False)
@@ -814,12 +804,10 @@ class ConnectionHandler:
return ActionResponse(action=Action.REQLLM, result="工具调用出错", response="") return ActionResponse(action=Action.REQLLM, result="工具调用出错", response="")
def _handle_function_result(self, result, function_call_data, text_index): def _handle_function_result(self, result, function_call_data):
if result.action == Action.RESPONSE: # 直接回复前端 if result.action == Action.RESPONSE: # 直接回复前端
text = result.response text = result.response
self.recode_first_last_text(text, text_index) self.tts.tts_one_sentence(self, ContentType.TEXT, content_detail=text)
future = self.executor.submit(self.speak_and_play, text, text_index)
self.tts_queue.put((future, text_index))
self.dialogue.put(Message(role="assistant", content=text)) self.dialogue.put(Message(role="assistant", content=text))
elif result.action == Action.REQLLM: # 调用函数后再请求llm生成回复 elif result.action == Action.REQLLM: # 调用函数后再请求llm生成回复
text = result.result text = result.result
@@ -853,111 +841,14 @@ class ConnectionHandler:
content=text, content=text,
) )
) )
self.chat_with_function_calling(text, tool_call=True) self.chat(text, tool_call=True)
elif result.action == Action.NOTFOUND or result.action == Action.ERROR: elif result.action == Action.NOTFOUND or result.action == Action.ERROR:
text = result.result text = result.result
self.recode_first_last_text(text, text_index) self.tts.tts_one_sentence(self, ContentType.TEXT, content_detail=text)
future = self.executor.submit(self.speak_and_play, text, text_index)
self.tts_queue.put((future, text_index))
self.dialogue.put(Message(role="assistant", content=text)) self.dialogue.put(Message(role="assistant", content=text))
else: else:
pass pass
def _tts_priority_thread(self):
while not self.stop_event.is_set():
text = None
try:
try:
item = self.tts_queue.get(timeout=1)
if item is None:
continue
future, text_index = item # 解包获取 Future 和 text_index
except queue.Empty:
if self.stop_event.is_set():
break
continue
if future is None:
continue
text = None
audio_datas, tts_file = [], None
try:
self.logger.bind(tag=TAG).debug("正在处理TTS任务...")
tts_timeout = int(self.config.get("tts_timeout", 10))
tts_file, text, _ = future.result(timeout=tts_timeout)
if text is None or len(text) <= 0:
self.logger.bind(tag=TAG).error(
f"TTS出错:{text_index}: tts text is empty"
)
elif tts_file is None:
self.logger.bind(tag=TAG).error(
f"TTS出错: file is empty: {text_index}: {text}"
)
else:
self.logger.bind(tag=TAG).debug(
f"TTS生成:文件路径: {tts_file}"
)
if os.path.exists(tts_file):
if self.audio_format == "pcm":
audio_datas, _ = self.tts.audio_to_pcm_data(tts_file)
else:
audio_datas, _ = self.tts.audio_to_opus_data(tts_file)
# 在这里上报TTS数据
enqueue_tts_report(self, text, audio_datas)
else:
self.logger.bind(tag=TAG).error(
f"TTS出错:文件不存在{tts_file}"
)
except TimeoutError:
self.logger.bind(tag=TAG).error("TTS超时")
except Exception as e:
self.logger.bind(tag=TAG).error(f"TTS出错: {e}")
if not self.client_abort:
# 如果没有中途打断就发送语音
self.audio_play_queue.put((audio_datas, text, text_index))
if (
self.tts.delete_audio_file
and tts_file is not None
and os.path.exists(tts_file)
):
os.remove(tts_file)
except Exception as e:
self.logger.bind(tag=TAG).error(f"TTS任务处理错误: {e}")
self.clearSpeakStatus()
asyncio.run_coroutine_threadsafe(
self.websocket.send(
json.dumps(
{
"type": "tts",
"state": "stop",
"session_id": self.session_id,
}
)
),
self.loop,
)
self.logger.bind(tag=TAG).error(
f"tts_priority priority_thread: {text} {e}"
)
def _audio_play_priority_thread(self):
while not self.stop_event.is_set():
text = None
try:
try:
audio_datas, text, text_index = self.audio_play_queue.get(timeout=1)
except queue.Empty:
if self.stop_event.is_set():
break
continue
future = asyncio.run_coroutine_threadsafe(
sendAudioMessage(self, audio_datas, text, text_index), self.loop
)
future.result()
except Exception as e:
self.logger.bind(tag=TAG).error(
f"audio_play_priority priority_thread: {text} {e}"
)
def _report_worker(self): def _report_worker(self):
"""聊天记录上报工作线程""" """聊天记录上报工作线程"""
while not self.stop_event.is_set(): while not self.stop_event.is_set():
@@ -966,17 +857,17 @@ class ConnectionHandler:
item = self.report_queue.get(timeout=1) item = self.report_queue.get(timeout=1)
if item is None: # 检测毒丸对象 if item is None: # 检测毒丸对象
break break
type, text, audio_data, report_time = item
type, text, audio_data = item
try: try:
# 执行上报(传入二进制数据) # 检查线程池状态
report(self, type, text, audio_data) if self.executor is None:
continue
# 提交任务到线程池
self.executor.submit(
self._process_report, type, text, audio_data, report_time
)
except Exception as e: except Exception as e:
self.logger.bind(tag=TAG).error(f"聊天记录上报线程异常: {e}") self.logger.bind(tag=TAG).error(f"聊天记录上报线程异常: {e}")
finally:
# 标记任务完成
self.report_queue.task_done()
except queue.Empty: except queue.Empty:
continue continue
except Exception as e: except Exception as e:
@@ -984,82 +875,79 @@ class ConnectionHandler:
self.logger.bind(tag=TAG).info("聊天记录上报线程已退出") self.logger.bind(tag=TAG).info("聊天记录上报线程已退出")
def speak_and_play(self, text, text_index=0): def _process_report(self, type, text, audio_data, report_time):
if text is None or len(text) <= 0: """处理上报任务"""
self.logger.bind(tag=TAG).info(f"无需tts转换,query为空,{text}") try:
return None, text, text_index # 执行上报(传入二进制数据)
tts_file = self.tts.to_tts(text) report(self, type, text, audio_data, report_time)
if tts_file is None: except Exception as e:
self.logger.bind(tag=TAG).error(f"tts转换失败,{text}") self.logger.bind(tag=TAG).error(f"上报处理异常: {e}")
return None, text, text_index finally:
self.logger.bind(tag=TAG).debug(f"TTS 文件生成完毕: {tts_file}") # 标记任务完成
if self.max_output_size > 0: self.report_queue.task_done()
add_device_output(self.headers.get("device-id"), len(text))
return tts_file, text, text_index
def clearSpeakStatus(self): def clearSpeakStatus(self):
self.client_is_speaking = False
self.logger.bind(tag=TAG).debug(f"清除服务端讲话状态") self.logger.bind(tag=TAG).debug(f"清除服务端讲话状态")
self.asr_server_receive = True
self.tts_last_text_index = -1
self.tts_first_text_index = -1
def recode_first_last_text(self, text, text_index=0):
if self.tts_first_text_index == -1:
self.logger.bind(tag=TAG).info(f"大模型说出第一句话: {text}")
self.tts_first_text_index = text_index
self.tts_last_text_index = text_index
async def close(self, ws=None): async def close(self, ws=None):
"""资源清理方法""" """资源清理方法"""
try:
# 取消超时任务
if self.timeout_task:
self.timeout_task.cancel()
self.timeout_task = None
# 取消超时任务 # 清理MCP资源
if self.timeout_task: if hasattr(self, "mcp_manager") and self.mcp_manager:
self.timeout_task.cancel() await self.mcp_manager.cleanup_all()
self.timeout_task = None
# 清理MCP资源 # 触发停止事件
if hasattr(self, "mcp_manager") and self.mcp_manager: if self.stop_event:
await self.mcp_manager.cleanup_all() self.stop_event.set()
# 触发停止事件 # 清空任务队列
if self.stop_event: self.clear_queues()
self.stop_event.set()
# 清空任务队列 # 关闭WebSocket连接
self.clear_queues() if ws:
await ws.close()
elif self.websocket:
await self.websocket.close()
# 关闭WebSocket连接 # 最后关闭线程池(避免阻塞)
if ws: if self.executor:
await ws.close() self.executor.shutdown(wait=False)
elif self.websocket: self.executor = None
await self.websocket.close()
# 最后关闭线程池(避免阻塞) self.logger.bind(tag=TAG).info("连接资源已释放")
if self.executor: except Exception as e:
self.executor.shutdown(wait=False) self.logger.bind(tag=TAG).error(f"关闭连接时出错: {e}")
self.executor = None
self.logger.bind(tag=TAG).info("连接资源已释放")
def clear_queues(self): def clear_queues(self):
"""清空所有任务队列""" """清空所有任务队列"""
self.logger.bind(tag=TAG).debug( if self.tts:
f"开始清理: TTS队列大小={self.tts_queue.qsize()}, 音频队列大小={self.audio_play_queue.qsize()}" self.logger.bind(tag=TAG).debug(
) f"开始清理: TTS队列大小={self.tts.tts_text_queue.qsize()}, 音频队列大小={self.tts.tts_audio_queue.qsize()}"
)
# 使用非阻塞方式清空队列 # 使用非阻塞方式清空队列
for q in [self.tts_queue, self.audio_play_queue]: for q in [
if not q: self.tts.tts_text_queue,
continue self.tts.tts_audio_queue,
while True: self.report_queue,
try: ]:
q.get_nowait() if not q:
except queue.Empty: continue
break while True:
try:
q.get_nowait()
except queue.Empty:
break
self.logger.bind(tag=TAG).debug( self.logger.bind(tag=TAG).debug(
f"清理结束: TTS队列大小={self.tts_queue.qsize()}, 音频队列大小={self.audio_play_queue.qsize()}" f"清理结束: TTS队列大小={self.tts.tts_text_queue.qsize()}, 音频队列大小={self.tts.tts_audio_queue.qsize()}"
) )
def reset_vad_states(self): def reset_vad_states(self):
self.client_audio_buffer = bytearray() self.client_audio_buffer = bytearray()
@@ -61,7 +61,7 @@ class FunctionHandler:
self.function_registry.register_function("plugin_loader") self.function_registry.register_function("plugin_loader")
self.function_registry.register_function("get_time") self.function_registry.register_function("get_time")
self.function_registry.register_function("get_lunar") self.function_registry.register_function("get_lunar")
self.function_registry.register_function("handle_speaker_volume_or_screen_brightness") # self.function_registry.register_function("handle_speaker_volume_or_screen_brightness")
def register_config_functions(self): def register_config_functions(self):
"""注册配置中的函数,可以不同客户端使用不同的配置""" """注册配置中的函数,可以不同客户端使用不同的配置"""
+39 -12
View File
@@ -1,11 +1,18 @@
import os
import time
import json import json
from core.handle.sendAudioHandle import send_stt_message import random
from core.utils.util import remove_punctuation_and_length
import shutil import shutil
import asyncio import asyncio
import os from core.handle.sendAudioHandle import send_stt_message
import random from core.utils.util import remove_punctuation_and_length
import time from core.providers.tts.dto.dto import ContentType, InterfaceType
from core.handle.mcpHandle import (
MCPClient,
send_mcp_initialize_message,
send_mcp_tools_list_request,
)
TAG = __name__ TAG = __name__
@@ -26,8 +33,20 @@ async def handleHelloMessage(conn, msg_json):
format = audio_params.get("format") format = audio_params.get("format")
conn.logger.bind(tag=TAG).info(f"客户端音频格式: {format}") conn.logger.bind(tag=TAG).info(f"客户端音频格式: {format}")
conn.audio_format = format conn.audio_format = format
conn.asr.set_audio_format(format) if conn.asr is not None:
conn.asr.set_audio_format(format)
conn.welcome_msg["audio_params"] = audio_params conn.welcome_msg["audio_params"] = audio_params
features = msg_json.get("features")
if features:
conn.logger.bind(tag=TAG).info(f"客户端特性: {features}")
conn.features = features
if features.get("mcp"):
conn.logger.bind(tag=TAG).info("客户端支持MCP")
conn.mcp_client = MCPClient()
# 发送初始化
asyncio.create_task(send_mcp_initialize_message(conn))
# 发送mcp消息,获取tools列表
asyncio.create_task(send_mcp_tools_list_request(conn))
await conn.websocket.send(json.dumps(conn.welcome_msg)) await conn.websocket.send(json.dumps(conn.welcome_msg))
@@ -36,6 +55,10 @@ async def checkWakeupWords(conn, text):
enable_wakeup_words_response_cache = conn.config[ enable_wakeup_words_response_cache = conn.config[
"enable_wakeup_words_response_cache" "enable_wakeup_words_response_cache"
] ]
"""是否用的是非流式tts"""
if conn.tts and conn.tts.interface_type != InterfaceType.NON_STREAM:
return False
"""是否开启唤醒词加速""" """是否开启唤醒词加速"""
if not enable_wakeup_words_response_cache: if not enable_wakeup_words_response_cache:
return False return False
@@ -43,19 +66,17 @@ async def checkWakeupWords(conn, text):
_, filtered_text = remove_punctuation_and_length(text) _, filtered_text = remove_punctuation_and_length(text)
if filtered_text in conn.config.get("wakeup_words"): if filtered_text in conn.config.get("wakeup_words"):
await send_stt_message(conn, text) await send_stt_message(conn, text)
conn.tts_first_text_index = 0
conn.tts_last_text_index = 0
conn.llm_finish_task = True
file = getWakeupWordFile(WAKEUP_CONFIG["file_name"]) file = getWakeupWordFile(WAKEUP_CONFIG["file_name"])
if file is None: if file is None:
asyncio.create_task(wakeupWordsResponse(conn)) asyncio.create_task(wakeupWordsResponse(conn))
return False return False
opus_packets, _ = conn.tts.audio_to_opus_data(file)
text_hello = WAKEUP_CONFIG["text"] text_hello = WAKEUP_CONFIG["text"]
if not text_hello: if not text_hello:
text_hello = text text_hello = text
conn.audio_play_queue.put((opus_packets, text_hello, 0)) conn.tts.tts_one_sentence(
conn, ContentType.FILE, content_file=file, content_detail=text_hello
)
if time.time() - WAKEUP_CONFIG["create_time"] > WAKEUP_CONFIG["refresh_time"]: if time.time() - WAKEUP_CONFIG["create_time"] > WAKEUP_CONFIG["refresh_time"]:
asyncio.create_task(wakeupWordsResponse(conn)) asyncio.create_task(wakeupWordsResponse(conn))
return True return True
@@ -87,7 +108,13 @@ async def wakeupWordsResponse(conn):
"""唤醒词响应""" """唤醒词响应"""
wakeup_word = random.choice(WAKEUP_CONFIG["words"]) wakeup_word = random.choice(WAKEUP_CONFIG["words"])
result = conn.llm.response_no_stream(conn.config["prompt"], wakeup_word) question = (
"此刻用户正在和你说```"
+ wakeup_word
+ "```。\n请你根据以上用户的内容,进行简短回复,文字内容控制在15个字以内。\n"
+ "请勿对这条内容本身进行任何解释和回应,仅返回对用户的内容的回复。"
)
result = conn.llm.response_no_stream(conn.config["prompt"], question)
if result is None or result == "": if result is None or result == "":
return return
tts_file = await asyncio.to_thread(conn.tts.to_tts, result) tts_file = await asyncio.to_thread(conn.tts.to_tts, result)
@@ -1,9 +1,9 @@
from config.logger import setup_logging
import json import json
import uuid import uuid
from core.handle.sendAudioHandle import send_stt_message from core.handle.sendAudioHandle import send_stt_message
from core.handle.helloHandle import checkWakeupWords from core.handle.helloHandle import checkWakeupWords
from core.utils.util import remove_punctuation_and_length from core.utils.util import remove_punctuation_and_length
from core.providers.tts.dto.dto import ContentType
from core.utils.dialogue import Message from core.utils.dialogue import Message
from plugins_func.register import Action from plugins_func.register import Action
from loguru import logger from loguru import logger
@@ -109,21 +109,21 @@ async def process_intent_result(conn, intent_result, original_text):
if result.action == Action.RESPONSE: # 直接回复前端 if result.action == Action.RESPONSE: # 直接回复前端
text = result.response text = result.response
if text is not None: if text is not None:
speak_and_play(conn, text) speak_txt(conn, text)
elif result.action == Action.REQLLM: # 调用函数后再请求llm生成回复 elif result.action == Action.REQLLM: # 调用函数后再请求llm生成回复
text = result.result text = result.result
conn.dialogue.put(Message(role="tool", content=text)) conn.dialogue.put(Message(role="tool", content=text))
llm_result = conn.intent.replyResult(text, original_text) llm_result = conn.intent.replyResult(text, original_text)
if llm_result is None: if llm_result is None:
llm_result = text llm_result = text
speak_and_play(conn, llm_result) speak_txt(conn, llm_result)
elif ( elif (
result.action == Action.NOTFOUND result.action == Action.NOTFOUND
or result.action == Action.ERROR or result.action == Action.ERROR
): ):
text = result.result text = result.result
if text is not None: if text is not None:
speak_and_play(conn, text) speak_txt(conn, text)
elif function_name != "play_music": elif function_name != "play_music":
# For backward compatibility with original code # For backward compatibility with original code
# 获取当前最新的文本索引 # 获取当前最新的文本索引
@@ -131,7 +131,7 @@ async def process_intent_result(conn, intent_result, original_text):
if text is None: if text is None:
text = result.result text = result.result
if text is not None: if text is not None:
speak_and_play(conn, text) speak_txt(conn, text)
# 将函数执行放在线程池中 # 将函数执行放在线程池中
conn.executor.submit(process_function_call) conn.executor.submit(process_function_call)
@@ -142,12 +142,6 @@ async def process_intent_result(conn, intent_result, original_text):
return False return False
def speak_and_play(conn, text): def speak_txt(conn, text):
text_index = ( conn.tts.tts_one_sentence(conn, ContentType.TEXT, content_detail=text)
conn.tts_last_text_index + 1 if hasattr(conn, "tts_last_text_index") else 0
)
conn.recode_first_last_text(text, text_index)
future = conn.executor.submit(conn.speak_and_play, text, text_index)
conn.llm_finish_task = True
conn.tts_queue.put((future, text_index))
conn.dialogue.put(Message(role="assistant", content=text)) conn.dialogue.put(Message(role="assistant", content=text))
@@ -0,0 +1,374 @@
import json
import asyncio
from concurrent.futures import Future
from core.utils.util import get_vision_url
from core.utils.auth import AuthToken
TAG = __name__
class MCPClient:
"""MCPClient,用于管理MCP状态和工具"""
def __init__(self):
self.tools = {} # Dictionary for O(1) lookup
self.ready = False
self.call_results = {} # To store Futures for tool call responses
self.next_id = 1
self.lock = asyncio.Lock()
self._cached_available_tools = None # Cache for get_available_tools
def has_tool(self, name: str) -> bool:
return name in self.tools
def get_available_tools(self) -> list:
# Check if the cache is valid
if self._cached_available_tools is not None:
return self._cached_available_tools
# If cache is not valid, regenerate the list
result = []
for tool_name, tool_data in self.tools.items():
function_def = {
"name": tool_data["name"],
"description": tool_data["description"],
"parameters": {
"type": tool_data["inputSchema"].get("type", "object"),
"properties": tool_data["inputSchema"].get("properties", {}),
"required": tool_data["inputSchema"].get("required", []),
},
}
result.append({"type": "function", "function": function_def})
self._cached_available_tools = result # Store the generated list in cache
return result
async def is_ready(self) -> bool:
async with self.lock:
return self.ready
async def set_ready(self, status: bool):
async with self.lock:
self.ready = status
async def add_tool(self, tool_data: dict):
async with self.lock:
self.tools[tool_data["name"]] = tool_data
self._cached_available_tools = (
None # Invalidate the cache when a tool is added
)
async def get_next_id(self) -> int:
async with self.lock:
current_id = self.next_id
self.next_id += 1
return current_id
async def register_call_result_future(self, id: int, future: Future):
async with self.lock:
self.call_results[id] = future
async def resolve_call_result(self, id: int, result: any):
async with self.lock:
if id in self.call_results:
future = self.call_results.pop(id)
if not future.done():
future.set_result(result)
async def reject_call_result(self, id: int, exception: Exception):
async with self.lock:
if id in self.call_results:
future = self.call_results.pop(id)
if not future.done():
future.set_exception(exception)
async def cleanup_call_result(self, id: int):
async with self.lock:
if id in self.call_results:
self.call_results.pop(id)
async def send_mcp_message(conn, payload: dict):
"""Helper to send MCP messages, encapsulating common logic."""
if not conn.features.get("mcp"):
conn.logger.bind(tag=TAG).warning("客户端不支持MCP,无法发送MCP消息")
return
message = json.dumps({"type": "mcp", "payload": payload})
try:
await conn.websocket.send(message)
conn.logger.bind(tag=TAG).info(f"成功发送MCP消息: {message}")
except Exception as e:
conn.logger.bind(tag=TAG).error(f"发送MCP消息失败: {e}")
async def handle_mcp_message(conn, mcp_client: MCPClient, payload: dict):
"""处理MCP消息,包括初始化、工具列表和工具调用响应等"""
conn.logger.bind(tag=TAG).info(f"处理MCP消息: {payload}")
if not isinstance(payload, dict):
conn.logger.bind(tag=TAG).error("MCP消息缺少payload字段或格式错误")
return
# Handle result
if "result" in payload:
result = payload["result"]
msg_id = int(payload.get("id", 0))
# Check for tool call response first
if msg_id in mcp_client.call_results:
conn.logger.bind(tag=TAG).debug(
f"收到工具调用响应,ID: {msg_id}, 结果: {result}"
)
await mcp_client.resolve_call_result(msg_id, result)
return
if msg_id == 1: # mcpInitializeID
conn.logger.bind(tag=TAG).debug("收到MCP初始化响应")
server_info = result.get("serverInfo")
if isinstance(server_info, dict):
name = server_info.get("name")
version = server_info.get("version")
conn.logger.bind(tag=TAG).info(
f"客户端MCP服务器信息: name={name}, version={version}"
)
await send_mcp_tools_list_request(
conn
) # After initialization, request tool list
return
elif msg_id == 2: # mcpToolsListID
conn.logger.bind(tag=TAG).debug("收到MCP工具列表响应")
if isinstance(result, dict) and "tools" in result:
tools_data = result["tools"]
if not isinstance(tools_data, list):
conn.logger.bind(tag=TAG).error("工具列表格式错误")
return
conn.logger.bind(tag=TAG).info(
f"客户端设备支持的工具数量: {len(tools_data)}"
)
for i, tool in enumerate(tools_data):
if not isinstance(tool, dict):
continue
name = tool.get("name", "")
description = tool.get("description", "")
input_schema = {"type": "object", "properties": {}, "required": []}
if "inputSchema" in tool and isinstance(tool["inputSchema"], dict):
schema = tool["inputSchema"]
input_schema["type"] = schema.get("type", "object")
input_schema["properties"] = schema.get("properties", {})
input_schema["required"] = [
s for s in schema.get("required", []) if isinstance(s, str)
]
new_tool = {
"name": name,
"description": description,
"inputSchema": input_schema,
}
await mcp_client.add_tool(new_tool)
conn.logger.bind(tag=TAG).debug(f"客户端工具 #{i+1}: {name}")
next_cursor = result.get("nextCursor", "")
if next_cursor:
conn.logger.bind(tag=TAG).info(
f"有更多工具,nextCursor: {next_cursor}"
)
await send_mcp_tools_list_continue_request(conn, next_cursor)
else:
await mcp_client.set_ready(True)
conn.logger.bind(tag=TAG).info("所有工具已获取,MCP客户端准备就绪")
return
# Handle method calls (requests from the client)
elif "method" in payload:
method = payload["method"]
conn.logger.bind(tag=TAG).info(f"收到MCP客户端请求: {method}")
elif "error" in payload:
error_data = payload["error"]
error_msg = error_data.get("message", "未知错误")
conn.logger.bind(tag=TAG).error(f"收到MCP错误响应: {error_msg}")
msg_id = int(payload.get("id", 0))
if msg_id in mcp_client.call_results:
await mcp_client.reject_call_result(
msg_id, Exception(f"MCP错误: {error_msg}")
)
# --- Outgoing MCP Messages ---
async def send_mcp_initialize_message(conn):
"""发送MCP初始化消息"""
vision_url = get_vision_url(conn.config)
# 密钥生成token
auth = AuthToken(conn.config["server"]["auth_key"])
token = auth.generate_token(conn.headers.get("device-id"))
vision = {
"url": vision_url,
"token": token,
}
conn.logger.bind(tag=TAG).info(f"视觉服务信息: {vision}")
payload = {
"jsonrpc": "2.0",
"id": 1, # mcpInitializeID
"method": "initialize",
"params": {
"protocolVersion": "2024-11-05",
"capabilities": {
"roots": {"listChanged": True},
"sampling": {},
"vision": vision,
},
"clientInfo": {
"name": "XiaozhiClient",
"version": "1.0.0",
},
},
}
conn.logger.bind(tag=TAG).info("发送MCP初始化消息")
await send_mcp_message(conn, payload)
async def send_mcp_tools_list_request(conn):
"""发送MCP工具列表请求"""
payload = {
"jsonrpc": "2.0",
"id": 2, # mcpToolsListID
"method": "tools/list",
}
conn.logger.bind(tag=TAG).debug("发送MCP工具列表请求")
await send_mcp_message(conn, payload)
async def send_mcp_tools_list_continue_request(conn, cursor: str):
"""发送带有cursor的MCP工具列表请求"""
payload = {
"jsonrpc": "2.0",
"id": 2, # mcpToolsListID (same ID for continuation)
"method": "tools/list",
"params": {"cursor": cursor},
}
conn.logger.bind(tag=TAG).info(f"发送带cursor的MCP工具列表请求: {cursor}")
await send_mcp_message(conn, payload)
async def call_mcp_tool(
conn, mcp_client: MCPClient, tool_name: str, args: str = "{}", timeout: int = 30
):
"""
调用指定的工具,并等待响应
"""
if not await mcp_client.is_ready():
raise RuntimeError("MCP客户端尚未准备就绪")
if not mcp_client.has_tool(tool_name):
raise ValueError(f"工具 {tool_name} 不存在")
tool_call_id = await mcp_client.get_next_id()
result_future = asyncio.Future()
await mcp_client.register_call_result_future(tool_call_id, result_future)
# 处理参数
try:
if isinstance(args, str):
# 确保字符串是有效的JSON
if not args.strip():
arguments = {}
else:
try:
# 尝试直接解析
arguments = json.loads(args)
except json.JSONDecodeError:
# 如果解析失败,尝试合并多个JSON对象
try:
# 使用正则表达式匹配所有JSON对象
import re
json_objects = re.findall(r"\{[^{}]*\}", args)
if len(json_objects) > 1:
# 合并所有JSON对象
merged_dict = {}
for json_str in json_objects:
try:
obj = json.loads(json_str)
if isinstance(obj, dict):
merged_dict.update(obj)
except json.JSONDecodeError:
continue
if merged_dict:
arguments = merged_dict
else:
raise ValueError(f"无法解析任何有效的JSON对象: {args}")
else:
raise ValueError(f"参数JSON解析失败: {args}")
except Exception as e:
conn.logger.bind(tag=TAG).error(
f"参数JSON解析失败: {str(e)}, 原始参数: {args}"
)
raise ValueError(f"参数JSON解析失败: {str(e)}")
elif isinstance(args, dict):
arguments = args
else:
raise ValueError(f"参数类型错误,期望字符串或字典,实际类型: {type(args)}")
# 确保参数是字典类型
if not isinstance(arguments, dict):
raise ValueError(f"参数必须是字典类型,实际类型: {type(arguments)}")
except Exception as e:
if not isinstance(e, ValueError):
raise ValueError(f"参数处理失败: {str(e)}")
raise e
payload = {
"jsonrpc": "2.0",
"id": tool_call_id,
"method": "tools/call",
"params": {"name": tool_name, "arguments": arguments},
}
conn.logger.bind(tag=TAG).info(
f"发送客户端mcp工具调用请求: {tool_name},参数: {args}"
)
await send_mcp_message(conn, payload)
try:
# Wait for response or timeout
raw_result = await asyncio.wait_for(result_future, timeout=timeout)
conn.logger.bind(tag=TAG).info(
f"客户端mcp工具调用 {tool_name} 成功,原始结果: {raw_result}"
)
if isinstance(raw_result, dict):
if raw_result.get("isError") is True:
error_msg = raw_result.get(
"error", "工具调用返回错误,但未提供具体错误信息"
)
raise RuntimeError(f"工具调用错误: {error_msg}")
content = raw_result.get("content")
if isinstance(content, list) and len(content) > 0:
if isinstance(content[0], dict) and "text" in content[0]:
# 直接返回文本内容,不进行JSON解析
return content[0]["text"]
# 如果结果不是预期的格式,将其转换为字符串
return str(raw_result)
except asyncio.TimeoutError:
await mcp_client.cleanup_call_result(tool_call_id)
raise TimeoutError("工具调用请求超时")
except Exception as e:
await mcp_client.cleanup_call_result(tool_call_id)
raise e
@@ -1,10 +1,9 @@
import time
import copy
from core.utils.util import remove_punctuation_and_length
from core.handle.sendAudioHandle import send_stt_message from core.handle.sendAudioHandle import send_stt_message
from core.handle.intentHandler import handle_user_intent from core.handle.intentHandler import handle_user_intent
from core.utils.output_counter import check_device_output_limit from core.utils.output_counter import check_device_output_limit
from core.handle.reportHandle import enqueue_asr_report from core.handle.abortHandle import handleAbortMessage
import time
from core.handle.sendAudioHandle import SentenceType
from core.utils.util import audio_to_data from core.utils.util import audio_to_data
TAG = __name__ TAG = __name__
@@ -12,45 +11,20 @@ TAG = __name__
async def handleAudioMessage(conn, audio): async def handleAudioMessage(conn, audio):
if conn.vad is None: if conn.vad is None:
conn.logger.bind(tag=TAG).warning("VAD模块未初始化,继续等待")
return return
if not conn.asr_server_receive: if conn.asr is None or not hasattr(conn.asr, "conn") or conn.asr.conn is None:
conn.logger.bind(tag=TAG).debug(f"前期数据处理中,暂停接收") conn.logger.bind(tag=TAG).warning("ASR模块未初始化或通道未就绪,继续等待")
return return
if conn.client_listen_mode == "auto" or conn.client_listen_mode == "realtime": # 当前片段是否有人说话
have_voice = conn.vad.is_vad(conn, audio) have_voice = conn.vad.is_vad(conn, audio)
else: if have_voice:
have_voice = conn.client_have_voice if conn.client_is_speaking:
await handleAbortMessage(conn)
# 如果本次没有声音,本段也没声音,就把声音丢弃了 # 设备长时间空闲检测,用于say goodbye
if have_voice == False and conn.client_have_voice == False: await no_voice_close_connect(conn, have_voice)
await no_voice_close_connect(conn) # 接收音频
conn.asr_audio.append(audio) await conn.asr.receive_audio(audio, have_voice)
conn.asr_audio = conn.asr_audio[
-10:
] # 保留最新的10帧音频内容,解决ASR句首丢字问题
return
conn.client_no_voice_last_time = 0.0
conn.asr_audio.append(audio)
# 如果本段有声音,且已经停止了
if conn.client_voice_stop:
conn.client_abort = False
conn.asr_server_receive = False
# 音频太短了,无法识别
if len(conn.asr_audio) < 15:
conn.asr_server_receive = True
else:
raw_text, _ = await conn.asr.speech_to_text(conn.asr_audio, conn.session_id) # 确保ASR模块返回原始文本
conn.logger.bind(tag=TAG).info(f"识别文本: {raw_text}")
text_len, _ = remove_punctuation_and_length(raw_text)
if text_len > 0:
# 使用自定义模块进行上报
enqueue_asr_report(conn, raw_text, copy.deepcopy(conn.asr_audio))
await startToChat(conn, raw_text)
else:
conn.asr_server_receive = True
conn.asr_audio.clear()
conn.reset_vad_states()
async def startToChat(conn, text): async def startToChat(conn, text):
@@ -65,25 +39,25 @@ async def startToChat(conn, text):
): ):
await max_out_size(conn) await max_out_size(conn)
return return
if conn.client_is_speaking:
await handleAbortMessage(conn)
# 首先进行意图分析 # 首先进行意图分析
intent_handled = await handle_user_intent(conn, text) intent_handled = await handle_user_intent(conn, text)
if intent_handled: if intent_handled:
# 如果意图已被处理,不再进行聊天 # 如果意图已被处理,不再进行聊天
conn.asr_server_receive = True
return return
# 意图未被处理,继续常规聊天流程 # 意图未被处理,继续常规聊天流程
await send_stt_message(conn, text) await send_stt_message(conn, text)
if conn.intent_type == "function_call": conn.executor.submit(conn.chat, text)
# 使用支持function calling的聊天方法
conn.executor.submit(conn.chat_with_function_calling, text)
else:
conn.executor.submit(conn.chat, text)
async def no_voice_close_connect(conn): async def no_voice_close_connect(conn, have_voice):
if have_voice:
conn.client_no_voice_last_time = 0.0
return
if conn.client_no_voice_last_time == 0.0: if conn.client_no_voice_last_time == 0.0:
conn.client_no_voice_last_time = time.time() * 1000 conn.client_no_voice_last_time = time.time() * 1000
else: else:
@@ -97,7 +71,6 @@ async def no_voice_close_connect(conn):
): ):
conn.close_after_chat = True conn.close_after_chat = True
conn.client_abort = False conn.client_abort = False
conn.asr_server_receive = False
end_prompt = conn.config.get("end_prompt", {}) end_prompt = conn.config.get("end_prompt", {})
if end_prompt and end_prompt.get("enable", True) is False: if end_prompt and end_prompt.get("enable", True) is False:
conn.logger.bind(tag=TAG).info("结束对话,无需发送结束提示语") conn.logger.bind(tag=TAG).info("结束对话,无需发送结束提示语")
@@ -105,19 +78,16 @@ async def no_voice_close_connect(conn):
return return
prompt = end_prompt.get("prompt") prompt = end_prompt.get("prompt")
if not prompt: if not prompt:
prompt = "请你以时间过得真快未来头,用富有感情、依依不舍的话来结束这场对话吧。!" prompt = "请你以```时间过得真快```未来头,用富有感情、依依不舍的话来结束这场对话吧。!"
await startToChat(conn, prompt) await startToChat(conn, prompt)
async def max_out_size(conn): async def max_out_size(conn):
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!" 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!"
await send_stt_message(conn, text) await send_stt_message(conn, text)
conn.tts_first_text_index = 0
conn.tts_last_text_index = 0
conn.llm_finish_task = True
file_path = "config/assets/max_output_size.wav" file_path = "config/assets/max_output_size.wav"
opus_packets, _ = audio_to_data(file_path) opus_packets, _ = audio_to_data(file_path)
conn.audio_play_queue.put((opus_packets, text, 0)) conn.tts.tts_audio_queue.put((SentenceType.LAST, opus_packets, text))
conn.close_after_chat = True conn.close_after_chat = True
@@ -132,14 +102,11 @@ async def check_bind_device(conn):
text = f"请登录控制面板,输入{conn.bind_code},绑定设备。" text = f"请登录控制面板,输入{conn.bind_code},绑定设备。"
await send_stt_message(conn, text) await send_stt_message(conn, text)
conn.tts_first_text_index = 0
conn.tts_last_text_index = 6
conn.llm_finish_task = True
# 播放提示音 # 播放提示音
music_path = "config/assets/bind_code.wav" music_path = "config/assets/bind_code.wav"
opus_packets, _ = audio_to_data(music_path) opus_packets, _ = audio_to_data(music_path)
conn.audio_play_queue.put((opus_packets, text, 0)) conn.tts.tts_audio_queue.put((SentenceType.FIRST, opus_packets, text))
# 逐个播放数字 # 逐个播放数字
for i in range(6): # 确保只播放6位数字 for i in range(6): # 确保只播放6位数字
@@ -147,16 +114,14 @@ async def check_bind_device(conn):
digit = conn.bind_code[i] digit = conn.bind_code[i]
num_path = f"config/assets/bind_code/{digit}.wav" num_path = f"config/assets/bind_code/{digit}.wav"
num_packets, _ = audio_to_data(num_path) num_packets, _ = audio_to_data(num_path)
conn.audio_play_queue.put((num_packets, None, i + 1)) conn.tts.tts_audio_queue.put((SentenceType.MIDDLE, num_packets, None))
except Exception as e: except Exception as e:
conn.logger.bind(tag=TAG).error(f"播放数字音频失败: {e}") conn.logger.bind(tag=TAG).error(f"播放数字音频失败: {e}")
continue continue
conn.tts.tts_audio_queue.put((SentenceType.LAST, [], None))
else: else:
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。" 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。"
await send_stt_message(conn, text) await send_stt_message(conn, text)
conn.tts_first_text_index = 0
conn.tts_last_text_index = 0
conn.llm_finish_task = True
music_path = "config/assets/bind_not_found.wav" music_path = "config/assets/bind_not_found.wav"
opus_packets, _ = audio_to_data(music_path) opus_packets, _ = audio_to_data(music_path)
conn.audio_play_queue.put((opus_packets, text, 0)) conn.tts.tts_audio_queue.put((SentenceType.LAST, opus_packets, text))
@@ -9,6 +9,8 @@ TTS上报功能已集成到ConnectionHandler类中。
具体实现请参考core/connection.py中的相关代码。 具体实现请参考core/connection.py中的相关代码。
""" """
import time
import opuslib_next import opuslib_next
from config.manage_api_client import report as manage_report from config.manage_api_client import report as manage_report
@@ -16,7 +18,7 @@ from config.manage_api_client import report as manage_report
TAG = __name__ TAG = __name__
def report(conn, type, text, opus_data): def report(conn, type, text, opus_data, report_time):
"""执行聊天记录上报操作 """执行聊天记录上报操作
Args: Args:
@@ -24,6 +26,7 @@ def report(conn, type, text, opus_data):
type: 上报类型,1为用户,2为智能体 type: 上报类型,1为用户,2为智能体
text: 合成文本 text: 合成文本
opus_data: opus音频数据 opus_data: opus音频数据
report_time: 上报时间
""" """
try: try:
if opus_data: if opus_data:
@@ -37,6 +40,7 @@ def report(conn, type, text, opus_data):
chat_type=type, chat_type=type,
content=text, content=text,
audio=audio_data, audio=audio_data,
report_time=report_time,
) )
except Exception as e: except Exception as e:
conn.logger.bind(tag=TAG).error(f"聊天记录上报失败: {e}") conn.logger.bind(tag=TAG).error(f"聊天记录上报失败: {e}")
@@ -104,12 +108,12 @@ def enqueue_tts_report(conn, text, opus_data):
try: try:
# 使用连接对象的队列,传入文本和二进制数据而非文件路径 # 使用连接对象的队列,传入文本和二进制数据而非文件路径
if conn.chat_history_conf == 2: if conn.chat_history_conf == 2:
conn.report_queue.put((2, text, opus_data)) conn.report_queue.put((2, text, opus_data, int(time.time())))
conn.logger.bind(tag=TAG).debug( conn.logger.bind(tag=TAG).debug(
f"TTS数据已加入上报队列: {conn.device_id}, 音频大小: {len(opus_data)} " f"TTS数据已加入上报队列: {conn.device_id}, 音频大小: {len(opus_data)} "
) )
else: else:
conn.report_queue.put((2, text, None)) conn.report_queue.put((2, text, None, int(time.time())))
conn.logger.bind(tag=TAG).debug( conn.logger.bind(tag=TAG).debug(
f"TTS数据已加入上报队列: {conn.device_id}, 不上报音频" f"TTS数据已加入上报队列: {conn.device_id}, 不上报音频"
) )
@@ -132,14 +136,14 @@ def enqueue_asr_report(conn, text, opus_data):
try: try:
# 使用连接对象的队列,传入文本和二进制数据而非文件路径 # 使用连接对象的队列,传入文本和二进制数据而非文件路径
if conn.chat_history_conf == 2: if conn.chat_history_conf == 2:
conn.report_queue.put((1, text, opus_data)) conn.report_queue.put((1, text, opus_data, int(time.time())))
conn.logger.bind(tag=TAG).debug( conn.logger.bind(tag=TAG).debug(
f"ASR数据已加入上报队列: {conn.device_id}, 音频大小: {len(opus_data)} " f"ASR数据已加入上报队列: {conn.device_id}, 音频大小: {len(opus_data)} "
) )
else: else:
conn.report_queue.put((1, text, None)) conn.report_queue.put((1, text, None, int(time.time())))
conn.logger.bind(tag=TAG).debug( conn.logger.bind(tag=TAG).debug(
f"ASR数据已加入上报队列: {conn.device_id}, 不上报音频" f"ASR数据已加入上报队列: {conn.device_id}, 不上报音频"
) )
except Exception as e: except Exception as e:
conn.logger.bind(tag=TAG).error(f"加入ASR上报队列失败: {text}, {e}") conn.logger.bind(tag=TAG).debug(f"加入ASR上报队列失败: {text}, {e}")
@@ -1,7 +1,9 @@
import json import json
import asyncio import asyncio
import time import time
from core.providers.tts.dto.dto import SentenceType
from core.utils.util import get_string_no_punctuation_or_emoji, analyze_emotion from core.utils.util import get_string_no_punctuation_or_emoji, analyze_emotion
from loguru import logger
TAG = __name__ TAG = __name__
@@ -30,7 +32,7 @@ emoji_map = {
} }
async def sendAudioMessage(conn, audios, text, text_index=0): async def sendAudioMessage(conn, sentenceType, audios, text):
# 发送句子开始消息 # 发送句子开始消息
if text is not None: if text is not None:
emotion = analyze_emotion(text) emotion = analyze_emotion(text)
@@ -45,25 +47,30 @@ async def sendAudioMessage(conn, audios, text, text_index=0):
} }
) )
) )
pre_buffer = False
if text_index == conn.tts_first_text_index: if conn.tts.tts_audio_first_sentence and text is not None:
conn.logger.bind(tag=TAG).info(f"发送第一段语音: {text}") conn.logger.bind(tag=TAG).info(f"发送第一段语音: {text}")
conn.tts.tts_audio_first_sentence = False
pre_buffer = True
await send_tts_message(conn, "sentence_start", text) await send_tts_message(conn, "sentence_start", text)
is_first_audio = text_index == conn.tts_first_text_index await sendAudio(conn, audios, pre_buffer)
await sendAudio(conn, audios, pre_buffer=is_first_audio)
await send_tts_message(conn, "sentence_end", text) await send_tts_message(conn, "sentence_end", text)
# 发送结束消息(如果是最后一个文本) # 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.llm_finish_task and sentenceType == SentenceType.LAST:
await send_tts_message(conn, "stop", None) await send_tts_message(conn, "stop", None)
conn.client_is_speaking = False
if conn.close_after_chat: if conn.close_after_chat:
await conn.close() await conn.close()
# 播放音频 # 播放音频
async def sendAudio(conn, audios, pre_buffer=True): async def sendAudio(conn, audios, pre_buffer=True):
if audios is None or len(audios) == 0:
return
# 流控参数优化 # 流控参数优化
frame_duration = 60 # 帧时长(毫秒),匹配 Opus 编码 frame_duration = 60 # 帧时长(毫秒),匹配 Opus 编码
start_time = time.perf_counter() start_time = time.perf_counter()
@@ -82,6 +89,7 @@ async def sendAudio(conn, audios, pre_buffer=True):
# 播放剩余音频帧 # 播放剩余音频帧
for opus_packet in remaining_audios: for opus_packet in remaining_audios:
if conn.client_abort: if conn.client_abort:
conn.client_abort = False
return return
# 每分钟重置一次计时器 # 每分钟重置一次计时器
@@ -125,9 +133,15 @@ async def send_tts_message(conn, state, text=None):
async def send_stt_message(conn, text): async def send_stt_message(conn, text):
end_prompt_str = conn.config.get("end_prompt", {}).get("prompt")
if end_prompt_str and end_prompt_str == text:
await send_tts_message(conn, "start")
return
"""发送 STT 状态消息""" """发送 STT 状态消息"""
stt_text = get_string_no_punctuation_or_emoji(text) stt_text = get_string_no_punctuation_or_emoji(text)
await conn.websocket.send( await conn.websocket.send(
json.dumps({"type": "stt", "text": stt_text, "session_id": conn.session_id}) json.dumps({"type": "stt", "text": stt_text, "session_id": conn.session_id})
) )
conn.client_is_speaking = True
await send_tts_message(conn, "start") await send_tts_message(conn, "start")
+11 -2
View File
@@ -1,6 +1,7 @@
import json import json
from core.handle.abortHandle import handleAbortMessage from core.handle.abortHandle import handleAbortMessage
from core.handle.helloHandle import handleHelloMessage from core.handle.helloHandle import handleHelloMessage
from core.handle.mcpHandle import handle_mcp_message
from core.utils.util import remove_punctuation_and_length, filter_sensitive_info from core.utils.util import remove_punctuation_and_length, filter_sensitive_info
from core.handle.receiveAudioHandle import startToChat, handleAudioMessage from core.handle.receiveAudioHandle import startToChat, handleAudioMessage
from core.handle.sendAudioHandle import send_stt_message, send_tts_message from core.handle.sendAudioHandle import send_stt_message, send_tts_message
@@ -41,12 +42,13 @@ async def handleTextMessage(conn, message):
if len(conn.asr_audio) > 0: if len(conn.asr_audio) > 0:
await handleAudioMessage(conn, b"") await handleAudioMessage(conn, b"")
elif msg_json["state"] == "detect": elif msg_json["state"] == "detect":
conn.asr_server_receive = False
conn.client_have_voice = False conn.client_have_voice = False
conn.asr_audio.clear() conn.asr_audio.clear()
if "text" in msg_json: if "text" in msg_json:
original_text = msg_json["text"] # 保留原始文本 original_text = msg_json["text"] # 保留原始文本
filtered_len, filtered_text = remove_punctuation_and_length(original_text) filtered_len, filtered_text = remove_punctuation_and_length(
original_text
)
# 识别是否是唤醒词 # 识别是否是唤醒词
is_wakeup_words = filtered_text in conn.config.get("wakeup_words") is_wakeup_words = filtered_text in conn.config.get("wakeup_words")
@@ -57,6 +59,7 @@ async def handleTextMessage(conn, message):
# 如果是唤醒词,且关闭了唤醒词回复,就不用回答 # 如果是唤醒词,且关闭了唤醒词回复,就不用回答
await send_stt_message(conn, original_text) await send_stt_message(conn, original_text)
await send_tts_message(conn, "stop", None) await send_tts_message(conn, "stop", None)
conn.client_is_speaking = False
elif is_wakeup_words: elif is_wakeup_words:
# 上报纯文字数据(复用ASR上报功能,但不提供音频数据) # 上报纯文字数据(复用ASR上报功能,但不提供音频数据)
enqueue_asr_report(conn, "嘿,你好呀", []) enqueue_asr_report(conn, "嘿,你好呀", [])
@@ -72,6 +75,10 @@ async def handleTextMessage(conn, message):
asyncio.create_task(handleIotDescriptors(conn, msg_json["descriptors"])) asyncio.create_task(handleIotDescriptors(conn, msg_json["descriptors"]))
if "states" in msg_json: if "states" in msg_json:
asyncio.create_task(handleIotStatus(conn, msg_json["states"])) asyncio.create_task(handleIotStatus(conn, msg_json["states"]))
elif msg_json["type"] == "mcp":
conn.logger.bind(tag=TAG).info(f"收到mcp消息:{message}")
if "payload" in msg_json:
asyncio.create_task(handle_mcp_message(conn, conn.mcp_client, msg_json["payload"]))
elif msg_json["type"] == "server": elif msg_json["type"] == "server":
# 记录日志时过滤敏感信息 # 记录日志时过滤敏感信息
conn.logger.bind(tag=TAG).info( conn.logger.bind(tag=TAG).info(
@@ -151,5 +158,7 @@ async def handleTextMessage(conn, message):
# 重启服务器 # 重启服务器
elif msg_json["action"] == "restart": elif msg_json["action"] == "restart":
await conn.handle_restart(msg_json) await conn.handle_restart(msg_json)
else:
conn.logger.bind(tag=TAG).error(f"收到未知类型消息:{message}")
except json.JSONDecodeError: except json.JSONDecodeError:
await conn.websocket.send(message) await conn.websocket.send(message)
+71
View File
@@ -0,0 +1,71 @@
import asyncio
from aiohttp import web
from config.logger import setup_logging
from core.api.ota_handler import OTAHandler
from core.api.vision_handler import VisionHandler
TAG = __name__
class SimpleHttpServer:
def __init__(self, config: dict):
self.config = config
self.logger = setup_logging()
self.ota_handler = OTAHandler(config)
self.vision_handler = VisionHandler(config)
def _get_websocket_url(self, local_ip: str, port: int) -> str:
"""获取websocket地址
Args:
local_ip: 本地IP地址
port: 端口号
Returns:
str: websocket地址
"""
server_config = self.config["server"]
websocket_config = server_config.get("websocket")
if websocket_config and "" not in websocket_config:
return websocket_config
else:
return f"ws://{local_ip}:{port}/xiaozhi/v1/"
async def start(self):
server_config = self.config["server"]
host = server_config.get("ip", "0.0.0.0")
port = int(server_config.get("http_port", 8003))
if port:
app = web.Application()
read_config_from_api = server_config.get("read_config_from_api", False)
if not read_config_from_api:
# 如果没有开启智控台,只是单模块运行,就需要再添加简单OTA接口,用于下发websocket接口
app.add_routes(
[
web.get("/xiaozhi/ota/", self.ota_handler.handle_get),
web.post("/xiaozhi/ota/", self.ota_handler.handle_post),
web.options("/xiaozhi/ota/", self.ota_handler.handle_post),
]
)
# 添加路由
app.add_routes(
[
web.get("/mcp/vision/explain", self.vision_handler.handle_get),
web.post("/mcp/vision/explain", self.vision_handler.handle_post),
web.options("/mcp/vision/explain", self.vision_handler.handle_post),
]
)
# 运行服务
runner = web.AppRunner(app)
await runner.setup()
site = web.TCPSite(runner, host, port)
await site.start()
# 保持服务运行
while True:
await asyncio.sleep(3600) # 每隔 1 小时检查一次
+12 -2
View File
@@ -92,11 +92,21 @@ class MCPClient:
args=self.config.get("args", []), args=self.config.get("args", []),
env=env, env=env,
) )
stdio_r, stdio_w = await stack.enter_async_context(stdio_client(params)) stdio_r, stdio_w = await stack.enter_async_context(
stdio_client(params)
)
read_stream, write_stream = stdio_r, stdio_w read_stream, write_stream = stdio_r, stdio_w
# 建立SSEClient # 建立SSEClient
elif "url" in self.config: elif "url" in self.config:
sse_r, sse_w = await stack.enter_async_context(sse_client(self.config["url"])) if "API_ACCESS_TOKEN" in self.config:
headers = {
"Authorization": f"Bearer {self.config['API_ACCESS_TOKEN']}"
}
else:
headers = {}
sse_r, sse_w = await stack.enter_async_context(
sse_client(self.config["url"], headers=headers)
)
read_stream, write_stream = sse_r, sse_w read_stream, write_stream = sse_r, sse_w
else: else:
@@ -2,9 +2,6 @@ import http.client
import json import json
import asyncio import asyncio
from typing import Optional, Tuple, List from typing import Optional, Tuple, List
import opuslib_next
import wave
import io
import os import os
import uuid import uuid
import hmac import hmac
@@ -16,6 +13,7 @@ import time
from datetime import datetime from datetime import datetime
from config.logger import setup_logging from config.logger import setup_logging
from core.providers.asr.base import ASRProviderBase from core.providers.asr.base import ASRProviderBase
from core.providers.asr.dto.dto import InterfaceType
TAG = __name__ TAG = __name__
logger = setup_logging() logger = setup_logging()
@@ -92,6 +90,7 @@ class AccessToken:
class ASRProvider(ASRProviderBase): class ASRProvider(ASRProviderBase):
def __init__(self, config: dict, delete_audio_file: bool): def __init__(self, config: dict, delete_audio_file: bool):
super().__init__() super().__init__()
self.interface_type = InterfaceType.NON_STREAM
"""阿里云ASR初始化""" """阿里云ASR初始化"""
# 新增空值判断逻辑 # 新增空值判断逻辑
self.access_key_id = config.get("access_key_id") self.access_key_id = config.get("access_key_id")
@@ -155,12 +154,6 @@ class ASRProvider(ASRProviderBase):
# f"剩余 {remaining:.2f}秒") # f"剩余 {remaining:.2f}秒")
return time.time() > self.expire_time return time.time() > self.expire_time
def generate_filename(self, extension=".wav"):
return os.path.join(
self.output_file,
f"tts-{__name__}{datetime.now().date()}@{uuid.uuid4().hex}{extension}",
)
def _construct_request_url(self) -> str: def _construct_request_url(self) -> str:
"""构造请求URL,包含参数""" """构造请求URL,包含参数"""
request = f"{self.base_url}?appkey={self.app_key}" request = f"{self.base_url}?appkey={self.app_key}"
@@ -171,21 +164,6 @@ class ASRProvider(ASRProviderBase):
request += "&enable_voice_detection=false" request += "&enable_voice_detection=false"
return request return request
def save_audio_to_file(self, pcm_data: List[bytes], session_id: str) -> str:
"""PCM数据保存为WAV文件"""
module_name = __name__.split(".")[-1]
file_name = f"asr_{module_name}_{session_id}_{uuid.uuid4()}.wav"
file_path = os.path.join(self.output_dir, file_name)
with wave.open(file_path, "wb") as wf:
wf.setnchannels(1) # 单声道
wf.setsampwidth(2) # 16-bit
wf.setframerate(self.sample_rate)
wf.writeframes(b"".join(pcm_data))
logger.bind(tag=TAG).debug(f"音频文件已保存至: {file_path}")
return file_path
async def _send_request(self, pcm_data: bytes) -> Optional[str]: async def _send_request(self, pcm_data: bytes) -> Optional[str]:
"""发送请求到阿里云ASR服务""" """发送请求到阿里云ASR服务"""
try: try:
@@ -1,18 +1,10 @@
import base64
import hashlib
import hmac
import json
import time import time
from datetime import datetime, timezone
import os import os
import uuid
from typing import Optional, Tuple, List from typing import Optional, Tuple, List
import wave
import opuslib_next
from aip import AipSpeech from aip import AipSpeech
from core.providers.asr.base import ASRProviderBase from core.providers.asr.base import ASRProviderBase
from config.logger import setup_logging from config.logger import setup_logging
from core.providers.asr.dto.dto import InterfaceType
TAG = __name__ TAG = __name__
logger = setup_logging() logger = setup_logging()
@@ -21,6 +13,7 @@ logger = setup_logging()
class ASRProvider(ASRProviderBase): class ASRProvider(ASRProviderBase):
def __init__(self, config: dict, delete_audio_file: bool = True): def __init__(self, config: dict, delete_audio_file: bool = True):
super().__init__() super().__init__()
self.interface_type = InterfaceType.NON_STREAM
self.app_id = config.get("app_id") self.app_id = config.get("app_id")
self.api_key = config.get("api_key") self.api_key = config.get("api_key")
self.secret_key = config.get("secret_key") self.secret_key = config.get("secret_key")
@@ -36,20 +29,6 @@ class ASRProvider(ASRProviderBase):
# 确保输出目录存在 # 确保输出目录存在
os.makedirs(self.output_dir, exist_ok=True) os.makedirs(self.output_dir, exist_ok=True)
def save_audio_to_file(self, pcm_data: List[bytes], session_id: str) -> str:
"""PCM数据保存为WAV文件"""
module_name = __name__.split(".")[-1]
file_name = f"asr_{module_name}_{session_id}_{uuid.uuid4()}.wav"
file_path = os.path.join(self.output_dir, file_name)
with wave.open(file_path, "wb") as wf:
wf.setnchannels(1)
wf.setsampwidth(2) # 2 bytes = 16-bit
wf.setframerate(16000)
wf.writeframes(b"".join(pcm_data))
return file_path
async def speech_to_text( async def speech_to_text(
self, opus_data: List[bytes], session_id: str self, opus_data: List[bytes], session_id: str
) -> Tuple[Optional[str], Optional[str]]: ) -> Tuple[Optional[str], Optional[str]]:
+87 -14
View File
@@ -1,7 +1,15 @@
from abc import ABC, abstractmethod import os
from typing import Optional, Tuple, List import time
import copy
import uuid
import wave
import opuslib_next import opuslib_next
from abc import ABC, abstractmethod
from config.logger import setup_logging from config.logger import setup_logging
from typing import Optional, Tuple, List
from core.utils.util import remove_punctuation_and_length
from core.handle.reportHandle import enqueue_asr_report
from core.handle.receiveAudioHandle import startToChat
TAG = __name__ TAG = __name__
logger = setup_logging() logger = setup_logging()
@@ -10,11 +18,66 @@ logger = setup_logging()
class ASRProviderBase(ABC): class ASRProviderBase(ABC):
def __init__(self): def __init__(self):
self.audio_format = "opus" self.audio_format = "opus"
self.conn = None
# 打开音频通道
# 这里默认是非流式的处理方式
# 流式处理方式请在子类中重写
async def open_audio_channels(self, conn):
self.conn = conn
# 接收音频
# 这里默认是非流式的处理方式
# 流式处理方式请在子类中重写
async def receive_audio(self, audio, audio_have_voice):
if (
self.conn.client_listen_mode == "auto"
or self.conn.client_listen_mode == "realtime"
):
have_voice = audio_have_voice
else:
have_voice = self.conn.client_have_voice
# 如果本次没有声音,本段也没声音,就把声音丢弃了
self.conn.asr_audio.append(audio)
if have_voice == False and self.conn.client_have_voice == False:
self.conn.asr_audio = self.conn.asr_audio[-10:]
return
# 如果本段有声音,且已经停止了
if self.conn.client_voice_stop:
asr_audio_task = copy.deepcopy(self.conn.asr_audio)
self.conn.asr_audio.clear()
# 音频太短了,无法识别
self.conn.reset_vad_states()
if len(asr_audio_task) > 15:
await self.handle_voice_stop(asr_audio_task)
# 处理语音停止
async def handle_voice_stop(self, asr_audio_task):
raw_text, _ = await self.speech_to_text(
asr_audio_task, self.conn.session_id
) # 确保ASR模块返回原始文本
self.conn.logger.bind(tag=TAG).info(f"识别文本: {raw_text}")
text_len, _ = remove_punctuation_and_length(raw_text)
if text_len > 0:
# 使用自定义模块进行上报
await startToChat(self.conn, raw_text)
enqueue_asr_report(self.conn, raw_text, asr_audio_task)
@abstractmethod
def save_audio_to_file(self, pcm_data: List[bytes], session_id: str) -> str: def save_audio_to_file(self, pcm_data: List[bytes], session_id: str) -> str:
"""PCM数据保存为WAV文件""" """PCM数据保存为WAV文件"""
pass module_name = __name__.split(".")[-1]
file_name = f"asr_{module_name}_{session_id}_{uuid.uuid4()}.wav"
file_path = os.path.join(self.output_dir, file_name)
with wave.open(file_path, "wb") as wf:
wf.setnchannels(1)
wf.setsampwidth(2) # 2 bytes = 16-bit
wf.setframerate(16000)
wf.writeframes(b"".join(pcm_data))
return file_path
@abstractmethod @abstractmethod
async def speech_to_text( async def speech_to_text(
@@ -30,15 +93,25 @@ class ASRProviderBase(ABC):
@staticmethod @staticmethod
def decode_opus(opus_data: List[bytes]) -> bytes: def decode_opus(opus_data: List[bytes]) -> bytes:
"""将Opus音频数据解码为PCM数据""" """将Opus音频数据解码为PCM数据"""
try:
decoder = opuslib_next.Decoder(16000, 1) # 16kHz, 单声道
pcm_data = []
buffer_size = 960 # 每次处理960个采样点
decoder = opuslib_next.Decoder(16000, 1) # 16kHz, 单声道 for opus_packet in opus_data:
pcm_data = [] try:
# 使用较小的缓冲区大小进行处理
pcm_frame = decoder.decode(opus_packet, buffer_size)
if pcm_frame:
pcm_data.append(pcm_frame)
except opuslib_next.OpusError as e:
logger.bind(tag=TAG).warning(f"Opus解码错误,跳过当前数据包: {e}")
continue
except Exception as e:
logger.bind(tag=TAG).error(f"音频处理错误: {e}", exc_info=True)
continue
for opus_packet in opus_data: return pcm_data
try: except Exception as e:
pcm_frame = decoder.decode(opus_packet, 960) # 960 samples = 60ms logger.bind(tag=TAG).error(f"音频解码过程发生错误: {e}", exc_info=True)
pcm_data.append(pcm_frame) return []
except opuslib_next.OpusError as e:
logger.bind(tag=TAG).error(f"Opus解码错误: {e}", exc_info=True)
return pcm_data
+470 -219
View File
@@ -1,283 +1,534 @@
import time
import io
import wave
import os
from typing import Optional, Tuple, List
import uuid
import websockets
import json import json
import gzip import gzip
import uuid
import asyncio
import websockets
import opuslib_next import opuslib_next
from core.providers.asr.base import ASRProviderBase from core.providers.asr.base import ASRProviderBase
from config.logger import setup_logging from config.logger import setup_logging
from core.providers.asr.dto.dto import InterfaceType
import threading
TAG = __name__ TAG = __name__
logger = setup_logging() logger = setup_logging()
CLIENT_FULL_REQUEST = 0b0001 CLIENT_FULL_REQUEST = 0b0001
CLIENT_AUDIO_ONLY_REQUEST = 0b0010 CLIENT_AUDIO_ONLY_REQUEST = 0b0010
NO_SEQUENCE = 0b0000
NEG_SEQUENCE = 0b0010
SERVER_FULL_RESPONSE = 0b1001 SERVER_FULL_RESPONSE = 0b1001
SERVER_ACK = 0b1011 SERVER_ACK = 0b1011
SERVER_ERROR_RESPONSE = 0b1111 SERVER_ERROR_RESPONSE = 0b1111
NO_SEQUENCE = 0b0000
NO_SERIALIZATION = 0b0000 NEG_SEQUENCE = 0b0010
JSON = 0b0001 JSON_SERIALIZATION = 0b0001
THRIFT = 0b0011 GZIP_COMPRESSION = 0b0001
CUSTOM_TYPE = 0b1111 PROTOCOL_VERSION = 0b0001
NO_COMPRESSION = 0b0000
GZIP = 0b0001
CUSTOM_COMPRESSION = 0b1111
def parse_response(res):
"""
protocol_version(4 bits), header_size(4 bits),
message_type(4 bits), message_type_specific_flags(4 bits)
serialization_method(4 bits) message_compression(4 bits)
reserved 8bits) 保留字段
header_extensions 扩展头(大小等于 8 * 4 * (header_size - 1) )
payload 类似与http 请求体
"""
protocol_version = res[0] >> 4
header_size = res[0] & 0x0F
message_type = res[1] >> 4
message_type_specific_flags = res[1] & 0x0F
serialization_method = res[2] >> 4
message_compression = res[2] & 0x0F
reserved = res[3]
header_extensions = res[4 : header_size * 4]
payload = res[header_size * 4 :]
result = {}
payload_msg = None
payload_size = 0
if message_type == SERVER_FULL_RESPONSE:
payload_size = int.from_bytes(payload[:4], "big", signed=True)
payload_msg = payload[4:]
elif message_type == SERVER_ACK:
seq = int.from_bytes(payload[:4], "big", signed=True)
result["seq"] = seq
if len(payload) >= 8:
payload_size = int.from_bytes(payload[4:8], "big", signed=False)
payload_msg = payload[8:]
elif message_type == SERVER_ERROR_RESPONSE:
code = int.from_bytes(payload[:4], "big", signed=False)
result["code"] = code
payload_size = int.from_bytes(payload[4:8], "big", signed=False)
payload_msg = payload[8:]
if payload_msg is None:
return result
if message_compression == GZIP:
payload_msg = gzip.decompress(payload_msg)
if serialization_method == JSON:
payload_msg = json.loads(str(payload_msg, "utf-8"))
elif serialization_method != NO_SERIALIZATION:
payload_msg = str(payload_msg, "utf-8")
result["payload_msg"] = payload_msg
result["payload_size"] = payload_size
return result
class ASRProvider(ASRProviderBase): class ASRProvider(ASRProviderBase):
def __init__(self, config: dict, delete_audio_file: bool): def __init__(self, config, delete_audio_file):
super().__init__() super().__init__()
self.appid = config.get("appid") self.interface_type = InterfaceType.STREAM
self.config = config
self.text = ""
self.max_retries = 3
self.retry_delay = 2 # 重试延迟秒数
self.recv_lock = asyncio.Lock() # 添加接收锁
self.reconnect_lock = asyncio.Lock() # 添加重连锁
self.last_reconnect_time = 0 # 上次重连时间
self.reconnect_cooldown = 1 # 增加重连冷却时间到10秒
self.reconnect_count = 0 # 当前重连次数
self.max_reconnect_count = 3 # 减少最大重连次数到3次
self.asr_thread = None # ASR监听线程
self.thread_lock = threading.Lock() # 线程管理锁
self.is_reconnecting = False # 添加重连状态标志
# 添加会话管理相关属性
self._session_lock = asyncio.Lock() # 会话操作的并发锁
self._current_session_id = None # 当前会话ID
self._session_started = False # 会话是否已开始
self._session_finished = False # 会话是否已结束
self._session_close_event = asyncio.Event() # 添加会话关闭事件
self.appid = str(config.get("appid"))
self.cluster = config.get("cluster") self.cluster = config.get("cluster")
self.access_token = config.get("access_token") self.access_token = config.get("access_token")
self.boosting_table_name = config.get("boosting_table_name", "") self.boosting_table_name = config.get("boosting_table_name", "")
self.correct_table_name = config.get("correct_table_name", "") self.correct_table_name = config.get("correct_table_name", "")
self.output_dir = config.get("output_dir") self.output_dir = config.get("output_dir", "temp/")
self.delete_audio_file = delete_audio_file self.delete_audio_file = delete_audio_file
self.host = "openspeech.bytedance.com" self.ws_url = "wss://openspeech.bytedance.com/api/v2/asr"
self.ws_url = f"wss://{self.host}/api/v2/asr" self.uid = config.get("uid", "streaming_asr_service")
self.success_code = 1000 self.workflow = config.get(
self.seg_duration = 15000 "workflow", "audio_in,resample,partition,vad,fe,decode,itn,nlu_punctuate"
)
self.result_type = config.get("result_type", "single")
self.format = config.get("format", "raw")
self.codec = config.get("codec", "pcm")
self.rate = config.get("sample_rate", 16000)
self.language = config.get("language", "zh-CN")
self.bits = config.get("bits", 16)
self.channel = config.get("channel", 1)
self.auth_method = config.get("auth_method", "token")
self.secret = config.get("secret", "access_secret")
self.decoder = opuslib_next.Decoder(16000, 1)
self.asr_ws = None
self.forward_task = None
self.conn = None
# 确保输出目录存在 ###################################################################################
os.makedirs(self.output_dir, exist_ok=True) # 豆包流式ASR重写父类的方法--开始
###################################################################################
async def open_audio_channels(self, conn):
await super().open_audio_channels(conn)
def save_audio_to_file(self, pcm_data: List[bytes], session_id: str) -> str: async with self._session_lock:
"""PCM数据保存为WAV文件""" # 如果正在重连,等待重连完成
module_name = __name__.split(".")[-1] if self.is_reconnecting:
file_name = f"asr_{module_name}_{session_id}_{uuid.uuid4()}.wav" logger.bind(tag=TAG).info("等待当前重连完成...")
file_path = os.path.join(self.output_dir, file_name) await self._session_close_event.wait()
self._session_close_event.clear()
with wave.open(file_path, "wb") as wf: # 如果已有会话未结束,先关闭它
wf.setnchannels(1) if self._session_started and not self._session_finished:
wf.setsampwidth(2) # 2 bytes = 16-bit logger.bind(tag=TAG).warning(
wf.setframerate(16000) f"发现未关闭的会话 {self._current_session_id},正在关闭..."
wf.writeframes(b"".join(pcm_data)) )
if self.asr_ws is not None:
try:
await self.asr_ws.close()
except Exception as e:
logger.bind(tag=TAG).warning(f"关闭旧连接时发生错误: {e}")
finally:
self.asr_ws = None
self._session_finished = True
self._session_close_event.set()
return file_path # 重置会话状态
self._current_session_id = str(uuid.uuid4())
self._session_started = True
self._session_finished = False
self.is_reconnecting = True
@staticmethod try:
def _generate_header( retry_count = 0
message_type=CLIENT_FULL_REQUEST, message_type_specific_flags=NO_SEQUENCE while retry_count < self.max_retries:
) -> bytearray: try:
"""Generate protocol header.""" headers = (
header = bytearray() self.token_auth() if self.auth_method == "token" else None
header_size = 1 )
header.append((0b0001 << 4) | header_size) # Protocol version self.asr_ws = await websockets.connect(
header.append((message_type << 4) | message_type_specific_flags) self.ws_url,
header.append((0b0001 << 4) | 0b0001) # JSON serialization & GZIP compression additional_headers=headers,
header.append(0x00) # reserved max_size=1000000000,
return header ping_interval=None,
ping_timeout=None,
close_timeout=10,
)
def _construct_request(self, reqid) -> dict: # 发送初始化请求
"""Construct the request payload.""" request_params = self.construct_request(
return { self._current_session_id
)
try:
payload_bytes = str.encode(json.dumps(request_params))
payload_bytes = gzip.compress(payload_bytes)
full_client_request = self.generate_header()
full_client_request.extend(
(len(payload_bytes)).to_bytes(4, "big")
)
full_client_request.extend(payload_bytes)
await self.asr_ws.send(full_client_request)
except Exception as e:
logger.bind(tag=TAG).error(f"发送初始化请求失败: {e}")
raise e
# 等待初始化响应
try:
init_res = await self.asr_ws.recv()
self.parse_response(init_res)
except Exception as e:
logger.bind(tag=TAG).error(f"ASR服务初始化失败: {e}")
raise e
# 启动接收ASR结果的异步任务
with self.thread_lock:
if (
self.asr_thread is None
or not self.asr_thread.is_alive()
):
logger.bind(tag=TAG).info("创建新的ASR监听线程...")
self.asr_thread = threading.Thread(
target=self._start_monitor_asr_response_thread,
daemon=True,
)
self.asr_thread.start()
# 等待一小段时间确保线程启动
await asyncio.sleep(0.1)
if not self.asr_thread.is_alive():
logger.bind(tag=TAG).error("ASR监听线程启动失败")
raise Exception("ASR监听线程启动失败")
logger.bind(tag=TAG).info("ASR监听线程已启动")
return
except websockets.exceptions.WebSocketException as e:
retry_count += 1
if retry_count < self.max_retries:
logger.bind(tag=TAG).warning(
f"WebSocket连接失败,正在进行第{retry_count}次重试: {e}"
)
await asyncio.sleep(self.retry_delay)
else:
logger.bind(tag=TAG).warning(
f"WebSocket连接失败,已达到最大重试次数: {e}"
)
raise
except Exception as e:
logger.bind(tag=TAG).error(f"WebSocket连接发生未知错误: {e}")
raise
finally:
self.is_reconnecting = False
self._session_close_event.set()
async def receive_audio(self, audio, _):
if not isinstance(audio, bytes):
return
try:
# 解码opus得到PCM数据
pcm_frame = self.decoder.decode(audio, 960)
payload = gzip.compress(pcm_frame)
audio_request = bytearray(self.generate_audio_default_header())
audio_request.extend(len(payload).to_bytes(4, "big"))
audio_request.extend(payload)
if self.asr_ws:
await self.asr_ws.send(audio_request)
except Exception as e:
logger.bind(tag=TAG).debug(f"发送音频数据时发生错误: {e}")
###################################################################################
# 豆包流式ASR重写父类的方法--结束
###################################################################################
def construct_request(self, reqid):
req = {
"app": { "app": {
"appid": f"{self.appid}", "appid": self.appid,
"cluster": self.cluster, "cluster": self.cluster,
"token": self.access_token, "token": self.access_token,
}, },
"user": { "user": {"uid": self.uid},
"uid": str(uuid.uuid4()),
},
"request": { "request": {
"reqid": reqid, "reqid": reqid,
"show_utterances": False, "workflow": self.workflow,
"show_utterances": True,
"result_type": self.result_type,
"sequence": 1, "sequence": 1,
"boosting_table_name": self.boosting_table_name, "boosting_table_name": self.boosting_table_name,
"correct_table_name": self.correct_table_name, "correct_table_name": self.correct_table_name,
}, },
"audio": { "audio": {
"format": "raw", "format": self.format,
"rate": 16000, "codec": self.codec,
"language": "zh-CN", "rate": self.rate,
"bits": 16, "language": self.language,
"channel": 1, "bits": self.bits,
"codec": "raw", "channel": self.channel,
}, },
} }
return req
async def _send_request( def token_auth(self):
self, audio_data: List[bytes], segment_size: int return {"Authorization": f"Bearer; {self.access_token}"}
) -> Optional[str]:
"""Send request to Volcano ASR service.""" def generate_header(
self,
version=PROTOCOL_VERSION,
message_type=CLIENT_FULL_REQUEST,
message_type_specific_flags=NO_SEQUENCE,
serial_method=JSON_SERIALIZATION,
compression_type=GZIP_COMPRESSION,
reserved_data=0x00,
extension_header: bytes = b"",
):
"""
生成协议头:
- 第1字节:高4位:协议版本,低4位:头部大小(单位 4 字节)
- 第2字节:高4位:消息类型,低4位:消息类型特定标志
- 第3字节:高4位:序列化方式,低4位:压缩方式
- 第4字节:保留字段
- 后续:扩展头(如果有)
"""
header = bytearray()
header_size = int(len(extension_header) / 4) + 1
header.append((version << 4) | header_size)
header.append((message_type << 4) | message_type_specific_flags)
header.append((serial_method << 4) | compression_type)
header.append(reserved_data)
header.extend(extension_header)
return header
def generate_full_default_header(self):
# full client request 默认头
return self.generate_header(
version=PROTOCOL_VERSION,
message_type=CLIENT_FULL_REQUEST,
message_type_specific_flags=NO_SEQUENCE,
serial_method=JSON_SERIALIZATION,
compression_type=GZIP_COMPRESSION,
)
def generate_audio_default_header(self):
# 普通音频片段请求
return self.generate_header(
version=PROTOCOL_VERSION,
message_type=CLIENT_AUDIO_ONLY_REQUEST,
message_type_specific_flags=NO_SEQUENCE,
serial_method=JSON_SERIALIZATION,
compression_type=GZIP_COMPRESSION,
)
def generate_last_audio_default_header(self):
# 最后一个音频片段标志
return self.generate_header(
version=PROTOCOL_VERSION,
message_type=CLIENT_AUDIO_ONLY_REQUEST,
message_type_specific_flags=NEG_SEQUENCE, # 用 NEG_SEQUENCE 表示结束
serial_method=JSON_SERIALIZATION,
compression_type=GZIP_COMPRESSION,
)
def _start_monitor_asr_response_thread(self):
# 初始化链接
try: try:
auth_header = {"Authorization": "Bearer; {}".format(self.access_token)} with self.thread_lock:
async with websockets.connect( if self.conn is None or self.conn.loop is None:
self.ws_url, additional_headers=auth_header logger.bind(tag=TAG).error(
) as websocket: "无法启动ASR监听线程:conn或loop未初始化"
# Prepare request data )
request_params = self._construct_request(str(uuid.uuid4())) return
payload_bytes = str.encode(json.dumps(request_params))
payload_bytes = gzip.compress(payload_bytes)
full_client_request = self._generate_header()
full_client_request.extend(
(len(payload_bytes)).to_bytes(4, "big")
) # payload size(4 bytes)
full_client_request.extend(payload_bytes) # payload
# Send header and metadata try:
# full_client_request logger.bind(tag=TAG).info("开始启动ASR监听...")
await websocket.send(full_client_request) asyncio.run_coroutine_threadsafe(
res = await websocket.recv() self._forward_asr_results(), loop=self.conn.loop
result = parse_response(res) )
if ( logger.bind(tag=TAG).info("ASR监听已启动")
"payload_msg" in result except Exception as e:
and result["payload_msg"]["code"] != self.success_code logger.bind(tag=TAG).error(f"启动ASR监听线程失败: {e}")
): except Exception as e:
logger.bind(tag=TAG).error(f"ASR error: {result}") logger.bind(tag=TAG).error(f"ASR监听线程发生未预期的错误: {e}")
return None
for seq, (chunk, last) in enumerate( async def _forward_asr_results(self):
self.slice_data(audio_data, segment_size), 1 try:
): while not self.conn.stop_event.is_set():
if last: try:
audio_only_request = self._generate_header( if self.asr_ws is None:
message_type=CLIENT_AUDIO_ONLY_REQUEST, # 检查是否需要重连
message_type_specific_flags=NEG_SEQUENCE, async with self.reconnect_lock:
current_time = asyncio.get_event_loop().time()
if (
current_time - self.last_reconnect_time
< self.reconnect_cooldown
):
await asyncio.sleep(1)
continue
if self.reconnect_count >= self.max_reconnect_count:
logger.bind(tag=TAG).error(
"达到最大重连次数限制,停止重连"
)
await asyncio.sleep(self.reconnect_cooldown)
self.reconnect_count = 0
continue
self.last_reconnect_time = current_time
self.reconnect_count += 1
logger.bind(tag=TAG).info(
f"尝试重新连接ASR服务... (第{self.reconnect_count}次)"
)
await self.open_audio_channels(self.conn)
continue
# 使用锁来确保同一时间只有一个协程在接收数据
async with self.recv_lock:
response = await self.asr_ws.recv()
result = self.parse_response(response)
# 检查是否需要重连
if result.get("need_reconnect", False):
logger.bind(tag=TAG).info(
"检测到需要重连的错误,准备重新连接..."
) )
else: if self.asr_ws is not None:
audio_only_request = self._generate_header( try:
message_type=CLIENT_AUDIO_ONLY_REQUEST await self.asr_ws.close()
) except Exception as e:
payload_bytes = gzip.compress(chunk) logger.bind(tag=TAG).warning(
audio_only_request.extend( f"关闭旧连接时发生错误: {e}"
(len(payload_bytes)).to_bytes(4, "big") )
) # payload size(4 bytes) finally:
audio_only_request.extend(payload_bytes) # payload self.asr_ws = None
# Send audio data continue
await websocket.send(audio_only_request)
# Receive response if "payload_msg" in result:
response = await websocket.recv() if "result" in result["payload_msg"]:
result = parse_response(response) # 检查是否有utterances并且definite为True
utterances = result["payload_msg"]["result"][0].get(
"utterances", []
)
for utterance in utterances:
if utterance.get("definite", False):
self.text = utterance["text"]
await self.handle_voice_stop(None)
break
if ( except websockets.ConnectionClosed:
"payload_msg" in result logger.bind(tag=TAG).debug("ASR服务连接已关闭,准备重连...")
and result["payload_msg"]["code"] == self.success_code # 确保关闭旧连接
): if self.asr_ws is not None:
if len(result["payload_msg"]["result"]) > 0: try:
return result["payload_msg"]["result"][0]["text"] await self.asr_ws.close()
return None except Exception as e:
else: logger.bind(tag=TAG).warning(f"关闭旧连接时发生错误: {e}")
logger.bind(tag=TAG).error(f"ASR error: {result}") finally:
return None self.asr_ws = None
# 等待冷却时间
await asyncio.sleep(self.reconnect_cooldown)
continue
except Exception as e:
if not self.conn.stop_event.is_set():
logger.bind(tag=TAG).error(f"ASR监听发生错误: {e}")
await asyncio.sleep(self.retry_delay)
continue
except Exception as e: except Exception as e:
logger.bind(tag=TAG).error(f"ASR request failed: {e}", exc_info=True) logger.bind(tag=TAG).error(f"ASR监听线程发生错误: {e}")
return None # 确保在发生严重错误时也能继续尝试重连
if not self.conn.stop_event.is_set():
await asyncio.sleep(self.retry_delay)
await self._forward_asr_results() # 递归重试
@staticmethod async def speech_to_text(self, opus_data, session_id):
def slice_data(data: bytes, chunk_size: int) -> (list, bool): result = self.text
self.text = "" # 清空text
return result, None
def parse_response(self, res: bytes) -> dict:
""" """
slice data 解析 ASR 服务返回的二进制响应。
:param data: wav data 根据协议格式解析头部和 payload,若采用 GZIP 压缩则先解压,再根据 JSON 反序列化。
:param chunk_size: the segment size in one request
:return: segment data, last flag
""" """
data_len = len(data) protocol_version = res[0] >> 4
offset = 0 header_size = res[0] & 0x0F
while offset + chunk_size < data_len: message_type = res[1] >> 4
yield data[offset : offset + chunk_size], False serialization_method = res[2] >> 4
offset += chunk_size message_compression = res[2] & 0x0F
payload = res[header_size * 4 :]
result = {}
payload_msg = None
payload_size = 0
if message_type == SERVER_FULL_RESPONSE:
payload_size = int.from_bytes(payload[:4], "big", signed=True)
payload_msg = payload[4:]
elif message_type == SERVER_ACK:
seq = int.from_bytes(payload[:4], "big", signed=True)
result["seq"] = seq
if len(payload) >= 8:
payload_size = int.from_bytes(payload[4:8], "big", signed=False)
payload_msg = payload[8:]
elif message_type == SERVER_ERROR_RESPONSE:
code = int.from_bytes(payload[:4], "big", signed=False)
result["code"] = code
payload_size = int.from_bytes(payload[4:8], "big", signed=False)
payload_msg = payload[8:]
if payload_msg is None:
return result
if message_compression == GZIP_COMPRESSION:
payload_msg = gzip.decompress(payload_msg)
if serialization_method == JSON_SERIALIZATION:
payload_msg = json.loads(payload_msg.decode("utf-8"))
else: else:
yield data[offset:data_len], True payload_msg = payload_msg.decode("utf-8")
result["payload_msg"] = payload_msg
result["payload_size"] = payload_size
async def speech_to_text( # 错误码处理
self, opus_data: List[bytes], session_id: str if "code" in result:
) -> Tuple[Optional[str], Optional[str]]: error_code = result["code"]
"""将语音数据转换为文本""" error_message = ""
file_path = None if error_code == 1000:
try: error_message = "成功"
# 合并所有opus数据包 elif error_code == 1001:
if self.audio_format == "pcm": error_message = "请求参数无效:请求参数缺失必需字段/字段值无效/重复请求"
pcm_data = opus_data elif error_code == 1002:
error_message = "无访问权限:token无效/过期/无权访问指定服务"
elif error_code == 1003:
error_message = "访问超频:当前appid访问QPS超出设定阈值"
elif error_code == 1004:
error_message = "访问超额:当前appid访问次数超出限制"
elif error_code == 1005:
error_message = "服务器繁忙:服务过载,无法处理当前请求"
elif error_code == 1010:
error_message = "音频过长:音频数据时长超出阈值"
elif error_code == 1011:
error_message = "音频过大:音频数据大小超出阈值"
elif error_code == 1012:
error_message = "音频格式无效:音频header有误/无法进行音频解码"
elif error_code == 1013:
error_message = "音频静音:音频未识别出任何文本结果"
elif error_code >= 1020 and error_code <= 1022:
error_message = "识别相关错误:需要重连"
if error_code == 1020:
error_message = "识别等待超时:等待下一包就绪超时"
elif error_code == 1021:
error_message = "识别处理超时:识别处理过程超时"
elif error_code == 1022:
error_message = "识别错误:识别过程中发生错误"
else: else:
pcm_data = self.decode_opus(opus_data) error_message = "未知错误:未归类错误"
combined_pcm_data = b"".join(pcm_data)
# 判断是否保存为WAV文件 logger.bind(tag=TAG).debug(
if self.delete_audio_file: f"ASR错误: {error_message} (错误码: {error_code})"
pass )
else:
file_path = self.save_audio_to_file(pcm_data, session_id)
# 直接使用PCM数据 # 如果是识别相关错误,标记需要重连
# 计算分段大小 (单声道, 16bit, 16kHz采样率) if error_code >= 1020 or error_code == 1001:
size_per_sec = 1 * 2 * 16000 # nchannels * sampwidth * framerate result["need_reconnect"] = True
segment_size = int(size_per_sec * self.seg_duration / 1000)
# 语音识别 return result
start_time = time.time()
text = await self._send_request(combined_pcm_data, segment_size) async def close_session(self):
if text: """关闭当前会话"""
logger.bind(tag=TAG).debug( async with self._session_lock:
f"语音识别耗时: {time.time() - start_time:.3f}s | 结果: {text}" if not self._session_started:
logger.bind(tag=TAG).warning("尝试关闭未开始的会话")
return
if self._session_finished:
logger.bind(tag=TAG).warning(
f"会话 {self._current_session_id} 已经关闭"
) )
return text, file_path return
return "", file_path
except Exception as e: try:
logger.bind(tag=TAG).error(f"语音识别失败: {e}", exc_info=True) if self.asr_ws is not None:
return "", file_path await self.asr_ws.close()
except Exception as e:
logger.bind(tag=TAG).warning(f"关闭WebSocket连接时发生错误: {e}")
finally:
self.asr_ws = None
self._session_finished = True
self._session_started = False
self._current_session_id = None
# 重置重连计数
self.reconnect_count = 0
async def close(self):
"""资源清理方法"""
await self.close_session()
@@ -0,0 +1,9 @@
from enum import Enum
from typing import Union, Optional
class InterfaceType(Enum):
# 接口类型
STREAM = "STREAM" # 流式接口
NON_STREAM = "NON_STREAM" # 非流式接口
LOCAL = "LOCAL" # 本地服务
@@ -1,18 +1,21 @@
import time import time
import wave
import os import os
import sys import sys
import io import io
from config.logger import setup_logging from config.logger import setup_logging
from typing import Optional, Tuple, List from typing import Optional, Tuple, List
import uuid
from core.providers.asr.base import ASRProviderBase from core.providers.asr.base import ASRProviderBase
from funasr import AutoModel from funasr import AutoModel
from funasr.utils.postprocess_utils import rich_transcription_postprocess from funasr.utils.postprocess_utils import rich_transcription_postprocess
import shutil
from core.providers.asr.dto.dto import InterfaceType
TAG = __name__ TAG = __name__
logger = setup_logging() logger = setup_logging()
MAX_RETRIES = 2
RETRY_DELAY = 1 # 重试延迟(秒)
# 捕获标准输出 # 捕获标准输出
class CaptureOutput: class CaptureOutput:
@@ -34,6 +37,7 @@ class CaptureOutput:
class ASRProvider(ASRProviderBase): class ASRProvider(ASRProviderBase):
def __init__(self, config: dict, delete_audio_file: bool): def __init__(self, config: dict, delete_audio_file: bool):
super().__init__() super().__init__()
self.interface_type = InterfaceType.LOCAL
self.model_dir = config.get("model_dir") self.model_dir = config.get("model_dir")
self.output_dir = config.get("output_dir") # 修正配置键名 self.output_dir = config.get("output_dir") # 修正配置键名
self.delete_audio_file = delete_audio_file self.delete_audio_file = delete_audio_file
@@ -49,65 +53,74 @@ class ASRProvider(ASRProviderBase):
# device="cuda:0", # 启用GPU加速 # device="cuda:0", # 启用GPU加速
) )
def save_audio_to_file(self, pcm_data: List[bytes], session_id: str) -> str:
"""PCM数据保存为WAV文件"""
module_name = __name__.split(".")[-1]
file_name = f"asr_{module_name}_{session_id}_{uuid.uuid4()}.wav"
file_path = os.path.join(self.output_dir, file_name)
with wave.open(file_path, "wb") as wf:
wf.setnchannels(1)
wf.setsampwidth(2) # 2 bytes = 16-bit
wf.setframerate(16000)
wf.writeframes(b"".join(pcm_data))
return file_path
async def speech_to_text( async def speech_to_text(
self, opus_data: List[bytes], session_id: str self, opus_data: List[bytes], session_id: str
) -> Tuple[Optional[str], Optional[str]]: ) -> Tuple[Optional[str], Optional[str]]:
"""语音转文本主处理逻辑""" """语音转文本主处理逻辑"""
file_path = None file_path = None
try: retry_count = 0
# 合并所有opus数据包
if self.audio_format == "pcm":
pcm_data = opus_data
else:
pcm_data = self.decode_opus(opus_data)
combined_pcm_data = b"".join(pcm_data) while retry_count < MAX_RETRIES:
try:
# 合并所有opus数据包
if self.audio_format == "pcm":
pcm_data = opus_data
else:
pcm_data = self.decode_opus(opus_data)
# 判断是否保存为WAV文件 combined_pcm_data = b"".join(pcm_data)
if self.delete_audio_file:
pass
else:
file_path = self.save_audio_to_file(pcm_data, session_id)
# 语音识别 # 检查磁盘空间
start_time = time.time() if not self.delete_audio_file:
result = self.model.generate( free_space = shutil.disk_usage(self.output_dir).free
input=combined_pcm_data, if free_space < len(combined_pcm_data) * 2: # 预留2倍空间
cache={}, raise OSError("磁盘空间不足")
language="auto",
use_itn=True,
batch_size_s=60,
)
text = rich_transcription_postprocess(result[0]["text"])
logger.bind(tag=TAG).debug(
f"语音识别耗时: {time.time() - start_time:.3f}s | 结果: {text}"
)
return text, file_path # 判断是否保存为WAV文件
if self.delete_audio_file:
pass
else:
file_path = self.save_audio_to_file(pcm_data, session_id)
except Exception as e: # 语音识别
logger.bind(tag=TAG).error(f"语音识别失败: {e}", exc_info=True) start_time = time.time()
return "", file_path result = self.model.generate(
input=combined_pcm_data,
cache={},
language="auto",
use_itn=True,
batch_size_s=60,
)
text = rich_transcription_postprocess(result[0]["text"])
logger.bind(tag=TAG).debug(
f"语音识别耗时: {time.time() - start_time:.3f}s | 结果: {text}"
)
# finally: return text, file_path
# # 文件清理逻辑
# if self.delete_audio_file and file_path and os.path.exists(file_path): except OSError as e:
# try: retry_count += 1
# os.remove(file_path) if retry_count >= MAX_RETRIES:
# logger.bind(tag=TAG).debug(f"已删除临时音频文件: {file_path}") logger.bind(tag=TAG).error(
# except Exception as e: f"语音识别失败(已重试{retry_count}次): {e}", exc_info=True
# logger.bind(tag=TAG).error(f"文件删除失败: {file_path} | 错误: {e}") )
return "", file_path
logger.bind(tag=TAG).warning(
f"语音识别失败,正在重试({retry_count}/{MAX_RETRIES}: {e}"
)
time.sleep(RETRY_DELAY)
except Exception as e:
logger.bind(tag=TAG).error(f"语音识别失败: {e}", exc_info=True)
return "", file_path
finally:
# 文件清理逻辑
if self.delete_audio_file and file_path and os.path.exists(file_path):
try:
os.remove(file_path)
logger.bind(tag=TAG).debug(f"已删除临时音频文件: {file_path}")
except Exception as e:
logger.bind(tag=TAG).error(
f"文件删除失败: {file_path} | 错误: {e}"
)
@@ -1,11 +1,8 @@
from typing import Optional, Tuple, List from typing import Optional, Tuple, List
import opuslib_next
from core.providers.asr.base import ASRProviderBase from core.providers.asr.base import ASRProviderBase
import os from core.providers.asr.dto.dto import InterfaceType
import ssl import ssl
import json import json
import uuid
import wave
import websockets import websockets
from config.logger import setup_logging from config.logger import setup_logging
import asyncio import asyncio
@@ -23,6 +20,7 @@ class ASRProvider(ASRProviderBase):
:param delete_audio_file: Boolean to indicate whether to delete audio files after processing. :param delete_audio_file: Boolean to indicate whether to delete audio files after processing.
""" """
super().__init__() super().__init__()
self.interface_type = InterfaceType.NON_STREAM
self.host = config.get("host", "localhost") self.host = config.get("host", "localhost")
self.port = config.get("port", 10095) self.port = config.get("port", 10095)
self.api_key = config.get("api_key", "none") self.api_key = config.get("api_key", "none")
@@ -43,20 +41,6 @@ class ASRProvider(ASRProviderBase):
self.ssl_context.check_hostname = False self.ssl_context.check_hostname = False
self.ssl_context.verify_mode = ssl.CERT_NONE self.ssl_context.verify_mode = ssl.CERT_NONE
def save_audio_to_file(self, pcm_data: List[bytes], session_id: str) -> str:
"""PCM数据保存为WAV文件"""
module_name = __name__.split(".")[-1]
file_name = f"asr_{module_name}_{session_id}_{uuid.uuid4()}.wav"
file_path = os.path.join(self.output_dir, file_name)
with wave.open(file_path, "wb") as wf:
wf.setnchannels(1)
wf.setsampwidth(2) # 2 bytes = 16-bit
wf.setframerate(16000)
wf.writeframes(b"".join(pcm_data))
return file_path
async def _receive_responses(self, ws) -> None: async def _receive_responses(self, ws) -> None:
""" """
Asynchronous generator to receive messages from the WebSocket. Asynchronous generator to receive messages from the WebSocket.
@@ -5,8 +5,7 @@ import sys
import io import io
from config.logger import setup_logging from config.logger import setup_logging
from typing import Optional, Tuple, List from typing import Optional, Tuple, List
import uuid from core.providers.asr.dto.dto import InterfaceType
import opuslib_next
from core.providers.asr.base import ASRProviderBase from core.providers.asr.base import ASRProviderBase
import numpy as np import numpy as np
@@ -38,6 +37,7 @@ class CaptureOutput:
class ASRProvider(ASRProviderBase): class ASRProvider(ASRProviderBase):
def __init__(self, config: dict, delete_audio_file: bool): def __init__(self, config: dict, delete_audio_file: bool):
super().__init__() super().__init__()
self.interface_type = InterfaceType.LOCAL
self.model_dir = config.get("model_dir") self.model_dir = config.get("model_dir")
self.output_dir = config.get("output_dir") self.output_dir = config.get("output_dir")
self.delete_audio_file = delete_audio_file self.delete_audio_file = delete_audio_file
@@ -84,20 +84,6 @@ class ASRProvider(ASRProviderBase):
use_itn=True, use_itn=True,
) )
def save_audio_to_file(self, pcm_data: List[bytes], session_id: str) -> str:
"""PCM数据保存为WAV文件"""
module_name = __name__.split(".")[-1]
file_name = f"asr_{module_name}_{session_id}_{uuid.uuid4()}.wav"
file_path = os.path.join(self.output_dir, file_name)
with wave.open(file_path, "wb") as wf:
wf.setnchannels(1)
wf.setsampwidth(2) # 2 bytes = 16-bit
wf.setframerate(16000)
wf.writeframes(b"".join(pcm_data))
return file_path
def read_wave(self, wave_filename: str) -> Tuple[np.ndarray, int]: def read_wave(self, wave_filename: str) -> Tuple[np.ndarray, int]:
""" """
Args: Args:
@@ -5,11 +5,8 @@ import json
import time import time
from datetime import datetime, timezone from datetime import datetime, timezone
import os import os
import uuid
from typing import Optional, Tuple, List from typing import Optional, Tuple, List
import wave from core.providers.asr.dto.dto import InterfaceType
import opuslib_next
import requests import requests
from core.providers.asr.base import ASRProviderBase from core.providers.asr.base import ASRProviderBase
from config.logger import setup_logging from config.logger import setup_logging
@@ -25,6 +22,7 @@ class ASRProvider(ASRProviderBase):
def __init__(self, config: dict, delete_audio_file: bool = True): def __init__(self, config: dict, delete_audio_file: bool = True):
super().__init__() super().__init__()
self.interface_type = InterfaceType.NON_STREAM
self.secret_id = config.get("secret_id") self.secret_id = config.get("secret_id")
self.secret_key = config.get("secret_key") self.secret_key = config.get("secret_key")
self.output_dir = config.get("output_dir") self.output_dir = config.get("output_dir")
@@ -33,20 +31,6 @@ class ASRProvider(ASRProviderBase):
# 确保输出目录存在 # 确保输出目录存在
os.makedirs(self.output_dir, exist_ok=True) os.makedirs(self.output_dir, exist_ok=True)
def save_audio_to_file(self, pcm_data: List[bytes], session_id: str) -> str:
"""PCM数据保存为WAV文件"""
module_name = __name__.split(".")[-1]
file_name = f"asr_{module_name}_{session_id}_{uuid.uuid4()}.wav"
file_path = os.path.join(self.output_dir, file_name)
with wave.open(file_path, "wb") as wf:
wf.setnchannels(1)
wf.setsampwidth(2) # 2 bytes = 16-bit
wf.setframerate(16000)
wf.writeframes(b"".join(pcm_data))
return file_path
async def speech_to_text( async def speech_to_text(
self, opus_data: List[bytes], session_id: str self, opus_data: List[bytes], session_id: str
) -> Tuple[Optional[str], Optional[str]]: ) -> Tuple[Optional[str], Optional[str]]:
@@ -72,6 +72,10 @@ class IntentProvider(IntentProviderBase):
'返回: {"function_call": {"name": "get_time"}}\n' '返回: {"function_call": {"name": "get_time"}}\n'
"```\n" "```\n"
"```\n" "```\n"
"用户: 当前电池电量是多少?\n"
'返回: {"function_call": {"name": "get_battery_level", "arguments": {"response_success": "当前电池电量为{value}%", "response_failure": "无法获取Battery的当前电量百分比"}}}\n'
"```\n"
"```\n"
"用户: 我想结束对话\n" "用户: 我想结束对话\n"
'返回: {"function_call": {"name": "handle_exit_intent", "arguments": {"say_goodbye": "goodbye"}}}\n' '返回: {"function_call": {"name": "handle_exit_intent", "arguments": {"say_goodbye": "goodbye"}}}\n'
"```\n" "```\n"
@@ -224,7 +228,8 @@ class IntentProvider(IntentProviderBase):
if function_name == "continue_chat": if function_name == "continue_chat":
# 保留非工具相关的消息 # 保留非工具相关的消息
clean_history = [ clean_history = [
msg for msg in conn.dialogue.dialogue msg
for msg in conn.dialogue.dialogue
if msg.role not in ["tool", "function"] if msg.role not in ["tool", "function"]
] ]
conn.dialogue.dialogue = clean_history conn.dialogue.dialogue = clean_history
@@ -10,7 +10,7 @@ class LLMProviderBase(ABC):
"""LLM response generator""" """LLM response generator"""
pass pass
def response_no_stream(self, system_prompt, user_prompt): def response_no_stream(self, system_prompt, user_prompt, **kwargs):
try: try:
# 构造对话格式 # 构造对话格式
dialogue = [ dialogue = [
@@ -18,7 +18,7 @@ class LLMProviderBase(ABC):
{"role": "user", "content": user_prompt} {"role": "user", "content": user_prompt}
] ]
result = "" result = ""
for part in self.response("", dialogue): for part in self.response("", dialogue, **kwargs):
result += part result += part
return result return result
@@ -30,7 +30,7 @@ class LLMProviderBase(ABC):
""" """
Default implementation for function calling (streaming) Default implementation for function calling (streaming)
This should be overridden by providers that support function calls This should be overridden by providers that support function calls
Returns: generator that yields either text tokens or a special function call token Returns: generator that yields either text tokens or a special function call token
""" """
# For providers that don't support functions, just return regular response # For providers that don't support functions, just return regular response
@@ -25,7 +25,7 @@ class LLMProvider(LLMProviderBase):
self.session_conversation_map = {} # 存储session_id和conversation_id的映射 self.session_conversation_map = {} # 存储session_id和conversation_id的映射
check_model_key("CozeLLM", self.personal_access_token) check_model_key("CozeLLM", self.personal_access_token)
def response(self, session_id, dialogue): def response(self, session_id, dialogue, **kwargs):
coze_api_token = self.personal_access_token coze_api_token = self.personal_access_token
coze_api_base = COZE_CN_BASE_URL coze_api_base = COZE_CN_BASE_URL
@@ -17,7 +17,7 @@ class LLMProvider(LLMProviderBase):
self.session_conversation_map = {} # 存储session_id和conversation_id的映射 self.session_conversation_map = {} # 存储session_id和conversation_id的映射
check_model_key("DifyLLM", self.api_key) check_model_key("DifyLLM", self.api_key)
def response(self, session_id, dialogue): def response(self, session_id, dialogue, **kwargs):
try: try:
# 取最后一条用户消息 # 取最后一条用户消息
last_msg = next(m for m in reversed(dialogue) if m["role"] == "user") last_msg = next(m for m in reversed(dialogue) if m["role"] == "user")
@@ -16,7 +16,7 @@ class LLMProvider(LLMProviderBase):
self.variables = config.get("variables", {}) self.variables = config.get("variables", {})
check_model_key("FastGPTLLM", self.api_key) check_model_key("FastGPTLLM", self.api_key)
def response(self, session_id, dialogue): def response(self, session_id, dialogue, **kwargs):
try: try:
# 取最后一条用户消息 # 取最后一条用户消息
last_msg = next(m for m in reversed(dialogue) if m["role"] == "user") last_msg = next(m for m in reversed(dialogue) if m["role"] == "user")
@@ -112,7 +112,7 @@ class LLMProvider(LLMProviderBase):
] ]
# Gemini文档提到,无需维护session-id,直接用dialogue拼接而成 # Gemini文档提到,无需维护session-id,直接用dialogue拼接而成
def response(self, session_id, dialogue): def response(self, session_id, dialogue, **kwargs):
yield from self._generate(dialogue, None) yield from self._generate(dialogue, None)
def response_with_functions(self, session_id, dialogue, functions=None): def response_with_functions(self, session_id, dialogue, functions=None):
@@ -14,7 +14,7 @@ class LLMProvider(LLMProviderBase):
self.base_url = config.get("base_url", config.get("url")) # 默认使用 base_url self.base_url = config.get("base_url", config.get("url")) # 默认使用 base_url
self.api_url = f"{self.base_url}/api/conversation/process" # 拼接完整的 API URL self.api_url = f"{self.base_url}/api/conversation/process" # 拼接完整的 API URL
def response(self, session_id, dialogue): def response(self, session_id, dialogue, **kwargs):
try: try:
# home assistant语音助手自带意图,无需使用xiaozhi ai自带的,只需要把用户说的话传递给home assistant即可 # home assistant语音助手自带意图,无需使用xiaozhi ai自带的,只需要把用户说的话传递给home assistant即可
@@ -18,13 +18,13 @@ class LLMProvider(LLMProviderBase):
self.client = OpenAI( self.client = OpenAI(
base_url=self.base_url, base_url=self.base_url,
api_key="ollama" # Ollama doesn't need an API key but OpenAI client requires one api_key="ollama", # Ollama doesn't need an API key but OpenAI client requires one
) )
# 检查是否是qwen3模型 # 检查是否是qwen3模型
self.is_qwen3 = self.model_name and self.model_name.lower().startswith("qwen3") self.is_qwen3 = self.model_name and self.model_name.lower().startswith("qwen3")
def response(self, session_id, dialogue): def response(self, session_id, dialogue, **kwargs):
try: try:
# 如果是qwen3模型,在用户最后一条消息中添加/no_think指令 # 如果是qwen3模型,在用户最后一条消息中添加/no_think指令
if self.is_qwen3: if self.is_qwen3:
@@ -35,7 +35,9 @@ class LLMProvider(LLMProviderBase):
for i in range(len(dialogue_copy) - 1, -1, -1): for i in range(len(dialogue_copy) - 1, -1, -1):
if dialogue_copy[i]["role"] == "user": if dialogue_copy[i]["role"] == "user":
# 在用户消息前添加/no_think指令 # 在用户消息前添加/no_think指令
dialogue_copy[i]["content"] = "/no_think " + dialogue_copy[i]["content"] dialogue_copy[i]["content"] = (
"/no_think " + dialogue_copy[i]["content"]
)
logger.bind(tag=TAG).debug(f"为qwen3模型添加/no_think指令") logger.bind(tag=TAG).debug(f"为qwen3模型添加/no_think指令")
break break
@@ -43,9 +45,7 @@ class LLMProvider(LLMProviderBase):
dialogue = dialogue_copy dialogue = dialogue_copy
responses = self.client.chat.completions.create( responses = self.client.chat.completions.create(
model=self.model_name, model=self.model_name, messages=dialogue, stream=True
messages=dialogue,
stream=True
) )
is_active = True is_active = True
# 用于处理跨chunk的标签 # 用于处理跨chunk的标签
@@ -53,29 +53,33 @@ class LLMProvider(LLMProviderBase):
for chunk in responses: for chunk in responses:
try: try:
delta = chunk.choices[0].delta if getattr(chunk, 'choices', None) else None delta = (
content = delta.content if hasattr(delta, 'content') else '' chunk.choices[0].delta
if getattr(chunk, "choices", None)
else None
)
content = delta.content if hasattr(delta, "content") else ""
if content: if content:
# 将内容添加到缓冲区 # 将内容添加到缓冲区
buffer += content buffer += content
# 处理缓冲区中的标签 # 处理缓冲区中的标签
while '<think>' in buffer and '</think>' in buffer: while "<think>" in buffer and "</think>" in buffer:
# 找到完整的<think></think>标签并移除 # 找到完整的<think></think>标签并移除
pre = buffer.split('<think>', 1)[0] pre = buffer.split("<think>", 1)[0]
post = buffer.split('</think>', 1)[1] post = buffer.split("</think>", 1)[1]
buffer = pre + post buffer = pre + post
# 处理只有开始标签的情况 # 处理只有开始标签的情况
if '<think>' in buffer: if "<think>" in buffer:
is_active = False is_active = False
buffer = buffer.split('<think>', 1)[0] buffer = buffer.split("<think>", 1)[0]
# 处理只有结束标签的情况 # 处理只有结束标签的情况
if '</think>' in buffer: if "</think>" in buffer:
is_active = True is_active = True
buffer = buffer.split('</think>', 1)[1] buffer = buffer.split("</think>", 1)[1]
# 如果当前处于活动状态且缓冲区有内容,则输出 # 如果当前处于活动状态且缓冲区有内容,则输出
if is_active and buffer: if is_active and buffer:
@@ -100,7 +104,9 @@ class LLMProvider(LLMProviderBase):
for i in range(len(dialogue_copy) - 1, -1, -1): for i in range(len(dialogue_copy) - 1, -1, -1):
if dialogue_copy[i]["role"] == "user": if dialogue_copy[i]["role"] == "user":
# 在用户消息前添加/no_think指令 # 在用户消息前添加/no_think指令
dialogue_copy[i]["content"] = "/no_think " + dialogue_copy[i]["content"] dialogue_copy[i]["content"] = (
"/no_think " + dialogue_copy[i]["content"]
)
logger.bind(tag=TAG).debug(f"为qwen3模型添加/no_think指令") logger.bind(tag=TAG).debug(f"为qwen3模型添加/no_think指令")
break break
@@ -119,9 +125,15 @@ class LLMProvider(LLMProviderBase):
for chunk in stream: for chunk in stream:
try: try:
delta = chunk.choices[0].delta if getattr(chunk, 'choices', None) else None delta = (
content = delta.content if hasattr(delta, 'content') else None chunk.choices[0].delta
tool_calls = delta.tool_calls if hasattr(delta, 'tool_calls') else None if getattr(chunk, "choices", None)
else None
)
content = delta.content if hasattr(delta, "content") else None
tool_calls = (
delta.tool_calls if hasattr(delta, "tool_calls") else None
)
# 如果是工具调用,直接传递 # 如果是工具调用,直接传递
if tool_calls: if tool_calls:
@@ -134,21 +146,21 @@ class LLMProvider(LLMProviderBase):
buffer += content buffer += content
# 处理缓冲区中的标签 # 处理缓冲区中的标签
while '<think>' in buffer and '</think>' in buffer: while "<think>" in buffer and "</think>" in buffer:
# 找到完整的<think></think>标签并移除 # 找到完整的<think></think>标签并移除
pre = buffer.split('<think>', 1)[0] pre = buffer.split("<think>", 1)[0]
post = buffer.split('</think>', 1)[1] post = buffer.split("</think>", 1)[1]
buffer = pre + post buffer = pre + post
# 处理只有开始标签的情况 # 处理只有开始标签的情况
if '<think>' in buffer: if "<think>" in buffer:
is_active = False is_active = False
buffer = buffer.split('<think>', 1)[0] buffer = buffer.split("<think>", 1)[0]
# 处理只有结束标签的情况 # 处理只有结束标签的情况
if '</think>' in buffer: if "</think>" in buffer:
is_active = True is_active = True
buffer = buffer.split('</think>', 1)[1] buffer = buffer.split("</think>", 1)[1]
# 如果当前处于活动状态且缓冲区有内容,则输出 # 如果当前处于活动状态且缓冲区有内容,则输出
if is_active and buffer: if is_active and buffer:
@@ -16,26 +16,37 @@ class LLMProvider(LLMProviderBase):
self.base_url = config.get("base_url") self.base_url = config.get("base_url")
else: else:
self.base_url = config.get("url") self.base_url = config.get("url")
max_tokens = config.get("max_tokens")
if max_tokens is None or max_tokens == "":
max_tokens = 500
try: param_defaults = {
max_tokens = int(max_tokens) "max_tokens": (500, int),
except (ValueError, TypeError): "temperature": (0.7, lambda x: round(float(x), 1)),
max_tokens = 500 "top_p": (1.0, lambda x: round(float(x), 1)),
self.max_tokens = max_tokens "frequency_penalty": (0, lambda x: round(float(x), 1))
}
for param, (default, converter) in param_defaults.items():
value = config.get(param)
try:
setattr(self, param, converter(value) if value not in (None, "") else default)
except (ValueError, TypeError):
setattr(self, param, default)
logger.debug(
f"意图识别参数初始化: {self.temperature}, {self.max_tokens}, {self.top_p}, {self.frequency_penalty}")
check_model_key("LLM", self.api_key) check_model_key("LLM", self.api_key)
self.client = openai.OpenAI(api_key=self.api_key, base_url=self.base_url) self.client = openai.OpenAI(api_key=self.api_key, base_url=self.base_url)
def response(self, session_id, dialogue): def response(self, session_id, dialogue, **kwargs):
try: try:
responses = self.client.chat.completions.create( responses = self.client.chat.completions.create(
model=self.model_name, model=self.model_name,
messages=dialogue, messages=dialogue,
stream=True, stream=True,
max_tokens=self.max_tokens, max_tokens=kwargs.get("max_tokens", self.max_tokens),
temperature=kwargs.get("temperature", self.temperature),
top_p=kwargs.get("top_p", self.top_p),
frequency_penalty=kwargs.get("frequency_penalty", self.frequency_penalty),
) )
is_active = True is_active = True
@@ -15,39 +15,45 @@ class LLMProvider(LLMProviderBase):
# 如果没有v1,增加v1 # 如果没有v1,增加v1
if not self.base_url.endswith("/v1"): if not self.base_url.endswith("/v1"):
self.base_url = f"{self.base_url}/v1" self.base_url = f"{self.base_url}/v1"
logger.bind(tag=TAG).info(f"Initializing Xinference LLM provider with model: {self.model_name}, base_url: {self.base_url}") logger.bind(tag=TAG).info(
f"Initializing Xinference LLM provider with model: {self.model_name}, base_url: {self.base_url}"
)
try: try:
self.client = OpenAI( self.client = OpenAI(
base_url=self.base_url, base_url=self.base_url,
api_key="xinference" # Xinference has a similar setup to Ollama where it doesn't need an actual key api_key="xinference", # Xinference has a similar setup to Ollama where it doesn't need an actual key
) )
logger.bind(tag=TAG).info("Xinference client initialized successfully") logger.bind(tag=TAG).info("Xinference client initialized successfully")
except Exception as e: except Exception as e:
logger.bind(tag=TAG).error(f"Error initializing Xinference client: {e}") logger.bind(tag=TAG).error(f"Error initializing Xinference client: {e}")
raise raise
def response(self, session_id, dialogue): def response(self, session_id, dialogue, **kwargs):
try: try:
logger.bind(tag=TAG).debug(f"Sending request to Xinference with model: {self.model_name}, dialogue length: {len(dialogue)}") logger.bind(tag=TAG).debug(
responses = self.client.chat.completions.create( f"Sending request to Xinference with model: {self.model_name}, dialogue length: {len(dialogue)}"
model=self.model_name,
messages=dialogue,
stream=True
) )
is_active=True responses = self.client.chat.completions.create(
model=self.model_name, messages=dialogue, stream=True
)
is_active = True
for chunk in responses: for chunk in responses:
try: try:
delta = chunk.choices[0].delta if getattr(chunk, 'choices', None) else None delta = (
content = delta.content if hasattr(delta, 'content') else '' chunk.choices[0].delta
if getattr(chunk, "choices", None)
else None
)
content = delta.content if hasattr(delta, "content") else ""
if content: if content:
if '<think>' in content: if "<think>" in content:
is_active = False is_active = False
content = content.split('<think>')[0] content = content.split("<think>")[0]
if '</think>' in content: if "</think>" in content:
is_active = True is_active = True
content = content.split('</think>')[-1] content = content.split("</think>")[-1]
if is_active: if is_active:
yield content yield content
except Exception as e: except Exception as e:
@@ -59,10 +65,14 @@ class LLMProvider(LLMProviderBase):
def response_with_functions(self, session_id, dialogue, functions=None): def response_with_functions(self, session_id, dialogue, functions=None):
try: try:
logger.bind(tag=TAG).debug(f"Sending function call request to Xinference with model: {self.model_name}, dialogue length: {len(dialogue)}") logger.bind(tag=TAG).debug(
f"Sending function call request to Xinference with model: {self.model_name}, dialogue length: {len(dialogue)}"
)
if functions: if functions:
logger.bind(tag=TAG).debug(f"Function calls enabled with: {[f.get('function', {}).get('name') for f in functions]}") logger.bind(tag=TAG).debug(
f"Function calls enabled with: {[f.get('function', {}).get('name') for f in functions]}"
)
stream = self.client.chat.completions.create( stream = self.client.chat.completions.create(
model=self.model_name, model=self.model_name,
messages=dialogue, messages=dialogue,
@@ -74,7 +84,7 @@ class LLMProvider(LLMProviderBase):
delta = chunk.choices[0].delta delta = chunk.choices[0].delta
content = delta.content content = delta.content
tool_calls = delta.tool_calls tool_calls = delta.tool_calls
if content: if content:
yield content, tool_calls yield content, tool_calls
elif tool_calls: elif tool_calls:
@@ -82,4 +92,7 @@ class LLMProvider(LLMProviderBase):
except Exception as e: except Exception as e:
logger.bind(tag=TAG).error(f"Error in Xinference function call: {e}") logger.bind(tag=TAG).error(f"Error in Xinference function call: {e}")
yield {"type": "content", "content": f"【Xinference服务响应异常: {str(e)}"} yield {
"type": "content",
"content": f"【Xinference服务响应异常: {str(e)}",
}
@@ -9,7 +9,13 @@ class MemoryProviderBase(ABC):
def __init__(self, config): def __init__(self, config):
self.config = config self.config = config
self.role_id = None self.role_id = None
self.llm = None
def set_llm(self, llm):
self.llm = llm
# 获取模型名称和类型信息
model_name = getattr(llm, "model_name", str(llm.__class__.__name__))
# 记录更详细的日志
logger.bind(tag=TAG).info(f"记忆总结设置LLM: {model_name}")
@abstractmethod @abstractmethod
async def save_memory(self, msgs): async def save_memory(self, msgs):
@@ -107,7 +107,7 @@ TAG = __name__
class MemoryProvider(MemoryProviderBase): class MemoryProvider(MemoryProviderBase):
def __init__(self, config, summary_memory): def __init__(self, config, summary_memory):
super().__init__(config) super().__init__(config)
self.short_momery = "" self.short_memory = ""
self.save_to_file = True self.save_to_file = True
self.memory_path = get_project_dir() + "data/.memory.yaml" self.memory_path = get_project_dir() + "data/.memory.yaml"
self.load_memory(summary_memory) self.load_memory(summary_memory)
@@ -122,7 +122,7 @@ class MemoryProvider(MemoryProviderBase):
def load_memory(self, summary_memory): def load_memory(self, summary_memory):
# api获取到总结记忆后直接返回 # api获取到总结记忆后直接返回
if summary_memory or not self.save_to_file: if summary_memory or not self.save_to_file:
self.short_momery = summary_memory self.short_memory = summary_memory
return return
all_memory = {} all_memory = {}
@@ -130,18 +130,21 @@ class MemoryProvider(MemoryProviderBase):
with open(self.memory_path, "r", encoding="utf-8") as f: with open(self.memory_path, "r", encoding="utf-8") as f:
all_memory = yaml.safe_load(f) or {} all_memory = yaml.safe_load(f) or {}
if self.role_id in all_memory: if self.role_id in all_memory:
self.short_momery = all_memory[self.role_id] self.short_memory = all_memory[self.role_id]
def save_memory_to_file(self): def save_memory_to_file(self):
all_memory = {} all_memory = {}
if os.path.exists(self.memory_path): if os.path.exists(self.memory_path):
with open(self.memory_path, "r", encoding="utf-8") as f: with open(self.memory_path, "r", encoding="utf-8") as f:
all_memory = yaml.safe_load(f) or {} all_memory = yaml.safe_load(f) or {}
all_memory[self.role_id] = self.short_momery all_memory[self.role_id] = self.short_memory
with open(self.memory_path, "w", encoding="utf-8") as f: with open(self.memory_path, "w", encoding="utf-8") as f:
yaml.dump(all_memory, f, allow_unicode=True) yaml.dump(all_memory, f, allow_unicode=True)
async def save_memory(self, msgs): async def save_memory(self, msgs):
# 打印使用的模型信息
model_info = getattr(self.llm, "model_name", str(self.llm.__class__.__name__))
logger.bind(tag=TAG).debug(f"使用记忆保存模型: {model_info}")
if self.llm is None: if self.llm is None:
logger.bind(tag=TAG).error("LLM is not set for memory provider") logger.bind(tag=TAG).error("LLM is not set for memory provider")
return None return None
@@ -155,31 +158,39 @@ class MemoryProvider(MemoryProviderBase):
msgStr += f"User: {msg.content}\n" msgStr += f"User: {msg.content}\n"
elif msg.role == "assistant": elif msg.role == "assistant":
msgStr += f"Assistant: {msg.content}\n" msgStr += f"Assistant: {msg.content}\n"
if self.short_momery and len(self.short_momery) > 0: if self.short_memory and len(self.short_memory) > 0:
msgStr += "历史记忆:\n" msgStr += "历史记忆:\n"
msgStr += self.short_momery msgStr += self.short_memory
# 当前时间 # 当前时间
time_str = time.strftime("%Y-%m-%d %H:%M:%S", time.localtime()) time_str = time.strftime("%Y-%m-%d %H:%M:%S", time.localtime())
msgStr += f"当前时间:{time_str}" msgStr += f"当前时间:{time_str}"
if self.save_to_file: if self.save_to_file:
result = self.llm.response_no_stream(short_term_memory_prompt, msgStr) result = self.llm.response_no_stream(
short_term_memory_prompt,
msgStr,
max_tokens=2000,
temperature=0.2,
)
json_str = extract_json_data(result) json_str = extract_json_data(result)
try: try:
json.loads(json_str) # 检查json格式是否正确 json.loads(json_str) # 检查json格式是否正确
self.short_momery = json_str self.short_memory = json_str
self.save_memory_to_file() self.save_memory_to_file()
except Exception as e: except Exception as e:
print("Error:", e) print("Error:", e)
else: else:
result = self.llm.response_no_stream( result = self.llm.response_no_stream(
short_term_memory_prompt_only_content, msgStr short_term_memory_prompt_only_content,
msgStr,
max_tokens=2000,
temperature=0.2,
) )
save_mem_local_short(self.role_id, result) save_mem_local_short(self.role_id, result)
logger.bind(tag=TAG).info(f"Save memory successful - Role: {self.role_id}") logger.bind(tag=TAG).info(f"Save memory successful - Role: {self.role_id}")
return self.short_momery return self.short_memory
async def query_memory(self, query: str) -> str: async def query_memory(self, query: str) -> str:
return self.short_momery return self.short_memory
@@ -1,4 +1,3 @@
import os
import uuid import uuid
import json import json
import hmac import hmac
@@ -8,61 +7,74 @@ import requests
from datetime import datetime from datetime import datetime
from core.providers.tts.base import TTSProviderBase from core.providers.tts.base import TTSProviderBase
import http.client
import urllib.parse
import time import time
import uuid import uuid
from urllib import parse from urllib import parse
class AccessToken: class AccessToken:
@staticmethod @staticmethod
def _encode_text(text): def _encode_text(text):
encoded_text = parse.quote_plus(text) encoded_text = parse.quote_plus(text)
return encoded_text.replace('+', '%20').replace('*', '%2A').replace('%7E', '~') return encoded_text.replace("+", "%20").replace("*", "%2A").replace("%7E", "~")
@staticmethod @staticmethod
def _encode_dict(dic): def _encode_dict(dic):
keys = dic.keys() keys = dic.keys()
dic_sorted = [(key, dic[key]) for key in sorted(keys)] dic_sorted = [(key, dic[key]) for key in sorted(keys)]
encoded_text = parse.urlencode(dic_sorted) encoded_text = parse.urlencode(dic_sorted)
return encoded_text.replace('+', '%20').replace('*', '%2A').replace('%7E', '~') return encoded_text.replace("+", "%20").replace("*", "%2A").replace("%7E", "~")
@staticmethod @staticmethod
def create_token(access_key_id, access_key_secret): def create_token(access_key_id, access_key_secret):
parameters = {'AccessKeyId': access_key_id, parameters = {
'Action': 'CreateToken', "AccessKeyId": access_key_id,
'Format': 'JSON', "Action": "CreateToken",
'RegionId': 'cn-shanghai', "Format": "JSON",
'SignatureMethod': 'HMAC-SHA1', "RegionId": "cn-shanghai",
'SignatureNonce': str(uuid.uuid1()), "SignatureMethod": "HMAC-SHA1",
'SignatureVersion': '1.0', "SignatureNonce": str(uuid.uuid1()),
'Timestamp': time.strftime("%Y-%m-%dT%H:%M:%SZ", time.gmtime()), "SignatureVersion": "1.0",
'Version': '2019-02-28'} "Timestamp": time.strftime("%Y-%m-%dT%H:%M:%SZ", time.gmtime()),
"Version": "2019-02-28",
}
# 构造规范化的请求字符串 # 构造规范化的请求字符串
query_string = AccessToken._encode_dict(parameters) query_string = AccessToken._encode_dict(parameters)
# print('规范化的请求字符串: %s' % query_string) # print('规范化的请求字符串: %s' % query_string)
# 构造待签名字符串 # 构造待签名字符串
string_to_sign = 'GET' + '&' + AccessToken._encode_text('/') + '&' + AccessToken._encode_text(query_string) string_to_sign = (
"GET"
+ "&"
+ AccessToken._encode_text("/")
+ "&"
+ AccessToken._encode_text(query_string)
)
# print('待签名的字符串: %s' % string_to_sign) # print('待签名的字符串: %s' % string_to_sign)
# 计算签名 # 计算签名
secreted_string = hmac.new(bytes(access_key_secret + '&', encoding='utf-8'), secreted_string = hmac.new(
bytes(string_to_sign, encoding='utf-8'), bytes(access_key_secret + "&", encoding="utf-8"),
hashlib.sha1).digest() bytes(string_to_sign, encoding="utf-8"),
hashlib.sha1,
).digest()
signature = base64.b64encode(secreted_string) signature = base64.b64encode(secreted_string)
# print('签名: %s' % signature) # print('签名: %s' % signature)
# 进行URL编码 # 进行URL编码
signature = AccessToken._encode_text(signature) signature = AccessToken._encode_text(signature)
# print('URL编码后的签名: %s' % signature) # print('URL编码后的签名: %s' % signature)
# 调用服务 # 调用服务
full_url = 'http://nls-meta.cn-shanghai.aliyuncs.com/?Signature=%s&%s' % (signature, query_string) full_url = "http://nls-meta.cn-shanghai.aliyuncs.com/?Signature=%s&%s" % (
signature,
query_string,
)
# print('url: %s' % full_url) # print('url: %s' % full_url)
# 提交HTTP GET请求 # 提交HTTP GET请求
response = requests.get(full_url) response = requests.get(full_url)
if response.ok: if response.ok:
root_obj = response.json() root_obj = response.json()
key = 'Token' key = "Token"
if key in root_obj: if key in root_obj:
token = root_obj[key]['Id'] token = root_obj[key]["Id"]
expire_time = root_obj[key]['ExpireTime'] expire_time = root_obj[key]["ExpireTime"]
return token, expire_time return token, expire_time
# print(response.text) # print(response.text)
return None, None return None, None
@@ -70,26 +82,36 @@ class AccessToken:
class TTSProvider(TTSProviderBase): class TTSProvider(TTSProviderBase):
def __init__(self, config, delete_audio_file): def __init__(self, config, delete_audio_file):
super().__init__(config, delete_audio_file) super().__init__(config, delete_audio_file)
# 新增空值判断逻辑 # 新增空值判断逻辑
self.access_key_id = config.get("access_key_id") self.access_key_id = config.get("access_key_id")
self.access_key_secret = config.get("access_key_secret") self.access_key_secret = config.get("access_key_secret")
self.appkey = config.get("appkey") self.appkey = config.get("appkey")
self.format = config.get("format", "wav") self.format = config.get("format", "wav")
self.sample_rate = config.get("sample_rate", 16000)
self.voice = config.get("voice", "xiaoyun") sample_rate = config.get("sample_rate", "16000")
self.volume = config.get("volume", 50) self.sample_rate = int(sample_rate) if sample_rate else 16000
self.speech_rate = config.get("speech_rate", 0)
self.pitch_rate = config.get("pitch_rate", 0) if config.get("private_voice"):
self.voice = config.get("private_voice")
else:
self.voice = config.get("voice", "xiaoyun")
volume = config.get("volume", "50")
self.volume = int(volume) if volume else 50
speech_rate = config.get("speech_rate", "0")
self.speech_rate = int(speech_rate) if speech_rate else 0
pitch_rate = config.get("pitch_rate", "0")
self.pitch_rate = int(pitch_rate) if pitch_rate else 0
self.host = config.get("host", "nls-gateway-cn-shanghai.aliyuncs.com") self.host = config.get("host", "nls-gateway-cn-shanghai.aliyuncs.com")
self.api_url = f"https://{self.host}/stream/v1/tts" self.api_url = f"https://{self.host}/stream/v1/tts"
self.header = { self.header = {"Content-Type": "application/json"}
"Content-Type": "application/json"
}
if self.access_key_id and self.access_key_secret: if self.access_key_id and self.access_key_secret:
# 使用密钥对生成临时token # 使用密钥对生成临时token
@@ -99,35 +121,30 @@ class TTSProvider(TTSProviderBase):
self.token = config.get("token") self.token = config.get("token")
self.expire_time = None self.expire_time = None
def _refresh_token(self): def _refresh_token(self):
"""刷新Token并记录过期时间""" """刷新Token并记录过期时间"""
if self.access_key_id and self.access_key_secret: if self.access_key_id and self.access_key_secret:
self.token, expire_time_str = AccessToken.create_token( self.token, expire_time_str = AccessToken.create_token(
self.access_key_id, self.access_key_id, self.access_key_secret
self.access_key_secret
) )
if not expire_time_str: if not expire_time_str:
raise ValueError("无法获取有效的Token过期时间") raise ValueError("无法获取有效的Token过期时间")
try: try:
#统一转换为字符串处理 # 统一转换为字符串处理
expire_str = str(expire_time_str).strip() expire_str = str(expire_time_str).strip()
if expire_str.isdigit(): if expire_str.isdigit():
expire_time = datetime.fromtimestamp(int(expire_str)) expire_time = datetime.fromtimestamp(int(expire_str))
else: else:
expire_time = datetime.strptime( expire_time = datetime.strptime(expire_str, "%Y-%m-%dT%H:%M:%SZ")
expire_str,
"%Y-%m-%dT%H:%M:%SZ"
)
self.expire_time = expire_time.timestamp() - 60 self.expire_time = expire_time.timestamp() - 60
except Exception as e: except Exception as e:
raise ValueError(f"无效的过期时间格式: {expire_str}") from e raise ValueError(f"无效的过期时间格式: {expire_str}") from e
else: else:
self.expire_time = None self.expire_time = None
if not self.token: if not self.token:
raise ValueError("无法获取有效的访问Token") raise ValueError("无法获取有效的访问Token")
@@ -142,8 +159,6 @@ class TTSProvider(TTSProviderBase):
# f"过期时间 {datetime.fromtimestamp(self.expire_time)} | " # f"过期时间 {datetime.fromtimestamp(self.expire_time)} | "
# f"剩余 {remaining:.2f}秒") # f"剩余 {remaining:.2f}秒")
return time.time() > self.expire_time return time.time() > self.expire_time
def generate_filename(self, extension=".wav"):
return os.path.join(self.output_file, f"tts-{__name__}{datetime.now().date()}@{uuid.uuid4().hex}{extension}")
async def text_to_speak(self, text, output_file): async def text_to_speak(self, text, output_file):
if self._is_token_expired(): if self._is_token_expired():
@@ -158,21 +173,27 @@ class TTSProvider(TTSProviderBase):
"voice": self.voice, "voice": self.voice,
"volume": self.volume, "volume": self.volume,
"speech_rate": self.speech_rate, "speech_rate": self.speech_rate,
"pitch_rate": self.pitch_rate "pitch_rate": self.pitch_rate,
} }
# print(self.api_url, json.dumps(request_json, ensure_ascii=False)) # print(self.api_url, json.dumps(request_json, ensure_ascii=False))
try: try:
resp = requests.post(self.api_url, json.dumps(request_json), headers=self.header) resp = requests.post(
self.api_url, json.dumps(request_json), headers=self.header
)
if resp.status_code == 401: # Token过期特殊处理 if resp.status_code == 401: # Token过期特殊处理
self._refresh_token() self._refresh_token()
resp = requests.post(self.api_url, json.dumps(request_json), headers=self.header) resp = requests.post(
self.api_url, json.dumps(request_json), headers=self.header
)
# 检查返回请求数据的mime类型是否是audio/***,是则保存到指定路径下;返回的是binary格式的 # 检查返回请求数据的mime类型是否是audio/***,是则保存到指定路径下;返回的是binary格式的
if resp.headers['Content-Type'].startswith('audio/'): if resp.headers["Content-Type"].startswith("audio/"):
with open(output_file, 'wb') as f: with open(output_file, "wb") as f:
f.write(resp.content) f.write(resp.content)
return output_file return output_file
else: else:
raise Exception(f"{__name__} status_code: {resp.status_code} response: {resp.content}") raise Exception(
f"{__name__} status_code: {resp.status_code} response: {resp.content}"
)
except Exception as e: except Exception as e:
raise Exception(f"{__name__} error: {e}") raise Exception(f"{__name__} error: {e}")
+286 -7
View File
@@ -1,9 +1,27 @@
import asyncio
from config.logger import setup_logging
import os import os
import queue
import uuid
import asyncio
import threading
from core.utils import p3
from datetime import datetime
from core.utils import textUtils
from abc import ABC, abstractmethod from abc import ABC, abstractmethod
from core.utils.tts import MarkdownCleaner from config.logger import setup_logging
from core.utils.util import audio_to_data from core.utils.util import audio_to_data
from core.utils.tts import MarkdownCleaner
from core.utils.output_counter import add_device_output
from core.handle.reportHandle import enqueue_tts_report
from core.handle.sendAudioHandle import sendAudioMessage
from core.providers.tts.dto.dto import (
TTSMessageDTO,
SentenceType,
ContentType,
InterfaceType,
)
import traceback
TAG = __name__ TAG = __name__
logger = setup_logging() logger = setup_logging()
@@ -11,12 +29,51 @@ logger = setup_logging()
class TTSProviderBase(ABC): class TTSProviderBase(ABC):
def __init__(self, config, delete_audio_file): def __init__(self, config, delete_audio_file):
self.interface_type = InterfaceType.NON_STREAM
self.conn = None
self.tts_timeout = 10
self.delete_audio_file = delete_audio_file self.delete_audio_file = delete_audio_file
self.output_file = config.get("output_dir") self.output_file = config.get("output_dir", "tmp/")
self.tts_text_queue = queue.Queue()
self.tts_audio_queue = queue.Queue()
self.tts_audio_first_sentence = True
@abstractmethod self.tts_text_buff = []
def generate_filename(self): self.punctuations = (
pass "",
".",
"",
"?",
"",
"!",
"",
";",
"",
)
self.first_sentence_punctuations = (
"",
"~",
"",
",",
"",
".",
"",
"?",
"",
"!",
"",
";",
"",
)
self.tts_stop_request = False
self.processed_chars = 0
self.is_first_sentence = True
def generate_filename(self, extension=".wav"):
return os.path.join(
self.output_file,
f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}",
)
def to_tts(self, text): def to_tts(self, text):
tmp_file = self.generate_filename() tmp_file = self.generate_filename()
@@ -60,3 +117,225 @@ class TTSProviderBase(ABC):
def audio_to_opus_data(self, audio_file_path): def audio_to_opus_data(self, audio_file_path):
"""音频文件转换为Opus编码""" """音频文件转换为Opus编码"""
return audio_to_data(audio_file_path, is_opus=True) return audio_to_data(audio_file_path, is_opus=True)
def tts_one_sentence(
self,
conn,
content_type,
content_detail=None,
content_file=None,
sentence_id=None,
):
"""发送一句话"""
if not sentence_id:
if conn.sentence_id:
sentence_id = conn.sentence_id
else:
sentence_id = str(uuid.uuid4()).replace("-", "")
conn.sentence_id = sentence_id
self.tts_text_queue.put(
TTSMessageDTO(
sentence_id=sentence_id,
sentence_type=SentenceType.FIRST,
content_type=ContentType.ACTION,
)
)
self.tts_text_queue.put(
TTSMessageDTO(
sentence_id=sentence_id,
sentence_type=SentenceType.MIDDLE,
content_type=content_type,
content_detail=content_detail,
content_file=content_file,
)
)
self.tts_text_queue.put(
TTSMessageDTO(
sentence_id=sentence_id,
sentence_type=SentenceType.LAST,
content_type=ContentType.ACTION,
)
)
async def open_audio_channels(self, conn):
self.conn = conn
self.tts_timeout = conn.config.get("tts_timeout", 10)
# tts 消化线程
self.tts_priority_thread = threading.Thread(
target=self.tts_text_priority_thread, daemon=True
)
self.tts_priority_thread.start()
# 音频播放 消化线程
self.audio_play_priority_thread = threading.Thread(
target=self._audio_play_priority_thread, daemon=True
)
self.audio_play_priority_thread.start()
# 这里默认是非流式的处理方式
# 流式处理方式请在子类中重写
def tts_text_priority_thread(self):
while not self.conn.stop_event.is_set():
try:
message = self.tts_text_queue.get(timeout=1)
if message.sentence_type == SentenceType.FIRST:
# 初始化参数
self.tts_stop_request = False
self.processed_chars = 0
self.tts_text_buff = []
self.is_first_sentence = True
self.tts_audio_first_sentence = True
elif ContentType.TEXT == message.content_type:
self.tts_text_buff.append(message.content_detail)
segment_text = self._get_segment_text()
if segment_text:
tts_file = self.to_tts(segment_text)
if tts_file:
audio_datas = self._process_audio_file(tts_file)
self.tts_audio_queue.put(
(message.sentence_type, audio_datas, segment_text)
)
elif ContentType.FILE == message.content_type:
self._process_remaining_text()
tts_file = message.content_file
if tts_file and os.path.exists(tts_file):
audio_datas = self._process_audio_file(tts_file)
self.tts_audio_queue.put(
(message.sentence_type, audio_datas, message.content_detail)
)
if message.sentence_type == SentenceType.LAST:
self._process_remaining_text()
self.tts_audio_queue.put(
(message.sentence_type, [], message.content_detail)
)
except queue.Empty:
continue
except Exception as e:
logger.bind(tag=TAG).error(
f"处理TTS文本失败: {str(e)}, 类型: {type(e).__name__}, 堆栈: {traceback.format_exc()}"
)
continue
def _audio_play_priority_thread(self):
while not self.conn.stop_event.is_set():
text = None
try:
try:
sentence_type, audio_datas, text = self.tts_audio_queue.get(
timeout=1
)
except queue.Empty:
if self.conn.stop_event.is_set():
break
continue
future = asyncio.run_coroutine_threadsafe(
sendAudioMessage(self.conn, sentence_type, audio_datas, text),
self.conn.loop,
)
future.result()
if self.conn.max_output_size > 0 and text:
add_device_output(self.conn.headers.get("device-id"), len(text))
enqueue_tts_report(self.conn, text, audio_datas)
except Exception as e:
logger.bind(tag=TAG).error(
f"audio_play_priority priority_thread: {text} {e}"
)
async def start_session(self, session_id):
pass
async def finish_session(self, session_id):
pass
async def close(self):
"""资源清理方法"""
if hasattr(self, "ws") and self.ws:
await self.ws.close()
def _get_segment_text(self):
# 合并当前全部文本并处理未分割部分
full_text = "".join(self.tts_text_buff)
current_text = full_text[self.processed_chars :] # 从未处理的位置开始
last_punct_pos = -1
# 根据是否是第一句话选择不同的标点符号集合
punctuations_to_use = (
self.first_sentence_punctuations
if self.is_first_sentence
else self.punctuations
)
for punct in punctuations_to_use:
pos = current_text.rfind(punct)
if (pos != -1 and last_punct_pos == -1) or (
pos != -1 and pos < last_punct_pos
):
last_punct_pos = pos
if last_punct_pos != -1:
segment_text_raw = current_text[: last_punct_pos + 1]
segment_text = textUtils.get_string_no_punctuation_or_emoji(
segment_text_raw
)
self.processed_chars += len(segment_text_raw) # 更新已处理字符位置
# 如果是第一句话,在找到第一个逗号后,将标志设置为False
if self.is_first_sentence:
self.is_first_sentence = False
return segment_text
elif self.tts_stop_request and current_text:
segment_text = current_text
self.is_first_sentence = True # 重置标志
return segment_text
else:
return None
def _process_audio_file(self, tts_file):
"""处理音频文件并转换为指定格式
Args:
tts_file: 音频文件路径
content_detail: 内容详情
Returns:
tuple: (sentence_type, audio_datas, content_detail)
"""
audio_datas = []
if tts_file.endswith(".p3"):
audio_datas, _ = p3.decode_opus_from_file(tts_file)
elif self.conn.audio_format == "pcm":
audio_datas, _ = self.audio_to_pcm_data(tts_file)
else:
audio_datas, _ = self.audio_to_opus_data(tts_file)
if (
self.delete_audio_file
and tts_file is not None
and os.path.exists(tts_file)
and tts_file.startswith(self.output_file)
):
os.remove(tts_file)
return audio_datas
def _process_remaining_text(self):
"""处理剩余的文本并生成语音
Returns:
bool: 是否成功处理了文本
"""
full_text = "".join(self.tts_text_buff)
remaining_text = full_text[self.processed_chars :]
if remaining_text:
segment_text = textUtils.get_string_no_punctuation_or_emoji(remaining_text)
if segment_text:
tts_file = self.to_tts(segment_text)
audio_datas = self._process_audio_file(tts_file)
self.tts_audio_queue.put(
(SentenceType.MIDDLE, audio_datas, segment_text)
)
self.processed_chars += len(full_text)
return True
return False
@@ -1,9 +1,4 @@
import os
import uuid
import json
import base64
import requests import requests
from datetime import datetime
from core.providers.tts.base import TTSProviderBase from core.providers.tts.base import TTSProviderBase
@@ -21,12 +16,6 @@ class TTSProvider(TTSProviderBase):
self.host = "api.coze.cn" self.host = "api.coze.cn"
self.api_url = f"https://{self.host}/v1/audio/speech" self.api_url = f"https://{self.host}/v1/audio/speech"
def generate_filename(self, extension=".wav"):
return os.path.join(
self.output_file,
f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}",
)
async def text_to_speak(self, text, output_file): async def text_to_speak(self, text, output_file):
request_json = { request_json = {
"model": self.model, "model": self.model,
@@ -47,4 +36,4 @@ class TTSProvider(TTSProviderBase):
file_to_save = open(output_file, "wb") file_to_save = open(output_file, "wb")
file_to_save.write(data) file_to_save.write(data)
except Exception as e: except Exception as e:
raise Exception(f"{__name__} error: {e}") raise Exception(f"{__name__} error: {e}")
@@ -0,0 +1,23 @@
import os
from config.logger import setup_logging
from core.providers.tts.base import TTSProviderBase
TAG = __name__
logger = setup_logging()
class DefaultTTS(TTSProviderBase):
def __init__(self, config, delete_audio_file=True):
super().__init__(config, delete_audio_file)
self.output_dir = config.get("output_dir", "output")
if not os.path.exists(self.output_dir):
os.makedirs(self.output_dir)
def generate_filename(self):
"""生成唯一的音频文件名"""
import uuid
return os.path.join(self.output_dir, f"{uuid.uuid4()}.wav")
async def text_to_speak(self, text, output_file):
logger.bind(tag=TAG).error(f"无法实例化 TTS 服务,请检查配置")
@@ -1,9 +1,7 @@
import os
import uuid import uuid
import json import json
import base64 import base64
import requests import requests
from datetime import datetime
from core.utils.util import check_model_key from core.utils.util import check_model_key
from core.providers.tts.base import TTSProviderBase from core.providers.tts.base import TTSProviderBase
from config.logger import setup_logging from config.logger import setup_logging
@@ -41,12 +39,6 @@ class TTSProvider(TTSProviderBase):
self.header = {"Authorization": f"{self.authorization}{self.access_token}"} self.header = {"Authorization": f"{self.authorization}{self.access_token}"}
check_model_key("TTS", self.access_token) check_model_key("TTS", self.access_token)
def generate_filename(self, extension=".wav"):
return os.path.join(
self.output_file,
f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}",
)
async def text_to_speak(self, text, output_file): async def text_to_speak(self, text, output_file):
request_json = { request_json = {
"app": { "app": {
@@ -0,0 +1,43 @@
from enum import Enum
from typing import Union, Optional
class SentenceType(Enum):
# 说话阶段
FIRST = "FIRST" # 首句话
MIDDLE = "MIDDLE" # 说话中
LAST = "LAST" # 最后一句
class ContentType(Enum):
# 内容类型
TEXT = "TEXT" # 文本内容
FILE = "FILE" # 文件内容
ACTION = "ACTION" # 动作内容
class InterfaceType(Enum):
# 接口类型
DUAL_STREAM = "DUAL_STREAM" # 双流式
SINGLE_STREAM = "SINGLE_STREAM" # 单流式
NON_STREAM = "NON_STREAM" # 非流式
class TTSMessageDTO:
def __init__(
self,
sentence_id: str,
# 说话阶段
sentence_type: SentenceType,
# 内容类型
content_type: ContentType,
# 内容详情,一般是需要转换的文本或者音频的歌词
content_detail: Optional[str] = None,
# 如果内容类型为文件,则需要传入文件路径
content_file: Optional[str] = None,
):
self.sentence_id = sentence_id
self.sentence_type = sentence_type
self.content_type = content_type
self.content_detail = content_detail
self.content_file = content_file
@@ -1,12 +1,9 @@
import base64 import base64
import os
import uuid
import requests import requests
import ormsgpack import ormsgpack
from pathlib import Path from pathlib import Path
from pydantic import BaseModel, Field, conint, model_validator from pydantic import BaseModel, Field, conint, model_validator
from typing_extensions import Annotated from typing_extensions import Annotated
from datetime import datetime
from typing import Literal from typing import Literal
from core.utils.util import check_model_key, parse_string_to_list from core.utils.util import check_model_key, parse_string_to_list
from core.providers.tts.base import TTSProviderBase from core.providers.tts.base import TTSProviderBase
@@ -133,12 +130,6 @@ class TTSProvider(TTSProviderBase):
self.seed = int(config.get("seed")) if config.get("seed") else None self.seed = int(config.get("seed")) if config.get("seed") else None
self.api_url = config.get("api_url", "http://127.0.0.1:8080/v1/tts") self.api_url = config.get("api_url", "http://127.0.0.1:8080/v1/tts")
def generate_filename(self, extension=".wav"):
return os.path.join(
self.output_file,
f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}",
)
async def text_to_speak(self, text, output_file): async def text_to_speak(self, text, output_file):
# Prepare reference data # Prepare reference data
byte_audios = [audio_to_bytes(ref_audio) for ref_audio in self.reference_audio] byte_audios = [audio_to_bytes(ref_audio) for ref_audio in self.reference_audio]
@@ -1,10 +1,5 @@
import os
import uuid
import json
import base64
import requests import requests
from config.logger import setup_logging from config.logger import setup_logging
from datetime import datetime
from core.providers.tts.base import TTSProviderBase from core.providers.tts.base import TTSProviderBase
from core.utils.util import parse_string_to_list from core.utils.util import parse_string_to_list
@@ -71,12 +66,6 @@ class TTSProvider(TTSProviderBase):
config.get("aux_ref_audio_paths") config.get("aux_ref_audio_paths")
) )
def generate_filename(self, extension=".wav"):
return os.path.join(
self.output_file,
f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}",
)
async def text_to_speak(self, text, output_file): async def text_to_speak(self, text, output_file):
request_json = { request_json = {
"text": text, "text": text,
@@ -1,8 +1,5 @@
import os
import uuid
import requests import requests
from config.logger import setup_logging from config.logger import setup_logging
from datetime import datetime
from core.providers.tts.base import TTSProviderBase from core.providers.tts.base import TTSProviderBase
from core.utils.util import parse_string_to_list from core.utils.util import parse_string_to_list
@@ -36,12 +33,6 @@ class TTSProvider(TTSProviderBase):
self.inp_refs = parse_string_to_list(config.get("inp_refs")) self.inp_refs = parse_string_to_list(config.get("inp_refs"))
self.if_sr = str(config.get("if_sr", False)).lower() in ("true", "1", "yes") self.if_sr = str(config.get("if_sr", False)).lower() in ("true", "1", "yes")
def generate_filename(self, extension=".wav"):
return os.path.join(
self.output_file,
f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}",
)
async def text_to_speak(self, text, output_file): async def text_to_speak(self, text, output_file):
request_params = { request_params = {
"refer_wav_path": self.refer_wav_path, "refer_wav_path": self.refer_wav_path,
@@ -67,4 +58,3 @@ class TTSProvider(TTSProviderBase):
error_msg = f"GPT_SoVITS_V3 TTS请求失败: {resp.status_code} - {resp.text}" error_msg = f"GPT_SoVITS_V3 TTS请求失败: {resp.status_code} - {resp.text}"
logger.bind(tag=TAG).error(error_msg) logger.bind(tag=TAG).error(error_msg)
raise Exception(error_msg) raise Exception(error_msg)
@@ -0,0 +1,739 @@
import os
import uuid
import json
import queue
import asyncio
import threading
import traceback
import websockets
import time
from config.logger import setup_logging
from core.utils import opus_encoder_utils
from core.utils.util import check_model_key
from core.providers.tts.base import TTSProviderBase
from core.providers.tts.dto.dto import SentenceType, ContentType, InterfaceType
TAG = __name__
logger = setup_logging()
PROTOCOL_VERSION = 0b0001
DEFAULT_HEADER_SIZE = 0b0001
# Message Type:
FULL_CLIENT_REQUEST = 0b0001
AUDIO_ONLY_RESPONSE = 0b1011
FULL_SERVER_RESPONSE = 0b1001
ERROR_INFORMATION = 0b1111
# Message Type Specific Flags
MsgTypeFlagNoSeq = 0b0000 # Non-terminal packet with no sequence
MsgTypeFlagPositiveSeq = 0b1 # Non-terminal packet with sequence > 0
MsgTypeFlagLastNoSeq = 0b10 # last packet with no sequence
MsgTypeFlagNegativeSeq = 0b11 # Payload contains event number (int32)
MsgTypeFlagWithEvent = 0b100
# Message Serialization
NO_SERIALIZATION = 0b0000
JSON = 0b0001
# Message Compression
COMPRESSION_NO = 0b0000
COMPRESSION_GZIP = 0b0001
EVENT_NONE = 0
EVENT_Start_Connection = 1
EVENT_FinishConnection = 2
EVENT_ConnectionStarted = 50 # 成功建连
EVENT_ConnectionFailed = 51 # 建连失败(可能是无法通过权限认证)
EVENT_ConnectionFinished = 52 # 连接结束
# 上行Session事件
EVENT_StartSession = 100
EVENT_FinishSession = 102
# 下行Session事件
EVENT_SessionStarted = 150
EVENT_SessionFinished = 152
EVENT_SessionFailed = 153
# 上行通用事件
EVENT_TaskRequest = 200
# 下行TTS事件
EVENT_TTSSentenceStart = 350
EVENT_TTSSentenceEnd = 351
EVENT_TTSResponse = 352
class Header:
def __init__(
self,
protocol_version=PROTOCOL_VERSION,
header_size=DEFAULT_HEADER_SIZE,
message_type: int = 0,
message_type_specific_flags: int = 0,
serial_method: int = NO_SERIALIZATION,
compression_type: int = COMPRESSION_NO,
reserved_data=0,
):
self.header_size = header_size
self.protocol_version = protocol_version
self.message_type = message_type
self.message_type_specific_flags = message_type_specific_flags
self.serial_method = serial_method
self.compression_type = compression_type
self.reserved_data = reserved_data
def as_bytes(self) -> bytes:
return bytes(
[
(self.protocol_version << 4) | self.header_size,
(self.message_type << 4) | self.message_type_specific_flags,
(self.serial_method << 4) | self.compression_type,
self.reserved_data,
]
)
class Optional:
def __init__(
self, event: int = EVENT_NONE, sessionId: str = None, sequence: int = None
):
self.event = event
self.sessionId = sessionId
self.errorCode: int = 0
self.connectionId: str | None = None
self.response_meta_json: str | None = None
self.sequence = sequence
# 转成 byte 序列
def as_bytes(self) -> bytes:
option_bytes = bytearray()
if self.event != EVENT_NONE:
option_bytes.extend(self.event.to_bytes(4, "big", signed=True))
if self.sessionId is not None:
session_id_bytes = str.encode(self.sessionId)
size = len(session_id_bytes).to_bytes(4, "big", signed=True)
option_bytes.extend(size)
option_bytes.extend(session_id_bytes)
if self.sequence is not None:
option_bytes.extend(self.sequence.to_bytes(4, "big", signed=True))
return option_bytes
class Response:
def __init__(self, header: Header, optional: Optional):
self.optional = optional
self.header = header
self.payload: bytes | None = None
def __str__(self):
return super().__str__()
class TTSProvider(TTSProviderBase):
def __init__(self, config, delete_audio_file):
super().__init__(config, delete_audio_file)
self.ws = None # 初始化ws属性
self.interface_type = InterfaceType.DUAL_STREAM
self.appId = config.get("appid")
self.access_token = config.get("access_token")
self.cluster = config.get("cluster")
self.resource_id = config.get("resource_id")
if config.get("private_voice"):
self.speaker = config.get("private_voice")
else:
self.speaker = config.get("speaker")
self.voice = config.get("voice")
self.ws_url = config.get("ws_url")
self.authorization = config.get("authorization")
self.header = {"Authorization": f"{self.authorization}{self.access_token}"}
self.enable_two_way = True
self.start_connection_flag = False
self.tts_text = ""
# 合成文字语音后,播放的音频文件列表
self.before_stop_play_files = []
self.opus_encoder = opus_encoder_utils.OpusEncoderUtils(
sample_rate=16000, channels=1, frame_size_ms=60
)
check_model_key("TTS", self.access_token)
# 添加会话状态控制
self._session_lock = asyncio.Lock() # 会话操作的并发锁
self._current_session_id = None # 当前会话ID
self._session_started = False # 会话是否已开始
self._session_finished = False # 会话是否已结束
self._connection_ready = False # 连接是否就绪
self._reconnect_attempts = 0 # 重连尝试次数
self._max_reconnect_attempts = 3 # 最大重连次数
###################################################################################
# 火山双流式TTS重写父类的方法--开始
###################################################################################
async def open_audio_channels(self, conn):
try:
await super().open_audio_channels(conn)
await self._ensure_connection()
tts_priority = threading.Thread(
target=self._start_monitor_tts_response_thread, daemon=True
)
tts_priority.start()
except Exception as e:
logger.bind(tag=TAG).error(f"Failed to open audio channels: {str(e)}")
self.ws = None
raise
async def _ensure_connection(self):
"""确保WebSocket连接可用"""
try:
if self.ws is None:
logger.bind(tag=TAG).info("WebSocket连接不存在,开始建立新连接...")
ws_header = {
"X-Api-App-Key": self.appId,
"X-Api-Access-Key": self.access_token,
"X-Api-Resource-Id": self.resource_id,
"X-Api-Connect-Id": uuid.uuid4(),
}
self.ws = await websockets.connect(
self.ws_url, additional_headers=ws_header, max_size=1000000000
)
self._connection_ready = True
self._reconnect_attempts = 0
logger.bind(tag=TAG).info("WebSocket连接建立成功")
else:
# 尝试发送ping来检查连接是否还活着
try:
logger.bind(tag=TAG).debug("检查WebSocket连接状态...")
pong_waiter = await self.ws.ping()
await asyncio.wait_for(pong_waiter, timeout=1.0)
logger.bind(tag=TAG).debug("WebSocket连接状态正常")
except (asyncio.TimeoutError, websockets.ConnectionClosed):
# 如果ping失败,重新建立连接
logger.bind(tag=TAG).warning("WebSocket连接已断开,准备重新连接...")
try:
await self.ws.close()
except:
pass
self.ws = None
self._connection_ready = False
# 重新建立连接
await self._ensure_connection()
except Exception as e:
logger.bind(tag=TAG).error(f"确保连接失败: {str(e)}")
self._connection_ready = False
self.ws = None
raise
def tts_text_priority_thread(self):
logger.bind(tag=TAG).info("TTS文本处理线程启动")
while not self.conn.stop_event.is_set():
try:
logger.bind(tag=TAG).debug("等待TTS文本队列消息...")
message = self.tts_text_queue.get(timeout=1)
logger.bind(tag=TAG).info(
f"收到TTS任务|{message.sentence_type.name} {message.content_type.name} | 会话ID: {self.conn.sentence_id}"
)
if message.sentence_type == SentenceType.FIRST:
# 初始化参数
try:
logger.bind(tag=TAG).info("开始启动TTS会话...")
future = asyncio.run_coroutine_threadsafe(
self.start_session(self.conn.sentence_id),
loop=self.conn.loop,
)
future.result()
self.tts_audio_first_sentence = True
self.before_stop_play_files.clear()
logger.bind(tag=TAG).info("TTS会话启动成功")
except Exception as e:
logger.bind(tag=TAG).error(f"启动TTS会话失败: {str(e)}")
# 直接跳过当前消息,不重新入队
time.sleep(1)
continue
elif ContentType.TEXT == message.content_type:
if message.content_detail:
try:
logger.bind(tag=TAG).info(
f"开始发送TTS文本: {message.content_detail}"
)
future = asyncio.run_coroutine_threadsafe(
self.text_to_speak(message.content_detail, None),
loop=self.conn.loop,
)
future.result()
logger.bind(tag=TAG).info("TTS文本发送成功")
except Exception as e:
logger.bind(tag=TAG).error(f"发送TTS文本失败: {str(e)}")
# 直接跳过当前消息,不重新入队
time.sleep(1)
continue
elif ContentType.FILE == message.content_type:
logger.bind(tag=TAG).info(
f"添加音频文件到待播放列表: {message.content_file}"
)
self.before_stop_play_files.append(
(message.content_file, message.content_detail)
)
if message.sentence_type == SentenceType.LAST:
try:
logger.bind(tag=TAG).info("开始结束TTS会话...")
future = asyncio.run_coroutine_threadsafe(
self.finish_session(self.conn.sentence_id),
loop=self.conn.loop,
)
future.result()
logger.bind(tag=TAG).info("TTS会话结束成功")
except Exception as e:
logger.bind(tag=TAG).error(f"结束TTS会话失败: {str(e)}")
# 直接跳过当前消息,不重新入队
time.sleep(1)
continue
except queue.Empty:
continue
except Exception as e:
logger.bind(tag=TAG).error(
f"处理TTS文本失败: {str(e)}, 类型: {type(e).__name__}, 堆栈: {traceback.format_exc()}"
)
# 如果是WebSocket连接关闭错误,等待一段时间后继续
if "non-exist session" in str(e):
time.sleep(1)
continue
async def text_to_speak(self, text, _):
"""发送文本到TTS服务"""
try:
# 确保WebSocket连接可用
if not self._connection_ready or self.ws is None:
logger.bind(tag=TAG).warning("WebSocket连接不可用,尝试重新连接...")
await self._ensure_connection()
# 发送文本
await self.send_text(self.speaker, text, self.conn.sentence_id)
return
except Exception as e:
logger.bind(tag=TAG).error(f"发送TTS文本失败: {str(e)}")
# 如果是连接问题,尝试重新连接
if isinstance(e, websockets.ConnectionClosed):
self._connection_ready = False
self.ws = None
await self._handle_connection_error()
raise
###################################################################################
# 火山双流式TTS重写父类的方法--结束
###################################################################################
def _start_monitor_tts_response_thread(self):
# 初始化链接
asyncio.run_coroutine_threadsafe(
self._start_monitor_tts_response(), loop=self.conn.loop
)
async def _start_monitor_tts_response(self):
opus_datas_cache = []
# 添加标志来区分是否是第一句话
is_first_sentence = True
while not self.conn.stop_event.is_set():
try:
# 确保 `recv()` 运行在同一个 event loop
msg = await self.ws.recv()
res = self.parser_response(msg)
self.print_response(res, "send_text res:")
if res.optional.event == EVENT_TTSSentenceStart:
json_data = json.loads(res.payload.decode("utf-8"))
self.tts_text = json_data.get("text", "")
logger.bind(tag=TAG).debug(f"句子语音生成开始: {self.tts_text}")
self.tts_audio_queue.put((SentenceType.FIRST, [], self.tts_text))
opus_datas_cache = []
elif (
res.optional.event == EVENT_TTSResponse
and res.header.message_type == AUDIO_ONLY_RESPONSE
):
logger.bind(tag=TAG).debug(f"推送数据到队列里面~~")
opus_datas = self.wav_to_opus_data_audio_raw(res.payload)
logger.bind(tag=TAG).debug(
f"推送数据到队列里面帧数~~{len(opus_datas)}"
)
if is_first_sentence:
# 第一句话直接发送
self.tts_audio_queue.put(
(SentenceType.MIDDLE, opus_datas, self.tts_text)
)
else:
# 后续句子缓存
opus_datas_cache = opus_datas_cache + opus_datas
elif res.optional.event == EVENT_TTSSentenceEnd:
logger.bind(tag=TAG).info(f"句子语音生成成功:{self.tts_text}")
if not is_first_sentence:
# 只有非第一句话才发送缓存的数据
self.tts_audio_queue.put(
(SentenceType.MIDDLE, opus_datas_cache, self.tts_text)
)
# 第一句话结束后,将标志设置为False
is_first_sentence = False
elif res.optional.event == EVENT_SessionFinished:
logger.bind(tag=TAG).debug(f"会话结束~~")
for tts_file, text in self.before_stop_play_files:
if tts_file and os.path.exists(tts_file):
audio_datas = self._process_audio_file(tts_file)
self.tts_audio_queue.put(
(SentenceType.MIDDLE, audio_datas, text)
)
self.before_stop_play_files.clear()
self.tts_audio_queue.put((SentenceType.LAST, [], None))
opus_datas_cache = []
is_first_sentence = True
continue
except websockets.ConnectionClosed:
break # 连接关闭时退出监听
except Exception as e:
logger.bind(tag=TAG).error(f"Error in _start_monitor_tts_response: {e}")
traceback.print_exc()
continue
async def send_event(
self, header: bytes, optional: bytes | None = None, payload: bytes = None
):
try:
full_client_request = bytearray(header)
if optional is not None:
full_client_request.extend(optional)
if payload is not None:
payload_size = len(payload).to_bytes(4, "big", signed=True)
full_client_request.extend(payload_size)
full_client_request.extend(payload)
await self.ws.send(full_client_request)
except websockets.ConnectionClosed:
if await self._handle_connection_error():
# 重连成功后重试发送
await self.ws.send(full_client_request)
else:
raise
async def send_text(self, speaker: str, text: str, session_id):
header = Header(
message_type=FULL_CLIENT_REQUEST,
message_type_specific_flags=MsgTypeFlagWithEvent,
serial_method=JSON,
).as_bytes()
optional = Optional(event=EVENT_TaskRequest, sessionId=session_id).as_bytes()
payload = self.get_payload_bytes(
event=EVENT_TaskRequest, text=text, speaker=speaker
)
return await self.send_event(header, optional, payload)
# 读取 res 数组某段 字符串内容
def read_res_content(self, res: bytes, offset: int):
content_size = int.from_bytes(res[offset : offset + 4], "big", signed=True)
offset += 4
content = str(res[offset : offset + content_size])
offset += content_size
return content, offset
# 读取 payload
def read_res_payload(self, res: bytes, offset: int):
payload_size = int.from_bytes(res[offset : offset + 4], "big", signed=True)
offset += 4
payload = res[offset : offset + payload_size]
offset += payload_size
return payload, offset
def parser_response(self, res) -> Response:
if isinstance(res, str):
raise RuntimeError(res)
response = Response(Header(), Optional())
# 解析结果
# header
header = response.header
num = 0b00001111
header.protocol_version = res[0] >> 4 & num
header.header_size = res[0] & 0x0F
header.message_type = (res[1] >> 4) & num
header.message_type_specific_flags = res[1] & 0x0F
header.serialization_method = res[2] >> num
header.message_compression = res[2] & 0x0F
header.reserved = res[3]
#
offset = 4
optional = response.optional
if header.message_type == FULL_SERVER_RESPONSE or AUDIO_ONLY_RESPONSE:
# read event
if header.message_type_specific_flags == MsgTypeFlagWithEvent:
optional.event = int.from_bytes(res[offset:8], "big", signed=True)
offset += 4
if optional.event == EVENT_NONE:
return response
# read connectionId
elif optional.event == EVENT_ConnectionStarted:
optional.connectionId, offset = self.read_res_content(res, offset)
elif optional.event == EVENT_ConnectionFailed:
optional.response_meta_json, offset = self.read_res_content(
res, offset
)
elif (
optional.event == EVENT_SessionStarted
or optional.event == EVENT_SessionFailed
or optional.event == EVENT_SessionFinished
):
optional.sessionId, offset = self.read_res_content(res, offset)
optional.response_meta_json, offset = self.read_res_content(
res, offset
)
else:
optional.sessionId, offset = self.read_res_content(res, offset)
response.payload, offset = self.read_res_payload(res, offset)
elif header.message_type == ERROR_INFORMATION:
optional.errorCode = int.from_bytes(
res[offset : offset + 4], "big", signed=True
)
offset += 4
response.payload, offset = self.read_res_payload(res, offset)
return response
async def start_connection(self):
header = Header(
message_type=FULL_CLIENT_REQUEST,
message_type_specific_flags=MsgTypeFlagWithEvent,
).as_bytes()
optional = Optional(event=EVENT_Start_Connection).as_bytes()
payload = str.encode("{}")
return await self.send_event(header, optional, payload)
def print_response(self, res, tag_msg: str):
logger.bind(tag=TAG).debug(f"===>{tag_msg} header:{res.header.__dict__}")
logger.bind(tag=TAG).debug(f"===>{tag_msg} optional:{res.optional.__dict__}")
def get_payload_bytes(
self,
uid="1234",
event=EVENT_NONE,
text="",
speaker="",
audio_format="pcm",
audio_sample_rate=16000,
):
return str.encode(
json.dumps(
{
"user": {"uid": uid},
"event": event,
"namespace": "BidirectionalTTS",
"req_params": {
"text": text,
"speaker": speaker,
"audio_params": {
"format": audio_format,
"sample_rate": audio_sample_rate,
},
},
}
)
)
async def finish_connection(self):
header = Header(
message_type=FULL_CLIENT_REQUEST,
message_type_specific_flags=MsgTypeFlagWithEvent,
serial_method=JSON,
).as_bytes()
optional = Optional(event=EVENT_FinishConnection).as_bytes()
payload = str.encode("{}")
await self.send_event(header, optional, payload)
return
async def start_session(self, session_id):
logger.bind(tag=TAG).info(f"开始会话~~{session_id}")
try:
async with self._session_lock:
try:
# 确保连接可用
logger.bind(tag=TAG).info("检查WebSocket连接状态...")
await asyncio.wait_for(self._ensure_connection(), timeout=5)
# 如果已有会话未结束,先关闭它
if self._session_started and not self._session_finished:
logger.bind(tag=TAG).warning(
f"发现未关闭的会话 {self._current_session_id},正在关闭..."
)
try:
await asyncio.wait_for(
self.finish_session(self._current_session_id), timeout=5
)
except Exception as e:
logger.bind(tag=TAG).error(f"关闭旧会话失败: {str(e)}")
# 强制重置会话状态
self._session_started = False
self._session_finished = True
self._current_session_id = None
# 重置会话状态
self._current_session_id = session_id
self._session_started = True
self._session_finished = False
logger.bind(tag=TAG).info(
f"会话状态已更新 - 开始: {self._session_started}, 结束: {self._session_finished}"
)
header = Header(
message_type=FULL_CLIENT_REQUEST,
message_type_specific_flags=MsgTypeFlagWithEvent,
serial_method=JSON,
).as_bytes()
optional = Optional(
event=EVENT_StartSession, sessionId=session_id
).as_bytes()
payload = self.get_payload_bytes(
event=EVENT_StartSession, speaker=self.speaker
)
await asyncio.wait_for(
self.send_event(header, optional, payload), timeout=5
)
logger.bind(tag=TAG).info("会话启动请求已发送")
except Exception as e:
logger.bind(tag=TAG).error(f"启动会话失败: {str(e)}")
self._session_started = False
self._session_finished = True
self._current_session_id = None
raise
except asyncio.TimeoutError:
logger.bind(tag=TAG).error(f"启动会话超时: {session_id}")
# 超时后强制重置会话状态
self._session_started = False
self._session_finished = True
self._current_session_id = None
# 尝试关闭WebSocket连接
if self.ws:
try:
await self.ws.close()
except:
pass
self.ws = None
except Exception as e:
logger.bind(tag=TAG).error(f"启动会话时发生未知错误: {str(e)}")
# 发生未知错误时也重置会话状态
self._session_started = False
self._session_finished = True
self._current_session_id = None
async def finish_session(self, session_id):
logger.bind(tag=TAG).info(f"关闭会话~~{session_id}")
try:
async with self._session_lock:
try:
# 检查会话状态
if not self._session_started:
logger.bind(tag=TAG).warning(
f"尝试关闭未开始的会话 {session_id}"
)
return
if self._session_finished:
logger.bind(tag=TAG).warning(f"会话 {session_id} 已经关闭")
return
if self._current_session_id != session_id:
logger.bind(tag=TAG).warning(
f"尝试关闭错误的会话 {session_id},当前会话为 {self._current_session_id}"
)
# 即使会话ID不匹配,也尝试关闭当前会话
if self._current_session_id:
session_id = self._current_session_id
# 确保WebSocket连接可用
if self.ws is None:
logger.bind(tag=TAG).warning(
"WebSocket连接不存在,尝试重新连接..."
)
await asyncio.wait_for(self._ensure_connection(), timeout=5)
header = Header(
message_type=FULL_CLIENT_REQUEST,
message_type_specific_flags=MsgTypeFlagWithEvent,
serial_method=JSON,
).as_bytes()
optional = Optional(
event=EVENT_FinishSession, sessionId=session_id
).as_bytes()
payload = str.encode("{}")
await asyncio.wait_for(
self.send_event(header, optional, payload), timeout=5
)
logger.bind(tag=TAG).info("会话结束请求已发送")
# 更新会话状态
self._session_finished = True
self._session_started = False
self._current_session_id = None
logger.bind(tag=TAG).info(
"会话状态已更新 - 开始: False, 结束: True"
)
except Exception as e:
logger.bind(tag=TAG).error(f"关闭会话失败: {str(e)}")
# 即使发生错误,也要重置会话状态
self._session_finished = True
self._session_started = False
self._current_session_id = None
raise
except asyncio.TimeoutError:
logger.bind(tag=TAG).error(f"关闭会话超时: {session_id}")
# 超时后强制重置会话状态
self._session_finished = True
self._session_started = False
self._current_session_id = None
# 尝试关闭WebSocket连接
if self.ws:
try:
await self.ws.close()
except:
pass
self.ws = None
except Exception as e:
logger.bind(tag=TAG).error(f"关闭会话时发生未知错误: {str(e)}")
# 发生未知错误时也重置会话状态
self._session_finished = True
self._session_started = False
self._current_session_id = None
async def reset(self):
# 关闭之前的对话
if self.start_connection_flag:
await self.finish_connection()
self.start_connection_flag = False
await self.start_connection()
self.start_connection_flag = True
async def close(self):
"""资源清理方法"""
await self.finish_connection()
await self.ws.close()
def wav_to_opus_data_audio_raw(self, raw_data_var, is_end=False):
opus_datas = self.opus_encoder.encode_pcm_to_opus(raw_data_var, is_end)
return opus_datas
async def _handle_connection_error(self):
"""处理连接错误"""
if self._reconnect_attempts < self._max_reconnect_attempts:
self._reconnect_attempts += 1
logger.bind(tag=TAG).warning(
f"尝试重新连接 (第{self._reconnect_attempts}次)"
)
try:
await self._ensure_connection()
return True
except Exception as e:
logger.bind(tag=TAG).error(f"重新连接失败: {str(e)}")
return False
else:
logger.bind(tag=TAG).error("达到最大重连次数,放弃重连")
return False
@@ -1,7 +1,4 @@
import os
import uuid
import requests import requests
from datetime import datetime
from core.utils.util import check_model_key from core.utils.util import check_model_key
from core.providers.tts.base import TTSProviderBase from core.providers.tts.base import TTSProviderBase
from config.logger import setup_logging from config.logger import setup_logging
@@ -29,12 +26,6 @@ class TTSProvider(TTSProviderBase):
self.output_file = config.get("output_dir", "tmp/") self.output_file = config.get("output_dir", "tmp/")
check_model_key("TTS", self.api_key) check_model_key("TTS", self.api_key)
def generate_filename(self, extension=".wav"):
return os.path.join(
self.output_file,
f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}",
)
async def text_to_speak(self, text, output_file): async def text_to_speak(self, text, output_file):
headers = { headers = {
"Authorization": f"Bearer {self.api_key}", "Authorization": f"Bearer {self.api_key}",
@@ -1,7 +1,4 @@
import os
import uuid
import requests import requests
from datetime import datetime
from core.providers.tts.base import TTSProviderBase from core.providers.tts.base import TTSProviderBase
@@ -22,12 +19,6 @@ class TTSProvider(TTSProviderBase):
self.host = "api.siliconflow.cn" self.host = "api.siliconflow.cn"
self.api_url = f"https://{self.host}/v1/audio/speech" self.api_url = f"https://{self.host}/v1/audio/speech"
def generate_filename(self, extension=".wav"):
return os.path.join(
self.output_file,
f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}",
)
async def text_to_speak(self, text, output_file): async def text_to_speak(self, text, output_file):
request_json = { request_json = {
"model": self.model, "model": self.model,
@@ -47,4 +38,4 @@ class TTSProvider(TTSProviderBase):
file_to_save = open(output_file, "wb") file_to_save = open(output_file, "wb")
file_to_save.write(data) file_to_save.write(data)
except Exception as e: except Exception as e:
raise Exception(f"{__name__} error: {e}") raise Exception(f"{__name__} error: {e}")
@@ -1,6 +1,5 @@
import hashlib import hashlib
import hmac import hmac
import os
import time import time
import uuid import uuid
import json import json
@@ -121,12 +120,6 @@ class TTSProvider(TTSProviderBase):
msg = msg.encode("utf-8") msg = msg.encode("utf-8")
return hmac.new(key, msg, hashlib.sha256).digest() return hmac.new(key, msg, hashlib.sha256).digest()
def generate_filename(self, extension=".wav"):
return os.path.join(
self.output_file,
f"tts-{datetime.now().date()}@{uuid.uuid4().hex}{extension}",
)
async def text_to_speak(self, text, output_file): async def text_to_speak(self, text, output_file):
# 构建请求体 # 构建请求体
request_json = { request_json = {
@@ -82,4 +82,4 @@ class TTSProvider(TTSProviderBase):
except Exception as e: except Exception as e:
print("error:", e) print("error:", e)
raise Exception(f"{__name__}: TTS请求失败") raise Exception(f"{__name__}: TTS请求失败")
@@ -12,13 +12,12 @@ logger = setup_logging()
class VADProvider(VADProviderBase): class VADProvider(VADProviderBase):
def __init__(self, config): def __init__(self, config):
logger.bind(tag=TAG).info("SileroVAD", config) logger.bind(tag=TAG).info("SileroVAD", config)
self.model, self.utils = torch.hub.load( self.model, _ = torch.hub.load(
repo_or_dir=config["model_dir"], repo_or_dir=config["model_dir"],
source="local", source="local",
model="silero_vad", model="silero_vad",
force_reload=False, force_reload=False,
) )
(get_speech_timestamps, _, _, _, _) = self.utils
self.decoder = opuslib_next.Decoder(16000, 1) self.decoder = opuslib_next.Decoder(16000, 1)
@@ -53,7 +52,7 @@ class VADProvider(VADProviderBase):
speech_prob = self.model(audio_tensor, 16000).item() speech_prob = self.model(audio_tensor, 16000).item()
client_have_voice = speech_prob >= self.vad_threshold client_have_voice = speech_prob >= self.vad_threshold
# 如果之前有声音,但本次没有声音,且与上次有声音的时间已经超过了静默阈值,则认为已经说完一句话 # 如果之前有声音,但本次没有声音,且与上次有声音的时间已经超过了静默阈值,则认为已经说完一句话
if conn.client_have_voice and not client_have_voice: if conn.client_have_voice and not client_have_voice:
stop_duration = ( stop_duration = (
time.time() * 1000 - conn.client_have_voice_last_time time.time() * 1000 - conn.client_have_voice_last_time
@@ -0,0 +1,12 @@
from abc import ABC, abstractmethod
from config.logger import setup_logging
TAG = __name__
logger = setup_logging()
class VLLMProviderBase(ABC):
@abstractmethod
def response(self, question, base64_image):
"""VLLM response generator"""
pass
@@ -0,0 +1,63 @@
import openai
import json
from config.logger import setup_logging
from core.utils.util import check_model_key
from core.providers.vllm.base import VLLMProviderBase
TAG = __name__
logger = setup_logging()
class VLLMProvider(VLLMProviderBase):
def __init__(self, config):
self.model_name = config.get("model_name")
self.api_key = config.get("api_key")
if "base_url" in config:
self.base_url = config.get("base_url")
else:
self.base_url = config.get("url")
param_defaults = {
"max_tokens": (500, int),
"temperature": (0.7, lambda x: round(float(x), 1)),
"top_p": (1.0, lambda x: round(float(x), 1)),
}
for param, (default, converter) in param_defaults.items():
value = config.get(param)
try:
setattr(
self,
param,
converter(value) if value not in (None, "") else default,
)
except (ValueError, TypeError):
setattr(self, param, default)
check_model_key("VLLM", self.api_key)
self.client = openai.OpenAI(api_key=self.api_key, base_url=self.base_url)
def response(self, question, base64_image):
try:
messages = [
{
"role": "user",
"content": [
{"type": "text", "text": question},
{
"type": "image_url",
"image_url": {"url": f"{base64_image}"},
},
],
}
]
response = self.client.chat.completions.create(
model=self.model_name, messages=messages, stream=False
)
return response.choices[0].message.content
except Exception as e:
logger.bind(tag=TAG).error(f"Error in response generation: {e}")
raise
+43
View File
@@ -0,0 +1,43 @@
import jwt
import time
from datetime import datetime, timedelta, timezone
from typing import Optional, Tuple
class AuthToken:
def __init__(self, secret_key: str):
self.secret_key = secret_key
def generate_token(self, device_id: str) -> str:
"""
生成JWT token
:param device_id: 设备ID
:return: JWT token字符串
"""
# 设置过期时间为1小时后
expire_time = datetime.now(timezone.utc) + timedelta(hours=1)
# 创建payload
payload = {"device_id": device_id, "exp": expire_time.timestamp()}
# 使用JWT进行编码
token = jwt.encode(payload, self.secret_key, algorithm="HS256")
return token
def verify_token(self, token: str) -> Tuple[bool, Optional[str]]:
"""
验证token
:param token: JWT token字符串
:return: (是否有效, 设备ID)
"""
try:
# 解码token
payload = jwt.decode(token, self.secret_key, algorithms=["HS256"])
# 检查是否过期
if payload["exp"] < time.time():
return False, None
return True, payload["device_id"]
except jwt.InvalidTokenError:
return False, None
@@ -0,0 +1,128 @@
from typing import Dict, Any
from config.logger import setup_logging
from core.utils import tts, llm, intent, memory, vad, asr
TAG = __name__
logger = setup_logging()
def initialize_modules(
logger,
config: Dict[str, Any],
init_vad=False,
init_asr=False,
init_llm=False,
init_tts=False,
init_memory=False,
init_intent=False,
) -> Dict[str, Any]:
"""
初始化所有模块组件
Args:
config: 配置字典
Returns:
Dict[str, Any]: 包含所有初始化后的模块的字典
"""
modules = {}
# 初始化TTS模块
if init_tts:
select_tts_module = config["selected_module"]["TTS"]
modules["tts"] = initialize_tts(config)
logger.bind(tag=TAG).info(f"初始化组件: tts成功 {select_tts_module}")
# 初始化LLM模块
if init_llm:
select_llm_module = config["selected_module"]["LLM"]
llm_type = (
select_llm_module
if "type" not in config["LLM"][select_llm_module]
else config["LLM"][select_llm_module]["type"]
)
modules["llm"] = llm.create_instance(
llm_type,
config["LLM"][select_llm_module],
)
logger.bind(tag=TAG).info(f"初始化组件: llm成功 {select_llm_module}")
# 初始化Intent模块
if init_intent:
select_intent_module = config["selected_module"]["Intent"]
intent_type = (
select_intent_module
if "type" not in config["Intent"][select_intent_module]
else config["Intent"][select_intent_module]["type"]
)
modules["intent"] = intent.create_instance(
intent_type,
config["Intent"][select_intent_module],
)
logger.bind(tag=TAG).info(f"初始化组件: intent成功 {select_intent_module}")
# 初始化Memory模块
if init_memory:
select_memory_module = config["selected_module"]["Memory"]
memory_type = (
select_memory_module
if "type" not in config["Memory"][select_memory_module]
else config["Memory"][select_memory_module]["type"]
)
modules["memory"] = memory.create_instance(
memory_type,
config["Memory"][select_memory_module],
config.get("summaryMemory", None),
)
logger.bind(tag=TAG).info(f"初始化组件: memory成功 {select_memory_module}")
# 初始化VAD模块
if init_vad:
select_vad_module = config["selected_module"]["VAD"]
vad_type = (
select_vad_module
if "type" not in config["VAD"][select_vad_module]
else config["VAD"][select_vad_module]["type"]
)
modules["vad"] = vad.create_instance(
vad_type,
config["VAD"][select_vad_module],
)
logger.bind(tag=TAG).info(f"初始化组件: vad成功 {select_vad_module}")
# 初始化ASR模块
if init_asr:
select_asr_module = config["selected_module"]["ASR"]
modules["asr"] = initialize_asr(config)
logger.bind(tag=TAG).info(f"初始化组件: asr成功 {select_asr_module}")
return modules
def initialize_tts(config):
select_tts_module = config["selected_module"]["TTS"]
tts_type = (
select_tts_module
if "type" not in config["TTS"][select_tts_module]
else config["TTS"][select_tts_module]["type"]
)
new_tts = tts.create_instance(
tts_type,
config["TTS"][select_tts_module],
str(config.get("delete_audio", True)).lower() in ("true", "1", "yes"),
)
return new_tts
def initialize_asr(config):
select_asr_module = config["selected_module"]["ASR"]
asr_type = (
select_asr_module
if "type" not in config["ASR"][select_asr_module]
else config["ASR"][select_asr_module]["type"]
)
new_asr = asr.create_instance(
asr_type,
config["ASR"][select_asr_module],
str(config.get("delete_audio", True)).lower() in ("true", "1", "yes"),
)
return new_asr
@@ -0,0 +1,136 @@
"""
Opus编码工具类
将PCM音频数据编码为Opus格式
"""
import logging
import traceback
import numpy as np
from typing import List, Optional
from opuslib_next import Encoder
from opuslib_next import constants
class OpusEncoderUtils:
"""PCM到Opus的编码器"""
def __init__(self, sample_rate: int, channels: int, frame_size_ms: int):
"""
初始化Opus编码器
Args:
sample_rate: 采样率 (Hz)
channels: 通道数 (1=单声道, 2=立体声)
frame_size_ms: 帧大小 (毫秒)
"""
self.sample_rate = sample_rate
self.channels = channels
self.frame_size_ms = frame_size_ms
# 计算每帧样本数 = 采样率 * 帧大小(毫秒) / 1000
self.frame_size = (sample_rate * frame_size_ms) // 1000
# 总帧大小 = 每帧样本数 * 通道数
self.total_frame_size = self.frame_size * channels
# 比特率和复杂度设置
self.bitrate = 24000 # bps
self.complexity = 10 # 最高质量
# 缓冲区初始化为空
self.buffer = np.array([], dtype=np.int16)
try:
# 创建Opus编码器
self.encoder = Encoder(
sample_rate, channels, constants.APPLICATION_AUDIO # 音频优化模式
)
self.encoder.bitrate = self.bitrate
self.encoder.complexity = self.complexity
self.encoder.signal = constants.SIGNAL_VOICE # 语音信号优化
except Exception as e:
logging.error(f"初始化Opus编码器失败: {e}")
raise RuntimeError("初始化失败") from e
def reset_state(self):
"""重置编码器状态"""
self.encoder.reset_state()
self.buffer = np.array([], dtype=np.int16)
def encode_pcm_to_opus(self, pcm_data: bytes, end_of_stream: bool) -> List[bytes]:
"""
将PCM数据编码为Opus格式
Args:
pcm_data: PCM字节数据
end_of_stream: 是否为流的结束
Returns:
Opus数据包列表
"""
# 将字节数据转换为short数组
new_samples = self._convert_bytes_to_shorts(pcm_data)
# 校验PCM数据
self._validate_pcm_data(new_samples)
# 将新数据追加到缓冲区
self.buffer = np.append(self.buffer, new_samples)
opus_packets = []
offset = 0
# 处理所有完整帧
while offset <= len(self.buffer) - self.total_frame_size:
frame = self.buffer[offset : offset + self.total_frame_size]
output = self._encode(frame)
if output:
opus_packets.append(output)
offset += self.total_frame_size
# 保留未处理的样本
self.buffer = self.buffer[offset:]
# 流结束时处理剩余数据
if end_of_stream and len(self.buffer) > 0:
# 创建最后一帧并用0填充
last_frame = np.zeros(self.total_frame_size, dtype=np.int16)
last_frame[: len(self.buffer)] = self.buffer
output = self._encode(last_frame)
if output:
opus_packets.append(output)
self.buffer = np.array([], dtype=np.int16)
return opus_packets
def _encode(self, frame: np.ndarray) -> Optional[bytes]:
"""编码一帧音频数据"""
try:
# 将numpy数组转换为bytes
frame_bytes = frame.tobytes()
# opuslib要求输入字节数必须是channels*2的倍数
encoded = self.encoder.encode(frame_bytes, self.frame_size)
return encoded
except Exception as e:
logging.error(f"Opus编码失败: {e}")
traceback.print_exc()
return None
def _convert_bytes_to_shorts(self, bytes_data: bytes) -> np.ndarray:
"""将字节数组转换为short数组 (16位PCM)"""
# 假设输入是小端字节序的16位PCM
return np.frombuffer(bytes_data, dtype=np.int16)
def _validate_pcm_data(self, pcm_shorts: np.ndarray) -> None:
"""验证PCM数据是否有效"""
# 16位PCM数据范围是 -32768 到 32767
if np.any((pcm_shorts < -32768) | (pcm_shorts > 32767)):
invalid_samples = pcm_shorts[(pcm_shorts < -32768) | (pcm_shorts > 32767)]
logging.warning(f"发现无效PCM样本: {invalid_samples[:5]}...")
# 在实际应用中可以选择裁剪而不是抛出异常
# np.clip(pcm_shorts, -32768, 32767, out=pcm_shorts)
def close(self):
"""关闭编码器并释放资源"""
# opuslib没有明确的关闭方法,Python的垃圾回收会处理
pass
@@ -0,0 +1,34 @@
def get_string_no_punctuation_or_emoji(s):
"""去除字符串首尾的空格、标点符号和表情符号"""
chars = list(s)
# 处理开头的字符
start = 0
while start < len(chars) and is_punctuation_or_emoji(chars[start]):
start += 1
# 处理结尾的字符
end = len(chars) - 1
while end >= start and is_punctuation_or_emoji(chars[end]):
end -= 1
return ''.join(chars[start:end + 1])
def is_punctuation_or_emoji(char):
"""检查字符是否为空格、指定标点或表情符号"""
# 定义需要去除的中英文标点(包括全角/半角)
punctuation_set = {
'', ',', # 中文逗号 + 英文逗号
'', '.', # 中文句号 + 英文句号
'', '!', # 中文感叹号 + 英文感叹号
'-', '', # 英文连字符 + 中文全角横线
'' # 中文顿号
}
if char.isspace() or char in punctuation_set:
return True
# 检查表情符号(保留原有逻辑)
code_point = ord(char)
emoji_ranges = [
(0x1F600, 0x1F64F), (0x1F300, 0x1F5FF),
(0x1F680, 0x1F6FF), (0x1F900, 0x1F9FF),
(0x1FA70, 0x1FAFF), (0x2600, 0x26FF),
(0x2700, 0x27BF)
]
return any(start <= code_point <= end for start, end in emoji_ranges)
+52 -113
View File
@@ -7,8 +7,6 @@ import numpy as np
import requests import requests
import opuslib_next import opuslib_next
from pydub import AudioSegment from pydub import AudioSegment
from typing import Dict, Any
from core.utils import tts, llm, intent, memory, vad, asr
import copy import copy
TAG = __name__ TAG = __name__
@@ -245,116 +243,6 @@ def extract_json_from_string(input_string):
return None return None
def initialize_modules(
logger,
config: Dict[str, Any],
init_vad=False,
init_asr=False,
init_llm=False,
init_tts=False,
init_memory=False,
init_intent=False,
) -> Dict[str, Any]:
"""
初始化所有模块组件
Args:
config: 配置字典
Returns:
Dict[str, Any]: 包含所有初始化后的模块的字典
"""
modules = {}
# 初始化TTS模块
if init_tts:
select_tts_module = config["selected_module"]["TTS"]
tts_type = (
select_tts_module
if "type" not in config["TTS"][select_tts_module]
else config["TTS"][select_tts_module]["type"]
)
modules["tts"] = tts.create_instance(
tts_type,
config["TTS"][select_tts_module],
str(config.get("delete_audio", True)).lower() in ("true", "1", "yes"),
)
logger.bind(tag=TAG).info(f"初始化组件: tts成功 {select_tts_module}")
# 初始化LLM模块
if init_llm:
select_llm_module = config["selected_module"]["LLM"]
llm_type = (
select_llm_module
if "type" not in config["LLM"][select_llm_module]
else config["LLM"][select_llm_module]["type"]
)
modules["llm"] = llm.create_instance(
llm_type,
config["LLM"][select_llm_module],
)
logger.bind(tag=TAG).info(f"初始化组件: llm成功 {select_llm_module}")
# 初始化Intent模块
if init_intent:
select_intent_module = config["selected_module"]["Intent"]
intent_type = (
select_intent_module
if "type" not in config["Intent"][select_intent_module]
else config["Intent"][select_intent_module]["type"]
)
modules["intent"] = intent.create_instance(
intent_type,
config["Intent"][select_intent_module],
)
logger.bind(tag=TAG).info(f"初始化组件: intent成功 {select_intent_module}")
# 初始化Memory模块
if init_memory:
select_memory_module = config["selected_module"]["Memory"]
memory_type = (
select_memory_module
if "type" not in config["Memory"][select_memory_module]
else config["Memory"][select_memory_module]["type"]
)
modules["memory"] = memory.create_instance(
memory_type,
config["Memory"][select_memory_module],
config.get("summaryMemory", None),
)
logger.bind(tag=TAG).info(f"初始化组件: memory成功 {select_memory_module}")
# 初始化VAD模块
if init_vad:
select_vad_module = config["selected_module"]["VAD"]
vad_type = (
select_vad_module
if "type" not in config["VAD"][select_vad_module]
else config["VAD"][select_vad_module]["type"]
)
modules["vad"] = vad.create_instance(
vad_type,
config["VAD"][select_vad_module],
)
logger.bind(tag=TAG).info(f"初始化组件: vad成功 {select_vad_module}")
# 初始化ASR模块
if init_asr:
select_asr_module = config["selected_module"]["ASR"]
asr_type = (
select_asr_module
if "type" not in config["ASR"][select_asr_module]
else config["ASR"][select_asr_module]["type"]
)
modules["asr"] = asr.create_instance(
asr_type,
config["ASR"][select_asr_module],
str(config.get("delete_audio", True)).lower() in ("true", "1", "yes"),
)
logger.bind(tag=TAG).info(f"初始化组件: asr成功 {select_asr_module}")
return modules
def analyze_emotion(text): def analyze_emotion(text):
""" """
分析文本情感并返回对应的emoji名称支持中英文 分析文本情感并返回对应的emoji名称支持中英文
@@ -882,7 +770,10 @@ def audio_to_data(audio_file_path, is_opus=True):
# 获取原始PCM数据(16位小端) # 获取原始PCM数据(16位小端)
raw_data = audio.raw_data raw_data = audio.raw_data
return pcm_to_data(raw_data, is_opus), duration
def pcm_to_data(raw_data, is_opus=True):
# 初始化Opus编码器 # 初始化Opus编码器
encoder = opuslib_next.Encoder(16000, 1, opuslib_next.APPLICATION_AUDIO) encoder = opuslib_next.Encoder(16000, 1, opuslib_next.APPLICATION_AUDIO)
@@ -910,7 +801,7 @@ def audio_to_data(audio_file_path, is_opus=True):
datas.append(frame_data) datas.append(frame_data)
return datas, duration return datas
def check_vad_update(before_config, new_config): def check_vad_update(before_config, new_config):
@@ -991,3 +882,51 @@ def filter_sensitive_info(config: dict) -> dict:
return filtered return filtered
return _filter_dict(copy.deepcopy(config)) return _filter_dict(copy.deepcopy(config))
def get_vision_url(config: dict) -> str:
"""获取 vision URL
Args:
config: 配置字典
Returns:
str: vision URL
"""
server_config = config["server"]
vision_explain = server_config.get("vision_explain", "")
if "你的" in vision_explain:
local_ip = get_local_ip()
port = int(server_config.get("http_port", 8003))
vision_explain = f"http://{local_ip}:{port}/mcp/vision/explain"
return vision_explain
def is_valid_image_file(file_data: bytes) -> bool:
"""
检查文件数据是否为有效的图片格式
Args:
file_data: 文件的二进制数据
Returns:
bool: 如果是有效的图片格式返回True否则返回False
"""
# 常见图片格式的魔数(文件头)
image_signatures = {
b"\xff\xd8\xff": "JPEG",
b"\x89PNG\r\n\x1a\n": "PNG",
b"GIF87a": "GIF",
b"GIF89a": "GIF",
b"BM": "BMP",
b"II*\x00": "TIFF",
b"MM\x00*": "TIFF",
b"RIFF": "WEBP",
}
# 检查文件头是否匹配任何已知的图片格式
for signature in image_signatures:
if file_data.startswith(signature):
return True
return False
+23
View File
@@ -0,0 +1,23 @@
import os
import sys
# 添加项目根目录到Python路径
current_dir = os.path.dirname(os.path.abspath(__file__))
project_root = os.path.abspath(os.path.join(current_dir, "..", ".."))
sys.path.insert(0, project_root)
from config.logger import setup_logging
import importlib
logger = setup_logging()
def create_instance(class_name, *args, **kwargs):
# 创建LLM实例
if os.path.exists(os.path.join("core", "providers", "vllm", f"{class_name}.py")):
lib_name = f"core.providers.vllm.{class_name}"
if lib_name not in sys.modules:
sys.modules[lib_name] = importlib.import_module(f"{lib_name}")
return sys.modules[lib_name].VLLMProvider(*args, **kwargs)
raise ValueError(f"不支持的VLLM类型: {class_name},请检查该配置的type是否设置正确")
+5 -7
View File
@@ -2,8 +2,9 @@ import asyncio
import websockets import websockets
from config.logger import setup_logging from config.logger import setup_logging
from core.connection import ConnectionHandler from core.connection import ConnectionHandler
from core.utils.util import initialize_modules, check_vad_update, check_asr_update
from config.config_loader import get_config_from_api from config.config_loader import get_config_from_api
from core.utils.modules_initialize import initialize_modules
from core.utils.util import check_vad_update, check_asr_update
TAG = __name__ TAG = __name__
@@ -19,16 +20,16 @@ class WebSocketServer:
"VAD" in self.config["selected_module"], "VAD" in self.config["selected_module"],
"ASR" in self.config["selected_module"], "ASR" in self.config["selected_module"],
"LLM" in self.config["selected_module"], "LLM" in self.config["selected_module"],
"TTS" in self.config["selected_module"], False,
"Memory" in self.config["selected_module"], "Memory" in self.config["selected_module"],
"Intent" in self.config["selected_module"], "Intent" in self.config["selected_module"],
) )
self._vad = modules["vad"] if "vad" in modules else None self._vad = modules["vad"] if "vad" in modules else None
self._asr = modules["asr"] if "asr" in modules else None self._asr = modules["asr"] if "asr" in modules else None
self._tts = modules["tts"] if "tts" in modules else None
self._llm = modules["llm"] if "llm" in modules else None self._llm = modules["llm"] if "llm" in modules else None
self._intent = modules["intent"] if "intent" in modules else None self._intent = modules["intent"] if "intent" in modules else None
self._memory = modules["memory"] if "memory" in modules else None self._memory = modules["memory"] if "memory" in modules else None
self.active_connections = set() self.active_connections = set()
async def start(self): async def start(self):
@@ -49,7 +50,6 @@ class WebSocketServer:
self._vad, self._vad,
self._asr, self._asr,
self._llm, self._llm,
self._tts,
self._memory, self._memory,
self._intent, self._intent,
self, # 传入server实例 self, # 传入server实例
@@ -98,7 +98,7 @@ class WebSocketServer:
update_vad, update_vad,
update_asr, update_asr,
"LLM" in new_config["selected_module"], "LLM" in new_config["selected_module"],
"TTS" in new_config["selected_module"], False,
"Memory" in new_config["selected_module"], "Memory" in new_config["selected_module"],
"Intent" in new_config["selected_module"], "Intent" in new_config["selected_module"],
) )
@@ -108,8 +108,6 @@ class WebSocketServer:
self._vad = modules["vad"] self._vad = modules["vad"]
if "asr" in modules: if "asr" in modules:
self._asr = modules["asr"] self._asr = modules["asr"]
if "tts" in modules:
self._tts = modules["tts"]
if "llm" in modules: if "llm" in modules:
self._llm = modules["llm"] self._llm = modules["llm"]
if "intent" in modules: if "intent" in modules:
+2 -2
View File
@@ -13,8 +13,8 @@ services:
ports: ports:
# ws服务端 # ws服务端
- "8000:8000" - "8000:8000"
# ota服务端 # http服务的端口,用于简单OTA接口(单服务部署),以及视觉分析接口
- "8002:8002" - "8003:8003"
volumes: volumes:
# 配置文件目录 # 配置文件目录
- ./data:/opt/xiaozhi-esp32-server/data - ./data:/opt/xiaozhi-esp32-server/data
@@ -15,6 +15,8 @@ services:
ports: ports:
# ws服务端 # ws服务端
- "8000:8000" - "8000:8000"
# http服务的端口,用于视觉分析接口
- "8003:8003"
security_opt: security_opt:
- seccomp:unconfined - seccomp:unconfined
environment: environment:
@@ -15,15 +15,26 @@ async def _get_device_status(conn, device_name, device_type, property_name):
return status return status
async def _set_device_property(conn, device_name, device_type, method_name, property_name, new_value=None, action=None, step=10): async def _set_device_property(
conn,
device_name,
device_type,
method_name,
property_name,
new_value=None,
action=None,
step=10,
):
"""设置设备属性""" """设置设备属性"""
current_value = await _get_device_status(conn, device_name, device_type, property_name) current_value = await _get_device_status(
conn, device_name, device_type, property_name
)
if action == 'raise': if action == "raise":
current_value += step current_value += step
elif action == 'lower': elif action == "lower":
current_value -= step current_value -= step
elif action == 'set': elif action == "set":
if new_value is None: if new_value is None:
raise Exception(f"缺少{property_name}参数") raise Exception(f"缺少{property_name}参数")
current_value = new_value current_value = new_value
@@ -37,8 +48,7 @@ async def _set_device_property(conn, device_name, device_type, method_name, prop
def _handle_device_action(conn, func, success_message, error_message, *args, **kwargs): def _handle_device_action(conn, func, success_message, error_message, *args, **kwargs):
"""处理设备操作的通用函数""" """处理设备操作的通用函数"""
future = asyncio.run_coroutine_threadsafe( future = asyncio.run_coroutine_threadsafe(func(conn, *args, **kwargs), conn.loop)
func(conn, *args, **kwargs), conn.loop)
try: try:
result = future.result() result = future.result()
logger.bind(tag=TAG).info(f"{success_message}: {result}") logger.bind(tag=TAG).info(f"{success_message}: {result}")
@@ -75,26 +85,41 @@ handle_device_function_desc = {
"device_type": { "device_type": {
"type": "string", "type": "string",
"description": "设备类型,**严格限定为Speaker(音量)或Screen(亮度)**,其他设备类型禁止调用此函数", "description": "设备类型,**严格限定为Speaker(音量)或Screen(亮度)**,其他设备类型禁止调用此函数",
"enum": ["Speaker", "Screen"] "enum": ["Speaker", "Screen"],
}, },
"action": { "action": {
"type": "string", "type": "string",
"description": "动作名称,可选值:get(获取),set(设置),raise(提高),lower(降低)" "description": "动作名称,可选值:get(获取),set(设置),raise(提高),lower(降低)",
}, },
"value": { "value": {
"type": "integer", "type": "integer",
"description": "值大小,可选值:0-100之间的整数" "description": "值大小,可选值:0-100之间的整数",
} },
}, },
"required": ["device_type", "action"] "required": ["device_type", "action"],
} },
} },
} }
@register_function('handle_speaker_volume_or_screen_brightness', handle_device_function_desc, ToolType.IOT_CTL) @register_function(
def handle_speaker_volume_or_screen_brightness(conn, device_type: str, action: str, value: int = None): "handle_speaker_volume_or_screen_brightness",
handle_device_function_desc,
ToolType.IOT_CTL,
)
def handle_speaker_volume_or_screen_brightness(
conn, device_type: str, action: str, value: int = None
):
# 检查value是否为中文值
if (
value is not None
and isinstance(value, str)
and any("\u4e00" <= char <= "\u9fff" for char in str(value))
):
raise Exception(
f"请直接告诉我要将{'音量' if device_type=='Speaker' else '亮度'}调整成多少"
)
if device_type == "Speaker": if device_type == "Speaker":
method_name, property_name, device_name = "SetVolume", "volume", "音量" method_name, property_name, device_name = "SetVolume", "volume", "音量"
elif device_type == "Screen": elif device_type == "Screen":
@@ -108,13 +133,25 @@ def handle_speaker_volume_or_screen_brightness(conn, device_type: str, action: s
if action == "get": if action == "get":
# get # get
return _handle_device_action( return _handle_device_action(
conn, _get_device_status, f"当前{device_name}", f"获取{device_name}失败", conn,
device_name=device_name, device_type=device_type, property_name=property_name, _get_device_status,
f"当前{device_name}",
f"获取{device_name}失败",
device_name=device_name,
device_type=device_type,
property_name=property_name,
) )
else: else:
# set, raise, lower # set, raise, lower
return _handle_device_action( return _handle_device_action(
conn, _set_device_property, f"{device_name}已调整到", f"{device_name}调整失败", conn,
device_name=device_name, device_type=device_type, method_name=method_name, _set_device_property,
property_name=property_name, new_value=value, action=action f"{device_name}已调整到",
f"{device_name}调整失败",
device_name=device_name,
device_type=device_type,
method_name=method_name,
property_name=property_name,
new_value=value,
action=action,
) )
@@ -11,6 +11,7 @@ from core.utils import p3
from core.handle.sendAudioHandle import send_stt_message from core.handle.sendAudioHandle import send_stt_message
from plugins_func.register import register_function, ToolType, ActionResponse, Action from plugins_func.register import register_function, ToolType, ActionResponse, Action
from core.utils.dialogue import Message from core.utils.dialogue import Message
from core.providers.tts.dto.dto import TTSMessageDTO, SentenceType, ContentType
TAG = __name__ TAG = __name__
@@ -216,23 +217,37 @@ async def play_local_music(conn, specific_file=None):
text = _get_random_play_prompt(selected_music) text = _get_random_play_prompt(selected_music)
await send_stt_message(conn, text) await send_stt_message(conn, text)
conn.dialogue.put(Message(role="assistant", content=text)) conn.dialogue.put(Message(role="assistant", content=text))
conn.tts_first_text_index = 0
conn.tts_last_text_index = 0
tts_file = await asyncio.to_thread(conn.tts.to_tts, text) conn.tts.tts_text_queue.put(
if tts_file is not None and os.path.exists(tts_file): TTSMessageDTO(
conn.tts_last_text_index = 1 sentence_id=conn.sentence_id,
opus_packets, _ = conn.tts.audio_to_opus_data(tts_file) sentence_type=SentenceType.FIRST,
conn.audio_play_queue.put((opus_packets, None, 0)) content_type=ContentType.ACTION,
os.remove(tts_file) )
)
conn.llm_finish_task = True conn.tts.tts_text_queue.put(
TTSMessageDTO(
if music_path.endswith(".p3"): sentence_id=conn.sentence_id,
opus_packets, _ = p3.decode_opus_from_file(music_path) sentence_type=SentenceType.MIDDLE,
else: content_type=ContentType.TEXT,
opus_packets, _ = conn.tts.audio_to_opus_data(music_path) content_detail=text,
conn.audio_play_queue.put((opus_packets, None, conn.tts_last_text_index)) )
)
conn.tts.tts_text_queue.put(
TTSMessageDTO(
sentence_id=conn.sentence_id,
sentence_type=SentenceType.MIDDLE,
content_type=ContentType.FILE,
content_file=music_path,
)
)
conn.tts.tts_text_queue.put(
TTSMessageDTO(
sentence_id=conn.sentence_id,
sentence_type=SentenceType.LAST,
content_type=ContentType.ACTION,
)
)
except Exception as e: except Exception as e:
conn.logger.bind(tag=TAG).error(f"播放音乐失败: {str(e)}") conn.logger.bind(tag=TAG).error(f"播放音乐失败: {str(e)}")
+1
View File
@@ -31,3 +31,4 @@ chardet==5.2.0
aioconsole==0.8.1 aioconsole==0.8.1
markitdown==0.1.1 markitdown==0.1.1
mcp-proxy==0.6.0 mcp-proxy==0.6.0
PyJWT==2.8.0
+92 -47
View File
@@ -14,7 +14,7 @@
} }
.container { .container {
max-width: 800px; max-width: 1000px;
margin: 0 auto; margin: 0 auto;
background-color: white; background-color: white;
border-radius: 10px; border-radius: 10px;
@@ -482,9 +482,10 @@
</span> </span>
</h2> </h2>
<div class="connection-controls"> <div class="connection-controls">
<input type="text" id="otaUrl" value="http://127.0.0.1:8002/xiaozhi/ota/" placeholder="OTA服务器地址" /> <input type="text" id="otaUrl" value="http://127.0.0.1:8002/xiaozhi/ota/"
placeholder="OTA服务器地址,如:http://127.0.0.1:8002/xiaozhi/ota/" />
<input type="text" id="serverUrl" value="ws://127.0.0.1:8000/xiaozhi/v1/" <input type="text" id="serverUrl" value="ws://127.0.0.1:8000/xiaozhi/v1/"
placeholder="WebSocket服务器地址" /> placeholder="WebSocket服务器地址,如:ws://127.0.0.1:8000/xiaozhi/v1/" />
<button id="connectButton">连接</button> <button id="connectButton">连接</button>
<button id="authTestButton">测试认证</button> <button id="authTestButton">测试认证</button>
</div> </div>
@@ -802,7 +803,10 @@
if (frameData && frameData.length > 0) { if (frameData && frameData.length > 0) {
// 转换为Float32 // 转换为Float32
const floatData = convertInt16ToFloat32(frameData); const floatData = convertInt16ToFloat32(frameData);
decodedSamples.push(...floatData); // 使用循环替代展开运算符
for (let i = 0; i < floatData.length; i++) {
decodedSamples.push(floatData[i]);
}
} }
} catch (error) { } catch (error) {
log("Opus解码失败: " + error.message, 'error'); log("Opus解码失败: " + error.message, 'error');
@@ -810,8 +814,10 @@
} }
if (decodedSamples.length > 0) { if (decodedSamples.length > 0) {
// 添加到解码队列 // 使用循环替代展开运算符
this.queue.push(...decodedSamples); for (let i = 0; i < decodedSamples.length; i++) {
this.queue.push(decodedSamples[i]);
}
this.totalSamples += decodedSamples.length; this.totalSamples += decodedSamples.length;
// 如果累积了至少0.2秒的音频,开始播放 // 如果累积了至少0.2秒的音频,开始播放
@@ -867,35 +873,39 @@
this.source = null; this.source = null;
this.playing = false; this.playing = false;
// 如果队列中还有数据或者缓冲区有新数据,继续播放 // 使用setTimeout避免递归调用
if (this.queue.length > 0) { setTimeout(() => {
setTimeout(() => this.startPlaying(), 10); // 如果队列中还有数据,继续播放
} else if (audioBufferQueue.length > 0) { if (this.queue.length > 0) {
// 缓冲区有新数据,进行解码 this.startPlaying();
const frames = [...audioBufferQueue]; } else if (audioBufferQueue.length > 0) {
audioBufferQueue = []; // 缓冲区有新数据,进行解码
this.decodeOpusFrames(frames); const frames = [...audioBufferQueue];
} else if (this.endOfStream) { audioBufferQueue = [];
// 流已结束且没有更多数据 this.decodeOpusFrames(frames);
log("音频播放完成", 'info'); } else if (this.endOfStream) {
isAudioPlaying = false; // 流已结束且没有更多数据
streamingContext = null; log("音频播放完成", 'info');
} else { isAudioPlaying = false;
// 等待更多数据 this.endOfStream = false;
setTimeout(() => { streamingContext = null;
// 如果仍然没有新数据,但有更多的包到达 } else {
if (this.queue.length === 0 && audioBufferQueue.length > 0) { // 等待更多数据
const frames = [...audioBufferQueue]; setTimeout(() => {
audioBufferQueue = []; // 如果仍然没有新数据,但有更多的包到达
this.decodeOpusFrames(frames); if (this.queue.length === 0 && audioBufferQueue.length > 0) {
} else if (this.queue.length === 0 && audioBufferQueue.length === 0) { const frames = [...audioBufferQueue];
// 真的没有更多数据了 audioBufferQueue = [];
log("音频播放完成 (超时)", 'info'); this.decodeOpusFrames(frames);
isAudioPlaying = false; } else if (this.queue.length === 0 && audioBufferQueue.length === 0) {
streamingContext = null; // 真的没有更多数据了
} log("音频播放完成 (超时)", 'info');
}, 500); // 500ms超时 isAudioPlaying = false;
} streamingContext = null;
}
}, 500); // 500ms超时
}
}, 10); // 10ms延迟,避免立即递归
}; };
this.source.start(); this.source.start();
@@ -1061,7 +1071,7 @@
} }
} }
// 初始化音频录制和处理 // 初始化音频录制和处理
async function initAudio() { async function initAudio() {
try { try {
// 请求麦克风权限 // 请求麦克风权限
@@ -1306,7 +1316,8 @@
// 先检查OTA状态 // 先检查OTA状态
log('正在检查OTA状态...', 'info'); log('正在检查OTA状态...', 'info');
const otaUrl = document.getElementById('otaUrl').value.trim(); const otaUrl = document.getElementById('otaUrl').value.trim();
localStorage.setItem('otaUrl', otaUrl);
localStorage.setItem('wsUrl', url);
try { try {
const otaResponse = await fetch(otaUrl, { const otaResponse = await fetch(otaUrl, {
method: 'POST', method: 'POST',
@@ -1473,6 +1484,23 @@
if (message.text && message.text !== '😊') { if (message.text && message.text !== '😊') {
addMessage(message.text); addMessage(message.text);
} }
}else if (message.type === 'mcp') {
const payload = message.payload || {};
log(`服务器下发: ${JSON.stringify(message)}`, 'info');
if (payload) {
// 模拟小智客户端行为
if(payload.method === 'tools/list'){
const replay_message = JSON.stringify({"session_id":"","type":"mcp","payload":{"jsonrpc":"2.0","id":2,"result":{"tools":[{"name":"self.get_device_status","description":"Provides the real-time information of the device, including the current status of the audio speaker, screen, battery, network, etc.\nUse this tool for: \n1. Answering questions about current condition (e.g. what is the current volume of the audio speaker?)\n2. As the first step to control the device (e.g. turn up / down the volume of the audio speaker, etc.)","inputSchema":{"type":"object","properties":{}}},{"name":"self.audio_speaker.set_volume","description":"Set the volume of the audio speaker. If the current volume is unknown, you must call `self.get_device_status` tool first and then call this tool.","inputSchema":{"type":"object","properties":{"volume":{"type":"integer","minimum":0,"maximum":100}},"required":["volume"]}},{"name":"self.screen.set_brightness","description":"Set the brightness of the screen.","inputSchema":{"type":"object","properties":{"brightness":{"type":"integer","minimum":0,"maximum":100}},"required":["brightness"]}},{"name":"self.screen.set_theme","description":"Set the theme of the screen. The theme can be 'light' or 'dark'.","inputSchema":{"type":"object","properties":{"theme":{"type":"string"}},"required":["theme"]}}]}}})
websocket.send(replay_message);
log(`回复MCP消息: ${replay_message}`, 'info');
} else if(payload.method === 'tools/call'){
// 模拟回复
const replay_message = JSON.stringify({"session_id":"9f261599","type":"mcp","payload":{"jsonrpc":"2.0","id": payload.id,"result":{"content":[{"type":"text","text":"true"}],"isError":false}}})
websocket.send(replay_message);
log(`回复MCP消息: ${replay_message}`, 'info');
}
}
} else { } else {
// 未知消息类型 // 未知消息类型
log(`未知消息类型: ${message.type}`, 'info'); log(`未知消息类型: ${message.type}`, 'info');
@@ -1512,7 +1540,10 @@
device_id: config.deviceId, device_id: config.deviceId,
device_name: config.deviceName, device_name: config.deviceName,
device_mac: config.deviceMac, device_mac: config.deviceMac,
token: config.token token: config.token,
features: {
mcp: true
}
}; };
log('发送hello握手消息', 'info'); log('发送hello握手消息', 'info');
@@ -1563,6 +1594,10 @@
const message = messageInput.value.trim(); const message = messageInput.value.trim();
if (message === '' || !websocket || websocket.readyState !== WebSocket.OPEN) return; if (message === '' || !websocket || websocket.readyState !== WebSocket.OPEN) return;
audioBufferQueue = [];
isAudioBuffering = false;
isAudioPlaying = false;
try { try {
// 直接发送listen消息,不需要重复发送hello // 直接发送listen消息,不需要重复发送hello
const listenMessage = { const listenMessage = {
@@ -1633,6 +1668,16 @@
// 初始更新显示值 // 初始更新显示值
updateDisplayValues(); updateDisplayValues();
const savedOtaUrl = localStorage.getItem('otaUrl');
if (savedOtaUrl) {
document.getElementById('otaUrl').value = savedOtaUrl;
}
const savedWsUrl = localStorage.getItem('wsUrl');
if (savedWsUrl) {
document.getElementById('serverUrl').value = savedWsUrl;
}
// 切换面板显示 // 切换面板显示
toggleButton.addEventListener('click', () => { toggleButton.addEventListener('click', () => {
const isExpanded = configPanel.classList.contains('expanded'); const isExpanded = configPanel.classList.contains('expanded');
@@ -1954,7 +1999,7 @@
this.buffer = new Int16Array(this.frameSize); this.buffer = new Int16Array(this.frameSize);
this.bufferIndex = 0; this.bufferIndex = 0;
this.isRecording = false; this.isRecording = false;
// 监听来自主线程的消息 // 监听来自主线程的消息
this.port.onmessage = (event) => { this.port.onmessage = (event) => {
if (event.data.command === 'start') { if (event.data.command === 'start') {
@@ -1962,7 +2007,7 @@
this.port.postMessage({ type: 'status', status: 'started' }); this.port.postMessage({ type: 'status', status: 'started' });
} else if (event.data.command === 'stop') { } else if (event.data.command === 'stop') {
this.isRecording = false; this.isRecording = false;
// 发送剩余的缓冲区 // 发送剩余的缓冲区
if (this.bufferIndex > 0) { if (this.bufferIndex > 0) {
const finalBuffer = this.buffer.slice(0, this.bufferIndex); const finalBuffer = this.buffer.slice(0, this.bufferIndex);
@@ -1972,18 +2017,18 @@
}); });
this.bufferIndex = 0; this.bufferIndex = 0;
} }
this.port.postMessage({ type: 'status', status: 'stopped' }); this.port.postMessage({ type: 'status', status: 'stopped' });
} }
}; };
} }
process(inputs, outputs, parameters) { process(inputs, outputs, parameters) {
if (!this.isRecording) return true; if (!this.isRecording) return true;
const input = inputs[0][0]; // 获取第一个输入通道 const input = inputs[0][0]; // 获取第一个输入通道
if (!input) return true; if (!input) return true;
// 将浮点采样转换为16位整数并存储 // 将浮点采样转换为16位整数并存储
for (let i = 0; i < input.length; i++) { for (let i = 0; i < input.length; i++) {
if (this.bufferIndex >= this.frameSize) { if (this.bufferIndex >= this.frameSize) {
@@ -1994,15 +2039,15 @@
}); });
this.bufferIndex = 0; this.bufferIndex = 0;
} }
// 转换为16位整数 (-32768到32767) // 转换为16位整数 (-32768到32767)
this.buffer[this.bufferIndex++] = Math.max(-32768, Math.min(32767, Math.floor(input[i] * 32767))); this.buffer[this.bufferIndex++] = Math.max(-32768, Math.min(32767, Math.floor(input[i] * 32767)));
} }
return true; return true;
} }
} }
registerProcessor('audio-recorder-processor', AudioRecorderProcessor); registerProcessor('audio-recorder-processor', AudioRecorderProcessor);
`; `;