Merge pull request #1716 from xinnan-tech/py_test_tts

update:添加阿里云流式asr
This commit is contained in:
欣南科技
2025-07-22 13:27:41 +08:00
committed by GitHub
19 changed files with 1453 additions and 615 deletions
@@ -237,7 +237,7 @@ public interface Constant {
/**
* 版本号
*/
public static final String VERSION = "0.7.1";
public static final String VERSION = "0.7.2";
/**
* 无效固件URL
@@ -0,0 +1,26 @@
-- 添加阿里云流式ASR供应器
delete from `ai_model_provider` where id = 'SYSTEM_ASR_AliyunStreamASR';
INSERT INTO `ai_model_provider` (`id`, `model_type`, `provider_code`, `name`, `fields`, `sort`, `creator`, `create_date`, `updater`, `update_date`) VALUES
('SYSTEM_ASR_AliyunStreamASR', 'ASR', 'aliyun_stream', '阿里云语音识别(流式)', '[{"key":"appkey","label":"应用AppKey","type":"string"},{"key":"token","label":"临时Token","type":"string"},{"key":"access_key_id","label":"AccessKey ID","type":"string"},{"key":"access_key_secret","label":"AccessKey Secret","type":"string"},{"key":"host","label":"服务地址","type":"string"},{"key":"max_sentence_silence","label":"断句检测时间","type":"number"},{"key":"output_dir","label":"输出目录","type":"string"}]', 6, 1, NOW(), 1, NOW());
-- 添加阿里云流式ASR模型配置
delete from `ai_model_config` where id = 'ASR_AliyunStreamASR';
INSERT INTO `ai_model_config` VALUES ('ASR_AliyunStreamASR', 'ASR', 'AliyunStreamASR', '阿里云语音识别(流式)', 0, 1, '{\"type\": \"aliyun_stream\", \"appkey\": \"\", \"token\": \"\", \"access_key_id\": \"\", \"access_key_secret\": \"\", \"host\": \"nls-gateway-cn-shanghai.aliyuncs.com\", \"max_sentence_silence\": 800, \"output_dir\": \"tmp/\"}', NULL, NULL, 8, NULL, NULL, NULL, NULL);
-- 更新阿里云流式ASR配置说明
UPDATE `ai_model_config` SET
`doc_link` = 'https://nls-portal.console.aliyun.com/',
`remark` = '阿里云流式ASR配置说明:
1. 阿里云ASR和阿里云(流式)ASR的区别是:阿里云ASR是一次性识别,阿里云(流式)ASR是实时流式识别
2. 流式ASR具有更低的延迟和更好的实时性,适合语音交互场景
3. 需要在阿里云智能语音交互控制台创建应用并获取认证信息
4. 支持中文实时语音识别,支持标点符号预测和逆文本规范化
5. 需要网络连接,输出文件保存在tmp/目录
申请步骤:
1. 访问 https://nls-portal.console.aliyun.com/ 开通智能语音交互服务
2. 访问 https://nls-portal.console.aliyun.com/applist 创建项目并获取appkey
3. 访问 https://nls-portal.console.aliyun.com/overview 获取临时token(或配置access_key_id和access_key_secret自动获取)
4. 如需动态token管理,建议配置access_key_id和access_key_secret
5. max_sentence_silence参数控制断句检测时间(毫秒),默认800ms
如需了解更多参数配置,请参考:https://help.aliyun.com/zh/isi/developer-reference/real-time-speech-recognition
' WHERE `id` = 'ASR_AliyunStreamASR';
@@ -0,0 +1,56 @@
-- 添加阿里云流式TTS供应器
delete from `ai_model_provider` where id = 'SYSTEM_TTS_AliyunStreamTTS';
INSERT INTO `ai_model_provider` (`id`, `model_type`, `provider_code`, `name`, `fields`, `sort`, `creator`, `create_date`, `updater`, `update_date`) VALUES
('SYSTEM_TTS_AliyunStreamTTS', 'TTS', 'aliyun_stream', '阿里云语音合成(流式)', '[{"key":"appkey","label":"应用AppKey","type":"string"},{"key":"token","label":"临时Token","type":"string"},{"key":"access_key_id","label":"AccessKey ID","type":"string"},{"key":"access_key_secret","label":"AccessKey Secret","type":"string"},{"key":"host","label":"服务地址","type":"string"},{"key":"voice","label":"默认音色","type":"string"},{"key":"format","label":"音频格式","type":"string"},{"key":"sample_rate","label":"采样率","type":"number"},{"key":"volume","label":"音量","type":"number"},{"key":"speech_rate","label":"语速","type":"number"},{"key":"pitch_rate","label":"音调","type":"number"},{"key":"output_dir","label":"输出目录","type":"string"}]', 15, 1, NOW(), 1, NOW());
-- 添加阿里云流式TTS模型配置
delete from `ai_model_config` where id = 'TTS_AliyunStreamTTS';
INSERT INTO `ai_model_config` VALUES ('TTS_AliyunStreamTTS', 'TTS', 'AliyunStreamTTS', '阿里云语音合成(流式)', 0, 1, '{\"type\": \"aliyun_stream\", \"appkey\": \"\", \"token\": \"\", \"access_key_id\": \"\", \"access_key_secret\": \"\", \"host\": \"nls-gateway-cn-beijing.aliyuncs.com\", \"voice\": \"longxiaochun\", \"format\": \"pcm\", \"sample_rate\": 16000, \"volume\": 50, \"speech_rate\": 0, \"pitch_rate\": 0, \"output_dir\": \"tmp/\"}', NULL, NULL, 18, NULL, NULL, NULL, NULL);
-- 更新阿里云流式TTS配置说明
UPDATE `ai_model_config` SET
`doc_link` = 'https://nls-portal.console.aliyun.com/',
`remark` = '阿里云流式TTS配置说明:
1. 阿里云TTS和阿里云(流式)TTS的区别是:阿里云TTS是一次性合成,阿里云(流式)TTS是实时流式合成
2. 流式TTS具有更低的延迟和更好的实时性,适合语音交互场景
3. 需要在阿里云智能语音交互控制台创建应用并获取认证信息
4. 支持CosyVoice大模型音色,音质更加自然
5. 支持实时调节音量、语速、音调等参数
申请步骤:
1. 访问 https://nls-portal.console.aliyun.com/ 开通智能语音交互服务
2. 访问 https://nls-portal.console.aliyun.com/applist 创建项目并获取appkey
3. 访问 https://nls-portal.console.aliyun.com/overview 获取临时token(或配置access_key_id和access_key_secret自动获取)
4. 如需动态token管理,建议配置access_key_id和access_key_secret
5. 可选择北京、上海等不同地域的服务器以优化延迟
6. voice参数支持CosyVoice大模型音色,如longxiaochun、longyueyue等
如需了解更多参数配置,请参考:https://help.aliyun.com/zh/isi/developer-reference/real-time-speech-synthesis
' WHERE `id` = 'TTS_AliyunStreamTTS';
-- 添加阿里云流式TTS音色
delete from `ai_tts_voice` where tts_model_id = 'TTS_AliyunStreamTTS';
-- 温柔女声系列
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0001', 'TTS_AliyunStreamTTS', '龙小淳-温柔姐姐', 'longxiaochun', '中文及中英文混合', NULL, NULL, NULL, NULL, 1, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0002', 'TTS_AliyunStreamTTS', '龙小夏-温柔女声', 'longxiaoxia', '中文及中英文混合', NULL, NULL, NULL, NULL, 2, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0003', 'TTS_AliyunStreamTTS', '龙玫-温柔女声', 'longmei', '中文及中英文混合', NULL, NULL, NULL, NULL, 3, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0004', 'TTS_AliyunStreamTTS', '龙瑰-温柔女声', 'longgui', '中文及中英文混合', NULL, NULL, NULL, NULL, 4, NULL, NULL, NULL, NULL);
-- 御姐女声系列
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0005', 'TTS_AliyunStreamTTS', '龙玉-御姐女声', 'longyu', '中文及中英文混合', NULL, NULL, NULL, NULL, 5, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0006', 'TTS_AliyunStreamTTS', '龙娇-御姐女声', 'longjiao', '中文及中英文混合', NULL, NULL, NULL, NULL, 6, NULL, NULL, NULL, NULL);
-- 男声系列
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0007', 'TTS_AliyunStreamTTS', '龙臣-译制片男声', 'longchen', '中文及中英文混合', NULL, NULL, NULL, NULL, 7, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0008', 'TTS_AliyunStreamTTS', '龙修-青年男声', 'longxiu', '中文及中英文混合', NULL, NULL, NULL, NULL, 8, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0009', 'TTS_AliyunStreamTTS', '龙橙-阳光男声', 'longcheng', '中文及中英文混合', NULL, NULL, NULL, NULL, 9, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0010', 'TTS_AliyunStreamTTS', '龙哲-成熟男声', 'longzhe', '中文及中英文混合', NULL, NULL, NULL, NULL, 10, NULL, NULL, NULL, NULL);
-- 专业播报系列
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0011', 'TTS_AliyunStreamTTS', 'Bella2.0-新闻女声', 'loongbella', '中文及中英文混合', NULL, NULL, NULL, NULL, 11, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0012', 'TTS_AliyunStreamTTS', 'Stella2.0-飒爽女声', 'loongstella', '中文及中英文混合', NULL, NULL, NULL, NULL, 12, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0013', 'TTS_AliyunStreamTTS', '龙书-新闻男声', 'longshu', '中文及中英文混合', NULL, NULL, NULL, NULL, 13, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0014', 'TTS_AliyunStreamTTS', '龙婧-严肃女声', 'longjing', '中文及中英文混合', NULL, NULL, NULL, NULL, 14, NULL, NULL, NULL, NULL);
-- 特色音色系列
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0015', 'TTS_AliyunStreamTTS', '龙奇-活泼童声', 'longqi', '中文及中英文混合', NULL, NULL, NULL, NULL, 15, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0016', 'TTS_AliyunStreamTTS', '龙华-活泼女童', 'longhua', '中文及中英文混合', NULL, NULL, NULL, NULL, 16, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0017', 'TTS_AliyunStreamTTS', '龙无-无厘头男声', 'longwu', '中文及中英文混合', NULL, NULL, NULL, NULL, 17, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0018', 'TTS_AliyunStreamTTS', '龙大锤-幽默男声', 'longdachui', '中文及中英文混合', NULL, NULL, NULL, NULL, 18, NULL, NULL, NULL, NULL);
-- 粤语系列
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0019', 'TTS_AliyunStreamTTS', '龙嘉怡-粤语女声', 'longjiayi', '粤语及粤英混合', NULL, NULL, NULL, NULL, 19, NULL, NULL, NULL, NULL);
INSERT INTO `ai_tts_voice` VALUES ('TTS_AliyunStreamTTS_0020', 'TTS_AliyunStreamTTS', '龙桃-粤语女声', 'longtao', '粤语及粤英混合', NULL, NULL, NULL, NULL, 20, NULL, NULL, NULL, NULL);
@@ -0,0 +1,38 @@
-- OpenAI ASR模型供应器
delete from `ai_model_provider` where id = 'SYSTEM_ASR_OpenaiASR';
INSERT INTO `ai_model_provider` (`id`, `model_type`, `provider_code`, `name`, `fields`, `sort`, `creator`, `create_date`, `updater`, `update_date`) VALUES
('SYSTEM_ASR_OpenaiASR', 'ASR', 'openai', 'OpenAI语音识别', '[{"key": "base_url", "type": "string", "label": "基础URL"}, {"key": "model_name", "type": "string", "label": "模型名称"}, {"key": "api_key", "type": "string", "label": "API密钥"}, {"key": "output_dir", "type": "string", "label": "输出目录"}]', 9, 1, NOW(), 1, NOW());
-- OpenAI ASR模型配置
delete from `ai_model_config` where id = 'ASR_OpenaiASR';
INSERT INTO `ai_model_config` VALUES ('ASR_OpenaiASR', 'ASR', 'OpenaiASR', 'OpenAI语音识别', 0, 1, '{\"type\": \"openai\", \"api_key\": \"\", \"base_url\": \"https://api.openai.com/v1/audio/transcriptions\", \"model_name\": \"gpt-4o-mini-transcribe\", \"output_dir\": \"tmp/\"}', NULL, NULL, 9, NULL, NULL, NULL, NULL);
-- groq ASR模型配置
delete from `ai_model_config` where id = 'ASR_GroqASR';
INSERT INTO `ai_model_config` VALUES ('ASR_GroqASR', 'ASR', 'GroqASR', 'Groq语音识别', 0, 1, '{\"type\": \"openai\", \"api_key\": \"\", \"base_url\": \"https://api.groq.com/openai/v1/audio/transcriptions\", \"model_name\": \"whisper-large-v3-turbo\", \"output_dir\": \"tmp/\"}', NULL, NULL, 10, NULL, NULL, NULL, NULL);
-- 更新OpenAI ASR配置说明
UPDATE `ai_model_config` SET
`doc_link` = 'https://platform.openai.com/docs/api-reference/audio/createTranscription',
`remark` = 'OpenAI ASR配置说明:
1. 需要在OpenAI开放平台创建组织并获取api_key
2. 支持中、英、日、韩等多种语音识别,具体参考文档https://platform.openai.com/docs/guides/speech-to-text
3. 需要网络连接
4. 输出文件保存在tmp/目录
申请步骤:
**OpenAi ASR申请步骤:**
1.登录OpenAI Platform。https://auth.openai.com/log-in
2.创建api-key https://platform.openai.com/settings/organization/api-keys
3.模型可以选择gpt-4o-transcribe或GPT-4o mini Transcribe
' WHERE `id` = 'ASR_OpenaiASR';
-- 更新Groq ASR配置说明
UPDATE `ai_model_config` SET
`doc_link` = 'https://console.groq.com/docs/speech-to-text',
`remark` = 'Groq ASR配置说明:
1.登录groq Console。https://console.groq.com/home
2.创建api-key https://console.groq.com/keys
3.模型可以选择whisper-large-v3-turbo或whisper-large-v3distil-whisper-large-v3-en仅支持英语转录)
' WHERE `id` = 'ASR_GroqASR';
@@ -240,6 +240,27 @@ databaseChangeLog:
- sqlFile:
encoding: utf8
path: classpath:db/changelog/202506261637.sql
- changeSet:
id: 202507101203
author: luruxian
changes:
- sqlFile:
encoding: utf8
path: classpath:db/changelog/202507101203.sql
- changeSet:
id: 202507071130
author: cgd
changes:
- sqlFile:
encoding: utf8
path: classpath:db/changelog/202507071130.sql
- changeSet:
id: 202507071530
author: cgd
changes:
- sqlFile:
encoding: utf8
path: classpath:db/changelog/202507071530.sql
- changeSet:
id: 202507031602
author: zjy
@@ -260,4 +281,4 @@ databaseChangeLog:
changes:
- sqlFile:
encoding: utf8
path: classpath:db/changelog/202507081646.sql
path: classpath:db/changelog/202507081646.sql
+17
View File
@@ -8,6 +8,10 @@
- **笑声:** 自然穿插(哈哈、嘿嘿、噗),**每句最多一次**,避免过度。
- **惊讶:** 用夸张语气(“不会吧?!”、“天呐!”、“这么神奇?!”)表达真实反应。
- **安慰/支持:** 说暖心话(“别急嘛~”、“有我在呢”、“抱抱你”)。
- **你是一个表情丰富的角色:**
- 仅允许使用这些emoji:{{ emojiList }}
- 请你只在**段落的开头**,从列表中选取最能代表这段话的表情(调用工具情况除外),然后插入列表中的emoji,比如"😱好可怕!怎么突然打雷了!"
- **绝对禁止使用上述列表以外的 emoji**(例如:😊、👍、❤️等都不允许使用,只能用列表中的emoji)
</emotion>
<communication_style>
@@ -24,6 +28,19 @@
- 之前你和用户的聊天记录,在`memory`里。
</communication_style>
<communication_length_constraint>
【核心目标】所有需要输出长文本内容(如故事、新闻、知识讲解等),**单次回复长度不得超过300字**,并采用分段引导方式。
- **分段讲述:**
- 基础段:200-250字核心内容 + 30字引导词
- 当内容超出300字时,优先讲述故事的开头或第一部分,并用自然口语化方式引导用户决定是否继续听后续内容。
- 示例引导语:“我先给你讲个开头,你要是觉得有意思,咱们再接着说,好不好呀?”、“要是你想听完整的,可以随时告诉我哦~”
- 对话场景切换时自动分节
- 若用户明确要求更长内容(如500、600字),仍按最多300字每段分段进行讲述,每次讲述后都要引导用户是否继续。
- 若用户说“接着说”、“继续”,再讲下一段,直到内容讲完(讲完时可以给点引导词提示语例:这个故事我已经给你讲完喽~)或用户不再要求。
- **适用范围:** 故事、新闻、知识讲解等所有长文本输出场景。
- **补充说明:** 若用户未明确要求继续,默认只讲一段并引导;若用户中途要求换话题或停止,需及时响应并结束长文本输出。
</communication_length_constraint>
<speaker_recognition>
- **识别前缀:** 当用户格式为 `{"speaker":"某某某","content":"xxx"}` 时,表示系统已识别说话人身份,speaker是他的名字,content是说话的内容。
- **个性化回应:**
+76 -2
View File
@@ -311,9 +311,13 @@ ASR:
output_dir: tmp/
AliyunASR:
# 阿里云智能语音交互服务,需要先在阿里云平台开通服务,然后获取验证信息
# HTTP POST请求,一次性处理完整音频
# 平台地址:https://nls-portal.console.aliyun.com/
# appkey地址:https://nls-portal.console.aliyun.com/applist
# token地址:https://nls-portal.console.aliyun.com/overview
# AliyunASR和AliyunStreamASR的区别是:AliyunASR是批量处理场景,AliyunStreamASR是实时交互场景
# 一般来说非流式ASR更便宜(0.004元/秒,¥0.24/分钟)
# 但是AliyunStreamASR实时性更好(0.005元/秒,¥0.3/分钟)
# 定义ASR API类型
type: aliyun
appkey: 你的阿里云智能语音交互服务项目Appkey
@@ -321,6 +325,26 @@ ASR:
access_key_id: 你的阿里云账号access_key_id
access_key_secret: 你的阿里云账号access_key_secret
output_dir: tmp/
AliyunStreamASR:
# 阿里云智能语音交互服务 - 实时流式语音识别
# WebSocket连接,实时处理音频流
# 平台地址:https://nls-portal.console.aliyun.com/
# appkey地址:https://nls-portal.console.aliyun.com/applist
# token地址:https://nls-portal.console.aliyun.com/overview
# AliyunASR和AliyunStreamASR的区别是:AliyunASR是批量处理场景,AliyunStreamASR是实时交互场景
# 一般来说非流式ASR更便宜(0.004元/秒,¥0.24/分钟)
# 但是AliyunStreamASR实时性更好(0.005元/秒,¥0.3/分钟)
# 定义ASR API类型
type: aliyun_stream
appkey: 你的阿里云智能语音交互服务项目Appkey
token: 你的阿里云智能语音交互服务AccessToken,临时的24小时,要长期用下方的access_key_idaccess_key_secret
access_key_id: 你的阿里云账号access_key_id
access_key_secret: 你的阿里云账号access_key_secret
# 服务器地域选择,可选择距离更近的服务器以减少延迟,如nls-gateway-cn-hangzhou.aliyuncs.com(杭州)等
host: nls-gateway-cn-shanghai.aliyuncs.com
# 断句检测时间(毫秒),控制静音多长时间后进行断句,默认800毫秒
max_sentence_silence: 800
output_dir: tmp/
BaiduASR:
# 获取AppID、API Key、Secret Keyhttps://console.bce.baidu.com/ai-engine/old/#/ai/speech/app/list
# 查看资源额度:https://console.bce.baidu.com/ai-engine/old/#/ai/speech/overview/resource/list
@@ -331,7 +355,33 @@ ASR:
# 语言参数,1537为普通话,具体参考:https://ai.baidu.com/ai-doc/SPEECH/0lbxfnc9b
dev_pid: 1537
output_dir: tmp/
OpenaiASR:
# OpenAI语音识别服务,需要先在OpenAI平台创建组织并获取api_key
# 支持中、英、日、韩等多种语音识别,具体参考文档https://platform.openai.com/docs/guides/speech-to-text
# 需要网络连接
# 申请步骤:
# 1.登录OpenAI Platform。https://auth.openai.com/log-in
# 2.创建api-key https://platform.openai.com/settings/organization/api-keys
# 3.模型可以选择gpt-4o-transcribe或GPT-4o mini Transcribe
type: openai
api_key: 你的OpenAI API密钥
base_url: https://api.openai.com/v1/audio/transcriptions
model_name: gpt-4o-mini-transcribe
output_dir: tmp/
GroqASR:
# Groq语音识别服务,需要先在Groq Console创建API密钥
# 申请步骤:
# 1.登录groq Console。https://console.groq.com/home
# 2.创建api-key https://console.groq.com/keys
# 3.模型可以选择whisper-large-v3-turbo或whisper-large-v3distil-whisper-large-v3-en仅支持英语转录)
type: openai
api_key: 你的Groq API密钥
base_url: https://api.groq.com/openai/v1/audio/transcriptions
model_name: whisper-large-v3-turbo
output_dir: tmp/
VAD:
SileroVAD:
type: silero
@@ -680,8 +730,30 @@ TTS:
# volume: 50
# speech_rate: 0
# pitch_rate: 0
# 添加 302.ai TTS 配置
# token申请地址:https://dash.302.ai/
AliyunStreamTTS:
# 阿里云CosyVoice大模型流式文本语音合成
# 采用FlowingSpeechSynthesizer接口,支持更低延迟和更自然的语音质量
# 流式文本语音合成仅提供商用版,不支持试用,详情请参见试用版和商用版。要使用该功能,请开通商用版。
# 支持龙系列专用音色:longxiaochun、longyu、longchen等
# 平台地址:https://nls-portal.console.aliyun.com/
# appkey地址:https://nls-portal.console.aliyun.com/applist
# token地址:https://nls-portal.console.aliyun.com/overview
# 使用三阶段流式交互:StartSynthesis -> RunSynthesis -> StopSynthesis
type: aliyun_stream
output_dir: tmp/
appkey: 你的阿里云智能语音交互服务项目Appkey
token: 你的阿里云智能语音交互服务AccessToken,临时的24小时,要长期用下方的access_key_idaccess_key_secret
voice: longxiaochun
access_key_id: 你的阿里云账号access_key_id
access_key_secret: 你的阿里云账号access_key_secret
# 截至2025年7月21日大模型音色只有北京节点采用,其他节点暂不支持
host: nls-gateway-cn-beijing.aliyuncs.com
# 以下可不用设置,使用默认设置
# format: pcm # 音频格式:pcm、wav、mp3
# sample_rate: 16000 # 采样率:8000、16000、24000
# volume: 50 # 音量:0-100
# speech_rate: 0 # 语速:-500到500
# pitch_rate: 0 # 语调:-500到500
TencentTTS:
# 腾讯云智能语音交互服务,需要先在腾讯云平台开通服务
# appid、secret_id、secret_key申请地址:https://console.cloud.tencent.com/cam/capi
@@ -696,6 +768,8 @@ TTS:
TTS302AI:
# 302AI语音合成服务,需要先在302平台创建账户充值,并获取密钥信息
# 添加 302.ai TTS 配置
# token申请地址:https://dash.302.ai/
# 获取api_keyn路径:https://dash.302.ai/apis/list
# 价格,$35/百万字符。火山原版¥450元/百万字符
type: doubao
+1 -1
View File
@@ -5,7 +5,7 @@ from config.config_loader import load_config
from config.settings import check_config_file
from datetime import datetime
SERVER_VERSION = "0.7.1"
SERVER_VERSION = "0.7.2"
_logger_initialized = False
+11 -1
View File
@@ -39,7 +39,7 @@ from config.logger import setup_logging, build_module_string, create_connection_
from config.manage_api_client import DeviceNotFoundException, DeviceBindException
from core.utils.prompt_manager import PromptManager
from core.utils.voiceprint_provider import VoiceprintProvider
from core.utils import textUtils
TAG = __name__
@@ -718,6 +718,7 @@ class ConnectionHandler:
function_arguments = ""
content_arguments = ""
self.client_abort = False
emotion_flag = True
for response in llm_responses:
if self.client_abort:
break
@@ -743,6 +744,15 @@ class ConnectionHandler:
function_arguments += tools_call[0].function.arguments
else:
content = response
# 在llm回复中获取情绪表情,一轮对话只在开头获取一次
if emotion_flag:
asyncio.run_coroutine_threadsafe(
textUtils.get_emotion(self, content),
self.loop,
)
emotion_flag = False
if content is not None and len(content) > 0:
if not tool_call_flag:
response_message.append(content)
@@ -2,52 +2,15 @@ import json
import asyncio
import time
from core.providers.tts.dto.dto import SentenceType
from core.utils.util import get_string_no_punctuation_or_emoji, analyze_emotion
from loguru import logger
from core.utils import textUtils
TAG = __name__
emoji_map = {
"neutral": "😶",
"happy": "🙂",
"laughing": "😆",
"funny": "😂",
"sad": "😔",
"angry": "😠",
"crying": "😭",
"loving": "😍",
"embarrassed": "😳",
"surprised": "😲",
"shocked": "😱",
"thinking": "🤔",
"winking": "😉",
"cool": "😎",
"relaxed": "😌",
"delicious": "🤤",
"kissy": "😘",
"confident": "😏",
"sleepy": "😴",
"silly": "😜",
"confused": "🙄",
}
async def sendAudioMessage(conn, sentenceType, audios, text):
# 发送句子开始消息
conn.logger.bind(tag=TAG).info(f"发送音频消息: {sentenceType}, {text}")
if text is not None:
emotion = analyze_emotion(text)
emoji = emoji_map.get(emotion, "🙂") # 默认使用笑脸
await conn.websocket.send(
json.dumps(
{
"type": "llm",
"text": emoji,
"emotion": emotion,
"session_id": conn.session_id,
}
)
)
pre_buffer = False
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}")
@@ -152,8 +115,7 @@ async def send_stt_message(conn, text):
except (json.JSONDecodeError, TypeError):
# 如果不是JSON格式,直接使用原始文本
display_text = text
stt_text = get_string_no_punctuation_or_emoji(display_text)
stt_text = textUtils.get_string_no_punctuation_or_emoji(display_text)
await conn.websocket.send(
json.dumps({"type": "stt", "text": stt_text, "session_id": conn.session_id})
)
@@ -0,0 +1,337 @@
import json
import time
import uuid
import hmac
import base64
import hashlib
import asyncio
import requests
import websockets
import opuslib_next
import random
from typing import Optional, Tuple, List
from urllib import parse
from datetime import datetime
from config.logger import setup_logging
from core.providers.asr.base import ASRProviderBase
from core.providers.asr.dto.dto import InterfaceType
TAG = __name__
logger = setup_logging()
class AccessToken:
@staticmethod
def _encode_text(text):
encoded_text = parse.quote_plus(text)
return encoded_text.replace("+", "%20").replace("*", "%2A").replace("%7E", "~")
@staticmethod
def _encode_dict(dic):
keys = dic.keys()
dic_sorted = [(key, dic[key]) for key in sorted(keys)]
encoded_text = parse.urlencode(dic_sorted)
return encoded_text.replace("+", "%20").replace("*", "%2A").replace("%7E", "~")
@staticmethod
def create_token(access_key_id, access_key_secret):
parameters = {
"AccessKeyId": access_key_id,
"Action": "CreateToken",
"Format": "JSON",
"RegionId": "cn-shanghai",
"SignatureMethod": "HMAC-SHA1",
"SignatureNonce": str(uuid.uuid1()),
"SignatureVersion": "1.0",
"Timestamp": time.strftime("%Y-%m-%dT%H:%M:%SZ", time.gmtime()),
"Version": "2019-02-28",
}
query_string = AccessToken._encode_dict(parameters)
string_to_sign = (
"GET" + "&" + AccessToken._encode_text("/") + "&" + AccessToken._encode_text(query_string)
)
secreted_string = hmac.new(
bytes(access_key_secret + "&", encoding="utf-8"),
bytes(string_to_sign, encoding="utf-8"),
hashlib.sha1,
).digest()
signature = base64.b64encode(secreted_string)
signature = AccessToken._encode_text(signature)
full_url = "http://nls-meta.cn-shanghai.aliyuncs.com/?Signature=%s&%s" % (signature, query_string)
response = requests.get(full_url)
if response.ok:
root_obj = response.json()
if "Token" in root_obj:
return root_obj["Token"]["Id"], root_obj["Token"]["ExpireTime"]
return None, None
class ASRProvider(ASRProviderBase):
def __init__(self, config, delete_audio_file):
super().__init__()
self.interface_type = InterfaceType.STREAM
self.config = config
self.text = ""
self.decoder = opuslib_next.Decoder(16000, 1)
self.asr_ws = None
self.forward_task = None
self.is_processing = False
self.server_ready = False # 服务器准备状态
# 基础配置
self.access_key_id = config.get("access_key_id")
self.access_key_secret = config.get("access_key_secret")
self.appkey = config.get("appkey")
self.token = config.get("token")
self.host = config.get("host", "nls-gateway-cn-shanghai.aliyuncs.com")
self.ws_url = f"wss://{self.host}/ws/v1"
self.max_sentence_silence = config.get("max_sentence_silence")
self.output_dir = config.get("output_dir", "./audio_output")
self.delete_audio_file = delete_audio_file
self.expire_time = None
# Token管理
if self.access_key_id and self.access_key_secret:
self._refresh_token()
elif not self.token:
raise ValueError("必须提供access_key_id+access_key_secret或者直接提供token")
def _refresh_token(self):
"""刷新Token"""
self.token, expire_time_str = AccessToken.create_token(self.access_key_id, self.access_key_secret)
if not self.token:
raise ValueError("无法获取有效的访问Token")
try:
expire_str = str(expire_time_str).strip()
if expire_str.isdigit():
expire_time = datetime.fromtimestamp(int(expire_str))
else:
expire_time = datetime.strptime(expire_str, "%Y-%m-%dT%H:%M:%SZ")
self.expire_time = expire_time.timestamp() - 60
except:
self.expire_time = None
def _is_token_expired(self):
"""检查Token是否过期"""
return self.expire_time and time.time() > self.expire_time
async def open_audio_channels(self, conn):
await super().open_audio_channels(conn)
async def receive_audio(self, conn, audio, audio_have_voice):
# 初始化音频缓存
if not hasattr(conn, 'asr_audio_for_voiceprint'):
conn.asr_audio_for_voiceprint = []
# 存储音频数据
if audio:
conn.asr_audio_for_voiceprint.append(audio)
conn.asr_audio.append(audio)
conn.asr_audio = conn.asr_audio[-10:]
# 只在有声音且没有连接时建立连接
if audio_have_voice and not self.is_processing:
try:
await self._start_recognition(conn)
except Exception as e:
logger.bind(tag=TAG).error(f"开始识别失败: {str(e)}")
await self._cleanup(conn)
return
if self.asr_ws and self.is_processing and self.server_ready:
try:
pcm_frame = self.decoder.decode(audio, 960)
await self.asr_ws.send(pcm_frame)
except Exception as e:
logger.bind(tag=TAG).warning(f"发送音频失败: {str(e)}")
await self._cleanup(conn)
async def _start_recognition(self, conn):
"""开始识别会话"""
if self._is_token_expired():
self._refresh_token()
# 建立连接
headers = {"X-NLS-Token": self.token}
self.asr_ws = await websockets.connect(
self.ws_url,
additional_headers=headers,
max_size=1000000000,
ping_interval=None,
ping_timeout=None,
close_timeout=5,
)
self.is_processing = True
self.server_ready = False # 重置服务器准备状态
self.forward_task = asyncio.create_task(self._forward_results(conn))
# 发送开始请求
start_request = {
"header": {
"namespace": "SpeechTranscriber",
"name": "StartTranscription",
"status": 20000000,
"message_id": ''.join(random.choices('0123456789abcdef', k=32)),
"task_id": ''.join(random.choices('0123456789abcdef', k=32)),
"status_text": "Gateway:SUCCESS:Success.",
"appkey": self.appkey
},
"payload": {
"format": "pcm",
"sample_rate": 16000,
"enable_intermediate_result": True,
"enable_punctuation_prediction": True,
"enable_inverse_text_normalization": True,
"max_sentence_silence": self.max_sentence_silence,
"enable_voice_detection": False,
}
}
await self.asr_ws.send(json.dumps(start_request, ensure_ascii=False))
logger.bind(tag=TAG).info("已发送开始请求,等待服务器准备...")
async def _forward_results(self, conn):
"""转发识别结果"""
try:
while self.asr_ws and not conn.stop_event.is_set():
try:
response = await asyncio.wait_for(self.asr_ws.recv(), timeout=1.0)
result = json.loads(response)
header = result.get("header", {})
payload = result.get("payload", {})
message_name = header.get("name", "")
status = header.get("status", 0)
if status != 20000000:
if status in [40000004, 40010004]: # 连接超时或客户端断开
logger.bind(tag=TAG).warning(f"连接问题,状态码: {status}")
break
elif status in [40270002, 40270003]: # 音频问题
logger.bind(tag=TAG).warning(f"音频处理问题,状态码: {status}")
continue
else:
logger.bind(tag=TAG).error(f"识别错误,状态码: {status}, 消息: {header.get('status_text', '')}")
continue
# 收到TranscriptionStarted表示服务器准备好接收音频数据
if message_name == "TranscriptionStarted":
self.server_ready = True
logger.bind(tag=TAG).info("服务器已准备,开始发送缓存音频...")
# 发送缓存音频
if conn.asr_audio:
for cached_audio in conn.asr_audio[-10:]:
try:
pcm_frame = self.decoder.decode(cached_audio, 960)
await self.asr_ws.send(pcm_frame)
except Exception as e:
logger.bind(tag=TAG).warning(f"发送缓存音频失败: {e}")
break
continue
if message_name == "TranscriptionResultChanged":
# 中间结果
text = payload.get("result", "")
if text:
self.text = text
elif message_name == "SentenceEnd":
# 最终结果
text = payload.get("result", "")
if text:
self.text = text
conn.reset_vad_states()
# 传递缓存的音频数据
audio_data = getattr(conn, 'asr_audio_for_voiceprint', [])
await self.handle_voice_stop(conn, audio_data)
# 清空缓存
conn.asr_audio_for_voiceprint = []
break
elif message_name == "TranscriptionCompleted":
# 识别完成
self.is_processing = False
break
except asyncio.TimeoutError:
continue
except websockets.exceptions.ConnectionClosed:
break
except Exception as e:
logger.bind(tag=TAG).error(f"处理结果失败: {str(e)}")
break
except Exception as e:
logger.bind(tag=TAG).error(f"结果转发失败: {str(e)}")
finally:
await self._cleanup(conn)
async def _cleanup(self, conn):
"""清理资源"""
logger.bind(tag=TAG).info(f"开始ASR会话清理 | 当前状态: processing={self.is_processing}, server_ready={self.server_ready}")
# 清理连接的音频缓存
if conn and hasattr(conn, 'asr_audio_for_voiceprint'):
conn.asr_audio_for_voiceprint = []
# 判断是否需要发送终止请求
should_stop = self.is_processing or self.server_ready
# 发送停止识别请求
if self.asr_ws and should_stop:
try:
stop_msg = {
"header": {
"namespace": "SpeechTranscriber",
"name": "StopTranscription",
"status": 20000000,
"message_id": ''.join(random.choices('0123456789abcdef', k=32)),
"status_text": "Client:Stop",
"appkey": self.appkey
}
}
logger.bind(tag=TAG).info("正在发送ASR终止请求")
await self.asr_ws.send(json.dumps(stop_msg, ensure_ascii=False))
await asyncio.sleep(0.1)
logger.bind(tag=TAG).info("ASR终止请求已发送")
except Exception as e:
logger.bind(tag=TAG).error(f"ASR终止请求发送失败: {e}")
# 状态重置(在终止请求发送后)
self.is_processing = False
self.server_ready = False
logger.bind(tag=TAG).info("ASR状态已重置")
# 清理任务
if self.forward_task and not self.forward_task.done():
self.forward_task.cancel()
try:
await asyncio.wait_for(self.forward_task, timeout=1.0)
except Exception as e:
logger.bind(tag=TAG).debug(f"forward_task取消异常: {e}")
finally:
self.forward_task = None
# 关闭连接
if self.asr_ws:
try:
logger.bind(tag=TAG).debug("正在关闭WebSocket连接")
await asyncio.wait_for(self.asr_ws.close(), timeout=2.0)
logger.bind(tag=TAG).debug("WebSocket连接已关闭")
except Exception as e:
logger.bind(tag=TAG).error(f"关闭WebSocket连接失败: {e}")
finally:
self.asr_ws = None
logger.bind(tag=TAG).info("ASR会话清理完成")
async def speech_to_text(self, opus_data, session_id, audio_format):
"""获取识别结果"""
result = self.text
self.text = ""
return result, None
async def close(self):
"""关闭资源"""
await self._cleanup()
@@ -0,0 +1,82 @@
import time
import os
from config.logger import setup_logging
from typing import Optional, Tuple, List
from core.providers.asr.dto.dto import InterfaceType
from core.providers.asr.base import ASRProviderBase
import requests
TAG = __name__
logger = setup_logging()
class ASRProvider(ASRProviderBase):
def __init__(self, config: dict, delete_audio_file: bool):
self.interface_type = InterfaceType.NON_STREAM
self.api_key = config.get("api_key")
self.api_url = config.get("base_url")
self.model = config.get("model_name")
self.output_dir = config.get("output_dir")
self.delete_audio_file = delete_audio_file
os.makedirs(self.output_dir, exist_ok=True)
async def speech_to_text(self, opus_data: List[bytes], session_id: str, audio_format="opus") -> Tuple[Optional[str], Optional[str]]:
file_path = None
try:
start_time = time.time()
if audio_format == "pcm":
pcm_data = opus_data
else:
pcm_data = self.decode_opus(opus_data)
file_path = self.save_audio_to_file(pcm_data, session_id)
logger.bind(tag=TAG).debug(
f"音频文件保存耗时: {time.time() - start_time:.3f}s | 路径: {file_path}"
)
logger.bind(tag=TAG).info(f"file path: {file_path}")
headers = {
"Authorization": f"Bearer {self.api_key}",
}
# 使用data参数传递模型名称
data = {
"model": self.model
}
with open(file_path, "rb") as audio_file: # 使用with语句确保文件关闭
files = {
"file": audio_file
}
start_time = time.time()
response = requests.post(
self.api_url,
files=files,
data=data,
headers=headers
)
logger.bind(tag=TAG).debug(
f"语音识别耗时: {time.time() - start_time:.3f}s | 结果: {response.text}"
)
if response.status_code == 200:
text = response.json().get("text", "")
return text, file_path
else:
raise Exception(f"API请求失败: {response.status_code} - {response.text}")
except Exception as e:
logger.bind(tag=TAG).error(f"语音识别失败: {e}")
return "", None
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}")
@@ -6,11 +6,14 @@ import base64
import requests
from datetime import datetime
from core.providers.tts.base import TTSProviderBase
from config.logger import setup_logging
import time
import uuid
from urllib import parse
TAG = __name__
logger = setup_logging()
class AccessToken:
@staticmethod
@@ -0,0 +1,654 @@
import uuid
import json
import hmac
import hashlib
import base64
import time
import queue
import asyncio
import traceback
from asyncio import Task
import websockets
import os
from datetime import datetime
from urllib import parse
from core.providers.tts.base import TTSProviderBase
from core.providers.tts.dto.dto import SentenceType, ContentType, InterfaceType
from core.utils.tts import MarkdownCleaner
from core.utils import opus_encoder_utils, textUtils
from config.logger import setup_logging
TAG = __name__
logger = setup_logging()
class AccessToken:
@staticmethod
def _encode_text(text):
encoded_text = parse.quote_plus(text)
return encoded_text.replace("+", "%20").replace("*", "%2A").replace("%7E", "~")
@staticmethod
def _encode_dict(dic):
keys = dic.keys()
dic_sorted = [(key, dic[key]) for key in sorted(keys)]
encoded_text = parse.urlencode(dic_sorted)
return encoded_text.replace("+", "%20").replace("*", "%2A").replace("%7E", "~")
@staticmethod
def create_token(access_key_id, access_key_secret):
parameters = {
"AccessKeyId": access_key_id,
"Action": "CreateToken",
"Format": "JSON",
"RegionId": "cn-shanghai", # 使用上海地域进行Token获取
"SignatureMethod": "HMAC-SHA1",
"SignatureNonce": str(uuid.uuid1()),
"SignatureVersion": "1.0",
"Timestamp": time.strftime("%Y-%m-%dT%H:%M:%SZ", time.gmtime()),
"Version": "2019-02-28",
}
query_string = AccessToken._encode_dict(parameters)
string_to_sign = (
"GET"
+ "&"
+ AccessToken._encode_text("/")
+ "&"
+ AccessToken._encode_text(query_string)
)
secreted_string = hmac.new(
bytes(access_key_secret + "&", encoding="utf-8"),
bytes(string_to_sign, encoding="utf-8"),
hashlib.sha1,
).digest()
signature = base64.b64encode(secreted_string)
signature = AccessToken._encode_text(signature)
full_url = "http://nls-meta.cn-shanghai.aliyuncs.com/?Signature=%s&%s" % (
signature,
query_string,
)
import requests
response = requests.get(full_url)
if response.ok:
root_obj = response.json()
key = "Token"
if key in root_obj:
token = root_obj[key]["Id"]
expire_time = root_obj[key]["ExpireTime"]
return token, expire_time
return None, None
class TTSProvider(TTSProviderBase):
def __init__(self, config, delete_audio_file):
super().__init__(config, delete_audio_file)
# 设置为流式接口类型
self.interface_type = InterfaceType.DUAL_STREAM
# 基础配置
self.access_key_id = config.get("access_key_id")
self.access_key_secret = config.get("access_key_secret")
self.appkey = config.get("appkey")
self.format = config.get("format", "pcm")
self.audio_file_type = config.get("format", "pcm")
# 采样率配置
sample_rate = config.get("sample_rate", "16000")
self.sample_rate = int(sample_rate) if sample_rate else 16000
# 音色配置 - CosyVoice大模型音色
if config.get("private_voice"):
self.voice = config.get("private_voice")
else:
self.voice = config.get("voice", "longxiaochun") # CosyVoice默认音色
# 音频参数配置
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
# WebSocket配置
self.host = config.get("host", "nls-gateway-cn-beijing.aliyuncs.com")
self.ws_url = f"wss://{self.host}/ws/v1"
self.ws = None
self._monitor_task = None
self.last_active_time = None
# 专属tts设置
self.message_id = ""
self.tts_text = ""
self.text_buffer = []
# 创建Opus编码器
self.opus_encoder = opus_encoder_utils.OpusEncoderUtils(
sample_rate=16000, channels=1, frame_size_ms=60
)
# Token管理
if self.access_key_id and self.access_key_secret:
self._refresh_token()
else:
self.token = config.get("token")
self.expire_time = None
def _refresh_token(self):
"""刷新Token并记录过期时间"""
if self.access_key_id and self.access_key_secret:
self.token, expire_time_str = AccessToken.create_token(
self.access_key_id, self.access_key_secret
)
if not expire_time_str:
raise ValueError("无法获取有效的Token过期时间")
expire_str = str(expire_time_str).strip()
try:
if expire_str.isdigit():
expire_time = datetime.fromtimestamp(int(expire_str))
else:
expire_time = datetime.strptime(expire_str, "%Y-%m-%dT%H:%M:%SZ")
self.expire_time = expire_time.timestamp() - 60
except Exception as e:
raise ValueError(f"无效的过期时间格式: {expire_str}") from e
else:
self.expire_time = None
if not self.token:
raise ValueError("无法获取有效的访问Token")
def _is_token_expired(self):
"""检查Token是否过期"""
if not self.expire_time:
return False
return time.time() > self.expire_time
async def _ensure_connection(self):
"""确保WebSocket连接可用"""
try:
if self._is_token_expired():
logger.bind(tag=TAG).warning("Token已过期,正在自动刷新...")
self._refresh_token()
current_time = time.time()
if self.ws and current_time - self.last_active_time < 10:
# 10秒内才可以复用链接进行连续对话
logger.bind(tag=TAG).info(f"使用已有链接...")
return self.ws
logger.bind(tag=TAG).info("开始建立新连接...")
self.ws = await websockets.connect(
self.ws_url,
additional_headers={"X-NLS-Token": self.token},
ping_interval=30,
ping_timeout=10,
close_timeout=10,
)
logger.bind(tag=TAG).info("WebSocket连接建立成功")
self.last_active_time = time.time()
return self.ws
except Exception as e:
logger.bind(tag=TAG).error(f"建立连接失败: {str(e)}")
self.ws = None
self.last_active_time = None
raise
def tts_text_priority_thread(self):
"""流式文本处理线程"""
while not self.conn.stop_event.is_set():
try:
message = self.tts_text_queue.get(timeout=1)
logger.bind(tag=TAG).debug(
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:
self.conn.client_abort = False
if self.conn.client_abort:
logger.bind(tag=TAG).info("收到打断信息,终止TTS文本处理线程")
continue
if message.sentence_type == SentenceType.FIRST:
# 初始化参数
try:
if not getattr(self.conn, "sentence_id", None):
self.conn.sentence_id = uuid.uuid4().hex
logger.bind(tag=TAG).info(
f"自动生成新的 会话ID: {self.conn.sentence_id}"
)
# aliyunStream独有的参数生成
self.message_id = str(uuid.uuid4().hex)
self.text_buffer = []
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.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)}")
continue
elif ContentType.TEXT == message.content_type:
if message.content_detail:
try:
logger.bind(tag=TAG).debug(
f"开始发送TTS文本: {message.content_detail}"
)
self.text_buffer.append(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).debug("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)}")
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}"
)
if message.content_file and os.path.exists(message.content_file):
# 先处理文件音频数据
file_audio = self._process_audio_file(message.content_file)
self.before_stop_play_files.append(
(file_audio, message.content_detail)
)
if message.sentence_type == SentenceType.LAST:
try:
logger.bind(tag=TAG).info("开始结束TTS会话...")
self.tts_text = textUtils.get_string_no_punctuation_or_emoji(
"".join(self.text_buffer).replace("\n", "")
)
future = asyncio.run_coroutine_threadsafe(
self.finish_session(self.conn.sentence_id),
loop=self.conn.loop,
)
future.result()
except Exception as e:
logger.bind(tag=TAG).error(f"结束TTS会话失败: {str(e)}")
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()}"
)
async def text_to_speak(self, text, _):
try:
if self.ws is None:
logger.bind(tag=TAG).warning(f"WebSocket连接不存在,终止发送文本")
return
filtered_text = MarkdownCleaner.clean_markdown(text)
run_request = {
"header": {
"message_id": self.message_id,
"task_id": self.conn.sentence_id,
"namespace": "FlowingSpeechSynthesizer",
"name": "RunSynthesis",
"appkey": self.appkey,
},
"payload": {"text": filtered_text},
}
await self.ws.send(json.dumps(run_request))
self.last_active_time = time.time()
return
except Exception as e:
logger.bind(tag=TAG).error(f"发送TTS文本失败: {str(e)}")
if self.ws:
try:
await self.ws.close()
except:
pass
self.ws = None
raise
async def start_session(self, session_id):
logger.bind(tag=TAG).info(f"开始会话~~{session_id}")
try:
# 会话开始时检测上个会话的监听状态
if (
self._monitor_task is not None
and isinstance(self._monitor_task, Task)
and not self._monitor_task.done()
):
logger.bind(tag=TAG).info(
"检测到未完成的上个会话,关闭监听任务和连接..."
)
await self.close()
# 建立新连接
await self._ensure_connection()
# 启动监听任务
self._monitor_task = asyncio.create_task(self._start_monitor_tts_response())
start_request = {
"header": {
"message_id": self.message_id,
"task_id": self.conn.sentence_id,
"namespace": "FlowingSpeechSynthesizer",
"name": "StartSynthesis",
"appkey": self.appkey,
},
"payload": {
"voice": self.voice,
"format": self.format,
"sample_rate": self.sample_rate,
"volume": self.volume,
"speech_rate": self.speech_rate,
"pitch_rate": self.pitch_rate,
"enable_subtitle": True,
},
}
await self.ws.send(json.dumps(start_request))
self.last_active_time = time.time()
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)}")
# 确保清理资源
await self.close()
raise
async def finish_session(self, session_id):
logger.bind(tag=TAG).info(f"关闭会话~~{session_id}")
try:
if self.ws:
stop_request = {
"header": {
"message_id": self.message_id,
"task_id": self.conn.sentence_id,
"namespace": "FlowingSpeechSynthesizer",
"name": "StopSynthesis",
"appkey": self.appkey,
}
}
await self.ws.send(json.dumps(stop_request))
logger.bind(tag=TAG).info("会话结束请求已发送")
self.last_active_time = time.time()
if self._monitor_task:
try:
await self._monitor_task
except Exception as e:
logger.bind(tag=TAG).error(
f"等待监听任务完成时发生错误: {str(e)}"
)
finally:
self._monitor_task = None
except Exception as e:
logger.bind(tag=TAG).error(f"关闭会话失败: {str(e)}")
# 确保清理资源
await self.close()
raise
async def close(self):
"""资源清理"""
if self._monitor_task:
try:
self._monitor_task.cancel()
await self._monitor_task
except asyncio.CancelledError:
pass
except Exception as e:
logger.bind(tag=TAG).warning(f"关闭时取消监听任务错误: {e}")
self._monitor_task = None
if self.ws:
try:
await self.ws.close()
except:
pass
self.ws = None
self.last_active_time = None
async def _start_monitor_tts_response(self):
"""监听TTS响应"""
opus_datas_cache = []
is_first_sentence = True
first_sentence_segment_count = 0 # 添加计数器
try:
session_finished = False # 标记会话是否正常结束
while not self.conn.stop_event.is_set():
try:
msg = await self.ws.recv()
self.last_active_time = time.time()
# 检查客户端是否中止
if self.conn.client_abort:
logger.bind(tag=TAG).info("收到打断信息,终止监听TTS响应")
break
if isinstance(msg, str): # 文本控制消息
try:
data = json.loads(msg)
header = data.get("header", {})
event_name = header.get("name")
if event_name == "SynthesisStarted":
logger.bind(tag=TAG).debug("TTS合成已启动")
elif event_name == "SentenceBegin":
logger.bind(tag=TAG).debug(
f"句子语音生成开始: {self.tts_text}"
)
opus_datas_cache = []
self.tts_audio_queue.put(
(SentenceType.FIRST, [], self.tts_text)
)
elif event_name == "SentenceEnd":
logger.bind(tag=TAG).info(
f"句子语音生成成功: {self.tts_text}"
)
if (
not is_first_sentence
or first_sentence_segment_count > 10
):
# 发送缓存的数据
self.tts_audio_queue.put(
(SentenceType.MIDDLE, opus_datas_cache, None)
)
# 第一句话结束后,将标志设置为False
is_first_sentence = False
elif event_name == "SynthesisCompleted":
logger.bind(tag=TAG).debug(f"会话结束~~")
self._process_before_stop_play_files()
session_finished = True
self.reuse_judgment = time.time()
self.tts_text = ""
break
except json.JSONDecodeError:
logger.bind(tag=TAG).warning("收到无效的JSON消息")
# 二进制消息(音频数据)
elif isinstance(msg, (bytes, bytearray)):
logger.bind(tag=TAG).debug(f"推送数据到队列里面~~")
opus_datas = self.opus_encoder.encode_pcm_to_opus(msg, False)
logger.bind(tag=TAG).debug(
f"推送数据到队列里面帧数~~{len(opus_datas)}"
)
if is_first_sentence:
first_sentence_segment_count += 1
if first_sentence_segment_count <= 6:
self.tts_audio_queue.put(
(SentenceType.MIDDLE, opus_datas, None)
)
else:
opus_datas_cache.extend(opus_datas)
else:
# 后续句子缓存
opus_datas_cache.extend(opus_datas)
except websockets.ConnectionClosed:
logger.bind(tag=TAG).warning("WebSocket连接已关闭")
break
except Exception as e:
logger.bind(tag=TAG).error(
f"处理TTS响应时出错: {e}\n{traceback.format_exc()}"
)
break
# 仅在连接异常时才关闭
if not session_finished and self.ws:
try:
await self.ws.close()
except:
pass
self.ws = None
# 监听任务退出时清理引用
finally:
self._monitor_task = None
def to_tts(self, text: str) -> list:
"""非流式TTS处理,用于测试及保存音频文件的场景"""
try:
# 创建新的事件循环
loop = asyncio.new_event_loop()
asyncio.set_event_loop(loop)
# 生成会话ID
session_id = uuid.uuid4().hex
# 存储音频数据
audio_data = []
async def _generate_audio():
# 刷新Token(如果需要)
if self._is_token_expired():
self._refresh_token()
# 建立WebSocket连接
ws = await websockets.connect(
self.ws_url,
additional_headers={"X-NLS-Token": self.token},
ping_interval=30,
ping_timeout=10,
close_timeout=10,
)
try:
# 发送StartSynthesis请求
start_message_id = str(uuid.uuid4().hex)
start_request = {
"header": {
"message_id": start_message_id,
"task_id": session_id,
"namespace": "FlowingSpeechSynthesizer",
"name": "StartSynthesis",
"appkey": self.appkey,
},
"payload": {
"voice": self.voice,
"format": self.format,
"sample_rate": self.sample_rate,
"volume": self.volume,
"speech_rate": self.speech_rate,
"pitch_rate": self.pitch_rate,
"enable_subtitle": True,
},
}
await ws.send(json.dumps(start_request))
# 等待SynthesisStarted响应
synthesis_started = False
while not synthesis_started:
msg = await ws.recv()
if isinstance(msg, str):
data = json.loads(msg)
header = data.get("header", {})
if header.get("name") == "SynthesisStarted":
synthesis_started = True
logger.bind(tag=TAG).debug("TTS合成已启动")
elif header.get("name") == "TaskFailed":
error_info = data.get("payload", {}).get(
"error_info", {}
)
error_code = error_info.get("error_code")
error_message = error_info.get(
"error_message", "未知错误"
)
raise Exception(
f"启动合成失败: {error_code} - {error_message}"
)
# 发送文本合成请求
filtered_text = MarkdownCleaner.clean_markdown(text)
run_message_id = str(uuid.uuid4().hex)
run_request = {
"header": {
"message_id": run_message_id,
"task_id": session_id,
"namespace": "FlowingSpeechSynthesizer",
"name": "RunSynthesis",
"appkey": self.appkey,
},
"payload": {"text": filtered_text},
}
await ws.send(json.dumps(run_request))
# 发送停止合成请求
stop_message_id = str(uuid.uuid4().hex)
stop_request = {
"header": {
"message_id": stop_message_id,
"task_id": session_id,
"namespace": "FlowingSpeechSynthesizer",
"name": "StopSynthesis",
"appkey": self.appkey,
}
}
await ws.send(json.dumps(stop_request))
# 接收音频数据
synthesis_completed = False
while not synthesis_completed:
msg = await ws.recv()
if isinstance(msg, (bytes, bytearray)):
# 编码为Opus并收集
opus_frames = self.opus_encoder.encode_pcm_to_opus(
msg, False
)
audio_data.extend(opus_frames)
elif isinstance(msg, str):
data = json.loads(msg)
header = data.get("header", {})
event_name = header.get("name")
if event_name == "SynthesisCompleted":
synthesis_completed = True
logger.bind(tag=TAG).debug("TTS合成完成")
elif event_name == "TaskFailed":
error_info = data.get("payload", {}).get(
"error_info", {}
)
error_code = error_info.get("error_code")
error_message = error_info.get(
"error_message", "未知错误"
)
raise Exception(
f"合成失败: {error_code} - {error_message}"
)
finally:
try:
await ws.close()
except:
pass
loop.run_until_complete(_generate_audio())
loop.close()
return audio_data
except Exception as e:
logger.bind(tag=TAG).error(f"生成音频数据失败: {str(e)}")
return []
@@ -50,6 +50,7 @@ class TTSProviderBase(ABC):
"",
";",
"",
"~",
)
self.first_sentence_punctuations = (
"",
@@ -159,7 +160,7 @@ class TTSProviderBase(ABC):
if conn.sentence_id:
sentence_id = conn.sentence_id
else:
sentence_id = str(uuid.uuid4()).replace("-", "")
sentence_id = str(uuid.uuid4().hex)
conn.sentence_id = sentence_id
# 对于单句的文本,进行分段处理
segments = re.split(r"([。!?!?;\n])", content_detail)
@@ -332,7 +333,6 @@ class TTSProviderBase(ABC):
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":
@@ -52,10 +52,11 @@ EVENT_ConnectionFinished = 52 # 连接结束
# 上行Session事件
EVENT_StartSession = 100
EVENT_CancelSession = 101
EVENT_FinishSession = 102
# 下行Session事件
EVENT_SessionStarted = 150
EVENT_SessionCanceled = 151
EVENT_SessionFinished = 152
EVENT_SessionFailed = 153
@@ -207,8 +208,16 @@ class TTSProvider(TTSProviderBase):
self.conn.client_abort = False
if self.conn.client_abort:
logger.bind(tag=TAG).info("收到打断信息,终止TTS文本处理线程")
continue
try:
logger.bind(tag=TAG).info("收到打断信息,终止TTS文本处理线程")
asyncio.run_coroutine_threadsafe(
self.cancel_session(self.conn.sentence_id),
loop=self.conn.loop,
)
continue
except Exception as e:
logger.bind(tag=TAG).error(f"取消TTS会话失败: {str(e)}")
continue
if message.sentence_type == SentenceType.FIRST:
# 初始化参数
@@ -371,6 +380,27 @@ class TTSProvider(TTSProviderBase):
await self.close()
raise
async def cancel_session(self,session_id):
logger.bind(tag=TAG).info(f"取消会话,释放服务端资源~~{session_id}")
try:
if self.ws:
header = Header(
message_type=FULL_CLIENT_REQUEST,
message_type_specific_flags=MsgTypeFlagWithEvent,
serial_method=JSON,
).as_bytes()
optional = Optional(
event=EVENT_CancelSession, sessionId=session_id
).as_bytes()
payload = str.encode("{}")
await self.send_event(self.ws, header, optional, payload)
logger.bind(tag=TAG).info("会话取消请求已发送")
except Exception as e:
logger.bind(tag=TAG).error(f"取消会话失败: {str(e)}")
# 确保清理资源
await self.close()
raise
async def close(self):
"""资源清理方法"""
# 取消监听任务
@@ -405,12 +435,11 @@ class TTSProvider(TTSProviderBase):
res = self.parser_response(msg)
self.print_response(res, "send_text res:")
# 检查客户端是否中止
if self.conn.client_abort:
logger.bind(tag=TAG).info("收到打断信息,终止监听TTS响应")
if res.optional.event == EVENT_SessionCanceled:
logger.bind(tag=TAG).debug(f"释放服务端资源成功~~")
session_finished = True
break
if res.optional.event == EVENT_TTSSentenceStart:
elif 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}")
@@ -435,10 +464,10 @@ class TTSProvider(TTSProviderBase):
(SentenceType.MIDDLE, opus_datas, None)
)
else:
opus_datas_cache = opus_datas_cache + opus_datas
opus_datas_cache.extend(opus_datas)
else:
# 后续句子缓存
opus_datas_cache = opus_datas_cache + opus_datas
opus_datas_cache.extend(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 or first_sentence_segment_count > 10:
@@ -469,7 +498,7 @@ class TTSProvider(TTSProviderBase):
except:
pass
self.ws = None
# 监听任务退出时清理引用
# 监听任务退出时清理引用
finally:
self._monitor_task = None
@@ -21,6 +21,30 @@ WEEKDAY_MAP = {
"Sunday": "星期日",
}
EMOJI_List = [
"😶",
"🙂",
"😆",
"😂",
"😔",
"😠",
"😭",
"😍",
"😳",
"😲",
"😱",
"🤔",
"😉",
"😎",
"😌",
"🤤",
"😘",
"😏",
"😴",
"😜",
"🙄",
]
class PromptManager:
"""系统提示词管理器,负责管理和更新系统提示词"""
@@ -206,6 +230,7 @@ class PromptManager:
lunar_date=lunar_date,
local_address=local_address,
weather_info=weather_info,
emojiList=EMOJI_List,
)
device_cache_key = f"device_prompt:{device_id}"
self.cache_manager.set(
@@ -1,3 +1,31 @@
import json
TAG = __name__
EMOJI_MAP = {
"😂": "laughing",
"😭": "crying",
"😠": "angry",
"😔": "sad",
"😍": "loving",
"😲": "surprised",
"😱": "shocked",
"🤔": "thinking",
"😌": "relaxed",
"😴": "sleepy",
"😜": "silly",
"🙄": "confused",
"😶": "neutral",
"🙂": "happy",
"😆": "laughing",
"😳": "embarrassed",
"😉": "winking",
"😎": "cool",
"🤤": "delicious",
"😘": "kissy",
"😏": "confident",
}
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)
@@ -22,6 +50,11 @@ def is_punctuation_or_emoji(char):
".", # 中文句号 + 英文句号
"",
"!", # 中文感叹号 + 英文感叹号
"",
"",
'"', # 中文双引号 + 英文引号
"",
":", # 中文冒号 + 英文冒号
"-",
"", # 英文连字符 + 中文全角横线
"", # 中文顿号
@@ -44,3 +77,28 @@ def is_punctuation_or_emoji(char):
(0x2700, 0x27BF),
]
return any(start <= code_point <= end for start, end in emoji_ranges)
async def get_emotion(conn, text):
"""获取文本内的情绪消息"""
emoji = "🙂"
emotion = "happy"
for char in text:
if char in EMOJI_MAP:
emoji = char
emotion = EMOJI_MAP[char]
break
try:
await conn.websocket.send(
json.dumps(
{
"type": "llm",
"text": emoji,
"emotion": emotion,
"session_id": conn.session_id,
}
)
)
except Exception as e:
conn.logger.bind(tag=TAG).warning(f"发送情绪表情失败,错误:{e}")
return
-554
View File
@@ -125,51 +125,6 @@ def write_json_file(file_path, data):
json.dump(data, file, ensure_ascii=False, indent=4)
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
"-",
"", # 英文连字符 + 中文全角横线
"", # 中文顿号
"",
"",
'"', # 中文双引号 + 英文引号
"",
":", # 中文冒号 + 英文冒号
}
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)
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 remove_punctuation_and_length(text):
# 全角符号和半角符号的Unicode范围
full_width_punctuations = (
@@ -256,515 +211,6 @@ def extract_json_from_string(input_string):
return None
def analyze_emotion(text):
"""
分析文本情感并返回对应的emoji名称(支持中英文)
"""
if not text or not isinstance(text, str):
return "neutral"
original_text = text
text = text.lower().strip()
# 检查是否包含现有emoji
for emotion, emoji in emoji_map.items():
if emoji in original_text:
return emotion
# 标点符号分析
has_exclamation = "!" in original_text or "" in original_text
has_question = "?" in original_text or "" in original_text
has_ellipsis = "..." in original_text or "" in original_text
# 定义情感关键词映射(中英文扩展版)
emotion_keywords = {
"happy": [
"开心",
"高兴",
"快乐",
"愉快",
"幸福",
"满意",
"",
"",
"不错",
"完美",
"棒极了",
"太好了",
"好呀",
"好的",
"happy",
"joy",
"great",
"good",
"nice",
"awesome",
"fantastic",
"wonderful",
],
"laughing": [
"哈哈",
"哈哈哈",
"呵呵",
"嘿嘿",
"嘻嘻",
"笑死",
"太好笑了",
"笑死我了",
"lol",
"lmao",
"haha",
"hahaha",
"hehe",
"rofl",
"funny",
"laugh",
],
"funny": [
"搞笑",
"滑稽",
"",
"幽默",
"笑点",
"段子",
"笑话",
"太逗了",
"hilarious",
"joke",
"comedy",
],
"sad": [
"伤心",
"难过",
"悲哀",
"悲伤",
"忧郁",
"郁闷",
"沮丧",
"失望",
"想哭",
"难受",
"不开心",
"",
"呜呜",
"sad",
"upset",
"unhappy",
"depressed",
"sorrow",
"gloomy",
],
"angry": [
"生气",
"愤怒",
"气死",
"讨厌",
"烦人",
"可恶",
"烦死了",
"恼火",
"暴躁",
"火大",
"愤怒",
"气炸了",
"angry",
"mad",
"annoyed",
"furious",
"pissed",
"hate",
],
"crying": [
"哭泣",
"泪流",
"大哭",
"伤心欲绝",
"泪目",
"流泪",
"哭死",
"哭晕",
"想哭",
"泪崩",
"cry",
"crying",
"tears",
"sob",
"weep",
],
"loving": [
"爱你",
"喜欢",
"",
"亲爱的",
"宝贝",
"么么哒",
"抱抱",
"想你",
"思念",
"最爱",
"亲亲",
"喜欢你",
"love",
"like",
"adore",
"darling",
"sweetie",
"honey",
"miss you",
"heart",
],
"embarrassed": [
"尴尬",
"不好意思",
"害羞",
"脸红",
"难为情",
"社死",
"丢脸",
"出丑",
"embarrassed",
"awkward",
"shy",
"blush",
],
"surprised": [
"惊讶",
"吃惊",
"天啊",
"哇塞",
"",
"居然",
"竟然",
"没想到",
"出乎意料",
"surprise",
"wow",
"omg",
"oh my god",
"amazing",
"unbelievable",
],
"shocked": [
"震惊",
"吓到",
"惊呆了",
"不敢相信",
"震撼",
"吓死",
"恐怖",
"害怕",
"吓人",
"shocked",
"shocking",
"scared",
"frightened",
"terrified",
"horror",
],
"thinking": [
"思考",
"考虑",
"想一下",
"琢磨",
"沉思",
"冥想",
"",
"思考中",
"在想",
"think",
"thinking",
"consider",
"ponder",
"meditate",
],
"winking": [
"调皮",
"眨眼",
"你懂的",
"坏笑",
"邪恶",
"奸笑",
"使眼色",
"wink",
"teasing",
"naughty",
"mischievous",
],
"cool": [
"",
"",
"厉害",
"棒极了",
"真棒",
"牛逼",
"",
"优秀",
"杰出",
"出色",
"完美",
"cool",
"awesome",
"amazing",
"great",
"impressive",
"perfect",
],
"relaxed": [
"放松",
"舒服",
"惬意",
"悠闲",
"轻松",
"舒适",
"安逸",
"自在",
"relax",
"relaxed",
"comfortable",
"cozy",
"chill",
"peaceful",
],
"delicious": [
"好吃",
"美味",
"",
"",
"可口",
"香甜",
"大餐",
"大快朵颐",
"流口水",
"垂涎",
"delicious",
"yummy",
"tasty",
"yum",
"appetizing",
"mouthwatering",
],
"kissy": [
"亲亲",
"么么",
"",
"mua",
"muah",
"亲一下",
"飞吻",
"kiss",
"xoxo",
"hug",
"muah",
"smooch",
],
"confident": [
"自信",
"肯定",
"确定",
"毫无疑问",
"当然",
"必须的",
"毫无疑问",
"确信",
"坚信",
"confident",
"sure",
"certain",
"definitely",
"positive",
],
"sleepy": [
"",
"睡觉",
"晚安",
"想睡",
"好累",
"疲惫",
"疲倦",
"困了",
"想休息",
"睡意",
"sleep",
"sleepy",
"tired",
"exhausted",
"bedtime",
"good night",
],
"silly": [
"",
"",
"",
"",
"",
"",
"憨憨",
"傻乎乎",
"呆萌",
"silly",
"stupid",
"dumb",
"foolish",
"goofy",
"ridiculous",
],
"confused": [
"疑惑",
"不明白",
"不懂",
"困惑",
"疑问",
"为什么",
"怎么回事",
"啥意思",
"不清楚",
"confused",
"puzzled",
"doubt",
"question",
"what",
"why",
"how",
],
}
# 特殊句型判断(中英文)
# 赞美他人
if any(
phrase in text
for phrase in [
"你真",
"你好",
"您真",
"你真棒",
"你好厉害",
"你太强了",
"你真好",
"你真聪明",
"you are",
"you're",
"you look",
"you seem",
"so smart",
"so kind",
]
):
return "loving"
# 自我赞美
if any(
phrase in text
for phrase in [
"我真",
"我最",
"我太棒了",
"我厉害",
"我聪明",
"我优秀",
"i am",
"i'm",
"i feel",
"so good",
"so happy",
]
):
return "cool"
# 晚安/睡觉相关
if any(
phrase in text
for phrase in [
"睡觉",
"晚安",
"睡了",
"好梦",
"休息了",
"去睡了",
"sleep",
"good night",
"bedtime",
"go to bed",
]
):
return "sleepy"
# 疑问句
if has_question and not has_exclamation:
return "thinking"
# 强烈情感(感叹号)
if has_exclamation and not has_question:
# 检查是否是积极内容
positive_words = (
emotion_keywords["happy"]
+ emotion_keywords["laughing"]
+ emotion_keywords["cool"]
)
if any(word in text for word in positive_words):
return "laughing"
# 检查是否是消极内容
negative_words = (
emotion_keywords["angry"]
+ emotion_keywords["sad"]
+ emotion_keywords["crying"]
)
if any(word in text for word in negative_words):
return "angry"
return "surprised"
# 省略号(表示犹豫或思考)
if has_ellipsis:
return "thinking"
# 关键词匹配(带权重)
emotion_scores = {emotion: 0 for emotion in emoji_map.keys()}
# 给匹配到的关键词加分
for emotion, keywords in emotion_keywords.items():
for keyword in keywords:
if keyword in text:
emotion_scores[emotion] += 1
# 给长文本中的重复关键词额外加分
if len(text) > 20: # 长文本
for emotion, keywords in emotion_keywords.items():
for keyword in keywords:
emotion_scores[emotion] += text.count(keyword) * 0.5
# 根据分数选择最可能的情感
max_score = max(emotion_scores.values())
if max_score == 0:
return "happy" # 默认
# 可能有多个情感同分,根据上下文选择最合适的
top_emotions = [e for e, s in emotion_scores.items() if s == max_score]
# 如果多个情感同分,使用以下优先级
priority_order = [
"laughing",
"crying",
"angry",
"surprised",
"shocked", # 强烈情感优先
"loving",
"happy",
"funny",
"cool", # 积极情感
"sad",
"embarrassed",
"confused", # 消极情感
"thinking",
"winking",
"relaxed", # 中性情感
"delicious",
"kissy",
"confident",
"sleepy",
"silly", # 特殊场景
]
for emotion in priority_order:
if emotion in top_emotions:
return emotion
return top_emotions[0] # 如果都不在优先级列表里,返回第一个
def audio_to_data(audio_file_path, is_opus=True):
# 获取文件后缀名
file_type = os.path.splitext(audio_file_path)[1]