Merge branch 'main' into add_news

This commit is contained in:
hrz
2025-05-09 15:25:15 +08:00
committed by GitHub
30 changed files with 432 additions and 313 deletions
@@ -177,5 +177,5 @@ public interface Constant {
/**
* 版本号
*/
public static final String VERSION = "0.4.1";
public static final String VERSION = "0.4.2";
}
@@ -0,0 +1,43 @@
-- 添加百度ASR模型配置
delete from `ai_model_config` where `id` = 'ASR_BaiduASR';
INSERT INTO `ai_model_config` VALUES ('ASR_BaiduASR', 'ASR', 'BaiduASR', '百度语音识别', 0, 1, '{\"type\": \"baidu\", \"app_id\": \"\", \"api_key\": \"\", \"secret_key\": \"\", \"dev_pid\": 1537, \"output_dir\": \"tmp/\"}', NULL, NULL, 7, NULL, NULL, NULL, NULL);
-- 添加百度ASR供应器
delete from `ai_model_provider` where `id` = 'SYSTEM_ASR_BaiduASR';
INSERT INTO `ai_model_provider` (`id`, `model_type`, `provider_code`, `name`, `fields`, `sort`, `creator`, `create_date`, `updater`, `update_date`) VALUES
('SYSTEM_ASR_BaiduASR', 'ASR', 'baidu', '百度语音识别', '[{"key":"app_id","label":"应用AppID","type":"string"},{"key":"api_key","label":"API Key","type":"string"},{"key":"secret_key","label":"Secret Key","type":"string"},{"key":"dev_pid","label":"语言参数","type":"number"},{"key":"output_dir","label":"输出目录","type":"string"}]', 7, 1, NOW(), 1, NOW());
-- 更新百度ASR配置说明
UPDATE `ai_model_config` SET
`doc_link` = 'https://console.bce.baidu.com/ai-engine/old/#/ai/speech/app/list',
`remark` = '百度ASR配置说明:
1. 访问 https://console.bce.baidu.com/ai-engine/old/#/ai/speech/app/list
2. 创建新应用
3. 获取AppID、API Key和Secret Key
4. 填入配置文件中
查看资源额度:https://console.bce.baidu.com/ai-engine/old/#/ai/speech/overview/resource/list
语言参数说明:https://ai.baidu.com/ai-doc/SPEECH/0lbxfnc9b
' WHERE `id` = 'ASR_BaiduASR';
-- 更新豆包供应器字段
update `ai_model_provider` set `fields` =
'[{"key":"appid","label":"应用ID","type":"string"},{"key":"access_token","label":"访问令牌","type":"string"},{"key":"cluster","label":"集群","type":"string"},{"key":"boosting_table_name","label":"热词文件名称","type":"string"},{"key":"correct_table_name","label":"替换词文件名称","type":"string"},{"key":"output_dir","label":"输出目录","type":"string"}]'
where `id` = 'SYSTEM_ASR_DoubaoASR';
-- 更新豆包ASR配置说明
UPDATE `ai_model_config` SET
`doc_link` = 'https://console.volcengine.com/speech/app',
`remark` = '豆包ASR配置说明:
1. 需要在火山引擎控制台创建应用并获取appid和access_token
2. 支持中文语音识别
3. 需要网络连接
4. 输出文件保存在tmp/目录
申请步骤:
1. 访问 https://console.volcengine.com/speech/app
2. 创建新应用
3. 获取appid和access_token
4. 填入配置文件中
如需设置热词,请参考:https://www.volcengine.com/docs/6561/155738
' WHERE `id` = 'ASR_DoubaoASR';
@@ -100,6 +100,13 @@ databaseChangeLog:
- sqlFile:
encoding: utf8
path: classpath:db/changelog/202505022134.sql
- changeSet:
id: 202505081146
author: hrz
changes:
- sqlFile:
encoding: utf8
path: classpath:db/changelog/202505081146.sql
- changeSet:
id: 202505091409
author: hrz
@@ -1,5 +1,5 @@
<template>
<el-dialog :visible="visible" @close="handleClose" width="400px" center>
<el-dialog :visible="visible" @close="handleClose" width="24%" center>
<div
style="margin: 0 10px 10px;display: flex;align-items: center;gap: 10px;font-weight: 700;font-size: 20px;text-align: left;color: #3d4566;">
<div
@@ -1,5 +1,5 @@
<template>
<el-dialog :visible="dialogVisible" @update:visible="handleVisibleChange" width="975px" center
<el-dialog :visible="dialogVisible" @update:visible="handleVisibleChange" width="57%" center
custom-class="custom-dialog" :show-close="false" class="center-dialog">
<div style="margin: 0 18px; text-align: left; padding: 10px; border-radius: 10px;">
<div style="font-size: 30px; color: #3d4566; margin-top: -10px; margin-bottom: 10px; text-align: center;">
@@ -54,7 +54,7 @@
</el-form-item>
<el-form-item label="备注" prop="remark" class="prop-remark">
<el-input v-model="formData.remark" type="textarea" :rows="3" placeholder="请输入模型备注"
<el-input v-model="formData.remark" type="textarea" :rows="3" placeholder="请输入模型备注" :autosize="{ minRows: 3, maxRows: 5 }"
class="custom-input-bg"></el-input>
</el-form-item>
</el-form>
@@ -271,7 +271,7 @@ export default {
}
.center-dialog .el-dialog {
margin: 4% 0 auto !important;
margin: 0 0 auto !important;
display: flex;
flex-direction: column;
}
@@ -1,5 +1,5 @@
<template>
<el-dialog :visible="visible" @close="handleClose" width="400px" center @open="handleOpen">
<el-dialog :visible="visible" @close="handleClose" width="25%" center @open="handleOpen">
<div
style="margin: 0 10px 10px;display: flex;align-items: center;gap: 10px;font-weight: 700;font-size: 20px;text-align: left;color: #3d4566;">
<div
@@ -10,7 +10,7 @@
</div>
<div style="height: 1px;background: #e8f0ff;" />
<div style="margin: 22px 15px;">
<div style="font-weight: 400;font-size: 14px;text-align: left;color: #3d4566;">
<div style="font-weight: 400;text-align: left;color: #3d4566;">
<div style="color: red;display: inline-block;">*</div> 智能体名称
</div>
<div class="input-46" style="margin-top: 12px;">
@@ -1,6 +1,6 @@
<template>
<form>
<el-dialog :visible.sync="value" width="400px" center>
<el-dialog :visible.sync="dialogVisible" width="24%" center>
<div
style="margin: 0 10px 10px;display: flex;align-items: center;gap: 10px;font-weight: 700;font-size: 20px;text-align: left;color: #3d4566;">
<div
@@ -60,11 +60,20 @@ export default {
},
data() {
return {
dialogVisible: this.value,
oldPassword: "",
newPassword: "",
confirmNewPassword: ""
}
},
watch: {
value(val) {
this.dialogVisible = val;
},
dialogVisible(val) {
this.$emit('input', val);
}
},
methods: {
...mapActions(['logout']), // 引入Vuex的logout action
confirm() {
@@ -101,7 +110,7 @@ export default {
this.$emit('input', false);
},
cancel() {
this.$emit('input', false);
this.dialogVisible = false;
this.resetForm();
},
resetForm() {
@@ -1,5 +1,5 @@
<template>
<el-dialog :title="title" :visible.sync="visible" width="500px" @close="handleClose">
<el-dialog :title="title" :visible.sync="dialogVisible" width="30%" @close="handleClose">
<el-form :model="form" :rules="rules" ref="form" label-width="100px">
<el-form-item label="字典标签" prop="dictLabel">
<el-input v-model="form.dictLabel" placeholder="请输入字典标签"></el-input>
@@ -41,6 +41,7 @@ export default {
},
data() {
return {
dialogVisible: this.visible,
form: {
id: null,
dictTypeId: null,
@@ -70,12 +71,18 @@ export default {
}
},
immediate: true
},
visible(val) {
this.dialogVisible = val;
},
dialogVisible(val) {
this.$emit('update:visible', val);
}
},
methods: {
handleClose() {
this.$emit('update:visible', false)
this.resetForm()
this.dialogVisible = false;
this.resetForm();
},
resetForm() {
this.form = {
@@ -102,4 +109,8 @@ export default {
.dialog-footer {
text-align: right;
}
:deep(.el-dialog) {
border-radius: 15px;
}
</style>
@@ -1,5 +1,5 @@
<template>
<el-dialog :title="title" :visible.sync="visible" width="500px" @close="handleClose">
<el-dialog :title="title" :visible.sync="dialogVisible" width="30%" @close="handleClose">
<el-form :model="form" :rules="rules" ref="form" label-width="120px">
<el-form-item label="字典类型名称" prop="dictName">
<el-input v-model="form.dictName" placeholder="请输入字典类型名称"></el-input>
@@ -34,6 +34,7 @@ export default {
},
data() {
return {
dialogVisible: this.visible,
form: {
id: null,
dictName: '',
@@ -46,6 +47,12 @@ export default {
}
},
watch: {
visible(val) {
this.dialogVisible = val;
},
dialogVisible(val) {
this.$emit('update:visible', val);
},
dictTypeData: {
handler(val) {
if (val) {
@@ -57,7 +64,7 @@ export default {
},
methods: {
handleClose() {
this.$emit('update:visible', false)
this.dialogVisible = false;
this.resetForm()
},
resetForm() {
@@ -83,4 +90,8 @@ export default {
.dialog-footer {
text-align: right;
}
:deep(.el-dialog) {
border-radius: 15px;
}
</style>
@@ -1,5 +1,5 @@
<template>
<el-dialog :title="title" :visible.sync="visible" width="500px" @close="handleClose" @open="handleOpen">
<el-dialog :title="title" :visible.sync="dialogVisible" width="30%" @close="handleClose" @open="handleOpen">
<el-form ref="form" :model="form" :rules="rules" label-width="100px">
<el-form-item label="固件名称" prop="firmwareName">
<el-input v-model="form.firmwareName" placeholder="请输入固件名称(板子+版本号)"></el-input>
@@ -59,11 +59,13 @@ export default {
default: () => []
}
},
data() {
return {
uploadProgress: 0,
uploadStatus: '',
isUploading: false,
dialogVisible: this.visible,
rules: {
firmwareName: [
{ required: true, message: '请输入固件名称(板子+版本号)', trigger: 'blur' }
@@ -90,10 +92,18 @@ export default {
created() {
// 移除 getDictDataByType 调用
},
watch: {
visible(val) {
this.dialogVisible = val;
},
dialogVisible(val) {
this.$emit('update:visible', val);
},
},
methods: {
// 移除 getFirmwareTypes 方法
handleClose() {
this.$refs.form.clearValidate();
this.dialogVisible = false;
this.$emit('cancel');
},
handleCancel() {
@@ -201,13 +211,17 @@ export default {
</script>
<style lang="scss" scoped>
::v-deep .el-dialog {
border-radius: 20px;
}
.upload-demo {
text-align: left;
}
.el-upload__tip {
line-height: 1.2;
padding-top: 5px;
padding-top: 2%;
color: #909399;
}
@@ -1,6 +1,6 @@
<template>
<el-dialog :visible.sync="dialogVisible" width="975px" center custom-class="custom-dialog" :show-close="false"
class="center-dialog">
<el-dialog :visible.sync="dialogVisible" width="57%" center custom-class="custom-dialog" :show-close="false"
class="center-dialog" >
<div style="margin: 0 18px; text-align: left; padding: 10px; border-radius: 10px;">
<div style="font-size: 30px; color: #3d4566; margin-top: -10px; margin-bottom: 10px; text-align: center;">
修改模型
@@ -53,7 +53,7 @@
</el-form-item>
<el-form-item label="备注" prop="remark" class="prop-remark">
<el-input v-model="form.remark" type="textarea" :rows="3" placeholder="请输入模型备注"
<el-input v-model="form.remark" type="textarea" :rows="3" placeholder="请输入模型备注" :autosize="{ minRows: 3, maxRows: 5 }"
class="custom-input-bg"></el-input>
</el-form-item>
</el-form>
@@ -296,7 +296,7 @@ export default {
};
</script>
<style scoped>
<style lang="scss" scoped>
.custom-dialog {
position: relative;
border-radius: 20px;
@@ -316,11 +316,6 @@ export default {
justify-content: center;
}
.center-dialog .el-dialog {
margin: 4% 0 auto !important;
display: flex;
flex-direction: column;
}
.custom-close-btn {
position: absolute;
+1 -8
View File
@@ -487,7 +487,7 @@ export default {
};
</script>
<style scoped>
<style lang="scss" scoped>
::v-deep .el-dialog {
border-radius: 8px !important;
@@ -648,12 +648,6 @@ export default {
margin: 0 auto;
}
/* 新增按钮组样式 */
.action-buttons {
bottom: 20px;
padding-top: 10px;
}
.action-buttons .el-button {
padding: 8px 15px;
font-size: 11px;
@@ -692,7 +686,6 @@ export default {
position: static;
padding: 15px 0;
background: white;
box-shadow: 0 -2px 12px rgba(0,0,0,0.05);
}
/* 输入框自适应 */
+32 -16
View File
@@ -49,9 +49,13 @@
v-loading="dictDataLoading" element-loading-text="拼命加载中"
element-loading-spinner="el-icon-loading"
element-loading-background="rgba(255, 255, 255, 0.7)"
@selection-change="handleDictDataSelectionChange" class="data-table"
class="data-table"
header-row-class-name="table-header">
<el-table-column type="selection" width="55" align="center"></el-table-column>
<el-table-column label="选择" align="center" width="55">
<template slot-scope="scope">
<el-checkbox v-model="scope.row.selected"></el-checkbox>
</template>
</el-table-column>
<el-table-column label="字典标签" prop="dictLabel" align="center"></el-table-column>
<el-table-column label="字典值" prop="dictValue" align="center"></el-table-column>
<el-table-column label="排序" prop="sort" align="center"></el-table-column>
@@ -153,7 +157,6 @@ export default {
// 字典数据相关
dictDataList: [],
dictDataLoading: false,
selectedDictData: [],
isAllDictDataSelected: false,
dictDataDialogVisible: false,
dictDataDialogTitle: '新增字典数据',
@@ -265,7 +268,10 @@ export default {
dictValue: ''
}, ({ data }) => {
if (data.code === 0) {
this.dictDataList = data.data.list
this.dictDataList = data.data.list.map(item => ({
...item,
selected: false
}))
this.total = data.data.total
} else {
this.$message.error(data.msg || '获取字典数据失败')
@@ -273,16 +279,11 @@ export default {
this.dictDataLoading = false
})
},
handleDictDataSelectionChange(val) {
this.selectedDictData = val
this.isAllDictDataSelected = val.length === this.dictDataList.length
},
selectAllDictData() {
if (this.isAllDictDataSelected) {
this.$refs.dictDataTable.clearSelection()
} else {
this.$refs.dictDataTable.toggleAllSelection()
}
this.isAllDictDataSelected = !this.isAllDictDataSelected
this.dictDataList.forEach(row => {
row.selected = this.isAllDictDataSelected
})
},
showAddDictDataDialog() {
if (!this.selectedDictType) {
@@ -329,17 +330,18 @@ export default {
})
},
batchDeleteDictData() {
if (this.selectedDictData.length === 0) {
const selectedRows = this.dictDataList.filter(row => row.selected)
if (selectedRows.length === 0) {
this.$message.warning('请选择要删除的字典数据')
return
}
this.$confirm('确定要删除选中的字典数据吗?', '提示', {
this.$confirm(`确定要删除选中的${selectedRows.length}字典数据吗?`, '提示', {
confirmButtonText: '确定',
cancelButtonText: '取消',
type: 'warning'
}).then(() => {
const ids = this.selectedDictData.map(item => item.id)
const ids = selectedRows.map(item => item.id)
dictApi.deleteDictData(ids, ({ data }) => {
if (data.code === 0) {
this.$message.success('删除成功')
@@ -832,4 +834,18 @@ export default {
flex: 1;
overflow: hidden;
}
:deep(.el-checkbox__inner) {
background-color: #eeeeee !important;
border-color: #cccccc !important;
}
:deep(.el-checkbox__inner:hover) {
border-color: #cccccc !important;
}
:deep(.el-checkbox__input.is-checked .el-checkbox__inner) {
background-color: #5f70f3 !important;
border-color: #5f70f3 !important;
}
</style>
+6 -7
View File
@@ -318,7 +318,6 @@ export default {
<style scoped>
.welcome {
min-width: 900px;
min-height: 506px;
height: 100vh;
display: flex;
position: relative;
@@ -334,7 +333,7 @@ export default {
display: flex;
justify-content: space-between;
align-items: center;
padding: 16px 24px;
padding: 1.5vh 24px;
}
.page-title {
@@ -344,7 +343,7 @@ export default {
}
.main-wrapper {
margin: 5px 22px;
margin: 1vh 22px;
border-radius: 15px;
height: calc(100vh - 24vh);
box-shadow: 0 2px 12px rgba(0, 0, 0, 0.1);
@@ -416,7 +415,7 @@ export default {
}
.form-content {
padding: 20px 0;
padding: 2vh 0;
}
.form-grid {
@@ -450,11 +449,11 @@ export default {
}
.template-item {
height: 37px;
height: 4vh;
width: 76px;
border-radius: 8px;
background: #e6ebff;
line-height: 37px;
line-height: 4vh;
font-weight: 400;
font-size: 11px;
text-align: center;
@@ -471,7 +470,7 @@ export default {
display: flex;
flex-wrap: wrap;
gap: 8px;
margin-top: 20px;
margin-top: 2vh;
align-items: center;
}
+1 -1
View File
@@ -4,7 +4,7 @@ from loguru import logger
from config.config_loader import load_config
from config.settings import check_config_file
SERVER_VERSION = "0.4.1"
SERVER_VERSION = "0.4.2"
def get_module_abbreviation(module_name, module_dict):
+16 -8
View File
@@ -138,6 +138,8 @@ class ConnectionHandler:
int(self.config.get("close_connection_no_voice_time", 120)) + 60
) # 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"
async def handle_connection(self, ws):
try:
# 获取并验证headers
@@ -241,9 +243,12 @@ class ConnectionHandler:
def _initialize_components(self):
"""初始化组件"""
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]}...")
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:
@@ -811,9 +816,12 @@ class ConnectionHandler:
f"TTS生成:文件路径: {tts_file}"
)
if os.path.exists(tts_file):
opus_datas, _ = self.tts.audio_to_opus_data(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(使用文件路径)
enqueue_tts_report(self, 2, text, opus_datas)
enqueue_tts_report(self, 2, text, audio_datas)
else:
self.logger.bind(tag=TAG).error(
f"TTS出错:文件不存在{tts_file}"
@@ -824,7 +832,7 @@ class ConnectionHandler:
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((opus_datas, text, text_index))
self.audio_play_queue.put((audio_datas, text, text_index))
if (
self.tts.delete_audio_file
and tts_file is not None
@@ -855,13 +863,13 @@ class ConnectionHandler:
text = None
try:
try:
opus_datas, text, text_index = self.audio_play_queue.get(timeout=1)
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, opus_datas, text, text_index), self.loop
sendAudioMessage(self, audio_datas, text, text_index), self.loop
)
future.result()
except Exception as e:
+10 -2
View File
@@ -1,5 +1,4 @@
import json
from config.logger import setup_logging
from core.handle.sendAudioHandle import send_stt_message
from core.utils.util import remove_punctuation_and_length
import shutil
@@ -20,7 +19,16 @@ WAKEUP_CONFIG = {
}
async def handleHelloMessage(conn):
async def handleHelloMessage(conn, msg_json):
"""处理hello消息"""
audio_params = msg_json.get("audio_params")
if audio_params:
format = audio_params.get("format")
conn.logger.bind(tag=TAG).info(f"客户端音频格式: {format}")
conn.audio_format = format
conn.asr.set_audio_format(format)
conn.welcome_msg["audio_params"] = audio_params
await conn.websocket.send(json.dumps(conn.welcome_msg))
@@ -5,7 +5,7 @@ from core.handle.sendAudioHandle import send_stt_message
from core.handle.intentHandler import handle_user_intent
from core.utils.output_counter import check_device_output_limit
from core.handle.ttsReportHandle import enqueue_tts_report
from core.providers.tts.base import audio_to_opus_data
from core.utils.util import audio_to_data
TAG = __name__
@@ -111,7 +111,7 @@ async def max_out_size(conn):
conn.tts_last_text_index = 0
conn.llm_finish_task = True
file_path = "config/assets/max_output_size.wav"
opus_packets, _ = audio_to_opus_data(file_path)
opus_packets, _ = audio_to_data(file_path)
conn.audio_play_queue.put((opus_packets, text, 0))
conn.close_after_chat = True
@@ -133,7 +133,7 @@ async def check_bind_device(conn):
# 播放提示音
music_path = "config/assets/bind_code.wav"
opus_packets, _ = audio_to_opus_data(music_path)
opus_packets, _ = audio_to_data(music_path)
conn.audio_play_queue.put((opus_packets, text, 0))
# 逐个播放数字
@@ -141,7 +141,7 @@ async def check_bind_device(conn):
try:
digit = conn.bind_code[i]
num_path = f"config/assets/bind_code/{digit}.wav"
num_packets, _ = audio_to_opus_data(num_path)
num_packets, _ = audio_to_data(num_path)
conn.audio_play_queue.put((num_packets, None, i + 1))
except Exception as e:
conn.logger.bind(tag=TAG).error(f"播放数字音频失败: {e}")
@@ -153,5 +153,5 @@ async def check_bind_device(conn):
conn.tts_last_text_index = 0
conn.llm_finish_task = True
music_path = "config/assets/bind_not_found.wav"
opus_packets, _ = audio_to_opus_data(music_path)
opus_packets, _ = audio_to_data(music_path)
conn.audio_play_queue.put((opus_packets, text, 0))
@@ -115,7 +115,7 @@ async def send_tts_message(conn, state, text=None):
stop_tts_notify_voice = conn.config.get(
"stop_tts_notify_voice", "config/assets/tts_notify.mp3"
)
audios, duration = conn.tts.audio_to_opus_data(stop_tts_notify_voice)
audios, _ = conn.tts.audio_to_opus_data(stop_tts_notify_voice)
await sendAudio(conn, audios)
# 清除服务端讲话状态
conn.clearSpeakStatus()
@@ -20,7 +20,7 @@ async def handleTextMessage(conn, message):
await conn.websocket.send(message)
return
if msg_json["type"] == "hello":
await handleHelloMessage(conn)
await handleHelloMessage(conn, msg_json)
elif msg_json["type"] == "abort":
await handleAbortMessage(conn)
elif msg_json["type"] == "listen":
@@ -20,64 +20,78 @@ from core.providers.asr.base import ASRProviderBase
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', '~')
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', '~')
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'}
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",
}
# 构造规范化的请求字符串
query_string = AccessToken._encode_dict(parameters)
# print('规范化的请求字符串: %s' % query_string)
# 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)
# 计算签名
secreted_string = hmac.new(bytes(access_key_secret + '&', encoding='utf-8'),
bytes(string_to_sign, encoding='utf-8'),
hashlib.sha1).digest()
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)
# print('签名: %s' % signature)
# 进行URL编码
signature = AccessToken._encode_text(signature)
# print('URL编码后的签名: %s' % signature)
# 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)
# 提交HTTP GET请求
response = requests.get(full_url)
if response.ok:
root_obj = response.json()
key = 'Token'
key = "Token"
if key in root_obj:
token = root_obj[key]['Id']
expire_time = root_obj[key]['ExpireTime']
token = root_obj[key]["Id"]
expire_time = root_obj[key]["ExpireTime"]
return token, expire_time
# print(response.text)
return None, None
class ASRProvider(ASRProviderBase):
def __init__(self, config: dict, delete_audio_file: bool):
super().__init__()
"""阿里云ASR初始化"""
# 新增空值判断逻辑
self.access_key_id = config.get("access_key_id")
@@ -102,28 +116,23 @@ class ASRProvider(ASRProviderBase):
# 确保输出目录存在
os.makedirs(self.output_dir, exist_ok=True)
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
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")
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()
if expire_str.isdigit():
expire_time = datetime.fromtimestamp(int(expire_str))
else:
expire_time = datetime.strptime(
expire_str,
"%Y-%m-%dT%H:%M:%SZ"
)
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
@@ -145,9 +154,12 @@ class ASRProvider(ASRProviderBase):
# f"过期时间 {datetime.fromtimestamp(self.expire_time)} | "
# f"剩余 {remaining:.2f}秒")
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 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:
"""构造请求URL,包含参数"""
@@ -159,20 +171,6 @@ class ASRProvider(ASRProviderBase):
request += "&enable_voice_detection=false"
return request
def decode_opus(self, opus_data: List[bytes], session_id: str) -> List[bytes]:
"""将Opus数据解码为PCM"""
decoder = opuslib_next.Decoder(16000, 1) # 16kHz, 单声道
pcm_data = []
for opus_packet in opus_data:
try:
pcm_frame = decoder.decode(opus_packet, 960) # 960 samples = 60ms
pcm_data.append(pcm_frame)
except opuslib_next.OpusError as e:
logger.bind(tag=TAG).error(f"Opus解码错误: {e}", exc_info=True)
return pcm_data
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]
@@ -183,7 +181,7 @@ class ASRProvider(ASRProviderBase):
wf.setnchannels(1) # 单声道
wf.setsampwidth(2) # 16-bit
wf.setframerate(self.sample_rate)
wf.writeframes(b''.join(pcm_data))
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
@@ -193,9 +191,9 @@ class ASRProvider(ASRProviderBase):
try:
# 设置HTTP头
headers = {
'X-NLS-Token': self.token,
'Content-type': 'application/octet-stream',
'Content-Length': str(len(pcm_data))
"X-NLS-Token": self.token,
"Content-type": "application/octet-stream",
"Content-Length": str(len(pcm_data)),
}
# 创建连接并发送请求
@@ -203,12 +201,12 @@ class ASRProvider(ASRProviderBase):
request_url = self._construct_request_url()
loop = asyncio.get_event_loop()
await loop.run_in_executor(None, lambda: conn.request(
method='POST',
url=request_url,
body=pcm_data,
headers=headers
))
await loop.run_in_executor(
None,
lambda: conn.request(
method="POST", url=request_url, body=pcm_data, headers=headers
),
)
# 获取响应
response = await loop.run_in_executor(None, conn.getresponse)
@@ -218,10 +216,10 @@ class ASRProvider(ASRProviderBase):
# 解析响应
try:
body_json = json.loads(body)
status = body_json.get('status')
status = body_json.get("status")
if status == 20000000:
result = body_json.get('result', '')
result = body_json.get("result", "")
logger.bind(tag=TAG).debug(f"ASR结果: {result}")
return result
else:
@@ -236,7 +234,9 @@ class ASRProvider(ASRProviderBase):
logger.bind(tag=TAG).error(f"ASR请求失败: {e}", exc_info=True)
return None
async def speech_to_text(self, opus_data: List[bytes], session_id: str) -> Tuple[Optional[str], Optional[str]]:
async def speech_to_text(
self, opus_data: List[bytes], session_id: str
) -> Tuple[Optional[str], Optional[str]]:
"""将语音数据转换为文本"""
if self._is_token_expired():
logger.warning("Token已过期,正在自动刷新...")
@@ -245,8 +245,11 @@ class ASRProvider(ASRProviderBase):
file_path = None
try:
# 解码Opus为PCM
pcm_data = self.decode_opus(opus_data, session_id)
combined_pcm_data = b''.join(pcm_data)
if self.audio_format == "pcm":
pcm_data = opus_data
else:
pcm_data = self.decode_opus(opus_data)
combined_pcm_data = b"".join(pcm_data)
# 判断是否保存为WAV文件
if self.delete_audio_file:
@@ -264,4 +267,4 @@ class ASRProvider(ASRProviderBase):
except Exception as e:
logger.bind(tag=TAG).error(f"语音识别失败: {e}", exc_info=True)
return "", file_path
return "", file_path
@@ -20,6 +20,7 @@ logger = setup_logging()
class ASRProvider(ASRProviderBase):
def __init__(self, config: dict, delete_audio_file: bool = True):
super().__init__()
self.app_id = config.get("app_id")
self.api_key = config.get("api_key")
self.secret_key = config.get("secret_key")
@@ -49,22 +50,6 @@ class ASRProvider(ASRProviderBase):
return file_path
@staticmethod
def decode_opus(opus_data: List[bytes]) -> bytes:
"""将Opus音频数据解码为PCM数据"""
decoder = opuslib_next.Decoder(16000, 1) # 16kHz, 单声道
pcm_data = []
for opus_packet in opus_data:
try:
pcm_frame = decoder.decode(opus_packet, 960) # 960 samples = 60ms
pcm_data.append(pcm_frame)
except opuslib_next.OpusError as e:
logger.bind(tag=TAG).error(f"Opus解码错误: {e}", exc_info=True)
return pcm_data
async def speech_to_text(
self, opus_data: List[bytes], session_id: str
) -> Tuple[Optional[str], Optional[str]]:
@@ -81,7 +66,10 @@ class ASRProvider(ASRProviderBase):
return None, file_path
# 将Opus音频数据解码为PCM
pcm_data = self.decode_opus(opus_data)
if self.audio_format == "pcm":
pcm_data = opus_data
else:
pcm_data = self.decode_opus(opus_data)
combined_pcm_data = b"".join(pcm_data)
# 判断是否保存为WAV文件
+27 -2
View File
@@ -1,6 +1,6 @@
from abc import ABC, abstractmethod
from typing import Optional, Tuple, List
import opuslib_next
from config.logger import setup_logging
TAG = __name__
@@ -8,12 +8,37 @@ logger = setup_logging()
class ASRProviderBase(ABC):
def __init__(self):
self.audio_format = "opus"
@abstractmethod
def save_audio_to_file(self, pcm_data: List[bytes], session_id: str) -> str:
"""PCM数据保存为WAV文件"""
pass
@abstractmethod
async def speech_to_text(self, opus_data: List[bytes], session_id: str) -> Tuple[Optional[str], Optional[str]]:
async def speech_to_text(
self, opus_data: List[bytes], session_id: str
) -> Tuple[Optional[str], Optional[str]]:
"""将语音数据转换为文本"""
pass
def set_audio_format(self, format: str) -> None:
"""设置音频格式"""
self.audio_format = format
@staticmethod
def decode_opus(opus_data: List[bytes]) -> bytes:
"""将Opus音频数据解码为PCM数据"""
decoder = opuslib_next.Decoder(16000, 1) # 16kHz, 单声道
pcm_data = []
for opus_packet in opus_data:
try:
pcm_frame = decoder.decode(opus_packet, 960) # 960 samples = 60ms
pcm_data.append(pcm_frame)
except opuslib_next.OpusError as e:
logger.bind(tag=TAG).error(f"Opus解码错误: {e}", exc_info=True)
return pcm_data
@@ -85,11 +85,12 @@ def parse_response(res):
class ASRProvider(ASRProviderBase):
def __init__(self, config: dict, delete_audio_file: bool):
super().__init__()
self.appid = config.get("appid")
self.cluster = config.get("cluster")
self.access_token = config.get("access_token")
self.boosting_table_name = config.get("boosting_table_name")
self.correct_table_name = config.get("correct_table_name")
self.boosting_table_name = config.get("boosting_table_name", "")
self.correct_table_name = config.get("correct_table_name", "")
self.output_dir = config.get("output_dir")
self.delete_audio_file = delete_audio_file
@@ -226,21 +227,6 @@ class ASRProvider(ASRProviderBase):
logger.bind(tag=TAG).error(f"ASR request failed: {e}", exc_info=True)
return None
@staticmethod
def decode_opus(opus_data: List[bytes], session_id: str) -> List[bytes]:
decoder = opuslib_next.Decoder(16000, 1) # 16kHz, 单声道
pcm_data = []
for opus_packet in opus_data:
try:
pcm_frame = decoder.decode(opus_packet, 960) # 960 samples = 60ms
pcm_data.append(pcm_frame)
except opuslib_next.OpusError as e:
logger.bind(tag=TAG).error(f"Opus解码错误: {e}", exc_info=True)
return pcm_data
@staticmethod
def slice_data(data: bytes, chunk_size: int) -> (list, bool):
"""
@@ -265,7 +251,10 @@ class ASRProvider(ASRProviderBase):
file_path = None
try:
# 合并所有opus数据包
pcm_data = self.decode_opus(opus_data, session_id)
if self.audio_format == "pcm":
pcm_data = opus_data
else:
pcm_data = self.decode_opus(opus_data)
combined_pcm_data = b"".join(pcm_data)
# 判断是否保存为WAV文件
@@ -6,9 +6,7 @@ import io
from config.logger import setup_logging
from typing import Optional, Tuple, List
import uuid
import opuslib_next
from core.providers.asr.base import ASRProviderBase
from funasr import AutoModel
from funasr.utils.postprocess_utils import rich_transcription_postprocess
@@ -35,6 +33,7 @@ class CaptureOutput:
class ASRProvider(ASRProviderBase):
def __init__(self, config: dict, delete_audio_file: bool):
super().__init__()
self.model_dir = config.get("model_dir")
self.output_dir = config.get("output_dir") # 修正配置键名
self.delete_audio_file = delete_audio_file
@@ -46,7 +45,7 @@ class ASRProvider(ASRProviderBase):
model=self.model_dir,
vad_kwargs={"max_single_segment_time": 30000},
disable_update=True,
hub="hf"
hub="hf",
# device="cuda:0", # 启用GPU加速
)
@@ -64,27 +63,18 @@ class ASRProvider(ASRProviderBase):
return file_path
@staticmethod
def decode_opus(opus_data: List[bytes], session_id: str) -> List[bytes]:
decoder = opuslib_next.Decoder(16000, 1) # 16kHz, 单声道
pcm_data = []
for opus_packet in opus_data:
try:
pcm_frame = decoder.decode(opus_packet, 960) # 960 samples = 60ms
pcm_data.append(pcm_frame)
except opuslib_next.OpusError as e:
logger.bind(tag=TAG).error(f"Opus解码错误: {e}", exc_info=True)
return pcm_data
async def speech_to_text(self, opus_data: List[bytes], session_id: str) -> Tuple[Optional[str], Optional[str]]:
async def speech_to_text(
self, opus_data: List[bytes], session_id: str
) -> Tuple[Optional[str], Optional[str]]:
"""语音转文本主处理逻辑"""
file_path = None
try:
# 合并所有opus数据包
pcm_data = self.decode_opus(opus_data, session_id)
if self.audio_format == "pcm":
pcm_data = opus_data
else:
pcm_data = self.decode_opus(opus_data)
combined_pcm_data = b"".join(pcm_data)
# 判断是否保存为WAV文件
@@ -103,7 +93,9 @@ class ASRProvider(ASRProviderBase):
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}")
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
@@ -9,22 +9,29 @@ import wave
import websockets
from config.logger import setup_logging
import asyncio
TAG = __name__
logger = setup_logging()
class ASRProvider(ASRProviderBase):
def __init__(self, config: dict, delete_audio_file: bool):
'''
"""
Initialize the ASRProvider with server configuration.
:param config: Dictionary containing 'host', 'port', and 'is_ssl'.
:param delete_audio_file: Boolean to indicate whether to delete audio files after processing.
'''
self.host = config.get('host', 'localhost')
self.port = config.get('port', 10095)
self.is_ssl = config.get('is_ssl', True)
"""
super().__init__()
self.host = config.get("host", "localhost")
self.port = config.get("port", 10095)
self.is_ssl = config.get("is_ssl", True)
self.output_dir = config.get("output_dir")
self.delete_audio_file = delete_audio_file
self.uri = f"wss://{self.host}:{self.port}" if self.is_ssl else f"ws://{self.host}:{self.port}"
self.uri = (
f"wss://{self.host}:{self.port}"
if self.is_ssl
else f"ws://{self.host}:{self.port}"
)
self.ssl_context = ssl.SSLContext() if self.is_ssl else None
if self.ssl_context:
self.ssl_context.check_hostname = False
@@ -44,28 +51,11 @@ class ASRProvider(ASRProviderBase):
return file_path
@staticmethod
def decode_opus(opus_data: List[bytes]) -> bytes:
"""将Opus音频数据解码为PCM数据"""
decoder = opuslib_next.Decoder(16000, 1) # 16kHz, 单声道
pcm_data = []
for opus_packet in opus_data:
try:
pcm_frame = decoder.decode(opus_packet, 960) # 960 samples = 60ms
pcm_data.append(pcm_frame)
except opuslib_next.OpusError as e:
logger.bind(tag=TAG).error(f"Opus解码错误: {e}", exc_info=True)
return pcm_data
async def _receive_responses(self, ws) -> None:
'''
"""
Asynchronous generator to receive messages from the WebSocket.
Yields each message as it is received.
'''
"""
text = ""
while True:
try:
@@ -78,30 +68,35 @@ class ASRProvider(ASRProviderBase):
else:
text += response_data.get("text", "")
except asyncio.TimeoutError:
logger.bind(tag=TAG).error("Timeout while waiting for response from WebSocket.")
logger.bind(tag=TAG).error(
"Timeout while waiting for response from WebSocket."
)
break
except websockets.exceptions.ConnectionClosed as e:
logger.bind(tag=TAG).error(f"WebSocket connection closed: {e}")
break
return text
async def _send_data(self, ws, pcm_data: bytes, session_id: str) -> tuple:
'''
"""
Internal method to handle WebSocket communication.
Reuses the persistent WebSocket connection if available.
:param pcm_data: PCM audio data to send.
:param session_id: Unique session identifier.
:return: Tuple containing recognized text and optional timestamp.
'''
"""
# Send initial configuration message
config_message = json.dumps({
"mode": "offline",
"chunk_size": [5, 10, 5],
"chunk_interval": 10,
"wav_name": session_id,
"is_speaking": True,
"itn": False
})
config_message = json.dumps(
{
"mode": "offline",
"chunk_size": [5, 10, 5],
"chunk_interval": 10,
"wav_name": session_id,
"is_speaking": True,
"itn": False,
}
)
await ws.send(config_message)
logger.bind(tag=TAG).debug(f"Sent configuration message: {config_message}")
@@ -114,16 +109,20 @@ class ASRProvider(ASRProviderBase):
await ws.send(end_message)
logger.bind(tag=TAG).debug(f"Sent end message: {end_message}")
async def speech_to_text(self, opus_data: List[bytes], session_id: str) -> Tuple[Optional[str], Optional[str]]:
'''
async def speech_to_text(
self, opus_data: List[bytes], session_id: str
) -> Tuple[Optional[str], Optional[str]]:
"""
Convert speech data to text using FunASR.
:param opus_data: List of Opus-encoded audio data chunks.
:param session_id: Unique session identifier.
:return: Tuple containing recognized text and optional timestamp.
'''
"""
file_path = None
pcm_data = self.decode_opus(opus_data)
if self.audio_format == "pcm":
pcm_data = opus_data
else:
pcm_data = self.decode_opus(opus_data)
combined_pcm_data = b"".join(pcm_data)
# 判断是否保存为WAV文件
@@ -132,16 +131,19 @@ class ASRProvider(ASRProviderBase):
else:
file_path = self.save_audio_to_file(pcm_data, session_id)
async with websockets.connect(self.uri, subprotocols=["binary"], ping_interval=None, ssl=self.ssl_context) as ws:
async with websockets.connect(
self.uri, subprotocols=["binary"], ping_interval=None, ssl=self.ssl_context
) as ws:
try:
# Use asyncio to handle WebSocket communication
send_task = asyncio.create_task(self._send_data(ws, combined_pcm_data, session_id))
send_task = asyncio.create_task(
self._send_data(ws, combined_pcm_data, session_id)
)
receive_task = asyncio.create_task(self._receive_responses(ws))
# Gather tasks with error handling
done, pending = await asyncio.wait(
[send_task, receive_task],
return_when=asyncio.FIRST_EXCEPTION
[send_task, receive_task], return_when=asyncio.FIRST_EXCEPTION
)
# Cancel any pending tasks
@@ -155,11 +157,16 @@ class ASRProvider(ASRProviderBase):
# Get the result from the receive task
result = receive_task.result()
return result, file_path # Return the recognized text and timestamp (if any)
return (
result,
file_path,
) # Return the recognized text and timestamp (if any)
except websockets.exceptions.ConnectionClosed as e:
logger.bind(tag=TAG).error(f"WebSocket connection closed: {e}")
return "", file_path
except Exception as e:
logger.bind(tag=TAG).error(f"Error during speech-to-text conversion: {e}", exc_info=True)
return "", file_path
logger.bind(tag=TAG).error(
f"Error during speech-to-text conversion: {e}", exc_info=True
)
return "", file_path
@@ -37,6 +37,7 @@ class CaptureOutput:
class ASRProvider(ASRProviderBase):
def __init__(self, config: dict, delete_audio_file: bool):
super().__init__()
self.model_dir = config.get("model_dir")
self.output_dir = config.get("output_dir")
self.delete_audio_file = delete_audio_file
@@ -97,21 +98,6 @@ class ASRProvider(ASRProviderBase):
return file_path
@staticmethod
def decode_opus(opus_data: List[bytes], session_id: str) -> List[bytes]:
decoder = opuslib_next.Decoder(16000, 1) # 16kHz, 单声道
pcm_data = []
for opus_packet in opus_data:
try:
pcm_frame = decoder.decode(opus_packet, 960) # 960 samples = 60ms
pcm_data.append(pcm_frame)
except opuslib_next.OpusError as e:
logger.bind(tag=TAG).error(f"Opus解码错误: {e}", exc_info=True)
return pcm_data
def read_wave(self, wave_filename: str) -> Tuple[np.ndarray, int]:
"""
Args:
@@ -144,7 +130,10 @@ class ASRProvider(ASRProviderBase):
try:
# 保存音频文件
start_time = time.time()
pcm_data = self.decode_opus(opus_data, session_id)
if self.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}"
@@ -17,12 +17,14 @@ from config.logger import setup_logging
TAG = __name__
logger = setup_logging()
class ASRProvider(ASRProviderBase):
API_URL = "https://asr.tencentcloudapi.com"
API_VERSION = "2019-06-14"
FORMAT = "pcm" # 支持的音频格式:pcm, wav, mp3
def __init__(self, config: dict, delete_audio_file: bool = True):
super().__init__()
self.secret_id = config.get("secret_id")
self.secret_key = config.get("secret_key")
self.output_dir = config.get("output_dir")
@@ -45,23 +47,9 @@ class ASRProvider(ASRProviderBase):
return file_path
@staticmethod
def decode_opus(opus_data: List[bytes]) -> bytes:
"""将Opus音频数据解码为PCM数据"""
decoder = opuslib_next.Decoder(16000, 1) # 16kHz, 单声道
pcm_data = []
for opus_packet in opus_data:
try:
pcm_frame = decoder.decode(opus_packet, 960) # 960 samples = 60ms
pcm_data.append(pcm_frame)
except opuslib_next.OpusError as e:
logger.bind(tag=TAG).error(f"Opus解码错误: {e}", exc_info=True)
return pcm_data
async def speech_to_text(self, opus_data: List[bytes], session_id: str) -> Tuple[Optional[str], Optional[str]]:
async def speech_to_text(
self, opus_data: List[bytes], session_id: str
) -> Tuple[Optional[str], Optional[str]]:
"""将语音数据转换为文本"""
if not opus_data:
logger.bind(tag=TAG).warn("音频数据为空!")
@@ -75,7 +63,10 @@ class ASRProvider(ASRProviderBase):
return None, file_path
# 将Opus音频数据解码为PCM
pcm_data = self.decode_opus(opus_data)
if self.audio_format == "pcm":
pcm_data = opus_data
else:
pcm_data = self.decode_opus(opus_data)
combined_pcm_data = b"".join(pcm_data)
# 判断是否保存为WAV文件
@@ -85,7 +76,7 @@ class ASRProvider(ASRProviderBase):
self.save_audio_to_file(pcm_data, session_id)
# 将音频数据转换为Base64编码
base64_audio = base64.b64encode(combined_pcm_data).decode('utf-8')
base64_audio = base64.b64encode(combined_pcm_data).decode("utf-8")
# 构建请求体
request_body = self._build_request_body(base64_audio)
@@ -98,7 +89,9 @@ class ASRProvider(ASRProviderBase):
result = self._send_request(request_body, timestamp, authorization)
if result:
logger.bind(tag=TAG).debug(f"腾讯云语音识别耗时: {time.time() - start_time:.3f}s | 结果: {result}")
logger.bind(tag=TAG).debug(
f"腾讯云语音识别耗时: {time.time() - start_time:.3f}s | 结果: {result}"
)
return result, file_path
@@ -115,7 +108,7 @@ class ASRProvider(ASRProviderBase):
"SourceType": 1, # 音频数据来源为语音文件
"VoiceFormat": self.FORMAT, # 音频格式
"Data": base64_audio, # Base64编码的音频数据
"DataLen": len(base64_audio) # 数据长度
"DataLen": len(base64_audio), # 数据长度
}
return json.dumps(request_map)
@@ -148,9 +141,11 @@ class ASRProvider(ASRProviderBase):
action = "SentenceRecognition" # 接口名称
# 构建规范头部信息,注意顺序和格式
canonical_headers = f"content-type:{content_type.lower()}\n" + \
f"host:{host.lower()}\n" + \
f"x-tc-action:{action.lower()}\n"
canonical_headers = (
f"content-type:{content_type.lower()}\n"
+ f"host:{host.lower()}\n"
+ f"x-tc-action:{action.lower()}\n"
)
signed_headers = "content-type;host;x-tc-action"
@@ -158,21 +153,25 @@ class ASRProvider(ASRProviderBase):
payload_hash = self._sha256_hex(request_body)
# 构建规范请求字符串
canonical_request = f"{http_request_method}\n" + \
f"{canonical_uri}\n" + \
f"{canonical_query_string}\n" + \
f"{canonical_headers}\n" + \
f"{signed_headers}\n" + \
f"{payload_hash}"
canonical_request = (
f"{http_request_method}\n"
+ f"{canonical_uri}\n"
+ f"{canonical_query_string}\n"
+ f"{canonical_headers}\n"
+ f"{signed_headers}\n"
+ f"{payload_hash}"
)
# 计算规范请求的哈希值
hashed_canonical_request = self._sha256_hex(canonical_request)
# 构建待签名字符串
string_to_sign = f"{algorithm}\n" + \
f"{timestamp}\n" + \
f"{credential_scope}\n" + \
f"{hashed_canonical_request}"
string_to_sign = (
f"{algorithm}\n"
+ f"{timestamp}\n"
+ f"{credential_scope}\n"
+ f"{hashed_canonical_request}"
)
# 计算签名密钥
secret_date = self._hmac_sha256(f"TC3{self.secret_key}", date)
@@ -180,13 +179,17 @@ class ASRProvider(ASRProviderBase):
secret_signing = self._hmac_sha256(secret_service, "tc3_request")
# 计算签名
signature = self._bytes_to_hex(self._hmac_sha256(secret_signing, string_to_sign))
signature = self._bytes_to_hex(
self._hmac_sha256(secret_signing, string_to_sign)
)
# 构建授权头
authorization = f"{algorithm} " + \
f"Credential={self.secret_id}/{credential_scope}, " + \
f"SignedHeaders={signed_headers}, " + \
f"Signature={signature}"
authorization = (
f"{algorithm} "
+ f"Credential={self.secret_id}/{credential_scope}, "
+ f"SignedHeaders={signed_headers}, "
+ f"Signature={signature}"
)
return timestamp, authorization
@@ -194,7 +197,9 @@ class ASRProvider(ASRProviderBase):
logger.bind(tag=TAG).error(f"生成认证头失败: {e}", exc_info=True)
raise RuntimeError(f"生成认证头失败: {e}")
def _send_request(self, request_body: str, timestamp: str, authorization: str) -> Optional[str]:
def _send_request(
self, request_body: str, timestamp: str, authorization: str
) -> Optional[str]:
"""发送请求到腾讯云API"""
headers = {
"Content-Type": "application/json; charset=utf-8",
@@ -203,7 +208,7 @@ class ASRProvider(ASRProviderBase):
"X-TC-Action": "SentenceRecognition",
"X-TC-Version": self.API_VERSION,
"X-TC-Timestamp": timestamp,
"X-TC-Region": "ap-shanghai"
"X-TC-Region": "ap-shanghai",
}
try:
@@ -234,16 +239,16 @@ class ASRProvider(ASRProviderBase):
def _sha256_hex(self, data: str) -> str:
"""计算字符串的SHA256哈希值"""
digest = hashlib.sha256(data.encode('utf-8')).digest()
digest = hashlib.sha256(data.encode("utf-8")).digest()
return self._bytes_to_hex(digest)
def _hmac_sha256(self, key, data: str) -> bytes:
"""计算HMAC-SHA256"""
if isinstance(key, str):
key = key.encode('utf-8')
key = key.encode("utf-8")
return hmac.new(key, data.encode('utf-8'), hashlib.sha256).digest()
return hmac.new(key, data.encode("utf-8"), hashlib.sha256).digest()
def _bytes_to_hex(self, bytes_data: bytes) -> str:
"""字节数组转十六进制字符串"""
return ''.join(f"{b:02x}" for b in bytes_data)
return "".join(f"{b:02x}" for b in bytes_data)
@@ -3,7 +3,7 @@ from config.logger import setup_logging
import os
from abc import ABC, abstractmethod
from core.utils.tts import MarkdownCleaner
from core.utils.util import audio_to_opus_data
from core.utils.util import audio_to_data
TAG = __name__
logger = setup_logging()
@@ -53,5 +53,10 @@ class TTSProviderBase(ABC):
async def text_to_speak(self, text, output_file):
pass
def audio_to_pcm_data(self, audio_file_path):
"""音频文件转换为PCM编码"""
return audio_to_data(audio_file_path, is_opus=False)
def audio_to_opus_data(self, audio_file_path):
return audio_to_opus_data(audio_file_path)
"""音频文件转换为Opus编码"""
return audio_to_data(audio_file_path, is_opus=True)
+11 -9
View File
@@ -862,8 +862,7 @@ def analyze_emotion(text):
return top_emotions[0] # 如果都不在优先级列表里,返回第一个
def audio_to_opus_data(audio_file_path):
"""音频文件转换为Opus编码"""
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]
if file_type:
@@ -889,7 +888,7 @@ def audio_to_opus_data(audio_file_path):
frame_duration = 60 # 60ms per frame
frame_size = int(16000 * frame_duration / 1000) # 960 samples/frame
opus_datas = []
datas = []
# 按帧处理所有音频数据(包括最后一帧可能补零)
for i in range(0, len(raw_data), frame_size * 2): # 16bit=2bytes/sample
# 获取当前帧的二进制数据
@@ -899,14 +898,17 @@ def audio_to_opus_data(audio_file_path):
if len(chunk) < frame_size * 2:
chunk += b"\x00" * (frame_size * 2 - len(chunk))
# 转换为numpy数组处理
np_frame = np.frombuffer(chunk, dtype=np.int16)
if is_opus:
# 转换为numpy数组处理
np_frame = np.frombuffer(chunk, dtype=np.int16)
# 编码Opus数据
frame_data = encoder.encode(np_frame.tobytes(), frame_size)
else:
frame_data = chunk if isinstance(chunk, bytes) else bytes(chunk)
# 编码Opus数据
opus_data = encoder.encode(np_frame.tobytes(), frame_size)
opus_datas.append(opus_data)
datas.append(frame_data)
return opus_datas, duration
return datas, duration
def check_vad_update(before_config, new_config):