tccli 3.0.1371.1__py2.py3-none-any.whl → 3.0.1373.1__py2.py3-none-any.whl

tccli/services/trtc/trtc_client.py

@@ -849,6 +849,58 @@ def doDescribeRecordingUsage(args, parsed_globals):
     FormatOutput.output("action", json_obj, g_param[OptionsDefine.Output], g_param[OptionsDefine.Filter])
 
 
+def doTextToSpeech(args, parsed_globals):
+    g_param = parse_global_arg(parsed_globals)
+
+    if g_param[OptionsDefine.UseCVMRole.replace('-', '_')]:
+        cred = credential.CVMRoleCredential()
+    elif g_param[OptionsDefine.RoleArn.replace('-', '_')] and g_param[OptionsDefine.RoleSessionName.replace('-', '_')]:
+        cred = credential.STSAssumeRoleCredential(
+            g_param[OptionsDefine.SecretId], g_param[OptionsDefine.SecretKey], g_param[OptionsDefine.RoleArn.replace('-', '_')],
+            g_param[OptionsDefine.RoleSessionName.replace('-', '_')], endpoint=g_param["sts_cred_endpoint"]
+        )
+    elif os.getenv(OptionsDefine.ENV_TKE_REGION)             and os.getenv(OptionsDefine.ENV_TKE_PROVIDER_ID)             and os.getenv(OptionsDefine.ENV_TKE_WEB_IDENTITY_TOKEN_FILE)             and os.getenv(OptionsDefine.ENV_TKE_ROLE_ARN):
+        cred = credential.DefaultTkeOIDCRoleArnProvider().get_credentials()
+    else:
+        cred = credential.Credential(
+            g_param[OptionsDefine.SecretId], g_param[OptionsDefine.SecretKey], g_param[OptionsDefine.Token]
+        )
+    http_profile = HttpProfile(
+        reqTimeout=60 if g_param[OptionsDefine.Timeout] is None else int(g_param[OptionsDefine.Timeout]),
+        reqMethod="POST",
+        endpoint=g_param[OptionsDefine.Endpoint],
+        proxy=g_param[OptionsDefine.HttpsProxy.replace('-', '_')]
+    )
+    profile = ClientProfile(httpProfile=http_profile, signMethod="HmacSHA256")
+    if g_param[OptionsDefine.Language]:
+        profile.language = g_param[OptionsDefine.Language]
+    mod = CLIENT_MAP[g_param[OptionsDefine.Version]]
+    client = mod.TrtcClient(cred, g_param[OptionsDefine.Region], profile)
+    client._sdkVersion += ("_CLI_" + __version__)
+    models = MODELS_MAP[g_param[OptionsDefine.Version]]
+    model = models.TextToSpeechRequest()
+    model.from_json_string(json.dumps(args))
+    start_time = time.time()
+    while True:
+        rsp = client.TextToSpeech(model)
+        result = rsp.to_json_string()
+        try:
+            json_obj = json.loads(result)
+        except TypeError as e:
+            json_obj = json.loads(result.decode('utf-8'))  # python3.3
+        if not g_param[OptionsDefine.Waiter] or search(g_param['OptionsDefine.WaiterInfo']['expr'], json_obj) == g_param['OptionsDefine.WaiterInfo']['to']:
+            break
+        cur_time = time.time()
+        if cur_time - start_time >= g_param['OptionsDefine.WaiterInfo']['timeout']:
+            raise ClientError('Request timeout, wait `%s` to `%s` timeout, last request is %s' %
+            (g_param['OptionsDefine.WaiterInfo']['expr'], g_param['OptionsDefine.WaiterInfo']['to'],
+            search(g_param['OptionsDefine.WaiterInfo']['expr'], json_obj)))
+        else:
+            print('Inquiry result is %s.' % search(g_param['OptionsDefine.WaiterInfo']['expr'], json_obj))
+        time.sleep(g_param['OptionsDefine.WaiterInfo']['interval'])
+    FormatOutput.output("action", json_obj, g_param[OptionsDefine.Output], g_param[OptionsDefine.Filter])
+
+
 def doStartStreamIngest(args, parsed_globals):
     g_param = parse_global_arg(parsed_globals)
 
@@ -2357,6 +2409,58 @@ def doDescribeUnusualEvent(args, parsed_globals):
     FormatOutput.output("action", json_obj, g_param[OptionsDefine.Output], g_param[OptionsDefine.Filter])
 
 
+def doVoiceClone(args, parsed_globals):
+    g_param = parse_global_arg(parsed_globals)
+
+    if g_param[OptionsDefine.UseCVMRole.replace('-', '_')]:
+        cred = credential.CVMRoleCredential()
+    elif g_param[OptionsDefine.RoleArn.replace('-', '_')] and g_param[OptionsDefine.RoleSessionName.replace('-', '_')]:
+        cred = credential.STSAssumeRoleCredential(
+            g_param[OptionsDefine.SecretId], g_param[OptionsDefine.SecretKey], g_param[OptionsDefine.RoleArn.replace('-', '_')],
+            g_param[OptionsDefine.RoleSessionName.replace('-', '_')], endpoint=g_param["sts_cred_endpoint"]
+        )
+    elif os.getenv(OptionsDefine.ENV_TKE_REGION)             and os.getenv(OptionsDefine.ENV_TKE_PROVIDER_ID)             and os.getenv(OptionsDefine.ENV_TKE_WEB_IDENTITY_TOKEN_FILE)             and os.getenv(OptionsDefine.ENV_TKE_ROLE_ARN):
+        cred = credential.DefaultTkeOIDCRoleArnProvider().get_credentials()
+    else:
+        cred = credential.Credential(
+            g_param[OptionsDefine.SecretId], g_param[OptionsDefine.SecretKey], g_param[OptionsDefine.Token]
+        )
+    http_profile = HttpProfile(
+        reqTimeout=60 if g_param[OptionsDefine.Timeout] is None else int(g_param[OptionsDefine.Timeout]),
+        reqMethod="POST",
+        endpoint=g_param[OptionsDefine.Endpoint],
+        proxy=g_param[OptionsDefine.HttpsProxy.replace('-', '_')]
+    )
+    profile = ClientProfile(httpProfile=http_profile, signMethod="HmacSHA256")
+    if g_param[OptionsDefine.Language]:
+        profile.language = g_param[OptionsDefine.Language]
+    mod = CLIENT_MAP[g_param[OptionsDefine.Version]]
+    client = mod.TrtcClient(cred, g_param[OptionsDefine.Region], profile)
+    client._sdkVersion += ("_CLI_" + __version__)
+    models = MODELS_MAP[g_param[OptionsDefine.Version]]
+    model = models.VoiceCloneRequest()
+    model.from_json_string(json.dumps(args))
+    start_time = time.time()
+    while True:
+        rsp = client.VoiceClone(model)
+        result = rsp.to_json_string()
+        try:
+            json_obj = json.loads(result)
+        except TypeError as e:
+            json_obj = json.loads(result.decode('utf-8'))  # python3.3
+        if not g_param[OptionsDefine.Waiter] or search(g_param['OptionsDefine.WaiterInfo']['expr'], json_obj) == g_param['OptionsDefine.WaiterInfo']['to']:
+            break
+        cur_time = time.time()
+        if cur_time - start_time >= g_param['OptionsDefine.WaiterInfo']['timeout']:
+            raise ClientError('Request timeout, wait `%s` to `%s` timeout, last request is %s' %
+            (g_param['OptionsDefine.WaiterInfo']['expr'], g_param['OptionsDefine.WaiterInfo']['to'],
+            search(g_param['OptionsDefine.WaiterInfo']['expr'], json_obj)))
+        else:
+            print('Inquiry result is %s.' % search(g_param['OptionsDefine.WaiterInfo']['expr'], json_obj))
+        time.sleep(g_param['OptionsDefine.WaiterInfo']['interval'])
+    FormatOutput.output("action", json_obj, g_param[OptionsDefine.Output], g_param[OptionsDefine.Filter])
+
+
 def doStartMCUMixTranscodeByStrRoomId(args, parsed_globals):
     g_param = parse_global_arg(parsed_globals)
 
@@ -3553,6 +3657,58 @@ def doDescribeTRTCRealTimeScaleMetricData(args, parsed_globals):
     FormatOutput.output("action", json_obj, g_param[OptionsDefine.Output], g_param[OptionsDefine.Filter])
 
 
+def doTextToSpeechSSE(args, parsed_globals):
+    g_param = parse_global_arg(parsed_globals)
+
+    if g_param[OptionsDefine.UseCVMRole.replace('-', '_')]:
+        cred = credential.CVMRoleCredential()
+    elif g_param[OptionsDefine.RoleArn.replace('-', '_')] and g_param[OptionsDefine.RoleSessionName.replace('-', '_')]:
+        cred = credential.STSAssumeRoleCredential(
+            g_param[OptionsDefine.SecretId], g_param[OptionsDefine.SecretKey], g_param[OptionsDefine.RoleArn.replace('-', '_')],
+            g_param[OptionsDefine.RoleSessionName.replace('-', '_')], endpoint=g_param["sts_cred_endpoint"]
+        )
+    elif os.getenv(OptionsDefine.ENV_TKE_REGION)             and os.getenv(OptionsDefine.ENV_TKE_PROVIDER_ID)             and os.getenv(OptionsDefine.ENV_TKE_WEB_IDENTITY_TOKEN_FILE)             and os.getenv(OptionsDefine.ENV_TKE_ROLE_ARN):
+        cred = credential.DefaultTkeOIDCRoleArnProvider().get_credentials()
+    else:
+        cred = credential.Credential(
+            g_param[OptionsDefine.SecretId], g_param[OptionsDefine.SecretKey], g_param[OptionsDefine.Token]
+        )
+    http_profile = HttpProfile(
+        reqTimeout=60 if g_param[OptionsDefine.Timeout] is None else int(g_param[OptionsDefine.Timeout]),
+        reqMethod="POST",
+        endpoint=g_param[OptionsDefine.Endpoint],
+        proxy=g_param[OptionsDefine.HttpsProxy.replace('-', '_')]
+    )
+    profile = ClientProfile(httpProfile=http_profile, signMethod="HmacSHA256")
+    if g_param[OptionsDefine.Language]:
+        profile.language = g_param[OptionsDefine.Language]
+    mod = CLIENT_MAP[g_param[OptionsDefine.Version]]
+    client = mod.TrtcClient(cred, g_param[OptionsDefine.Region], profile)
+    client._sdkVersion += ("_CLI_" + __version__)
+    models = MODELS_MAP[g_param[OptionsDefine.Version]]
+    model = models.TextToSpeechSSERequest()
+    model.from_json_string(json.dumps(args))
+    start_time = time.time()
+    while True:
+        rsp = client.TextToSpeechSSE(model)
+        result = rsp.to_json_string()
+        try:
+            json_obj = json.loads(result)
+        except TypeError as e:
+            json_obj = json.loads(result.decode('utf-8'))  # python3.3
+        if not g_param[OptionsDefine.Waiter] or search(g_param['OptionsDefine.WaiterInfo']['expr'], json_obj) == g_param['OptionsDefine.WaiterInfo']['to']:
+            break
+        cur_time = time.time()
+        if cur_time - start_time >= g_param['OptionsDefine.WaiterInfo']['timeout']:
+            raise ClientError('Request timeout, wait `%s` to `%s` timeout, last request is %s' %
+            (g_param['OptionsDefine.WaiterInfo']['expr'], g_param['OptionsDefine.WaiterInfo']['to'],
+            search(g_param['OptionsDefine.WaiterInfo']['expr'], json_obj)))
+        else:
+            print('Inquiry result is %s.' % search(g_param['OptionsDefine.WaiterInfo']['expr'], json_obj))
+        time.sleep(g_param['OptionsDefine.WaiterInfo']['interval'])
+    FormatOutput.output("action", json_obj, g_param[OptionsDefine.Output], g_param[OptionsDefine.Filter])
+
+
 def doStopMCUMixTranscode(args, parsed_globals):
     g_param = parse_global_arg(parsed_globals)
 
@@ -3632,6 +3788,7 @@ ACTION_MAP = {
     "DescribeTRTCRealTimeScaleData": doDescribeTRTCRealTimeScaleData,
     "DescribeAIConversation": doDescribeAIConversation,
     "DescribeRecordingUsage": doDescribeRecordingUsage,
+    "TextToSpeech": doTextToSpeech,
     "StartStreamIngest": doStartStreamIngest,
     "DescribeCloudModeration": doDescribeCloudModeration,
     "UpdateStreamIngest": doUpdateStreamIngest,
@@ -3661,6 +3818,7 @@ ACTION_MAP = {
     "StopMCUMixTranscodeByStrRoomId": doStopMCUMixTranscodeByStrRoomId,
     "StopStreamIngest": doStopStreamIngest,
     "DescribeUnusualEvent": doDescribeUnusualEvent,
+    "VoiceClone": doVoiceClone,
     "StartMCUMixTranscodeByStrRoomId": doStartMCUMixTranscodeByStrRoomId,
     "RemoveUserByStrRoomId": doRemoveUserByStrRoomId,
     "DescribeMixTranscodingUsage": doDescribeMixTranscodingUsage,
@@ -3684,6 +3842,7 @@ ACTION_MAP = {
     "DeleteCloudSliceTask": doDeleteCloudSliceTask,
     "ModifyCloudModeration": doModifyCloudModeration,
     "DescribeTRTCRealTimeScaleMetricData": doDescribeTRTCRealTimeScaleMetricData,
+    "TextToSpeechSSE": doTextToSpeechSSE,
     "StopMCUMixTranscode": doStopMCUMixTranscode,
 
 }

tccli/services/trtc/v20190722/api.json

@@ -455,6 +455,20 @@
       "output": "StopWebRecordResponse",
       "status": "online"
     },
+    "TextToSpeech": {
+      "document": "语音合成接口",
+      "input": "TextToSpeechRequest",
+      "name": "文本转语音",
+      "output": "TextToSpeechResponse",
+      "status": "online"
+    },
+    "TextToSpeechSSE": {
+      "document": "SSE流式文本转语音",
+      "input": "TextToSpeechSSERequest",
+      "name": "SSE流式文本转语音",
+      "output": "TextToSpeechSSEResponse",
+      "status": "online"
+    },
     "UpdateAIConversation": {
       "document": "更新AIConversation参数",
       "input": "UpdateAIConversationRequest",
@@ -482,6 +496,13 @@
       "name": "更新声纹信息",
       "output": "UpdateVoicePrintResponse",
       "status": "online"
+    },
+    "VoiceClone": {
+      "document": "声音克隆",
+      "input": "VoiceCloneRequest",
+      "name": "声音克隆",
+      "output": "VoiceCloneResponse",
+      "status": "online"
     }
   },
   "metadata": {
@@ -893,6 +914,23 @@
       ],
       "usage": "in"
     },
+    "AudioFormat": {
+      "document": "TTS音频输出的格式",
+      "members": [
+        {
+          "disabled": false,
+          "document": "生成的音频格式，默认pcm，目前支持的格式列表：[pcm]。",
+          "example": "pcm",
+          "member": "string",
+          "name": "Format",
+          "output_required": false,
+          "required": false,
+          "type": "string",
+          "value_allowed_null": false
+        }
+      ],
+      "usage": "both"
+    },
     "AudioParams": {
       "document": "录制音频转码参数。",
       "members": [
@@ -2204,7 +2242,7 @@
         {
           "disabled": false,
           "document": "用户SdkAppId（如：1400xxxxxx）。",
-          "example": "1400xxxxxx",
+          "example": "1400123456",
           "member": "uint64",
           "name": "SdkAppId",
           "required": true,
@@ -2213,7 +2251,7 @@
         {
           "disabled": false,
           "document": "需查询的用户数组，默认不填返回6个用户。",
-          "example": "[\"a\",\"b\"]",
+          "example": "[\"user_a\",\"user_b\"]",
           "member": "string",
           "name": "UserIds",
           "required": false,
@@ -2221,7 +2259,7 @@
         },
         {
           "disabled": false,
-          "document": "需查询的指标，不填则只返回用户列表，填all则返回所有指标。\nappCpu：APP CPU使用率；\nsysCpu：系统 CPU使用率；\naBit：上/下行音频码率；单位：bps\naBlock：音频卡顿时长；单位：ms\nbigvBit：上/下行视频码率；单位：bps\nbigvCapFps：视频采集帧率；\nbigvEncFps：视频发送帧率；\nbigvDecFps：渲染帧率；\nbigvBlock：视频卡顿时长；单位：ms\naLoss：上/下行音频丢包率；\nbigvLoss：上/下行视频丢包率；\nbigvWidth：上/下行分辨率宽；\nbigvHeight：上/下行分辨率高；\naCapEnergy：音频采集能量；\naPlayEnergy：音频播放能量；\nrtt：SDK到云端的往返延时；单位: ms",
+          "document": "需查询的指标，不填则只返回用户列表，填all则返回所有指标。\nappCpu：APP CPU使用率；\nsysCpu：系统 CPU使用率；\naBit：上/下行音频码率；单位：bps\naBlock：音频卡顿时长；单位：ms\nbigvBit：上/下行视频码率；单位：bps\nbigvCapFps：视频采集帧率；\nbigvEncFps：视频发送帧率；\nbigvDecFps：渲染帧率；\nbigvBlock：视频卡顿时长；单位：ms\naLoss：上/下行音频丢包率；\nbigvLoss：上/下行视频丢包率；\nbigvWidth：上/下行分辨率宽；\nbigvHeight：上/下行分辨率高；\naCapEnergy：音频采集能量；\naPlayEnergy：音频播放能量；\nrtt：SDK到云端的往返延时；单位: ms\nbigvRecFps: 云端送达帧率；",
           "example": "[\"bigvCapFps\"]",
           "member": "string",
           "name": "DataType",
@@ -6990,7 +7028,7 @@
         {
           "disabled": false,
           "document": "**发起模糊识别为高级版能力,默认按照高级版收费**\n注意：不支持填写\"zh-dialect\", \"16k_zh_edu\", \"16k_zh_medical\", \"16k_zh_court\", \"8k_zh_large\", \"16k_zh_large\",\"16k_multi_lang\", \"16k_zh_en\"\n",
-          "example": "无",
+          "example": "[\"zh\", \"en\"]",
           "member": "string",
           "name": "AlternativeLanguage",
           "output_required": false,
@@ -7001,7 +7039,7 @@
         {
           "disabled": false,
           "document": "自定义参数，联系后台使用\n",
-          "example": "无",
+          "example": "{\"key\": \"xx\"}",
           "member": "string",
           "name": "CustomParam",
           "output_required": false,
@@ -7012,7 +7050,7 @@
         {
           "disabled": false,
           "document": "语音识别vad的时间，范围为240-2000，默认为1000，单位为ms。更小的值会让语音识别分句更快。",
-          "example": "无",
+          "example": "500",
           "member": "uint64",
           "name": "VadSilenceTime",
           "output_required": false,
@@ -7484,7 +7522,7 @@
         {
           "disabled": false,
           "document": "机器人参数",
-          "example": "{     \"UserId\": \"user_12345\",     \"UserSig\": \"user_signature_example\",     \"MaxIdleTime\": 120,     \"TargetUserId\": \"target_user_54321\"   }",
+          "example": "无",
           "member": "AgentConfig",
           "name": "AgentConfig",
           "required": true,
@@ -7511,7 +7549,7 @@
         {
           "disabled": false,
           "document": "语音识别配置。",
-          "example": " {     \"Language\": \"zh\"}",
+          "example": "无",
           "member": "STTConfig",
           "name": "STTConfig",
           "required": false,
@@ -7519,7 +7557,7 @@
         },
         {
           "disabled": false,
-          "document": "LLM配置。需符合openai规范，为JSON字符串，示例如下：\n<pre> { <br> &emsp;  \"LLMType\": \"大模型类型\",  // String 必填，如：\"openai\" <br> &emsp;  \"Model\": \"您的模型名称\", // String 必填，指定使用的模型<br>    \"APIKey\": \"您的LLM API密钥\", // String 必填 <br> &emsp;  \"APIUrl\": \"https://api.xxx.com/chat/completions\", // String 必填，LLM API访问的URL<br> &emsp;  \"Streaming\": true // Boolean 非必填，指定是否使用流式传输<br> &emsp;} </pre>\n",
+          "document": "LLM配置。需符合openai规范，为JSON字符串，示例如下：\n<pre> { <br> &emsp;  \"LLMType\": \"大模型类型\",  // String 必填，如：\"openai\" <br> &emsp;  \"Model\": \"您的模型名称\", // String 必填，指定使用的模型<br>    \"APIKey\": \"您的LLM API密钥\", // String 必填 <br> &emsp;  \"APIUrl\": \"https://api.xxx.com/chat/completions\", // String 必填，LLM API访问的URL<br> &emsp;  \"History\": 10, // Integer 选填，设置 LLM 的上下文轮次，默认值为0，最大值50<br> &emsp;  \"HistoryMode\": 1, // Integer 选填，1表示LLM上下文中的内容会和播放音频做同步，没有播放的音频对应的文本不会出现在上下文中。0表示不会做同步，默认值为0<br> &emsp;  \"Streaming\": true // Boolean 非必填，指定是否使用流式传输<br> &emsp;} </pre>\n",
           "example": "{\\\"LLMType\\\": \\\"openai\\\", \\\"Model\\\": \\\"gpt-xxx\", \\\"APIKey\\\": \\\"xxxxxxxx\\\", \\\"APIUrl\\\": \\\"https://api.xxx.com/v1/chat/completions\\\", \\\"Streaming\\\": true}",
           "member": "string",
           "name": "LLMConfig",
@@ -8777,6 +8815,142 @@
       ],
       "usage": "in"
     },
+    "TextToSpeechRequest": {
+      "document": "TextToSpeech请求参数结构体",
+      "members": [
+        {
+          "disabled": false,
+          "document": "需要转语音的文字内容，长度范围：[1, 255]",
+          "example": "你好呀",
+          "member": "string",
+          "name": "Text",
+          "required": true,
+          "type": "string"
+        },
+        {
+          "disabled": false,
+          "document": "文本转语音的声音配置",
+          "example": "无",
+          "member": "Voice",
+          "name": "Voice",
+          "required": true,
+          "type": "object"
+        },
+        {
+          "disabled": false,
+          "document": "TRTC的SdkAppId",
+          "example": "14600000000",
+          "member": "uint64",
+          "name": "SdkAppId",
+          "required": true,
+          "type": "int"
+        },
+        {
+          "disabled": false,
+          "document": "文本转语音的输出音频的格式",
+          "example": "无",
+          "member": "AudioFormat",
+          "name": "AudioFormat",
+          "required": false,
+          "type": "object"
+        },
+        {
+          "disabled": false,
+          "document": "TTS的API密钥",
+          "example": "sk-xxxxxx",
+          "member": "string",
+          "name": "APIKey",
+          "required": false,
+          "type": "string"
+        }
+      ],
+      "type": "object"
+    },
+    "TextToSpeechResponse": {
+      "document": "TextToSpeech返回参数结构体",
+      "members": [
+        {
+          "disabled": false,
+          "document": "Base64编码的音频数据",
+          "example": "ejujainengoxpam",
+          "member": "string",
+          "name": "Audio",
+          "output_required": false,
+          "type": "string",
+          "value_allowed_null": false
+        },
+        {
+          "document": "唯一请求 ID，由服务端生成，每次请求都会返回（若请求因其他原因未能抵达服务端，则该次请求不会获得 RequestId）。定位问题时需要提供该次请求的 RequestId。",
+          "member": "string",
+          "name": "RequestId",
+          "type": "string"
+        }
+      ],
+      "type": "object"
+    },
+    "TextToSpeechSSERequest": {
+      "document": "TextToSpeechSSE请求参数结构体",
+      "members": [
+        {
+          "disabled": false,
+          "document": "需要转语音的文字内容，长度范围：[1, 255]",
+          "example": "你好呀",
+          "member": "string",
+          "name": "Text",
+          "required": true,
+          "type": "string"
+        },
+        {
+          "disabled": false,
+          "document": "文本转语音的声音配置",
+          "example": "无",
+          "member": "Voice",
+          "name": "Voice",
+          "required": true,
+          "type": "object"
+        },
+        {
+          "disabled": false,
+          "document": "TRTC的SdkAppId",
+          "example": "14600000000",
+          "member": "uint64",
+          "name": "SdkAppId",
+          "required": true,
+          "type": "int"
+        },
+        {
+          "disabled": false,
+          "document": "文本转语音的输出音频的格式",
+          "example": "无",
+          "member": "AudioFormat",
+          "name": "AudioFormat",
+          "required": false,
+          "type": "object"
+        },
+        {
+          "disabled": false,
+          "document": "TTS的API密钥",
+          "example": "sk-xxxxxxx",
+          "member": "string",
+          "name": "APIKey",
+          "required": false,
+          "type": "string"
+        }
+      ],
+      "type": "object"
+    },
+    "TextToSpeechSSEResponse": {
+      "document": "TextToSpeechSSE返回参数结构体",
+      "members": [
+        {
+          "document": "唯一请求 ID，由服务端生成，每次请求都会返回（若请求因其他原因未能抵达服务端，则该次请求不会获得 RequestId）。定位问题时需要提供该次请求的 RequestId。本接口为流式响应接口，当请求成功时，RequestId 会被放在 HTTP 响应的 Header \"X-TC-RequestId\" 中。",
+          "member": "string",
+          "name": "RequestId",
+          "type": "string"
+        }
+      ],
+      "type": "object"
+    },
     "TimeValue": {
       "document": "返回的质量数据，时间:值",
       "members": [
@@ -9537,6 +9711,96 @@
       ],
       "usage": "in"
     },
+    "Voice": {
+      "document": "TTS的声音参数",
+      "members": [
+        {
+          "disabled": false,
+          "document": "TTS的声音的ID",
+          "example": "256",
+          "member": "string",
+          "name": "VoiceId",
+          "output_required": false,
+          "required": true,
+          "type": "string",
+          "value_allowed_null": false
+        }
+      ],
+      "usage": "both"
+    },
+    "VoiceCloneRequest": {
+      "document": "VoiceClone请求参数结构体",
+      "members": [
+        {
+          "disabled": false,
+          "document": "TRTC的SdkAppId",
+          "example": "14600000000",
+          "member": "uint64",
+          "name": "SdkAppId",
+          "required": true,
+          "type": "int"
+        },
+        {
+          "disabled": false,
+          "document": "TTS的API密钥",
+          "example": "sk-xxxxxxx",
+          "member": "string",
+          "name": "APIKey",
+          "required": true,
+          "type": "string"
+        },
+        {
+          "disabled": false,
+          "document": "声音克隆的名称, 只允许使用数字、字母、下划线，不能超过36位",
+          "example": "xiaoming_01",
+          "member": "string",
+          "name": "VoiceName",
+          "required": true,
+          "type": "string"
+        },
+        {
+          "disabled": false,
+          "document": "声音克隆的参考音频，必须为16k单声道的wav的base64字符串， 长度在5秒～12秒之间",
+          "example": "ejiwnorinofnODSw",
+          "member": "string",
+          "name": "PromptAudio",
+          "required": true,
+          "type": "string"
+        },
+        {
+          "disabled": false,
+          "document": "声音克隆的参考文本，为参考音频对应的文字。",
+          "example": "你好，我要复刻一个我的音色。",
+          "member": "string",
+          "name": "PromptText",
+          "required": false,
+          "type": "string"
+        }
+      ],
+      "type": "object"
+    },
+    "VoiceCloneResponse": {
+      "document": "VoiceClone返回参数结构体",
+      "members": [
+        {
+          "disabled": false,
+          "document": "克隆出的音色ID，可以用此id进行语音合成",
+          "example": "xiaoming_01_ejwiqnd",
+          "member": "string",
+          "name": "VoiceId",
+          "output_required": false,
+          "type": "string",
+          "value_allowed_null": false
+        },
+        {
+          "document": "唯一请求 ID，由服务端生成，每次请求都会返回（若请求因其他原因未能抵达服务端，则该次请求不会获得 RequestId）。定位问题时需要提供该次请求的 RequestId。",
+          "member": "string",
+          "name": "RequestId",
+          "type": "string"
+        }
+      ],
+      "type": "object"
+    },
     "VoicePrint": {
       "document": "声纹配置参数",
       "members": [

tccli/services/trtc/v20190722/examples.json

@@ -556,6 +556,22 @@
         "title": "停止页面录制任务"
       }
     ],
+    "TextToSpeech": [
+      {
+        "document": "",
+        "input": "POST / HTTP/1.1\nHost: trtc.tencentcloudapi.com\nContent-Type: application/json\nX-TC-Action: TextToSpeech\n<公共请求参数>\n\n{\n  \"Text\": \"你好呀\",\n  \"SdkAppId\": 1466000000,\n  \"Voice\": {\n     \"VoiceId\": \"256\"\n  }\n}\n\n",
+        "output": "{\n    \"Response\": {\n        \"Audio\": \"inxxx\",\n        \"RequestId\": \"3c140219-cfe9-470e-b241-907877d6fb03\"\n    }\n}",
+        "title": "开始一个语音合成"
+      }
+    ],
+    "TextToSpeechSSE": [
+      {
+        "document": "启动一个SSE流式语音合成，注意在解析时，音频的Type为chunk，其他的Type 非音频",
+        "input": "POST / HTTP/1.1\nHost: trtc.tencentcloudapi.com\nContent-Type: application/json\nX-TC-Action: TextToSpeechSSE\n<公共请求参数>\n\n{\n   \"Text\": \"你好呀\",\n    \"SdkAppId\": 14600000000,\n    \"Voice\": {\n        \"VoiceId\": \"256\"\n    }\n}\n\n",
+        "output": "data: {\"Type\": \"chunk\", \"RequestId\": \"456e7890-e12b-34d5-a678-426614174000\", \"ChunkId\": \"chunk_0\", \"Audio\": \"base64编码的音频数据...\", \"Seq\": 0, \"IsEnd\": false}\n\ndata: {\"Type\": \"chunk\", \"RequestId\": \"456e7890-e12b-34d5-a678-426614174000\", \"ChunkId\": \"chunk_1\", \"Audio\": \"base64编码的音频数据...\", \"Seq\": 1, \"IsEnd\": false}\n\ndata: {\"Type\": \"chunk\", \"RequestId\": \"456e7890-e12b-34d5-a678-426614174000\", \"ChunkId\": \"chunk_2\", \"Audio\": \"base64编码的音频数据...\", \"Seq\": 2, \"IsEnd\": true}",
+        "title": "启动一个SSE流式语音合成"
+      }
+    ],
     "UpdateAIConversation": [
       {
         "document": "在对话过程中，想要动态更新tts的音色",
@@ -593,6 +609,14 @@
         "output": "{\n    \"Response\": {\n        \"RequestId\": \"40e323b2-f0e7-402c-be5d-b02aef40d887\"\n    }\n}",
         "title": "更新jackson的音频信息"
       }
+    ],
+    "VoiceClone": [
+      {
+        "document": "",
+        "input": "POST / HTTP/1.1\nHost: trtc.tencentcloudapi.com\nContent-Type: application/json\nX-TC-Action: VoiceClone\n<公共请求参数>\n\n{\n  \"SdkAppId\": 1466000000,\n  \"VoiceName\": \"xiaoming_02\",\n  \"APIKey\": \"sk-xxxxxx\",\n  \"PromptAudio\": \"ejxxxxxx\"\n}\n\n",
+        "output": "{\n    \"Response\": {\n        \"VoiceId\": \"xiaoming_02_aidniwqndq\",\n        \"RequestId\": \"3c140219-cfe9-470e-b241-907877d6fb03\"\n    }\n}",
+        "title": "创建一个声音克隆"
+      }
     ]
   },
   "version": "1.0"