@tencent-rtc/mcp 1.6.4 → 1.6.6

package/resource/knowledge/trtc/c++/ITRTCStatistics.md

@@ -33,21 +33,21 @@ Function: the TRTC SDK reports to you the current real-time audio/video metrics
 | --- | --- |
 | audioBitrate | Field description: local audio bitrate (Kbps) |
 | audioBlockRate | Field description: audio playback lag rate (%)<br>Audio playback lag rate (audioBlockRate) = cumulative audio playback lag duration (audioTotalBlockTime)/total audio playback duration |
-| audioPacketLoss | Field description: total packet loss rate (%) of the audio stream<br><code> audioPacketLoss </code> represents the packet loss rate eventually calculated on the audience side after the audio/video stream goes through the complete transfer linkage of "anchor -> cloud -> audience".<br>The smaller the <code> audioPacketLoss </code>, the better. The packet loss rate of 0 indicates that all data of the audio stream has entirely reached the audience.<br>If <code> downLoss </code> is <code> 0 </code> but <code> audioPacketLoss </code> isn't, there is no packet loss on the linkage of "cloud -> audience" for the audiostream, but there are unrecoverable packet losses on the linkage of "anchor -> cloud". |
+| audioPacketLoss | Field description: total packet loss rate (%) of the audio stream<br><code>audioPacketLoss</code>  represents the packet loss rate eventually calculated on the audience side after the audio/video stream goes through the complete transfer linkage of "anchor -> cloud -> audience".<br>The smaller the  <code>audioPacketLoss</code> , the better. The packet loss rate of 0 indicates that all data of the audio stream has entirely reached the audience.<br>If  <code>downLoss</code>  is  <code>0</code>  but  <code>audioPacketLoss</code>  isn't, there is no packet loss on the linkage of "cloud -> audience" for the audiostream, but there are unrecoverable packet losses on the linkage of "anchor -> cloud". |
 | audioSampleRate | Field description: local audio sample rate (Hz) |
 | audioTotalBlockTime | Field description: cumulative audio playback lag duration (ms) |
 | finalLoss | Field description: total packet loss rate (%) of the audio/video stream<br>Deprecated, please use audioPacketLoss and videoPacketLoss instead. |
 | frameRate | Field description: remote video frame rate (fps) |
 | height | Field description: remote video height in px |
-| jitterBufferDelay | Field description: playback delay (ms)<br>In order to avoid audio/video lags caused by network jitters and network packet disorders, TRTC maintains a playback buffer on the playback side to organize the received network data packets.<br>The size of the buffer is adaptively adjusted according to the current network quality and converted to the length of time in milliseconds, i.e., <code> jitterBufferDelay </code>. |
-| point2PointDelay | Field description: end-to-end delay (ms)<br><code> point2PointDelay </code> represents the delay of "anchor -> cloud -> audience". To be more precise, it represents the delay of the entire linkage of "collection -> encoding -> network transfer -> receiving -> buffering -> decoding -> playback".<br><code> point2PointDelay </code> works only if both the local and remote SDKs are on version 8.5 or above. If the remote SDK is on a version below 8.5, this value will always be 0 and thus meaningless. |
-| remoteNetworkRTT | Field description: round-trip delay (ms) from the SDK to cloud<br>This value represents the total time it takes to send a network packet from the SDK to the cloud and then send a network packet back from the cloud to the SDK, i.e., the total time it takes for a network packet to go through the linkage of "SDK -> cloud -> SDK".<br>The smaller the value, the better. If <code> remoteNetworkRTT </code> is below 50 ms, it means a short audio/video call delay; if <code> remoteNetworkRTT </code> is above 200 ms, it means a long audio/video call delay.<br>It should be explained that <code> remoteNetworkRTT </code> represents the total time spent on the linkage of "SDK -> cloud -> SDK"; therefore, there is no need to distinguish between <code> remoteNetworkUpRTT </code> and <code> remoteNetworkDownRTT </code>. |
-| remoteNetworkUplinkLoss | Field description: upstream packet loss rate (%) from the SDK to cloud<br>The smaller the value, the better. If <code> remoteNetworkUplinkLoss </code> is <code> 0% </code>, the upstream network quality is very good, and the data packets uploaded to the cloud are basically not lost.<br>If <code> remoteNetworkUplinkLoss </code> is <code> 30% </code>, 30% of the audio/video data packets sent to the cloud by the SDK are lost on the transfer linkage. |
+| jitterBufferDelay | Field description: playback delay (ms)<br>In order to avoid audio/video lags caused by network jitters and network packet disorders, TRTC maintains a playback buffer on the playback side to organize the received network data packets.<br>The size of the buffer is adaptively adjusted according to the current network quality and converted to the length of time in milliseconds, i.e.,  <code>jitterBufferDelay</code> . |
+| point2PointDelay | Field description: end-to-end delay (ms)<br><code>point2PointDelay</code>  represents the delay of "anchor -> cloud -> audience". To be more precise, it represents the delay of the entire linkage of "collection -> encoding -> network transfer -> receiving -> buffering -> decoding -> playback".<br><code>point2PointDelay</code>  works only if both the local and remote SDKs are on version 8.5 or above. If the remote SDK is on a version below 8.5, this value will always be 0 and thus meaningless. |
+| remoteNetworkRTT | Field description: round-trip delay (ms) from the SDK to cloud<br>This value represents the total time it takes to send a network packet from the SDK to the cloud and then send a network packet back from the cloud to the SDK, i.e., the total time it takes for a network packet to go through the linkage of "SDK -> cloud -> SDK".<br>The smaller the value, the better. If  <code>remoteNetworkRTT</code>  is below 50 ms, it means a short audio/video call delay; if  <code>remoteNetworkRTT</code>  is above 200 ms, it means a long audio/video call delay.<br>It should be explained that  <code>remoteNetworkRTT</code>  represents the total time spent on the linkage of "SDK -> cloud -> SDK"; therefore, there is no need to distinguish between  <code>remoteNetworkUpRTT</code>  and  <code>remoteNetworkDownRTT</code> . |
+| remoteNetworkUplinkLoss | Field description: upstream packet loss rate (%) from the SDK to cloud<br>The smaller the value, the better. If  <code>remoteNetworkUplinkLoss</code>  is  <code>0%</code> , the upstream network quality is very good, and the data packets uploaded to the cloud are basically not lost.<br>If  <code>remoteNetworkUplinkLoss</code>  is  <code>30%</code> , 30% of the audio/video data packets sent to the cloud by the SDK are lost on the transfer linkage. |
 | streamType | Field description: video stream type (HD big image \\| smooth small image \\| substream image) |
 | userId | Field description: user ID |
 | videoBitrate | Field description: remote video bitrate (Kbps) |
 | videoBlockRate | Field description: video playback lag rate (%)<br>Video playback lag rate (videoBlockRate) = cumulative video playback lag duration (videoTotalBlockTime)/total video playback duration |
-| videoPacketLoss | Field description: total packet loss rate (%) of the video stream<br><code> videoPacketLoss </code> represents the packet loss rate eventually calculated on the audience side after the audio/video stream goes through the complete transfer linkage of "anchor -> cloud -> audience".<br>The smaller the <code> videoPacketLoss </code>, the better. The packet loss rate of 0 indicates that all data of the video stream has entirely reached the audience.<br>If <code> downLoss </code> is <code> 0 </code> but <code> videoPacketLoss </code> isn't, there is no packet loss on the linkage of "cloud -> audience" for the video stream, but there are unrecoverable packet losses on the linkage of "anchor -> cloud". |
+| videoPacketLoss | Field description: total packet loss rate (%) of the video stream<br><code>videoPacketLoss</code>  represents the packet loss rate eventually calculated on the audience side after the audio/video stream goes through the complete transfer linkage of "anchor -> cloud -> audience".<br>The smaller the  <code>videoPacketLoss</code> , the better. The packet loss rate of 0 indicates that all data of the video stream has entirely reached the audience.<br>If  <code>downLoss</code>  is  <code>0</code>  but  <code>videoPacketLoss</code>  isn't, there is no packet loss on the linkage of "cloud -> audience" for the video stream, but there are unrecoverable packet losses on the linkage of "anchor -> cloud". |
 | videoTotalBlockTime | Field description: cumulative video playback lag duration (ms) |
 | width | Field description: remote video width in px |
 
@@ -58,16 +58,16 @@ Function: the TRTC SDK reports to you the current real-time audio/video metrics
 | --- | --- |
 | appCpu | Field description: CPU utilization (%) of the current application, Android 8.0 and above systems are not supported |
 | appMemoryUsageInMB | Field description: Memory usage size (MB) of current application |
-| downLoss | Field description: downstream packet loss rate (%) from cloud to the SDK<br>The smaller the value, the better. If <code> downLoss </code> is <code> 0% </code>, the downstream network quality is very good, and the data packets received from the cloud are basically not lost.<br>If <code> downLoss </code> is <code> 30% </code>, 30% of the audio/video data packets sent to the SDK by the cloud are lost on the transfer linkage. |
-| gatewayRtt | Field description: round-trip delay (ms) from the SDK to gateway<br>This value represents the total time it takes to send a network packet from the SDK to the gateway and then send a network packet back from the gateway to the SDK, i.e., the total time it takes for a network packet to go through the linkage of "SDK -> gateway -> SDK".<br>The smaller the value, the better. If <code> gatewayRtt </code> is below 50 ms, it means a short audio/video call delay; if <code> gatewayRtt </code> is above 200 ms, it means a long audio/video call delay.<br>It should be explained that <code> gatewayRtt </code> is invalid for cellular network. |
+| downLoss | Field description: downstream packet loss rate (%) from cloud to the SDK<br>The smaller the value, the better. If  <code>downLoss</code>  is  <code>0%</code> , the downstream network quality is very good, and the data packets received from the cloud are basically not lost.<br>If  <code>downLoss</code>  is  <code>30%</code> , 30% of the audio/video data packets sent to the SDK by the cloud are lost on the transfer linkage. |
+| gatewayRtt | Field description: round-trip delay (ms) from the SDK to gateway<br>This value represents the total time it takes to send a network packet from the SDK to the gateway and then send a network packet back from the gateway to the SDK, i.e., the total time it takes for a network packet to go through the linkage of "SDK -> gateway -> SDK".<br>The smaller the value, the better. If  <code>gatewayRtt</code>  is below 50 ms, it means a short audio/video call delay; if  <code>gatewayRtt</code>  is above 200 ms, it means a long audio/video call delay.<br>It should be explained that  <code>gatewayRtt</code>  is invalid for cellular network. |
 | localStatisticsArray | Field description: local audio/video statistics<br>As there may be three local audio/video streams (i.e., HD big image, smooth small image, and substream image), the local audio/video statistics are an array. |
-| localStatisticsArraySize | Field description: <code> localStatisticsArray </code> array size |
+| localStatisticsArraySize | Field description:  <code>localStatisticsArray</code>  array size |
 | receivedBytes | Field description: total number of received bytes (including signaling data and audio/video data) |
 | remoteStatisticsArray | Field description: remote audio/video statistics<br>As there may be multiple concurrent remote users, and each of them may have multiple concurrent audio/video streams (i.e., HD big image, smooth small image, and substream image), the remote audio/video statistics are an array. |
-| remoteStatisticsArraySize | Field description: <code> remoteStatisticsArray </code> array size |
-| rtt | Field description: round-trip delay (ms) from the SDK to cloud<br>This value represents the total time it takes to send a network packet from the SDK to the cloud and then send a network packet back from the cloud to the SDK, i.e., the total time it takes for a network packet to go through the linkage of "SDK -> cloud -> SDK".<br>The smaller the value, the better. If <code> rtt </code> is below 50 ms, it means a short audio/video call delay; if <code> rtt </code> is above 200 ms, it means a long audio/video call delay.<br>It should be explained that <code> rtt </code> represents the total time spent on the linkage of "SDK -> cloud -> SDK"; therefore, there is no need to distinguish between <code> upRtt </code> and <code> downRtt </code>. |
+| remoteStatisticsArraySize | Field description:  <code>remoteStatisticsArray</code>  array size |
+| rtt | Field description: round-trip delay (ms) from the SDK to cloud<br>This value represents the total time it takes to send a network packet from the SDK to the cloud and then send a network packet back from the cloud to the SDK, i.e., the total time it takes for a network packet to go through the linkage of "SDK -> cloud -> SDK".<br>The smaller the value, the better. If  <code>rtt</code>  is below 50 ms, it means a short audio/video call delay; if  <code>rtt</code>  is above 200 ms, it means a long audio/video call delay.<br>It should be explained that  <code>rtt</code>  represents the total time spent on the linkage of "SDK -> cloud -> SDK"; therefore, there is no need to distinguish between  <code>upRtt</code>  and  <code>downRtt</code> . |
 | sentBytes | Field description: total number of sent bytes (including signaling data and audio/video data) |
 | systemCpu | Field description: CPU utilization (%) of the current system, Android 8.0 and above systems are not supported |
 | systemMemoryInMB | Field description: Memory size (MB) of current system |
 | systemMemoryUsageInMB | Field description: Memory usage size (MB) of current system , iOS and MAC are not supported |
-| upLoss | Field description: upstream packet loss rate (%) from the SDK to cloud<br>The smaller the value, the better. If <code> upLoss </code> is <code> 0% </code>, the upstream network quality is very good, and the data packets uploaded to the cloud are basically not lost.<br>If <code> upLoss </code> is <code> 30% </code>, 30% of the audio/video data packets sent to the cloud by the SDK are lost on the transfer linkage. |
+| upLoss | Field description: upstream packet loss rate (%) from the SDK to cloud<br>The smaller the value, the better. If  <code>upLoss</code>  is  <code>0%</code> , the upstream network quality is very good, and the data packets uploaded to the cloud are basically not lost.<br>If  <code>upLoss</code>  is  <code>30%</code> , 30% of the audio/video data packets sent to the cloud by the SDK are lost on the transfer linkage. |

package/resource/knowledge/trtc/c++/ITXAudioEffectManager.md

@@ -111,11 +111,10 @@ Given that not all phones deliver excellent in-ear monitoring effects, we have b
 
 |Param|DESC|
 |---------|---------|
-|enable|` true: ` enable; ` false `: disable|
+|enable| `true:`  enable;  `false` : disable|
 
 > **Note**
 >
-
 > In-ear monitoring can be enabled only when earphones are used. Please remind anchors to use wired earphones.
 >
 
@@ -131,7 +130,6 @@ This API is used to set the volume of in-ear monitoring.
 
 > **Note**
 >
-
 > The volume of the ear monitor is one of the most important factors affecting the user experience. Considering that different users have different sensitivities to volume, it is strongly recommended to provide a ear monitor volume adjustment slider on the UI interface. The recommended adjustment range is between 0 and 150. The gain effect of 100-150 is very helpful for some Android phones with low recording volume.
 >
 
@@ -143,7 +141,6 @@ This API is used to set reverb effects for human voice. For the effects supporte
 
 > **Note**
 >
-
 > Effects become invalid after room exit. If you want to use the same effect after you enter the room again, you need to set the effect again using this API.
 >
 
@@ -155,7 +152,6 @@ This API is used to set voice changing effects. For the effects supported, pleas
 
 > **Note**
 >
-
 > Effects become invalid after room exit. If you want to use the same effect after you enter the room again, you need to set the effect again using this API.
 >
 
@@ -171,7 +167,6 @@ This API is used to set the volume of speech. It is often used together with the
 
 > **Note**
 >
-
 > If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
 >
 
@@ -194,11 +189,10 @@ Before playing background music, please use this API to set the music callback,
 |Param|DESC|
 |---------|---------|
 |musicId|Music ID|
-|observer|For more information, please see the APIs defined in ` ITXMusicPlayObserver `.|
+|observer|For more information, please see the APIs defined in  `ITXMusicPlayObserver` .|
 
 > **Note**
 >
-
 > 1. If the ID does not need to be used, the observer can be set to NULL to release it completely.
 >
 
@@ -214,13 +208,10 @@ You must assign an ID to each music track so that you can start, stop, or set th
 
 > **Note**
 >
-
 > 1. If you play the same music track multiple times, please use the same ID instead of a separate ID for each playback.
 >
-
 > 2. If you want to play different music tracks at the same time, use different IDs for them.
 >
-
 > 3. If you use the same ID to play a music track different from the current one, the SDK will stop the current one before playing the new one.
 >
 
@@ -263,7 +254,6 @@ This API is used to set the local and remote playback volume of background music
 
 > **Note**
 >
-
 > If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
 >
 
@@ -280,7 +270,6 @@ This API is used to control the remote playback volume (the volume heard by audi
 
 > **Note**
 >
-
 > If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
 >
 
@@ -297,7 +286,6 @@ This API is used to control the local playback volume (the volume heard by ancho
 
 > **Note**
 >
-
 > If 100 is still not loud enough for you, you can set the volume to up to 150, but there may be side effects.
 >
 
@@ -352,13 +340,10 @@ The length of the specified music file is returned. -1 indicates failure to get
 
 > **Note**
 >
-
 > Do not call this API frequently as the music file may be read and written to each time the API is called, which can be time-consuming.
 >
-
 > Wait till users finish dragging the progress bar before you call this API.
 >
-
 > The progress bar controller on the UI tends to update the progress at a high frequency as users drag the progress bar. This will result in poor user experience unless you limit the frequency.
 >
 
@@ -373,7 +358,6 @@ The length of the specified music file is returned. -1 indicates failure to get
 
 > **Note**
 >
-
 > Precondition preloadMusic succeeds.
 >
 
@@ -399,14 +383,11 @@ You must assign an ID to each music track so that you can start, stop, or set th
 
 > **Note**
 >
-
 > 1. Preload supports up to 2 preloads with different IDs at the same time, and the preload time does not exceed 10 minutes,you need to call stopPlayMusic API after use, otherwise the memory will not be released.
 >
-
 > 2. If the music corresponding to the ID is being played, the preloading fails, and stopPlayMusic must be called first.
 >
-
-> 3. When the ` musicParam ` passed to startPlayMusic is exactly the same, preloading works.
+> 3. When the  `musicParam`  passed to startPlayMusic is exactly the same, preloading works.
 >
 
 ## getMusicTrackCount
@@ -428,7 +409,6 @@ You must assign an ID to each music track so that you can start, stop, or set th
 
 > **Note**
 >
-
 > The total number of tracks can be obtained through the [getMusicTrackCount interface.
 >
 
@@ -493,10 +473,10 @@ The information, including playback ID, file path, and loop times, is passed in
 3. If you use the same ID to play a music track different from the current one, the SDK will stop the current one before playing the new one.
 | EnumType | DESC |
 | --- | --- |
-| endTimeMS | <code> Field description: </code> the point in time in milliseconds for ending music playback. 0 indicates that playback continues till the end of the music track. |
-| id | <code> Field description: </code> music ID<br><strong>Note</strong><br>the SDK supports playing multiple music tracks. IDs are used to distinguish different music tracks and control their start, end, volume, etc. |
-| isShortFile | <code> Field description: </code> whether the music played is a short music track<br><code> Valid values: </code><code> true </code>: short music track that needs to be looped; <code> false </code> (default): normal-length music track |
-| loopCount | <code> Field description: </code> number of times the music track is looped<br><code> Valid values: </code>0 or any positive integer. 0 (default) indicates that the music is played once, 1 twice, and so on. |
-| path | <code> Field description: </code> absolute path of the music file or url.the mp3,aac,m4a,wav supported. |
-| publish | <code> Field description: </code> whether to send the music to remote users<br><code> Valid values: </code><code> true </code>: remote users can hear the music played locally; <code> false </code> (default): only the local user can hear the music. |
-| startTimeMS | <code> Field description: </code> the point in time in milliseconds for starting music playback |
+| endTimeMS | <code>Field description:</code>  the point in time in milliseconds for ending music playback. 0 indicates that playback continues till the end of the music track. |
+| id | <code>Field description:</code>  music ID<br><strong>Note</strong><br>the SDK supports playing multiple music tracks. IDs are used to distinguish different music tracks and control their start, end, volume, etc. |
+| isShortFile | <code>Field description:</code>  whether the music played is a short music track<br><code>Valid values:</code>  <code>true</code> : short music track that needs to be looped;  <code>false</code>  (default): normal-length music track |
+| loopCount | <code>Field description:</code>  number of times the music track is looped<br><code>Valid values:</code> 0 or any positive integer. 0 (default) indicates that the music is played once, 1 twice, and so on. |
+| path | <code>Field description:</code>  absolute path of the music file or url.the mp3,aac,m4a,wav supported. |
+| publish | <code>Field description:</code>  whether to send the music to remote users<br><code>Valid values:</code>  <code>true</code> : remote users can hear the music played locally;  <code>false</code>  (default): only the local user can hear the music. |
+| startTimeMS | <code>Field description:</code>  the point in time in milliseconds for starting music playback |

package/resource/knowledge/trtc/c++/ITXDeviceManager.md

@@ -107,9 +107,9 @@ This API can be used to achieve the following:
 
 > **Note**
 >
-
 > Before using this API, you must first disable auto focus using enableCameraAutoFocus.
 >
+
 **Return Desc:**
 
 0: operation successful; negative number: operation failed.
@@ -138,9 +138,9 @@ If the audio route is set to the speaker, the volume is relatively high, and the
 
 > **Note**
 >
-> -  To ensure that the SDK can manage the lifecycle of the ` ITXDeviceCollection ` object, after using this API, please call the ` release ` method to release the resources.
-> -  Do not use ` delete ` to release the Collection object returned as deleting the ITXDeviceCollection* pointer will cause crash.
-> -  The valid values of ` type ` are ` TXMediaDeviceTypeMic `, ` TXMediaDeviceTypeSpeaker `, and ` TXMediaDeviceTypeCamera `.
+> -  To ensure that the SDK can manage the lifecycle of the  `ITXDeviceCollection`  object, after using this API, please call the  `release`  method to release the resources.
+> -  Do not use  `delete`  to release the Collection object returned as deleting the ITXDeviceCollection* pointer will cause crash.
+> -  The valid values of  `type`  are  `TXMediaDeviceTypeMic` ,  `TXMediaDeviceTypeSpeaker` , and  `TXMediaDeviceTypeCamera` .
 > -  This API can be used only on macOS and Windows.
 
 ## setCurrentDevice
@@ -150,7 +150,7 @@ If the audio route is set to the speaker, the volume is relatively high, and the
 |Param|DESC|
 |---------|---------|
 |deviceId|Device ID. You can get the ID of a device using the getDevicesList API.|
-|type|Device type. For details, please see the definition of ` TXMediaDeviceType `.|
+|type|Device type. For details, please see the definition of  `TXMediaDeviceType` .|
 **Return Desc:**
 
 0: operation successful; negative number: operation failed.
@@ -195,7 +195,7 @@ This API is used to set the microphone and speaker types. Camera following the s
 | Param | DESC |
 | --- | --- |
 | enable | Whether to follow the system default audio device.<br>-  true: following. When the default audio device of the system is changed or new audio device is plugged in, the SDK immediately switches the audio device.<br>-  false：not following. When the default audio device of the system is changed or new audio device is plugged in, the SDK doesn't switch the audio device. |
-| type | Device type. For details, please see the definition of <code> TXMediaDeviceType </code>. |
+| type | Device type. For details, please see the definition of  <code>TXMediaDeviceType</code> . |
 
 ## startCameraDeviceTest
 
@@ -203,7 +203,6 @@ This API is used to set the microphone and speaker types. Camera following the s
 
 > **Note**
 >
-
 > You can use the setCurrentDevice API to switch between cameras during testing.
 >
 
@@ -223,7 +222,6 @@ This API is used to test whether the mic functions properly. The mic volume dete
 
 > **Note**
 >
-
 > When this interface is called, the sound recorded by the microphone will be played back to the speakers by default.
 >
 
@@ -236,7 +234,7 @@ This API is used to test whether the mic functions properly. The mic volume dete
 |Param|DESC|
 |---------|---------|
 |interval|Interval of volume callbacks, in milliseconds.|
-|playback|Whether to play back the microphone sound. The user will hear his own sound when testing the microphone if ` playback ` is true.|
+|playback|Whether to play back the microphone sound. The user will hear his own sound when testing the microphone if  `playback`  is true.|
 
 ## stopMicDeviceTest
 
@@ -260,7 +258,7 @@ This API is used to test whether the audio playback device functions properly by
 
 **Starting camera testing (for desktop OS).**
 
-This API supports custom rendering, meaning that you can use the callback API ` ITRTCVideoRenderCallback ` to get the images captured by the camera for custom rendering.
+This API supports custom rendering, meaning that you can use the callback API  `ITRTCVideoRenderCallback`  to get the images captured by the camera for custom rendering.
 
 ## setApplicationPlayVolume
 
@@ -290,7 +288,7 @@ This API supports custom rendering, meaning that you can use the callback API `
 
 **Setting the system volume type (for mobile OS).**
 
-@deprecated This API is not recommended after v9.5. Please use the ` startLocalAudio(quality) ` API in ` TRTCCloud ` instead, which param ` quality ` is used to decide audio quality.
+@deprecated This API is not recommended after v9.5. Please use the  `startLocalAudio(quality)`  API in  `TRTCCloud`  instead, which param  `quality`  is used to decide audio quality.
 
 ## TXSystemVolumeType(Deprecated)