Global

Aspect ratio: 1:1; resolution: 120x120; recommended bitrate (VideoCall): 80 Kbps; recommended bitrate (LIVE): 120 Kbps.

Aspect ratio: 1:1; resolution: 160x160; recommended bitrate (VideoCall): 100 Kbps; recommended bitrate (LIVE): 150 Kbps.

Aspect ratio: 1:1; resolution: 270x270; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.

Aspect ratio: 1:1; resolution: 480x480; recommended bitrate (VideoCall): 350 Kbps; recommended bitrate (LIVE): 500 Kbps.

Aspect ratio: 4:3; resolution: 160x120; recommended bitrate (VideoCall): 100 Kbps; recommended bitrate (LIVE): 150 Kbps.

Aspect ratio: 4:3; resolution: 240x180; recommended bitrate (VideoCall): 150 Kbps; recommended bitrate (LIVE): 250 Kbps.

Aspect ratio: 4:3; resolution: 280x210; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.

Aspect ratio: 4:3; resolution: 320x240; recommended bitrate (VideoCall): 250 Kbps; recommended bitrate (LIVE): 375 Kbps.

Aspect ratio: 4:3; resolution: 400x300; recommended bitrate (VideoCall): 300 Kbps; recommended bitrate (LIVE): 450 Kbps.

Aspect ratio: 4:3; resolution: 480x360; recommended bitrate (VideoCall): 400 Kbps; recommended bitrate (LIVE): 600 Kbps.

Aspect ratio: 4:3; resolution: 640x480; recommended bitrate (VideoCall): 600 Kbps; recommended bitrate (LIVE): 900 Kbps.

Aspect ratio: 4:3; resolution: 960x720; recommended bitrate (VideoCall): 1000 Kbps; recommended bitrate (LIVE): 1500 Kbps.

Aspect ratio: 16:9; resolution: 160x90; recommended bitrate (VideoCall): 150 Kbps; recommended bitrate (LIVE): 250 Kbps.

Aspect ratio: 16:9; resolution: 256x144; recommended bitrate (VideoCall): 200 Kbps; recommended bitrate (LIVE): 300 Kbps.

Aspect ratio: 16:9; resolution: 320x180; recommended bitrate (VideoCall): 250 Kbps; recommended bitrate (LIVE): 400 Kbps.

Aspect ratio: 16:9; resolution: 480x270; recommended bitrate (VideoCall): 350 Kbps; recommended bitrate (LIVE): 550 Kbps.

Aspect ratio: 16:9; resolution: 640x360; recommended bitrate (VideoCall): 500 Kbps; recommended bitrate (LIVE): 900 Kbps.

Aspect ratio: 16:9; resolution: 960x540; recommended bitrate (VideoCall): 850 Kbps; recommended bitrate (LIVE): 1300 Kbps.

Aspect ratio: 16:9; resolution: 1280x720; recommended bitrate (VideoCall): 1200 Kbps; recommended bitrate (LIVE): 1800 Kbps. If used for screen share, recommended bitrate 1000Kbps for LD(Low Definition), 1600Kbps for HD(High Definition)

Aspect ratio: 16:9; resolution: 1920x1080; recommended bitrate (VideoCall): 2000 Kbps; recommended bitrate (LIVE): 3000 Kbps.

横屏分辨率

竖屏分辨率

HD big image: it is generally used to transfer video data from the camera.

Smooth small image: it has the same content as the big image, but with lower resolution and bitrate and thus lower definition.

Substream image: it is generally used for screen sharing. Only one user in the room is allowed to publish the substream video image at any time, while other users must wait for this user to close the substream before they can publish their own substream.

Undefined

Excellent

Good

Fair

Bad

Very bad

Down: Network cannot meet the minimum requirements of TRTC

Fill mode: the video image will be centered and scaled to fill the entire display area, where parts that exceed the area will be cropped. The displayed image may be incomplete in this mode.

Fit mode: the video image will be scaled based on its long side to fit the display area, where the short side will be filled with black bars. The displayed image is complete in this mode, but there may be black bars.

No rotation

Clockwise rotation by 90 degrees

Clockwise rotation by 180 degrees

Clockwise rotation by 270 degrees

Smooth style, which uses a more radical algorithm for more obvious effect and is suitable for show live streaming.

Natural style, which retains more facial details for more natural effect and is suitable for most live streaming use cases.

Undefined format

YUV420P (I420) format

OpenGL 2D texture format

BGRA32 format

RGBA32 format

Undefined transfer method

Use memory buffer to transfer video data. iOS: PixelBuffer; Android: Direct Buffer for JNI layer; Windows: memory data block.

Auto mode: mirror the front camera's image but not the rear camera's image(for mobile devices only).

Mirror the images of both the front and rear cameras.

Disable mirroring for both the front and rear cameras.

Record audio only.

Record video only.

Record both audio and video.

In the video call scenario, 720p and 1080p HD image quality is supported. A single room can sustain up to 300 concurrent online users, and up to 50 of them can speak simultaneously. Use cases: [one-to-one video call], [video conferencing with up to 300 participants], [online medical diagnosis], [small class], [video interview], etc.

In the interactive video live streaming scenario, mic can be turned on/off smoothly without waiting for switchover, and the anchor latency is as low as less than 300 ms. Live streaming to hundreds of thousands of concurrent users in the audience role is supported with the playback latency down to 1,000 ms. Use cases: [low-latency interactive live streaming], [big class], [anchor competition], [video dating room], [online interactive classroom], [remote training], [large-scale conferencing], etc.

Notice: In this scenario, you must use the role field in TRTCParams to specify the role of the current user.

Audio call scenario, where the SPEECH sound quality is used by default. A single room can sustain up to 300 concurrent online users, and up to 50 of them can speak simultaneously. Use cases: [one-to-one audio call], [audio conferencing with up to 300 participants], [audio chat], [online Werewolf], etc.

In the interactive audio live streaming scenario, mic can be turned on/off smoothly without waiting for switchover, and the anchor latency is as low as less than 300 ms. Live streaming to hundreds of thousands of concurrent users in the audience role is supported with the playback latency down to 1,000 ms. Use cases: [audio club], [online karaoke room], [music live room], [FM radio], etc.

Notice: In this scenario, you must use the role field in TRTCParams to specify the role of the current user.

Anchor

Audience

ModeClient(Only for SDK internal test and debug)

ModeServer(Default)

Ensuring smoothness

Ensuring clarity (default value)

Undefined

Audio data in PCM format.

Undefined

The screen sharing target is the window of an application

The screen sharing target is the entire screen

The screen sharing target is a user-defined data source

Speech mode: sample rate: 16 kHz; mono channel; bitrate: 16 Kbps. This mode has the best resistance among all modes and is suitable for audio call scenarios, such as online meeting and audio call.

Default mode: sample rate: 48 kHz; mono channel; bitrate: 50 Kbps. This mode is between the speech mode and the music mode as the default mode in the SDK and is recommended.

Music mode: sample rate: 48 kHz; full-band stereo; bitrate: 128 Kbps. This mode is suitable for scenarios where Hi-Fi music transfer is required, such as online karaoke and music live streaming.

Output logs at all levels

Output logs at the DEBUG, INFO, WARNING, ERROR, and FATAL levels

Output logs at the INFO, WARNING, ERROR, and FATAL levels

Output logs at the WARNING, ERROR, and FATAL levels

Output logs at the ERROR and FATAL levels

Output logs at the FATAL level

Do not output any SDK logs

The device has been plugged in

The device has been removed

The device has been enabled

System default device changed

undefined device type

microphone

speaker or earpiece

camera

Auto adjustment of camera acquisition parameters. SDK selects the appropriate camera output parameters according to the actual acquisition device performance and network situation, and maintains a balance between device performance and video preview quality.

Give priority to equipment performance. SDK selects the closest camera output parameters according to the user's encoder resolution and frame rate, so as to ensure the performance of the device.

Give priority to the quality of video preview. SDK selects higher camera output parameters to improve the quality of preview video. In this case, it will consume more CPU and memory to do video preprocessing.

Allows the user to set the width and height of the video captured by the local camera.

Path of the image file, which can be in BMP, GIF, JPEG, PNG, TIFF, Exif, WMF, or EMF format

Memory block in BGRA32 format

Memory block in RGBA32 format

Undefined

Manual layout mode In this mode, you need to specify the precise position of each video image. This mode has the highest degree of freedom, but its ease of use is the worst:

• You need to enter all the parameters in TRTCTranscodingConfig, including the position coordinates of each video image (TRTCMixUser).
• You need to listen on the onUserVideoAvailable() and onUserAudioAvailable() event callbacks in TRTCCloudDelegate and constantly adjust the mixUsers parameter according to the audio/video status of each user with mic on in the current room.

Pure audio mode This mode is suitable for pure audio scenarios such as audio call (AudioCall) and audio chat room (VoiceChatRoom).

• You only need to set it once through the setMixTranscodingConfig() API after room entry, and then the SDK will automatically mix the audio of all mic-on users in the room into the current user's live stream.
• You don't need to set the mixUsers parameter in TRTCTranscodingConfig; instead, you only need to set the audioSampleRate, audioBitrate and audioChannels parameters.

Preset layout mode This is the most popular layout mode, because it allows you to set the position of each video image in advance through placeholders, and then the SDK automatically adjusts it dynamically according to the number of video images in the room. In this mode, you still need to set the mixUsers parameter, but you can set userId as a "placeholder". Placeholder values include:

• "$PLACE_HOLDER_REMOTE$": image of remote user. Multiple images can be set.
• "$PLACE_HOLDER_LOCAL_MAIN$": local camera image. Only one image can be set.
• "$PLACE_HOLDER_LOCAL_SUB$": local screen sharing image. Only one image can be set. In this mode, you don't need to listen on the onUserVideoAvailable() and onUserAudioAvailable() callbacks in TRTCCloudDelegate to make real-time adjustments. Instead, you only need to call setMixTranscodingConfig() once after successful room entry. Then, the SDK will automatically populate the placeholders you set with real userId values.

Screen sharing mode This mode is suitable for screen sharing-based use cases such as online education and supported only by the SDKs for Windows and macOS. In this mode, the SDK will first build a canvas according to the target resolution you set (through the videoWidth and videoHeight parameters).

• Before the teacher enables screen sharing, the SDK will scale up the teacher's camera image and draw it onto the canvas.
• After the teacher enables screen sharing, the SDK will draw the video image shared on the screen onto the same canvas. The purpose of this layout mode is to ensure consistency in the output resolution of the mixtranscoding module and avoid problems with blurred screen during course replay and webpage playback (web players don't support adjustable resolution). Meanwhile, the audio of mic-on students will be mixed into the teacher's audio/video stream by default.
  Video content is primarily the shared screen in teaching mode, and it is a waste of bandwidth to transfer camera image and screen image at the same time. Therefore, the recommended practice is to directly draw the camera image onto the current screen through the setLocalVideoRenderCallback API. In this mode, you don't need to set the mixUsers parameter in TRTCTranscodingConfig, and the SDK will not mix students' images so as not to interfere with the screen sharing effect.
  You can set width x height in TRTCTranscodingConfig to 0 px x 0 px, and the SDK will automatically calculate a suitable resolution based on the aspect ratio of the user's current screen.
• If the teacher's current screen width is less than or equal to 1920 px, the SDK will use the actual resolution of the teacher's current screen.
• If the teacher's current screen width is greater than 1920 px, the SDK will select one of the three resolutions of 1920x1080 (16:9), 1920x1200 (16:10), and 1920x1440 (4:3) according to the current screen aspect ratio.

Unspecified. The SDK will determine the stream mix input type according to the value of the pureAudio parameter

Mix both audio and video

Mix video only

Mix audio only

Mix watermark. In this case, you don't need to specify the userId parameter, but you need to specify the image parameter. It is recommended to use png format.

Record both local and remote audio

Record local audio only

Record remote audio only

Undefined mode

You can set this parameter to publish the main stream in your room (TRTCVideoStreamTypeBig) to Tencent or third-party live streaming CDN service providers (only standard RTMP protocol is supported).

You can set this parameter to publish the auxiliary stream in your room (TRTCVideoStreamTypeSub) to Tencent or third-party live streaming CDN service providers (only standard RTMP protocol is supported).

You can set this parameter, in conjunction with the encoding output parameters (TRTCStreamEncoderParam) and the mixing and transcoding parameters (TRTCStreamMixingConfig), to transcode and publish your specified multiple audio and video streams to Tencent or third-party live streaming CDN service providers (only standard RTMP protocol is supported).

You can set this parameter, in conjunction with the media stream encoding output parameters (TRTCStreamEncoderParam) and the mixing and transcoding parameters (TRTCStreamMixingConfig), to transcode and publish your specified multiple audio and video streams to the room you specify. Specify the robot information of the push-back room through TRTCUser in TRTCPublishTarget.

disable

KTV

small room

great hall

deep voice

loud voice

metallic sound

magnetic sound

ethereal

studio

melodious

studio2

disable

Naughty child

Lolita

Uncle

Heavy Metal

Influenza

Foreign Language

Trapped Beast

Plump

Heavy Current

Heavy Machinery

Intangible

Video processing plugin type, useful for customized beauty such as AI beauty

Media encryption and decryption, useful for customized security

audio processing plugin type, useful for custom audio processing

[Field Meaning] Maximum number of parallel playback. Default value: 0.

• If maxCount > 0 and the actual number of people > maxCount, the SDK will intelligently select maxCount routes of data for playback in real-time, which will greatly reduce performance consumption.
• If maxCount = 0, the SDK does not limit the number of parallel playback. This may trigger performance issues in rooms with many people on the mic.

[Field Meaning] Specifies users who must be able to play back in parallel.
[Special Instructions] Specifies the user ID list that must be played back in parallel. These users do not participate in intelligent selection.

• The number of includeUsers must be less than maxCount, otherwise the parallel playback setting will be invalid.
• includeUsers is only valid when maxCount > 0. When includeUsers takes effect, the maximum number of parallel playback participants in intelligent selection = maxCount - the number of valid includeUsers.

Acquisition mode

Acquisition width

Acquisition height

plugin ID. User should be sure that it is unique for each plugin.

camera device ID. Required when you open multi-camera for customized beauty. Default is empty string.

enable current plugin. Success when return 0, otherwise failed.

disable current plugin. Success when return 0, otherwise failed.

set parameter to control plugin. Success when return 0, otherwise failed. The parameter is a JSON string and will be passed to C++ plugin's setParameter function.

Whether to enable

Pixel format

Whether to enable

Whether to enable

[Field Meaning] Specify the streaming address (RTMP format) of this audio and video stream on Tencent or third-party live streaming service providers.
[Recommended Value] The streaming address rules of each service provider vary greatly. Please fill in a valid streaming URL according to the requirements of the target service provider. TRTC's backend server will push standard format audio and video streams to third-party service providers according to the URL you fill in.
[Special Note] The streaming URL must be in RTMP format and must comply with the specification requirements of your target live streaming service provider, otherwise the target service provider will reject the streaming request from TRTC's backend service.

[Field Meaning] Specify whether this audio and video stream is published to Tencent Cloud.
[Recommended Value] Default value: true.
[Special Note] If your target live streaming service provider is Tencent, please set this parameter to true. At this time, the backend billing system will not calculate the forwarding service fee for this.

[Field Meaning] User identifier, the current user's userId, equivalent to the username, using UTF-8 encoding.
[Recommended Value] If a user's ID in your account system is "mike", then userId can be set to "mike".

[Field Meaning] Numeric room number, needs to match the room number type in your room entry parameters (TRTCParams).
[Recommended Value] Value range: 1 - 4294967294.
[Special Note] intRoomId and strRoomId are mutually exclusive. If you choose strRoomId in your room entry parameters, then intRoomId needs to be set to 0. If both are filled, the SDK will preferentially use intRoomId.

[Field Meaning] String room number, needs to match the room number type in your room entry parameters (TRTCParams).
[Special Note] intRoomId and strRoomId are mutually exclusive. If you choose roomId in your room entry parameters, then strRoomId does not need to be filled. If both are filled, the SDK will preferentially use intRoomId.
[Recommended Value] Limited to a length of 64 bytes. The following is the supported character set range (a total of 89 characters):

• Uppercase and lowercase English letters (a-zA-Z);
  
• Digits (0-9);
  
• Space, !,#,$,%,&,(,),+,-,:,;,<,=,.,>,?,@,[,],^,_,{,},|,~,,.

[Field Meaning] Media stream publishing mode.
[Recommended Value] Please choose according to your business scenario requirements. TRTC supports forwarding, transcoding, and pushing back to RTC room modes.
[Special Note] If your business scenario requires multiple publishing modes, you can call the media stream publishing interface (startPublishMediaStream) multiple times and set different TRTCPublishTarget respectively.
[Special Note] The same mode should correspond to one media stream publishing interface (startPublishMediaStream), and use updatePublishMediaStream for updates when needed.

[Field Meaning] Streaming address (RTMP format) published to Tencent or third-party live streaming service providers.
[Special Note] If your mode is set to TRTCPublishMixStreamToRoom, you do not need to set this parameter.

[Field Meaning] Push-back room robot information.
[Special Note] Only when your mode is set to TRTCPublishMixStreamToRoom, you need to set this parameter.
[Special Note] After setting, the transcoded audio and video data will be pushed back to the room you specify. It is recommended to set a special userId to avoid confusion between the push-back robot and the anchor who enters the room through the TRTC SDK.
[Special Note] Users participating in the mix stream do not support subscribing to this transcoded stream.
[Special Note] When the subscription mode (setDefaultStreamRecvMode) you set before entering the room is manual, you need to manage the audio and video streams you want to pull yourself (usually when you pull the transcoded stream of the push-back room, you should unsubscribe from the corresponding audio and video single streams involved in transcoding).
[Special Note] When the subscription mode (setDefaultStreamRecvMode) you set before entering the room is automatic, users who do not participate in transcoding will automatically receive the transcoded stream issued by the backend and will no longer continue to receive the audio and video single streams involved in transcoding. Unless you explicitly unsubscribe (muteRemoteVideoStream and muteRemoteAudio), the transcoded stream data will continue to be issued.

[Field Meaning] Specifies the target resolution (width) of the media publishing stream.
[Recommended Value] Unit: pixel value, recommended value: 368. If you only mix audio streams, set both width and height to 0, otherwise there will be a black background in the live stream after mixing and transcoding.

[Field Meaning] Specifies the target resolution (height) of the media publishing stream.
[Recommended Value] Unit: pixel value, recommended value: 640. If you only mix audio streams, set both width and height to 0, otherwise there will be a black background in the live stream after mixing and transcoding.

[Field Meaning] Specifies the target video frame rate (FPS) of the media publishing stream.
[Recommended Value] Recommended value: 20fps, value range is (0,30].

[Field Meaning] Specifies the target video key frame interval (GOP) of the media publishing stream.
[Recommended Value] Recommended value: 3, unit: seconds, value range is [1,5].

[Field Meaning] Specifies the target video bitrate (kbps) of the media publishing stream.
[Recommended Value] If you fill in 0, TRTC will estimate a reasonable bitrate value based on videoEncodedWidth and videoEncodedHeight. You can also refer to the recommended bitrate value in the video resolution enumeration definition (see the comment part).

[Field Meaning] Specifies the target audio sampling rate of the media publishing stream.
[Recommended Value] Default value: 48000Hz. Value range is [48000, 44100, 32000, 24000, 16000, 8000], unit is Hz.

[Field Meaning] Specifies the target number of audio channels of the media publishing stream.
[Recommended Value] Default value: 1, representing mono. There are only two possible values: 1-mono, 2-stereo.

[Field Meaning] Specifies the target audio bitrate (kbps) of the media publishing stream.
[Recommended Value] Default value: 50kbps, value range is [32, 192].

[Field Meaning] Specifies the target audio encoding type of the media publishing stream.
[Recommended Value] Default value: 0, representing LC-AAC. There are only three possible values: 0 - LC-AAC, 1 - HE-AAC, 2 - HE-AACv2.
[Special Note] HE-AAC and HE-AACv2 support output stream audio sampling rates ranging from [48000, 44100, 32000, 24000, 16000].
[Special Note] When the audio encoding is set to HE-AACv2, only stereo audio channels are supported.

[Field Meaning] Specifies the target video encoding type of the media publishing stream.
[Recommended Value] Default value: 0, representing H264. There are only two possible values: 0 - H264, 1 - H265.

[Field Meaning] Mixing SEI parameters, not filled in by default.
[Special Note] The parameter is passed in as a JSON string, as shown below:

{
"payLoadContent":"xxx",
"payloadType":5,
"payloadUuid":"1234567890abcdef1234567890abcdef",
"interval":1000,
"followIdr":false
}

Currently supported fields and meanings:

• payloadContent: Required. The payload content of the transmitted sei, which cannot be empty;
• payloadType: Required. The type of the sei message, value range: 5 or integers within [100, 254] (except 244, which is an internally customized timestamp sei);
• payloadUuid: Must be filled in when payloadType is 5, otherwise it will be ignored. This value must be a 32-character hexadecimal number;
• interval: Optional, default 1000. The sending interval of the sei, in milliseconds;
• followIdr: Optional, default false. When this value is true, it ensures that the sei is carried when sending keyframes, otherwise it does not guarantee.

[Field Meaning] Specifies the coordinate area (unit: pixels) of this screen.

[Field Meaning] Specifies the level (value range: 0 - 15, non-repetitive) of this screen.

[Field Meaning] Image filling mode.
[Recommended Value] Fill (the image may be stretched and cropped) or adapt (there may be black edges), default value: TRTCVideoFillMode_Fill.

[Field Meaning] Specifies the background color of the mixed screen.
[Recommended Value] Default value: 0x000000 represents black. The format is a hexadecimal number, such as "0x61B9F1" represents RGB values of (97,158,241).

[Field Meaning] Placeholder image URL, that is, when a specified user temporarily only uploads audio, Tencent Cloud's backend server will mix the image specified by this URL into the final screen.
[Recommended Value] Default value: null, that is, no placeholder image is set.
[Special Note] At this time, you need to specify the userId in the fixedVideoUser user information.
[Special Note]

• The URL link length limit is 512 bytes. The image size limit does not exceed 2MB.
• Image formats support png, jpg, jpeg, bmp formats. It is recommended to use a semi-transparent png image as a placeholder.

[Field Meaning] Information of users participating in transcoding.
[Special Note] User information (TRTCUser) can be left blank (that is, userId, intRoomId and strRoomId are not filled in). At this time, when there is audio and video data uploaded by the anchor in the room where the mix stream is initiated, TRTC's backend server will automatically fill the corresponding anchor's audio and video into the layout you specified.

[Field Meaning] Specifies whether this screen is a main screen (TRTCVideoStreamTypeBig) or a secondary screen (TRTCVideoStreamTypeSub).

[Field Meaning] Watermark URL, Tencent Cloud's backend server will mix the image specified by this URL into the final image.
[Special Note]

• The URL link length limit is 512 bytes. The image size limit does not exceed 2MB.
• Image formats support png, jpg, jpeg, bmp formats. It is recommended to use a semi-transparent png image as a watermark.

[Field Meaning] Specifies the coordinate area (unit: pixels) of this watermark image.

[Field Meaning] Specifies the level (value range: 0 - 15, non-repetitive) of this watermark image.

[Field Meaning] Specifies the background color of the mixed screen.
[Recommended Value] Default value: 0x000000 represents black. The format is a hexadecimal number, such as "0x61B9F1" representing RGB values of (97, 158, 241).

[Field Meaning] Specifies the background image URL of the mixed screen. Tencent Cloud's backend server will mix the image specified by this URL into the final image.
[Recommended Value] Default value: null, that is, no background image is set.
[Special Note]

• The URL link length limit is 512 bytes. The image size limit does not exceed 2MB.
• Image formats support png, jpg, jpeg, bmp formats. It is recommended to use a semi-transparent png image as the background image.

[Field Meaning] Specifies the position, size, layer, and stream type information of each video screen in the mixed screen.
[Recommended Value] This field is an array of TRTCVideoLayout types. Each element in the array represents the information of each screen.

[Field Meaning] Specifies the information of each input audio stream in the transcoded stream.
[Recommended Value] This field is an array of TRTCUser types. Each element in the array represents the information of each input audio stream.
[Special Note] User information can be left blank (that is, audioMixUserList is empty). At this time, if the audio-related encoding output parameters in TRTCStreamEncoderParam are set, TRTC's backend server will automatically mix all anchors' audio for output (currently only supports up to 16 channels of audio and video input).

[Field Meaning] Specifies the position, size, and layer information of each watermark screen in the mixed screen.
[Recommended Value] This field is an array of TRTCWatermark types. Each element in the array represents the information of each watermark.

Background music started event

Background music playing progress event

Background music completed event

Background music preloading progress event

Background music preloading error

Audio data captured by the local mic and pre-processed by the audio module.

Audio data captured by the local mic, pre-processed by the audio module, effect-processed and BGM-mixed.

Audio data of each remote user before audio mixing.

Data mixed from each channel before being submitted to the system for playback.

Data mixed from all the captured and to-be-played audio in the SDK.