Interface

The number of sample data contained in each channel of the audio data for each callback can only be set to the following values: 256, 512, 1024, 2048, 4096, 8192, 16384. Default is 4096.

Specifies the sampling rate of the callback audio data conversion. It can only be set to the following values: 8000, 16000, 22050, 24000, 32000, 44100, 48000. The default is to set it according to the original sampling rate of AudioContext.

Specifies the number of channels of the callback audio data. Only the following values ​​can be set: 1, 2. The default is based on the number of channels of the original data.

The audio data bit depth can only be set to the following values: 32 for 32-bit float, 16 for 16-bit. The default value is 32.

背景图片对象

The lower limit of the buffer interval. The optional range of minInterval is in the [0, 1000] ms interval. The default is 0.

The upper limit of the buffer interval. The optional range of maxInterval is in the [1000, 10000] ms interval. The default is 2000 ms.

Optional parameter, other options are ignored when present. Strictly specify the final output value of the acquisition device. If the device does not support the specified value, the acquisition will fail.

Optional parameter, expect the final output value of the collection device. If the device does not support the specified value, it will try to output the closest value.

Optional parameter, the upper limit of the final output value of the collection device.

Optional parameter, the lower limit of the final output value of the collection device.

Event code

Information description

Video frame width

Video frame height

The original frame rate of the video resource

Audio sample rate

Video resource duration, the usual value for CDN live streams is -1.

The number of audio channels. By default, binaural audio will be used to render audio.

Video encoding format name

Audio encoding format name

Media packaging format name

decodeType

video bitrate, KB/s

audio bitrate，KB/s

video decode bitrate，KB/s

audio decode bitrate，KB/s

has audio

has audio

video render fps

audio render fps

audio is muted

video target render frame rate

audio target render frame rate

network quality grade (0~100

Total number of bytes of data received

Number of bytes of video data received

Number of bytes of audio data received

The number of bytes of SEI data received

Audio bit rate

Audio encoding format "opus"

Network jitter

volume

Number of lost packets

Packet loss rate

Stream audio quality，- 1 for unknown, 0 for excellent, 1 for good, 2 for medium, 3 for poor, 4 for very poor

Sampling Rate

Whether the audio track is turned off

Audio frame rate

Audio transmission energy

Audio encoding format

Detail description: The cumulative audio decoding time, in milliseconds (supported by version 2.19.0 and above).

Detail description: The cumulative video freeze duration, in milliseconds (supported by version 2.19.0 and above).

Detail description: The cumulative video freezing rate, the unit is 1, and the value range is [0,1] (supported by version 2.19.0 and above).

End to end delay, in ms.

End to end packet loss rate, unit: 1.

Audio bit rate

Audio encoding format

Audio packet loss

Audio packet loss rate

Audio encoding format

Whether the audio track is turned off

Audio frame rate

audio quality, - 1 for unknown, 0 for excellent, 1 for good, 2 for medium, 3 for poor, 4 for very poor

Set whether to keep on when the page is hidden to get the sound wave and call back the sound wave callback. The default is off in versions after 2.18.0.

Autoplay failed event.

Fires when the user can start playing the video.

Fires when the user can start playing audio.

Specify the address to load the online audio resource. If the audioEffectID has already loaded a sound effect, the loaded sound effect will be used directly.

1. Online audio files need to comply with the Browser's Same Origin Policy.
2. MP3, AAC, and other audio formats supported by the browser are supported.

Audio raw data, the array is the audio raw data of each channel.

channel count

Audio sample rate.

Mixed-flow task id (customer-defined, must be unique), required, maximum 256 characters, only supports numbers, English characters and'~','!','@','#','$', ' ','^','&','*','(',')','_','+','=','-',',';',''',', ', '

The room ID of the automatic streaming task.

Audio configuration of automatic stream mixing tasks, including audio bitrate, audio channel, encoding ID, and multi-channel audio stream mixing mode.

Use case: If there are special requirements for the audio of automatic stream mixing tasks, such as adjusting the audio bitrate, you can adjust this parameter as needed, otherwise no configuration is required.

The output stream list of the automatic stream mixing task, which contains URLs or stream IDs. If it is in URL format, it currently only supports the RTMP URL format: rtmp://xxxxxxxx.

Enables the mixed flow sound wave callback notification, which is used to obtain the sound wave of each audio input in the mixed flow. When postmixing is enabled, the sound information of each single stream can be received through the on[autoMixerSoundLevelUpdate] callback.

Stream alignment mode, 0 means off, 1 means on.

Set the lower limit of the interval range for adaptive adjustment of the stream cache of the mixing server. In the real-time chorus KTV scenario, slight fluctuations in the network on the push end may cause the mixed stream to freeze. At this time, when the audience pulls the mixed stream, there is a high probability of freezing. By adjusting the lower limit of the interval range for adaptive adjustment of the stream cache on the mixing server, the freezing problem that occurs when the audience pulls the mixed stream can be optimized, but the delay will be increased. Not set by default, that is, the server uses its own configuration value. It will only take effect on new input streams and will not take effect on input streams that have already started mixing.

Whether to enable automatic switching of camera devices

Whether to enable automatic switching of microphone devices

Send message user

Message content

Send time

Message ID

Send message user

Message content

Send time

Message ID

Whether audio is needed, default is true

Audio input device, the default device selected for the browser

Audio bit rate, the default is 48kbps

Whether video is required, the default is true.

Video input devices, pass the default device selected for the browser.

Video quality level, the default quality level is 4, (the default level for versions before 2.21.0 is 2): 1: Resolution 320 * 240, frame rate 15, code rate 300 kbps. 2: Resolution 640 * 480, frame rate: 15, code rate 800 kbps. 3: Resolution 1280 * 720, frame rate: 20, code rate 1500 kbps. 4: When videoQuality is 4, the four parameters of width, height, frameRate and bitrate can be customized and modified. If not passed, the scene configuration value shall prevail. The default is the common scene value: resolution 640 * 480, frame rate: 15, Bit rate 800 kbps.

Camera orientation, "user" means the front camera, "environment" means the rear camera. If not passed, the default is to use the device specified by videoInput or the default camera device of the operating system.

Number of channels, 1 is mono, 2 is two channels, the default is mono. If you turn on the microphone collection when pulling streams on the Safari browser, the stereo effect will be lost. You need to avoid turning on the microphone when pulling streams.

Whether to turn on noise reduction or not will be determined based on the browser's capabilities. It is usually turned on by default.

Whether to turn on automatic gain or not will be determined based on the browser's capabilities. It is usually turned on by default.

Whether to turn on echo cancellation or not will be determined based on the browser's capabilities. It is usually turned on by default.

Wide resolution, effective when videoQuality is 4.

High resolution, effective when videoQuality is 4

Bit rate, unit kb/s, effective when videoQuality is 4.

Frame rate, effective when videoQuality is 4

The starting bit rate, the default is "default", environmental requirements: Google kernel browser and SDK version 2.7.0 and above. A value of "default" means to use webrtc's default start bitrate of 300kbps and then slowly increase; a value of "target" means to quickly increase to the set target bitrate.

Video optimization mode, default to "default";

A value of "motion" indicates smooth mode. In most cases, the SDK will not reduce the frame rate, but may reduce the transmission resolution.

A value of "detail" indicates clear mode. In most cases, the SDK will not reduce the transmission resolution, but may reduce the frame rate.

The minimum streaming bit rate. The minimum bit rate for acceptable video quality degradation when the network is poor.

The length of the video keyframe interval, in seconds.

Whether to support the webRTC protocol, true means that the webRTC protocol can be used to transmit the stream

Whether to support custom streams (other streams not captured by camera or screen capture, such as mp4 played by video tags, etc.)

Does the camera have permission to call

Does the microphone have permission to call

Whether to support screen sharing function

The video encoding format supported by the browser; it should be noted that some browsers will support a certain encoding format, but the actual system castrated this function, so if a certain encoding format returns false, it must not be supported, true is not necessarily 100 %stand by

Error information related to detection failure, device-related error description can refer to MediaDevices.getUserMedia The exception description of the interface document.

Judge whether it is supported or not according to the type passed in

error name

error description information

For specific error descriptions, please refer to MediaDevices.getUserMedia interface documentation The exception description.

For specific error descriptions, please refer to MediaDevices.getUserMedia interface documentation The exception description.

Video input devices, pass the default device selected for the browser.

Video quality level, the default quality level is 4, (the default level for versions before 2.21.0 is 2): 1: Resolution 320 * 240, frame rate 15, code rate 300 kbps. 2: Resolution 640 * 480, frame rate: 15, code rate 800 kbps. 3: Resolution 1280 * 720, frame rate: 20, code rate 1500 kbps. 4: When quality is 4, the four parameters of width, height, frameRate and bitrate can be customized and modified. If not passed, the scene configuration value shall prevail. The default is the common scene value: resolution 640 * 480, frame rate: 15, code rate 800 kbps.

Camera orientation, "user" means the front camera, "environment" means the rear camera, the default is the device specified by the input parameter or the default device selected by the browser.

Wide resolution, effective when quality is 4.

High resolution, effective when quality is 4.

Frame rate, effective when quality is 4.

Video optimization mode, default to "default";

A value of "motion" indicates smooth mode. In most cases, the SDK will not reduce the frame rate, but may reduce the transmission resolution.

A value of "detail" indicates clear mode. In most cases, the SDK will not reduce the transmission resolution, but may reduce the frame rate.

The length of the video keyframe interval, in seconds.

Audio input device, the default device selected for the browser

Number of channels, 1 is mono, 2 is two channels, the default is mono.

Whether to turn on noise reduction or not will be determined based on the browser's capabilities. It is usually turned on by default.

Whether to turn on the automatic gain or not to decide whether to turn on the browser's ability, usually the default is to open.

Whether to turn on echo cancellation or not will be decided based on the browser's capabilities. It is usually turned on by default.

Whether to turn on noise reduction. It is turned off by default.

Whether to turn on the automatic gain. It is turned off by default.

Whether echo cancellation is enabled or not. It is enabled by default.

Generally speaking, the choice of screen sharing video quality should be determined according to the actual situation and scene. The quality in screen provides three preset values:

1: The frame rate is larger, which is suitable for scenes that require high fluency. 2: Suitable for scenes that strike a balance between fluency and clarity. 3: Large bit rate, suitable for scenes with high definition requirements. In addition to the above three preset values, quality also provides a custom value of 4, which can customize the frame rate and bitrate. The screen adds frameRate attributes. This attributes need to be passed to the SDK when used.

Video quality level, the default quality level is 2 1 Frame rate: 20 Code rate: 800K 2 Frame rate: 15 Code rate: 1500K 3 Frame rate: 5 Code rate: 2000K 4 Frame rate: frameRate

wide resolution

High resolution

frameRate

Video optimization mode, default to "default";

A value of "motion" indicates smooth mode. In most cases, the SDK will not reduce the frame rate, but may reduce the transmission resolution.

A value of "detail" indicates clear mode. In most cases, the SDK will not reduce the transmission resolution, but may reduce the frame rate

Specify the screen shared screen ID, only the Chrome kernel browser is used

The length of the video key frame interval, in seconds. Screen sharing can reduce the performance pressure of video encoding by increasing the interval of keyframes.

Screen sharing is supported, and videos only share DOM element areas.

Test whether the browser supports webRTC

Check whether the browser supports custom stream

Check whether the camera has permission to call

Check if the microphone has permission to call

Check whether screen sharing is supported

Check whether H264 encoding format is supported

Check whether VP8 encoding format is supported

Check if H265 encoding format is supported

User object instance, configure user ID and username. that the user ID set here needs to be consistent with the user ID set when logging into the room, otherwise the request for copyright music background service will fail.

Error code, please refer to Common error code document for details.

Song link list. If the style is downloaded, the first link is the style and the second is the original song.

Loading song or accompaniment progress callback.

Start scoring the song, and complete the callback for scoring each row of the song. You can use this callback to obtain the score information for this row. When calling the startScore interface, you need to pass the ZegoStartScoreParams.enableSongLineScoreCompleteEvent value as true to enable this event callback.

Error code, please refer to Common error code document for details.

Lyrics in lrc format, dedicated to progressive lyrics scene, please refer to https://doc-zh.zego.im/online-ktv-electron/client-api/apis-to-obtain-songs-and-lyrics for details.

Error code, please refer to Common error code document for details.

Lyrics in lrc format, dedicated to progressive lyrics scene, please refer to https://doc-zh.zego.im/online-ktv-electron/client-api/apis-to-obtain-songs-and-lyrics for details.

Error code, please refer to Common error code document for details.

标准音高线数据。详情请参考 Android：/online-ktv-android/zego-content-center/scoring-component, iOS：/online-ktv-ios/zego-content-center/scoring-component

Song ID.

Vendor.

billing mode.

Host ID, which must be passed when the billing mode is billed by host. Indicate according to which homeowner to order song/accompaniment/climax clip.

Scenario ID, indicating the actual business, please consult ZEGO technical support for details.

Error code, please refer to Common error code document for details.

The JSON returned by the song order service contains song resource information, please refer to https://doc-zh.zego.im/online-ktv-electron/client-api/apis-to-obtain-songs-and-lyrics#1_1 for details.

Error code, please refer to Common error code document for details.

result

Third-party source, <video> or <audio> media object or MediaStream

Video bitrate

Audio bit rate, the default is 48kbps

Number of channels, 1 is mono, 2 is two channels, the default is mono. If you turn on the microphone collection when pulling streams on the Safari browser, the stereo effect will be lost. You need to avoid turning on the microphone when pulling streams.

The starting bit rate, the default is "default", environmental requirements: Google kernel browser and SDK version 2.7.0 and above. A value of "default" means to use webrtc's default start bitrate of 300kbps and then slowly increase; a value of "target" means to quickly increase to the set target bitrate.

Video optimization mode, default to "default";

A value of "motion" indicates smooth mode. In most cases, the SDK will not reduce the frame rate, but may reduce the transmission resolution.

A value of "detail" indicates clear mode. In most cases, the SDK will not reduce the transmission resolution, but may reduce the frame rate

The minimum streaming bit rate. The minimum bit rate for acceptable video quality degradation when the network is poor.

The length of the video keyframe interval, in seconds.

Third-party source, <video> or <audio> media object or MediaStream

Number of channels, 1 is mono, 2 is two channels, the default is mono

The resource type selected for pulling the stream before starting to push the stream. 0 means pulling traffic only from RTC; 1 means pulling traffic only from CDN; 2 means pulling traffic only from L3.

The resource type selected for pull streaming during push streaming. 0 means pulling traffic only from RTC; 1 means pulling traffic only from CDN; 2 means pulling traffic only from L3.

The resource type selected for pulling after stopping pushing. 0 means pulling traffic only from RTC; 1 means pulling traffic only from CDN; 2 means pulling traffic only from L3.

Third-party source, <video> or <audio> media object or MediaStream

Video optimization mode, default to "default";

A value of "motion" indicates smooth mode. In most cases, the SDK will not reduce the frame rate, but may reduce the transmission resolution.

A value of "detail" indicates clear mode. In most cases, the SDK will not reduce the transmission resolution, but may reduce the frame rate

The length of the video keyframe interval, in seconds.

Real time data receiving update event.

Whitening intensity, the value range is [0,100], the default value is 50.

Ruddy intensity, the value range is [0,100], and the default value is 50.

Smooth intensity, the value range is [0,100], the default value is 50.

Sharpening intensity, the value range is [0,100], and the default value is 50.

id

screen name

Screen shot

Detailed description: Container for placing player rendering components. You can set the relevant attributes on the div tag of the specified container:

• zg-width, the width of the player component, optional. If only the width is set, the height of the rendered component will be scaled proportionally to the video ratio.
• zg-height, the height of the player component, optional. If only the height is set, the width of the rendered component will be scaled proportionally to the video ratio.

Playback mode currently only supports "live" mode, which is the live streaming mode.

Optional value, the type of engine instance passed into the SDK, optional "express" and "liveroom", if no value is passed, "express" is used by default, which means that the engine instance of express sdk is passed in for initialization. If the option "liveroom" is set, it means that the specified player plug-in is compatible with the engine instance of the liveroom sdk.

Optional value, specifying the decoding type that the player will prefer to use for decoding. The optional value is the string "WebCodecs", "wasm" or "MSE". If no value is passed, the player will default to using the "WebCodecs" type for decoding.

The "WebCodecs" decoding type indicates a preference for using the browser's WebCodecs API for hardware decoding, which results in faster decoding rates compared to software decoding. It is important to note that this feature is only supported on PC Chrome browsers and a limited number of Chrome browsers on certain Android devices. If run on unsupported browsers, the player will automatically fallback to using wasm software decoding for playback.

The "MSE" decoding type means that the browser's MSE (Media Source Extensions) API decoding is used first. The decoding speed will be faster than the soft decoding, and the browser compatibility is better than the WebCodecs solution. It supports mobile IOS. external browser. However, compared to WebCodecs decoding and wasm decoding, the player is limited by the MSE API solution, and the quality monitoring data collected will be relatively less.

The "wasm" decoding type signifies a preference for using software decoding, but with slightly lower decoding efficiency compared to WebCodecs decoding.

When the player decodes, the default priority order of "WebCodecs hard decoding > MSE hard decoding > wasm soft decoding" is followed for decoding and playback. If the browser does not support a certain decoding method or the decoding fails midway, it will automatically fall back to the next method.

error code

Detailed error description

The pop-up window for auto-play failure is enabled by default. When auto-play fails, the SDK will pop up the window to guide the user to click on the page to resume audio and video playback. Can be set to false to disable.

Used to relay signaling data packets between SDK and ZEGO backend. The format is $\{domain name\}:$\{port\}/$\{path\}. The port and path are filled in as needed according to the proxy configuration.

Set up a log reporting agent for troubleshooting. Please be sure to set this agent. The format is $\{domain name\}:$\{port\}/$\{path\}. The port and path are filled in as needed according to the proxy configuration.

Actively report to the log service, optional parameters. The format is $\{domain name\}:$\{port\}/$\{path\}. The port and path are filled in as needed according to the proxy configuration.

Camera microphone collection stream related configuration

Third-party stream collection related configuration

Event callback after calling [playVideo] or [playAudio] playback failure.player Indicates the audio or video DOM node that plays media.

Triggered when the user invokes [playVideo] to start playing video.

Triggered when the user invokes [playAudio] to start playing audio.

Event callback that failed to play after calling [playCaptureVideo] or [playCaptureAudio]. player Indicates the audio or video DOM node that plays media.

After the user calls [playCaptureVideo], it can be triggered when the video can start playing.

Triggered when the user invokes [playCaptureAudio] to start playing audio.

Detailed description: The local streaming view track update callback. The videoTrackChanged callback gets the visual track after pre-processing, while captureVideoTrackChanged gets the acquisition visual track before pre-processing.

If there is no need for special processing of audio and video rendering, it is not recommended to use this callback. It is recommended to use ZegoStreamView for rendering in most scenarios.

Detailed description: Local streaming track update callback. The audioTrackChanged callback gets the audio tracks after pre-processing, while captureAudioTrackChanged gets the audio track before pre-processing.

If there is no need for special processing of audio and video rendering, it is not recommended to use this callback. It is recommended to use ZegoStreamView for rendering in most scenarios.

Detailed description: Local stream acquisition visual track update callback. The videoTrackChanged callback gets the visual track after pre-processing, while captureVideoTrackChanged gets the acquisition visual track before pre-processing. If there is no need for special processing of audio and video rendering, it is not recommended to use this callback. It is recommended to use ZegoStreamView for rendering in most scenarios.

Detailed description: Local stream acquisition track update callback. The audioTrackChanged callback gets the audio tracks after pre-processing, while captureAudioTrackChanged gets the audio track before pre-processing. If there is no need for special processing of audio and video rendering, it is not recommended to use this callback. It is recommended to use ZegoStreamView for rendering in most scenarios.

Mirror display, true is on, false is off, the default is off.

Video screen display mode, "cover" is to cover the container; "contain" is to display all contents; "fill" is to stretch and fill.

The pop-up window for auto-play failure is enabled by default. When auto-play fails, the SDK will pop up the window to guide the user to click on the page to resume audio and video playback. Can be set to false to disable.

Whether to remove the last frame when the video data is interrupted. true to clear, false to keep, the default is true.

Local log level, the higher the level, the fewer logs will be printed

Number of log files. The default is 3. The range of values is [3, 20].

Used to customize the maximum size of log files, with a default value of 5MB (5 * 1024 * 1024 bytes) and a minimum size of 1MB (1 * 1024 * 1024 bytes). A value of 0 indicates that no logs need to be written.

Mixed stream background color; backgroundColor is hexadecimal RGB, and the input format must be 0xRRGGBB00. For example, pure white will pass 0xffffff00.

Mixed-flow background image; backgroundImage needs to be preset in the background image ID in advance, and the input format is preset-id://xxx

Mixed-stream video encoding,'vp8' (string) or 'h264' (string), default 'h264'(string)

User-defined data, the length does not exceed 1000 bytes. After setting, the SEI content can be obtained by listening to the callback of [onPlayerRecvSEI]. that it must be created using the [ByteBuffer.allocateDirect] function, otherwise the data cannot be passed to the SDK

Mixed-flow task id (customer-defined, must be unique), required, maximum 256 characters, only supports numbers, English characters and'~','!','@','#','$', ' ','^','&','*','(',')','_','+','=','-',',';',''',', ', '

Muxed input stream list

Muxing output stream list

Mixed flow output configuration

Mix stream watermark

Enables the mixed flow sound wave callback notification, which is used to obtain the sound wave of each audio input in the mixed flow. When postmixing is enabled, the sound information of each single stream can be received through the [mixerSoundLevelUpdate] callback. Mini program streaming does not currently support the [mixerSoundLevelUpdate] callback.

Stream alignment mode, 0 means off, 1 means on.

Mixed whiteboard input info

Input stream ID

Mixed stream content type; contentType value is'VIDEO' (audio and video),'AUDIO' (pure audio), "VIDEO_ONLY"(pure video),the default is'VIDEO'

The layout of the stream on the output canvas. When the contentType is "AUDIO", the layout parameter can not be passed.

Render mode, 0 for cropping mode and 1 for scaling mode.

User image information.

Text watermark.

Video frame corner radius, in px.

Mixer sound wave ID, used to find the sound wave value corresponding to the input stream in [mixerSoundLevelUpdate].

Input stream volume, valid range [0, 200], default is 100.

Whether the current input stream has focus voice enabled. If enabled, the sound of this stream will be highlighted. Default value is false.

Target location, up

Target location, left

Target location, down

Target location, right

Mixed output stream ID or URL

Mixed stream output video settings.

Mixed stream output video bit rate, kbps Value (Required and greater than 0)

Mixed stream output video frame rate

Mixed stream output video resolution width

Mixed stream output video resolution width...

Mixed stream output audio encoding outputAudioCodecID optional 0: HE-AAC, 1: AAC-LC, 2: MP3, 3: OPULS; default is 0. If you use CDN to record, please select AAC-LC for audio encoding. This is because some browsers (such as Google Chrome and Microsoft Edge) are not compatible with the HE-AAC audio encoding format, causing the recorded files to fail to play.

Mixed stream output audio bit rate, kbps

Number of mixed stream output channels

Multi-channel audio stream mixing mode. If [ZegoAudioMixMode] is set to [Focused], the SDK will select 4 input streams with [isAudioFocus] set as the focused voice. If no or less than 4 input streams are selected, the SDK will automatically make up to 4.

Audio bitrate, in kbps, the default is 48 kbps, and cannot be modified after starting the mixing task

Audio channel, default is Mono

Audio Codec ID

The image path, if not empty, the image will be displayed, otherwise, the video will be displayed. JPG and PNG formats are supported. There are 2 ways to use it: 1. URI: Provide the picture to ZEGO technical support for configuration. After the configuration is complete, the picture URI will be provided, for example: preset-id://xxx.jpg. 2. URL: Only HTTP protocol is supported.

Image display mode. 0: Default value. When the url is not empty, the video content is overwritten and the image is displayed. 1: According to the status of the camera, determine whether to display the image. The camera is turned off and the image is displayed. The camera turns on and displays the video content (without manually emptying the url parameters). 2: Judge whether to display image according to whether the input stream is empty. When the input stream was empty for 3 seconds, the image was displayed. The default time for judging empty flow is 3 seconds. If you need additional configuration, please contact ZEGO technical support. When the input stream has video data, the video content is displayed.

Mixed stream output video encoding format, supports H.264, H.265 encoding

The bitrate of the mixed output video, in kbps. The default value is the bitrate configured in [ZegoMixStreamOutputConfig].

The encoding profile of the mixed stream output video. The default value is [ZegoEncodeProfileDefault].

The encoding delay of the mixed stream output video, the valid value range is [0, 2000], the unit is milliseconds. The default value is 0.

Enable high definition low code, the default value is false

room scenario

Audio bit rate

Audio encoding format "opus"

Network jitter

volume

Number of lost packets

Packet loss rate

Stream audio quality，- 1 for unknown, 0 for excellent, 1 for good, 2 for medium, 3 for poor, 4 for very poor

Sampling Rate

Whether the audio track is turned off

Audio frame rate

Audio transmission energy

Audio encoding format

Detail description: The cumulative audio decoding time, in milliseconds (supported by version 2.19.0 and above).

Detail description: The cumulative audio freeze duration, in milliseconds (supported by version 2.19.0 and above).

Detail description: The cumulative audio freezing rate, the unit is 1, and the value range is [0,1] (supported by version 2.19.0 and above).

Video related description

Audio related description

Peer to peer delay in ms.

Peer to peer packet loss rate, unit 1. (2.13.0 and earlier versions are in %)

The name of the concrete implementation encoder.

The number of seconds it takes for data to be sent by this peer to the remote peer and back over the connection described by this pair of ICE candidates.

Available bandwidth for sending video data (Field unique to Google Chrome and obsolete in new versions).

The value indicating how many times the receiver sent a NACK packet to the sender after detecting that one or more packets were lost during transport.

The value indicating the number of times a PLI packet was sent by the stream's receiver to the sender

Sample time window.

This value indicates the total number of seconds which have elapsed between sending out STUN connectivity and consent check requests and receiving their responses.

Receive video high

Receive video width

Video encoding format

Whether the video track is turned off

Video bitrate

Video decoding frame rate

Video packet loss

Video loss rate

Video receiving frame rate

Video quality， 1 for unknown, 0 for excellent, 1 for good, 2 for medium, 3 for poor, 4 for very poor

Total video decoding size

The current actual received video frame loss

Cumulative video decoding time, in milliseconds (supported by version 2.19.0 and above).

Detail description: Cumulative video freeze duration, in milliseconds (supported by version 2.19.0 and above)

Detail description: Cumulative video freeze rate, unit is percentage, 0.0 ~ 1.0 (supported by version 2.9.0 and above)

Stream ID

NO_PUBLISH: not playing state, PLAY_REQUESTING: requesting play state, PLAYING: playing state

error code

Extended Information

domain name. Under cloud proxy: Proxy domain name provided by ZEGO.

Proxy service port number.

Audio bit rate

Audio encoding format

Audio packet loss

Audio packet loss rate

Audio encoding format

Whether the audio track is turned off

Audio frame rate

audio quality, - 1 for unknown, 0 for excellent, 1 for good, 2 for medium, 3 for poor, 4 for very poor

Push streaming video data

Push streaming audio data

The name of the concrete implementation encoder.

The number of seconds it takes for data to be sent by this peer to the remote peer and back over the connection described by this pair of ICE candidates.

The actual output bit rate of the video encoder(Field unique to Google Chrome and obsolete in new versions).

Available bandwidth for sending video data(Field unique to Google Chrome and obsolete in new versions).

Whether to reduce the video resolution sent due to limited bandwidth(Field unique to Google Chrome and obsolete in new versions).

Whether to reduce the sent video resolution due to insufficient CPU (Field unique to Google Chrome and obsolete in new versions).

Set height of sender resolution (Field unique to Google Chrome and obsolete in new versions).

The initial frame rate set by the sender (Field unique to Google Chrome and obsolete in new versions).

Set width of sender resolution (Field unique to Google Chrome and obsolete in new versions).

Target bitrate for video encoding (Field unique to Google Chrome and obsolete in new versions).

Whether to mute.

The value indicating how many times the receiver sent a NACK packet to the sender after detecting that one or more packets were lost during transport.

Pause or not

The value indicating the number of times a PLI packet was sent by the stream's receiver to the sender

Reasons for limiting flow quality.

ID of the output audio device

This value indicates the total number of seconds which have elapsed between sending out STUN connectivity and consent check requests and receiving their responses.

The value of volume.

Whether to turn on noise reduction

Whether to enable automatic gain

Whether to enable echo cancellation

wide resolution

High resolution

frameRate

max bitrate

video quality level, default level is 2

Capture video high

Capture video width

Video encoding format

Whether the video track is turned off

Video bitrate

Video encoding frame rate

Video packet loss

Video loss rate

Video sending frame rate

video quality, - 1 for unknown, 0 for excellent, 1 for good, 2 for medium, 3 for poor, 4 for very poor

Whether to enable hard coding, true is on, false is off, undefined is unknown.

Stream ID

NO_PUBLISH: not pushing state, PUBLISH_REQUESTING: requesting push state, PUBLISHING: pushing state

error code

Extended Information

Push stream quality callback

Push stream status callback

Screen sharing interrupt callback

Pull stream quality callback, each callback interval is 3s

Receive additional stream information set by the peer

Pull stream status callback

Trigger when the remote camera is changed

Trigger when the remote microphone state of the soil flow

Sound wave callback

Monitoring video device status change

Monitoring audio device status change

 abnormal monitoring device

Locally collected audio sound level callback

An abnormal error occurred in the beauty function.

Notifications of increased or decreased streams pushed by other users in the same room.

Listen to the notification of the exception or return to normal event of the acquisition encoding decoding of the push and pull stream.

The collection page is stopped or the shared page has no audio. Therefore, you are not advised to use the exceptionUpdate data as reference.

The network quality callback for the in-room user who is pushing streams

Sound update notification for each single stream in the a mixed stream.

Sound update notification for each single stream in the automatic mixed stream.

An abnormal error occurred in background process.

Experimental API callbacks

Mixed streaming and forwarding CDN status callback

Triggered when the viewport of the stream changes.

Triggered when the streamed audio track changes.

push stream encoding update callback.

Parameters

Details

Detailed description: Subscription push quality callback

Trigger condition: after successful push

Limit frequency: None

Follow callback: None

Important reminder: Call back every 3 seconds

Supported version: 1.0.0

Obsolete time: None

Parameters

Details

Detailed description: State notification of push streaming, push streaming is an asynchronous process, and the state switching in the middle is called back through this interface. Listen to the callback, collect error logs if the retry still fails, and remind the customer

Trigger condition: After starting to push

Limit frequency: None

Follow callback: None

Important reminder: Must listen before pushing

Supported version: 1.0.0

Obsolete time: None

Parameters

Details

Detailed description: Monitor the passive shutdown of the screen sharing stream caused by clicking the system stop button, or closing the capture process, etc.

Trigger condition: After screen sharing, click the system to stop or close the capture process, etc.

Limit frequency: None

Important reminder: Must listen before screen sharing

Supported version: 1.0.0

Obsolete time: None

Parameters

Details

Detailed description: The network quality callback of the user who is streaming in the room.

Trigger condition: The streaming quality starts updating.

Limit frequency: None

Important reminder: Need to bind before push pull flow.

Supported version: 2.24.0

Obsolete time: None

Parameters

Details

Detailed description: The network quality callback of the user who is streaming in the room.

scene: Mixed streaming is often used when multiple video frames need to be combined into one video, such as in education, live broadcasting of teachers and students.

Caoutions: This feature requires contacting technical support to enable the corresponding feature configuration on the mixed stream server.

Supported version: 3.0.0

Obsolete time: None

Parameters

Details

Detailed description: The network quality callback of the user who is streaming in the room.

scene: Mixed streaming is often used when multiple video frames need to be combined into one video, such as in education, live broadcasting of teachers and students.

Cautions: This function requires contacting ZEGO technical support to activate the corresponding function configuration of the mixed-stream server.

Supported version: 3.0.0

Obsolete time: None

Parameters

Details

Detailed description: monitor the changes of the stream in the logged-in room (add, delete)

Trigger condition: Add or delete stream in the room

Limit frequency: None

Important note: need to be called before logging in to the room

Supported version: 1.0.0

Obsolete time: None

Parameters

Details

Detailed description: Subscription pull quality callback

Trigger condition: after successful pull

Limit frequency: None

Follow callback: None