This page provides the release notes for the Agora Video SDK.
v3.7.1 was released on August 4, 2022.
1. Super resolution (beta)
To effectively boost the resolution of a remote user's video seen by the local user, this release adds support for the super resolution feature. If the original resolution of a remote user's video is a × b, the local user's device can render the remote video at a resolution of 2a × 2b after you enable this feature.
You can call enableRemoteSuperResolution
to enable this feature. The SDK triggers the onUserSuperResolutionEnabled
callback to report whether this feature has been successfully enabled. If it is not enabled successfully, use the reason
code to troubleshoot issues.
libagora_super_resolution_extension.dll
dynamical library. For other requirements, see the enableRemoteSuperResolution documentation.1. Co-hosting across channels
This release enhances the connection mechanism between the SDK and the server of co-hosting across channels and therefore reduces the failure rate.
This release fixed the following issues:
onRemoteAudioStateChanged(2,6)
callback after disconnecting from and reconnecting to a network even though the remote user had not changed the audio state.gatewayRtt
reported by the onRtcStats
callback was inaccurate when the network latency was too high or the router did not respond to ICMP packets.onNetworkQuality
callback was inaccurate for the user who was sharing a screen.Added
enableRemoteSuperResolution
VIRTUAL_BACKGROUND_SOURCE_STATE_REASON_INSUFFICIENT_PERFORMANCE(4)
in VIRTUAL_BACKGROUND_SOURCE_STATE_REASON
v3.7.0.2 was released on June 6, 2022.
This release fixed the issue where native clients saw video freeze on web clients when web clients published H.264 video streams.
v3.7.0 was released on April 11, 2022.
1. JND library integration change
As of v3.7.0, the JND library is statically compiled in the SDK by default and no longer provided as an extension library. If you have integrated the JND library, remove the JND extension library from your project dependencies and recompile your project when upgrading to v3.7.0. For the description of the JND library, see extension libraries.
2. Screen Sharing
You can use excludeWindowList
to block the specified windows during screen sharing. This release modifies the window blocking behavior on devices with multiple graphics cards as follows:
ERR_NOT_SUPPORTED_MUTI_GPU_EXCLUDE_WINDOW(1736)
to indicate that the window blocking feature is not supported on devices with multiple graphics cards.1. Spatial audio effect
This release adds the feature of spatial audio effect, which can add a sense of space to remote users' audio and simulate the audio transmission process in the real world. This enables the local user to hear remote users with the spatial audio effect.
To use this feature, contact support@agora.io.
2. Screen sharing
This release adds the setScreenCaptureScenario
method for setting the screen sharing scenario, including sharing documents or videos. The SDK can adjust the video quality and experience of screen sharing according to the scenario you set.
3. Local voice pitch
This release adds the enableLocalVoicePitchCallback
method and the onLocalVoicePitchInHz
callback, to allow the SDK to report the voice pitch of the local user at the set time interval to the app.
4. Reporting the first remote video frame rendered
This release adds the onFirstRemoteVideoFrame
callback to the IChannelEventHandler
class to report to the local user that the first video frame of the remote user has been rendered in a multichannel scenario. You can also get the remote user ID, the width and height (in pixels) of the video, and the time (in milliseconds) to render the first video frame from this callback.
5. Reporting the user role switch failure
This release adds the onClientRoleChangeFailed
callback to report the reason for a user role switch failure and the current user role in the interactive live streaming.
6. Reasons for connection state changes
To help users better understand the cause of connection state changes, this release adds the following enumerators in CONNECTION_CHANGED_REASON_TYPE
:
CONNECTION_CHANGED_SAME_UID_LOGIN(19)
: Join the same channel from different devices using the same user ID.CONNECTION_CHANGED_TOO_MANY_BROADCASTERS(20)
: The number of hosts in the channel reaches the upper limit. This enumerator is reported only after the support for 128 hosts is enabled.1. Channel capacity improvement
As of this release, a single channel can support up to 128 concurrent online hosts, who can publish audio and video streams at the same time. The number of audience members in a channel is unlimited. Each host or audience member can subscribe to a maximum of 50 hosts at the same time.
To experience this improvement, contact support@agora.io.
2. Virtual background
Test results and user feedback show that the effect of the virtual background feature improves when a user is against a complex background and adopts a more casual pose. As a result, virtual background is converted from a beta feature to an official feature as of v3.7.0.
3. A new version of noise suppression
Agora has added the option of using a new version of noise suppression to further improve audio quality. To experience this feature, contact support@agora.io
4. playEffect improvements
This release improves the internal implementation logic of playEffect
to avoid blocking and reduce freezing when playing an audio effect file.
5. Audio experience optimization
This release reduces audio stuttering under network conditions with burst jitter and high packet loss.
6. Transmission upgrade
This release upgrades transmission protocols and algorithms to enhance the SDK's ability to counter poor network conditions. At the same time, this release optimizes the scalable video coding capability for multi-user video scenarios to improve user video experience.
Added
setScreenCaptureScenario
enableLocalVoicePitchCallback
onLocalVoicePitchInHz
onFirstRemoteVideoFrame
in the IChannelEventHandler
classonClientRoleChangeFailed
CONNECTION_CHANGED_SAME_UID_LOGIN(19)
and CONNECTION_CHANGED_TOO_MANY_BROADCASTERS(20)
in CONNECTION_CHANGED_REASON_TYPE
ERR_NOT_SUPPORTED_MUTI_GPU_EXCLUDE_WINDOW(1736)
Deprecated
startScreenCaptureByScreenRect
. Use startScreenCaptureByDisplayId
instead. ERR_SET_CLIENT_ROLE_NOT_AUTHORIZED(119)
. Use CLIENT_ROLE_CHANGE_FAILED_REASON
instead.LOCAL_VIDEO_STREAM_ERROR_SCREEN_CAPTURE_WINDOW_NOT_SUPPORTED(20)
v3.6.2 was released on February 22, 2022.
Screen sharing
This release renames several fields reported by the onScreenCaptureInfoUpdated
callback:
cardType
is renamed to graphicsCardType
.FILT_WINDOW_ERROR
is renamed to EXCLUDE_WINDOW_ERROR
.FILT_WINDOW_ERROR_FAIL
is renamed to EXCLUDE_WINDOW_ERROR_FAIL
.FILT_WINDOW_ERROR_NONE
is renamed to EXCLUDE_WINDOW_ERROR_NONE
.Video enhancement
In addition to the image enhancement feature, Agora adds support for more video enhancement features:
setVideoDenoiserOptions
method to enable/disable video noise reduction and set the options of the video noise reduction effect.setLowLightEnhanceOptions
method to enable/disable low-light enhancement and set the options of the low-light enhancement effect.setColorEnhanceOptions
method to enable/disable color enhancement and set the options of the color enhancement effect.libagora_video_process_extension.dll
dynamical library is integrated.1. Cloud proxy
To enrich application scenarios of the Agora Cloud Proxy, this release updates the cloud proxy types (CLOUD_PROXY_TYPE
) as follows:
NONE_PROXY(0)
from "Do not use the cloud proxy" to "Automatic mode". In this mode, the SDK attempts a direct connection to SD-RTN™ and automatically switches to TLS 443 if the attempt fails. As of v3.6.2, the SDK has this mode enabled by default.TCP_PROXY(2)
, which is the TCP (encrypted) mode. In this mode, the SDK always transmits data over TLS 443.In addition, this release adds the onProxyConnected
callback to report the proxy connection state. For example, when a user calls setCloudProxy
and joins a channel successfully, the SDK triggers this callback to report the user ID, the proxy type connected, and the time elapsed from the user calling joinChannel
until this callback is triggered.
2. Audio device test
To support testing audio devices after joining a channel, this release improves the following methods and callbacks:
onAudioDeviceTestVolumeIndication
, which reports the volume information of the audio device being tested.startRecordingDeviceTest
, startPlaybackDeviceTest
, and startAudioDeviceLoopbackTest
. Before v3.6.2, you can call these methods only before joining a channel; as of v3.6.2, you can call these methods either before or after joining a channel. After a successful method call, the SDK triggers the following callbacks:onAudioVolumeIndication
and onAudioDeviceTestVolumeIndication
when you call the methods before joining a channel. Both callbacks report the same volume information. If you upgrade the SDK to v3.6.2 or later, Agora recommendeds using onAudioDeviceTestVolumeIndication
.onAudioDeviceTestVolumeIndication
when you call the methods after joining a channel.startRecordingDeviceTest
, startPlaybackDeviceTest
, or startAudioDeviceLoopbackTest
after joining a channel tests the audio devices that the SDK is using.3. Audio recording
This release extends startAudioRecording
with support for setting recording with dual channels and higher audio quality.
recordingChannel
to AudioRecordingConfiguration
, it enables you to set the recorded audio channel to be mono or dual. Because the actual recorded audio channel is related to the captured audio channel and the integration scheme affects the final recorded audio channel, contact technical support for assistance with stereo recording using startAudioRecording
.AUDIO_RECORDING_QUALITY_ULTRA_HIGH(3)
to AudioRecordingConfiguration.recordingQuality
, it enables you to set the recorded audio quality to ultra-high. Ultra-high quality is the highest quality available in recordingQuality
. When you record a 10-minute AAC audio file at a sample rate of 32,000 Hz and ultra-high quality, the file size is approximately 7.5 MB.4. Music file playback
This release optimizes the experience of calling startAudioMixing
to play music files as follows:
getAudioFileInfo
.If you want to experience these improvements, ensure that you have integrated the libagora-full-audio-format-extension.dll
dynamic library. Given the large size of this library, if you have limitations on app size and do not need these improvements, you can remove this dynamic library when integrating the SDK. For more instructions, see Reduce App Size.
This release fixed the following issues:
Added
setVideoDenoiserOptions
setLowLightEnhanceOptions
setColorEnhanceOptions
onAudioDeviceTestVolumeIndication
recordingChannel
in AudioRecordingConfiguration
AUDIO_RECORDING_QUALITY_ULTRA_HIGH(3)
in AudioRecordingConfiguration.recordingQuality
onProxyConnected
TCP_PROXY(2)
in CLOUD_PROXY_TYPE
Modified
onScreenCaptureInfoUpdated
callback. startRecordingDeviceTest
, startPlaybackDeviceTest
, and startAudioDeviceLoopbackTest
method calls. NONE_PROXY(0)
in CLOUD_PROXY_TYPE
.v3.6.1.1 was released on January 16, 2022.
This release fixed the following audio issue:
When an end user muted and unmuted audio from an application using the Agora Web SDK, end users using an application with Native SDK version 3.6.0 might not hear audio from this end user.
v3.6.1 was released on January 14, 2022.
In screen sharing scenarios, some users cannot use the excludeWindowList
attribute to block the specified window in a shared screen because their hardware does not support it. This release adds the onScreenCaptureInfoUpdated
callback, which is triggered when the window blocking fails and reports the error code of the failure (errCode
) and the user's graphics card type (cardType
).
This release provides the following improvements:
leaveChannel
method call, network interruption, or other reasons, the local user sees the remote view stay on the last video frame instead of the remote view directly disappearing as in previous versions.This release fixed the following issues:
updateScreenCaptureParameters
to update the screen-sharing configuration, windowFocus
did not work.v3.6.0 was released on December 7, 2021.
1. Media Push
To reduce the difficulty of integrating Media Push, this release optimizes the API design of Media Push and improves the handling of network issues within Media Push clients and servers. You can experience the optimized Media Push functionality with the following new methods:
startRtmpStreamWithoutTranscoding
: Starts pushing media streams to a CDN without transcoding. This method works the same as the old method addPublishStreamUrl(false)
.startRtmpStreamWithTranscoding
: Starts pushing media streams to a CDN and sets the transcoding configuration. This method works the same as calling the old methods setLiveTranscoding
and addPublishStreamUrl(true)
in sequence.updateRtmpTranscoding
: Updates the transcoding configuration. This method works the same as the non-first call to the old method setLiveTranscoding
.stopRtmpStream
: Stops pushing media streams to a CDN. This method works the same as the old method removePublishStreamUrl
.addPublishStreamUrl
, setLiveTranscoding
, and removePublishStreamUrl
. Agora recommends that you use the new methods for Media Push and update your code logic by referring to Media Push.Also, as of this release, whether you use the new or old methods for Media Push, you can experience the following improvements:
Reporting streaming states, errors, and events:
In the RTMP_STREAM_PUBLISH_STATE
state code, adding RTMP_STREAM_PUBLISH_STATE_DISCONNECTING(5)
: The SDK is disconnecting from the Agora streaming server and CDN. When you call remove
or stop
to stop the streaming normally, the SDK reports the streaming state as DISCONNECTING
, IDLE
in sequence.
Renaming RTMP_STREAM_PUBLISH_ERROR
to RTMP_STREAM_PUBLISH_ERROR_TYPE
and adding the following to the RTMP_STREAM_PUBLISH_ERROR_TYPE
error code:
RTMP_STREAM_PUBLISH_ERROR_NOT_BROADCASTER(11)
: The user role is not host, so the user cannot use the Media Push function. Check your application code logic.RTMP_STREAM_PUBLISH_ERROR_TRANSCODING_NO_MIX_STREAM(13)
: The update
or setLiveTranscoding
method is called to update the transcoding configuration in a scenario where there is streaming without transcoding. Check your application code logic.RTMP_STREAM_PUBLISH_ERROR_NET_DOWN(14)
: Errors occurred in the host's network.RTMP_STREAM_PUBLISH_ERROR_INVALID_APPID(15)
: Your App ID does not have permission to use the Media Push function. Refer to Prerequisites to enable the Media Push permission.Adding the following to the RTMP_STREAMING_EVENT
event code:
RTMP_STREAMING_EVENT_ADVANCED_FEATURE_NOT_SUPPORT(3)
: The feature is not supported.RTMP_STREAMING_EVENT_REQUEST_TOO_OFTEN(4)
: Reserved.Adding multiple watermarks and background images during streaming with transcoding: This is achieved by adding watermarkCount
and backgroundImageCount
to the LiveTranscoding
structure.
Setting the layer number and transparency of each watermark and background image during streaming with transcoding: This is achieved by adding zOrder
and alpha
to the RtcImage
structure.
Using the HE-AAC v2 audio codec during streaming with transcoding: This is achieved by adding AUDIO_CODEC_PROFILE_HE_AAC_V2(2)
to AUDIO_CODEC_PROFILE_TYPE
.
2. Following the system default audio device
To help developers more flexibly manage audio device switching, this release adds the following methods in the IAudioDeviceCollection
class:
getDefaultDevice
: Gets the default audio device of the system.followSystemPlaybackDevice
: Sets the audio playback device used by the SDK to follow the system default audio playback device.followSystemRecordingDevice
: Sets the audio recording device used by the SDK to follow the system default audio recording device.On Windows, after initialized, the SDK uses the system default audio device to play and capture audio by default. This release optimizes the default processing logic of the SDK when the system default audio device changes, as follows:
With versions earlier than v3.6.0, the SDK follows the system default audio device only when the following occurs:
A user connects a new audio device, and the system default audio device changes to the newly connected audio device.
A user disconnects or disables an audio device, and that audio device is being used by the SDK.
With v3.6.0 and later versions, the SDK always follows the system default audio device. If you do not need the SDK to follow the system default audio device, call followSystemPlaybackDevice(false)
and followSystemRecordingDevice(false)
.
1. Image enhancement
This release updates the Agora image enhancement algorithm to improve the image enhancement effects that you can enable by calling setBeautyEffectOptions
. To balance these image enhancement effects, this release changes the default value of lighteningLevel
from 0.7 to 0.6 and adds sharpnessLevel
to the BeautyOptions
structure to support setting the sharpness level.
libagora_video_process_extension.dll
dynamic library before calling setBeautyEffectOptions
; otherwise, you cannot experience the optimized image enhancement effects and sharpness adjustment brought out by the updated Agora image enhancement algorithm, but only the image enhancement effects brought out by the older Agora image enhancement algorithm.2. Rendering of the first remote video frame
This release speeds up rendering of the first remote video frame under poor network conditions to enhance user video experience.
This release fixed the following issues:
enableLoopbackRecording
to enable loopback audio capturing on some devices, remote users heard the audio being played back twice.Added
startRtmpStreamWithoutTranscoding
startRtmpStreamWithTranscoding
updateRtmpTranscoding
stopRtmpStream
RTMP_STREAM_PUBLISH_STATE_DISCONNECTING(5)
in RTMP_STREAM_PUBLISH_STATE
RTMP_STREAM_PUBLISH_ERROR_TYPE
:RTMP_STREAM_PUBLISH_ERROR_NOT_BROADCASTER(11)
RTMP_STREAM_PUBLISH_ERROR_TRANSCODING_NO_MIX_STREAM(13)
RTMP_STREAM_PUBLISH_ERROR_NET_DOWN(14)
RTMP_STREAM_PUBLISH_ERROR_INVALID_APPID(15)
RTMP_STREAMING_EVENT
:RTMP_STREAMING_EVENT_ADVANCED_FEATURE_NOT_SUPPORT(3)
RTMP_STREAMING_EVENT_REQUEST_TOO_OFTEN(4)
watermarkCount
and backgroundImageCount
in the LiveTranscoding
structurezOrder
and alpha
in the RtcImage
structureAUDIO_CODEC_PROFILE_HE_AAC_V2(2)
in AUDIO_CODEC_PROFILE_TYPE
IAudioDeviceCollection
class:getDefaultDevice
followSystemPlaybackDevice
followSystemRecordingDevice
sharpnessLevel
in the BeautyOptions
structureModified
RTMP_STREAM_PUBLISH_ERROR
to RTMP_STREAM_PUBLISH_ERROR_TYPE
lighteningLevel
from 0.7 to 0.6v3.5.2 was released on November 25, 2021.
Error codes returned when the request to join a channel is rejected
To more accurately report the reason for the failure to join a channel, as of this release, the SDK returns the error code -17 (ERR_JOIN_CHANNEL_REJECTED)
in the return value of the joining channel method in the following situations:
IRtcEngine
channel calls the joining channel method of the IRtcEngine
class with a valid channel name.IChannel
channel calls the joining channel method of this IChannel
object.In the SDK earlier than v3.5.2, when the above errors occur, the SDK might report the error code 17
through the onError
callback, or return the error code -5 (ERR_REFUSED)
in the return value of the joining channel method.
1. Audio and video recording on the client
To enable users to record local audio and video on their client, this release adds the IMediaRecorder
class, which supports recording the following content:
The generated recording files are saved in the local path specified by the user.
After successfully initializing the IRtcEngine
object, you can call the following methods and callbacks in the IMediaRecorder
class to implement the local audio and video recording function:
getMediaRecorder
: Registers IMediaRecorderObserver
and gets the IMediaRecorder
object.startRecording
: Sets the recording content, duration, format of generated files, and callback interval, and enables the local audio and video recording.stopRecording
: Stops recording the local audio and video.releaseRecorder
: Releases the IMediaRecorder
object.onRecorderStateChanged
: Occurs when the recording state changes.onRecorderInfoUpdated
: Occurs when the recording information is updated.COMMUNICATION
channel profile, this function is incompatible with versions of the SDK earlier than v3.0.0.2. Video snapshot
This release adds the takeSnapshot
method for taking a snapshot of a video stream from the specified user, generating a JPG image, and saving it to the specified path. After a successful method call, the SDK triggers the onSnapshotTaken
callback to report whether the snapshot is successfully taken as well as the details of the snapshot taken.
3. Testing audio and video call loop
To check whether the local audio device, video device, and network conditions can guarantee the proper sending and receiving of audio and video streams, this release adds the startEchoTest
[3/3] method. You can call this method before joining a channel to test whether the loop of a user's audio and video devices and network conditions are working properly.
Compared to startEchoTest
[2/3], startEchoTest
[3/3] can test video devices and secure the test, but cannot support setting the time interval for reporting test results.
1. Screen sharing
This release adds the getScreenCaptureSources
method for you to get a list of all the shareable screens and windows. Users can visually select a particular screen or window to share by using thumbnails in the IScreenCaptureSourceList
list. In IScreenCaptureSourceList
, you can also use getSourceInfo
to get detailed information about the shared target, including the type of the shared target (window or screen), the window ID or screen ID of the shared target, and the thumbnail and icon of the shared target.
Previously, you needed to call startScreenCaptureByScreenRect
to share a screen by the Rectangle
coordinates of the display on the virtual screen. As of v3.5.2, you can also call startScreenCaptureByDisplayId
to start screen sharing by passing in a display ID.
startScreenCaptureByDisplayId
to start sharing instead of using startScreenCaptureByScreenRect
.2. Local video errors
To facilitate troubleshooting, this release adds the following error codes into LOCAL_VIDEO_STREAM_ERROR
:
LOCAL_VIDEO_STREAM_ERROR_DEVICE_INVALID_ID(10)
: The SDK cannot find the video device in the video device list. Check whether the ID of the video device is valid.LOCAL_VIDEO_STREAM_ERROR_SCREEN_CAPTURE_WINDOW_OCCLUDED(13)
: The window being shared is overlapped by another window, so the overlapped area is blacked out by the SDK during window sharing.LOCAL_VIDEO_STREAM_ERROR_SCREEN_CAPTURE_WINDOW_NOT_SUPPORTED(20)
: The SDK does not support sharing this type of window.3. Performance improvements
This release slightly reduces CPU usage in video scenarios by optimizing the FEC (Forward Error Correction) codec.
4. Experience improvements
This release improves video sharpness and smoothness under poor network conditions by optimizing the congestion control algorithm.
This release fixed the following issues:
startAudioMixing
on some Android devices.enableDualStreamMode(false)
, a dual stream (high-quality and low-quality video stream) was nevertheless sent when the user switched from custom video capture using setVideoSource
with the custom video source in the SCREEN
type to SDK capture.This release adds the following APIs:
startEchoTest
[3/3]getScreenCaptureSources
startScreenCaptureByDisplayId
getMediaRecorder
onRecorderStateChanged
onRecorderInfoUpdated
startRecording
stopRecording
releaseRecorder
takeSnapshot
onSnapshotTaken
LOCAL_VIDEO_STREAM_ERROR
:LOCAL_VIDEO_STREAM_ERROR_DEVICE_INVALID_ID(10)
LOCAL_VIDEO_STREAM_ERROR_SCREEN_CAPTURE_WINDOW_OCCLUDED(13)
LOCAL_VIDEO_STREAM_ERROR_SCREEN_CAPTURE_WINDOW_NOT_SUPPORTED(20)
v3.5.1 was released on October 14, 2021.
1. Background blurring
To enrich the virtual background effects, this release adds support for setting background blur when a user enables the virtual background function by calling enableVirtualBackground
. You need to set the type of the custom background image as BACKGROUND_BLUR
and set blur_degree
according to the degree of blurring that the user desires.
2. Pausing and resuming the media stream relay across channels
In order to enable the host to quickly pause or resume a cross-channel media stream relay, this release adds the following methods and event code:
pauseAllChannelMediaRelay
: Pauses the media stream relay to all destination channels.resumeAllChannelMediaRelay
: Resumes the media stream relay to all destination channels.RELAY_EVENT_PAUSE_SEND_PACKET_TO_DEST_CHANNEL_SUCCESS(12)
: The SDK successfully pauses relaying the media stream to destination channels.RELAY_EVENT_PAUSE_SEND_PACKET_TO_DEST_CHANNEL_FAILED(13)
: The SDK fails to pause relaying the media stream to destination channels.RELAY_EVENT_RESUME_SEND_PACKET_TO_DEST_CHANNEL_SUCCESS(14)
: The SDK successfully resumes relaying the media stream to destination channels.RELAY_EVENT_RESUME_SEND_PACKET_TO_DEST_CHANNEL_FAILED(15)
: The SDK fails to resume relaying the media stream to destination channels.After a successful method call of pauseAllChannelMediaRelay
or resumeAllChannelMediaRelay
, the SDK triggers the onChannelMediaRelayEvent
callback to report whether the media stream relay is successfully paused or resumed.
3. Pushing the external audio frame to a specified position
To meet the different processing requirements of external audio frames in different scenarios, this release deprecates pushAudioFrame
[2/3] and adds pushAudioFrame
[3/3] with sourcePos
instead. You can push the external audio frame to one of three positions: after audio capture, before audio encoding, or before local playback. For example, in the KTV scenario, you can push the singing voice to after audio capture, and push the accompaniment to before audio encoding, so that the singing voice is processed by the SDK audio module, but the accompaniment is not affected by the SDK audio module.
This release also adds setExternalAudioSourceVolume
, which enables you to set the volume of external audio frames at a specified position.
4. Advanced settings of the music file
To set the playback speed, audio track, and channel mode of a music file, this release adds the following methods:
setAudioMixingPlaybackSpeed
: Sets the playback speed of the current music file. Agora recommends a value range of [50,400], where 100 represents the original speed.getAudioTrackCount
: Gets the number of audio tracks of the current music file.selectAudioTrack
: Specifies the playback audio track of the current music file. The range of index is [0, getAudioTrackCount()).setAudioMixingDualMonoMode
: Sets the channel mode of the current music file to original mode, left channel mode, right channel mode, or mixed channel mode.5. Getting audio file information
To get the information of any audio file, this release deprecates getAudioMixingDuration
and adds getAudioFileInfo
instead. After joining a channel, you can call getAudioFileInfo
and get information such as duration for the specified audio file in onRequestAudioFileInfo
.
1. Audio device errors
This release adds the following error codes in LOCAL_AUDIO_STREAM_ERROR
:
LOCAL_AUDIO_STREAM_ERROR_RECORD_INVALID_ID(9)
, which indicates that the ID of the audio capture device is invalid.LOCAL_AUDIO_STREAM_ERROR_PLAYOUT_INVALID_ID(10)
, which indicates that the ID of the audio playback device is invalid.2. Identification and quality testing for 5G mobile networks
This release adds identification and connection quality testing for 5G mobile networks:
NETWORK_TYPE_MOBILE_5G(6)
to NETWORK_TYPE
. When the local network changes to 5G, the SDK triggers the onNetworkTypeChanged
callback to report this network connection type.enableLastmileTest
or startLastmileProbeTest
to test the connection quality of the 5G mobile network.3. Other improvements
This release also provides the following improvements:
This release has fixed the following issues:
GAME_STREAMING
scenario, occasional echo or noise that was caused by inaccurate music detection occurred.enableLocalVideo(false)
, enableLocalVideo(true)
to disable and enable local video capture or called muteLocalVideoStream(true)
, muteLocalVideoStream(false)
to stop and resume sending local video in sequence within a few seconds.muteAllRemoteAudioStreams
and received onRemoteAudioStateChanged(REMOTE_AUDIO_REASON_LOCAL_MUTED)
, onRemoteAudioStateChanged(REMOTE_AUDIO_REASON_REMOTE_MUTED)
was incorrectly received after 15 seconds.onAudioDeviceStateChanged
callback when the user connected or disconnected an audio device.joinChannel
multiple times in a row.setExternalAudioSource
before joining the channel did not take effect.onLocalAudioStateChanged
callback to report LOCAL_AUDIO_STREAM_STATE_CAPTURING(1)
when you called enableLocalAudio(false)
after joining a channel.Added
blur_degree
and BACKGROUND_BLUR_DEGREE
in VirtualBackgroundSource
BACKGROUND_BLUR
in BACKGROUND_SOURCE_TYPE
pushAudioFrame
[3/3]setExternalAudioSourceVolume
setAudioMixingPlaybackSpeed
getAudioTrackCount
selectAudioTrack
setAudioMixingDualMonoMode
pauseAllChannelMediaRelay
resumeAllChannelMediaRelay
CHANNEL_MEDIA_RELAY_EVENT
:RELAY_EVENT_PAUSE_SEND_PACKET_TO_DEST_CHANNEL_SUCCESS(12)
RELAY_EVENT_PAUSE_SEND_PACKET_TO_DEST_CHANNEL_FAILED(13)
RELAY_EVENT_RESUME_SEND_PACKET_TO_DEST_CHANNEL_SUCCESS(14)
RELAY_EVENT_RESUME_SEND_PACKET_TO_DEST_CHANNEL_FAILED(15)
getAudioFileInfo
onRequestAudioFileInfo
LOCAL_AUDIO_STREAM_ERROR_RECORD_INVALID_ID(9)
and LOCAL_AUDIO_STREAM_ERROR_PLAYOUT_INVALID_ID(10)
in LOCAL_AUDIO_STREAM_ERROR
NETWORK_TYPE_MOBILE_5G(6)
in NETWORK_TYPE
Deprecated
pushAudioFrame
[2/3]getAudioMixingDuration
v3.5.0.4 was released on September 26, 2021.
This release fixed the following issues:
startAudioDeviceLoopbackTest
to start the audio device loopback test after joining and then immediately leaving a channel, the content of the AudioVolumeInfo
array reported by the onAudioVolumeIndication
callback was incorrect.onRenderVideoFrame
and onRenderVideoFrameEx
callbacks when capturing a local video frame to be rendered.onCaptureVideoFrame
callback and caused video freezes for remote users.getObservedFramePosition
as POSITION_PRE_ENCODER (1 << 2)
, the SDK incorrectly triggered a onCaptureVideoFrame
callback that did not match the set observation position.v3.5.0.3 was released on September 6, 2021 and improved the stability of the SDK.
v3.5.0.2 was released on August 11, 2021.
This release fixed black screen issues that occurred when setupLocalVideo
or setupRemoteVideo
was called under certain circumstances, such as the following:
setupLocalVideo(null)
or setupRemoteVideo(null)
was called to disable the local or remote video view, and then setupLocalVideo
or setupRemoteVideo
was called to initialize the local or remote video view.setupRemoteVideo
was called in the onRemoteVideoStateChanged
callback to initialize the remote video view.v3.5.0.1 was released on August 4, 2021. This release fixed the issue where the user could not receive the onRenderVideoFrame
callback from the registered raw video data observer when the setupLocalVideo
and setupRemoteVideo
methods were not called.
v3.5.0 was released on July 20, 2021.
Custom video renderer
During a real-time audio and video interaction, the SDK enables the default renderer to render video. Starting from this release, Agora adds support for custom video rendering via the MediaIO method. If you want to customize the video rendering, you can use the IVideoSink
class to customize the video renderer, and then call setLocalVideoRenderer
or setRemoteVideoRenderer
to render the local or remote video.
1. Video quality
The following improvements have been made to the quality of captured and received video in this release:
2. Virtual background (beta)
This release optimizes the portrait edge effect of the virtual background.
3. Audio error code
To help troubleshoot audio issues, this release adds the ERR_ADM_WIN_CORE_SERVRE_SHUT_DOWN (1735)
error code, which reports that the Windows Audio service is disabled. You need to either enable the Windows Audio service or restart the device.
4. Additional improvements
This release also improves the following:
This release fixed the following issues:
enableLoopbackRecording
, occasionally the local user heard an echo of their own audio.muteLocalAudioStream(true)
to stop publishing a local audio stream, the local user could hear remote users when the local user joined a channel for the first time. After leaving the channel and joining another channel, occasionally the local user could not hear remote users.rxQuality
) of remote users reported by the SDK was not accurate.This release adds the following APIs:
setLocalVideoRenderer
setRemoteVideoRenderer
ERR_ADM_WIN_CORE_SERVRE_SHUT_DOWN (1735)
v3.4.6 was released on July 15, 2021 and improved the stability of the SDK.
v3.4.5 was released on June 22, 2021.
1. Support for GCM2 encryption
To further improve the security during real-time audio and video transmission process, this release adds the following:
AES_128_GCM2 (7)
and AES_256_GCM2 (8)
encryption modes in ENCRYPTION_MODE
. The new GCM encryption modes use a more secure KDF (Key Derivation Function) and support setting the key and salt.encryptionKdfSalt
member in EncryptionConfig
to add the salt for the AES_128_GCM2 (7)
and AES_256_GCM2 (8)
encryption modes.This release also changes the default encryption mode from AES_128_XTS (1)
to AES_128_GCM2 (7)
. If you use the default encryption mode in earlier versions, after upgrading the SDK to v3.4.5, ensure that you call enableEncryption
and set ENCRYPTION_MODE
to AES_128_XTS (1)
.
2. Media stream publishing behavior changes
To flexibly control the publishing state in multiple channels, this release optimizes the IChannel
class as follows:
publish
and unpublish
methods, and adds muteLocalAudioStream
and muteLocalVideoStream
instead. You can set the publishing state of the audio stream and video stream separately.publishLocalAudio
and publishLocalVideo
members in ChannelMediaOptions
. The default value is true
. You can set the publishing state when joining a channel. If a user publishes streams in a channel, regardless of whether the user is a host or an audience member, you need to set publishLocalAudio
and publishLocalVideo
to false when the user joins other channels; otherwise, the user fails to join the channel.setClientRole(BROADCASTER)
, the local user publishes audio and video streams by default. You no longer need to call publish
.publishLocalAudio = false
serves the same function as muteLocalAudioStream(true)
.publishLocalVideo = false
serves the same function as muteLocalVideoStream(true)
.The above improvements bring the following changes in the IRtcEngine
class:
muteLocalAudioStream
and muteLocalVideoStream
of IRtcEngine
do not take effect for channels created by the IChannel
class, so you need to use the muteLocalAudioStream
and muteLocalVideoStream
of the IChannel
class instead.joinChannel
with the options
parameter, you can set the publishing state.joinChannel
with the options
parameter, muteLocalAudioStream
or muteLocalVideoStream
only takes effect when it is called after joinChannel
.If you upgrade the SDK to v3.4.5 or later, to avoid affecting your service, Agora recommends modifying the implementation of muteLocalAudioStream
, muteLocalVideoStream
, publish
, and unpublish
.
See Set the Publishing State and Join Multiple Channels.
3. startAudioMixing changes
To avoid blocking, this release changes the startAudioMixing
from a synchronous call to an asynchronous call.
1. Virtual background (beta)
This release adds the enableVirtualBackground
method, with which you can enable and set a virtual background for users as either a solid color or an image in PNG or JPG format. You can use the onVirtualBackgroundSourceEnabled
callback to learn whether the virtual background is successfully enabled or the reason for an error.
2. Local encoded video data
This release adds the registerVideoEncodedFrameObserver
method, which you can use to register the local encoded video data observer. You can get the local encoded video data from the onVideoEncodedFrame
callback function to implement custom functions, such as recording video, taking screenshots, and image enhancement.
1. Performance optimization
Agora has optimized internal threads of the SDK, which reduces CPU usage by about 20% on low-end devices.
2. Media Push
To be more transparent to users about errors and events in Media Push, this release adds the following codes:
onRtmpStreamingStateChanged
: Error code RTMP_STREAM_UNPUBLISH_ERROR_OK (100)
, which reports that the streaming has been stopped normally. After you call removePublishStreamUrl
to stop streaming, the SDK returns this error code and state code RTMP_STREAM_PUBLISH_STATE_IDLE (0)
.onRtmpStreamingEvent
: Event code RTMP_STREAMING_EVENT_URL_ALREADY_IN_USE (2)
, which reports that the streaming URL is already being used for Media Push. If you want to start new streaming, use a new streaming URL.3. Custom video source
You can use the setVideoSource
method to customize a video source, that is, use the MediaIO method for custom video capture. To enhance the flexibility of video capture, as of this release, Agora supports switching from custom video capture by MediaIO to SDK capture directly in the channel. See How can I switch from custom video capture to SDK capture.
4. Media device state
To report media device states more accurately, this release optimizes the enumerations of MEDIA_DEVICE_STATE_TYPE
as follows:
MEDIA_DEVICE_STATE_ACTIVE (1)
from "The device is ready for use" to "The device is in use", which is more consistent with its name.MEDIA_DEVICE_STATE_IDLE (0)
, which reports the device is ready for use.MEDIA_DEVICE_STATE_TYPE
is reported in the onAudioDeviceStateChanged
and onVideoDeviceStateChanged
callbacks.
5. Music file state
When you call startAudioMixing
after pauseAudioMixing
, this release adds the onAudioMixingStateChanged(AUDIO_MIXING_STATE_STOPPED, AUDIO_MIXING_REASON_STOPPED_BY_USER)
state, which indicates the music file is stopped, before reporting onAudioMixingStateChanged(AUDIO_MIXING_STATE_PLAYING, AUDIO_MIXING_REASON_STARTED_BY_USER)
.
This release fixed the following issues:
startAudioMixing
to play an online music file, occasionally the SDK response time was too long and caused a freeze.AUDIO_SCENARIO_CHATROOM_GAMING
, after the local user played a video through the system media player, started screen sharing, and used a sound card to sample audio streams, the video sound heard by remote users contained noise.Added
enableVirtualBackground
onVirtualBackgroundSourceEnabled
RTMP_STREAM_UNPUBLISH_ERROR_OK (100)
in RTMP_STREAM_PUBLISH_ERROR
RTMP_STREAMING_EVENT_URL_ALREADY_IN_USE (2)
in RTMP_STREAMING_EVENT
AES_128_GCM2 (7)
and AES_256_GCM2 (8)
in ENCRYPTION_MODE
encryptionKdfSalt
in EncryptionConfig
muteLocalAudioStream
and muteLocalVideoStream
in IChannel
publishLocalAudio
and publishLocalVideo
in ChannelMediaOptions
MEDIA_DEVICE_STATE_IDLE (0)
in MEDIA_DEVICE_STATE_TYPE
registerVideoEncodedFrameObserver
onVideoEncodedFrame
Modified
Deprecated
publish
and unpublish
in IChannel
v3.4.3 was released on June 16, 2021. This release fixed the following issues:
startAudioMixing
method could not play the MP3 local music files with the .m4a
filename extension.v3.4.2 was released on May 12, 2021. This release fixed the following issues:
COMMUNICATION
channel profile, users of the v3.4.0 or v3.4.1 Native SDK experienced video freezes if they subscribed to videos from users of the Native SDK with versions earlier than v3.0.0.LIVE_BROADCASTING
channel profile, users of the v3.4.0 or v3.4.1 Native SDK experienced video freezes if they subscribed to videos from users of the Windows or macOS SDK with versions earlier than v3.0.0.libagora_ai_denoise_extension.dll
library.v3.4.1 was released on April 22, 2021. This release fixed the following issues:
v3.4.0 was released on April 16, 2021.
Integration changes
This release adds the libagora-wgc.dll
dynamic library, which is related to window sharing. To upgrade the SDK to v3.4.0 or later, ensure that you have copied this dynamic library to the folder where the agora_rtc_sdk.dll
file is located.
To reduce the app size after integrating the SDK, this release packages some features as extension libraries (with the Extension
suffix). For details, see Extension libraries. If you do not need to use the related features, you can remove the corresponding extension libraries and recompile the project.
Behavior changes
To monitor the reason why the playback state of a music file changes, this release modifies the onAudioMixingStateChanged
callback as follows:
errorCode
parameter with the reason
parameter.AUDIO_MIXING_ERROR_TYPE
and replaces it with AUDIO_MIXING_REASON_TYPE
instead. Using AUDIO_MIXING_REASON_TYPE
, you can get the reason for the change of the playback state, such as start, pause, stop or fail.AUDIO_MIXING_STATE_TYPE
. For example, as of this release, when looping music, the SDK triggers the AUDIO_MIXING_STATE_PLAYING
state when each loop is completed or starts.If you upgrade the SDK to v3.4.0 or later, to avoid affecting your service, Agora recommends that you modify the implementation of onAudioMixingStateChanged
.
As of this release, the adjustRecordingSignalVolume
method only adjusts the volume of the signal captured by the microphone. If you need to adjust the volume of the signal captured by the sound card, use the adjustLoopbackRecordingSignalVolume
method instead.
1. Volume control of sound cards
This release adds the adjustLoopbackRecordingSignalVolume
method. You can call this method after calling enableLoopbackRecording
to adjust the volume of the signal captured by the sound card.
2. Playback progress of audio effect files
To control the playback progress of audio effect files, this release adds the following methods:
playEffect
[2/2]: Sets the playback position when starting playback of an audio effect file by using the startPos
parameter.setEffectPosition
: Sets the playback position after starting playback of an audio effect file.getEffectDuration
: Gets the duration of a local audio effect file.getEffectCurrentPosition
: Gets current playback progress of an audio effect file.Also, this release deprecates the playEffect
[1/2] method. You can use playEffect
[2/2] instead.
1. Video encoding
The Agora SDK allows you to use degradationPreference
to set the degradation preference of local video encoding under limited bandwidth, for example, reducing the video frame rate while maintaining video quality, or reducing the video quality while maintaining the video frame rate. As of this release, degradationPreference
can be set as MAINTAIN_BALANCED
, which reduces the video frame rate and video quality simultaneously to strike a balance between smoothness and video quality. MAINTAIN_BALANCED
is suitable for scenarios where both smoothness and video quality are a priority, such as one-to-one calls, one-to-one online teaching, and multi-user online meetings.
2. Raw video data
The getRotationApplied
and getMirrorApplied
callbacks can rotate and mirror the raw video data. To improve the user experience, this release expands the video-data formats supported by these callback functions from RGBA to RGBA and YUV 420.
3. Window sharing
This release optimizes window sharing as follows:
startScreenCaptureByWindowId
method supports sharing most UWP (Universal Windows Platform) application windows and windows rendered using DirectX. During window sharing, a yellow border appears around the shared window, and calling captureMouseCursor(false)
does not take effect.4. Playback progress of music files
To control the playback progress of music files, this release is optimized as follows:
startAudioMixing
method and adds a new method with the same name in its place. The new startAudioMixing
method allows you to set the playback position when starting playback of a music file by using the startPos
parameter.getAudioMixingDuration
method and adds a new method with the same name in its place. The new getAudioMixingDuration
method allows you to get the duration of a local music file before playing it.5. Audio recording
To set the recording configuration during audio recording, this release adds a new startAudioRecording
method and deprecates the old method with the same name. The config
parameter of the new startAudioRecording
method allows you to set the audio recording quality, content, sample rate, and storage path of the recording file.
This release also adds a new error code: ERR_ALREADY_IN_RECORDING(160)
. If you call the new startAudioRecording
method again before the current recording ends, the SDK reports this error code.
6. Media device errors
To help users better understand the cause of local video or audio errors, this release adds the following new error codes:
LOCAL_VIDEO_STREAM_ERROR
: LOCAL_VIDEO_STREAM_ERROR_DEVICE_NOT_FOUND(8)
, which indicates that the SDK cannot find the local video-capture device.LOCAL_AUDIO_STREAM_ERROR
:LOCAL_AUDIO_STREAM_ERROR_NO_RECORDING_DEVICE(6)
, which indicates that the SDK cannot find the local audio-capture device.LOCAL_AUDIO_STREAM_ERROR_NO_PLAYOUT_DEVICE(7)
, which indicates that the SDK cannot find the local audio-playback device.This release fixed the following issues:
getMirrorApplied
to set the mirror mode under certain circumstances.onPreEncodeVideoFrame
.getMirrorApplied
and getRotationApplied
to process the local video data before encoding.onAudioVolumeIndication
callback returned incorrect volume information after you called enableSoundPositionIndication(true)
.Added
adjustLoopbackRecordingSignalVolume
getAudioMixingDuration
[2/2]startAudioRecording
[3/3]getEffectDuration
setEffectPosition
getEffectCurrentPosition
playEffect
[2/2]startAudioMixing
[2/2]LOCAL_VIDEO_STREAM_ERROR_DEVICE_NOT_FOUND(8)
in LOCAL_VIDEO_STREAM_ERROR
LOCAL_AUDIO_STREAM_ERROR_NO_RECORDING_DEVICE(6)
and LOCAL_AUDIO_STREAM_ERROR_NO_PLAYOUT_DEVICE(7)
in LOCAL_AUDIO_STREAM_ERROR
ERR_ALREADY_IN_RECORDING(160)
Modified
Deprecated
getAudioMixingDuration
[1/2]startAudioRecording
[2/3]playEffect
[1/2]startAudioMixing
[1/2]AUDIO_MIXING_ERROR_TYPE
v3.3.2 was released on March 29, 2021. This release fixed the issue where remote users saw a distorted image when the local user minimized the shared screen with the genie effect on Mac devices.
v3.3.1 was released on March 4, 2021.
Voice Conversion
This release adds the setVoiceConversionPreset
method to set a voice conversion effect (to disguise a user's voice). You can use this method to make a male-sounding voice sound steady or deep, and a female-sounding voice sound gender-neutral or sweet. See Set the Voice Effect.
1. AES-GCM encryption mode
For scenarios requiring high security, to guarantee the confidentiality, integrity, and authenticity of data, and to improve the computational efficiency of data encryption, this release adds the following enumerators in ENCRYPTION_MODE
:
AES_128_GCM
: 128-bit AES encryption, GCM mode.AES_256_GCM
: 256-bit AES encryption, GCM mode.2. Remote audio statistics
To monitor quality of experience (QoE) of the local user when receiving a remote audio stream, this release adds mosValue
to onRemoteAudioStats
, which reports the quality of the remote audio stream as determined by the Agora real-time audio MOS (Mean Opinion Score) measurement system.
This release adds the following APIs:
setVoiceConversionPreset
AES_128_GCM
and AES_256_GCM
in ENCRYPTION_MODE
mosValue
in RemoteAudioStats
v3.3.0 was released on January 22, 2021.
1. Integration change
This release adds the following frameworks:
libagora-core.dll
: The Agora basic calculation framework. av1.dll
: The AV1 framework for screen sharing.To integrate the SDK into your project, see Integrate the SDK.
2. Changes to subscribing behavior
This release deprecates setDefaultMuteAllRemoteAudioStreams
and setDefaultMuteAllRemoteVideoStreams
and changes the behavior of mute
-related methods as follows:
mute
-related methods must be called after joining or switching to a channel; otherwise, the method call does not take effect.muteAll
are no longer the master switch, and each mute
-related method independently controls the user's subscribing state. When you call methods with the prefixes muteAll
and muteRemote
together, the method that is called later takes effect.muteAll
set whether to subscribe to the audio or video streams of all remote users, including all subsequent users, which means methods with the prefix muteAll
contain the function of methods with the prefix setDefaultMute
. Agora recommends not calling methods with the prefixes muteAll
and setDefaultMute
together; otherwise, the setting may not take effect.See details in Set the Subscribing State.
3. Changes to voice activity behavior
In the remote users' onAudioVolumeIndication
callback, this release changes the value of the vad
member from always 0
to always 1
.
1. Channel media options
To help developers control media subscription more flexibly, this release adds the joinChannel
[2/2] and switchChannel
[2/2] methods to set whether users subscribe to all remote audio or video streams in a channel when joining and switching channels.
2. Cloud proxy
To improve the usability of the Agora Cloud Proxy, this release adds the setCloudProxy
method to set the cloud proxy and allows you to select a cloud proxy that uses the UDP protocol. For details, see Cloud Proxy.
3. Noise suppression
To eliminate non-stationary noise based on traditional noise reduction, this release adds enableDeepLearningDenoise
to enable noise suppression.
libagora_ai_denoise_extension.dll
dynamic library into your project files.4. Singing beautifier
To beautify the voice and add reverberation effects in a singing scenario, this release adds the setVoiceBeautifierParameters
method and adds the SINGING_BEAUTIFIER
enumeration value to VOICE_BEAUTIFIER_PRESET
.
You can call setVoiceBeautifierPreset(SINGING_BEAUTIFIER)
to beautify the male voice and add the reverberation effect for a voice in a small room. For more settings, you can call setVoiceBeautifierParameters
(SINGING_BEAUTIFIER
, param1
, param2
) to beautify male or female voices and add reverberation effects for a voice in a small room, large room, or hall.
5. Log files
To ensure the integrity of log content, this release adds the logConfig
member variable to RtcEngineContext
. You can use logConfig
to set the log files output by the Agora SDK when you initialize RtcEngine
. See How can I set the log file? for details.
As of v3.3.0, Agora does not recommend using the setLogFile
, setLogFileSize
, or setLogFilter
methods to set the log files.
6. Quality of captured video
To control the quality of video captured by the local camera, this release adds support for customizing the capture resolution and listening for abnormalities.
setCameraCapturerConfiguration
method to set the capture preference to MANUAL(3)
and set the width and height of the captured video image.captureBrightnessLevel
in the onLocalVideoStats
callback.onLocalVideoStateChanged(LOCAL_VIDEO_STREAM_STATE_FAILED, LOCAL_VIDEO_STREAM_ERROR_CAPTURE_FAILURE)
callback.onLocalVideoStateChanged(LOCAL_VIDEO_STREAM_STATE_CAPTURING, LOCAL_VIDEO_STREAM_ERROR_CAPTURE_FAILURE)
callback.7. Data streams
To support scenarios such as lyrics synchronization and courseware synchronization, this release deprecates the previous createDataStream
method and replaces it with a new method of the same name. You can use this new method to create a data stream and set whether to synchronize the data stream with the audio stream sent to the Agora channel and whether the received data is ordered.
1. Screen Sharing
As of v3.3.0, the SDK automatically enables DXGI (DirectX Graphics Infrastructure) to capture the screen as appropriate, which significantly increases the video frame rate and resolution supported by screen sharing.
2. Remote audio statistics
To monitor quality of experience (QoE) of the local user when receiving a remote audio stream, this release adds qoeQuality
and qualityChangedReason
to onRemoteAudioStats
, which report QoE of the local user and the reason for poor QoE, respectively.
This release fixed the following issues:
Added
setVoiceBeautifierParameters
SINGING_BEAUTIFIER
in the VOICE_BEAUTIFIER_PRESET
enumenableDeepLearningDenoise
joinChannel
[2/2]switchChannel
[2/2]createDataStream
logConfig
in the RtcEngineContext
structqoeQuality
and qualityChangedReason
in the RemoteAudioStats
structsetCloudProxy
captureBrightnessLevel
in the LocalVideoStats
structcaptureWidth
and captureHeight
in the CameraCapturerConfiguration
structCAPTURER_OUTPUT_PREFERENCE_MANUAL(3)
in the CAPTURER_OUTPUT_PREFERENCE
enumERR_MODULE_NOT_FOUND(157)
Deprecated
setDefaultMuteAllRemoteVideoStreams
setDefaultMuteAllRemoteAudioStreams
setLogFile
setLogFileSize
setLogFilter
createDataStream
v3.2.1 was released on December 17, 2020. This release fixed the following issues:
enableEncryption
, the SDK did not trigger the onFirstLocalVideoFramePublished
callback.v3.2.0 was released on November 30, 2020.
1. Integration change
Since v3.2.0, the following files have been added to the SDK package:
libagora-fdkaac.dll
: The Fraunhofer FDK AAC dynamic library.libagora-mpg123.dll
: The mpg123 dynamic library.libagora-soundtouch.dll
: The SoundTouch dynamic library.libagora-ffmpeg.dll
: The FFmpeg dynamic library.If you upgrade the SDK to v3.2.0 or later, ensure that you have copied the above files to the folder where the libagora-rtc-sdk.dll
file is located.
2. Cloud proxy
This release optimizes the Agora cloud proxy architecture. If you are already using cloud proxy, to avoid compatibility issues between the new SDK and the old cloud proxy, please contact support@agora.io before upgrading the SDK. See Cloud Proxy.
3. Security and compliance
Agora has passed ISO 27001, ISO 27017, and ISO 27018 international certifications, providing safe and reliable real-time interactive cloud services for users worldwide. See ISO Certificates.
This release supports transport layer encryption by adding TLS (Transport Layer Security) and UDP (User Datagram Protocol) encryption methods.
Interactive Live Streaming Standard
This release adds setClientRole
for setting the latency level of an audience member. You can use this method to switch between Interactive Live Streaming Premium and Interactive Live Streaming Standard as follows:
For details, see the product overview of Interactive Live Streaming Standard.
1. Meeting scenario
To improve the user experience in the meeting scenario, this release adds the following:
AUDIO_SCENARIO_MEETING(8)
in setAudioProfile
.LOCAL_VIDEO_STREAM_ERROR_SCREEN_CAPTURE_WINDOW_CLOSED(12)
error code, notifying you that a window shared by the window ID has been closed, or a full-screen window shared by the window ID has exited full-screen mode.CONTENT_HINT_DETAILS
type in a poor network environment.2. Voice beautifier and audio effects
To improve the usability of the APIs related to voice beautifier and audio effects, this release deprecates setLocalVoiceChanger
and setLocalVoiceReverbPreset
, and adds the following methods instead:
setVoiceBeautifierPreset
: Compared with setLocalVoiceChanger
, this method deletes audio effects such as a little boy’s voice and a more spatially resonant voice.setAudioEffectPreset
: Compared with setLocalVoiceReverbPreset
, this method adds audio effects such as the 3D voice, the pitch correction, a little boy’s voice and a more spatially resonant voice.setAudioEffectParameters
: This method sets detailed parameters for a specified audio effect. In this release, the supported audio effects are the 3D voice and pitch correction.3. Media Push with transcoding
This release adds the videoCodecType
member to the LiveTranscoding struct for setting the codec type of the transcoded video stream to H.264 or H.265.
4. Interactive streaming delay
This release reduces the latency on the audience's client during an interactive live streaming by about 500 ms.
Added
setAudioEffectPreset
setVoiceBeautifierPreset
setAudioEffectParameters
AUDIO_SCENARIO_MEETING(8)
in AUDIO_SCENARIO_TYPE
enum
LOCAL_VIDEO_STREAM_ERROR_SCREEN_CAPTURE_WINDOW_CLOSED(12)
in LOCAL_VIDEO_STREAM_ERROR
enum
setClientRole
videoCodecType
in the LiveTranscoding
structDeprecated
setLocalVoiceChanger
setLocalVoiceReverbPreset
getSmoothRenderingEnabled
(C++)v3.1.2 was released on September 15, 2020.
This release fixed the issue that the trigger timing of the onFirstLocalVideoFrame
and onFirstRemoteVideoFrame
callbacks is not accurate.
v3.1.1 was released on August 28, 2020.
This release changes the AREA_CODE
for regional connection. The latest area codes are as follows:
AREA_CODE_CN
: Mainland China.AREA_CODE_NA
: North America.AREA_CODE_EU
: Europe.AREA_CODE_AS
: Asia, excluding Mainland China.AREA_CODE_JP
: Japan.AREA_CODE_IN
: India.AREA_CODE_GLOB
: (Default) Global.If you have specified a region for connection when calling initialize
, ensure that you use the latest area code when migrating from an earlier SDK version.
v3.1.0 was released on Aug 11, 2020.
1. Publishing and subscription states
This release adds the following callbacks to report the current publishing and subscribing states:
onAudioPublishStateChanged
: Reports the change of the audio publishing state.onVideoPublishStateChanged
: Reports the change of the video publishing state.onAudioSubscribeStateChanged
: Reports the change of the audio subscribing state.onVideoSubscribeStateChanged
: Reports the change of the video subscribing state.2. First local frame published callback
This release adds the onFirstLocalAudioFramePublished
and onFirstLocalVideoFramePublished
callbacks to report that the first audio or video frame is published. The onFirstLocalAudioFrame
callback is deprecated from v3.1.0.
3. Custom data report
This release adds the sendCustomReportMessage
method for reporting customized messages. To try out this function, contact support@agora.io and discuss the format of customized messages with us.
1. Regional connection
This release adds the following regions for regional connection. After you specify the region for connection, your app that integrates the Agora SDK connects to the Agora servers within that region.
AREA_CODE_JAPAN
: Japan.AREA_CODE_INDIA
: India.2. Advanced screen sharing
This release adds the following features for screen sharing:
windowFocus
member to the ScreenCaptureParameters
struct.excludeWindowList
member to the ScreenCaptureParameters
struct.LOCAL_VIDEO_STREAM_STATE_CAPTURING(1)
and the error code LOCAL_VIDEO_STREAM_ERROR_SCREEN_CAPTURE_WINDOW_MINIMIZED(11)
to the onLocalVideoStateChanged
callback.IVideoSource
class, which defines the custom video source.3. Media Push
To improve the user experience in Media Push, this release adds the onRtmpStreamingEvent
callback to report events during Media Push, such as failure to add a background image or watermark image.
4. Encryption
This release adds the enableEncryption
method for enabling built-in encryption, and deprecates the following methods:
setEncryptionSecret
setEncryptionMode
5. More in-call statistics
This release adds the following attributes to provide more in-call statistics:
txPacketLossRate
in LocalAudioStats
, which represents the audio packet loss rate (%) from the local client to the Agora edge server before applying anti-packet loss strategies.LocalVideoStats
:txPacketLossRate
: The video packet loss rate (%) from the local client to the Agora edge server before applying anti-packet loss strategies.captureFrameRate
: The capture frame rate (fps) of the local video.publishDuration
in RemoteAudioStats
and RemoteVideoStats
, which represents the total publish duration (ms) of the remote media stream.6. Audio profile
To improve audio performance, this release adjusts the maximum audio bitrate of each audio profile as follows:
Profile | v3.1.0 | Earlier than v3.1.0 |
---|---|---|
AUDIO_PROFILE_DEFAULT |
||
AUDIO_PROFILE_SPEECH_STANDARD |
18 Kbps | 18 Kbps |
AUDIO_PROFILE_MUSIC_STANDARD |
64 Kbps | 48 Kbps |
AUDIO_PROFILE_MUSIC_STANDARD_STEREO |
80 Kbps | 56 Kbps |
AUDIO_PROFILE_MUSIC_HIGH_QUALITY |
96 Kbps | 128 Kbps |
AUDIO_PROFILE_MUSIC_HIGH_QUALITY_STEREO |
128 Kbps | 192 Kbps |
7. Log files
This release increases the default number of log files that the Agora SDK outputs from 2 to 5, and increases the default size of each log file from 512 KB to 1024 KB. By default, the SDK outputs five log files, agorasdk.log
, agorasdk_1.log
, agorasdk_2.log
, agorasdk_3.log
, agorasdk_4.log
. The SDK writes the latest logs in agorasdk.log
. When agorasdk.log
is full, the SDK deletes the log file with the earliest modification time among the other four, renames agorasdk.log
to the name of the deleted log file, and creates a new agorasdk.log
to record the latest logs.
8. Video experience under poor network conditions
This release fixed the occasional howling issue on some Huawei laptops.
Added
onAudioPublishStateChanged
onVideoPublishStateChanged
onAudioSubscribeStateChanged
onVideoSubscribeStateChanged
onFirstLocalAudioFramePublished
onFirstLocalVideoFramePublished
enableEncryption
txPacketLossRate
in the LocalAudioStats
structtxPacketLossRate
and captureFrameRate
in the LocalVideoStats
structpublishDuration
in the RemoteAudioStats
and RemoteVideoStats
structswindowFocus
and excludeWindowList
in the ScreenCaptureParameters
structLOCAL_VIDEO_STREAM_STATE_CAPTURING(1)
in the LOCAL_VIDEO_STREAM_STATE
enumLOCAL_VIDEO_STREAM_ERROR_SCREEN_CAPTURE_WINDOW_MINIMIZED(11)
in the LOCAL_VIDEO_STREAM_ERROR
enumsetVideoSource
IVideoSource
classIVideoFrameConsumer
classonRtmpStreamingEvent
ERR_NO_SERVER_RESOURCES(103)
WARN_APM_RESIDUAL_ECHO(1053)
Deprecated
setEncryptionSecret
setEncryptionMode
onFirstLocalAudioFrame
Deleted
WARN_ADM_IMPROPER_SETTINGS(1053)
v3.0.1.1 was released on Jun 18, 2020. This release fixed the following issues:
registerVideoRenderFactory
(deprecated) method.v3.0.1 was released on May 27, 2020.
Frame position for the video observer
As of this release, to get the video frame from the onPreEncodeVideoFrame
callback, you must set POSITION_PRE_ENCODER(1<<2)
in getObserverFramePosition
as the frame position to observe, as well as implementing the onPreEncodeVideoFrame
callback.
1. Audio mixing pitch
To set the pitch of the local music file during audio mixing, this release adds setAudioMixingPitch
. You can set the pitch
parameter to increase or decrease the pitch of the music file. This method sets the pitch of the local music file only. It does not affect the pitch of a human voice.
2. Voice enhancement
To improve the audio quality, this release adds the following enumerate elements in setLocalVoiceChanger
and setLocalVoiceReverbPreset
:
VOICE_CHANGER_PRESET
adds several elements that have the prefixes VOICE_BEAUTY
and GENERAL_BEAUTY_VOICE
. The VOICE_BEAUTY
elements enhance the local voice, and the GENERAL_BEAUTY_VOICE
enumerations add gender-based enhancement effects.AUDIO_REVERB_PRESET
adds the enumeration AUDIO_VIRTUAL_STEREO
and several enumerations that have the prefix AUDIO_REVERB_FX
. The AUDIO_VIRTUAL_STEREO
enumeration implements reverberation in the virtual stereo, and the AUDIO_REVERB_FX
enumerations implement additional enhanced reverberation effects.See the advanced guide Set the Voice Changer and Reverberation Effects for more information.
3. Fill mode
To improve the user experience of watching videos, this release adds a video display mode RENDER_MODE_FILL(4)
. This mode zooms and stretches the video to fill the display window. You can select this mode when calling the following methods:
setupLocalVideo
setupRemoteVideo
setLocalRenderMode
setRemoteRenderMode
4. Data post-processing in multiple channels
This release adds support for post-processing remote audio and video data in a multi-channel scenario by adding the following C++ methods:
IAudioFrameObserver
class: isMultipleChannelFrameWanted
and onPlaybackAudioFrameBeforeMixingEx
.IVideoFrameObserver
class: isMultipleChannelFrameWanted
and onRenderVideoFrameEx
.After successfully registering the audio or video observer, if you set the return value of isMultipleChannelFrameWanted
as true
, you can get the corresponding audio or video data from onPlaybackAudioFrameBeforeMixingEx
or onRenderVideoFrameEx
. In a multi-channel scenario, Agora recommends setting the return value as true.
Frame position
After successfully registering the video observer, you can observe and get the video frame at each node of video processing. To conserve power consumption, this release enables customizing the frame position for the video observer. Set the return value of the getObservedFramePosition
callback to set the position to observe:
onClientRoleChanged
callback, authentication with an App ID and token, and a garbled log directory.This release adds the following APIs:
setAudioMixingPitch
VOICE_CHANGER_PRESET
, such as VOICE_BEAUTY_VIGOROUS
AUDIO_REVERB_PRESET
, such as AUDIO_REVERB_FX_KTV
RENDER_MODE_FILL(4)
in the RENDER_MODE_TYPE
enumisMultipleChannelFrameWanted
and onPlaybackAudioFrameBeforeMixingEx
in the IAudioFrameObserver
classisMultipleChannelFrameWanted
and onRenderVideoFrameEx
in the IVideoFrameObserver
classgetObservedFramePosition
totalActiveTime
member in the RemoteAudioStats
structtotalActiveTime
member in the RemoteVideoStats
structWARN_ADM_WINDOWS_NO_DATA_READY_EVENT(1040)
and WARN_ADM_INCONSISTENT_AUDIO_DEVICE(1042)
in the warning codesv3.0.0.2 was released on Apr 22, 2020.
Specifying the area of connection
This release adds areaCode
member in the RtcEngineContext
struct for specifying the area of connection when creating an IRtcEngine
instance. This advanced feature applies to scenarios that have regional restrictions. You can choose from areas including Mainland China, North America, Europe, Asia (excluding Mainland China), and global (default).
After specifying the area of connection:
This release fixed the occasional black screen issue when a co-host shares the screen.
Added
areaCode
member in the RtcEngineContext
struct
v3.0.0 was released on Mar 5, 2020.
In this release, Agora improves the user experience under poor network conditions for both the COMMUNICATION
and LIVE_BROADCASTING
profiles through the following measures:
COMMUNICATION
profile.COMMUNICATION
and LIVE_BROADCASTING
profiles, which enhances the SDK's anti-packet-loss capacity by maximizing the net bitrate when the uplink and downlink bandwidth are insufficient.To deal with any incompatibility issues caused by the architecture change, Agora uses the fallback mechanism to ensure that users of different versions of the SDKs can communicate with each other: if a user joins the channel from a client using a previous version, all clients using v3.0.0 automatically fall back to the older version. This has the effect that none of the users in the channel can enjoy the improved experience. Therefore we strongly recommend upgrading all your clients to v3.0.0.
We also upgrade the On-premise Recording SDK to v3.0.0. Ensure that you upgrade your On-premise Recording SDK to v3.0.0 so that all users can enjoy the improvements brought by the new architecture and network strategy.
1. Dual-stream mode not enabled in the COMMUNICATION
profile
As of v3.0.0, the native SDK does not enable the dual-stream mode by default in the COMMUNICATION
profile. Call the enableDualStreamMode(true)
method after joining the channel to enable it. In video scenarios with multiple users, we recommend enabling the dual-stream mode.
1. Multiple channel management
To enable a user to join an unlimited number of channels at a time, this release adds the IChannel
and IChannelEventHandler
classes. By creating multiple IChannel
objects, a user can join the corresponding channels at the same time.
After joining multiple channels, users can receive the audio and video streams of all the channels, but publish one stream to only one channel at a time. This feature applies to scenarios where users need to receive streams from multiple channels, or frequently switch between channels to publish streams. See Join Multiple Channels for details.
2. Raw video data
Adds the following C++ callbacks to the IVideoFrameObserver
class to provide raw video data at different video transmission stages, and to accommodate more scenarios.
onPreEncodeVideo
: Gets the video data after pre-processing and prior to encoding. This method applies to the scenarios where you need to pre-process the video data.getSmoothRenderingEnabled
: Sets whether to smooth the acquired video frames. The smoothed video frames are more evenly spaced, providing a better rendering experience.3. Adjusting the playback volume of the specified remote user
Adds adjustUserPlaybackSignalVolume
for adjusting the playback volume of a specified remote user. You can call this method as many times as necessary in a call or interactive live streaming to adjust the playback volume of different remote users, or to repeatedly adjust the playback volume of the same remote user.
4. Image enhancement
Adds setBeautyEffectOptions
for enabling image enhancement in scenarios such as video social networking, an online class, or interactive live streaming. You can call this method to set parameters including contrast, brightness, smoothness, red saturation, and so on. See Image Enhancement for details.
1. Audio profiles
To meet the need for higher audio quality, this release adjusts the corresponding audio profile of AUDIO_PROFILE_DEFAULT (0)
in the LIVE_BROADCASTING
profile.
SDK | AUDIO_PROFILE_DEFAULT (0) |
---|---|
v3.0.0 | A sample rate of 48 kHz, music encoding, mono, and a bitrate of up to 52 Kbps. |
Earlier than v3.0.0 | 3A sample rate of 32 kHz, music encoding, mono, and a bitrate of up to 64 Kbps. |
2. Mirror mode
This release enables you to set a mirror effect for the stream to be encoded or for the streams to be rendered.
mirrorMode
member to the VideoEncoderConfiguration
struct for setting a mirror effect for the stream to be encoded and transmitted.setLocalRenderMode
and setRemoteRenderMode
methods, both of which take an extra mirrorMode
parameter. During a call, you can use setLocalRenderMode
to update the mirror effect of the local view or setRemoteRenderMode
to update the mirror effect of the remote view on the local device.mirrorMode
member to the VideoCanvas
struct. You can use setupLocalVideo
to set a mirror effect for the local view, or use setupRemoteVideo
to set a mirror effect for the remote view on the local device.3. Quality statistics
Adds the following members in the RtcStats
class for providing more in-call statistics, making it easier to monitor the call quality and memory usage in real time:
gatewayRtt
memoryAppUsageRatio
memoryTotalUsageRatio
memoryAppUsageInKbytes
4. Screen sharing
This release enables window sharing of UWP (Universal Windows Platform) applications when you call startScreenCaptureByWindowId
.
5. Others
This release enables interoperability between the RTC Native SDK and the RTC Web SDK by default, and deprecates the enableWebSdkInteroperability
method.
Added
setBeautyEffectOptions
onPreEncodeVideoFrame
getSmoothRenderingEnabled
setLocalRenderMode
setRemoteRenderMode
mirrorMode
member in the VideoEncoderConfiguration
structmirrorMode
and channelId
members in the VideoCanvas
structchannelId
member in the AudioVolumeInfo
structcreateChannel
IChannel
classIChannelEventHandler
classgatewayRtt
, memoryAppUsageRatio
, memoryTotalUsageRatio
and memoryAppUsageInKbytes
members in the RtcStats
classDeprecated
RtcEngineParameters
classenableWebSdkInteroperability
setLocalRenderMode¹
setRemoteRenderMode¹
setLocalVideoMirrorMode
onFirstRemoteAudioDecoded
, and onFirstRemoteAudioFrame
, replaced by onRemoteAudioStateChanged
onStreamPublished
and onStreamUnpublished
, replaced by onRtmpStreamingStateChanged
v2.9.3 was released on Feb 10, 2020.
This release fixed the following issues:
setRemoteSubscribeFallbackOption
method, which should work in the LIVE_BROADCASTING
profile only, also works in the COMMUNICATION
profile.v2.9.1 is released on Sep 19, 2019.
1. Detecting local voice activity
This release adds the report_vad(bool)
parameter to the enableAudioVolumeIndication
method to enable local voice activity detection. Once it is enabled, you can check the AudioVolumeInfo
struct of the onAudioVolumeIndication
callback for the voice activity status of the local user.
2. Supporting RGBA raw video data
This release supports RGBA raw video data. Use the getVideoFormatPreference
method to set the format of the raw video data format.
You can also rotate or mirror the RGBA raw data using the getRotationApplied
or getMirrorApplied
methods respectively.
1. Improving the watermark function in Live Interactive Streaming
This release adds a new addVideoWatermark
method with the following settings:
visibleInPreview
member sets whether the watermark is visible in the local preview.positionInLandscapeMode
/positionInPortraitMode
member sets the watermark position when the encoding video is in landscape/portrait mode.This release optimizes the watermark function, reducing the CPU usage by 5% to 20%.
The original addVideoWatermark
method is deprecated.
2. Supporting more audio sample rates for recording
To enable more audio sample rate options for recording, this release adds a new startAudioRecording
method with a sampleRate
parameter. In the new method, you can set the sample rate as 16, 32, 44.1 or 48 kHz. The original method supports only a fixed sample rate of 32 kHz and is deprecated.
Miscellaneous
A typo in the IAgoraRtcEngine.h file.
To improve the user experience, we made the following changes in v2.9.1:
Added
startAudioRecording
addVideoWatermark
getVideoFormatPreference
getRotationApplied
getMirrorApplied
report_vad
parameter in the enableAudioVolumeIndication
methodvad
member in the AudioVolumeInfo
classDeprecated
startAudioRecording
addVideoWatermark
v2.9.0 is released on Aug 16, 2019.
1. Media Push
In this release, we deleted the following methods:
configPublisher
setVideoCompositingLayout
clearVideoCompositingLayout
If your app implements Media Push with the methods above, ensure that you upgrade the SDK to the latest version and use the following methods for the same function:
For how to implement the new methods, see Media Push.
2. Reporting the state of the remote video
This release extends the onRemoteVideoStateChanged
callback with more states of the remote video: STOPPED(0), STARTING(1), DECODING(2), FROZEN(3), and FAILED(4). It adds a reason parameter to the callback to indicate why the remote video state changes. The original onRemoteVideoStateChanged
callback is deleted. If you upgrade your Native SDK to the latest version, ensure that you re-implement the onRemoteVideoStateChanged
callback.
onRemoteVideoStateChanged
callback is triggered only when the remote video state has changed.3. Disabling/enabling the local audio
To improve the audio quality in the COMMUNICATION
profile, this release sets the system volume to the media volume after you call the enableLocalAudio
(true) method. Calling enableLocalAudio
(false) switches the system volume back to the in-call volume.
1. Faster switching to another channel
This release adds the switchChannel
method to enable the audience in an interactive live streaming channel to quickly switch to another channel. With this method, you can achieve a much faster switch than with the leaveChannel
and joinChannel
methods. After the audience successfully switches to another channel by calling the switchChannel
method, the SDK triggers the onLeaveChannel
and onJoinChannelSuccess
callbacks to indicate that the audience has left the original channel and joined a new one.
2. Channel media stream relay
This release adds the following methods to relay the media streams of a host from a source channel to a destination channel. This feature applies to scenarios such as online singing contests, where hosts of different LIVE_BROADCASTING
channels interact with each other.
During the media stream relay, the SDK reports the states and events of the relay with the onChannelMediaRelayStateChanged
and onChannelMediaRelayEvent
callbacks.
For more information on the implementation, API call sequence, sample code, and considerations, see Co-host across Channels.
3. Reporting the local and remote audio state
This release adds the onLocalAudioStateChanged
and onRemoteAudioStateChanged
callbacks to report the local and remote audio states. With these callbacks, the SDK reports the following states for the local and remote audio:
error
parameter for troubleshooting.reason
parameter for why the remote audio state changes.4. Reporting the local audio statistics
This release adds the onLocalAudioStats
callback to report the statistics of the local audio during a call, including the number of channels, the sending sample rate, and the average sending bitrate of the local audio.
1. Reporting more statistics of the in-call quality
This release adds the following statistics in the RtcStats
, LocalVideoStats
, and RemoteVideoStats
classes:
RtcStats
: The total number of the sent audio bytes, sent video bytes, received audio bytes, and received video bytes during a session.LocalVideoStats
: The encoding bitrate, the width and height of the encoding frame, the number of frames, and the codec type of the local video.RemoteVideoStats
: The packet loss rate of the remote video.2. Improving the video quality of the interactive live streaming
This release minimizes the video freeze rate under poor network conditions, improves the video sharpness, and optimizes the video smoothness when the packet loss rate is high.
3. Improving the screen sharing quality
This release improves the sharpness of text during screen sharing in the COMMUNICATION
profile, particularly when the network condition is poor. Note that this improvement takes effect only when you set ContentHint as Details(2).
4. Supporting hot-swappable devices
To improve the experience in using video devices on Windows, this release adds support for hot-swappable devices with the context
member in the RtcEngineContext
class. Setting this member also achieves the following functions:
5. Other Improvements
COMMUNICATION
profile.Audio
muteRemoteAudioStream
method.Video
onRemoteVideoStateChanged
callback behaves unexpectedly.Miscellaneous
To improve the user experience, we made the following changes in v2.9.0:
Added
onLocalAudioStateChanged
onRemoteAudioStateChanged
onRemoteVideoStateChanged
onLocalAudioStats
switchChannel
startChannelMediaRelay
updateChannelMediaRelay
stopChannelMediaRelay
onChannelMediaRelayStateChanged
onChannelMediaRelayEvent
RtcEngineContext
: contextRtcStats
: txAudioBytes
, txVideoBytes
, rxAudioBytes
, and rxVideoBytes
LocalVideoStats
: encodedBitrate
, encodedFrameWidth
, encodedFrameHeight
, encodedFrameCount
, and codedType
RemoteVideoStats
: packetLossRate
Deprecated
onMicrophoneEnabled
. Use LOCAL_AUDIO_STREAM_STATE_CHANGED(0) or LOCAL_AUDIO_STREAM_STATE_RECORDING(1) in the onLocalAudioStateChanged
callback instead.onRemoteAudioTransportStats
. Use the onRemoteAudioStats
callback instead.onRemoteVideoTransportStats
. Use the onRemoteVideoStats
callback instead.Deleted
configPublisher
setVideoCompositingLayout
clearVideoCompositingLayout
onRemoteVideoStateChanged
v2.8.0 is released on Jul. 8, 2019.
1. Supporting string user IDs
Many apps use string user IDs. This release adds the following methods to enable apps to join an Agora channel directly with string user IDs as user accounts:
For other methods, Agora uses the integer uid parameter. The Agora Engine maintains a mapping table that contains the user ID and string user account, and you can get the corresponding user account or ID by calling the getUserInfoByUid or getUserInfoByUserAccount method.
To ensure smooth call, use the same parameter type to identify all users within a channel, that is, all users should use either the integer user ID or the string user account to join a channel.
Do not mix parameter types within the same channel. The following Agora SDKs support string user accounts:
If you use SDKs that do not support string user accounts, only integer user IDs can be used in the channel.
If you change your user IDs into string user accounts, ensure that all app clients are upgraded to the latest version.
If you use string user accounts, ensure that the token generation script on your server is updated to the latest version. If you join the channel with a user account, ensure that you use the same user account or its corresponding integer user ID to generate a token. Call the getUserInfoByUserAccount
method to get the user ID that corresponds to the user account.
2. Adding remote audio and video statistics
To monitor the audio and video transmission quality during a call or interactive live streaming, this release adds the totalFrozenTime
and frozenRate
members in the RemoteAudioStats and RemoteVideoStats classes, to report the audio and video freeze time and freeze rate of the remote user.
This release also adds the numChannels
, receivedSampleRate
, and receivedBitrate
members in the RemoteAudioStats class.
This release adds a CONNECTION_CHANGED_KEEP_ALIVE_TIMEOUT(14)
member to the reason
parameter of the onConnectionStateChanged callback. This member indicates a connection state change caused by the timeout of the connection keep-alive between the SDK and Agora's edge server.
To improve your experience, we made the following changes to the APIs:
Added
numChannels
, receivedSampleRate
, receivedBitrate
, totalFrozenTime
, and frozenRate
members in the RemoteAudioStats classtotalFrozenTime
and frozenRate
members in the RemoteVideoStats classDeprecated
lowLatency
member in the LiveTranscoding classV2.4.1 is released on Jun 12th, 2019.
Ensure that you read the following SDK behavior changes if you migrate from an earlier SDK version.
1. Publishing streams to the RTMP
To improve the usability of the Media Push service, v2.4.1 defines the following parameter limits:
Class / Interface | Parameter Limit |
---|---|
LiveTranscoding | |
RtcImage | url: The maximum length of this parameter is **1024 bytes. |
addPublishStreamUrl | url : The maximum length of this parameter is **1024 bytes. |
removePublishStreamUrl | url : The maximum length of this parameter is **1024 bytes. |
This release also adds the audioCodecProfile parameter in the LiveTranscoding
class to set the audio codec profile type. The default type is LC-AAC, which means the low-complexity audio codec profile.
v2.4.1 also adds five error codes to the error
parameter in the onStreamPublished method for quick troubleshooting.
2. Renaming the receivedFrameRate parameter in the RemoteVideoStats class
v2.4.1 renames the receivedFrameRate
parameter to rendererOutputFrameRate in the RemoteVideoStats class to more accurately describe the statistics of the remote video stream.
1. Adding media metadata
In interactive live streaming scenarios, the host can send shopping links, digital coupons, and online quizzes to the audience for more diversified interactive live streaming interactions. v2.4.1 adds the registerMediaMetadataObserver interface and the IMediaMetadataObserver class, allowing the host to add metadata to the output video and to send media attached information.
2. Optimized screen sharing
To avoid image cropping and distortion in screen sharing, v2.4.1 optimizes the encoding algorithms. In this release Agora applies the following encoding algorithms:
Suppose the value of dimensions is 1920 x 1080 pixels, that is, 2073600 pixels:
dimensions
is lower than that of the encoding dimensions, for example, 1000 x 1000 pixels, the SDK uses 1000 x 1000 pixels for encoding.dimensions
is higher than that of the encoding dimensions, for example, 2000 x 2000 pixels, the SDK uses the maximum value under 1920 x 1080 pixels with the aspect ratio of the screen dimension (1:1) for encoding, that is, 1440 x 1440 pixels.Agora uses the dimensions value in theScreenCaptureParameters class to calculate the charges. If you do not set the value of **dimensions, the SDK uses the default value of 1920 x 1080 to calculate the charges.
You can also choose whether or not to capture the mouse cursor when sharing the screen. v2.4.1 adds the captureMouseCursor parameter in the ScreenCaptureParameters
class and captures the mouse by default.
3. State of the local video
v2.4.1 adds the onLocalVideoStateChanged callback to indicate the local video state. In this callback, the SDK returns the STOPPED
,CAPTURING
, ENCODING
, or FAILED
state. When the state is FAILED
, you can use the error code for troubleshooting. This callback indicates whether or not the interruption is caused by capturing or encoding. This release deprecates the onCameraReady
and onVideoStopped
callbacks.
4. State of the Media Push
v2.4.1 adds the onRtmpStreamingStateChanged callback to indicate the state of the Media Push and help you troubleshoot issues when exceptions occur. In this callback, the SDK returns the IDLE, CONNECTING
, RUNNING
, RECOVERING
, or FAILURE
state. When the state is FAILURE
, you can use the error code for troubleshooting. You can still use the onStreamPublished
and onStreamUnpublished
callbacks, but we do not recommend using them.
5. More reasons for a network connection state change
In the onConnectionStateChanged callback, v2.4.1 adds error codes to the reason
parameter to help you troubleshoot issues when exceptions occur. The SDK returns the onConnectionStateChanged
callback whenever the connection state changes. This release also deprecates WARN_LOOK_UP_CHANNEL_REJECTED(105)
, ERR_TOKEN_EXPIRED(109)
, and ERR_INVALID_TOKEN(110)
.
6. State of the local network type
v2.4.1 adds the onNetworkTypeChanged callback to indicate the local network type. In this callback, the SDK returns the UNKNOWN
, DISCONNECTED
, LAN
, WIFI
, 2G
, 3G
, or 4G
type. When the network connection is interrupted, this callback indicates whether or not the interruption is caused by a network type change or poor network conditions.
7. Getting the audio mixing volume
v2.4.1 adds the getAudioMixingPlayoutVolume and getAudioMixingPublishVolume methods, which respectively gets the audio mixing volume for local playback and remote playback, to help you troubleshoot audio volume related issues.
8. Reporting when the first remote audio frame is received and decoded
To get the more accurate time of the first audio frame from a specified remote user, v2.4.1 adds the onFirstRemoteAudioDecoded callback to report to the app that the SDK decodes first remote audio. This callback is triggered in either of the following scenarios:
The difference between the onFirstRemoteAudioDecoded
and onFirstRemoteAudioFrame
callbacks is that the onFirstRemoteAudioFrame
callback occurs when the SDK receives the first audio packet. It occurs before the onFirstRemoteAudioDecoded
callback.
9. Miscellaneous
v2.4.1 supports 64-bit operation systems.
1. Reporting more statistics
v2.4.1 adds the txPacketLossRate and rxPacketLossRate parameters in the RtcStats class. These parameters return the packet loss rate from the local client to the server and vice versa.
To provide more accurate statistics of the local and remote video, v2.4.1 makes the following changes to the following classes:
2. Miscellaneous
enableAudioQualityIndication
method.Audio
Video
Miscellaneous
onNetworkQuality
callback after leaving the channel.To improve your experience, we made the following changes to the APIs:
Unified the C++ interface for all platforms
v2.4.1 unifies the behavior of the C++ interfaces across different platforms so that you can apply the same code logic on different platforms. v2.4.1 implements the methods of the RtcEngineParameters
class in the IRtcEngine
class. Refer to Agora C++ API Reference for All Platforms home page for the applicable platforms and considerations of each interface.
Added
LiveTranscoding
classScreenCaptureParameters
classRtcStats
classLocalVideoStats
classreceivedRemoteRate
) parameters in the RemoteVideoStats
classDeprecated
enableAudioQualityIndication
onCameraReady
. Use LOCAL_VIDEO_STREAM_STATE_CAPTURING(1) in the onLocalVideoStateChanged callback instead.onVideoStopped
. Use LOCAL_VIDEO_STREAM_STATE_STOPPED(0) in the onLocalVideoStateChanged callback instead.WARN_LOOKUP_CHANNEL_REJECTED(105)
warning code. Use CONNECTION_CHANGED_REJECTED_BY_SERVER(10) in the onConnectionStateChanged callback instead.ERR_TOKEN_EXPIRED(109)
error code. Use CONNECTION_CHANGED_TOKEN_EXPIRED(9) in the onConnectionStateChanged callback instead.ERR_INVALID_TOKEN(110)
error code. Use CONNECTION_CHANGED_INVALID_TOKEN(8) in the onConnectionStateChanged callback instead.ERR_START_CAMERA(1003)
error code. Use LOCAL_VIDEO_STREAM_ERROR_CAPTURE_FAILURE(4) in the onLocalVideoStateChanged callback instead.ERR_VDM_WIM_DEVICE_IN_USE(1502)
error code. Use LOCAL_VIDEO_STREAM_ERROR_DEVICE_BUSY(3) in the in the onLocalVideoStateChanged callback instead.v2.4.0
v2.4.0 is released on April 1, 2019.
v2.4.0 upgrades screen sharing and provides the following advanced functions:
startScreenCaptureByScreenRect
).startScreenCaptureByWindowId
) .setScreenCaptureContentHint
).updateScreenCaptureParameters
).v2.4.0 deprecates the startScreenCapture
method. We recommend using the new methods for screen sharing. With the new methods, developers need to design the code logic to obtain the screenRect
and windowId
. For more information, see Share the Screen.
Adding voice changer and reverberation effects in an audio chat room brings much more fun. v2.4.0 adds the setLocalVoiceChanger
and setLocalVoiceReverbPreset
methods, allowing you to change your voice or reverberation by choosing from the preset options.
v2.4.0 adds the enableSoundPositionIndication
and setRemoteVoicePosition
. Call the enableSoundPositionIndication
method before joining a channel to enable stereo panning for the remote users, and then you can call the setRemoteVoicePosition
method to track the position of a remote user.
Conducting a last-mile probe test before joining the channel helps the local user to evaluate or predict the uplink network conditions. v2.4.0 adds the startLastmileProbeTest
, stopLastmileProbeTest
, and onLastmileProbeResult
APIs, allowing you to get the uplink and downlink last-mile network statistics, including the bandwidth, packet loss, jitter, and round-trip time (RTT).
v2.4.0 adds the startAudioDeviceLoopbackTest
and stopAudioDeviceLoopbackTest
methods for testing whether the local audio devices are working properly. The test involves only the local audio devices and does not report the network condition.
v2.4.0 adds the setRemoteUserPriority
method for setting the priority of a remote user. You can use this method with the setRemoteSubscribeFallbackOption
method. If the fallback function is enabled for a remote stream, the SDK ensures the high-priority user gets the best possible stream quality.
v2.4.0 adds the onAudioMixingStateChanged
callback to report any change of the audio-mixing file playback state (playback succeeds or fails) and the corresponding reason. This release also adds the warning code 701, which is triggered if the local audio-mixing file does not exist, or if the SDK does not support the file format or cannot access the music file URL when playing the audio-mixing file.
The SDK has two log files, each with a default size of 512 KB. In case some customers require more than the default size, v2.4.0 adds the setLogFileSize
method for setting the log file size (KB).
v2.4.0 adds the backgroundImage
parameter in the LiveTranscoding
Class in Windows to set the background image in the combined video of the interactive live streaming.
Adds the cloud proxy service. See Use Cloud Proxy for details.
LocalVideoStats
class: targetBitrate
for setting the target bitrate of the current encoder, targetFrameRate
for setting the target frame rate, and qualityAdaptIndication
for reporting the quality of the local video since last count.v2.4.0 provides the following options for setting video encoder preferences:
VideoEncoderConfiguration
class: minFrameRate and degradationPrefer. You can use these parameters together to set the minimum video encoder frame rate and the video encoding degradation preference under limited bandwidth.setCameraCapturerConfiguration
method, allowing you to set the camera capture preference. You can choose system performance over video quality or vice versa as needed. For more information, see the API Reference.enableLocalAudio
method disconnects all connected Bluetooth devices.renderMode
setting, the video stretches due to a mismatch with the display.To improve your experience, we made the following changes to the APIs:
setBeautyEffectOptions
startScreenCaptureByScreenRect
startScreenCaptureByWindowId
updateScreenCaptureParameters
setScreenCaptureContentHint
setLocalVoiceChanger
setLocalVoiceReverbPreset
enableSoundPositionIndication
setRemoteVoicePosition
startLastmileProbeTest
stopLastmileProbeTest
setRemoteUserPriority
startEchoTest
startAudioDeviceLoopbackTest
stopAudioDeviceLoopbackTest
setCameraCapturerConfiguration
setLogFileSize
onAudioMixingStateChanged
onLastmileProbeResult
startEchoTest
startScreenCapture
setVideoQualityParameters
v2.3.3
v2.3.3 is released on January 24, 2019.
v2.3.3 optimizes the screen-sharing algorithm for different scenarios. The video smoothness and quality are enhanced when a user presents slides or browses websites. v2.3.3 also improves the initial image quality in the COMMUNICATION
profile.
Occasional inaccurate statistics returned in the onNetworkQuality
callback.
v2.3.2
v2.3.2 is released on January 16, 2019.
Besides the new features and improvements mentioned below, it is worth noting that v2.3.2:
LIVE_BROADCASTING
profile.Before upgrading your SDK, ensure that your version is:
If you using an Agora SDK version v2.0.8 and wish to migrate the v2.3.2, refer to the Migration Guide for major API changes.
v2.3.2 adds the following methods, allowing you to use the external video during a call or in interactive live streaming:
setExternalVideoSource
: Configures the external video source.pushVideoFrame
: Pushes the external video frames.The application can encapsulate the external video frames in the AgoraVideoFrame
format and push them to the SDK for encoding and transmitting.
This feature applies to scenarios where a user captures and renders the external video and then pushes the video frames to the SDK for transmission (the application processes the rendering).
v2.3.2 adds the following methods to improve the user experience in playing the external audio:
setExternalAudioSink
: Sets the external audio sink.pullAudioFrame
: Pulls the external audio frame.The application pulls the decoded audio frames (mixed from the remote users) from the media engine for external playback.
The difference between pullAudioFrame
and onPlaybackAudioFrame
:
pullAudioFrame
: The application pulls the audio frames and:
Specifies the number of audio samples for playback.
Adjusts the frame buffer.
Allows for a delay in processing the audio frame.
This method avoids problems caused by jitter in the external audio, such as an unsynchronized audio playback.
onPlaybackAudioFrame
: The SDK pushes the audio frames to the application once every 10 ms. Any delay in processing the audio frames may result in audio jitter.
v2.3.2 adds the minBitrate parameter (minimum encoding bitrate) in the setVideoEncoderConfiguration method. The SDK automatically adjusts the encoding bitrate to adapt to the network conditions. Using a value greater than the default value forces the video encoder to output high-quality images but may cause more packet loss and hence sacrifice the smoothness of the video transmission. Agora does not recommend changing this value unless you have special requirements for image quality.
v2.3.2 adds the adjustAudioMixingPlayoutVolume
and adjustAudioMixingPublishVolume
methods to complement the adjustAudioMixingVolume
method, allowing you to independently adjust the audio mixing volume for local playback and remote publishing.
This release also changes the behavior of the adjustPlaybackSignalVolume method to control only the voice volume. Therefore, to mute the local audio playback, call both the adjustPlaybackSignalVolume(0)
and adjustAudioMixingVolume(0)
methods.
See Adjust the Volume for the scenarios and corresponding APIs.
Unreliable network conditions affect the overall quality of the interactive live streaming. v2.3.2 adds the setLocalPublishFallbackOption
and setRemoteSubscribeFallbackOption
methods to allow the SDK to:
The SDK triggers the onLocalPublishFallbackToAudioOnly
or onRemoteSubscribeFallbackToAudioOnly
callback when the stream falls back to audio-only or switches back to the video.
v2.3.2 adds the onRemoteAudioTransportStats
and onRemoteVideoTransportStats
callbacks to provide the upstream and downstream statistics of each remote user/host. During a call or interactive live streaming, the SDK triggers these callbacks once every two seconds after the local user receives audio/video packets from a remote user. The callbacks return the user ID, received audio/video bitrate, packet loss rate, and network time delay (ms).
To support video rotation scenarios and improve the quality of the custom video source, v2.3.2 deprecates the setVideoProfile
method and replaces it with the setVideoEncoderConfiguration
method to set the video encoder configurations. The VideoEncoderConfiguration
class provides a set of configurable video parameters, including the dimension, frame rate, bitrate, and orientation. You can still use the setVideoProfile
method, but Agora recommends using the setVideoEncoderConfiguration
method to set the video profile.
v2.3.2 adds the deviceName
parameter in the enableLoopbackRecording
method, allowing you to use a virtual sound card for audio recording:
deviceName
as the name of the virtual card.v2.3.2 deprecates the onAudioQuality
callback and replaces it with the onRemoteAudioStats
callback to improve the accuracy of the call quality statistics. The onRemoteAudioStats
callback returns parameters such as the audio frame loss rate, end-to-end audio delay, and jitter buffer delay at the receiver, which are more closely linked to the real user experience. In addition, v2.3.2 optimizes the algorithm of the onNetworkQuality
callback for the uplink and downlink network qualities.
onRemoteAudioStats
: Reports the statistics of the remote audio stream from each user/host. This callback replaces the onAudioQuality callback.onNetworkQuality
: Reports the last mile network quality of each user in the channel.Agora plans to improve the following callback in subsequent versions:
onLastmileQuality
: Reports the last mile network quality of the local user before the user joins a channel.v2.3.2 adds the following API method and callback to get the current network connection state and reason for a connection state change:
getConnectionState
: Gets the connection state of the SDK.onConnectionStateChanged
: Occurs when the connection state of the SDK to the server changes.v2.3.2 deprecates the onConnectionInterrupted
and onConnectionBanned
callbacks.
In the new API method, the network connection states are "disconnected", "connecting", "connected", "reconnecting", and "failed". The SDK triggers the onConnectionStateChanged
callback when the network connection state changes. The SDK also triggers the onConnectionInterrupted
and onConnectionBanned
callbacks under certain circumstances, but Agora does not recommend using them.
v2.3.2 changes the rating parameter in the rate
method to "1 to 5" to encourage more feedback from end-users on the quality of a call or interactive live streaming. Application developers can use this feedback for future product improvement. Agora strongly recommends integrating this method in your application.
v2.3.2 optimizes the audio device selection to fix the no audio issue when a user switches the audio device during a call or interactive live streaming.
LIVE_BROADCASTING
profile.The following issues are fixed in v2.3.2:
startAudioMixing
method to play music files.setVideoProfile
method to specify the video resolution.To improve the user experience, Agora has made the following changes to the APIs:
setVideoEncoderConfiguration
setExternalVideoSource
pushVideoFrame
setExternalAudioSink
pullAudioFrame
setLocalPublishFallbackOption
setRemoteSubscribeFallbackOption
getConnectionState
adjustAudioMixingPlayoutVolume
adjustAudioMixingPublishVolume
onConnectionStateChanged
onLocalPublishFallbackToAudioOnly
onRemoteSubscribeFallbackToAudioOnly
onRemoteAudioStats
onRemoteAudioTransportStats
onRemoteVideoTransportStats
v2.2.2
v2.2.2 is released on June 21, 2018.
v2.2.1
v2.2.0 is released on May 30, 2018 and improves the internal code implementation.
v2.2.0
v2.2.0 is released on May 4, 2018.
Adds a publish
parameter in the playEffect
method to enable the remote user in the channel to hear the audio effect played locally.
If your SDK is upgraded to v2.2 from a previous version, pay attention to the functional changes of this API.
Agora provides a proxy package for enterprise users with corporate firewalls to deploy before accessing the services of Agora.
Adds the remoteVideoStateChangedOfUid
method to get the state of the remote video stream.
Adds the watermark function for users to add a PNG file to the local or RTMP live streaming as a watermark. Adds the addVideoWatermark
and clearVideoWatermarks
methods to add and delete watermarks in a local live streaming. Adds the watermark
parameter in the LiveTranscoding
interface to add watermarks in Media Push.
Improves the enableAudioVolumeIndication
method. This method once enabled, sends the audio volume indication of the speaker in its callback at set intervals, regardless of whether anyone is speaking in the channel.
To meet the need for real-time network quality detection in the channel, the onNetworkQuality
callback improves its data accuracy.
To test if the network condition can support audio or video calls before joining a channel, the onLastmileQuality
callback changes its detection base from a fixed bitrate to the bitrate set in videoProfile
to improve data accuracy. When the network condition is unknown, the SDK still triggers this callback once every 2 seconds.
Improves the audio quality in scenarios that involve music playback.
LIVE_BROADCASTING
channel.v2.1.3
v2.1.3 is released on April 19, 2018.
In v2.1.3, Agora updates the bitrate values of the setVideoProfile
method in the LIVE_BROADCASTING
profile. The bitrate values in v2.1.3 stay consistent with those in v2.0.
Occasional recording failures on some phones when a user leaves a channel and turns on the built-in recording device.
Improves the performance of screen sharing by shortening the time interval between which users switch from screen sharing to a normal COMMUNICATION
or LIVE_BROADCASTING
profile.
v2.1.2
v2.1.2 is released on April 2, 2018.
If you upgrade the SDK to v2.1.2 from a previous version, the video quality of the interactive live streaming will be better than the video quality of the call in the same resolutions, resulting in the interactive live streaming using more bandwidth.
COMMUNICATION
profile than in the LIVE_BROADCASTING
profile.v2.1.1
v2.1.1 is released on March 16, 2018.
Agora has identified a critical bug in SDK v2.1. Upgrade to v2.1.1 if you are using Agora SDK v2.1.
v2.1.0
v2.1.0 is released on March 7, 2018.
Adds a scenario for the game chat room to reduce the bandwidth and cancel the noise with the setAudioProfile
method.
In the interactive live streaming scenario, the host can enhance the local audio effects from the built-in microphone with the setLocalVoiceEqualization
and setLocalVoiceReverb
methods to implement the expected voice equalization and reverberation effects.
Adds RESTful APIs to check the status of the users in the channel, the channel list of a specific company, and whether the user is an audience or a host. See:
Adds the support of 17-way video in the interactive live streaming scenario
Adds the function of injecting an external video stream to an ongoing interactive streaming.
Adds the enableLoopbackRecording
method to collect all local sounds.
Improvement | Description |
---|---|
Video Freeze Rate | Reduces the video freeze rate in the audience mode and for specific devices. |
Authentication | Supports a new authentication mechanism. Each legacy Dynamic Key (Channel Key) corresponds to a single privilege (for example, joining a channel), but each token in the new authentication mechanism includes all privileges (for example, joining a channel, host-in, and stream-pushing). |
Billing Optimization | Small video resolutions are charged according to the voice-only mode. For example, 16 × 16. |
v2.0
v2.0 is released on November 21, 2017.
Developers,
Due to the upgrade of Agora products, Windows SDK 2.0 no longer supports the Agora Recording Server version 1.8.2 and before, or relevant APIs.
This may affect:
Two solutions are available:
Migrate from the Agora Recording Server v1.8.2 or before to the Agora Recording SDK v1.12 or later. The Agora Recording SDK does not require recording on the client side and does not affect subsequent upgrades of the Windows SDK (recommended).
If you want to continue using the Agora Recording Server, you can use your current Windows SDK version (v1.14 and before). Contact Technical Support for any questions.
Adds the setRemoteVideoStreamType
and enableDualStreamMode
methods in the COMMUNICATION
profile to support dual streams.
Supports the external audio source in the COMMUNICATION
and LIVE_BROADCASTING
profiles by adding the following API methods:
Name | Description |
---|---|
setExternalAudioSource |
Sets the external audio source parameters, including the sample rate and channel number. |
pushExternalAudioFrame |
Pushes the external audio frame to the Agora SDK. |
Provides a set of RESTful APIs to ban a peer user from the server in the COMMUNICATION
and LIVE_BROADCASTING
profiles. Contact support@agora.io to enable the function, if required.
Supports Windows (x64).
Adds API methods such as setting the volume, mute states, and system volume settings.
Name | Description |
---|---|
setPlaybackDeviceVolume |
Sets the volume of the playback device. |
getPlaybackDeviceVolume |
Gets the volume of the playback device. |
setRecordingDeviceVolume |
Sets the volume of the microphone. |
getRecordingDeviceVolume |
Gets the volume of the microphone. |
setPlaybackDeviceMute |
Mutes the playback device. |
getPlaybackDeviceMute |
Gets the mute status of the playback device. |
setRecordingDeviceMute |
Mutes the microphone. |
getRecordingDeviceMute |
Gets the mute status of the microphone. |
onAudioDeviceVolumeChanged |
Notifies the application when the volume of the playback device, recording device, or application changes. |
v1.14
v1.14 is released on October 20, 2017.
setAudioProfile
method to set the audio parameters and scenarios.setLocalVoicePitch
method to set the local voice pitch.setInEarMonitoringVolume
method to adjust the volume of the in-ear monitor.v1.13.1
v1.13.1 is released on September 28, 2017 and optimizes the echo issue under certain circumstances.
v1.13
v1.13 is released on September 4, 2017.
onClientRoleChanged
callback to report on a user role switch between the host and audience in the interactive live streaming.Occasional crashes.
v1.12
v1.12 is released on July 25, 2017.
injectStream
method to inject an RTMP stream into the current channel in the LIVE_BROADCASTING
profile.aes-128-ecb
encryption mode in the setEncryptionMode
method.quality
parameter in the startAudioRecording
method to set the recording audio quality.ActiveSpeaker
method to report on the active speaker in the current channel.setScreenCaptureWindow
method, and updated the startScreenCapture
method to share the whole screen and specify the window or region in the COMMUNICATION
profile.COMMUNICATION
profile.COMMUNICATION
and LIVE_BROADCASTING
profiles. See Release Notes.Recording: Adds APIs for real-time video mixing and web recording. See Release Notes for the Recording SDK.
Android/iOS/macOS/Windows: In the COMMUNICATION
profile, an improvement for the 320 × 180 resolution profile: