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Release Notes

Last updated 2022/11/08 06:58:39

This page provides the release notes for the Agora Voice SDK for Windows.

v3.7.1

v3.7.1 was released on August 4, 2022.

Improvements

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.

Fixed issues

This release fixed the following issues:

  • Occasionally, the app got stuck after a user joined a channel.
  • The local user received the 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.
  • Occasionally, it took users longer than normal to join a channel for the first time.

v3.7.0.2

v3.7.0.2 was released on June 6, 2022 and improved the stability of the SDK.

v3.7.0

v3.7.0 was released on April 11, 2022.

New features

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. 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.

3. 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.

4. 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.

Improvements

1. Channel capacity improvement

As of this release, a single channel can support up to 128 concurrent online hosts, who can publish audio 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. 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

3. playEffect improvements

This release improves the internal implementation logic of playEffect to avoid blocking and reduce freezing when playing an audio effect file.

4. Audio experience optimization

This release reduces audio stuttering under network conditions with burst jitter and high packet loss.

5. Transmission upgrade

This release upgrades transmission protocols and algorithms to enhance the SDK's ability to counter poor network conditions.

API changes

Added

  • enableLocalVoicePitchCallback
  • onLocalVoicePitchInHz
  • onClientRoleChangeFailed
  • CONNECTION_CHANGED_SAME_UID_LOGIN(19) and CONNECTION_CHANGED_TOO_MANY_BROADCASTERS(20) in CONNECTION_CHANGED_REASON_TYPE

Deprecated

  • Error code ERR_SET_CLIENT_ROLE_NOT_AUTHORIZED(119). Use CLIENT_ROLE_CHANGE_FAILED_REASON instead.

v3.6.2

v3.6.2 was released on February 22, 2022.

Improvements

1. Cloud proxy

To enrich application scenarios of the Agora Cloud Proxy, this release updates the cloud proxy types (CLOUD_PROXY_TYPE) as follows:

  • Changes the meaning of 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.
  • Adds 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:

  • Adds onAudioDeviceTestVolumeIndication, which reports the volume information of the audio device being tested.
  • Changes call sequence of 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.
Calling 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.

  • By adding 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.
  • By adding 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:

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.

Fixed issues

This release fixed the issue that remote users could hear the local user with echoes when the local user was in a specific reverb environment.

API changes

Added

  • onAudioDeviceTestVolumeIndication
  • recordingChannel in AudioRecordingConfiguration
  • AUDIO_RECORDING_QUALITY_ULTRA_HIGH(3) in AudioRecordingConfiguration.recordingQuality
  • onProxyConnected
  • TCP_PROXY(2) in CLOUD_PROXY_TYPE

Modified

  • Timing restrictions for the startRecordingDeviceTest, startPlaybackDeviceTest, and startAudioDeviceLoopbackTest method calls.
  • The behavior of NONE_PROXY(0) in CLOUD_PROXY_TYPE.

v3.6.1.1

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

v3.6.1 was released on January 14, 2022.

This release fixed the following issues:

  • The CDN live streaming failed because of the abnormal resolution of the RTMP streaming URL with a port number.
  • Crashes occurred on some devices running on Windows 7.

v3.6.0

v3.6.0 was released on December 7, 2021.

Compatibility changes

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.
This release deprecates three old methods for Media Push: 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.
      To ensure the quality of the Media Push, make sure to handle the business logic according to the state code, error code, and event code.

  • 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).

Fixed issues

This release fixed the issue: When the local user called enableLoopbackRecording to enable loopback audio capturing on some devices, remote users heard the audio being played back twice.

API changes

Added

  • startRtmpStreamWithoutTranscoding
  • startRtmpStreamWithTranscoding
  • updateRtmpTranscoding
  • stopRtmpStream
  • RTMP_STREAM_PUBLISH_STATE_DISCONNECTING(5) in RTMP_STREAM_PUBLISH_STATE
  • The following enumerators in 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)
  • The following enumerators in RTMP_STREAMING_EVENT:
    • RTMP_STREAMING_EVENT_ADVANCED_FEATURE_NOT_SUPPORT(3)
    • RTMP_STREAMING_EVENT_REQUEST_TOO_OFTEN(4)
  • AUDIO_CODEC_PROFILE_HE_AAC_V2(2) in AUDIO_CODEC_PROFILE_TYPE
  • The following methods in the IAudioDeviceCollection class:
    • getDefaultDevice
    • followSystemPlaybackDevice
    • followSystemRecordingDevice

Modified

  • Renamed the enumeration RTMP_STREAM_PUBLISH_ERROR to RTMP_STREAM_PUBLISH_ERROR_TYPE

v3.5.2

v3.5.2 was released on November 25, 2021.

Compatibility changes

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:

  • When a user who has already joined an IRtcEngine channel calls the joining channel method of the IRtcEngine class with a valid channel name.
  • When a user who has already joined an 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.

New features

Testing audio call loop

To check whether the local audio device and network conditions can guarantee the proper sending and receiving of audio 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 device and network conditions are working properly.

Compared to startEchoTest [2/3], startEchoTest [3/3] can secure the test, but cannot support setting the time interval for reporting test results.

Fixed issues

This release fixed the following issues:

  • Remote users heard slightly unsynchronized voice and music when a user called startAudioMixing on some Android devices.
  • The system version of Windows was incorrectly recognized on some devices.

API changes

This release adds the startEchoTest [3/3] method.

v3.5.1

v3.5.1 was released on October 14, 2021.

New features

1. 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.

2. 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.

3. 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.

4. 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.

Improvements

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:

  • Adds 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.
  • Supports calling enableLastmileTest or startLastmileProbeTest to test the connection quality of the 5G mobile network.

3. Other improvements

This release also provides the following improvements:

  • Adds continuous optimization of the portrait edge effect of the virtual background.
  • In scenarios with high CPU usage, the audio experience is improved when the sound card is used for audio capture.
  • Adds support for audio playback devices with S/PDIF (Sony/Philips Digital Interface).
  • After enabling noise suppression, the audio experience is improved when using some Bluetooth headsets.

Fixed issues

This release has fixed the following issues:

  • In the GAME_STREAMING scenario, occasional echo or noise that was caused by inaccurate music detection occurred.
  • When the local user called muteAllRemoteAudioStreams and received onRemoteAudioStateChanged(REMOTE_AUDIO_REASON_LOCAL_MUTED), onRemoteAudioStateChanged(REMOTE_AUDIO_REASON_REMOTE_MUTED) was incorrectly received after 15 seconds.
  • The SDK repeatedly triggered the onAudioDeviceStateChanged callback when the user connected or disconnected an audio device.
  • As of v3.4.5, the return value was inaccurate when you called joinChannel multiple times in a row.
  • The method call of setExternalAudioSource before joining the channel did not take effect.
  • Echoes could occur when you used a fiber optic speaker.
  • The SDK additionally triggered the onLocalAudioStateChanged callback to report LOCAL_AUDIO_STREAM_STATE_CAPTURING(1) when you called enableLocalAudio(false) after joining a channel.

API changes

Added

Deprecated

  • pushAudioFrame[2/3]
  • getAudioMixingDuration

v3.5.0.4

v3.5.0.4 was released on September 26, 2021.

This release fixed the following issues:

  • When you called 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.
  • When using previously released 3.5.x versions of the SDK, the local user could not be heard by remote users when the local user used custom audio capture by the Push method.

v3.5.0.3

v3.5.0.3 was released on September 6, 2021 and improved the stability of the SDK.

v3.5.0.1

v3.5.0.1 was released on August 4, 2021 and improved the stability of the SDK.

v.3.5.0

v3.5.0 was released on July 20, 2021.

Improvements

1. 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.

2. Additional improvement

In scenarios where the AGC is enabled, as of this release, the convergence time for noise reduction is shortened, which reduces noise fluctuations.

Fixed issues

This release fixed the following issues:

  • After a remote user called enableLoopbackRecording, occasionally the local user heard an echo of their own audio.
  • After calling 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.
  • When the local network quality was poor, the downlink network quality rating (rxQuality) of remote users reported by the SDK was not accurate.

API changes

This release added the error code: ERR_ADM_WIN_CORE_SERVRE_SHUT_DOWN (1735).

v3.4.6

v3.4.6 was released on July 15, 2021 and improved the stability of the SDK.

v3.4.5

v3.4.5 was released on June 22, 2021.

Compatibility changes

1. Support for GCM2 encryption

To further improve the security during real-time audio 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.
  • The 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).

See Media Stream Encryption.

Once you enable the built-in encryption, all users in the same channel must use the same encryption mode and key, including the server-side users such as the Agora recording service.

2. Audio stream publishing behavior changes

To flexibly control the publishing state in multiple channels, this release optimizes the IChannel class as follows:

  • Deprecates the publish and unpublish methods, and adds muteLocalAudioStream instead. You can set the publishing state of the audio stream.
  • Adds the publishLocalAudio member 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 to false when the user joins other channels; otherwise, the user fails to join the channel.
  • After calling setClientRole(BROADCASTER), the local user publishes audio stream by default. You no longer need to call publish.
publishLocalAudio = false serves the same function as muteLocalAudioStream(true). If you call them together, the one called later takes effect.

The above improvements bring the following changes in the IRtcEngine class:

  • muteLocalAudioStream of IRtcEngine does not take effect for channels created by the IChannel class, so you need to use the muteLocalAudioStream of the IChannel class instead.
  • When a user joins a channel by using joinChannel with the options parameter, you can set the publishing state.
  • If a user joins a channel by calling joinChannel with the options parameter, muteLocalAudioStream 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, 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.

Improvements

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:

  • To 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).
  • To 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. Media device state

To report media device states more accurately, this release optimizes the enumerations of MEDIA_DEVICE_STATE_TYPE as follows:

  • Changes the definition of 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.
  • Adds MEDIA_DEVICE_STATE_IDLE (0), which reports the device is ready for use.

MEDIA_DEVICE_STATE_TYPE is reported in the onAudioDeviceStateChanged callback.

4. 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).

Fixed issues

This release fixed the following issues:

  • Under poor network conditions, when you called startAudioMixing to play an online music file, occasionally the SDK response time was too long and caused a freeze.
  • Under poor network conditions, users were occasionally unable to switch roles.

API changes

Added

Modified

Deprecated

  • publish and unpublish in IChannel

v3.4.3

v3.4.3 was released on June 16, 2021. This release fixed the following issues:

  • The startAudioMixing method could not play the MP3 local music files with the .m4a filename extension.
  • After a user of the SDK v3.0.0+ left a communication channel that had any user of the SDK versions earlier than 3.0.0 and switched the channel profile to live streaming, the user of the SDK v3.0.0+ set user role to host before joining the next channel, but the client role setting did not take effect.
  • Under certain circumstances, an audio delay occasionally occurred 10 minutes after a user joined the channel.

v3.4.2

v3.4.2 was released on May 12, 2021. This release fixed occasional crashes after integrating the libagora_ai_denoise_extension.dll library.

v3.4.1

v3.4.1 was released on April 22, 2021. This release fixed this issue: Crashes occurred in a certain voice conversion scenario.

v3.4.0

v3.4.0 was released on April 16, 2021.

Compatibility changes

Integration changes

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

  1. Music file playback state

To monitor the reason why the playback state of a music file changes, this release modifies the onAudioMixingStateChanged callback as follows:

  • Replaces the errorCode parameter with the reason parameter.
  • Deprecates 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.
  • Modifies some logic of reporting 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.

  1. Volume control of microphone

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.

New features

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.

Improvements

1. Playback progress of music files

To control the playback progress of music files, this release is optimized as follows:

  • Deprecates the 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.
  • Deprecates the 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.

2. 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.

3. Media device errors

To help users better understand the cause of local audio errors, this release adds the following new error codes to 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.

Fixed issues

This release fixed this issues: the onAudioVolumeIndication callback returned incorrect volume information after you called enableSoundPositionIndication(true).

API changes

Added

Modified

Deprecated

  • getAudioMixingDuration[1/2]
  • startAudioRecording[2/3]
  • playEffect[1/2]
  • startAudioMixing[1/2]
  • AUDIO_MIXING_ERROR_TYPE

v3.3.2

v3.3.2 was released on March 29, 2021. This release fixed the issue where the local user heard noises from the remote user when a remote user was silent for a long time under certain circumstances.

v3.3.1

v3.3.1 was released on March 4, 2021.

New features

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.

Improvements

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.
Once you enable the built-in encryption, all users in the same channel must use the same encryption mode and key, including the server-side users such as the Agora recording service.

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.

API changes

This release adds the following APIs:

v3.3.0

v3.3.0 was released on January 22, 2021.

Compatibility changes

1. Integration change

This release adds the Agora basic calculation framework libagora-core.dll. To integrate the SDK into your project, see Integrate the SDK.

2. Changes to subscribing behavior

This release deprecates setDefaultMuteAllRemoteAudioStreams 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.
  • Methods with the prefix 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.
  • Methods with the prefix muteAll set whether to subscribe to the audio 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.

New features

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 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.

Before enabling noise suppression, integrate the 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. 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.

Improvements

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.

Fixed issues

This release fixed the following issues:

  • Occasionally, local users could not play remote audio streams.
  • 360 Brain of Security (online shopping mode) automatically blocked the application program from running.

API changes

Added

Deprecated

  • setDefaultMuteAllRemoteAudioStreams
  • setLogFile
  • setLogFileSize
  • setLogFilter
  • createDataStream

v3.2.1

v3.2.1 was released on December 17, 2020. This release fixed the following issues:

  • Crashes occurred when using IPv6.
  • When Native clients integrated with the Agora Voice SDK interoperated with Web clients, the Web clients continuously reported the Client.on(disable-local-video) or Client.on(mute-video) callback.

v3.2.0

v3.2.0 was released on November 30, 2020.

Compatibility changes

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.

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.

Transport layer encryption affects the size of the SDK package. For details, see Product Overview.

New features

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:

  • Set the latency level as ultra low latency to use Interactive Live Streaming Premium.
  • Set the latency level as low latency to use Interactive Live Streaming Standard.

For details, see the product overview of Interactive Live Streaming Standard.

Improvements

1. Meeting scenario

To improve the audio experience for multi-person meetings, this release adds AUDIO_SCENARIO_MEETING(8) in setAudioProfile.

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. Interactive streaming delay

This release reduces the latency on the audience's client during an interactive live streaming by about 500 ms.

API changes

Added

Deprecated

  • setLocalVoiceChanger
  • setLocalVoiceReverbPreset

v3.1.1

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

v3.1.0 was released on Aug 11, 2020.

New features

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.
  • onAudioSubscribeStateChanged: Reports the change of the audio subscribing state.

2. First local frame published callback

This release adds the onFirstLocalAudioFramePublished callback to report that the first audio 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.

Improvement

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. 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.

3. Encryption

This release adds the enableEncryption method for enabling built-in encryption, and deprecates the following methods:

  • setEncryptionSecret
  • setEncryptionMode

4. More in-call statistics

This release adds the following attributes to provide more in-call statistics:

  • Adds 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.
  • Adds publishDuration in RemoteAudioStats, which represents the total publish duration (ms) of the remote media stream.

5. 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
  • For the interactive streaming profile: 64 Kbps
  • For the communication profile: 16 Kbps
    		•
  • For the interactive streaming profile: 52 Kbps
  • For the communication profile: 16 Kbps
    		•
    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

    6. 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.

    Issues fixed

    This release fixed the occasional howling issue on some Huawei laptops.

    API changes

    Added

    Deprecated

    • setEncryptionSecret
    • setEncryptionMode
    • onFirstLocalAudioFrame

    Deleted

    • Warning code: WARN_ADM_IMPROPER_SETTINGS(1053)

    v3.0.1

    v3.0.1 was released on May 27, 2020.

    New features

    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. Data post-processing in multiple channels

    This release adds support for post-processing remote audio data in a multi-channel scenario by adding isMultipleChannelFrameWanted and onPlaybackAudioFrameBeforeMixingEx in the IAudioFrameObserver class.

    After successfully registering the audio observer, if you set the return value of isMultipleChannelFrameWanted as true, you can get the corresponding audio data from onPlaybackAudioFrameBeforeMixingEx. In a multi-channel scenario, Agora recommends setting the return value as true.

    Fixed issues

    • Audio mixing issues and abnormal loopback test results.
    • Inaccurate report of the onClientRoleChanged callback, authentication with an App ID and token, and a garbled log directory.

    API changes

    Added

    v3.0.0.2

    v3.0.0.2 was released on Apr 22, 2020.

    New features

    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:

    • When the app that integrates the Agora SDK is used within the specified area, it connects to the Agora servers within the specified area under normal circumstances.
    • When the app that integrates the Agora SDK is used out of the specified area, it connects to the Agora servers either in the specified area or in the area where the SDK is located.

    API changes

    Added

    areaCode member in the RtcEngineContext struct

    v3.0.0

    v3.0.0 was released on Mar 5, 2020.

    In this release, Agora improves the user experience under poor network conditions for both the COMMUNICATIONand LIVE_BROADCASTING profiles through the following measures:

    • Adopting a new architecture for the COMMUNICATION profile.
    • Upgrading the last-mile network strategy for both the COMMUNICATIONand 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.

    New features

    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. 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.

    Improvements

    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. 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

    3. Others

    This release enables interoperability between the RTC Native SDK and the RTC Web SDK by default, and deprecates the enableWebSdkInteroperability method.

    Issues fixed

    • Audio issues relating to audio mixing, audio encoding, and echoing.
    • Other issues relating to app crashes, log file, and unstable service during Media Push.

    API changes

    Added

    Deprecated

    v2.9.1

    As of December 8, 2021, Agora begins implementing the retirement process for the Voice SDK and Video SDK for native platforms and third-party frameworks earlier than v3.0.0. See Service Sunset Plans.

    v2.9.1 is released on Sep 19, 2019.

    New features

    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.

    Improvements

    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.

    Issues fixed

    Miscellaneous

    A typo in the IAgoraRtcEngine.h file.

    API changes

    To improve the user experience, we made the following changes in v2.9.1:

    Added

    Deprecated

    • startAudioRecording

    v2.9.0

    v2.9.0 is released on Aug 16, 2019.

    Compatibility changes

    1. Media Push

    In this release, we deleted the following methods:

    • configPublisher

    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:

    2. 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.

    New features

    1. Faster switching to another channel

    This release adds the switchChannel method to enable the audience in a LIVE_BROADCASTING 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.

    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:

    • The local audio: STOPPED(0), RECORDING(1), ENCODING(2), or FAILED(3). When the state is FAILED(3), see the error parameter for troubleshooting.
    • The remote audio: STOPPED(0), STARTING(1), DECODING(2), FROZEN(3), or FAILED(4). See the 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.

    Improvements

    1. Reporting more statistics of the in-call quality

    This release adds the following statistics in the RtcStats class:

    • RtcStats: The total number of the sent audio bytes and received audio bytes during a session.

    2. Other Improvements

    • Improves the audio quality when the audio scenario is set to Game Streaming.
    • Improves the audio quality after the user disables the microphone in the COMMUNICATION profile.

    Issues fixed

    Audio

    • When interoperating with a Web app, voice distortion occurs after the native app enables the remote sound position indication.
    • Invalid call of the muteRemoteAudioStream method.
    • Occasionally no audio.

    Miscellaneous

    • Occasionally mixed streams in Media Push.
    • Occasional crashes occur.
    • Failure to join the channel.

    API changes

    To improve the user experience, we made the following changes in v2.9.0:

    Added

    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.

    Deleted

    • configPublisher

    v2.8.0

    v2.8.0 is released on Jul. 8, 2019.

    New features

    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.

    Note:

    • Do not mix parameter types within the same channel. The following Agora SDKs support string user accounts:

      • The Native SDK: v2.8.0 and later.
      • The Web SDK: v2.5.0 and later.

      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 statistics

    To monitor the audio transmission quality during a call interactive live streaming, this release adds the totalFrozenTime and frozenRate members in the RemoteAudioStats class, to report the audio freeze time and freeze rate of the remote user.

    This release also adds the numChannels, receivedSampleRate, and receivedBitrate members in the RemoteAudioStats class.

    Improvements

    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.

    API changes

    To improve your experience, we made the following changes to the APIs:

    Added

    Deprecated

    v2.4.1

    V2.4.1 is released on Jun 12th, 2019.

    This is the first release of the Agora Voice SDK for Windows. Refer to the following guides to quickly integrate the SDK and enable real-time communication in your project.

    If you migrate to this SDK from the Windows Video SDK, refer to the Release notes for the Windows video SDK for audio improvements.