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

Overview

The Voice SDK supports the following scenarios:

  • Voice communication
  • Live audio broadcast

For the key features included in each scenario, see Voice Overview and Audio Broadcast Overview.

Known Issues and Limitations

Privacy changes

If your app targets Android 9, you should keep the following behavior changes in mind. These updates to device serial and DNS information enhance user privacy.

Build serial number deprecation

In Android 9, Build.SERIAL is always set to "UNKNOWN" to protect users' privacy.
If your app needs to access a device's hardware serial number, you should instead request the READ_PHONE_STATE permission, then call getSerial().

DNS privacy

Apps targeting Android 9 should honor the private DNS APIs. In particular, apps should ensure that, if the system resolver is doing DNS-over-TLS, any built-in DNS client either uses encrypted DNS to the same hostname as the system, or is disabled in favor of the system resolver.

For more information about privacy changes, see Android Privacy Changes.

v3.0.0

v3.0.0 was released on Mar 4, 2020.

On Mar 24, 2020, we fixed occasional issues relating to no audio, audio mixing, multiple onClientRoleChanged callbacks, and SDK crashes.

In this release, Agora improves the user experience under poor network conditions for both the Communication and Live-broadcast profiles through the following measures:

  • Adopting a new architecture for the Communication profile.
  • Upgrading the last-mile network strategy for both the Communication and Live-broadcast 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.

Compatibility changes

Default log file path change

To avoid privilege issues, this release changes the default log file path from /storage/emulated/0/<package name>/ to /storage/emulated/0/Android/data/<package name>/files/.

New features

1. Multiple channel management

To enable a user to join an unlimited number of channels at a time, this release adds the RtcChannel and IRtcChannelEventHandler classes. By creating multiple RtcChannel 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 a live broadcast to adjust the playback volume of different remote users, or to repeatedly adjust the playback volume of the same remote user.

3. Agora Mediaplayer Kit

To enrich the playability of a live broadcast, Agora releases the Mediaplayer Kit plug-in, which supports the broadcaster playing local or online media resources and sharing them with all users in the channel during a live broadcast. See Mediaplayer Kit release notes for details.

Improvement

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-Broadcast 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 A sample rate of 32 KHz, music encoding, mono, and a bitrate of up to 44 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

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 CDN live streaming.

API changes

Behavior change

  • Calling enableLocalAudio(false) does not change the in-call volume to media volume.
  • When the device is connected to the earpiece or Bluetooth, calling setEnableSpeakerphone(true) does not route the audio to the speakerphone.

Added

Deprecated

v2.9.4

v2.9.4 was released on Feb 17, 2020.

This release fixed some abnormal behaviors on Android devices.

v2.9.2

v2.9.2 is released on Oct 18, 2019.

This release fixed crashes on some Android device.

v2.9.1

v2.9.1 is released on Sep 19, 2019.

New features

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. Removing the event handler

This release adds the removeHandler method to remove specified IRtcEngineEventHandler objects when you want to stop listening for specific events.

Improvements

1. 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.

2. Adding error codes

This release adds the following error codes in the ErrorCode class:

  • ERR_ALREADY_IN_USE(19)
  • ERR_WATERMARK_PATH(125)
  • ERR_INVALID_USER_ACCOUNT(134)
  • ERR_AUDIO_BT_SCO_FAILED(1030)
  • ERR_ADM_NO_RECORDING_DEVICE(1359)
  • ERR_VCM_UNKNOWN_ERROR(1600)
  • ERR_VCM_ENCODER_INIT_ERROR(1601)
  • ERR_VCM_ENCODER_ENCODE_ERROR(1602)
  • ERR_VCM_ENCODER_SET_ERROR(1603)

For detailed descriptions for each error, see Error Codes.

Issues fixed

Audio

  • A user makes a call after connecting to a Bluetooth device. After the call ends, the user watches YouTube and cannot hear any sound.
  • The audio route is different from the settings in the setEnableSpeakerphone method when Bluetooth is connected in the Communication profile.
  • Exceptions occur in the audio route when a user is in the channel.
  • The app crashes when using external audio sources in the push mode.
  • Audio freezes.
  • After turning off the Bluetooth headset, the audio route becomes the earpiece instead of the loudspeaker.
  • Echos occur when a user is in the channel.
  • Occasional noise occurs in the Live Broadcast profile.
  • The app fails to play online music files using the startAudioMixing method on devices running Android 10.

Miscellaneous

  • The OpenSSL version is outdated.

API Changes

Added

Deprecated

  • startAudioRecording

v2.9.0

v2.9.0 is released on Aug 16, 2019.

Compatibility changes

1. RTMP streaming

In this release, we deleted the following methods:

  • configPublisher

If your app implements CDN streaming 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 Push Streams to the CDN.

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 Broadcast 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 Broadcast 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:

  • 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.

5. Pulling the remote audio data

To improve the experience in audio playback, this release adds the following methods to pull the remote audio data. After getting the audio data, you can process it and play it with the audio effects that you want.

The difference between the onPlaybackFrame callback and the pullPlaybackAudioFrame method is as follows:

  • onPlaybackFrame: The SDK sends the audio data to the app once every 10 ms. Any delay in processing the audio frames may result in an audio delay.
  • pullPlaybackAudioFrame: The app pulls the remote audio data. After setting the audio data parameters, the SDK adjusts the frame buffer and avoids problems caused by jitter in external audio playback.

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

  • Reduces the earpiece delay.
  • 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.
  • The audience cannot hear the host after the host sets the in-ear monitoring volume to 0.
  • Failure to play the audio file by calling the startAudioMixing method.
  • The audio route cannot be set to Bluetooth on some devices.
  • Crashes occur when using the raw audio data.
  • The audio route does not conform to the default settings after the device disconnects from Bluetooth.

Miscellaneous

  • Occasionally mixed streams in CDN streaming.
  • Occasional crashes occur after joining the channel on some devices.

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

v2.8.2

v2.8.2 is released on Aug 1, 2019.

This release fixed the interoperating problem with the Agora Web SDK.

v2.8.1

v2.8.1 is released on Jul. 20, 2019.

New features

  • Support for the x86-64 architecture.
  • Support for Android Q.

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 communication, 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 or live broadcast, 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.

Issues Fixed

Audio

  • Occasional audio freezes.

Miscellaneous

  • When the log file path specified in the setLogFile method does not exist, no log file is generated and the default log file is incomplete.

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 12, 2019.

Compatibility changes

Ensure that you read the following SDK behavior changes if you migrate from an earlier SDK version.

Publishing streams to the CDN

To improve the usability of the CDN streaming service, v2.4.1 defines the following parameter limits:

Class / Interface Parameter Limit
LiveTranscoding
  • videoFramerRate: Frame rate (fps) of the CDN live output video stream. The value range is [0, 30], and the default value is 15. Agora adjusts all values over 30 to 30.
  • videoBitrate: Bitrate (Kbps) of the CDN live output video stream. The default value is 400. Set this parameter according to the Video Bitrate Table. If you set a bitrate beyond the proper range, the SDK automatically adapts it to a value within the range.
  • videoCodecProfile: The video codec profile. Set it as BASELINE, MAIN, or HIGH (default). If you set this parameter to other values, Agora adjusts it to the default value of HIGH.
  • width and height: Pixel of the video. The minimum value of width x height is 16 x 16.
    		•
    AgoraImage 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.

    New features

    1. State of the RTMP streaming

    v2.4.1 adds the onRtmpStreamingStateChanged callback to indicate the state of the RTMP streaming 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.

    2. 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).

    3. 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.

    4. 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.

    5. 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 remote user joins the channel and sends the audio stream.
    • The remote user stops sending the audio stream and re-sends it after 15 seconds.

    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.

    Improvements

    1. Playing multiple online audio effect files simultaneously

    v2.4.1 adds the support for playing multiple online audio effect files simultaneously by allowing you to call the playEffect method multiple times with the URLs of the online audio effect files.

    2. Reporting more statistics

    3. Miscellaneous

    • Improved the sound quality of the GAME_STREAMING audio scenario.
    • Reduced the audio latency.
    • Reduced the SDK package size by 0.5 M.
    • Improved the accuracy of the network quality after users change the video bitrate.
    • Enabled the audio quality notification callback by default, that is, enabled the onRemoteAudioStats callback without calling the enableAudioVolumeIndication method.
    • Improved the stability of CDN streaming services.

    Issues fixed

    Audio

    • The audio stream is played through the loudspeaker even after the user plugs in the earphone.
    • The user cannot hear the audio mixing file through Bluetooth in the single-broadcaster scenario.
    • Exceptions occur when playing the audio mixing file in the Live Broadcast profile.

    Miscellaneous

    • Users still receive the onNetworkQuality callback after leaving the channel.
    • Occasional crashes.
    • The app quits after calling joinChannel.

    API changes

    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

    Deprecated

    • enableAudioQualityIndication
    • The WARN_LOOKUP_CHANNEL_REJECTED(105) warning code. Use CONNECTION_CHANGED_REJECTED_BY_SERVER(10) in the onConnectionStateChanged callback instead.
    • The ERR_TOKEN_EXPIRED(109) error code. Use CONNECTION_CHANGED_TOKEN_EXPIRED(9) in the onConnectionStateChanged callback instead.
    • The ERR_INVALID_TOKEN(110) error code. Use CONNECTION_CHANGED_INVALID_TOKEN(8) in the onConnectionStateChanged callback instead.

    v2.4.0 and Earlier

    v2.4.0

    v2.4.0 is released on April 1, 2019.

    New features

    1. Voice changer and voice reverberation

    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. See Adjust the pitch and tone.

    2. Tracking the sound position of a remote user

    v2.4.0 adds the enableSoundPositionIndication and setRemoteVoicePosition methods. 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.

    3. Pre-call last-mile network probe test

    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).

    4. State of an audio mixing file

    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.

    5. Setting the log file size

    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).

    Improvements

    1. Accuracy of call quality statistics
    • v2.4.0 adds the intervalInSeconds parameter to the startEchoTest method, allowing you to set the interval between when you speak and when the recording plays back.
    • v2.4.0 adds three parameters to the 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.
    2. Core quality improvements
    • Reduces the audio delay.
    • Improves the video quality and stability.
    • Shortens the time to render the first remote video frame.
    • Reduces the time delay when playing through the earpiece and minimizes the echo.

    Issues Fixed

    Audio
    • Calling the enableLocalAudio method disconnects all connected Bluetooth devices.
    • The SDK does not support audio mixing URLs with Chinese characters.
    • Volume levels of the high-pitch sound are lowered.
    • Sounds are occasionally played fast.
    • The app cannot adjust the volume on some devices.
    Miscellaneous
    • The user drop-offline time between Android and iOS is not unified.
    • The SEI information does not synchronize with the media stream when publishing transcoded streams to the CDN.

    API Changes

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

    Added
    Deprecated
    • startEchoTest

    v2.3.3

    v2.3.3 is released on January 24, 2019.

    Issues Fixed

    • Occasional inaccurate statistics returned in the onNetworkQuality callback.
    • Occasional crashes on Huawei P9.

    v2.3.2

    v2.3.2 is released on January 16, 2019.

    Compatibility changes

    Besides the new features and improvements mentioned below, it is worth noting that v2.3.2:

    • Improves the SDK's ability to counter packet loss under unreliable network conditions.
    • Improves the communication smoothness.
    • Reduces video freezes in the Live Broadcast profile.

    Before upgrading your SDK, ensure that the version is:

    • Native SDK v1.11 or later.
    • Web SDK v2.1 or later.

    New features

    Independent audio mixing volume adjustments for local playback and remote publishing

    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.

    Improvements

    1. Improves the accuracy of the call quality statistics

    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.

    We plan 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.

    For the list of API methods related to the call quality statistics and on how and when to use them, see Report In-call Statistics.

    2. New network connection policy

    v2.3.2 adds the following API method and callback to get the current network connection state and the reason for a connection state change:

    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 we do not recommend using them.

    3. Improves the call rating system

    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 live broadcast. You can use this feedback for future product improvement. We strongly recommend integrating this method in your app.

    4. Other improvements
    • Minimizes packet loss under unreliable network conditions in the Live Broadcast profile.
    • Improves the stability in pushing streams.
    • Improves the performance of the SDK on Android 6.0 or later.
    • Optimizes the API calling threads.
    • Checks the headset and Bluetooth device connection.
    • Reduces the audio delay.

    Issues Fixed

    The following issues are fixed in v2.3.2:

    SDK
    • Crashes on emulators, such as Yeshen and mumu.
    • Crashes on Android 6.0+ due to x86 .so relocation.
    Audio
    • A user joins a live broadcast with a Bluetooth headset. The audio is not played through the Bluetooth headset when the user leaves the channel and opens another app.
    • Crashes when calling the startAudioMixing method to play music files.
    • A previously disabled microphone becomes enabled when the device connects to a headset.
    • On Huawei Mate 20 X, a remote user cannot hear any voice when the app switches to the background and the user opens another app.
    • Echo on Pixel 2.
    • Cannot adjust the volume of the speaker when users change roles, join and leave channels, or a system phone or Siri interrupts.
    • Users do not hear any voice for a while when an app switches back from the background.

    API Changes

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

    Added:
    Deprecated

    v2.3.1

    v2.3.1 is released on October 12, 2018.

    New features

    Disables/Re-enables the Local Audio Function

    When a user joins a channel, the audio function is enabled by default.
    To receive audio streams without sending any audio stream after joining a channel, this version adds the enableLocalAudio method is to disable or re-enable the local audio function.
    Once the local audio function is disabled or re-enabled, the SDK returns the onMicrophoneEnabled callback, and the local audio capturing stops.
    This method does not affect receiving or playing the remote audio streams.

    The difference between this method and the muteLocalAudioStream method is that the enableLocalAudio method does not capture or send any audio stream, while the muteLocalAudioStream method captures but does not send audio streams.

    Issues Fixed

    • Live-broadcast profile: Delay at the client due to incorrect statistics.
    • Live-broadcast profile: Occasional crashes on some Android devices after a user repeats the process of switching roles between BROADCASTER and AUDIENCE.
    • Occasionally on some Android devices, a user hears a popping sound if the user leaves the channel at the same time another user is speaking.

    v2.3.0

    v2.3.0 is released on August 31, 2018.

    Compatibility changes

    • From v2.3.0, the LiveTranscoding class is moved from the io.agora.live package to the io.agora.rtc.live package.

    • Fixed a typo in the constants.java API in v2.3.0.

      • Before:
      public static final int SOFEWARE_ENCODER = 1;
      • After:
      public static final int SOFTWARE_ENCODER = 1;

    • The security keys are improved and updated in v2.1.0. If you are using an Agora SDK version below v2.1.0 and wish to migrate to the latest version, see Token Migration Guide.

    New features

    1. Notifies the user that the token expires in 30 seconds

    The SDK returns the onTokenPrivilegeWillExpire callback 30 seconds before a token expires to notify the app to renew it. When this callback is received, you need to generate a new token on your server and call the renewToken method to pass the newly-generated token to the SDK.

    2. Returns user-specific upstream and downstream statistics, including the bitrate, frame rate, packet loss rate and time delay

    The onRemoteAudioTransportStats callback is added to provide user-specific upstream and downstream statistics, including the bitrate, frame rate, and packet loss rate. During a call or a live broadcast, the SDK triggers these callbacks once every two seconds after the user receives audio/video packets from a remote user. The callbacks include the user ID, audio bitrate at the receiver, packet loss rate, and time delay (ms).

    Improvements

    • Improves the quality for one-on-one voice/video scenarios with optimized latency and smoothness, especially for areas like Southeast Asia, South America, Africa, and the Middle East.
    • Improves the audio encoder efficiency in a live broadcast to reduce user traffic while ensuring the call quality.
    • Improves the audio quality during a call or a live broadcast using the deep-learning algorithm.

    Issues Fixed

    • Excessive increase in memory usage when multiple delegated hosts broadcast in the channel.
    • Occasional crashes on some Android devices.
    • Occasional crashes after interoperating with devices of other platforms for some Android devices.
    • Excessive increase in the memory usage for the host when the host frequently joins and leaves a channel that has multiple delegated hosts.
    • Occasionally, the remote user cannot hear the host when the host switches between AUDIENCE and BROADCASTER.
    • Occasionally, the destroy method does not respond after a user enables the video and joins a channel.
    • Occasional crashes on Android devices when remote users frequently join and leave the channel.
    • Occasionally, the audience cannot adjust the channel volume.
    • Occasional crashes when one of the two broadcasters mutes or disables the local audio while playing the background music.
    • Occasional crashes on some devices when preloading the sound effects.
    • Failure to enable the hardware encoder on some Android devices.
    • The host cannot receive the audio/video stream of the delegated host on some Android devices.
    • Occasional crashes on some Android devices when a user frequently changes the token.
    • Occasional inter-operational failures between SIP devices and the SDK.
    • Occasional echo issues when using a specific audio card.

    API Changes

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

    To avoid adding too many users with the same uid into the CDN publishing channel, the following API method is deleted in v2.3.0, and the return value type of addUser is changed from void to int.

    • setUser

    The following API methods are deleted and no longer supported in v2.3.0. Agora provides the Recording SDK for better recording services. For more information on the Recording SDK, see Release Notes for Agora Recording SDK.

    • startRecordingService
    • stopRecordingService
    • refreshRecordingServiceStatus

    The following deprecated API methods are deleted and no longer supported from v2.3.0:

    • monitorConnectionEvent
    • monitorBluetoothHeadsetEvent
    • monitorHeadsetEvent
    • setPreferHeadset
    • switchView
    • setSpeakerphoneVolume

    v2.2.3

    v2.2.3 is released on July 5, 2018.

    Compatibility changes

    The security keys are improved and updated in v2.1.0. If you are using an Agora SDK version below v2.1.0 and wish to migrate to the latest version, see Token Migration Guide.

    Issues Fixed

    • Occasional online statistics crashes.
    • The broadcaster’s voice distorts occasionally on some Android devices.
    • Occasional crashes during a live broadcast.
    • Excessive increase in the memory usage when multiple delegated hosts broadcast in the channel.
    • Receiving the onLeaveChannel callback long after a user has left the channel on some Android devices.
    • Failing to report the uid and volume of the speaker in a channel.
    • Unsteady voice volume of the broadcaster in a live broadcast.

    v2.2.2

    v2.2.2 is released on June 21, 2018.

    Issues Fixed

    • Fixed occasional online statistics crashes.
    • Fixed occasional audio crashes on some Android devices.
    • Fixed the issue that the broadcaster’s voice distorts occasionally on some Android devices.
    • Fixed the issue of failing to report the uid and volume of the speaker in a channel.
    • Fixed the issue of receiving the onLeaveChannel callback long after a user has left the channel on some Android devices.

    v2.2.1

    v2.2.1 is released on May 30, 2018.

    Issues Fixed

    • Occasional crashes during gaming on some Android devices.
    • The soundtrack pointer cannot be retrieved on some Android devices.
    • Occasional crashes on some Android devices.
    • The audio volume on some Android devices cannot be adjusted after a headset is plugged in.

    v2.2.0

    v2.2.0 is released on May 4, 2018.

    New features

    1. Play the audio effect in the channel

    Adds a publish parameter in the playEffect method for 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.

    2. Deploy the proxy at the server

    We provide a proxy package for enterprise users with corporate firewalls to deploy before accessing our services.

    Improvements

    1. Audio volume indication

    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.

    2. Network quality detection during a session

    To meet the customers’ need for real-time network quality detection in the channel, the onNetworkQuality method improves its data accuracy.

    3. Last mile network quality detection before joining a channel

    To test if the customers’ network condition can support voice or video calls before joining the channel, the onLastmileQuality callback changes the detection from a fixed bitrate to the bitrate set by the customer in the setVideoProfile method to improve data accuracy. When the network condition is unknown, the SDK triggers this callback once every two seconds.

    4. Audio quality enhancement

    Improves the audio quality in scenarios that involve music playback.

    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-broadcast profile. The bitrate values in v2.1.3 stay consistent with those in v2.0.

    Issues Fixed

    Occasional recording failures on some phones when a user leaves a channel and turns on the built-in recording device.

    v2.1.2

    v2.1.2 is released on April 2, 2018.

    Issues Fixed

    Video freeze in DTX + AAC mode.

    v2.1.1

    v2.1.1 is released on March 16, 2018.

    Agora has identified a critical issue 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.

    New features

    1. Voice Optimization

    Adds a scenario for the game chat room to reduce the bandwidth and cancel the noise with the setAudioProfile method.

    2. Enhance the audio effect input from the built-in microphone

    In an interactive broadcast scenario, the host can enhance the local audio effects from the built-in microphone with the setLocalVoiceEqualization and setLocalVoiceReverb methods by implementing the voice equalization and reverberation effects.

    3. Online statistics query

    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. For details, see Online Statistics Query API.

    Improvements

    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, hosting in, and stream-pushing).

    Issues Fixed

    • Occasional playback noise on specific devices.
    • Occasional crackling voice playback on specific devices.
    • Occasional crashes.

    v2.0.2

    v2.0.2 is released on December 15, 2017, and fixes occasional audio routing issues.

    v2.0

    v2.0 is released on December 6, 2017.

    New Features

    • Supports external audio sources in the Communication and Live-broadcast profiles by adding the following API methods:

      Name Description
      setExternalAudioSource Enables the external audio source function.
      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-broadcast profiles. Contact sales-us@agora.io to enable this function, if required.

    • Supports the following Android emulators: NOX, Lightning, and Xiaoyao.

    Issues Fixed

    • Audio routing and Bluetooth issues.
    • Optimizes the volume balance control.

    v1.14

    v1.14 is released on October 20, 2017.

    New Features

    • Adds the setAudioProfile method to set the audio parameters and scenarios
    • Adds the setLocalVoicePitch method to set the local voice pitch
    • Live Broadcast: Adds the setInEarMonitoringVolume method to adjust the volume of the in-ear monitor

    Improvements

    • Optimizes the audio at high bitrates.
    • Live Broadcast: The audience can view the host within one second in a single-stream mode (226 ms on average, and 204 ms under good network conditions).
    • Adds the ability to reduce the bandwidth.
      • Before v1.14: If you muted the audio of a specific user, the network still sent the stream.
      • Starting from v1.14: If you mute the audio of a specific user, the network will not send the stream of the user to reduce the bandwidth.

    Issues Fixed

    Camera related issues on Android devices.

    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.

    New Features

    • Adds the function to dynamically enable and disable acquiring the sound card in a live broadcast.
    • Adds the function to disable the audio playback.
    • Adds the onClientRoleChanged callback to report to the app on a user role switch between the host and the audience in a live broadcast.
    • Supports the push-stream failure callback on the server side.

    Issues Fixed:

    Occasional crashes on some devices.

    v1.12

    v1.12 is released on July 25, 2017.

    New Features:

    • Adds the aes-128-ecb encryption mode in the setEncryptionMode method.
    • Adds the quality parameter in the startAudioRecording method to set the recording audio quality.
    • Adds a set of APIs for audio effect management.

    Issues Fixed:

    • Android: Bluetooth issues related to audio routing.
    • Android/iOS/Mac/Windows: Occasional crashes.