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

Overview

The Voice SDK supports the following scenarios:

  • Voice call
  • Live interactive audio streaming

For the key features included in each scenario, see Agora Voice Call Overview and Agora Live Interactive Audio Streaming Overview.

v3.0.1

v3.0.1 was released on May 27, 2020.

Compatibility changes

Dynamic library

This release replaces the static library with a dynamic library for the following reasons:

  • Improving overall security.
  • Avoiding incompatibility issues with other third-party libraries.
  • Making it easier to upload the app to the App Store.

To upgrade the RTC Native SDK, you must re-integrate the dynamic library, AgoraRtcKit.framework. This process should take no more than five minutes. See Migration Guide.

Apple supports the dynamic library on iOS 13.4 and later.

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:

  • AgoraAudioVoiceChanger adds several elements that have the prefixes AgoraAudioVoiceBeauty and AgoraAudioGeneralBeautyVoice. The AgoraAudioVoiceBeauty elements enhance the local voice, and the AgoraAudioGeneralBeautyVoice enumerations add gender-based enhancement effects.
  • AgoraAudioReverbPreset adds the enumeration AgoraAudioReverbPresetVirtualStereo and several enumerations that have the prefix AgoraAudioReverbPresetFx. The AgoraAudioReverbPresetVirtualStereo enumeration implements reverberation in the virtual stereo, and the AgoraAudioReverbPresetFx enumerations implement additional enhanced reverberation effects.

See Set the Voice Changer and Reverberation Effects for more information.

3. Data post-processing in multiple channels (C++)

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.

Improvements

  • Improves in-call audio quality. When multiple users speak at the same time, the SDK does not decrease volume of any speaker.
  • Reduces overall CPU usage of the device.

Fixed issues

  • This release fixed issues with inaccurate report of the didRemoteAudioStateChanged callback, no audio, audio mixing and audio freezing.
  • This release fixed issues that failure to end a call, and inaccurate report of the didClientRoleChanged callback.

API changes

This release adds the following APIs:

v3.0.0.2

v3.0.0.2 was released on Apr 22, 2020.

New features

Specify the area of connection

This release adds sharedEngineWithConfig for specifying the area of connection when creating an AgoraRtcEngineKit 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.

Issues fixed

This release fixed the occasional failure to connect to a Bluetooth headset.

API changes

Added

sharedEngineWithConfig

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 didClientRoleChanged callbacks, and SDK crashes.

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

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

Renaming the static library and adding support for dynamic library

To unify the library names across platforms, this release renames the library from AgoraRtcEngineKit.framework to AgoraRtcKit.framework. If you upgrade your SDK to v3.0.0, you must re-import the AgoraRtcKit class. For details, see Import the class in the Quickstart.

To improve your development experience, this release also adds support for the dynamic library. You can integrate either the static or the dynamic library in your project, and the name of the dynamic library package is Agora_Native_SDK_for_iOS_v3_0_0_VOICE_Dynamic.

Integrating the dynamic library has the following advantages:

  • The overall security level is improved.
  • Incompatibility issues with other third-party libraries are avoided.
  • Uploading the app onto App Store is easier.

If you prefer the dynamic library, you need to re-integrate the SDK and re-import the AgoraRtcKit class. This process should take no more than five minutes. See Integrate the SDK and Import the class in the Quickstart.

The following table shows the difference in the file size when generating ipa files with a dynamic and static library:
Library type ipa size (M) Decompressed ipa size (M) Frameworks folder size (M) Binary file size (M) Total size of frameworks folder + binary file (M)
Dynamic library 27 56.6 44 2.4 46.4
Static library 26.5 55.3 30.1 15.1 45.2

The dynamic library is located in the framework folder as an independent library. Note that the corresponding binary file size does not include the SDK size. Overall, this decreases the binary file size by 12.7 M and increases the framework folder size by 13.9 M.

New features

1. Multiple channel management

To enable a user to join an unlimited number of channels at a time, this release adds the AgoraRtcChannel and AgoraRtcChannelDelegate classes. By creating multiple AgoraRtcChannel objects, a user can join the corresponding channels at the same time.

After joining multiple channels, users can receive the audio 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 live interactive streaming 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 the live interactive streaming, Agora releases the Mediaplayer Kit plug-in, which supports the host playing local or online media resources and sharing them with all users in the channel during the interactive streaming. See Mediaplayer Kit release notes for details.

Improvements

1. Audio profiles

To meet the need for higher audio quality, this release adjusts the corresponding audio profile of AgoraAudioProfileDefault(0) in the LiveBroadcasting profile.

SDK AgoraAudioProfileDefault(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 52 Kbps.

2. Quality statistics

Adds the following members in the AgoraChannelStats 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 Native SDK and the Web SDK by default, and deprecates the enableWebSdkInteroperability method.

Issues fixed

  • Audio issues concerning audio mixing, audio encoding, and echo.
  • Other issues related to app crashes, log file, and unstable service when pushing streams to the CDN.

API changes

Behavior change

Calling enableLocalAudio (NO) does not change the in-call volume to media volume.

Added

Deprecated

v2.9.1

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 AgoraRtcAudioVolumeInfo struct of the reportAudioVolumeIndicationOfSpeakers 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

Audio

  • Audio freezes.
  • Abnormal audio when a user rejoins the channel after a third-party app interrupts the call.
  • Echoes occur when a user is in a channel.

Miscellaneous

  • The remote users do not receive the didUpdatedUserInfo callback when the local user switches the network connection before joining the channel and calls the joinChannelByUserAccount method.
  • Mixing streams occur in RTMP streaming.

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. RTMP streaming

In this release, we deleted the following methods:

  • configPublisher

If your app implements RTMP 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 switchChannelByToken method to enable the audience in an interactive streaming channel to quickly switch to another channel. With this method, you can achieve a much faster switch than with the leaveChannel and joinChannelByToken methods. After the audience successfully switches to another channel by calling the switchChannelByToken method, the SDK triggers the didLeaveChannelWithStats and didJoinChannel 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 interactive streaming channels interact with each other.

During the media stream relay, the SDK reports the states and events of the relay with the channelMediaRelayStateDidChange and didReceiveChannelMediaRelayEvent 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 localAudioStateChange and remoteAudioStateChangedOfUid 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 localAudioStats 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 onPlaybackAudioFrame callback and the pullPlaybackAudioFrameRawData / pullPlaybackAudioFrameSampleBufferByLengthInByte method is as follows:

  • onPlaybackAudioFrame: 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.
  • pullPlaybackAudioFrameRawData / pullPlaybackAudioFrameSampleBufferByLengthInByte: 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 AgoraChannelStats class:

  • AgoraChannelStats: 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 GameStreaming.
  • 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.
  • Crashes occur when using the Voice SDK.
  • Audio freezes when the audience in the LiveBroadcasting profile sets the audio session category to playback.
  • No audio after a Web app joins the channel when the remote sound position indication is enabled.

Miscellaneous

  • Occasionally mixed streams in RTMP streaming.
  • Crashes occur after calling the leaveChannel method.

API changes

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

Added

Deprecated

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 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 interactive streaming, this release adds the totalFrozenTime and frozenRate members in the AgoraRtcRemoteAudioStats 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 AgoraRtcRemoteAudioStats class.

Improvements

This release adds a AgoraConnectionChangedKeepAliveTimeout(14) member to the AgoraConnectionChangedReason parameter of the connectionChangedToState 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.

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 RTMP streaming service, v2.4.1 defines the following parameter limits:

Class / Interface Parameter Limit
AgoraLiveTranscoding
  • videoFrameRate: 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 RTMP 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.
  • size: Pixel of the video. The minimum value of size 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 streamPublishedWithUrl method for quick troubleshooting.

    New features

    1. State of the RTMP streaming

    v2.4.1 adds the rtmpStreamingChangedToState 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, Runing, Recovering, or Failure state. When the state is Failure, you can use the error code for troubleshooting. You can still use the streamPublishedWithUrl and streamUnpublishedWithUrl 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 connectionChangedToState callback whenever the connection state changes. This release also deprecates AgoraWarningCodeLookupChannelRejected(105), AgoraErrorCodeTokenExpired(109), and AgoraErrorCodeInvalidToken(110).

    3. State of the local network type

    v2.4.1 adds the networkTypeChangedToType 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 firstRemoteAudioFrameDecodedOfUid 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 GameStreaming 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 remoteAudioStats callback without calling the enableAudioVolumeIndication method.
    • Improved the stability of RTMP streaming.

    Issues fixed

    Audio

    • The audio stream is interrupted by Siri and does not resume.

    Miscellaneous

    • Users still receive the onNetworkQuality callback after leaving the channel.
    • Occasional crashes.

    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 AgoraWarningCodeLookupChannelRejected(105) warning code
    • The AgoraErrorCodeTokenExpired(109) error code
    • The AgoraErrorCodeInvalidToken(110) error code

    v2.4.0 and Earlier

    v2.4.0

    v2.4.0 is released on April 1, 2019.

    Compatibility changes

    • Agora Voice SDK for iOS adds a library dependency on CoreML.framework in v2.4.0. Ensure that you add this library when integrating the SDK. For details, see Integrate the SDK.
    • If you integrate the SDK by using CocoaPods,ensure that you run pod update in your Terminal before pod install. If you prefer to specify the SDK version to obtain the latest release, ensure that you specify it as 'AgoraRtcEngine_iOS', '2.4.0.1' in the Podfile.

    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 Voice Changer.

    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 lastmileProbeResult 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 localAudioMixingStateDidChanged 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).

    6. Cloud proxy

    Supports the cloud proxy service. See Use Cloud Proxy for details.

    Improvements

    1. Accuracy of call quality statistics
    2. Core quality improvements
    • Reduces the audio delay.
    • Improves the video quality and stability.
    • Shortens the time to render the first remote video frame.

    Issues fixed

    Audio
    • Calling the enableLocalAudio method disconnects all connected Bluetooth devices.
    • The SDK does not support audio mixing URLs with Chinese characters.
    • The SDK does not return YES after the pushExternalAudioFrameSampleBuffer method call succeeds.
    • Volume levels of the high-pitch sound are lowered.
    • Sounds are occasionally played fast.
    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 networkQuality callback.

    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 LiveBroadcasting 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 audioQualityOfUid callback and replaces it with the remoteAudioStats callback to improve the accuracy of the call quality statistics. The remoteAudioStats 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 networkQuality callback for the uplink and downlink network qualities.

    • remoteAudioStats: Reports the statistics of the remote audio stream from each user/host. This callback replaces the onAudioQuality callback.
    • networkQuality: Reports the last mile network quality of each user in the channel.

    Agora plans to improve the following callback in subsequent versions:

    • lastmileQuality: 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 reason for a connection state change:

    v2.3.2 deprecates the rtcEngineConnectionDidInterrupted and rtcEngineConnectionDidBanned callbacks.

    In the new API method, the network connection states are "disconnected", "connecting", "connected", "reconnecting", and "failed". The SDK triggers the connectionChangedToState callback when the network connection state changes. The SDK also triggers the rtcEngineConnectionDidInterrupted and rtcEngineConnectionDidBanned callbacks under certain circumstances, but Agora does 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 interactive streaming. Application developers can use this feedback for future product improvement. Agora strongly recommends integrating this method in your application.

    4. Improves the audio quality in music scenarios

    v2.3.2 adds high-quality audio in scenarios such as music education. In the setAudioProfile method, you can set AgoraAudioProfile as MusicHighQuality(4) and AgoraAudioScenario as GameStreaming(3) to minimize echo and noise while maintaining the quality of the music.

    5. Other improvements
    • Improves the stability in pushing streams.
    • Optimizes the API calling threads.
    • Improves the performance of the SDK on mid-end and low-end iOS devices.
    • Checks the headset and Bluetooth device connection.
    • Reduces the audio delay.

    Issues fixed

    The following issues are fixed in v2.3.2.

    Audio
    • A user joins an interactive streaming channel with a Bluetooth headset. The audio is not played through the Bluetooth headset when the user leaves the channel and opens another application.
    • Crashes when calling the startAudioMixing method to play music files.
    • A previously disabled microphone becomes enabled when the device connects to a headset.
    • 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 application 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 September 28, 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 streams after joining a channel, this version adds the enableLocalAudio method to disable or re-enable the local audio function.
    Once the local audio function is disabled or re-enabled, the SDK returns the didMicrophoneEnabled 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 muteLocalAudioStream is that enableLocalAudio does not capture or send any audio stream, while muteLocalAudioStream captures but does not send audio streams.

    Improvements

    • Reduces the CPU consumption in the audio communication profile on some low-end iOS devices.

    Issues fixed

    • Occasional crashes.
    • LiveBroadcasting profile: Delay at the client due to incorrect statistics.

    v2.3.0

    v2.3.0 is released on August 31, 2018.

    Compatibility changes

    • An Accelerate.framework library is added to the SDK in v2.3.0, which is capable of large-scale mathematical computations and image calculations, optimized for high performance.
    • 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 tokenPrivilegeWillExpire 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 audioTransportStatsOfUid callback is added to provide user-specific upstream and downstream statistics, including the bitrate, frame rate, and packet loss rate. During a call or the live interactive streaming, the SDK triggers these callbacks once every two seconds after the local 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).

    3. Sets the SDK’s control over an audio session

    The SDK and app both have control over the audio session. However, the app may restrict the SDK’s control over an audio session and allow another app or a third-party component to control it by using the setAudioSessionOperationRestriction method. You can implement different levels of control by choosing the corresponding restriction. You can call this method before or after joining the channel.

    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 the live interactive streaming to reduce user traffic while ensuring the call quality.
    • Improves the audio quality during a call or the live interactive streaming using the deep-learning algorithm.

    Issues fixed

    • Excessive increase in the memory usage when multiple delegated hosts in the channel.
    • Occasional app crashes on some iOS devices.
    • Crashes after publishing streams from some iOS devices.
    • Crashes when a user frequently mutes and resumes all sound effects on some iOS 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 on some iOS devices, a user fails to hear any sound after returning to the channel from a system phone call.
    • Occasionally, the audience cannot adjust the channel volume.
    • Occasional crashes when a user frequently joins and leaves the channel.
    • Occasional crashes when one of the two hosts mutes or disables the local audio while playing the background music.
    • Occasional crashes on the iOS device when the device interoperates with the Web and when a web user frequently joins and leaves a channel.
    • Occasional crashes on some devices when preloading the sound effects.
    • Occasional inter-operational failures between an iOS and a macOS device.
    • Occasional crashes on some iOS devices when a user leaves the interactive streaming channel while playing music using a third-party application.
    • Occasional crashes on some iOS devices when leaving the channel.
    • On iOS, when a host injects a stream to the interactive streaming channel, other hosts can still inject a second stream to the channel.
    • Occasional inter-operational failures between SIP devices and the SDK.
    • Occasional echo issues when using a specific audio card.
    • Failure to adjust the volume on some iOS devices.

    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 methods are added in v2.3.0:

    • addUser
    • removeUser

    The following API methods are deleted and no longer supported in v2.3.0. Agora provides the Recording SDK for better recording services.

    • startRecordingService
    • stopRecordingService
    • refreshRecordingServiceStatus

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

    • 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.
    • Occasional crashes during the live interactive streaming.
    • Excessive increase in the memory usage when multiple delegated hosts start streaming in the channel.
    • Failure to report the uid and volume of the speaker in a channel.
    • Unsteady voice volume of the host's in the live interactive streaming.

    v2.2.2

    v2.2.2 is released on June 21, 2018.

    Issues fixed

    • Fixed occasional online statistics crashes.
    • Fixed the issue that the media and the signaling services cannot be accessed at the same time on some iOS devices.
    • Fixed the issue of failing to report the uid and volume of the speaker in a channel.

    v2.2.1

    v2.2.1 is released on May 30, 2018.

    Issues fixed

    • Occasional crashes on some iOS devices.
    • Occasional memory leak on some iOS devices.
    • Occasional app crashes when the app starts audio mixing on some iOS devices.

    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 audio or video calls before joining the channel, the onLastmileQuality callback changes its detection base from a fixed bitrate to the bitrate set by the customer in videoProfile to improve data accuracy. When the network condition is unknown, the SDK triggers this callback once every 2 seconds.

    4. Audio quality enhancement

    Improves the audio quality in scenarios that involve music playback. To achieve high-fidelity music playback, you can set Scenario:AgoraAudioScenarioGameStreaming = 3 in the setAudioProfile method.

    5. Bitcode support

    Adds support for Bitcode, which enables App optimization and thinning by the App Store. The package size of the Bitcode SDK is 2.5 times that of the normal one.

    Issues fixed

    • Fixed occasional echo issues caused by some iOS device.

    v2.1.3

    v2.1.3 is released on April 19, 2018.

    Issues fixed

    • Block callbacks are occasionally not received if the delegate is not set.
    • NSAssertionHandler appears in external links to the SDK.
    • Occasional recording failures on some phones when a user leaves a channel and turns on the built-in recording device.
    • Occasional crashes during a Communication or LiveBroadcasting session.

    v2.1.2

    v2.1.2 is released on April 2, 2018.

    Issues fixed

    • Occasional crashes on iOS 11.

    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 streaming, 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:

    Improvement

    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).
    API Naming Optimization Modifies a set of names for the API attributes and enumeration values.

    Issues fixed

    • Occasional crashes.
    • Occasionally, no voice after turning off the microphone.

    v2.0.2

    v2.0.2 was released on December 15, 2017, and fixes the FFmpeg symbol conflict.

    v2.0

    v2.0 is released on December 6, 2017.

    New features

    • Updates the following callbacks for audio mixing and sound effects:

      Name Description
      rtcEngineMediaEngineDidAudioMixingFinish Removed. Replaced by rtcEngineLocalAudioMixingDidFinish.
      rtcEngineDidAudioEffectFinish Added. Notifies the app when the local audio effect playback stops.
      rtcEngineRemoteAudioMixingDidStart Added. Notifies the app when the remote user starts audio mixing.
      rtcEngineRemoteAudioMixingDidFinish Added. Notifies the app when the remote user stops audio mixing.
    • Supports the external audio source in the Communication and LiveBroadcasting profiles by adding the following API methods:

      Name Description
      enableExternalAudioSourceWithSampleRate Enables the external audio source.
      disableExternalAudioSource Disables the external audio source.
      pushExternalAudioFrameRawData Pushes the external audio frame.
    • Provides a set of RESTful APIs to ban a peer user from the server in the Communication and LiveBroadcasting profiles. Contact support@agora.io to enable this function, if required.

    Issues fixed

    Audio routing and Bluetooth issues.

    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.
    • LiveBroadcasting: Adds the setInEarMonitoringVolume method to adjust the volume of the in-ear monitor.

    Improvements

    • Optimizes the audio at high bitrates.
    • LiveBroadcasting: The audience can view the host within one second in a single-stream mode (938 ms on average, and 734 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:

    Occasional crashes on iOS devices.

    v1.13.1

    v1.13.1 is released on September 28, 2017.

    • Fixes the issue of unable to adjust the volume in the speaker mode on iOS 11 with iPhone 7 or later.
    • 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 the live interactive streaming.
    • Adds the function to disable the audio playback.
    • Adds the module map for the SDK, which means that bridging header files are not necessary for Swift projects.
    • Supports the profile configuration for stream-pushing on the client side.
    • Adds the didClientRoleChanged method to report on a user role switch between the host and the audience in the live interactive streaming.
    • Supports the push-stream failure callback on the server side.

    Issues fixed:

    Occasional crashes.

    v1.12

    v1.12 is released on July 25, 2017.

    New features:

    • Adds the injectStream method to inject an RTMP stream into the current channel in the live interactive streaming.
    • 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 to manage the audio effect.

    Issues fixed:

    Occasional crashes.