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5.0 Changelog: Twilio Video Android

The Twilio Programmable Video SDKs use Semantic Versioning.

5.0.0-beta3 (September 17th, 2019)

Dominant Speaker Detection API

The Dominant Speaker Detection API sends events to your application every time the dominant speaker changes. You can use those events to improve the end user's experience by, for example, highlighting which participant is currently talking.

The Dominant Speaker Detection API is only available for Group Rooms. To enable dominant speaker detection, set the ConnectOptions.dominantSpeakerEnabled property to true. Use Room.getDominantSpeaker() to determine the current dominant speaker. Implement Room.Listener.onDominantSpeakerChanged() method to receive callbacks when the dominant speaker changes.

For more information, refer to the API docs and to the dominant speaker tutorial

 ConnectOptions connectOptions =
                new ConnectOptions.Builder(token)
                        .roomName(roomName)
                        .enableDominantSpeaker(true)
                        .build();
Room room = Video.connect(context, connectOptions, roomListener);

@Override
void onDominantSpeakerChanged(
                @NonNull Room room, @Nullable RemoteParticipant remoteParticipant) {
                // Handle dominant speaker change
        }

API の変更点

  • Introduced TwilioException.SIGNALING_DNS_RESOLUTION_ERROR_EXCEPTION, which is now raised instead of TwilioException.SIGNALING_CONNECTION_ERROR_EXCEPTION in the following scenarios:
  • The device has misconfigured DNS Server(s) on its active network interface.
  • The region provided in ConnectOptions was invalid.
  • The device lost its Internet connection before the query could complete.

改善点

  • Reduced connection times by removing a round trip when:
  • Reconnecting after a signaling connection failure
  • Connecting with ConnectOptions.iceOptions, and overridden Servers
  • Connecting with default ICE servers (us1 only)

バグ修正

  • Fixed crash that occurred when rapidly connecting and disconnecting from a room.
  • Fixed updating CameraCapturer.State when error occurs.
  • Setting ConnectOptions.region to an empty or null value results in the default region being used.
  • Fixed a bug where native memory was leaked after disconnecting from a Room.
  • Fixed a bug where network monitoring would continue on closed connections in a Peer-to-Peer Room.

既知の問題

  • In rare cases, the SDK might timeout during a TCP handshake and should be more aggressive at establishing a connection.
  • Only Constrained Baseline Profile is supported when H.264 is the preferred video codec.
  • Network handoff, and subsequent connection renegotiation is not supported for IPv6 networks #72
  • The SDK is not side-by-side compatible with other WebRTC based libraries #340
  • Codec preferences do not function correctly in a hybrid codec Group Room with the following codecs:
    • ISAC
    • PCMA
    • G722
    • VP9
  • Unpublishing and republishing a LocalAudioTrack or LocalVideoTrack might not be seen by Participants. As a result, tracks published after a Room.State.RECONNECTED event might not be subscribed to by a RemoteParticipant.
  • Server side deflate compression is disabled due to occasional errors when reading messages.
  • Using Camera2Capturer with a camera ID that does not support ImageFormat.PRIVATE capture outputs results in a runtime exception. Reference this issue for guidance on a temporary work around.

Size Report

ABI APK Size Impact
universal 22.8MB
armeabi-v7a 5MB
arm64-v8a 5.9MB
x86 6.2MB
x86_64 6.3MB

5.0.0-beta2 (August 7th, 2019)

改善点

  • The Participant's signaling connection now conforms to Twilio's TLS & Cipher Suite Policy. Support for TLS versions older than 1.2 has been removed.
  • Adjusted the buffer sizes used for signaling messages to reduce network fragmentation.
  • Setting video::LogModule::kSignaling enables logging of low-level connection events.

バグ修正

  • WebSocket errors are handled immediately, rather than waiting for a timeout to occur.
  • Handle rare exceptions when constructing a WebSocket.

既知の問題

  • Future 5.0.0-beta releases will reduce the number of round-trips required to connect to a Room.
  • In rare cases, the SDK might timeout during a TCP handshake and should be more aggressive at establishing a connection.
  • Only Constrained Baseline Profile is supported when H.264 is the preferred video codec.
  • Network handoff, and subsequent connection renegotiation is not supported for IPv6 networks #72
  • The SDK is not side-by-side compatible with other WebRTC based libraries #340
  • Codec preferences do not function correctly in a hybrid codec Group Room with the following codecs:
    • ISAC
    • PCMA
    • G722
    • VP9
  • Unpublishing and republishing a LocalAudioTrack or LocalVideoTrack might not be seen by Participants. As a result, tracks published after a Room.State.RECONNECTED event might not be subscribed to by a RemoteParticipant.
  • Server side deflate compression is disabled due to occasional errors when reading messages.
  • Rapidly connecting and disconnecting from a Room may cause a crash.

5.0.0-beta1 (July 16th, 2019)

改善点

  • The SDK uses a new WebSocket based signaling transport, and communicates with globally available signaling Servers over IPv4 and IPv6 networks.
  • Added a ConnectOptions.region property. By default, the Client will connect to the nearest signaling Server determined by latency based routing. Setting a value other than "gll" bypasses routing and guarantees that signaling traffic will be terminated in the region that you prefer.
  • Participants are considered to be reconnecting within 15 seconds, and are disconnected from a Room after 45 seconds of lost connectivity. #80
  • Added and updated public API nullability annotations.

バグ修正

  • Participants can send messages that are larger than 16 KB.
  • The “roomimpl.worker” thread is no longer needed.

既知の問題

  • Setting LogModule.SIGNALING does not produce any logging.
  • Future 5.0.0-beta releases will reduce the number of round-trips required to connect to a Room.
  • In rare cases, the SDK might timeout during a TCP handshake and should be more aggressive at establishing a connection.
  • Only Constrained Baseline Profile is supported when H.264 is the preferred video codec.
  • Network handoff, and subsequent connection renegotiation is not supported for IPv6 networks #72
  • The SDK is not side-by-side compatible with other WebRTC based libraries #340
  • Codec preferences do not function correctly in a hybrid codec Group Room with the following codecs:
    • ISAC
    • PCMA
    • G722
    • VP9
  • Unpublishing and republishing a LocalAudioTrack or LocalVideoTrack might not be seen by Participants. As a result, tracks published after a Room.State.RECONNECTED event might not be subscribed to by a RemoteParticipant.
  • Server side deflate compression is disabled due to occasional errors when reading messages.
  • Rapidly connecting and disconnecting from a Room may cause a crash.

Size Report

ABI APK Size Impact
universal 22.7MB
armeabi-v7a 5MB
arm64-v8a 5.9MB
x86 6.2MB
x86_64 6.3MB
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