When i try to build release on ios (Product -> archive), the build fails with 'WebRTC/WebRTC.h' file not found error. When i downgrade the dependency to 0.6.7, it works without errors, but i need the changes in the latest version for improved iOS audio handling.

version that fails: 0.6.10+hotfix.1 version that works: 0.6.7

Oh and another thing that i noticed while running pod update is that flutter_webrtc.podspec still states s.version = '0.2.2' for almost all the versions, it gets confusing sometimes to check if xcode is using the right version dependency or not 😅

I added a desktop-specific capture interface, which uses the desktop unified interface for Windows/macOS/Linux.

https://github.com/flutter-webrtc/webrtc-interface/blob/feat/desktop-cpaturer/lib/src/desktop_capturer.dart

enum SourceType {
  kEntireScreen,
  kWindow,
}

class ThumbnailSize {
  ThumbnailSize(this.width, this.height);
  int width;
  int height;
}

abstract class DesktopCapturerSource {
  /// The identifier of a window or screen that can be used as a
  /// chromeMediaSourceId constraint when calling
  String get id;

  /// A screen source will be named either Entire Screen or Screen <index>,
  /// while the name of a window source will match the window title.
  String get name;

  ///A thumbnail image of the source. ARGB
  ByteBuffer get thumbnail;

  /// specified in the options passed to desktopCapturer.getSources.
  /// The actual size depends on the scale of the screen or window.
  ThumbnailSize get thumbnailSize;

  /// The type of the source.
  SourceType get type;
}

abstract class DesktopCapturer {
  Future<List<DesktopCapturerSource>> getSources(
      {List<SourceType> types, ThumbnailSize thumbnailSize});
}

The enumeration is as follows

var sources = await navigator.desktopCapturer.getSources(
        types: [kScreen, kWindow], 
        thumbnailSize:  ThumbnailSize(width: 320, heigth: 180)
        );

DesktopCapturerSource selectedSource;
sources.forEach((item) => {
     // print item.id, item.title
     // display thumbnail
     selectedSource = item;
});

get default display media.

var mediaStream = await navigator.mediaDevices.getDisplayMedia({video: true});

get display media by source id.

var mediaStream = await navigator.mediaDevices.getDisplayMedia({
      video: {
               deviceId: {
                     exact: selectedSource.id
              }
              mandatory: {
                    minWidth: 1280,
                    minHeight: 720,
                    frameRate: 30.0
              }
      }
})

Hi, I can switch between front and back cameras using the following function on mobile:

stream.getVideoTracks()[0].switchCamera();

Can it be possible to iterate through all available cameras on Chrome (web) using the same function or an additional function?

Or can you add a function to get a list of available cameras (and also available audio input sources) and select one of them using an index?

On Chrome WebRTC, one can select from available video and audio sources even during a conversation.

But on Firefox, the user chooses the video/audio source before the conversation starts and it's not possible to switch to another source afterwards.

I'm not sure how this difference can be addressed.

PS: My app works fine on the latest versions of Chrome and Firefox but not on Safari. Do you have any plans to support Safari?

Hello,

Did anyone know how can we add camera effects and transmit them to peer connection?

For example Snapchat like filters or open GL filters

Appreciate it if anyone able to give a solution or guiding to how to do it

TIA

Texture widget for the embedder seems to be coming soon: https://github.com/google/flutter-desktop-embedding/issues/107#issuecomment-444272681, let's port it to the desktop embedder and let the plugin extend to mac/linux/windows. 😄

I/org.webrtc.Logging(28548): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4005 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4004 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4005 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4004 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4004 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4003 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4005 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4004 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4005 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4005 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4004 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4004 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4004 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(28548): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.

** During stream audio working fine, but the video shows a black screen for S22 ultra, other device working normally. **

Make a call from S22 ultra and the incoming call device picked up the black screen show. If make a call from other devices to s22 ultra video and audio work well as usual.

I/ViewRootImpl@f81f857MainActivity: ViewPostIme pointer 0 I/ViewRootImpl@f81f857MainActivity: ViewPostIme pointer 1 I/DecorView(32251): notifyKeepScreenOnChanged: keepScreenOn=true I/ViewRootImpl@f81f857MainActivity: Relayout returned: old=(0,0,1080,2316) new=(0,0,1080,2316) req=(1080,2316)0 dur=4 res=0x1 s={true -5476376610960537264} ch=false fn=3 I/ViewRootImpl@f81f857MainActivity: updateBoundsLayer: t = android.view.SurfaceControl$Transaction@73115d3 sc = Surface(name=Bounds for - com.futureteen.chat/com.example.myapp.MainActivity@0)/@0x8bec510 frame = 3 I/org.webrtc.Logging(32251): NativeLibrary: Loading native library: jingle_peerconnection_so I/org.webrtc.Logging(32251): NativeLibrary: Loading library: jingle_peerconnection_so I/org.webrtc.Logging(32251): PeerConnectionFactory: PeerConnectionFactory was initialized without an injected Loggable. Any existing Loggable will be deleted. I/org.webrtc.Logging(32251): EglBase14Impl: SDK version: 31. isEGL14Supported: true I/org.webrtc.Logging(32251): EglBase14Impl: Using OpenGL ES version 2 I/org.webrtc.Logging(32251): WebRtcAudioManagerExternal: Sample rate is set to 48000 Hz I/org.webrtc.Logging(32251): WebRtcAudioManagerExternal: Sample rate is set to 48000 Hz I/org.webrtc.Logging(32251): JavaAudioDeviceModule: createAudioDeviceModule I/org.webrtc.Logging(32251): JavaAudioDeviceModule: HW NS will be used. I/org.webrtc.Logging(32251): JavaAudioDeviceModule: HW AEC will be used. I/org.webrtc.Logging(32251): WebRtcAudioEffectsExternal: ctor@[name=main, id=2] I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: ctor@[name=main, id=2] I/org.webrtc.Logging(32251): WebRtcAudioTrackExternal: ctor@[name=main, id=2] W/AudioCapabilities(32251): Unsupported mime audio/x-ape W/AudioCapabilities(32251): Unsupported mime audio/x-ima W/AudioCapabilities(32251): Unsupported mime audio/mpeg-L1 W/AudioCapabilities(32251): Unsupported mime audio/mpeg-L2 W/VideoCapabilities(32251): Unsupported mime video/wvc1 W/VideoCapabilities(32251): Unsupported mime video/x-ms-wmv W/VideoCapabilities(32251): Unsupported mime video/avc-wfd W/VideoCapabilities(32251): Unsupported mime video/avc-wfd W/VideoCapabilities(32251): Unsupported mime video/hevc-wfd W/VideoCapabilities(32251): Unsupported mime video/hevc-wfd W/VideoCapabilities(32251): Unsupported mime image/vnd.android.heic W/VideoCapabilities(32251): Unsupported mime image/vnd.android.heic I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: enableBuiltInAEC(true) I/org.webrtc.Logging(32251): WebRtcAudioEffectsExternal: setAEC(true) I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: enableBuiltInNS(true) I/org.webrtc.Logging(32251): WebRtcAudioEffectsExternal: setNS(true) I/org.webrtc.Logging(32251): PeerConnectionFactory: onNetworkThreadReady I/org.webrtc.Logging(32251): PeerConnectionFactory: onWorkerThreadReady I/org.webrtc.Logging(32251): PeerConnectionFactory: onSignalingThreadReady I/org.webrtc.Logging(32251): EglRenderer: Initializing EglRenderer I/org.webrtc.Logging(32251): EglRenderer: EglBase.create shared context I/org.webrtc.Logging(32251): EglBase14Impl: Using OpenGL ES version 2 I/org.webrtc.Logging(32251): EglRenderer: Initializing EglRenderer I/org.webrtc.Logging(32251): EglRenderer: EglBase.create shared context I/org.webrtc.Logging(32251): EglBase14Impl: Using OpenGL ES version 2 I/FlutterWebRTCPlugin(32251): getUserMedia(audio): mandatory: [], optional: [googNoiseSuppression: true, googEchoCancellation: true, echoCancellation: true, googEchoCancellation2: true, googDAEchoCancellation: true] I/FlutterWebRTCPlugin(32251): getUserMedia(video): ConstraintsMap{mMap={facingMode=user}} I/CameraManagerGlobal(32251): Connecting to camera service D/VendorTagDescriptor(32251): addVendorDescriptor: vendor tag id 14172875900359437128 added I/CameraManagerGlobal(32251): Camera 0 facing CAMERA_FACING_BACK state now CAMERA_STATE_CLOSED for client com.samsung.android.visionintelligence API Level 2 I/CameraManagerGlobal(32251): Camera 1 facing CAMERA_FACING_FRONT state now CAMERA_STATE_CLOSED for client com.futureteen.chat API Level 2 I/CameraManagerGlobal(32251): Camera 2 facing CAMERA_FACING_BACK state now CAMERA_STATE_CLOSED for client android.system API Level 2 D/FlutterWebRTCPlugin(32251): Creating video capturer using Camera2 API. I/CameraManagerGlobal(32251): Camera 20 facing CAMERA_FACING_BACK state now CAMERA_STATE_CLOSED for client com.sec.android.app.camera API Level 2 I/CameraManagerGlobal(32251): Camera 21 facing CAMERA_FACING_BACK state now CAMERA_STATE_CLOSED for client android.system API Level 2 I/CameraManagerGlobal(32251): Camera 23 facing CAMERA_FACING_BACK state now CAMERA_STATE_CLOSED for client android.system API Level 2 I/CameraManagerGlobal(32251): Camera 3 facing CAMERA_FACING_FRONT state now CAMERA_STATE_CLOSED for client android.system API Level 2 D/FlutterWebRTCPlugin(32251): Create front camera 1 succeeded I/CameraManagerGlobal(32251): Camera 4 facing CAMERA_FACING_FRONT state now CAMERA_STATE_CLOSED for client vendor.client.pid<1907> API Level 2 I/CameraManagerGlobal(32251): Camera 52 facing CAMERA_FACING_BACK state now CAMERA_STATE_CLOSED for client android.system API Level 2 I/CameraManagerGlobal(32251): Camera 54 facing CAMERA_FACING_BACK state now CAMERA_STATE_CLOSED for client android.system API Level 2 I/org.webrtc.Logging(32251): EglBase14Impl: Using OpenGL ES version 2 I/CameraManagerGlobal(32251): Camera 56 facing CAMERA_FACING_BACK state now CAMERA_STATE_CLOSED for client com.sec.android.app.camera API Level 2 I/CameraManagerGlobal(32251): Camera 58 facing CAMERA_FACING_BACK state now CAMERA_STATE_CLOSED for client android.system API Level 2 I/CameraManagerGlobal(32251): Camera 90 facing CAMERA_FACING_FRONT state now CAMERA_STATE_CLOSED for client android.system API Level 2 I/org.webrtc.Logging(32251): CameraCapturer: startCapture: 1280x720@30 D/FlutterWebRTCPlugin(32251): changeCaptureFormat: 1280x720@30 I/org.webrtc.Logging(32251): Camera2Session: Create new camera2 session on camera 1 I/org.webrtc.Logging(32251): Camera2Session: start D/FlutterWebRTCPlugin(32251): MediaStream id: 3d840e6c-64e7-4c1c-abd9-7e34575c00b3 I/flutter (32251): MediaStreamTrack:enableSpeakerphone true I/flutter (32251): MediaStreamTrack:enableSpeakerphone true I/org.webrtc.Logging(32251): Camera2Session: Available preview sizes: [3648x2736, 3648x2048, 3648x1640, 2736x2736, 2400x1080, 2320x1080, 1920x1440, 1920x1080, 1920x824, 1440x1080, 1280x720, 1088x1088, 960x720, 720x480, 640x480, 640x360, 352x288, 320x240, 176x144] I/org.webrtc.Logging(32251): Camera2Session: Available fps ranges: [[10.0:10.0], [7.0:15.0], [15.0:15.0], [24.0:24.0], [7.0:30.0], [30.0:30.0]] I/org.webrtc.Logging(32251): Camera2Session: Using capture format: 1280x720@[7.0:30.0] I/org.webrtc.Logging(32251): Camera2Session: Opening camera 1 D/FlutterWebRTCPlugin(32251): CameraEventsHandler.onCameraOpening: cameraName=1 D/RTCAudioManager(32251): ctor D/RTCBluetoothManager(32251): create@[name=main, id=2] D/RTCBluetoothManager(32251): ctor D/RTCAudioManager(32251): useSpeakerphone: auto D/RTCProximitySensor(32251): RTCProximitySensor@[name=main, id=2] D/RTCAudioManager(32251): defaultAudioDevice: SPEAKER_PHONE D/RTCAudioManager(32251): Android SDK: 31, Release: 12, Brand: samsung, Device: b0q, Id: SP1A.210812.016, Hardware: qcom, Manufacturer: samsung, Model: SM-S908E, Product: b0qxxx D/RTCAudioManager(32251): start D/RTCAudioManager(32251): AudioManager starts... W/AudioManager(32251): Use of stream types is deprecated for operations other than volume control W/AudioManager(32251): See the documentation of requestAudioFocus() for what to use instead with android.media.AudioAttributes to qualify your playback use case D/RTCAudioManager(32251): Audio focus request granted for VOICE_CALL streams D/RTCBluetoothManager(32251): start I/BluetoothAdapter(32251): BluetoothAdapter() : com.futureteen.chat D/RTCBluetoothManager(32251): BluetoothAdapter: enabled=false, state=OFF, name=Sokpheng's S22 Ultra, address=02:00:00:00:00:00 I/org.webrtc.Logging(32251): Camera2Session: Camera opened. D/BluetoothHeadset(32251): BTStateChangeCB is registed by 32251 @ com.futureteen.chat D/RTCBluetoothManager(32251): HEADSET profile state: DISCONNECTED D/RTCBluetoothManager(32251): Bluetooth proxy for headset profile has started D/RTCBluetoothManager(32251): start done: BT state=HEADSET_UNAVAILABLE D/RTCAudioManager(32251): --- updateAudioDeviceState: wired headset=false, BT state=HEADSET_UNAVAILABLE D/RTCAudioManager(32251): Device status: available=[], selected=NONE, user selected=NONE D/RTCAudioManager(32251): setAudioDeviceInternal(device=SPEAKER_PHONE) I/CameraManagerGlobal(32251): Camera 1 facing CAMERA_FACING_FRONT state now CAMERA_STATE_OPEN for client com.futureteen.chat API Level 2 D/RTCAudioManager(32251): New device status: available=[EARPIECE, SPEAKER_PHONE], selected=SPEAKER_PHONE D/FlutterWebRTCPlugin(32251): onAudioManagerDevicesChanged: [EARPIECE, SPEAKER_PHONE], selected: SPEAKER_PHONE D/RTCAudioManager(32251): --- updateAudioDeviceState done D/RTCAudioManager(32251): AudioManager started D/RTCAudioManager(32251): WiredHeadsetReceiver.onReceive@[name=main, id=2]: a=android.intent.action.HEADSET_PLUG, s=unplugged, m=mic, n=null, sb=true D/RTCAudioManager(32251): --- updateAudioDeviceState: wired headset=false, BT state=HEADSET_UNAVAILABLE D/RTCAudioManager(32251): Device status: available=[EARPIECE, SPEAKER_PHONE], selected=SPEAKER_PHONE, user selected=NONE D/RTCAudioManager(32251): --- updateAudioDeviceState done W/FlutterWebRTCPlugin(32251): FlutterRTCVideoRenderer.setVideoTrack, set video track to 9f56a813-5dab-4455-b6ff-9254b07e01a3 I/org.webrtc.Logging(32251): EglRenderer: Releasing. I/org.webrtc.Logging(32251): EglRenderer: eglBase detach and release. I/org.webrtc.Logging(32251): EglRenderer: Quitting render thread. I/org.webrtc.Logging(32251): EglRenderer: Releasing done. I/org.webrtc.Logging(32251): EglRenderer: Initializing EglRenderer I/org.webrtc.Logging(32251): EglRenderer: EglBase.create shared context I/org.webrtc.Logging(32251): EglBase14Impl: Using OpenGL ES version 2 I/org.webrtc.Logging(32251): Camera2Session: Camera capture session configured. I/org.webrtc.Logging(32251): Camera2Session: Using video stabilization. I/org.webrtc.Logging(32251): Camera2Session: Using continuous video auto-focus. I/org.webrtc.Logging(32251): Camera2Session: Camera device successfully started. I/org.webrtc.Logging(32251): CameraCapturer: Create session done. Switch state: IDLE I/org.webrtc.Logging(32251): SurfaceTextureHelper: Setting listener to org.webrtc.-$$Lambda$Camera2Session$CaptureSessionCallback$_XbqUL6ihfYRgME_EfDLZtLmkGA@83fe3b I/CameraManagerGlobal(32251): Camera 1 facing CAMERA_FACING_FRONT state now CAMERA_STATE_ACTIVE for client com.futureteen.chat API Level 2 I/BufferQueueProducer(32251): SurfaceTexture-1-32251-2 queueBuffer: queued for the first time. D/FlutterWebRTCPlugin(32251): CameraEventsHandler.onFirstFrameAvailable I/BufferQueueProducer(32251): SurfaceTexture-0-32251-0 queueBuffer: queued for the first time. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 26. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4004 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4004 ms. Frames received: 107. Dropped: 0. Rendered: 107. Render fps: 26.7. Average render time: 1560 us. Average swapBuffer time: 774 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/ViewRootImpl@f81f857MainActivity: ViewPostIme pointer 0 I/ViewRootImpl@f81f857MainActivity: ViewPostIme pointer 1 D/ThumbnailPlugin(32251): buildThumbnailFile( format:0, maxh:0, maxw:0, timeMs:0, quality:5 ) D/ThumbnailPlugin(32251): buildThumbnailData( format:0, maxh:0, maxw:0, timeMs:0, quality:5 ) V/MediaMetadataRetriever(32251): constructor V/MediaHTTPService(32251): MediaHTTPService(android.media.MediaHTTPService@844dab1): Cookies: null V/MediaMetadataRetriever(32251): setDataSource V/MediaHTTPService(32251): makeHTTPConnection: CookieManager created: java.net.CookieManager@5d93b96 V/MediaHTTPService(32251): makeHTTPConnection(android.media.MediaHTTPService@844dab1): cookieHandler: java.net.CookieManager@5d93b96 Cookies: null I/MediaHTTPConnection(32251): setReadTimeout and setConnectTimeout with 8000ms I/MediaHTTPConnection(32251): response code = 200 I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. D/MediaConstraintsUtils(32251): mandatory constraints are not a map D/MediaConstraintsUtils(32251): optional constraints are not an array I/org.webrtc.Logging(32251): NetworkMonitor: Start monitoring with native observer -5476376614449608560 D/ConnectivityManager(32251): StackLog: [android.net.ConnectivityManager.sendRequestForNetwork(ConnectivityManager.java:3937)] [android.net.ConnectivityManager.sendRequestForNetwork(ConnectivityManager.java:3979)] [android.net.ConnectivityManager.requestNetwork(ConnectivityManager.java:4117)] [android.net.ConnectivityManager.requestNetwork(ConnectivityManager.java:4095)] [org.webrtc.NetworkMonitorAutoDetect$ConnectivityManagerDelegate.requestMobileNetwork(NetworkMonitorAutoDetect.java:404)] [org.webrtc.NetworkMonitorAutoDetect.(NetworkMonitorAutoDetect.java:597)] [org.webrtc.NetworkMonitor$1.create(NetworkMonitor.java:50)] [org.webrtc.NetworkMonitor.createNetworkChangeDetector(NetworkMonitor.java:181)] [org.webrtc.NetworkMonitor.startMonitoring(NetworkMonitor.java:108)] [org.webrtc.NetworkMonitor.startMonitoring(NetworkMonitor.java:129)] D/ConnectivityManager(32251): StackLog: [android.net.ConnectivityManager.sendRequestForNetwork(ConnectivityManager.java:3937)] [android.net.ConnectivityManager.sendRequestForNetwork(ConnectivityManager.java:3979)] [android.net.ConnectivityManager.registerNetworkCallback(ConnectivityManager.java:4361)] [android.net.ConnectivityManager.registerNetworkCallback(ConnectivityManager.java:4331)] [org.webrtc.NetworkMonitorAutoDetect$ConnectivityManagerDelegate.registerNetworkCallback(NetworkMonitorAutoDetect.java:391)] [org.webrtc.NetworkMonitorAutoDetect.(NetworkMonitorAutoDetect.java:604)] [org.webrtc.NetworkMonitor$1.create(NetworkMonitor.java:50)] [org.webrtc.NetworkMonitor.createNetworkChangeDetector(NetworkMonitor.java:181)] [org.webrtc.NetworkMonitor.startMonitoring(NetworkMonitor.java:108)] [org.webrtc.NetworkMonitor.startMonitoring(NetworkMonitor.java:129)] D/FlutterWebRTCPlugin(32251): onIceGatheringChangeGATHERING D/FlutterWebRTCPlugin(32251): onIceCandidate D/FlutterWebRTCPlugin(32251): onIceCandidate D/FlutterWebRTCPlugin(32251): onIceCandidate D/FlutterWebRTCPlugin(32251): onIceCandidate D/FlutterWebRTCPlugin(32251): onIceCandidate D/FlutterWebRTCPlugin(32251): onIceCandidate I/org.webrtc.Logging(32251): NetworkMonitorAutoDetect: Network becomes available: 180 I/org.webrtc.Logging(32251): NetworkMonitorAutoDetect: capabilities changed: [ Transports: WIFI Capabilities: NOT_METERED&INTERNET&NOT_RESTRICTED&TRUSTED&NOT_VPN&VALIDATED&NOT_ROAMING&FOREGROUND&NOT_CONGESTED&NOT_SUSPENDED&NOT_VCN_MANAGED LinkUpBandwidth>=76233Kbps LinkDnBandwidth>=38894Kbps TransportInfo: <SSID: , BSSID: 02:00:00:00:00:00, MAC: 02:00:00:00:00:00, Security type: 2, Supplicant state: COMPLETED, Wi-Fi standard: 4, RSSI: -62, Link speed: 52Mbps, Tx Link speed: 52Mbps, Max Supported Tx Link speed: 300Mbps, Rx Link speed: 117Mbps, Max Supported Rx Link speed: 300Mbps, Frequency: 2452MHz, Net ID: -1, Metered hint: false, score: 60, CarrierMerged: false, SubscriptionId: -1, IsPrimary: -1> SignalStrength: -62 UnderlyingNetworks: Null] I/org.webrtc.Logging(32251): NetworkMonitorAutoDetect: link properties changed I/flutter (32251): singaling RTCSignalingState.RTCSignalingStateHaveLocalOffer D/FlutterWebRTCPlugin(32251): onIceCandidate D/FlutterWebRTCPlugin(32251): onIceCandidate D/FlutterWebRTCPlugin(32251): onIceCandidate D/FlutterWebRTCPlugin(32251): onIceCandidate V/MediaMetadataRetriever(32251): setDataSource return(0) V/MediaMetadataRetriever(32251): getFrameAtTime: time(0 us) option(3) colorFormat(4) metaOnly(0) V/MediaMetadataRetriever(32251): destructor V/MediaMetadataRetriever(32251): disconnect D/ThumbnailPlugin(32251): buildThumbnailFile( written:9479 ) D/FlutterWebRTCPlugin(32251): onIceCandidate I/org.webrtc.Logging(32251): WebRtcAudioTrackExternal: initPlayout(sampleRate=48000, channels=1, bufferSizeFactor=1.0) I/org.webrtc.Logging(32251): WebRtcAudioTrackExternal: byteBuffer.capacity: 960 I/org.webrtc.Logging(32251): WebRtcAudioTrackExternal: minBufferSizeInBytes: 9620 I/org.webrtc.Logging(32251): WebRtcAudioTrackExternal: createAudioTrackOnLollipopOrHigher I/org.webrtc.Logging(32251): WebRtcAudioTrackExternal: nativeOutputSampleRate: 48000 I/org.webrtc.Logging(32251): WebRtcAudioTrackExternal: AudioTrack: session ID: 13529, channels: 1, sample rate: 48000, max gain: 1.0 I/org.webrtc.Logging(32251): WebRtcAudioTrackExternal: AudioTrack: buffer size in frames: 4810 I/org.webrtc.Logging(32251): WebRtcAudioTrackExternal: AudioTrack: buffer capacity in frames: 4810 I/org.webrtc.Logging(32251): VolumeLogger: start@[name=worker_thread - 32454, id=154] I/org.webrtc.Logging(32251): VolumeLogger: audio mode is: MODE_NORMAL I/org.webrtc.Logging(32251): WebRtcAudioTrackExternal: startPlayout I/org.webrtc.Logging(32251): WebRtcAudioTrackExternal: AudioTrackThread@[name=AudioTrackJavaThread, id=170] I/org.webrtc.Logging(32251): WebRtcAudioTrackExternal: doAudioTrackStateCallback: 0 I/org.webrtc.Logging(32251): AndroidVideoDecoder: ctor name: c2.qti.avc.decoder type: H264 color format: 19 context: org.webrtc.EglBase14Impl$Context@6c18591 I/org.webrtc.Logging(32251): AndroidVideoDecoder: ctor name: c2.android.avc.decoder type: H264 color format: 19 context: org.webrtc.EglBase14Impl$Context@6c18591 D/FlutterWebRTCPlugin(32251): onAddTrack D/FlutterWebRTCPlugin(32251): onAddTrack D/FlutterWebRTCPlugin(32251): onConnectionChangeCONNECTING D/FlutterWebRTCPlugin(32251): onIceCandidate I/flutter (32251): singaling RTCSignalingState.RTCSignalingStateStable D/FlutterWebRTCPlugin(32251): onIceGatheringChangeCOMPLETE D/FlutterWebRTCPlugin(32251): onSelectedCandidatePairChanged I/flutter (32251): onIceConnectionState RTCIceConnectionState.RTCIceConnectionStateChecking W/FlutterWebRTCPlugin(32251): FlutterRTCVideoRenderer.setVideoTrack, set video track to 5c5648e7-f470-4365-84ba-923585071cdc I/org.webrtc.Logging(32251): EglRenderer: Releasing. I/flutter (32251): onConnectionState RTCPeerConnectionState.RTCPeerConnectionStateConnecting D/FlutterWebRTCPlugin(32251): onConnectionChangeCONNECTED I/org.webrtc.Logging(32251): EglRenderer: eglBase detach and release. I/org.webrtc.Logging(32251): EglRenderer: Quitting render thread. I/org.webrtc.Logging(32251): EglRenderer: Releasing done. I/org.webrtc.Logging(32251): EglRenderer: Initializing EglRenderer I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: initRecording(sampleRate=48000, channels=1) I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: byteBuffer.capacity: 960 I/org.webrtc.Logging(32251): EglRenderer: EglBase.create shared context I/org.webrtc.Logging(32251): EglBase14Impl: Using OpenGL ES version 2 I/org.webrtc.Logging(32251): HardwareVideoEncoder: initEncode: 1280 x 720. @ 300kbps. Fps: 60 Use surface mode: true I/CCodec (32251): state->set(ALLOCATING) I/CCodec (32251): allocate(c2.qti.avc.encoder) I/Codec2Client(32251): Available Codec2 services: "default" "default0" "software" I/CCodec (32251): setting up 'default' as default (vendor) store I/CCodec (32251): Created component [c2.qti.avc.encoder] I/CCodec (32251): state->set(ALLOCATED) D/CCodecConfig(32251): read media type: video/avc D/ReflectedParamUpdater(32251): ignored struct field coding.qp.values D/ReflectedParamUpdater(32251): extent() != 1 for single value type: vendor.qti-ext-enc-temporal-layer-bitrate.layerBitrates D/ReflectedParamUpdater(32251): ignored struct field coding.gop.values D/ReflectedParamUpdater(32251): extent() != 1 for single value type: output.buffers.pool-ids.values D/ReflectedParamUpdater(32251): extent() != 1 for single value type: vendor.qti-ext-enc-info-metadata-cvp.reserved D/CCodecConfig(32251): ignoring local param raw.color (0xc2001809) as it is already supported I/CCodecConfig(32251): query failed after returning 23 values (BAD_INDEX) D/CCodecConfig(32251): c2 config diff is Dict { D/CCodecConfig(32251): c2::u32 algo.bitrate-mode.value = 3 D/CCodecConfig(32251): c2::u32 algo.complexity.value = 100 D/CCodecConfig(32251): c2::i32 algo.priority.value = 0 D/CCodecConfig(32251): c2::float algo.rate.value = 30 D/CCodecConfig(32251): c2::u32 algo.secure-mode.value = 0 D/CCodecConfig(32251): c2::u32 coded.bitrate.value = 20000000 D/CCodecConfig(32251): c2::float coded.frame-rate.value = 30 D/CCodecConfig(32251): c2::u32 coded.pl.level = 0 D/CCodecConfig(32251): c2::u32 coded.pl.profile = 20484 D/CCodecConfig(32251): c2::u32 coded.vui.color.matrix = 255 D/CCodecConfig(32251): c2::u32 coded.vui.color.primaries = 0 D/CCodecConfig(32251): c2::u32 coded.vui.color.range = 0 D/CCodecConfig(32251): c2::u32 coded.vui.color.transfer = 255 D/CCodecConfig(32251): c2::u32 coding.intra-refresh.mode = 0 D/CCodecConfig(32251): c2::float coding.intra-refresh.period = 0 D/CCodecConfig(32251): c2::u32 coding.request-sync-frame.value = 0 D/CCodecConfig(32251): c2::i64 coding.sync-frame-interval.value = 1000000 D/CCodecConfig(32251): Buffer coding.temporal-layering = { D/CCodecConfig(32251): 00000000: 10 00 00 00 07 20 01 52 00 00 00 00 00 00 00 00 ..... .R........ D/CCodecConfig(32251): } D/CCodecConfig(32251): c2::u32 default.color.matrix = 0 D/CCodecConfig(32251): c2::u32 default.color.primaries = 0 D/CCodecConfig(32251): c2::u32 default.color.range = 0 D/CCodecConfig(32251): c2::u32 default.color.transfer = 0 D/CCodecConfig(32251): c2::u32 input.delay.value = 0 D/CCodecConfig(32251): string input.media-type.value = W/ColorUtils(32251): expected specified color aspects (0:0:255:255) I/org.webrtc.Logging(32251): HardwareVideoEncoder: Format: {color-format=2130708361, i-frame-interval=20, mime=video/avc, width=1280, bitrate-mode=2, bitrate=300000, frame-rate=60.0, height=720} I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: AudioRecord.getMinBufferSize: 3840 I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: bufferSizeInBytes: 7680 I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: createAudioRecordOnMOrHigher D/CodecSeeding(32251): Seed: codec c2.qti.avc.encoder, mediatype video/avc, overrideable 1 D/CodecProperties(32251): setTuningValue(vq-target-bpp,0) D/CodecProperties(32251): setTuningValue(vq-target-bpp-1080p,1.90) D/CodecProperties(32251): setTuningValue(vq-target-bpp-720p,2.25) D/CodecProperties(32251): setTuningValue(vq-target-bpp-540p,2.65) D/CodecProperties(32251): setTuningValue(vq-target-bpp-480p,3.00) D/CodecProperties(32251): setTuningValue(vq-target-bpp-320x240,0) D/CodecProperties(32251): setTuningValue(vq-target-qpmax,-1) D/CodecProperties(32251): setTuningValue(vq-target-qpmax-1080p,45) D/CodecProperties(32251): setTuningValue(vq-target-qpmax-720p,43) D/CodecProperties(32251): setTuningValue(vq-target-qpmax-540p,42) D/CodecProperties(32251): setTuningValue(vq-target-qpmax-480p,38) D/CodecProperties(32251): setTuningValue(vq-bitrate-phaseout,1.75) D/CodecProperties(32251): setTuningValue(vq-boost-missing-qp,0.20) D/CodecProperties(32251): setFeatureValue(intra-refresh,0) D/CodecProperties(32251): setFeatureValue(video-minimum-quality,0) D/CodecProperties(32251): setFeatureValue(_vq_eligible.device,1) D/CodecProperties(32251): setFeatureValue(_quality.target,1) D/CodecSeeding(32251): Seed: codec c2.qti.avc.encoder, mediatype video/avc, overrideable 0 D/VQApply (32251): minquality: applies only to VBR encoding D/MediaCodec(32251): shapeMediaFormat: deltas(0): AMessage(what = 0x00000000) = { D/MediaCodec(32251): } D/CCodec (32251): [c2.qti.avc.encoder] buffers are bound to CCodec for this session I/CCodec (32251): appPid(32251) width(1280) height(720) I/CCodec (32251): app-name: com.futureteen.chat I/CCodec (32251): Set content adaptive mode (0) I/CCodec (32251): VideoController::setValues() isSet [0] W/CCodec (32251): can't get ro.hardware.chipname I/CCodec (32251): set values for [taro] D/CCodecConfig(32251): no c2 equivalents for color-format D/CCodecConfig(32251): no c2 equivalents for flags D/CCodecConfig(32251): no c2 equivalents for encoder D/CCodecConfig(32251): c2 config diff is c2::u32 algo.bitrate-mode.value = 1 D/CCodecConfig(32251): c2::u32 coded.bitrate.value = 300000 D/CCodecConfig(32251): c2::float coded.frame-rate.value = 60 D/CCodecConfig(32251): c2::i64 coding.sync-frame-interval.value = 20000000 D/CCodecConfig(32251): c2::u32 raw.pixel-format.value = 34 D/CCodecConfig(32251): c2::u32 raw.size.height = 720 D/CCodecConfig(32251): c2::u32 raw.size.width = 1280 W/ColorUtils(32251): expected specified color aspects (0:0:255:255) W/Codec2Client(32251): query -- param skipped: index = 3254781982. D/CCodec (32251): setup formats input: AMessage(what = 0x00000000) = { D/CCodec (32251): int32_t android._color-format = 2130708361 D/CCodec (32251): Rect crop(0, 0, 1279, 719) D/CCodec (32251): int32_t color-standard = 130816 D/CCodec (32251): int32_t color-range = 0 D/CCodec (32251): int32_t color-transfer = 65791 D/CCodec (32251): int32_t width = 1280 D/CCodec (32251): int32_t android._dataspace = 12648448 D/CCodec (32251): int32_t feature-secure-playback = 0 D/CCodec (32251): int32_t frame-rate = 60 D/CCodec (32251): int32_t height = 720 D/CCodec (32251): int32_t intra-refresh-period = 0 D/CCodec (32251): string mime = "video/raw" D/CCodec (32251): int32_t prepend-sps-pps-to-idr-frames = 0 D/CCodec (32251): int32_t priority = 0 D/CCodec (32251): Buffer hdr-static-info = { D/CCodec (32251): 00000000: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................ D/CCodec (32251): 00000010: 00 00 00 00 00 00 00 00 00 ......... D/CCodec (32251): } D/CCodec (32251): int64_t android._C2MemoryUsage = 4503599631631104 D/CCodec (32251): int32_t color-format = 2130708361 D/CCodec (32251): } D/CCodec (32251): setup formats output: AMessage(what = 0x00000000) = { D/CCodec (32251): int32_t bitrate = 300000 D/CCodec (32251): int32_t bitrate-mode = 2 D/CCodec (32251): Rect crop(0, 0, 1279, 719) D/CCodec (32251): int32_t width = 1280 D/CCodec (32251): int32_t color-standard = 130816 D/CCodec (32251): int32_t color-range = 0 D/CCodec (32251): int32_t color-transfer = 65791 D/CCodec (32251): int32_t feature-secure-playback = 0 D/CCodec (32251): int32_t frame-rate = 60 D/CCodec (32251): int32_t height = 720 D/CCodec (32251): int32_t intra-refresh-period = 0 D/CCodec (32251): int32_t max-bitrate = 300000 D/CCodec (32251): string mime = "video/avc" D/CCodec (32251): int32_t prepend-sps-pps-to-idr-frames = 0 D/CCodec (32251): int32_t priority = 0 D/CCodec (32251): int32_t profile = 8 D/CCodec (32251): } I/CCodecConfig(32251): query failed after returning 24 values (BAD_INDEX) I/org.webrtc.Logging(32251): EglBase14Impl: Using OpenGL ES version 2 D/CCodec (32251): input format changed to AMessage(what = 0x00000000) = { D/CCodec (32251): int32_t android._color-format = 2130708361 D/CCodec (32251): Rect crop(0, 0, 1279, 719) D/CCodec (32251): int32_t color-standard = 1 D/CCodec (32251): int32_t color-range = 2 D/CCodec (32251): int32_t color-transfer = 3 D/CCodec (32251): int32_t width = 1280 D/CCodec (32251): int32_t android._dataspace = 260 D/CCodec (32251): int32_t feature-secure-playback = 0 D/CCodec (32251): int32_t frame-rate = 60 D/CCodec (32251): int32_t height = 720 D/CCodec (32251): int32_t intra-refresh-period = 0 D/CCodec (32251): string mime = "video/raw" D/CCodec (32251): int32_t prepend-sps-pps-to-idr-frames = 0 D/CCodec (32251): int32_t priority = 0 D/CCodec (32251): Buffer hdr-static-info = { D/CCodec (32251): 00000000: 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 00 ................ D/CCodec (32251): 00000010: 00 00 00 00 00 00 00 00 00 ......... D/CCodec (32251): } D/CCodec (32251): int64_t android._C2MemoryUsage = 4503599631631104 D/CCodec (32251): int32_t color-format = 2130708361 D/CCodec (32251): } D/CCodec (32251): ISConfig not changed I/CCodec (32251): state->set(STARTING) W/Codec2Client(32251): query -- param skipped: index = 1342179345. W/Codec2Client(32251): query -- param skipped: index = 2415921170. W/Codec2Client(32251): query -- param skipped: index = 1610614798. D/CCodecBufferChannel(32251): [c2.qti.avc.encoder#634] Query input allocators returned 0 params => BAD_INDEX (6) D/CCodecBufferChannel(32251): [c2.qti.avc.encoder#634] Using basic input block pool with poolID 1 => got 1 - OK (0) D/CCodecBufferChannel(32251): [c2.qti.avc.encoder#634] Query output allocators returned 0 params => BAD_INDEX (6) I/CCodecBufferChannel(32251): [c2.qti.avc.encoder#634] Created output block pool with allocatorID 16 => poolID 989 - OK D/CCodecBufferChannel(32251): [c2.qti.avc.encoder#634] Configured output block pool ids 989 => OK I/CCodec (32251): state->set(RUNNING) I/MediaCodec(32251): setCodecState state(0), called in 6 I/org.webrtc.Logging(32251): WebRtcAudioEffectsExternal: enable(audioSession=13537) I/org.webrtc.Logging(32251): WebRtcAudioEffectsExternal: AcousticEchoCanceler: was enabled, enable: true, is now: enabled I/org.webrtc.Logging(32251): WebRtcAudioEffectsExternal: NoiseSuppressor: was disabled, enable: true, is now: enabled I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: AudioRecord: session ID: 13537, channels: 1, sample rate: 48000 I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: AudioRecord: buffer size in frames: 3840 I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: Number of active recording sessions: 0 I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: startRecording I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: scheduleLogRecordingConfigurationsTask I/Choreographer(32251): Skipped 47 frames! The application may be doing too much work on its main thread. I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: AudioRecordThread@[name=AudioRecordJavaThread, id=180] I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: doAudioRecordStateCallback: START I/flutter (32251): onConnectionState RTCPeerConnectionState.RTCPeerConnectionStateConnected I/flutter (32251): MediaStreamTrack:enableSpeakerphone true I/flutter (32251): onIceConnectionState RTCIceConnectionState.RTCIceConnectionStateConnected W/futureteen.cha(32251): Long monitor contention with owner Binder:32251_6 (631) at int android.media.AudioManager.updateAudioPortCache(java.util.ArrayList, java.util.ArrayList, java.util.ArrayList)(AudioManager.java:6525) waiters=0 in int android.media.AudioManager.resetAudioPortGeneration() for 727ms I/org.webrtc.Logging(32251): HardwareVideoEncoder: Releasing MediaCodec on output thread I/CCodec (32251): state->set(STOPPING) I/CCodec (32251): state->set(ALLOCATED) I/CCodec (32251): state->set(RELEASING) I/CCodec (32251): [c2.qti.avc.encoder] release(1) I/CCodec (32251): state->set(RELEASED) I/MediaCodec(32251): Codec shutdown complete I/hw-BpHwBinder(32251): onLastStrongRef automatically unlinking death recipients I/org.webrtc.Logging(32251): HardwareVideoEncoder: Release on output thread done E/org.webrtc.Logging(32251): HardwareVideoEncoder: MediaCodec is only tested with resolutions that are 16x16 aligned. E/org.webrtc.Logging(32251): HardwareVideoEncoder: MediaCodec is only tested with resolutions that are 16x16 aligned. I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: Number of active recording sessions: 1 I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: AudioRecordingConfigurations: I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: client audio source=VOICE_COMMUNICATION, client session id=13537 (13537) I/org.webrtc.Logging(32251): Device AudioFormat: channel count=1, channel index mask=0, channel mask=IN_MONO, encoding=PCM_16BIT, sample rate=48000 I/org.webrtc.Logging(32251): Client AudioFormat: channel count=1, channel index mask=0, channel mask=IN_MONO, encoding=PCM_16BIT, sample rate=48000 I/org.webrtc.Logging(32251): AudioDevice: type=TYPE_BUILTIN_MIC, id=25 I/org.webrtc.Logging(32251): WebRtcAudioRecordExternal: verifyAudioConfig: PASS I/org.webrtc.Logging(32251): EglRenderer: Duration: 4003 ms. Frames received: 119. Dropped: 0. Rendered: 119. Render fps: 29.7. Average render time: 1388 us. Average swapBuffer time: 766 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): NetworkMonitorAutoDetect: Network becomes available: 200 I/org.webrtc.Logging(32251): NetworkMonitorAutoDetect: capabilities changed: [ Transports: CELLULAR Capabilities: SUPL&INTERNET&NOT_RESTRICTED&TRUSTED&NOT_VPN&NOT_ROAMING&FOREGROUND&NOT_CONGESTED&NOT_SUSPENDED&NOT_VCN_MANAGED LinkUpBandwidth>=1800Kbps LinkDnBandwidth>=4300Kbps Specifier: <TelephonyNetworkSpecifier [mSubId = 1]> UnderlyingNetworks: Null] I/org.webrtc.Logging(32251): NetworkMonitorAutoDetect: link properties changed I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): NetworkMonitorAutoDetect: capabilities changed: [ Transports: CELLULAR Capabilities: SUPL&INTERNET&NOT_RESTRICTED&TRUSTED&NOT_VPN&NOT_ROAMING&FOREGROUND&NOT_CONGESTED&NOT_SUSPENDED&NOT_VCN_MANAGED LinkUpBandwidth>=1705Kbps LinkDnBandwidth>=4305Kbps Specifier: <TelephonyNetworkSpecifier [mSubId = 1]> UnderlyingNetworks: Null] I/org.webrtc.Logging(32251): EglRenderer: Duration: 4003 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1634 us. Average swapBuffer time: 972 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): NetworkMonitorAutoDetect: capabilities changed: [ Transports: CELLULAR Capabilities: SUPL&INTERNET&NOT_RESTRICTED&TRUSTED&NOT_VPN&VALIDATED&NOT_ROAMING&FOREGROUND&NOT_CONGESTED&NOT_SUSPENDED&NOT_VCN_MANAGED LinkUpBandwidth>=1705Kbps LinkDnBandwidth>=4305Kbps Specifier: <TelephonyNetworkSpecifier [mSubId = 1]> UnderlyingNetworks: Null] I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. E/org.webrtc.Logging(32251): HardwareVideoEncoder: MediaCodec is only tested with resolutions that are 16x16 aligned. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. E/org.webrtc.Logging(32251): HardwareVideoEncoder: MediaCodec is only tested with resolutions that are 16x16 aligned. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4003 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1776 us. Average swapBuffer time: 1032 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4004 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): NetworkMonitorAutoDetect: capabilities changed: [ Transports: CELLULAR Capabilities: SUPL&INTERNET&NOT_RESTRICTED&TRUSTED&NOT_VPN&VALIDATED&NOT_ROAMING&FOREGROUND&NOT_CONGESTED&NOT_SUSPENDED&NOT_VCN_MANAGED LinkUpBandwidth>=1800Kbps LinkDnBandwidth>=4300Kbps Specifier: <TelephonyNetworkSpecifier [mSubId = 1]> UnderlyingNetworks: Null] I/org.webrtc.Logging(32251): EglRenderer: Duration: 4003 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1718 us. Average swapBuffer time: 1022 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. E/org.webrtc.Logging(32251): HardwareVideoEncoder: MediaCodec is only tested with resolutions that are 16x16 aligned. E/org.webrtc.Logging(32251): HardwareVideoEncoder: MediaCodec is only tested with resolutions that are 16x16 aligned. I/flutter (32251): onIceConnectionState RTCIceConnectionState.RTCIceConnectionStateCompleted I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4003 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4002 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1669 us. Average swapBuffer time: 997 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4003 ms. Frames received: 119. Dropped: 0. Rendered: 119. Render fps: 29.7. Average render time: 1681 us. Average swapBuffer time: 1029 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. E/org.webrtc.Logging(32251): HardwareVideoEncoder: MediaCodec is only tested with resolutions that are 16x16 aligned. E/org.webrtc.Logging(32251): HardwareVideoEncoder: MediaCodec is only tested with resolutions that are 16x16 aligned. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): NetworkMonitorAutoDetect: capabilities changed: [ Transports: CELLULAR Capabilities: SUPL&INTERNET&NOT_RESTRICTED&TRUSTED&NOT_VPN&VALIDATED&NOT_ROAMING&FOREGROUND&NOT_CONGESTED&NOT_SUSPENDED&NOT_VCN_MANAGED LinkUpBandwidth>=1705Kbps LinkDnBandwidth>=4305Kbps Specifier: <TelephonyNetworkSpecifier [mSubId = 1]> UnderlyingNetworks: Null] I/org.webrtc.Logging(32251): EglRenderer: Duration: 4006 ms. Frames received: 121. Dropped: 0. Rendered: 121. Render fps: 30.2. Average render time: 1647 us. Average swapBuffer time: 976 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 31. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4006 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1702 us. Average swapBuffer time: 1004 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 31. I/org.webrtc.Logging(32251): VolumeLogger: VOICE_CALL stream volume: 8 (max=8) I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4006 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1707 us. Average swapBuffer time: 1025 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4012 ms. Frames received: 121. Dropped: 0. Rendered: 121. Render fps: 30.2. Average render time: 1566 us. Average swapBuffer time: 963 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4003 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4006 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1778 us. Average swapBuffer time: 1169 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4002 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1616 us. Average swapBuffer time: 989 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4008 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4003 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1737 us. Average swapBuffer time: 1071 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4005 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4004 ms. Frames received: 121. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1623 us. Average swapBuffer time: 974 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4007 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4002 ms. Frames received: 120. Dropped: 0. Rendered: 121. Render fps: 30.2. Average render time: 1712 us. Average swapBuffer time: 1040 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 31. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): NetworkMonitorAutoDetect: capabilities changed: [ Transports: CELLULAR Capabilities: SUPL&INTERNET&NOT_RESTRICTED&TRUSTED&NOT_VPN&VALIDATED&NOT_ROAMING&FOREGROUND&NOT_CONGESTED&NOT_SUSPENDED&NOT_VCN_MANAGED LinkUpBandwidth>=1800Kbps LinkDnBandwidth>=4300Kbps Specifier: <TelephonyNetworkSpecifier [mSubId = 1]> UnderlyingNetworks: Null] I/org.webrtc.Logging(32251): VolumeLogger: VOICE_CALL stream volume: 8 (max=8) I/org.webrtc.Logging(32251): EglRenderer: Duration: 4004 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4002 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1772 us. Average swapBuffer time: 1060 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4003 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4002 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1799 us. Average swapBuffer time: 1085 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4003 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4004 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1575 us. Average swapBuffer time: 941 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4007 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4004 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1668 us. Average swapBuffer time: 1025 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4004 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4005 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1579 us. Average swapBuffer time: 951 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4003 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4003 ms. Frames received: 121. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1650 us. Average swapBuffer time: 1017 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4007 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4011 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 29.9. Average render time: 1633 us. Average swapBuffer time: 1023 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4005 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4004 ms. Frames received: 120. Dropped: 0. Rendered: 121. Render fps: 30.2. Average render time: 1591 us. Average swapBuffer time: 953 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): VolumeLogger: VOICE_CALL stream volume: 8 (max=8) I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 31. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4010 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4004 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1740 us. Average swapBuffer time: 1035 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4005 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1663 us. Average swapBuffer time: 1037 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4005 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4003 ms. Frames received: 121. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1629 us. Average swapBuffer time: 958 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4003 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4003 ms. Frames received: 120. Dropped: 0. Rendered: 121. Render fps: 30.2. Average render time: 1591 us. Average swapBuffer time: 948 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4002 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1722 us. Average swapBuffer time: 1093 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4010 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4003 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1715 us. Average swapBuffer time: 1068 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4007 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA. I/org.webrtc.Logging(32251): EglRenderer: Duration: 4003 ms. Frames received: 120. Dropped: 0. Rendered: 120. Render fps: 30.0. Average render time: 1778 us. Average swapBuffer time: 1054 us. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): CameraStatistics: Camera fps: 30. I/org.webrtc.Logging(32251): VolumeLogger: VOICE_CALL stream volume: 8 (max=8) I/org.webrtc.Logging(32251): EglRenderer: Duration: 4006 ms. Frames received: 0. Dropped: 0. Rendered: 0. Render fps: .0. Average render time: NA. Average swapBuffer time: NA.

Platform information

• Flutter version: 2.8.1
• Plugin version: ^0.8.3
• OS: Android
• OS version: Android 12

Hello,

Is it possible to list all audio sources : headphones, speaker, bluetooth, ... available on the device ? An how can I change from one to another during a call ?

Thank you, Luc

Need help, please tell me:

1. Logs say that the default codec is H264. How to specify VP9 video codec and OPUS audio codec in configuration?
2. And another question: how can you quickly switch the established connection to the "audio only" mode, when we do not send and do not receive the video track? And then re-enable video conferencing?

Describe the bug I think there are several problems.

1. List of remote streams in PeerConnection contains duplicate of each stream;
2. Remote streams in the same List do not disappearing after stopping stream;
3. Looks like C++ lib crashing (check "To Peproduce" for case).

To Reproduce If opposite user turn off and turn on stream (for example webcamera stream) for some amount of time I got the error;

E/rtc (17935): #
E/rtc (17935): # Fatal error in: ../../../../usr/local/google/home/sakal/code/webrtc-aar-release/src/rtc_base/task_queue_libevent.cc, line 178
E/rtc (17935): # last system error: 24
E/rtc (17935): # Check failed: pipe(fds) == 0
E/rtc (17935): #
F/libc (17935): Fatal signal 6 (SIGABRT), code -6 in tid 18053 (worker_thread -)

Expected behavior I expect no crash... Please <3

Platform information Linux 20.04

• Flutter version:

[✓] Flutter (Channel master, 1.23.0-8.0.pre.105, on Linux, locale en_US.UTF-8)
    • Flutter version 1.23.0-8.0.pre.105 at
      /home/firstname_lastname/snap/flutter/common/flutter
    • Framework revision f1013e6873 (6 weeks ago), 2020-10-02 09:50:33 +0800
    • Engine revision 0522ff22cc
    • Dart version 2.11.0 (build 2.11.0-181.0.dev)

 
[✓] Android toolchain - develop for Android devices (Android SDK version 30.0.2)
    • Android SDK at /home/firstname_lastname/Android/Sdk
    • Platform android-30, build-tools 30.0.2
    • Java binary at: /usr/local/android-studio/jre/bin/java
    • Java version OpenJDK Runtime Environment (build
      1.8.0_242-release-1644-b3-6222593)
    • All Android licenses accepted.

[✓] Chrome - develop for the web
    • Chrome at google-chrome

[✓] Android Studio (version 4.0)
    • Android Studio at /usr/local/android-studio
    • Flutter plugin version 50.0.1
    • Dart plugin version 193.7547
    • Java version OpenJDK Runtime Environment (build
      1.8.0_242-release-1644-b3-6222593)

[✓] VS Code (version 1.50.1)
    • VS Code at /usr/share/code
    • Flutter extension version 3.16.0

[✓] Connected device (3 available)
    • Nexus 7 (mobile) • 08f4ceaf   • android-arm    • Android 6.0.1 (API 23)
    • Web Server (web) • web-server • web-javascript • Flutter Tools
    • Chrome (web)     • chrome     • web-javascript • Google Chrome
      86.0.4240.183

• No issues found!

• Plugin version: 0.4.0
• OS: Android&iOS
• OS version:

This migrates the Android side of the plugin to the new Embedding V2 API.

• The V2 APIs make lifecycle a lot more explicit to plugins, thus should allow for improve code structure and predictable memory usage
• There may have been a memory leak observed due to missing cleanup methods in the previous implementation (I've been testing with "leakcanary" and "Don't Keep Activities" both set to on
• The plugin will now lazily initialize the audio manager and most of the RTC classes on the first method call invocation (important when in a multi-engine environment, e.g. when a App utilizes firebase_messaging & flutter-webrtc)
• Small refactors and decouplings have been done

Functionality has been tested locally on a few phones, but I'll appreciate if others can test as well.

Note: My Android Studio installation probably formats the code different than other contributors. Could the project agree on a standard set of rules? Most of the official flutter code uses google-java-format (a plugin in Android Studio) with the Google rules. Maybe this project could use it, too.

cmake windows error.

-- Building for: Visual Studio 16 2019 -- Selecting Windows SDK version 10.0.19041.0 to target Windows 10.0.19045. -- The CXX compiler identification is MSVC 19.29.30146.0 -- Check for working CXX compiler: C:/Program Files (x86)/Microsoft Visual Studio/2019/Community/VC/Tools/MSVC/14.29.30133/bin/Hostx64/x64/cl.exe -- Check for working CXX compiler: C:/Program Files (x86)/Microsoft Visual Studio/2019/Community/VC/Tools/MSVC/14.29.30133/bin/Hostx64/x64/cl.exe -- works -- Detecting CXX compiler ABI info -- Detecting CXX compiler ABI info - done -- Detecting CXX compile features -- Detecting CXX compile features - done CMake Error at CMakeLists.txt:31 (apply_standard_settings): Unknown CMake command "apply_standard_settings".

-- Configuring incomplete, errors occurred!

How to compile flutter-webrtc under windows?

Describe the bug Some null parameters are ignored when trying to use them with the method setParameters, e.g. when trying to discard the limitations of the maxBitrate that was set before. To Reproduce My method for changing bandwidth:

  void setMaxBandwidth(int? bandwidth) {
    _peerConnection?.senders.then((senders) {
      senders.forEach((sender) async {
        var parameters = sender.parameters;
        var encodings = parameters.encodings;

        if (encodings == null || encodings.isEmpty) {
          encodings = List.of([RTCRtpEncoding()]);
        }

        encodings.forEach((encoding) {
          if (bandwidth == null || bandwidth == 0) {
            encoding.maxBitrate = null;
          } else {
            encoding.maxBitrate = bandwidth * 1000;
          }
        });

        parameters.encodings = encodings;
        sender.setParameters(parameters);
      });
    });
  }

It works perfectly when using some numerical value. But it doesn't work if set a null value. It happens because the lib ignores null parameters. If remove checks on the null statement there https://github.com/flutter-webrtc/flutter-webrtc/blob/main/common/darwin/Classes/FlutterWebRTCPlugin.m#L1804-L1822

and there https://github.com/flutter-webrtc/flutter-webrtc/blob/main/android/src/main/java/com/cloudwebrtc/webrtc/PeerConnectionObserver.java#L765-L776

it starts to work.

But I can't fix it for other platforms such as Linux and Windows. I found these checks there https://github.com/flutter-webrtc/flutter-webrtc/blob/main/common/cpp/src/flutter_peerconnection.cc#L586-L615 but the code gets stuck earlier there https://github.com/flutter-webrtc/flutter-webrtc/blob/main/common/cpp/src/flutter_webrtc.cc#L798 and it looks like the lib has an issue with parsing incoming parameters in the C code. I don't have experience with C language and can't fix it by myself.

@cloudwebrtc could you please check it and make the required fixes?

Expected behavior The method setParameters should accept null values for some parameters.

Platform information

• Flutter version:

[✓] Flutter (Channel stable, 3.3.2, on macOS 13.1 22C65 darwin-x64, locale en-UA)
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• No issues found!

• Plugin version: 0.9.18
• OS: Android / iOS / Linux / Windows
• OS version:
  • Android 13
  • iOS 16.2
  • Linux Ubuntu 22.04
  • Windows 11 Pro

I get the issue Unable to RTCPeerConnection::createAnswer: peerConnectionCreateAnswer(): WEBRTC_CREATE_ANSWER_ERROR: PeerConnection cannot create an answer in a state other than have-remote-offer or have-local-pranswer. when I call for create answer like this:

void createAnswer() async {
    RTCSessionDescription? description =
        await _peerConnection?.createAnswer({'offerToReceiveVideo': 1});
    await _peerConnection?.setLocalDescription(description!);
    var data = {
      'type': 'answer',
      'sdp': description?.sdp.toString(),
    };
    sendMessage(data);
  }

can anyone help to solve this issue?

Describe the bug

When building on Linux I get this error:

[  +96 ms] Building Linux application... (completed in 6.1s)
[  +26 ms] (fluffychat:230187): Gtk-WARNING **: 20:33:18.576: Locale not supported by C library.
[        ]      Using the fallback 'C' locale.
[ +131 ms] #
[        ] # Fatal error in: ../../media/engine/adm_helpers.cc, line 39
[        ] # last system error: 0
[        ] # Check failed: 0 == adm->Init() (0 vs. -1)
[  +84 ms] # Failed to initialize the ADM.
[   +2 ms] Error waiting for a debug connection: The log reader stopped unexpectedly, or never started.
[   +1 ms] Error launching application on Linux.

To Reproduce

Build on Ubuntu 22.10 with Flutter 3.3.9

Expected behavior

Just builds

Platform information

• Flutter version: 3.3.9
• Plugin version: 0.9.18
• OS: Ubuntu
• OS version: 22.10

@cloudwebrtc Describe the bug when i stop from controller is working fine To Reproduce

Expected behavior

Platform information

• Flutter version:
• Plugin version:
• OS:
• ios
• OS version:
• 16.2