Fix crash after video call
This commit is contained in:
ganfra
parent
3a683fc4e9
commit
de64df3cdb
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@ -0,0 +1 @@
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Fix crash after video call.
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@ -83,9 +83,9 @@ import java.util.concurrent.TimeUnit
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import javax.inject.Provider
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import kotlin.coroutines.CoroutineContext
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private const val STREAM_ID = "ARDAMS"
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private const val AUDIO_TRACK_ID = "ARDAMSa0"
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private const val VIDEO_TRACK_ID = "ARDAMSv0"
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private const val STREAM_ID = "userMedia"
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private const val AUDIO_TRACK_ID = "${STREAM_ID}a0"
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private const val VIDEO_TRACK_ID = "${STREAM_ID}v0"
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private val DEFAULT_AUDIO_CONSTRAINTS = MediaConstraints()
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class WebRtcCall(
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@ -274,12 +274,77 @@ class WebRtcCall(
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peerConnection = peerConnectionFactory.createPeerConnection(rtcConfig, PeerConnectionObserver(this))
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}
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/**
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* Without consultation
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*/
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fun transferToUser(targetUserId: String, targetRoomId: String?) {
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sessionScope?.launch(dispatcher) {
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mxCall.transfer(
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targetUserId = targetUserId,
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targetRoomId = targetRoomId,
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createCallId = CallIdGenerator.generate(),
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awaitCallId = null
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)
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endCall(sendEndSignaling = false)
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}
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}
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/**
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* With consultation
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*/
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fun transferToCall(transferTargetCall: WebRtcCall) {
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sessionScope?.launch(dispatcher) {
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val newCallId = CallIdGenerator.generate()
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transferTargetCall.mxCall.transfer(
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targetUserId = mxCall.opponentUserId,
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targetRoomId = null,
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createCallId = null,
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awaitCallId = newCallId
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)
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mxCall.transfer(
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targetUserId = transferTargetCall.mxCall.opponentUserId,
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targetRoomId = null,
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createCallId = newCallId,
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awaitCallId = null
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)
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endCall(sendEndSignaling = false)
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transferTargetCall.endCall(sendEndSignaling = false)
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}
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}
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fun acceptIncomingCall() {
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sessionScope?.launch {
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Timber.v("## VOIP acceptIncomingCall from state ${mxCall.state}")
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if (mxCall.state == CallState.LocalRinging) {
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internalAcceptIncomingCall()
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}
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}
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}
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/**
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* Sends a DTMF digit to the other party
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* @param digit The digit (nb. string - '#' and '*' are dtmf too)
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*/
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fun sendDtmfDigit(digit: String) {
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sessionScope?.launch {
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for (sender in peerConnection?.senders.orEmpty()) {
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if (sender.track()?.kind() == "audio" && sender.dtmf()?.canInsertDtmf() == true) {
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try {
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sender.dtmf()?.insertDtmf(digit, 100, 70)
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return@launch
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} catch (failure: Throwable) {
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Timber.v("Fail to send Dtmf digit")
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}
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}
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}
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}
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}
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fun attachViewRenderers(localViewRenderer: SurfaceViewRenderer?, remoteViewRenderer: SurfaceViewRenderer, mode: String?) {
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sessionScope?.launch(dispatcher) {
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Timber.v("## VOIP attachViewRenderers localRendeder $localViewRenderer / $remoteViewRenderer")
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localSurfaceRenderers.addIfNeeded(localViewRenderer)
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remoteSurfaceRenderers.addIfNeeded(remoteViewRenderer)
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sessionScope?.launch(dispatcher) {
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when (mode) {
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VectorCallActivity.INCOMING_ACCEPT -> {
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internalAcceptIncomingCall()
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@ -299,67 +364,31 @@ class WebRtcCall(
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}
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}
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/**
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* Without consultation
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*/
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suspend fun transferToUser(targetUserId: String, targetRoomId: String?) {
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mxCall.transfer(
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targetUserId = targetUserId,
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targetRoomId = targetRoomId,
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createCallId = CallIdGenerator.generate(),
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awaitCallId = null
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)
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endCall(sendEndSignaling = false)
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}
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/**
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* With consultation
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*/
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suspend fun transferToCall(transferTargetCall: WebRtcCall) {
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val newCallId = CallIdGenerator.generate()
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transferTargetCall.mxCall.transfer(
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targetUserId = mxCall.opponentUserId,
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targetRoomId = null,
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createCallId = null,
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awaitCallId = newCallId
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)
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mxCall.transfer(
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targetUserId = transferTargetCall.mxCall.opponentUserId,
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targetRoomId = null,
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createCallId = newCallId,
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awaitCallId = null
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)
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endCall(sendEndSignaling = false)
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transferTargetCall.endCall(sendEndSignaling = false)
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}
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fun acceptIncomingCall() {
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sessionScope?.launch {
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Timber.v("## VOIP acceptIncomingCall from state ${mxCall.state}")
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if (mxCall.state == CallState.LocalRinging) {
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internalAcceptIncomingCall()
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}
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private suspend fun attachViewRenderersInternal() = withContext(dispatcher) {
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// render local video in pip view
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localSurfaceRenderers.forEach { renderer ->
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renderer.get()?.let { pipSurface ->
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pipSurface.setMirror(cameraInUse?.type == CameraType.FRONT)
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// no need to check if already added, addSink is checking that
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localVideoTrack?.addSink(pipSurface)
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}
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}
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/**
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* Sends a DTMF digit to the other party
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* @param digit The digit (nb. string - '#' and '*' are dtmf too)
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*/
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fun sendDtmfDigit(digit: String) {
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for (sender in peerConnection?.senders.orEmpty()) {
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if (sender.track()?.kind() == "audio" && sender.dtmf()?.canInsertDtmf() == true) {
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try {
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sender.dtmf()?.insertDtmf(digit, 100, 70)
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return
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} catch (failure: Throwable) {
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Timber.v("Fail to send Dtmf digit")
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}
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// If remote track exists, then sink it to surface
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remoteSurfaceRenderers.forEach { renderer ->
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renderer.get()?.let { participantSurface ->
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remoteVideoTrack?.addSink(participantSurface)
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}
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}
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}
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fun detachRenderers(renderers: List<SurfaceViewRenderer>?) {
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sessionScope?.launch(dispatcher) {
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detachRenderersInternal(renderers)
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}
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}
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private suspend fun detachRenderersInternal(renderers: List<SurfaceViewRenderer>?) = withContext(dispatcher) {
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Timber.v("## VOIP detachRenderers")
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if (renderers.isNullOrEmpty()) {
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// remove all sinks
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@ -452,24 +481,6 @@ class WebRtcCall(
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})
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}
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private fun attachViewRenderersInternal() {
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// render local video in pip view
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localSurfaceRenderers.forEach { renderer ->
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renderer.get()?.let { pipSurface ->
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pipSurface.setMirror(this.cameraInUse?.type == CameraType.FRONT)
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// no need to check if already added, addSink is checking that
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localVideoTrack?.addSink(pipSurface)
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}
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}
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// If remote track exists, then sink it to surface
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remoteSurfaceRenderers.forEach { renderer ->
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renderer.get()?.let { participantSurface ->
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remoteVideoTrack?.addSink(participantSurface)
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}
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}
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}
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private suspend fun getTurnServer(): TurnServerResponse? {
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return tryOrNull {
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sessionProvider.get()?.callSignalingService()?.getTurnServer()
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@ -580,10 +591,12 @@ class WebRtcCall(
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}
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fun setCaptureFormat(format: CaptureFormat) {
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sessionScope?.launch(dispatcher) {
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Timber.v("## VOIP setCaptureFormat $format")
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videoCapturer?.changeCaptureFormat(format.width, format.height, format.fps)
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currentCaptureFormat = format
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}
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}
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private fun updateMuteStatus() {
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val micShouldBeMuted = micMuted || remoteOnHold
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@ -645,14 +658,18 @@ class WebRtcCall(
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}
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fun muteCall(muted: Boolean) {
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sessionScope?.launch(dispatcher) {
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micMuted = muted
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updateMuteStatus()
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}
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}
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fun enableVideo(enabled: Boolean) {
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sessionScope?.launch(dispatcher) {
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videoMuted = !enabled
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updateMuteStatus()
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}
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}
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fun canSwitchCamera(): Boolean {
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return availableCamera.size > 1
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@ -668,9 +685,10 @@ class WebRtcCall(
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}
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fun switchCamera() {
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sessionScope?.launch(dispatcher) {
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Timber.v("## VOIP switchCamera")
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if (mxCall.state is CallState.Connected && mxCall.isVideoCall) {
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val oppositeCamera = getOppositeCameraIfAny() ?: return
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val oppositeCamera = getOppositeCameraIfAny() ?: return@launch
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videoCapturer?.switchCamera(
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object : CameraVideoCapturer.CameraSwitchHandler {
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// Invoked on success. |isFrontCamera| is true if the new camera is front facing.
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@ -692,6 +710,7 @@ class WebRtcCall(
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)
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}
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}
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}
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private suspend fun createAnswer(): SessionDescription? {
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Timber.w("## VOIP createAnswer")
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@ -718,11 +737,12 @@ class WebRtcCall(
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return currentCaptureFormat
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}
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private fun release() {
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private suspend fun release() {
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listeners.clear()
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mxCall.removeListener(this)
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timer.stop()
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timer.tickListener = null
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detachRenderersInternal(null)
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videoCapturer?.stopCapture()
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videoCapturer?.dispose()
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videoCapturer = null
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@ -736,6 +756,8 @@ class WebRtcCall(
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localAudioTrack = null
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localVideoSource = null
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localVideoTrack = null
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remoteAudioTrack = null
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remoteVideoTrack = null
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cameraAvailabilityCallback = null
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}
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@ -745,7 +767,7 @@ class WebRtcCall(
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if (stream.audioTracks.size > 1 || stream.videoTracks.size > 1) {
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Timber.e("## VOIP StreamObserver weird looking stream: $stream")
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// TODO maybe do something more??
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mxCall.hangUp()
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endCall(true)
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return@launch
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}
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if (stream.audioTracks.size == 1) {
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@ -774,8 +796,9 @@ class WebRtcCall(
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}
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fun endCall(sendEndSignaling: Boolean = true, reason: CallHangupContent.Reason? = null) {
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sessionScope?.launch(dispatcher) {
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if (mxCall.state == CallState.Terminated) {
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return
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return@launch
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}
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// Close tracks ASAP
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localVideoTrack?.setEnabled(false)
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@ -786,10 +809,8 @@ class WebRtcCall(
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}
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val wasRinging = mxCall.state is CallState.LocalRinging
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mxCall.state = CallState.Terminated
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sessionScope?.launch(dispatcher) {
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release()
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onCallEnded(callId)
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}
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if (sendEndSignaling) {
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if (wasRinging) {
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mxCall.reject()
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@ -798,6 +819,7 @@ class WebRtcCall(
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}
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}
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}
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}
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// Call listener
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