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https://github.com/mozilla/gecko-dev.git
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Bug 1156472 - Part 5 - Add MediaEngineWebRTCAudioCaptureSource as a new audio source, and "audioCapture" as a new MediaSource. r=jesup,bz
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d95134e72e
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6f08789e18
@ -300,7 +300,8 @@ protected:
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NS_IMPL_ISUPPORTS(MediaDevice, nsIMediaDevice)
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MediaDevice::MediaDevice(MediaEngineSource* aSource, bool aIsVideo)
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: mSource(aSource)
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: mMediaSource(aSource->GetMediaSource())
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, mSource(aSource)
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, mIsVideo(aIsVideo)
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{
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mSource->GetName(mName);
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@ -311,9 +312,7 @@ MediaDevice::MediaDevice(MediaEngineSource* aSource, bool aIsVideo)
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VideoDevice::VideoDevice(MediaEngineVideoSource* aSource)
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: MediaDevice(aSource, true)
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{
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mMediaSource = aSource->GetMediaSource();
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}
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{}
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/**
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* Helper functions that implement the constraints algorithm from
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@ -439,6 +438,8 @@ MediaDevice::GetMediaSource(nsAString& aMediaSource)
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{
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if (mMediaSource == dom::MediaSourceEnum::Microphone) {
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aMediaSource.Assign(NS_LITERAL_STRING("microphone"));
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} else if (mMediaSource == dom::MediaSourceEnum::AudioCapture) {
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aMediaSource.Assign(NS_LITERAL_STRING("audioCapture"));
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} else if (mMediaSource == dom::MediaSourceEnum::Window) { // this will go away
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aMediaSource.Assign(NS_LITERAL_STRING("window"));
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} else { // all the rest are shared
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@ -784,11 +785,52 @@ public:
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}
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}
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#endif
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// Create a media stream.
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nsRefPtr<nsDOMUserMediaStream> trackunion =
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nsDOMUserMediaStream::CreateTrackUnionStream(window, mListener,
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mAudioSource, mVideoSource);
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if (!trackunion || sInShutdown) {
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MediaStreamGraph* msg = MediaStreamGraph::GetInstance();
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nsRefPtr<SourceMediaStream> stream = msg->CreateSourceStream(nullptr);
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nsRefPtr<DOMLocalMediaStream> domStream;
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// AudioCapture is a special case, here, in the sense that we're not really
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// using the audio source and the SourceMediaStream, which acts as
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// placeholders. We re-route a number of stream internaly in the MSG and mix
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// them down instead.
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if (mAudioSource &&
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mAudioSource->GetMediaSource() == dom::MediaSourceEnum::AudioCapture) {
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domStream = DOMLocalMediaStream::CreateAudioCaptureStream(window);
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msg->RegisterCaptureStreamForWindow(
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mWindowID, domStream->GetStream()->AsProcessedStream());
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window->SetAudioCapture(true);
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} else {
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// Normal case, connect the source stream to the track union stream to
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// avoid us blocking
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nsRefPtr<nsDOMUserMediaStream> trackunion =
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nsDOMUserMediaStream::CreateTrackUnionStream(window, mListener,
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mAudioSource, mVideoSource);
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trackunion->GetStream()->AsProcessedStream()->SetAutofinish(true);
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nsRefPtr<MediaInputPort> port = trackunion->GetStream()->AsProcessedStream()->
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AllocateInputPort(stream, MediaInputPort::FLAG_BLOCK_OUTPUT);
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trackunion->mSourceStream = stream;
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trackunion->mPort = port.forget();
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// Log the relationship between SourceMediaStream and TrackUnion stream
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// Make sure logger starts before capture
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AsyncLatencyLogger::Get(true);
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LogLatency(AsyncLatencyLogger::MediaStreamCreate,
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reinterpret_cast<uint64_t>(stream.get()),
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reinterpret_cast<int64_t>(trackunion->GetStream()));
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nsCOMPtr<nsIPrincipal> principal;
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if (mPeerIdentity) {
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principal = nsNullPrincipal::Create();
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trackunion->SetPeerIdentity(mPeerIdentity.forget());
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} else {
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principal = window->GetExtantDoc()->NodePrincipal();
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}
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trackunion->CombineWithPrincipal(principal);
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domStream = trackunion.forget();
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}
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if (!domStream || sInShutdown) {
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nsCOMPtr<nsIDOMGetUserMediaErrorCallback> onFailure = mOnFailure.forget();
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LOG(("Returning error for getUserMedia() - no stream"));
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@ -802,36 +844,6 @@ public:
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}
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return NS_OK;
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}
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trackunion->AudioConfig(aec_on, (uint32_t) aec,
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agc_on, (uint32_t) agc,
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noise_on, (uint32_t) noise,
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playout_delay);
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MediaStreamGraph* gm = MediaStreamGraph::GetInstance();
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nsRefPtr<SourceMediaStream> stream = gm->CreateSourceStream(nullptr);
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// connect the source stream to the track union stream to avoid us blocking
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trackunion->GetStream()->AsProcessedStream()->SetAutofinish(true);
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nsRefPtr<MediaInputPort> port = trackunion->GetStream()->AsProcessedStream()->
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AllocateInputPort(stream, MediaInputPort::FLAG_BLOCK_OUTPUT);
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trackunion->mSourceStream = stream;
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trackunion->mPort = port.forget();
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// Log the relationship between SourceMediaStream and TrackUnion stream
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// Make sure logger starts before capture
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AsyncLatencyLogger::Get(true);
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LogLatency(AsyncLatencyLogger::MediaStreamCreate,
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reinterpret_cast<uint64_t>(stream.get()),
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reinterpret_cast<int64_t>(trackunion->GetStream()));
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nsCOMPtr<nsIPrincipal> principal;
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if (mPeerIdentity) {
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principal = nsNullPrincipal::Create();
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trackunion->SetPeerIdentity(mPeerIdentity.forget());
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} else {
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principal = window->GetExtantDoc()->NodePrincipal();
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}
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trackunion->CombineWithPrincipal(principal);
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// The listener was added at the beginning in an inactive state.
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// Activate our listener. We'll call Start() on the source when get a callback
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@ -841,7 +853,7 @@ public:
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// Note: includes JS callbacks; must be released on MainThread
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TracksAvailableCallback* tracksAvailableCallback =
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new TracksAvailableCallback(mManager, mOnSuccess, mWindowID, trackunion);
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new TracksAvailableCallback(mManager, mOnSuccess, mWindowID, domStream);
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mListener->AudioConfig(aec_on, (uint32_t) aec,
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agc_on, (uint32_t) agc,
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@ -852,11 +864,11 @@ public:
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// because that can take a while.
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// Pass ownership of trackunion to the MediaOperationTask
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// to ensure it's kept alive until the MediaOperationTask runs (at least).
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MediaManager::PostTask(FROM_HERE,
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new MediaOperationTask(MEDIA_START, mListener, trackunion,
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tracksAvailableCallback,
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mAudioSource, mVideoSource, false, mWindowID,
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mOnFailure.forget()));
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MediaManager::PostTask(
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FROM_HERE, new MediaOperationTask(MEDIA_START, mListener, domStream,
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tracksAvailableCallback, mAudioSource,
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mVideoSource, false, mWindowID,
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mOnFailure.forget()));
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// We won't need mOnFailure now.
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mOnFailure = nullptr;
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@ -2075,7 +2087,7 @@ StopSharingCallback(MediaManager *aThis,
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listener->Invalidate();
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}
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listener->Remove();
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listener->StopScreenWindowSharing();
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listener->StopSharing();
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}
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aListeners->Clear();
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aThis->RemoveWindowID(aWindowID);
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@ -2398,7 +2410,7 @@ MediaManager::Observe(nsISupports* aSubject, const char* aTopic,
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uint64_t windowID = PromiseFlatString(Substring(data, strlen("screen:"))).ToInteger64(&rv);
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MOZ_ASSERT(NS_SUCCEEDED(rv));
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if (NS_SUCCEEDED(rv)) {
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LOG(("Revoking Screeen/windowCapture access for window %llu", windowID));
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LOG(("Revoking Screen/windowCapture access for window %llu", windowID));
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StopScreensharing(windowID);
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}
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} else {
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@ -2579,7 +2591,7 @@ StopScreensharingCallback(MediaManager *aThis,
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if (aListeners) {
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auto length = aListeners->Length();
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for (size_t i = 0; i < length; ++i) {
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aListeners->ElementAt(i)->StopScreenWindowSharing();
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aListeners->ElementAt(i)->StopSharing();
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}
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}
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}
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@ -2741,7 +2753,7 @@ GetUserMediaCallbackMediaStreamListener::Invalidate()
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// Doesn't kill audio
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// XXX refactor to combine with Invalidate()?
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void
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GetUserMediaCallbackMediaStreamListener::StopScreenWindowSharing()
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GetUserMediaCallbackMediaStreamListener::StopSharing()
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{
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NS_ASSERTION(NS_IsMainThread(), "Only call on main thread");
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if (mVideoSource && !mStopped &&
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@ -2754,6 +2766,13 @@ GetUserMediaCallbackMediaStreamListener::StopScreenWindowSharing()
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this, nullptr, nullptr,
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nullptr, mVideoSource,
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mFinished, mWindowID, nullptr));
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} else if (mAudioSource &&
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mAudioSource->GetMediaSource() == dom::MediaSourceEnum::AudioCapture) {
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nsCOMPtr<nsPIDOMWindow> window = nsGlobalWindow::GetInnerWindowWithId(mWindowID);
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MOZ_ASSERT(window);
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window->SetAudioCapture(false);
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MediaStreamGraph::GetInstance()->UnregisterCaptureStreamForWindow(mWindowID);
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mStream->Destroy();
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}
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}
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@ -103,7 +103,7 @@ public:
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return mStream->AsSourceStream();
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}
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void StopScreenWindowSharing();
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void StopSharing();
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void StopTrack(TrackID aID, bool aIsAudio);
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@ -291,6 +291,13 @@ MediaEngineWebRTC::EnumerateAudioDevices(dom::MediaSourceEnum aMediaSource,
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// We spawn threads to handle gUM runnables, so we must protect the member vars
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MutexAutoLock lock(mMutex);
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if (aMediaSource == dom::MediaSourceEnum::AudioCapture) {
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nsRefPtr<MediaEngineWebRTCAudioCaptureSource> audioCaptureSource =
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new MediaEngineWebRTCAudioCaptureSource(nullptr);
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aASources->AppendElement(audioCaptureSource);
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return;
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}
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#ifdef MOZ_WIDGET_ANDROID
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jobject context = mozilla::AndroidBridge::Bridge()->GetGlobalContextRef();
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@ -358,7 +365,7 @@ MediaEngineWebRTC::EnumerateAudioDevices(dom::MediaSourceEnum aMediaSource,
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strcpy(uniqueId,deviceName); // safe given assert and initialization/error-check
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}
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nsRefPtr<MediaEngineWebRTCMicrophoneSource> aSource;
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nsRefPtr<MediaEngineAudioSource> aSource;
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NS_ConvertUTF8toUTF16 uuid(uniqueId);
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if (mAudioSources.Get(uuid, getter_AddRefs(aSource))) {
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// We've already seen this device, just append.
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@ -384,9 +391,8 @@ ClearVideoSource (const nsAString&, // unused
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}
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static PLDHashOperator
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ClearAudioSource (const nsAString&, // unused
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MediaEngineWebRTCAudioSource* aData,
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void *userArg)
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ClearAudioSource(const nsAString &, // unused
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MediaEngineAudioSource *aData, void *userArg)
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{
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if (aData) {
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aData->Shutdown();
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@ -133,6 +133,67 @@ private:
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void GetCapability(size_t aIndex, webrtc::CaptureCapability& aOut) override;
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};
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class MediaEngineWebRTCAudioCaptureSource : public MediaEngineAudioSource
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{
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public:
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NS_DECL_THREADSAFE_ISUPPORTS
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explicit MediaEngineWebRTCAudioCaptureSource(const char* aUuid)
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: MediaEngineAudioSource(kReleased)
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{
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}
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void GetName(nsAString& aName) override;
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void GetUUID(nsACString& aUUID) override;
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nsresult Allocate(const dom::MediaTrackConstraints& aConstraints,
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const MediaEnginePrefs& aPrefs,
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const nsString& aDeviceId) override
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{
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// Nothing to do here, everything is managed in MediaManager.cpp
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return NS_OK;
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}
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nsresult Deallocate() override
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{
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// Nothing to do here, everything is managed in MediaManager.cpp
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return NS_OK;
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}
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void Shutdown() override
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{
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// Nothing to do here, everything is managed in MediaManager.cpp
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}
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nsresult Start(SourceMediaStream* aMediaStream, TrackID aId) override;
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nsresult Stop(SourceMediaStream* aMediaStream, TrackID aId) override;
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void SetDirectListeners(bool aDirect) override
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{}
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nsresult Config(bool aEchoOn, uint32_t aEcho, bool aAgcOn,
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uint32_t aAGC, bool aNoiseOn, uint32_t aNoise,
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int32_t aPlayoutDelay) override
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{
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return NS_OK;
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}
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void NotifyPull(MediaStreamGraph* aGraph, SourceMediaStream* aSource,
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TrackID aID, StreamTime aDesiredTime) override
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{}
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const dom::MediaSourceEnum GetMediaSource() override
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{
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return dom::MediaSourceEnum::AudioCapture;
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}
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bool IsFake() override
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{
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return false;
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}
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nsresult TakePhoto(PhotoCallback* aCallback) override
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{
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return NS_ERROR_NOT_IMPLEMENTED;
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}
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uint32_t GetBestFitnessDistance(
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const nsTArray<const dom::MediaTrackConstraintSet*>& aConstraintSets,
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const nsString& aDeviceId) override;
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protected:
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virtual ~MediaEngineWebRTCAudioCaptureSource() { Shutdown(); }
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nsCString mUUID;
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};
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class MediaEngineWebRTCMicrophoneSource : public MediaEngineAudioSource,
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public webrtc::VoEMediaProcess,
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private MediaConstraintsHelper
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@ -297,8 +358,7 @@ private:
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// Store devices we've already seen in a hashtable for quick return.
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// Maps UUID to MediaEngineSource (one set for audio, one for video).
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nsRefPtrHashtable<nsStringHashKey, MediaEngineVideoSource> mVideoSources;
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nsRefPtrHashtable<nsStringHashKey, MediaEngineWebRTCMicrophoneSource>
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mAudioSources;
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nsRefPtrHashtable<nsStringHashKey, MediaEngineAudioSource> mAudioSources;
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};
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}
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@ -44,6 +44,7 @@ extern PRLogModuleInfo* GetMediaManagerLog();
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* Webrtc microphone source source.
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*/
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NS_IMPL_ISUPPORTS0(MediaEngineWebRTCMicrophoneSource)
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NS_IMPL_ISUPPORTS0(MediaEngineWebRTCAudioCaptureSource)
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// XXX temp until MSG supports registration
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StaticRefPtr<AudioOutputObserver> gFarendObserver;
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@ -620,4 +621,55 @@ MediaEngineWebRTCMicrophoneSource::Process(int channel,
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return;
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}
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void
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MediaEngineWebRTCAudioCaptureSource::GetName(nsAString &aName)
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{
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aName.AssignLiteral("AudioCapture");
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}
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void
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MediaEngineWebRTCAudioCaptureSource::GetUUID(nsACString &aUUID)
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{
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nsID uuid;
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char uuidBuffer[NSID_LENGTH];
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nsCString asciiString;
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ErrorResult rv;
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rv = nsContentUtils::GenerateUUIDInPlace(uuid);
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if (rv.Failed()) {
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aUUID.AssignLiteral("");
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return;
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}
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uuid.ToProvidedString(uuidBuffer);
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asciiString.AssignASCII(uuidBuffer);
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// Remove {} and the null terminator
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aUUID.Assign(Substring(asciiString, 1, NSID_LENGTH - 3));
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}
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nsresult
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MediaEngineWebRTCAudioCaptureSource::Start(SourceMediaStream *aMediaStream,
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TrackID aId)
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{
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aMediaStream->AddTrack(aId, 0, new AudioSegment());
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return NS_OK;
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}
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nsresult
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MediaEngineWebRTCAudioCaptureSource::Stop(SourceMediaStream *aMediaStream,
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TrackID aId)
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{
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aMediaStream->EndAllTrackAndFinish();
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return NS_OK;
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}
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uint32_t
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MediaEngineWebRTCAudioCaptureSource::GetBestFitnessDistance(
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const nsTArray<const dom::MediaTrackConstraintSet*>& aConstraintSets,
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const nsString& aDeviceId)
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{
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// There is only one way of capturing audio for now, and it's always adequate.
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return 0;
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}
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}
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@ -25,6 +25,7 @@ enum MediaSourceEnum {
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"window",
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"browser",
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"microphone",
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"audioCapture",
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"other"
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};
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