From 6c2138c080e8646e22587c0f1f127914b2655e0b Mon Sep 17 00:00:00 2001 From: Paul Adenot Date: Fri, 24 Jul 2015 14:28:16 +0200 Subject: [PATCH] Bug 1156472 - Part 2 - Rename MediaEngineWebRTCAudioSource to MediaEngineWebRTCMicrophoneSource. r=jesup There are now two different possible audio source, so this was getting confusing. --- dom/media/webrtc/MediaEngineWebRTC.cpp | 7 ++- dom/media/webrtc/MediaEngineWebRTC.h | 18 +++++--- dom/media/webrtc/MediaEngineWebRTCAudio.cpp | 48 +++++++++++---------- 3 files changed, 39 insertions(+), 34 deletions(-) diff --git a/dom/media/webrtc/MediaEngineWebRTC.cpp b/dom/media/webrtc/MediaEngineWebRTC.cpp index 8771ef79af7c..34adf582ed88 100644 --- a/dom/media/webrtc/MediaEngineWebRTC.cpp +++ b/dom/media/webrtc/MediaEngineWebRTC.cpp @@ -358,15 +358,14 @@ MediaEngineWebRTC::EnumerateAudioDevices(dom::MediaSourceEnum aMediaSource, strcpy(uniqueId,deviceName); // safe given assert and initialization/error-check } - nsRefPtr aSource; + nsRefPtr aSource; NS_ConvertUTF8toUTF16 uuid(uniqueId); if (mAudioSources.Get(uuid, getter_AddRefs(aSource))) { // We've already seen this device, just append. aASources->AppendElement(aSource.get()); } else { - aSource = new MediaEngineWebRTCAudioSource( - mThread, mVoiceEngine, i, deviceName, uniqueId - ); + aSource = new MediaEngineWebRTCMicrophoneSource(mThread, mVoiceEngine, i, + deviceName, uniqueId); mAudioSources.Put(uuid, aSource); // Hashtable takes ownership. aASources->AppendElement(aSource); } diff --git a/dom/media/webrtc/MediaEngineWebRTC.h b/dom/media/webrtc/MediaEngineWebRTC.h index 105cca560048..8b05480718b6 100644 --- a/dom/media/webrtc/MediaEngineWebRTC.h +++ b/dom/media/webrtc/MediaEngineWebRTC.h @@ -133,13 +133,16 @@ private: void GetCapability(size_t aIndex, webrtc::CaptureCapability& aOut) override; }; -class MediaEngineWebRTCAudioSource : public MediaEngineAudioSource, - public webrtc::VoEMediaProcess, - private MediaConstraintsHelper +class MediaEngineWebRTCMicrophoneSource : public MediaEngineAudioSource, + public webrtc::VoEMediaProcess, + private MediaConstraintsHelper { public: - MediaEngineWebRTCAudioSource(nsIThread* aThread, webrtc::VoiceEngine* aVoiceEnginePtr, - int aIndex, const char* name, const char* uuid) + MediaEngineWebRTCMicrophoneSource(nsIThread* aThread, + webrtc::VoiceEngine* aVoiceEnginePtr, + int aIndex, + const char* name, + const char* uuid) : MediaEngineAudioSource(kReleased) , mVoiceEngine(aVoiceEnginePtr) , mMonitor("WebRTCMic.Monitor") @@ -207,7 +210,7 @@ public: virtual void Shutdown() override; protected: - ~MediaEngineWebRTCAudioSource() { Shutdown(); } + ~MediaEngineWebRTCMicrophoneSource() { Shutdown(); } private: void Init(); @@ -294,7 +297,8 @@ private: // Store devices we've already seen in a hashtable for quick return. // Maps UUID to MediaEngineSource (one set for audio, one for video). nsRefPtrHashtable mVideoSources; - nsRefPtrHashtable mAudioSources; + nsRefPtrHashtable + mAudioSources; }; } diff --git a/dom/media/webrtc/MediaEngineWebRTCAudio.cpp b/dom/media/webrtc/MediaEngineWebRTCAudio.cpp index c85c5710c193..2aca1ecbbf99 100644 --- a/dom/media/webrtc/MediaEngineWebRTCAudio.cpp +++ b/dom/media/webrtc/MediaEngineWebRTCAudio.cpp @@ -41,9 +41,9 @@ extern PRLogModuleInfo* GetMediaManagerLog(); #define LOG_FRAMES(msg) MOZ_LOG(GetMediaManagerLog(), mozilla::LogLevel::Verbose, msg) /** - * Webrtc audio source. + * Webrtc microphone source source. */ -NS_IMPL_ISUPPORTS0(MediaEngineWebRTCAudioSource) +NS_IMPL_ISUPPORTS0(MediaEngineWebRTCMicrophoneSource) // XXX temp until MSG supports registration StaticRefPtr gFarendObserver; @@ -177,7 +177,7 @@ AudioOutputObserver::InsertFarEnd(const AudioDataValue *aBuffer, uint32_t aFrame } void -MediaEngineWebRTCAudioSource::GetName(nsAString& aName) +MediaEngineWebRTCMicrophoneSource::GetName(nsAString& aName) { if (mInitDone) { aName.Assign(mDeviceName); @@ -187,7 +187,7 @@ MediaEngineWebRTCAudioSource::GetName(nsAString& aName) } void -MediaEngineWebRTCAudioSource::GetUUID(nsACString& aUUID) +MediaEngineWebRTCMicrophoneSource::GetUUID(nsACString& aUUID) { if (mInitDone) { aUUID.Assign(mDeviceUUID); @@ -197,10 +197,10 @@ MediaEngineWebRTCAudioSource::GetUUID(nsACString& aUUID) } nsresult -MediaEngineWebRTCAudioSource::Config(bool aEchoOn, uint32_t aEcho, - bool aAgcOn, uint32_t aAGC, - bool aNoiseOn, uint32_t aNoise, - int32_t aPlayoutDelay) +MediaEngineWebRTCMicrophoneSource::Config(bool aEchoOn, uint32_t aEcho, + bool aAgcOn, uint32_t aAGC, + bool aNoiseOn, uint32_t aNoise, + int32_t aPlayoutDelay) { LOG(("Audio config: aec: %d, agc: %d, noise: %d", aEchoOn ? aEcho : -1, @@ -281,9 +281,9 @@ uint32_t MediaEngineWebRTCAudioSource::GetBestFitnessDistance( } nsresult -MediaEngineWebRTCAudioSource::Allocate(const dom::MediaTrackConstraints &aConstraints, - const MediaEnginePrefs &aPrefs, - const nsString& aDeviceId) +MediaEngineWebRTCMicrophoneSource::Allocate(const dom::MediaTrackConstraints &aConstraints, + const MediaEnginePrefs &aPrefs, + const nsString& aDeviceId) { if (mState == kReleased) { if (mInitDone) { @@ -309,7 +309,7 @@ MediaEngineWebRTCAudioSource::Allocate(const dom::MediaTrackConstraints &aConstr } nsresult -MediaEngineWebRTCAudioSource::Deallocate() +MediaEngineWebRTCMicrophoneSource::Deallocate() { bool empty; { @@ -331,7 +331,8 @@ MediaEngineWebRTCAudioSource::Deallocate() } nsresult -MediaEngineWebRTCAudioSource::Start(SourceMediaStream* aStream, TrackID aID) +MediaEngineWebRTCMicrophoneSource::Start(SourceMediaStream *aStream, + TrackID aID) { if (!mInitDone || !aStream) { return NS_ERROR_FAILURE; @@ -384,7 +385,7 @@ MediaEngineWebRTCAudioSource::Start(SourceMediaStream* aStream, TrackID aID) } nsresult -MediaEngineWebRTCAudioSource::Stop(SourceMediaStream *aSource, TrackID aID) +MediaEngineWebRTCMicrophoneSource::Stop(SourceMediaStream *aSource, TrackID aID) { { MonitorAutoLock lock(mMonitor); @@ -421,17 +422,17 @@ MediaEngineWebRTCAudioSource::Stop(SourceMediaStream *aSource, TrackID aID) } void -MediaEngineWebRTCAudioSource::NotifyPull(MediaStreamGraph* aGraph, - SourceMediaStream *aSource, - TrackID aID, - StreamTime aDesiredTime) +MediaEngineWebRTCMicrophoneSource::NotifyPull(MediaStreamGraph *aGraph, + SourceMediaStream *aSource, + TrackID aID, + StreamTime aDesiredTime) { // Ignore - we push audio data LOG_FRAMES(("NotifyPull, desired = %ld", (int64_t) aDesiredTime)); } void -MediaEngineWebRTCAudioSource::Init() +MediaEngineWebRTCMicrophoneSource::Init() { mVoEBase = webrtc::VoEBase::GetInterface(mVoiceEngine); @@ -496,7 +497,7 @@ MediaEngineWebRTCAudioSource::Init() } void -MediaEngineWebRTCAudioSource::Shutdown() +MediaEngineWebRTCMicrophoneSource::Shutdown() { if (!mInitDone) { // duplicate these here in case we failed during Init() @@ -551,9 +552,10 @@ MediaEngineWebRTCAudioSource::Shutdown() typedef int16_t sample; void -MediaEngineWebRTCAudioSource::Process(int channel, - webrtc::ProcessingTypes type, sample* audio10ms, - int length, int samplingFreq, bool isStereo) +MediaEngineWebRTCMicrophoneSource::Process(int channel, + webrtc::ProcessingTypes type, + sample *audio10ms, int length, + int samplingFreq, bool isStereo) { // On initial capture, throw away all far-end data except the most recent sample // since it's already irrelevant and we want to keep avoid confusing the AEC far-end