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

@@ -358,15 +358,14 @@ MediaEngineWebRTC::EnumerateAudioDevices(dom::MediaSourceEnum aMediaSource,
       strcpy(uniqueId,deviceName); // safe given assert and initialization/error-check
     }
 
-    nsRefPtr<MediaEngineWebRTCAudioSource> aSource;
+    nsRefPtr<MediaEngineWebRTCMicrophoneSource> 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(
+      aSource = new MediaEngineWebRTCMicrophoneSource(mThread, mVoiceEngine, i,
-        mThread, mVoiceEngine, i, deviceName, uniqueId
+                                                      deviceName, uniqueId);
-      );
       mAudioSources.Put(uuid, aSource); // Hashtable takes ownership.
       aASources->AppendElement(aSource);
     }

dom/media/webrtc/MediaEngineWebRTC.h

@@ -133,13 +133,16 @@ private:
   void GetCapability(size_t aIndex, webrtc::CaptureCapability& aOut) override;
 };
 
-class MediaEngineWebRTCAudioSource : public MediaEngineAudioSource,
+class MediaEngineWebRTCMicrophoneSource : public MediaEngineAudioSource,
-                                     public webrtc::VoEMediaProcess,
+                                          public webrtc::VoEMediaProcess,
-                                     private MediaConstraintsHelper
+                                          private MediaConstraintsHelper
 {
 public:
-  MediaEngineWebRTCAudioSource(nsIThread* aThread, webrtc::VoiceEngine* aVoiceEnginePtr,
+  MediaEngineWebRTCMicrophoneSource(nsIThread* aThread,
-                               int aIndex, const char* name, const char* uuid)
+                                    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<nsStringHashKey, MediaEngineVideoSource> mVideoSources;
-  nsRefPtrHashtable<nsStringHashKey, MediaEngineWebRTCAudioSource> mAudioSources;
+  nsRefPtrHashtable<nsStringHashKey, MediaEngineWebRTCMicrophoneSource>
+  mAudioSources;
 };
 
 }

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<AudioOutputObserver> 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,
+MediaEngineWebRTCMicrophoneSource::Config(bool aEchoOn, uint32_t aEcho,
-                                     bool aAgcOn, uint32_t aAGC,
+                                          bool aAgcOn, uint32_t aAGC,
-                                     bool aNoiseOn, uint32_t aNoise,
+                                          bool aNoiseOn, uint32_t aNoise,
-                                     int32_t aPlayoutDelay)
+                                          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,
+MediaEngineWebRTCMicrophoneSource::Allocate(const dom::MediaTrackConstraints &aConstraints,
-                                       const MediaEnginePrefs &aPrefs,
+                                            const MediaEnginePrefs &aPrefs,
-                                       const nsString& aDeviceId)
+                                            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,
+MediaEngineWebRTCMicrophoneSource::NotifyPull(MediaStreamGraph *aGraph,
-                                         SourceMediaStream *aSource,
+                                              SourceMediaStream *aSource,
-                                         TrackID aID,
+                                              TrackID aID,
-                                         StreamTime aDesiredTime)
+                                              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,
+MediaEngineWebRTCMicrophoneSource::Process(int channel,
-  webrtc::ProcessingTypes type, sample* audio10ms,
+                                           webrtc::ProcessingTypes type,
-  int length, int samplingFreq, bool isStereo)
+                                           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