Merge branch 'master' of https://github.com/MoonchildProductions/UXP into _testBranch_test_1

5 files changed, 4 insertions, 11 deletions

diff --git a/dom/media/webrtc/MediaEngine.h b/dom/media/webrtc/MediaEngine.h
index ff2a6e25a..6a6988544 100644
--- a/dom/media/webrtc/MediaEngine.h
+++ b/dom/media/webrtc/MediaEngine.h
@@ -54,11 +54,8 @@ public:
   static const int DEFAULT_169_VIDEO_WIDTH = 1280;
   static const int DEFAULT_169_VIDEO_HEIGHT = 720;
 
-#ifndef MOZ_B2G
   static const int DEFAULT_SAMPLE_RATE = 32000;
-#else
-  static const int DEFAULT_SAMPLE_RATE = 16000;
-#endif
+
   // This allows using whatever rate the graph is using for the
   // MediaStreamTrack. This is useful for microphone data, we know it's already
   // at the correct rate for insertion in the MSG.
diff --git a/dom/media/webrtc/MediaEngineCameraVideoSource.cpp b/dom/media/webrtc/MediaEngineCameraVideoSource.cpp
index a0f31d937..e1e572724 100644
--- a/dom/media/webrtc/MediaEngineCameraVideoSource.cpp
+++ b/dom/media/webrtc/MediaEngineCameraVideoSource.cpp
@@ -325,7 +325,7 @@ MediaEngineCameraVideoSource::SetName(nsString aName)
   VideoFacingModeEnum facingMode = VideoFacingModeEnum::User;
 
   // Set facing mode based on device name.
-#if defined(ANDROID) && !defined(MOZ_WIDGET_GONK)
+#if defined(ANDROID)
   // Names are generated. Example: "Camera 0, Facing back, Orientation 90"
   //
   // See media/webrtc/trunk/webrtc/modules/video_capture/android/java/src/org/
diff --git a/dom/media/webrtc/MediaEngineDefault.cpp b/dom/media/webrtc/MediaEngineDefault.cpp
index 9c97d197f..eb0ac2b6f 100644
--- a/dom/media/webrtc/MediaEngineDefault.cpp
+++ b/dom/media/webrtc/MediaEngineDefault.cpp
@@ -192,7 +192,7 @@ MediaEngineDefaultVideoSource::Start(SourceMediaStream* aStream, TrackID aID,
   mTrackID = aID;
 
   // Start timer for subsequent frames
-#if (defined(MOZ_WIDGET_GONK) || defined(MOZ_WIDGET_ANDROID)) && defined(DEBUG)
+#if defined(MOZ_WIDGET_ANDROID) && defined(DEBUG)
 // emulator debug is very, very slow and has problems dealing with realtime audio inputs
   mTimer->InitWithCallback(this, (1000 / mOpts.mFPS)*10, nsITimer::TYPE_REPEATING_SLACK);
 #else
diff --git a/dom/media/webrtc/MediaEngineWebRTC.cpp b/dom/media/webrtc/MediaEngineWebRTC.cpp
index 1a2dc9a04..a77800424 100644
--- a/dom/media/webrtc/MediaEngineWebRTC.cpp
+++ b/dom/media/webrtc/MediaEngineWebRTC.cpp
@@ -335,7 +335,7 @@ MediaEngineWebRTC::EnumerateAudioDevices(dom::MediaSourceEnum aMediaSource,
   int nDevices = 0;
   mAudioInput->GetNumOfRecordingDevices(nDevices);
   int i;
-#if defined(MOZ_WIDGET_ANDROID) || defined(MOZ_WIDGET_GONK)
+#if defined(MOZ_WIDGET_ANDROID)
   i = 0; // Bug 1037025 - let the OS handle defaulting for now on android/b2g
 #else
   // -1 is "default communications device" depending on OS in webrtc.org code
diff --git a/dom/media/webrtc/MediaEngineWebRTCAudio.cpp b/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
index 0b8796aa8..1e2e13d01 100644
--- a/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
+++ b/dom/media/webrtc/MediaEngineWebRTCAudio.cpp
@@ -741,9 +741,6 @@ MediaEngineWebRTCMicrophoneSource::AllocChannel()
 
       // Check for availability.
       if (!mAudioInput->SetRecordingDevice(mCapIndex)) {
-#ifndef MOZ_B2G
-        // Because of the permission mechanism of B2G, we need to skip the status
-        // check here.
         bool avail = false;
         mAudioInput->GetRecordingDeviceStatus(avail);
         if (!avail) {
@@ -752,7 +749,6 @@ MediaEngineWebRTCMicrophoneSource::AllocChannel()
           }
           return false;
         }
-#endif // MOZ_B2G
 
         // Set "codec" to PCM, 32kHz on 1 channel
         ScopedCustomReleasePtr<webrtc::VoECodec> ptrVoECodec(webrtc::VoECodec::GetInterface(mVoiceEngine));