From b7d9dad58e5a3f87a6c767412941700bc8010044 Mon Sep 17 00:00:00 2001 From: wolfbeast Date: Sat, 12 May 2018 14:32:03 +0200 Subject: Remove MOZ_B2G leftovers and some dead B2G-only components. --- dom/media/webrtc/MediaEngine.h | 5 +---- dom/media/webrtc/MediaEngineWebRTCAudio.cpp | 4 ---- 2 files changed, 1 insertion(+), 8 deletions(-) (limited to 'dom/media/webrtc') 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/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 ptrVoECodec(webrtc::VoECodec::GetInterface(mVoiceEngine)); -- cgit v1.2.3 From e16bcd08aae85a7d9c2de5a4b1c733280cb81112 Mon Sep 17 00:00:00 2001 From: wolfbeast Date: Sun, 13 May 2018 00:08:52 +0200 Subject: Remove MOZ_WIDGET_GONK [2/2] Tag #288 --- dom/media/webrtc/MediaEngineCameraVideoSource.cpp | 2 +- dom/media/webrtc/MediaEngineDefault.cpp | 2 +- dom/media/webrtc/MediaEngineWebRTC.cpp | 2 +- 3 files changed, 3 insertions(+), 3 deletions(-) (limited to 'dom/media/webrtc') 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 -- cgit v1.2.3