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-rw-r--r--dom/media/webrtc/MediaEngine.h5
-rw-r--r--dom/media/webrtc/MediaEngineWebRTCAudio.cpp4
2 files changed, 1 insertions, 8 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/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));