Bug 953265: Update webrtc/getUserMedia default audio capture rate to 32KHz r=padenot

dom/media/webrtc/MediaEngine.h

@@ -54,6 +54,12 @@ public:
   static const int DEFAULT_169_VIDEO_HEIGHT = 720;
   static const int DEFAULT_AUDIO_TIMER_MS = 10;
 
+#ifndef MOZ_B2G
+  static const int DEFAULT_SAMPLE_RATE = 32000;
+#else
+  static const int DEFAULT_SAMPLE_RATE = 16000;
+#endif
+
   /* Populate an array of video sources in the nsTArray. Also include devices
    * that are currently unavailable. */
   virtual void EnumerateVideoDevices(dom::MediaSourceEnum,

dom/media/webrtc/MediaEngineDefault.cpp

@@ -26,7 +26,7 @@
 #include "YuvStamper.h"
 #endif
 
-#define AUDIO_RATE 16000
+#define AUDIO_RATE mozilla::MediaEngine::DEFAULT_SAMPLE_RATE
 #define AUDIO_FRAME_LENGTH ((AUDIO_RATE * MediaEngine::DEFAULT_AUDIO_TIMER_MS) / 1000)
 namespace mozilla {
 

dom/media/webrtc/MediaEngineWebRTC.h

@@ -137,6 +137,7 @@ public:
     , mChannel(-1)
     , mInitDone(false)
     , mStarted(false)
+    , mSampleFrequency(MediaEngine::DEFAULT_SAMPLE_RATE)
     , mEchoOn(false), mAgcOn(false), mNoiseOn(false)
     , mEchoCancel(webrtc::kEcDefault)
     , mAGC(webrtc::kAgcDefault)
@@ -225,6 +226,7 @@ private:
   nsString mDeviceName;
   nsCString mDeviceUUID;
 
+  uint32_t mSampleFrequency;
   bool mEchoOn, mAgcOn, mNoiseOn;
   webrtc::EcModes  mEchoCancel;
   webrtc::AgcModes mAGC;

dom/media/webrtc/MediaEngineWebRTCAudio.cpp

@@ -19,9 +19,8 @@
 #define ENCODING "L16"
 #define DEFAULT_PORT 5555
 
-#define SAMPLE_RATE 256000
-#define SAMPLE_FREQUENCY 16000
-#define SAMPLE_LENGTH ((SAMPLE_FREQUENCY*10)/1000)
+#define SAMPLE_RATE(freq) ((freq)*2*8) // bps, 16-bit samples
+#define SAMPLE_LENGTH(freq) (((freq)*10)/1000)
 
 // These are restrictions from the webrtc.org code
 #define MAX_CHANNELS 2
@@ -345,7 +344,7 @@ MediaEngineWebRTCMicrophoneSource::Start(SourceMediaStream *aStream,
   }
 
   AudioSegment* segment = new AudioSegment();
-  aStream->AddAudioTrack(aID, SAMPLE_FREQUENCY, 0, segment, SourceMediaStream::ADDTRACK_QUEUED);
+  aStream->AddAudioTrack(aID, mSampleFrequency, 0, segment, SourceMediaStream::ADDTRACK_QUEUED);
 
   // XXX Make this based on the pref.
   aStream->RegisterForAudioMixing();
@@ -470,6 +469,9 @@ MediaEngineWebRTCMicrophoneSource::Init()
     return;
   }
 
+  mSampleFrequency = MediaEngine::DEFAULT_SAMPLE_RATE;
+  LOG(("%s: sampling rate %u", __FUNCTION__, mSampleFrequency));
+
   // Check for availability.
   ScopedCustomReleasePtr<webrtc::VoEHardware> ptrVoEHw(webrtc::VoEHardware::GetInterface(mVoiceEngine));
   if (!ptrVoEHw || ptrVoEHw->SetRecordingDevice(mCapIndex)) {
@@ -495,9 +497,10 @@ MediaEngineWebRTCMicrophoneSource::Init()
   webrtc::CodecInst codec;
   strcpy(codec.plname, ENCODING);
   codec.channels = CHANNELS;
-  codec.rate = SAMPLE_RATE;
-  codec.plfreq = SAMPLE_FREQUENCY;
-  codec.pacsize = SAMPLE_LENGTH;
+  MOZ_ASSERT(mSampleFrequency == 16000 || mSampleFrequency == 32000);
+  codec.rate = SAMPLE_RATE(mSampleFrequency);
+  codec.plfreq = mSampleFrequency;
+  codec.pacsize = SAMPLE_LENGTH(mSampleFrequency);
   codec.pltype = 0; // Default payload type
 
   if (!ptrVoECodec->SetSendCodec(mChannel, codec)) {

media/webrtc/signaling/src/mediapipeline/MediaPipeline.cpp

@@ -48,6 +48,9 @@
 
 #include "logging.h"
 
+// Should come from MediaEngineWebRTC.h, but that's a pain to include here
+#define DEFAULT_SAMPLE_RATE 32000
+
 using namespace mozilla;
 using namespace mozilla::gfx;
 using namespace mozilla::layers;
@@ -1332,7 +1335,7 @@ void GenericReceiveListener::AddSelf(MediaSegment* segment) {
 MediaPipelineReceiveAudio::PipelineListener::PipelineListener(
     SourceMediaStream * source, TrackID track_id,
     const RefPtr<MediaSessionConduit>& conduit, bool queue_track)
-  : GenericReceiveListener(source, track_id, 16000, queue_track), // XXX rate assumption
+  : GenericReceiveListener(source, track_id, DEFAULT_SAMPLE_RATE, queue_track), // XXX rate assumption
     conduit_(conduit)
 {
   MOZ_ASSERT(track_rate_%100 == 0);