Bug 1487057 - Part 5 - Remove MediaEngineWebRTCMicrophoneSource::mStarted. r=pehrsons

Differential Revision: https://phabricator.services.mozilla.com/D5439

--HG--
extra : rebase_source : 31a0b36e2bc6431a6c7aa387baddfd9d3e35f8f8

dom/media/webrtc/MediaEngineWebRTCAudio.cpp

@@ -128,7 +128,6 @@ MediaEngineWebRTCMicrophoneSource::MediaEngineWebRTCMicrophoneSource(
   , mMutex("WebRTCMic::Mutex")
   , mDelayAgnostic(aDelayAgnostic)
   , mExtendedFilter(aExtendedFilter)
-  , mStarted(false)
   , mDeviceName(aDeviceName)
   , mDeviceUUID(aDeviceUUID)
   , mSettings(
@@ -1004,13 +1003,6 @@ MediaEngineWebRTCMicrophoneSource::PacketizeAndProcess(MediaStreamGraphImpl* aGr
       new AudioPacketizer<AudioDataValue, float>(aRate/100, aChannels);
   }
 
-  // On initial capture, throw away all far-end data except the most recent
-  // sample since it's already irrelevant and we want to avoid confusing the AEC
-  // far-end input code with "old" audio.
-  if (!mStarted) {
-    mStarted  = true;
-  }
-
   // Packetize our input data into 10ms chunks, deinterleave into planar channel
   // buffers, process, and append to the right MediaStreamTrack.
   mPacketizerInput->Input(aBuffer, static_cast<uint32_t>(aFrames));

dom/media/webrtc/MediaEngineWebRTCAudio.h

@@ -266,7 +266,6 @@ private:
 
   bool mDelayAgnostic;
   bool mExtendedFilter;
-  bool mStarted;
 
   const nsString mDeviceName;
   const nsCString mDeviceUUID;

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

@@ -2038,6 +2038,10 @@ private:
   }
 
   RefPtr<MediaSessionConduit> mConduit;
+  // This conduit's sampling rate. This is either 16, 32, 44.1 or 48kHz, and
+  // tries to be the same as the graph rate. If the graph rate is higher than
+  // 48kHz, mRate is capped to 48kHz. If mRate does not match the graph rate,
+  // audio is resampled to the graph rate.
   const TrackRate mRate;
   const RefPtr<TaskQueue> mTaskQueue;
 };