Bug 1397793 - Move away from VoEExternalMedia "external" API in AudioConduit.cpp. r=dminor

This forces us to do a copy. It's not the end of the world but could be avoided.
The number of channels received is now explicit (via
`AudioFrame::num_channels_`), instead of being guessed based on the number of
samples (considering we're always dealing with 10ms of audio, and we know the
rate).

It's still coupled a bit with audio devices, but we cheat, and use a "fake audio
device", which isn't going to touch actual OS APIs.

MozReview-Commit-ID: 1Tfajkv1HQR

--HG--
extra : rebase_source : f9ed6f1beeb3745dc17c4e6264808d1918e8906c
extra : histedit_source : 4338aea961b861462caa79afab66ebaea06e40b2

media/webrtc/signaling/src/media-conduit/AudioConduit.h

@@ -17,6 +17,7 @@
 
 // Audio Engine Includes
 #include "webrtc/common_types.h"
+#include "webrtc/modules/audio_device/include/fake_audio_device.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_volume_control.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
@@ -174,6 +175,7 @@ public:
 
   explicit WebrtcAudioConduit():
                       mVoiceEngine(nullptr),
+                      mFakeAudioDevice(new webrtc::FakeAudioDeviceModule()),
                       mTransportMonitor("WebrtcAudioConduit"),
                       mTransmitterTransport(nullptr),
                       mReceiverTransport(nullptr),
@@ -297,6 +299,7 @@ private:
   void DumpCodecDB() const;
 
   webrtc::VoiceEngine* mVoiceEngine;
+  UniquePtr<webrtc::FakeAudioDeviceModule> mFakeAudioDevice;
   mozilla::ReentrantMonitor mTransportMonitor;
   RefPtr<TransportInterface> mTransmitterTransport;
   RefPtr<TransportInterface> mReceiverTransport;
@@ -334,6 +337,8 @@ private:
 
   uint32_t mLastTimestamp;
 
+  webrtc::AudioFrame mAudioFrame; // for output pulls
+
   uint32_t mSamples;
   uint32_t mLastSyncLog;
 

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

@@ -2104,7 +2104,7 @@ public:
       if (source_->AppendToTrack(track_id_, &segment)) {
         played_ticks_ += frames;
         if (MOZ_LOG_TEST(AudioLogModule(), LogLevel::Debug)) {
-          if (played_ticks_ > last_log_ + graph->GraphRate()) { // ~ 1 second
+          if (played_ticks_ > last_log_ + rate) { // ~ 1 second
             MOZ_LOG(AudioLogModule(), LogLevel::Debug,
                     ("%p: Inserting %zu samples into track %d, total = %" PRIu64,
                      (void*) this, frames, track_id_, played_ticks_));

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

@@ -27,7 +27,6 @@
 
 // Should come from MediaEngine.h, but that's a pain to include here
 // because of the MOZILLA_EXTERNAL_LINKAGE stuff.
-#define WEBRTC_DEFAULT_SAMPLE_RATE 32000
 #define WEBRTC_MAX_SAMPLE_RATE 48000
 
 class nsIPrincipal;

media/webrtc/signaling/src/peerconnection/PeerConnectionImpl.cpp

@@ -2269,6 +2269,7 @@ PeerConnectionImpl::InsertAudioLevelForContributingSource(
       break;
     }
   }
+
   return NS_OK;
 }