Support for remote audio into tracks

talk/media/base/fakemediaengine.h

Index: talk/media/base/fakemediaengine.h
diff --git a/talk/media/base/fakemediaengine.h b/talk/media/base/fakemediaengine.h
index b1f09aaf64908fb4e62bf001dc42d7ca282c82dc..63d8134fc0b161a51c828820c56ec8de01e81192 100644
--- a/talk/media/base/fakemediaengine.h
+++ b/talk/media/base/fakemediaengine.h
@@ -35,12 +35,13 @@
 #include <vector>
 
 #include "talk/media/base/audiorenderer.h"
+#include "talk/media/base/audiorenderer.h"

[the sun, 2015/12/11 16:32:04]

one too many

[tommi (sloooow) - chröme, 2015/12/11 17:51:18]

Acknowledged.

 #include "talk/media/base/mediaengine.h"
 #include "talk/media/base/rtputils.h"
 #include "talk/media/base/streamparams.h"
-#include "webrtc/p2p/base/sessiondescription.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/stringutils.h"
+#include "webrtc/p2p/base/sessiondescription.h"
 
 namespace cricket {
 
@@ -346,6 +347,12 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
 
   virtual bool GetStats(VoiceMediaInfo* info) { return false; }
 
+  virtual void SetRawAudioSink(
+      uint32_t ssrc,
+      rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
+    sink_ = std::move(sink);

[the sun, 2015/12/11 16:32:04]

If you're doing tests in a separate CL, can you skip adding this infrastructure
until then?

[tommi (sloooow) - chröme, 2015/12/11 17:51:18]

This runs in existing tests.  The code isn't _completely_ without test coverage
:)  Anything that tests a remote track, runs through most of these layers. 
Audio is even sent to the track.  However, there isn't a test that tests the
libjingle interfaces (basically verifying that audio comes out of
AudioTrackSinkInterface).

[the sun, 2015/12/11 19:46:28]

Ah, thanks for explaining. I hadn't looked at the upper libjingle layers in too
much detail.

+  }
+
  private:
   class VoiceChannelAudioSink : public AudioRenderer::Sink {
    public:
@@ -418,6 +425,7 @@ class FakeVoiceMediaChannel : public RtpHelper<VoiceMediaChannel> {
   int time_since_last_typing_;
   AudioOptions options_;
   std::map<uint32_t, VoiceChannelAudioSink*> local_renderers_;
+  rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_;
 };
 
 // A helper function to compare the FakeVoiceMediaChannel::DtmfInfo.