Support for remote audio into tracks

talk/media/webrtc/fakewebrtccall.h

 /*
  * libjingle
  * Copyright 2015 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 88 matching lines @@
     return true;
   }
   bool DeliverRtp(const uint8_t* packet,
                   size_t length,
                   const webrtc::PacketTime& packet_time) override {
     return true;
   }
 
   // webrtc::AudioReceiveStream implementation.
   webrtc::AudioReceiveStream::Stats GetStats() const override;
+  void SetSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override;
 
   webrtc::AudioReceiveStream::Config config_;
   webrtc::AudioReceiveStream::Stats stats_;
   int received_packets_;
+  rtc::scoped_ptr<webrtc::AudioSinkInterface> sink_;

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

again, I'd rather not add any extra gunk unless it is used or you have a CL
already in the pipe to use it.

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

Already used in existing tests.  Before, there was a class, AudioTrack, which
was used for both local and remote tracks (albeit close to no functionality for
the remote case).  I've split that up into AudioTrack (which continues to do
very little) and RemoteAudioTrack, which does a lot more.  All tests that
exercise the 'remote audio track' code path, do run this code, and it's several
tests.

 };
 
 class FakeVideoSendStream final : public webrtc::VideoSendStream,
                                   public webrtc::VideoCaptureInput {
  public:
   FakeVideoSendStream(const webrtc::VideoSendStream::Config& config,
                       const webrtc::VideoEncoderConfig& encoder_config);
   webrtc::VideoSendStream::Config GetConfig() const;
   webrtc::VideoEncoderConfig GetEncoderConfig() const;
   std::vector<webrtc::VideoStream> GetVideoStreams();
@@ skipping 134 matching lines @@
   std::vector<FakeAudioSendStream*> audio_send_streams_;
   std::vector<FakeVideoReceiveStream*> video_receive_streams_;
   std::vector<FakeAudioReceiveStream*> audio_receive_streams_;
 
   int num_created_send_streams_;
   int num_created_receive_streams_;
 };
 
 }  // namespace cricket
 #endif  // TALK_MEDIA_WEBRTC_WEBRTCVIDEOENGINE2_UNITTEST_H_