Support for remote audio into tracks

talk/app/webrtc/rtpsenderreceiver_unittest.cc

 /*
  * libjingle
  * Copyright 2012 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,
  *     this list of conditions and the following disclaimer in the documentation
  *     and/or other materials provided with the distribution.
  *  3. The name of the author may not be used to endorse or promote products
  *     derived from this software without specific prior written permission.
  *
  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
  * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
  * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
  * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
 #include <string>
+#include <utility>
 
 #include "talk/app/webrtc/audiotrack.h"
 #include "talk/app/webrtc/mediastream.h"
 #include "talk/app/webrtc/remoteaudiosource.h"
+#include "talk/app/webrtc/remoteaudiotrack.h"
 #include "talk/app/webrtc/rtpreceiver.h"
 #include "talk/app/webrtc/rtpsender.h"
 #include "talk/app/webrtc/streamcollection.h"
 #include "talk/app/webrtc/videosource.h"
 #include "talk/app/webrtc/videotrack.h"
 #include "talk/media/base/fakevideocapturer.h"
 #include "talk/media/base/mediachannel.h"
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/gunit.h"
 
 using ::testing::_;
 using ::testing::Exactly;
 
 static const char kStreamLabel1[] = "local_stream_1";
 static const char kVideoTrackId[] = "video_1";
 static const char kAudioTrackId[] = "audio_1";
 static const uint32_t kVideoSsrc = 98;
 static const uint32_t kVideoSsrc2 = 100;
 static const uint32_t kAudioSsrc = 99;
 static const uint32_t kAudioSsrc2 = 101;
 
 namespace webrtc {
 
 // Helper class to test RtpSender/RtpReceiver.
 class MockAudioProvider : public AudioProviderInterface {
  public:
-  virtual ~MockAudioProvider() {}
+  ~MockAudioProvider() override {}
+
   MOCK_METHOD2(SetAudioPlayout,
                void(uint32_t ssrc,
                     bool enable));
   MOCK_METHOD4(SetAudioSend,
                void(uint32_t ssrc,
                     bool enable,
                     const cricket::AudioOptions& options,
                     cricket::AudioRenderer* renderer));
   MOCK_METHOD2(SetAudioPlayoutVolume, void(uint32_t ssrc, double volume));
+
+  void SetRawAudioSink(uint32_t,
+                       rtc::scoped_ptr<AudioSinkInterface> sink) override {
+    sink_ = std::move(sink);
+  }
+
+ private:
+  rtc::scoped_ptr<AudioSinkInterface> sink_;
 };
 
 // Helper class to test RtpSender/RtpReceiver.
 class MockVideoProvider : public VideoProviderInterface {
  public:
   virtual ~MockVideoProvider() {}
   MOCK_METHOD2(SetCaptureDevice,
                bool(uint32_t ssrc, cricket::VideoCapturer* camera));
   MOCK_METHOD3(SetVideoPlayout,
                void(uint32_t ssrc,
@@ skipping 64 matching lines @@
     audio_rtp_sender_ = nullptr;
   }
 
   void DestroyVideoRtpSender() {
     EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc, NULL)).Times(1);
     EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc, false, _)).Times(1);
     video_rtp_sender_ = nullptr;
   }
 
   void CreateAudioRtpReceiver() {
-    audio_track_ =
-        AudioTrack::Create(kAudioTrackId, RemoteAudioSource::Create().get());
+    audio_track_ = RemoteAudioTrack::Create(
+        kAudioTrackId, RemoteAudioSource::Create(kAudioSsrc, NULL));
     EXPECT_TRUE(stream_->AddTrack(audio_track_));
     EXPECT_CALL(audio_provider_, SetAudioPlayout(kAudioSsrc, true));
     audio_rtp_receiver_ = new AudioRtpReceiver(stream_->GetAudioTracks()[0],
                                                kAudioSsrc, &audio_provider_);
   }
 
   void CreateVideoRtpReceiver() {
     EXPECT_CALL(video_provider_,
                 SetVideoPlayout(kVideoSsrc, true,
                                 video_track_->GetSource()->FrameInput()));
@@ skipping 315 matching lines @@
                                video_track_->GetSource()->GetVideoCapturer()));
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, true, _));
   sender->SetSsrc(kVideoSsrc2);
 
   // Calls expected from destructor.
   EXPECT_CALL(video_provider_, SetCaptureDevice(kVideoSsrc2, nullptr)).Times(1);
   EXPECT_CALL(video_provider_, SetVideoSend(kVideoSsrc2, false, _)).Times(1);
 }
 
 }  // namespace webrtc