Prevent a voice channel from sending data before a renderer is set.

webrtc/api/rtpsenderreceiver_unittest.cc

 /*
  *  Copyright 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 32 matching lines @@
  public:
   ~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));
+                    cricket::AudioSource* source));
   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_;
 };
@@ skipping 426 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