Removed calls to VoE::SetPlayout() from WebRTCVoiceEngine.

webrtc/media/engine/fakewebrtccall.h

 /*
  *  Copyright (c) 2015 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 61 matching lines @@
 
   const webrtc::AudioReceiveStream::Config& GetConfig() const;
   void SetStats(const webrtc::AudioReceiveStream::Stats& stats);
   int received_packets() const { return received_packets_; }
   bool VerifyLastPacket(const uint8_t* data, size_t length) const;
   const webrtc::AudioSinkInterface* sink() const { return sink_.get(); }
   float gain() const { return gain_; }
   bool DeliverRtp(const uint8_t* packet,
                   size_t length,
                   const webrtc::PacketTime& packet_time);
+  bool started() const { return started_; }
 
  private:
   // webrtc::AudioReceiveStream implementation.
-  void Start() override {}
-  void Stop() override {}
+  void Start() override { started_ = true; }
+  void Stop() override { started_ = false; }
 
   webrtc::AudioReceiveStream::Stats GetStats() const override;
   void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
   void SetGain(float gain) override;
 
   webrtc::AudioReceiveStream::Config config_;
   webrtc::AudioReceiveStream::Stats stats_;
   int received_packets_ = 0;
   std::unique_ptr<webrtc::AudioSinkInterface> sink_;
   float gain_ = 1.0f;
   rtc::Buffer last_packet_;
+  bool started_ = false;
 };
 
 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 137 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_