Add task queue to Call.

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 82 matching lines @@
   webrtc::AudioReceiveStream::Stats stats_;
   int received_packets_ = 0;
   std::unique_ptr<webrtc::AudioSinkInterface> sink_;
   float gain_ = 1.0f;
   rtc::Buffer last_packet_;
 };
 
 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;
+  FakeVideoSendStream(webrtc::VideoSendStream::Config config,
+                      webrtc::VideoEncoderConfig encoder_config);
+  const webrtc::VideoSendStream::Config& GetConfig() const;
+  const webrtc::VideoEncoderConfig& GetEncoderConfig() const;
   std::vector<webrtc::VideoStream> GetVideoStreams();
 
   bool IsSending() const;
   bool GetVp8Settings(webrtc::VideoCodecVP8* settings) const;
   bool GetVp9Settings(webrtc::VideoCodecVP9* settings) const;
 
   int GetNumberOfSwappedFrames() const;
   int GetLastWidth() const;
   int GetLastHeight() const;
   int64_t GetLastTimestamp() const;
   void SetStats(const webrtc::VideoSendStream::Stats& stats);
   int num_encoder_reconfigurations() const {
     return num_encoder_reconfigurations_;
   }
 
  private:
   void IncomingCapturedFrame(const webrtc::VideoFrame& frame) override;
 
   // webrtc::VideoSendStream implementation.
   void Start() override;
   void Stop() override;
   webrtc::VideoSendStream::Stats GetStats() override;
-  void ReconfigureVideoEncoder(
-      const webrtc::VideoEncoderConfig& config) override;
+  void ReconfigureVideoEncoder(webrtc::VideoEncoderConfig config) override;
   webrtc::VideoCaptureInput* Input() override;
 
   bool sending_;
   webrtc::VideoSendStream::Config config_;
   webrtc::VideoEncoderConfig encoder_config_;
   bool codec_settings_set_;
   union VpxSettings {
     webrtc::VideoCodecVP8 vp8;
     webrtc::VideoCodecVP9 vp9;
   } vpx_settings_;
@@ skipping 58 matching lines @@
   webrtc::AudioSendStream* CreateAudioSendStream(
       const webrtc::AudioSendStream::Config& config) override;
   void DestroyAudioSendStream(webrtc::AudioSendStream* send_stream) override;
 
   webrtc::AudioReceiveStream* CreateAudioReceiveStream(
       const webrtc::AudioReceiveStream::Config& config) override;
   void DestroyAudioReceiveStream(
       webrtc::AudioReceiveStream* receive_stream) override;
 
   webrtc::VideoSendStream* CreateVideoSendStream(
-      const webrtc::VideoSendStream::Config& config,
-      const webrtc::VideoEncoderConfig& encoder_config) override;
+      webrtc::VideoSendStream::Config config,
+      webrtc::VideoEncoderConfig encoder_config) override;
   void DestroyVideoSendStream(webrtc::VideoSendStream* send_stream) override;
 
   webrtc::VideoReceiveStream* CreateVideoReceiveStream(
       webrtc::VideoReceiveStream::Config config) override;
   void DestroyVideoReceiveStream(
       webrtc::VideoReceiveStream* receive_stream) override;
   webrtc::PacketReceiver* Receiver() override;
 
   DeliveryStatus DeliverPacket(webrtc::MediaType media_type,
                                const uint8_t* packet,
@@ skipping 24 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_