Added large room scenario to full-stack tests. Added thumbnail streams functionality to call test/v…

webrtc/test/call_test.h

 /*
  *  Copyright (c) 2014 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.
  */
 #ifndef WEBRTC_TEST_CALL_TEST_H_
@@ skipping 56 matching lines @@
   void CreateCalls(const Call::Config& sender_config,
                    const Call::Config& receiver_config);
   void CreateSenderCall(const Call::Config& config);
   void CreateReceiverCall(const Call::Config& config);
   void DestroyCalls();
 
   void CreateSendConfig(size_t num_video_streams,
                         size_t num_audio_streams,
                         size_t num_flexfec_streams,
                         Transport* send_transport);
+
+  void CreateMatchingVideoReceiveConfigs(
+      const VideoSendStream::Config& video_send_config,
+      Transport* rtcp_send_transport);
+
+  // Creates receive configs for video_send_config_,
+  // for flexfec stream and for audio stream.
   void CreateMatchingReceiveConfigs(Transport* rtcp_send_transport);
 
   void CreateFrameGeneratorCapturerWithDrift(Clock* drift_clock,
                                              float speed,
                                              int framerate,
                                              int width,
                                              int height);
   void CreateFrameGeneratorCapturer(int framerate, int width, int height);
   void CreateFakeAudioDevices();
 
@@ skipping 119 matching lines @@
  public:
   explicit EndToEndTest(unsigned int timeout_ms);
 
   bool ShouldCreateReceivers() const override;
 };
 
 }  // namespace test
 }  // namespace webrtc
 
 #endif  // WEBRTC_TEST_CALL_TEST_H_