Make sure VideoReceiveStream can be restarted

webrtc/video/end_to_end_tests.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include <algorithm>
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   CreateMatchingReceiveConfigs(&transport);
 
   CreateVideoStreams();
 
   video_receive_streams_[0]->Stop();
   video_receive_streams_[0]->Stop();
 
   DestroyStreams();
 }
 
+TEST_F(EndToEndTest, ReceiverCanBeStoppedAndRestarted) {
+  CreateCalls(Call::Config(&event_log_), Call::Config(&event_log_));
+
+  test::NullTransport transport;
+  CreateSendConfig(1, 0, &transport);
+  CreateMatchingReceiveConfigs(&transport);
+
+  CreateVideoStreams();
+
+  video_receive_streams_[0]->Stop();
+  video_receive_streams_[0]->Start();
+  video_receive_streams_[0]->Stop();
+
+  DestroyStreams();
+}
+
 TEST_F(EndToEndTest, RendersSingleDelayedFrame) {
   static const int kWidth = 320;
   static const int kHeight = 240;
   // This constant is chosen to be higher than the timeout in the video_render
   // module. This makes sure that frames aren't dropped if there are no other
   // frames in the queue.
   static const int kDelayRenderCallbackMs = 1000;
 
   class Renderer : public rtc::VideoSinkInterface<VideoFrame> {
    public:
@@ skipping 3753 matching lines @@
     std::unique_ptr<VideoEncoder> encoder_;
     std::unique_ptr<VideoDecoder> decoder_;
     rtc::CriticalSection crit_;
     int recorded_frames_ GUARDED_BY(crit_);
   } test(this);
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc