Add logging statements to places where the frame might be dropped in WebRTC pipeline.

webrtc/test/fake_videorenderer.h

 /*
  *  Copyright (c) 2016 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 "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
+#ifndef WEBRTC_TEST_FAKE_VIDEORENDERER_H_
+#define WEBRTC_TEST_FAKE_VIDEORENDERER_H_
+
+#include "webrtc/media/base/videosinkinterface.h"
+#include "webrtc/video_frame.h"
 
 namespace webrtc {
 namespace test {
 
-bool AudioSinkFork::WriteArray(const int16_t* audio, size_t num_samples) {
-  return left_sink_->WriteArray(audio, num_samples) &&
-         right_sink_->WriteArray(audio, num_samples);
-}
+class FakeVideoRenderer : public rtc::VideoSinkInterface<webrtc::VideoFrame> {
+ public:
+  void OnFrame(const webrtc::VideoFrame& frame) override {}
+};
+
 }  // namespace test
 }  // namespace webrtc
+
+#endif  // WEBRTC_TEST_FAKE_VIDEORENDERER_H_