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

webrtc/test/fake_videorenderer.h

Index: webrtc/test/fake_videorenderer.h
diff --git a/webrtc/modules/audio_coding/neteq/tools/audio_sink.cc b/webrtc/test/fake_videorenderer.h
similarity index 57%
copy from webrtc/modules/audio_coding/neteq/tools/audio_sink.cc
copy to webrtc/test/fake_videorenderer.h
index fb848bdbd9b53e62af0c2a8cfb7fb3668a8e75c5..ff43fc09e475a03dcd1ff9dcef8b934dd03e105d 100644
--- a/webrtc/modules/audio_coding/neteq/tools/audio_sink.cc
+++ b/webrtc/test/fake_videorenderer.h
@@ -8,14 +8,21 @@
  *  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_