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

webrtc/test/test.gyp

 # Copyright (c) 2011 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.
 
 # TODO(andrew): consider moving test_support to src/base/test.
 {
@@ skipping 155 matching lines @@
        'encoder_settings.cc',
        'encoder_settings.h',
        'fake_audio_device.cc',
        'fake_audio_device.h',
        'fake_decoder.cc',
        'fake_decoder.h',
        'fake_encoder.cc',
        'fake_encoder.h',
        'fake_network_pipe.cc',
        'fake_network_pipe.h',
+       'fake_videorenderer.h',
        'frame_generator_capturer.cc',
        'frame_generator_capturer.h',
        'layer_filtering_transport.cc',
        'layer_filtering_transport.h',
        'mock_transport.h',
        'mock_voe_channel_proxy.h',
        'mock_voice_engine.h',
        'null_transport.cc',
        'null_transport.h',
        'rtp_rtcp_observer.h',
@@ skipping 97 matching lines @@
                '-framework OpenGL',
                '-framework CoreVideo',
              ],
            },
          }],
        ],
      },
     },
   ],
 }