Add logging statements to places where the frame might be dropped in WebRTC pipeline.
BUG=b/31645554
Review-Url: https://codereview.webrtc.org/2361803003
Cr-Commit-Position: refs/heads/master@{#14457}
diff --git a/webrtc/test/call_test.h b/webrtc/test/call_test.h
index d8019e5..4667e08 100644
--- a/webrtc/test/call_test.h
+++ b/webrtc/test/call_test.h
@@ -17,6 +17,7 @@
#include "webrtc/test/fake_audio_device.h"
#include "webrtc/test/fake_decoder.h"
#include "webrtc/test/fake_encoder.h"
+#include "webrtc/test/fake_videorenderer.h"
#include "webrtc/test/frame_generator_capturer.h"
#include "webrtc/test/rtp_rtcp_observer.h"
@@ -103,6 +104,7 @@
size_t num_video_streams_;
size_t num_audio_streams_;
rtc::scoped_refptr<AudioDecoderFactory> decoder_factory_;
+ test::FakeVideoRenderer fake_renderer_;
private:
// TODO(holmer): Remove once VoiceEngine is fully refactored to the new API.
