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

webrtc/test/call_test.cc

Index: webrtc/test/call_test.cc
diff --git a/webrtc/test/call_test.cc b/webrtc/test/call_test.cc
index d6cbaa6d7ee02a0c097f62761ea8824edc028df7..a2133fc01fb208536d9e47c27f41b6601c3f5a0e 100644
--- a/webrtc/test/call_test.cc
+++ b/webrtc/test/call_test.cc
@@ -210,6 +210,7 @@ void CallTest::CreateMatchingReceiveConfigs(Transport* rtcp_send_transport) {
     video_config.rtp.local_ssrc = kReceiverLocalVideoSsrc;
     for (const RtpExtension& extension : video_send_config_.rtp.extensions)
       video_config.rtp.extensions.push_back(extension);
+    video_config.renderer = &fake_renderer_;
     for (size_t i = 0; i < video_send_config_.rtp.ssrcs.size(); ++i) {
       VideoReceiveStream::Decoder decoder =
           test::CreateMatchingDecoder(video_send_config_.encoder_settings);