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

webrtc/call/bitrate_estimator_tests.cc

Index: webrtc/call/bitrate_estimator_tests.cc
diff --git a/webrtc/call/bitrate_estimator_tests.cc b/webrtc/call/bitrate_estimator_tests.cc
index 2896c67f37d0dad443f05918c68b0770b8f111e0..84181d26ff5c470ed43a5ce2b7f96a6fc1073385 100644
--- a/webrtc/call/bitrate_estimator_tests.cc
+++ b/webrtc/call/bitrate_estimator_tests.cc
@@ -208,6 +208,7 @@ class BitrateEstimatorTest : public test::CallTest {
         test_->receive_config_.rtp.remote_ssrc =
             test_->video_send_config_.rtp.ssrcs[0];
         test_->receive_config_.rtp.local_ssrc++;
+        test_->receive_config_.renderer = &test->fake_renderer_;
         video_receive_stream_ = test_->receiver_call_->CreateVideoReceiveStream(
             test_->receive_config_.Copy());
         video_receive_stream_->Start();