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

webrtc/call/bitrate_estimator_tests.cc

 /*
  *  Copyright (c) 2013 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.
  */
 #include <functional>
@@ skipping 190 matching lines @@
         decoder.decoder = &fake_decoder_;
         decoder.payload_type =
             test_->video_send_config_.encoder_settings.payload_type;
         decoder.payload_name =
             test_->video_send_config_.encoder_settings.payload_name;
         test_->receive_config_.decoders.clear();
         test_->receive_config_.decoders.push_back(decoder);
         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();
       }
       is_sending_receiving_ = true;
     }
 
     ~Stream() {
       EXPECT_FALSE(is_sending_receiving_);
       test_->sender_call_->DestroyVideoSendStream(send_stream_);
@@ skipping 103 matching lines @@
       RtpExtension(RtpExtension::kTimestampOffsetUri, kTOFExtensionId);
   receiver_log_.PushExpectedLogLine(kAbsSendTimeLog);
   receiver_log_.PushExpectedLogLine(
       "WrappingBitrateEstimator: Switching to transmission time offset RBE.");
   streams_.push_back(new Stream(this, false));
   streams_[0]->StopSending();
   streams_[1]->StopSending();
   EXPECT_TRUE(receiver_log_.Wait());
 }
 }  // namespace webrtc