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

webrtc/video/end_to_end_tests.cc

Index: webrtc/video/end_to_end_tests.cc
diff --git a/webrtc/video/end_to_end_tests.cc b/webrtc/video/end_to_end_tests.cc
index 2f7b7463a041ff82f6e5c3b8a4de69443b469903..29d6c464995e1b753d2f2b0c3987da512a3cced3 100644
--- a/webrtc/video/end_to_end_tests.cc
+++ b/webrtc/video/end_to_end_tests.cc
@@ -24,6 +24,7 @@
 #include "webrtc/call.h"
 #include "webrtc/call/transport_adapter.h"
 #include "webrtc/common_video/include/frame_callback.h"
+#include "webrtc/media/base/fakevideorenderer.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
@@ -1585,6 +1586,7 @@ TEST_F(EndToEndTest, AssignsTransportSequenceNumbers) {
       receive_config->rtp.extensions.clear();
       receive_config->rtp.extensions.push_back(RtpExtension(
           RtpExtension::kTransportSequenceNumberUri, kExtensionId));
+      receive_config->renderer = &fake_renderer_;
     }
 
     test::DirectTransport* CreateSendTransport(Call* sender_call) override {
@@ -1594,6 +1596,7 @@ TEST_F(EndToEndTest, AssignsTransportSequenceNumbers) {
     }
 
    private:
+    test::FakeVideoRenderer fake_renderer_;
     uint32_t first_media_ssrc_;
     std::map<uint32_t, uint32_t> rtx_to_media_ssrcs_;
     RtpExtensionHeaderObserver* observer_;
@@ -2086,6 +2089,7 @@ TEST_F(EndToEndTest, VerifyNackStats) {
         VideoEncoderConfig* encoder_config) override {
       send_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       (*receive_configs)[0].rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
+      (*receive_configs)[0].renderer = &fake_renderer_;
     }
 
     void OnVideoStreamsCreated(
@@ -2099,6 +2103,7 @@ TEST_F(EndToEndTest, VerifyNackStats) {
       EXPECT_TRUE(Wait()) << "Timed out waiting for packet to be NACKed.";
     }
 
+    test::FakeVideoRenderer fake_renderer_;
     rtc::CriticalSection crit_;
     uint64_t sent_rtp_packets_;
     uint16_t dropped_rtp_packet_ GUARDED_BY(&crit_);