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

webrtc/video/end_to_end_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 <algorithm>
 #include <list>
 #include <map>
 #include <memory>
 #include <sstream>
 #include <string>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/optional.h"
 #include "webrtc/base/rate_limiter.h"
 #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"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/nack.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/rapid_resync_request.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
 #include "webrtc/modules/video_coding/codecs/vp8/include/vp8.h"
 #include "webrtc/modules/video_coding/codecs/vp9/include/vp9.h"
@@ skipping 1541 matching lines @@
         first_media_ssrc_ = send_config->rtp.ssrcs[0];
     }
 
     void UpdateReceiveConfig(
         size_t stream_index,
         VideoReceiveStream::Config* receive_config) override {
       receive_config->rtp.nack.rtp_history_ms = kNackRtpHistoryMs;
       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 {
       observer_ = new RtpExtensionHeaderObserver(sender_call, first_media_ssrc_,
                                                  rtx_to_media_ssrcs_);
       return observer_;
     }
 
    private:
+    test::FakeVideoRenderer fake_renderer_;
     uint32_t first_media_ssrc_;
     std::map<uint32_t, uint32_t> rtx_to_media_ssrcs_;
     RtpExtensionHeaderObserver* observer_;
   } tester;
 
   tester.RunTest();
 }
 
 class TransportFeedbackTester : public test::EndToEndTest {
  public:
@@ skipping 472 matching lines @@
       int64_t elapsed_sec = (now - start_runtime_ms_) / 1000;
       return elapsed_sec > metrics::kMinRunTimeInSeconds;
     }
 
     void ModifyVideoConfigs(
         VideoSendStream::Config* send_config,
         std::vector<VideoReceiveStream::Config>* receive_configs,
         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(
         VideoSendStream* send_stream,
         const std::vector<VideoReceiveStream*>& receive_streams) override {
       send_stream_ = send_stream;
       receive_streams_ = receive_streams;
     }
 
     void PerformTest() override {
       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_);
     bool dropped_rtp_packet_requested_ GUARDED_BY(&crit_);
     std::vector<VideoReceiveStream*> receive_streams_;
     VideoSendStream* send_stream_;
     int64_t start_runtime_ms_;
   } test;
 
   metrics::Reset();
@@ skipping 1629 matching lines @@
    private:
     bool video_observed_;
     bool audio_observed_;
     SequenceNumberUnwrapper unwrapper_;
     std::set<int64_t> received_packet_ids_;
   } test;
 
   RunBaseTest(&test);
 }
 }  // namespace webrtc