Measure encoding time on encode callbacks.

webrtc/video/video_quality_test.cc

 /*
  *  Copyright (c) 2015 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 <stdio.h>
@@ skipping 26 matching lines @@
 namespace webrtc {
 
 static const int kSendStatsPollingIntervalMs = 1000;
 static const int kPayloadTypeVP8 = 123;
 static const int kPayloadTypeVP9 = 124;
 
 class VideoAnalyzer : public PacketReceiver,
                       public Transport,
                       public VideoRenderer,
                       public VideoCaptureInput,
-                      public EncodedFrameObserver,
-                      public EncodingTimeObserver {
+                      public EncodedFrameObserver {
  public:
   VideoAnalyzer(test::LayerFilteringTransport* transport,
                 const std::string& test_label,
                 double avg_psnr_threshold,
                 double avg_ssim_threshold,
                 int duration_frames,
                 FILE* graph_data_output_file,
                 const std::string& graph_title,
                 uint32_t ssrc_to_analyze)
       : input_(nullptr),
@@ skipping 60 matching lines @@
     parser.Parse(&header);
     {
       rtc::CritScope lock(&crit_);
       recv_times_[header.timestamp - rtp_timestamp_delta_] =
           Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
     }
 
     return receiver_->DeliverPacket(media_type, packet, length, packet_time);
   }
 
-  // EncodingTimeObserver.
-  void OnReportEncodedTime(int64_t ntp_time_ms, int encode_time_ms) override {
+  void OnEncodeTiming(int64_t ntp_time_ms, int encode_time_ms) override {
     rtc::CritScope crit(&comparison_lock_);
     samples_encode_time_ms_[ntp_time_ms] = encode_time_ms;
   }
 
   void IncomingCapturedFrame(const VideoFrame& video_frame) override {
     VideoFrame copy = video_frame;
     copy.set_timestamp(copy.ntp_time_ms() * 90);
 
     {
       rtc::CritScope lock(&crit_);
@@ skipping 55 matching lines @@
       AddFrameComparison(frames_.front(), last_rendered_frame_, true,
                          render_time_ms);
       frames_.pop_front();
     }
 
     VideoFrame reference_frame = frames_.front();
     frames_.pop_front();
     assert(!reference_frame.IsZeroSize());
     if (send_timestamp == reference_frame.timestamp() - 1) {
       // TODO(ivica): Make this work for > 2 streams.
-      // Look at rtp_sender.c:RTPSender::BuildRTPHeader.
+      // Look at RTPSender::BuildRTPHeader.
       ++send_timestamp;
     }
     EXPECT_EQ(reference_frame.timestamp(), send_timestamp);
     assert(reference_frame.timestamp() == send_timestamp);
 
     AddFrameComparison(reference_frame, video_frame, false, render_time_ms);
 
     last_rendered_frame_ = video_frame;
   }
 
@@ skipping 743 matching lines @@
       disable_quality_check ? -1.1 : params_.analyzer.avg_ssim_threshold,
       params_.analyzer.test_durations_secs * params_.common.fps,
       graph_data_output_file, graph_title,
       kVideoSendSsrcs[params_.ss.selected_stream]);
 
   analyzer.SetReceiver(receiver_call_->Receiver());
   send_transport.SetReceiver(&analyzer);
   recv_transport.SetReceiver(sender_call_->Receiver());
 
   SetupCommon(&analyzer, &recv_transport);
-  video_send_config_.encoding_time_observer = &analyzer;
   video_receive_configs_[params_.ss.selected_stream].renderer = &analyzer;
   for (auto& config : video_receive_configs_)
     config.pre_decode_callback = &analyzer;
 
   if (params_.screenshare.enabled)
     SetupScreenshare();
 
   CreateVideoStreams();
   analyzer.input_ = video_send_stream_->Input();
   analyzer.send_stream_ = video_send_stream_;
@@ skipping 79 matching lines @@
   video_send_stream_->Stop();
   receive_stream->Stop();
 
   call->DestroyVideoReceiveStream(receive_stream);
   call->DestroyVideoSendStream(video_send_stream_);
 
   transport.StopSending();
 }
 
 }  // namespace webrtc