Retransmitted packets are now counted in receive time

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 "webrtc/video/video_quality_test.h"
@@ skipping 116 matching lines @@
                       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,
+                uint32_t rtx_ssrc_to_analyze,
                 uint32_t selected_width,
                 uint32_t selected_height)
       : transport_(transport),
         receiver_(nullptr),
         send_stream_(nullptr),
         receive_stream_(nullptr),
         captured_frame_forwarder_(this),
         test_label_(test_label),
         graph_data_output_file_(graph_data_output_file),
         graph_title_(graph_title),
         ssrc_to_analyze_(ssrc_to_analyze),
+        rtx_ssrc_to_analyze_(rtx_ssrc_to_analyze),
         selected_width_(selected_width),
         selected_height_(selected_height),
         pre_encode_proxy_(this),
         encode_timing_proxy_(this),
         frames_to_process_(duration_frames),
         frames_recorded_(0),
         frames_processed_(0),
         dropped_frames_(0),
         dropped_frames_before_first_encode_(0),
         dropped_frames_before_rendering_(0),
@@ skipping 65 matching lines @@
                                const PacketTime& packet_time) override {
     // Ignore timestamps of RTCP packets. They're not synchronized with
     // RTP packet timestamps and so they would confuse wrap_handler_.
     if (RtpHeaderParser::IsRtcp(packet, length)) {
       return receiver_->DeliverPacket(media_type, packet, length, packet_time);
     }
 
     RtpUtility::RtpHeaderParser parser(packet, length);
     RTPHeader header;
     parser.Parse(&header);
-    if (!IsFlexfec(header.payloadType) && header.ssrc == ssrc_to_analyze_) {
+    if (!IsFlexfec(header.payloadType) &&
+        (header.ssrc == ssrc_to_analyze_ ||
+         header.ssrc == rtx_ssrc_to_analyze_)) {
       // Ignore FlexFEC timestamps, to avoid collisions with media timestamps.
       // (FlexFEC and media are sent on different SSRCs, which have different
       // timestamps spaces.)
-      // Also ignore packets from wrong SSRC.
+      // Also ignore packets from wrong SSRC, but include retransmits.
       rtc::CritScope lock(&crit_);
       int64_t timestamp =
           wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
       recv_times_[timestamp] =
           Clock::GetRealTimeClock()->CurrentNtpInMilliseconds();
     }
 
     return receiver_->DeliverPacket(media_type, packet, length, packet_time);
   }
 
@@ skipping 38 matching lines @@
       rtc::CritScope lock(&crit_);
       if (rtp_timestamp_delta_ == 0 && header.ssrc == ssrc_to_analyze_) {
         RTC_CHECK(static_cast<bool>(first_sent_timestamp_));
         rtp_timestamp_delta_ = header.timestamp - *first_sent_timestamp_;
       }
 
       if (!IsFlexfec(header.payloadType) && header.ssrc == ssrc_to_analyze_) {
         // Ignore FlexFEC timestamps, to avoid collisions with media timestamps.
         // (FlexFEC and media are sent on different SSRCs, which have different
         // timestamps spaces.)
-        // Also ignore packets from wrong SSRC.
+        // Also ignore packets from wrong SSRC and retransmits.
         int64_t timestamp =
             wrap_handler_.Unwrap(header.timestamp - rtp_timestamp_delta_);
         send_times_[timestamp] = current_time;
         if (!transport_->DiscardedLastPacket() &&
             header.ssrc == ssrc_to_analyze_) {
           encoded_frame_sizes_[timestamp] +=
               length - (header.headerLength + header.paddingLength);
         }
       }
     }
@@ skipping 531 matching lines @@
     frames_.push_back(video_frame);
   }
 
   VideoSendStream* send_stream_;
   VideoReceiveStream* receive_stream_;
   CapturedFrameForwarder captured_frame_forwarder_;
   const std::string test_label_;
   FILE* const graph_data_output_file_;
   const std::string graph_title_;
   const uint32_t ssrc_to_analyze_;
+  const uint32_t rtx_ssrc_to_analyze_;
   const uint32_t selected_width_;
   const uint32_t selected_height_;
   PreEncodeProxy pre_encode_proxy_;
   OnEncodeTimingProxy encode_timing_proxy_;
   std::vector<Sample> samples_ GUARDED_BY(comparison_lock_);
   std::map<int64_t, int> samples_encode_time_ms_ GUARDED_BY(comparison_lock_);
   test::Statistics sender_time_ GUARDED_BY(comparison_lock_);
   test::Statistics receiver_time_ GUARDED_BY(comparison_lock_);
   test::Statistics psnr_ GUARDED_BY(comparison_lock_);
   test::Statistics ssim_ GUARDED_BY(comparison_lock_);
@@ skipping 476 matching lines @@
             "not implemented yet! Skipping PSNR and SSIM calculations!\n");
   }
 
   VideoAnalyzer analyzer(
       &send_transport, params_.analyzer.test_label,
       disable_quality_check ? -1.1 : params_.analyzer.avg_psnr_threshold,
       disable_quality_check ? -1.1 : params_.analyzer.avg_ssim_threshold,
       params_.analyzer.test_durations_secs * params_.video.fps,
       graph_data_output_file, graph_title,
       kVideoSendSsrcs[params_.ss.selected_stream],
+      kSendRtxSsrcs[params_.ss.selected_stream],
       static_cast<uint32_t>(selected_stream.width),
       static_cast<uint32_t>(selected_stream.height));
 
   analyzer.SetReceiver(receiver_call_->Receiver());
   send_transport.SetReceiver(&analyzer);
   recv_transport.SetReceiver(sender_call_->Receiver());
 
   SetupVideo(&analyzer, &recv_transport);
   video_receive_configs_[params_.ss.selected_stream].renderer = &analyzer;
   video_send_config_.pre_encode_callback = analyzer.pre_encode_proxy();
@@ skipping 237 matching lines @@
     std::ostringstream str;
     str << receive_logs_++;
     std::string path =
         params_.video.encoded_frame_base_path + "." + str.str() + ".recv.ivf";
     stream->EnableEncodedFrameRecording(rtc::CreatePlatformFile(path),
                                         10000000);
   }
 }
 
 }  // namespace webrtc