Rename webrtc::PacketInfo to webrtc::PacketFeedback

webrtc/tools/event_log_visualizer/analyzer.cc

 /*
  *  Copyright (c) 2016 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.
  */
 
@@ skipping 1037 matching lines @@
   int64_t last_update_us = 0;
   while (time_us != std::numeric_limits<int64_t>::max()) {
     clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
     if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
       const LoggedRtcpPacket& rtcp = *rtcp_iterator->second;
       if (rtcp.type == kRtcpTransportFeedback) {
         TransportFeedbackObserver* observer = cc.GetTransportFeedbackObserver();
         observer->OnTransportFeedback(*static_cast<rtcp::TransportFeedback*>(
             rtcp.packet.get()));
-        std::vector<PacketInfo> feedback =
+        std::vector<PacketFeedback> feedback =
             observer->GetTransportFeedbackVector();
         rtc::Optional<uint32_t> bitrate_bps;
         if (!feedback.empty()) {
-          for (const PacketInfo& packet : feedback)
+          for (const PacketFeedback& packet : feedback)
             acked_bitrate.Update(packet.payload_size, packet.arrival_time_ms);
           bitrate_bps = acked_bitrate.Rate(feedback.back().arrival_time_ms);
         }
         uint32_t y = 0;
         if (bitrate_bps)
           y = *bitrate_bps / 1000;
         float x = static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
                   1000000;
         acked_time_series.points.emplace_back(x, y);
       }
@@ skipping 110 matching lines @@
 
   int64_t time_us = std::min(NextRtpTime(), NextRtcpTime());
   while (time_us != std::numeric_limits<int64_t>::max()) {
     clock.AdvanceTimeMicroseconds(time_us - clock.TimeInMicroseconds());
     if (clock.TimeInMicroseconds() >= NextRtcpTime()) {
       RTC_DCHECK_EQ(clock.TimeInMicroseconds(), NextRtcpTime());
       const LoggedRtcpPacket& rtcp = *rtcp_iterator->second;
       if (rtcp.type == kRtcpTransportFeedback) {
         feedback_adapter.OnTransportFeedback(
             *static_cast<rtcp::TransportFeedback*>(rtcp.packet.get()));
-        std::vector<PacketInfo> feedback =
-          feedback_adapter.GetTransportFeedbackVector();
-        for (const PacketInfo& packet : feedback) {
+        std::vector<PacketFeedback> feedback =
+            feedback_adapter.GetTransportFeedbackVector();
+        for (const PacketFeedback& packet : feedback) {
           int64_t y = packet.arrival_time_ms - packet.send_time_ms;
           float x =
               static_cast<float>(clock.TimeInMicroseconds() - begin_time_) /
               1000000;
           estimated_base_delay_ms = std::min(y, estimated_base_delay_ms);
           time_series.points.emplace_back(x, y);
         }
       }
       ++rtcp_iterator;
     }
@@ skipping 175 matching lines @@
         return rtc::Optional<float>();
       });
   plot->series_list_.back().label = "Audio encoder number of channels";
   plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
   plot->SetSuggestedYAxis(0, 1, "Number of channels (1 (mono)/2 (stereo))",
                           kBottomMargin, kTopMargin);
   plot->SetTitle("Reported audio encoder number of channels");
 }
 }  // namespace plotting
 }  // namespace webrtc