Renamed fields in rtp_rtcp_defines.h/RTCPReportBlock

webrtc/modules/rtp_rtcp/source/rtcp_receiver.cc

 /*
  *  Copyright (c) 2012 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 440 matching lines @@
 
   // |report_block.source_ssrc()| is the SSRC identifier of the source to
   // which the information in this reception report block pertains.
 
   // Filter out all report blocks that are not for us.
   if (registered_ssrcs_.count(report_block.source_ssrc()) == 0)
     return;
 
   ReportBlockWithRtt* report_block_info =
       &received_report_blocks_[report_block.source_ssrc()][remote_ssrc];
-  report_block_info->report_block.remoteSSRC = remote_ssrc;
-  report_block_info->report_block.sourceSSRC = report_block.source_ssrc();
-  report_block_info->report_block.fractionLost = report_block.fraction_lost();
-  report_block_info->report_block.cumulativeLost =
-      report_block.cumulative_lost();
+  report_block_info->report_block.sender_ssrc = remote_ssrc;
+  report_block_info->report_block.source_ssrc = report_block.source_ssrc();
+  report_block_info->report_block.fraction_lost = report_block.fraction_lost();
+  report_block_info->report_block.packets_lost = report_block.cumulative_lost();
   if (report_block.extended_high_seq_num() >
-      report_block_info->report_block.extendedHighSeqNum) {
+      report_block_info->report_block.extended_highest_sequence_number) {
     // We have successfully delivered new RTP packets to the remote side after
     // the last RR was sent from the remote side.
     last_increased_sequence_number_ms_ = clock_->TimeInMilliseconds();
   }
-  report_block_info->report_block.extendedHighSeqNum =
+  report_block_info->report_block.extended_highest_sequence_number =
       report_block.extended_high_seq_num();
   report_block_info->report_block.jitter = report_block.jitter();
-  report_block_info->report_block.delaySinceLastSR =
+  report_block_info->report_block.delay_since_last_sender_report =
       report_block.delay_since_last_sr();
-  report_block_info->report_block.lastSR = report_block.last_sr();
+  report_block_info->report_block.last_sender_report_timestamp =
+      report_block.last_sr();
 
   int64_t rtt_ms = 0;
   uint32_t send_time_ntp = report_block.last_sr();
   // RFC3550, section 6.4.1, LSR field discription states:
   // If no SR has been received yet, the field is set to zero.
   // Receiver rtp_rtcp module is not expected to calculate rtt using
   // Sender Reports even if it accidentally can.
   if (!receiver_only_ && send_time_ntp != 0) {
     uint32_t delay_ntp = report_block.delay_since_last_sr();
     // Local NTP time.
@@ skipping 497 matching lines @@
       packet_information.target_bitrate_allocation) {
     bitrate_allocation_observer_->OnBitrateAllocationUpdated(
         *packet_information.target_bitrate_allocation);
   }
 
   if (!receiver_only_) {
     rtc::CritScope cs(&feedbacks_lock_);
     if (stats_callback_) {
       for (const auto& report_block : packet_information.report_blocks) {
         RtcpStatistics stats;
-        stats.packets_lost = report_block.cumulativeLost;
+        stats.packets_lost = report_block.packets_lost;
         stats.extended_highest_sequence_number =
-            report_block.extendedHighSeqNum;
-        stats.fraction_lost = report_block.fractionLost;
+            report_block.extended_highest_sequence_number;
+        stats.fraction_lost = report_block.fraction_lost;
         stats.jitter = report_block.jitter;
 
-        stats_callback_->StatisticsUpdated(stats, report_block.sourceSSRC);
+        stats_callback_->StatisticsUpdated(stats, report_block.source_ssrc);
       }
     }
   }
 }
 
 int32_t RTCPReceiver::CNAME(uint32_t remoteSSRC,
                             char cName[RTCP_CNAME_SIZE]) const {
   RTC_DCHECK(cName);
 
   rtc::CritScope lock(&rtcp_receiver_lock_);
@@ skipping 22 matching lines @@
       } else {
         candidates.push_back(it->second.tmmbr_item);
         ++it;
       }
     }
   }
   return candidates;
 }
 
 }  // namespace webrtc