Replace rtcp parser in rtc event log handlers.

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 11 matching lines @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/rate_statistics.h"
 #include "webrtc/call.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/bitrate_controller/include/bitrate_controller.h"
 #include "webrtc/modules/congestion_controller/include/congestion_controller.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp_defines.h"
 #include "webrtc/modules/rtp_rtcp/source/rtp_utility.h"
-#include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
+#include "webrtc/modules/rtp_rtcp/source/rtcp_packet/common_header.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_packet/transport_feedback.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
 namespace webrtc {
 namespace plotting {
 
 namespace {
 
 std::string SsrcToString(uint32_t ssrc) {
@@ skipping 342 matching lines @@
         rtp_packets_[stream].push_back(
             LoggedRtpPacket(timestamp, parsed_header, total_length));
         break;
       }
       case ParsedRtcEventLog::RTCP_EVENT: {
         uint8_t packet[IP_PACKET_SIZE];
         MediaType media_type;
         parsed_log_.GetRtcpPacket(i, &direction, &media_type, packet,
                                   &total_length);
 
-        RtpUtility::RtpHeaderParser rtp_parser(packet, total_length);
-        RTPHeader parsed_header;
-        RTC_CHECK(rtp_parser.ParseRtcp(&parsed_header));
-        uint32_t ssrc = parsed_header.ssrc;
-
-        RTCPUtility::RTCPParserV2 rtcp_parser(packet, total_length, true);
-        RTC_CHECK(rtcp_parser.IsValid());
-
-        RTCPUtility::RTCPPacketTypes packet_type = rtcp_parser.Begin();
-        while (packet_type != RTCPUtility::RTCPPacketTypes::kInvalid) {
-          switch (packet_type) {
-            case RTCPUtility::RTCPPacketTypes::kTransportFeedback: {
-              // Currently feedback is logged twice, both for audio and video.
-              // Only act on one of them.
-              if (media_type == MediaType::VIDEO) {
-                std::unique_ptr<rtcp::RtcpPacket> rtcp_packet(
-                    rtcp_parser.ReleaseRtcpPacket());
+        // Currently feedback is logged twice, both for audio and video.
+        // Only act on one of them.
+        if (media_type == MediaType::VIDEO) {
+          rtcp::CommonHeader header;
+          const uint8_t* packet_end = packet + total_length;
+          for (const uint8_t* block = packet; block < packet_end;
+               block = header.NextPacket()) {
+            RTC_CHECK(header.Parse(block, packet_end - block));
+            if (header.type() == rtcp::TransportFeedback::kPacketType &&
+                header.fmt() == rtcp::TransportFeedback::kFeedbackMessageType) {
+              std::unique_ptr<rtcp::TransportFeedback> rtcp_packet(
+                  new rtcp::TransportFeedback());
+              if (rtcp_packet->Parse(header)) {
+                uint32_t ssrc = rtcp_packet->sender_ssrc();
                 StreamId stream(ssrc, direction);
                 uint64_t timestamp = parsed_log_.GetTimestamp(i);
                 rtcp_packets_[stream].push_back(LoggedRtcpPacket(
                     timestamp, kRtcpTransportFeedback, std::move(rtcp_packet)));
               }
-              break;
             }
-            default:
-              break;
           }
-          rtcp_parser.Iterate();
-          packet_type = rtcp_parser.PacketType();
         }
         break;
       }
       case ParsedRtcEventLog::LOG_START: {
         break;
       }
       case ParsedRtcEventLog::LOG_END: {
         break;
       }
       case ParsedRtcEventLog::BWE_PACKET_LOSS_EVENT: {
@@ skipping 733 matching lines @@
     point.y -= estimated_base_delay_ms;
   // Add the data set to the plot.
   plot->series_list_.push_back(std::move(time_series));
 
   plot->SetXAxis(0, call_duration_s_, "Time (s)", kLeftMargin, kRightMargin);
   plot->SetSuggestedYAxis(0, 10, "Delay (ms)", kBottomMargin, kTopMargin);
   plot->SetTitle("Network Delay Change.");
 }
 }  // namespace plotting
 }  // namespace webrtc