Ensure the RtcEventLog parser is backwards compatible

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 313 matching lines @@
         event_type != ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT &&
         event_type != ParsedRtcEventLog::LOG_START &&
         event_type != ParsedRtcEventLog::LOG_END) {
       uint64_t timestamp = parsed_log_.GetTimestamp(i);
       first_timestamp = std::min(first_timestamp, timestamp);
       last_timestamp = std::max(last_timestamp, timestamp);
     }
 
     switch (parsed_log_.GetEventType(i)) {
       case ParsedRtcEventLog::VIDEO_RECEIVER_CONFIG_EVENT: {
-        rtclog::StreamConfig config;
-        parsed_log_.GetVideoReceiveConfig(i, &config);
+        rtclog::StreamConfig config = parsed_log_.GetVideoReceiveConfig(i);
         StreamId stream(config.remote_ssrc, kIncomingPacket);
         extension_maps[stream] = RtpHeaderExtensionMap(config.rtp_extensions);
         video_ssrcs_.insert(stream);
         StreamId rtx_stream(config.rtx_ssrc, kIncomingPacket);
         extension_maps[rtx_stream] =
             RtpHeaderExtensionMap(config.rtp_extensions);
         video_ssrcs_.insert(rtx_stream);
         rtx_ssrcs_.insert(rtx_stream);
         break;
       }
       case ParsedRtcEventLog::VIDEO_SENDER_CONFIG_EVENT: {
-        rtclog::StreamConfig config;
-        parsed_log_.GetVideoSendConfig(i, &config);
-        StreamId stream(config.local_ssrc, kOutgoingPacket);
-        extension_maps[stream] = RtpHeaderExtensionMap(config.rtp_extensions);
-        video_ssrcs_.insert(stream);
-        StreamId rtx_stream(config.rtx_ssrc, kOutgoingPacket);
-        extension_maps[rtx_stream] =
-            RtpHeaderExtensionMap(config.rtp_extensions);
-        video_ssrcs_.insert(rtx_stream);
-        rtx_ssrcs_.insert(rtx_stream);
+        std::vector<rtclog::StreamConfig> configs =
+            parsed_log_.GetVideoSendConfig(i);
+        for (size_t j = 0; j < configs.size(); j++) {
+          StreamId stream(configs[i].local_ssrc, kOutgoingPacket);
+          extension_maps[stream] =
+              RtpHeaderExtensionMap(configs[i].rtp_extensions);
+          video_ssrcs_.insert(stream);
+          StreamId rtx_stream(configs[i].rtx_ssrc, kOutgoingPacket);
+          extension_maps[rtx_stream] =
+              RtpHeaderExtensionMap(configs[i].rtp_extensions);
+          video_ssrcs_.insert(rtx_stream);
+          rtx_ssrcs_.insert(rtx_stream);
+        }
         break;
       }
       case ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: {
-        rtclog::StreamConfig config;
-        parsed_log_.GetAudioReceiveConfig(i, &config);
+        rtclog::StreamConfig config = parsed_log_.GetAudioReceiveConfig(i);
         StreamId stream(config.remote_ssrc, kIncomingPacket);
         extension_maps[stream] = RtpHeaderExtensionMap(config.rtp_extensions);
         audio_ssrcs_.insert(stream);
         break;
       }
       case ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT: {
-        rtclog::StreamConfig config;
-        parsed_log_.GetAudioSendConfig(i, &config);
+        rtclog::StreamConfig config = parsed_log_.GetAudioSendConfig(i);
         StreamId stream(config.local_ssrc, kOutgoingPacket);
         extension_maps[stream] = RtpHeaderExtensionMap(config.rtp_extensions);
         audio_ssrcs_.insert(stream);
         break;
       }
       case ParsedRtcEventLog::RTP_EVENT: {
         parsed_log_.GetRtpHeader(i, &direction, header, &header_length,
                                  &total_length);
         // Parse header to get SSRC.
         RtpUtility::RtpHeaderParser rtp_parser(header, header_length);
@@ skipping 1011 matching lines @@
       },
       audio_network_adaptation_events_, begin_time_, &time_series);
   plot->AppendTimeSeries(std::move(time_series));
   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