Fix indexing error in event log analyzer.

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 327 matching lines @@
         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: {
         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);
+        for (const auto& config : configs) {
+          StreamId stream(config.local_ssrc, kOutgoingPacket);
           extension_maps[stream] =
-              RtpHeaderExtensionMap(configs[i].rtp_extensions);
+              RtpHeaderExtensionMap(config.rtp_extensions);
           video_ssrcs_.insert(stream);
-          StreamId rtx_stream(configs[i].rtx_ssrc, kOutgoingPacket);
+          StreamId rtx_stream(config.rtx_ssrc, kOutgoingPacket);
           extension_maps[rtx_stream] =
-              RtpHeaderExtensionMap(configs[i].rtp_extensions);
+              RtpHeaderExtensionMap(config.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);
         StreamId stream(config.remote_ssrc, kIncomingPacket);
         extension_maps[stream] = RtpHeaderExtensionMap(config.rtp_extensions);
         audio_ssrcs_.insert(stream);
@@ skipping 1025 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