commit e0928d8002cb7d25f314c54c8d6ce20ab859aa4d
author ivoc <ivoc@webrtc.org> Mon Oct 10 05:12:51 2016 -0700
committer Commit bot <commit-bot@chromium.org> Mon Oct 10 12:12:57 2016 +0000
tree af598d3eaa037f9940dbaa562293641cf924a069
parent fc9414ab513941028309d15a2baf711ef38f93a7

Added logging for audio send/receive stream configs.

BUG=webrtc:4741,webrtc:6399

Review-Url: https://codereview.webrtc.org/2353543003
Cr-Commit-Position: refs/heads/master@{#14585}

webrtc/tools/event_log_visualizer/analyzer.cc

diff --git a/webrtc/tools/event_log_visualizer/analyzer.cc b/webrtc/tools/event_log_visualizer/analyzer.cc
index 4ec1a29..cc6425f 100644
--- a/webrtc/tools/event_log_visualizer/analyzer.cc
+++ b/webrtc/tools/event_log_visualizer/analyzer.cc
@@ -353,12 +353,20 @@
       }
       case ParsedRtcEventLog::AUDIO_RECEIVER_CONFIG_EVENT: {
         AudioReceiveStream::Config config;
-        // TODO(terelius): Parse the audio configs once we have them.
+        parsed_log_.GetAudioReceiveConfig(i, &config);
+        StreamId stream(config.rtp.remote_ssrc, kIncomingPacket);
+        RegisterHeaderExtensions(config.rtp.extensions,
+                                 &extension_maps[stream]);
+        audio_ssrcs_.insert(stream);
         break;
       }
       case ParsedRtcEventLog::AUDIO_SENDER_CONFIG_EVENT: {
         AudioSendStream::Config config(nullptr);
-        // TODO(terelius): Parse the audio configs once we have them.
+        parsed_log_.GetAudioSendConfig(i, &config);
+        StreamId stream(config.rtp.ssrc, kOutgoingPacket);
+        RegisterHeaderExtensions(config.rtp.extensions,
+                                 &extension_maps[stream]);
+        audio_ssrcs_.insert(stream);
         break;
       }
       case ParsedRtcEventLog::RTP_EVENT: {