Adding jitter buffer plots for all SSRCs in event log visualizer.

Bug: webrtc:9147
Change-Id: I64291666d329c026f35ecf1c4245b192794441fe
Reviewed-on: https://webrtc-review.googlesource.com/84745
Commit-Queue: Minyue Li <minyue@webrtc.org>
Reviewed-by: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#23726}
diff --git a/rtc_tools/event_log_visualizer/main.cc b/rtc_tools/event_log_visualizer/main.cc
index 7d9d45e..8a303b2 100644
--- a/rtc_tools/event_log_visualizer/main.cc
+++ b/rtc_tools/event_log_visualizer/main.cc
@@ -335,12 +335,12 @@
           "audio_processing/conversational_speech/EN_script2_F_sp2_B1", "wav");
     }
     auto neteq_stats = analyzer.SimulateNetEq(wav_path, 48000);
-
-    if (!neteq_stats.empty()) {
-      analyzer.CreateAudioJitterBufferGraph(neteq_stats,
+    for (webrtc::EventLogAnalyzer::NetEqStatsGetterMap::const_iterator it =
+             neteq_stats.cbegin();
+         it != neteq_stats.cend(); ++it) {
+      analyzer.CreateAudioJitterBufferGraph(it->first, it->second.get(),
                                             collection->AppendNewPlot());
     }
-
     analyzer.CreateNetEqStatsGraph(
         neteq_stats,
         [](const webrtc::NetEqNetworkStatistics& stats) {