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

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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}
This commit is contained in:
Minyue Li
2018-06-21 21:17:19 +02:00
committed by Commit Bot
parent 91280e4d04
commit 01d2a67a70
3 changed files with 10 additions and 12 deletions
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8
rtc_tools/event_log_visualizer/main.cc
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@ -335,12 +335,12 @@ int main(int argc, char* argv[]) {
"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) {