Estimate RTP clock frequency and plot capture-send delay.
Bug: webrtc:8450
Change-Id: Idea093854a644f3018a565168425583dc4783ce9
Reviewed-on: https://webrtc-review.googlesource.com/15480
Reviewed-by: Patrik Höglund <phoglund@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Commit-Queue: Björn Terelius <terelius@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20442}diff --git a/rtc_tools/event_log_visualizer/main.cc b/rtc_tools/event_log_visualizer/main.cc
index aef040e..e028a0f 100644
--- a/rtc_tools/event_log_visualizer/main.cc
+++ b/rtc_tools/event_log_visualizer/main.cc
@@ -88,6 +88,11 @@
true,
"Plot packet loss in percent for outgoing packets (as perceived by "
"the send-side bandwidth estimator).");
+DEFINE_bool(plot_pacer_delay,
+ false,
+ "Plot the time each sent packet has spent in the pacer (based on "
+ "the difference between the RTP timestamp and the send "
+ "timestamp).");
DEFINE_bool(plot_timestamps,
false,
"Plot the rtp timestamps of all rtp and rtcp packets over time.");
@@ -265,6 +270,9 @@
if (FLAG_plot_timestamps) {
analyzer.CreateTimestampGraph(collection->AppendNewPlot());
}
+ if (FLAG_plot_pacer_delay) {
+ analyzer.CreatePacerDelayGraph(collection->AppendNewPlot());
+ }
if (FLAG_plot_audio_encoder_bitrate_bps) {
analyzer.CreateAudioEncoderTargetBitrateGraph(collection->AppendNewPlot());
}