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Estimate RTP clock frequency and plot capture-send delay.

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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}
This commit is contained in:
Bjorn Terelius
2017-10-25 17:42:41 +02:00
committed by Commit Bot
parent 30915a742d
commit 0295a967c0
4 changed files with 104 additions and 1 deletions
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8
rtc_tools/event_log_visualizer/main.cc
View File

@ -88,6 +88,11 @@ DEFINE_bool(plot_fraction_loss_feedback,
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 @@ int main(int argc, char* argv[]) {
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());
}