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/analyzer.h b/rtc_tools/event_log_visualizer/analyzer.h
index 5504282..15f1fa7 100644
--- a/rtc_tools/event_log_visualizer/analyzer.h
+++ b/rtc_tools/event_log_visualizer/analyzer.h
@@ -94,6 +94,7 @@
void CreateReceiveSideBweSimulationGraph(Plot* plot);
void CreateNetworkDelayFeedbackGraph(Plot* plot);
+ void CreatePacerDelayGraph(Plot* plot);
void CreateTimestampGraph(Plot* plot);
void CreateAudioEncoderTargetBitrateGraph(Plot* plot);
@@ -144,7 +145,10 @@
bool IsAudioSsrc(StreamId stream_id) const;
- std::string GetStreamName(StreamId) const;
+ std::string GetStreamName(StreamId stream_id) const;
+
+ rtc::Optional<uint32_t> EstimateRtpClockFrequency(
+ const std::vector<LoggedRtpPacket>& packets) const;
const ParsedRtcEventLog& parsed_log_;