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![]() The RTC event log analyzer would previously only plot network latency for incoming video streams. (The latency is computed from the capture time in the RTP header, and the packet receive time.) This CL adds support for audio packets, which requires estimating the RTP clock frequency for the incoming packets. Bug: None Change-Id: Idf1ff9febfdd4097976b22a61f1c5679deb6068c Reviewed-on: https://webrtc-review.googlesource.com/c/108784 Reviewed-by: Sebastian Jansson <srte@webrtc.org> Commit-Queue: Björn Terelius <terelius@webrtc.org> Cr-Commit-Position: refs/heads/master@{#25580} |
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.. | ||
analyzer.cc | ||
analyzer.h | ||
chart.proto | ||
main.cc | ||
OWNERS | ||
plot_base.cc | ||
plot_base.h | ||
plot_protobuf.cc | ||
plot_protobuf.h | ||
plot_python.cc | ||
plot_python.h | ||
triage_notifications.h |