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author | Erwin Jansen <jansene@google.com> | 2021-06-30 07:29:26 +0000 |
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committer | Gerrit Code Review <noreply-gerritcodereview@google.com> | 2021-06-30 07:29:26 +0000 |
commit | 059cdc5996938f5f6b5343b6c969c12098275587 (patch) | |
tree | 6eacaffe4bebf8e00c290c1e1839e084b0c52e88 /logging/g3doc/rtc_event_log.md | |
parent | 97e54a7e73c7b24e464ef06ef3c3b3716f21bb15 (diff) | |
parent | 16be34ae72cdb525c88c2b31b21b976f35fe36d8 (diff) | |
download | webrtc-059cdc5996938f5f6b5343b6c969c12098275587.tar.gz |
Merge "Merge upstream-master and enable ARM64" into emu-master-devemu-31-stable-releaseemu-31-release
Diffstat (limited to 'logging/g3doc/rtc_event_log.md')
-rw-r--r-- | logging/g3doc/rtc_event_log.md | 85 |
1 files changed, 85 insertions, 0 deletions
diff --git a/logging/g3doc/rtc_event_log.md b/logging/g3doc/rtc_event_log.md new file mode 100644 index 0000000000..c7996e0b42 --- /dev/null +++ b/logging/g3doc/rtc_event_log.md @@ -0,0 +1,85 @@ +# RTC event log + +<?% config.freshness.owner = 'terelius' %?> +<?% config.freshness.reviewed = '2021-06-02' %?> + +## Overview + +RTC event logs can be enabled to capture in-depth inpformation about sent and +received packets and the internal state of some WebRTC components. The logs are +useful to understand network behavior and to debug issues around connectivity, +bandwidth estimation and audio jitter buffers. + +The contents include: + +* Sent and received RTP headers +* Full RTCP feedback +* ICE candidates, pings and responses +* Bandwidth estimator events, including loss-based estimate, delay-based + estimate, probe results and ALR state +* Audio network adaptation settings +* Audio playout events + +## Binary wire format + +No guarantees are made on the wire format, and the format may change without +prior notice. To maintain compatibility with past and future formats, analysis +tools should be built on top of the provided +[rtc_event_log_parser.h](https://source.chromium.org/chromium/chromium/src/+/master:third_party/webrtc/logging/rtc_event_log/rtc_event_log_parser.h) + +In particular, an analysis tool should *not* read the log as a protobuf. + +## Visualization + +Since the logs contain a substantial amount of data, it is usually convenient to +get an overview by visualizing them as a set of plots. Use the command: + +``` +out/Default/event_log_visualizer /path/to/log_file | python +``` + +This visualization requires matplotlib to be installed. The tool is capable of +producing a substantial number of plots, of which only a handful are generated +by default. You can select which plots are generated though the `--plot=` +command line argument. For example, the command + +``` +out/Default/event_log_visualizer \ + --plot=incoming_packet_sizes,incoming_stream_bitrate \ + /path/to/log_file | python +``` + +plots the sizes of incoming packets and the bitrate per incoming stream. + +You can get a full list of options for the `--plot` argument through + +``` +out/Default/event_log_visualizer --list_plots /path/to/log_file +``` + +You can also synchronize the x-axis between all plots (so zooming or +panning in one plot affects all of them), by adding the command line +argument `--shared_xaxis`. + + +## Viewing the raw log contents as text + +If you know which format version the log file uses, you can view the raw +contents as text. For version 1, you can use the command + +``` +out/Default/protoc --decode webrtc.rtclog.EventStream \ + ./logging/rtc_event_log/rtc_event_log.proto < /path/to/log_file +``` + +Similarly, you can use + +``` +out/Default/protoc --decode webrtc.rtclog2.EventStream \ + ./logging/rtc_event_log/rtc_event_log2.proto < /path/to/log_file +``` + +for logs that use version 2. However, note that not all of the contents will be +human readable. Some fields are based on the raw RTP format or may be encoded as +deltas relative to previous fields. Such fields will be printed as a list of +bytes. |