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author | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:47:29 +0000 |
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committer | Daniel Baumann <daniel.baumann@progress-linux.org> | 2024-04-19 01:47:29 +0000 |
commit | 0ebf5bdf043a27fd3dfb7f92e0cb63d88954c44d (patch) | |
tree | a31f07c9bcca9d56ce61e9a1ffd30ef350d513aa /third_party/libwebrtc/logging/g3doc | |
parent | Initial commit. (diff) | |
download | firefox-esr-0ebf5bdf043a27fd3dfb7f92e0cb63d88954c44d.tar.xz firefox-esr-0ebf5bdf043a27fd3dfb7f92e0cb63d88954c44d.zip |
Adding upstream version 115.8.0esr.upstream/115.8.0esr
Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
Diffstat (limited to 'third_party/libwebrtc/logging/g3doc')
-rw-r--r-- | third_party/libwebrtc/logging/g3doc/rtc_event_log.md | 85 |
1 files changed, 85 insertions, 0 deletions
diff --git a/third_party/libwebrtc/logging/g3doc/rtc_event_log.md b/third_party/libwebrtc/logging/g3doc/rtc_event_log.md new file mode 100644 index 0000000000..7ffeb68840 --- /dev/null +++ b/third_party/libwebrtc/logging/g3doc/rtc_event_log.md @@ -0,0 +1,85 @@ +<!-- go/cmark --> +<!--* freshness: {owner: 'terelius' reviewed: '2021-06-02'} *--> + +# RTC event log + +## 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/+/main: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. |