a617867a45
This reverts commit 0f2ce5cc1c779f9bf33f51f29bfffbcbe105d1b1. Reason for revert: Downstream infrastructure should be ready now Original change's description: > Revert "Migrate WebRTC documentation to new renderer" > > This reverts commit 3eceaf46695518f25bef43f155f82ed174827197. > > Reason for revert: > > Original change's description: > > Migrate WebRTC documentation to new renderer > > > > Bug: b/258408932 > > Change-Id: Ib96f39fe0c3912f9746bcc09d079097a145d6115 > > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/290987 > > Reviewed-by: Harald Alvestrand <hta@webrtc.org> > > Commit-Queue: Artem Titov <titovartem@webrtc.org> > > Cr-Commit-Position: refs/heads/main@{#39205} > > Bug: b/258408932 > Change-Id: I16cb4088bee3fc15c2bb88bd692c592b3a7db9fe > No-Presubmit: true > No-Tree-Checks: true > No-Try: true > Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291560 > Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> > Owners-Override: Artem Titov <titovartem@webrtc.org> > Commit-Queue: Artem Titov <titovartem@webrtc.org> > Cr-Commit-Position: refs/heads/main@{#39209} Bug: b/258408932 Change-Id: Ia172e4a6ad1cc7953b48eed08776e9d1e44eb074 Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/291660 Owners-Override: Artem Titov <titovartem@webrtc.org> Reviewed-by: Harald Alvestrand <hta@webrtc.org> Bot-Commit: rubber-stamper@appspot.gserviceaccount.com <rubber-stamper@appspot.gserviceaccount.com> Commit-Queue: Artem Titov <titovartem@webrtc.org> Cr-Commit-Position: refs/heads/main@{#39231}
86 lines
2.8 KiB
Markdown
86 lines
2.8 KiB
Markdown
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<!--* freshness: {owner: 'terelius' reviewed: '2021-06-02'} *-->
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# RTC event log
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## Overview
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RTC event logs can be enabled to capture in-depth inpformation about sent and
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received packets and the internal state of some WebRTC components. The logs are
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useful to understand network behavior and to debug issues around connectivity,
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bandwidth estimation and audio jitter buffers.
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The contents include:
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* Sent and received RTP headers
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* Full RTCP feedback
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* ICE candidates, pings and responses
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* Bandwidth estimator events, including loss-based estimate, delay-based
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estimate, probe results and ALR state
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* Audio network adaptation settings
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* Audio playout events
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## Binary wire format
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No guarantees are made on the wire format, and the format may change without
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prior notice. To maintain compatibility with past and future formats, analysis
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tools should be built on top of the provided
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[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)
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In particular, an analysis tool should *not* read the log as a protobuf.
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## Visualization
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Since the logs contain a substantial amount of data, it is usually convenient to
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get an overview by visualizing them as a set of plots. Use the command:
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```
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out/Default/event_log_visualizer /path/to/log_file | python
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```
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This visualization requires matplotlib to be installed. The tool is capable of
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producing a substantial number of plots, of which only a handful are generated
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by default. You can select which plots are generated though the `--plot=`
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command line argument. For example, the command
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```
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out/Default/event_log_visualizer \
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--plot=incoming_packet_sizes,incoming_stream_bitrate \
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/path/to/log_file | python
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```
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plots the sizes of incoming packets and the bitrate per incoming stream.
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You can get a full list of options for the `--plot` argument through
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```
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out/Default/event_log_visualizer --list_plots /path/to/log_file
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```
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You can also synchronize the x-axis between all plots (so zooming or
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panning in one plot affects all of them), by adding the command line
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argument `--shared_xaxis`.
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## Viewing the raw log contents as text
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If you know which format version the log file uses, you can view the raw
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contents as text. For version 1, you can use the command
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```
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out/Default/protoc --decode webrtc.rtclog.EventStream \
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./logging/rtc_event_log/rtc_event_log.proto < /path/to/log_file
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```
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Similarly, you can use
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```
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out/Default/protoc --decode webrtc.rtclog2.EventStream \
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./logging/rtc_event_log/rtc_event_log2.proto < /path/to/log_file
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```
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for logs that use version 2. However, note that not all of the contents will be
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human readable. Some fields are based on the raw RTP format or may be encoded as
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deltas relative to previous fields. Such fields will be printed as a list of
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bytes.
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