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@vpalmisano/webrtcperf-js

v1.1.17

Published

WebRTCPerf javascript browser library

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Links

Git

Maintainers

vpalmisanovpalmisano

Keywords

Readme

WebRTC Perf javascript browser library

GitHub page | Documentation

A browser library used by the webrtcperf tool to capture the RTC logs and run page inspection/automation. It can also be used stand-alone importing the javascript package into the page before loading the other javascript sources. It contains some utilities to debug the page RTC connections, the getUserMedia and getDisplayMedia, evaluate the end-to-end delay, etc.

Import the library

<head>
  <script type="text/javascript" src="https://unpkg.com/@vpalmisano/webrtcperf-js/dist/webrtcperf.js"></script>
</head>

Userscript usage

The library can be used directly into a regular Google Chome browser session.

  1. Install Tampermonkey
  2. Install the User Script

Examples

Access the RTC PeerConnections objects

webrtcperf.PeerConnections

Get the current RTC stats

await webrtcperf.collectPeerConnectionStats(true)

Use a video as getUserMedia default source

Visit https://webrtc.github.io/samples/src/content/getusermedia/resolution/ with the tampermonkey script activated:

webrtcperf.config.MEDIA_URL = 'https://commondatastorage.googleapis.com/gtv-videos-bucket/sample/BigBuckBunny.mp4'
// Select one of the resolutions in the sample page.

To activate the option every time the page is loaded, you can set the following localStorage value:

localStorage.setItem('webrtcperf.config', JSON.stringify({
  MEDIA_URL: 'https://commondatastorage.googleapis.com/gtv-videos-bucket/sample/BigBuckBunny.mp4',
}))

Use a fake screenshare as getDisplayMedia source

await webrtcperf.startFakeScreenshare()
webrtcperf.overrides.getDisplayMedia = constraints => Object.assign(constraints, { preferCurrentTab: true })
// Start a screen sharing with getDisplayMedia.

Measure the audio and video end-to-end delay

Visit https://webrtc.github.io/samples/src/content/peerconnection/pc1/ with the tampermonkey script activated:

webrtcperf.config.MEDIA_URL = 'https://commondatastorage.googleapis.com/gtv-videos-bucket/sample/BigBuckBunny.mp4'
webrtcperf.params.timestampWatermarkVideo = true
webrtcperf.params.timestampWatermarkAudio = true
// Output:
// [webrtcperf-0] [e2e-video-stats] rx text=0-1743947998586 delay=44ms confidence=74 elapsed=73ms
// [webrtcperf-0] [e2e-audio-stats] rx delay: 71.33333333333326ms rxFrames: 62 rxFramesDuration: 1322.6666666666667ms

// Access the collected stats:
webrtcperf.videoEndToEndDelayStats
webrtcperf.audioEndToEndDelayStats

The library adds a video timestamp overlay at sender side and it recognizes the video at receiver side using Tesseract.js. The audio track at sender side is mixed with signals generated with the ggwave library; at receiver side, the same library is used do decode the received signals with the encoded sender timestamp. Sender and receiver machine clocks must be syncronized.