libav.js
webrtc-for-the-curious
libav.js | webrtc-for-the-curious | |
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2 | 13 | |
271 | 1,884 | |
- | 0.6% | |
9.3 | 5.2 | |
30 days ago | 7 months ago | |
JavaScript | Python | |
- | Creative Commons Zero v1.0 Universal |
Stars - the number of stars that a project has on GitHub. Growth - month over month growth in stars.
Activity is a relative number indicating how actively a project is being developed. Recent commits have higher weight than older ones.
For example, an activity of 9.0 indicates that a project is amongst the top 10% of the most actively developed projects that we are tracking.
libav.js
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Replacing WebRTC: real-time latency with WebTransport and WebCodecs
I'm currently working with WebCodecs to get (the long awaited) frame-by-frame seeking and reverse playback working in the browser. And it even seems to work, albeit the VideoDecoder queuing logic seems to give some grief for this. Any tips on figuring out how many chunks have to be queued for a specific VideoFrame to pop out?
An aside: to work with video/container files, be sure to check the libav.js project that can be used to demux streams (WebCodecs don't do this) and even used as a polyfill decoder for browsers without WebCodec support!
https://github.com/Yahweasel/libav.js/
- Libav.js – FFmpeg libraries compiled for JavaScript
webrtc-for-the-curious
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Dive into Web RTC or write SFU on your own
Here I will briefly go over the basics of how Web RTC works; for those who are interested in going a little deeper, I’ll leave the link here. In order for two peers to be able to provide themselves with RTCPeerConnection, the SDP (Session Description Protocol) protocol is used. The protocol has a key-value structure and is essentially a description of a single peer (the name speaks for itself).
- WebRTC for the Curious
- Show HN: Bring phone calls into the browser (sip-to-WebRTC)
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Ask HN: What side projects landed you a job?
I have worked four jobs related to https://github.com/pion/webrtc and one for https://webrtcforthecurious.com
Two companies used Pion. The other two were just using the protocol (WebRTC)
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Replacing WebRTC: real-time latency with WebTransport and WebCodecs
For the WebRTC jargon check out https://webrtcforthecurious.com/
If that still doesn’t cover enough I would love to hear! Always trying to make it better.
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OBS Merges WebRTC Support
It is pretty easy to get a one way trip time for packets that is sub-second! You see it with conferencing and other real-time communication things.
If you are curious on the 'how' of WebRTC I wrote a Free/Open Source book that goes into the details https://webrtcforthecurious.com/. Happy to answer any particular questions you have.
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Help: I'm facing an issue developing a webrtc app
Read this thoroughly: https://webrtcforthecurious.com
What are some alternatives?
web-codecs - WebCodecs is a flexible web API for encoding and decoding audio and video.
broadcast-box - A broadcast, in a box.
ws-tcp-proxy - Simple websocket tcp proxy.
offline-browser-communication - Demonstration of a browser connecting to Pion WebRTC without a signaling server.
webrtc-rtptransport - Repository for the RTPTransport specification of the WebRTC Working Group
direct-sockets - Direct Sockets API for the web platform
ebook-diffuser - An end to end, customizable, ebook automation tool
libdweb - Extension containing an experimental libdweb APIs
WebKit - Home of the WebKit project, the browser engine used by Safari, Mail, App Store and many other applications on macOS, iOS and Linux.
goby - Goby - Yet another programming language written in Go
obs-studio - OBS Studio - Free and open source software for live streaming and screen recording