obs-studio
broadcast-box
obs-studio | broadcast-box | |
---|---|---|
2,362 | 10 | |
55,593 | 519 | |
1.0% | 4.4% | |
9.8 | 8.9 | |
6 days ago | 1 day ago | |
C | Go | |
GNU General Public License v3.0 only | MIT License |
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.
obs-studio
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Software Engineering Workflow
OBS
- Open Broadcaster Software
- OBS merges AV1 support for WebRTC
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Ask HN: Has anyone achieved Douglas Engelbart's Vision?
Any specific area?
unix,telnet, uucp/news groups/email, linux, sequel/postgres, AI (chatgpt), video/hardware emulation with or/without VM layer. software defined radio, open broadcaster software[0], etc.
[0] obs : https://obsproject.com/
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My Rules for Being a Tech Speaker
OBS Studio - For recording
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Show HN: Bring phone calls into the browser (sip-to-WebRTC)
I don't! But when adding WebRTC support to OBS I would see 120ms https://github.com/obsproject/obs-studio/pull/7926
This is me (in Ohio) going to a Digital Ocean Droplet (NYC) and back.
- Denied OBS PR regarding Kick support lights up
- OBS with AV1 Support is now in the AUR
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Do not update to OBS 30.0.1 (MacOS)
Thankfully, you can simply delete the app, then reinstall the previous version here : OBS 30.0.0
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Obs Studio 30.0.1 Crashing when launched on MacOS Sonoma
Nvm, i found it after a bit more investigation: https://github.com/obsproject/obs-studio/releases
broadcast-box
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Show HN: Bring phone calls into the browser (sip-to-WebRTC)
> [1] https://github.com/Glimesh/broadcast-box
This is really interesting! Any latency benchmarks comparing it to gstreamer UDP, P2P primarily?
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OBS Studio 30.0
This release includes WebRTC support. I would really appreciate if people could try it! I have so many things I want to improve with broadcasting.
——
* Serverless Streaming - WebRTC is P2P so you can video right into your browser. You don’t have to stand up a server anymore to stream for a small audience.
* Sub-Second Latency - Create content and interact with viewers instantly. There is something magical about having a real conversation with your viewers.
* Multitrack Input - Upload your transcodes instead of generating then server side. Give viewers multiple video tracks to see action from all sides.
* Mobility - WebRTC lets you switch networks at any time. Go from WiFi -> Mobile with zero interruptions.
* E2E Authentication - WebRTC lets you add authentication on the sender and receiver. Broadcasters could sign their video streams so viewers can be sure it wasn’t modified.
Try it out with Broadcast Box. A reference server implementation. https://github.com/glimesh/broadcast-box
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Show HN: AV1 Broadcasting from OBS
This is a PR (with builds) to add AV1 support to the WebRTC output from OBS.
We also have been working on a reference server implementation with https://github.com/Glimesh/broadcast-box/ to make it easy to self host.
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Show HN: OBS with Simulcast Support
Simulcast builds of OBS are available. Simulcast is when a client generates multiple encodes of a video. Testers and feedback would be greatly appreciated!
Design Discussion: https://github.com/obsproject/rfcs/pull/55
Use it against your favorite WHIP ingest. If you don't have one check out https://github.com/glimesh/broadcast-box. It is a reference implementation of a WHIP/WHEP server.
I hope/believe the impact of this will be much greater then WebRTC. This could benefit all broadcast workflows.
* Higher Quality Videos - Decoding and re-encoding causes quality loss. Generating all the renditions from the source video is going to be a big improvement for viewers.
* Lower Latency - Removing the additional encoding/decoding allows video to be delivered faster.
* Reduce complexity - With Simulcast setting up a streaming server becomes dramatically easier. It is much easier for an individual to afford and manage a single broadcast server. RTMP -> Transcode -> HLS gets complicated and expensive quickly.
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OBS merges WebRTC support
One of the devs commented on HN that you can use Broadcast Box handle the initial connection. That's open source so you can self-host that too
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OBS Merges WebRTC Support
Try it out today with Broadcast Box. A reference server implementation. https://github.com/glimesh/broadcast-box. The PR to add WebRTC to OBS https://github.com/obsproject/obs-studio/pull/7926
I don't have an idea on how to do cascading OBS instances. I should write some software to make this easy though. I want to make 'self hosting' as easy as possible. I think https://github.com/glimesh/broadcast-box will fill that gap for now.
For my own usage I run WebRTC servers and have them 'replicate' the video traffic. LiveKit did a write up on this with some good graphics https://blog.livekit.io/scaling-webrtc-with-distributed-mesh...
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WebRTC support being added to FFmpeg
I am really excited for this PR to land. Being able to use WebRTC for Broadcast and Playback is going to be huge.
Instead of doing fixed interval keyframe we can use receiver feedback (Massive reduction in bandwidth)
Instead of server side generated transcodes we can use Simulcast. Will be better quality AND massively reduced server load.
If anyone wants to use with Pion check out https://github.com/Glimesh/broadcast-box
Then run `ffmpeg -re -f lavfi -i testsrc=s=1280x720:r=30 -f lavfi -i sine=f=440:b=4 -vcodec libx264 -pix_fmt yuv420p -profile:v baseline -r 25 -g 50 -acodec libopus -ar 48000 -ac 2 -f rtc -authorization "STREAM_NAME" "http://localhost:8080/whip"
What are some alternatives?
obsninja - VDO.Ninja is a powerful tool that lets you bring remote video feeds into OBS or other studio software via WebRTC.
go2rtc - Ultimate camera streaming application with support RTSP, RTMP, HTTP-FLV, WebRTC, MSE, HLS, MP4, MJPEG, HomeKit, FFmpeg, etc.
obs-StreamFX - StreamFX is a plugin for OBS® Studio which adds many new effects, filters, sources, transitions and encoders! Be it 3D Transform, Blur, complex Masking, or even custom shaders, you'll find it all here.
str0m - A synchronous sans I/O WebRTC implementation in Rust.
ShareX - ShareX is a free and open source program that lets you capture or record any area of your screen and share it with a single press of a key. It also allows uploading images, text or other types of files to many supported destinations you can choose from.
ffmpeg - FFmpeg Zig package
jellyfin-ffmpeg - FFmpeg for Jellyfin
webrtc-for-the-curious - WebRTC for the Curious: Go beyond the APIs
Kodi Home Theater Software - Kodi is an award-winning free and open source home theater/media center software and entertainment hub for digital media. With its beautiful interface and powerful skinning engine, it's available for Android, BSD, Linux, macOS, iOS, tvOS and Windows.
webrtc-echoes - Simple useful interoperability tests for WebRTC libraries. If you are a WebRTC library developer we'd love to include you!
openshot-qt - OpenShot Video Editor is an award-winning free and open-source video editor for Linux, Mac, and Windows, and is dedicated to delivering high quality video editing and animation solutions to the world.
rfcs - RFCs for changes to OBS Studio (and supporting toolset)