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Author SHA1 Message Date
shadowdaoandClaude Sonnet 5 af7d2c6d24 fix: pin video quality to stop OBS source resizing; add audio-only mode
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Live testing (2026-09-07) showed two real problems in one root cause:
LiveKit's default subscriber behavior lets the SFU switch simulcast
layers on its own bandwidth/adaptive logic, and this plugin never told
it not to. For a real camera, that showed up as the OBS source's
received frame size visibly hopping between 320x180/640x360/1280x720
mid-show -- OBS's async video source resizes to match, breaking any
manual crop/position a director had set up. For the soundboard (a
Camera-source track that exists only to satisfy RTMP's video
requirement -- Soundboard.tsx -- with no real visual content), the
same instability, plus the video showing at all, was pure noise: there
was no way to pull just its audio.

Both come from RemoteTrackPublication (livekit/remote_track_publication.h
in the pinned SDK), on the exact publication object TrackSubscribedEvent
and attachExistingTracks already hand this code:

  - setVideoQuality(VideoQuality::HIGH) on every wanted video track,
    unconditionally, so the SFU always sends the top simulcast layer
    instead of switching layers underneath a source with no
    rendered-size hint to give it (this is a native subscriber, not a
    sized <video> element).
  - A new SessionConfig::subscribe_video (mirrors subscribe_audio):
    when false, the wanted video track is never attached, and its
    publication is explicitly setEnabled(false) -- the SFU stops
    sending it, not just "decoded and discarded here". Wired to a new
    "Audio only (no video)" checkbox in the source's properties.

Both call sites (a fresh TrackSubscribedEvent, and attachExistingTracks
sweeping tracks already up when the session starts watching) go
through one new handleWantedVideoTrack() so they can't drift apart.

Not unit-testable without a real LiveKit connection (RemoteTrackPublication
isn't fakeable, matching why test_integration_livekit.cpp already needs a
real server) -- verified instead by a full local build against real
libobs-dev + the pinned SDK (clean compile, all 6 existing tests still
pass) and CI. The actual behavioral fix -- stable resolution, no video
for an audio-only source -- needs the same real-OBS verification every
other claim in this repo's "What is verified, and how" section does.

Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01RL8abRmgFXkVASHkkqiJbE
2026-09-07 10:43:54 -07:00
shadowdaoandClaude Sonnet 5 969b8db94a license: relicense first-party code from GPL-2.0-or-later to Apache-2.0
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Owner sign-off: replace root LICENSE with Apache License 2.0, add a root
NOTICE file, and swap the GPL-2.0 boilerplate header in every first-party
core/ and obs-adapter/ source file for a short Apache-2.0 notice.

This resolves review finding C2 (GPLv2 top-level LICENSE vs. the vendored
Apache-2.0 LiveKit SDK is a license-compatibility violation): the whole
repo is now Apache-2.0, matching LiveKit, so there's no GPL/Apache clash
left. Updated the README Status gate and the CI workflow comment to reflect
that C2 is resolved, while leaving the C1 WebRTC/OpenH264 patent/royalty
gate untouched -- that question is still open and still blocks release.

third_party/ stays under its own upstream licenses; only this project's own
code changed hands. All 6 CTest suites still pass after the header swap.

Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01RL8abRmgFXkVASHkkqiJbE
2026-09-07 04:44:16 -07:00
shadowdaoandClaude Sonnet 5 80904a3e85 Add the LiveKit session wrapper, verified end-to-end against a real room
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stplugin::LiveKitSession wraps livekit::Room for exactly one subscribed slot:
connect with the wsUrl/lkToken the API client minted, find the chosen
participant's camera (and microphone), and hand decoded frames to
callback-shaped handlers the OBS adapter can consume directly.

Two architectural decisions worth recording, both forced by reading the SDK
rather than guessed:

1. Frames come from VideoStream/AudioStream::fromTrack with our own reader
   threads, NOT from Room::setOnVideoFrameCallback. The dispatcher API is
   keyed by (participant identity, track NAME), which we cannot know before
   the track is published -- and disassembling liblivekit.so confirms that
   both Room::setOnVideoFrameCallback and the dispatcher's own
   setOnVideoFrameCallback merely record the registration: neither starts a
   reader for a track that is already subscribed. Registering after the
   subscription event, which is the only time the track name exists, would
   therefore have silently produced no video. Taking the shared_ptr<Track>
   straight off the TrackSubscribedEvent sidesteps the name entirely, and
   lets us pick the camera by TrackSource (streamer-tools publishes cameras
   as Source.Camera and screenshares separately -- apps/web/src/avatar/
   publish.ts), which is what we actually mean.

2. Every stream operation runs on one owned worker thread, never on a room
   event thread. The SDK documents that Room::disconnect() from inside a
   delegate callback deadlocks, and Room's own event dispatch holds a mutex,
   so delegate callbacks only ever enqueue a command here.

VideoStream::Options::capacity is set (3 frames) so the SDK's queue is a
drop-oldest ring buffer: a stalled consumer can only fall three frames
behind, and what it then sees is the newest frame rather than a backlog.
That is the structural answer to the stale-media bug that motivated this
plugin.

The pure decision-making -- the state machine, track selection, frame
geometry validation -- lives in session_types.h/.cpp with no LiveKit or OBS
types, so it is unit-testable headlessly (81 checks in test_session,
including the publisher-swap and reconnect transitions, plus the real
connect() failure paths against the real SDK: unreachable host, garbage
token, incomplete config, and destruction mid-connect).

test_integration_livekit is the test that proves media actually flows. It
publishes a synthetic camera and microphone into a real LiveKit room using
the same SDK, subscribes through LiveKitSession, and asserts on the exact
fields the OBS adapter will dereference. It skips (exit 0) unless
STPLUGIN_IT_* is set, so the three build runners stay green;
scripts/livekit-dev-room.py mints the tokens for a local `livekit-server
--dev`.

Verified locally against livekit-server 1.13.6 in dev mode:

  integration_livekit: 36 video frames, 323 audio frames, 10 state changes
  integration_livekit: 32 checks passed

covering: connect; subscribe to the named participant's camera; 320x240
I420 frames with three planes, non-null plane pointers and strides >= the
frame's own width; 48kHz audio; unpublish -> hasVideo() false, state stays
Connected (a dark camera is the placeholder state, never an error) and NO
further frames arrive from the dead publisher; republish -> video resumes;
clean disconnect.

One real finding from that run, now handled: WebRTC ramps a new subscription
up from a downscaled spatial layer, so the first frames after (re)subscribing
legitimately arrive smaller than what is being published. The OBS adapter
must cope with a mid-stream resolution change; the test asserts per-frame
geometry rather than the publisher's, and separately asserts the stream does
reach full size.

Full suite: ctest -> 6/6 passed.

Co-Authored-By: Claude Sonnet 5 <noreply@anthropic.com>
Claude-Session: https://claude.ai/code/session_01RL8abRmgFXkVASHkkqiJbE
2026-09-06 21:39:49 -07:00