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WIP: Google/libwebrtc calls migration #29

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dinoxim commented 2026-07-07 13:03:09 +02:00
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Planning and audit branch for replacing the current GStreamer RTP call engine with a real Google/libwebrtc engine.

Do not merge yet. This PR exists to make the long-running feature branch explicit in Forgejo and to avoid the ambiguous "new pull request" banner.

Current contents:

  • Google/libwebrtc migration plan
  • Slice 1 audit of the current GStreamer/WebRTC-like call code

Runtime code is unchanged in the current commit.

Planning and audit branch for replacing the current GStreamer RTP call engine with a real Google/libwebrtc engine. Do not merge yet. This PR exists to make the long-running feature branch explicit in Forgejo and to avoid the ambiguous "new pull request" banner. Current contents: - Google/libwebrtc migration plan - Slice 1 audit of the current GStreamer/WebRTC-like call code Runtime code is unchanged in the current commit.
Route calls, device handling, and media-message recording through the external libwebrtc worker. Add VP8 decoding, Jingle interoperability, Linux packaging, Windows runtime scaffolding, and recorder lifecycle regression coverage.
Defer heavy preferences content until after first paint, keep OpenPGP loading feedback visible, and add focused UI timing markers. Apply Windows backup restores after DinoX exits, document the validated WebRTC distribution flow, and trim temporary recorder diagnostics.
Defer server administration command row population to keep the dialog responsive.

Preserve cached OpenPGP preference widgets, show missing-backup results reliably, and refresh the encryption account picker after live account additions.
Route MUC room-state notifications through account-and-room keyed observers, avoid duplicate conversation selection, and release quiet live member snapshots without waiting on separate join bookkeeping.
Complete the Slice 5 contact-model work with bounded list factories, account-scoped live roster updates, authoritative roster replacement, and focused regression coverage. Keep conversation selection queries bounded and retain server-confirmed contact removal semantics.
- make occupant snapshots, bookmarks and MUC PM routing lifecycle-aware
- validate room administration, configuration and timed bans
- stabilize member UI, call teardown and Windows input handling
- expand XMPP, OpenPGP and UI regression coverage
Apply verified room configuration deltas to live occupant state and keep roles, hats, avatars, JID visibility, and invitations aligned with server behavior.\n\nPersist voice requests in the chat UI, fence their lifecycle by account and room, and retract stale desktop actions across notification providers.\n\nAdd focused XMPP, service, UI, snapshot, and data-form regression coverage.
build(release): refresh libwebrtc runtime hashes
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This pull request is marked as a work in progress.
This branch is out-of-date with the base branch
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From your project repository, check out a new branch and test the changes.
git fetch -u origin feature/google-webrtc-calls:feature/google-webrtc-calls
git switch feature/google-webrtc-calls

Merge

Merge the changes and update on Forgejo.

Warning: The "Autodetect manual merge" setting is not enabled for this repository, you will have to mark this pull request as manually merged afterwards.

git switch master
git merge --no-ff feature/google-webrtc-calls
git switch feature/google-webrtc-calls
git rebase master
git switch master
git merge --ff-only feature/google-webrtc-calls
git switch feature/google-webrtc-calls
git rebase master
git switch master
git merge --no-ff feature/google-webrtc-calls
git switch master
git merge --squash feature/google-webrtc-calls
git switch master
git merge --ff-only feature/google-webrtc-calls
git switch master
git merge feature/google-webrtc-calls
git push origin master
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