Keep RTMP source when an ingest joins a large call

[PratimMallick]

Goal

Fixes AND-1597 — livestream viewers stay on the host camera because an RTMP ingest is stored as PARTICIPANT_SOURCE_WEBRTC_UNSPECIFIED.

LivestreamPlayer already prefers RTMP via bySourcePriority (#1684). That sort only works when the stored source is actually RTMP. Two paths drop it.

Ingest joins an 8+ call the viewer is already in. Under 8 participants, ParticipantJoined copies participant.source. At 8 or more, that event is debounced. TrackPublished runs immediately and used to construct the participant with no source. The later flush in getOrCreateParticipants also dropped it.source. source is a constructor value, and the get branch of getOrCreateParticipant returns the entry already in the map, so nothing after that can repair it.

OBS drops and comes back as a new session. The call stays up because the browser camera never left, so both platforms fall back to the camera while the ingest has no video. iOS builds the new session with toCallParticipant(), which copies source from the protobuf, and returns to OBS. Android 1.33 created that new session through the same 8+ path and stayed on the camera. A same-session pause would have kept Android's source, so the "iOS returned, Android stayed" report is this rejoin, not a separate pause bug.

Viewers who join after the ingest is already live still see OBS. The join snapshot copies source. The bug is the viewer who is already in the large call when the ingest joins or rejoins.

Implementation

• getOrCreateParticipants passes it.source when the debounced join creates the participant.
• TrackPublished and TrackUnpublished in CallState, including the deprecated livestreamFlow collector, pass event.participant?.source when the participant is present. Those events can create the participant before the debounced join runs.
• ParticipantState.source stays a constructor value. The get branch of getOrCreateParticipant is unchanged.
• Events that carry only a session id and user id (AudioLevelChanged, ConnectionQualityChange, DominantSpeakerChanged) are unchanged. If one of those creates the session first, a later event still cannot repair the source.
• pendingParticipantsJoined is not cleared here. That is AND-1606.

Testing

Manual, Pixel 8, same call livestream:zEGoHSaCAz9v3SLLOZg90 with 9 local participants. Host camera was host-yYuTuF4MkPV00b7ByWNB3 (role host). OBS was a different user, viewer-5Ibbgm23ji6qicptXmC2Q (role user), publishing over RTMP. The phone was already in the call before OBS connected.

• Previous versions : the SFU sent PARTICIPANT_SOURCE_RTMP and the SDK stored PARTICIPANT_SOURCE_WEBRTC_UNSPECIFIED. The phone still showed OBS, because the ingest was publishing audio and that comparison runs before host role. That is the 1.30.0 outcome where some viewers see OBS and some stay on the camera, depending on who wins the fallback sort.
• Debug dogfood 1.35.0-DEBUG from this branch, before OBS published video: each join stored PARTICIPANT_SOURCE_RTMP and sorted the ingest ahead of the host, then the session left a second later. There was no TrackPublished, so the player stayed on the host camera. That is the only video still in the call.
• Same debug build, once session 461ffa30 published video and audio: the SDK kept PARTICIPANT_SOURCE_RTMP, hid the host viewport, and subscribed to the OBS track (720×1280, then 1080×2205). The host stayed role host and the ingest stayed role user, so the switch was the source sort.

The interruption path was confirmed from the 1.33 event handling and the iOS toCallParticipant() path. It was not re-run as a drop-and-return on this debug build.

./gradlew spotlessCheck and ./gradlew apiCheck passed in the pre-push hook. detekt and testDebugUnitTest were not run.

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Summary by CodeRabbit

• Bug Fixes
  • Participant source information is now preserved when handling track publication and participant creation events. When source metadata is unavailable, a default source is used.

[PratimMallick]

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PR checklist ✅

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• Title length is OK (or ignored by label).
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Walkthrough

CallState now passes participant source metadata to participant state creation from track events and bulk participant creation. Track event handlers use the unspecified WebRTC source when participant metadata is absent.

Changes

Participant Source Propagation

Layer / File(s)	Summary
Forward participant source stream-video-android-core/src/main/kotlin/io/getstream/video/android/core/CallState.kt	Track-published and track-unpublished handlers pass the event participant’s source, with an unspecified WebRTC fallback. Bulk participant creation passes each participant’s source.

Priority: ➖ Normal

Estimated code review effort: 2 (Simple) | ~5 minutes

Change: Bug fix

Suggested reviewers: aleksandar-apostolov

Merge Risk: 🔵 Low · up to 1d5a7

This change fixes the main case where an RTMP ingest joining a large call was shown as the host camera. One event order can still leave the participant marked with an unspecified source. The change is reasonable to merge, with a follow-up to update the source on an existing participant.

🚥 Pre-merge checks | ✅ 4 | ❌ 1

❌ Failed checks (1 warning)

Check name	Status	Explanation	Resolution
Docstring Coverage	⚠️ Warning	Docstring coverage is 0.00% which is insufficient. The required threshold is 80.00%. Docstring coverage is scoped to functions touched by this diff. Analyzed 2 functions across 1 files.	Write docstrings for the functions missing them to satisfy the coverage threshold.

✅ Passed checks (4 passed)

Check name	Status	Explanation
Linked Issues check	✅ Passed	Check skipped because no linked issues were found for this pull request.
Out of Scope Changes check	✅ Passed	Check skipped because no linked issues were found for this pull request.
Title check	✅ Passed	The title clearly and concisely describes preserving the RTMP source when an ingest joins a large call, which is the main change.
Description check	✅ Passed	The description covers the goal, implementation, and testing. It also explains relevant limitations and distinguishes the affected rejoin path. UI materials and checklist items are omitted, but they a…

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Inline comments:
Review comments at
@stream-video-android-core/src/main/kotlin/io/getstream/video/android/core/CallState.kt:
- Around line 424-425: Update getOrCreateParticipant and the event handling
paths that assign ParticipantState.source so authoritative source metadata
replaces WEBRTC_UNSPECIFIED on an existing participant; preserve the unspecified
fallback when metadata is absent.

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SDK Size Comparison 📏

SDK	Before	After	Difference	Status
stream-video-android-core	12.38 MB	12.38 MB	0.00 MB	🟢
stream-video-android-ui-xml	5.66 MB	5.68 MB	0.02 MB	🟢
stream-video-android-ui-compose	6.19 MB	6.19 MB	0.00 MB	🟢

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Quality Gate failed

Failed conditions
14.6% Coverage on New Code (required ≥ 80%)

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🚀 Available in v1.36.0