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A session's jitterbuffer can release a run of packets from different
frames on one PTS, as it does when a publisher joins a slot. The
slot's h264parse takes a PTS only when it differs from the previous
buffer's, so the later frames of the run leave it with none. Browser
video carries no frame timing to fill one in from, and the GPU vision
mixer fails the whole flow on a frame without a timestamp ("Need
timestamped buffers!").
When the slot bridge forwards a video packet from a new RTP frame, give
it at least 1 ns more than the previous frame's PTS. Packets of one
frame keep sharing theirs.
Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
A new frame moves past the previous frame's PTS only when its own is at most 1 ms behind it, the case a jitterbuffer run on one PTS produces. A larger step back is real timing and keeps its PTS: an encoder with B-frames sends a frame after the one it precedes, and moving it past that frame would bunch the decoded video. A frame far ahead does not hold back the frames after it, which would freeze the seat until real time caught up. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
srperens
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Verdict: Approve — the mechanism is traced end to end (one PTS across several frames, then baseparse keeps only a PTS that changed). The fix is bounded to that one shape, and B-frames and outliers keep their timing. The three new tests drive the real forward_sample_to_slot and all three ran in CI. It changes the same function as #759; either order works, see Overlaps.
Non-blocking notes
- The RTP read fails silently:
backend/src/blocks/builtin/whip.rs:1361—adjusted = order.order(rtp.timestamp(), adjusted);runs only if the header maps. That is right as long as the bridge carries RTP, which the rig anddecode=false's own description (outputs RTP (application/x-rtp)) both say it does. Butbackend/src/blocks/builtin/whip.rs:1201—// whipserversrc autoplugs RTP depayloaders inside its own bin, so thissays the opposite. If that comment is stale, correcting it here would stop the next reader doubting this fix. If an upstream change ever depayloads before the bridge, the fix turns into a silent no-op. - The body says the packets of one frame "keep sharing a PTS". The code enforces that:
backend/src/blocks/builtin/whip.rs:1271—if last_rtptime == u64::from(rtptime) {returns the first packet's PTS for every later packet of the frame. With rtpbin's default slave mode these already match, so in practice nothing changes. It is still a behaviour worth one line in the doc comment.
Claims
| Claim | Verdict | Evidence |
|---|---|---|
| Only a step back of at most 1 ms is moved; B-frames and far-ahead frames pass | CONFIRMED |
backend/src/blocks/builtin/whip.rs:1275 — last_rtptime != u64::MAX && pts <= last_pts && last_pts - pts <= MAX_PTS_NUDGE_NS;, and the tests video_b_frames_keep_their_pts and video_frame_far_ahead_does_not_hold_back_later_frames pin both sides |
| The regression test fails with the ordering bypassed | CONFIRMED |
video_frames_on_one_pts_reach_the_slot_in_order asserts out[2] > out[1] for packets (4000, 1000) after (1000, 1000). Without order() both leave at 1000 ms |
| No lock and no new allocation on the per-packet path | CONFIRMED |
Relaxed atomics plus a read-only RTPBuffer map. The copy happens only when the PTS changed, as it already did whenever offset_ns != 0 |
baseparse ignores a repeated PTS; vtdec/glvideomixer fail on PTS NONE |
EXTERNAL |
upstream behaviour, taken from the body's traces |
Diagnosis — Right layer. The bridge is the last point that still sees RTP timestamps before h264parse loses the PTS. Deriving PTS from RTP-time deltas would be the obvious alternative, and it is worse: a backlog released at once would be stamped ahead of real time, and the frames after it would then step backwards. The +1 ns nudge avoids that. Coverage is BOUNDED to runs of at most ~1e6 frames on one PTS, which in practice is every run.
Radius — LOCAL: the video bridge of every WHIP Input slot, codec-agnostic, for decode=true and decode=false. Audio is untouched. VideoPtsOrder is created per session pad, so a reused slot starts clean.
Overlaps — #759 (Approve at 9ba9941) changes the same code path: it moves forward_sample_to_slot into SessionBridge. Landing #759 first is the smaller rebase, because VideoPtsOrder moves as one unit. #854 (draft) regenerates after either. #901 restarts the slot decodebin, which is a different mechanism: a fresh h264parse still drops the repeated PTS.
Tests & CI — All Linux, WASM and API checks green at e90f52b. The job log shows all three new whip::tests::video_* tests ok, and pipeline_lifecycle_test ran. macOS/Windows skipped; nothing here is platform-specific.
Confidence: HIGH
Eyevinn#756, Eyevinn#792 and Eyevinn#805 drop AtomicU64 from whip.rs's imports once nothing else there uses it. Naming it by path keeps this struct compiling in either merge order. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com>
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Written by Claude (AI) on behalf of @srperens. This fix is now part of #957, which consolidates the WHIP Input fixes into one series against current main. Your commits are in it with you as author. Adapted: Suggest closing this once #957 merges. |
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Closing in favour of #957, which carries this work (see the comment above). |
… and #943 (#957) * fix(whip): stop an unstamped buffer at the session bridge before qtmux sees it A WHIP session's appsink -> appsrc bridge forwarded buffers with no PTS. A recorder downstream answers one with "Buffer has no PTS" from qtmux and GST_FLOW_ERROR, which takes the whole flow down: on a live rig, a second participant joining killed the first. The slot bridge now goes through gst::pipeline_bridge::SessionBridge: a buffer with no PTS is dropped and counted (warned on the first and every hundredth), the shared A/V offset is set once per session with compare_exchange and taken from the first stamped buffer, DTS is shifted with PTS, and a sample without caps no longer panics. From #759 (cfc4518, f62e777, 9ba9941), adapted to main: - The slot callback keeps the session_finished gate from #753 and the ingress liveness stamp from #756, wrapped in install_slot_bridge so the wiring test drives the real callback. - The Media Player half of #759 is left out. #945 replaced that bridge's timing with its own, so SessionBridge no longer fits there. Co-Authored-By: Per Enstedt <per.enstedt@eyevinn.se> Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * fix(whip): keep a slot's video PTS moving forward frame to frame A WHIP publisher joining a slot can take the whole flow down with a GPU vision mixer downstream: "Need timestamped buffers!" from videoaggregator. The session's jitterbuffer can release a run of packets from different frames on one PTS; h264parse takes a PTS only when it differs from the previous one, so the later frames of the run leave it with none, and a browser stream has no framerate to derive one from. A video packet that starts a new RTP frame and whose PTS is at most 1 ms behind the previous frame's now gets that PTS plus 1 ns. Packets of one frame keep sharing a PTS, and any other PTS (a B-frame, a step in the sender's clock, a frame far ahead) passes through. On the rig, 5 of 8 runs of publish / switch camera / leave / rejoin died before, 0 of 8 after. From #893 (745ce02, e90f52b, 2540522), adapted to the SessionBridge from #759: the ordering runs on the session PTS before the bridge rebases it, and copies the buffer only when the PTS moves. Co-Authored-By: Per Enstedt <per.enstedt@eyevinn.se> Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * fix(whip): stop a session's second stream stalling below PLAYING A WHIP session pipeline is already PLAYING when whipserversrc adds its pads, and each pad gets a `tee -> fakesink + appsink` branch synced into it. The appsink ran with `GstBaseSink`'s default `async=true`, so it answered that state change with ASYNC and waited for a preroll. The pads arrive one after the other within a millisecond, so the second branch lands while the first is still waiting, and the second is left below PLAYING: it takes a single buffer and then blocks its streaming thread for good. That seat's audio or video then never reaches the slot's appsrc, so its decodebin exposes no pad, and the flow's recorder and mixer wait on a track that will never arrive. Nothing in the log says so — `sync_state_with_parent` reports the state change as accepted either way. Turn `async` off, as the fakesink beside it already does. Measured on a five-seat meeting rig, 18 fresh-server trials per arm: 7/18 seats failed to start before, 0/18 after. The branch construction moves into `attach_session_branch` so the regression test builds the real thing rather than a copy of it. Co-Authored-By: Claude Opus 5 (1M context) <noreply@anthropic.com> * fix(thumbnail): turn off async on the thumbnail tap appsink The thumbnail branch is attached to a flow pipeline that is already running, and its appsink ran with GstBaseSink's default async=true, so it answered that state change with ASYNC and waited for a preroll. The activation guard admits the pipeline in PAUSED with PLAYING pending, which is that wait, and a branch attached inside one is left below PLAYING: it takes a single buffer and blocks its streaming thread. Activation returns without waiting for a preroll, and a branch is created per block from an HTTP request, so a client asking for several blocks' thumbnails serializes on one mutex but not on the async cycles behind it. Same defect and same fix as the WHIP session branch. Co-Authored-By: Claude Opus 5 (1M context) <noreply@anthropic.com> * fix(whip): give a reused slot a fresh video decoder A WHIP Input slot keeps its decode chain while publishers come and go. The video decoder that decodebin plugged for the first session kept that session's reference frames, reorder queue and configuration, and a later session with equal caps sends none to reconfigure it. On macOS, VideoToolbox then decoded a second Safari publisher at a few frames per second and flagged every frame of a third as an error, so the slot showed no video while its audio played. When a session claims a slot whose video decodebin has already run, restart that decodebin from an IDLE probe on the slot appsrc: unlink, set it to NULL, relink so the sticky events are sent again, and sync it with the pipeline. The existing pad-added handler links the new chain when data arrives. Audio keeps its running chain. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * test(whip): guard a codec change on a reused slot The test reuses a slot with a VP8 publisher after an H.264 one and asserts the VP8 session decodes. With the restart disabled, the kept decoder refuses the VP8 caps and the slot's source stops with not-negotiated. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * fix(whip): ask for a keyframe when a running session's video breaks A WHIP session asked its publisher for a keyframe only until video first decoded. After that, nothing asked again. A browser sends a keyframe only when asked, so a loss the jitterbuffer could not repair left the slot's decoder broken for the rest of the session: smeared pictures, or every frame rejected with an error flag, until the watchdog tore the session down and the guest rejoined. The slot's decodebin now reports damage through a per-slot flag: - a gap: the depayloader marks the first access unit after a sequence gap DISCONT; damaged until a keyframe with no gap in front of it leaves the depayloader (a repair keyframe can lose a packet too). - a silent decoder: data lost with no gap (the depayloader drops a fragment it cannot place) makes the decoder reject every frame without a bus message. 30 access units in with no picture out counts as damage, cleared when pictures come out again. Both are buffer probes that touch atomics only. A per-session thread on the session side reads the flag and sends the keyframe request on the whipserversrc pad, where it reaches the publisher: at once, then at most once a second, 10 times per damage episode, and never on healthy video. Each episode logs its recovery time and request count. Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> * test(whip): guard that a reused slot reports its video as decoding The keyframe requester stops once the slot's video_decoding flag is set, and that flag is set from decodebin's pad-added. Before the decoder restart on reuse, pad-added fired once per flow, so every reconnecting publisher was asked for the full request budget while its video decoded fine (#805). The restart brings pad-added back for each session. The reuse test now re-arms the flag as a new session does and asserts the second session sets it again. With the restart disabled it fails with "the second session decoded, but the slot never reported its video as decoding". Co-Authored-By: Claude Opus 5.5 <noreply@anthropic.com> --------- Co-authored-by: Peter Wagenet <peter.wagenet@gmail.com> Co-authored-by: Claude Opus 5.5 <noreply@anthropic.com>
Opened by Claude (AI), on behalf of @wagenet.
Current behaviour
A WHIP publisher joining a slot, including a reconnect or a second device on a reused slot, can take the whole flow down. With a GPU vision mixer downstream, the flow stops with
Internal data stream errorfromappsrc_overlay,queue_overlayand the slot'sappsrc_video_0. The GStreamer log shows the cause:On a local rig (browser publisher, same flow shape as the production that hit it, with the aspect fix #894 applied) this happened in 5 of 8 runs of "publish, switch camera, leave, second publisher joins".
Fix
When the slot bridge forwards a video packet that starts a new RTP frame, and its PTS is at most 1 ms behind the previous frame's, the packet gets the previous frame's PTS plus 1 ns. Packets of one frame keep sharing a PTS, and any other PTS passes through unchanged. With the fix, the same rig had 0 of 8 deaths.
Why frames lose their timestamp
rtpbasedepayload:6: seqnums 22859-22882, eight RTP timestamps, one PTS0:00:21.537999057.rtph264depaypasses that PTS on to every access unit.h264parse(baseparse) takes an input PTS only when it differs from the previous one (gstbaseparse.c,prev_pts != pts), so every later frame of the run leaves the parser with no PTS.framerate=0/1), so the decoder cannot interpolate one.vtdecpassed PTS NONE through (videodecoder:6:chain PTS 99:99:99.999999999for every frame after the second), andglvideomixerreturnsGST_FLOW_ERRORon the first frame without a timestamp.A standalone check (x264 → RTP → appsink/appsrc bridge →
rtph264depay ! h264parse), with frames 10-25 forced onto one PTS: 15 of 60 frames leaveh264parsewithout a PTS; with the ordering rule, 0.Only a PTS at most 1 ms behind the previous frame's is moved. A larger step back is real timing and passes through: a publisher whose encoder uses B-frames sends a frame after the one it precedes. Browsers never send B-frames, but other WHIP clients can. A frame far ahead does not hold back the frames after it.
How to review
VideoPtsOrderand the reworked tail offorward_sample_to_slotinbackend/src/blocks/builtin/whip.rs. This runs per RTP packet on the session's appsink thread: atomics and a read-only RTP header map, no lock and no allocation beyond the buffer copy the bridge already made when rebasing.SlotMediareplaces themedia: &strargument so the video ordering state can travel with it (clippy's argument limit).gst::pipeline_bridge::SessionBridgeand drops buffers that arrive without a PTS. This PR covers buffers that arrive with one and lose it inside the slot's decodebin; the production rig runs with fix(whip): stop an unstamped buffer at the session bridge before qtmux sees it #759 included and still hit this. Whichever lands second needs a rebase;VideoPtsOrderwould move into the bridge. refactor(whip): split whip.rs and whip_session_manager.rs into sub-modules #854 (generated split) regenerates after either.Tests
whip::tests::video_frames_on_one_pts_reach_the_slot_in_orderdrivesforward_sample_to_slotwith real RTP packets. With the ordering bypassed it fails ([1000000000, 1000000000, 1000000000, 1000000000, 1100000000]); with it, it passes.video_b_frames_keep_their_ptsandvideo_frame_far_ahead_does_not_hold_back_later_framesfail if every late frame is moved past the previous one (the rule without the 1 ms bound).cargo test --workspaceon this branch, macOS: 1063 passed, 0 failed./player/whip-ingestpage. Not run: a phone over cellular.🤖 Generated with Claude Code