fix: clear pre-live capture ring backlog at StartLive (2.2s A/V sync lag)
AudioMixer.Start() begins mic/loopback capture at app startup to feed the level meters, so both 2s ring buffers fill with pre-live audio. StartLive() drained from the oldest tail sample, putting every recorded event ~2.2s late in the audio track (confirmed by clap analysis + cross-correlation on two takes: +2.11 to +2.22s). Fix: AudioMixer.StartLive() now runs _micBuffer.Clear() + _loopbackBuffer.Clear() immediately after the pipe starts, before the drain task runs. Recording now begins at go-live; the <=10ms in-flight chunk evicted by Clear() is imperceptible. Regression test: StartLive_DiscardsPreLiveBacklog_SoFirstAudioIsCurrent saturates the loopback ring with stale 0.8 pre-live audio, then asserts the wire carries fresh post-live 0.2 (max < 0.3). Reference (external, per derivative-work rule): OBS 'Audio mixer' keeps its buffers fed continuously and syncs the stream start timestamp at record time rather than replaying pre-live capture; a go-live flush of the capture buffer is the accepted pattern for live tools restarting a stream. Docs: HANDOFF.md (fix shipped), MyMistakes.md (A/V sync measurement recipe).
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@@ -1,152 +1,126 @@
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# HANDOFF — 2026-09-12 (webcam FIXED; audio-silence FIXED; TRUNCATION ROOT-CAUSED + FIXED for static scenes)
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# HANDOFF — 2026-09-14 (AUDIO−VIDEO SYNC FIX SHIPPED: StartLive now clears the pre-live ring backlog)
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## Branch / Commit State
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`main` HEAD = `a62a283`. Ahead of origin by 25 commits. **NOT pushing** — user
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ruling: no push until web-overlay transparency AND audio-silence are addressed.
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`main` HEAD = `5ba4d70`. Ahead of origin by **27 commits**. Working tree **dirty**:
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the sync fix is implemented + tests green below, HANDOFF/MyMistakes updated — NOT yet
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committed. **NOT pushing** — the prior no-push ruling (transparency + audio-silence) is
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clear on the first gate; the second was the sync issue, whose fix is now in the tree
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but the user has not greenlit a push. On-board next: verify a take, then decide.
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Uncommitted (all belong to the current work, none committed yet):
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Heads-up for any reader: the previous HANDOFF (a62a283 era) described webcam/audio/
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truncation fixes as **uncommitted** — that file was stale on arrival. They were
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actually already committed:
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```
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M .gitignore
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M HANDOFF.md (this file)
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M MyMistakes.md (webcam collateral recipe added)
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M Services/Audio/AudioSyncDelay.cs (audio-silence fix)
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M Services/Compositor/SceneCompositor.cs (webcam cbW/cbH fix)
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M Services/MediaCaptureFrameSource.cs (diagnostics REMOVED — clean)
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M ytLive.Tests/AudioSyncDelayTests.cs (silence regression test, proven both ways)
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M ytLive.Tests/SceneCompositorTests.cs (webcam regression test, proven both ways)
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M Services/Encoder/FramePump.cs (slow-render probe, see TRUNCATION below)
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```
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- `724af14` fix: audio silence (idempotent `AudioSyncDelay.Configure`), webcam gray
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block (cbW/cbH default), truncated videos (SceneGraph split/static bake docs + FramePump probe)
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- `5ba4d70` docs: audio-sync offset measured ~367ms w/ 300 offset → "set SyncOffsetMs=0"
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Build is green (0 warnings); 295 tests pass (audio + compositor + pump suites).
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Build was green (0 warnings) and 295 tests passing at the prior session end; no code
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changed this session (analysis only).
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Audio.SyncOffsetMs changed DB-side to 0 (see AUDIO section) — HANDOFF-only change,
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nothing new to compile.
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## ✅ SHIPPED — pre-live capture ring backlog fix (2026-09-14)
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## ✅ TRANSPARENCY — CLOSED (pushed gate #1)
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**Root cause:** `AudioMixer.Start()` begins mic and loopback capture at **app startup**
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for the level meters. The 2-second ring buffers (`_micBuffer` = 96k floats = 2.0s mono;
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`_loopbackBuffer` = 192k floats = 2.0s stereo) fill continuously with pre-live audio.
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`StartLive` → `LiveLoopAsync` drained from the ring's tail — the **oldest** sample —
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so every recorded event landed ~2.0s late in the audio track (fixed lag, scaled with
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time-since-app-launch, capped at ring depth).
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User confirmed transparency is fixed (`efe88b8` era). Push gate #1 clears.
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**Fix:** `AudioMixer.StartLive()` now runs `_micBuffer.Clear(); _loopbackBuffer.Clear();`
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immediately after the pipe starts, before the drain task runs — the in-flight ≤10ms
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chunk loss is imperceptible and correct ("recording begins at go-live").
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## ✅ WEBCAM GRAY BLOCK — CLOSED
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**Regression test (the ONE integration test for this change):**
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`StartLive_DiscardsPreLiveBacklog_SoFirstAudioIsCurrent` in
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`ytLive.Tests/AudioPipelineTests.cs` — saturates the loopback ring with stale 0.8
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(simulated pre-live meters), StartLive, then emits fresh 0.2; asserts the wire carries
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~0.2 (max < 0.3), failing loudly if the stale 0.8 backlog survived the clear.
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Root cause: `SceneCompositor.BlitContentRaw` defaulted `cbW=cbH=0` when
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`src.CropBounds` is null — regression from `ed9d7c1`. Every webcam frame (no crop
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bounds) blitted to an empty 0×0 source rect → nothing drawn where the webcam
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should be. Fix: default `cbW=src.Width`, `cbH=src.Height`. Regression test
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`BlitContentRaw_NoCropBounds_SamplesAcrossFullSource_NotJustOrigin` proven BOTH
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ways (fails on old code, passes on new). User confirmed correct webcam render in
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the latest recording. All temp diagnostics stripped from
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`MediaCaptureFrameSource.cs` and `SceneCompositor.cs`.
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**Test collateral:** the 3 existing pipe-emit tests pre-filled the rings BEFORE
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StartLive (relying on the bug as a reservoir). Their emits now land right after
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StartLive (post-clear, pre-connect — the pipe drops pre-connect writes, so this keeps
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the ring primed for the client) with the 2026-09-14 comments updated. All passed.
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## 🟡 AUDIO SILENCE — ROOT-CAUSED AND FIXED, pending final user take
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Verified: `AudioPipelineTests` 26/26 green. Room-native build 0 warnings. Full gate
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pending (verify.sh clean build + full suite + scope check).
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**Symptom:** mic/desktop levels showed real values in preview and in the
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per-5s `Audio live:` telemetry, but `peakMix` stayed exactly `0.000` and saved
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recordings were digital silence (-91 dB, verified with `ffprobe -af
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volumedetect`).
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### Read/Write semantics confirmed (AudioRingBuffer.cs)
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**Root cause:** `AudioMixer.FillAndMix` calls `_syncDelay.Configure(...)` on
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EVERY ~10ms tick (it re-reads the live UI setting each mix). The old
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`AudioSyncDelay.Configure` unconditionally reallocated and zeroed the delay
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buffer AND reset `_writePos=0` on every call — discarding the just-written
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audio before the delay offset could ever read it back. With offset 0 the early
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passthrough masked it; any non-zero saved offset = total silence. The DB has
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`Audio.SyncOffsetMs = 500` (left over from TASK 22 testing) → that's what
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triggered it.
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Thread-safe via `_sync`: Write evicts oldest when full; Read returns from
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`tail = (_head - _count + C) % C` — when full `tail == _head`, so every drained sample
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is one lap behind the last write. `Clear()` is locked; it is the right seam.
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**Fix:** `Configure` now early-returns when the delay samples haven't actually
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changed (still reallocates/flushes on a *change*, so the live slider still
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clicks rather than smearing). Regression test
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`RepeatedConfigureWithSameDelay_EveryTick_StillDelivers_TheMarker` mirrors the
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mixer's exact calling pattern and was proven BOTH ways (marker lost = silence
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on old code; marker survives → expected 960-sample shift on new).
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**Root cause (confirmed two independent takes):** `AudioMixer.Start()` begins mic and
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loopback capture at **app startup** for the level meters. The 2-second ring buffers
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(`_micBuffer` = 96k floats = 2.0s mono; `_loopbackBuffer` = 192k floats = 2.0s stereo)
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fill continuously with pre-live audio. `StartLive` → `LiveLoopAsync` drains from the
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ring's tail — the **oldest** sample, which is the moment-of-go-live sample minus the
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full ring depth — so every recorded event appears ~2.0s late in the audio track. The
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lag is **fixed** for the entire recording and scales linearly with "how long the app was
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open before you hit record", capped at the 2.0s ring depth (plus the configured sync
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offset plus ~0.1s encoder pipeline).
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**Telemtery proof the fix works:** latest take (ty-20260912-1135-0000-2.mp4,
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recorded WITH the fix, sync offset still 500): `peakMix` = 0.434–0.523 across
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all five 5s windows (was 0.000 before). **Audio is NOT silence anymore.**
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**Read/Write semantics confirmed** in `AudioRingBuffer.cs` (thread-safe via `_sync`):
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Write evicts oldest samples when full (`_count == _buffer.Length`); Read returns from
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`tail = (_head - _count + C) % C` — when full, `tail == _head`, meaning every drained
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sample is exactly one lap behind what was last written. `Clear()` exists and is locked;
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it is the right seam.
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**Follow-up:** sync offset `Audio.SyncOffsetMs` measured and SET TO **0** (2026-09-12, take ty-...-1358):
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**Only one Clear call exists** in the capture path: `RestartMic()` (AudioMixer.cs:132)
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clears `_micBuffer` on device swap. **Neither buffer is cleared at `StartLive`** — this
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is the bug.
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Measurement method tri-wins instead of clap deltas: `silencedetect` on the audio
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gave the count-word starts (one≈0, two@4.12, three@6.48 — audio PTS, both streams
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start at 0, no mux offset), and per-frame video motion (`tblend=difference` +
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`signalstats YAVG`) cross-correlated against the audio envelope peaks at shift
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**+22 frames → audio LAGS video by ~367ms with the 300ms offset active** → the
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pipeline's own lag is only ~70ms. The saved 300ms (TASK 22 leftover) is what made
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it read as "one word late" (370ms ≈ a third of the ~2.3s pause ≈ perceptually a
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full beat behind). Fix = offset 0. Expected residual ≤~70ms (imperceptible).
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**Taking:** the lag also explains the earlier 13:58 take that measured only ~367ms
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with 300ms offset active: the app had just been restarted for that take, so the rings
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were only partially pre-filled (lag = min(2.0s, time-since-app-launch)). Two takes
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hours later (rings fully saturated) reproduce the full ~2.0s backlog consistently.
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## ✅ TRUNCATED VIDEO — ROOT-CAUSED AND FIXED (for baked/static scenes)
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---
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**Symptom:** recordings ran ~1s short at first, then HALF-length: 22.8s wall →
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10.48s file. Not a mux/`-shortest` cut — the pump simply produces frames slower
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than the declared 60fps.
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**Take 1** (ty-20260912-1923-0000-2.mp4, talking + 3 claps):
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- Audio 17.25s, video 17.05s (1023 frames), no frame drops — NOT truncated.
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- Per-clap offsets: +2.16 / +2.18 / +2.15s (audio later). Cross-correlation peak:
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**+133 frames (+2.217s)**, sharp and unique. Fixed offset, no drift.
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- DB: `Audio.SyncOffsetMs = 300` (not 0 as the 5ba4d70 commit message claimed —
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the write never actually landed).
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**Mechanism (proven):** ffmpeg rawvideo timestamps frames by ARRIVAL at declared
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`-r 60`. Pump deadline pacing (OBS libobs pattern) yields each frame when its
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~17ms slot lands; if the render stage for that tick takes longer than the slot,
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the pump can only do ~1 frame per 35ms → ~27–28fps of *submitted* frames → file
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length ≈ submittedFrames/60. So a take that *should* be 22.8s muxes to ~10.5s,
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and logs show `avg render 35–38ms` with 136/300 per 5s. FramePump had NO way to
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tell which compositor pass ate a slow frame.
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**Take 2** (ty-20260914-1104-0000-2.mp4, clean clap test):
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- Audio 22.31s, video 22.10s (1326 frames), no frame drops.
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- Audio clap peak at **14.04s**; video motion peak at **11.93s** (yavg 2.75 vs
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background ~0.1–0.5). Offset: **+2.11s** — confirms the same mechanism.
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- Tiny pre-clap blips at 0.2/1.5/4.5/13.5s are faint artifacts (< 100 RMS);
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the 11617-RMS clap is unambiguous.
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**Root cause of the 35ms:** the scene was NOT actually fully static. Any dynamic
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element still in the scene — **including hidden ones**, which still count as
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Dynamic in `SceneGraph.GetSplitPoint` — forces the per-frame composite path. If
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it sits at split 0, `GetBakedBase` returns null and every tick is a FULL render
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(no baked base). The 12:59/13:39 takes logged `resolve 1.0ms` per frame — a live
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element was still resolving, confirming dynamic content was present.
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**Theory of the "cut off at the end":** audio events land ~2s late; when the video
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ends the audio track is still playing the final 2s of real-time content, giving the
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impression of a cut. The audio track's slightly longer duration (17.25 vs 17.05;
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22.31 vs 22.10) is the tail of that delayed signal — real audio outlives the
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pump-stopped video.
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**The fix that matters (no code changed in the renderer):** a truly static scene
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renders in <1ms. New slow-render probe (`ProbeRender` in FramePump.cs) names the
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path; the 13:53 take of a 1-element static scene logs
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`render=fully-static split=1 dynamic=0 bakeMs=246 totalMs=246` on the FIRST
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frame (cold raster of the background at 1080p), then `0.8→0.0ms` avg render,
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`301/300` frames per 5s, 0 stalls → **12.46s file from 12.29s wall, 735 frames
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@ 60fps. No truncation.**
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**Theories RULED OUT by data:**
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- ❌ Frame truncation / slow full-render — video is real-time; frame counts match
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wall-clock (FramePump log: 301/300 per 5s, dropped 0, avg render 14ms).
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- ❌ Named-pipe-connect drop — would *shorten* audio (events earlier), opposite sign.
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Pipe connected near-instantly in both takes.
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- ❌ AudioSyncDelay alone — capped at 0..500ms; 300ms is a subset, not the whole.
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- ❌ Audio drift/resampler rate — offset is constant, not growing.
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**Rule for recording:** every scene used for RECORDING should be ≥1 static
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element first (background image) with all web/cam/chat widgets AFTER it in
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layer order (split ≥ 1 → bake feasible) — a scene whose FIRST layer is dynamic
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at recording time will truncate. If a take with widgets comes back truncated
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again, the probe now points at whether it took `fully-static` (bake), or
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`base+layers` / `full-render` (composite) and how long.
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## Open threads
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**Probe left in place** (≈0 overhead: one Stopwatch, string built only when a
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render ≥20ms): it is the diagnostic that named this bug; keep it until the
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dynamic-scene path is proven fast too, then remove.
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## 🔴 OPEN — user-reported "video seems truncated ~1s" (unconfirmed)
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Recording ty-20260912-1135-0000-2.mp4: format duration 26.370s, video 26.167s
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(1570 frames @ 60fps), audio 26.370s. FramePump stats for the whole take show
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`dropped 0` and 300/300 frames every 5s window. End-frames (n=1500 vs 1560 vs
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1569) differ — not a frozen/repeat-last-frame truncation. Log shows recording
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window 11:35:01.386 → 11:35:27.562 = 26.2s of video: matches the file. No
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evidence of truncation found in data. Possibly user perceived the 500ms audio
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delay (audio trailing video start / leading video end) as truncation — worth
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re-checking with offset zeroed. If a real freeze/truncation reappears: pull
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last-3-frames diffs and confirm_frame timestamps next.
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## Other threads (paused)
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- **Audio-silence verification** — the FIXED take has real peakMix; final =
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user re-records with the finalized offset and confirms audible audio in playback.
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- **Audio sync offset final value** — set to 0 via DB (measured residual ~70ms).
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NEEDS user re-take to confirm (the running app may still hold 300 in memory —
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if unsynced on the re-take, use the UI slider to 0, or restart the app BEFORE
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recording so the 0 loads).
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- **Truncation with DYNAMIC scenes** — proven mechanism; static scenes now
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record full-length. The 13:58 take was a 6-element/4-dynamic scene
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(`split=0 → full-render`, ~34ms, 20 stalls/5s): fully full-length (10.98s)
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but JITTERY (judder from the 34ms hard frames). If a widget-heavy scene reads
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badly again, the ProbeRender log names the stage; the next move is getting a
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baked base by reordering static elements first or optimizing full-render.
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- **Webcam MJPG missing** — "MJPG negotiation refused (being used by another
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process)". Queued.
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- **Web capture speed** — ~10-14Hz effective. Revisit only on request.
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- **Layer order** — SortOrder not persisting on drag. Queued.
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- **Audio-silence verification** — FIXED code (idempotent `Configure`, committed
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`724af14`); the 19:23 take had real peakMix. Final user confirmation still pending.
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- **Audio sync offset DB value** — `Audio.SyncOffsetMs = 300` (the "set to 0" write
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in 5ba4d70 never actually landed). Fix for the ~2s ring backlog is now SHIPPED (local,
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uncommitted). Next: re-record the clap pattern, run the cross-correlation recipe → expect
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residual ≈ 400ms (300ms configured sync + ~100ms pipeline); then optionally zero the
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slider to hit ~100ms.
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- **Truncation with DYNAMIC scenes** — proven mechanism + probe in place; static
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scenes full-length. 19:23 take confirms dynamic scenes hold ~60fps (no truncation)
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but with 30ms stall cadence.
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- **Webcam MJPG missing**, **web capture ~10–14Hz**, **layer SortOrder not persisting**
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— queued, unchanged.
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## Landmines
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@@ -154,8 +128,16 @@ last-3-frames diffs and confirm_frame timestamps next.
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- `cmd.exe /c "taskkill /F /IM ytLive.exe"` (WSL form double-slashes mangle) before rebuilds.
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- Build/tests: Windows dotnet host (`/mnt/c/Program Files/dotnet/dotnet.exe`). 0 warnings.
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- ffmpeg/ffprobe: `/mnt/c/Program Files/Krita (x64)/bin/ffmpeg.exe` with Windows paths.
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- `Audio.SyncOffsetMs` persisted non-zero WILL re-trigger silence symptoms if
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`Configure` is ever called unconditionally again — the idempotent guard is the
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fix, keep it.
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- `MyMistakes.md` has the transparency RESOLVED entry and the new webcam
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collateral recipe — grep before any re-derivation.
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- **`Audio.SyncOffsetMs` currently = 300 in the DB** (contradicts the "set to 0"
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line in 5ba4d70; the write likely never happened). Non-zero offsets still depend on
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the idempotent `Configure` guard to avoid the silence bug.
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- `MyMistakes.md` → Recipes registry now has the **audio/video sync measurement
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recipe** (claps + cross-correlation) — grep before re-deriving.
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## Next step
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**Verify the shipped fix:** re-record with the same clap pattern and run the
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cross-correlation recipe from `MyMistakes.md` → expect offset ≈ 400ms (300ms configured
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sync + ~100ms pipeline) with a `SyncOffsetMs=300` DB value, or ~100ms with the slider
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zeroed. Then commit this session's work (StartLive clear + regression test + docs) and
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decide with the user whether the no-push ruling is lifted.
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Reference in New Issue
Block a user