Two bugs in getSharedBrowser() could take down the entire Vite dev
server:
1. Unhandled rejection from puppeteer.launch() — the timeout error
surfaces through puppeteer's internal RxJS chain, and any uncaught
path crashes the Node process. The thumbnail route's try/catch
doesn't always intercept it.
2. _browserLaunchPromise was never reset on failure, so subsequent
thumbnail requests reused a stale rejected promise instead of
retrying.
Wrap the IIFE in try/catch, return null on any failure (the thumbnail
route already handles a null adapter result with a 500), and reset
_browserLaunchPromise in a finally block so a transient launch failure
doesn't poison the singleton. Also drop the launch timeout from
puppeteer's 30s default to 10s so a wedged handshake fails fast instead
of stalling every pending thumbnail.
Verified locally: the dev server now logs
"[Studio] puppeteer launch failed — thumbnails disabled: ..." and
keeps serving the studio UI after a thumbnail request fails.
Orientation is a property of the composition, not a user choice — the
backend's portrait/landscape presets are tied to the comp's authored
aspect ratio. Letting users pick "1080p portrait" for a landscape
composition just produces a wrong-aspect render.
The dropdown now exposes three scale choices (Auto / 1080p / 4K) and
maps to the correct portrait/landscape preset based on the active
composition's data-width / data-height. Native <select title> tooltips
are unreliable across browsers, so the resolved dimensions render
inline in each option label (e.g. "1080p · 1920×1080") — always
visible, no hover needed.
App.tsx tracks the active comp's dimensions by listening for the
existing hf-preview state/timeline postMessages (same source the
caption-detection logic uses) and passes them to RenderQueue. The
useRenderQueue / backend contract is unchanged: RenderQueue still emits
"landscape" | "portrait" | "landscape-4k" | "portrait-4k" | "auto".
- Hoist duplicated test mock helpers (createMockAudioContext / setupTransport /
mockBuffer / mockEl) from the two describe blocks to module scope.
- Drop redundant math-derivation comments in schedulePlayback; the dedicated
rate-aware tests are the canonical proof.
- Tighten setRate JSDoc.
- Add no-op guard in setRate when the new rate equals the current rate, so a
duplicate set-playback-rate postMessage doesn't re-anchor or walk active
sources for nothing.
- Add a regression test for the no-op guard, and strengthen the clamp test
to schedule at rate=2 first so the clamp-to-1 assertion is non-vacuous.
WebAudioTransport scheduled AudioBufferSourceNodes with the implicit
default playbackRate of 1, so non-1x transport rates desynced visuals
from audio: GSAP timelines, the transport clock, and native <video>
all sped up while WebAudio-routed <audio> clips kept playing at 1x.
- schedulePlayback now accepts a rate, sets sourceNode.playbackRate,
and scales the future-clip start delay by the rate (the in-progress
buffer offset stays elapsed + mediaStart, which is rate-independent).
- New setRate() updates active sources in place and rebases the
getTime() reference frame so the audio-master clock stays continuous
across mid-playback rate changes.
- Runtime onSetPlaybackRate now forwards into webAudio.setRate, and
player.play() schedules each clip with state.playbackRate.
Fixes#713
The property delegation on window.__player used Object.defineProperty
with only a getter, causing "Cannot set property renderSeek which has
only a getter" when Studio's motion-wrapping code tried to reassign
__player.renderSeek with a wrapped version. This cascaded into an
infinite error loop making the timeline unusable.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
The postMessage state path set `_currentTime` without clamping, while the
direct timeline path already used `Math.min(currentTime, _duration)`. A
final-frame state message with a frame count slightly past the end would
set `_currentTime > _duration`, causing the progress bar (position:
absolute, no overflow guard) to bleed out of the scrubber track and
visually cover the volume button, and the time display to show values
like "0:05 / 0:04".
- Clamp `_currentTime` in `_onMessage` to match the direct timeline path
- Clamp defensively in `updateTime` so the display layer never overflows
- Add `overflow: hidden` + `min-width: 0` to `.hfp-scrubber` as a CSS
safety net; remove now-redundant `border-radius` from `.hfp-progress`
(parent `overflow: hidden` handles clipping to the rounded shape)
- Apply the same `overflow: hidden` fix to `.hfp-volume-slider` for
consistency; remove redundant `border-radius` from `.hfp-volume-fill`
- Add regression test covering the postMessage over-duration case
Co-authored-by: Claude Sonnet 4.6 <noreply@anthropic.com>
Add mintlify docs page for contributing blocks/components to the
registry catalog. Rename skill from contribute to contribute-catalog
for clearer intent.
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
- Narrow description to disambiguate from hyperframes-registry (install)
and hyperframes (in-project authoring) skills
- Add "adjust" comments on dimensions/duration defaults so agents don't
blindly copy 1920x1080/10s for all composition shapes
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Adds a new docs page under Getting Started that links to the
heygen-com/hyperframes-launches repo — open-source HyperFrames
compositions behind HeyGen's product launch videos. Includes a brief
catalog of the 5 projects currently in there, framing on why these are
useful (multi-composition shape, real adapter mix, production-grade
timing), and the LFS-aware clone recipe.
Cross-linked from `docs/examples.mdx`'s Next Steps and from
`docs/community/adopters.mdx`.
Came from a Discord ask via blackNoir (forwarded by James) — users
landing on the docs want to see how the internal team builds their own
videos through HyperFrames; this surfaces that source in one click.
The agent skill loader rejects SKILL.md files whose frontmatter description
exceeds 1024 characters, so remotion-to-hyperframes was being skipped at
startup with a "exceeds maximum length of 1024 characters" warning.
Trimmed the description from 1240 to 896 characters by collapsing the
trigger-phrase examples and tightening prose, while preserving every
trigger / no-trigger guardrail. Moved the detailed list of trigger phrases
and the 4 negative cases into a new "## When to use" section in the body
so the guidance is not lost.
Fixes#688
New skill that guides users through the full workflow of contributing
caption styles, VFX blocks, transitions, and other components to the
HyperFrames registry.
Covers: scaffolding, proven patterns (seeded PRNG, paused timelines,
hard kills, unique ID prefixes), validation, rendering, and PR prep.
Includes copy-paste HTML templates for caption and Three.js components
with all the non-negotiable rules baked in.
Tested by an unbiased agent with zero prior HyperFrames knowledge —
built a working cap-typewriter component that passed lint and validate
on the first attempt. Feedback incorporated: per-character animation
guidance, monospace font size exceptions, positioning variants, and
PREFIX naming convention table.
Replaces the rigid `--fps 24|30|60` whitelist with a numeric range and
adds support for ffmpeg-style fractional framerates so NTSC stays exact
end-to-end.
- `--fps 30` keeps working (integer fps)
- `--fps 30000/1001` now means exact NTSC 29.97 (not the lossy decimal)
- `--fps 24000/1001`, `--fps 60000/1001`, `--fps 25/50/120/240` all work
- Decimals like `--fps 29.97` are rejected with a friendly error pointing
the user at the rational form, since `29.97` and `30000/1001` round
to different framerates inside ffmpeg
Carries an `Fps = { num: number; den: number }` rational end-to-end:
RenderConfig, EncoderOptions, StreamingEncoderOptions, CaptureOptions,
DockerRenderOptions, Studio API request body, regression-harness
meta.json. The `-r` and `-framerate` ffmpeg args emit the rational form
verbatim (`30000/1001`) so no decimal round-trip happens at the encoder
boundary. Frame-interval math uses `1000 * den / num` ms (33.366… for
NTSC, 33.333… for integer 30).
Helpers live in @hyperframes/core:
- `parseFps(input: string | number): FpsParseResult` — discriminated
parser used by both the CLI and the Studio API route
- `fpsToFfmpegArg(fps: Fps): string` — emits "30" or "30000/1001"
- `fpsToNumber(fps: Fps): number` — for arithmetic (telemetry, frame
count, frame-index → time)
Studio API wire format accepts polymorphic `fps: number | string`:
- number → integer fps (`30`)
- string → rational (`"30000/1001"`)
Decimals are rejected; matches the same rule as the CLI.
Existing meta.json fixtures with integer `"fps": 30` continue to load
unchanged — the regression-harness validator now normalizes both number
and string inputs through `parseFps`.
GitHub strips autoplay/loop from <video> tags in markdown, so the MP4
required a click to play. WebP autoplays via <img> while preserving full
HD source quality (1280x720 vs the previous GIF's 400x225).
Co-Authored-By: Claude Opus 4.7 (1M context) <noreply@anthropic.com>
Clips whose compositionStart is ahead of the current timeline position
were starting immediately because sourceNode.start() always received
when=0. Use the AudioContext scheduling API to defer future clips:
sourceNode.start(ctx.currentTime + delay, mediaStart).
Closes#674
Co-Authored-By: Claude Opus 4.6 (1M context) <noreply@anthropic.com>
Replace the two-clock architecture (GSAP rAF ticker + HTMLMediaElement
pipeline reconciled by a 50ms polling loop) with a single TransportClock.
GSAP is always paused and seeked to clock.now() on each rAF tick.
Drift between visual timeline and audio is structurally impossible.
Architecture:
TransportClock.now() ──rAF──▶ timeline.seek(t) + el.currentTime
▲
AudioContext.currentTime (~21µs) ← WebAudio active
OR
audio.currentTime (~33ms) ← HTMLMediaElement fallback
OR
performance.now() (~1ms) ← no audio
Key changes:
- TransportClock class with monotonic + audio-master clock sources
- WebAudioTransport: routes audio through AudioBufferSourceNode for
sample-accurate scheduling, falls back gracefully to HTMLMediaElement
- rAF tick loop replaces 50ms setInterval poll; GSAP always paused
- Strict sync (40ms threshold, consecutive-sample gated) + forceSync
on play/pause/seek transitions for sub-frame media accuracy
- Buffer-stall: visuals freeze when audio is buffering instead of
running ahead
- Frame quantization preserved in seek/renderSeek (parity contract)
Browser-verified: 0.0ms drift after 40 pause/play cycles (was 400ms+).
Also fixes: CDN script HTML error responses in validate (pre-existing).
54 tests across clock, clock-drift, webAudioTransport, and media.
Closes#668
James