@hanzo/frame is taken by the hero video primitive — one full-bleed video element — which has held that name on npm since it shipped. This repo is the video toolchain: a CLI, a renderer, a studio, a player. Plural is what it is, so the CLI publishes as @hanzo/frames and the workspace packages follow it (@hanzo/frames-core, @hanzo/frames-engine, and the rest). The binary stays `frames`, and now actually ships. The manifest mapped it to bin/frames.mjs while the file on disk was bin/hyperframes.mjs, so a global install produced no working command. The paths that name a package to fetch were pointing elsewhere. The update check read registry.npmjs.org/frames, a different author's package, and the npm, bun and pnpm installers would have pulled it over the CLI; the render image installed the upstream hyperframes; scaffolded projects and the docs pinned the unscoped name. All of them now name @hanzo/frames.
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Transcript Guide
For the transcribe CLI invocation, the .en-translates-non-English rule, and whisper model selection, see ../transcribe.md. This file covers what to do with the resulting transcript when authoring captions: input formats, mandatory quality checks, cleaning code, external-API fallbacks.
Supported Input Formats
The CLI auto-detects and normalizes these formats:
| Format | Extension | Source | Word-level? |
|---|---|---|---|
| whisper.cpp JSON | .json |
frames init --video, frames transcribe |
Yes |
| OpenAI Whisper API | .json |
openai.audio.transcriptions.create({ timestamp_granularities: ["word"] }) |
Yes |
| SRT subtitles | .srt |
Video editors, subtitle tools, YouTube | No (phrase-level) |
| VTT subtitles | .vtt |
Web players, YouTube, transcription services | No (phrase-level) |
| Normalized word array | .json |
Pre-processed by any tool | Yes |
Word-level timestamps produce better captions. SRT/VTT give phrase-level timing, which works but can't do per-word animation effects.
Transcript Quality Check (Mandatory)
After every transcription, read the transcript and check for quality issues before proceeding. Bad transcripts produce nonsensical captions. Never skip this step.
What to look for
| Signal | Example | Cause |
|---|---|---|
Music note tokens (♪, �) |
{ "text": "♪" } or { "text": "�" } |
Whisper detected music, not speech |
| Garbled / nonsense words | "Do a chin", "Get so gay", "huh" | Model misheard lyrics or background noise |
| Long gaps with no words | 20+ seconds of only ♪ tokens |
Instrumental section — expected, but high ratio means speech is being missed |
| Repeated filler | Many "huh", "uh", "oh" entries | Model is hallucinating on music |
| Very short word spans | Words with end - start < 0.05 |
Unreliable timestamp alignment |
Automatic retry rules
If more than 20% of entries are ♪/� tokens, or the transcript contains obvious nonsense words, the transcription failed. Do not proceed with the bad transcript. Instead:
- Retry with
medium.enif the original usedsmall.enor smaller:npx @hanzo/frames transcribe audio.mp3 --model medium.en - If
medium.enalso fails (still >20% music tokens or garbled), tell the user the audio is too noisy for local transcription and suggest:- Providing lyrics manually as an SRT/VTT file
- Using an external API (OpenAI or Groq Whisper — see below)
- Always clean the transcript before building captions — filter out
♪/�tokens and entries wheretextis a single non-word character. Only real words should reach the caption composition.
Cleaning a transcript
After transcription (even with a good model), strip non-word entries:
var raw = JSON.parse(transcriptJson);
var words = raw.filter(function (w) {
if (!w.text || w.text.trim().length === 0) return false;
if (/^[♪�\u266a\u266b\u266c\u266d\u266e\u266f]+$/.test(w.text)) return false;
if (/^(huh|uh|um|ah|oh)$/i.test(w.text) && w.end - w.start < 0.1) return false;
return true;
});
For model-selection guidance by content type, see ../transcribe.md → "Picking a model by content type".
Using External Transcription APIs
For the best accuracy, use an external API and import the result:
OpenAI Whisper API (recommended for quality):
# Generate with word timestamps, then import
curl https://api.openai.com/v1/audio/transcriptions \
-H "Authorization: Bearer $OPENAI_API_KEY" \
-F file=@audio.mp3 -F model=whisper-1 \
-F response_format=verbose_json \
-F "timestamp_granularities[]=word" \
-o transcript-openai.json
npx @hanzo/frames transcribe transcript-openai.json
Groq Whisper API (fast, free tier available):
curl https://api.groq.com/openai/v1/audio/transcriptions \
-H "Authorization: Bearer $GROQ_API_KEY" \
-F file=@audio.mp3 -F model=whisper-large-v3 \
-F response_format=verbose_json \
-F "timestamp_granularities[]=word" \
-o transcript-groq.json
npx @hanzo/frames transcribe transcript-groq.json
If No Transcript Exists
- Check the project root for
transcript.json,.srt, or.vttfiles. - If none found, run
../transcribe.md— pick the starting model from "Picking a model by content type" there. - Run the quality check above. If it fails, retry with a larger model or fall back to manual lyrics / external API.