Files
hyperframes/skills/hyperframes/references/transcript-guide.md
T
James 4413d25b0e refactor(skills): consolidate tts/whisper guidance into hyperframes-media
Code review found the new hyperframes-media skill was parallel
content with skills/hyperframes/references/tts.md and the "Whisper
Model Guide" section of transcript-guide.md — same voice table, same
.en-translates-non-English warning, same TTS→transcribe chain in
both places. Plus some scope creep in hyperframes-media (audio/video
HTML snippets that duplicate the canonical track docs in
hyperframes/SKILL.md:265+).

Consolidation:

- hyperframes-media is now the single source of truth for CLI
  invocation, voice selection, multilingual phonemization, whisper
  model selection, and the .en gotcha. Picked up the multilingual
  prefix decoding from the deleted tts.md.
- skills/hyperframes/references/tts.md deleted; the bullet in
  hyperframes/SKILL.md is removed (no replacement — agents land on
  hyperframes-media via its own description).
- skills/hyperframes/references/transcript-guide.md keeps only the
  caption-side concerns: input-format table, mandatory quality
  check, cleaning JS, external-API import path, and the
  "if no transcript exists" flow. The intro bash recipe and Whisper
  Model Guide section both moved to hyperframes-media. Top of the
  file now points to hyperframes-media for CLI/model details.

Other tightening in hyperframes-media:

- Dropped WHAT-narration filler and the inline <audio>/<video> HTML
  snippets — they duplicate the canonical track-attribute docs in
  hyperframes/SKILL.md.
- Added the `id` field (`w0`, `w1`, ...) to the transcript output
  shape — the actual Word interface in
  packages/cli/src/whisper/normalize.ts includes it (optional for
  backwards compat), used by caption override logic.
- Compressed the TTS → Transcribe → Captions chain section.

Net: hyperframes-media 147 → 136 lines, transcript-guide.md 152 →
106 lines, tts.md gone (-75 lines).
2026-05-04 22:06:02 +00:00

5.7 KiB

Transcript Guide

For the transcribe CLI invocation, the .en-translates-non-English rule, and whisper model selection, see the hyperframes-media skill. 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 hyperframes init --video, hyperframes 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:

  1. Retry with medium.en if the original used small.en or smaller:
    npx hyperframes transcribe audio.mp3 --model medium.en
    
  2. If medium.en also 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)
  3. Always clean the transcript before building captions — filter out / tokens and entries where text is 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;
});

When to use which model (decision tree)

  1. Is this speech over silence/light background?small.en is fine
  2. Is this speech over music, or music with vocals? → Start with medium.en
  3. Is this a produced music track (vocals + full instrumentation)? → Start with medium.en, expect to need manual lyrics or an external API
  4. Is this multilingual? → Use medium or large-v3 (no .en suffix)

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 hyperframes 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 hyperframes transcribe transcript-groq.json

If No Transcript Exists

  1. Check the project root for transcript.json, .srt, or .vtt files
  2. If none found, run transcription — pick the starting model based on the content type:
    • Speech/voiceover → small.en
    • Music with vocals → medium.en
    npx hyperframes transcribe <audio-or-video-file> --model medium.en
    
  3. Read the transcript and run the quality check (see above). If it fails, retry with a larger model or suggest manual lyrics.