whisper_skill
- Go
934
GitHub Stars
1
Bundled Files
2 months ago
Catalog Refreshed
4 months ago
First Indexed
Readme & install
Copy the install command, review bundled files from the catalogue, and read any extended description pulled from the listing source.
Installation
Preview and clipboard use veilstrat where the catalogue uses aiagentskills.
npx veilstrat add skill trpc-group/trpc-agent-go --skill whisper- SKILL.md1.9 KB
Overview
This skill transcribes audio files to text using OpenAI Whisper models. It supports common audio formats, automatic language detection for 90+ languages, optional timestamps, multiple model sizes, and text or JSON output. It is optimized for practical transcription workflows with speed/accuracy trade-offs across model sizes.
How this skill works
You provide an audio file and target output path; the skill runs a selected Whisper model to produce a transcript. Options let you choose model size (tiny to large), force or auto-detect language, include word- or segment-level timestamps, and export plain text or structured JSON with metadata. FFmpeg is used to preprocess and normalize audio so many file types are accepted.
When to use it
- Transcribing interviews, meetings, podcasts, or lectures into text.
- Generating captions or subtitles with optional timestamps for video content.
- Quickly transcribing short voice notes or long multi-hour recordings depending on model choice.
- Processing multilingual audio where automatic language detection is helpful.
- Converting audio archives into searchable text for indexing or compliance.
Best practices
- Choose model size based on trade-off: tiny/base for speed and low memory, medium/large for best accuracy on noisy or complex audio.
- Preprocess noisy recordings with noise reduction and ensure consistent sampling rate via FFmpeg for better accuracy.
- Specify language when known to improve transcription quality and reduce mis-detections.
- Use timestamps when you need alignment for subtitles or highlight extraction; omit them for faster, smaller outputs.
- Export JSON when downstream tooling needs structured segments, speaker labels, or timing metadata.
Example use cases
- Produce meeting minutes from a recorded team call with medium model and timestamps for quick reference.
- Create subtitles for a multilingual webinar using automatic language detection and JSON export for editing.
- Batch-transcribe a podcast series using the small model for cost/speed balance and plain text output for publishing.
- Index archived customer support calls by transcribing audio into searchable text for QA and training.
- Transcribe voice memos on mobile devices with tiny or base model for near-real-time feedback.
FAQ
Most formats are supported (MP3, WAV, M4A, FLAC, OGG, AAC, WMA) because FFmpeg is used to decode and normalize audio.
Which model should I pick for accuracy vs. speed?
Use tiny/base for fastest, low-memory tasks; small for a balance; medium or large when you need higher accuracy on noisy or complex speech.