Automatic Speech Recognition (ASR)
Agent skill frombenchflow-ai/skillsbench
Transcribe audio segments to text using Whisper models. Use larger models (small, base, medium, large-v3) for better accuracy, or faster-whisper for optimized performance. Always align transcription timestamps with diarization segments for accurate speaker-labeled subtitles.
How to install Automatic Speech Recognition (ASR)
install Automatic Speech Recognition (ASR)
# install the skillsbench pack with the skills CLI
npx skills add benchflow-ai/skillsbench# or add only this skill to Claude Code: copy the folder
tasks-extra/speaker-diarization-subtitles/environment/skills/automatic-speech-recognition/
# into
~/.claude/skills/automatic-speech-recognition/ (all projects)
.claude/skills/automatic-speech-recognition/ (one project)
tasks-extra/speaker-diarization-subtitles/environment/skills/automatic-speech-recognition/SKILL.md
Apache-2.0Full file1# Automatic Speech Recognition (ASR)23## Overview45After speaker diarization, you need to transcribe each speech segment to text. Whisper is the current state-of-the-art for ASR, with multiple model sizes offering different trade-offs between accuracy and speed.67## When to Use89- After speaker diarization is complete10- Need to generate speaker-labeled transcripts11- Creating subtitles from audio segments12- Converting speech segments to text1314## Whisper Model Selection1516### Model Size Comparison1718| Model | Size | Speed | Accuracy | Best For |19|-------|------|-------|----------|----------|20| tiny | 39M | Fastest | Lowest | Quick testing, low accuracy needs |21| base | 74M | Fast | Low | Fast processing with moderate accuracy |22| small | 244M | Medium | Good | **Recommended balance** |23| medium | 769M | Slow | Very Good | High accuracy needs |24| large-v3 | 1550M | Slowest | **Best** | **Maximum accuracy** |2526### Recommended: Use `small` or `large-v3`2728**For best accuracy (recommended for this task):**29```python30import whisper3132model = whisper.load_model("large-v3") # Best accuracy33result = model.transcribe(audio_path)34```3536**For balanced performance:**37```python38import whisper3940model = whisper.load_model("small") # Good balance41result = model.transcribe(audio_path)42```4344## Faster-Whisper (Optimized Alternative)4546For faster processing with similar accuracy, use `faster-whisper`:4748```python49from faster_whisper import WhisperModel5051# Use small model with CPU int8 quantization52…
Excerpt shown under the repository's Apache-2.0 license.
