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SKILL.md

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 file
1# Automatic Speech Recognition (ASR)
2
3## Overview
4
5After 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.
6
7## When to Use
8
9- After speaker diarization is complete
10- Need to generate speaker-labeled transcripts
11- Creating subtitles from audio segments
12- Converting speech segments to text
13
14## Whisper Model Selection
15
16### Model Size Comparison
17
18| 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** |
25
26### Recommended: Use `small` or `large-v3`
27
28**For best accuracy (recommended for this task):**
29```python
30import whisper
31
32model = whisper.load_model("large-v3") # Best accuracy
33result = model.transcribe(audio_path)
34```
35
36**For balanced performance:**
37```python
38import whisper
39
40model = whisper.load_model("small") # Good balance
41result = model.transcribe(audio_path)
42```
43
44## Faster-Whisper (Optimized Alternative)
45
46For faster processing with similar accuracy, use `faster-whisper`:
47
48```python
49from faster_whisper import WhisperModel
50
51# Use small model with CPU int8 quantization
52…

Excerpt shown under the repository's Apache-2.0 license.

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