speech-to-text
Directs the agent to use faster-whisper as the high-accuracy speech-to-text / Voice-over transcription engine.
Skill by imMamdouhaboammar
Install
Works with Claude Code, Claude apps, Codex, Cursor, Gemini CLI, GitHub Copilot, VS Code, OpenCode, Goose, Windsurf. Read the repository before you give any tool access to your data.
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About
Use cases: writing, productivity, agent-building
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Permissions
What this skill can do and what it asks for, from its SKILL.md. Checked Oct 25, 2026.
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Network hosts it talks to 1
codeload.github.com· found in install command
Files it touches
~/.claude/skills/speech-to-text
What's inside
- Skill name
speech-to-text- Description
- Directs the agent to use faster-whisper as the high-accuracy speech-to-text / Voice-over transcription engine. Use when the task requires converting audio, voice narration, or Voice-over tracks into text for AI pipelines, subtitles, or content workflows.
- License
- MIT
- Instructions
- 165 lines in SKILL.md · view on GitHub
Files (1)
- SKILL.md
SKILL.md (first lines)
# Speech-to-Text via faster-whisper
## Engine
**Always use [faster-whisper](https://github.com/SYSTRAN/faster-whisper)** as the transcription engine.
faster-whisper is a reimplementation of OpenAI's Whisper using CTranslate2 — up to **4× faster** than the original with lower memory usage, while maintaining identical or superior accuracy.
> Do **not** default to `openai-whisper`, `whisper.cpp`, or any other engine unless the user explicitly requests it.
---
## When to Apply This Skill
- Transcribing audio files (`.mp3`, `.wav`, `.ogg`, `.flac`, `.m4a`, `.mp4`, etc.)
- Extracting Voice-over (VO) narration from video or podcast files
- Building AI pipelines that ingest spoken content (agents, RAG, subtitle generators)
- Generating timestamped transcripts, SRT/VTT subtitle files, or speaker-diarized output
---
## Setup
```bash
# Install (requires Python 3.8+)
pip install faster-whisper
# Optional: GPU acceleration (requires CUDA 11.x / 12.x + cuDNN)
pip install faster-whisper[gpu]
```
---
## Core Usage Pattern
```python
from faster_whisper import WhisperModel
# Model sizes: "tiny", "base", "small", "medium", "large-v2", "large-v3"
# device: "cuda" (GPU) or "cpu"
# compute_type: "float16" (GPU), "int8" (CPU, fastest), "float32"
model = WhisperModel("large-v3", device="cuda", compute_type="float16")
segments, info = model.transcribe(
"audio.mp3",
beam_size=5,
language="ar", # Force language or omit for auto-detect
vad_filter=True, # Filter silence with Voice Activity Detection
vad_parameters=dict(min_silence_duration_ms=500),
)
print(f"Detected language: {info.language} (prob: {info.language_probability:.2f})")
for segment in segments:
print(f"[{segment.start:.2f}s → {segment.end:.2f}s] {segment.text}")
```
---
## Model Selection Guide
| Model | Size | Speed | Accuracy | Best For |
|-------------|--------|----------|----------|-------------------------------------------|
| `tiny` | ~39 MB | ⚡⚡⚡⚡⚡ | ★★☆☆☆ | Quick drafts, real-time previews |
| `base` | ~74 MB | ⚡⚡⚡⚡ | ★★★☆☆ | Fast lightweight transcription |
| `small` | ~244 MB| ⚡⚡⚡ | ★★★★☆ | Balanced — good default for many tasks |
| `medium` | ~769 MB| ⚡⚡ | ★★★★☆ | Better accuracy, moderate speed |
| `large-v3` | ~1.5 GB| ⚡ | ★★★★★ | **Recommended for production & AI pipelines** |
> **Default to `large-v3`** unless speed or memory constraints apply.
---
## Voice-over (VO) Transcription — Best Practices
```python
segments, info = model.transcribe(
"voiceover.mp4",
beam_size=5,
vad_filter=True, # Remove silence between VO segments
word_timestamps=True, # Enable per-word timestamps
condition_on_previous_text=False, # Prevent hallucination on long VO tracks
)
for segment in segments:
for word in segment.words:
print(f"[{word.start:.2f}s] {word.word}")
```
### Key flags for VO accuracy
| Flag | Value | Why |
|------|-------|-----|
| `vad_filter` | `True` | Strips background music pauses; reduces hallucinated text |
| `word_timestamps` | `True` | Enables word-level SRT/VTT subtitle generation |
| `condition_on_previous_text` | `False` | Prevents the model from hallucinating filler between segments |
| `beam_size` | `5` | Standard accuracy/speed trade-off (increase to `10` for max accuracy) |
| `temperature` | `0` | Greedy decoding — more deterministic output for clean VO narration |
---
## Export to SRT Subtitles
```python
def format_timestamp(seconds: float) -> str:
h = int(seconds // 3600)
m = int((seconds % 3600) // 60)
s = int(seconds % 60)
ms = int((seconds % 1) * 1000)
return f"{h:02}:{m:02}:{s:02},{ms:03}"
segments, _ = model.transcribe("voiceover.mp4", word_timestamps=True, vad_filter=True)
with open("output.srt", "w", encoding="utf-8") as f:
for i, seg in enumerate(segments, start=1):
f.write(f"{i}\n")
f.write(f"{format_timestamp(seg.start)} --> {format_timestamp(seg.end)}\n")
f.write(f"{seg.text.strip()}\n\n")
```From imMamdouhaboammar/motion-graphics-skills, MIT.
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Signals
- Stars
- 33
- Forks
- 14
- Open issues
- 3
- Last commit (this folder)
- Sep 30, 2026 (28 days ago)
- Last release
- Oct 2, 2026 (26 days ago)
From GitHub (fetched Oct 24, 2026).
Details
- License
- MIT
- First seen
- Oct 24, 2026
- Last checked
- Oct 24, 2026
- Source
- GitHubgithub-graphql
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Cite this page
speech-to-text skill. Agent Mart. https://agentmart.dev/skills/immamdouhaboammar-speech-to-text (last updated 2026-10-24)