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whisper

by @davila7v
4.6(60)

OpenAIのWhisper多言語音声認識モデルを提供し、99言語での音声テキスト変換、ポッドキャスト/ビデオの文字起こし、会議議事録作成に利用します。

OpenAI WhisperSpeech-to-TextASRAudio TranscriptionNatural Language ProcessingGitHub
インストール方法
npx skills add davila7/claude-code-templates --skill whisper
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Before / After 効果比較

1
使用前

従来の音声認識ツールでは、多言語環境や騒がしい環境での転写精度が低く、時間がかかり、大量の手動校正が必要で、コストが高額でした。

使用後

OpenAI Whisperモデルを採用することで、高精度な多言語音声認識を実現し、手動での介入や校正時間を大幅に削減し、コストを削減しました。

description SKILL.md

whisper

Whisper - Robust Speech Recognition

OpenAI's multilingual speech recognition model.

When to use Whisper

Use when:

  • Speech-to-text transcription (99 languages)

  • Podcast/video transcription

  • Meeting notes automation

  • Translation to English

  • Noisy audio transcription

  • Multilingual audio processing

Metrics:

  • 72,900+ GitHub stars

  • 99 languages supported

  • Trained on 680,000 hours of audio

  • MIT License

Use alternatives instead:

  • AssemblyAI: Managed API, speaker diarization

  • Deepgram: Real-time streaming ASR

  • Google Speech-to-Text: Cloud-based

Quick start

Installation

# Requires Python 3.8-3.11
pip install -U openai-whisper

# Requires ffmpeg
# macOS: brew install ffmpeg
# Ubuntu: sudo apt install ffmpeg
# Windows: choco install ffmpeg

Basic transcription

import whisper

# Load model
model = whisper.load_model("base")

# Transcribe
result = model.transcribe("audio.mp3")

# Print text
print(result["text"])

# Access segments
for segment in result["segments"]:
    print(f"[{segment['start']:.2f}s - {segment['end']:.2f}s] {segment['text']}")

Model sizes

# Available models
models = ["tiny", "base", "small", "medium", "large", "turbo"]

# Load specific model
model = whisper.load_model("turbo")  # Fastest, good quality

Model Parameters English-only Multilingual Speed VRAM

tiny 39M ✓ ✓ ~32x ~1 GB

base 74M ✓ ✓ ~16x ~1 GB

small 244M ✓ ✓ ~6x ~2 GB

medium 769M ✓ ✓ ~2x ~5 GB

large 1550M ✗ ✓ 1x ~10 GB

turbo 809M ✗ ✓ ~8x ~6 GB

Recommendation: Use turbo for best speed/quality, base for prototyping

Transcription options

Language specification

# Auto-detect language
result = model.transcribe("audio.mp3")

# Specify language (faster)
result = model.transcribe("audio.mp3", language="en")

# Supported: en, es, fr, de, it, pt, ru, ja, ko, zh, and 89 more

Task selection

# Transcription (default)
result = model.transcribe("audio.mp3", task="transcribe")

# Translation to English
result = model.transcribe("spanish.mp3", task="translate")
# Input: Spanish audio → Output: English text

Initial prompt

# Improve accuracy with context
result = model.transcribe(
    "audio.mp3",
    initial_prompt="This is a technical podcast about machine learning and AI."
)

# Helps with:
# - Technical terms
# - Proper nouns
# - Domain-specific vocabulary

Timestamps

# Word-level timestamps
result = model.transcribe("audio.mp3", word_timestamps=True)

for segment in result["segments"]:
    for word in segment["words"]:
        print(f"{word['word']} ({word['start']:.2f}s - {word['end']:.2f}s)")

Temperature fallback

# Retry with different temperatures if confidence low
result = model.transcribe(
    "audio.mp3",
    temperature=(0.0, 0.2, 0.4, 0.6, 0.8, 1.0)
)

Command line usage

# Basic transcription
whisper audio.mp3

# Specify model
whisper audio.mp3 --model turbo

# Output formats
whisper audio.mp3 --output_format txt     # Plain text
whisper audio.mp3 --output_format srt     # Subtitles
whisper audio.mp3 --output_format vtt     # WebVTT
whisper audio.mp3 --output_format json    # JSON with timestamps

# Language
whisper audio.mp3 --language Spanish

# Translation
whisper spanish.mp3 --task translate

Batch processing

import os

audio_files = ["file1.mp3", "file2.mp3", "file3.mp3"]

for audio_file in audio_files:
    print(f"Transcribing {audio_file}...")
    result = model.transcribe(audio_file)

    # Save to file
    output_file = audio_file.replace(".mp3", ".txt")
    with open(output_file, "w") as f:
        f.write(result["text"])

Real-time transcription

# For streaming audio, use faster-whisper
# pip install faster-whisper

from faster_whisper import WhisperModel

model = WhisperModel("base", device="cuda", compute_type="float16")

# Transcribe with streaming
segments, info = model.transcribe("audio.mp3", beam_size=5)

for segment in segments:
    print(f"[{segment.start:.2f}s -> {segment.end:.2f}s] {segment.text}")

GPU acceleration

import whisper

# Automatically uses GPU if available
model = whisper.load_model("turbo")

# Force CPU
model = whisper.load_model("turbo", device="cpu")

# Force GPU
model = whisper.load_model("turbo", device="cuda")

# 10-20× faster on GPU

Integration with other tools

Subtitle generation

# Generate SRT subtitles
whisper video.mp4 --output_format srt --language English

# Output: video.srt

With LangChain

from langchain.document_loaders import WhisperTranscriptionLoader

loader = WhisperTranscriptionLoader(file_path="audio.mp3")
docs = loader.load()

# Use transcription in RAG
from langchain_chroma import Chroma
from langchain_openai import OpenAIEmbeddings

vectorstore = Chroma.from_documents(docs, OpenAIEmbeddings())

Extract audio from video

# Use ffmpeg to extract audio
ffmpeg -i video.mp4 -vn -acodec pcm_s16le audio.wav

# Then transcribe
whisper audio.wav

Best practices

  • Use turbo model - Best speed/quality for English

  • Specify language - Faster than auto-detect

  • Add initial prompt - Improves technical terms

  • Use GPU - 10-20× faster

  • Batch process - More efficient

  • Convert to WAV - Better compatibility

  • Split long audio - <30 min chunks

  • Check language support - Quality varies by language

  • Use faster-whisper - 4× faster than openai-whisper

  • Monitor VRAM - Scale model size to hardware

Performance

Model Real-time factor (CPU) Real-time factor (GPU)

tiny ~0.32 ~0.01

base ~0.16 ~0.01

turbo ~0.08 ~0.01

large ~1.0 ~0.05

Real-time factor: 0.1 = 10× faster than real-time

Language support

Top-supported languages:

  • English (en)

  • Spanish (es)

  • French (fr)

  • German (de)

  • Italian (it)

  • Portuguese (pt)

  • Russian (ru)

  • Japanese (ja)

  • Korean (ko)

  • Chinese (zh)

Full list: 99 languages total

Limitations

  • Hallucinations - May repeat or invent text

  • Long-form accuracy - Degrades on >30 min audio

  • Speaker identification - No diarization

  • Accents - Quality varies

  • Background noise - Can affect accuracy

  • Real-time latency - Not suitable for live captioning

Resources

Weekly Installs559Repositorydavila7/claude-…emplatesGitHub Stars23.1KFirst SeenJan 21, 2026Security AuditsGen Agent Trust HubPassSocketPassSnykPassInstalled onopencode490gemini-cli476codex455cursor441github-copilot434amp387

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統計データ

インストール数1.5K
評価4.6 / 5.0
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更新日2026年3月17日
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対応プラットフォーム

🔧Claude Code
🔧OpenClaw
🔧OpenCode
🔧Codex
🔧Gemini CLI
🔧GitHub Copilot
🔧Amp
🔧Kimi CLI

タイムライン

作成2026年3月17日
最終更新2026年3月17日