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Whisper

Free

Transcribe and translate speech in 99 languages.

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Free · Opens the source repo

What Whisper does

Whisper is an advanced speech recognition model developed by OpenAI, designed to handle multilingual audio transcription and translation tasks. It supports transcription in 99 languages, making it a versatile tool for developers and designers who need to convert spoken content into text. With over 680,000 hours of training data, Whisper can accurately transcribe audio from various sources, including podcasts, videos, and meetings, even in noisy environments.

The tool offers a range of model sizes, from 'tiny' for lightweight applications to 'turbo' for high-speed processing, allowing users to choose the best fit based on their hardware capabilities and performance needs. Whisper can auto-detect the language of the audio or allow users to specify it for faster processing. Additionally, it includes features for generating timestamps and handling technical vocabulary through initial prompts, making it suitable for specialized content.

Whisper is particularly useful for automating meeting notes, creating subtitles, and translating audio content into English. The command-line interface and Python API facilitate easy integration into existing workflows, enabling batch processing and real-time transcription for streaming audio. The flexibility of the tool, combined with its robust performance, makes it an essential resource for anyone involved in content creation, translation, or audio processing.

When to use it

Use Whisper when you need to transcribe or translate audio content in multiple languages, especially in noisy environments or for automation tasks like meeting notes.

When not to use it

Avoid Whisper if you require real-time streaming capabilities without additional setup, or if you need features like speaker diarization that are not supported.

What you can build with it

Transcribing Podcasts

Use Whisper to automate the transcription of podcast episodes, making them accessible in text format for your audience.

Meeting Notes Automation

Leverage Whisper to transcribe meeting discussions and generate notes automatically, saving time and improving documentation.

Multilingual Video Subtitles

Generate subtitles for videos in multiple languages using Whisper, enhancing viewer engagement and accessibility.

How to install Whisper

View source

1. Install with the skills CLI

npx skills add nousresearch/hermes-agent/whisper --agent claude-code

2. Or install it manually

Download the skill folder and drop it into ~/.claude/skills/ for all projects, or .claude/skills/ to scope it to one repo. Restart Claude Code so it picks up the new skill.

Anthropic's agentic coding CLI, and the reference implementation of Agent Skills. Drop a skill folder into ~/.claude/skills and Claude Code loads it automatically whenever a task matches the skill's description. Claude Code docs

Inside SKILL.md

Written by nousresearch

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
ModelParametersEnglish-onlyMultilingualSpeedVRAM
tiny39M~32x~1 GB
base74M~16x~1 GB
small244M~6x~2 GB
medium769M~2x~5 GB
large1550M1x~10 GB
turbo809M~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

  1. Use turbo model - Best speed/quality for English
  2. Specify language - Faster than auto-detect
  3. Add initial prompt - Improves technical terms
  4. Use GPU - 10-20× faster
  5. Batch process - More efficient
  6. Convert to WAV - Better compatibility
  7. Split long audio - <30 min chunks
  8. Check language support - Quality varies by language
  9. Use faster-whisper - 4× faster than openai-whisper
  10. Monitor VRAM - Scale model size to hardware

Performance

ModelReal-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

  1. Hallucinations - May repeat or invent text
  2. Long-form accuracy - Degrades on >30 min audio
  3. Speaker identification - No diarization
  4. Accents - Quality varies
  5. Background noise - Can affect accuracy
  6. Real-time latency - Not suitable for live captioning

Resources

Frequently asked questions about Whisper

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