Whisper

orchestra-research/ai-research-skills/18-multimodal/whisper

作者 orchestra-research773a52944ba4MIT13K 個星標收錄於 2026年10月8日更新於 2026年10月8日儲存庫3 個月前更新

OpenAI's general-purpose speech recognition model. Supports 99 languages, transcription, translation to English, and language identification. Six model sizes from tiny (39M params) to large (1550M params). Use for speech-to-text, podcast transcription, or multilingual audio processing. Best for robust, multilingual ASR.

僅含說明AI & Agents

AI 產生的概覽

指導使用 OpenAI 的 Whisper 語音辨識模型進行轉錄、翻譯成英文與語言辨識。

功能: 這個技能說明如何執行 OpenAI 的 Whisper 語音辨識模型，用於語音轉文字、翻譯成英文，以及辨識 99 種語言。內容涵蓋模型大小、轉錄選項（例如指定語言、初始提示、字詞層級時間戳與溫度回退），以及命令列用法和 txt、srt、vtt、json 等輸出格式。也介紹批次處理、GPU 加速、字幕產生，以及與 LangChain 等工具的整合。
適用情境: 適用於語音轉文字、Podcast 或影片轉錄、會議記錄自動化、將音訊翻譯成英文，以及多語言與嘈雜音訊處理。適合需要穩健多語言 ASR，而非即時串流轉錄或講者分離的情境。
執行需求: 需要 Python 3.8-3.11、openai-whisper 套件與 ffmpeg；文件也提到 transformers 和 torch 作為相依套件，以及用於串流處理的 faster-whisper。GPU 加速為選用。此技能不附帶指令碼，只有說明文件與一份語言參考檔案。

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

bash

# Requires Python 3.8-3.11pip install -U openai-whisper
# Requires ffmpeg# macOS: brew install ffmpeg# Ubuntu: sudo apt install ffmpeg# Windows: choco install ffmpeg

# Requires Python 3.8-3.11pip install -U openai-whisper
# Requires ffmpeg# macOS: brew install ffmpeg# Ubuntu: sudo apt install ffmpeg# Windows: choco install ffmpeg

Basic transcription

python

import whisper
# Load modelmodel = whisper.load_model("base")
# Transcriberesult = model.transcribe("audio.mp3")
# Print textprint(result["text"])
# Access segmentsfor segment in result["segments"]:    print(f"[{segment['start']:.2f}s - {segment['end']:.2f}s] {segment['text']}")

import whisper
# Load modelmodel = whisper.load_model("base")
# Transcriberesult = model.transcribe("audio.mp3")
# Print textprint(result["text"])
# Access segmentsfor segment in result["segments"]:    print(f"[{segment['start']:.2f}s - {segment['end']:.2f}s] {segment['text']}")

Model sizes

python

# Available modelsmodels = ["tiny", "base", "small", "medium", "large", "turbo"]
# Load specific modelmodel = whisper.load_model("turbo")  # Fastest, good quality

# Available modelsmodels = ["tiny", "base", "small", "medium", "large", "turbo"]
# Load specific modelmodel = whisper.load_model("turbo")  # Fastest, good quality

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

Transcription options

Language specification

python

# Auto-detect languageresult = 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

# Auto-detect languageresult = 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

python

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

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

Initial prompt

python

# Improve accuracy with contextresult = 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

# Improve accuracy with contextresult = 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

python

# Word-level timestampsresult = 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)")

# Word-level timestampsresult = 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

python

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

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

Command line usage

bash

# Basic transcriptionwhisper audio.mp3
# Specify modelwhisper audio.mp3 --model turbo
# Output formatswhisper audio.mp3 --output_format txt     # Plain textwhisper audio.mp3 --output_format srt     # Subtitleswhisper audio.mp3 --output_format vtt     # WebVTTwhisper audio.mp3 --output_format json    # JSON with timestamps
# Languagewhisper audio.mp3 --language Spanish
# Translationwhisper spanish.mp3 --task translate

# Basic transcriptionwhisper audio.mp3
# Specify modelwhisper audio.mp3 --model turbo
# Output formatswhisper audio.mp3 --output_format txt     # Plain textwhisper audio.mp3 --output_format srt     # Subtitleswhisper audio.mp3 --output_format vtt     # WebVTTwhisper audio.mp3 --output_format json    # JSON with timestamps
# Languagewhisper audio.mp3 --language Spanish
# Translationwhisper spanish.mp3 --task translate

Batch processing

python

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"])

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

python

# 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 streamingsegments, info = model.transcribe("audio.mp3", beam_size=5)
for segment in segments:    print(f"[{segment.start:.2f}s -> {segment.end:.2f}s] {segment.text}")

# 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 streamingsegments, 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

python

import whisper
# Automatically uses GPU if availablemodel = whisper.load_model("turbo")
# Force CPUmodel = whisper.load_model("turbo", device="cpu")
# Force GPUmodel = whisper.load_model("turbo", device="cuda")
# 10-20× faster on GPU

import whisper
# Automatically uses GPU if availablemodel = whisper.load_model("turbo")
# Force CPUmodel = whisper.load_model("turbo", device="cpu")
# Force GPUmodel = whisper.load_model("turbo", device="cuda")
# 10-20× faster on GPU

Integration with other tools

Subtitle generation

bash

# Generate SRT subtitleswhisper video.mp4 --output_format srt --language English
# Output: video.srt

# Generate SRT subtitleswhisper video.mp4 --output_format srt --language English
# Output: video.srt

With LangChain

python

from langchain.document_loaders import WhisperTranscriptionLoader
loader = WhisperTranscriptionLoader(file_path="audio.mp3")docs = loader.load()
# Use transcription in RAGfrom langchain_chroma import Chromafrom langchain_openai import OpenAIEmbeddings
vectorstore = Chroma.from_documents(docs, OpenAIEmbeddings())

from langchain.document_loaders import WhisperTranscriptionLoader
loader = WhisperTranscriptionLoader(file_path="audio.mp3")docs = loader.load()
# Use transcription in RAGfrom langchain_chroma import Chromafrom langchain_openai import OpenAIEmbeddings
vectorstore = Chroma.from_documents(docs, OpenAIEmbeddings())

Extract audio from video

bash

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

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

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

GitHub: https://github.com/openai/whisper ⭐ 72,900+
Paper: https://arxiv.org/abs/2212.04356
Model Card: https://github.com/openai/whisper/blob/main/model-card.md
Colab: Available in repo
License: MIT

來源與署名

來源：orchestra-research/ai-research-skills位於18-multimodal/whisper提交773a529

授權條款： MIT

內容歸原作者所有。SourceWeft 從公開儲存庫中收錄這些內容。

檢舉或申請下架

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

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

Whisper

Whisper - Robust Speech Recognition

When to use Whisper

Quick start

Installation

Basic transcription

Model sizes

Transcription options

Language specification

Task selection

Initial prompt

Timestamps

Temperature fallback

Command line usage

Batch processing

Real-time transcription

GPU acceleration

Integration with other tools

Subtitle generation

With LangChain

Extract audio from video

Best practices

Performance

Language support

Limitations

Resources

來源與署名

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