whisper README

The project's README from github.com/openai/whisper. As one markdown file: README.md.

Whisper

[Blog] [Paper] [Model card] [Colab example]

Whisper is a general-purpose speech recognition model. It is trained on a large dataset of diverse audio and is also a multitasking model that can perform multilingual speech recognition, speech translation, and language identification.

Approach

Approach

A Transformer sequence-to-sequence model is trained on various speech processing tasks, including multilingual speech recognition, speech translation, spoken language identification, and voice activity detection. These tasks are jointly represented as a sequence of tokens to be predicted by the decoder, allowing a single model to replace many stages of a traditional speech-processing pipeline. The multitask training format uses a set of special tokens that serve as task specifiers or classification targets.

Setup

We used Python 3.9.9 and PyTorch 1.10.1 to train and test our models, but the codebase is expected to be compatible with Python 3.8-3.11 and recent PyTorch versions. The codebase also depends on a few Python packages, most notably OpenAI's tiktoken for their fast tokenizer implementation. You can download and install (or update to) the latest release of Whisper with the following command:

pip install -U openai-whisper

Alternatively, the following command will pull and install the latest commit from this repository, along with its Python dependencies:

pip install git+https://github.com/openai/whisper.git 

To update the package to the latest version of this repository, please run:

pip install --upgrade --no-deps --force-reinstall git+https://github.com/openai/whisper.git

It also requires the command-line tool ffmpeg to be installed on your system, which is available from most package managers:

# on Ubuntu or Debian
sudo apt update && sudo apt install ffmpeg

# on Arch Linux
sudo pacman -S ffmpeg

# on MacOS using Homebrew (https://brew.sh/)
brew install ffmpeg

# on Windows using Chocolatey (https://chocolatey.org/)
choco install ffmpeg

# on Windows using Scoop (https://scoop.sh/)
scoop install ffmpeg

You may need rust installed as well, in case tiktoken does not provide a pre-built wheel for your platform. If you see installation errors during the pip install command above, please follow the Getting started page to install Rust development environment. Additionally, you may need to configure the PATH environment variable, e.g. export PATH="$HOME/.cargo/bin:$PATH". If the installation fails with No module named 'setuptools_rust', you need to install setuptools_rust, e.g. by running:

pip install setuptools-rust

Available models and languages

There are six model sizes, four with English-only versions, offering speed and accuracy tradeoffs. Below are the names of the available models and their approximate memory requirements and inference speed relative to the large model. The relative speeds below are measured by transcribing English speech on a A100, and the real-world speed may vary significantly depending on many factors including the language, the speaking speed, and the available hardware.

Size Parameters English-only model Multilingual model Required VRAM Relative speed
tiny 39 M tiny.en tiny ~1 GB ~10x
base 74 M base.en base ~1 GB ~7x
small 244 M small.en small ~2 GB ~4x
medium 769 M medium.en medium ~5 GB ~2x
large 1550 M N/A large ~10 GB 1x
turbo 809 M N/A turbo ~6 GB ~8x

The .en models for English-only applications tend to perform better, especially for the tiny.en and base.en models. We observed that the difference becomes less significant for the small.en and medium.en models. Additionally, the turbo model is an optimized version of large-v3 that offers faster transcription speed with a minimal degradation in accuracy.

Whisper's performance varies widely depending on the language. The figure below shows a performance breakdown of large-v3 and large-v2 models by language, using WERs (word error rates) or CER (character error rates, shown in Italic) evaluated on the Common Voice 15 and Fleurs datasets. Additional WER/CER metrics corresponding to the other models and datasets can be found in Appendix D.1, D.2, and D.4 of the paper, as well as the BLEU (Bilingual Evaluation Understudy) scores for translation in Appendix D.3.

WER breakdown by language

Command-line usage

The following command will transcribe speech in audio files, using the turbo model:

whisper audio.flac audio.mp3 audio.wav --model turbo

The default setting (which selects the turbo model) works well for transcribing English. However, the turbo model is not trained for translation tasks. If you need to translate non-English speech into English, use one of the multilingual models (tiny, base, small, medium, large) instead of turbo.

For example, to transcribe an audio file containing non-English speech, you can specify the language:

whisper japanese.wav --language Japanese

To translate speech into English, use:

whisper japanese.wav --model medium --language Japanese --task translate

Note: The turbo model will return the original language even if --task translate is specified. Use medium or large for the best translation results.

Run the following to view all available options:

whisper --help

See tokenizer.py for the list of all available languages.

Python usage

Transcription can also be performed within Python:

import whisper

model = whisper.load_model("turbo")
result = model.transcribe("audio.mp3")
print(result["text"])

Internally, the transcribe() method reads the entire file and processes the audio with a sliding 30-second window, performing autoregressive sequence-to-sequence predictions on each window.

Below is an example usage of whisper.detect_language() and whisper.decode() which provide lower-level access to the model.

import whisper

model = whisper.load_model("turbo")

# load audio and pad/trim it to fit 30 seconds
audio = whisper.load_audio("audio.mp3")
audio = whisper.pad_or_trim(audio)

# make log-Mel spectrogram and move to the same device as the model
mel = whisper.log_mel_spectrogram(audio, n_mels=model.dims.n_mels).to(model.device)

# detect the spoken language
_, probs = model.detect_language(mel)
print(f"Detected language: {max(probs, key=probs.get)}")

# decode the audio
options = whisper.DecodingOptions()
result = whisper.decode(model, mel, options)

# print the recognized text
print(result.text)

More examples

Please use the 🙌 Show and tell category in Discussions for sharing more example usages of Whisper and third-party extensions such as web demos, integrations with other tools, ports for different platforms, etc.

License

Whisper's code and model weights are released under the MIT License. See LICENSE for further details.

whisper (openai/whisper)

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110.7K ·GitHub
这是什么?Whisper 是一个通用语音识别模型,基于大规模、多样化的数据并采用弱监督方式训练而成。它可以进行多语言转写、把语音翻译成英文,还能识别说话所用的语言。
这是独立应用,还是我可以在代码里调用的库?它是一个 Python 库,附带一个命令行工具。你可以用 pip 安装它,然后对音频文件运行 'whisper' 命令,或者在自己的 Python 代码中调用它。
我该怎么安装和运行它?用 'pip install -U openai-whisper' 安装(如果想要最新提交,可以用 'pip install git+...')。然后在命令行中运行 'whisper audio.mp3' 来转写文件,默认使用 turbo 模型。
使用前需要先安装什么?你需要 Python 3.8-3.11 和 PyTorch,并在系统中安装 ffmpeg 命令行工具。如果 tiktoken 没有适用于你所用平台的预编译包,可能还需要 Rust 工具链。
最快的模型也能做翻译吗?不能。默认的 turbo 模型针对速度做了优化,但没有经过翻译训练。即使你指定 translate 任务,它仍会输出原始语言。如需翻译,请使用 medium 或 large 多语言模型。
运行一个模型需要多少显存?这取决于模型大小:tiny 和 base 约需 1GB,small 约需 2GB,medium 约需 5GB,large 最高需要 10GB,turbo 约需 6GB。速度是在 A100 GPU 上测得的。
有哪些模型尺寸可选?共有六种尺寸:tiny(39M 参数)、base(74M)、small(244M)、medium(769M)、large(1550M)和 turbo(809M),各自在速度与准确率之间有所取舍。tiny 到 medium 还提供仅支持英语的 '.en' 版本。
我为什么要选仅英语(.en)模型?在只处理英语时,.en 模型通常比多语言版本表现更好,其中 tiny.en 和 base.en 的提升最为明显。
turbo 模型有什么不同?turbo 是 large-v3 的优化版本,转写速度更快,准确率仅略有下降。但它不支持其他多语言模型具备的翻译功能。
它如何处理长音频文件?transcribe() 方法会读取整个文件,并通过 30 秒的滑动窗口进行处理,对每个窗口运行自回归的序列到序列(sequence-to-sequence)预测。
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