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[flake8]
per-file-ignores =
*/__init__.py: F401

3
.gitattributes vendored
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# Override jupyter in Github language stats for more accurate estimate of repo code languages
# reference: https://github.com/github/linguist/blob/master/docs/overrides.md#generated-code
*.ipynb linguist-generated

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# Keep GitHub Actions up to date with GitHub's Dependabot...
# https://docs.github.com/en/code-security/dependabot/working-with-dependabot/keeping-your-actions-up-to-date-with-dependabot
# https://docs.github.com/en/code-security/dependabot/dependabot-version-updates/configuration-options-for-the-dependabot.yml-file#package-ecosystem
version: 2
updates:
- package-ecosystem: github-actions
directory: /
groups:
github-actions:
patterns:
- "*" # Group all Actions updates into a single larger pull request
schedule:
interval: weekly

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@ -8,23 +8,23 @@ jobs:
deploy:
runs-on: ubuntu-latest
steps:
- uses: actions/checkout@v4
- uses: actions/checkout@v2
- uses: actions-ecosystem/action-regex-match@v2
id: regex-match
with:
text: ${{ github.event.head_commit.message }}
regex: '^Release ([^ ]+)'
- name: Set up Python
uses: actions/setup-python@v5
uses: actions/setup-python@v2
with:
python-version: '3.12'
python-version: '3.8'
- name: Install dependencies
run: |
python -m pip install --upgrade pip
pip install setuptools wheel twine build
pip install setuptools wheel twine
- name: Release
if: ${{ steps.regex-match.outputs.match != '' }}
uses: softprops/action-gh-release@v2
uses: softprops/action-gh-release@v1
with:
tag_name: v${{ steps.regex-match.outputs.group1 }}
- name: Build and publish
@ -33,5 +33,5 @@ jobs:
TWINE_USERNAME: __token__
TWINE_PASSWORD: ${{ secrets.PYPI_API_TOKEN }}
run: |
python -m build --sdist
python setup.py sdist
twine upload dist/*

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@ -6,74 +6,21 @@ on:
pull_request:
branches:
- main
jobs:
pre-commit:
runs-on: ubuntu-latest
steps:
- uses: actions/checkout@v4
- name: Fetch base branch
run: git fetch origin ${{ github.base_ref }}
- uses: actions/setup-python@v5
with:
python-version: "3.9"
architecture: x64
- name: Get pip cache dir
id: pip-cache
run: |
echo "dir=$(pip cache dir)" >> $GITHUB_OUTPUT
- name: pip/pre-commit cache
uses: actions/cache@v4
with:
path: |
${{ steps.pip-cache.outputs.dir }}
~/.cache/pre-commit
key: ${{ runner.os }}-pip-pre-commit-${{ hashFiles('**/.pre-commit-config.yaml') }}
restore-keys: |
${{ runner.os }}-pip-pre-commit
- name: pre-commit
run: |
pip install --upgrade pre-commit
pre-commit install --install-hooks
pre-commit run --all-files
whisper-test:
needs: pre-commit
runs-on: ubuntu-latest
strategy:
fail-fast: false
matrix:
include:
- python-version: '3.8'
pytorch-version: 1.10.1
numpy-requirement: "'numpy<2'"
- python-version: '3.8'
pytorch-version: 1.13.1
numpy-requirement: "'numpy<2'"
- python-version: '3.8'
pytorch-version: 2.0.1
numpy-requirement: "'numpy<2'"
- python-version: '3.9'
pytorch-version: 2.1.2
numpy-requirement: "'numpy<2'"
python-version: ['3.8', '3.9', '3.10']
pytorch-version: [1.10.2, 1.13.1]
exclude:
- python-version: '3.10'
pytorch-version: 2.2.2
numpy-requirement: "'numpy<2'"
- python-version: '3.11'
pytorch-version: 2.3.1
numpy-requirement: "'numpy'"
- python-version: '3.12'
pytorch-version: 2.4.1
numpy-requirement: "'numpy'"
- python-version: '3.12'
pytorch-version: 2.5.1
numpy-requirement: "'numpy'"
- python-version: '3.13'
pytorch-version: 2.5.1
numpy-requirement: "'numpy'"
pytorch-version: 1.10.2
steps:
- uses: conda-incubator/setup-miniconda@v3
- run: conda install -n test ffmpeg python=${{ matrix.python-version }}
- uses: actions/checkout@v4
- uses: conda-incubator/setup-miniconda@v2
- run: conda install -n test ffmpeg python=${{ matrix.python-version }} pytorch=${{ matrix.pytorch-version }} cpuonly -c pytorch
- uses: actions/checkout@v2
- run: echo "$CONDA/envs/test/bin" >> $GITHUB_PATH
- run: pip3 install .["dev"] ${{ matrix.numpy-requirement }} torch==${{ matrix.pytorch-version }}+cpu --index-url https://download.pytorch.org/whl/cpu --extra-index-url https://pypi.org/simple
- run: pytest --durations=0 -vv -k 'not test_transcribe or test_transcribe[tiny] or test_transcribe[tiny.en]' -m 'not requires_cuda'
- run: pip install pytest
- run: pip install .
- run: pytest -k 'not test_transcribe or test_transcribe[tiny]'

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@ -1,28 +0,0 @@
repos:
- repo: https://github.com/pre-commit/pre-commit-hooks
rev: v5.0.0
hooks:
- id: check-json
- id: end-of-file-fixer
types: [file, python]
- id: trailing-whitespace
types: [file, python]
- id: mixed-line-ending
- id: check-added-large-files
args: [--maxkb=4096]
- repo: https://github.com/psf/black
rev: 25.1.0
hooks:
- id: black
- repo: https://github.com/pycqa/isort
rev: 6.0.0
hooks:
- id: isort
name: isort (python)
args: ["--profile", "black", "-l", "88", "--trailing-comma", "--multi-line", "3"]
- repo: https://github.com/pycqa/flake8.git
rev: 7.1.1
hooks:
- id: flake8
types: [python]
args: ["--max-line-length", "88", "--ignore", "E203,E501,W503,W504"]

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@ -1,121 +0,0 @@
# CHANGELOG
## [v20250625](https://github.com/openai/whisper/releases/tag/v20250625)
* Fix: Update torch.load to use weights_only=True to prevent security w… ([#2451](https://github.com/openai/whisper/pull/2451))
* Fix: Ensure DTW cost tensor is on the same device as input tensor ([#2561](https://github.com/openai/whisper/pull/2561))
* docs: updated README to specify translation model limitation ([#2547](https://github.com/openai/whisper/pull/2547))
* Fixed triton kernel update to support latest triton versions ([#2588](https://github.com/openai/whisper/pull/2588))
* Fix: GitHub display errors for Jupyter notebooks ([#2589](https://github.com/openai/whisper/pull/2589))
* Bump the github-actions group with 3 updates ([#2592](https://github.com/openai/whisper/pull/2592))
* Keep GitHub Actions up to date with GitHub's Dependabot ([#2486](https://github.com/openai/whisper/pull/2486))
* pre-commit: Upgrade black v25.1.0 and isort v6.0.0 ([#2514](https://github.com/openai/whisper/pull/2514))
* GitHub Actions: Add Python 3.13 to the testing ([#2487](https://github.com/openai/whisper/pull/2487))
* PEP 621: Migrate from setup.py to pyproject.toml ([#2435](https://github.com/openai/whisper/pull/2435))
* pre-commit autoupdate && pre-commit run --all-files ([#2484](https://github.com/openai/whisper/pull/2484))
* Upgrade GitHub Actions ([#2430](https://github.com/openai/whisper/pull/2430))
* Bugfix: Illogical "Avoid computing higher temperatures on no_speech" ([#1903](https://github.com/openai/whisper/pull/1903))
* Updating README and doc strings to reflect that n_mels can now be 128 ([#2049](https://github.com/openai/whisper/pull/2049))
* fix typo data/README.md ([#2433](https://github.com/openai/whisper/pull/2433))
* Update README.md ([#2379](https://github.com/openai/whisper/pull/2379))
* Add option to carry initial_prompt with the sliding window ([#2343](https://github.com/openai/whisper/pull/2343))
* more pytorch versions in tests ([#2408](https://github.com/openai/whisper/pull/2408))
## [v20240930](https://github.com/openai/whisper/releases/tag/v20240930)
* allowing numpy 2 in tests ([#2362](https://github.com/openai/whisper/pull/2362))
* large-v3-turbo model ([#2361](https://github.com/openai/whisper/pull/2361))
* test on python/pytorch versions up to 3.12 and 2.4.1 ([#2360](https://github.com/openai/whisper/pull/2360))
* using sdpa if available ([#2359](https://github.com/openai/whisper/pull/2359))
## [v20240927](https://github.com/openai/whisper/releases/tag/v20240927)
* pinning numpy<2 in tests ([#2332](https://github.com/openai/whisper/pull/2332))
* Relax triton requirements for compatibility with pytorch 2.4 and newer ([#2307](https://github.com/openai/whisper/pull/2307))
* Skip silence around hallucinations ([#1838](https://github.com/openai/whisper/pull/1838))
* Fix triton env marker ([#1887](https://github.com/openai/whisper/pull/1887))
## [v20231117](https://github.com/openai/whisper/releases/tag/v20231117)
* Relax triton requirements for compatibility with pytorch 2.1 and newer ([#1802](https://github.com/openai/whisper/pull/1802))
## [v20231106](https://github.com/openai/whisper/releases/tag/v20231106)
* large-v3 ([#1761](https://github.com/openai/whisper/pull/1761))
## [v20231105](https://github.com/openai/whisper/releases/tag/v20231105)
* remove tiktoken pin ([#1759](https://github.com/openai/whisper/pull/1759))
* docs: Disambiguation of the term "relative speed" in the README ([#1751](https://github.com/openai/whisper/pull/1751))
* allow_pickle=False while loading of mel matrix IN audio.py ([#1511](https://github.com/openai/whisper/pull/1511))
* handling transcribe exceptions. ([#1682](https://github.com/openai/whisper/pull/1682))
* Add new option to generate subtitles by a specific number of words ([#1729](https://github.com/openai/whisper/pull/1729))
* Fix exception when an audio file with no speech is provided ([#1396](https://github.com/openai/whisper/pull/1396))
## [v20230918](https://github.com/openai/whisper/releases/tag/v20230918)
* Add .pre-commit-config.yaml ([#1528](https://github.com/openai/whisper/pull/1528))
* fix doc of TextDecoder ([#1526](https://github.com/openai/whisper/pull/1526))
* Update model-card.md ([#1643](https://github.com/openai/whisper/pull/1643))
* word timing tweaks ([#1559](https://github.com/openai/whisper/pull/1559))
* Avoid rearranging all caches ([#1483](https://github.com/openai/whisper/pull/1483))
* Improve timestamp heuristics. ([#1461](https://github.com/openai/whisper/pull/1461))
* fix condition_on_previous_text ([#1224](https://github.com/openai/whisper/pull/1224))
* Fix numba depreceation notice ([#1233](https://github.com/openai/whisper/pull/1233))
* Updated README.md to provide more insight on BLEU and specific appendices ([#1236](https://github.com/openai/whisper/pull/1236))
* Avoid computing higher temperatures on no_speech segments ([#1279](https://github.com/openai/whisper/pull/1279))
* Dropped unused execute bit from mel_filters.npz. ([#1254](https://github.com/openai/whisper/pull/1254))
* Drop ffmpeg-python dependency and call ffmpeg directly. ([#1242](https://github.com/openai/whisper/pull/1242))
* Python 3.11 ([#1171](https://github.com/openai/whisper/pull/1171))
* Update decoding.py ([#1219](https://github.com/openai/whisper/pull/1219))
* Update decoding.py ([#1155](https://github.com/openai/whisper/pull/1155))
* Update README.md to reference tiktoken ([#1105](https://github.com/openai/whisper/pull/1105))
* Implement max line width and max line count, and make word highlighting optional ([#1184](https://github.com/openai/whisper/pull/1184))
* Squash long words at window and sentence boundaries. ([#1114](https://github.com/openai/whisper/pull/1114))
* python-publish.yml: bump actions version to fix node warning ([#1211](https://github.com/openai/whisper/pull/1211))
* Update tokenizer.py ([#1163](https://github.com/openai/whisper/pull/1163))
## [v20230314](https://github.com/openai/whisper/releases/tag/v20230314)
* abort find_alignment on empty input ([#1090](https://github.com/openai/whisper/pull/1090))
* Fix truncated words list when the replacement character is decoded ([#1089](https://github.com/openai/whisper/pull/1089))
* fix github language stats getting dominated by jupyter notebook ([#1076](https://github.com/openai/whisper/pull/1076))
* Fix alignment between the segments and the list of words ([#1087](https://github.com/openai/whisper/pull/1087))
* Use tiktoken ([#1044](https://github.com/openai/whisper/pull/1044))
## [v20230308](https://github.com/openai/whisper/releases/tag/v20230308)
* kwargs in decode() for convenience ([#1061](https://github.com/openai/whisper/pull/1061))
* fix all_tokens handling that caused more repetitions and discrepancy in JSON ([#1060](https://github.com/openai/whisper/pull/1060))
* fix typo in CHANGELOG.md
## [v20230307](https://github.com/openai/whisper/releases/tag/v20230307)
* Fix the repetition/hallucination issue identified in #1046 ([#1052](https://github.com/openai/whisper/pull/1052))
* Use triton==2.0.0 ([#1053](https://github.com/openai/whisper/pull/1053))
* Install triton in x86_64 linux only ([#1051](https://github.com/openai/whisper/pull/1051))
* update setup.py to specify python >= 3.8 requirement
## [v20230306](https://github.com/openai/whisper/releases/tag/v20230306)
* remove auxiliary audio extension ([#1021](https://github.com/openai/whisper/pull/1021))
* apply formatting with `black`, `isort`, and `flake8` ([#1038](https://github.com/openai/whisper/pull/1038))
* word-level timestamps in `transcribe()` ([#869](https://github.com/openai/whisper/pull/869))
* Decoding improvements ([#1033](https://github.com/openai/whisper/pull/1033))
* Update README.md ([#894](https://github.com/openai/whisper/pull/894))
* Fix infinite loop caused by incorrect timestamp tokens prediction ([#914](https://github.com/openai/whisper/pull/914))
* drop python 3.7 support ([#889](https://github.com/openai/whisper/pull/889))
## [v20230124](https://github.com/openai/whisper/releases/tag/v20230124)
* handle printing even if sys.stdout.buffer is not available ([#887](https://github.com/openai/whisper/pull/887))
* Add TSV formatted output in transcript, using integer start/end time in milliseconds ([#228](https://github.com/openai/whisper/pull/228))
* Added `--output_format` option ([#333](https://github.com/openai/whisper/pull/333))
* Handle `XDG_CACHE_HOME` properly for `download_root` ([#864](https://github.com/openai/whisper/pull/864))
* use stdout for printing transcription progress ([#867](https://github.com/openai/whisper/pull/867))
* Fix bug where mm is mistakenly replaced with hmm in e.g. 20mm ([#659](https://github.com/openai/whisper/pull/659))
* print '?' if a letter can't be encoded using the system default encoding ([#859](https://github.com/openai/whisper/pull/859))
## [v20230117](https://github.com/openai/whisper/releases/tag/v20230117)
The first versioned release available on [PyPI](https://pypi.org/project/openai-whisper/)

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@ -2,4 +2,6 @@ include requirements.txt
include README.md
include LICENSE
include whisper/assets/*
include whisper/assets/gpt2/*
include whisper/assets/multilingual/*
include whisper/normalizers/english.json

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@ -5,19 +5,19 @@
[[Model card]](https://github.com/openai/whisper/blob/main/model-card.md)
[[Colab example]](https://colab.research.google.com/github/openai/whisper/blob/master/notebooks/LibriSpeech.ipynb)
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.
Whisper is a general-purpose speech recognition model. It is trained on a large dataset of diverse audio and is also a multi-task model that can perform multilingual speech recognition as well as speech translation and language identification.
## Approach
![Approach](https://raw.githubusercontent.com/openai/whisper/main/approach.png)
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.
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. All of these tasks are jointly represented as a sequence of tokens to be predicted by the decoder, allowing for a single model to replace many different 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](https://pytorch.org/) 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](https://github.com/openai/tiktoken) for their fast tokenizer implementation. You can download and install (or update to) the latest release of Whisper with the following command:
We used Python 3.9.9 and [PyTorch](https://pytorch.org/) 1.10.1 to train and test our models, but the codebase is expected to be compatible with Python 3.7 or later and recent PyTorch versions. The codebase also depends on a few Python packages, most notably [HuggingFace Transformers](https://huggingface.co/docs/transformers/index) for their fast tokenizer implementation and [ffmpeg-python](https://github.com/kkroening/ffmpeg-python) for reading audio files. You can download and install (or update to) the latest release of Whisper with the following command:
pip install -U openai-whisper
@ -48,7 +48,7 @@ choco install ffmpeg
scoop install ffmpeg
```
You may need [`rust`](http://rust-lang.org) installed as well, in case [tiktoken](https://github.com/openai/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](https://www.rust-lang.org/learn/get-started) 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:
You may need [`rust`](http://rust-lang.org) installed as well, in case [tokenizers](https://pypi.org/project/tokenizers/) 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](https://www.rust-lang.org/learn/get-started) 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:
```bash
pip install setuptools-rust
@ -57,55 +57,42 @@ 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.
There are five 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 relative speed.
| 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 |
| tiny | 39 M | `tiny.en` | `tiny` | ~1 GB | ~32x |
| base | 74 M | `base.en` | `base` | ~1 GB | ~16x |
| small | 244 M | `small.en` | `small` | ~2 GB | ~6x |
| 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.
For English-only applications, the `.en` models 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.
Whisper's performance varies widely depending on the language. The figure below shows a WER (Word Error Rate) breakdown by languages of Fleurs dataset, using the `large-v2` model. More WER and BLEU scores corresponding to the other models and datasets can be found in Appendix D in [the paper](https://arxiv.org/abs/2212.04356). The smaller is better.
![WER breakdown by language](https://raw.githubusercontent.com/openai/whisper/main/language-breakdown.svg)
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](https://arxiv.org/abs/2212.04356), as well as the BLEU (Bilingual Evaluation Understudy) scores for translation in Appendix D.3.
![WER breakdown by language](https://github.com/openai/whisper/assets/266841/f4619d66-1058-4005-8f67-a9d811b77c62)
## Command-line usage
The following command will transcribe speech in audio files, using the `turbo` model:
The following command will transcribe speech in audio files, using the `medium` model:
```bash
whisper audio.flac audio.mp3 audio.wav --model turbo
```
whisper audio.flac audio.mp3 audio.wav --model medium
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`.
The default setting (which selects the `small` model) works well for transcribing English. To transcribe an audio file containing non-English speech, you can specify the language using the `--language` option:
For example, to transcribe an audio file containing non-English speech, you can specify the language:
whisper japanese.wav --language Japanese
```bash
whisper japanese.wav --language Japanese
```
Adding `--task translate` will translate the speech into English:
To **translate** speech into English, use:
```bash
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.
whisper japanese.wav --language Japanese --task translate
Run the following to view all available options:
```bash
whisper --help
```
whisper --help
See [tokenizer.py](https://github.com/openai/whisper/blob/main/whisper/tokenizer.py) for the list of all available languages.
@ -117,7 +104,7 @@ Transcription can also be performed within Python:
```python
import whisper
model = whisper.load_model("turbo")
model = whisper.load_model("base")
result = model.transcribe("audio.mp3")
print(result["text"])
```
@ -129,14 +116,14 @@ Below is an example usage of `whisper.detect_language()` and `whisper.decode()`
```python
import whisper
model = whisper.load_model("turbo")
model = whisper.load_model("base")
# 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)
mel = whisper.log_mel_spectrogram(audio).to(model.device)
# detect the spoken language
_, probs = model.detect_language(mel)
@ -157,4 +144,4 @@ Please use the [🙌 Show and tell](https://github.com/openai/whisper/discussion
## License
Whisper's code and model weights are released under the MIT License. See [LICENSE](https://github.com/openai/whisper/blob/main/LICENSE) for further details.
The code and the model weights of Whisper are released under the MIT License. See [LICENSE](https://github.com/openai/whisper/blob/main/LICENSE) for further details.

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@ -45,7 +45,7 @@ We downloaded the [CHiME-5 dataset](https://spandh.dcs.shef.ac.uk//chime_challen
### AMI-IHM, AMI-SDM1
We preprocessed the [AMI Corpus](https://groups.inf.ed.ac.uk/ami/corpus/overview.shtml) by following the stage 0 and 2 of the [s5b recipe](https://github.com/kaldi-asr/kaldi/tree/master/egs/ami/s5b).
We preprocessed the [AMI Corpus](https://groups.inf.ed.ac.uk/ami/corpus/overview.shtml) by following the stage 0 ad 2 of the [s5b recipe](https://github.com/kaldi-asr/kaldi/tree/master/egs/ami/s5b).
## Long-form English-only datasets

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@ -16,15 +16,13 @@ The Whisper models are trained for speech recognition and translation tasks, cap
| small | 244 M | ✓ | ✓ |
| medium | 769 M | ✓ | ✓ |
| large | 1550 M | | ✓ |
| turbo | 798 M | | ✓ |
In December 2022, we [released an improved large model named `large-v2`](https://github.com/openai/whisper/discussions/661), and `large-v3` in November 2023.
Additionally, we've added a `turbo` model in September 2024 which is optimized for inference speed.
In December 2022, we [released an improved large model named `large-v2`](https://github.com/openai/whisper/discussions/661).
### Release date
September 2022 (original series), December 2022 (`large-v2`), November 2023 (`large-v3`), September 2024 (`large-v3-turbo`)
September 2022 (original series) and December 2022 (`large-v2`)
### Model type
@ -39,7 +37,7 @@ Sequence-to-sequence ASR (automatic speech recognition) and speech translation m
### Evaluated Use
The primary intended users of these models are AI researchers studying the robustness, generalization, capabilities, biases, and constraints of the current model. However, Whisper is also potentially quite useful as an ASR solution for developers, especially for English speech recognition. We recognize that once models are released, it is impossible to restrict access to only “intended” uses or to draw reasonable guidelines around what is or is not research.
The primary intended users of these models are AI researchers studying robustness, generalization, capabilities, biases, and constraints of the current model. However, Whisper is also potentially quite useful as an ASR solution for developers, especially for English speech recognition. We recognize that once models are released, it is impossible to restrict access to only “intended” uses or to draw reasonable guidelines around what is or is not research.
The models are primarily trained and evaluated on ASR and speech translation to English tasks. They show strong ASR results in ~10 languages. They may exhibit additional capabilities, particularly if fine-tuned on certain tasks like voice activity detection, speaker classification, or speaker diarization but have not been robustly evaluated in these areas. We strongly recommend that users perform robust evaluations of the models in a particular context and domain before deploying them.
@ -55,17 +53,17 @@ As discussed in [the accompanying paper](https://arxiv.org/abs/2212.04356), we s
## Performance and Limitations
Our studies show that, over many existing ASR systems, the models exhibit improved robustness to accents, background noise, and technical language, as well as zero-shot translation from multiple languages into English; and that accuracy on speech recognition and translation is near the state-of-the-art level.
Our studies show that, over many existing ASR systems, the models exhibit improved robustness to accents, background noise, technical language, as well as zero shot translation from multiple languages into English; and that accuracy on speech recognition and translation is near the state-of-the-art level.
However, because the models are trained in a weakly supervised manner using large-scale noisy data, the predictions may include texts that are not actually spoken in the audio input (i.e. hallucination). We hypothesize that this happens because, given their general knowledge of language, the models combine trying to predict the next word in audio with trying to transcribe the audio itself.
Our models perform unevenly across languages, and we observe lower accuracy on low-resource and/or low-discoverability languages or languages where we have less training data. The models also exhibit disparate performance on different accents and dialects of particular languages, which may include a higher word error rate across speakers of different genders, races, ages, or other demographic criteria. Our full evaluation results are presented in [the paper accompanying this release](https://arxiv.org/abs/2212.04356).
Our models perform unevenly across languages, and we observe lower accuracy on low-resource and/or low-discoverability languages or languages where we have less training data. The models also exhibit disparate performance on different accents and dialects of particular languages, which may include higher word error rate across speakers of different genders, races, ages, or other demographic criteria. Our full evaluation results are presented in [the paper accompanying this release](https://arxiv.org/abs/2212.04356).
In addition, the sequence-to-sequence architecture of the model makes it prone to generating repetitive texts, which can be mitigated to some degree by beam search and temperature scheduling but not perfectly. Further analysis of these limitations is provided in [the paper](https://arxiv.org/abs/2212.04356). It is likely that this behavior and hallucinations may be worse in lower-resource and/or lower-discoverability languages.
In addition, the sequence-to-sequence architecture of the model makes it prone to generating repetitive texts, which can be mitigated to some degree by beam search and temperature scheduling but not perfectly. Further analysis on these limitations are provided in [the paper](https://arxiv.org/abs/2212.04356). It is likely that this behavior and hallucinations may be worse on lower-resource and/or lower-discoverability languages.
## Broader Implications
We anticipate that Whisper models transcription capabilities may be used for improving accessibility tools. While Whisper models cannot be used for real-time transcription out of the box their speed and size suggest that others may be able to build applications on top of them that allow for near-real-time speech recognition and translation. The real value of beneficial applications built on top of Whisper models suggests that the disparate performance of these models may have real economic implications.
There are also potential dual-use concerns that come with releasing Whisper. While we hope the technology will be used primarily for beneficial purposes, making ASR technology more accessible could enable more actors to build capable surveillance technologies or scale up existing surveillance efforts, as the speed and accuracy allow for affordable automatic transcription and translation of large volumes of audio communication. Moreover, these models may have some capabilities to recognize specific individuals out of the box, which in turn presents safety concerns related both to dual use and disparate performance. In practice, we expect that the cost of transcription is not the limiting factor of scaling up surveillance projects.
There are also potential dual use concerns that come with releasing Whisper. While we hope the technology will be used primarily for beneficial purposes, making ASR technology more accessible could enable more actors to build capable surveillance technologies or scale up existing surveillance efforts, as the speed and accuracy allow for affordable automatic transcription and translation of large volumes of audio communication. Moreover, these models may have some capabilities to recognize specific individuals out of the box, which in turn presents safety concerns related both to dual use and disparate performance. In practice, we expect that the cost of transcription is not the limiting factor of scaling up surveillance projects.

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@ -949,8 +949,7 @@
"style": "IPY_MODEL_039b53f2702c4179af7e0548018d0588",
"value": " 164/164 [05:08&lt;00:00, 1.86s/it]"
}
},
"state": {}
}
}
}
},

View File

@ -4219,8 +4219,7 @@
"_view_name": "StyleView",
"description_width": ""
}
},
"state": {}
}
}
}
},

View File

@ -1,54 +0,0 @@
[build-system]
build-backend = "setuptools.build_meta"
requires = [ "setuptools>=61.2" ]
[project]
name = "openai-whisper"
description = "Robust Speech Recognition via Large-Scale Weak Supervision"
readme.content-type = "text/markdown"
readme.file = "README.md"
license = { text = "MIT" }
authors = [ { name = "OpenAI" } ]
requires-python = ">=3.8"
classifiers = [
"Programming Language :: Python :: 3 :: Only",
"Programming Language :: Python :: 3.8",
"Programming Language :: Python :: 3.9",
"Programming Language :: Python :: 3.10",
"Programming Language :: Python :: 3.11",
"Programming Language :: Python :: 3.12",
"Programming Language :: Python :: 3.13",
]
dynamic = [ "version" ]
dependencies = [
"more-itertools",
"numba",
"numpy",
"tiktoken",
"torch",
"tqdm",
"triton>=2; (platform_machine=='x86_64' and sys_platform=='linux') or sys_platform=='linux2'",
]
optional-dependencies.dev = [ "black", "flake8", "isort", "pytest", "scipy" ]
urls = { Homepage = "https://github.com/openai/whisper" }
scripts.whisper = "whisper.transcribe:cli"
[tool.setuptools]
py-modules = [ "whisper" ]
include-package-data = true
[tool.setuptools.dynamic]
version = { attr = "whisper.version.__version__" }
[tool.setuptools.packages.find]
exclude = [ "tests*" ]
namespaces = false
[tool.black]
[tool.isort]
profile = "black"
include_trailing_comma = true
line_length = 88
multi_line_output = 3

View File

@ -1,7 +1,6 @@
numba
numpy
torch
tqdm
more-itertools
tiktoken
triton>=2.0.0;platform_machine=="x86_64" and sys_platform=="linux" or sys_platform=="linux2"
transformers>=4.19.0
ffmpeg-python==0.2.0

36
setup.py Normal file
View File

@ -0,0 +1,36 @@
import os
import pkg_resources
from setuptools import setup, find_packages
def read_version(fname="whisper/version.py"):
exec(compile(open(fname, encoding="utf-8").read(), fname, "exec"))
return locals()["__version__"]
setup(
name="openai-whisper",
py_modules=["whisper"],
version=read_version(),
description="Robust Speech Recognition via Large-Scale Weak Supervision",
long_description=open("README.md", encoding="utf-8").read(),
long_description_content_type="text/markdown",
readme="README.md",
python_requires=">=3.7",
author="OpenAI",
url="https://github.com/openai/whisper",
license="MIT",
packages=find_packages(exclude=["tests*"]),
install_requires=[
str(r)
for r in pkg_resources.parse_requirements(
open(os.path.join(os.path.dirname(__file__), "requirements.txt"))
)
],
entry_points={
"console_scripts": ["whisper=whisper.transcribe:cli"],
},
include_package_data=True,
extras_require={"dev": ["pytest"]},
)

View File

@ -1,14 +0,0 @@
import random as rand
import numpy
import pytest
def pytest_configure(config):
config.addinivalue_line("markers", "requires_cuda")
@pytest.fixture
def random():
rand.seed(42)
numpy.random.seed(42)

View File

@ -2,7 +2,7 @@ import os.path
import numpy as np
from whisper.audio import SAMPLE_RATE, load_audio, log_mel_spectrogram
from whisper.audio import load_audio, log_mel_spectrogram, SAMPLE_RATE
def test_audio():

View File

@ -1,10 +1,7 @@
import pytest
from whisper.normalizers import EnglishTextNormalizer
from whisper.normalizers.english import (
EnglishNumberNormalizer,
EnglishSpellingNormalizer,
)
from whisper.normalizers.english import EnglishNumberNormalizer, EnglishSpellingNormalizer
@pytest.mark.parametrize("std", [EnglishNumberNormalizer(), EnglishTextNormalizer()])
@ -87,7 +84,6 @@ def test_text_normalizer():
assert std("he's like") == "he is like"
assert std("she's been like") == "she has been like"
assert std("10km") == "10 km"
assert std("10mm") == "10 mm"
assert std("RC232") == "rc 232"
assert (

View File

@ -1,96 +0,0 @@
import numpy as np
import pytest
import scipy.ndimage
import torch
from whisper.timing import dtw_cpu, dtw_cuda, median_filter
sizes = [
(10, 20),
(32, 16),
(123, 1500),
(234, 189),
]
shapes = [
(10,),
(1, 15),
(4, 5, 345),
(6, 12, 240, 512),
]
@pytest.mark.parametrize("N, M", sizes)
def test_dtw(N: int, M: int):
steps = np.concatenate([np.zeros(N - 1), np.ones(M - 1)])
np.random.shuffle(steps)
x = np.random.random((N, M)).astype(np.float32)
i, j, k = 0, 0, 0
trace = []
while True:
x[i, j] -= 1
trace.append((i, j))
if k == len(steps):
break
if k + 1 < len(steps) and steps[k] != steps[k + 1]:
i += 1
j += 1
k += 2
continue
if steps[k] == 0:
i += 1
if steps[k] == 1:
j += 1
k += 1
trace = np.array(trace).T
dtw_trace = dtw_cpu(x)
assert np.allclose(trace, dtw_trace)
@pytest.mark.requires_cuda
@pytest.mark.parametrize("N, M", sizes)
def test_dtw_cuda_equivalence(N: int, M: int):
x_numpy = np.random.randn(N, M).astype(np.float32)
x_cuda = torch.from_numpy(x_numpy).cuda()
trace_cpu = dtw_cpu(x_numpy)
trace_cuda = dtw_cuda(x_cuda)
assert np.allclose(trace_cpu, trace_cuda)
@pytest.mark.parametrize("shape", shapes)
def test_median_filter(shape):
x = torch.randn(*shape)
for filter_width in [3, 5, 7, 13]:
filtered = median_filter(x, filter_width)
# using np.pad to reflect-pad, because Scipy's behavior is different near the edges.
pad_width = filter_width // 2
padded_x = np.pad(
x, [(0, 0)] * (x.ndim - 1) + [(pad_width, pad_width)], mode="reflect"
)
scipy_filtered = scipy.ndimage.median_filter(
padded_x, [1] * (x.ndim - 1) + [filter_width]
)
scipy_filtered = scipy_filtered[..., pad_width:-pad_width]
assert np.allclose(filtered, scipy_filtered)
@pytest.mark.requires_cuda
@pytest.mark.parametrize("shape", shapes)
def test_median_filter_equivalence(shape):
x = torch.randn(*shape)
for filter_width in [3, 5, 7, 13]:
filtered_cpu = median_filter(x, filter_width)
filtered_gpu = median_filter(x.cuda(), filter_width).cpu()
assert np.allclose(filtered_cpu, filtered_gpu)

View File

@ -1,17 +1,7 @@
import pytest
from whisper.tokenizer import get_tokenizer
@pytest.mark.parametrize("multilingual", [True, False])
def test_tokenizer(multilingual):
tokenizer = get_tokenizer(multilingual=False)
assert tokenizer.sot in tokenizer.sot_sequence
assert len(tokenizer.all_language_codes) == len(tokenizer.all_language_tokens)
assert all(c < tokenizer.timestamp_begin for c in tokenizer.all_language_tokens)
def test_multilingual_tokenizer():
def test_tokenizer():
gpt2_tokenizer = get_tokenizer(multilingual=False)
multilingual_tokenizer = get_tokenizer(multilingual=True)
@ -22,13 +12,3 @@ def test_multilingual_tokenizer():
assert gpt2_tokenizer.decode(gpt2_tokens) == text
assert multilingual_tokenizer.decode(multilingual_tokens) == text
assert len(gpt2_tokens) > len(multilingual_tokens)
def test_split_on_unicode():
multilingual_tokenizer = get_tokenizer(multilingual=True)
tokens = [8404, 871, 287, 6, 246, 526, 3210, 20378]
words, word_tokens = multilingual_tokenizer.split_tokens_on_unicode(tokens)
assert words == [" elle", " est", " l", "'", "\ufffd", "é", "rit", "oire"]
assert word_tokens == [[8404], [871], [287], [6], [246], [526], [3210], [20378]]

View File

@ -4,7 +4,6 @@ import pytest
import torch
import whisper
from whisper.tokenizer import get_tokenizer
@pytest.mark.parametrize("model_name", whisper.available_models())
@ -14,29 +13,10 @@ def test_transcribe(model_name: str):
audio_path = os.path.join(os.path.dirname(__file__), "jfk.flac")
language = "en" if model_name.endswith(".en") else None
result = model.transcribe(
audio_path, language=language, temperature=0.0, word_timestamps=True
)
result = model.transcribe(audio_path, language=language, temperature=0.0)
assert result["language"] == "en"
assert result["text"] == "".join([s["text"] for s in result["segments"]])
transcription = result["text"].lower()
assert "my fellow americans" in transcription
assert "your country" in transcription
assert "do for you" in transcription
tokenizer = get_tokenizer(model.is_multilingual, num_languages=model.num_languages)
all_tokens = [t for s in result["segments"] for t in s["tokens"]]
assert tokenizer.decode(all_tokens) == result["text"]
assert tokenizer.decode_with_timestamps(all_tokens).startswith("<|0.00|>")
timing_checked = False
for segment in result["segments"]:
for timing in segment["words"]:
assert timing["start"] < timing["end"]
if timing["word"].strip(" ,") == "Americans":
assert timing["start"] <= 1.8
assert timing["end"] >= 1.8
timing_checked = True
assert timing_checked

View File

@ -10,10 +10,11 @@ from tqdm import tqdm
from .audio import load_audio, log_mel_spectrogram, pad_or_trim
from .decoding import DecodingOptions, DecodingResult, decode, detect_language
from .model import ModelDimensions, Whisper
from .model import Whisper, ModelDimensions
from .transcribe import transcribe
from .version import __version__
_MODELS = {
"tiny.en": "https://openaipublic.azureedge.net/main/whisper/models/d3dd57d32accea0b295c96e26691aa14d8822fac7d9d27d5dc00b4ca2826dd03/tiny.en.pt",
"tiny": "https://openaipublic.azureedge.net/main/whisper/models/65147644a518d12f04e32d6f3b26facc3f8dd46e5390956a9424a650c0ce22b9/tiny.pt",
@ -25,29 +26,7 @@ _MODELS = {
"medium": "https://openaipublic.azureedge.net/main/whisper/models/345ae4da62f9b3d59415adc60127b97c714f32e89e936602e85993674d08dcb1/medium.pt",
"large-v1": "https://openaipublic.azureedge.net/main/whisper/models/e4b87e7e0bf463eb8e6956e646f1e277e901512310def2c24bf0e11bd3c28e9a/large-v1.pt",
"large-v2": "https://openaipublic.azureedge.net/main/whisper/models/81f7c96c852ee8fc832187b0132e569d6c3065a3252ed18e56effd0b6a73e524/large-v2.pt",
"large-v3": "https://openaipublic.azureedge.net/main/whisper/models/e5b1a55b89c1367dacf97e3e19bfd829a01529dbfdeefa8caeb59b3f1b81dadb/large-v3.pt",
"large": "https://openaipublic.azureedge.net/main/whisper/models/e5b1a55b89c1367dacf97e3e19bfd829a01529dbfdeefa8caeb59b3f1b81dadb/large-v3.pt",
"large-v3-turbo": "https://openaipublic.azureedge.net/main/whisper/models/aff26ae408abcba5fbf8813c21e62b0941638c5f6eebfb145be0c9839262a19a/large-v3-turbo.pt",
"turbo": "https://openaipublic.azureedge.net/main/whisper/models/aff26ae408abcba5fbf8813c21e62b0941638c5f6eebfb145be0c9839262a19a/large-v3-turbo.pt",
}
# base85-encoded (n_layers, n_heads) boolean arrays indicating the cross-attention heads that are
# highly correlated to the word-level timing, i.e. the alignment between audio and text tokens.
_ALIGNMENT_HEADS = {
"tiny.en": b"ABzY8J1N>@0{>%R00Bk>$p{7v037`oCl~+#00",
"tiny": b"ABzY8bu8Lr0{>%RKn9Fp%m@SkK7Kt=7ytkO",
"base.en": b"ABzY8;40c<0{>%RzzG;p*o+Vo09|#PsxSZm00",
"base": b"ABzY8KQ!870{>%RzyTQH3`Q^yNP!>##QT-<FaQ7m",
"small.en": b"ABzY8>?_)10{>%RpeA61k&I|OI3I$65C{;;pbCHh0B{qLQ;+}v00",
"small": b"ABzY8DmU6=0{>%Rpa?J`kvJ6qF(V^F86#Xh7JUGMK}P<N0000",
"medium.en": b"ABzY8usPae0{>%R7<zz_OvQ{)4kMa0BMw6u5rT}kRKX;$NfYBv00*Hl@qhsU00",
"medium": b"ABzY8B0Jh+0{>%R7}kK1fFL7w6%<-Pf*t^=N)Qr&0RR9",
"large-v1": b"ABzY8r9j$a0{>%R7#4sLmoOs{s)o3~84-RPdcFk!JR<kSfC2yj",
"large-v2": b"ABzY8zd+h!0{>%R7=D0pU<_bnWW*tkYAhobTNnu$jnkEkXqp)j;w1Tzk)UH3X%SZd&fFZ2fC2yj",
"large-v3": b"ABzY8gWO1E0{>%R7(9S+Kn!D~%ngiGaR?*L!iJG9p-nab0JQ=-{D1-g00",
"large": b"ABzY8gWO1E0{>%R7(9S+Kn!D~%ngiGaR?*L!iJG9p-nab0JQ=-{D1-g00",
"large-v3-turbo": b"ABzY8j^C+e0{>%RARaKHP%t(lGR*)0g!tONPyhe`",
"turbo": b"ABzY8j^C+e0{>%RARaKHP%t(lGR*)0g!tONPyhe`",
"large": "https://openaipublic.azureedge.net/main/whisper/models/81f7c96c852ee8fc832187b0132e569d6c3065a3252ed18e56effd0b6a73e524/large-v2.pt",
}
@ -66,18 +45,10 @@ def _download(url: str, root: str, in_memory: bool) -> Union[bytes, str]:
if hashlib.sha256(model_bytes).hexdigest() == expected_sha256:
return model_bytes if in_memory else download_target
else:
warnings.warn(
f"{download_target} exists, but the SHA256 checksum does not match; re-downloading the file"
)
warnings.warn(f"{download_target} exists, but the SHA256 checksum does not match; re-downloading the file")
with urllib.request.urlopen(url) as source, open(download_target, "wb") as output:
with tqdm(
total=int(source.info().get("Content-Length")),
ncols=80,
unit="iB",
unit_scale=True,
unit_divisor=1024,
) as loop:
with tqdm(total=int(source.info().get("Content-Length")), ncols=80, unit='iB', unit_scale=True, unit_divisor=1024) as loop:
while True:
buffer = source.read(8192)
if not buffer:
@ -88,9 +59,7 @@ def _download(url: str, root: str, in_memory: bool) -> Union[bytes, str]:
model_bytes = open(download_target, "rb").read()
if hashlib.sha256(model_bytes).hexdigest() != expected_sha256:
raise RuntimeError(
"Model has been downloaded but the SHA256 checksum does not not match. Please retry loading the model."
)
raise RuntimeError("Model has been downloaded but the SHA256 checksum does not not match. Please retry loading the model.")
return model_bytes if in_memory else download_target
@ -100,12 +69,7 @@ def available_models() -> List[str]:
return list(_MODELS.keys())
def load_model(
name: str,
device: Optional[Union[str, torch.device]] = None,
download_root: str = None,
in_memory: bool = False,
) -> Whisper:
def load_model(name: str, device: Optional[Union[str, torch.device]] = None, download_root: str = None, in_memory: bool = False) -> Whisper:
"""
Load a Whisper ASR model
@ -130,32 +94,24 @@ def load_model(
if device is None:
device = "cuda" if torch.cuda.is_available() else "cpu"
if download_root is None:
default = os.path.join(os.path.expanduser("~"), ".cache")
download_root = os.path.join(os.getenv("XDG_CACHE_HOME", default), "whisper")
download_root = os.getenv(
"XDG_CACHE_HOME",
os.path.join(os.path.expanduser("~"), ".cache", "whisper")
)
if name in _MODELS:
checkpoint_file = _download(_MODELS[name], download_root, in_memory)
alignment_heads = _ALIGNMENT_HEADS[name]
elif os.path.isfile(name):
checkpoint_file = open(name, "rb").read() if in_memory else name
alignment_heads = None
else:
raise RuntimeError(
f"Model {name} not found; available models = {available_models()}"
)
raise RuntimeError(f"Model {name} not found; available models = {available_models()}")
with (
io.BytesIO(checkpoint_file) if in_memory else open(checkpoint_file, "rb")
) as fp:
kwargs = {"weights_only": True} if torch.__version__ >= "1.13" else {}
checkpoint = torch.load(fp, map_location=device, **kwargs)
with (io.BytesIO(checkpoint_file) if in_memory else open(checkpoint_file, "rb")) as fp:
checkpoint = torch.load(fp, map_location=device)
del checkpoint_file
dims = ModelDimensions(**checkpoint["dims"])
model = Whisper(dims)
model.load_state_dict(checkpoint["model_state_dict"])
if alignment_heads is not None:
model.set_alignment_heads(alignment_heads)
return model.to(device)

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@ -1,3 +1,4 @@
from .transcribe import cli
cli()

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whisper/assets/gpt2/merges.txt Normal file

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@ -0,0 +1 @@
{"bos_token": "<|endoftext|>", "eos_token": "<|endoftext|>", "unk_token": "<|endoftext|>"}

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@ -0,0 +1 @@
{"unk_token": "<|endoftext|>", "bos_token": "<|endoftext|>", "eos_token": "<|endoftext|>", "add_prefix_space": false, "model_max_length": 1024, "special_tokens_map_file": null, "name_or_path": "gpt2", "tokenizer_class": "GPT2Tokenizer"}

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whisper/assets/mel_filters.npz Normal file → Executable file

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@ -0,0 +1 @@
{"<|endoftext|>": 50257}

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@ -0,0 +1 @@
{"bos_token": "<|endoftext|>", "eos_token": "<|endoftext|>", "unk_token": "<|endoftext|>"}

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@ -0,0 +1 @@
{"unk_token": {"content": "<|endoftext|>", "single_word": false, "lstrip": false, "rstrip": false, "normalized": true, "__type": "AddedToken"}, "bos_token": {"content": "<|endoftext|>", "single_word": false, "lstrip": false, "rstrip": false, "normalized": true, "__type": "AddedToken"}, "eos_token": {"content": "<|endoftext|>", "single_word": false, "lstrip": false, "rstrip": false, "normalized": true, "__type": "AddedToken"}, "add_prefix_space": false, "model_max_length": 1024, "special_tokens_map_file": null, "name_or_path": "multilingual", "errors": "replace", "tokenizer_class": "GPT2Tokenizer"}

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@ -1,8 +1,8 @@
import os
from functools import lru_cache
from subprocess import CalledProcessError, run
from typing import Optional, Union
from typing import Union
import ffmpeg
import numpy as np
import torch
import torch.nn.functional as F
@ -12,14 +12,11 @@ from .utils import exact_div
# hard-coded audio hyperparameters
SAMPLE_RATE = 16000
N_FFT = 400
N_MELS = 80
HOP_LENGTH = 160
CHUNK_LENGTH = 30
N_SAMPLES = CHUNK_LENGTH * SAMPLE_RATE # 480000 samples in a 30-second chunk
N_FRAMES = exact_div(N_SAMPLES, HOP_LENGTH) # 3000 frames in a mel spectrogram input
N_SAMPLES_PER_TOKEN = HOP_LENGTH * 2 # the initial convolutions has stride 2
FRAMES_PER_SECOND = exact_div(SAMPLE_RATE, HOP_LENGTH) # 10ms per audio frame
TOKENS_PER_SECOND = exact_div(SAMPLE_RATE, N_SAMPLES_PER_TOKEN) # 20ms per audio token
N_SAMPLES = CHUNK_LENGTH * SAMPLE_RATE # 480000: number of samples in a chunk
N_FRAMES = exact_div(N_SAMPLES, HOP_LENGTH) # 3000: number of frames in a mel spectrogram input
def load_audio(file: str, sr: int = SAMPLE_RATE):
@ -38,25 +35,15 @@ def load_audio(file: str, sr: int = SAMPLE_RATE):
-------
A NumPy array containing the audio waveform, in float32 dtype.
"""
# This launches a subprocess to decode audio while down-mixing
# and resampling as necessary. Requires the ffmpeg CLI in PATH.
# fmt: off
cmd = [
"ffmpeg",
"-nostdin",
"-threads", "0",
"-i", file,
"-f", "s16le",
"-ac", "1",
"-acodec", "pcm_s16le",
"-ar", str(sr),
"-"
]
# fmt: on
try:
out = run(cmd, capture_output=True, check=True).stdout
except CalledProcessError as e:
# This launches a subprocess to decode audio while down-mixing and resampling as necessary.
# Requires the ffmpeg CLI and `ffmpeg-python` package to be installed.
out, _ = (
ffmpeg.input(file, threads=0)
.output("-", format="s16le", acodec="pcm_s16le", ac=1, ar=sr)
.run(cmd=["ffmpeg", "-nostdin"], capture_stdout=True, capture_stderr=True)
)
except ffmpeg.Error as e:
raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e
return np.frombuffer(out, np.int16).flatten().astype(np.float32) / 32768.0
@ -68,9 +55,7 @@ def pad_or_trim(array, length: int = N_SAMPLES, *, axis: int = -1):
"""
if torch.is_tensor(array):
if array.shape[axis] > length:
array = array.index_select(
dim=axis, index=torch.arange(length, device=array.device)
)
array = array.index_select(dim=axis, index=torch.arange(length, device=array.device))
if array.shape[axis] < length:
pad_widths = [(0, 0)] * array.ndim
@ -89,7 +74,7 @@ def pad_or_trim(array, length: int = N_SAMPLES, *, axis: int = -1):
@lru_cache(maxsize=None)
def mel_filters(device, n_mels: int) -> torch.Tensor:
def mel_filters(device, n_mels: int = N_MELS) -> torch.Tensor:
"""
load the mel filterbank matrix for projecting STFT into a Mel spectrogram.
Allows decoupling librosa dependency; saved using:
@ -97,22 +82,14 @@ def mel_filters(device, n_mels: int) -> torch.Tensor:
np.savez_compressed(
"mel_filters.npz",
mel_80=librosa.filters.mel(sr=16000, n_fft=400, n_mels=80),
mel_128=librosa.filters.mel(sr=16000, n_fft=400, n_mels=128),
)
"""
assert n_mels in {80, 128}, f"Unsupported n_mels: {n_mels}"
filters_path = os.path.join(os.path.dirname(__file__), "assets", "mel_filters.npz")
with np.load(filters_path, allow_pickle=False) as f:
assert n_mels == 80, f"Unsupported n_mels: {n_mels}"
with np.load(os.path.join(os.path.dirname(__file__), "assets", "mel_filters.npz")) as f:
return torch.from_numpy(f[f"mel_{n_mels}"]).to(device)
def log_mel_spectrogram(
audio: Union[str, np.ndarray, torch.Tensor],
n_mels: int = 80,
padding: int = 0,
device: Optional[Union[str, torch.device]] = None,
):
def log_mel_spectrogram(audio: Union[str, np.ndarray, torch.Tensor], n_mels: int = N_MELS):
"""
Compute the log-Mel spectrogram of
@ -122,17 +99,11 @@ def log_mel_spectrogram(
The path to audio or either a NumPy array or Tensor containing the audio waveform in 16 kHz
n_mels: int
The number of Mel-frequency filters, only 80 and 128 are supported
padding: int
Number of zero samples to pad to the right
device: Optional[Union[str, torch.device]]
If given, the audio tensor is moved to this device before STFT
The number of Mel-frequency filters, only 80 is supported
Returns
-------
torch.Tensor, shape = (n_mels, n_frames)
torch.Tensor, shape = (80, n_frames)
A Tensor that contains the Mel spectrogram
"""
if not torch.is_tensor(audio):
@ -140,10 +111,6 @@ def log_mel_spectrogram(
audio = load_audio(audio)
audio = torch.from_numpy(audio)
if device is not None:
audio = audio.to(device)
if padding > 0:
audio = F.pad(audio, (0, padding))
window = torch.hann_window(N_FFT).to(audio.device)
stft = torch.stft(audio, N_FFT, HOP_LENGTH, window=window, return_complex=True)
magnitudes = stft[..., :-1].abs() ** 2

View File

@ -1,5 +1,5 @@
from dataclasses import dataclass, field, replace
from typing import TYPE_CHECKING, Dict, Iterable, List, Optional, Sequence, Tuple, Union
from dataclasses import dataclass, field
from typing import Dict, List, Tuple, Iterable, Optional, Sequence, Union, TYPE_CHECKING
import numpy as np
import torch
@ -16,9 +16,7 @@ if TYPE_CHECKING:
@torch.no_grad()
def detect_language(
model: "Whisper", mel: Tensor, tokenizer: Tokenizer = None
) -> Tuple[Tensor, List[dict]]:
def detect_language(model: "Whisper", mel: Tensor, tokenizer: Tokenizer = None) -> Tuple[Tensor, List[dict]]:
"""
Detect the spoken language in the audio, and return them as list of strings, along with the ids
of the most probable language tokens and the probability distribution over all language tokens.
@ -32,16 +30,9 @@ def detect_language(
list of dictionaries containing the probability distribution over all languages.
"""
if tokenizer is None:
tokenizer = get_tokenizer(
model.is_multilingual, num_languages=model.num_languages
)
if (
tokenizer.language is None
or tokenizer.language_token not in tokenizer.sot_sequence
):
raise ValueError(
"This model doesn't have language tokens so it can't perform lang id"
)
tokenizer = get_tokenizer(model.is_multilingual)
if tokenizer.language is None or tokenizer.language_token not in tokenizer.sot_sequence:
raise ValueError(f"This model doesn't have language tokens so it can't perform lang id")
single = mel.ndim == 2
if single:
@ -79,36 +70,31 @@ def detect_language(
@dataclass(frozen=True)
class DecodingOptions:
# whether to perform X->X "transcribe" or X->English "translate"
task: str = "transcribe"
# language that the audio is in; uses detected language if None
language: Optional[str] = None
task: str = "transcribe" # whether to perform X->X "transcribe" or X->English "translate"
language: Optional[str] = None # language that the audio is in; uses detected language if None
# sampling-related options
temperature: float = 0.0
sample_len: Optional[int] = None # maximum number of tokens to sample
best_of: Optional[int] = None # number of independent sample trajectories, if t > 0
beam_size: Optional[int] = None # number of beams in beam search, if t == 0
patience: Optional[float] = None # patience in beam search (arxiv:2204.05424)
best_of: Optional[int] = None # number of independent samples to collect, when t > 0
beam_size: Optional[int] = None # number of beams in beam search, when t == 0
patience: Optional[float] = None # patience in beam search (https://arxiv.org/abs/2204.05424)
# "alpha" in Google NMT, or None for length norm, when ranking generations
# to select which to return among the beams or best-of-N samples
length_penalty: Optional[float] = None
# options for ranking generations (either beams or best-of-N samples)
length_penalty: Optional[float] = None # "alpha" in Google NMT, None defaults to length norm
# text or tokens to feed as the prompt or the prefix; for more info:
# https://github.com/openai/whisper/discussions/117#discussioncomment-3727051
prompt: Optional[Union[str, List[int]]] = None # for the previous context
prefix: Optional[Union[str, List[int]]] = None # to prefix the current context
# prompt, prefix, and token suppression
prompt: Optional[Union[str, List[int]]] = None # text or tokens for the previous context
prefix: Optional[Union[str, List[int]]] = None # text or tokens to prefix the current context
suppress_blank: bool = True # this will suppress blank outputs
# list of tokens ids (or comma-separated token ids) to suppress
# "-1" will suppress a set of symbols as defined in `tokenizer.non_speech_tokens()`
suppress_tokens: Optional[Union[str, Iterable[int]]] = "-1"
suppress_blank: bool = True # this will suppress blank outputs
# timestamp sampling options
without_timestamps: bool = False # use <|notimestamps|> to sample text tokens only
max_initial_timestamp: Optional[float] = 1.0
without_timestamps: bool = False # use <|notimestamps|> to sample text tokens only
max_initial_timestamp: Optional[float] = 1.0 # the initial timestamp cannot be later than this
# implementation details
fp16: bool = True # use fp16 for most of the calculation
@ -148,10 +134,6 @@ class PyTorchInference(Inference):
self.kv_cache = {}
self.hooks = []
key_modules = [block.attn.key for block in self.model.decoder.blocks]
value_modules = [block.attn.value for block in self.model.decoder.blocks]
self.kv_modules = key_modules + value_modules
def logits(self, tokens: Tensor, audio_features: Tensor) -> Tensor:
if not self.kv_cache:
self.kv_cache, self.hooks = self.model.install_kv_cache_hooks()
@ -170,16 +152,13 @@ class PyTorchInference(Inference):
self.hooks = []
def rearrange_kv_cache(self, source_indices):
if source_indices != list(range(len(source_indices))):
for module in self.kv_modules:
# update the key/value cache to contain the selected sequences
self.kv_cache[module] = self.kv_cache[module][source_indices].detach()
for module, tensor in self.kv_cache.items():
# update the key/value cache to contain the selected sequences
self.kv_cache[module] = tensor[source_indices].detach()
class SequenceRanker:
def rank(
self, tokens: List[List[Tensor]], sum_logprobs: List[List[float]]
) -> List[int]:
def rank(self, tokens: List[List[Tensor]], sum_logprobs: List[List[float]]) -> List[int]:
"""
Given a list of groups of samples and their cumulative log probabilities,
return the indices of the samples in each group to select as the final result
@ -217,9 +196,7 @@ class TokenDecoder:
def reset(self):
"""Initialize any stateful variables for decoding a new sequence"""
def update(
self, tokens: Tensor, logits: Tensor, sum_logprobs: Tensor
) -> Tuple[Tensor, bool]:
def update(self, tokens: Tensor, logits: Tensor, sum_logprobs: Tensor) -> Tuple[Tensor, bool]:
"""Specify how to select the next token, based on the current trace and logits
Parameters
@ -274,13 +251,12 @@ class GreedyDecoder(TokenDecoder):
self.temperature = temperature
self.eot = eot
def update(
self, tokens: Tensor, logits: Tensor, sum_logprobs: Tensor
) -> Tuple[Tensor, bool]:
if self.temperature == 0:
def update(self, tokens: Tensor, logits: Tensor, sum_logprobs: Tensor) -> Tuple[Tensor, bool]:
temperature = self.temperature
if temperature == 0:
next_tokens = logits.argmax(dim=-1)
else:
next_tokens = Categorical(logits=logits / self.temperature).sample()
next_tokens = Categorical(logits=logits / temperature).sample()
logprobs = F.log_softmax(logits.float(), dim=-1)
current_logprobs = logprobs[torch.arange(logprobs.shape[0]), next_tokens]
@ -299,13 +275,7 @@ class GreedyDecoder(TokenDecoder):
class BeamSearchDecoder(TokenDecoder):
def __init__(
self,
beam_size: int,
eot: int,
inference: Inference,
patience: Optional[float] = None,
):
def __init__(self, beam_size: int, eot: int, inference: Inference, patience: Optional[float] = None):
self.beam_size = beam_size
self.eot = eot
self.inference = inference
@ -313,16 +283,12 @@ class BeamSearchDecoder(TokenDecoder):
self.max_candidates: int = round(beam_size * self.patience)
self.finished_sequences = None
assert (
self.max_candidates > 0
), f"Invalid beam size ({beam_size}) or patience ({patience})"
assert self.max_candidates > 0, f"Invalid beam size ({beam_size}) or patience ({patience})"
def reset(self):
self.finished_sequences = None
def update(
self, tokens: Tensor, logits: Tensor, sum_logprobs: Tensor
) -> Tuple[Tensor, bool]:
def update(self, tokens: Tensor, logits: Tensor, sum_logprobs: Tensor) -> Tuple[Tensor, bool]:
if tokens.shape[0] % self.beam_size != 0:
raise ValueError(f"{tokens.shape}[0] % {self.beam_size} != 0")
@ -366,9 +332,7 @@ class BeamSearchDecoder(TokenDecoder):
# add newly finished sequences to self.finished_sequences
assert len(self.finished_sequences) == len(finished_sequences)
for previously_finished, newly_finished in zip(
self.finished_sequences, finished_sequences
):
for previously_finished, newly_finished in zip(self.finished_sequences, finished_sequences):
for seq in sorted(newly_finished, key=newly_finished.get, reverse=True):
if len(previously_finished) >= self.max_candidates:
break # the candidate list is full
@ -376,8 +340,7 @@ class BeamSearchDecoder(TokenDecoder):
# mark as completed if all audio has enough number of samples
completed = all(
len(sequences) >= self.max_candidates
for sequences in self.finished_sequences
len(sequences) >= self.max_candidates for sequences in self.finished_sequences
)
return tokens, completed
@ -385,9 +348,7 @@ class BeamSearchDecoder(TokenDecoder):
# collect all finished sequences, including patience, and add unfinished ones if not enough
sum_logprobs = sum_logprobs.cpu()
for i, sequences in enumerate(self.finished_sequences):
if (
len(sequences) < self.beam_size
): # when not enough sequences are finished
if len(sequences) < self.beam_size: # when not enough sequences are finished
for j in list(np.argsort(sum_logprobs[i]))[::-1]:
sequence = preceding_tokens[i, j].tolist() + [self.eot]
sequences[tuple(sequence)] = sum_logprobs[i][j].item()
@ -395,8 +356,7 @@ class BeamSearchDecoder(TokenDecoder):
break
tokens: List[List[Tensor]] = [
[torch.tensor(seq) for seq in sequences.keys()]
for sequences in self.finished_sequences
[torch.tensor(seq) for seq in sequences.keys()] for sequences in self.finished_sequences
]
sum_logprobs: List[List[float]] = [
list(sequences.values()) for sequences in self.finished_sequences
@ -440,10 +400,7 @@ class SuppressTokens(LogitFilter):
class ApplyTimestampRules(LogitFilter):
def __init__(
self,
tokenizer: Tokenizer,
sample_begin: int,
max_initial_timestamp_index: Optional[int],
self, tokenizer: Tokenizer, sample_begin: int, max_initial_timestamp_index: Optional[int]
):
self.tokenizer = tokenizer
self.sample_begin = sample_begin
@ -456,14 +413,9 @@ class ApplyTimestampRules(LogitFilter):
# timestamps have to appear in pairs, except directly before EOT; mask logits accordingly
for k in range(tokens.shape[0]):
sampled_tokens = tokens[k, self.sample_begin :]
seq = [t for t in sampled_tokens.tolist()]
last_was_timestamp = (
len(seq) >= 1 and seq[-1] >= self.tokenizer.timestamp_begin
)
penultimate_was_timestamp = (
len(seq) < 2 or seq[-2] >= self.tokenizer.timestamp_begin
)
seq = [t for t in tokens[k, self.sample_begin :].tolist()]
last_was_timestamp = len(seq) >= 1 and seq[-1] >= self.tokenizer.timestamp_begin
penultimate_was_timestamp = len(seq) < 2 or seq[-2] >= self.tokenizer.timestamp_begin
if last_was_timestamp:
if penultimate_was_timestamp: # has to be non-timestamp
@ -471,35 +423,19 @@ class ApplyTimestampRules(LogitFilter):
else: # cannot be normal text tokens
logits[k, : self.tokenizer.eot] = -np.inf
timestamps = sampled_tokens[
sampled_tokens.ge(self.tokenizer.timestamp_begin)
]
if timestamps.numel() > 0:
# timestamps shouldn't decrease; forbid timestamp tokens smaller than the last
# also force each segment to have a nonzero length, to prevent infinite looping
if last_was_timestamp and not penultimate_was_timestamp:
timestamp_last = timestamps[-1]
else:
timestamp_last = timestamps[-1] + 1
logits[k, self.tokenizer.timestamp_begin : timestamp_last] = -np.inf
if tokens.shape[1] == self.sample_begin:
# suppress generating non-timestamp tokens at the beginning
logits[:, : self.tokenizer.timestamp_begin] = -np.inf
# apply the `max_initial_timestamp` option
if self.max_initial_timestamp_index is not None:
last_allowed = (
self.tokenizer.timestamp_begin + self.max_initial_timestamp_index
)
last_allowed = self.tokenizer.timestamp_begin + self.max_initial_timestamp_index
logits[:, last_allowed + 1 :] = -np.inf
# if sum of probability over timestamps is above any other token, sample timestamp
logprobs = F.log_softmax(logits.float(), dim=-1)
for k in range(tokens.shape[0]):
timestamp_logprob = logprobs[k, self.tokenizer.timestamp_begin :].logsumexp(
dim=-1
)
timestamp_logprob = logprobs[k, self.tokenizer.timestamp_begin :].logsumexp(dim=-1)
max_text_token_logprob = logprobs[k, : self.tokenizer.timestamp_begin].max()
if timestamp_logprob > max_text_token_logprob:
logits[k, : self.tokenizer.timestamp_begin] = -np.inf
@ -515,12 +451,7 @@ class DecodingTask:
self.model = model
language = options.language or "en"
tokenizer = get_tokenizer(
model.is_multilingual,
num_languages=model.num_languages,
language=language,
task=options.task,
)
tokenizer = get_tokenizer(model.is_multilingual, language=language, task=options.task)
self.tokenizer: Tokenizer = tokenizer
self.options: DecodingOptions = self._verify_options(options)
@ -560,13 +491,9 @@ class DecodingTask:
precision = CHUNK_LENGTH / model.dims.n_audio_ctx # usually 0.02 seconds
max_initial_timestamp_index = None
if options.max_initial_timestamp:
max_initial_timestamp_index = round(
self.options.max_initial_timestamp / precision
)
max_initial_timestamp_index = round(self.options.max_initial_timestamp / precision)
self.logit_filters.append(
ApplyTimestampRules(
tokenizer, self.sample_begin, max_initial_timestamp_index
)
ApplyTimestampRules(tokenizer, self.sample_begin, max_initial_timestamp_index)
)
def _verify_options(self, options: DecodingOptions) -> DecodingOptions:
@ -577,38 +504,30 @@ class DecodingTask:
raise ValueError("best_of with greedy sampling (T=0) is not compatible")
if options.patience is not None and options.beam_size is None:
raise ValueError("patience requires beam_size to be given")
if options.length_penalty is not None and not (
0 <= options.length_penalty <= 1
):
if options.length_penalty is not None and not (0 <= options.length_penalty <= 1):
raise ValueError("length_penalty (alpha) should be a value between 0 and 1")
return options
def _get_initial_tokens(self) -> Tuple[int]:
tokens = list(self.sot_sequence)
prefix = self.options.prefix
prompt = self.options.prompt
if prefix := self.options.prefix:
if prefix:
prefix_tokens = (
self.tokenizer.encode(" " + prefix.strip())
if isinstance(prefix, str)
else prefix
self.tokenizer.encode(" " + prefix.strip()) if isinstance(prefix, str) else prefix
)
if self.sample_len is not None:
max_prefix_len = self.n_ctx // 2 - self.sample_len
prefix_tokens = prefix_tokens[-max_prefix_len:]
tokens = tokens + prefix_tokens
if prompt := self.options.prompt:
if prompt:
prompt_tokens = (
self.tokenizer.encode(" " + prompt.strip())
if isinstance(prompt, str)
else prompt
)
tokens = (
[self.tokenizer.sot_prev]
+ prompt_tokens[-(self.n_ctx // 2 - 1) :]
+ tokens
self.tokenizer.encode(" " + prompt.strip()) if isinstance(prompt, str) else prompt
)
tokens = [self.tokenizer.sot_prev] + prompt_tokens[-(self.n_ctx // 2 - 1) :] + tokens
return tuple(tokens)
@ -627,13 +546,7 @@ class DecodingTask:
assert isinstance(suppress_tokens, list), "suppress_tokens must be a list"
suppress_tokens.extend(
[
self.tokenizer.transcribe,
self.tokenizer.translate,
self.tokenizer.sot,
self.tokenizer.sot_prev,
self.tokenizer.sot_lm,
]
[self.tokenizer.sot, self.tokenizer.sot_prev, self.tokenizer.sot_lm]
)
if self.tokenizer.no_speech is not None:
# no-speech probability is collected separately
@ -645,21 +558,14 @@ class DecodingTask:
if self.options.fp16:
mel = mel.half()
if mel.shape[-2:] == (
self.model.dims.n_audio_ctx,
self.model.dims.n_audio_state,
):
if mel.shape[-2:] == (self.model.dims.n_audio_ctx, self.model.dims.n_audio_state):
# encoded audio features are given; skip audio encoding
audio_features = mel
else:
audio_features = self.model.encoder(mel)
if audio_features.dtype != (
torch.float16 if self.options.fp16 else torch.float32
):
return TypeError(
f"audio_features has an incorrect dtype: {audio_features.dtype}"
)
if audio_features.dtype != (torch.float16 if self.options.fp16 else torch.float32):
return TypeError(f"audio_features has an incorrect dtype: {audio_features.dtype}")
return audio_features
@ -668,9 +574,7 @@ class DecodingTask:
lang_probs = None
if self.options.language is None or self.options.task == "lang_id":
lang_tokens, lang_probs = self.model.detect_language(
audio_features, self.tokenizer
)
lang_tokens, lang_probs = self.model.detect_language(audio_features, self.tokenizer)
languages = [max(probs, key=probs.get) for probs in lang_probs]
if self.options.language is None:
tokens[:, self.sot_index + 1] = lang_tokens # write language tokens
@ -678,6 +582,7 @@ class DecodingTask:
return languages, lang_probs
def _main_loop(self, audio_features: Tensor, tokens: Tensor):
assert audio_features.shape[0] == tokens.shape[0]
n_batch = tokens.shape[0]
sum_logprobs: Tensor = torch.zeros(n_batch, device=audio_features.device)
no_speech_probs = [np.nan] * n_batch
@ -686,9 +591,7 @@ class DecodingTask:
for i in range(self.sample_len):
logits = self.inference.logits(tokens, audio_features)
if (
i == 0 and self.tokenizer.no_speech is not None
): # save no_speech_probs
if i == 0 and self.tokenizer.no_speech is not None: # save no_speech_probs
probs_at_sot = logits[:, self.sot_index].float().softmax(dim=-1)
no_speech_probs = probs_at_sot[:, self.tokenizer.no_speech].tolist()
@ -722,15 +625,12 @@ class DecodingTask:
languages, language_probs = self._detect_language(audio_features, tokens)
if self.options.task == "lang_id":
return [
DecodingResult(
audio_features=features, language=language, language_probs=probs
)
for features, language, probs in zip(
audio_features, languages, language_probs
)
DecodingResult(audio_features=features, language=language, language_probs=probs)
for features, language, probs in zip(audio_features, languages, language_probs)
]
# repeat text tensors by the group size, for beam search or best-of-n sampling
# repeat the audio & text tensors by the group size, for beam search or best-of-n sampling
audio_features = audio_features.repeat_interleave(self.n_group, dim=0)
tokens = tokens.repeat_interleave(self.n_group, dim=0).to(audio_features.device)
# call the main sampling loop
@ -747,8 +647,7 @@ class DecodingTask:
# get the final candidates for each group, and slice between the first sampled token and EOT
tokens, sum_logprobs = self.decoder.finalize(tokens, sum_logprobs)
tokens: List[List[Tensor]] = [
[t[self.sample_begin : (t == tokenizer.eot).nonzero()[0, 0]] for t in s]
for s in tokens
[t[self.sample_begin : (t == tokenizer.eot).nonzero()[0, 0]] for t in s] for s in tokens
]
# select the top-ranked sample in each group
@ -757,18 +656,9 @@ class DecodingTask:
texts: List[str] = [tokenizer.decode(t).strip() for t in tokens]
sum_logprobs: List[float] = [lp[i] for i, lp in zip(selected, sum_logprobs)]
avg_logprobs: List[float] = [
lp / (len(t) + 1) for t, lp in zip(tokens, sum_logprobs)
]
avg_logprobs: List[float] = [lp / (len(t) + 1) for t, lp in zip(tokens, sum_logprobs)]
fields = (
texts,
languages,
tokens,
audio_features,
avg_logprobs,
no_speech_probs,
)
fields = (texts, languages, tokens, audio_features, avg_logprobs, no_speech_probs)
if len(set(map(len, fields))) != 1:
raise RuntimeError(f"inconsistent result lengths: {list(map(len, fields))}")
@ -783,19 +673,12 @@ class DecodingTask:
temperature=self.options.temperature,
compression_ratio=compression_ratio(text),
)
for text, language, tokens, features, avg_logprob, no_speech_prob in zip(
*fields
)
for text, language, tokens, features, avg_logprob, no_speech_prob in zip(*fields)
]
@torch.no_grad()
def decode(
model: "Whisper",
mel: Tensor,
options: DecodingOptions = DecodingOptions(),
**kwargs,
) -> Union[DecodingResult, List[DecodingResult]]:
def decode(model: "Whisper", mel: Tensor, options: DecodingOptions = DecodingOptions()) -> Union[DecodingResult, List[DecodingResult]]:
"""
Performs decoding of 30-second audio segment(s), provided as Mel spectrogram(s).
@ -815,12 +698,13 @@ def decode(
result: Union[DecodingResult, List[DecodingResult]]
The result(s) of decoding contained in `DecodingResult` dataclass instance(s)
"""
if single := mel.ndim == 2:
single = mel.ndim == 2
if single:
mel = mel.unsqueeze(0)
if kwargs:
options = replace(options, **kwargs)
result = DecodingTask(model, options).run(mel)
if single:
result = result[0]
return result[0] if single else result
return result

View File

@ -1,25 +1,15 @@
import base64
import gzip
from contextlib import contextmanager
from dataclasses import dataclass
from typing import Dict, Iterable, Optional, Tuple
from typing import Dict
from typing import Iterable, Optional
import numpy as np
import torch
import torch.nn.functional as F
from torch import Tensor, nn
from torch import Tensor
from torch import nn
from .decoding import decode as decode_function
from .decoding import detect_language as detect_language_function
from .transcribe import transcribe as transcribe_function
try:
from torch.nn.functional import scaled_dot_product_attention
SDPA_AVAILABLE = True
except (ImportError, RuntimeError, OSError):
scaled_dot_product_attention = None
SDPA_AVAILABLE = False
from .decoding import detect_language as detect_language_function, decode as decode_function
@dataclass
@ -44,16 +34,12 @@ class LayerNorm(nn.LayerNorm):
class Linear(nn.Linear):
def forward(self, x: Tensor) -> Tensor:
return F.linear(
x,
self.weight.to(x.dtype),
None if self.bias is None else self.bias.to(x.dtype),
x, self.weight.to(x.dtype), None if self.bias is None else self.bias.to(x.dtype)
)
class Conv1d(nn.Conv1d):
def _conv_forward(
self, x: Tensor, weight: Tensor, bias: Optional[Tensor]
) -> Tensor:
def _conv_forward(self, x: Tensor, weight: Tensor, bias: Optional[Tensor]) -> Tensor:
return super()._conv_forward(
x, weight.to(x.dtype), None if bias is None else bias.to(x.dtype)
)
@ -68,19 +54,7 @@ def sinusoids(length, channels, max_timescale=10000):
return torch.cat([torch.sin(scaled_time), torch.cos(scaled_time)], dim=1)
@contextmanager
def disable_sdpa():
prev_state = MultiHeadAttention.use_sdpa
try:
MultiHeadAttention.use_sdpa = False
yield
finally:
MultiHeadAttention.use_sdpa = prev_state
class MultiHeadAttention(nn.Module):
use_sdpa = True
def __init__(self, n_state: int, n_head: int):
super().__init__()
self.n_head = n_head
@ -111,32 +85,20 @@ class MultiHeadAttention(nn.Module):
wv, qk = self.qkv_attention(q, k, v, mask)
return self.out(wv), qk
def qkv_attention(
self, q: Tensor, k: Tensor, v: Tensor, mask: Optional[Tensor] = None
) -> Tuple[torch.Tensor, Optional[torch.Tensor]]:
def qkv_attention(self, q: Tensor, k: Tensor, v: Tensor, mask: Optional[Tensor] = None):
n_batch, n_ctx, n_state = q.shape
scale = (n_state // self.n_head) ** -0.25
q = q.view(*q.shape[:2], self.n_head, -1).permute(0, 2, 1, 3)
k = k.view(*k.shape[:2], self.n_head, -1).permute(0, 2, 1, 3)
q = q.view(*q.shape[:2], self.n_head, -1).permute(0, 2, 1, 3) * scale
k = k.view(*k.shape[:2], self.n_head, -1).permute(0, 2, 3, 1) * scale
v = v.view(*v.shape[:2], self.n_head, -1).permute(0, 2, 1, 3)
if SDPA_AVAILABLE and MultiHeadAttention.use_sdpa:
a = scaled_dot_product_attention(
q, k, v, is_causal=mask is not None and n_ctx > 1
)
out = a.permute(0, 2, 1, 3).flatten(start_dim=2)
qk = None
else:
qk = (q * scale) @ (k * scale).transpose(-1, -2)
if mask is not None:
qk = qk + mask[:n_ctx, :n_ctx]
qk = qk.float()
qk = q @ k
if mask is not None:
qk = qk + mask[:n_ctx, :n_ctx]
qk = qk.float()
w = F.softmax(qk, dim=-1).to(q.dtype)
out = (w @ v).permute(0, 2, 1, 3).flatten(start_dim=2)
qk = qk.detach()
return out, qk
w = F.softmax(qk, dim=-1).to(q.dtype)
return (w @ v).permute(0, 2, 1, 3).flatten(start_dim=2), qk.detach()
class ResidualAttentionBlock(nn.Module):
@ -146,15 +108,11 @@ class ResidualAttentionBlock(nn.Module):
self.attn = MultiHeadAttention(n_state, n_head)
self.attn_ln = LayerNorm(n_state)
self.cross_attn = (
MultiHeadAttention(n_state, n_head) if cross_attention else None
)
self.cross_attn = MultiHeadAttention(n_state, n_head) if cross_attention else None
self.cross_attn_ln = LayerNorm(n_state) if cross_attention else None
n_mlp = n_state * 4
self.mlp = nn.Sequential(
Linear(n_state, n_mlp), nn.GELU(), Linear(n_mlp, n_state)
)
self.mlp = nn.Sequential(Linear(n_state, n_mlp), nn.GELU(), Linear(n_mlp, n_state))
self.mlp_ln = LayerNorm(n_state)
def forward(
@ -172,9 +130,7 @@ class ResidualAttentionBlock(nn.Module):
class AudioEncoder(nn.Module):
def __init__(
self, n_mels: int, n_ctx: int, n_state: int, n_head: int, n_layer: int
):
def __init__(self, n_mels: int, n_ctx: int, n_state: int, n_head: int, n_layer: int):
super().__init__()
self.conv1 = Conv1d(n_mels, n_state, kernel_size=3, padding=1)
self.conv2 = Conv1d(n_state, n_state, kernel_size=3, stride=2, padding=1)
@ -205,19 +161,14 @@ class AudioEncoder(nn.Module):
class TextDecoder(nn.Module):
def __init__(
self, n_vocab: int, n_ctx: int, n_state: int, n_head: int, n_layer: int
):
def __init__(self, n_vocab: int, n_ctx: int, n_state: int, n_head: int, n_layer: int):
super().__init__()
self.token_embedding = nn.Embedding(n_vocab, n_state)
self.positional_embedding = nn.Parameter(torch.empty(n_ctx, n_state))
self.blocks: Iterable[ResidualAttentionBlock] = nn.ModuleList(
[
ResidualAttentionBlock(n_state, n_head, cross_attention=True)
for _ in range(n_layer)
]
[ResidualAttentionBlock(n_state, n_head, cross_attention=True) for _ in range(n_layer)]
)
self.ln = LayerNorm(n_state)
@ -228,23 +179,18 @@ class TextDecoder(nn.Module):
"""
x : torch.LongTensor, shape = (batch_size, <= n_ctx)
the text tokens
xa : torch.Tensor, shape = (batch_size, n_audio_ctx, n_audio_state)
xa : torch.Tensor, shape = (batch_size, n_mels, n_audio_ctx)
the encoded audio features to be attended on
"""
offset = next(iter(kv_cache.values())).shape[1] if kv_cache else 0
x = (
self.token_embedding(x)
+ self.positional_embedding[offset : offset + x.shape[-1]]
)
x = self.token_embedding(x) + self.positional_embedding[offset : offset + x.shape[-1]]
x = x.to(xa.dtype)
for block in self.blocks:
x = block(x, xa, mask=self.mask, kv_cache=kv_cache)
x = self.ln(x)
logits = (
x @ torch.transpose(self.token_embedding.weight.to(x.dtype), 0, 1)
).float()
logits = (x @ torch.transpose(self.token_embedding.weight.to(x.dtype), 0, 1)).float()
return logits
@ -267,22 +213,6 @@ class Whisper(nn.Module):
self.dims.n_text_head,
self.dims.n_text_layer,
)
# use the last half among the decoder layers for time alignment by default;
# to use a specific set of heads, see `set_alignment_heads()` below.
all_heads = torch.zeros(
self.dims.n_text_layer, self.dims.n_text_head, dtype=torch.bool
)
all_heads[self.dims.n_text_layer // 2 :] = True
self.register_buffer("alignment_heads", all_heads.to_sparse(), persistent=False)
def set_alignment_heads(self, dump: bytes):
array = np.frombuffer(
gzip.decompress(base64.b85decode(dump)), dtype=bool
).copy()
mask = torch.from_numpy(array).reshape(
self.dims.n_text_layer, self.dims.n_text_head
)
self.register_buffer("alignment_heads", mask.to_sparse(), persistent=False)
def embed_audio(self, mel: torch.Tensor):
return self.encoder(mel)
@ -290,9 +220,7 @@ class Whisper(nn.Module):
def logits(self, tokens: torch.Tensor, audio_features: torch.Tensor):
return self.decoder(tokens, audio_features)
def forward(
self, mel: torch.Tensor, tokens: torch.Tensor
) -> Dict[str, torch.Tensor]:
def forward(self, mel: torch.Tensor, tokens: torch.Tensor) -> Dict[str, torch.Tensor]:
return self.decoder(tokens, self.encoder(mel))
@property
@ -301,11 +229,7 @@ class Whisper(nn.Module):
@property
def is_multilingual(self):
return self.dims.n_vocab >= 51865
@property
def num_languages(self):
return self.dims.n_vocab - 51765 - int(self.is_multilingual)
return self.dims.n_vocab == 51865
def install_kv_cache_hooks(self, cache: Optional[dict] = None):
"""
@ -325,9 +249,8 @@ class Whisper(nn.Module):
hooks = []
def save_to_cache(module, _, output):
if module not in cache or output.shape[1] > self.dims.n_text_ctx:
# save as-is, for the first token or cross attention
cache[module] = output
if module not in cache or output.shape[1] > self.decoder.positional_embedding.shape[0]:
cache[module] = output # save as-is, for the first token or cross attention
else:
cache[module] = torch.cat([cache[module], output], dim=1).detach()
return cache[module]

View File

@ -1,2 +1,2 @@
from .basic import BasicTextNormalizer as BasicTextNormalizer
from .english import EnglishTextNormalizer as EnglishTextNormalizer
from .basic import BasicTextNormalizer
from .english import EnglishTextNormalizer

View File

@ -30,19 +30,15 @@ def remove_symbols_and_diacritics(s: str, keep=""):
and drop any diacritics (category 'Mn' and some manual mappings)
"""
return "".join(
(
c
if c in keep
else (
ADDITIONAL_DIACRITICS[c]
if c in ADDITIONAL_DIACRITICS
else (
""
if unicodedata.category(c) == "Mn"
else " " if unicodedata.category(c)[0] in "MSP" else c
)
)
)
c
if c in keep
else ADDITIONAL_DIACRITICS[c]
if c in ADDITIONAL_DIACRITICS
else ""
if unicodedata.category(c) == "Mn"
else " "
if unicodedata.category(c)[0] in "MSP"
else c
for c in unicodedata.normalize("NFKD", s)
)
@ -52,16 +48,13 @@ def remove_symbols(s: str):
Replace any other markers, symbols, punctuations with a space, keeping diacritics
"""
return "".join(
" " if unicodedata.category(c)[0] in "MSP" else c
for c in unicodedata.normalize("NFKC", s)
" " if unicodedata.category(c)[0] in "MSP" else c for c in unicodedata.normalize("NFKC", s)
)
class BasicTextNormalizer:
def __init__(self, remove_diacritics: bool = False, split_letters: bool = False):
self.clean = (
remove_symbols_and_diacritics if remove_diacritics else remove_symbols
)
self.clean = remove_symbols_and_diacritics if remove_diacritics else remove_symbols
self.split_letters = split_letters
def __call__(self, s: str):
@ -73,8 +66,6 @@ class BasicTextNormalizer:
if self.split_letters:
s = " ".join(regex.findall(r"\X", s, regex.U))
s = re.sub(
r"\s+", " ", s
) # replace any successive whitespace characters with a space
s = re.sub(r"\s+", " ", s) # replace any successive whitespace characters with a space
return s

View File

@ -1737,5 +1737,6 @@
"yoghurt": "yogurt",
"yoghurts": "yogurts",
"mhm": "hmm",
"mm": "hmm",
"mmm": "hmm"
}

View File

@ -84,8 +84,7 @@ class EnglishNumberNormalizer:
name.replace("y", "ies"): (value, "s") for name, value in self.tens.items()
}
self.tens_ordinal = {
name.replace("y", "ieth"): (value, "th")
for name, value in self.tens.items()
name.replace("y", "ieth"): (value, "th") for name, value in self.tens.items()
}
self.tens_suffixed = {**self.tens_plural, **self.tens_ordinal}
@ -109,10 +108,7 @@ class EnglishNumberNormalizer:
self.multipliers_ordinal = {
name + "th": (value, "th") for name, value in self.multipliers.items()
}
self.multipliers_suffixed = {
**self.multipliers_plural,
**self.multipliers_ordinal,
}
self.multipliers_suffixed = {**self.multipliers_plural, **self.multipliers_ordinal}
self.decimals = {*self.ones, *self.tens, *self.zeros}
self.preceding_prefixers = {
@ -132,8 +128,7 @@ class EnglishNumberNormalizer:
"cents": "¢",
}
self.prefixes = set(
list(self.preceding_prefixers.values())
+ list(self.following_prefixers.values())
list(self.preceding_prefixers.values()) + list(self.following_prefixers.values())
)
self.suffixers = {
"per": {"cent": "%"},
@ -223,9 +218,7 @@ class EnglishNumberNormalizer:
if value is None:
value = ones
elif isinstance(value, str) or prev in self.ones:
if (
prev in self.tens and ones < 10
): # replace the last zero with the digit
if prev in self.tens and ones < 10: # replace the last zero with the digit
assert value[-1] == "0"
value = value[:-1] + str(ones)
else:
@ -529,14 +522,14 @@ class EnglishTextNormalizer:
s = re.sub(r"[<\[][^>\]]*[>\]]", "", s) # remove words between brackets
s = re.sub(r"\(([^)]+?)\)", "", s) # remove words between parenthesis
s = re.sub(self.ignore_patterns, "", s)
s = re.sub(r"\s+'", "'", s) # when there's a space before an apostrophe
s = re.sub(r"\s+'", "'", s) # standardize when there's a space before an apostrophe
for pattern, replacement in self.replacers.items():
s = re.sub(pattern, replacement, s)
s = re.sub(r"(\d),(\d)", r"\1\2", s) # remove commas between digits
s = re.sub(r"\.([^0-9]|$)", r" \1", s) # remove periods not followed by numbers
s = remove_symbols_and_diacritics(s, keep=".%$¢€£") # keep numeric symbols
s = remove_symbols_and_diacritics(s, keep=".%$¢€£") # keep some symbols for numerics
s = self.standardize_numbers(s)
s = self.standardize_spellings(s)
@ -545,6 +538,6 @@ class EnglishTextNormalizer:
s = re.sub(r"[.$¢€£]([^0-9])", r" \1", s)
s = re.sub(r"([^0-9])%", r"\1 ", s)
s = re.sub(r"\s+", " ", s) # replace any successive whitespaces with a space
s = re.sub(r"\s+", " ", s) # replace any successive whitespace characters with a space
return s

View File

@ -1,388 +0,0 @@
import itertools
import subprocess
import warnings
from dataclasses import dataclass
from typing import TYPE_CHECKING, List
import numba
import numpy as np
import torch
import torch.nn.functional as F
from .audio import HOP_LENGTH, SAMPLE_RATE, TOKENS_PER_SECOND
from .tokenizer import Tokenizer
if TYPE_CHECKING:
from .model import Whisper
def median_filter(x: torch.Tensor, filter_width: int):
"""Apply a median filter of width `filter_width` along the last dimension of `x`"""
pad_width = filter_width // 2
if x.shape[-1] <= pad_width:
# F.pad requires the padding width to be smaller than the input dimension
return x
if (ndim := x.ndim) <= 2:
# `F.pad` does not support 1D or 2D inputs for reflect padding but supports 3D and 4D
x = x[None, None, :]
assert (
filter_width > 0 and filter_width % 2 == 1
), "`filter_width` should be an odd number"
result = None
x = F.pad(x, (filter_width // 2, filter_width // 2, 0, 0), mode="reflect")
if x.is_cuda:
try:
from .triton_ops import median_filter_cuda
result = median_filter_cuda(x, filter_width)
except (RuntimeError, subprocess.CalledProcessError):
warnings.warn(
"Failed to launch Triton kernels, likely due to missing CUDA toolkit; "
"falling back to a slower median kernel implementation..."
)
if result is None:
# sort() is faster than torch.median (https://github.com/pytorch/pytorch/issues/51450)
result = x.unfold(-1, filter_width, 1).sort()[0][..., filter_width // 2]
if ndim <= 2:
result = result[0, 0]
return result
@numba.jit(nopython=True)
def backtrace(trace: np.ndarray):
i = trace.shape[0] - 1
j = trace.shape[1] - 1
trace[0, :] = 2
trace[:, 0] = 1
result = []
while i > 0 or j > 0:
result.append((i - 1, j - 1))
if trace[i, j] == 0:
i -= 1
j -= 1
elif trace[i, j] == 1:
i -= 1
elif trace[i, j] == 2:
j -= 1
else:
raise ValueError("Unexpected trace[i, j]")
result = np.array(result)
return result[::-1, :].T
@numba.jit(nopython=True, parallel=True)
def dtw_cpu(x: np.ndarray):
N, M = x.shape
cost = np.ones((N + 1, M + 1), dtype=np.float32) * np.inf
trace = -np.ones((N + 1, M + 1), dtype=np.float32)
cost[0, 0] = 0
for j in range(1, M + 1):
for i in range(1, N + 1):
c0 = cost[i - 1, j - 1]
c1 = cost[i - 1, j]
c2 = cost[i, j - 1]
if c0 < c1 and c0 < c2:
c, t = c0, 0
elif c1 < c0 and c1 < c2:
c, t = c1, 1
else:
c, t = c2, 2
cost[i, j] = x[i - 1, j - 1] + c
trace[i, j] = t
return backtrace(trace)
def dtw_cuda(x, BLOCK_SIZE=1024):
from .triton_ops import dtw_kernel
M, N = x.shape
assert M < BLOCK_SIZE, f"M should be smaller than {BLOCK_SIZE=}"
x_skew = (
F.pad(x, (0, M + 1), value=np.inf).flatten()[: M * (N + M)].reshape(M, N + M)
)
x_skew = x_skew.T.contiguous()
cost = torch.ones(N + M + 2, M + 2) * np.inf
cost[0, 0] = 0
cost = cost.to(x.device)
trace = torch.zeros_like(cost, dtype=torch.int32)
dtw_kernel[(1,)](
cost,
trace,
x_skew,
x_skew.stride(0),
cost.stride(0),
trace.stride(0),
N,
M,
BLOCK_SIZE=BLOCK_SIZE,
)
trace = trace.T.flatten()[: (M + 1) * (M + N + 3)].reshape(M + 1, M + N + 3)[
:, : N + 1
]
return backtrace(trace.cpu().numpy())
def dtw(x: torch.Tensor) -> np.ndarray:
if x.is_cuda:
try:
return dtw_cuda(x)
except (RuntimeError, subprocess.CalledProcessError):
warnings.warn(
"Failed to launch Triton kernels, likely due to missing CUDA toolkit; "
"falling back to a slower DTW implementation..."
)
return dtw_cpu(x.double().cpu().numpy())
@dataclass
class WordTiming:
word: str
tokens: List[int]
start: float
end: float
probability: float
def find_alignment(
model: "Whisper",
tokenizer: Tokenizer,
text_tokens: List[int],
mel: torch.Tensor,
num_frames: int,
*,
medfilt_width: int = 7,
qk_scale: float = 1.0,
) -> List[WordTiming]:
if len(text_tokens) == 0:
return []
tokens = torch.tensor(
[
*tokenizer.sot_sequence,
tokenizer.no_timestamps,
*text_tokens,
tokenizer.eot,
]
).to(model.device)
# install hooks on the cross attention layers to retrieve the attention weights
QKs = [None] * model.dims.n_text_layer
hooks = [
block.cross_attn.register_forward_hook(
lambda _, ins, outs, index=i: QKs.__setitem__(index, outs[-1][0])
)
for i, block in enumerate(model.decoder.blocks)
]
from .model import disable_sdpa
with torch.no_grad(), disable_sdpa():
logits = model(mel.unsqueeze(0), tokens.unsqueeze(0))[0]
sampled_logits = logits[len(tokenizer.sot_sequence) :, : tokenizer.eot]
token_probs = sampled_logits.softmax(dim=-1)
text_token_probs = token_probs[np.arange(len(text_tokens)), text_tokens]
text_token_probs = text_token_probs.tolist()
for hook in hooks:
hook.remove()
# heads * tokens * frames
weights = torch.stack([QKs[_l][_h] for _l, _h in model.alignment_heads.indices().T])
weights = weights[:, :, : num_frames // 2]
weights = (weights * qk_scale).softmax(dim=-1)
std, mean = torch.std_mean(weights, dim=-2, keepdim=True, unbiased=False)
weights = (weights - mean) / std
weights = median_filter(weights, medfilt_width)
matrix = weights.mean(axis=0)
matrix = matrix[len(tokenizer.sot_sequence) : -1]
text_indices, time_indices = dtw(-matrix)
words, word_tokens = tokenizer.split_to_word_tokens(text_tokens + [tokenizer.eot])
if len(word_tokens) <= 1:
# return on eot only
# >>> np.pad([], (1, 0))
# array([0.])
# This results in crashes when we lookup jump_times with float, like
# IndexError: arrays used as indices must be of integer (or boolean) type
return []
word_boundaries = np.pad(np.cumsum([len(t) for t in word_tokens[:-1]]), (1, 0))
jumps = np.pad(np.diff(text_indices), (1, 0), constant_values=1).astype(bool)
jump_times = time_indices[jumps] / TOKENS_PER_SECOND
start_times = jump_times[word_boundaries[:-1]]
end_times = jump_times[word_boundaries[1:]]
word_probabilities = [
np.mean(text_token_probs[i:j])
for i, j in zip(word_boundaries[:-1], word_boundaries[1:])
]
return [
WordTiming(word, tokens, start, end, probability)
for word, tokens, start, end, probability in zip(
words, word_tokens, start_times, end_times, word_probabilities
)
]
def merge_punctuations(alignment: List[WordTiming], prepended: str, appended: str):
# merge prepended punctuations
i = len(alignment) - 2
j = len(alignment) - 1
while i >= 0:
previous = alignment[i]
following = alignment[j]
if previous.word.startswith(" ") and previous.word.strip() in prepended:
# prepend it to the following word
following.word = previous.word + following.word
following.tokens = previous.tokens + following.tokens
previous.word = ""
previous.tokens = []
else:
j = i
i -= 1
# merge appended punctuations
i = 0
j = 1
while j < len(alignment):
previous = alignment[i]
following = alignment[j]
if not previous.word.endswith(" ") and following.word in appended:
# append it to the previous word
previous.word = previous.word + following.word
previous.tokens = previous.tokens + following.tokens
following.word = ""
following.tokens = []
else:
i = j
j += 1
def add_word_timestamps(
*,
segments: List[dict],
model: "Whisper",
tokenizer: Tokenizer,
mel: torch.Tensor,
num_frames: int,
prepend_punctuations: str = "\"'“¿([{-",
append_punctuations: str = "\"'.。,!?::”)]}、",
last_speech_timestamp: float,
**kwargs,
):
if len(segments) == 0:
return
text_tokens_per_segment = [
[token for token in segment["tokens"] if token < tokenizer.eot]
for segment in segments
]
text_tokens = list(itertools.chain.from_iterable(text_tokens_per_segment))
alignment = find_alignment(model, tokenizer, text_tokens, mel, num_frames, **kwargs)
word_durations = np.array([t.end - t.start for t in alignment])
word_durations = word_durations[word_durations.nonzero()]
median_duration = np.median(word_durations) if len(word_durations) > 0 else 0.0
median_duration = min(0.7, float(median_duration))
max_duration = median_duration * 2
# hack: truncate long words at sentence boundaries.
# a better segmentation algorithm based on VAD should be able to replace this.
if len(word_durations) > 0:
sentence_end_marks = ".。!?"
# ensure words at sentence boundaries are not longer than twice the median word duration.
for i in range(1, len(alignment)):
if alignment[i].end - alignment[i].start > max_duration:
if alignment[i].word in sentence_end_marks:
alignment[i].end = alignment[i].start + max_duration
elif alignment[i - 1].word in sentence_end_marks:
alignment[i].start = alignment[i].end - max_duration
merge_punctuations(alignment, prepend_punctuations, append_punctuations)
time_offset = segments[0]["seek"] * HOP_LENGTH / SAMPLE_RATE
word_index = 0
for segment, text_tokens in zip(segments, text_tokens_per_segment):
saved_tokens = 0
words = []
while word_index < len(alignment) and saved_tokens < len(text_tokens):
timing = alignment[word_index]
if timing.word:
words.append(
dict(
word=timing.word,
start=round(time_offset + timing.start, 2),
end=round(time_offset + timing.end, 2),
probability=timing.probability,
)
)
saved_tokens += len(timing.tokens)
word_index += 1
# hack: truncate long words at segment boundaries.
# a better segmentation algorithm based on VAD should be able to replace this.
if len(words) > 0:
# ensure the first and second word after a pause is not longer than
# twice the median word duration.
if words[0]["end"] - last_speech_timestamp > median_duration * 4 and (
words[0]["end"] - words[0]["start"] > max_duration
or (
len(words) > 1
and words[1]["end"] - words[0]["start"] > max_duration * 2
)
):
if (
len(words) > 1
and words[1]["end"] - words[1]["start"] > max_duration
):
boundary = max(words[1]["end"] / 2, words[1]["end"] - max_duration)
words[0]["end"] = words[1]["start"] = boundary
words[0]["start"] = max(0, words[0]["end"] - max_duration)
# prefer the segment-level start timestamp if the first word is too long.
if (
segment["start"] < words[0]["end"]
and segment["start"] - 0.5 > words[0]["start"]
):
words[0]["start"] = max(
0, min(words[0]["end"] - median_duration, segment["start"])
)
else:
segment["start"] = words[0]["start"]
# prefer the segment-level end timestamp if the last word is too long.
if (
segment["end"] > words[-1]["start"]
and segment["end"] + 0.5 < words[-1]["end"]
):
words[-1]["end"] = max(
words[-1]["start"] + median_duration, segment["end"]
)
else:
segment["end"] = words[-1]["end"]
last_speech_timestamp = segment["end"]
segment["words"] = words

View File

@ -1,11 +1,11 @@
import base64
import os
import string
from dataclasses import dataclass, field
from functools import cached_property, lru_cache
from typing import Dict, List, Optional, Tuple
from dataclasses import dataclass
from functools import lru_cache
from typing import List, Optional, Tuple, Union
import tiktoken
import numpy as np
import torch
from transformers import GPT2TokenizerFast
LANGUAGES = {
"en": "english",
@ -107,7 +107,6 @@ LANGUAGES = {
"ba": "bashkir",
"jw": "javanese",
"su": "sundanese",
"yue": "cantonese",
}
# language code lookup by name, with a few language aliases
@ -124,121 +123,117 @@ TO_LANGUAGE_CODE = {
"moldovan": "ro",
"sinhalese": "si",
"castilian": "es",
"mandarin": "zh",
}
@dataclass
@dataclass(frozen=True)
class Tokenizer:
"""A thin wrapper around `tiktoken` providing quick access to special tokens"""
"""A thin wrapper around `GPT2TokenizerFast` providing quick access to special tokens"""
encoding: tiktoken.Encoding
num_languages: int
language: Optional[str] = None
task: Optional[str] = None
sot_sequence: Tuple[int] = ()
special_tokens: Dict[str, int] = field(default_factory=dict)
def __post_init__(self):
for special in self.encoding.special_tokens_set:
special_token = self.encoding.encode_single_token(special)
self.special_tokens[special] = special_token
sot: int = self.special_tokens["<|startoftranscript|>"]
translate: int = self.special_tokens["<|translate|>"]
transcribe: int = self.special_tokens["<|transcribe|>"]
langs = tuple(LANGUAGES.keys())[: self.num_languages]
sot_sequence = [sot]
if self.language is not None:
sot_sequence.append(sot + 1 + langs.index(self.language))
if self.task is not None:
task_token: int = transcribe if self.task == "transcribe" else translate
sot_sequence.append(task_token)
self.sot_sequence = tuple(sot_sequence)
tokenizer: "GPT2TokenizerFast"
language: Optional[str]
sot_sequence: Tuple[int]
def encode(self, text, **kwargs):
return self.encoding.encode(text, **kwargs)
return self.tokenizer.encode(text, **kwargs)
def decode(self, token_ids: List[int], **kwargs) -> str:
token_ids = [t for t in token_ids if t < self.timestamp_begin]
return self.encoding.decode(token_ids, **kwargs)
def decode(self, token_ids: Union[int, List[int], np.ndarray, torch.Tensor], **kwargs):
return self.tokenizer.decode(token_ids, **kwargs)
def decode_with_timestamps(self, token_ids: List[int], **kwargs) -> str:
def decode_with_timestamps(self, tokens) -> str:
"""
Timestamp tokens are above other special tokens' id range and are ignored by `decode()`.
Timestamp tokens are above the special tokens' id range and are ignored by `decode()`.
This method decodes given tokens with timestamps tokens annotated, e.g. "<|1.08|>".
"""
return self.encoding.decode(token_ids, **kwargs)
outputs = [[]]
for token in tokens:
if token >= self.timestamp_begin:
timestamp = f"<|{(token - self.timestamp_begin) * 0.02:.2f}|>"
outputs.append(timestamp)
outputs.append([])
else:
outputs[-1].append(token)
outputs = [s if isinstance(s, str) else self.tokenizer.decode(s) for s in outputs]
return "".join(outputs)
@cached_property
@property
@lru_cache()
def eot(self) -> int:
return self.encoding.eot_token
return self.tokenizer.eos_token_id
@cached_property
def transcribe(self) -> int:
return self.special_tokens["<|transcribe|>"]
@cached_property
def translate(self) -> int:
return self.special_tokens["<|translate|>"]
@cached_property
@property
@lru_cache()
def sot(self) -> int:
return self.special_tokens["<|startoftranscript|>"]
return self._get_single_token_id("<|startoftranscript|>")
@cached_property
@property
@lru_cache()
def sot_lm(self) -> int:
return self.special_tokens["<|startoflm|>"]
return self._get_single_token_id("<|startoflm|>")
@cached_property
@property
@lru_cache()
def sot_prev(self) -> int:
return self.special_tokens["<|startofprev|>"]
return self._get_single_token_id("<|startofprev|>")
@cached_property
@property
@lru_cache()
def no_speech(self) -> int:
return self.special_tokens["<|nospeech|>"]
return self._get_single_token_id("<|nospeech|>")
@cached_property
@property
@lru_cache()
def no_timestamps(self) -> int:
return self.special_tokens["<|notimestamps|>"]
return self._get_single_token_id("<|notimestamps|>")
@cached_property
@property
@lru_cache()
def timestamp_begin(self) -> int:
return self.special_tokens["<|0.00|>"]
return self.tokenizer.all_special_ids[-1] + 1
@cached_property
@property
@lru_cache()
def language_token(self) -> int:
"""Returns the token id corresponding to the value of the `language` field"""
if self.language is None:
raise ValueError("This tokenizer does not have language token configured")
raise ValueError(f"This tokenizer does not have language token configured")
return self.to_language_token(self.language)
additional_tokens = dict(
zip(
self.tokenizer.additional_special_tokens,
self.tokenizer.additional_special_tokens_ids,
)
)
candidate = f"<|{self.language}|>"
if candidate in additional_tokens:
return additional_tokens[candidate]
def to_language_token(self, language):
if token := self.special_tokens.get(f"<|{language}|>", None):
return token
raise KeyError(f"Language {self.language} not found in tokenizer.")
raise KeyError(f"Language {language} not found in tokenizer.")
@cached_property
@property
@lru_cache()
def all_language_tokens(self) -> Tuple[int]:
result = []
for token, token_id in self.special_tokens.items():
for token, token_id in zip(
self.tokenizer.additional_special_tokens,
self.tokenizer.additional_special_tokens_ids,
):
if token.strip("<|>") in LANGUAGES:
result.append(token_id)
return tuple(result)[: self.num_languages]
return tuple(result)
@cached_property
@property
@lru_cache()
def all_language_codes(self) -> Tuple[str]:
return tuple(self.decode([_l]).strip("<|>") for _l in self.all_language_tokens)
return tuple(self.decode([l]).strip("<|>") for l in self.all_language_tokens)
@cached_property
@property
@lru_cache()
def sot_sequence_including_notimestamps(self) -> Tuple[int]:
return tuple(list(self.sot_sequence) + [self.no_timestamps])
@cached_property
@property
@lru_cache()
def non_speech_tokens(self) -> Tuple[int]:
"""
Returns the list of tokens to suppress in order to avoid any speaker tags or non-speech
@ -250,10 +245,8 @@ class Tokenizer:
keeping basic punctuations like commas, periods, question marks, exclamation points, etc.
"""
symbols = list('"#()*+/:;<=>@[\\]^_`{|}~「」『』')
symbols += (
"<< >> <<< >>> -- --- -( -[ (' (\" (( )) ((( ))) [[ ]] {{ }} ♪♪ ♪♪♪".split()
)
symbols = list("\"#()*+/:;<=>@[\\]^_`{|}~「」『』")
symbols += "<< >> <<< >>> -- --- -( -[ (' (\" (( )) ((( ))) [[ ]] {{ }} ♪♪ ♪♪♪".split()
# symbols that may be a single token or multiple tokens depending on the tokenizer.
# In case they're multiple tokens, suppress the first token, which is safe because:
@ -263,113 +256,47 @@ class Tokenizer:
assert all(0x2640 <= ord(c) <= 0x267F for c in miscellaneous)
# allow hyphens "-" and single quotes "'" between words, but not at the beginning of a word
result = {self.encoding.encode(" -")[0], self.encoding.encode(" '")[0]}
result = {self.tokenizer.encode(" -")[0], self.tokenizer.encode(" '")[0]}
for symbol in symbols + list(miscellaneous):
for tokens in [
self.encoding.encode(symbol),
self.encoding.encode(" " + symbol),
]:
for tokens in [self.tokenizer.encode(symbol), self.tokenizer.encode(" " + symbol)]:
if len(tokens) == 1 or symbol in miscellaneous:
result.add(tokens[0])
return tuple(sorted(result))
def split_to_word_tokens(self, tokens: List[int]):
if self.language in {"zh", "ja", "th", "lo", "my", "yue"}:
# These languages don't typically use spaces, so it is difficult to split words
# without morpheme analysis. Here, we instead split words at any
# position where the tokens are decoded as valid unicode points
return self.split_tokens_on_unicode(tokens)
return self.split_tokens_on_spaces(tokens)
def split_tokens_on_unicode(self, tokens: List[int]):
decoded_full = self.decode_with_timestamps(tokens)
replacement_char = "\ufffd"
words = []
word_tokens = []
current_tokens = []
unicode_offset = 0
for token in tokens:
current_tokens.append(token)
decoded = self.decode_with_timestamps(current_tokens)
if (
replacement_char not in decoded
or decoded_full[unicode_offset + decoded.index(replacement_char)]
== replacement_char
):
words.append(decoded)
word_tokens.append(current_tokens)
current_tokens = []
unicode_offset += len(decoded)
return words, word_tokens
def split_tokens_on_spaces(self, tokens: List[int]):
subwords, subword_tokens_list = self.split_tokens_on_unicode(tokens)
words = []
word_tokens = []
for subword, subword_tokens in zip(subwords, subword_tokens_list):
special = subword_tokens[0] >= self.eot
with_space = subword.startswith(" ")
punctuation = subword.strip() in string.punctuation
if special or with_space or punctuation or len(words) == 0:
words.append(subword)
word_tokens.append(subword_tokens)
else:
words[-1] = words[-1] + subword
word_tokens[-1].extend(subword_tokens)
return words, word_tokens
def _get_single_token_id(self, text) -> int:
tokens = self.tokenizer.encode(text)
assert len(tokens) == 1, f"{text} is not encoded as a single token"
return tokens[0]
@lru_cache(maxsize=None)
def get_encoding(name: str = "gpt2", num_languages: int = 99):
vocab_path = os.path.join(os.path.dirname(__file__), "assets", f"{name}.tiktoken")
ranks = {
base64.b64decode(token): int(rank)
for token, rank in (line.split() for line in open(vocab_path) if line)
}
n_vocab = len(ranks)
special_tokens = {}
def build_tokenizer(name: str = "gpt2"):
os.environ["TOKENIZERS_PARALLELISM"] = "false"
path = os.path.join(os.path.dirname(__file__), "assets", name)
tokenizer = GPT2TokenizerFast.from_pretrained(path)
specials = [
"<|endoftext|>",
"<|startoftranscript|>",
*[f"<|{lang}|>" for lang in list(LANGUAGES.keys())[:num_languages]],
*[f"<|{lang}|>" for lang in LANGUAGES.keys()],
"<|translate|>",
"<|transcribe|>",
"<|startoflm|>",
"<|startofprev|>",
"<|nospeech|>",
"<|notimestamps|>",
*[f"<|{i * 0.02:.2f}|>" for i in range(1501)],
]
for token in specials:
special_tokens[token] = n_vocab
n_vocab += 1
return tiktoken.Encoding(
name=os.path.basename(vocab_path),
explicit_n_vocab=n_vocab,
pat_str=r"""'s|'t|'re|'ve|'m|'ll|'d| ?\p{L}+| ?\p{N}+| ?[^\s\p{L}\p{N}]+|\s+(?!\S)|\s+""",
mergeable_ranks=ranks,
special_tokens=special_tokens,
)
tokenizer.add_special_tokens(dict(additional_special_tokens=specials))
return tokenizer
@lru_cache(maxsize=None)
def get_tokenizer(
multilingual: bool,
*,
num_languages: int = 99,
language: Optional[str] = None,
task: Optional[str] = None, # Literal["transcribe", "translate", None]
language: Optional[str] = None,
) -> Tokenizer:
if language is not None:
language = language.lower()
@ -380,16 +307,25 @@ def get_tokenizer(
raise ValueError(f"Unsupported language: {language}")
if multilingual:
encoding_name = "multilingual"
language = language or "en"
tokenizer_name = "multilingual"
task = task or "transcribe"
language = language or "en"
else:
encoding_name = "gpt2"
language = None
tokenizer_name = "gpt2"
task = None
language = None
encoding = get_encoding(name=encoding_name, num_languages=num_languages)
tokenizer = build_tokenizer(name=tokenizer_name)
all_special_ids: List[int] = tokenizer.all_special_ids
sot: int = all_special_ids[1]
translate: int = all_special_ids[-6]
transcribe: int = all_special_ids[-5]
return Tokenizer(
encoding=encoding, num_languages=num_languages, language=language, task=task
)
langs = tuple(LANGUAGES.keys())
sot_sequence = [sot]
if language is not None:
sot_sequence.append(sot + 1 + langs.index(language))
if task is not None:
sot_sequence.append(transcribe if task == "transcribe" else translate)
return Tokenizer(tokenizer=tokenizer, language=language, sot_sequence=tuple(sot_sequence))

View File

@ -1,35 +1,16 @@
import argparse
import os
import traceback
import warnings
from typing import TYPE_CHECKING, List, Optional, Tuple, Union
from typing import List, Optional, Tuple, Union, TYPE_CHECKING
import numpy as np
import torch
import tqdm
from .audio import (
FRAMES_PER_SECOND,
HOP_LENGTH,
N_FRAMES,
N_SAMPLES,
SAMPLE_RATE,
log_mel_spectrogram,
pad_or_trim,
)
from .audio import SAMPLE_RATE, N_FRAMES, HOP_LENGTH, pad_or_trim, log_mel_spectrogram
from .decoding import DecodingOptions, DecodingResult
from .timing import add_word_timestamps
from .tokenizer import LANGUAGES, TO_LANGUAGE_CODE, get_tokenizer
from .utils import (
exact_div,
format_timestamp,
get_end,
get_writer,
make_safe,
optional_float,
optional_int,
str2bool,
)
from .utils import exact_div, format_timestamp, optional_int, optional_float, str2bool, write_txt, write_vtt, write_srt
if TYPE_CHECKING:
from .model import Whisper
@ -45,13 +26,6 @@ def transcribe(
logprob_threshold: Optional[float] = -1.0,
no_speech_threshold: Optional[float] = 0.6,
condition_on_previous_text: bool = True,
initial_prompt: Optional[str] = None,
carry_initial_prompt: bool = False,
word_timestamps: bool = False,
prepend_punctuations: str = "\"'“¿([{-",
append_punctuations: str = "\"'.。,!?::”)]}、",
clip_timestamps: Union[str, List[float]] = "0",
hallucination_silence_threshold: Optional[float] = None,
**decode_options,
):
"""
@ -88,37 +62,9 @@ def transcribe(
disabling may make the text inconsistent across windows, but the model becomes less prone to
getting stuck in a failure loop, such as repetition looping or timestamps going out of sync.
word_timestamps: bool
Extract word-level timestamps using the cross-attention pattern and dynamic time warping,
and include the timestamps for each word in each segment.
prepend_punctuations: str
If word_timestamps is True, merge these punctuation symbols with the next word
append_punctuations: str
If word_timestamps is True, merge these punctuation symbols with the previous word
initial_prompt: Optional[str]
Optional text to provide as a prompt for the first window. This can be used to provide, or
"prompt-engineer" a context for transcription, e.g. custom vocabularies or proper nouns
to make it more likely to predict those word correctly.
carry_initial_prompt: bool
If carry_initial_prompt is True, `initial_prompt` is prepended to the prompt of each internal
`decode()` call. If there is not enough context space at the start of the prompt, it is
left-sliced to make space.
decode_options: dict
Keyword arguments to construct `DecodingOptions` instances
clip_timestamps: Union[str, List[float]]
Comma-separated list start,end,start,end,... timestamps (in seconds) of clips to process.
The last end timestamp defaults to the end of the file.
hallucination_silence_threshold: Optional[float]
When word_timestamps is True, skip silent periods longer than this threshold (in seconds)
when a possible hallucination is detected
Returns
-------
A dictionary containing the resulting text ("text") and segment-level details ("segments"), and
@ -135,56 +81,26 @@ def transcribe(
if dtype == torch.float32:
decode_options["fp16"] = False
# Pad 30-seconds of silence to the input audio, for slicing
mel = log_mel_spectrogram(audio, model.dims.n_mels, padding=N_SAMPLES)
content_frames = mel.shape[-1] - N_FRAMES
content_duration = float(content_frames * HOP_LENGTH / SAMPLE_RATE)
mel = log_mel_spectrogram(audio)
if decode_options.get("language", None) is None:
if not model.is_multilingual:
decode_options["language"] = "en"
else:
if verbose:
print(
"Detecting language using up to the first 30 seconds. Use `--language` to specify the language"
)
mel_segment = pad_or_trim(mel, N_FRAMES).to(model.device).to(dtype)
_, probs = model.detect_language(mel_segment)
print("Detecting language using up to the first 30 seconds. Use `--language` to specify the language")
segment = pad_or_trim(mel, N_FRAMES).to(model.device).to(dtype)
_, probs = model.detect_language(segment)
decode_options["language"] = max(probs, key=probs.get)
if verbose is not None:
print(
f"Detected language: {LANGUAGES[decode_options['language']].title()}"
)
print(f"Detected language: {LANGUAGES[decode_options['language']].title()}")
language: str = decode_options["language"]
task: str = decode_options.get("task", "transcribe")
tokenizer = get_tokenizer(
model.is_multilingual,
num_languages=model.num_languages,
language=language,
task=task,
)
if isinstance(clip_timestamps, str):
clip_timestamps = [
float(ts) for ts in (clip_timestamps.split(",") if clip_timestamps else [])
]
seek_points: List[int] = [round(ts * FRAMES_PER_SECOND) for ts in clip_timestamps]
if len(seek_points) == 0:
seek_points.append(0)
if len(seek_points) % 2 == 1:
seek_points.append(content_frames)
seek_clips: List[Tuple[int, int]] = list(zip(seek_points[::2], seek_points[1::2]))
punctuation = "\"'“¿([{-\"'.。,!?::”)]}、"
if word_timestamps and task == "translate":
warnings.warn("Word-level timestamps on translations may not be reliable.")
language = decode_options["language"]
task = decode_options.get("task", "transcribe")
tokenizer = get_tokenizer(model.is_multilingual, language=language, task=task)
def decode_with_fallback(segment: torch.Tensor) -> DecodingResult:
temperatures = (
[temperature] if isinstance(temperature, (int, float)) else temperature
)
temperatures = [temperature] if isinstance(temperature, (int, float)) else temperature
decode_result = None
for t in temperatures:
@ -201,30 +117,17 @@ def transcribe(
decode_result = model.decode(segment, options)
needs_fallback = False
if (
compression_ratio_threshold is not None
and decode_result.compression_ratio > compression_ratio_threshold
):
if compression_ratio_threshold is not None and decode_result.compression_ratio > compression_ratio_threshold:
needs_fallback = True # too repetitive
if (
logprob_threshold is not None
and decode_result.avg_logprob < logprob_threshold
):
if logprob_threshold is not None and decode_result.avg_logprob < logprob_threshold:
needs_fallback = True # average log probability is too low
if (
no_speech_threshold is not None
and decode_result.no_speech_prob > no_speech_threshold
and logprob_threshold is not None
and decode_result.avg_logprob < logprob_threshold
):
needs_fallback = False # silence
if not needs_fallback:
break
return decode_result
clip_idx = 0
seek = seek_clips[clip_idx][0]
seek = 0
input_stride = exact_div(
N_FRAMES, model.dims.n_audio_ctx
) # mel frames per output token: 2
@ -235,303 +138,123 @@ def transcribe(
all_segments = []
prompt_reset_since = 0
remaining_prompt_length = model.dims.n_text_ctx // 2 - 1
if initial_prompt is not None:
initial_prompt_tokens = tokenizer.encode(" " + initial_prompt.strip())
all_tokens.extend(initial_prompt_tokens)
remaining_prompt_length -= len(initial_prompt_tokens)
else:
initial_prompt_tokens = []
initial_prompt = decode_options.pop("initial_prompt", None) or []
if initial_prompt:
initial_prompt = tokenizer.encode(" " + initial_prompt.strip())
all_tokens.extend(initial_prompt)
def new_segment(
*, start: float, end: float, tokens: torch.Tensor, result: DecodingResult
def add_segment(
*, start: float, end: float, text_tokens: torch.Tensor, result: DecodingResult
):
tokens = tokens.tolist()
text_tokens = [token for token in tokens if token < tokenizer.eot]
return {
"seek": seek,
"start": start,
"end": end,
"text": tokenizer.decode(text_tokens),
"tokens": tokens,
"temperature": result.temperature,
"avg_logprob": result.avg_logprob,
"compression_ratio": result.compression_ratio,
"no_speech_prob": result.no_speech_prob,
}
text = tokenizer.decode([token for token in text_tokens if token < tokenizer.eot])
if len(text.strip()) == 0: # skip empty text output
return
# show the progress bar when verbose is False (if True, transcribed text will be printed)
with tqdm.tqdm(
total=content_frames, unit="frames", disable=verbose is not False
) as pbar:
last_speech_timestamp = 0.0
# NOTE: This loop is obscurely flattened to make the diff readable.
# A later commit should turn this into a simpler nested loop.
# for seek_clip_start, seek_clip_end in seek_clips:
# while seek < seek_clip_end
while clip_idx < len(seek_clips):
seek_clip_start, seek_clip_end = seek_clips[clip_idx]
if seek < seek_clip_start:
seek = seek_clip_start
if seek >= seek_clip_end:
clip_idx += 1
if clip_idx < len(seek_clips):
seek = seek_clips[clip_idx][0]
continue
time_offset = float(seek * HOP_LENGTH / SAMPLE_RATE)
window_end_time = float((seek + N_FRAMES) * HOP_LENGTH / SAMPLE_RATE)
segment_size = min(N_FRAMES, content_frames - seek, seek_clip_end - seek)
mel_segment = mel[:, seek : seek + segment_size]
segment_duration = segment_size * HOP_LENGTH / SAMPLE_RATE
mel_segment = pad_or_trim(mel_segment, N_FRAMES).to(model.device).to(dtype)
all_segments.append(
{
"id": len(all_segments),
"seek": seek,
"start": start,
"end": end,
"text": text,
"tokens": text_tokens.tolist(),
"temperature": result.temperature,
"avg_logprob": result.avg_logprob,
"compression_ratio": result.compression_ratio,
"no_speech_prob": result.no_speech_prob,
}
)
if verbose:
print(f"[{format_timestamp(start)} --> {format_timestamp(end)}] {text}")
if carry_initial_prompt:
nignored = max(len(initial_prompt_tokens), prompt_reset_since)
remaining_prompt = all_tokens[nignored:][-remaining_prompt_length:]
decode_options["prompt"] = initial_prompt_tokens + remaining_prompt
else:
decode_options["prompt"] = all_tokens[prompt_reset_since:]
# show the progress bar when verbose is False (otherwise the transcribed text will be printed)
num_frames = mel.shape[-1]
previous_seek_value = seek
result: DecodingResult = decode_with_fallback(mel_segment)
with tqdm.tqdm(total=num_frames, unit='frames', disable=verbose is not False) as pbar:
while seek < num_frames:
timestamp_offset = float(seek * HOP_LENGTH / SAMPLE_RATE)
segment = pad_or_trim(mel[:, seek:], N_FRAMES).to(model.device).to(dtype)
segment_duration = segment.shape[-1] * HOP_LENGTH / SAMPLE_RATE
decode_options["prompt"] = all_tokens[prompt_reset_since:]
result: DecodingResult = decode_with_fallback(segment)
tokens = torch.tensor(result.tokens)
if no_speech_threshold is not None:
# no voice activity check
should_skip = result.no_speech_prob > no_speech_threshold
if (
logprob_threshold is not None
and result.avg_logprob > logprob_threshold
):
if logprob_threshold is not None and result.avg_logprob > logprob_threshold:
# don't skip if the logprob is high enough, despite the no_speech_prob
should_skip = False
if should_skip:
seek += segment_size # fast-forward to the next segment boundary
seek += segment.shape[-1] # fast-forward to the next segment boundary
continue
previous_seek = seek
current_segments = []
# anomalous words are very long/short/improbable
def word_anomaly_score(word: dict) -> float:
probability = word.get("probability", 0.0)
duration = word["end"] - word["start"]
score = 0.0
if probability < 0.15:
score += 1.0
if duration < 0.133:
score += (0.133 - duration) * 15
if duration > 2.0:
score += duration - 2.0
return score
def is_segment_anomaly(segment: Optional[dict]) -> bool:
if segment is None or not segment["words"]:
return False
words = [w for w in segment["words"] if w["word"] not in punctuation]
words = words[:8]
score = sum(word_anomaly_score(w) for w in words)
return score >= 3 or score + 0.01 >= len(words)
def next_words_segment(segments: List[dict]) -> Optional[dict]:
return next((s for s in segments if s["words"]), None)
timestamp_tokens: torch.Tensor = tokens.ge(tokenizer.timestamp_begin)
single_timestamp_ending = timestamp_tokens[-2:].tolist() == [False, True]
consecutive = torch.where(timestamp_tokens[:-1] & timestamp_tokens[1:])[0]
consecutive.add_(1)
if len(consecutive) > 0:
# if the output contains two consecutive timestamp tokens
slices = consecutive.tolist()
if single_timestamp_ending:
slices.append(len(tokens))
consecutive = torch.where(timestamp_tokens[:-1] & timestamp_tokens[1:])[0].add_(1)
if len(consecutive) > 0: # if the output contains two consecutive timestamp tokens
last_slice = 0
for current_slice in slices:
for current_slice in consecutive:
sliced_tokens = tokens[last_slice:current_slice]
start_timestamp_pos = (
start_timestamp_position = (
sliced_tokens[0].item() - tokenizer.timestamp_begin
)
end_timestamp_pos = (
end_timestamp_position = (
sliced_tokens[-1].item() - tokenizer.timestamp_begin
)
current_segments.append(
new_segment(
start=time_offset + start_timestamp_pos * time_precision,
end=time_offset + end_timestamp_pos * time_precision,
tokens=sliced_tokens,
result=result,
)
add_segment(
start=timestamp_offset + start_timestamp_position * time_precision,
end=timestamp_offset + end_timestamp_position * time_precision,
text_tokens=sliced_tokens[1:-1],
result=result,
)
last_slice = current_slice
if single_timestamp_ending:
# single timestamp at the end means no speech after the last timestamp.
seek += segment_size
else:
# otherwise, ignore the unfinished segment and seek to the last timestamp
last_timestamp_pos = (
tokens[last_slice - 1].item() - tokenizer.timestamp_begin
)
seek += last_timestamp_pos * input_stride
last_timestamp_position = (
tokens[last_slice - 1].item() - tokenizer.timestamp_begin
)
seek += last_timestamp_position * input_stride
all_tokens.extend(tokens[: last_slice + 1].tolist())
else:
duration = segment_duration
timestamps = tokens[timestamp_tokens.nonzero().flatten()]
if (
len(timestamps) > 0
and timestamps[-1].item() != tokenizer.timestamp_begin
):
if len(timestamps) > 0 and timestamps[-1].item() != tokenizer.timestamp_begin:
# no consecutive timestamps but it has a timestamp; use the last one.
last_timestamp_pos = (
timestamps[-1].item() - tokenizer.timestamp_begin
)
duration = last_timestamp_pos * time_precision
# single timestamp at the end means no speech after the last timestamp.
last_timestamp_position = timestamps[-1].item() - tokenizer.timestamp_begin
duration = last_timestamp_position * time_precision
current_segments.append(
new_segment(
start=time_offset,
end=time_offset + duration,
tokens=tokens,
result=result,
)
)
seek += segment_size
if word_timestamps:
add_word_timestamps(
segments=current_segments,
model=model,
tokenizer=tokenizer,
mel=mel_segment,
num_frames=segment_size,
prepend_punctuations=prepend_punctuations,
append_punctuations=append_punctuations,
last_speech_timestamp=last_speech_timestamp,
add_segment(
start=timestamp_offset,
end=timestamp_offset + duration,
text_tokens=tokens,
result=result,
)
if not single_timestamp_ending:
last_word_end = get_end(current_segments)
if last_word_end is not None and last_word_end > time_offset:
seek = round(last_word_end * FRAMES_PER_SECOND)
# skip silence before possible hallucinations
if hallucination_silence_threshold is not None:
threshold = hallucination_silence_threshold
if not single_timestamp_ending:
last_word_end = get_end(current_segments)
if last_word_end is not None and last_word_end > time_offset:
remaining_duration = window_end_time - last_word_end
if remaining_duration > threshold:
seek = round(last_word_end * FRAMES_PER_SECOND)
else:
seek = previous_seek + segment_size
# if first segment might be a hallucination, skip leading silence
first_segment = next_words_segment(current_segments)
if first_segment is not None and is_segment_anomaly(first_segment):
gap = first_segment["start"] - time_offset
if gap > threshold:
seek = previous_seek + round(gap * FRAMES_PER_SECOND)
continue
# skip silence before any possible hallucination that is surrounded
# by silence or more hallucinations
hal_last_end = last_speech_timestamp
for si in range(len(current_segments)):
segment = current_segments[si]
if not segment["words"]:
continue
if is_segment_anomaly(segment):
next_segment = next_words_segment(
current_segments[si + 1 :]
)
if next_segment is not None:
hal_next_start = next_segment["words"][0]["start"]
else:
hal_next_start = time_offset + segment_duration
silence_before = (
segment["start"] - hal_last_end > threshold
or segment["start"] < threshold
or segment["start"] - time_offset < 2.0
)
silence_after = (
hal_next_start - segment["end"] > threshold
or is_segment_anomaly(next_segment)
or window_end_time - segment["end"] < 2.0
)
if silence_before and silence_after:
seek = round(
max(time_offset + 1, segment["start"])
* FRAMES_PER_SECOND
)
if content_duration - segment["end"] < threshold:
seek = content_frames
current_segments[si:] = []
break
hal_last_end = segment["end"]
last_word_end = get_end(current_segments)
if last_word_end is not None:
last_speech_timestamp = last_word_end
if verbose:
for segment in current_segments:
start, end, text = segment["start"], segment["end"], segment["text"]
line = f"[{format_timestamp(start)} --> {format_timestamp(end)}] {text}"
print(make_safe(line))
# if a segment is instantaneous or does not contain text, clear it
for i, segment in enumerate(current_segments):
if segment["start"] == segment["end"] or segment["text"].strip() == "":
segment["text"] = ""
segment["tokens"] = []
segment["words"] = []
all_segments.extend(
[
{"id": i, **segment}
for i, segment in enumerate(
current_segments, start=len(all_segments)
)
]
)
all_tokens.extend(
[token for segment in current_segments for token in segment["tokens"]]
)
seek += segment.shape[-1]
all_tokens.extend(tokens.tolist())
if not condition_on_previous_text or result.temperature > 0.5:
# do not feed the prompt tokens if a high temperature was used
prompt_reset_since = len(all_tokens)
# update progress bar
pbar.update(min(content_frames, seek) - previous_seek)
pbar.update(min(num_frames, seek) - previous_seek_value)
previous_seek_value = seek
return dict(
text=tokenizer.decode(all_tokens[len(initial_prompt_tokens) :]),
segments=all_segments,
language=language,
)
return dict(text=tokenizer.decode(all_tokens[len(initial_prompt):]), segments=all_segments, language=language)
def cli():
from . import available_models
def valid_model_name(name):
if name in available_models() or os.path.exists(name):
return name
raise ValueError(
f"model should be one of {available_models()} or path to a model checkpoint"
)
# fmt: off
parser = argparse.ArgumentParser(formatter_class=argparse.ArgumentDefaultsHelpFormatter)
parser.add_argument("audio", nargs="+", type=str, help="audio file(s) to transcribe")
parser.add_argument("--model", default="turbo", type=valid_model_name, help="name of the Whisper model to use")
parser.add_argument("--model", default="small", choices=available_models(), help="name of the Whisper model to use")
parser.add_argument("--model_dir", type=str, default=None, help="the path to save model files; uses ~/.cache/whisper by default")
parser.add_argument("--device", default="cuda" if torch.cuda.is_available() else "cpu", help="device to use for PyTorch inference")
parser.add_argument("--output_dir", "-o", type=str, default=".", help="directory to save the outputs")
parser.add_argument("--output_format", "-f", type=str, default="all", choices=["txt", "vtt", "srt", "tsv", "json", "all"], help="format of the output file; if not specified, all available formats will be produced")
parser.add_argument("--verbose", type=str2bool, default=True, help="whether to print out the progress and debug messages")
parser.add_argument("--task", type=str, default="transcribe", choices=["transcribe", "translate"], help="whether to perform X->X speech recognition ('transcribe') or X->English translation ('translate')")
@ -545,8 +268,6 @@ def cli():
parser.add_argument("--suppress_tokens", type=str, default="-1", help="comma-separated list of token ids to suppress during sampling; '-1' will suppress most special characters except common punctuations")
parser.add_argument("--initial_prompt", type=str, default=None, help="optional text to provide as a prompt for the first window.")
parser.add_argument("--carry_initial_prompt", type=str2bool, default=False, help="if True, prepend initial_prompt to every internal decode() call. May reduce the effectiveness of condition_on_previous_text")
parser.add_argument("--condition_on_previous_text", type=str2bool, default=True, help="if True, provide the previous output of the model as a prompt for the next window; disabling may make the text inconsistent across windows, but the model becomes less prone to getting stuck in a failure loop")
parser.add_argument("--fp16", type=str2bool, default=True, help="whether to perform inference in fp16; True by default")
@ -554,70 +275,51 @@ def cli():
parser.add_argument("--compression_ratio_threshold", type=optional_float, default=2.4, help="if the gzip compression ratio is higher than this value, treat the decoding as failed")
parser.add_argument("--logprob_threshold", type=optional_float, default=-1.0, help="if the average log probability is lower than this value, treat the decoding as failed")
parser.add_argument("--no_speech_threshold", type=optional_float, default=0.6, help="if the probability of the <|nospeech|> token is higher than this value AND the decoding has failed due to `logprob_threshold`, consider the segment as silence")
parser.add_argument("--word_timestamps", type=str2bool, default=False, help="(experimental) extract word-level timestamps and refine the results based on them")
parser.add_argument("--prepend_punctuations", type=str, default="\"\'“¿([{-", help="if word_timestamps is True, merge these punctuation symbols with the next word")
parser.add_argument("--append_punctuations", type=str, default="\"\'.。,!?::”)]}、", help="if word_timestamps is True, merge these punctuation symbols with the previous word")
parser.add_argument("--highlight_words", type=str2bool, default=False, help="(requires --word_timestamps True) underline each word as it is spoken in srt and vtt")
parser.add_argument("--max_line_width", type=optional_int, default=None, help="(requires --word_timestamps True) the maximum number of characters in a line before breaking the line")
parser.add_argument("--max_line_count", type=optional_int, default=None, help="(requires --word_timestamps True) the maximum number of lines in a segment")
parser.add_argument("--max_words_per_line", type=optional_int, default=None, help="(requires --word_timestamps True, no effect with --max_line_width) the maximum number of words in a segment")
parser.add_argument("--threads", type=optional_int, default=0, help="number of threads used by torch for CPU inference; supercedes MKL_NUM_THREADS/OMP_NUM_THREADS")
parser.add_argument("--clip_timestamps", type=str, default="0", help="comma-separated list start,end,start,end,... timestamps (in seconds) of clips to process, where the last end timestamp defaults to the end of the file")
parser.add_argument("--hallucination_silence_threshold", type=optional_float, help="(requires --word_timestamps True) skip silent periods longer than this threshold (in seconds) when a possible hallucination is detected")
# fmt: on
args = parser.parse_args().__dict__
model_name: str = args.pop("model")
model_dir: str = args.pop("model_dir")
output_dir: str = args.pop("output_dir")
output_format: str = args.pop("output_format")
device: str = args.pop("device")
os.makedirs(output_dir, exist_ok=True)
if model_name.endswith(".en") and args["language"] not in {"en", "English"}:
if args["language"] is not None:
warnings.warn(
f"{model_name} is an English-only model but receipted '{args['language']}'; using English instead."
)
warnings.warn(f"{model_name} is an English-only model but receipted '{args['language']}'; using English instead.")
args["language"] = "en"
temperature = args.pop("temperature")
if (increment := args.pop("temperature_increment_on_fallback")) is not None:
temperature = tuple(np.arange(temperature, 1.0 + 1e-6, increment))
temperature_increment_on_fallback = args.pop("temperature_increment_on_fallback")
if temperature_increment_on_fallback is not None:
temperature = tuple(np.arange(temperature, 1.0 + 1e-6, temperature_increment_on_fallback))
else:
temperature = [temperature]
if (threads := args.pop("threads")) > 0:
threads = args.pop("threads")
if threads > 0:
torch.set_num_threads(threads)
from . import load_model
model = load_model(model_name, device=device, download_root=model_dir)
writer = get_writer(output_format, output_dir)
word_options = [
"highlight_words",
"max_line_count",
"max_line_width",
"max_words_per_line",
]
if not args["word_timestamps"]:
for option in word_options:
if args[option]:
parser.error(f"--{option} requires --word_timestamps True")
if args["max_line_count"] and not args["max_line_width"]:
warnings.warn("--max_line_count has no effect without --max_line_width")
if args["max_words_per_line"] and args["max_line_width"]:
warnings.warn("--max_words_per_line has no effect with --max_line_width")
writer_args = {arg: args.pop(arg) for arg in word_options}
for audio_path in args.pop("audio"):
try:
result = transcribe(model, audio_path, temperature=temperature, **args)
writer(result, audio_path, **writer_args)
except Exception as e:
traceback.print_exc()
print(f"Skipping {audio_path} due to {type(e).__name__}: {str(e)}")
result = transcribe(model, audio_path, temperature=temperature, **args)
audio_basename = os.path.basename(audio_path)
# save TXT
with open(os.path.join(output_dir, audio_basename + ".txt"), "w", encoding="utf-8") as txt:
write_txt(result["segments"], file=txt)
# save VTT
with open(os.path.join(output_dir, audio_basename + ".vtt"), "w", encoding="utf-8") as vtt:
write_vtt(result["segments"], file=vtt)
# save SRT
with open(os.path.join(output_dir, audio_basename + ".srt"), "w", encoding="utf-8") as srt:
write_srt(result["segments"], file=srt)
if __name__ == "__main__":
if __name__ == '__main__':
cli()

View File

@ -1,117 +0,0 @@
from functools import lru_cache
import numpy as np
import torch
try:
import triton
import triton.language as tl
except ImportError:
raise RuntimeError("triton import failed; try `pip install --pre triton`")
@triton.jit
def dtw_kernel(
cost, trace, x, x_stride, cost_stride, trace_stride, N, M, BLOCK_SIZE: tl.constexpr
):
offsets = tl.arange(0, BLOCK_SIZE)
mask = offsets < M
for k in range(1, N + M + 1): # k = i + j
tl.debug_barrier()
p0 = cost + (k - 1) * cost_stride
p1 = cost + k * cost_stride
p2 = cost + k * cost_stride + 1
c0 = tl.load(p0 + offsets, mask=mask)
c1 = tl.load(p1 + offsets, mask=mask)
c2 = tl.load(p2 + offsets, mask=mask)
x_row = tl.load(x + (k - 1) * x_stride + offsets, mask=mask, other=0)
cost_row = x_row + tl.minimum(tl.minimum(c0, c1), c2)
cost_ptr = cost + (k + 1) * cost_stride + 1
tl.store(cost_ptr + offsets, cost_row, mask=mask)
trace_ptr = trace + (k + 1) * trace_stride + 1
tl.store(trace_ptr + offsets, 2, mask=mask & (c2 <= c0) & (c2 <= c1))
tl.store(trace_ptr + offsets, 1, mask=mask & (c1 <= c0) & (c1 <= c2))
tl.store(trace_ptr + offsets, 0, mask=mask & (c0 <= c1) & (c0 <= c2))
@lru_cache(maxsize=None)
def median_kernel(filter_width: int):
@triton.jit
def kernel(
y, x, x_stride, y_stride, BLOCK_SIZE: tl.constexpr
): # x.shape[-1] == filter_width
row_idx = tl.program_id(0)
offsets = tl.arange(0, BLOCK_SIZE)
mask = offsets < y_stride
x_ptr = x + row_idx * x_stride # noqa: F841
y_ptr = y + row_idx * y_stride
LOAD_ALL_ROWS_HERE # noqa: F821
BUBBLESORT_HERE # noqa: F821
tl.store(y_ptr + offsets, MIDDLE_ROW_HERE, mask=mask) # noqa: F821
kernel = triton.JITFunction(kernel.fn)
new_kernel = kernel.src.replace(
" LOAD_ALL_ROWS_HERE",
"\n".join(
[
f" row{i} = tl.load(x_ptr + offsets + {i}, mask=mask)"
for i in range(filter_width)
]
),
)
new_kernel = new_kernel.replace(
" BUBBLESORT_HERE",
"\n\n".join(
[
"\n\n".join(
[
"\n".join(
[
f" smaller = tl.where(row{j} < row{j + 1}, row{j}, row{j + 1})",
f" larger = tl.where(row{j} > row{j + 1}, row{j}, row{j + 1})",
f" row{j} = smaller",
f" row{j + 1} = larger",
]
)
for j in range(filter_width - i - 1)
]
)
for i in range(filter_width // 2 + 1)
]
),
)
new_kernel = new_kernel.replace("MIDDLE_ROW_HERE", f"row{filter_width // 2}")
if hasattr(kernel, "_unsafe_update_src") is True:
kernel._unsafe_update_src(new_kernel)
kernel.hash = None
else:
kernel.src = new_kernel
return kernel
def median_filter_cuda(x: torch.Tensor, filter_width: int):
"""Apply a median filter of given width along the last dimension of x"""
slices = x.contiguous().unfold(-1, filter_width, 1)
grid = np.prod(slices.shape[:-2])
kernel = median_kernel(filter_width)
y = torch.empty_like(slices[..., 0])
BLOCK_SIZE = 1 << (y.stride(-2) - 1).bit_length()
kernel[(grid,)](y, x, x.stride(-2), y.stride(-2), BLOCK_SIZE=BLOCK_SIZE)
return y

View File

@ -1,24 +1,5 @@
import json
import os
import re
import sys
import zlib
from typing import Callable, List, Optional, TextIO
system_encoding = sys.getdefaultencoding()
if system_encoding != "utf-8":
def make_safe(string):
# replaces any character not representable using the system default encoding with an '?',
# avoiding UnicodeEncodeError (https://github.com/openai/whisper/discussions/729).
return string.encode(system_encoding, errors="replace").decode(system_encoding)
else:
def make_safe(string):
# utf-8 can encode any Unicode code point, so no need to do the round-trip encoding
return string
from typing import Iterator, TextIO
def exact_div(x, y):
@ -47,9 +28,7 @@ def compression_ratio(text) -> float:
return len(text_bytes) / len(zlib.compress(text_bytes))
def format_timestamp(
seconds: float, always_include_hours: bool = False, decimal_marker: str = "."
):
def format_timestamp(seconds: float, always_include_hours: bool = False, decimal_marker: str = '.'):
assert seconds >= 0, "non-negative timestamp expected"
milliseconds = round(seconds * 1000.0)
@ -63,256 +42,47 @@ def format_timestamp(
milliseconds -= seconds * 1_000
hours_marker = f"{hours:02d}:" if always_include_hours or hours > 0 else ""
return (
f"{hours_marker}{minutes:02d}:{seconds:02d}{decimal_marker}{milliseconds:03d}"
)
return f"{hours_marker}{minutes:02d}:{seconds:02d}{decimal_marker}{milliseconds:03d}"
def get_start(segments: List[dict]) -> Optional[float]:
return next(
(w["start"] for s in segments for w in s["words"]),
segments[0]["start"] if segments else None,
)
def write_txt(transcript: Iterator[dict], file: TextIO):
for segment in transcript:
print(segment['text'].strip(), file=file, flush=True)
def get_end(segments: List[dict]) -> Optional[float]:
return next(
(w["end"] for s in reversed(segments) for w in reversed(s["words"])),
segments[-1]["end"] if segments else None,
)
class ResultWriter:
extension: str
def __init__(self, output_dir: str):
self.output_dir = output_dir
def __call__(
self, result: dict, audio_path: str, options: Optional[dict] = None, **kwargs
):
audio_basename = os.path.basename(audio_path)
audio_basename = os.path.splitext(audio_basename)[0]
output_path = os.path.join(
self.output_dir, audio_basename + "." + self.extension
)
with open(output_path, "w", encoding="utf-8") as f:
self.write_result(result, file=f, options=options, **kwargs)
def write_result(
self, result: dict, file: TextIO, options: Optional[dict] = None, **kwargs
):
raise NotImplementedError
class WriteTXT(ResultWriter):
extension: str = "txt"
def write_result(
self, result: dict, file: TextIO, options: Optional[dict] = None, **kwargs
):
for segment in result["segments"]:
print(segment["text"].strip(), file=file, flush=True)
class SubtitlesWriter(ResultWriter):
always_include_hours: bool
decimal_marker: str
def iterate_result(
self,
result: dict,
options: Optional[dict] = None,
*,
max_line_width: Optional[int] = None,
max_line_count: Optional[int] = None,
highlight_words: bool = False,
max_words_per_line: Optional[int] = None,
):
options = options or {}
max_line_width = max_line_width or options.get("max_line_width")
max_line_count = max_line_count or options.get("max_line_count")
highlight_words = highlight_words or options.get("highlight_words", False)
max_words_per_line = max_words_per_line or options.get("max_words_per_line")
preserve_segments = max_line_count is None or max_line_width is None
max_line_width = max_line_width or 1000
max_words_per_line = max_words_per_line or 1000
def iterate_subtitles():
line_len = 0
line_count = 1
# the next subtitle to yield (a list of word timings with whitespace)
subtitle: List[dict] = []
last: float = get_start(result["segments"]) or 0.0
for segment in result["segments"]:
chunk_index = 0
words_count = max_words_per_line
while chunk_index < len(segment["words"]):
remaining_words = len(segment["words"]) - chunk_index
if max_words_per_line > len(segment["words"]) - chunk_index:
words_count = remaining_words
for i, original_timing in enumerate(
segment["words"][chunk_index : chunk_index + words_count]
):
timing = original_timing.copy()
long_pause = (
not preserve_segments and timing["start"] - last > 3.0
)
has_room = line_len + len(timing["word"]) <= max_line_width
seg_break = i == 0 and len(subtitle) > 0 and preserve_segments
if (
line_len > 0
and has_room
and not long_pause
and not seg_break
):
# line continuation
line_len += len(timing["word"])
else:
# new line
timing["word"] = timing["word"].strip()
if (
len(subtitle) > 0
and max_line_count is not None
and (long_pause or line_count >= max_line_count)
or seg_break
):
# subtitle break
yield subtitle
subtitle = []
line_count = 1
elif line_len > 0:
# line break
line_count += 1
timing["word"] = "\n" + timing["word"]
line_len = len(timing["word"].strip())
subtitle.append(timing)
last = timing["start"]
chunk_index += max_words_per_line
if len(subtitle) > 0:
yield subtitle
if len(result["segments"]) > 0 and "words" in result["segments"][0]:
for subtitle in iterate_subtitles():
subtitle_start = self.format_timestamp(subtitle[0]["start"])
subtitle_end = self.format_timestamp(subtitle[-1]["end"])
subtitle_text = "".join([word["word"] for word in subtitle])
if highlight_words:
last = subtitle_start
all_words = [timing["word"] for timing in subtitle]
for i, this_word in enumerate(subtitle):
start = self.format_timestamp(this_word["start"])
end = self.format_timestamp(this_word["end"])
if last != start:
yield last, start, subtitle_text
yield start, end, "".join(
[
(
re.sub(r"^(\s*)(.*)$", r"\1<u>\2</u>", word)
if j == i
else word
)
for j, word in enumerate(all_words)
]
)
last = end
else:
yield subtitle_start, subtitle_end, subtitle_text
else:
for segment in result["segments"]:
segment_start = self.format_timestamp(segment["start"])
segment_end = self.format_timestamp(segment["end"])
segment_text = segment["text"].strip().replace("-->", "->")
yield segment_start, segment_end, segment_text
def format_timestamp(self, seconds: float):
return format_timestamp(
seconds=seconds,
always_include_hours=self.always_include_hours,
decimal_marker=self.decimal_marker,
def write_vtt(transcript: Iterator[dict], file: TextIO):
print("WEBVTT\n", file=file)
for segment in transcript:
print(
f"{format_timestamp(segment['start'])} --> {format_timestamp(segment['end'])}\n"
f"{segment['text'].strip().replace('-->', '->')}\n",
file=file,
flush=True,
)
class WriteVTT(SubtitlesWriter):
extension: str = "vtt"
always_include_hours: bool = False
decimal_marker: str = "."
def write_result(
self, result: dict, file: TextIO, options: Optional[dict] = None, **kwargs
):
print("WEBVTT\n", file=file)
for start, end, text in self.iterate_result(result, options, **kwargs):
print(f"{start} --> {end}\n{text}\n", file=file, flush=True)
class WriteSRT(SubtitlesWriter):
extension: str = "srt"
always_include_hours: bool = True
decimal_marker: str = ","
def write_result(
self, result: dict, file: TextIO, options: Optional[dict] = None, **kwargs
):
for i, (start, end, text) in enumerate(
self.iterate_result(result, options, **kwargs), start=1
):
print(f"{i}\n{start} --> {end}\n{text}\n", file=file, flush=True)
class WriteTSV(ResultWriter):
def write_srt(transcript: Iterator[dict], file: TextIO):
"""
Write a transcript to a file in TSV (tab-separated values) format containing lines like:
<start time in integer milliseconds>\t<end time in integer milliseconds>\t<transcript text>
Write a transcript to a file in SRT format.
Using integer milliseconds as start and end times means there's no chance of interference from
an environment setting a language encoding that causes the decimal in a floating point number
to appear as a comma; also is faster and more efficient to parse & store, e.g., in C++.
Example usage:
from pathlib import Path
from whisper.utils import write_srt
result = transcribe(model, audio_path, temperature=temperature, **args)
# save SRT
audio_basename = Path(audio_path).stem
with open(Path(output_dir) / (audio_basename + ".srt"), "w", encoding="utf-8") as srt:
write_srt(result["segments"], file=srt)
"""
extension: str = "tsv"
def write_result(
self, result: dict, file: TextIO, options: Optional[dict] = None, **kwargs
):
print("start", "end", "text", sep="\t", file=file)
for segment in result["segments"]:
print(round(1000 * segment["start"]), file=file, end="\t")
print(round(1000 * segment["end"]), file=file, end="\t")
print(segment["text"].strip().replace("\t", " "), file=file, flush=True)
class WriteJSON(ResultWriter):
extension: str = "json"
def write_result(
self, result: dict, file: TextIO, options: Optional[dict] = None, **kwargs
):
json.dump(result, file)
def get_writer(
output_format: str, output_dir: str
) -> Callable[[dict, TextIO, dict], None]:
writers = {
"txt": WriteTXT,
"vtt": WriteVTT,
"srt": WriteSRT,
"tsv": WriteTSV,
"json": WriteJSON,
}
if output_format == "all":
all_writers = [writer(output_dir) for writer in writers.values()]
def write_all(
result: dict, file: TextIO, options: Optional[dict] = None, **kwargs
):
for writer in all_writers:
writer(result, file, options, **kwargs)
return write_all
return writers[output_format](output_dir)
for i, segment in enumerate(transcript, start=1):
# write srt lines
print(
f"{i}\n"
f"{format_timestamp(segment['start'], always_include_hours=True, decimal_marker=',')} --> "
f"{format_timestamp(segment['end'], always_include_hours=True, decimal_marker=',')}\n"
f"{segment['text'].strip().replace('-->', '->')}\n",
file=file,
flush=True,
)

View File

@ -1 +1 @@
__version__ = "20250625"
__version__ = "20230117"