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13
.github/dependabot.yml vendored Normal file
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@ -0,0 +1,13 @@
# 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@v3
- uses: actions/checkout@v4
- 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@v4
uses: actions/setup-python@v5
with:
python-version: '3.8'
python-version: '3.12'
- name: Install dependencies
run: |
python -m pip install --upgrade pip
pip install setuptools wheel twine
pip install setuptools wheel twine build
- name: Release
if: ${{ steps.regex-match.outputs.match != '' }}
uses: softprops/action-gh-release@v1
uses: softprops/action-gh-release@v2
with:
tag_name: v${{ 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 setup.py sdist
python -m build --sdist
twine upload dist/*

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@ -11,19 +11,19 @@ jobs:
pre-commit:
runs-on: ubuntu-latest
steps:
- uses: actions/checkout@v3
- uses: actions/checkout@v4
- name: Fetch base branch
run: git fetch origin ${{ github.base_ref }}
- uses: actions/setup-python@v4
- uses: actions/setup-python@v5
with:
python-version: "3.8"
python-version: "3.9"
architecture: x64
- name: Get pip cache dir
id: pip-cache
run: |
echo "dir=$(pip cache dir)" >> $GITHUB_OUTPUT
- name: pip/pre-commit cache
uses: actions/cache@v3
uses: actions/cache@v4
with:
path: |
${{ steps.pip-cache.outputs.dir }}
@ -33,24 +33,47 @@ jobs:
${{ runner.os }}-pip-pre-commit
- name: pre-commit
run: |
pip install -U pre-commit
pip install --upgrade pre-commit
pre-commit install --install-hooks
pre-commit run --all-files
whisper-test:
needs: pre-commit
runs-on: ubuntu-latest
strategy:
fail-fast: false
matrix:
python-version: ['3.8', '3.9', '3.10', '3.11']
pytorch-version: [1.13.1, 2.0.0]
exclude:
- python-version: '3.11'
include:
- python-version: '3.8'
pytorch-version: 1.10.1
numpy-requirement: "'numpy<2'"
- python-version: '3.8'
pytorch-version: 1.13.1
numpy-requirement: "'numpy<2'"
- python-version: '3.8'
pytorch-version: 2.0.1
numpy-requirement: "'numpy<2'"
- python-version: '3.9'
pytorch-version: 2.1.2
numpy-requirement: "'numpy<2'"
- python-version: '3.10'
pytorch-version: 2.2.2
numpy-requirement: "'numpy<2'"
- python-version: '3.11'
pytorch-version: 2.3.1
numpy-requirement: "'numpy'"
- python-version: '3.12'
pytorch-version: 2.4.1
numpy-requirement: "'numpy'"
- python-version: '3.12'
pytorch-version: 2.5.1
numpy-requirement: "'numpy'"
- python-version: '3.13'
pytorch-version: 2.5.1
numpy-requirement: "'numpy'"
steps:
- uses: conda-incubator/setup-miniconda@v2
- uses: conda-incubator/setup-miniconda@v3
- run: conda install -n test ffmpeg python=${{ matrix.python-version }}
- run: pip3 install torch==${{ matrix.pytorch-version }}+cpu --index-url https://download.pytorch.org/whl/cpu
- uses: actions/checkout@v3
- uses: actions/checkout@v4
- run: echo "$CONDA/envs/test/bin" >> $GITHUB_PATH
- run: pip install .["dev"]
- 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'

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@ -1,6 +1,6 @@
repos:
- repo: https://github.com/pre-commit/pre-commit-hooks
rev: v4.0.1
rev: v5.0.0
hooks:
- id: check-json
- id: end-of-file-fixer
@ -11,17 +11,17 @@ repos:
- id: check-added-large-files
args: [--maxkb=4096]
- repo: https://github.com/psf/black
rev: 23.7.0
rev: 25.1.0
hooks:
- id: black
- repo: https://github.com/pycqa/isort
rev: 5.12.0
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: 6.0.0
rev: 7.1.1
hooks:
- id: flake8
types: [python]

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@ -1,5 +1,57 @@
# 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))

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@ -57,42 +57,55 @@ pip install setuptools-rust
## Available models and languages
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.
There are six model sizes, four with English-only versions, offering speed and accuracy tradeoffs.
Below are the names of the available models and their approximate memory requirements and inference speed relative to the large model.
The relative speeds below are measured by transcribing English speech on a A100, and the real-world speed may vary significantly depending on many factors including the language, the speaking speed, and the available hardware.
| Size | Parameters | English-only model | Multilingual model | Required VRAM | Relative speed |
|:------:|:----------:|:------------------:|:------------------:|:-------------:|:--------------:|
| tiny | 39 M | `tiny.en` | `tiny` | ~1 GB | ~32x |
| base | 74 M | `base.en` | `base` | ~1 GB | ~16x |
| small | 244 M | `small.en` | `small` | ~2 GB | ~6x |
| tiny | 39 M | `tiny.en` | `tiny` | ~1 GB | ~10x |
| base | 74 M | `base.en` | `base` | ~1 GB | ~7x |
| small | 244 M | `small.en` | `small` | ~2 GB | ~4x |
| medium | 769 M | `medium.en` | `medium` | ~5 GB | ~2x |
| large | 1550 M | N/A | `large` | ~10 GB | 1x |
| turbo | 809 M | N/A | `turbo` | ~6 GB | ~8x |
The `.en` models for English-only applications tend to perform better, especially for the `tiny.en` and `base.en` models. We observed that the difference becomes less significant for the `small.en` and `medium.en` models.
Additionally, the `turbo` model is an optimized version of `large-v3` that offers faster transcription speed with a minimal degradation in accuracy.
Whisper's performance varies widely depending on the language. The figure below shows a WER (Word Error Rate) breakdown by languages of the Fleurs dataset using the `large-v2` model (The smaller the numbers, the better the performance). Additional WER scores corresponding to the other models and datasets can be found in Appendix D.1, D.2, and D.4. Meanwhile, more BLEU (Bilingual Evaluation Understudy) scores can be found in Appendix D.3. Both are found in [the paper](https://arxiv.org/abs/2212.04356).
![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 `medium` model:
The following command will transcribe speech in audio files, using the `turbo` model:
whisper audio.flac audio.mp3 audio.wav --model medium
```bash
whisper audio.flac audio.mp3 audio.wav --model 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:
The default setting (which selects the `turbo` model) works well for transcribing English. However, **the `turbo` model is not trained for translation tasks**. If you need to **translate non-English speech into English**, use one of the **multilingual models** (`tiny`, `base`, `small`, `medium`, `large`) instead of `turbo`.
For example, to transcribe an audio file containing non-English speech, you can specify the language:
```bash
whisper japanese.wav --language Japanese
```
Adding `--task translate` will translate the speech into English:
To **translate** speech into English, use:
whisper japanese.wav --language Japanese --task translate
```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.
Run the following to view all available options:
```bash
whisper --help
```
See [tokenizer.py](https://github.com/openai/whisper/blob/main/whisper/tokenizer.py) for the list of all available languages.
@ -104,7 +117,7 @@ Transcription can also be performed within Python:
```python
import whisper
model = whisper.load_model("base")
model = whisper.load_model("turbo")
result = model.transcribe("audio.mp3")
print(result["text"])
```
@ -116,14 +129,14 @@ Below is an example usage of `whisper.detect_language()` and `whisper.decode()`
```python
import whisper
model = whisper.load_model("base")
model = whisper.load_model("turbo")
# load audio and pad/trim it to fit 30 seconds
audio = whisper.load_audio("audio.mp3")
audio = whisper.pad_or_trim(audio)
# make log-Mel spectrogram and move to the same device as the model
mel = whisper.log_mel_spectrogram(audio).to(model.device)
mel = whisper.log_mel_spectrogram(audio, n_mels=model.dims.n_mels).to(model.device)
# detect the spoken language
_, probs = model.detect_language(mel)

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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 ad 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 and 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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@ -16,13 +16,15 @@ 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).
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.
### Release date
September 2022 (original series) and December 2022 (`large-v2`)
September 2022 (original series), December 2022 (`large-v2`), November 2023 (`large-v3`), September 2024 (`large-v3-turbo`)
### Model type

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

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

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@ -1,3 +1,50 @@
[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]
@ -5,4 +52,3 @@ profile = "black"
include_trailing_comma = true
line_length = 88
multi_line_output = 3

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@ -3,4 +3,5 @@ numpy
torch
tqdm
more-itertools
tiktoken==0.3.3
tiktoken
triton>=2.0.0;platform_machine=="x86_64" and sys_platform=="linux" or sys_platform=="linux2"

View File

@ -1,43 +0,0 @@
import os
import platform
import sys
import pkg_resources
from setuptools import find_packages, setup
def read_version(fname="whisper/version.py"):
exec(compile(open(fname, encoding="utf-8").read(), fname, "exec"))
return locals()["__version__"]
requirements = []
if sys.platform.startswith("linux") and platform.machine() == "x86_64":
requirements.append("triton==2.0.0")
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=">=3.8",
author="OpenAI",
url="https://github.com/openai/whisper",
license="MIT",
packages=find_packages(exclude=["tests*"]),
install_requires=requirements
+ [
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", "scipy", "black", "flake8", "isort"]},
)

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@ -1,7 +1,17 @@
import pytest
from whisper.tokenizer import get_tokenizer
def test_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():
gpt2_tokenizer = get_tokenizer(multilingual=False)
multilingual_tokenizer = get_tokenizer(multilingual=True)
@ -20,5 +30,5 @@ def test_split_on_unicode():
tokens = [8404, 871, 287, 6, 246, 526, 3210, 20378]
words, word_tokens = multilingual_tokenizer.split_tokens_on_unicode(tokens)
assert words == [" elle", " est", " l", "'", "<EFBFBD>", "é", "rit", "oire"]
assert words == [" elle", " est", " l", "'", "\ufffd", "é", "rit", "oire"]
assert word_tokens == [[8404], [871], [287], [6], [246], [526], [3210], [20378]]

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@ -25,7 +25,7 @@ def test_transcribe(model_name: str):
assert "your country" in transcription
assert "do for you" in transcription
tokenizer = get_tokenizer(model.is_multilingual)
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|>")

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@ -28,7 +28,10 @@ _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": "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
@ -44,7 +47,10 @@ _ALIGNMENT_HEADS = {
"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": 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`",
}
@ -150,7 +156,8 @@ def load_model(
with (
io.BytesIO(checkpoint_file) if in_memory else open(checkpoint_file, "rb")
) as fp:
checkpoint = torch.load(fp, map_location=device)
kwargs = {"weights_only": True} if torch.__version__ >= "1.13" else {}
checkpoint = torch.load(fp, map_location=device, **kwargs)
del checkpoint_file
dims = ModelDimensions(**checkpoint["dims"])

Binary file not shown.

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@ -12,7 +12,6 @@ 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
@ -90,7 +89,7 @@ def pad_or_trim(array, length: int = N_SAMPLES, *, axis: int = -1):
@lru_cache(maxsize=None)
def mel_filters(device, n_mels: int = N_MELS) -> torch.Tensor:
def mel_filters(device, n_mels: int) -> torch.Tensor:
"""
load the mel filterbank matrix for projecting STFT into a Mel spectrogram.
Allows decoupling librosa dependency; saved using:
@ -98,18 +97,19 @@ def mel_filters(device, n_mels: int = N_MELS) -> 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 == 80, f"Unsupported n_mels: {n_mels}"
with np.load(
os.path.join(os.path.dirname(__file__), "assets", "mel_filters.npz")
) as f:
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:
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 = N_MELS,
n_mels: int = 80,
padding: int = 0,
device: Optional[Union[str, torch.device]] = None,
):
@ -122,7 +122,7 @@ 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 is supported
The number of Mel-frequency filters, only 80 and 128 are supported
padding: int
Number of zero samples to pad to the right
@ -132,7 +132,7 @@ def log_mel_spectrogram(
Returns
-------
torch.Tensor, shape = (80, n_frames)
torch.Tensor, shape = (n_mels, n_frames)
A Tensor that contains the Mel spectrogram
"""
if not torch.is_tensor(audio):

View File

@ -32,7 +32,9 @@ def detect_language(
list of dictionaries containing the probability distribution over all languages.
"""
if tokenizer is None:
tokenizer = get_tokenizer(model.is_multilingual)
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
@ -514,7 +516,10 @@ class DecodingTask:
language = options.language or "en"
tokenizer = get_tokenizer(
model.is_multilingual, language=language, task=options.task
model.is_multilingual,
num_languages=model.num_languages,
language=language,
task=options.task,
)
self.tokenizer: Tokenizer = tokenizer
self.options: DecodingOptions = self._verify_options(options)

View File

@ -1,7 +1,8 @@
import base64
import gzip
from contextlib import contextmanager
from dataclasses import dataclass
from typing import Dict, Iterable, Optional
from typing import Dict, Iterable, Optional, Tuple
import numpy as np
import torch
@ -12,6 +13,14 @@ 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
@dataclass
class ModelDimensions:
@ -59,7 +68,19 @@ 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
@ -92,20 +113,30 @@ class MultiHeadAttention(nn.Module):
def qkv_attention(
self, q: Tensor, k: Tensor, v: Tensor, mask: Optional[Tensor] = None
):
) -> Tuple[torch.Tensor, Optional[torch.Tensor]]:
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) * scale
k = k.view(*k.shape[:2], self.n_head, -1).permute(0, 2, 3, 1) * scale
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)
v = v.view(*v.shape[:2], self.n_head, -1).permute(0, 2, 1, 3)
qk = q @ k
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()
w = F.softmax(qk, dim=-1).to(q.dtype)
return (w @ v).permute(0, 2, 1, 3).flatten(start_dim=2), qk.detach()
out = (w @ v).permute(0, 2, 1, 3).flatten(start_dim=2)
qk = qk.detach()
return out, qk
class ResidualAttentionBlock(nn.Module):
@ -236,7 +267,8 @@ class Whisper(nn.Module):
self.dims.n_text_head,
self.dims.n_text_layer,
)
# use the last half layers for alignment by default; see `set_alignment_heads()` below
# 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
)
@ -269,7 +301,11 @@ class Whisper(nn.Module):
@property
def is_multilingual(self):
return self.dims.n_vocab == 51865
return self.dims.n_vocab >= 51865
@property
def num_languages(self):
return self.dims.n_vocab - 51765 - int(self.is_multilingual)
def install_kv_cache_hooks(self, cache: Optional[dict] = None):
"""

View File

@ -30,15 +30,19 @@ 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]
else (
ADDITIONAL_DIACRITICS[c]
if c in ADDITIONAL_DIACRITICS
else ""
else (
""
if unicodedata.category(c) == "Mn"
else " "
if unicodedata.category(c)[0] in "MSP"
else c
else " " if unicodedata.category(c)[0] in "MSP" else c
)
)
)
for c in unicodedata.normalize("NFKD", s)
)

View File

@ -117,7 +117,7 @@ def dtw_cuda(x, BLOCK_SIZE=1024):
x_skew = x_skew.T.contiguous()
cost = torch.ones(N + M + 2, M + 2) * np.inf
cost[0, 0] = 0
cost = cost.cuda()
cost = cost.to(x.device)
trace = torch.zeros_like(cost, dtype=torch.int32)
dtw_kernel[(1,)](
@ -191,7 +191,9 @@ def find_alignment(
for i, block in enumerate(model.decoder.blocks)
]
with torch.no_grad():
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)
@ -299,6 +301,7 @@ def add_word_timestamps(
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.

View File

@ -107,6 +107,7 @@ LANGUAGES = {
"ba": "bashkir",
"jw": "javanese",
"su": "sundanese",
"yue": "cantonese",
}
# language code lookup by name, with a few language aliases
@ -123,6 +124,7 @@ TO_LANGUAGE_CODE = {
"moldovan": "ro",
"sinhalese": "si",
"castilian": "es",
"mandarin": "zh",
}
@ -131,6 +133,7 @@ class Tokenizer:
"""A thin wrapper around `tiktoken` providing quick access to special tokens"""
encoding: tiktoken.Encoding
num_languages: int
language: Optional[str] = None
task: Optional[str] = None
sot_sequence: Tuple[int] = ()
@ -145,7 +148,7 @@ class Tokenizer:
translate: int = self.special_tokens["<|translate|>"]
transcribe: int = self.special_tokens["<|transcribe|>"]
langs = tuple(LANGUAGES.keys())
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))
@ -211,10 +214,13 @@ class Tokenizer:
if self.language is None:
raise ValueError("This tokenizer does not have language token configured")
if token := self.special_tokens.get(f"<|{self.language}|>", None):
return self.to_language_token(self.language)
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
def all_language_tokens(self) -> Tuple[int]:
@ -222,7 +228,7 @@ class Tokenizer:
for token, token_id in self.special_tokens.items():
if token.strip("<|>") in LANGUAGES:
result.append(token_id)
return tuple(result)
return tuple(result)[: self.num_languages]
@cached_property
def all_language_codes(self) -> Tuple[str]:
@ -269,7 +275,7 @@ class Tokenizer:
return tuple(sorted(result))
def split_to_word_tokens(self, tokens: List[int]):
if self.language in {"zh", "ja", "th", "lo", "my"}:
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
@ -322,7 +328,7 @@ class Tokenizer:
@lru_cache(maxsize=None)
def get_encoding(name: str = "gpt2"):
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)
@ -334,7 +340,7 @@ def get_encoding(name: str = "gpt2"):
specials = [
"<|endoftext|>",
"<|startoftranscript|>",
*[f"<|{lang}|>" for lang in LANGUAGES.keys()],
*[f"<|{lang}|>" for lang in list(LANGUAGES.keys())[:num_languages]],
"<|translate|>",
"<|transcribe|>",
"<|startoflm|>",
@ -361,6 +367,7 @@ def get_encoding(name: str = "gpt2"):
def get_tokenizer(
multilingual: bool,
*,
num_languages: int = 99,
language: Optional[str] = None,
task: Optional[str] = None, # Literal["transcribe", "translate", None]
) -> Tokenizer:
@ -381,6 +388,8 @@ def get_tokenizer(
language = None
task = None
encoding = get_encoding(name=encoding_name)
encoding = get_encoding(name=encoding_name, num_languages=num_languages)
return Tokenizer(encoding=encoding, language=language, task=task)
return Tokenizer(
encoding=encoding, num_languages=num_languages, language=language, task=task
)

View File

@ -1,7 +1,8 @@
import argparse
import os
import traceback
import warnings
from typing import TYPE_CHECKING, Optional, Tuple, Union
from typing import TYPE_CHECKING, List, Optional, Tuple, Union
from importlib.util import find_spec
import numpy as np
@ -23,6 +24,7 @@ from .tokenizer import LANGUAGES, TO_LANGUAGE_CODE, get_tokenizer
from .utils import (
exact_div,
format_timestamp,
get_end,
get_writer,
make_safe,
optional_float,
@ -45,9 +47,12 @@ def transcribe(
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,
):
"""
@ -99,9 +104,22 @@ def transcribe(
"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
@ -121,8 +139,9 @@ def transcribe(
decode_options["fp16"] = False
# Pad 30-seconds of silence to the input audio, for slicing
mel = log_mel_spectrogram(audio, padding=N_SAMPLES)
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)
if decode_options.get("language", None) is None:
if not model.is_multilingual:
@ -142,7 +161,25 @@ def transcribe(
language: str = decode_options["language"]
task: str = decode_options.get("task", "transcribe")
tokenizer = get_tokenizer(model.is_multilingual, language=language, task=task)
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.")
@ -180,6 +217,8 @@ def transcribe(
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:
@ -187,7 +226,8 @@ def transcribe(
return decode_result
seek = 0
clip_idx = 0
seek = seek_clips[clip_idx][0]
input_stride = exact_div(
N_FRAMES, model.dims.n_audio_ctx
) # mel frames per output token: 2
@ -198,9 +238,11 @@ 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 = []
@ -226,14 +268,33 @@ def transcribe(
total=content_frames, unit="frames", disable=verbose is not False
) as pbar:
last_speech_timestamp = 0.0
while seek < content_frames:
# 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)
mel_segment = mel[:, seek : seek + N_FRAMES]
segment_size = min(N_FRAMES, content_frames - seek)
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)
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:]
result: DecodingResult = decode_with_fallback(mel_segment)
tokens = torch.tensor(result.tokens)
@ -254,6 +315,30 @@ def transcribe(
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]
@ -327,17 +412,71 @@ def transcribe(
append_punctuations=append_punctuations,
last_speech_timestamp=last_speech_timestamp,
)
word_end_timestamps = [
w["end"] for s in current_segments for w in s["words"]
]
if len(word_end_timestamps) > 0:
last_speech_timestamp = word_end_timestamps[-1]
if not single_timestamp_ending and len(word_end_timestamps) > 0:
seek_shift = round(
(word_end_timestamps[-1] - time_offset) * FRAMES_PER_SECOND
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 seek_shift > 0:
seek = previous_seek + seek_shift
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:
@ -381,10 +520,17 @@ def transcribe(
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="small", choices=available_models(), help="name of the Whisper model to use")
parser.add_argument("--model", default="turbo", type=valid_model_name, 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=None, help="device to use for PyTorch inference")
parser.add_argument("--output_dir", "-o", type=str, default=".", help="directory to save the outputs")
@ -402,6 +548,8 @@ 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")
@ -415,7 +563,10 @@ def cli():
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__
@ -447,17 +598,28 @@ def cli():
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"]
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)
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)}")
if __name__ == "__main__":

View File

@ -60,7 +60,7 @@ def median_kernel(filter_width: int):
tl.store(y_ptr + offsets, MIDDLE_ROW_HERE, mask=mask) # noqa: F821
kernel = triton.JITFunction(kernel.fn)
kernel.src = kernel.src.replace(
new_kernel = kernel.src.replace(
" LOAD_ALL_ROWS_HERE",
"\n".join(
[
@ -69,7 +69,8 @@ def median_kernel(filter_width: int):
]
),
)
kernel.src = kernel.src.replace(
new_kernel = new_kernel.replace(
" BUBBLESORT_HERE",
"\n\n".join(
[
@ -90,7 +91,14 @@ def median_kernel(filter_width: int):
]
),
)
kernel.src = kernel.src.replace("MIDDLE_ROW_HERE", f"row{filter_width // 2}")
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

View File

@ -3,7 +3,7 @@ import os
import re
import sys
import zlib
from typing import Callable, Optional, TextIO
from typing import Callable, List, Optional, TextIO
system_encoding = sys.getdefaultencoding()
@ -68,13 +68,29 @@ def format_timestamp(
)
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 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: dict):
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(
@ -82,16 +98,20 @@ class ResultWriter:
)
with open(output_path, "w", encoding="utf-8") as f:
self.write_result(result, file=f, options=options)
self.write_result(result, file=f, options=options, **kwargs)
def write_result(self, result: dict, file: TextIO, options: dict):
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: dict):
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)
@ -100,26 +120,53 @@ class SubtitlesWriter(ResultWriter):
always_include_hours: bool
decimal_marker: str
def iterate_result(self, result: dict, options: dict):
raw_max_line_width: Optional[int] = options["max_line_width"]
max_line_count: Optional[int] = options["max_line_count"]
highlight_words: bool = options["highlight_words"]
max_line_width = 1000 if raw_max_line_width is None else raw_max_line_width
preserve_segments = max_line_count is None or raw_max_line_width is None
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 = result["segments"][0]["words"][0]["start"]
subtitle: List[dict] = []
last: float = get_start(result["segments"]) or 0.0
for segment in result["segments"]:
for i, original_timing in enumerate(segment["words"]):
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
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:
if (
line_len > 0
and has_room
and not long_pause
and not seg_break
):
# line continuation
line_len += len(timing["word"])
else:
@ -142,10 +189,11 @@ class SubtitlesWriter(ResultWriter):
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 "words" in result["segments"][0]:
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"])
@ -161,9 +209,11 @@ class SubtitlesWriter(ResultWriter):
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)
]
)
@ -190,9 +240,11 @@ class WriteVTT(SubtitlesWriter):
always_include_hours: bool = False
decimal_marker: str = "."
def write_result(self, result: dict, file: TextIO, options: dict):
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):
for start, end, text in self.iterate_result(result, options, **kwargs):
print(f"{start} --> {end}\n{text}\n", file=file, flush=True)
@ -201,9 +253,11 @@ class WriteSRT(SubtitlesWriter):
always_include_hours: bool = True
decimal_marker: str = ","
def write_result(self, result: dict, file: TextIO, options: dict):
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), start=1
self.iterate_result(result, options, **kwargs), start=1
):
print(f"{i}\n{start} --> {end}\n{text}\n", file=file, flush=True)
@ -220,7 +274,9 @@ class WriteTSV(ResultWriter):
extension: str = "tsv"
def write_result(self, result: dict, file: TextIO, options: dict):
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")
@ -231,7 +287,9 @@ class WriteTSV(ResultWriter):
class WriteJSON(ResultWriter):
extension: str = "json"
def write_result(self, result: dict, file: TextIO, options: dict):
def write_result(
self, result: dict, file: TextIO, options: Optional[dict] = None, **kwargs
):
json.dump(result, file)
@ -249,9 +307,11 @@ def get_writer(
if output_format == "all":
all_writers = [writer(output_dir) for writer in writers.values()]
def write_all(result: dict, file: TextIO, options: dict):
def write_all(
result: dict, file: TextIO, options: Optional[dict] = None, **kwargs
):
for writer in all_writers:
writer(result, file, options)
writer(result, file, options, **kwargs)
return write_all

View File

@ -1 +1 @@
__version__ = "20230918"
__version__ = "20250625"