Merge pull request #99 from JSchmie/develop

Update Major Release v0.2.0

.dockerignore

@@ -0,0 +1,6 @@
+scraibe/*__pycache__
+scraibe/app/*__pycache__
+scraibe/.pyannotetoken
+.git
+.gitignore
+.github

.github/workflows/check_semver.yaml

@@ -0,0 +1,90 @@
+name: Check and Add Version in Changelog
+
+on:
+  pull_request:
+    branches:
+      - main
+      - develop
+
+jobs:
+  check-and-add-version:
+    runs-on: ubuntu-latest
+    steps:
+      - name: Checkout Repository
+        uses: actions/checkout@v4
+        with:
+          fetch-depth: 0
+
+      - name: Check if Source Branch is docs
+        id: check_docs_branch
+        run: |
+          pr_head_ref="${{ github.event.pull_request.head.ref }}"
+          if [[ "$pr_head_ref" == "docs" ]]; then
+            echo "This is a docs branch merge. Exiting without creating a tag."
+            echo "is_docs_branch=true" >> $GITHUB_ENV
+            exit 0
+          else
+            echo "is_docs_branch=false" >> $GITHUB_ENV
+          fi
+
+      - name: Extract and Determine Version
+        if: env.is_docs_branch != 'true'
+        id: extract_version
+        run: |
+          # Fetch the latest tags from the remote
+          git fetch --tags
+
+          # Get the latest tag, or initialize to v0.0.0 if no tags are found
+          latest_tag=$(git describe --tags `git rev-list --tags --max-count=1` 2>/dev/null || echo "v0.0.0")
+
+          # Extract version from PR title or body
+          pr_body="${{ github.event.pull_request.body }}"
+          pr_title="${{ github.event.pull_request.title }}"
+          version_regex="v([0-9]+)\.([0-9]+)\.([0-9]+)"
+
+          if [[ $pr_body =~ $version_regex ]]; then
+            major=${BASH_REMATCH[1]}
+            minor=${BASH_REMATCH[2]}
+            patch=${BASH_REMATCH[3]}
+            new_tag="v$major.$minor.$patch"
+          elif [[ $pr_title =~ $version_regex ]]; then
+            major=${BASH_REMATCH[1]}
+            minor=${BASH_REMATCH[2]}
+            patch=${BASH_REMATCH[3]}
+            new_tag="v$major.$minor.$patch"
+          else
+            # Split the latest tag into parts
+            IFS='.' read -r -a parts <<< "${latest_tag#v}"
+            major=${parts[0]}
+            minor=${parts[1]}
+            patch=${parts[2]}
+            patch=$((patch + 1))
+            new_tag="v$major.$minor.$patch"
+          fi
+
+          clean_version="${new_tag#v}"
+          echo "version=$clean_version" >> $GITHUB_ENV
+          echo "Version determined: $clean_version"
+
+      - name: Check if Version Already Exists in Tags
+        if: env.is_docs_branch != 'true'
+        run: |
+          version="${{ env.version }}"
+          if git tag --list | grep -q "^$version$"; then
+            echo "Version $version already exists in tags."
+            exit 1
+          else
+            echo "Version $version does not exist in tags."
+          fi
+
+      - name: Check Version in CHANGELOG
+        if: env.is_docs_branch != 'true'
+        id: check_version
+        run: |
+          version="${{ env.version }}"
+          if ! grep -q "^## \[$version\]" CHANGELOG.md; then
+            echo "Version $version not found in CHANGELOG.md."
+            exit 1
+          else
+            echo "Version $version found in CHANGELOG.md."
+          fi

.github/workflows/documentation.yml

@@ -0,0 +1,44 @@
+name: documentation
+
+on:
+  push:
+    branches:
+      - main
+  workflow_dispatch:
+
+permissions:
+  contents: write
+
+jobs:
+  docs:
+    runs-on: ubuntu-latest
+    steps:
+      - uses: actions/checkout@v4
+      - uses: actions/setup-python@v3
+        with:
+          python-version: 3.9
+      - name: Install dependencies
+        run: |
+          sudo apt-get install libsndfile1-dev
+          pip install --upgrade pip
+          pip install -r requirements.txt
+          pip install --upgrade sphinx sphinx_rtd_theme myst-parser
+          pip install --upgrade markdown-it-py[plugins]
+          pip install --upgrade mdit-py-plugins
+          
+      - name: Sphinx build
+        run: |
+          cp README.md ./source/README.md
+          cp LICENSE ./source/LICENSE
+          cp -r Pictures ./source/Pictures
+          sphinx-apidoc -o source scraibe/  
+          sphinx-build -M html source docs
+          make html
+      - name: Deploy to GitHub Pages
+        uses: peaceiris/actions-gh-pages@v3
+        if: ${{ github.event_name == 'push' && github.ref == 'refs/heads/sphinx_action' }}
+        with:
+          publish_branch: gh-pages
+          github_token: ${{ secrets.TOKEN_GH }}
+          publish_dir: ./docs/html
+          force_orphan: true

.github/workflows/mirror_to_gitlab.yml

@@ -0,0 +1,23 @@
+name: Mirror and run GitLab CI
+
+on: [push, delete]
+
+jobs:
+  build:
+    runs-on: ubuntu-latest
+    steps:
+    - uses: actions/checkout@v3
+      with:
+        fetch-depth: 0 
+    - name: Mirror + trigger CI
+      uses: SvanBoxel/gitlab-mirror-and-ci-action@master
+      with:
+        args: "https://git-dmz.thuenen.de/kida/i2-skills-beratungsstelle/active-service-requests/scraibe/scraibe"
+      env:
+        FOLLOW_TAGS: "true"
+        FORCE_PUSH: "true"
+        GITLAB_HOSTNAME: "git-dmz.thuenen.de"
+        GITLAB_USERNAME: ${{ secrets.GITLAB_USERNAME }} 
+        GITLAB_PASSWORD: ${{ secrets.GITLAB_PASSWORD }}
+        GITLAB_PROJECT_ID: ${{ secrets.GITLAB_PROJECT_ID }}
+        GITHUB_TOKEN: ${{ secrets.GH_TOKEN }}

.github/workflows/pypi.yml

@@ -0,0 +1,75 @@
+name: Publish Python 🐍 distribution 📦 to PyPI and TestPyPI
+
+on:
+  push:
+    tags:
+      - v* # Push tags to trigger the workflow
+  pull_request:
+    types: [closed]
+    branches:
+      - develop
+  workflow_dispatch:
+    inputs:
+      test:
+        description: "Push to TestPyPI not PyPI"
+        default: true
+        type: boolean
+
+jobs: 
+  Build-and-publish-to-Test-PyPI:
+    runs-on: ubuntu-latest
+    steps: 
+      - uses: actions/checkout@v4
+        with:
+          fetch-depth: '0'
+      - name: Set up Poetry 📦
+        uses: JRubics/poetry-publish@v1.16
+        with:
+          pypi_token: ${{ secrets.TEST_PYPI_API_TOKEN }}
+          plugins: "poetry-dynamic-versioning"
+          repository_name: "scraibe"
+          repository_url: "https://test.pypi.org/legacy/"
+
+  test-install:
+    name: Test Installation from TestPyPI
+    needs: Build-and-publish-to-Test-PyPI
+    runs-on: ubuntu-latest
+    strategy:
+      matrix:
+        python-version: [3.9, 3.11, 3.12]
+    steps:
+      - uses: actions/checkout@v4
+      - name: Set up Python ${{ matrix.python-version }}
+        uses: actions/setup-python@v5
+        with:
+          python-version: ${{ matrix.python-version }}
+      - name: Install package
+        run: |
+          pip install -U setuptools
+          pip install -r requirements.txt
+          python3 -m pip install --no-deps --pre --index-url https://test.pypi.org/simple/ scraibe>=0.1.3
+          python3 -c "import scraibe; print(scraibe.__version__)"
+
+  publish-to-pypi:
+    name: Publish to PyPI
+    needs: test-install
+    runs-on: ubuntu-latest
+    steps: 
+      - name: Checkout Repository Tags
+        uses: actions/checkout@v4
+        if: github.ref == 'refs/tags/v*'
+        with:
+          fetch-depth: '0'
+          branch: 'main'
+      - name: Checkout Repository (Develop)
+        uses: actions/checkout@v4
+        if: github.ref == 'refs/heads/develop'
+        with:
+          fetch-depth: '0'
+          branch: 'develop'
+      - name: Set up Poetry 📦
+        uses: JRubics/poetry-publish@v1.16
+        with:
+          pypi_token: ${{ secrets.PYPI_API_TOKEN }}
+          plugins: "poetry-dynamic-versioning"
+          repository_name: "scraibe"

.github/workflows/pytest.yaml

@@ -0,0 +1,39 @@
+name: Run Tests
+
+on:
+  pull_request:
+    branches: ['main', 'develop']
+  workflow_dispatch:
+
+jobs:
+  pytest:
+    runs-on: ubuntu-latest
+
+    steps:
+      - name: Checkout
+        uses: actions/checkout@v3
+        with:
+          fetch-depth: 0
+
+      - name: Setup Python
+        uses: actions/setup-python@v3
+        with:
+          python-version: 3.9
+
+      - name: Install Dependencies
+        run: |
+
+             sudo apt update && sudo apt upgrade
+             python -m pip install --upgrade pip
+             pip install -r requirements.txt
+             pip install .
+             sudo apt-get install libsndfile1-dev 
+             sudo apt-get install ffmpeg 
+             pip install pytest
+             
+      - name: Run pytest
+        env: 
+          HF_TOKEN : ${{ secrets.HF_TOKEN }}
+        run: |
+            pytest 
+           

.github/workflows/python-publish.yml

@@ -1,39 +0,0 @@
-# This workflow will upload a Python Package using Twine when a release is created
-# For more information see: https://docs.github.com/en/actions/automating-builds-and-tests/building-and-testing-python#publishing-to-package-registries
-
-# This workflow uses actions that are not certified by GitHub.
-# They are provided by a third-party and are governed by
-# separate terms of service, privacy policy, and support
-# documentation.
-
-name: Upload Python Package
-
-on:
-  release:
-    types: [published]
-
-permissions:
-  contents: read
-
-jobs:
-  deploy:
-
-    runs-on: ubuntu-latest
-
-    steps:
-    - uses: actions/checkout@v3
-    - name: Set up Python
-      uses: actions/setup-python@v3
-      with:
-        python-version: '3.x'
-    - name: Install dependencies
-      run: |
-        python -m pip install --upgrade pip
-        pip install build
-    - name: Build package
-      run: python -m build
-    - name: Publish package
-      uses: pypa/gh-action-pypi-publish@27b31702a0e7fc50959f5ad993c78deac1bdfc29
-      with:
-        user: __token__
-        password: ${{ secrets.PYPI_API_TOKEN }}

.github/workflows/ruff.yml

@@ -0,0 +1,9 @@
+name: Ruff
+on: push
+jobs:
+  ruff:
+    runs-on: ubuntu-latest
+    if: ${{ github.event_name == 'pull_request' || (github.event_name == 'push') }}
+    steps:
+      - uses: actions/checkout@v4
+      - uses: chartboost/ruff-action@v1

.github/workflows/semver.yml

@@ -0,0 +1,111 @@
+name: Semantic Versioning for Tags
+
+on:
+  pull_request:
+    types: [closed]
+    branches:
+      - main
+
+jobs:
+  bump-version:
+    if: ${{ github.event.pull_request.merged == true && github.event.pull_request.base.ref == 'main' }}
+    runs-on: ubuntu-latest
+    steps:
+      - name: Checkout Repository
+        uses: actions/checkout@v4
+        with:
+          fetch-depth: 0
+
+      - name: Check if Source Branch is docs
+        id: check_docs_branch
+        run: |
+          pr_head_ref="${{ github.event.pull_request.head.ref }}"
+          if [[ "$pr_head_ref" == "docs" ]]; then
+            echo "is_docs_branch=true" >> $GITHUB_ENV
+            echo "This is a docs branch merge. Exiting without creating a tag."
+            exit 0
+          else
+            echo "is_docs_branch=false" >> $GITHUB_ENV
+          fi
+
+      - name: Bump Version and Tag
+        if: env.is_docs_branch != 'true'
+        id: bump_version
+        env:
+          GITHUB_TOKEN: ${{ secrets.GH_TOKEN }}
+        run: |
+          # Fetch the latest tags from the remote
+          git fetch --tags
+
+          # Get the latest tag, or initialize to v0.0.0 if no tags are found
+          latest_tag=$(git describe --tags `git rev-list --tags --max-count=1` 2>/dev/null || echo "v0.0.0")
+
+          # Extract version from PR title or body
+          pr_body="${{ github.event.pull_request.body }}"
+          pr_title="${{ github.event.pull_request.title }}"
+          version_regex="v([0-9]+)\.([0-9]+)\.([0-9]+)"
+
+          if [[ $pr_body =~ $version_regex ]]; then
+            major=${BASH_REMATCH[1]}
+            minor=${BASH_REMATCH[2]}
+            patch=${BASH_REMATCH[3]}
+            new_tag="v$major.$minor.$patch"
+          elif [[ $pr_title =~ $version_regex ]]; then
+            major=${BASH_REMATCH[1]}
+            minor=${BASH_REMATCH[2]}
+            patch=${BASH_REMATCH[3]}
+            new_tag="v$major.$minor.$patch"
+          else
+            # Split the latest tag into parts
+            IFS='.' read -r -a parts <<< "${latest_tag#v}"
+            major=${parts[0]}
+            minor=${parts[1]}
+            patch=${parts[2]}
+            patch=$((patch + 1))
+            new_tag="v$major.$minor.$patch"
+          fi
+
+          echo "Bumping version from $latest_tag to $new_tag"
+
+          # Set the new tag as an environment variable
+          echo "new_tag=$new_tag" >> $GITHUB_ENV
+
+          # Tag the new version
+          git tag $new_tag
+
+          # Configure GitHub token authentication
+          git remote set-url origin https://x-access-token:${{ secrets.GH_TOKEN }}@github.com/${{ github.repository }}.git
+
+          # Push the new tag to the remote repository
+          git push origin $new_tag
+
+      - name: Extract Release Notes
+        if: env.is_docs_branch != 'true'
+        id: extract_notes
+        run: |
+          version="${{ env.new_tag }}"
+          clean_version="${version#v}"
+          release_notes=$(awk -v version="$clean_version" '
+            BEGIN { flag=0 }
+            # Start flagging when the version section is found
+            /^## \[.*\]/ {
+              if (flag) exit  # Exit when the next section starts
+            }
+            /^## \['"$clean_version"'\]/ { flag=1; next }  # Start printing after the header
+            flag { print }  # Print lines while flag is 1
+          ' CHANGELOG.md)
+          echo "RELEASE_NOTES<<EOF" >> $GITHUB_ENV
+          echo "$release_notes" >> $GITHUB_ENV
+          echo "EOF" >> $GITHUB_ENV
+
+      - name: Create Release
+        if: env.is_docs_branch != 'true'
+        uses: actions/create-release@v1
+        env:
+          GITHUB_TOKEN: ${{ secrets.GH_TOKEN }}
+        with:
+          tag_name: ${{ env.new_tag }}
+          release_name: Release ${{ env.new_tag }}
+          body: ${{ env.RELEASE_NOTES }}
+          draft: false
+          prerelease: false

.gitignore

@@ -0,0 +1,242 @@
+transcibe.py
+scraibe/*__pycache__
+scraibe/app/*__pycache__
+scraibe/.pyannotetoken
+
+
+# Created by https://www.toptal.com/developers/gitignore/api/python,visualstudiocode,linux,windows
+# Edit at https://www.toptal.com/developers/gitignore?templates=python,visualstudiocode,linux,windows
+
+### Linux ###
+*~
+
+# temporary files which can be created if a process still has a handle open of a deleted file
+.fuse_hidden*
+
+# KDE directory preferences
+.directory
+
+# Linux trash folder which might appear on any partition or disk
+.Trash-*
+
+# .nfs files are created when an open file is removed but is still being accessed
+.nfs*
+
+### Python ###
+# Byte-compiled / optimized / DLL files
+__pycache__/
+*.py[cod]
+*$py.class
+
+# C extensions
+*.so
+
+# Distribution / packaging
+.Python
+build/
+develop-eggs/
+dist/
+downloads/
+eggs/
+.eggs/
+lib/
+lib64/
+parts/
+sdist/
+var/
+wheels/
+share/python-wheels/
+*.egg-info/
+.installed.cfg
+*.egg
+MANIFEST
+
+# PyInstaller
+#  Usually these files are written by a python script from a template
+#  before PyInstaller builds the exe, so as to inject date/other infos into it.
+*.manifest
+*.spec
+
+# Installer logs
+pip-log.txt
+pip-delete-this-directory.txt
+
+# Unit test / coverage reports
+htmlcov/
+.tox/
+.nox/
+.coverage
+.coverage.*
+.cache
+nosetests.xml
+coverage.xml
+*.cover
+*.py,cover
+.hypothesis/
+.pytest_cache/
+cover/
+
+# Translations
+*.mo
+*.pot
+
+# Django stuff:
+*.log
+local_settings.py
+db.sqlite3
+db.sqlite3-journal
+
+# Flask stuff:
+instance/
+.webassets-cache
+
+# Scrapy stuff:
+.scrapy
+
+# Sphinx documentation
+docs/_build/
+
+# PyBuilder
+.pybuilder/
+target/
+
+# Jupyter Notebook
+.ipynb_checkpoints
+
+# IPython
+profile_default/
+ipython_config.py
+
+# pyenv
+#   For a library or package, you might want to ignore these files since the code is
+#   intended to run in multiple environments; otherwise, check them in:
+# .python-version
+
+# pipenv
+#   According to pypa/pipenv#598, it is recommended to include Pipfile.lock in version control.
+#   However, in case of collaboration, if having platform-specific dependencies or dependencies
+#   having no cross-platform support, pipenv may install dependencies that don't work, or not
+#   install all needed dependencies.
+#Pipfile.lock
+
+# poetry
+#   Similar to Pipfile.lock, it is generally recommended to include poetry.lock in version control.
+#   This is especially recommended for binary packages to ensure reproducibility, and is more
+#   commonly ignored for libraries.
+#   https://python-poetry.org/docs/basic-usage/#commit-your-poetrylock-file-to-version-control
+#poetry.lock
+
+# pdm
+#   Similar to Pipfile.lock, it is generally recommended to include pdm.lock in version control.
+#pdm.lock
+#   pdm stores project-wide configurations in .pdm.toml, but it is recommended to not include it
+#   in version control.
+#   https://pdm.fming.dev/#use-with-ide
+.pdm.toml
+
+# PEP 582; used by e.g. github.com/David-OConnor/pyflow and github.com/pdm-project/pdm
+__pypackages__/
+
+# Celery stuff
+celerybeat-schedule
+celerybeat.pid
+
+# SageMath parsed files
+*.sage.py
+
+# Environments
+.env
+.venv
+env/
+venv/
+ENV/
+env.bak/
+venv.bak/
+
+# Spyder project settings
+.spyderproject
+.spyproject
+
+# Rope project settings
+.ropeproject
+
+# mkdocs documentation
+/site
+
+# mypy
+.mypy_cache/
+.dmypy.json
+dmypy.json
+
+# Pyre type checker
+.pyre/
+
+# pytype static type analyzer
+.pytype/
+
+# Cython debug symbols
+cython_debug/
+
+# PyCharm
+#  JetBrains specific template is maintained in a separate JetBrains.gitignore that can
+#  be found at https://github.com/github/gitignore/blob/main/Global/JetBrains.gitignore
+#  and can be added to the global gitignore or merged into this file.  For a more nuclear
+#  option (not recommended) you can uncomment the following to ignore the entire idea folder.
+#.idea/
+
+### Python Patch ###
+# Poetry local configuration file - https://python-poetry.org/docs/configuration/#local-configuration
+poetry.toml
+
+# ruff
+.ruff_cache/
+
+# LSP config files
+pyrightconfig.json
+
+### VisualStudioCode ###
+.vscode/*
+!.vscode/settings.json
+!.vscode/tasks.json
+!.vscode/launch.json
+!.vscode/extensions.json
+!.vscode/*.code-snippets
+
+# Local History for Visual Studio Code
+.history/
+
+# Built Visual Studio Code Extensions
+*.vsix
+
+### VisualStudioCode Patch ###
+# Ignore all local history of files
+.history
+.ionide
+
+### Windows ###
+# Windows thumbnail cache files
+Thumbs.db
+Thumbs.db:encryptable
+ehthumbs.db
+ehthumbs_vista.db
+
+# Dump file
+*.stackdump
+
+# Folder config file
+[Dd]esktop.ini
+
+# Recycle Bin used on file shares
+$RECYCLE.BIN/
+
+# Windows Installer files
+*.cab
+*.msi
+*.msix
+*.msm
+*.msp
+
+# Windows shortcuts
+*.lnk
+
+# End of https://www.toptal.com/developers/gitignore/api/python,visualstudiocode,linux,windows

CHANGELOG.md

@@ -0,0 +1,34 @@
+# Changelog
+
+All notable changes to this project will be documented in this file.
+
+## [0.2.0] - 2024-05-28
+
+### Added
+
+- **Python Usage Section**: Detailed instructions on how to use ScrAIbe with Python, including examples for Whisper models, WhisperX, and keyword arguments.
+- **Command-line Usage Section**: Enhanced instructions for using ScrAIbe via the command-line interface, including examples and key options.
+- **Documentation Section**: Expanded the documentation section with highlights on installation guides, usage examples, API reference, troubleshooting tips, and advanced configuration.
+- **Getting Started Section**: Added detailed prerequisites and installation instructions for both stable and development versions of ScrAIbe.
+- **WhisperX Support**: Added support for the WhisperX backend.
+
+### Changed
+
+- **Model Customization**: Clarified the use of various keywords to customize Whisper models, Pyannote diarization models, and WhisperX.
+- **Example Enhancements**: Improved examples to illustrate the usage of different features and options in ScrAIbe.
+- **Formatting and Clarity**: Improved formatting and clarity across all sections to enhance readability and user experience.
+- **Backend Robustness**: Enhanced the backend to be more robust, removing the need for a HuggingFace token for basic usage.
+- **CLI**: to Work without Gradio
+
+### Removed
+
+- **Docker Build**: Removed Docker build support.
+- **Gradio App**: Removed the Gradio App integration.
+
+Both the Docker Build and the Gradio App are now Available under [ScrAIbe-WebUI](https://github.com/JSchmie/ScrAIbe-WebUI)
+
+### Documentation
+
+- **Documentation Page Link**: Updated the documentation section with a direct link to the [ScrAIbe documentation page](https://jschmie.github.io/ScrAIbe/).
+
+**Note**: This changelog might be incomplete, but we promise to improve it in the future. Thank you for your understanding and support.

CONTRIBUTING.md

@@ -0,0 +1,59 @@
+# Contributing to ScrAIbe
+
+Thank you for your interest in contributing to ScrAIbe! We appreciate your efforts to improve the project. Before making any changes, please discuss them with the project maintainers via an issue, email, or any other method.
+
+Please note that we have a code of conduct, and we ask you to adhere to it in all your interactions with the project.
+
+## Pull Request Process
+
+1. **Dependency Management**: Ensure any install or build dependencies are removed before the end of the layer when doing a build.
+2. **Documentation Updates**: Update the `README.md` with details of changes to the interface, including new environment variables, exposed ports, useful file locations, and container parameters.
+3. **Versioning**: Increase the version numbers in any example files and the `README.md` to the new version that this Pull Request would represent. We use the [SemVer](http://semver.org/) versioning scheme.
+4. **Review and Merge**: You may merge the Pull Request once you have the sign-off of two other developers. If you do not have permission to merge, request a second reviewer to merge it for you.
+
+## Code of Conduct
+
+### Our Pledge
+
+In the interest of fostering an open and welcoming environment, we as contributors and maintainers pledge to make participation in our project and our community a harassment-free experience for everyone, regardless of age, body size, disability, ethnicity, gender identity and expression, level of experience, nationality, personal appearance, race, religion, or sexual identity and orientation.
+
+### Our Standards
+
+Examples of behavior that contributes to creating a positive environment include:
+
+* Using welcoming and inclusive language
+* Being respectful of differing viewpoints and experiences
+* Gracefully accepting constructive criticism
+* Focusing on what is best for the community
+* Showing empathy towards other community members
+
+Examples of unacceptable behavior by participants include:
+
+* The use of sexualized language or imagery and unwelcome sexual attention or advances
+* Trolling, insulting/derogatory comments, and personal or political attacks
+* Public or private harassment
+* Publishing others' private information, such as a physical or electronic address, without explicit permission
+* Other conduct which could reasonably be considered inappropriate in a professional setting
+
+### Our Responsibilities
+
+Project maintainers are responsible for clarifying the standards of acceptable behavior and are expected to take appropriate and fair corrective action in response to any instances of unacceptable behavior.
+
+Project maintainers have the right and responsibility to remove, edit, or reject comments, commits, code, wiki edits, issues, and other contributions that are not aligned to this Code of Conduct, or to ban temporarily or permanently any contributor for other behaviors that they deem inappropriate, threatening, offensive, or harmful.
+
+### Scope
+
+This Code of Conduct applies both within project spaces and in public spaces when an individual is representing the project or its community. Examples include using an official project email address, posting via an official social media account, or acting as an appointed representative at an online or offline event. Representation of a project may be further defined and clarified by project maintainers.
+
+### Enforcement
+
+Instances of abusive, harassing, or otherwise unacceptable behavior may be reported by contacting the project team at [INSERT EMAIL ADDRESS]. All complaints will be reviewed and investigated and will result in a response that is deemed necessary and appropriate to the circumstances. The project team is obligated to maintain confidentiality with regard to the reporter of an incident. Further details of specific enforcement policies may be posted separately.
+
+Project maintainers who do not follow or enforce the Code of Conduct in good faith may face temporary or permanent repercussions as determined by other members of the project's leadership.
+
+### Attribution
+
+This Code of Conduct is adapted from the [Contributor Covenant][homepage], version 1.4, available at [http://contributor-covenant.org/version/1/4][version].
+
+[homepage]: http://contributor-covenant.org
+[version]: http://contributor-covenant.org/version/1/4/

Dockerfile

@@ -1,8 +1,20 @@
 #pytorch Image
 FROM pytorch/pytorch:1.11.0-cuda11.3-cudnn8-runtime
+
+# Labels
+
+LABEL maintainer="Jacob Schmieder"
+LABEL email="Jacob.Schmieder@dbfz.de"
+LABEL version="0.1.1.dev"
+LABEL description="Scraibe is a tool for automatic speech recognition and speaker diarization. \
+                    It is based on the Hugging Face Transformers library and the Pyannote library. \
+                    It is designed to be used with the Whisper model, a lightweight model for automatic \
+                    speech recognition and speaker diarization."
+LABEL url="https://github.com/JSchmie/ScrAIbe"
+
 # Install dependencies
 WORKDIR /app
-ARG hf_token
+ARG model_name=medium
 #Enviorment Dependncies
 ENV TRANSFORMERS_CACHE /app/models
 ENV HF_HOME /app/models
@@ -10,17 +22,25 @@ ENV AUTOT_CACHE /app/models
 ENV PYANNOTE_CACHE /app/models/pyannote
 #Copy all necessary files 
 COPY requirements.txt /app/requirements.txt
-COPY scraibe /app/Scraibe
+COPY README.md /app/README.md
+COPY models /app/models
+COPY scraibe /app/scraibe
 COPY setup.py /app/setup.py
+
 #Installing all necessary Dependencies and Running the Application with a personalised Hugging-Face-Token
+RUN apt update && apt-get install -y libsm6 libxrender1 libfontconfig1
+RUN conda update --all
+
 RUN conda install pip
 RUN conda install -y ffmpeg 
 RUN conda install -c conda-forge libsndfile
 RUN pip install torchaudio==0.11.0+cu113 -f https://download.pytorch.org/whl/torch_stable.html
-RUN pip install /app/ 
+RUN pip install -r requirements.txt
 RUN pip install markupsafe==2.0.1 --force-reinstall
-RUN Scraibe  --hf_token $hf_token
+
+RUN python3 -m 'scraibe.cli' --whisper-model-name $model_name
 # Expose port
 EXPOSE 7860
 # Run the application
-ENTRYPOINT ["scraibe"]  
+
+ENTRYPOINT ["python3", "-m",  "scraibe.cli" ,"--whisper-model-name", "$model_name"]  

Makefile

@@ -0,0 +1,20 @@
+# Minimal makefile for Sphinx documentation
+#
+
+# You can set these variables from the command line, and also
+# from the environment for the first two.
+SPHINXOPTS    ?=
+SPHINXBUILD   ?= sphinx-build
+SOURCEDIR     = source
+BUILDDIR      = build
+
+# Put it first so that "make" without argument is like "make help".
+help:
+	@$(SPHINXBUILD) -M help "$(SOURCEDIR)" "$(BUILDDIR)" $(SPHINXOPTS) $(O)
+
+.PHONY: help Makefile
+
+# Catch-all target: route all unknown targets to Sphinx using the new
+# "make mode" option.  $(O) is meant as a shortcut for $(SPHINXOPTS).
+%: Makefile
+	@$(SPHINXBUILD) -M $@ "$(SOURCEDIR)" "$(BUILDDIR)" $(SPHINXOPTS) $(O)

README.md

@@ -1,199 +1,173 @@
+# `ScrAIbe: Streamlined Conversation Recording with Automated Intelligence Based Environment` 🎙️🧠
 
-# `ScrAIbe: Streamlined Conversation Recording with Automated Intelligence Based Environment`
+Welcome to `ScrAIbe`, a state-of-the-art, [PyTorch](https://pytorch.org/) based multilingual speech-to-text framework designed to generate fully automated transcriptions. 
 
-`ScrAIbe` is a state-of-the-art,  [PyTorch](https://pytorch.org/) based multilingual speech-to-text framework to generate fully automated transcriptions. 
+Beyond transcription, ScrAIbe supports advanced functions such as speaker diarization and speaker recognition. 🚀
 
-Beyond transcription, ScrAIbe supports advanced functions, such as speaker diarization and speaker recognition.
+Designed as a comprehensive AI toolkit, it uses multiple powerful AI models:
 
-Designed as a comprehensive AI toolkit, it uses multiple AI models:
+- **[Whisper](https://github.com/openai/whisper)**: A general-purpose speech recognition model.
-
+- **[WhisperX](https://github.com/m-bain/whisperX)**: A faster, quantized version of Whisper for enhanced performance on CPU. ⚡
-- [whisper](https://github.com/openai/whisper): A general-purpose speech recognition model.
+- **[Pyannote-Audio](https://github.com/pyannote/pyannote-audio)**: An open-source toolkit for speaker diarization. 🗣️
-- [payannote-audio](https://github.com/pyannote/pyannote-audio): An open-source toolkit for speaker diarization.
 
 The framework utilizes a PyanNet-inspired pipeline, with the `Pyannote` library for speaker diarization and `VoxCeleb` for speaker embedding.
 
-During post-diarization, each audio segment is processed by the OpenAI `Whisper` model, in a transformer encoder-decoder structure. Initially, a CNN mitigates noise and enhances speech. Before transcription, `VoxLingua` identifies the language segment, facilitating Whisper's role in both transcription and text translation.
+During post-diarization, each audio segment is processed by the OpenAI `Whisper` model in a transformer encoder-decoder structure. Initially, a CNN mitigates noise and enhances speech. Before transcription, `VoxLingua` identifies the language segment, facilitating Whisper's role in both transcription and text translation. 🌍✨
 
 The following graphic illustrates the whole pipeline:
 
-![Pipeline](Pictures/pipeline.png#gh-dark-mode-only) 
+<div style="text-align:center;">
-![Pipeline](Pictures/pipeline_light.png#gh-light-mode-only) 
+  <img src="./Pictures/pipeline.png#gh-dark-mode-only" style="width: 60%;" />
+  <img src="./Pictures/pipeline_light.png#gh-light-mode-only" style="width: 60%;" />
+</div>
 
-## Install `ScrAIbe` : 
+## Getting Started 🚀
 
-The following command will pull and install the latest commit from this repository, along with its Python dependencies.
+### Prerequisites
 
-    pip install scraibe
+Before installing ScrAIbe, ensure you have the following prerequisites:
 
-- **Python version**: Python 3.8
+- **Python**: Version 3.9 or later.
-- **PyTorch version**: Python 1.11.0
+- **PyTorch**: Version 2.0 or later.
-- **CUDA version**: Cuda-toolkit 11.3.1
+- **CUDA**: A compatible version with your PyTorch Version if you want to use GPU acceleration.
 
+**Note:** PyTorch should be automatically installed with the pip installer. However, if you encounter any issues, you should consider installing it manually by following the instructions on the [PyTorch website](https://pytorch.org/get-started/locally/).
 
-Important: For the `Pyannote` model, you need to be granted access to Hugging Face.
+### Install ScrAIbe 
-Check the [Pyannote model page](https://huggingface.co/pyannote/speaker-diarization) to get access to the model.
 
-Additionally, you need to generate a [Hugging Face token](https://huggingface.co/docs/hub/security-tokens). 
+Install ScrAIbe on your local machine with ease using PyPI.
+
+```bash
+pip install scraibe
+```
+
+If you want to install the development version, you can do so by installing it from GitHub:
+
+```bash
+pip install git+https://github.com/JSchmie/ScrAIbe.git@develop
+```
+
+or from PyPI using our latest pre-release:
+
+```bash
+pip install --pre scraibe
+```
+
+Get started with ScrAIbe today and experience seamless, automated transcription and diarization.
 
 ## Usage
 
 We've developed ScrAIbe with several access points to cater to diverse user needs.
 
-### Python usage
+### Python Usage
 
-It enables full control over the functionalities as well as process customization. 
+Gain full control over the functionalities as well as process customization.
 
 ```python
 from scraibe import Scraibe
 
-model = Scraibe(use_auth_token = "hf_yourhftoken")
+model = Scraibe()
 
 text = model.autotranscribe("audio.wav")
 
 print(f"Transcription: \n{text}")
 ```
-The `Scraibe` Class is taking care of the models being properly loaded. Therefore, you can choose the other [whisper](https://github.com/openai/whisper/blob/main/model-card.md) models using the `whisper_model` keyword. 
-You can also change the `pyannote` diarization model using the `dia_model` keyword.
 
+The `Scraibe` class ensures the models are properly loaded. You can customize the models with various keywords:
 
-As input, `autoranscribe` accepts every format which is compatible with [ffmgeg](https://ffmpeg.org/ffmpeg-formats.html). Examples therefore are `.mp4 .mp3 .wav .ogg .flac` and many more.
+- **Whisper Models**: Use the `whisper_model` keyword to specify models like `tiny`, `base`, `small`, `medium`, or `large` (`large-v2`, `large-v3`) depending on your accuracy and speed needs.
+- **Pyannote Diarization Model**: Use the `dia_model` keyword to change the diarization model.
+- **WhisperX**: Set the `whisper_type` to `"whisperX"` for enhanced performance on CPU and use their enhanced models. (Model names are the same)
+- **Keyword Arguments**: A variety of different `kwargs` are available:
+  - `use_auth_token`: Pass a Hugging Face token to the Pyannote backend if you want to use one of the models hosted on their Hugging Face.
+  - `verbose`: Enable this to add an additional level of verbosity.
   
-To further control the pipeline of `ScrAIbe` you can parse almost any keyword you also cloud parsed towards `whisper` or `pyannote` if you need more option, try to check out the documentations  tows two Frameworks, you might have a good chance that these keywords will work here as well. 
+  In general, you should be able to input any `kwargs` that you can input in the original Whisper (WhisperX) and Pyannote Python APIs.
-Here's are some examples regarding the `diarization` (which relies on the `pyannote` pipeline):
 
-- `num_speakers` Number of speakers in the audio file
+As input, `autotranscribe` accepts every format compatible with [FFmpeg](https://ffmpeg.org/ffmpeg-formats.html). Examples include `.mp4`, `.mp3`, `.wav`, `.ogg`, `.flac`, and many more.
-- `min_speakers` Minimal Number of speakers in the audio file 
-- `max_speakers` maximal Number of speakers in the audio file
 
-Then there are arguments about the transcription process, which uses the "whisper" model.
+To further control the pipeline of `ScrAIbe`, you can pass almost any keyword argument that is accepted by `Whisper` or `Pyannote`. For more options, refer to the documentation of these frameworks, as their keywords are likely to work here as well. 
 
-- `language` Specify the language ([list to supported languages](https://github.com/openai/whisper/blob/main/language-breakdown.svg)) 
+Here are some examples regarding `diarization` (which relies on the `pyannote` pipeline):
-- `task` can be just `transcribe` or `translate`. If `translate` is selected, the transcribed audio will be translated to English.
+
+- `num_speakers`: Number of speakers in the audio file
+- `min_speakers`: Minimum number of speakers in the audio file
+- `max_speakers`: Maximum number of speakers in the audio file
+
+Then there are arguments for the transcription process, which uses the "Whisper" model:
+
+- `language`: Specify the language ([list of supported languages](https://github.com/openai/whisper/blob/main/language-breakdown.svg))
+- `task`: Can be either `transcribe` or `translate`. If `translate` is selected, the transcribed audio will be translated to English.
 
 For example:
 
-```
+```python
 text = model.autotranscribe("audio.wav", language="german", num_speakers = 2)
 ```
 
-`Scraibe` also contains the option to just do a transcription
+`Scraibe` also contains the option to just do a transcription:
+
 ```python
 transcription = model.transcribe("audio.wav")
 ``` 
+
 or just do a diarization:
 
 ```python
-diarization = model.diarize("audio.wav")
+diarization = model.diarization("audio.wav")
 ```
 
+Start exploring the powerful features of ScrAIbe and customize it to fit your specific transcription and diarization needs!
+
 ### Command-line usage
 
 Next to the Pyhton interface, you can also run ScrAIbe using the command-line interface:
 
-    scraibe -f "audio.wav" --hf-token "hf_yourhftoken" --language "german" --num_speakers 2
+```bash
+scraibe -f "audio.wav" --language "german" --num_speakers 2
+```
 
 For the full list of options, run:
 
-    scraibe -h
+```bash
-
+scraibe -h
-### Gradio App
-
-The Gradio App is a user-friendly interface for ScrAIbe. It enables you to run the model without any coding knowledge. Therefore, you can run the app in your browser and upload your audio file, or you can make the Framework avail on your network and run it on your local machine.
-
-#### Running the Gradio App on your local machine
-
-To run the Gradio App on your local machine, just use the following command:
-
-```
-scraibe --start_server --port 7860 --hf_token hf_yourhftoken
 ```
 
-- `--start_server`: Command to start the Gradio App.
+This will display a comprehensive list of all command-line options, allowing you to tailor ScrAIbe’s functionality to your specific needs.
-- `--port`: Flag for connecting the container internal port to the port on your local machine.
-- `--hf_token`: Flag for entering your personal HuggingFace token in the container.
 
-When the app is running, it will show you at which address you can access it.
+## Gradio App 🌐
-The default address is: http://127.0.0.1:7860 or http://0.0.0.0:7860
 
-After the app is running, you can upload your audio file and select the desired options.
+The Gradio App is now part of ScrAIbe-WebUI! This user-friendly interface enables you to run the model without any coding knowledge. You can easily run the app in your browser and upload your audio files, or make the framework available on your network and run it on your local machine. 🚀
-An example is shown below:
 
-![Gradio App](Pictures/gradio_app.png)
+All functionalities previously available in the Gradio App are now part of the ScrAIbe-WebUI. For more information and detailed instructions, visit the [ScrAIbe-WebUI GitHub repository](https://github.com/JSchmie/ScrAIbe-WebUI).
+
+## Docker Container 🐳
+
+ScrAIbe's Docker containers have also moved to ScrAIbe-WebUI! This option is especially useful if you want to run the model on a server or if you would like to use the GPU without dealing with CUDA.
+
+All Docker container functionalities are now part of ScrAIbe-WebUI. For more information and detailed instructions on how to use the Docker containers, please visit the [ScrAIbe-WebUI GitHub repository](https://github.com/JSchmie/ScrAIbe-WebUI).
+
+---
+
+With these changes, ScrAIbe focuses on its core functionalities while the enhanced Gradio App and related Docker containers are now part of ScrAIbe-WebUI. Enjoy a more streamlined and powerful transcription experience! 🎉
+
+## Documentation 📚
+
+For comprehensive guides, detailed instructions, and advanced usage tips, visit our [documentation page](https://jschmie.github.io/ScrAIbe/). Here, you will find everything you need to make the most out of ScrAIbe.
+
+### Contributions 🤝
+
+We warmly welcome contributions from the community! Whether you’re fixing bugs, adding new features, or improving documentation, your help is invaluable. Please see our [Contributing Guidelines](./CONTRIBUTING.md) for more information on how to get involved and make your mark on ScrAIbe-WebUI.
 
 
-### Running a Docker container
+### License 📜
 
-Another option to run ScrAIbe is to use a Docker container. This option is especially useful if you want to run the model on a server or if you would like to use the GPU without dealing with CUDA.
+ScrAIbe-WebUI is proudly open source and licensed under the GPL-3.0 license. This promotes a collaborative and transparent development process. For more details, see the [LICENSE](./LICENSE) file in this repository.
-After you have installed Docker, you can execute the following commands in the terminal.
-
-First, you need to build the Docker image. Therefore, you need to enter your HuggingFace token and the image name.
-
-```
-docker build . --build-arg="hf_token=[enter your HuggingFace token] " -t scraibe
-```
-
-After the image is built, you can run the container with the following command:
-
-```
-sudo docker run -it  -p 7860:7860  --name [container name][image name]  --hf_token [enter your HuggingFace token] --start_server
-
-```
--  `-p`: Flag for connecting the container internal port to the port on your local machine.
--  `--hf_token`: Flag for entering your personal HuggingFace token in the container.
-- `--start_server`: Command to start the Gradio App.
-
-Inside the container, the `cli` is used. Therefore, you can use the same commands as in the command-line interface.
-
-#### Enabling GPU usage
-
-To use the GPU, ensure your Docker installation supports GPU usage.
-For further information, check: https://docs.nvidia.com/datacenter/cloud-native/container-toolkit/install-guide.html#docker
-To enable GPU usage, you need to add the following flag to the `docker run` command:
-
-```
-docker run -it  -p 7860:7860 --gpus 'all,capabilities=utility'  --name [container name][image name]  --hf_token [enter your HuggingFace token] --start_server
-```
-
-For further guidance, check: https://blog.roboflow.com/use-the-gpu-in-docker/ 
-
-## Documentation 
-
-For further insights, check the [documentation page]().
-
-## Contributions
-
-We are happy to have any interest in contributing and about feedback: In order to do that, create an issue with your feedback or feel free to contact us.
-
-## Roadmap
-
-The following milestones are planned for further releases of ScrAIbe:
-
-- Model quantization   
-Quantization to empower memory and computational efficiency.
-
-- Model fine-tuning  
-In order to be able to cover a variety of linguistic phenomena.
-
-For example, currently ScrAIbe is able to transcribe word by word, but ignores filler words or speech pauses. 
-These phenomena can be addressed by fine-tuning with the corresponding data.
-
-- Implementation of LLMs   
-One example is the implementation of a summarization or extraction model, which enables ScrAIbe to automatically summarize or retrieve the key information out of a generated transcription, which could be the minutes of a meeting.
-
-- Executable for Windows
-
-## Contact
-
-For queries contact [Jacob Schmieder](Jacob.Schmieder@dbfz.de)
-
-## License 
-
-ScrAIbe is licensed under GNU General Public License.
 
 ## Acknowledgments
 
-Special thanks go to the KIDA project and the BMEL (Bundesministerium für Ernährung und Landwirtschaft), especially to the AI Consultancy Team.
+Special thanks go to the [KIDA](https://www.kida-bmel.de/) project and the [BMEL (Bundesministerium für Ernährung und Landwirtschaft)](https://www.bmel.de/EN/Home/home_node.html), especially to the AI Consultancy Team.
 
-![KIDA](Pictures/kida_dark.png#gh-dark-mode-only)    &nbsp;    ![BMEL](Pictures/BMEL_dark.png#gh-dark-mode-only) &nbsp;&nbsp;&nbsp;&nbsp; ![DBFZ](Pictures/DBFZ_dark.png#gh-dark-mode-only)   &nbsp;  &nbsp;&nbsp;&nbsp;    ![MRI](Pictures/MRI.png#gh-dark-mode-only)   
+---
 
-![KIDA](Pictures/kida.png#gh-light-mode-only)    &nbsp;    ![BMEL](Pictures/BMEL.jpg#gh-light-mode-only) &nbsp;&nbsp;&nbsp;&nbsp; ![DBFZ](Pictures/DBFZ.png#gh-light-mode-only)   &nbsp;  &nbsp;&nbsp;&nbsp;    ![MRI](Pictures/MRI.png#gh-light-mode-only)  
+Join us in making ScrAIbe even better! 🚀

pyproject.toml

@@ -0,0 +1,73 @@
+[build-system]
+requires = ["poetry-core>=1.8.0", "poetry-dynamic-versioning>=1.0.0,<2.0.0"]
+build-backend = "poetry_dynamic_versioning.backend"
+
+[tool.poetry]
+name = "scraibe"
+version = "0.0.0"
+description = "Transcription tool for audio files based on Whisper and Pyannote"
+authors = ["Schmieder, Jacob <jacob.schmieder@dbfz.de>"]
+license = "GPL-3.0-or-later"
+readme = ["README.md", "LICENSE"]
+repository = "https://github.com/JSchmie/ScAIbe"
+documentation = "https://jschmie.github.io/ScrAIbe/"
+keywords = ["transcription", "audio", "whisper", "pyannote", "speech-to-text", "speech-recognition"]
+classifiers = [
+                'Development Status :: 4 - Beta',
+                'Intended Audience :: Developers',
+                'License :: OSI Approved :: GNU General Public License v3 (GPLv3)',
+                'Programming Language :: Python :: 3.8',
+                'Programming Language :: Python :: 3.9',
+                'Programming Language :: Python :: 3.10',
+                'Programming Language :: Python :: 3.11',
+                'Environment :: GPU :: NVIDIA CUDA :: 12 :: 12.1',
+                'Topic :: Scientific/Engineering :: Artificial Intelligence'
+                ]
+packages = [{include = "scraibe"}]
+exclude =[
+        "__pycache__",
+        "*.pyc",
+        "test"
+        ]
+[tool.poetry.dependencies]
+python = "^3.9"
+tqdm = "^4.66.4"
+numpy = "^1.26.4"
+openai-whisper = "^20231117"
+whisperx = "^3.1.3"
+"pyannote.audio" = "^3.1.1"
+torch = "^2.3.0"
+
+[tool.poetry.group.dev.dependencies]
+pytest = "^8.1.1"
+
+[tool.poetry-dynamic-versioning]
+enable = true
+vcs = "git"
+strict = true
+format-jinja = """
+    {%- if distance == 0 -%}
+        {{ serialize_pep440(base) }}
+    {%- elif branch == 'develop' -%}
+        {{ serialize_pep440(bump_version(base), dev = distance) }}
+    {%- else -%}
+        {{ serialize_pep440(bump_version(base), dev=distance, metadata=[commit]) }}
+    {%- endif -%}
+"""
+
+[tool.poetry.group.docs.dependencies]
+sphinx = "^7.3.7"
+sphinx-rtd-theme = "^2.0.0"
+markdown-it-py = {version = "~3.0.0", extras = ["plugins"]}
+myst-parser = "^3.0.1"
+mdit-py-plugins = "^0.4.1"
+
+[tool.poetry.scripts]
+scraibe = "scraibe.cli:cli"
+
+[tool.poetry.extras]
+app = ["scraibe-webui"]
+
+[tool.ruff.lint.extend-per-file-ignores]
+"__init__.py" = ["E402","F403",'F401']
+"scraibe/misc.py" = ["E722"]

requirements.txt

@@ -1,25 +1,14 @@
-openai-whisper==20230314
-
-pyannote.audio~=2.1.1
-pyannote.core~=4.5
-pyannote.database~=4.1.3
-pyannote.metrics~=3.2.1
-pyannote.pipeline~=2.3
-
-setuptools~=69.0.3
-setuptools-rust~=1.8.1
-
 tqdm>=4.65.0
+numpy>=1.26.4
 
-gradio~=3.36.1
+openai-whisper==20231117
-gradio-client~=0.2.7
+whisperx~=3.1.3
 
-# add pytorch to override the one installed by pyannote.audio
+pyannote.audio~=3.1.1
-
+pyannote.core~=5.0.0
-torch~=1.11.0
+pyannote.database~=5.0.1 
-torchvision~=0.12.0
+pyannote.metrics~=3.2.1
-torchaudio~=0.11.0
+pyannote.pipeline~=3.0.1
-#optional: 
-#sphinx~=5.0.2 
 
+torch>=2.0.0
 

scraibe/.pyannotetoken

@@ -1 +0,0 @@
-

scraibe/__init__.py

@@ -4,12 +4,8 @@ from .audio import *
 from .transcript_exporter import *
 from .diarisation import *
 
-from .version import get_version as _get_version
 from .misc import *
 
-from .app.gradio_app import *
-from .app.qtfaststart import *
-
 from .cli import *
 
-__version__ = _get_version()
+from ._version import __version__

scraibe/_version.py

@@ -0,0 +1 @@
+__version__ = "0.0.0"

scraibe/app/__init__.py

@@ -1,2 +0,0 @@
-from .qtfaststart import *
-from .gradio_app import *

scraibe/app/gradio_app.py

@@ -1,438 +0,0 @@
-"""
-Gradio Audio Transcription App.
---------------------------------
-
-This module provides an interface to transcribe audio files using the 
-Scraibe model. Users can either upload an audio file or record their speech 
-live for transcription. The application supports multiple languages and provides 
-options to specify the number of speakers and the language of the audio.
-
-Attributes:
-    LANGUAGES (list): A list of supported languages for transcription.
-
-Usage:
-    Run this script to start the Gradio web interface for audio transcription.
-    
-"""
-
-"""
-Gradio Audio Transcription App.
---------------------------------
-
-This module provides an interface to transcribe audio files using the 
-Scraibe model. Users can either upload an audio file or record their speech 
-live for transcription. The application supports multiple languages and provides 
-options to specify the number of speakers and the language of the audio.
-
-Attributes:
-    LANGUAGES (list): A list of supported languages for transcription.
-
-Usage:
-    Run this script to start the Gradio web interface for audio transcription.
-    
-"""
-
-import json
-import os
-
-
-import gradio as gr
-from tqdm import tqdm
-
-from scraibe import Scraibe, Transcript
-
-theme = gr.themes.Soft(
-    primary_hue="green",
-    secondary_hue='orange',
-    neutral_hue="gray",  
-)
-
-LANGUAGES = [
-    "Afrikaans", "Arabic", "Armenian", "Azerbaijani", "Belarusian",
-    "Bosnian", "Bulgarian", "Catalan", "Chinese", "Croatian",
-    "Czech", "Danish", "Dutch", "English", "Estonian",
-    "Finnish", "French", "Galician", "German", "Greek",
-    "Hebrew", "Hindi", "Hungarian", "Icelandic", "Indonesian",
-    "Italian", "Japanese", "Kannada", "Kazakh", "Korean",
-    "Latvian", "Lithuanian", "Macedonian", "Malay", "Marathi",
-    "Maori", "Nepali", "Norwegian", "Persian", "Polish",
-    "Portuguese", "Romanian", "Russian", "Serbian", "Slovak",
-    "Slovenian", "Spanish", "Swahili", "Swedish", "Tagalog",
-    "Tamil", "Thai", "Turkish", "Ukrainian", "Urdu",
-    "Vietnamese", "Welsh"
-]
-
-CURRENT_PATH = os.path.dirname(os.path.realpath(__file__))
-
-class GradioTranscriptionInterface:
-    """
-    Interface handling the interaction between Gradio UI and the Audio Transcription system.
-    """
-
-    def __init__(self, model: Scraibe):
-        """
-        Initializes the GradioTranscriptionInterface with a transcription model.
-
-        Args:
-            model (Scraibe): Model responsible for audio transcription tasks.
-        """
-        self.model = model
-
-    def auto_transcribe(self, source,
-                        num_speakers : int,
-                        translation : bool,
-                        language : str):
-        """
-        Shortcut method for the Scraibe task.
-
-        Returns:
-            tuple: Transcribed text (str), JSON output (dict)
-        """
-        
-        kwargs = {
-            "num_speakers": num_speakers if num_speakers != 0 else None,
-            "language": language if language != "None" else None,
-            "task": 'translate' if translation else None
-        }
-        if isinstance(source, str):
-            try:
-                result = self.model.autotranscribe(source, **kwargs)
-            except ValueError:
-                raise gr.Error("Couldn't detect any speech in the provided audio. \
-                        Please try again!")
-            
-            return str(result), result.get_json()
-        
-        elif isinstance(source, list):
-            source_names = [s.split("/")[-1] for s in source]
-            result = []
-            for s in tqdm(source, total=len(source),desc = "Transcribing audio files"):
-                try:
-                    res = self.model.autotranscribe(s, **kwargs)
-                except ValueError:
-                    _name = s.split("/")[-1]
-                    res = f"NO TRANSCRIPT FOUND FOR {_name}"
-                    gr.Warning(f"Couldn't detect any speech in {_name} will skip this file.")
-                result.append(res)
-            
-            out = ''
-            out_dict = {}
-            for i, r in enumerate(result):
-                out += f"TRANSCRIPT {i} FOR ({source_names[i]}):\n\n"
-                out += str(r)
-                out += "\n\n"
-                
-                if isinstance(r, str):
-                    out_dict[source_names[i]] = r
-                else:
-                    out_dict[source_names[i]] = r.get_dict()
-              
-            return out, json.dumps(out_dict, indent=4)
-        
-        else:
-            raise gr.Error("Please provide a valid audio file.")
-
-
-    def transcribe(self, source, translation, language):
-        """
-        Shortcut method for the Transcribe task.
-
-        Returns:
-            str: Transcribed text.
-        """
-        kwargs = {
-            "language": language if language != "None" else None,
-            "task": 'translate' if translation == "Yes" else None
-        }
-        
-        if isinstance(source, str):
-            result = self.model.transcribe(source, **kwargs)
-            
-            return str(result)
-        
-        elif isinstance(source, list):
-            source_names = [s.split("/")[-1] for s in source]
-            result = []
-            for s in tqdm(source, total=len(source),desc = "Transcribing audio files"):
-                res = self.model.transcribe(s, **kwargs)
-                result.append(res)
-            
-            out = ''
-            for i, res in enumerate(result):
-                out += f"TRANSCRIPT {i} FOR ({source_names[i]}):\n\n"
-                out += str(res)
-                out += "\n\n"
-            
-            return out
-        
-        else:
-            raise gr.Error("Please provide a valid audio file.")
-
-    def perform_diarisation(self, source, num_speakers):
-        """
-        Shortcut method for the Diarisation task.
-
-        Returns:
-            str: JSON output of diarisation result.
-        """
-        kwargs = {
-            "num_speakers": num_speakers if num_speakers != 0 else None,
-        }
-        
-        if isinstance(source, str):
-            try:
-                result = self.model.diarization(source, **kwargs)
-            except ValueError:
-                raise gr.Error("Couldn't detect any speech in the provided audio. \
-                        Please try again!")
-        
-            return json.dumps(result, indent=2)
-        elif isinstance(source, list):
-            source_names = [s.split("/")[-1] for s in source]
-            result = []
-            for s in tqdm(source, total=len(source),desc = "Performing diarisation"):
-                try:
-                    res = self.model.diarization(s, **kwargs)
-                except ValueError:
-                    res = f"NO DIARISATION FOUND FOR {s}"
-                    gr.Warning(f"Couldn't detect any speech in {s} will skip this file.")
-                result.append(res)
-            
-            out = {}
-            
-            for i, res in enumerate(result):
-                out[source_names[i]] = res
-                
-            return json.dumps(out, indent=4)
-        
-        else:
-            gr.Error("Please provide a valid audio file.")
-            
-
-####
-# Gradio Interface
-####
-
-def gradio_Interface(model : Scraibe = None):
-    
-    if model is None:
-        model = Scraibe()
-        
-    pipe = GradioTranscriptionInterface(model)
-
-    def select_task(choice):
-        if choice == 'Auto Transcribe':
-            
-            return (gr.update(visible = True),
-                    gr.update(visible = True),
-                    gr.update(visible = True))
-                    
-            
-        elif choice == 'Transcribe':
-            
-            return (gr.update(visible = False),
-                    gr.update(visible = True),
-                    gr.update(visible = True))
-
-            
-        elif choice == 'Diarisation':
-            
-            return (gr.update(visible = True),
-                    gr.update(visible = False),
-                    gr.update(visible = False))
-        
-    def select_origin(choice):
-        if choice == "Upload Audio":
-            
-            return (gr.update(visible = True),
-                    gr.update(visible = False, value = None),
-                    gr.update(visible = False, value = None),
-                    gr.update(visible = False, value = None),
-                    gr.update(visible = False, value = None))
-        
-        elif choice == "Record Audio":
-            
-            return (gr.update(visible = False, value = None),
-                    gr.update(visible = True),
-                    gr.update(visible = False, value = None),
-                    gr.update(visible = False, value = None),
-                    gr.update(visible = False, value = None))
-
-        elif choice == "Upload Video":
-            
-            return (gr.update(visible = False, value = None),
-                    gr.update(visible = False, value = None),
-                    gr.update(visible = True),
-                    gr.update(visible = False, value = None),
-                    gr.update(visible = False, value = None))
-        
-        elif choice == "Record Video":
-            
-            return (gr.update(visible = False, value = None),
-                    gr.update(visible = False, value = None),
-                    gr.update(visible = False, value = None),
-                    gr.update(visible = True),
-                    gr.update(visible = False, value = None))
-            
-        elif choice == "File or Files":
-            
-            return (gr.update(visible = False, value = None),
-                    gr.update(visible = False, value = None),
-                    gr.update(visible = False, value = None),
-                    gr.update(visible = False, value = None),
-                    gr.update(visible = True))
-
-    def run_scribe(task,
-                   num_speakers,
-                   translate,
-                   language,
-                   audio1,
-                   audio2,
-                   video1,
-                   video2,
-                   file_in,
-                   progress = gr.Progress(track_tqdm= True)):
-        # get *args which are not None
-        progress(0, desc='Starting task...')
-        source = audio1 or audio2 or video1 or video2 or file_in
-        
-        if isinstance(source, list):
-            source = [s.name for s in source]
-            if len(source) == 1:
-                source = source[0]
- 
-        if task == 'Auto Transcribe':
-    
-            out_str , out_json = pipe.auto_transcribe(source = source,
-                                num_speakers = num_speakers,
-                                translation = translate,
-                                language = language)
-            
-            if isinstance(source, str):
-                return (gr.update(value = out_str, visible = True),
-                        gr.update(value = out_json, visible = True),
-                        gr.update(visible = True),
-                        gr.update(visible = True))      
-            else:
-                return (gr.update(value = out_str, visible = True),
-                        gr.update(value = out_json, visible = True),
-                        gr.update(visible = False),
-                        gr.update(visible = False))  
-            
-        elif task == 'Transcribe':
-            
-            out = pipe.transcribe(source = source,
-                                translation = translate,
-                                language = language)
-            
-            return (gr.update(value = out, visible = True),
-                    gr.update(value = None, visible = False),
-                    gr.update(visible = False),
-                    gr.update(visible = False))
-            
-        elif task == 'Diarisation':
-            
-            out = pipe.perform_diarisation(source = source,
-                                num_speakers = num_speakers)
-            
-            return (gr.update(value = None, visible = False),
-                    gr.update(value = out, visible = True),
-                    gr.update(visible = False),
-                    gr.update(visible = False))
-        
-    def annotate_output(annoation : str, out_json : dict):
-        # get *args which are not None
-        
-        trans = Transcript.from_json(out_json)
-        trans = trans.annotate(*annoation.split(","))
-
-        return gr.update(value = str(trans)),gr.update(value = trans.get_json())
-        
-        
-    with gr.Blocks(theme=theme,title='ScrAIbe: Automatic Audio Transcription') as demo:
-            
-        # Define components
-        hname = os.path.join(CURRENT_PATH, "header.html")
-        header = open(hname, "r").read()
-        gr.HTML(header, visible= True, show_label=False)
-        
-        with gr.Row():
-            
-            with gr.Column():
-            
-                task = gr.Radio(["Auto Transcribe", "Transcribe", "Diarisation"], label="Task",
-                                value= 'Auto Transcribe')
-                
-                num_speakers = gr.Number(value=0, label= "Number of speakers (optional)", 
-                                info = "Number of speakers in the audio file. If you don't know,\
-                                    leave it at 0.", visible= True)
-                
-                translate = gr.Checkbox(label="Translation", choices=[True, False], value = False,
-                                info="Select 'Yes' to have the output translated into English.",
-                                visible= True)
-                
-                language = gr.Dropdown(LANGUAGES,
-                                label="Language (optional)", value = "None",
-                                info="Language of the audio file. If you don't know,\
-                                    leave it at None.", visible= True)
-                
-                input = gr.Radio(["Upload Audio", "Record Audio", "Upload Video","Record Video" 
-                                    ,"File or Files"], label="Input Type", value="Upload Audio")
-                
-                audio1 = gr.Audio(source="upload", type="filepath", label="Upload Audio",
-                                    interactive= True, visible= True)
-                audio2 = gr.Audio(source="microphone", label="Record Audio", type="filepath",
-                                    interactive= True, visible= False)
-                video1 = gr.Video(source="upload", type="filepath", label="Upload Video",
-                                    interactive= True, visible= False)
-                video2 = gr.Video(source="webcam", label="Record Video", type="filepath",
-                                    interactive= True, visible= False)
-                file_in = gr.Files(label="Upload File or Files", interactive= True, visible= False)
-                
-                submit = gr.Button()
-            
-            with gr.Column():
-                
-                out_txt = gr.Textbox(label="Output",
-                                        visible= True, show_copy_button=True)
-                
-                out_json = gr.JSON(label="JSON Output",
-                                    visible= False, show_copy_button=True)
-                
-                annoation = gr.Textbox(label="Name your speaker's",
-                                    info= "Please provide a list of the speakers arranged \
-                                    in the order in which they appear in the input. Use comma ',' \
-                                    as a seperator. Be aware that the first name is given \
-                                        to SPEAKER_00 the second to SPEAKER_01 and so on.",
-                                    visible= False, interactive= True)
-                
-                annotate = gr.Button(value="Annotate", visible= False, interactive= True)
-            
-        # Define usage of components
-        input.change(fn=select_origin, inputs=[input],
-                        outputs=[audio1, audio2, video1, video2, file_in])
-        
-        task.change(fn=select_task, inputs=[task],
-                    outputs=[num_speakers, translate, language])
-        
-        translate.change(fn= lambda x : gr.update(value = x),
-                            inputs=[translate], outputs=[translate])
-        num_speakers.change(fn= lambda x : gr.update(value = x),
-                            inputs=[num_speakers], outputs=[num_speakers])
-        language.change(fn= lambda x : gr.update(value = x), 
-                        inputs=[language], outputs=[language])
-        
-        submit.click(fn = run_scribe, 
-                        inputs=[task, num_speakers, translate, language, audio1,
-                                audio2, video1, video2, file_in],
-                        outputs=[out_txt, out_json, annoation, annotate])
-        
-        annotate.click(fn = annotate_output, inputs=[annoation, out_json],
-                        outputs=[out_txt, out_json])
-        
-    return demo
-
-    
-if __name__ == "__main__":
-    
-    gradio_Interface().queue().launch()

scraibe/app/header.html

@@ -1,66 +0,0 @@
-<!-- Importing Cormorant Garamond font from Google Fonts -->
-<link href="https://fonts.googleapis.com/css2?family=Cormorant+Garamond:wght@400;700&display=swap" rel="stylesheet">
-
-<style>
-    .header-container {
-        display: flex;
-        align-items: center;
-        justify-content: center;
-        position: relative;
-        padding-top: 30px;
-    }
-    .logo-container {
-        position: absolute;
-        top: 50%;
-        right: 20px;
-        transform: translateY(-50%);
-        width: 300px;
-    }
-    .logo {
-        width: 100%;
-        height: auto;
-    }
-    h1 {
-        font-family: 'Cormorant Garamond', serif;
-        font-size: 50px !important; /* Increased font size */
-        font-weight: bold;
-        color: #50AF31;
-        margin: 0;
-        position: relative;
-        padding: 0.5em 0;
-    }
-    h1::before, h1::after {
-        content: "";
-        position: absolute;
-        height: 2px;
-        width: 80%;
-        background-color: #50AF31;
-        left: 10%;
-    }
-    h1::before {
-        top: 0.5em;
-    }
-    h1::after {
-        bottom: 0.5em;
-    }
-    p, h2 {
-        font-size: 16px;
-        margin: 10px 0;
-        line-height: 1.4;
-    }
-</style>
-
-<div class="header-container">
-    <h1>ScrAIbe</h1>
-    <div class="logo-container">
-        <a href="https://www.kida-bmel.de/"> <!-- Replace with your actual URL -->
-            <img src="file/Logo_KIDA_bmel_green.svg" alt="KIDA Logo" class="logo">
-        </a>
-    </div>
-</div>
-<div style="text-align: center; padding: 20px 10%;">
-    <p>
-        Upload, record, or provide a video with audio for transcription. Our toolkit is designed to transcribe content from multiple languages accurately. The integrated speaker diarisation feature identifies different speakers, ensuring a smooth transcription experience. For optimal results, indicate the number of speakers and the original language of the content.
-    </p>
-    <h2 style="font-weight: bold; color: #50AF31;">What would you like to do next?</h2>
-</div>

scraibe/app/qtfaststart.py

@@ -1,319 +0,0 @@
-"""
-This file contains a modified version of qtfaststart by qtfaststart
-https://github.com/danielgtaylor/qtfaststart/tree/master
-
-All credit goes to the original author.
-Copyright (C) 2008 - 2013 Daniel G. Taylor <dan@programmer-art.org>
-Permission is hereby granted, free of charge, to any person obtaining a copy of this
-software and associated documentation files (the "Software"),
-to deal in the Software without restriction, including without limitation the rights to 
-use, copy, modify, merge, publish, distribute, sublicense, and/or sell copies of the 
-Software, and to permit persons to whom the Software is furnished to do so, 
-subject to the following conditions:
-
-The above copyright notice and this permission notice shall be included in all copies
-or substantial portions of the Software.
-
-THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR IMPLIED,
-INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY, 
-FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT.
-IN NO EVENT SHALL THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM,
-DAMAGES OR OTHER LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE,
-ARISING FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER DEALINGS 
-IN THE SOFTWARE.
-"""
-
-import logging
-import os
-import struct
-import collections
-import io
-
-# define error classes
-class FastStartException(Exception):
-    """
-    Raised when something bad happens during processing.
-    """
-    pass
-
-class FastStartSetupError(FastStartException):
-    """
-    Rasised when asked to process a file that does not need processing
-    """
-    pass
-
-class MalformedFileError(FastStartException):
-    """
-    Raised when the input file is setup in an unexpected way
-    """
-    pass
-
-class UnsupportedFormatError(FastStartException):
-    """
-    Raised when a movie file is recognized as a format not supported.
-    """
-    pass
-
-# define constants
-CHUNK_SIZE = 8192
-
-log = logging.getLogger("qtfaststart")
-
-# Older versions of Python require this to be defined
-if not hasattr(os, 'SEEK_CUR'):
-    os.SEEK_CUR = 1
-
-Atom = collections.namedtuple('Atom', 'name position size')
-
-def read_atom(datastream):
-    """
-        Read an atom and return a tuple of (size, type) where size is the size
-        in bytes (including the 8 bytes already read) and type is a "fourcc"
-        like "ftyp" or "moov".
-    """
-    size, type = struct.unpack(">L4s", datastream.read(8))
-    type = type.decode('ascii')
-    return size, type
-
-
-def _read_atom_ex(datastream):
-    """
-    Read an Atom from datastream
-    """
-    pos = datastream.tell()
-    atom_size, atom_type = read_atom(datastream)
-    if atom_size == 1:
-        atom_size, = struct.unpack(">Q", datastream.read(8))
-    return Atom(atom_type, pos, atom_size)
-
-
-def get_index(datastream):
-    """
-        Return an index of top level atoms, their absolute byte-position in the
-        file and their size in a list:
-
-        index = [
-            ("ftyp", 0, 24),
-            ("moov", 25, 2658),
-            ("free", 2683, 8),
-            ...
-        ]
-
-        The tuple elements will be in the order that they appear in the file.
-    """
-    log.debug("Getting index of top level atoms...")
-
-    index = list(_read_atoms(datastream))
-    _ensure_valid_index(index)
-
-    return index
-
-
-def _read_atoms(datastream):
-    """
-    Read atoms until an error occurs
-    """
-    while datastream:
-        try:
-            atom = _read_atom_ex(datastream)
-            log.debug("%s: %s" % (atom.name, atom.size))
-        except:
-            break
-
-        yield atom
-
-        if atom.size == 0:
-            if atom.name == "mdat":
-                # Some files may end in mdat with no size set, which generally
-                # means to seek to the end of the file. We can just stop indexing
-                # as no more entries will be found!
-                break
-            else:
-                # Weird, but just continue to try to find more atoms
-                continue
-
-        datastream.seek(atom.position + atom.size)
-
-
-def _ensure_valid_index(index):
-    """
-    Ensure the minimum viable atoms are present in the index.
-
-    Raise FastStartException if not.
-    """
-    top_level_atoms = set([item.name for item in index])
-    for key in ["moov", "mdat"]:
-        if key not in top_level_atoms:
-            log.error("%s atom not found, is this a valid MOV/MP4 file?" % key)
-            raise FastStartException()
-
-
-def find_atoms(size, datastream):
-    """
-    Compatibilty interface for _find_atoms_ex
-    """
-    fake_parent = Atom('fake', datastream.tell()-8, size+8)
-    for atom in _find_atoms_ex(fake_parent, datastream):
-        yield atom.name
-
-
-def _find_atoms_ex(parent_atom, datastream):
-    """
-        Yield either "stco" or "co64" Atoms from datastream.
-        datastream will be 8 bytes into the stco or co64 atom when the value
-        is yielded.
-
-        It is assumed that datastream will be at the end of the atom after
-        the value has been yielded and processed.
-
-        parent_atom is the parent atom, a 'moov' or other ancestor of CO
-        atoms in the datastream.
-    """
-    stop = parent_atom.position + parent_atom.size
-
-    while datastream.tell() < stop:
-        try:
-            atom = _read_atom_ex(datastream)
-        except:
-            log.exception("Error reading next atom!")
-            raise FastStartException()
-
-        if atom.name in ["trak", "mdia", "minf", "stbl"]:
-            # Known ancestor atom of stco or co64, search within it!
-            for res in _find_atoms_ex(atom, datastream):
-                yield res
-        elif atom.name in ["stco", "co64"]:
-            yield atom
-        else:
-            # Ignore this atom, seek to the end of it.
-            datastream.seek(atom.position + atom.size)
-
-
-def process(infilename, limit=float('inf')):
-    """
-        Convert a Quicktime/MP4 file for streaming by moving the metadata to
-        the front of the file. This method writes a new file.
-
-        If limit is set to something other than zero it will be used as the
-        number of bytes to write of the atoms following the moov atom. This
-        is very useful to create a small sample of a file with full headers,
-        which can then be used in bug reports and such.
-    """
-    if isinstance(infilename, str):
-        datastream = open(infilename, "rb")
-    elif isinstance(infilename, bytes):
-        datastream = io.BytesIO(infilename)
-    else:
-        raise TypeError("infilename must be a filename, bytes or file-like object")
-    # Get the top level atom index
-    index = get_index(datastream)
-
-    mdat_pos = 999999
-    free_size = 0
-
-    # Make sure moov occurs AFTER mdat, otherwise no need to run!
-    for atom in index:
-        # The atoms are guaranteed to exist from get_index above!
-        if atom.name == "moov":
-            moov_atom = atom
-            moov_pos = atom.position
-        elif atom.name == "mdat":
-            mdat_pos = atom.position
-        elif atom.name == "free" and atom.position < mdat_pos:
-            # This free atom is before the mdat!
-            free_size += atom.size
-            log.info("Removing free atom at %d (%d bytes)" % (atom.position, atom.size))
-        elif atom.name == "\x00\x00\x00\x00" and atom.position < mdat_pos:
-            # This is some strange zero atom with incorrect size
-            free_size += 8
-            log.info("Removing strange zero atom at %s (8 bytes)" % atom.position)
-
-    # Offset to shift positions
-    offset = moov_atom.size - free_size
-
-    if moov_pos < mdat_pos:
-        # moov appears to be in the proper place, don't shift by moov size
-        offset -= moov_atom.size
-        if not free_size:
-            # No free atoms and moov is correct, we are done!
-            log.error("This file appears to already be setup for streaming!")
-            # Stupid hack to retrun the non-processed file:
-            if isinstance(infilename, str):
-                return open(infilename, "rb").read()
-            elif isinstance(infilename, bytes):
-                return io.BytesIO(infilename).read()
-            
-    # Read and fix moov
-    moov = _patch_moov(datastream, moov_atom, offset)
-
-    log.info("Writing output...")
-    outfile = b''
-
-    # Write ftype
-    for atom in index:
-        if atom.name == "ftyp":
-            log.debug("Writing ftyp... (%d bytes)" % atom.size)
-            datastream.seek(atom.position)
-            outfile += datastream.read(atom.size)
-
-    # Write moov
-    _bytes = moov.getvalue()
-    log.debug("Writing moov... (%d bytes)" % len(_bytes))
-    outfile += _bytes
-
-    # Write the rest
-    atoms = [item for item in index if item.name not in ["ftyp", "moov", "free"]]
-    for atom in atoms:
-        log.debug("Writing %s... (%d bytes)" % (atom.name, atom.size))
-        datastream.seek(atom.position)
-
-        # for compatability, allow '0' to mean no limit
-        cur_limit = limit or float('inf')
-        cur_limit = min(cur_limit, atom.size)
-
-        for chunk in get_chunks(datastream, CHUNK_SIZE, cur_limit):
-            outfile += chunk
-
-    return outfile
-
-
-def _patch_moov(datastream, atom, offset):
-    datastream.seek(atom.position)
-    moov = io.BytesIO(datastream.read(atom.size))
-
-    # reload the atom from the fixed stream
-    atom = _read_atom_ex(moov)
-
-    for atom in _find_atoms_ex(atom, moov):
-        # Read either 32-bit or 64-bit offsets
-        ctype, csize = dict(
-            stco=('L', 4),
-            co64=('Q', 8),
-        )[atom.name]
-
-        # Get number of entries
-        version, entry_count = struct.unpack(">2L", moov.read(8))
-
-        log.info("Patching %s with %d entries" % (atom.name, entry_count))
-
-        entries_pos = moov.tell()
-
-        struct_fmt = ">%(entry_count)s%(ctype)s" % vars()
-
-        # Read entries
-        entries = struct.unpack(struct_fmt, moov.read(csize * entry_count))
-
-        # Patch and write entries
-        offset_entries = [entry + offset for entry in entries]
-        moov.seek(entries_pos)
-        moov.write(struct.pack(struct_fmt, *offset_entries))
-    return moov
-
-def get_chunks(stream, chunk_size, limit):
-    remaining = limit
-    while remaining:
-        chunk = stream.read(min(remaining, chunk_size))
-        if not chunk:
-            return
-        remaining -= len(chunk)
-        yield chunk

scraibe/audio.py

@@ -28,6 +28,7 @@ import torch
 SAMPLE_RATE = 16000
 NORMALIZATION_FACTOR = 32768.0
 
+
 class AudioProcessor:
     """
     Audio Processor class that leverages PyTorchaudio to provide functionalities
@@ -40,9 +41,8 @@ class AudioProcessor:
             The sample rate of the audio.
     """
 
-    def __init__(self, waveform: torch.Tensor, sr : int = SAMPLE_RATE,
+    def __init__(self, waveform: torch.Tensor, sr: int = SAMPLE_RATE,
                  *args, **kwargs) -> None:
-        
         """
         Initialize the AudioProcessor object.
 
@@ -57,13 +57,14 @@ class AudioProcessor:
             ValueError: If the provided sample rate is not of type int.
         """
 
-        device = kwargs.get("device", "cuda" if torch.cuda.is_available() else "cpu")
+        device = kwargs.get(
+            "device", "cuda" if torch.cuda.is_available() else "cpu")
 
         self.waveform = waveform.to(device)
         self.sr = sr
 
         if not isinstance(self.sr, int):
-            raise ValueError("Sample rate should be a single value of type int," \
+            raise ValueError("Sample rate should be a single value of type int,"
                              f"not {len(self.sr)} and type {type(self.sr)}")
 
     @classmethod
@@ -78,13 +79,12 @@ class AudioProcessor:
             AudioProcessor: An instance of the AudioProcessor class containing the loaded audio.
         """
 
-        audio, sr = cls.load_audio(file , *args, **kwargs)
+        audio, sr = cls.load_audio(file, *args, **kwargs)
 
         audio = torch.from_numpy(audio)
 
         return cls(audio, sr)
 
-    
     def cut(self, start: float, end: float) -> torch.Tensor:
         """
         Cut a segment from the audio waveform between the specified start and end times.
@@ -140,11 +140,13 @@ class AudioProcessor:
         try:
             out = run(cmd, capture_output=True, check=True).stdout
         except CalledProcessError as e:
-            raise RuntimeError(f"Failed to load audio: {e.stderr.decode()}") from e
+            raise RuntimeError(
+                f"Failed to load audio: {e.stderr.decode()}") from e
 
-        out = np.frombuffer(out, np.int16).flatten().astype(np.float32) / NORMALIZATION_FACTOR
+        out = np.frombuffer(out, np.int16).flatten().astype(
+            np.float32) / NORMALIZATION_FACTOR
 
-        return out , sr
+        return out, sr
 
     def __repr__(self) -> str:
         return f'TorchAudioProcessor(waveform={len(self.waveform)}, sr={int(self.sr)})'

scraibe/autotranscript.py

@@ -38,7 +38,7 @@ from tqdm import trange
 # Application-Specific Imports
 from .audio import AudioProcessor
 from .diarisation import Diariser
-from .transcriber import Transcriber, whisper
+from .transcriber import Transcriber, load_transcriber, whisper
 from .transcript_exporter import Transcript
 
 
@@ -62,26 +62,36 @@ class Scraibe:
         remove_audio_file: Removes the original audio file to avoid disk space issues or ensure data privacy.
         get_audio_file: Gets an audio file as an AudioProcessor object.
     """
+
     def __init__(self,
-                whisper_model: Union[bool, str, whisper] = None,
+                 whisper_model: Union[bool, str, whisper] = None,
-                dia_model : Union[bool, str, DiarisationType] = None,
+                 whisper_type: str = "whisper",
-                **kwargs) -> None:
+                 dia_model: Union[bool, str, DiarisationType] = None,
+                 **kwargs) -> None:
         """Initializes the Scraibe class.
 
         Args:
             whisper_model (Union[bool, str, whisper], optional): 
                                 Path to whisper model or whisper model itself.
+            whisper_type (str):
+                                Type of whisper model to load. "whisper" or "whisperx".
             diarisation_model (Union[bool, str, DiarisationType], optional): 
                                 Path to pyannote diarization model or model itself.
             **kwargs: Additional keyword arguments for whisper
                         and pyannote diarization models.
+                    e.g.:
+
+                    - verbose: If True, the class will print additional information.
+                    - save_kwargs: If True, the keyword arguments will be saved
+                                    for autotranscribe. So you can unload the class and reload it again.
         """
 
-        
         if whisper_model is None:
-            self.transcriber = Transcriber.load_model("medium", **kwargs)    
+            self.transcriber = load_transcriber(
+                "medium", whisper_type, **kwargs)
         elif isinstance(whisper_model, str):
-            self.transcriber = Transcriber.load_model(whisper_model, **kwargs)
+            self.transcriber = load_transcriber(
+                whisper_model, whisper_type, **kwargs)
         else:
             self.transcriber = whisper_model
 
@@ -90,7 +100,7 @@ class Scraibe:
         elif isinstance(dia_model, str):
             self.diariser = Diariser.load_model(dia_model, **kwargs)
         else:
-            self.diariser = dia_model
+            self.diariser: Diariser = dia_model
 
         if kwargs.get("verbose"):
             print("Scraibe initialized all models successfully loaded.")
@@ -98,9 +108,17 @@ class Scraibe:
         else:
             self.verbose = False
 
-    def autotranscribe(self, audio_file : Union[str, torch.Tensor, ndarray],
+        # Save kwargs for autotranscribe if you want to unload the class and load it again.
-                   remove_original : bool = False,
+        if kwargs.get('save_setup'):
-                   **kwargs) -> Transcript:
+            self.params = dict(whisper_model=whisper_model,
+                               dia_model=dia_model,
+                               **kwargs)
+        else:
+            self.params = {}
+
+    def autotranscribe(self, audio_file: Union[str, torch.Tensor, ndarray],
+                       remove_original: bool = False,
+                       **kwargs) -> Transcript:
         """
         Transcribes an audio file using the whisper model and pyannote diarization model.
 
@@ -119,13 +137,13 @@ class Scraibe:
         if kwargs.get("verbose"):
             self.verbose = kwargs.get("verbose")
         # Get audio file as an AudioProcessor object
-        audio_file = self.get_audio_file(audio_file)
+        audio_file: AudioProcessor = self.get_audio_file(audio_file)
 
         # Prepare waveform and sample rate for diarization
         dia_audio = {
-            "waveform" : audio_file.waveform.reshape(1,len(audio_file.waveform)), 
+            "waveform": audio_file.waveform.reshape(1, len(audio_file.waveform)),
             "sample_rate": audio_file.sr
-            }
+        }
 
         if self.verbose:
             print("Starting diarisation.")
@@ -135,23 +153,25 @@ class Scraibe:
         if not diarisation["segments"]:
             print("No segments found. Try to run transcription without diarisation.")
 
-            transcript = self.transcriber.transcribe(audio_file.waveform, **kwargs)
+            transcript = self.transcriber.transcribe(
+                audio_file.waveform, **kwargs)
 
-            final_transcript= {0 : {"speakers" : 'SPEAKER_01',
+            final_transcript = {0: {"speakers": 'SPEAKER_01',
-                                   "segments" : [0, len(audio_file.waveform)],
+                                    "segments": [0, len(audio_file.waveform)],
-                                   "text" : transcript}}
+                                    "text": transcript}}
 
             return Transcript(final_transcript)
 
         if self.verbose:
             print("Diarisation finished. Starting transcription.")
 
-        audio_file.sr = torch.Tensor([audio_file.sr]).to(audio_file.waveform.device)
+        audio_file.sr = torch.Tensor([audio_file.sr]).to(
+            audio_file.waveform.device)
 
         # Transcribe each segment and store the results
         final_transcript = dict()
 
-        for i in trange(len(diarisation["segments"]), desc= "Transcribing", disable = not self.verbose):
+        for i in trange(len(diarisation["segments"]), desc="Transcribing", disable=not self.verbose):
 
             seg = diarisation["segments"][i]
 
@@ -159,9 +179,9 @@ class Scraibe:
 
             transcript = self.transcriber.transcribe(audio, **kwargs)
 
-            final_transcript[i] = {"speakers" : diarisation["speakers"][i],
+            final_transcript[i] = {"speakers": diarisation["speakers"][i],
-                                   "segments" : seg,
+                                   "segments": seg,
-                                   "text" : transcript}
+                                   "text": transcript}
 
         # Remove original file if needed
         if remove_original:
@@ -172,7 +192,7 @@ class Scraibe:
 
         return Transcript(final_transcript)
 
-    def diarization(self, audio_file : Union[str, torch.Tensor, ndarray],
+    def diarization(self, audio_file: Union[str, torch.Tensor, ndarray],
                     **kwargs) -> dict:
         """
         Perform diarization on an audio file using the pyannote diarization model.
@@ -189,13 +209,13 @@ class Scraibe:
         """
 
         # Get audio file as an AudioProcessor object
-        audio_file = self.get_audio_file(audio_file)
+        audio_file: AudioProcessor = self.get_audio_file(audio_file)
 
         # Prepare waveform and sample rate for diarization
         dia_audio = {
-            "waveform" : audio_file.waveform.reshape(1,len(audio_file.waveform)), 
+            "waveform": audio_file.waveform.reshape(1, len(audio_file.waveform)),
             "sample_rate": audio_file.sr
-            }
+        }
 
         print("Starting diarisation.")
 
@@ -203,8 +223,8 @@ class Scraibe:
 
         return diarisation
 
-    def transcribe(self, audio_file : Union[str, torch.Tensor, ndarray],
+    def transcribe(self, audio_file: Union[str, torch.Tensor, ndarray],
-                    **kwargs):
+                   **kwargs):
         """
             Transcribe the provided audio file.
 
@@ -218,12 +238,60 @@ class Scraibe:
                 str:
                     The transcribed text from the audio source.
         """
-        audio_file = self.get_audio_file(audio_file)
+        audio_file: AudioProcessor = self.get_audio_file(audio_file)
 
         return self.transcriber.transcribe(audio_file.waveform, **kwargs)
+
+    def update_transcriber(self, whisper_model: Union[str, whisper], **kwargs) -> None:
+        """
+        Update the transcriber model.
+
+        Args:
+            whisper_model (Union[str, whisper]):
+                The new whisper model to use for transcription.
+            **kwargs:
+                Additional keyword arguments for the transcriber model.
+
+            Returns:
+                None
+        """
+        _old_model = self.transcriber.model_name
+
+        if isinstance(whisper_model, str):
+            self.transcriber = load_transcriber(whisper_model, **kwargs)
+        elif isinstance(whisper_model, Transcriber):
+            self.transcriber = whisper_model
+        else:
+            warn(
+                f"Invalid model type. Please provide a valid model. Fallback to old {_old_model} Model.", RuntimeWarning)
+
+        return None
+
+    def update_diariser(self, dia_model: Union[str, DiarisationType], **kwargs) -> None:
+        """
+        Update the diariser model.
+
+        Args:
+            dia_model (Union[str, DiarisationType]):
+                The new diariser model to use for diarization.
+            **kwargs:
+                Additional keyword arguments for the diariser model.
+
+            Returns:
+                None
+        """
+        if isinstance(dia_model, str):
+            self.diariser = Diariser.load_model(dia_model, **kwargs)
+        elif isinstance(dia_model, Diariser):
+            self.diariser = dia_model
+        else:
+            warn("Invalid model type. Please provide a valid model. Fallback to old Model.", RuntimeWarning)
+
+        return None
+
     @staticmethod
-    def remove_audio_file(audio_file : str,
+    def remove_audio_file(audio_file: str,
-                          shred : bool = False) -> None:
+                          shred: bool = False) -> None:
         """
         Removes the original audio file to avoid disk space issues or ensure data privacy.
 
@@ -248,17 +316,15 @@ class Scraibe:
             for file in gen:
                 print(f'shredding {file} now\n')
 
-                run(cmd , check=True)
+                run(cmd, check=True)
 
         else:
             os.remove(audio_file)
             print(f"Audiofile {audio_file} removed.")
 
-        
-    
     @staticmethod
-    def get_audio_file(audio_file : Union[str, torch.Tensor, ndarray],
+    def get_audio_file(audio_file: Union[str, torch.Tensor, ndarray],
-                        *args, **kwargs) -> AudioProcessor:
+                       *args, **kwargs) -> AudioProcessor:
         """Gets an audio file as TorchAudioProcessor.
 
         Args:
@@ -279,11 +345,12 @@ class Scraibe:
             audio_file = AudioProcessor(audio_file[0], audio_file[1])
         elif isinstance(audio_file, ndarray):
             audio_file = AudioProcessor(torch.Tensor(audio_file[0]),
-                                       audio_file[1])
+                                        audio_file[1])
 
         if not isinstance(audio_file, AudioProcessor):
-            raise ValueError(f'Audiofile must be of type AudioProcessor,' \
+            raise ValueError(f'Audiofile must be of type AudioProcessor,'
                              f'not {type(audio_file)}')
+
         return audio_file
 
     def __repr__(self):

scraibe/cli.py

@@ -5,16 +5,12 @@ The function includes arguments for specifying the audio files, model paths,
 output formats, and other options necessary for transcription.
 """
 import os
-from argparse import ArgumentParser, ArgumentDefaultsHelpFormatter
 import json
-
+from argparse import ArgumentParser, ArgumentDefaultsHelpFormatter
-from .autotranscript import Scraibe
+from whisper.tokenizer import LANGUAGES, TO_LANGUAGE_CODE
-from .app.gradio_app import gradio_Interface
-
-from whisper.tokenizer import LANGUAGES , TO_LANGUAGE_CODE
 from torch.cuda import is_available
 from torch import set_num_threads
-
+from .autotranscript import Scraibe
 
 def cli():
     """
@@ -31,34 +27,28 @@ def cli():
         if string in str2val:
             return str2val[string]
         else:
-            raise ValueError(f"Expected one of {set(str2val.keys())}, got {string}")
+            raise ValueError(
+                f"Expected one of {set(str2val.keys())}, got {string}")
 
-    parser = ArgumentParser(formatter_class = ArgumentDefaultsHelpFormatter)
+    parser = ArgumentParser(formatter_class=ArgumentDefaultsHelpFormatter)
 
-    group = parser.add_mutually_exclusive_group()
+    parser.add_argument("-f", "--audio-files", nargs="+", type=str, default=None,
-    
-    parser.add_argument("-f","--audio-files", nargs="+", type=str, default=None,
                         help="List of audio files to transcribe.")
 
-    group.add_argument('--start-server', action='store_true',
+    parser.add_argument("--whisper-type", type=str, default="whisper",
-                        help='Start the Gradio app.')
+                        choices=["whisper", "whisperx"],
-    
+                        help="Type of Whisper model to use ('whisper' or 'whisperx').")
-    parser.add_argument("--port", type=int, default= None,
-                        help="Port to run the Gradio app on. Defaults to 7860.")
-    
-    parser.add_argument("--server-name", type=str, default= None,
-                        help="Name of the Gradio app. If empty 127.0.0.1 or 0.0.0.0 will be used.")
     
     parser.add_argument("--whisper-model-name", default="medium",
                         help="Name of the Whisper model to use.")
 
-    parser.add_argument("--whisper-model-directory", type=str, default= None,
+    parser.add_argument("--whisper-model-directory", type=str, default=None,
                         help="Path to save Whisper model files; defaults to ./models/whisper.")
 
-    parser.add_argument("--diarization-directory", type=str, default= None,
+    parser.add_argument("--diarization-directory", type=str, default=None,
                         help="Path to the diarization model directory.")
 
-    parser.add_argument("--hf-token", default= None, type=str,
+    parser.add_argument("--hf-token", default=None, type=str,
                         help="HuggingFace token for private model download.")
 
     parser.add_argument("--inference-device",
@@ -66,7 +56,8 @@ def cli():
                         help="Device to use for PyTorch inference.")
 
     parser.add_argument("--num-threads", type=int, default=0,
-                        help="Number of threads used by torch for CPU inference; overrides MKL_NUM_THREADS/OMP_NUM_THREADS.")
+                        help="Number of threads used by torch for CPU inference; '\
+                            'overrides MKL_NUM_THREADS/OMP_NUM_THREADS.")
 
     parser.add_argument("--output-directory", "-o", type=str, default=".",
                         help="Directory to save the transcription outputs.")
@@ -78,14 +69,15 @@ def cli():
     parser.add_argument("--verbose-output", type=str2bool, default=True,
                         help="Enable or disable progress and debug messages.")
 
-    parser.add_argument("--task", type=str, default= 'autotranscribe', # unifinished code
+    parser.add_argument("--task", type=str, default='autotranscribe',
                         choices=["autotranscribe", "diarization",
                                  "autotranscribe+translate", "translate", 'transcribe'],
                         help="Choose to perform transcription, diarization, or translation. \
                         If set to translate, the output will be translated to English.")
 
     parser.add_argument("--language", type=str, default=None,
-                        choices=sorted(LANGUAGES.keys()) + sorted([k.title() for k in TO_LANGUAGE_CODE.keys()]),
+                        choices=sorted(
+                            LANGUAGES.keys()) + sorted([k.title() for k in TO_LANGUAGE_CODE.keys()]),
                         help="Language spoken in the audio. Specify None to perform language detection.")
 
     args = parser.parse_args()
@@ -98,23 +90,22 @@ def cli():
 
     out_format = arg_dict.pop("output_format")
 
-    # seup server arg: 
-    start_server = arg_dict.pop("start_server")
-    
     task = arg_dict.pop("task")
 
     if args.num_threads > 0:
         set_num_threads(arg_dict.pop("num_threads"))
 
-    class_kwargs = {'whisper_model' : arg_dict.pop("whisper_model_name"),
+    class_kwargs = {'whisper_model': arg_dict.pop("whisper_model_name"),
+                    'whisper_type':arg_dict.pop("whisper_type"),
                     'dia_model': arg_dict.pop("diarization_directory"),
-                    'use_auth_token' : arg_dict.pop("hf_token")}
+                    'use_auth_token': arg_dict.pop("hf_token"),
+                    }
 
     if arg_dict["whisper_model_directory"]:
         class_kwargs["download_root"] = arg_dict.pop("whisper_model_directory")
         
-    model = Scraibe(**class_kwargs)
 
+    model = Scraibe(**class_kwargs)
 
     if arg_dict["audio_files"]:
         audio_files = arg_dict.pop("audio_files")
@@ -126,15 +117,17 @@ def cli():
                 else:
                     task = "transcribe"
 
-                out = model.autotranscribe(audio,task = task, language=arg_dict.pop("language"), verbose = arg_dict.pop("verbose_output"))
+                out = model.autotranscribe(audio, task=task, language=arg_dict.pop(
+                    "language"), verbose=arg_dict.pop("verbose_output"))
                 basename = audio.split("/")[-1].split(".")[0]
                 print(f'Saving {basename}.{out_format} to {out_folder}')
-                out.save(os.path.join(out_folder, f"{basename}.{out_format}"))
+                out.save(os.path.join(
+                    out_folder, f"{basename}.{out_format}"))
 
         elif task == "diarization":
             for audio in audio_files:
                 if arg_dict.pop("verbose_output"):
-                    print(f"Verbose not implemented for diarization.")
+                    print("Verbose not implemented for diarization.")
 
                 out = model.diarization(audio)
                 basename = audio.split("/")[-1].split(".")[0]
@@ -143,25 +136,19 @@ def cli():
                 print(f'Saving {basename}.{out_format} to {out_folder}')
 
                 with open(path, "w") as f:
-                    json.dump(json.dumps(out, indent= 1), f)
+                    json.dump(json.dumps(out, indent=1), f)
 
         elif task == "transcribe" or task == "translate":
 
             for audio in audio_files:
 
-                out = model.transcribe(audio, task = task,
+                out = model.transcribe(audio, task=task,
-                                       language= arg_dict.pop("language"),
+                                        language=arg_dict.pop("language"),
-                                       verbose = arg_dict.pop("verbose_output"))
+                                        verbose=arg_dict.pop("verbose_output"))
                 basename = audio.split("/")[-1].split(".")[0]
                 path = os.path.join(out_folder, f"{basename}.{out_format}")
                 with open(path, "w") as f:
                     f.write(out)
 
-    
-    if start_server: # unfinished code
-        
-        gradio_Interface(model).queue().launch(server_port=args.port, server_name=args.server_name)
-        
-
 if __name__ == "__main__":
     cli()

scraibe/diarisation.py

@@ -27,19 +27,26 @@ Usage:
     diarisation_output = model.diarization("path/to/audiofile.wav")
 """
 
+import warnings
 import os
+import yaml
 from pathlib import Path
 from typing import TypeVar, Union
 
 from pyannote.audio import Pipeline
 from pyannote.audio.pipelines.speaker_diarization import SpeakerDiarization
 from torch import Tensor
+from torch import device as torch_device
+from torch.cuda import is_available
+from huggingface_hub import HfApi
+from huggingface_hub.utils import RepositoryNotFoundError
 
 from .misc import PYANNOTE_DEFAULT_PATH, PYANNOTE_DEFAULT_CONFIG
 Annotation = TypeVar('Annotation')
 
 TOKEN_PATH = os.path.join(os.path.dirname(
-            os.path.realpath(__file__)), '.pyannotetoken')
+    os.path.realpath(__file__)), '.pyannotetoken')
+
 
 class Diariser:
     """
@@ -54,7 +61,7 @@ class Diariser:
 
         self.model = model
 
-    def diarization(self, audiofile : Union[str, Tensor, dict] ,
+    def diarization(self, audiofile: Union[str, Tensor, dict],
                     *args, **kwargs) -> Annotation:
         """
         Perform speaker diarization on the provided audio file, 
@@ -74,14 +81,14 @@ class Diariser:
         """
         kwargs = self._get_diarisation_kwargs(**kwargs)
 
-        diarization = self.model(audiofile,*args, **kwargs)
+        diarization = self.model(audiofile, *args, **kwargs)
 
         out = self.format_diarization_output(diarization)
 
         return out
 
     @staticmethod
-    def format_diarization_output(dia : Annotation) -> dict:
+    def format_diarization_output(dia: Annotation) -> dict:
         """
         Formats the raw diarization output into a more usable structure for this project.
 
@@ -93,7 +100,7 @@ class Diariser:
                   as keys and a list of tuples representing segments as values.
         """
 
-        dia_list  = list(dia.itertracks(yield_label=True))
+        dia_list = list(dia.itertracks(yield_label=True))
         diarization_output = {"speakers": [], "segments": []}
 
         normalized_output = []
@@ -120,24 +127,23 @@ class Diariser:
                     index_end_speaker = i - 1
 
                     normalized_output.append([index_start_speaker,
-                                            index_end_speaker,
+                                              index_end_speaker,
-                                            current_speaker])
+                                              current_speaker])
 
                     index_start_speaker = i
                     current_speaker = speaker
 
-            
                 if i == len(dia_list) - 1:
 
                     index_end_speaker = i
 
                     normalized_output.append([index_start_speaker,
-                                            index_end_speaker, 
+                                              index_end_speaker,
-                                            current_speaker])
+                                              current_speaker])
 
         for outp in normalized_output:
-            start =  dia_list[outp[0]][0].start 
+            start = dia_list[outp[0]][0].start
-            end =  dia_list[outp[1]][0].end
+            end = dia_list[outp[1]][0].end
 
             diarization_output["segments"].append([start, end])
             diarization_output["speakers"].append(outp[2])
@@ -160,9 +166,9 @@ class Diariser:
             with open(TOKEN_PATH, 'r', encoding="utf-8") as file:
                 token = file.read()
         else:
-            raise ValueError('No token found.' \
+            raise ValueError('No token found.'
-                'Please create a token at https://huggingface.co/settings/token' \
+                             'Please create a token at https://huggingface.co/settings/token'
-                f'and save it in a file called {TOKEN_PATH}')
+                             f'and save it in a file called {TOKEN_PATH}')
         return token
 
     @staticmethod
@@ -179,50 +185,110 @@ class Diariser:
 
     @classmethod
     def load_model(cls,
-                    model: str = PYANNOTE_DEFAULT_CONFIG, 
+                   model: str = PYANNOTE_DEFAULT_CONFIG,
-                    use_auth_token: str = None,
+                   use_auth_token: str = None,
-                    cache_token: bool = True,
+                   cache_token: bool = False,
-                    cache_dir: Union[Path, str] = PYANNOTE_DEFAULT_PATH,
+                   cache_dir: Union[Path, str] = PYANNOTE_DEFAULT_PATH,
-                    hparams_file: Union[str, Path] = None,
+                   hparams_file: Union[str, Path] = None,
-                    *args, **kwargs
+                   device: str = None,
-                    ) -> Pipeline:
+                   *args, **kwargs
-        
+                   ) -> Pipeline:
         """
         Loads a pretrained model from pyannote.audio, 
-        either from a local cache or online repository.
+        either from a local cache or some online repository.
 
         Args:
             model: Path or identifier for the pyannote model.
-                default: /models/pyannote/speaker_diarization/config.yaml
+                default: '/home/[user]/.cache/torch/models/pyannote/config.yaml'
+                or one of 'jaikinator/scraibe', 'pyannote/speaker-diarization-3.1'
             token: Optional HUGGINGFACE_TOKEN for authenticated access.
             cache_token: Whether to cache the token locally for future use.
             cache_dir: Directory for caching models.
             hparams_file: Path to a YAML file containing hyperparameters.
+            device: Device to load the model on.
             args: Additional arguments only to avoid errors.
             kwargs: Additional keyword arguments only to avoid errors.
 
         Returns:
             Pipeline: A pyannote.audio Pipeline object, encapsulating the loaded model.
         """
+        if isinstance(model, str) and os.path.exists(model):
+            # check if model can be found locally nearby the config file
+            with open(model, 'r') as file:
+                config = yaml.safe_load(file)
 
-        if cache_token and use_auth_token is not None:
+            path_to_model = config['pipeline']['params']['segmentation']
-            cls._save_token(use_auth_token)
 
-        if not os.path.exists(model) and use_auth_token is None:
+            if not os.path.exists(path_to_model):
-            use_auth_token = cls._get_token()
+                warnings.warn(f"Model not found at {path_to_model}. "
-            model = 'pyannote/speaker-diarization'
+                              "Trying to find it nearby the config file.")
-        elif not os.path.exists(model) and use_auth_token is not None:
-            model = 'pyannote/speaker-diarization'
 
-        _model =  Pipeline.from_pretrained(model,
+                pwd = model.split("/")[:-1]
-                                           use_auth_token = use_auth_token,
+                pwd = "/".join(pwd)
-                                           cache_dir = cache_dir,
-                                           hparams_file = hparams_file,)
 
+                path_to_model = os.path.join(pwd, "pytorch_model.bin")
+
+                if not os.path.exists(path_to_model):
+                    warnings.warn(f"Model not found at {path_to_model}. \
+                        'Trying to find it nearby .bin files instead.")
+                    warnings.warn(
+                        'Searching for nearby files in a folder path is '
+                        'deprecated and will be removed in future versions.',
+                        category=DeprecationWarning)
+                    # list elementes with the ending .bin
+                    bin_files = [f for f in os.listdir(
+                        pwd) if f.endswith(".bin")]
+                    if len(bin_files) == 1:
+                        path_to_model = os.path.join(pwd, bin_files[0])
+                    else:
+                        warnings.warn("Found more than one .bin file. "
+                                      "or none. Please specify the path to the model "
+                                      "or setup a huggingface token.")
+                        raise FileNotFoundError
+
+                warnings.warn(
+                    f"Found model at {path_to_model} overwriting config file.")
+
+                config['pipeline']['params']['segmentation'] = path_to_model
+
+                with open(model, 'w') as file:
+                    yaml.dump(config, file)
+        elif isinstance(model, tuple):
+            try:
+                _model = model[0]
+                HfApi().model_info(_model)
+                model = _model
+                use_auth_token = None
+            except RepositoryNotFoundError:
+                print(f'{model[0]} not found on Huggingface, \
+                      trying {model[1]}')
+                _model = model[1]
+                HfApi().model_info(_model)
+                model = _model
+                if cache_token and use_auth_token is not None:
+                    cls._save_token(use_auth_token)
+
+                if use_auth_token is None:
+                    use_auth_token = cls._get_token()
+        else:
+            raise FileNotFoundError(
+                f'No local model or directory found at {model}.')
+
+        _model = Pipeline.from_pretrained(model,
+                                          use_auth_token=use_auth_token,
+                                          cache_dir=cache_dir,
+                                          hparams_file=hparams_file,)
         if _model is None:
-            raise ValueError('Unable to load model either from local cache' \
+            raise ValueError('Unable to load model either from local cache'
-                'or from huggingface.co models. Please check your token' \
+                             'or from huggingface.co models. Please check your token'
-                'or your local model path')
+                             'or your local model path')
+
+        # try to move the model to the device
+        if device is None:
+            device = "cuda" if is_available() else "cpu"
+
+        # torch_device is renamed from torch.device to avoid name conflict
+        _model = _model.to(torch_device(device))
 
         return cls(_model)
 
@@ -239,7 +305,8 @@ class Diariser:
         """
         _possible_kwargs = SpeakerDiarization.apply.__code__.co_varnames
 
-        diarisation_kwargs = {k: v for k, v in kwargs.items() if k in _possible_kwargs}
+        diarisation_kwargs = {k: v for k,
+                              v in kwargs.items() if k in _possible_kwargs}
 
         return diarisation_kwargs
 

scraibe/hallucinations.py

@@ -0,0 +1,95 @@
+# List of known hallucinations - adapted from:
+# https://github.com/openai/whisper/discussions/928
+KNOWN_HALLUCINATIONS = [
+    # en
+    " www.mooji.org"
+    # nl
+    " Ondertitels ingediend door de Amara.org gemeenschap",
+    " Ondertiteld door de Amara.org gemeenschap",
+    " Ondertiteling door de Amara.org gemeenschap"
+    # de
+    " Untertitelung aufgrund der Amara.org-Community"
+    " Untertitelung im Auftrag des ZDF für funk, 2016",
+    " Untertitelung im Auftrag des ZDF f\u00fcr funk, 2016",
+    " Untertitel im Auftrag des ZDF für funk, 2017",
+    " Untertitel im Auftrag des ZDF f\u00fcr funk, 2017",
+    " Untertitel im Auftrag des ZDF für funk, 2018",
+    " Untertitel von Stephanie Geiges",
+    " Untertitel der Amara.org-Community",
+    " Untertitel im Auftrag des ZDF, 2017",
+    " Untertitel im Auftrag des ZDF, 2018",
+    " Untertitel im Auftrag des ZDF, 2019",
+    " Untertitel im Auftrag des ZDF, 2020",
+    " Untertitel im Auftrag des ZDF, 2021",
+    " Untertitelung im Auftrag des ZDF, 2021",
+    " Copyright WDR 2021",
+    " Copyright WDR 2020",
+    " Copyright WDR 2019",
+    " SWR 2021",
+    " SWR 2020",
+    # fr
+    " Sous-titres réalisés para la communauté d'Amara.org",
+    " Sous-titres réalisés par la communauté d'Amara.org",
+    " Sous-titres fait par Sous-titres par Amara.org",
+    " Sous-titres réalisés par les SousTitres d'Amara.org",
+    " Sous-titres par Amara.org",
+    " Sous-titres par la communauté d'Amara.org",
+    " Sous-titres réalisés pour la communauté d'Amara.org",
+    " Sous-titres réalisés par la communauté de l'Amara.org",
+    " Sous-Titres faits par la communauté d'Amara.org",
+    " Sous-titres par l'Amara.org",
+    " Sous-titres fait par la communauté d'Amara.org"
+    " Sous-titrage ST' 501",
+    " Sous-titrage ST'501",
+    " Cliquez-vous sur les sous-titres et abonnez-vous à la chaîne d'Amara.org",
+    " ❤️ par SousTitreur.com",
+    # it
+    " Sottotitoli creati dalla comunità Amara.org",
+    " Sottotitoli di Sottotitoli di Amara.org",
+    " Sottotitoli e revisione al canale di Amara.org",
+    " Sottotitoli e revisione a cura di Amara.org",
+    " Sottotitoli e revisione a cura di QTSS",
+    " Sottotitoli e revisione a cura di QTSS.",
+    " Sottotitoli a cura di QTSS",
+    " Subtítulos realizados por la comunidad de Amara.org",
+    " Subtitulado por la comunidad de Amara.org",
+    " Subtítulos por la comunidad de Amara.org",
+    " Subtítulos creados por la comunidad de Amara.org",
+    " Subtítulos en español de Amara.org",
+    " Subtítulos hechos por la comunidad de Amara.org",
+    " Subtitulos por la comunidad de Amara.org"
+    " Más información www.alimmenta.com",
+    " www.mooji.org",
+    # gl
+    " Subtítulos realizados por la comunidad de Amara.org"
+    # pt
+    " Legendas pela comunidade Amara.org",
+    " Legendas pela comunidade de Amara.org",
+    " Legendas pela comunidade do Amara.org",
+    " Legendas pela comunidade das Amara.org",
+    " Transcrição e Legendas pela comunidade de Amara.org"
+    # la
+    " Sottotitoli creati dalla comunità Amara.org",
+    " Sous-titres réalisés para la communauté d'Amara.org"
+    # ln
+    " Sous-titres réalisés para la communauté d'Amara.org"
+    # pl
+    " Napisy stworzone przez społeczność Amara.org",
+    " Napisy wykonane przez społeczność Amara.org",
+    " Zdjęcia i napisy stworzone przez społeczność Amara.org",
+    " napisy stworzone przez społeczność Amara.org",
+    " Tłumaczenie i napisy stworzone przez społeczność Amara.org",
+    " Napisy stworzone przez społeczności Amara.org",
+    " Tłumaczenie stworzone przez społeczność Amara.org",
+    " Napisy robione przez społeczność Amara.org"
+    " www.multi-moto.eu",
+    # ru
+    " Редактор субтитров А.Синецкая Корректор А.Егорова"
+    # tr
+    " Yorumlarınızıza abone olmayı unutmayın.",
+    # su
+    " Sottotitoli creati dalla comunità Amara.org"
+    # zh
+    "字幕由Amara.org社区提供",
+    "小編字幕由Amara.org社區提供"
+]

scraibe/misc.py

@@ -1,6 +1,8 @@
 import os
 import yaml
 from pyannote.audio.core.model import CACHE_DIR as PYANNOTE_CACHE_DIR
+from argparse import Action
+from ast import literal_eval
 
 CACHE_DIR = os.getenv(
     "AUTOT_CACHE",
@@ -12,7 +14,10 @@ if CACHE_DIR != PYANNOTE_CACHE_DIR:
 
 WHISPER_DEFAULT_PATH = os.path.join(CACHE_DIR, "whisper")
 PYANNOTE_DEFAULT_PATH = os.path.join(CACHE_DIR, "pyannote")
-PYANNOTE_DEFAULT_CONFIG = os.path.join(PYANNOTE_DEFAULT_PATH, "config.yaml")
+PYANNOTE_DEFAULT_CONFIG = os.path.join(PYANNOTE_DEFAULT_PATH, "config.yaml") \
+    if os.path.exists(os.path.join(PYANNOTE_DEFAULT_PATH, "config.yaml")) \
+    else ('jaikinator/scraibe', 'pyannote/speaker-diarization-3.1')
+
 
 def config_diarization_yaml(file_path: str, path_to_segmentation: str = None) -> None:
     """Configure diarization pipeline from a YAML file.
@@ -30,11 +35,29 @@ def config_diarization_yaml(file_path: str, path_to_segmentation: str = None) ->
     with open(file_path, "r") as stream:
         yml = yaml.safe_load(stream)
 
-    segmentation_path = path_to_segmentation or os.path.join(PYANNOTE_DEFAULT_PATH, "pytorch_model.bin")
+    segmentation_path = path_to_segmentation or os.path.join(
+        PYANNOTE_DEFAULT_PATH, "pytorch_model.bin")
     yml["pipeline"]["params"]["segmentation"] = segmentation_path
 
     if not os.path.exists(segmentation_path):
-        raise FileNotFoundError(f"Segmentation model not found at {segmentation_path}")
+        raise FileNotFoundError(
+            f"Segmentation model not found at {segmentation_path}")
 
     with open(file_path, "w") as stream:
         yaml.dump(yml, stream)
+
+
+class ParseKwargs(Action):
+    """
+    Custom argparse action to parse keyword arguments.
+    """
+
+    def __call__(self, parser, namespace, values, option_string=None):
+        setattr(namespace, self.dest, dict())
+        for value in values:
+            key, value = value.split('=')
+            try:
+                value = literal_eval(value)
+            except:
+                pass
+            getattr(namespace, self.dest)[key] = value

scraibe/transcriber.py

@@ -24,18 +24,22 @@ Usage:
     >>> transcriber.save_transcript(transcript, "path/to/save.txt")
 """
 
-from whisper import Whisper, load_model
+from whisper import Whisper
-from typing import TypeVar , Union , Optional
+from whisper import load_model as whisper_load_model
+from whisperx.asr import WhisperModel
+from whisperx import load_model as whisperx_load_model
+from typing import TypeVar, Union, Optional
 from torch import Tensor, device
+from torch.cuda import is_available as cuda_is_available
 from numpy import ndarray
-
+from inspect import signature
+from abc import abstractmethod
+import warnings
 
 from .misc import WHISPER_DEFAULT_PATH
 whisper = TypeVar('whisper')
 
 
-
-
 class Transcriber:
     """
     Transcriber Class
@@ -64,16 +68,117 @@ class Transcriber:
         The class supports various sizes and versions of Whisper models. Please refer to
         the load_model method for available options.
     """
-    def __init__(self, model: whisper ) -> None:
+
+    def __init__(self, model: whisper, model_name: str) -> None:
         """
         Initialize the Transcriber class with a Whisper model.
 
         Args:
             model (whisper): The Whisper model to use for transcription.
+            model_name (str): The name of the model.
         """
+
         self.model = model
 
-    def transcribe(self, audio : Union[str, Tensor, ndarray] ,
+        self.model_name = model_name
+
+    @abstractmethod
+    def transcribe(self, audio: Union[str, Tensor, ndarray],
+                   *args, **kwargs) -> str:
+        """
+        Transcribe an audio file.
+
+        Args:
+            audio (Union[str, Tensor, nparray]): The audio file to transcribe.
+            *args: Additional arguments.
+            **kwargs: Additional keyword arguments, 
+                        such as the language of the audio file.
+
+        Returns:
+            str: The transcript as a string.
+        """
+        pass
+
+    @staticmethod
+    def save_transcript(transcript: str, save_path: str) -> None:
+        """
+        Save a transcript to a file.
+
+        Args:
+            transcript (str): The transcript as a string.
+            save_path (str): The path to save the transcript.
+
+        Returns:
+            None
+        """
+
+        with open(save_path, 'w') as f:
+            f.write(transcript)
+
+        print(f'Transcript saved to {save_path}')
+
+    @classmethod
+    @abstractmethod
+    def load_model(cls,
+                   model: str = "medium",
+                   whisper_type: str = 'whisper',
+                   download_root: str = WHISPER_DEFAULT_PATH,
+                   device: Optional[Union[str, device]] = None,
+                   in_memory: bool = False,
+                   *args, **kwargs
+                   ) -> None:
+        """
+        Load whisper model.
+
+        Args:
+            model (str): Whisper model. Available models include:
+                        - 'tiny.en'
+                        - 'tiny'
+                        - 'base.en'
+                        - 'base'
+                        - 'small.en'
+                        - 'small'
+                        - 'medium.en'
+                        - 'medium'
+                        - 'large-v1'
+                        - 'large-v2'
+                        - 'large-v3'
+                        - 'large'
+            whisper_type (str):
+                                Type of whisper model to load. "whisper" or "whisperx".
+            download_root (str, optional): Path to download the model.
+                                            Defaults to WHISPER_DEFAULT_PATH.
+            device (Optional[Union[str, torch.device]], optional): 
+                                        Device to load model on. Defaults to None.
+            in_memory (bool, optional): Whether to load model in memory. 
+                                        Defaults to False.
+            args: Additional arguments only to avoid errors.
+            kwargs: Additional keyword arguments only to avoid errors.
+
+        Returns:
+            None: abscract method.
+        """
+        pass
+
+    @staticmethod
+    def _get_whisper_kwargs(**kwargs) -> dict:
+        """
+        Get kwargs for whisper model. Ensure that kwargs are valid.
+
+        Returns:
+            dict: Keyword arguments for whisper model.
+        """
+        pass
+
+    def __repr__(self) -> str:
+        return f"Transcriber(model_name={self.model_name}, model={self.model})"
+
+
+class WhisperTranscriber(Transcriber):
+    def __init__(self, model: whisper, model_name: str) -> None:
+        super().__init__(model, model_name)
+
+    def transcribe(self, audio: Union[str, Tensor, ndarray],
                    *args, **kwargs) -> str:
         """
         Transcribe an audio file.
@@ -96,32 +201,14 @@ class Transcriber:
         result = self.model.transcribe(audio, *args, **kwargs)
         return result["text"]
 
-    @staticmethod
-    def save_transcript(transcript : str , save_path : str) -> None:
-        """
-        Save a transcript to a file.
-
-        Args:
-            transcript (str): The transcript as a string.
-            save_path (str): The path to save the transcript.
-
-        Returns:
-            None
-        """
-
-        with open(save_path, 'w') as f:
-            f.write(transcript)
-            
-        print(f'Transcript saved to {save_path}')
-
     @classmethod
     def load_model(cls,
-                    model: str = "medium", 
+                   model: str = "medium",
-                    download_root: str = WHISPER_DEFAULT_PATH,
+                   download_root: str = WHISPER_DEFAULT_PATH,
-                    device: Optional[Union[str, device]] = None,
+                   device: Optional[Union[str, device]] = None,
-                    in_memory: bool = False,
+                   in_memory: bool = False,
-                    *args, **kwargs
+                   *args, **kwargs
-                    ) -> 'Transcriber':
+                   ) -> 'WhisperTranscriber':
         """
         Load whisper model.
 
@@ -137,6 +224,7 @@ class Transcriber:
                         - 'medium'
                         - 'large-v1'
                         - 'large-v2'
+                        - 'large-v3'
                         - 'large'
 
             download_root (str, optional): Path to download the model.
@@ -153,10 +241,10 @@ class Transcriber:
             Transcriber: A Transcriber object initialized with the specified model.
         """
 
-        _model = load_model(model, download_root=download_root,
+        _model = whisper_load_model(model, download_root=download_root,
-                            device=device, in_memory=in_memory)
+                                    device=device, in_memory=in_memory)
 
-        return cls(_model)
+        return cls(_model, model_name=model)
 
     @staticmethod
     def _get_whisper_kwargs(**kwargs) -> dict:
@@ -166,9 +254,11 @@ class Transcriber:
         Returns:
             dict: Keyword arguments for whisper model.
         """
-        _possible_kwargs = Whisper.transcribe.__code__.co_varnames
+        # _possible_kwargs = WhisperModel.transcribe.__code__.co_varnames
+        _possible_kwargs = signature(Whisper.transcribe).parameters.keys()
 
-        whisper_kwargs = {k: v for k, v in kwargs.items() if k in _possible_kwargs}
+        whisper_kwargs = {k: v for k,
+                          v in kwargs.items() if k in _possible_kwargs}
 
         if (task := kwargs.get("task")):
             whisper_kwargs["task"] = task
@@ -179,4 +269,162 @@ class Transcriber:
         return whisper_kwargs
 
     def __repr__(self) -> str:
-        return f"Transcriber(model={self.model})"
+        return f"WhisperTranscriber(model_name={self.model_name}, model={self.model})"
+
+
+class WhisperXTranscriber(Transcriber):
+    def __init__(self, model: whisper, model_name: str) -> None:
+        super().__init__(model, model_name)
+
+    def transcribe(self, audio: Union[str, Tensor, ndarray],
+                   *args, **kwargs) -> str:
+        """
+        Transcribe an audio file.
+
+        Args:
+            audio (Union[str, Tensor, nparray]): The audio file to transcribe.
+            *args: Additional arguments.
+            **kwargs: Additional keyword arguments, 
+                        such as the language of the audio file.
+
+        Returns:
+            str: The transcript as a string.
+        """
+        kwargs = self._get_whisper_kwargs(**kwargs)
+
+        if isinstance(audio, Tensor):
+            audio = audio.cpu().numpy()
+        result = self.model.transcribe(audio, *args, **kwargs)
+        text = ""
+        for seg in result['segments']:
+            text += seg['text']
+        return text
+
+    @classmethod
+    def load_model(cls,
+                   model: str = "medium",
+                   download_root: str = WHISPER_DEFAULT_PATH,
+                   device: Optional[Union[str, device]] = None,
+                   *args, **kwargs
+                   ) -> 'WhisperXTranscriber':
+        """
+        Load whisper model.
+
+        Args:
+            model (str): Whisper model. Available models include:
+                        - 'tiny.en'
+                        - 'tiny'
+                        - 'base.en'
+                        - 'base'
+                        - 'small.en'
+                        - 'small'
+                        - 'medium.en'
+                        - 'medium'
+                        - 'large-v1'
+                        - 'large-v2'
+                        - 'large-v3'
+                        - 'large'
+
+            download_root (str, optional): Path to download the model.
+                                            Defaults to WHISPER_DEFAULT_PATH.
+
+            device (Optional[Union[str, torch.device]], optional): 
+                                        Device to load model on. Defaults to None.
+            in_memory (bool, optional): Whether to load model in memory. 
+                                        Defaults to False.
+            args: Additional arguments only to avoid errors.
+            kwargs: Additional keyword arguments only to avoid errors.
+
+        Returns:
+            Transcriber: A Transcriber object initialized with the specified model.
+        """
+        if device is None:
+            device = "cuda" if cuda_is_available() else "cpu"
+        if not isinstance(device, str):
+            device = str(device)
+        compute_type = kwargs.get('compute_type', 'float16')
+        if device == 'cpu' and compute_type == 'float16':
+            warnings.warn(f'Compute type {compute_type} not compatible with '
+                          f'device {device}! Changing compute type to int8.')
+            compute_type = 'int8'
+        _model = whisperx_load_model(model, download_root=download_root,
+                                     device=device, compute_type=compute_type)
+
+        return cls(_model, model_name=model)
+
+    @staticmethod
+    def _get_whisper_kwargs(**kwargs) -> dict:
+        """
+        Get kwargs for whisper model. Ensure that kwargs are valid.
+
+        Returns:
+            dict: Keyword arguments for whisper model.
+        """
+        # _possible_kwargs = WhisperModel.transcribe.__code__.co_varnames
+        _possible_kwargs = signature(WhisperModel.transcribe).parameters.keys()
+
+        whisper_kwargs = {k: v for k,
+                          v in kwargs.items() if k in _possible_kwargs}
+
+        if (task := kwargs.get("task")):
+            whisper_kwargs["task"] = task
+
+        if (language := kwargs.get("language")):
+            whisper_kwargs["language"] = language
+
+        return whisper_kwargs
+
+    def __repr__(self) -> str:
+        return f"WhisperXTranscriber(model_name={self.model_name}, model={self.model})"
+
+
+def load_transcriber(model: str = "medium",
+                     whisper_type: str = 'whisper',
+                     download_root: str = WHISPER_DEFAULT_PATH,
+                     device: Optional[Union[str, device]] = None,
+                     in_memory: bool = False,
+                     *args, **kwargs
+                     ) -> Union[WhisperTranscriber, WhisperXTranscriber]:
+    """
+    Load whisper model.
+
+    Args:
+        model (str): Whisper model. Available models include:
+                    - 'tiny.en'
+                    - 'tiny'
+                    - 'base.en'
+                    - 'base'
+                    - 'small.en'
+                    - 'small'
+                    - 'medium.en'
+                    - 'medium'
+                    - 'large-v1'
+                    - 'large-v2'
+                    - 'large-v3'
+                    - 'large'
+        whisper_type (str):
+                            Type of whisper model to load. "whisper" or "whisperx".
+        download_root (str, optional): Path to download the model.
+                                        Defaults to WHISPER_DEFAULT_PATH.
+        device (Optional[Union[str, torch.device]], optional): 
+                                    Device to load model on. Defaults to None.
+        in_memory (bool, optional): Whether to load model in memory. 
+                                    Defaults to False.
+        args: Additional arguments only to avoid errors.
+        kwargs: Additional keyword arguments only to avoid errors.
+
+    Returns:
+        Union[WhisperTranscriber, WhisperXTranscriber]: 
+        One of the Whisper variants as Transcrbier object initialized with the specified model.
+    """
+    if whisper_type.lower() == 'whisper':
+        _model = WhisperTranscriber.load_model(
+            model, download_root, device, in_memory, *args, **kwargs)
+        return _model
+    elif whisper_type.lower() == 'whisperx':
+        _model = WhisperXTranscriber.load_model(
+            model, download_root, device, *args, **kwargs)
+        return _model
+    else:
+        raise ValueError(f'Model type not recognized, exptected "whisper" '
+                         f'or "whisperx", got {whisper_type}.')

scraibe/transcript_exporter.py

@@ -1,10 +1,11 @@
 import json
 import time
-from traceback import print_stack
+from json.decoder import JSONDecodeError
-
 
 from typing import Union
 
+from .hallucinations import KNOWN_HALLUCINATIONS
+
 ALPHABET = [*"abcdefghijklmnopqrstuvwxyz"]
 
 
@@ -25,6 +26,7 @@ class Transcript:
         """
 
         self.transcript = transcript
+        self._remove_hallucinations()
         self.speakers = self._extract_speakers()
         self.segments = self._extract_segments()
         self.annotation = {}
@@ -47,7 +49,8 @@ class Transcript:
 
         annotations = {}
         if args and len(args) != len(self.speakers):
-            raise ValueError("Number of speaker names does not match number of speakers")
+            raise ValueError(
+                "Number of speaker names does not match number of speakers")
 
         if args:
             for arg, speaker in zip(args, sorted(self.speakers)):
@@ -56,14 +59,32 @@ class Transcript:
 
         invalid_speakers = set(kwargs.keys()) - set(self.speakers)
         if invalid_speakers:
-            raise ValueError(f"These keys are not speakers: {', '.join(invalid_speakers)}")
+            raise ValueError(
+                f"These keys are not speakers: {', '.join(invalid_speakers)}")
 
-        annotations.update({key: kwargs[key] for key in self.speakers if key in kwargs})
+        annotations.update({key: kwargs[key]
+                           for key in self.speakers if key in kwargs})
 
         self.annotation = annotations
 
         return self
 
+    def _remove_hallucinations(self) -> None:
+        """
+        Removes all occurances of known hallucinations from all segments of the transcript.
+        Segments that are identical to empty strings afterwards are removed from the transcript.
+        """
+        segments_to_drop = []
+        for id in self.transcript:
+            for snippet in KNOWN_HALLUCINATIONS:
+                self.transcript[id]['text'] = self.transcript[id]['text'].replace(
+                    snippet, '')
+            if self.transcript[id]['text'] == '':
+                segments_to_drop.append(id)
+
+        for id in segments_to_drop:
+            del self.transcript[id]
+
     def _extract_speakers(self) -> list:
         """
         Extracts the unique speaker names from the transcript.
@@ -103,8 +124,8 @@ class Transcript:
                 speaker = seq["speakers"]
 
             segm = seq["segments"]
-            sseg = time.strftime("%H:%M:%S",time.gmtime(segm[0]))
+            sseg = time.strftime("%H:%M:%S", time.gmtime(segm[0]))
-            eseg = time.strftime("%H:%M:%S",time.gmtime(segm[1]))
+            eseg = time.strftime("%H:%M:%S", time.gmtime(segm[1]))
 
             fstring += f"{speaker} ({sseg} ; {eseg}):\t{seq['text']}\n"
 
@@ -117,7 +138,7 @@ class Transcript:
             str: A string that provides an informative description of the object.
         """
         return f"Transcript(speakers = {self.speakers},"\
-                f"segments = {self.segments}, annotation = {self.annotation})"
+            f"segments = {self.segments}, annotation = {self.annotation})"
 
     def get_dict(self) -> dict:
         """
@@ -129,7 +150,7 @@ class Transcript:
 
         return self.transcript
 
-    def get_json(self, *args, use_annotation : bool = True, **kwargs) -> str:
+    def get_json(self, *args, use_annotation: bool = True, **kwargs) -> str:
         """
         Get transcript as json string
         :return: transcript as json string
@@ -177,12 +198,12 @@ class Transcript:
 
             self.annotate(*ALPHABET[:len(self.speakers)])
 
-        fstring ="\\begin{drama}"
+        fstring = "\\begin{drama}"
 
         for speaker in self.speakers:
 
-            fstring += "\n\t\\Character{"+ str(self.annotation[speaker]) + "}" \
+            fstring += "\n\t\\Character{" + str(self.annotation[speaker]) + "}" \
-                "{"+ str(self.annotation[speaker]) + "}"
+                "{" + str(self.annotation[speaker]) + "}"
 
         for id in self.transcript:
             seq = self.transcript[id]
@@ -193,8 +214,7 @@ class Transcript:
 
         return fstring
 
-            
+    def to_json(self, path, *args, **kwargs) -> None:
-    def to_json(self,path, *args, **kwargs) -> None:
         """Save transcript as json file
 
         Args:
@@ -280,7 +300,7 @@ class Transcript:
             raise ValueError("Unknown file format")
 
     @classmethod
-    def from_json(cls, json: Union[dict, str]) -> "Transcript":
+    def from_json(cls, _json: Union[dict, str]) -> "Transcript":
         """Load transcript from json file
 
         Args:
@@ -289,15 +309,13 @@ class Transcript:
         Returns:
             Transcript: Transcript object
         """
-        if isinstance(json, dict):
+        if isinstance(_json, dict):
-            return cls(json)
+            return cls(_json)
         else:
             try:
-                transcript = json.loads(json)
+                transcript = json.loads(_json)
-            except:
+            except (TypeError, JSONDecodeError):
-                with open(json, "r") as f:
+                with open(_json, "r") as f:
                     transcript = json.load(f)
 
             return cls(transcript)
-
-    

scraibe/version.py

@@ -1,69 +0,0 @@
-import os
-import subprocess as sp
-
-MAJOR = 0
-MINOR = 1
-MICRO = 0
-MICRO_POST = 0
-ISRELEASED = False
-VERSION = '%d.%d.%d.%d' % (MAJOR, MINOR, MICRO, MICRO_POST)
-
-# Return the git revision as a string
-# taken from numpy/numpy
-def git_version():
-    def _minimal_ext_cmd(cmd):
-        # construct minimal environment
-        env = {}
-        for k in ['SYSTEMROOT', 'PATH', 'HOME']:
-            v = os.environ.get(k)
-            if v is not None:
-                env[k] = v
-
-        # LANGUAGE is used on win32
-        env['LANGUAGE'] = 'C'
-        env['LANG'] = 'C'
-        env['LC_ALL'] = 'C'
-
-        out = sp.Popen(cmd, stdout=sp.PIPE, stderr=sp.PIPE, env=env).communicate()[0]
-        return out
-
-    try:
-        out = _minimal_ext_cmd(['git', 'rev-parse', 'HEAD'])
-        GIT_REVISION = out.strip().decode('ascii')
-    except OSError:
-        GIT_REVISION = "Unknown"
-
-    return GIT_REVISION
-
-def _get_git_version():
-    cwd = os.getcwd()
-
-    # go to the main directory
-    fdir = os.path.dirname(os.path.abspath(__file__))
-    maindir = os.path.abspath(os.path.join(fdir, ".."))
-    # maindir = fdir # os.path.join(fdir, "..")
-    os.chdir(maindir)
-
-    # get git version
-    res = git_version()
-
-    # restore the cwd
-    os.chdir(cwd)
-    return res
-
-def get_version(build_version=False):
-    if ISRELEASED:
-        return VERSION
-
-    # unreleased version
-    GIT_REVISION = _get_git_version()
-
-    if build_version:
-        import datetime as dt
-        date = dt.date.strftime(dt.datetime.now(), "%Y%m%d%H%M%S")
-        return VERSION + ".dev" + date
-    else:
-        return VERSION + ".dev0+" + GIT_REVISION[:7]
-
-
-

setup.cfg

@@ -1,31 +0,0 @@
-[metadata]
-name = scraibe
-version = attr: scraibe.__version__
-author = Jacob Schmieder
-author_email = Jacob.Schmieder@dbfz.de
-description = My package description
-long_description = file: README.md, LICENSE
-platforms = Linux
-keywords = transcription speech recognition whisper pyannote audio speech-to-text speech-to-text transcription speech-to-text recognition voice-to-speech
-license = GPL-3.0
-classifiers =
-    Development Status :: 3 - Alpha
-    Environment :: GPU :: NVIDIA CUDA :: 11.2
-    License :: OSI Approved :: Open Software License 3.0 (OSL-3.0)
-    Topic :: Scientific/Engineering :: Artificial Intelligence
-    Programming Language :: Python :: 3.8
-    Programming Language :: Python :: 3.9
-    Programming Language :: Python :: 3.10
-
-[options]
-zip_safe = False
-include_package_data = True
-packages = find:
-python_requires = >=3.7
-install_requires =
-    requests
-    importlib-metadata; python_version<"3.8"
-
-[options.entry_points]
-console_scripts =
-    executable-name = scraibe.cli:cli

setup.py

@@ -1,60 +0,0 @@
-from calendar import c
-import pkg_resources
-import os
-from setuptools import setup, find_packages
-
-module_name = "scraibe"
-github_url = "https://github.com/JSchmie/ScAIbe"
-
-file_dir = os.path.dirname(os.path.realpath(__file__))
-absdir = lambda p: os.path.join(file_dir, p)
-
-############### versioning ###############
-verfile = os.path.abspath(os.path.join(module_name, "version.py"))
-version = {"__file__": verfile}
-
-with open(verfile, "r") as fp:
-    exec(fp.read(), version)
-
-
-############### setup ###############
-
-build_version = "SCRAIBE_BUILD" in os.environ
-
-if __name__ == "__main__":
-
-    setup(
-        name=module_name,
-        version=version["get_version"](build_version),
-        packages=find_packages(),
-        python_requires=">=3.8",
-        readme="README.md",
-        install_requires = [str(r) for r in pkg_resources.parse_requirements(
-                open(os.path.join(os.path.dirname(__file__), "requirements.txt"))
-            )
-        ],
-        dependency_links=[
-            'https://download.pytorch.org/whl/cu113',
-            ],
-        url= github_url,
-        
-        license='GPL-3',
-        author='Jacob Schmieder',
-        author_email='Jacob.Schmieder@dbfz.de',
-        description='Transcription tool for audio files based on Whisper and Pyannote',
-        classifiers=[
-            'Development Status :: 3 - Alpha',
-            'Environment :: GPU :: NVIDIA CUDA :: 11.2',
-            'License :: OSI Approved :: Open Software License 3.0 (OSL-3.0)',
-            'Topic :: Scientific/Engineering :: Artificial Intelligence',
-            'Programming Language :: Python :: 3.8',
-            'Programming Language :: Python :: 3.9',
-            'Programming Language :: Python :: 3.10'],
-        keywords = ['transcription', 'speech recognition', 'whisper', 'pyannote', 'audio',
-                    'speech-to-text', 'speech-to-text transcription', 'speech-to-text recognition',
-                    'voice-to-speech'],
-        package_data={'scraibe.app' : ["*.html", "*.svg"]},
-        entry_points={'console_scripts':
-            ['scraibe = scraibe.cli:cli']}
-        
-    )

source/conf.py

@@ -0,0 +1,84 @@
+# Configuration file for the Sphinx documentation builder.
+#
+# This file only contains a selection of the most common options. For a full
+# list see the documentation:
+# https://www.sphinx-doc.org/en/master/usage/configuration.html
+
+# -- Path setup --------------------------------------------------------------
+
+# If extensions (or modules to document with autodoc) are in another directory,
+# add these directories to sys.path here. If the directory is relative to the
+# documentation root, use os.path.abspath to make it absolute, like shown here.
+#
+import os
+import sys
+sys.path.insert(0, os.path.abspath('../'))
+
+
+# -- Project information -----------------------------------------------------
+
+project = 'ScrAIbe: Streamlined Conversation Recording with Automated Intelligence Based Environment'
+copyright = '2023, Jacob Schmieder'
+author = 'Jacob Schmieder'
+
+# The full version, including alpha/beta/rc tags
+release = '0.1.1'
+
+
+# -- General configuration ---------------------------------------------------
+
+# Add any Sphinx extension module names here, as strings. They can be
+# extensions coming with Sphinx (named 'sphinx.ext.*') or your custom
+# ones.
+extensions = ['sphinx.ext.autodoc',
+              'sphinx.ext.doctest',
+              'sphinx.ext.intersphinx',
+              'sphinx.ext.todo',
+              'sphinx.ext.coverage',
+              'sphinx.ext.mathjax',
+              'sphinx.ext.ifconfig',
+              'sphinx.ext.viewcode',
+              'sphinx.ext.githubpages',
+              'sphinx.ext.napoleon',
+              'myst_parser']
+
+# Napoleon settings
+napoleon_google_docstring = True
+napoleon_numpy_docstring = True
+napoleon_include_init_with_doc = True
+napoleon_include_private_with_doc = True
+napoleon_include_special_with_doc = True
+napoleon_use_admonition_for_examples = False
+napoleon_use_admonition_for_notes = False
+napoleon_use_admonition_for_references = False
+napoleon_use_ivar = False
+napoleon_use_param = True
+napoleon_use_rtype = True
+
+# Add any paths that contain templates here, relative to this directory.
+templates_path = ['_templates']
+
+# List of patterns, relative to source directory, that match files and
+# directories to ignore when looking for source files.
+# This pattern also affects html_static_path and html_extra_path.
+exclude_patterns = []
+
+# Add source file parsers
+source_suffix = {
+    '.rst': 'restructuredtext',
+    '.txt': 'markdown',
+    '.md': 'markdown',
+}
+
+
+# -- Options for HTML output -------------------------------------------------
+
+# The theme to use for HTML and HTML Help pages.  See the documentation for
+# a list of builtin themes.
+#
+html_theme = 'sphinx_rtd_theme'
+
+# Add any paths that contain custom static files (such as style sheets) here,
+# relative to this directory. They are copied after the builtin static files,
+# so a file named "default.css" will overwrite the builtin "default.css".
+html_static_path = ['_static']

source/index.rst

@@ -0,0 +1,21 @@
+Welcome to ScrAIbe: Streamlined Conversation Recording with Automated Intelligence Based Environment's documentation!
+=====================================================================================================================
+
+.. automodule:: scraibe
+   :members: 
+
+.. toctree::
+   :maxdepth: 2
+   :caption: Contents:
+
+   ../README.md
+
+   modules
+
+
+Indices and tables
+==================
+
+* :ref:`genindex`
+* :ref:`modindex`
+* :ref:`search`

source/modules.rst

@@ -0,0 +1,7 @@
+scraibe
+=======
+
+.. toctree::
+   :maxdepth: 4
+
+   scraibe

test/test_audio.py

@@ -0,0 +1,96 @@
+import pytest
+from scraibe.audio import AudioProcessor
+import torch
+
+
+DEVICE = torch.device("cuda" if torch.cuda.is_available() else "cpu")
+TEST_WAVEFORM = torch.sin(torch.randn(160000)).to(DEVICE)
+TEST_SR = 16000
+SAMPLE_RATE = 16000
+NORMALIZATION_FACTOR = 32768
+
+
+@pytest.fixture
+def probe_audio_processor():
+    """Fixture for creating an instance of the AudioProcessor class with test waveform and sample rate. 
+
+    This fixture is used to create an instance of the AudioProcessor class with a predfined test waveform and sample rate (TEST_SR). It returns the instantiated AudioProcessor , which can bes used as a 
+    dependency in other test functions.
+
+
+    Returns:
+        AudioProcessor (obj): An instance of the AudioProcessor class with the test waveform and sample rate.
+    """
+    return AudioProcessor(TEST_WAVEFORM, TEST_SR)
+
+
+def test_AudioProcessor_init(probe_audio_processor):
+    """
+    Test the initialization of the AudioProcessor class.
+
+    This test verifies that the AUdioProcessor class is correctly initialized with the provided waveform and sample rate. It checks whether the instantiated AhdioProcessor object has the correct attributes
+    and whether the waveform and sample rate match the expected values.
+
+    Args:
+        probe_audio_processor (obj): An instance of the AudioProcessor class to be tested.
+
+
+    Returns:
+           None
+
+
+
+
+    """
+    assert isinstance(probe_audio_processor, AudioProcessor)
+    assert probe_audio_processor.waveform.device == TEST_WAVEFORM.device
+    assert torch.equal(probe_audio_processor.waveform, TEST_WAVEFORM)
+    assert probe_audio_processor.sr == TEST_SR
+
+
+def test_cut(probe_audio_processor):
+    """Test the cut function of the AudioProcessor class.
+
+    This test verifies that the cut function correctly extracts a segment of audio data from
+     the waveform, given start and end indices. It checks whether the size of the extracted segment matches
+     the expected size based on the provided start and end indices and the sample rate.
+
+     Returns:
+            None
+
+
+    """
+
+    start = 4
+    end = 7
+    trimmed_waveform = probe_audio_processor.cut(start, end)
+    expected_size = int((end - start) * TEST_SR)
+    real_size = trimmed_waveform.size(0)
+    assert real_size == expected_size
+    # assert AudioProcessor(TEST_WAVEFORM, TEST_SR).cut(start, end).size() == int((end - start) * TEST_SR)
+
+
+def test_audio_processor_invalid_sr():
+    """Test the behavior of AudioProcessor when an invalid smaple rate is provided.
+
+    This test verifies that the AudioProcessor constructor raises a ValueError when an invalid sample rate is provided. It uses the pytest.raises context manager to check if the ValueError is raised when initializing an 
+    AudioProcessor object with an invalid sample rate.
+
+    Returns:
+           None
+    """
+    with pytest.raises(ValueError):
+        AudioProcessor(TEST_WAVEFORM, [44100, 48000])
+
+
+def test_audio_processor_SAMPLE_RATE():
+    """Test the default sample rate of the AudioProcessor class.
+
+    This test verifies that the default sample rate of the AudioProcessor class matches the expected value defined by the constant SAMPLE_RATE. It instantiates an AudioProcessor object with a test waveform
+    and checks whether the sample rate attribute (sr) of the AudioProcessor object equals the predefined constant SAMPLE_RATE.
+
+    Returns:
+           None
+    """
+    probe_audio_processor = AudioProcessor(TEST_WAVEFORM)
+    assert probe_audio_processor.sr == SAMPLE_RATE

test/test_autotranscript.py

@@ -0,0 +1,52 @@
+import pytest
+from scraibe import Scraibe, Diariser, Transcriber, Transcript
+import os
+
+
+@pytest.fixture
+def create_scraibe_instance():
+    if "HF_TOKEN" in os.environ:
+        return Scraibe(use_auth_token=os.environ["HF_TOKEN"])
+    else:
+        return Scraibe()
+
+
+def test_scraibe_init(create_scraibe_instance):
+    model = create_scraibe_instance
+    assert isinstance(model.transcriber, Transcriber)
+    assert isinstance(model.diariser, Diariser)
+
+
+def test_scraibe_autotranscribe(create_scraibe_instance):
+    model = create_scraibe_instance
+    transcript = model.autotranscribe('test/audio_test_2.mp4')
+    assert isinstance(transcript, Transcript)
+
+
+def test_scraibe_diarization(create_scraibe_instance):
+    model = create_scraibe_instance
+    diarisation_result = model.diarization('test/audio_test_2.mp4')
+    assert isinstance(diarisation_result, dict)
+
+
+def test_scraibe_transcribe(create_scraibe_instance):
+    model = create_scraibe_instance
+    transcription_result = model.transcribe('test/audio_test_2.mp4')
+    assert isinstance(transcription_result, str)
+
+
+""" def test_remove_audio_file(create_scraibe_instance):
+    model = create_scraibe_instance
+    with pytest.raises(ValueError):
+        model.remove_audio_file("non_existing_audio_file")
+
+    model.remove_audio_file("audio_test_2.mp4")
+    assert not os.path.exists("audio_test_2.mp4")   """
+
+
+""" def test_get_audio_file(create_scraibe_instance):
+    model = create_scraibe_instance
+    audio_file = os.path.exist("audio_test_2.mp4")
+    assert isinstance(audio_file, AudioProcessor)
+    assert isinstance(audio_file.waveform, torch.Tensor)
+    assert isinstance(audio_file.sr, torch.Tensor)   """

test/test_diarisation.py

@@ -0,0 +1,32 @@
+import pytest
+from scraibe import Diariser
+
+
+@pytest.fixture
+def diariser_instance():
+    """Fixture for creating an instance of the Diariser class with mocked token.
+
+    This fixture is used to create an instance of the the Diariser class with a mocked token returned by the _get_token method. It patches the _get_token method of the Diariser class
+    using unit.test.mock.patch.object, ensuring that it returns a predetrmined value ('personal Hugging-Face token'). The mocked Diariser object is retunrned and can be used as a dependency in otehr tests.
+
+    Returns:
+        Diariser(Obj): An instance of the Diariser class with a mocked token.
+    """
+    # with mock.patch.object(Diariser, '_get_token', return_value = 'HF_TOKEN' ):
+    return Diariser('pyannote')
+
+
+def test_Diariser_init(diariser_instance):
+    """Test the initialization of the Diariser class.
+
+    This test verifies that the Diariser class is correctly initialized with the specified model.
+    It checks whether the 'model' attribute of the instantiated Diariser object equals 'pyannote'.
+
+
+    Args:
+        diariser_instance (obj): instance of the Diariser class
+
+    Returns:
+           None
+    """
+    assert diariser_instance.model == 'pyannote'

test/test_transcriber.py

@@ -0,0 +1,80 @@
+import pytest
+from scraibe import (Transcriber, WhisperTranscriber,
+                     WhisperXTranscriber, load_transcriber)
+import torch
+
+
+DEVICE = torch.device("cuda" if torch.cuda.is_available() else "cpu")
+TEST_WAVEFORM = "Hello World"
+
+"""
+@pytest.mark.parametrize("audio_file, expected_transcription",[("path_to_test_audiofile", "test_transcription")] )
+@patch("scraibe.Transcriber.load_model")
+
+def test_transcriber(mock_load_model, audio_file, expected_transcription):
+
+
+    Args:
+        mock_load_model (_type_): _description_
+        audio_file (_type_): _description_
+        expected_transcription (_type_): _description_
+
+    mock_model = mock_load_model.return_value
+    mock_model.transcribe.return_value ={"text": expected_transcription}
+
+    transcriber = Transcriber.load_model(model="medium")
+
+    transcription_result = transcriber.transcribe(audio=audio_file)
+
+    assert transcription_result == expected_transcription """
+
+
+@pytest.fixture
+def whisper_instance():
+    return load_transcriber('medium', whisper_type='whisper')
+
+
+@pytest.fixture
+def whisperx_instance():
+    return load_transcriber('medium', whisper_type='whisperx')
+
+
+def test_whisper_base_initialization(whisper_instance):
+    assert isinstance(whisper_instance, Transcriber)
+
+
+def test_whisperx_base_initialization(whisperx_instance):
+    assert isinstance(whisperx_instance, Transcriber)
+
+
+def test_whisper_transcriber_initialization(whisper_instance):
+    assert isinstance(whisper_instance, WhisperTranscriber)
+
+
+def test_whisperx_transcriber_initialization(whisperx_instance):
+    assert isinstance(whisperx_instance, WhisperXTranscriber)
+
+
+def test_wrong_transcriber_initialization():
+    with pytest.raises(ValueError):
+        load_transcriber('medium', whisper_type='wrong_whisper')
+
+
+def test_get_whisper_kwargs():
+    kwargs = {"arg1": 1, "arg3": 3}
+    valid_kwargs = Transcriber._get_whisper_kwargs(**kwargs)
+    assert not valid_kwargs == {"arg1": 1, "arg3": 3}
+
+
+def test_whisper_transcribe(whisper_instance):
+    model = whisper_instance
+    # mocker.patch.object(transcriber_instance.model, 'transcribe', return_value={'Hello, World !'} )
+    transcript = model.transcribe('test/audio_test_2.mp4')
+    assert isinstance(transcript, str)
+
+
+def test_whisperx_transcribe(whisperx_instance):
+    model = whisperx_instance
+    # mocker.patch.object(transcriber_instance.model, 'transcribe', return_value={'Hello, World !'} )
+    transcript = model.transcribe('test/audio_test_2.mp4')
+    assert isinstance(transcript, str)

tests/test_autotranscript.py

@@ -1,120 +0,0 @@
-import pytest
-from scraibe import Transcriber
-from unittest.mock import patch, mock_open
-import os
-
-def test_load_pyannote_model():
-    """
-    Test load_pyannote_test
-    """
-    from pyannote.audio.pipelines.speaker_diarization import SpeakerDiarization
-    from pyannote.audio import Pipeline
-
-    pipeline = Pipeline.from_pretrained("models/pyannote/speaker_diarization/config.yaml")
-    assert isinstance(pipeline, SpeakerDiarization)
-
-# Test Transcribtion class
-
-
-@pytest.fixture
-def transcriber():
-    """
-    Prepare Transcriber for testing
-    Returns: Transcriber Object
-    """
-
-    return Transcriber.load_model("medium", local=True)
-
-
-def test_Transcriber_init(transcriber):
-    """
-    Test Transcriber initialization with a whisper model 
-    """
-    
-    assert isinstance(transcriber, Transcriber)
-
-def test_transcription(transcriber):
-    """
-    Test transcription
-    """
-
-    transcript = transcriber.transcribe("tests/test.wav") 
-    assert isinstance(transcript, str)
-    
-def test_save_transcript_to_file(transcriber):
-    """
-    Test save_transcript_to_file
-    """
-    transcript = transcriber.transcribe("tests/test.wav")
-
-    Transcriber.save_transcript(transcript, "tests/output.txt")
-    
-    assert os.path.exists("tests/output.txt")
-
-    os.remove("tests/output.txt")
-    
-# Test Diaraization class
-
-from scraibe import Diariser
-
-@pytest.fixture
-def diarisation():
-    """
-    Prepare Diarisation for testing
-    Returns: Diarisation Object
-    """
-
-    return Diariser.load_model("models/pyannote/speaker_diarization/config.yaml", local=True)
-
-def test_Diarisation_init(diarisation):
-    """
-    Test Diarisation initialization with a pyannote model 
-    """
-    
-    assert isinstance(diarisation, Diariser)
-
-def test_diarisation(diarisation):
-    """
-    Test diarisation
-    """
-
-    diarisation = diarisation.diarization("tests/test.wav") 
-    assert isinstance(diarisation, dict)
-
-# Test AudioProcessor
-
-from scraibe import AudioProcessor , TorchAudioProcessor
-
-
-def test_AudioProcessor_init():
-    """
-    Test AudioProcessor initialization
-    """
-    audio = AudioProcessor("tests/test.wav")
-    assert isinstance(audio, AudioProcessor)
-
-def test_AudioProcessor_convert():
-    """
-    Test AudioProcessor convert
-    """
-    audio = AudioProcessor("tests/test.wav")
-    audio.convert_audio("tests/test.mp3", format="mp3")
-    assert os.path.exists("tests/test.mp3")
-    
-def test_TorchAudioProcessor_from_file():
-    """
-    Test TorchAudioProcessor initialization
-    """
-    audio = TorchAudioProcessor.from_file("tests/test.wav")
-    
-    assert isinstance(audio, TorchAudioProcessor)
-    
-    os.remove("tests/test.mp3")
-
-
-def test_TorchAudioProcessor_from_ffmpeg():
-    """
-    Test TorchAudioProcessor initialization
-    """
-    audio = TorchAudioProcessor.from_ffmpeg("tests/test.wav")
-    assert isinstance(audio, TorchAudioProcessor)