Fix merge conflicts

.github/workflows/build-push.yml

@@ -1,87 +0,0 @@
-name: Docker Build and push
-
-on:
-  push:
-    tags: [ 'v*.*.*' ]
-    paths-ignore:
-      - '**.md'
-      - 'docs/**'
-  workflow_dispatch:
-    inputs:
-      version:
-        description: 'Version to build and publish (e.g. v0.2.0)'
-        required: true
-        type: string
-
-jobs:
-  prepare-release:
-    runs-on: ubuntu-latest
-    outputs:
-      version: ${{ steps.get-version.outputs.version }}
-      is_prerelease: ${{ steps.check-prerelease.outputs.is_prerelease }}
-    steps:
-      - name: Checkout repository
-        uses: actions/checkout@v4
-
-      - name: Get version
-        id: get-version
-        run: |
-          if [ "${{ github.event_name }}" = "workflow_dispatch" ]; then
-            echo "version=${{ inputs.version }}" >> $GITHUB_OUTPUT
-          else
-            echo "version=$(cat VERSION)" >> $GITHUB_OUTPUT
-          fi
-
-      - name: Check if prerelease
-        id: check-prerelease
-        run: |
-          echo "is_prerelease=${{ contains(steps.get-version.outputs.version, '-pre') }}" >> $GITHUB_OUTPUT
-
-  build-images:
-    needs: prepare-release
-    runs-on: ubuntu-latest
-    permissions:
-      packages: write
-    env:
-      DOCKER_BUILDKIT: 1
-      BUILDKIT_STEP_LOG_MAX_SIZE: 10485760
-      # This environment variable will override the VERSION variable in your HCL file.
-      VERSION: ${{ needs.prepare-release.outputs.version }}
-    steps:
-      - name: Checkout repository
-        uses: actions/checkout@v4
-
-      - name: Free disk space
-        run: |
-          echo "Listing current disk space"
-          df -h
-          echo "Cleaning up disk space..."
-          sudo rm -rf /usr/share/dotnet
-          sudo rm -rf /usr/local/lib/android
-          sudo rm -rf /opt/ghc
-          sudo rm -rf /opt/hostedtoolcache
-          docker system prune -af
-          echo "Disk space after cleanup"
-          df -h
-
-      - name: Set up QEMU
-        uses: docker/setup-qemu-action@v2
-
-      - name: Set up Docker Buildx
-        uses: docker/setup-buildx-action@v2
-        with:
-          driver-opts: |
-            image=moby/buildkit:latest
-            network=host
-
-      - name: Log in to GitHub Container Registry
-        uses: docker/login-action@v2
-        with:
-          registry: ghcr.io
-          username: ${{ github.actor }}
-          password: ${{ secrets.GITHUB_TOKEN }}
-
-      - name: Build and push images
-        run: |
-          # No need to override VERSION via --set; the env var does the job.
-          docker buildx bake --push

.github/workflows/docker-publish.yml

@@ -1,102 +0,0 @@
-name: Docker Build and Publish
-
-on:
-  push:
-    tags: [ 'v*.*.*' ]
-    paths-ignore:
-      - '**.md'
-      - 'docs/**'
-  workflow_dispatch:
-    inputs:
-      version:
-        description: 'Version to release (e.g. v0.2.0)'
-        required: true
-        type: string
-
-jobs:
-  prepare-release:
-    runs-on: ubuntu-latest
-    outputs:
-      version: ${{ steps.get-version.outputs.version }}
-      is_prerelease: ${{ steps.check-prerelease.outputs.is_prerelease }}
-    steps:
-      - name: Checkout repository
-        uses: actions/checkout@v4
-      
-      - name: Get version
-        id: get-version
-        run: |
-          if [ "${{ github.event_name }}" = "workflow_dispatch" ]; then
-            echo "version=${{ inputs.version }}" >> $GITHUB_OUTPUT
-          else
-            echo "version=$(cat VERSION)" >> $GITHUB_OUTPUT
-          fi
-      
-      - name: Check if prerelease
-        id: check-prerelease
-        run: echo "is_prerelease=${{ contains(steps.get-version.outputs.version, '-pre') }}" >> $GITHUB_OUTPUT
-
-  build-images:
-    needs: prepare-release
-    runs-on: ubuntu-latest
-    permissions:
-      packages: write
-    steps:
-      - name: Checkout repository
-        uses: actions/checkout@v4
-
-      - name: Free disk space
-        run: |
-          echo "Listing current disk space"
-          df -h
-          echo "Cleaning up disk space..."
-          sudo rm -rf /usr/share/dotnet
-          sudo rm -rf /usr/local/lib/android
-          sudo rm -rf /opt/ghc
-          sudo rm -rf /opt/hostedtoolcache
-          docker system prune -af
-          echo "Disk space after cleanup"
-          df -h
-
-      - name: Set up QEMU
-        uses: docker/setup-qemu-action@v2
-
-      - name: Set up Docker Buildx
-        uses: docker/setup-buildx-action@v2
-        with:
-          driver-opts: |
-            image=moby/buildkit:latest
-            network=host
-
-      - name: Log in to GitHub Container Registry
-        uses: docker/login-action@v2
-        with:
-          registry: ghcr.io
-          username: ${{ github.actor }}
-          password: ${{ secrets.GITHUB_TOKEN }}
-
-      - name: Build and push images
-        env:
-          DOCKER_BUILDKIT: 1
-          BUILDKIT_STEP_LOG_MAX_SIZE: 10485760
-          VERSION: ${{ needs.prepare-release.outputs.version }}
-        run: docker buildx bake --push
-
-  create-release:
-    needs: [prepare-release, build-images]
-    runs-on: ubuntu-latest
-    permissions:
-      contents: write
-    steps:
-      - name: Checkout repository
-        uses: actions/checkout@v4
-        with:
-          fetch-depth: 0
-
-      - name: Create Release
-        uses: softprops/action-gh-release@v1
-        with:
-          tag_name: ${{ needs.prepare-release.outputs.version }}
-          generate_release_notes: true
-          draft: true
-          prerelease: ${{ needs.prepare-release.outputs.is_prerelease }}

.github/workflows/release.yml

@@ -0,0 +1,110 @@
+name: Create Release and Publish Docker Images
+
+on:
+  push:
+    branches:
+      - release # Trigger when commits are pushed to the release branch (e.g., after merging master)
+    paths-ignore:
+      - '**.md'
+      - 'docs/**'
+
+jobs:
+  prepare-release:
+    runs-on: ubuntu-latest
+    outputs:
+      version: ${{ steps.get-version.outputs.version }}
+      version_tag: ${{ steps.get-version.outputs.version_tag }}
+    steps:
+      - name: Checkout repository
+        uses: actions/checkout@v4
+
+      - name: Get version from VERSION file
+        id: get-version
+        run: |
+          VERSION_PLAIN=$(cat VERSION)
+          echo "version=${VERSION_PLAIN}" >> $GITHUB_OUTPUT
+          echo "version_tag=v${VERSION_PLAIN}" >> $GITHUB_OUTPUT # Add 'v' prefix for tag
+
+  build-images:
+    needs: prepare-release
+    runs-on: ubuntu-latest
+    permissions:
+      packages: write # Needed to push images to GHCR
+    env:
+      DOCKER_BUILDKIT: 1
+      BUILDKIT_STEP_LOG_MAX_SIZE: 10485760
+      # This environment variable will override the VERSION variable in docker-bake.hcl
+      VERSION: ${{ needs.prepare-release.outputs.version_tag }} # Use tag version (vX.Y.Z) for bake
+    steps:
+      - name: Checkout repository
+        uses: actions/checkout@v4
+        with:
+          fetch-depth: 0 # Needed to check for existing tags
+
+      - name: Check if tag already exists
+        run: |
+          TAG_NAME="${{ needs.prepare-release.outputs.version_tag }}"
+          echo "Checking for existing tag: $TAG_NAME"
+          # Fetch tags explicitly just in case checkout didn't get them all
+          git fetch --tags
+          if git rev-parse "$TAG_NAME" >/dev/null 2>&1; then
+            echo "::error::Tag $TAG_NAME already exists. Please increment the version in the VERSION file."
+            exit 1
+          else
+            echo "Tag $TAG_NAME does not exist. Proceeding with release."
+          fi
+
+      - name: Free disk space # Optional: Keep as needed for large builds
+        run: |
+          echo "Listing current disk space"
+          df -h
+          echo "Cleaning up disk space..."
+          sudo rm -rf /usr/share/dotnet /usr/local/lib/android /opt/ghc /opt/hostedtoolcache
+          docker system prune -af
+          echo "Disk space after cleanup"
+          df -h
+
+      - name: Set up QEMU
+        uses: docker/setup-qemu-action@v3 # Use v3
+
+      - name: Set up Docker Buildx
+        uses: docker/setup-buildx-action@v3 # Use v3
+        with:
+          driver-opts: |
+            image=moby/buildkit:latest
+            network=host
+
+      - name: Log in to GitHub Container Registry
+        uses: docker/login-action@v3 # Use v3
+        with:
+          registry: ghcr.io
+          username: ${{ github.actor }}
+          password: ${{ secrets.GITHUB_TOKEN }}
+
+      - name: Build and push images using Docker Bake
+        run: |
+          echo "Building and pushing images for version ${{ needs.prepare-release.outputs.version_tag }}"
+          # The VERSION env var above sets the tag for the bake file targets
+          docker buildx bake --push
+
+  create-release:
+    needs: [prepare-release, build-images]
+    runs-on: ubuntu-latest
+    permissions:
+      contents: write # Needed to create releases
+    steps:
+      - name: Checkout repository
+        uses: actions/checkout@v4
+        with:
+          fetch-depth: 0 # Fetch all history for release notes generation
+
+      - name: Create GitHub Release
+        uses: softprops/action-gh-release@v2 # Use v2
+        with:
+          tag_name: ${{ needs.prepare-release.outputs.version_tag }} # Use vX.Y.Z tag
+          name: Release ${{ needs.prepare-release.outputs.version_tag }}
+          generate_release_notes: true # Auto-generate release notes
+          draft: false # Publish immediately
+          prerelease: false # Mark as a stable release
+        env:
+          GITHUB_TOKEN: ${{ secrets.GITHUB_TOKEN }}

CHANGELOG.md

@@ -2,6 +2,37 @@
 
 Notable changes to this project will be documented in this file.
 
+## [v0.3.0] - 2025-04-04
+### Added
+- Apple Silicon (MPS) acceleration support for macOS users.
+- Voice subtraction capability for creating unique voice effects.
+- Windows PowerShell start scripts (`start-cpu.ps1`, `start-gpu.ps1`).
+- Automatic model downloading integrated into all start scripts.
+- Example Helm chart values for Azure AKS and Nvidia GPU Operator deployments.
+- `CONTRIBUTING.md` guidelines for developers.
+
+### Changed
+- Version bump of underlying Kokoro and Misaki libraries
+- Default API port reverted to 8880.
+- Docker containers now run as a non-root user for enhanced security.
+- Improved text normalization for numbers, currency, and time formats.
+- Updated and improved Helm chart configurations and documentation.
+- Enhanced temporary file management with better error tracking.
+- Web UI dependencies (Siriwave) are now served locally.
+- Standardized environment variable handling across shell/PowerShell scripts.
+
+### Fixed
+- Corrected an issue preventing download links from being returned when `streaming=false`.
+- Resolved errors in Windows PowerShell scripts related to virtual environment activation order.
+- Addressed potential segfaults during inference.
+- Fixed various Helm chart issues related to health checks, ingress, and default values.
+- Corrected audio quality degradation caused by incorrect bitrate settings in some cases.
+- Ensured custom phonemes provided in input text are preserved.
+- Fixed a 'MediaSource' error affecting playback stability in the web player.
+
+### Removed
+- Obsolete GitHub Actions build workflow, build and publish now occurs on merge to `Release` branch
+
 ## [v0.2.0post1] - 2025-02-07
 - Fix: Building Kokoro from source with adjustments, to avoid CUDA lock 
 - Fixed ARM64 compatibility on Spacy dep to avoid emulation slowdown

CONTRIBUTING.md

@@ -0,0 +1,86 @@
+# Contributing to Kokoro-FastAPI
+
+Always appreciate community involvement in making this project better. 
+
+## Development Setup
+
+We use `uv` for managing Python environments and dependencies, and `ruff` for linting and formatting.
+
+1.  **Clone the repository:**
+    ```bash
+    git clone https://github.com/remsky/Kokoro-FastAPI.git
+    cd Kokoro-FastAPI
+    ```
+
+2.  **Install `uv`:**
+    Follow the instructions on the [official `uv` documentation](https://docs.astral.sh/uv/install/).
+
+3.  **Create a virtual environment and install dependencies:**
+    It's recommended to use a virtual environment. `uv` can create one for you. Install the base dependencies along with the `test` and `cpu` extras (needed for running tests locally).
+    ```bash
+    # Create and activate a virtual environment (e.g., named .venv)
+    uv venv
+    source .venv/bin/activate # On Linux/macOS
+    # .venv\Scripts\activate # On Windows
+
+    # Install dependencies including test requirements
+    uv pip install -e ".[test,cpu]"
+    ```
+    *Note: If you have an NVIDIA GPU and want to test GPU-specific features locally, you can install `.[test,gpu]` instead, ensuring you have the correct CUDA toolkit installed.*
+
+    *Note: If running via uv locally, you will have to install espeak and handle any pathing issues that arise. The Docker images handle this automatically*
+
+4.  **Install `ruff` (if not already installed globally):**
+    While `ruff` might be included via dependencies, installing it explicitly ensures you have it available.
+    ```bash
+    uv pip install ruff
+    ```
+
+## Running Tests
+
+Before submitting changes, please ensure all tests pass as this is a automated requirement. The tests are run using `pytest`. 
+```bash
+# Make sure your virtual environment is activated
+uv run pytest
+```
+*Note: The CI workflow runs tests using `uv run pytest api/tests/ --asyncio-mode=auto --cov=api --cov-report=term-missing`. Running `uv run pytest` locally should cover the essential checks.*
+
+## Testing with Docker Compose
+
+In addition to local `pytest` runs, test your changes using Docker Compose to ensure they work correctly within the containerized environment. If you aren't able to test on CUDA hardware, make note so it can be tested by another maintainer
+
+```bash
+
+docker compose -f docker/cpu/docker-compose.yml up --build
++
+docker compose -f docker/gpu/docker-compose.yml up --build
+```
+This command will build the Docker images (if they've changed) and start the services defined in the respective compose file. Verify the application starts correctly and test the relevant functionality.
+
+## Code Formatting and Linting
+
+We use `ruff` to maintain code quality and consistency. Please format and lint your code before committing. 
+
+1.  **Format the code:**
+    ```bash
+    # Make sure your virtual environment is activated
+    ruff format .
+    ```
+
+2.  **Lint the code (and apply automatic fixes):**
+    ```bash
+    # Make sure your virtual environment is activated
+    ruff check . --fix
+    ```
+    Review any changes made by `--fix` and address any remaining linting errors manually.
+
+## Submitting Changes
+
+0.  Clone the repo
+1.  Create a new branch for your feature or bug fix.
+2.  Make your changes, following setup, testing, and formatting guidelines above.
+3.  Please try to keep your changes inline with the current design, and modular. Large-scale changes will take longer to review and integrate, and have less chance of being approved outright.
+4.  Push your branch to your fork.
+5.  Open a Pull Request against the `master` branch of the main repository.
+
+Thank you for contributing!

README.md

@@ -3,12 +3,12 @@
 </p>
 
 # <sub><sub>_`FastKoko`_ </sub></sub>
-[![Tests](https://img.shields.io/badge/tests-69%20passed-darkgreen)]()
-[![Coverage](https://img.shields.io/badge/coverage-52%25-tan)]()
+[![Tests](https://img.shields.io/badge/tests-69-darkgreen)]()
+[![Coverage](https://img.shields.io/badge/coverage-54%25-tan)]()
 [![Try on Spaces](https://img.shields.io/badge/%F0%9F%A4%97%20Try%20on-Spaces-blue)](https://huggingface.co/spaces/Remsky/Kokoro-TTS-Zero)
 
-[![Kokoro](https://img.shields.io/badge/kokoro-v0.7.9::31a2b63-BB5420)](https://github.com/hexgrad/kokoro)
-[![Misaki](https://img.shields.io/badge/misaki-v0.7.9::ebc76c2-B8860B)](https://github.com/hexgrad/misaki)
+[![Kokoro](https://img.shields.io/badge/kokoro-0.9.2-BB5420)](https://github.com/hexgrad/kokoro)
+[![Misaki](https://img.shields.io/badge/misaki-0.9.3-B8860B)](https://github.com/hexgrad/misaki)
 
 [![Tested at Model Commit](https://img.shields.io/badge/last--tested--model--commit-1.0::9901c2b-blue)](https://huggingface.co/hexgrad/Kokoro-82M/commit/9901c2b79161b6e898b7ea857ae5298f47b8b0d6)
 
@@ -24,10 +24,6 @@ Dockerized FastAPI wrapper for [Kokoro-82M](https://huggingface.co/hexgrad/Kokor
 ### Integration Guides
  [![Helm Chart](https://img.shields.io/badge/Helm%20Chart-black?style=flat&logo=helm&logoColor=white)](https://github.com/remsky/Kokoro-FastAPI/wiki/Setup-Kubernetes) [![DigitalOcean](https://img.shields.io/badge/DigitalOcean-black?style=flat&logo=digitalocean&logoColor=white)](https://github.com/remsky/Kokoro-FastAPI/wiki/Integrations-DigitalOcean) [![SillyTavern](https://img.shields.io/badge/SillyTavern-black?style=flat&color=red)](https://github.com/remsky/Kokoro-FastAPI/wiki/Integrations-SillyTavern)
 [![OpenWebUI](https://img.shields.io/badge/OpenWebUI-black?style=flat&color=white)](https://github.com/remsky/Kokoro-FastAPI/wiki/Integrations-OpenWebUi)
-
-
-
-
 ## Get Started
 
 <details>
@@ -38,11 +34,12 @@ Pre built images are available to run, with arm/multi-arch support, and baked in
 Refer to the core/config.py file for a full list of variables which can be managed via the environment
 
 ```bash
-# the `latest` tag can be used, but should not be considered stable as it may include `nightly` branch builds
-# it may have some bonus features however, and feedback/testing is welcome
+# the `latest` tag can be used, though it may have some unexpected bonus features which impact stability.
+ Named versions should be pinned for your regular usage.
+ Feedback/testing is always welcome
 
-docker run -p 8880:8880 ghcr.io/remsky/kokoro-fastapi-cpu:v0.2.2 # CPU, or:
-docker run --gpus all -p 8880:8880 ghcr.io/remsky/kokoro-fastapi-gpu:v0.2.2  #NVIDIA GPU
+docker run -p 8880:8880 ghcr.io/remsky/kokoro-fastapi-cpu:latest # CPU, or:
+docker run --gpus all -p 8880:8880 ghcr.io/remsky/kokoro-fastapi-gpu:latest  #NVIDIA GPU
 ```
 
 
@@ -66,7 +63,7 @@ docker run --gpus all -p 8880:8880 ghcr.io/remsky/kokoro-fastapi-gpu:v0.2.2  #NV
         # *Note for Apple Silicon (M1/M2) users:
         # The current GPU build relies on CUDA, which is not supported on Apple Silicon.  
         # If you are on an M1/M2/M3 Mac, please use the `docker/cpu` setup.  
-        # MPS (Apple’s GPU acceleration) support is planned but not yet available.
+        # MPS (Apple's GPU acceleration) support is planned but not yet available.
 
         # Models will auto-download, but if needed you can manually download:
         python docker/scripts/download_model.py --output api/src/models/v1_0
@@ -139,8 +136,8 @@ with client.audio.speech.with_streaming_response.create(
 
 </details>
 
-
 ## Features 
+
 <details>
 <summary>OpenAI-Compatible Speech Endpoint</summary>
 
@@ -550,13 +547,15 @@ for chunk in response.iter_content(chunk_size=1024):
 <details>
 <summary>Versioning & Development</summary>
 
-I'm doing what I can to keep things stable, but we are on an early and rapid set of build cycles here.
-If you run into trouble, you may have to roll back a version on the release tags if something comes up, or build up from source and/or troubleshoot + submit a PR. Will leave the branch up here for the last known stable points:
+**Branching Strategy:**
+*   **`release` branch:** Contains the latest stable build, recommended for production use. Docker images tagged with specific versions (e.g., `v0.3.0`) are built from this branch.
+*   **`master` branch:** Used for active development. It may contain experimental features, ongoing changes, or fixes not yet in a stable release. Use this branch if you want the absolute latest code, but be aware it might be less stable. The `latest` Docker tag often points to builds from this branch.
 
-`v0.1.4`
-`v0.0.5post1`
+Note: This is a *development* focused project at its core. 
 
-Free and open source is a community effort, and I love working on this project, though there's only really so many hours in a day. If you'd like to support the work, feel free to open a PR, buy me a coffee, or report any bugs/features/etc you find during use.
+If you run into trouble, you may have to roll back a version on the release tags if something comes up, or build up from source and/or troubleshoot + submit a PR.
+
+Free and open source is a community effort, and there's only really so many hours in a day. If you'd like to support the work, feel free to open a PR, buy me a coffee, or report any bugs/features/etc you find during use.
 
   <a href="https://www.buymeacoffee.com/remsky" target="_blank">
     <img 

VERSION

@@ -1 +1 @@
-v0.2.1
+0.3.0

api/src/core/config.py

@@ -1,5 +1,5 @@
-from pydantic_settings import BaseSettings
 import torch
+from pydantic_settings import BaseSettings
 
 
 class Settings(BaseSettings):
@@ -14,9 +14,13 @@ class Settings(BaseSettings):
     output_dir: str = "output"
     output_dir_size_limit_mb: float = 500.0  # Maximum size of output directory in MB
     default_voice: str = "af_heart"
-    default_voice_code: str | None = None  # If set, overrides the first letter of voice name, though api call param still takes precedence
+    default_voice_code: str | None = (
+        None  # If set, overrides the first letter of voice name, though api call param still takes precedence
+    )
     use_gpu: bool = True  # Whether to use GPU acceleration if available
-    device_type: str | None = None  # Will be auto-detected if None, can be "cuda", "mps", or "cpu"
+    device_type: str | None = (
+        None  # Will be auto-detected if None, can be "cuda", "mps", or "cpu"
+    )
     allow_local_voice_saving: bool = (
         False  # Whether to allow saving combined voices locally
     )
@@ -31,12 +35,21 @@ class Settings(BaseSettings):
     target_min_tokens: int = 175  # Target minimum tokens per chunk
     target_max_tokens: int = 250  # Target maximum tokens per chunk
     absolute_max_tokens: int = 450  # Absolute maximum tokens per chunk
-    advanced_text_normalization: bool = True # Preproesses the text before misiki
-    voice_weight_normalization: bool = True # Normalize the voice weights so they add up to 1
+    advanced_text_normalization: bool = True  # Preproesses the text before misiki
+    voice_weight_normalization: bool = (
+        True  # Normalize the voice weights so they add up to 1
+    )
 
-    gap_trim_ms: int = 1  # Base amount to trim from streaming chunk ends in milliseconds
-    dynamic_gap_trim_padding_ms: int = 410 # Padding to add to dynamic gap trim
-    dynamic_gap_trim_padding_char_multiplier: dict[str,float] = {".":1,"!":0.9,"?":1,",":0.8}
+    gap_trim_ms: int = (
+        1  # Base amount to trim from streaming chunk ends in milliseconds
+    )
+    dynamic_gap_trim_padding_ms: int = 410  # Padding to add to dynamic gap trim
+    dynamic_gap_trim_padding_char_multiplier: dict[str, float] = {
+        ".": 1,
+        "!": 0.9,
+        "?": 1,
+        ",": 0.8,
+    }
 
     # Web Player Settings
     enable_web_player: bool = True  # Whether to serve the web player UI
@@ -69,5 +82,4 @@ class Settings(BaseSettings):
         return "cpu"
 
 
-
 settings = Settings()

api/src/inference/base.py

@@ -1,34 +1,41 @@
 """Base interface for Kokoro inference."""
 
 from abc import ABC, abstractmethod
-from typing import AsyncGenerator, Optional, Tuple, Union, List
+from typing import AsyncGenerator, List, Optional, Tuple, Union
 
 import numpy as np
 import torch
 
+
 class AudioChunk:
     """Class for audio chunks returned by model backends"""
 
-    def __init__(self,
-                 audio: np.ndarray,
-                 word_timestamps: Optional[List]=[],
-                 output: Optional[Union[bytes,np.ndarray]]=b""
-                 ):
-        self.audio=audio
-        self.word_timestamps=word_timestamps
-        self.output=output
+    def __init__(
+        self,
+        audio: np.ndarray,
+        word_timestamps: Optional[List] = [],
+        output: Optional[Union[bytes, np.ndarray]] = b"",
+    ):
+        self.audio = audio
+        self.word_timestamps = word_timestamps
+        self.output = output
 
     @staticmethod
     def combine(audio_chunk_list: List):
-        output=AudioChunk(audio_chunk_list[0].audio,audio_chunk_list[0].word_timestamps)
+        output = AudioChunk(
+            audio_chunk_list[0].audio, audio_chunk_list[0].word_timestamps
+        )
 
         for audio_chunk in audio_chunk_list[1:]:
-            output.audio=np.concatenate((output.audio,audio_chunk.audio),dtype=np.int16)
+            output.audio = np.concatenate(
+                (output.audio, audio_chunk.audio), dtype=np.int16
+            )
             if output.word_timestamps is not None:
-                output.word_timestamps+=audio_chunk.word_timestamps
+                output.word_timestamps += audio_chunk.word_timestamps
 
         return output
 
+
 class ModelBackend(ABC):
     """Abstract base class for model inference backend."""
 

api/src/inference/kokoro_v1.py

@@ -11,9 +11,10 @@ from loguru import logger
 from ..core import paths
 from ..core.config import settings
 from ..core.model_config import model_config
-from .base import BaseModelBackend
-from .base import AudioChunk
 from ..structures.schemas import WordTimestamp
+from .base import AudioChunk, BaseModelBackend
+
+
 class KokoroV1(BaseModelBackend):
     """Kokoro backend with controlled resource management."""
 
@@ -50,7 +51,9 @@ class KokoroV1(BaseModelBackend):
             self._model = KModel(config=config_path, model=model_path).eval()
             # For MPS, manually move ISTFT layers to CPU while keeping rest on MPS
             if self._device == "mps":
-                logger.info("Moving model to MPS device with CPU fallback for unsupported operations")
+                logger.info(
+                    "Moving model to MPS device with CPU fallback for unsupported operations"
+                )
                 self._model = self._model.to(torch.device("mps"))
             elif self._device == "cuda":
                 self._model = self._model.cuda()
@@ -145,11 +148,11 @@ class KokoroV1(BaseModelBackend):
             voice_path = temp_path
 
             # Use provided lang_code, settings voice code override, or first letter of voice name
-            if lang_code: # api is given priority
+            if lang_code:  # api is given priority
                 pipeline_lang_code = lang_code
-            elif settings.default_voice_code: # settings is next priority
+            elif settings.default_voice_code:  # settings is next priority
                 pipeline_lang_code = settings.default_voice_code
-            else: # voice name is default/fallback
+            else:  # voice name is default/fallback
                 pipeline_lang_code = voice_name[0].lower()
 
             pipeline = self._get_pipeline(pipeline_lang_code)
@@ -244,7 +247,15 @@ class KokoroV1(BaseModelBackend):
             voice_path = temp_path
 
             # Use provided lang_code, settings voice code override, or first letter of voice name
-            pipeline_lang_code = lang_code if lang_code else (settings.default_voice_code if settings.default_voice_code else voice_name[0].lower())
+            pipeline_lang_code = (
+                lang_code
+                if lang_code
+                else (
+                    settings.default_voice_code
+                    if settings.default_voice_code
+                    else voice_name[0].lower()
+                )
+            )
             pipeline = self._get_pipeline(pipeline_lang_code)
 
             logger.debug(
@@ -255,16 +266,19 @@ class KokoroV1(BaseModelBackend):
             ):
                 if result.audio is not None:
                     logger.debug(f"Got audio chunk with shape: {result.audio.shape}")
-                    word_timestamps=None
-                    if return_timestamps and hasattr(result, "tokens") and result.tokens:
-                        word_timestamps=[]
-                        current_offset=0.0
+                    word_timestamps = None
+                    if (
+                        return_timestamps
+                        and hasattr(result, "tokens")
+                        and result.tokens
+                    ):
+                        word_timestamps = []
+                        current_offset = 0.0
                         logger.debug(
-                                    f"Processing chunk timestamps with {len(result.tokens)} tokens"
-                                )
+                            f"Processing chunk timestamps with {len(result.tokens)} tokens"
+                        )
                         if result.pred_dur is not None:
                             try:
-
                                 # Add timestamps with offset
                                 for token in result.tokens:
                                     if not all(
@@ -285,7 +299,7 @@ class KokoroV1(BaseModelBackend):
                                         WordTimestamp(
                                             word=str(token.text).strip(),
                                             start_time=start_time,
-                                            end_time=end_time
+                                            end_time=end_time,
                                         )
                                     )
                                     logger.debug(
@@ -297,8 +311,9 @@ class KokoroV1(BaseModelBackend):
                                     f"Failed to process timestamps for chunk: {e}"
                                 )
 
-
-                    yield AudioChunk(result.audio.numpy(),word_timestamps=word_timestamps)
+                    yield AudioChunk(
+                        result.audio.numpy(), word_timestamps=word_timestamps
+                    )
                 else:
                     logger.warning("No audio in chunk")
 
@@ -329,7 +344,7 @@ class KokoroV1(BaseModelBackend):
             torch.cuda.synchronize()
         elif self._device == "mps":
             # Empty cache if available (future-proofing)
-            if hasattr(torch.mps, 'empty_cache'):
+            if hasattr(torch.mps, "empty_cache"):
                 torch.mps.empty_cache()
 
     def unload(self) -> None:

api/src/routers/debug.py

@@ -3,8 +3,8 @@ import time
 from datetime import datetime
 
 import psutil
-from fastapi import APIRouter
 import torch
+from fastapi import APIRouter
 
 try:
     import GPUtil
@@ -115,12 +115,12 @@ async def get_system_info():
     # GPU Info if available
     gpu_info = None
     if torch.backends.mps.is_available():
-            gpu_info = {
-                "type": "MPS",
-                "available": True,
-                "device": "Apple Silicon",
-                "backend": "Metal"
-            }
+        gpu_info = {
+            "type": "MPS",
+            "available": True,
+            "device": "Apple Silicon",
+            "backend": "Metal",
+        }
     elif GPU_AVAILABLE:
         try:
             gpus = GPUtil.getGPUs()

api/src/routers/development.py

@@ -156,6 +156,7 @@ async def generate_from_phonemes(
             },
         )
 
+
 @router.post("/dev/captioned_speech")
 async def create_captioned_speech(
     request: CaptionedSpeechRequest,
@@ -184,7 +185,9 @@ async def create_captioned_speech(
         # Check if streaming is requested (default for OpenAI client)
         if request.stream:
             # Create generator but don't start it yet
-            generator = stream_audio_chunks(tts_service, request, client_request, writer)
+            generator = stream_audio_chunks(
+                tts_service, request, client_request, writer
+            )
 
             # If download link requested, wrap generator with temp file writer
             if request.return_download_link:
@@ -211,20 +214,31 @@ async def create_captioned_speech(
                         # Write chunks to temp file and stream
                         async for chunk_data in generator:
                             # The timestamp acumulator is only used when word level time stamps are generated but no audio is returned.
-                            timestamp_acumulator=[]
+                            timestamp_acumulator = []
 
                             if chunk_data.output:  # Skip empty chunks
                                 await temp_writer.write(chunk_data.output)
-                                base64_chunk= base64.b64encode(chunk_data.output).decode("utf-8")
+                                base64_chunk = base64.b64encode(
+                                    chunk_data.output
+                                ).decode("utf-8")
 
                                 # Add any chunks that may be in the acumulator into the return word_timestamps
-                                chunk_data.word_timestamps=timestamp_acumulator + chunk_data.word_timestamps
-                                timestamp_acumulator=[]
+                                chunk_data.word_timestamps = (
+                                    timestamp_acumulator + chunk_data.word_timestamps
+                                )
+                                timestamp_acumulator = []
 
-                                yield CaptionedSpeechResponse(audio=base64_chunk,audio_format=content_type,timestamps=chunk_data.word_timestamps)
+                                yield CaptionedSpeechResponse(
+                                    audio=base64_chunk,
+                                    audio_format=content_type,
+                                    timestamps=chunk_data.word_timestamps,
+                                )
                             else:
-                                if chunk_data.word_timestamps is not None and len(chunk_data.word_timestamps) > 0:
-                                    timestamp_acumulator+=chunk_data.word_timestamps
+                                if (
+                                    chunk_data.word_timestamps is not None
+                                    and len(chunk_data.word_timestamps) > 0
+                                ):
+                                    timestamp_acumulator += chunk_data.word_timestamps
 
                         # Finalize the temp file
                         await temp_writer.finalize()
@@ -246,25 +260,36 @@ async def create_captioned_speech(
             async def single_output():
                 try:
                     # The timestamp acumulator is only used when word level time stamps are generated but no audio is returned.
-                    timestamp_acumulator=[]
+                    timestamp_acumulator = []
 
                     # Stream chunks
                     async for chunk_data in generator:
                         if chunk_data.output:  # Skip empty chunks
                             # Encode the chunk bytes into base 64
-                            base64_chunk= base64.b64encode(chunk_data.output).decode("utf-8")
+                            base64_chunk = base64.b64encode(chunk_data.output).decode(
+                                "utf-8"
+                            )
 
                             # Add any chunks that may be in the acumulator into the return word_timestamps
                             if chunk_data.word_timestamps != None:
-                                chunk_data.word_timestamps = timestamp_acumulator + chunk_data.word_timestamps
+                                chunk_data.word_timestamps = (
+                                    timestamp_acumulator + chunk_data.word_timestamps
+                                )
                             else:
                                 chunk_data.word_timestamps = []
-                            timestamp_acumulator=[]
+                            timestamp_acumulator = []
 
-                            yield CaptionedSpeechResponse(audio=base64_chunk,audio_format=content_type,timestamps=chunk_data.word_timestamps)
+                            yield CaptionedSpeechResponse(
+                                audio=base64_chunk,
+                                audio_format=content_type,
+                                timestamps=chunk_data.word_timestamps,
+                            )
                         else:
-                            if chunk_data.word_timestamps is not None and len(chunk_data.word_timestamps) > 0:
-                                timestamp_acumulator+=chunk_data.word_timestamps
+                            if (
+                                chunk_data.word_timestamps is not None
+                                and len(chunk_data.word_timestamps) > 0
+                            ):
+                                timestamp_acumulator += chunk_data.word_timestamps
 
                 except Exception as e:
                     logger.error(f"Error in single output streaming: {e}")
@@ -309,11 +334,15 @@ async def create_captioned_speech(
                 writer,
                 is_last_chunk=True,
             )
-            output=audio_data.output + final.output
+            output = audio_data.output + final.output
 
-            base64_output= base64.b64encode(output).decode("utf-8")
+            base64_output = base64.b64encode(output).decode("utf-8")
 
-            content=CaptionedSpeechResponse(audio=base64_output,audio_format=content_type,timestamps=audio_data.word_timestamps).model_dump()
+            content = CaptionedSpeechResponse(
+                audio=base64_output,
+                audio_format=content_type,
+                timestamps=audio_data.word_timestamps,
+            ).model_dump()
 
             writer.close()
 

api/src/routers/openai_compatible.py

@@ -10,18 +10,18 @@ from urllib import response
 
 import aiofiles
 import numpy as np
-from ..services.streaming_audio_writer import StreamingAudioWriter
 import torch
 from fastapi import APIRouter, Depends, Header, HTTPException, Request, Response
 from fastapi.responses import FileResponse, StreamingResponse
 from loguru import logger
-from ..structures.schemas import CaptionedSpeechRequest
 
 from ..core.config import settings
 from ..inference.base import AudioChunk
 from ..services.audio import AudioService
+from ..services.streaming_audio_writer import StreamingAudioWriter
 from ..services.tts_service import TTSService
 from ..structures import OpenAISpeechRequest
+from ..structures.schemas import CaptionedSpeechRequest
 
 
 # Load OpenAI mappings
@@ -80,7 +80,9 @@ def get_model_name(model: str) -> str:
     return base_name + ".pth"
 
 
-async def process_and_validate_voices(voice_input: Union[str, List[str]], tts_service: TTSService) -> str:
+async def process_and_validate_voices(
+    voice_input: Union[str, List[str]], tts_service: TTSService
+) -> str:
     """Process voice input, handling both string and list formats
 
     Returns:
@@ -107,22 +109,35 @@ async def process_and_validate_voices(voice_input: Union[str, List[str]], tts_se
         mapped_voice = list(map(str.strip, mapped_voice))
 
         if len(mapped_voice) > 2:
-            raise ValueError(f"Voice '{voices[voice_index]}' contains too many weight items")
+            raise ValueError(
+                f"Voice '{voices[voice_index]}' contains too many weight items"
+            )
 
         if mapped_voice.count(")") > 1:
-            raise ValueError(f"Voice '{voices[voice_index]}' contains too many weight items")
+            raise ValueError(
+                f"Voice '{voices[voice_index]}' contains too many weight items"
+            )
 
-        mapped_voice[0] = _openai_mappings["voices"].get(mapped_voice[0], mapped_voice[0])
+        mapped_voice[0] = _openai_mappings["voices"].get(
+            mapped_voice[0], mapped_voice[0]
+        )
 
         if mapped_voice[0] not in available_voices:
-            raise ValueError(f"Voice '{mapped_voice[0]}' not found. Available voices: {', '.join(sorted(available_voices))}")
+            raise ValueError(
+                f"Voice '{mapped_voice[0]}' not found. Available voices: {', '.join(sorted(available_voices))}"
+            )
 
         voices[voice_index] = "(".join(mapped_voice)
 
     return "".join(voices)
 
 
-async def stream_audio_chunks(tts_service: TTSService, request: Union[OpenAISpeechRequest, CaptionedSpeechRequest], client_request: Request, writer: StreamingAudioWriter) -> AsyncGenerator[AudioChunk, None]:
+async def stream_audio_chunks(
+    tts_service: TTSService,
+    request: Union[OpenAISpeechRequest, CaptionedSpeechRequest],
+    client_request: Request,
+    writer: StreamingAudioWriter,
+) -> AsyncGenerator[AudioChunk, None]:
     """Stream audio chunks as they're generated with client disconnect handling"""
     voice_name = await process_and_validate_voices(request.voice, tts_service)
     unique_properties = {"return_timestamps": False}
@@ -193,7 +208,9 @@ async def create_speech(
         # Check if streaming is requested (default for OpenAI client)
         if request.stream:
             # Create generator but don't start it yet
-            generator = stream_audio_chunks(tts_service, request, client_request, writer)
+            generator = stream_audio_chunks(
+                tts_service, request, client_request, writer
+            )
 
             # If download link requested, wrap generator with temp file writer
             if request.return_download_link:
@@ -245,7 +262,9 @@ async def create_speech(
                         writer.close()
 
                 # Stream with temp file writing
-                return StreamingResponse(dual_output(), media_type=content_type, headers=headers)
+                return StreamingResponse(
+                    dual_output(), media_type=content_type, headers=headers
+                )
 
             async def single_output():
                 try:
@@ -285,7 +304,13 @@ async def create_speech(
                 lang_code=request.lang_code,
             )
 
-            audio_data = await AudioService.convert_audio(audio_data, request.response_format, writer, is_last_chunk=False, trim_audio=False)
+            audio_data = await AudioService.convert_audio(
+                audio_data,
+                request.response_format,
+                writer,
+                is_last_chunk=False,
+                trim_audio=False,
+            )
 
             # Convert to requested format with proper finalization
             final = await AudioService.convert_audio(
@@ -384,7 +409,6 @@ async def create_speech(
         )
 
 
-
 @router.get("/download/{filename}")
 async def download_audio_file(filename: str):
     """Download a generated audio file from temp storage"""
@@ -392,7 +416,9 @@ async def download_audio_file(filename: str):
         from ..core.paths import _find_file, get_content_type
 
         # Search for file in temp directory
-        file_path = await _find_file(filename=filename, search_paths=[settings.temp_file_dir])
+        file_path = await _find_file(
+            filename=filename, search_paths=[settings.temp_file_dir]
+        )
 
         # Get content type from path helper
         content_type = await get_content_type(file_path)
@@ -425,9 +451,24 @@ async def list_models():
     try:
         # Create standard model list
         models = [
-            {"id": "tts-1", "object": "model", "created": 1686935002, "owned_by": "kokoro"},
-            {"id": "tts-1-hd", "object": "model", "created": 1686935002, "owned_by": "kokoro"},
-            {"id": "kokoro", "object": "model", "created": 1686935002, "owned_by": "kokoro"},
+            {
+                "id": "tts-1",
+                "object": "model",
+                "created": 1686935002,
+                "owned_by": "kokoro",
+            },
+            {
+                "id": "tts-1-hd",
+                "object": "model",
+                "created": 1686935002,
+                "owned_by": "kokoro",
+            },
+            {
+                "id": "kokoro",
+                "object": "model",
+                "created": 1686935002,
+                "owned_by": "kokoro",
+            },
         ]
 
         return {"object": "list", "data": models}
@@ -449,14 +490,36 @@ async def retrieve_model(model: str):
     try:
         # Define available models
         models = {
-            "tts-1": {"id": "tts-1", "object": "model", "created": 1686935002, "owned_by": "kokoro"},
-            "tts-1-hd": {"id": "tts-1-hd", "object": "model", "created": 1686935002, "owned_by": "kokoro"},
-            "kokoro": {"id": "kokoro", "object": "model", "created": 1686935002, "owned_by": "kokoro"},
+            "tts-1": {
+                "id": "tts-1",
+                "object": "model",
+                "created": 1686935002,
+                "owned_by": "kokoro",
+            },
+            "tts-1-hd": {
+                "id": "tts-1-hd",
+                "object": "model",
+                "created": 1686935002,
+                "owned_by": "kokoro",
+            },
+            "kokoro": {
+                "id": "kokoro",
+                "object": "model",
+                "created": 1686935002,
+                "owned_by": "kokoro",
+            },
         }
 
         # Check if requested model exists
         if model not in models:
-            raise HTTPException(status_code=404, detail={"error": "model_not_found", "message": f"Model '{model}' not found", "type": "invalid_request_error"})
+            raise HTTPException(
+                status_code=404,
+                detail={
+                    "error": "model_not_found",
+                    "message": f"Model '{model}' not found",
+                    "type": "invalid_request_error",
+                },
+            )
 
         # Return the specific model
         return models[model]
@@ -541,7 +604,9 @@ async def combine_voices(request: Union[str, List[str]]):
         available_voices = await tts_service.list_voices()
         for voice in voices:
             if voice not in available_voices:
-                raise ValueError(f"Base voice '{voice}' not found. Available voices: {', '.join(sorted(available_voices))}")
+                raise ValueError(
+                    f"Base voice '{voice}' not found. Available voices: {', '.join(sorted(available_voices))}"
+                )
 
         # Combine voices
         combined_tensor = await tts_service.combine_voices(voices=voices)

api/src/services/audio.py

@@ -1,12 +1,12 @@
 """Audio conversion service"""
 
+import math
 import struct
 import time
-from typing import Tuple
 from io import BytesIO
+from typing import Tuple
 
 import numpy as np
-import math
 import scipy.io.wavfile as wavfile
 import soundfile as sf
 from loguru import logger
@@ -14,8 +14,9 @@ from pydub import AudioSegment
 from torch import norm
 
 from ..core.config import settings
-from .streaming_audio_writer import StreamingAudioWriter
 from ..inference.base import AudioChunk
+from .streaming_audio_writer import StreamingAudioWriter
+
 
 class AudioNormalizer:
     """Handles audio normalization state for a single stream"""
@@ -24,9 +25,16 @@ class AudioNormalizer:
         self.chunk_trim_ms = settings.gap_trim_ms
         self.sample_rate = 24000  # Sample rate of the audio
         self.samples_to_trim = int(self.chunk_trim_ms * self.sample_rate / 1000)
-        self.samples_to_pad_start= int(50 * self.sample_rate / 1000)
+        self.samples_to_pad_start = int(50 * self.sample_rate / 1000)
 
-    def find_first_last_non_silent(self,audio_data: np.ndarray, chunk_text: str, speed: float, silence_threshold_db: int = -45, is_last_chunk: bool = False) -> tuple[int, int]:
+    def find_first_last_non_silent(
+        self,
+        audio_data: np.ndarray,
+        chunk_text: str,
+        speed: float,
+        silence_threshold_db: int = -45,
+        is_last_chunk: bool = False,
+    ) -> tuple[int, int]:
         """Finds the indices of the first and last non-silent samples in audio data.
 
         Args:
@@ -40,37 +48,55 @@ class AudioNormalizer:
             A tuple with the start of the non silent portion and with the end of the non silent portion
         """
 
-        pad_multiplier=1
-        split_character=chunk_text.strip()
+        pad_multiplier = 1
+        split_character = chunk_text.strip()
         if len(split_character) > 0:
-            split_character=split_character[-1]
+            split_character = split_character[-1]
             if split_character in settings.dynamic_gap_trim_padding_char_multiplier:
-                pad_multiplier=settings.dynamic_gap_trim_padding_char_multiplier[split_character]
+                pad_multiplier = settings.dynamic_gap_trim_padding_char_multiplier[
+                    split_character
+                ]
 
         if not is_last_chunk:
-            samples_to_pad_end= max(int((settings.dynamic_gap_trim_padding_ms * self.sample_rate * pad_multiplier) / 1000) - self.samples_to_pad_start, 0)
+            samples_to_pad_end = max(
+                int(
+                    (
+                        settings.dynamic_gap_trim_padding_ms
+                        * self.sample_rate
+                        * pad_multiplier
+                    )
+                    / 1000
+                )
+                - self.samples_to_pad_start,
+                0,
+            )
         else:
-            samples_to_pad_end=self.samples_to_pad_start
+            samples_to_pad_end = self.samples_to_pad_start
         # Convert dBFS threshold to amplitude
-        amplitude_threshold = np.iinfo(audio_data.dtype).max * (10 ** (silence_threshold_db / 20))
+        amplitude_threshold = np.iinfo(audio_data.dtype).max * (
+            10 ** (silence_threshold_db / 20)
+        )
         # Find the first samples above the silence threshold at the start and end of the audio
-        non_silent_index_start, non_silent_index_end = None,None 
+        non_silent_index_start, non_silent_index_end = None, None
 
-        for X in range(0,len(audio_data)):
+        for X in range(0, len(audio_data)):
             if audio_data[X] > amplitude_threshold:
-                non_silent_index_start=X
+                non_silent_index_start = X
                 break
 
         for X in range(len(audio_data) - 1, -1, -1):
             if audio_data[X] > amplitude_threshold:
-                non_silent_index_end=X
+                non_silent_index_end = X
                 break
 
         # Handle the case where the entire audio is silent
         if non_silent_index_start == None or non_silent_index_end == None:
             return 0, len(audio_data)
 
-        return max(non_silent_index_start - self.samples_to_pad_start,0), min(non_silent_index_end + math.ceil(samples_to_pad_end / speed),len(audio_data))
+        return max(non_silent_index_start - self.samples_to_pad_start, 0), min(
+            non_silent_index_end + math.ceil(samples_to_pad_end / speed),
+            len(audio_data),
+        )
 
     def normalize(self, audio_data: np.ndarray) -> np.ndarray:
         """Convert audio data to int16 range
@@ -85,6 +111,7 @@ class AudioNormalizer:
             return np.clip(audio_data * 32767, -32768, 32767).astype(np.int16)
         return audio_data
 
+
 class AudioService:
     """Service for audio format conversions with streaming support"""
 
@@ -147,24 +174,24 @@ class AudioService:
             audio_chunk.audio = normalizer.normalize(audio_chunk.audio)
 
             if trim_audio == True:
-                audio_chunk = AudioService.trim_audio(audio_chunk,chunk_text,speed,is_last_chunk,normalizer)
+                audio_chunk = AudioService.trim_audio(
+                    audio_chunk, chunk_text, speed, is_last_chunk, normalizer
+                )
 
             # Write audio data first
             if len(audio_chunk.audio) > 0:
                 chunk_data = writer.write_chunk(audio_chunk.audio)
 
-            
-
             # Then finalize if this is the last chunk
             if is_last_chunk:
                 final_data = writer.write_chunk(finalize=True)
 
                 if final_data:
-                    audio_chunk.output=final_data
+                    audio_chunk.output = final_data
                 return audio_chunk
 
             if chunk_data:
-                audio_chunk.output=chunk_data
+                audio_chunk.output = chunk_data
             return audio_chunk
 
         except Exception as e:
@@ -172,8 +199,15 @@ class AudioService:
             raise ValueError(
                 f"Failed to convert audio stream to {output_format}: {str(e)}"
             )
+
     @staticmethod
-    def trim_audio(audio_chunk: AudioChunk, chunk_text: str = "", speed: float = 1, is_last_chunk: bool = False, normalizer: AudioNormalizer = None) -> AudioChunk:
+    def trim_audio(
+        audio_chunk: AudioChunk,
+        chunk_text: str = "",
+        speed: float = 1,
+        is_last_chunk: bool = False,
+        normalizer: AudioNormalizer = None,
+    ) -> AudioChunk:
         """Trim silence from start and end
 
         Args:
@@ -189,23 +223,26 @@ class AudioService:
         if normalizer is None:
             normalizer = AudioNormalizer()
 
-        audio_chunk.audio=normalizer.normalize(audio_chunk.audio)
+        audio_chunk.audio = normalizer.normalize(audio_chunk.audio)
 
-        trimed_samples=0
+        trimed_samples = 0
         # Trim start and end if enough samples
         if len(audio_chunk.audio) > (2 * normalizer.samples_to_trim):
-            audio_chunk.audio = audio_chunk.audio[normalizer.samples_to_trim : -normalizer.samples_to_trim]
-            trimed_samples+=normalizer.samples_to_trim
+            audio_chunk.audio = audio_chunk.audio[
+                normalizer.samples_to_trim : -normalizer.samples_to_trim
+            ]
+            trimed_samples += normalizer.samples_to_trim
 
         # Find non silent portion and trim
-        start_index,end_index=normalizer.find_first_last_non_silent(audio_chunk.audio,chunk_text,speed,is_last_chunk=is_last_chunk)
+        start_index, end_index = normalizer.find_first_last_non_silent(
+            audio_chunk.audio, chunk_text, speed, is_last_chunk=is_last_chunk
+        )
 
-        audio_chunk.audio=audio_chunk.audio[start_index:end_index]
-        trimed_samples+=start_index
+        audio_chunk.audio = audio_chunk.audio[start_index:end_index]
+        trimed_samples += start_index
 
         if audio_chunk.word_timestamps is not None:
             for timestamp in audio_chunk.word_timestamps:
-                timestamp.start_time-=trimed_samples / 24000
-                timestamp.end_time-=trimed_samples / 24000
+                timestamp.start_time -= trimed_samples / 24000
+                timestamp.end_time -= trimed_samples / 24000
         return audio_chunk
-    

api/src/services/streaming_audio_writer.py

@@ -4,11 +4,12 @@ import struct
 from io import BytesIO
 from typing import Optional
 
+import av
 import numpy as np
 import soundfile as sf
 from loguru import logger
 from pydub import AudioSegment
-import av
+
 
 class StreamingAudioWriter:
     """Handles streaming audio format conversions"""
@@ -18,15 +19,29 @@ class StreamingAudioWriter:
         self.sample_rate = sample_rate
         self.channels = channels
         self.bytes_written = 0
-        self.pts=0
+        self.pts = 0
 
-        codec_map = {"wav":"pcm_s16le","mp3":"mp3","opus":"libopus","flac":"flac", "aac":"aac"}
+        codec_map = {
+            "wav": "pcm_s16le",
+            "mp3": "mp3",
+            "opus": "libopus",
+            "flac": "flac",
+            "aac": "aac",
+        }
         # Format-specific setup
-        if self.format in ["wav","flac","mp3","pcm","aac","opus"]:
+        if self.format in ["wav", "flac", "mp3", "pcm", "aac", "opus"]:
             if self.format != "pcm":
                 self.output_buffer = BytesIO()
-                self.container = av.open(self.output_buffer, mode="w", format=self.format if self.format != "aac" else "adts")
-                self.stream = self.container.add_stream(codec_map[self.format],sample_rate=self.sample_rate,layout='mono' if self.channels == 1 else 'stereo')
+                self.container = av.open(
+                    self.output_buffer,
+                    mode="w",
+                    format=self.format if self.format != "aac" else "adts",
+                )
+                self.stream = self.container.add_stream(
+                    codec_map[self.format],
+                    sample_rate=self.sample_rate,
+                    layout="mono" if self.channels == 1 else "stereo",
+                )
                 self.stream.bit_rate = 128000
         else:
             raise ValueError(f"Unsupported format: {format}")
@@ -54,7 +69,7 @@ class StreamingAudioWriter:
                 for packet in packets:
                     self.container.mux(packet)
 
-                data=self.output_buffer.getvalue()
+                data = self.output_buffer.getvalue()
                 self.close()
                 return data
 
@@ -65,9 +80,12 @@ class StreamingAudioWriter:
             # Write raw bytes
             return audio_data.tobytes()
         else:
-            frame = av.AudioFrame.from_ndarray(audio_data.reshape(1, -1), format='s16', layout='mono' if self.channels == 1 else 'stereo')
-            frame.sample_rate=self.sample_rate
-
+            frame = av.AudioFrame.from_ndarray(
+                audio_data.reshape(1, -1),
+                format="s16",
+                layout="mono" if self.channels == 1 else "stereo",
+            )
+            frame.sample_rate = self.sample_rate
 
             frame.pts = self.pts
             self.pts += frame.samples
@@ -80,4 +98,3 @@ class StreamingAudioWriter:
             self.output_buffer.seek(0)
             self.output_buffer.truncate(0)
             return data
-

api/src/services/text_processing/normalizer.py

@@ -6,12 +6,13 @@ Converts them into a format suitable for text-to-speech processing.
 
 import re
 from functools import lru_cache
+
 import inflect
 from numpy import number
-from torch import mul
-from ...structures.schemas import NormalizationOptions
-
 from text_to_num import text2num
+from torch import mul
+
+from ...structures.schemas import NormalizationOptions
 
 # Constants
 VALID_TLDS = [
@@ -54,26 +55,81 @@ VALID_TLDS = [
     "uk",
     "us",
     "io",
+    "co",
 ]
 
 VALID_UNITS = {
-    "m":"meter", "cm":"centimeter", "mm":"millimeter", "km":"kilometer", "in":"inch", "ft":"foot", "yd":"yard", "mi":"mile",  # Length
-    "g":"gram", "kg":"kilogram", "mg":"miligram",      # Mass
-    "s":"second", "ms":"milisecond", "min":"minutes", "h":"hour", # Time
-    "l":"liter", "ml":"mililiter", "cl":"centiliter", "dl":"deciliter",  # Volume
-    "kph":"kilometer per hour", "mph":"mile per hour","mi/h":"mile per hour", "m/s":"meter per second", "km/h":"kilometer per hour", "mm/s":"milimeter per second","cm/s":"centimeter per second", "ft/s":"feet per second","cm/h":"centimeter per day", # Speed
-    "°c":"degree celsius","c":"degree celsius", "°f":"degree fahrenheit","f":"degree fahrenheit", "k":"kelvin",     # Temperature
-    "pa":"pascal", "kpa":"kilopascal", "mpa":"megapascal", "atm":"atmosphere",  # Pressure
-    "hz":"hertz", "khz":"kilohertz", "mhz":"megahertz", "ghz":"gigahertz", # Frequency
-    "v":"volt", "kv":"kilovolt", "mv":"mergavolt",      # Voltage
-    "a":"amp", "ma":"megaamp", "ka":"kiloamp",      # Current
-    "w":"watt", "kw":"kilowatt", "mw":"megawatt",      # Power
-    "j":"joule", "kj":"kilojoule", "mj":"megajoule",      # Energy
-    "Ω":"ohm", "kΩ":"kiloohm", "mΩ":"megaohm",      # Resistance (Ohm)
-    "f":"farad", "µf":"microfarad", "nf":"nanofarad", "pf":"picofarad", # Capacitance
-    "b":"bit", "kb":"kilobit", "mb":"megabit", "gb":"gigabit", "tb":"terabit", "pb":"petabit", # Data size
-    "kbps":"kilobit per second","mbps":"megabit per second","gbps":"gigabit per second","tbps":"terabit per second",
-    "px":"pixel"  # CSS units
+    "m": "meter",
+    "cm": "centimeter",
+    "mm": "millimeter",
+    "km": "kilometer",
+    "in": "inch",
+    "ft": "foot",
+    "yd": "yard",
+    "mi": "mile",  # Length
+    "g": "gram",
+    "kg": "kilogram",
+    "mg": "milligram",  # Mass
+    "s": "second",
+    "ms": "millisecond",
+    "min": "minutes",
+    "h": "hour",  # Time
+    "l": "liter",
+    "ml": "mililiter",
+    "cl": "centiliter",
+    "dl": "deciliter",  # Volume
+    "kph": "kilometer per hour",
+    "mph": "mile per hour",
+    "mi/h": "mile per hour",
+    "m/s": "meter per second",
+    "km/h": "kilometer per hour",
+    "mm/s": "milimeter per second",
+    "cm/s": "centimeter per second",
+    "ft/s": "feet per second",
+    "cm/h": "centimeter per day",  # Speed
+    "°c": "degree celsius",
+    "c": "degree celsius",
+    "°f": "degree fahrenheit",
+    "f": "degree fahrenheit",
+    "k": "kelvin",  # Temperature
+    "pa": "pascal",
+    "kpa": "kilopascal",
+    "mpa": "megapascal",
+    "atm": "atmosphere",  # Pressure
+    "hz": "hertz",
+    "khz": "kilohertz",
+    "mhz": "megahertz",
+    "ghz": "gigahertz",  # Frequency
+    "v": "volt",
+    "kv": "kilovolt",
+    "mv": "mergavolt",  # Voltage
+    "a": "amp",
+    "ma": "megaamp",
+    "ka": "kiloamp",  # Current
+    "w": "watt",
+    "kw": "kilowatt",
+    "mw": "megawatt",  # Power
+    "j": "joule",
+    "kj": "kilojoule",
+    "mj": "megajoule",  # Energy
+    "Ω": "ohm",
+    "kΩ": "kiloohm",
+    "mΩ": "megaohm",  # Resistance (Ohm)
+    "f": "farad",
+    "µf": "microfarad",
+    "nf": "nanofarad",
+    "pf": "picofarad",  # Capacitance
+    "b": "bit",
+    "kb": "kilobit",
+    "mb": "megabit",
+    "gb": "gigabit",
+    "tb": "terabit",
+    "pb": "petabit",  # Data size
+    "kbps": "kilobit per second",
+    "mbps": "megabit per second",
+    "gbps": "gigabit per second",
+    "tbps": "terabit per second",
+    "px": "pixel",  # CSS units
 }
 
 
@@ -88,11 +144,19 @@ URL_PATTERN = re.compile(
     re.IGNORECASE,
 )
 
-UNIT_PATTERN = re.compile(r"((?<!\w)([+-]?)(\d{1,3}(,\d{3})*|\d+)(\.\d+)?)\s*(" + "|".join(sorted(list(VALID_UNITS.keys()),reverse=True)) + r"""){1}(?=[^\w\d]{1}|\b)""",re.IGNORECASE)
+UNIT_PATTERN = re.compile(
+    r"((?<!\w)([+-]?)(\d{1,3}(,\d{3})*|\d+)(\.\d+)?)\s*("
+    + "|".join(sorted(list(VALID_UNITS.keys()), reverse=True))
+    + r"""){1}(?=[^\w\d]{1}|\b)""",
+    re.IGNORECASE,
+)
 
-TIME_PATTERN = re.compile(r"([0-9]{2} ?: ?[0-9]{2}( ?: ?[0-9]{2})?)( ?(pm|am)\b)?", re.IGNORECASE)
+TIME_PATTERN = re.compile(
+    r"([0-9]{2} ?: ?[0-9]{2}( ?: ?[0-9]{2})?)( ?(pm|am)\b)?", re.IGNORECASE
+)
+
+INFLECT_ENGINE = inflect.engine()
 
-INFLECT_ENGINE=inflect.engine()
 
 def split_num(num: re.Match[str]) -> str:
     """Handle number splitting for various formats"""
@@ -118,29 +182,32 @@ def split_num(num: re.Match[str]) -> str:
             return f"{left} oh {right}{s}"
     return f"{left} {right}{s}"
 
+
 def handle_units(u: re.Match[str]) -> str:
     """Converts units to their full form"""
-    unit_string=u.group(6).strip() 
-    unit=unit_string
+    unit_string = u.group(6).strip()
+    unit = unit_string
 
     if unit_string.lower() in VALID_UNITS:
-        unit=VALID_UNITS[unit_string.lower()].split(" ")
+        unit = VALID_UNITS[unit_string.lower()].split(" ")
 
         # Handles the B vs b case
         if unit[0].endswith("bit"):
-            b_case=unit_string[min(1,len(unit_string) - 1)]
+            b_case = unit_string[min(1, len(unit_string) - 1)]
             if b_case == "B":
-                unit[0]=unit[0][:-3] + "byte"
+                unit[0] = unit[0][:-3] + "byte"
 
-        number=u.group(1).strip()
-        unit[0]=INFLECT_ENGINE.no(unit[0],number)
+        number = u.group(1).strip()
+        unit[0] = INFLECT_ENGINE.no(unit[0], number)
     return " ".join(unit)
 
+
 def conditional_int(number: float, threshold: float = 0.00001):
     if abs(round(number) - number) < threshold:
         return int(round(number))
     return number
 
+
 def handle_money(m: re.Match[str]) -> str:
     """Convert money expressions to spoken form"""
 
@@ -166,6 +233,7 @@ def handle_money(m: re.Match[str]) -> str:
 
     return text_number
 
+
 def handle_decimal(num: re.Match[str]) -> str:
     """Convert decimal numbers to spoken form"""
     a, b = num.group().split(".")
@@ -229,34 +297,41 @@ def handle_url(u: re.Match[str]) -> str:
     # Clean up extra spaces
     return re.sub(r"\s+", " ", url).strip()
 
-def handle_phone_number(p: re.Match[str]) -> str:
-    p=list(p.groups())
 
-    country_code=""
+def handle_phone_number(p: re.Match[str]) -> str:
+    p = list(p.groups())
+
+    country_code = ""
     if p[0] is not None:
-        p[0]=p[0].replace("+","")
+        p[0] = p[0].replace("+", "")
         country_code += INFLECT_ENGINE.number_to_words(p[0])
 
-    area_code=INFLECT_ENGINE.number_to_words(p[2].replace("(","").replace(")",""),group=1,comma="")
+    area_code = INFLECT_ENGINE.number_to_words(
+        p[2].replace("(", "").replace(")", ""), group=1, comma=""
+    )
 
-    telephone_prefix=INFLECT_ENGINE.number_to_words(p[3],group=1,comma="")
+    telephone_prefix = INFLECT_ENGINE.number_to_words(p[3], group=1, comma="")
 
-    line_number=INFLECT_ENGINE.number_to_words(p[4],group=1,comma="")
+    line_number = INFLECT_ENGINE.number_to_words(p[4], group=1, comma="")
+
+    return ",".join([country_code, area_code, telephone_prefix, line_number])
 
-    return ",".join([country_code,area_code,telephone_prefix,line_number])
 
 def handle_time(t: re.Match[str]) -> str:
-    t=t.groups()
+    t = t.groups()
 
-    numbers = " ".join([INFLECT_ENGINE.number_to_words(X.strip()) for X in t[0].split(":")])
+    numbers = " ".join(
+        [INFLECT_ENGINE.number_to_words(X.strip()) for X in t[0].split(":")]
+    )
 
-    half=""
+    half = ""
     if t[2] is not None:
-        half=t[2].strip()
+        half = t[2].strip()
 
     return numbers + half
 
-def normalize_text(text: str,normalization_options: NormalizationOptions) -> str:
+
+def normalize_text(text: str, normalization_options: NormalizationOptions) -> str:
     """Normalize text for TTS processing"""
     # Handle email addresses first if enabled
     if normalization_options.email_normalization:
@@ -268,15 +343,19 @@ def normalize_text(text: str,normalization_options: NormalizationOptions) -> str
 
     # Pre-process numbers with units if enabled
     if normalization_options.unit_normalization:
-        text=UNIT_PATTERN.sub(handle_units,text)
+        text = UNIT_PATTERN.sub(handle_units, text)
 
     # Replace optional pluralization
     if normalization_options.optional_pluralization_normalization:
-        text = re.sub(r"\(s\)","s",text)
+        text = re.sub(r"\(s\)", "s", text)
 
     # Replace phone numbers:
     if normalization_options.phone_normalization:
-        text = re.sub(r"(\+?\d{1,2})?([ .-]?)(\(?\d{3}\)?)[\s.-](\d{3})[\s.-](\d{4})",handle_phone_number,text)
+        text = re.sub(
+            r"(\+?\d{1,2})?([ .-]?)(\(?\d{3}\)?)[\s.-](\d{3})[\s.-](\d{4})",
+            handle_phone_number,
+            text,
+        )
 
     # Replace quotes and brackets
     text = text.replace(chr(8216), "'").replace(chr(8217), "'")
@@ -288,7 +367,10 @@ def normalize_text(text: str,normalization_options: NormalizationOptions) -> str
         text = text.replace(a, b + " ")
 
     # Handle simple time in the format of HH:MM:SS
-    text = TIME_PATTERN.sub(handle_time, text, )
+    text = TIME_PATTERN.sub(
+        handle_time,
+        text,
+    )
 
     # Clean up whitespace
     text = re.sub(r"[^\S \n]", " ", text)
@@ -328,6 +410,6 @@ def normalize_text(text: str,normalization_options: NormalizationOptions) -> str
     text = re.sub(
         r"(?:[A-Za-z]\.){2,} [a-z]", lambda m: m.group().replace(".", "-"), text
     )
-    text = re.sub( r"(?i)(?<=[A-Z])\.(?=[A-Z])", "-", text)
+    text = re.sub(r"(?i)(?<=[A-Z])\.(?=[A-Z])", "-", text)
 
     return text.strip()

api/src/services/text_processing/text_processor.py

@@ -7,14 +7,15 @@ from typing import AsyncGenerator, Dict, List, Tuple
 from loguru import logger
 
 from ...core.config import settings
+from ...structures.schemas import NormalizationOptions
 from .normalizer import normalize_text
 from .phonemizer import phonemize
 from .vocabulary import tokenize
-from ...structures.schemas import NormalizationOptions
 
 # Pre-compiled regex patterns for performance
 CUSTOM_PHONEMES = re.compile(r"(\[([^\]]|\n)*?\])(\(\/([^\/)]|\n)*?\/\))")
 
+
 def process_text_chunk(
     text: str, language: str = "a", skip_phonemize: bool = False
 ) -> List[int]:
@@ -41,9 +42,7 @@ def process_text_chunk(
         t1 = time.time()
 
         t0 = time.time()
-        phonemes = phonemize(
-            text, language, normalize=False
-        )  # Already normalized
+        phonemes = phonemize(text, language, normalize=False)  # Already normalized
         t1 = time.time()
 
         t0 = time.time()
@@ -88,7 +87,9 @@ def process_text(text: str, language: str = "a") -> List[int]:
     return process_text_chunk(text, language)
 
 
-def get_sentence_info(text: str, custom_phenomes_list: Dict[str, str]) -> List[Tuple[str, List[int], int]]:
+def get_sentence_info(
+    text: str, custom_phenomes_list: Dict[str, str]
+) -> List[Tuple[str, List[int], int]]:
     """Process all sentences and return info."""
     sentences = re.split(r"([.!?;:])(?=\s|$)", text)
     phoneme_length, min_value = len(custom_phenomes_list), 0
@@ -99,10 +100,11 @@ def get_sentence_info(text: str, custom_phenomes_list: Dict[str, str]) -> List[T
         for replaced in range(min_value, phoneme_length):
             current_id = f"</|custom_phonemes_{replaced}|/>"
             if current_id in sentence:
-                sentence = sentence.replace(current_id, custom_phenomes_list.pop(current_id))
+                sentence = sentence.replace(
+                    current_id, custom_phenomes_list.pop(current_id)
+                )
                 min_value += 1
 
-            
         punct = sentences[i + 1] if i + 1 < len(sentences) else ""
 
         if not sentence:
@@ -114,16 +116,18 @@ def get_sentence_info(text: str, custom_phenomes_list: Dict[str, str]) -> List[T
 
     return results
 
-def handle_custom_phonemes(s: re.Match[str], phenomes_list: Dict[str,str]) -> str:
+
+def handle_custom_phonemes(s: re.Match[str], phenomes_list: Dict[str, str]) -> str:
     latest_id = f"</|custom_phonemes_{len(phenomes_list)}|/>"
     phenomes_list[latest_id] = s.group(0).strip()
     return latest_id
 
+
 async def smart_split(
     text: str,
     max_tokens: int = settings.absolute_max_tokens,
     lang_code: str = "a",
-    normalization_options: NormalizationOptions = NormalizationOptions()
+    normalization_options: NormalizationOptions = NormalizationOptions(),
 ) -> AsyncGenerator[Tuple[str, List[int]], None]:
     """Build optimal chunks targeting 300-400 tokens, never exceeding max_tokens."""
     start_time = time.time()
@@ -135,11 +139,15 @@ async def smart_split(
     # Normalize text
     if settings.advanced_text_normalization and normalization_options.normalize:
         print(lang_code)
-        if lang_code in ["a","b","en-us","en-gb"]:
-            text = CUSTOM_PHONEMES.sub(lambda s: handle_custom_phonemes(s, custom_phoneme_list), text)
-            text=normalize_text(text,normalization_options)
+        if lang_code in ["a", "b", "en-us", "en-gb"]:
+            text = CUSTOM_PHONEMES.sub(
+                lambda s: handle_custom_phonemes(s, custom_phoneme_list), text
+            )
+            text = normalize_text(text, normalization_options)
         else:
-            logger.info("Skipping text normalization as it is only supported for english")
+            logger.info(
+                "Skipping text normalization as it is only supported for english"
+            )
 
     # Process all sentences
     sentences = get_sentence_info(text, custom_phoneme_list)

api/src/services/tts_service.py

@@ -8,7 +8,6 @@ import time
 from typing import AsyncGenerator, List, Optional, Tuple, Union
 
 import numpy as np
-from .streaming_audio_writer import StreamingAudioWriter
 import torch
 from kokoro import KPipeline
 from loguru import logger
@@ -20,6 +19,7 @@ from ..inference.model_manager import get_manager as get_model_manager
 from ..inference.voice_manager import get_manager as get_voice_manager
 from ..structures.schemas import NormalizationOptions
 from .audio import AudioNormalizer, AudioService
+from .streaming_audio_writer import StreamingAudioWriter
 from .text_processing import tokenize
 from .text_processing.text_processor import process_text_chunk, smart_split
 
@@ -69,7 +69,9 @@ class TTSService:
                         yield AudioChunk(np.array([], dtype=np.int16), output=b"")
                         return
                     chunk_data = await AudioService.convert_audio(
-                        AudioChunk(np.array([], dtype=np.float32)),  # Dummy data for type checking
+                        AudioChunk(
+                            np.array([], dtype=np.float32)
+                        ),  # Dummy data for type checking
                         output_format,
                         writer,
                         speed,
@@ -114,13 +116,22 @@ class TTSService:
                             except Exception as e:
                                 logger.error(f"Failed to convert audio: {str(e)}")
                         else:
-                            chunk_data = AudioService.trim_audio(chunk_data, chunk_text, speed, is_last, normalizer)
+                            chunk_data = AudioService.trim_audio(
+                                chunk_data, chunk_text, speed, is_last, normalizer
+                            )
                             yield chunk_data
                         chunk_index += 1
                 else:
                     # For legacy backends, load voice tensor
-                    voice_tensor = await self._voice_manager.load_voice(voice_name, device=backend.device)
-                    chunk_data = await self.model_manager.generate(tokens, voice_tensor, speed=speed, return_timestamps=return_timestamps)
+                    voice_tensor = await self._voice_manager.load_voice(
+                        voice_name, device=backend.device
+                    )
+                    chunk_data = await self.model_manager.generate(
+                        tokens,
+                        voice_tensor,
+                        speed=speed,
+                        return_timestamps=return_timestamps,
+                    )
 
                     if chunk_data.audio is None:
                         logger.error("Model generated None for audio chunk")
@@ -146,7 +157,9 @@ class TTSService:
                         except Exception as e:
                             logger.error(f"Failed to convert audio: {str(e)}")
                     else:
-                        trimmed = AudioService.trim_audio(chunk_data, chunk_text, speed, is_last, normalizer)
+                        trimmed = AudioService.trim_audio(
+                            chunk_data, chunk_text, speed, is_last, normalizer
+                        )
                         yield trimmed
             except Exception as e:
                 logger.error(f"Failed to process tokens: {str(e)}")
@@ -178,7 +191,9 @@ class TTSService:
             # If it is only once voice there is no point in loading it up, doing nothing with it, then saving it
             if len(split_voice) == 1:
                 # Since its a single voice the only time that the weight would matter is if voice_weight_normalization is off
-                if ("(" not in voice and ")" not in voice) or settings.voice_weight_normalization == True:
+                if (
+                    "(" not in voice and ")" not in voice
+                ) or settings.voice_weight_normalization == True:
                     path = await self._voice_manager.get_voice_path(voice)
                     if not path:
                         raise RuntimeError(f"Voice not found: {voice}")
@@ -206,13 +221,19 @@ class TTSService:
 
             # Load the first voice as the starting point for voices to be combined onto
             path = await self._voice_manager.get_voice_path(split_voice[0][0])
-            combined_tensor = await self._load_voice_from_path(path, split_voice[0][1] / total_weight)
+            combined_tensor = await self._load_voice_from_path(
+                path, split_voice[0][1] / total_weight
+            )
 
             # Loop through each + or - in split_voice so they can be applied to combined voice
             for operation_index in range(1, len(split_voice) - 1, 2):
                 # Get the voice path of the voice 1 index ahead of the operator
-                path = await self._voice_manager.get_voice_path(split_voice[operation_index + 1][0])
-                voice_tensor = await self._load_voice_from_path(path, split_voice[operation_index + 1][1] / total_weight)
+                path = await self._voice_manager.get_voice_path(
+                    split_voice[operation_index + 1][0]
+                )
+                voice_tensor = await self._load_voice_from_path(
+                    path, split_voice[operation_index + 1][1] / total_weight
+                )
 
                 # Either add or subtract the voice from the current combined voice
                 if split_voice[operation_index] == "+":
@@ -255,10 +276,16 @@ class TTSService:
 
             # Use provided lang_code or determine from voice name
             pipeline_lang_code = lang_code if lang_code else voice[:1].lower()
-            logger.info(f"Using lang_code '{pipeline_lang_code}' for voice '{voice_name}' in audio stream")
+            logger.info(
+                f"Using lang_code '{pipeline_lang_code}' for voice '{voice_name}' in audio stream"
+            )
 
             # Process text in chunks with smart splitting
-            async for chunk_text, tokens in smart_split(text, lang_code=pipeline_lang_code, normalization_options=normalization_options):
+            async for chunk_text, tokens in smart_split(
+                text,
+                lang_code=pipeline_lang_code,
+                normalization_options=normalization_options,
+            ):
                 try:
                     # Process audio for chunk
                     async for chunk_data in self._process_chunk(
@@ -286,10 +313,14 @@ class TTSService:
                             yield chunk_data
 
                         else:
-                            logger.warning(f"No audio generated for chunk: '{chunk_text[:100]}...'")
+                            logger.warning(
+                                f"No audio generated for chunk: '{chunk_text[:100]}...'"
+                            )
                         chunk_index += 1
                 except Exception as e:
-                    logger.error(f"Failed to process audio for chunk: '{chunk_text[:100]}...'. Error: {str(e)}")
+                    logger.error(
+                        f"Failed to process audio for chunk: '{chunk_text[:100]}...'. Error: {str(e)}"
+                    )
                     continue
 
             # Only finalize if we successfully processed at least one chunk
@@ -332,7 +363,16 @@ class TTSService:
         audio_data_chunks = []
 
         try:
-            async for audio_stream_data in self.generate_audio_stream(text, voice, writer, speed=speed, normalization_options=normalization_options, return_timestamps=return_timestamps, lang_code=lang_code, output_format=None):
+            async for audio_stream_data in self.generate_audio_stream(
+                text,
+                voice,
+                writer,
+                speed=speed,
+                normalization_options=normalization_options,
+                return_timestamps=return_timestamps,
+                lang_code=lang_code,
+                output_format=None,
+            ):
                 if len(audio_stream_data.audio) > 0:
                     audio_data_chunks.append(audio_stream_data)
 
@@ -384,11 +424,15 @@ class TTSService:
                 result = None
                 # Use provided lang_code or determine from voice name
                 pipeline_lang_code = lang_code if lang_code else voice[:1].lower()
-                logger.info(f"Using lang_code '{pipeline_lang_code}' for voice '{voice_name}' in phoneme pipeline")
+                logger.info(
+                    f"Using lang_code '{pipeline_lang_code}' for voice '{voice_name}' in phoneme pipeline"
+                )
 
                 try:
                     # Use backend's pipeline management
-                    for r in backend._get_pipeline(pipeline_lang_code).generate_from_tokens(
+                    for r in backend._get_pipeline(
+                        pipeline_lang_code
+                    ).generate_from_tokens(
                         tokens=phonemes,  # Pass raw phonemes string
                         voice=voice_path,
                         speed=speed,
@@ -406,7 +450,9 @@ class TTSService:
                 processing_time = time.time() - start_time
                 return result.audio.numpy(), processing_time
             else:
-                raise ValueError("Phoneme generation only supported with Kokoro V1 backend")
+                raise ValueError(
+                    "Phoneme generation only supported with Kokoro V1 backend"
+                )
 
         except Exception as e:
             logger.error(f"Error in phoneme audio generation: {str(e)}")

api/src/structures/custom_responses.py

@@ -24,16 +24,12 @@ class JSONStreamingResponse(StreamingResponse, JSONResponse):
         else:
             self._content_iterable = iterate_in_threadpool(content)
 
-        
-
         async def body_iterator() -> AsyncIterable[bytes]:
             async for content_ in self._content_iterable:
                 if isinstance(content_, BaseModel):
                     content_ = content_.model_dump()
                 yield self.render(content_)
 
-        
-        
         self.body_iterator = body_iterator()
         self.status_code = status_code
         if media_type is not None:
@@ -42,10 +38,13 @@ class JSONStreamingResponse(StreamingResponse, JSONResponse):
         self.init_headers(headers)
 
     def render(self, content: typing.Any) -> bytes:
-            return (json.dumps(
+        return (
+            json.dumps(
                 content,
                 ensure_ascii=False,
                 allow_nan=False,
                 indent=None,
                 separators=(",", ":"),
-            ) + "\n").encode("utf-8")
+            )
+            + "\n"
+        ).encode("utf-8")

api/src/structures/schemas.py

@@ -35,16 +35,38 @@ class CaptionedSpeechResponse(BaseModel):
 
     audio: str = Field(..., description="The generated audio data encoded in base 64")
     audio_format: str = Field(..., description="The format of the output audio")
-    timestamps: Optional[List[WordTimestamp]] = Field(..., description="Word-level timestamps")
+    timestamps: Optional[List[WordTimestamp]] = Field(
+        ..., description="Word-level timestamps"
+    )
+
 
 class NormalizationOptions(BaseModel):
     """Options for the normalization system"""
-    normalize: bool = Field(default=True, description="Normalizes input text to make it easier for the model to say")
-    unit_normalization: bool = Field(default=False,description="Transforms units like 10KB to 10 kilobytes")
-    url_normalization: bool = Field(default=True, description="Changes urls so they can be properly pronouced by kokoro")
-    email_normalization: bool = Field(default=True, description="Changes emails so they can be properly pronouced by kokoro")
-    optional_pluralization_normalization: bool = Field(default=True, description="Replaces (s) with s so some words get pronounced correctly")
-    phone_normalization: bool = Field(default=True, description="Changes phone numbers so they can be properly pronouced by kokoro")
+
+    normalize: bool = Field(
+        default=True,
+        description="Normalizes input text to make it easier for the model to say",
+    )
+    unit_normalization: bool = Field(
+        default=False, description="Transforms units like 10KB to 10 kilobytes"
+    )
+    url_normalization: bool = Field(
+        default=True,
+        description="Changes urls so they can be properly pronounced by kokoro",
+    )
+    email_normalization: bool = Field(
+        default=True,
+        description="Changes emails so they can be properly pronouced by kokoro",
+    )
+    optional_pluralization_normalization: bool = Field(
+        default=True,
+        description="Replaces (s) with s so some words get pronounced correctly",
+    )
+    phone_normalization: bool = Field(
+        default=True,
+        description="Changes phone numbers so they can be properly pronouced by kokoro",
+    )
+
 
 class OpenAISpeechRequest(BaseModel):
     """Request schema for OpenAI-compatible speech endpoint"""
@@ -62,9 +84,11 @@ class OpenAISpeechRequest(BaseModel):
         default="mp3",
         description="The format to return audio in. Supported formats: mp3, opus, flac, wav, pcm. PCM format returns raw 16-bit samples without headers. AAC is not currently supported.",
     )
-    download_format: Optional[Literal["mp3", "opus", "aac", "flac", "wav", "pcm"]] = Field(
-        default=None,
-        description="Optional different format for the final download. If not provided, uses response_format.",
+    download_format: Optional[Literal["mp3", "opus", "aac", "flac", "wav", "pcm"]] = (
+        Field(
+            default=None,
+            description="Optional different format for the final download. If not provided, uses response_format.",
+        )
     )
     speed: float = Field(
         default=1.0,
@@ -85,8 +109,8 @@ class OpenAISpeechRequest(BaseModel):
         description="Optional language code to use for text processing. If not provided, will use first letter of voice name.",
     )
     normalization_options: Optional[NormalizationOptions] = Field(
-        default= NormalizationOptions(),
-        description= "Options for the normalization system"
+        default=NormalizationOptions(),
+        description="Options for the normalization system",
     )
 
 
@@ -129,6 +153,6 @@ class CaptionedSpeechRequest(BaseModel):
         description="Optional language code to use for text processing. If not provided, will use first letter of voice name.",
     )
     normalization_options: Optional[NormalizationOptions] = Field(
-        default= NormalizationOptions(),
-        description= "Options for the normalization system"
+        default=NormalizationOptions(),
+        description="Options for the normalization system",
     )

api/tests/conftest.py

@@ -69,4 +69,3 @@ async def tts_service(mock_model_manager, mock_voice_manager):
 def test_voice():
     """Return a test voice name."""
     return "voice1"
-

api/tests/test_audio_service.py

@@ -5,9 +5,11 @@ from unittest.mock import patch
 import numpy as np
 import pytest
 
-from api.src.services.audio import AudioNormalizer, AudioService
 from api.src.inference.base import AudioChunk
+from api.src.services.audio import AudioNormalizer, AudioService
 from api.src.services.streaming_audio_writer import StreamingAudioWriter
+
+
 @pytest.fixture(autouse=True)
 def mock_settings():
     """Mock settings for all tests"""
@@ -64,7 +66,9 @@ async def test_convert_to_mp3(sample_audio):
     assert isinstance(audio_chunk, AudioChunk)
     assert len(audio_chunk.output) > 0
     # Check MP3 header (ID3 or MPEG frame sync)
-    assert audio_chunk.output.startswith(b"ID3") or audio_chunk.output.startswith(b"\xff\xfb")
+    assert audio_chunk.output.startswith(b"ID3") or audio_chunk.output.startswith(
+        b"\xff\xfb"
+    )
 
 
 @pytest.mark.asyncio
@@ -76,7 +80,7 @@ async def test_convert_to_opus(sample_audio):
     writer = StreamingAudioWriter("opus", sample_rate=24000)
 
     audio_chunk = await AudioService.convert_audio(
-        AudioChunk(audio_data), "opus",writer
+        AudioChunk(audio_data), "opus", writer
     )
 
     writer.close()
@@ -125,7 +129,9 @@ async def test_convert_to_aac(sample_audio):
     assert isinstance(audio_chunk, AudioChunk)
     assert len(audio_chunk.output) > 0
     # Check ADTS header (AAC)
-    assert audio_chunk.output.startswith(b"\xff\xf0") or audio_chunk.output.startswith(b"\xff\xf1")
+    assert audio_chunk.output.startswith(b"\xff\xf0") or audio_chunk.output.startswith(
+        b"\xff\xf1"
+    )
 
 
 @pytest.mark.asyncio
@@ -150,7 +156,7 @@ async def test_convert_to_pcm(sample_audio):
 @pytest.mark.asyncio
 async def test_convert_to_invalid_format_raises_error(sample_audio):
     """Test that converting to an invalid format raises an error"""
-    #audio_data, sample_rate = sample_audio
+    # audio_data, sample_rate = sample_audio
     with pytest.raises(ValueError, match="Unsupported format: invalid"):
         writer = StreamingAudioWriter("invalid", sample_rate=24000)
 
@@ -212,7 +218,6 @@ async def test_different_sample_rates(sample_audio):
     sample_rates = [8000, 16000, 44100, 48000]
 
     for rate in sample_rates:
-
         writer = StreamingAudioWriter("wav", sample_rate=rate)
 
         audio_chunk = await AudioService.convert_audio(

api/tests/test_development.py

@@ -1,8 +1,10 @@
-import pytest
-from unittest.mock import patch, MagicMock
-import requests
 import base64
 import json
+from unittest.mock import MagicMock, patch
+
+import pytest
+import requests
+
 
 def test_generate_captioned_speech():
     """Test the generate_captioned_speech function with mocked responses"""
@@ -12,14 +14,15 @@ def test_generate_captioned_speech():
 
     mock_timestamps_response = MagicMock()
     mock_timestamps_response.status_code = 200
-    mock_timestamps_response.content = json.dumps({
-        "audio":base64.b64encode(b"mock audio data").decode("utf-8"),
-        "timestamps":[{"word": "test", "start_time": 0.0, "end_time": 1.0}]
-    })
+    mock_timestamps_response.content = json.dumps(
+        {
+            "audio": base64.b64encode(b"mock audio data").decode("utf-8"),
+            "timestamps": [{"word": "test", "start_time": 0.0, "end_time": 1.0}],
+        }
+    )
 
     # Patch the HTTP requests
-    with patch('requests.post', return_value=mock_timestamps_response):
-        
+    with patch("requests.post", return_value=mock_timestamps_response):
         # Import here to avoid module-level import issues
         from examples.captioned_speech_example import generate_captioned_speech
 

api/tests/test_normalizer.py

@@ -5,27 +5,48 @@ import pytest
 from api.src.services.text_processing.normalizer import normalize_text
 from api.src.structures.schemas import NormalizationOptions
 
+
 def test_url_protocols():
     """Test URL protocol handling"""
     assert (
-        normalize_text("Check out https://example.com",normalization_options=NormalizationOptions())
+        normalize_text(
+            "Check out https://example.com",
+            normalization_options=NormalizationOptions(),
+        )
         == "Check out https example dot com"
     )
-    assert normalize_text("Visit http://site.com",normalization_options=NormalizationOptions()) == "Visit http site dot com"
     assert (
-        normalize_text("Go to https://test.org/path",normalization_options=NormalizationOptions())
+        normalize_text(
+            "Visit http://site.com", normalization_options=NormalizationOptions()
+        )
+        == "Visit http site dot com"
+    )
+    assert (
+        normalize_text(
+            "Go to https://test.org/path", normalization_options=NormalizationOptions()
+        )
         == "Go to https test dot org slash path"
     )
 
 
 def test_url_www():
     """Test www prefix handling"""
-    assert normalize_text("Go to www.example.com",normalization_options=NormalizationOptions()) == "Go to www example dot com"
     assert (
-        normalize_text("Visit www.test.org/docs",normalization_options=NormalizationOptions()) == "Visit www test dot org slash docs"
+        normalize_text(
+            "Go to www.example.com", normalization_options=NormalizationOptions()
+        )
+        == "Go to www example dot com"
     )
     assert (
-        normalize_text("Check www.site.com?q=test",normalization_options=NormalizationOptions())
+        normalize_text(
+            "Visit www.test.org/docs", normalization_options=NormalizationOptions()
+        )
+        == "Visit www test dot org slash docs"
+    )
+    assert (
+        normalize_text(
+            "Check www.site.com?q=test", normalization_options=NormalizationOptions()
+        )
         == "Check www site dot com question-mark q equals test"
     )
 
@@ -33,15 +54,21 @@ def test_url_www():
 def test_url_localhost():
     """Test localhost URL handling"""
     assert (
-        normalize_text("Running on localhost:7860",normalization_options=NormalizationOptions())
+        normalize_text(
+            "Running on localhost:7860", normalization_options=NormalizationOptions()
+        )
         == "Running on localhost colon 78 60"
     )
     assert (
-        normalize_text("Server at localhost:8080/api",normalization_options=NormalizationOptions())
+        normalize_text(
+            "Server at localhost:8080/api", normalization_options=NormalizationOptions()
+        )
         == "Server at localhost colon 80 80 slash api"
     )
     assert (
-        normalize_text("Test localhost:3000/test?v=1",normalization_options=NormalizationOptions())
+        normalize_text(
+            "Test localhost:3000/test?v=1", normalization_options=NormalizationOptions()
+        )
         == "Test localhost colon 3000 slash test question-mark v equals 1"
     )
 
@@ -49,48 +76,104 @@ def test_url_localhost():
 def test_url_ip_addresses():
     """Test IP address URL handling"""
     assert (
-        normalize_text("Access 0.0.0.0:9090/test",normalization_options=NormalizationOptions())
+        normalize_text(
+            "Access 0.0.0.0:9090/test", normalization_options=NormalizationOptions()
+        )
         == "Access 0 dot 0 dot 0 dot 0 colon 90 90 slash test"
     )
     assert (
-        normalize_text("API at 192.168.1.1:8000",normalization_options=NormalizationOptions())
+        normalize_text(
+            "API at 192.168.1.1:8000", normalization_options=NormalizationOptions()
+        )
         == "API at 192 dot 168 dot 1 dot 1 colon 8000"
     )
-    assert normalize_text("Server 127.0.0.1",normalization_options=NormalizationOptions()) == "Server 127 dot 0 dot 0 dot 1"
+    assert (
+        normalize_text("Server 127.0.0.1", normalization_options=NormalizationOptions())
+        == "Server 127 dot 0 dot 0 dot 1"
+    )
 
 
 def test_url_raw_domains():
     """Test raw domain handling"""
     assert (
-        normalize_text("Visit google.com/search",normalization_options=NormalizationOptions()) == "Visit google dot com slash search"
+        normalize_text(
+            "Visit google.com/search", normalization_options=NormalizationOptions()
+        )
+        == "Visit google dot com slash search"
     )
     assert (
-        normalize_text("Go to example.com/path?q=test",normalization_options=NormalizationOptions())
+        normalize_text(
+            "Go to example.com/path?q=test",
+            normalization_options=NormalizationOptions(),
+        )
         == "Go to example dot com slash path question-mark q equals test"
     )
-    assert normalize_text("Check docs.test.com",normalization_options=NormalizationOptions()) == "Check docs dot test dot com"
+    assert (
+        normalize_text(
+            "Check docs.test.com", normalization_options=NormalizationOptions()
+        )
+        == "Check docs dot test dot com"
+    )
 
 
 def test_url_email_addresses():
     """Test email address handling"""
     assert (
-        normalize_text("Email me at user@example.com",normalization_options=NormalizationOptions())
+        normalize_text(
+            "Email me at user@example.com", normalization_options=NormalizationOptions()
+        )
         == "Email me at user at example dot com"
     )
-    assert normalize_text("Contact admin@test.org",normalization_options=NormalizationOptions()) == "Contact admin at test dot org"
     assert (
-        normalize_text("Send to test.user@site.com",normalization_options=NormalizationOptions())
+        normalize_text(
+            "Contact admin@test.org", normalization_options=NormalizationOptions()
+        )
+        == "Contact admin at test dot org"
+    )
+    assert (
+        normalize_text(
+            "Send to test.user@site.com", normalization_options=NormalizationOptions()
+        )
         == "Send to test dot user at site dot com"
     )
 
+
 def test_money():
     """Test that money text is normalized correctly"""
-    assert normalize_text("He lost $5.3 thousand.",normalization_options=NormalizationOptions()) == "He lost five point three thousand dollars."
-    assert normalize_text("To put it weirdly -$6.9 million",normalization_options=NormalizationOptions()) == "To put it weirdly minus six point nine million dollars"
-    assert normalize_text("It costs $50.3.",normalization_options=NormalizationOptions()) == "It costs fifty dollars and thirty cents."
+    assert (
+        normalize_text(
+            "He lost $5.3 thousand.", normalization_options=NormalizationOptions()
+        )
+        == "He lost five point three thousand dollars."
+    )
+    assert (
+        normalize_text(
+            "To put it weirdly -$6.9 million",
+            normalization_options=NormalizationOptions(),
+        )
+        == "To put it weirdly minus six point nine million dollars"
+    )
+    assert (
+        normalize_text("It costs $50.3.", normalization_options=NormalizationOptions())
+        == "It costs fifty dollars and thirty cents."
+    )
+
 
 def test_non_url_text():
     """Test that non-URL text is unaffected"""
-    assert normalize_text("This is not.a.url text",normalization_options=NormalizationOptions()) == "This is not-a-url text"
-    assert normalize_text("Hello, how are you today?",normalization_options=NormalizationOptions()) == "Hello, how are you today?"
-    assert normalize_text("It costs $50.",normalization_options=NormalizationOptions()) == "It costs fifty dollars."
+    assert (
+        normalize_text(
+            "This is not.a.url text", normalization_options=NormalizationOptions()
+        )
+        == "This is not-a-url text"
+    )
+    assert (
+        normalize_text(
+            "Hello, how are you today?", normalization_options=NormalizationOptions()
+        )
+        == "Hello, how are you today?"
+    )
+    assert (
+        normalize_text("It costs $50.", normalization_options=NormalizationOptions())
+        == "It costs fifty dollars."
+    )

api/tests/test_openai_endpoints.py

@@ -4,20 +4,19 @@ import os
 from typing import AsyncGenerator, Tuple
 from unittest.mock import AsyncMock, MagicMock, patch
 
-from api.src.services.streaming_audio_writer import StreamingAudioWriter
-
-from api.src.inference.base import AudioChunk
 import numpy as np
 import pytest
 from fastapi.testclient import TestClient
 
 from api.src.core.config import settings
+from api.src.inference.base import AudioChunk
 from api.src.main import app
 from api.src.routers.openai_compatible import (
     get_tts_service,
     load_openai_mappings,
     stream_audio_chunks,
 )
+from api.src.services.streaming_audio_writer import StreamingAudioWriter
 from api.src.services.tts_service import TTSService
 from api.src.structures.schemas import OpenAISpeechRequest
 
@@ -114,7 +113,6 @@ def test_retrieve_model(mock_openai_mappings):
     assert error["detail"]["type"] == "invalid_request_error"
 
 
-
 @pytest.mark.asyncio
 async def test_get_tts_service_initialization():
     """Test TTSService initialization"""
@@ -147,7 +145,7 @@ async def test_stream_audio_chunks_client_disconnect():
 
     async def mock_stream(*args, **kwargs):
         for i in range(5):
-            yield AudioChunk(np.ndarray([],np.int16),output=b"chunk")
+            yield AudioChunk(np.ndarray([], np.int16), output=b"chunk")
 
     mock_service.generate_audio_stream = mock_stream
     mock_service.list_voices.return_value = ["test_voice"]
@@ -243,10 +241,10 @@ def mock_tts_service(mock_audio_bytes):
     """Mock TTS service for testing."""
     with patch("api.src.routers.openai_compatible.get_tts_service") as mock_get:
         service = AsyncMock(spec=TTSService)
-        service.generate_audio.return_value = AudioChunk(np.zeros(1000,np.int16))
+        service.generate_audio.return_value = AudioChunk(np.zeros(1000, np.int16))
 
         async def mock_stream(*args, **kwargs) -> AsyncGenerator[AudioChunk, None]:
-            yield AudioChunk(np.ndarray([],np.int16),output=mock_audio_bytes)
+            yield AudioChunk(np.ndarray([], np.int16), output=mock_audio_bytes)
 
         service.generate_audio_stream = mock_stream
         service.list_voices.return_value = ["test_voice", "voice1", "voice2"]
@@ -263,8 +261,10 @@ def test_openai_speech_endpoint(
 ):
     """Test the OpenAI-compatible speech endpoint with basic MP3 generation"""
     # Configure mocks
-    mock_tts_service.generate_audio.return_value = AudioChunk(np.zeros(1000,np.int16))
-    mock_convert.return_value = AudioChunk(np.zeros(1000,np.int16),output=mock_audio_bytes)
+    mock_tts_service.generate_audio.return_value = AudioChunk(np.zeros(1000, np.int16))
+    mock_convert.return_value = AudioChunk(
+        np.zeros(1000, np.int16), output=mock_audio_bytes
+    )
 
     response = client.post(
         "/v1/audio/speech",

api/tests/test_text_processor.py

@@ -44,9 +44,12 @@ def test_get_sentence_info():
         assert count == len(tokens)
         assert count > 0
 
+
 def test_get_sentence_info_phenomoes():
     """Test sentence splitting and info extraction."""
-    text = "This is sentence one. This is </|custom_phonemes_0|/> two! What about three?"
+    text = (
+        "This is sentence one. This is </|custom_phonemes_0|/> two! What about three?"
+    )
     results = get_sentence_info(text, {"</|custom_phonemes_0|/>": r"sˈɛntᵊns"})
 
     assert len(results) == 3
@@ -58,6 +61,7 @@ def test_get_sentence_info_phenomoes():
         assert count == len(tokens)
         assert count > 0
 
+
 @pytest.mark.asyncio
 async def test_smart_split_short_text():
     """Test smart splitting with text under max tokens."""

charts/kokoro-fastapi/Chart.yaml

@@ -2,8 +2,8 @@ apiVersion: v2
 name: kokoro-fastapi
 description: A Helm chart for deploying the Kokoro FastAPI TTS service to Kubernetes
 type: application
-version: 0.2.0
-appVersion: "0.2.0"
+version: 0.3.0
+appVersion: "0.3.0"
 
 keywords:
   - tts

dev/Test Threads.py

@@ -17,35 +17,44 @@ import base64
 import concurrent.futures
 import json
 import os
-import requests
+import sys
 import time
 import wave
-import sys
 from pathlib import Path
 
+import requests
+
 
 def setup_args():
     """Parse command line arguments"""
     parser = argparse.ArgumentParser(description="Test Kokoro TTS for race conditions")
-    parser.add_argument("--url", default="http://localhost:8880",
-                      help="Base URL of the Kokoro TTS service")
-    parser.add_argument("--threads", type=int, default=8,
-                      help="Number of concurrent threads to use")
-    parser.add_argument("--iterations", type=int, default=5,
-                      help="Number of iterations per thread")
-    parser.add_argument("--voice", default="af_heart",
-                      help="Voice to use for TTS")
-    parser.add_argument("--output-dir", default="./tts_test_output",
-                      help="Directory to save output files")
-    parser.add_argument("--debug", action="store_true",
-                      help="Enable debug logging")
+    parser.add_argument(
+        "--url",
+        default="http://localhost:8880",
+        help="Base URL of the Kokoro TTS service",
+    )
+    parser.add_argument(
+        "--threads", type=int, default=8, help="Number of concurrent threads to use"
+    )
+    parser.add_argument(
+        "--iterations", type=int, default=5, help="Number of iterations per thread"
+    )
+    parser.add_argument("--voice", default="af_heart", help="Voice to use for TTS")
+    parser.add_argument(
+        "--output-dir",
+        default="./tts_test_output",
+        help="Directory to save output files",
+    )
+    parser.add_argument("--debug", action="store_true", help="Enable debug logging")
     return parser.parse_args()
 
 
 def generate_test_sentence(thread_id, iteration):
     """Generate a simple test sentence with numbers to make mismatches easily identifiable"""
-    return f"This is test sentence number {thread_id}-{iteration}. " \
-           f"If you hear this sentence, you should hear the numbers {thread_id}-{iteration}."
+    return (
+        f"This is test sentence number {thread_id}-{iteration}. "
+        f"If you hear this sentence, you should hear the numbers {thread_id}-{iteration}."
+    )
 
 
 def log_message(message, debug=False, is_error=False):
@@ -73,7 +82,9 @@ def request_tts(url, test_id, text, voice, output_dir, debug=False):
             f.write(text)
         log_message(f"Thread {test_id}: Successfully saved text file", debug)
     except Exception as e:
-        log_message(f"Thread {test_id}: Error saving text file: {str(e)}", debug, is_error=True)
+        log_message(
+            f"Thread {test_id}: Error saving text file: {str(e)}", debug, is_error=True
+        )
 
     # Make the TTS request
     try:
@@ -85,56 +96,102 @@ def request_tts(url, test_id, text, voice, output_dir, debug=False):
                 "model": "kokoro",
                 "input": text,
                 "voice": voice,
-                "response_format": "wav"
+                "response_format": "wav",
             },
             headers={"Accept": "audio/wav"},
-            timeout=60  # Increase timeout to 60 seconds
+            timeout=60,  # Increase timeout to 60 seconds
         )
 
-        log_message(f"Thread {test_id}: Response status code: {response.status_code}", debug)
-        log_message(f"Thread {test_id}: Response content type: {response.headers.get('Content-Type', 'None')}", debug)
-        log_message(f"Thread {test_id}: Response content length: {len(response.content)} bytes", debug)
+        log_message(
+            f"Thread {test_id}: Response status code: {response.status_code}", debug
+        )
+        log_message(
+            f"Thread {test_id}: Response content type: {response.headers.get('Content-Type', 'None')}",
+            debug,
+        )
+        log_message(
+            f"Thread {test_id}: Response content length: {len(response.content)} bytes",
+            debug,
+        )
 
         if response.status_code != 200:
-            log_message(f"Thread {test_id}: API error: {response.status_code} - {response.text}", debug, is_error=True)
+            log_message(
+                f"Thread {test_id}: API error: {response.status_code} - {response.text}",
+                debug,
+                is_error=True,
+            )
             return False
 
         # Check if we got valid audio data
-        if len(response.content) < 100:  # Sanity check - WAV files should be larger than this
-            log_message(f"Thread {test_id}: Received suspiciously small audio data: {len(response.content)} bytes", debug, is_error=True)
-            log_message(f"Thread {test_id}: Content (base64): {base64.b64encode(response.content).decode('utf-8')}", debug, is_error=True)
+        if (
+            len(response.content) < 100
+        ):  # Sanity check - WAV files should be larger than this
+            log_message(
+                f"Thread {test_id}: Received suspiciously small audio data: {len(response.content)} bytes",
+                debug,
+                is_error=True,
+            )
+            log_message(
+                f"Thread {test_id}: Content (base64): {base64.b64encode(response.content).decode('utf-8')}",
+                debug,
+                is_error=True,
+            )
             return False
 
         # Save the audio output with explicit error handling
         try:
             with open(output_file, "wb") as f:
                 bytes_written = f.write(response.content)
-                log_message(f"Thread {test_id}: Wrote {bytes_written} bytes to {output_file}", debug)
+                log_message(
+                    f"Thread {test_id}: Wrote {bytes_written} bytes to {output_file}",
+                    debug,
+                )
 
             # Verify the WAV file exists and has content
             if os.path.exists(output_file):
                 file_size = os.path.getsize(output_file)
-                log_message(f"Thread {test_id}: Verified file exists with size: {file_size} bytes", debug)
+                log_message(
+                    f"Thread {test_id}: Verified file exists with size: {file_size} bytes",
+                    debug,
+                )
 
                 # Validate WAV file by reading its headers
                 try:
-                    with wave.open(output_file, 'rb') as wav_file:
+                    with wave.open(output_file, "rb") as wav_file:
                         channels = wav_file.getnchannels()
                         sample_width = wav_file.getsampwidth()
                         framerate = wav_file.getframerate()
                         frames = wav_file.getnframes()
-                        log_message(f"Thread {test_id}: Valid WAV file - channels: {channels}, "
-                                   f"sample width: {sample_width}, framerate: {framerate}, frames: {frames}", debug)
+                        log_message(
+                            f"Thread {test_id}: Valid WAV file - channels: {channels}, "
+                            f"sample width: {sample_width}, framerate: {framerate}, frames: {frames}",
+                            debug,
+                        )
                 except Exception as wav_error:
-                    log_message(f"Thread {test_id}: Invalid WAV file: {str(wav_error)}", debug, is_error=True)
+                    log_message(
+                        f"Thread {test_id}: Invalid WAV file: {str(wav_error)}",
+                        debug,
+                        is_error=True,
+                    )
             else:
-                log_message(f"Thread {test_id}: File was not created: {output_file}", debug, is_error=True)
+                log_message(
+                    f"Thread {test_id}: File was not created: {output_file}",
+                    debug,
+                    is_error=True,
+                )
         except Exception as save_error:
-            log_message(f"Thread {test_id}: Error saving audio file: {str(save_error)}", debug, is_error=True)
+            log_message(
+                f"Thread {test_id}: Error saving audio file: {str(save_error)}",
+                debug,
+                is_error=True,
+            )
             return False
 
         end_time = time.time()
-        log_message(f"Thread {test_id}: Saved output to {output_file} (time: {end_time - start_time:.2f}s)", debug)
+        log_message(
+            f"Thread {test_id}: Saved output to {output_file} (time: {end_time - start_time:.2f}s)",
+            debug,
+        )
         return True
 
     except requests.exceptions.Timeout:
@@ -151,10 +208,16 @@ def worker_task(thread_id, args):
         iteration = i + 1
         test_id = f"{thread_id:02d}_{iteration:02d}"
         text = generate_test_sentence(thread_id, iteration)
-        success = request_tts(args.url, test_id, text, args.voice, args.output_dir, args.debug)
+        success = request_tts(
+            args.url, test_id, text, args.voice, args.output_dir, args.debug
+        )
 
         if not success:
-            log_message(f"Thread {thread_id}: Iteration {iteration} failed", args.debug, is_error=True)
+            log_message(
+                f"Thread {thread_id}: Iteration {iteration} failed",
+                args.debug,
+                is_error=True,
+            )
 
         # Small delay between iterations to avoid overwhelming the API
         time.sleep(0.1)
@@ -171,9 +234,14 @@ def run_test(args):
         with open(test_file, "w") as f:
             f.write("Testing write access\n")
         os.remove(test_file)
-        log_message(f"Successfully verified write access to output directory: {args.output_dir}")
+        log_message(
+            f"Successfully verified write access to output directory: {args.output_dir}"
+        )
     except Exception as e:
-        log_message(f"Warning: Cannot write to output directory {args.output_dir}: {str(e)}", is_error=True)
+        log_message(
+            f"Warning: Cannot write to output directory {args.output_dir}: {str(e)}",
+            is_error=True,
+        )
         log_message(f"Current directory: {os.getcwd()}", is_error=True)
         log_message(f"Directory contents: {os.listdir('.')}", is_error=True)
 
@@ -183,13 +251,21 @@ def run_test(args):
         if response.status_code == 200:
             log_message(f"Successfully connected to Kokoro TTS service at {args.url}")
         else:
-            log_message(f"Warning: Kokoro TTS service health check returned status {response.status_code}", is_error=True)
+            log_message(
+                f"Warning: Kokoro TTS service health check returned status {response.status_code}",
+                is_error=True,
+            )
     except Exception as e:
-        log_message(f"Warning: Cannot connect to Kokoro TTS service at {args.url}: {str(e)}", is_error=True)
+        log_message(
+            f"Warning: Cannot connect to Kokoro TTS service at {args.url}: {str(e)}",
+            is_error=True,
+        )
 
     # Record start time
     start_time = time.time()
-    log_message(f"Starting test with {args.threads} threads, {args.iterations} iterations per thread")
+    log_message(
+        f"Starting test with {args.threads} threads, {args.iterations} iterations per thread"
+    )
 
     # Create and start worker threads
     with concurrent.futures.ThreadPoolExecutor(max_workers=args.threads) as executor:
@@ -202,7 +278,9 @@ def run_test(args):
             try:
                 future.result()
             except Exception as e:
-                log_message(f"Thread execution failed: {str(e)}", args.debug, is_error=True)
+                log_message(
+                    f"Thread execution failed: {str(e)}", args.debug, is_error=True
+                )
 
     # Record end time and print summary
     end_time = time.time()
@@ -213,8 +291,12 @@ def run_test(args):
     log_message(f"Average time per request: {total_time / total_requests:.2f} seconds")
     log_message(f"Requests per second: {total_requests / total_time:.2f}")
     log_message(f"Output files saved to: {os.path.abspath(args.output_dir)}")
-    log_message("To verify, listen to the audio files and check if they match the text files")
-    log_message("If you hear audio describing a different test number than the filename, you've found a race condition")
+    log_message(
+        "To verify, listen to the audio files and check if they match the text files"
+    )
+    log_message(
+        "If you hear audio describing a different test number than the filename, you've found a race condition"
+    )
 
 
 def analyze_audio_files(output_dir):
@@ -226,30 +308,34 @@ def analyze_audio_files(output_dir):
     log_message(f"Found {len(wav_files)} WAV files and {len(txt_files)} TXT files")
 
     if len(wav_files) == 0:
-        log_message("No WAV files found! This indicates the TTS service requests may be failing.", is_error=True)
-        log_message("Check the connection to the TTS service and the response status codes above.", is_error=True)
+        log_message(
+            "No WAV files found! This indicates the TTS service requests may be failing.",
+            is_error=True,
+        )
+        log_message(
+            "Check the connection to the TTS service and the response status codes above.",
+            is_error=True,
+        )
 
     file_stats = []
     for wav_path in wav_files:
         try:
-            with wave.open(str(wav_path), 'rb') as wav_file:
+            with wave.open(str(wav_path), "rb") as wav_file:
                 frames = wav_file.getnframes()
                 rate = wav_file.getframerate()
                 duration = frames / rate
 
                 # Get corresponding text
-                text_path = wav_path.with_suffix('.txt')
+                text_path = wav_path.with_suffix(".txt")
                 if text_path.exists():
-                    with open(text_path, 'r') as text_file:
+                    with open(text_path, "r") as text_file:
                         text = text_file.read().strip()
                 else:
                     text = "N/A"
 
-                file_stats.append({
-                    'filename': wav_path.name,
-                    'duration': duration,
-                    'text': text
-                })
+                file_stats.append(
+                    {"filename": wav_path.name, "duration": duration, "text": text}
+                )
         except Exception as e:
             log_message(f"Error analyzing {wav_path}: {str(e)}", False, is_error=True)
 
@@ -259,12 +345,17 @@ def analyze_audio_files(output_dir):
         log_message(f"{'Filename':<20}{'Duration':<12}{'Text':<60}")
         log_message("-" * 92)
         for stat in file_stats:
-            log_message(f"{stat['filename']:<20}{stat['duration']:<12.2f}{stat['text'][:57]+'...' if len(stat['text']) > 60 else stat['text']:<60}")
+            log_message(
+                f"{stat['filename']:<20}{stat['duration']:<12.2f}{stat['text'][:57] + '...' if len(stat['text']) > 60 else stat['text']:<60}"
+            )
 
     # List missing WAV files where text files exist
     missing_wavs = set(p.stem for p in txt_files) - set(p.stem for p in wav_files)
     if missing_wavs:
-        log_message(f"\nFound {len(missing_wavs)} text files without corresponding WAV files:", is_error=True)
+        log_message(
+            f"\nFound {len(missing_wavs)} text files without corresponding WAV files:",
+            is_error=True,
+        )
         for stem in sorted(list(missing_wavs))[:10]:  # Limit to 10 for readability
             log_message(f"  - {stem}.txt (no WAV file)", is_error=True)
         if len(missing_wavs) > 10:
@@ -279,5 +370,9 @@ if __name__ == "__main__":
     log_message("\nNext Steps:")
     log_message("1. Listen to the generated audio files")
     log_message("2. Verify if each audio correctly says its ID number")
-    log_message("3. Check for any mismatches between the audio content and the text files")
-    log_message("4. If mismatches are found, you've successfully reproduced the race condition")
+    log_message(
+        "3. Check for any mismatches between the audio content and the text files"
+    )
+    log_message(
+        "4. If mismatches are found, you've successfully reproduced the race condition"
+    )

dev/Test copy.py

@@ -1,8 +1,10 @@
-import requests
 import base64
 import json
+
 import pydub
-text="""Delving into the Abyss: A Deeper Exploration of Meaning in 5 Seconds of Summer's "Jet Black Heart"
+import requests
+
+text = """Delving into the Abyss: A Deeper Exploration of Meaning in 5 Seconds of Summer's "Jet Black Heart"
 
 5 Seconds of Summer, initially perceived as purveyors of upbeat, radio-friendly pop-punk, embarked on a significant artistic evolution with their album Sounds Good Feels Good. Among its tracks, "Jet Black Heart" stands out as a powerful testament to this shift, moving beyond catchy melodies and embracing a darker, more emotionally complex sound. Released in 2015, the song transcends the typical themes of youthful exuberance and romantic angst, instead plunging into the depths of personal turmoil and the corrosive effects of inner darkness on interpersonal relationships. "Jet Black Heart" is not merely a song about heartbreak; it is a raw and vulnerable exploration of internal struggle, self-destructive patterns, and the precarious flicker of hope that persists even in the face of profound emotional chaos. Through potent metaphors, starkly honest lyrics, and a sonic landscape that mirrors its thematic weight, the song offers a profound meditation on the human condition, grappling with the shadows that reside within us all and their far-reaching consequences.
 
@@ -23,7 +25,7 @@ In conclusion, "Jet Black Heart" by 5 Seconds of Summer is far more than a typic
 5 Seconds of Summer, initially perceived as purveyors of upbeat, radio-friendly pop-punk, embarked on a significant artistic evolution with their album Sounds Good Feels Good. Among its tracks, "Jet Black Heart" stands out as a powerful testament to this shift, moving beyond catchy melodies and embracing a darker, more emotionally complex sound. Released in 2015, the song transcends the typical themes of youthful exuberance and romantic angst, instead plunging into the depths of personal turmoil and the corrosive effects of inner darkness on interpersonal relationships. "Jet Black Heart" is not merely a song about heartbreak; it is a raw and vulnerable exploration of internal struggle, self-destructive patterns, and the precarious flicker of hope that persists even in the face of profound emotional chaos."""
 
 
-Type="wav"
+Type = "wav"
 response = requests.post(
     "http://localhost:8880/dev/captioned_speech",
     json={
@@ -34,30 +36,34 @@ response = requests.post(
         "response_format": Type,
         "stream": True,
     },
-    stream=True
+    stream=True,
 )
 
-f=open(f"outputstream.{Type}","wb")
+f = open(f"outputstream.{Type}", "wb")
 for chunk in response.iter_lines(decode_unicode=True):
     if chunk:
-        temp_json=json.loads(chunk)
+        temp_json = json.loads(chunk)
         if temp_json["timestamps"] != []:
-            chunk_json=temp_json
+            chunk_json = temp_json
 
         # Decode base 64 stream to bytes
-        chunk_audio=base64.b64decode(temp_json["audio"].encode("utf-8"))
+        chunk_audio = base64.b64decode(temp_json["audio"].encode("utf-8"))
 
         # Process streaming chunks
         f.write(chunk_audio)
 
         # Print word level timestamps
-last_chunks={"start_time":chunk_json["timestamps"][-10]["start_time"],"end_time":chunk_json["timestamps"][-3]["end_time"],"word":" ".join([X["word"] for X in chunk_json["timestamps"][-10:-3]])}
+last_chunks = {
+    "start_time": chunk_json["timestamps"][-10]["start_time"],
+    "end_time": chunk_json["timestamps"][-3]["end_time"],
+    "word": " ".join([X["word"] for X in chunk_json["timestamps"][-10:-3]]),
+}
 
 print(f"CUTTING TO {last_chunks['word']}")
 
-audioseg=pydub.AudioSegment.from_file(f"outputstream.{Type}",format=Type)
-audioseg=audioseg[last_chunks["start_time"]*1000:last_chunks["end_time"] * 1000]
-audioseg.export(f"outputstreamcut.{Type}",format=Type)
+audioseg = pydub.AudioSegment.from_file(f"outputstream.{Type}", format=Type)
+audioseg = audioseg[last_chunks["start_time"] * 1000 : last_chunks["end_time"] * 1000]
+audioseg.export(f"outputstreamcut.{Type}", format=Type)
 
 
 """

dev/Test money.py

@@ -1,13 +1,14 @@
-import requests
 import base64
 import json
 
-text="""the administration has offered up a platter of repression for more than a year and is still slated to lose $400 million.
+import requests
+
+text = """the administration has offered up a platter of repression for more than a year and is still slated to lose $400 million.
 
 Columbia is the largest private landowner in New York City and boasts an endowment of $14.8 billion;"""
 
 
-Type="wav"
+Type = "wav"
 
 response = requests.post(
     "http://localhost:8880/v1/audio/speech",
@@ -19,7 +20,7 @@ response = requests.post(
         "response_format": Type,
         "stream": False,
     },
-    stream=True
+    stream=True,
 )
 
 with open(f"outputnostreammoney.{Type}", "wb") as f:

dev/Test num.py

@@ -1,6 +1,7 @@
-from text_to_num import text2num
 import re
+
 import inflect
+from text_to_num import text2num
 from torch import mul
 
 INFLECT_ENGINE = inflect.engine()
@@ -11,6 +12,7 @@ def conditional_int(number: float, threshold: float = 0.00001):
         return int(round(number))
     return number
 
+
 def handle_money(m: re.Match[str]) -> str:
     """Convert money expressions to spoken form"""
 

dev/Test.py

@@ -1,8 +1,9 @@
-import requests
 import base64
 import json
 
-text="""Delving into the Abyss: A Deeper Exploration of Meaning in 5 Seconds of Summer's "Jet Black Heart"
+import requests
+
+text = """Delving into the Abyss: A Deeper Exploration of Meaning in 5 Seconds of Summer's "Jet Black Heart"
 
 5 Seconds of Summer, initially perceived as purveyors of upbeat, radio-friendly pop-punk, embarked on a significant artistic evolution with their album Sounds Good Feels Good. Among its tracks, "Jet Black Heart" stands out as a powerful testament to this shift, moving beyond catchy melodies and embracing a darker, more emotionally complex sound. Released in 2015, the song transcends the typical themes of youthful exuberance and romantic angst, instead plunging into the depths of personal turmoil and the corrosive effects of inner darkness on interpersonal relationships. "Jet Black Heart" is not merely a song about heartbreak; it is a raw and vulnerable exploration of internal struggle, self-destructive patterns, and the precarious flicker of hope that persists even in the face of profound emotional chaos. Through potent metaphors, starkly honest lyrics, and a sonic landscape that mirrors its thematic weight, the song offers a profound meditation on the human condition, grappling with the shadows that reside within us all and their far-reaching consequences.
 
@@ -18,12 +19,12 @@ Beyond the lyrical content, the musical elements of "Jet Black Heart" contribute
 
 In conclusion, "Jet Black Heart" by 5 Seconds of Summer is far more than a typical pop song; it is a poignant and deeply resonant exploration of inner darkness, self-destructive tendencies, and the fragile yet persistent hope for human connection and redemption. Through its powerful central metaphor of the "jet black heart," its unflinching portrayal of internal turmoil, and its subtle yet potent message of vulnerability and potential transformation, the song resonates with anyone who has grappled with their own inner demons and the complexities of human relationships. It is a reminder that even in the deepest darkness, a flicker of hope can endure, and that true healing and connection often emerge from the courageous act of confronting and sharing our most vulnerable selves. "Jet Black Heart" stands as a testament to 5 Seconds of Summer's artistic growth, showcasing their capacity to delve into profound emotional territories and create music that is not only catchy and engaging but also deeply meaningful and emotionally resonant, solidifying their position as a band capable of capturing the complexities of the human experience."""
 
-text="""Delving into the Abyss: A Deeper Exploration of Meaning in 5 Seconds of Summer's "Jet Black Heart"
+text = """Delving into the Abyss: A Deeper Exploration of Meaning in 5 Seconds of Summer's "Jet Black Heart"
 
 5 Seconds of Summer, initially perceived as purveyors of upbeat, radio-friendly pop-punk, embarked on a significant artistic evolution with their album Sounds Good Feels Good. Among its tracks, "Jet Black Heart" stands out as a powerful testament to this shift, moving beyond catchy melodies and embracing a darker, more emotionally complex sound. Released in 2015, the song transcends the typical themes of youthful exuberance and romantic angst, instead plunging into the depths of personal turmoil and the corrosive effects of inner darkness on interpersonal relationships. "Jet Black Heart" is not merely a song about heartbreak; it is a raw and vulnerable exploration of internal struggle, self-destructive patterns, and the precarious flicker of hope that persists even in the face of profound emotional chaos."""
 
 
-Type="wav"
+Type = "wav"
 
 
 response = requests.post(
@@ -36,11 +37,11 @@ response = requests.post(
         "response_format": Type,
         "stream": True,
     },
-    stream=True
+    stream=True,
 )
 
 
-f=open(f"outputstream.{Type}","wb")
+f = open(f"outputstream.{Type}", "wb")
 for chunk in response.iter_content():
     if chunk:
         # Process streaming chunks
@@ -56,7 +57,7 @@ response = requests.post(
         "response_format": Type,
         "stream": False,
     },
-    stream=True
+    stream=True,
 )
 
 with open(f"outputnostream.{Type}", "wb") as f:

docker/cpu/docker-compose.yml

@@ -20,7 +20,7 @@ services:
       
   # # Gradio UI service [Comment out everything below if you don't need it]
   # gradio-ui:
-  #   image: ghcr.io/remsky/kokoro-fastapi-ui:v0.2.0
+  #   image: ghcr.io/remsky/kokoro-fastapi-ui:v${VERSION}
   #   # Uncomment below (and comment out above) to build from source instead of using the released image
   #   build:
   #     context: ../../ui

docker/gpu/docker-compose.yml

@@ -1,7 +1,7 @@
 name: kokoro-tts-gpu
 services:
   kokoro-tts:
-    # image: ghcr.io/remsky/kokoro-fastapi-gpu:v0.2.0
+    # image: ghcr.io/remsky/kokoro-fastapi-gpu:v${VERSION}
     build:
       context: ../..
       dockerfile: docker/gpu/Dockerfile
@@ -24,7 +24,7 @@ services:
 
   # # Gradio UI service
   # gradio-ui:
-  #   image: ghcr.io/remsky/kokoro-fastapi-ui:v0.2.0
+  #   image: ghcr.io/remsky/kokoro-fastapi-ui:v${VERSION}
   #   # Uncomment below to build from source instead of using the released image
   #   # build:
   #     # context: ../../ui

docker/scripts/download_model.py

@@ -91,9 +91,7 @@ def main():
 
     parser = argparse.ArgumentParser(description="Download Kokoro v1.0 model")
     parser.add_argument(
-        "--output",
-        required=True,
-        help="Output directory for model files"
+        "--output", required=True, help="Output directory for model files"
     )
 
     args = parser.parse_args()

examples/stream_tts_playback.py

@@ -123,7 +123,7 @@ def main():
     with open(wells_path, "r", encoding="utf-8") as f:
         full_text = f.read()
         # Take first few paragraphs
-        text = " ".join(full_text.split("\n\n")[:2])
+        text = " ".join(full_text.split("\n\n")[1:3])
 
     print("\nStarting TTS stream playback...")
     print(f"Text length: {len(text)} characters")

pyproject.toml

@@ -1,6 +1,6 @@
 [project]
 name = "kokoro-fastapi"
-version = "0.1.4"
+version = "0.3.0"
 description = "FastAPI TTS Service"
 readme = "README.md"
 requires-python = ">=3.10"
@@ -31,10 +31,11 @@ dependencies = [
     "matplotlib>=3.10.0",
     "mutagen>=1.47.0",
     "psutil>=6.1.1",
-    "kokoro @ git+https://github.com/hexgrad/kokoro.git@31a2b6337b8c1b1418ef68c48142328f640da938",
-    'misaki[en,ja,ko,zh]',
-    "spacy==3.7.2",
-    "en-core-web-sm @ https://github.com/explosion/spacy-models/releases/download/en_core_web_sm-3.7.1/en_core_web_sm-3.7.1-py3-none-any.whl",
+    "espeakng-loader==0.2.4",
+    "kokoro==0.9.2",
+    "misaki[en,ja,ko,zh]==0.9.3",
+    "spacy==3.8.5",
+    "en-core-web-sm @ https://github.com/explosion/spacy-models/releases/download/en_core_web_sm-3.8.0/en_core_web_sm-3.8.0-py3-none-any.whl",
     "inflect>=7.5.0",
     "phonemizer-fork>=3.3.2",
     "av>=14.2.0",
@@ -57,8 +58,8 @@ test = [
     "pytest-cov==6.0.0",
     "httpx==0.26.0",
     "pytest-asyncio==0.25.3",
-    "openai>=1.59.6",
     "tomli>=2.0.1",
+    "jinja2>=3.1.6"
 ]
 
 [tool.uv]

scripts/fix_misaki.py

@@ -0,0 +1,46 @@
+"""
+Patch for misaki package to fix the EspeakWrapper.set_data_path issue.
+"""
+
+import importlib.util
+import os
+import sys
+
+# Find the misaki package
+try:
+    import misaki
+
+    misaki_path = os.path.dirname(misaki.__file__)
+    print(f"Found misaki package at: {misaki_path}")
+except ImportError:
+    print("Misaki package not found. Make sure it's installed.")
+    sys.exit(1)
+
+# Path to the espeak.py file
+espeak_file = os.path.join(misaki_path, "espeak.py")
+
+if not os.path.exists(espeak_file):
+    print(f"Could not find {espeak_file}")
+    sys.exit(1)
+
+# Read the current content
+with open(espeak_file, "r") as f:
+    content = f.read()
+
+# Check if the problematic line exists
+if "EspeakWrapper.set_data_path(espeakng_loader.get_data_path())" in content:
+    # Replace the problematic line
+    new_content = content.replace(
+        "EspeakWrapper.set_data_path(espeakng_loader.get_data_path())",
+        "# Fixed line to use data_path attribute instead of set_data_path method\n"
+        "EspeakWrapper.data_path = espeakng_loader.get_data_path()",
+    )
+
+    # Write the modified content back
+    with open(espeak_file, "w") as f:
+        f.write(new_content)
+
+    print(f"Successfully patched {espeak_file}")
+else:
+    print(f"The problematic line was not found in {espeak_file}")
+    print("The file may have already been patched or the issue is different.")

scripts/update_badges.py

@@ -1,54 +1,51 @@
 import re
 import subprocess
-import tomli
 from pathlib import Path
 
+import tomli
+
+
 def extract_dependency_info():
-    """Extract version and commit hash for kokoro and misaki from pyproject.toml"""
+    """Extract version for kokoro and misaki from pyproject.toml"""
     with open("pyproject.toml", "rb") as f:
         pyproject = tomli.load(f)
 
     deps = pyproject["project"]["dependencies"]
     info = {}
+    kokoro_found = False
+    misaki_found = False
 
-    # Extract kokoro info
     for dep in deps:
-        if dep.startswith("kokoro @"):
-            # Extract version from the dependency string if available
-            version_match = re.search(r"kokoro @ git\+https://github\.com/hexgrad/kokoro\.git@", dep)
-            if version_match:
-                # If no explicit version, use v0.7.9 as shown in the README
-                version = "v0.7.9"
-            commit_match = re.search(r"@([a-f0-9]{7})", dep)
-            if commit_match:
-                info["kokoro"] = {
-                    "version": version,
-                    "commit": commit_match.group(1)
-                }
-        elif dep.startswith("misaki["):
-            # Extract version from the dependency string if available
-            version_match = re.search(r"misaki\[.*?\] @ git\+https://github\.com/hexgrad/misaki\.git@", dep)
-            if version_match:
-                # If no explicit version, use v0.7.9 as shown in the README
-                version = "v0.7.9"
-            commit_match = re.search(r"@([a-f0-9]{7})", dep)
-            if commit_match:
-                info["misaki"] = {
-                    "version": version,
-                    "commit": commit_match.group(1)
-                }
+        # Match kokoro==version
+        kokoro_match = re.match(r"^kokoro==(.+)$", dep)
+        if kokoro_match:
+            info["kokoro"] = {"version": kokoro_match.group(1)}
+            kokoro_found = True
+
+        # Match misaki[...] ==version or misaki==version
+        misaki_match = re.match(r"^misaki(?:\[.*?\])?==(.+)$", dep)
+        if misaki_match:
+            info["misaki"] = {"version": misaki_match.group(1)}
+            misaki_found = True
+
+        # Stop if both found
+        if kokoro_found and misaki_found:
+            break
+
+    if not kokoro_found:
+        raise ValueError("Kokoro version not found in pyproject.toml dependencies")
+    if not misaki_found:
+        raise ValueError("Misaki version not found in pyproject.toml dependencies")
 
     return info
 
+
 def run_pytest_with_coverage():
     """Run pytest with coverage and return the results"""
     try:
         # Run pytest with coverage
         result = subprocess.run(
-            ["pytest", "--cov=api", "-v"],
-            capture_output=True,
-            text=True,
-            check=True
+            ["pytest", "--cov=api", "-v"], capture_output=True, text=True, check=True
         )
 
         # Extract test results
@@ -57,10 +54,7 @@ def run_pytest_with_coverage():
 
         # Extract coverage from .coverage file
         coverage_output = subprocess.run(
-            ["coverage", "report"],
-            capture_output=True,
-            text=True,
-            check=True
+            ["coverage", "report"], capture_output=True, text=True, check=True
         ).stdout
 
         # Extract total coverage percentage
@@ -73,6 +67,7 @@ def run_pytest_with_coverage():
         print(f"Output: {e.output}")
         return 0, "0"
 
+
 def update_readme_badges(passed_tests, coverage_percentage, dep_info):
     """Update the badges in the README file"""
     readme_path = Path("README.md")
@@ -84,37 +79,42 @@ def update_readme_badges(passed_tests, coverage_percentage, dep_info):
 
     # Update tests badge
     content = re.sub(
-        r'!\[Tests\]\(https://img\.shields\.io/badge/tests-\d+%20passed-[a-zA-Z]+\)',
-        f'![Tests](https://img.shields.io/badge/tests-{passed_tests}%20passed-darkgreen)',
-        content
+        r"!\[Tests\]\(https://img\.shields\.io/badge/tests-\d+%20passed-[a-zA-Z]+\)",
+        f"![Tests](https://img.shields.io/badge/tests-{passed_tests}%20passed-darkgreen)",
+        content,
     )
 
     # Update coverage badge
     content = re.sub(
-        r'!\[Coverage\]\(https://img\.shields\.io/badge/coverage-\d+%25-[a-zA-Z]+\)',
-        f'![Coverage](https://img.shields.io/badge/coverage-{coverage_percentage}%25-tan)',
-        content
+        r"!\[Coverage\]\(https://img\.shields\.io/badge/coverage-\d+%25-[a-zA-Z]+\)",
+        f"![Coverage](https://img.shields.io/badge/coverage-{coverage_percentage}%25-tan)",
+        content,
     )
 
     # Update kokoro badge
     if "kokoro" in dep_info:
+        # Find badge like kokoro-v0.9.2::abcdefg-BB5420 or kokoro-v0.9.2-BB5420
+        kokoro_version = dep_info["kokoro"]["version"]
         content = re.sub(
-            r'!\[Kokoro\]\(https://img\.shields\.io/badge/kokoro-[^)]+\)',
-            f'![Kokoro](https://img.shields.io/badge/kokoro-{dep_info["kokoro"]["version"]}::{dep_info["kokoro"]["commit"]}-BB5420)',
-            content
+            r"(!\[Kokoro\]\(https://img\.shields\.io/badge/kokoro-)[^)-]+(-BB5420\))",
+            lambda m: f"{m.group(1)}{kokoro_version}{m.group(2)}",
+            content,
         )
 
     # Update misaki badge
     if "misaki" in dep_info:
+        # Find badge like misaki-v0.9.3::abcdefg-B8860B or misaki-v0.9.3-B8860B
+        misaki_version = dep_info["misaki"]["version"]
         content = re.sub(
-            r'!\[Misaki\]\(https://img\.shields\.io/badge/misaki-[^)]+\)',
-            f'![Misaki](https://img.shields.io/badge/misaki-{dep_info["misaki"]["version"]}::{dep_info["misaki"]["commit"]}-B8860B)',
-            content
+            r"(!\[Misaki\]\(https://img\.shields\.io/badge/misaki-)[^)-]+(-B8860B\))",
+            lambda m: f"{m.group(1)}{misaki_version}{m.group(2)}",
+            content,
         )
 
     readme_path.write_text(content)
     return True
 
+
 def main():
     # Get dependency info
     dep_info = extract_dependency_info()
@@ -128,11 +128,12 @@ def main():
         print(f"- Tests: {passed_tests} passed")
         print(f"- Coverage: {coverage_percentage}%")
         if "kokoro" in dep_info:
-            print(f"- Kokoro: {dep_info['kokoro']['version']}::{dep_info['kokoro']['commit']}")
+            print(f"- Kokoro: {dep_info['kokoro']['version']}")
         if "misaki" in dep_info:
-            print(f"- Misaki: {dep_info['misaki']['version']}::{dep_info['misaki']['commit']}")
+            print(f"- Misaki: {dep_info['misaki']['version']}")
     else:
         print("Failed to update badges")
 
+
 if __name__ == "__main__":
     main()

scripts/update_version.py

@@ -0,0 +1,234 @@
+#!/usr/bin/env python3
+"""
+Version Update Script
+
+This script reads the version from the VERSION file and updates references
+in pyproject.toml, the Helm chart, and README.md.
+"""
+
+import re
+from pathlib import Path
+
+import yaml
+
+# Get the project root directory
+ROOT_DIR = Path(__file__).parent.parent
+
+# --- Configuration ---
+VERSION_FILE = ROOT_DIR / "VERSION"
+PYPROJECT_FILE = ROOT_DIR / "pyproject.toml"
+HELM_CHART_FILE = ROOT_DIR / "charts" / "kokoro-fastapi" / "Chart.yaml"
+README_FILE = ROOT_DIR / "README.md"
+# --- End Configuration ---
+
+
+def update_pyproject(version: str):
+    """Updates the version in pyproject.toml"""
+    if not PYPROJECT_FILE.exists():
+        print(f"Skipping: {PYPROJECT_FILE} not found.")
+        return
+
+    try:
+        content = PYPROJECT_FILE.read_text()
+        # Regex to find and capture current version = "X.Y.Z" under [project]
+        pattern = r'(^\[project\]\s*(?:.*\s)*?version\s*=\s*)"([^"]+)"'
+        match = re.search(pattern, content, flags=re.MULTILINE)
+
+        if not match:
+            print(f"Warning: Version pattern not found in {PYPROJECT_FILE}")
+            return
+
+        current_version = match.group(2)
+        if current_version == version:
+            print(f"Already up-to-date: {PYPROJECT_FILE} (version {version})")
+        else:
+            # Perform replacement
+            new_content = re.sub(
+                pattern, rf'\1"{version}"', content, count=1, flags=re.MULTILINE
+            )
+            PYPROJECT_FILE.write_text(new_content)
+            print(f"Updated {PYPROJECT_FILE} from {current_version} to {version}")
+
+    except Exception as e:
+        print(f"Error processing {PYPROJECT_FILE}: {e}")
+
+
+def update_helm_chart(version: str):
+    """Updates the version and appVersion in the Helm chart"""
+    if not HELM_CHART_FILE.exists():
+        print(f"Skipping: {HELM_CHART_FILE} not found.")
+        return
+
+    try:
+        content = HELM_CHART_FILE.read_text()
+        original_content = content
+        updated_count = 0
+
+        # Update 'version:' line (unquoted)
+        # Looks for 'version:' followed by optional whitespace and the version number
+        version_pattern = r"^(version:\s*)(\S+)"
+        current_version_match = re.search(version_pattern, content, flags=re.MULTILINE)
+        if current_version_match and current_version_match.group(2) != version:
+            content = re.sub(
+                version_pattern,
+                rf"\g<1>{version}",
+                content,
+                count=1,
+                flags=re.MULTILINE,
+            )
+            print(
+                f"Updating 'version' in {HELM_CHART_FILE} from {current_version_match.group(2)} to {version}"
+            )
+            updated_count += 1
+        elif current_version_match:
+            print(f"Already up-to-date: 'version' in {HELM_CHART_FILE} is {version}")
+        else:
+            print(f"Warning: 'version:' pattern not found in {HELM_CHART_FILE}")
+
+        # Update 'appVersion:' line (quoted or unquoted)
+        # Looks for 'appVersion:' followed by optional whitespace, optional quote, the version, optional quote
+        app_version_pattern = r"^(appVersion:\s*)(\"?)([^\"\s]+)(\"?)"
+        current_app_version_match = re.search(
+            app_version_pattern, content, flags=re.MULTILINE
+        )
+
+        if current_app_version_match:
+            leading_whitespace = current_app_version_match.group(
+                1
+            )  # e.g., "appVersion: "
+            opening_quote = current_app_version_match.group(2)  # e.g., '"' or ''
+            current_app_ver = current_app_version_match.group(3)  # e.g., '0.2.0'
+            closing_quote = current_app_version_match.group(4)  # e.g., '"' or ''
+
+            # Check if quotes were consistent (both present or both absent)
+            if opening_quote != closing_quote:
+                print(
+                    f"Warning: Inconsistent quotes found for appVersion in {HELM_CHART_FILE}. Skipping update for this line."
+                )
+            elif (
+                current_app_ver == version and opening_quote == '"'
+            ):  # Check if already correct *and* quoted
+                print(
+                    f"Already up-to-date: 'appVersion' in {HELM_CHART_FILE} is \"{version}\""
+                )
+            else:
+                # Always replace with the quoted version
+                replacement = f'{leading_whitespace}"{version}"'  # Ensure quotes
+                original_display = f"{opening_quote}{current_app_ver}{closing_quote}"  # How it looked before
+                target_display = f'"{version}"'  # How it should look
+
+                # Only report update if the displayed value actually changes
+                if original_display != target_display:
+                    content = re.sub(
+                        app_version_pattern,
+                        replacement,
+                        content,
+                        count=1,
+                        flags=re.MULTILINE,
+                    )
+                    print(
+                        f"Updating 'appVersion' in {HELM_CHART_FILE} from {original_display} to {target_display}"
+                    )
+                    updated_count += 1
+                else:
+                    # It matches the target version but might need quoting fixed silently if we didn't update
+                    # Or it was already correct. Check if content changed. If not, report up-to-date.
+                    if not (
+                        content != original_content and updated_count > 0
+                    ):  # Avoid double message if version also changed
+                        print(
+                            f"Already up-to-date: 'appVersion' in {HELM_CHART_FILE} is {target_display}"
+                        )
+
+        else:
+            print(f"Warning: 'appVersion:' pattern not found in {HELM_CHART_FILE}")
+
+        # Write back only if changes were made
+        if content != original_content:
+            HELM_CHART_FILE.write_text(content)
+            # Confirmation message printed above during the specific update
+        elif updated_count == 0 and current_version_match and current_app_version_match:
+            # If no updates were made but patterns were found, confirm it's up-to-date overall
+            print(f"Already up-to-date: {HELM_CHART_FILE} (version {version})")
+
+    except Exception as e:
+        print(f"Error processing {HELM_CHART_FILE}: {e}")
+
+
+def update_readme(version_with_v: str):
+    """Updates Docker image tags in README.md"""
+    if not README_FILE.exists():
+        print(f"Skipping: {README_FILE} not found.")
+        return
+
+    try:
+        content = README_FILE.read_text()
+        # Regex to find and capture current ghcr.io/.../kokoro-fastapi-(cpu|gpu):vX.Y.Z
+        pattern = r"(ghcr\.io/remsky/kokoro-fastapi-(?:cpu|gpu)):(v\d+\.\d+\.\d+)"
+        matches = list(re.finditer(pattern, content))  # Find all occurrences
+
+        if not matches:
+            print(f"Warning: Docker image tag pattern not found in {README_FILE}")
+        else:
+            updated_needed = False
+            for match in matches:
+                current_tag = match.group(2)
+                if current_tag != version_with_v:
+                    updated_needed = True
+                    break  # Only need one mismatch to trigger update
+
+            if updated_needed:
+                # Perform replacement on all occurrences
+                new_content = re.sub(pattern, rf"\1:{version_with_v}", content)
+                README_FILE.write_text(new_content)
+                print(f"Updated Docker image tags in {README_FILE} to {version_with_v}")
+            else:
+                print(
+                    f"Already up-to-date: Docker image tags in {README_FILE} (version {version_with_v})"
+                )
+
+        # Check for ':latest' tag usage remains the same
+        if ":latest" in content:
+            print(
+                f"Warning: Found ':latest' tag in {README_FILE}. Consider updating manually if needed."
+            )
+
+    except Exception as e:
+        print(f"Error processing {README_FILE}: {e}")
+
+
+def main():
+    # Read the version from the VERSION file
+    if not VERSION_FILE.exists():
+        print(f"Error: {VERSION_FILE} not found.")
+        return
+
+    try:
+        version = VERSION_FILE.read_text().strip()
+        if not re.match(r"^\d+\.\d+\.\d+$", version):
+            print(
+                f"Error: Invalid version format '{version}' in {VERSION_FILE}. Expected X.Y.Z"
+            )
+            return
+    except Exception as e:
+        print(f"Error reading {VERSION_FILE}: {e}")
+        return
+
+    print(f"Read version: {version} from {VERSION_FILE}")
+    print("-" * 20)
+
+    # Prepare versions (with and without 'v')
+    version_plain = version
+    version_with_v = f"v{version}"
+
+    # Update files
+    update_pyproject(version_plain)
+    update_helm_chart(version_plain)
+    update_readme(version_with_v)
+
+    print("-" * 20)
+    print("Version update script finished.")
+
+
+if __name__ == "__main__":
+    main()

slim.report.json

@@ -1,49 +0,0 @@
-{
-  "document": "doc.report.command",
-  "version": "ov/command/slim/1.1",
-  "engine": "linux/amd64|ALP|x.1.42.2|29e62e7836de7b1004607c51c502537ffe1969f0|2025-01-16_07:48:54AM|x",
-  "containerized": false,
-  "host_distro": {
-    "name": "Ubuntu",
-    "version": "22.04",
-    "display_name": "Ubuntu 22.04.5 LTS"
-  },
-  "type": "slim",
-  "state": "error",
-  "target_reference": "kokoro-fastapi:latest",
-  "system": {
-    "type": "",
-    "release": "",
-    "distro": {
-      "name": "",
-      "version": "",
-      "display_name": ""
-    }
-  },
-  "source_image": {
-    "identity": {
-      "id": ""
-    },
-    "size": 0,
-    "size_human": "",
-    "create_time": "",
-    "architecture": "",
-    "container_entry": {
-      "exe_path": ""
-    }
-  },
-  "minified_image_size": 0,
-  "minified_image_size_human": "",
-  "minified_image": "",
-  "minified_image_id": "",
-  "minified_image_digest": "",
-  "minified_image_has_data": false,
-  "minified_by": 0,
-  "artifact_location": "",
-  "container_report_name": "",
-  "seccomp_profile_name": "",
-  "apparmor_profile_name": "",
-  "image_stack": null,
-  "image_created": false,
-  "image_build_engine": ""
-}

start-cpu.sh

@@ -10,9 +10,17 @@ export PYTHONPATH=$PROJECT_ROOT:$PROJECT_ROOT/api
 export MODEL_DIR=src/models
 export VOICES_DIR=src/voices/v1_0
 export WEB_PLAYER_PATH=$PROJECT_ROOT/web
+# Set the espeak-ng data path to your location
+export ESPEAK_DATA_PATH=/usr/lib/x86_64-linux-gnu/espeak-ng-data
 
 # Run FastAPI with CPU extras using uv run
 # Note: espeak may still require manual installation,
 uv pip install -e ".[cpu]"
 uv run --no-sync python docker/scripts/download_model.py --output api/src/models/v1_0
+
+# Apply the misaki patch to fix possible EspeakWrapper issue in older versions
+# echo "Applying misaki patch..."
+# python scripts/fix_misaki.py
+
+# Start the server
 uv run --no-sync uvicorn api.src.main:app --host 0.0.0.0 --port 8880

start-gpu_mac.sh

@@ -3,13 +3,6 @@
 # Get project root directory
 PROJECT_ROOT=$(pwd)
 
-# Create mps-specific venv directory
-VENV_DIR="$PROJECT_ROOT/.venv-mps"
-if [ ! -d "$VENV_DIR" ]; then
-    echo "Creating MPS-specific virtual environment..."
-    python3 -m venv "$VENV_DIR"
-fi
-
 # Set other environment variables
 export USE_GPU=true
 export USE_ONNX=false
@@ -18,18 +11,11 @@ export MODEL_DIR=src/models
 export VOICES_DIR=src/voices/v1_0
 export WEB_PLAYER_PATH=$PROJECT_ROOT/web
 
-# Set environment variables
-export USE_GPU=true
-export USE_ONNX=false
-export PYTHONPATH=$PROJECT_ROOT:$PROJECT_ROOT/api
-export MODEL_DIR=src/models
-export VOICES_DIR=src/voices/v1_0
-export WEB_PLAYER_PATH=$PROJECT_ROOT/web
-
 export DEVICE_TYPE=mps
 # Enable MPS fallback for unsupported operations
 export PYTORCH_ENABLE_MPS_FALLBACK=1
 
 # Run FastAPI with GPU extras using uv run
 uv pip install -e .
+uv run --no-sync python docker/scripts/download_model.py --output api/src/models/v1_0
 uv run --no-sync uvicorn api.src.main:app --host 0.0.0.0 --port 8880

ui/depr_tests/conftest.py

@@ -1,6 +1,7 @@
-import pytest
 from unittest.mock import AsyncMock, Mock
 
+import pytest
+
 from api.src.services.tts_service import TTSService
 
 
@@ -30,8 +31,11 @@ async def mock_tts_service(mock_model_manager, mock_voice_manager):
 
 
 @pytest.fixture(autouse=True)
-async def setup_mocks(monkeypatch, mock_model_manager, mock_voice_manager, mock_tts_service):
+async def setup_mocks(
+    monkeypatch, mock_model_manager, mock_voice_manager, mock_tts_service
+):
     """Setup global mocks for UI tests"""
+
     async def mock_get_model():
         return mock_model_manager
 
@@ -43,4 +47,6 @@ async def setup_mocks(monkeypatch, mock_model_manager, mock_voice_manager, mock_
 
     monkeypatch.setattr("api.src.inference.model_manager.get_manager", mock_get_model)
     monkeypatch.setattr("api.src.inference.voice_manager.get_manager", mock_get_voice)
-    monkeypatch.setattr("api.src.services.tts_service.TTSService.create", mock_create_service)
+    monkeypatch.setattr(
+        "api.src.services.tts_service.TTSService.create", mock_create_service
+    )

ui/depr_tests/test_api.py

@@ -1,4 +1,4 @@
-from unittest.mock import patch, mock_open
+from unittest.mock import mock_open, patch
 
 import pytest
 import requests
@@ -59,9 +59,11 @@ def test_check_api_status_connection_error():
 
 def test_text_to_speech_success(mock_response, tmp_path):
     """Test successful speech generation"""
-    with patch("requests.post", return_value=mock_response({})), patch(
-        "ui.lib.api.OUTPUTS_DIR", str(tmp_path)
-    ), patch("builtins.open", mock_open()) as mock_file:
+    with (
+        patch("requests.post", return_value=mock_response({})),
+        patch("ui.lib.api.OUTPUTS_DIR", str(tmp_path)),
+        patch("builtins.open", mock_open()) as mock_file,
+    ):
         result = api.text_to_speech("test text", "voice1", "mp3", 1.0)
 
         assert result is not None
@@ -116,9 +118,11 @@ def test_text_to_speech_api_params(mock_response, tmp_path):
     ]
 
     for input_voice, expected_voice in test_cases:
-        with patch("requests.post") as mock_post, patch(
-            "ui.lib.api.OUTPUTS_DIR", str(tmp_path)
-        ), patch("builtins.open", mock_open()):
+        with (
+            patch("requests.post") as mock_post,
+            patch("ui.lib.api.OUTPUTS_DIR", str(tmp_path)),
+            patch("builtins.open", mock_open()),
+        ):
             mock_post.return_value = mock_response({})
             api.text_to_speech("test text", input_voice, "mp3", 1.5)
 
@@ -149,11 +153,15 @@ def test_text_to_speech_output_filename(mock_response, tmp_path):
     ]
 
     for input_voice, filename_check in test_cases:
-        with patch("requests.post", return_value=mock_response({})), patch(
-            "ui.lib.api.OUTPUTS_DIR", str(tmp_path)
-        ), patch("builtins.open", mock_open()) as mock_file:
+        with (
+            patch("requests.post", return_value=mock_response({})),
+            patch("ui.lib.api.OUTPUTS_DIR", str(tmp_path)),
+            patch("builtins.open", mock_open()) as mock_file,
+        ):
             result = api.text_to_speech("test text", input_voice, "mp3", 1.0)
 
             assert result is not None
-            assert filename_check(result), f"Expected voice pattern not found in filename: {result}"
+            assert filename_check(result), (
+                f"Expected voice pattern not found in filename: {result}"
+            )
             mock_file.assert_called_once()

ui/depr_tests/test_components.py

@@ -1,9 +1,9 @@
 import gradio as gr
 import pytest
 
-from ui.lib.config import AUDIO_FORMATS
 from ui.lib.components.model import create_model_column
 from ui.lib.components.output import create_output_column
+from ui.lib.config import AUDIO_FORMATS
 
 
 def test_create_model_column_structure():

ui/depr_tests/test_files.py

@@ -15,8 +15,9 @@ def mock_dirs(tmp_path):
     inputs_dir.mkdir()
     outputs_dir.mkdir()
 
-    with patch("ui.lib.files.INPUTS_DIR", str(inputs_dir)), patch(
-        "ui.lib.files.OUTPUTS_DIR", str(outputs_dir)
+    with (
+        patch("ui.lib.files.INPUTS_DIR", str(inputs_dir)),
+        patch("ui.lib.files.OUTPUTS_DIR", str(outputs_dir)),
     ):
         yield inputs_dir, outputs_dir
 

ui/depr_tests/test_interface.py

@@ -62,8 +62,9 @@ def test_interface_html_links():
 def test_update_status_available(mock_timer):
     """Test status update when service is available"""
     voices = ["voice1", "voice2"]
-    with patch("ui.lib.api.check_api_status", return_value=(True, voices)), patch(
-        "gradio.Timer", return_value=mock_timer
+    with (
+        patch("ui.lib.api.check_api_status", return_value=(True, voices)),
+        patch("gradio.Timer", return_value=mock_timer),
     ):
         demo = create_interface()
 
@@ -81,8 +82,9 @@ def test_update_status_available(mock_timer):
 
 def test_update_status_unavailable(mock_timer):
     """Test status update when service is unavailable"""
-    with patch("ui.lib.api.check_api_status", return_value=(False, [])), patch(
-        "gradio.Timer", return_value=mock_timer
+    with (
+        patch("ui.lib.api.check_api_status", return_value=(False, [])),
+        patch("gradio.Timer", return_value=mock_timer),
     ):
         demo = create_interface()
         update_fn = mock_timer.events[0].fn
@@ -97,9 +99,10 @@ def test_update_status_unavailable(mock_timer):
 
 def test_update_status_error(mock_timer):
     """Test status update when an error occurs"""
-    with patch(
-        "ui.lib.api.check_api_status", side_effect=Exception("Test error")
-    ), patch("gradio.Timer", return_value=mock_timer):
+    with (
+        patch("ui.lib.api.check_api_status", side_effect=Exception("Test error")),
+        patch("gradio.Timer", return_value=mock_timer),
+    ):
         demo = create_interface()
         update_fn = mock_timer.events[0].fn
 
@@ -113,8 +116,9 @@ def test_update_status_error(mock_timer):
 
 def test_timer_configuration(mock_timer):
     """Test timer configuration"""
-    with patch("ui.lib.api.check_api_status", return_value=(False, [])), patch(
-        "gradio.Timer", return_value=mock_timer
+    with (
+        patch("ui.lib.api.check_api_status", return_value=(False, [])),
+        patch("gradio.Timer", return_value=mock_timer),
     ):
         demo = create_interface()
 

ui/lib/api.py

@@ -1,6 +1,6 @@
-import os
 import datetime
-from typing import List, Tuple, Optional
+import os
+from typing import List, Optional, Tuple
 
 import requests
 

ui/lib/components/input.py

@@ -13,9 +13,7 @@ def create_input_column(disable_local_saving: bool = False) -> Tuple[gr.Column,
         )
 
         # Always show file upload but handle differently based on disable_local_saving
-        file_upload = gr.File(
-            label="Upload Text File (.txt)", file_types=[".txt"]
-        )
+        file_upload = gr.File(label="Upload Text File (.txt)", file_types=[".txt"])
 
         if not disable_local_saving:
             # Show full interface with tabs when saving is enabled
@@ -24,7 +22,9 @@ def create_input_column(disable_local_saving: bool = False) -> Tuple[gr.Column,
                 tabs.selected = 0
                 # Direct Input Tab
                 with gr.TabItem("Direct Input"):
-                    text_submit_direct = gr.Button("Generate Speech", variant="primary", size="lg")
+                    text_submit_direct = gr.Button(
+                        "Generate Speech", variant="primary", size="lg"
+                    )
 
                 # File Input Tab
                 with gr.TabItem("From File"):
@@ -48,7 +48,9 @@ def create_input_column(disable_local_saving: bool = False) -> Tuple[gr.Column,
                         )
         else:
             # Just show the generate button when saving is disabled
-            text_submit_direct = gr.Button("Generate Speech", variant="primary", size="lg")
+            text_submit_direct = gr.Button(
+                "Generate Speech", variant="primary", size="lg"
+            )
             tabs = None
             input_files_list = None
             file_preview = None

ui/lib/components/model.py

@@ -1,4 +1,4 @@
-from typing import Tuple, Optional
+from typing import Optional, Tuple
 
 import gradio as gr
 

ui/lib/components/output.py

@@ -12,7 +12,7 @@ def create_output_column(disable_local_saving: bool = False) -> Tuple[gr.Column,
         audio_output = gr.Audio(
             label="Generated Speech",
             type="filepath",
-            waveform_options={"waveform_color": "#4C87AB"}
+            waveform_options={"waveform_color": "#4C87AB"},
         )
 
         # Create file-related components with visible=False when local saving is disabled

ui/lib/files.py

@@ -1,8 +1,8 @@
-import os
 import datetime
-from typing import List, Tuple, Optional
+import os
+from typing import List, Optional, Tuple
 
-from .config import INPUTS_DIR, OUTPUTS_DIR, AUDIO_FORMATS
+from .config import AUDIO_FORMATS, INPUTS_DIR, OUTPUTS_DIR
 
 
 def list_input_files() -> List[str]:

ui/lib/handlers.py

@@ -58,17 +58,21 @@ def setup_event_handlers(components: dict, disable_local_saving: bool = False):
 
     def handle_file_upload(file):
         if file is None:
-            return "" if disable_local_saving else [gr.update(choices=files.list_input_files())]
+            return (
+                ""
+                if disable_local_saving
+                else [gr.update(choices=files.list_input_files())]
+            )
 
         try:
             # Read the file content
-            with open(file.name, 'r', encoding='utf-8') as f:
+            with open(file.name, "r", encoding="utf-8") as f:
                 text_content = f.read()
 
             if disable_local_saving:
                 # When saving is disabled, put content directly in text input
                 # Normalize whitespace by replacing newlines with spaces
-                normalized_text = ' '.join(text_content.split())
+                normalized_text = " ".join(text_content.split())
                 return normalized_text
             else:
                 # When saving is enabled, save file and update dropdown
@@ -88,7 +92,11 @@ def setup_event_handlers(components: dict, disable_local_saving: bool = False):
 
         except Exception as e:
             print(f"Error handling file: {e}")
-            return "" if disable_local_saving else [gr.update(choices=files.list_input_files())]
+            return (
+                ""
+                if disable_local_saving
+                else [gr.update(choices=files.list_input_files())]
+            )
 
     def generate_from_text(text, voice, format, speed):
         """Generate speech from direct text input"""
@@ -203,7 +211,11 @@ def setup_event_handlers(components: dict, disable_local_saving: bool = False):
         components["input"]["file_upload"].upload(
             fn=handle_file_upload,
             inputs=[components["input"]["file_upload"]],
-            outputs=[components["input"]["text_input"] if disable_local_saving else components["input"]["file_select"]],
+            outputs=[
+                components["input"]["text_input"]
+                if disable_local_saving
+                else components["input"]["file_select"]
+            ],
         )
 
     if components["output"]["play_btn"] is not None:

ui/lib/interface.py

@@ -1,9 +1,10 @@
-import gradio as gr
 import os
 
+import gradio as gr
+
 from . import api
-from .handlers import setup_event_handlers
 from .components import create_input_column, create_model_column, create_output_column
+from .handlers import setup_event_handlers
 
 
 def create_interface():