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# Kokoro TTS API [![Tests](https://img.shields.io/badge/tests-105%20passed-darkgreen)]() [![Coverage](https://img.shields.io/badge/coverage-74%25-darkgreen)]() [![Tested at Model Commit](https://img.shields.io/badge/last--tested--model--commit-a67f113-blue)](https://huggingface.co/hexgrad/Kokoro-82M/tree/c3b0d86e2a980e027ef71c28819ea02e351c2667) [![Try on Spaces](https://img.shields.io/badge/%F0%9F%A4%97%20Try%20on-Spaces-blue)](https://huggingface.co/spaces/Remsky/Kokoro-TTS-Zero) Dockerized FastAPI wrapper for [Kokoro-82M](https://huggingface.co/hexgrad/Kokoro-82M) text-to-speech model - OpenAI-compatible Speech endpoint, with inline voice combination functionality - NVIDIA GPU accelerated inference (or CPU) option - very fast generation time - 35x+ real time speed via 4060Ti, ~300ms latency - 5x+ real time spead via M3 Pro CPU, ~1000ms latency - streaming support w/ variable chunking to control latency & artifacts - simple audio generation web ui utility ## Quick Start The service can be accessed through either the API endpoints or the Gradio web interface. 1. Install prerequisites: - Install [Docker Desktop](https://www.docker.com/products/docker-desktop/) + [Git](https://git-scm.com/downloads) - Clone and start the service: ```bash git clone https://github.com/remsky/Kokoro-FastAPI.git cd Kokoro-FastAPI docker compose up --build ``` 2. Run locally as an OpenAI-Compatible Speech Endpoint ```python from openai import OpenAI client = OpenAI( base_url="http://localhost:8880", api_key="not-needed" ) response = client.audio.speech.create( model="kokoro", voice="af_sky+af_bella", #single or multiple voicepack combo input="Hello world!", response_format="mp3" ) response.stream_to_file("output.mp3") ``` or visit http://localhost:7860

Voice Analysis Comparison

## Features
OpenAI-Compatible Speech Endpoint ```python # Using OpenAI's Python library from openai import OpenAI client = OpenAI(base_url="http://localhost:8880", api_key="not-needed") response = client.audio.speech.create( model="kokoro", # Not used but required for compatibility, also accepts library defaults voice="af_bella+af_sky", input="Hello world!", response_format="mp3" ) response.stream_to_file("output.mp3") ``` Or Via Requests: ```python import requests response = requests.get("http://localhost:8880/v1/audio/voices") voices = response.json()["voices"] # Generate audio response = requests.post( "http://localhost:8880/v1/audio/speech", json={ "model": "kokoro", # Not used but required for compatibility "input": "Hello world!", "voice": "af_bella", "response_format": "mp3", # Supported: mp3, wav, opus, flac "speed": 1.0 } ) # Save audio with open("output.mp3", "wb") as f: f.write(response.content) ``` Quick tests (run from another terminal): ```bash python examples/test_openai_tts.py # Test OpenAI Compatibility python examples/test_all_voices.py # Test all available voices ```
Voice Combination - Averages model weights of any existing voicepacks - Saves generated voicepacks for future use - (new) Available through any endpoint, simply concatenate desired packs with "+" Combine voices and generate audio: ```python import requests response = requests.get("http://localhost:8880/v1/audio/voices") voices = response.json()["voices"] # Create combined voice (saves locally on server) response = requests.post( "http://localhost:8880/v1/audio/voices/combine", json=[voices[0], voices[1]] ) combined_voice = response.json()["voice"] # Generate audio with combined voice (or, simply pass multiple directly with `+` ) response = requests.post( "http://localhost:8880/v1/audio/speech", json={ "input": "Hello world!", "voice": combined_voice, # or skip the above step with f"{voices[0]}+{voices[1]}" "response_format": "mp3" } ) ```

Voice Analysis Comparison

Multiple Output Audio Formats - mp3 - wav - opus - flac - aac - pcm

Audio Format Comparison

Gradio Web Utility Access the interactive web UI at http://localhost:7860 after starting the service. Features include: - Voice/format/speed selection - Audio playback and download - Text file or direct input If you only want the API, just comment out everything in the docker-compose.yml under and including `gradio-ui` Currently, voices created via the API are accessible here, but voice combination/creation has not yet been added *Note: Recent updates for streaming could lead to temporary glitches. If so, pull from the most recent stable release v0.0.2 to restore*
Streaming Support ```python # OpenAI-compatible streaming from openai import OpenAI client = OpenAI( base_url="http://localhost:8880", api_key="not-needed") # Stream to file with client.audio.speech.with_streaming_response.create( model="kokoro", voice="af_bella", input="Hello world!" ) as response: response.stream_to_file("output.mp3") # Stream to speakers (requires PyAudio) import pyaudio player = pyaudio.PyAudio().open( format=pyaudio.paInt16, channels=1, rate=24000, output=True ) with client.audio.speech.with_streaming_response.create( model="kokoro", voice="af_bella", response_format="pcm", input="Hello world!" ) as response: for chunk in response.iter_bytes(chunk_size=1024): player.write(chunk) ``` Or via requests: ```python import requests response = requests.post( "http://localhost:8880/v1/audio/speech", json={ "input": "Hello world!", "voice": "af_bella", "response_format": "pcm" }, stream=True ) for chunk in response.iter_content(chunk_size=1024): if chunk: # Process streaming chunks pass ```

GPU First Token Timeline CPU First Token Timeline

Key Streaming Metrics: - First token latency @ chunksize - ~300ms (GPU) @ 400 - ~3500ms (CPU) @ 200 - Adjustable chunking settings for real-time playback *Note: Artifacts in intonation can increase with smaller chunks*
## Processing Details
Performance Benchmarks Benchmarking was performed on generation via the local API using text lengths up to feature-length books (~1.5 hours output), measuring processing time and realtime factor. Tests were run on: - Windows 11 Home w/ WSL2 - NVIDIA 4060Ti 16gb GPU @ CUDA 12.1 - 11th Gen i7-11700 @ 2.5GHz - 64gb RAM - WAV native output - H.G. Wells - The Time Machine (full text)

Processing Time Realtime Factor

Key Performance Metrics: - Realtime Speed: Ranges between 25-50x (generation time to output audio length) - Average Processing Rate: 137.67 tokens/second (cl100k_base)
GPU Vs. CPU ```bash # GPU: Requires NVIDIA GPU with CUDA 12.1 support (~35x realtime speed) docker compose up --build # CPU: ONNX optimized inference (~2.4x realtime speed) docker compose -f docker-compose.cpu.yml up --build ``` *Note: Overall speed may have reduced somewhat with the structural changes to accomodate streaming. Looking into it*
Natural Boundary Detection - Automatically splits and stitches at sentence boundaries - Helps to reduce artifacts and allow long form processing as the base model is only currently configured for approximately 30s output
## Model and License
Model This API uses the [Kokoro-82M](https://huggingface.co/hexgrad/Kokoro-82M) model from HuggingFace. Visit the model page for more details about training, architecture, and capabilities. I have no affiliation with any of their work, and produced this wrapper for ease of use and personal projects.
License This project is licensed under the Apache License 2.0 - see below for details: - The Kokoro model weights are licensed under Apache 2.0 (see [model page](https://huggingface.co/hexgrad/Kokoro-82M)) - The FastAPI wrapper code in this repository is licensed under Apache 2.0 to match - The inference code adapted from StyleTTS2 is MIT licensed The full Apache 2.0 license text can be found at: https://www.apache.org/licenses/LICENSE-2.0