Open-source alternatives to Openutau

DiffSinger is an AI vocal synthesizer and neural audio generator designed to produce high-fidelity singing and speech. It functions as a text-to-speech system and a diffusion-based singing voice synthesis tool that transforms text and pitch into audible audio. The system utilizes a shallow diffusion mechanism and iterative noise refinement to generate realistic vocal performances. It incorporates specialized sampling plugins and numerical solvers to accelerate inference and reduce the time required to generate synthetic voices. The project covers acoustic modeling, mel-spectrogram synthesis,

Muzic is a deep learning platform and framework for AI-driven music analysis, composition, and synthesis. It functions as a music generation framework and analysis tool, utilizing large language models and autonomous agents to orchestrate the creation and interpretation of symbolic and audio music. The project is distinguished by its cross-modal capabilities, mapping natural language and symbolic music into a shared joint embedding space for zero-shot classification and information retrieval. It employs a variety of specialized architectures, including diffusion frameworks for audio synthesis

Ardour is a digital audio workstation, multitrack audio mixer, and MIDI sequencer. It functions as a non-linear audio editor and a plugin host for running third-party effects and instruments. The system provides specialized capabilities for post-production audio scoring through video frame synchronization, as well as live performance sequencing for triggering clips and patterns in real-time. It also supports tactile mixing via control surface mapping and hardware controller configuration. The software covers a broad range of audio production needs, including multitrack recording, MIDI sequen

Kokoro-FastAPI is a text-to-speech API and LLM speech synthesis server that generates spoken audio from text via a REST interface. It functions as a Kubernetes-native deployment designed for orchestrated speech synthesis. The system includes a voice blending engine that creates unique vocal profiles by mixing multiple existing voices using custom weight ratios. The service provides real-time audio streaming to reduce latency and generates word-level timestamps for speech synchronization. It manages hardware efficiency through on-demand model loading to optimize VRAM usage and includes system

EmotiVoice is an emotional text-to-speech engine and bilingual speech synthesizer designed to generate synthetic audio in English and Chinese. It utilizes a deep learning architecture to produce high-fidelity speech with controllable emotional states and timbres. The project includes a voice cloning framework for replicating specific speaker identities by training custom acoustic models on personal audio datasets. It employs a jointly-trained acoustic-vocoder pipeline and style-embedding-based synthesis to manage expression and reduce audio artifacts. The system covers a broad range of speec

MegaTTS3 is a bilingual speech synthesis system that generates natural-sounding speech in Chinese and English, including seamless code-switching within a single utterance. It functions as a text-to-speech engine, voice cloning system, and speech-to-text alignment tool, built around an acoustic latent compression model that encodes high-resolution audio into compact representations for efficient processing. The system distinguishes itself through accent intensity control, allowing adjustment of a speaker's accent strength in generated speech, and voice cloning from short audio samples for pers

This project is an AI singing voice conversion system and vocal processor used for training generative voice models and converting vocal recordings or live input into a target voice. It functions as a VITS model trainer and a real-time voice changer that transforms vocal timbre and pitch to change the identity of a singer. The system provides a graphical management dashboard for controlling training hyperparameters and voice conversion presets. It supports low-latency audio streaming for live microphone input and employs pitch estimation to ensure precise matching between source and target vo

espeak-ng is a multilingual text-to-speech engine and C-based library that converts written text into spoken audio across various languages, accents, and regional dialects. It functions as both a programmatic interface for embedding synthesis capabilities into external applications and a phonetic text converter that translates written text into phoneme codes. The system utilizes multiple synthesis methods, including formant synthesis to generate vocal sounds mathematically and diphone synthesis to produce audio by concatenating pre-recorded phonetic segments. It incorporates a speech processo

Kokoro is a lightweight neural text-to-speech engine that converts written text into spoken audio using a compact model designed for fast inference. It supports multiple languages through language-specific grapheme-to-phoneme conversion pipelines, and offers voice profile selection to change the character of the generated speech. The engine provides GPU acceleration on Apple Silicon hardware by setting a single environment variable, enabling faster inference on Mac M-series machines. It also includes pattern-based text segmentation, allowing input text to be split at user-defined delimiters t

This project is an AI voice training framework and singing voice conversion tool. It uses VITS and SoftVC technologies to transform the timbre of singing and spoken audio recordings, allowing a user to change the vocal characteristics of a recording to match a specific target speaker. The system provides a web-based voice converter interface for managing model checkpoints and performing timbre transformation and pitch shifting. It supports exporting trained models to the ONNX format for use in external interfaces and lightweight runtimes. The framework covers the full production pipeline, in

Amphion is an audio generation toolkit designed for the research and development of models that synthesize speech, music, and environmental sound effects. It provides a standardized framework for reproducible audio synthesis, incorporating a text-to-speech engine and a voice conversion framework. The project specializes in transforming audio identities, allowing for the modification of speaker accents and voice identities while preserving original rhythm and style. It also includes capabilities for singing voice synthesis and the generation of environmental soundscapes from text descriptions

ESPnet is a comprehensive speech processing toolkit and PyTorch-based trainer designed for building end-to-end speech recognition, synthesis, and translation models. It provides a structured framework for developing automatic speech recognition systems using transducer and encoder-decoder architectures, alongside engines for text-to-speech synthesis and speech translation pipelines. The project distinguishes itself through a recipe-based workflow execution system that ensures experimental reproducibility by running standardized sequences of scripts for data preparation and model training. It

AudioGPT is an LLM-driven audio framework and processing suite that uses large language models to orchestrate neural audio pipelines. It functions as a multimodal audio generator and processing system, integrating a collection of pretrained models to handle speech synthesis, sound generation, and audio manipulation. The system is distinguished by its ability to generate audio from diverse inputs, including text and images, and its capacity to produce synchronized talking head videos. It also operates as a neural speech translator, converting spoken language between different tongues while pre

Basic-pitch is a neural network audio transcriber and polyphonic pitch detector. It functions as an audio-to-MIDI converter that transforms polyphonic audio recordings into MIDI note events and pitch bend data. The system preserves musical expression by tracking continuous frequency fluctuations to convert glides and vibrato into MIDI pitch bend events. It utilizes a pluggable inference engine that allows the model runtime to be configured based on the operating system or hardware acceleration needs. The project provides a command line interface for batch audio processing and a programmatic

Logoly is a browser-based logo editor that generates logos mimicking the signature visual style of popular platforms. It renders logo previews by drawing vector graphics directly onto an HTML canvas element, managing all editor state entirely within the browser. The tool provides real-time customization of logo text, color, and font size through an interactive on-screen editor, with instant preview updates as adjustments are made. It applies visual styles by mapping user-selected colors and fonts to CSS-like properties on SVG elements, and uses predefined layout templates to position text and

Moonshine is a complete on-device voice interface toolkit that provides speech recognition, text-to-speech synthesis, phonetic processing, speaker diarization, and intent recognition, all running locally on edge hardware without any cloud dependency. It executes quantized neural networks for speech and language tasks directly on the device, enabling fully offline conversational AI capabilities. The toolkit distinguishes itself by orchestrating multi-turn spoken exchanges through a conversational flow manager that maintains context across interactions and manages branching dialog flows. It inc

THIS REPO IS NOT MAINTAINED ANYMORE. Please see https://codeberg.org/tenacityteam/tenacity for Tenacity, which is maintained.

ACE Step 1.5 is a local text-to-music generation and audio editing system that runs on consumer hardware. It transforms plain-language descriptions into full-length songs with lyrics, and can edit existing audio through cover generation, vocal removal, track separation, and selective repainting. The system supports multilingual prompts and lyrics in over 50 languages, and provides precise control over musical structure including duration, BPM, key, and time signature. The project distinguishes itself through a dual-stream diffusion architecture that processes separate latent streams for vocal

VoxCPM is a multilingual speech synthesis system and text-to-speech inference server. It functions as an AI voice cloning tool and a synthetic voice designer, capable of generating natural speech across global languages and regional dialects using a GPU-accelerated audio generator. The project features a speech model fine-tuning framework that supports both full parameter updates and low-rank adaptation for customizing voice characteristics. It enables high-fidelity voice cloning from reference audio, including cross-lingual voice transfer and acoustic environment mimicry, as well as the crea

This project provides a React component and a set of utilities for implementing interactive image cropping. It enables users to define selection areas over media elements using a draggable and resizable interface. The library supports fixed aspect ratio locking and the definition of custom crop shapes, such as circles. It utilizes a percentage-based coordinate system to ensure the selection interface remains responsive across different screen sizes and layouts. The toolset includes capabilities for real-time crop previewing and the generation of cropped regions using canvas-based rendering.

This is a Chinese text segmentation library that converts Chinese characters into their phonetic pinyin representation. It functions as a polyphone disambiguation tool, resolving ambiguous pronunciations for multi-sound characters using word segmentation and context analysis, and also serves as a pinyin sorting utility for ordering Chinese strings alphabetically. The library distinguishes itself through surname-aware pronunciation switching, applying specialized phonetic rules for Chinese surnames with non-standard pronunciations in name contexts. It supports pluggable word segmentation algor

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This project is a neural text-to-speech system and voice trainer that converts written text into spoken audio across a variety of global languages and regional dialects. It functions as an ONNX-based engine capable of performing fast offline inference and uses a phoneme-based controller to manage precise pronunciation. The system distinguishes itself through a comprehensive toolkit for neural voice training, allowing for the creation of custom single-speaker or multi-speaker models. It supports the export of these models to a standardized open format and provides hardware acceleration via gra

ChatTTS is a conversational text-to-speech generative model designed to convert written dialogue into natural sounding audio. It functions as a multilingual speech synthesis framework capable of producing human-like audio across different languages and speaker profiles. The system is distinguished by its ability to generate interactive dialogue with realistic vocal nuances. It utilizes a speech nuance controller to insert specific tokens that trigger non-verbal elements, such as laughter, pauses, and interjections, during the synthesis process. The project includes a streaming audio generato

ace-step-ui is an AI music production workspace and interface for generating, editing, and organizing synthetic audio tracks and vocals. It provides a technical control panel for managing prompts, seeds, and style parameters to produce high-quality audio. The project includes a digital audio workstation interface for trimming and fading files, alongside an audio stem separation tool that splits mixed tracks into individual components such as drums, bass, and vocals. It also features a music video creator for generating visual content and procedural album art to accompany generated music. The

wavesurfer.js is a WebAudio playback library and interactive waveform visualizer that renders audio data onto an HTML5 canvas. It enables users to see and navigate sound files through a visual representation of audio peaks, allowing for direct seeking and playback control within a web browser. The project is distinguished by its flexible rendering model, which can use precomputed peak data to display waveforms without downloading or decoding the full audio file. It utilizes a plugin-based extension model to integrate advanced tools such as spectrograms, interactive audio timelines, and real-t

Audacity is a cross-platform digital audio workstation and multi-track audio editor. It serves as a comprehensive suite for capturing live audio input, refining sound files through splicing and effects, and mixing multi-track audio files using a non-destructive waveform interface. The project functions as a VST3 plugin host, providing a software environment to load and execute audio effects and virtual instruments for real-time signal processing. It also includes an audio spectrum analyzer for visualizing frequencies and waveforms to identify specific sonic characteristics. The software cove