Projects sharing features with Ultimatevocalremovergui

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

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

Vocal Remover is a deep learning application designed for audio source separation. It functions as a command-line utility that decomposes complex audio signals into individual components, specifically isolating vocals and instrumental tracks from mixed recordings. The software utilizes a symmetric encoder-decoder neural network architecture to process audio spectrograms. By applying learned magnitude masks to the original signal phase, the system reconstructs output audio while maintaining temporal coherence. It supports both the execution of pre-trained models for track extraction and the tr

SpleeterGui is a graphical interface for the Spleeter machine learning library, serving as an AI source separation tool and audio stem extractor. It allows users to separate mixed audio files into individual source tracks, such as vocals, drums, and bass, using a visual application. The project functions as a wrapper for the Spleeter engine, removing the requirement to use command line tools for music stem isolation and audio source separation. It provides a visual method for managing audio source isolation and preparing instrument tracks. The interface includes tools for output directory ma

ncmdump is a proprietary audio cache converter and binary stream decryption utility. It decrypts raw audio data from specialized music cache files and transforms them into standard audio formats. The project functions both as a standalone tool and a cross-language conversion library. It exposes its internal decryption and conversion logic as a dynamic library, allowing the capabilities to be embedded into external applications written in different programming languages. The utility provides batch media processing through recursive directory traversal, enabling the identification and conversi

This project is a cross-platform machine learning inference engine designed to execute pre-trained models across diverse operating systems and hardware environments. It functions as a standardized execution framework that manages the entire lifecycle of model inference, from loading and graph optimization to hardware-accelerated execution and generative sequence management. The runtime distinguishes itself through a highly modular architecture that decouples model logic from hardware-specific kernels. By utilizing an execution provider abstraction, it enables developers to offload computation

Matchering is an audio mastering tool and Python library designed to match the frequency balance and loudness of a target track to a specific reference track. It functions as a reference-based mastering system that aligns a target signal's spectral envelope, RMS, and peak amplitude with those of a chosen reference file. The project utilizes a multi-stage processing pipeline featuring an FFT spectral matching engine to adjust frequency response. It ensures output quality through the use of a brickwall limiter to prevent signal clipping while preserving the original waveform shape. The tool pr

Librosa is a Python audio analysis library and digital signal processing framework. It functions as a feature extraction suite and music information retrieval tool designed to analyze the structural and sonic characteristics of audio signals. The library provides specialized capabilities for music analysis, including dynamic tempo tracking to identify rhythmic pulses and spectral feature extraction to compute harmonic spectra, chroma variants, and onset points. It also serves as a time-series audio processor for synchronizing audio streams. The system covers a broad range of audio processing

Sherpa-ONNX is an ONNX-based speech processing toolkit that provides a local speech recognition engine, an on-device voice synthesis tool, and a speaker identification framework. It is designed as a cross-platform speech API that enables speech-to-text, text-to-speech, and speaker verification tasks to be executed locally on a device without requiring network access. The project is distinguished by its ability to perform zero-shot voice cloning and speaker diarization on-device. It supports a wide range of hardware accelerations, including GPU and various NPU architectures, and provides a Web

Aubio is an audio analysis and digital signal processing library designed for music information retrieval. It provides a suite of tools for extracting musical features, estimating fundamental frequencies, and tracking rhythmic pulses in audio streams. The library specializes in the detection of pitch and beat, enabling the extraction of musical notes and the estimation of overall tempo. It also includes capabilities for automatic onset detection to identify the start of sonic events and the separation of audio signals into percussive transients and steady-state tonal components. The system c

LMMS is a digital audio workstation and MIDI sequencer designed for composing, arranging, and mixing music. It functions as a comprehensive production environment that integrates a MIDI sequencer, a sample-based synthesizer, and an audio mixing console. The project distinguishes itself through a versatile synthesis engine that includes additive synthesis, wavetable generation, and emulations of vintage hardware such as NES audio and FM chips. It also serves as a VST plugin host, allowing for the integration of third-party virtual instruments and audio effects via a standardized interface. Be

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

JUCE is a comprehensive C++ audio framework and digital signal processing library used to build cross-platform audio applications, audio plug-ins, and high-performance user interfaces. It serves as a development kit for creating audio processors compatible with industry-standard plugin formats for digital audio workstations, as well as a tool for MIDI and Open Sound Control communication between musical hardware and software. The framework is distinguished by its ability to maintain a single codebase for native desktop and mobile applications across multiple operating systems. It provides a f

This project is a scalable, containerized pipeline designed to transform digital documents and image-based ebooks into narrated audiobooks. It functions as an end-to-end production platform that integrates text-to-speech synthesis, optical character recognition, and automated workflow management to convert various file formats into spoken audio. The system distinguishes itself through advanced linguistic analysis and voice synthesis capabilities, including the ability to identify characters within a text and assign them distinct voice profiles for multi-speaker narration. Users can further pe

Frigate is a self-hosted network video recorder that functions as a private, local AI-powered vision engine. It manages video streams by performing real-time object detection, tracking, and classification directly on local hardware, ensuring that security monitoring and activity recording remain independent of cloud services. The system distinguishes itself through a modular, hardware-accelerated video pipeline that offloads intensive decoding and machine learning inference to dedicated GPUs, NPUs, or specialized accelerators like Coral TPUs and Hailo modules. It utilizes state-based object t

NAudio is a .NET audio library that provides playback, recording, format conversion, and signal processing capabilities for Windows applications. It serves as an audio format converter between common formats like WAV, MP3, AAC, and AIFF using ACM or Media Foundation codecs, and functions as an audio playback and recording API through Windows audio APIs including WaveOut, DirectSound, ASIO, and WASAPI. The library includes an audio stream processor that can mix, trim, loop, fade, resample, and adjust pitch or channel count of audio streams with a floating-point engine. It also provides an audi

Whisper.cpp is a high-performance, local-first speech recognition engine designed to run large-scale machine learning models on consumer hardware. It functions as a portable library that converts audio into text, supporting both static file transcription and real-time stream processing. By utilizing a lightweight inference engine and weight quantization, the project minimizes memory and compute overhead, allowing for efficient execution without reliance on external cloud APIs or internet connectivity. The project distinguishes itself through a hardware-agnostic compute abstraction that offloa

MAME is a vintage hardware emulation platform designed to recreate the circuitry of arcade games, computers, and consoles to run original software on modern devices. It functions as a retro gaming preservation framework for managing, verifying, and archiving ROM sets and disk images to ensure long-term software accessibility. The project features a system debugging tool for inspecting emulated memory, CPU registers, and execution flow via breakpoints and disassembly. It also includes a Lua-based automation layer that exposes core system state and hardware controls for custom behavior and anal

Recorder is a JavaScript library for capturing audio from microphones and encoding the resulting data into multiple formats, including MP3, WAV, and WebM. It provides a unified interface for cross-platform audio recording across web browsers and mobile applications. The library functions as an audio signal processing tool and waveform visualizer, offering real-time visual feedback through frequency histograms and waveforms. It includes specialized capabilities for encoding and decoding dual-tone multi-frequency telephone keypad tones. The project covers a broad range of audio manipulation ca

Gyroflow is a gyroscope video stabilization software and IMU telemetry processor designed to remove camera shake from video files. It functions as a hardware-accelerated video renderer and lens calibration tool, utilizing embedded or external gyroscope and accelerometer data to perform pixel-level stabilization. The system is distinguished by its ability to integrate with professional non-linear video editing software via plugins, allowing stabilization to be applied directly to timelines without transcoding original footage. It supports diverse telemetry ingestion from camera brands, flight

Vocal-separate is an audio processing tool designed to isolate vocal and instrumental tracks from audio and video files. It functions as a local artificial intelligence engine that performs source separation directly on the user's machine, ensuring data privacy by eliminating the need for external server connectivity. The system provides a browser-based control interface for managing media uploads and monitoring processing tasks. To handle intensive signal decomposition, it utilizes hardware-accelerated tensor processing, which offloads complex mathematical calculations to dedicated graphics

Spleeter is an AI audio source separation library and deep learning toolkit designed to split mixed music files into individual audio stems, such as vocals and drums. It provides a suite of pretrained models for isolating different instruments and voices from a recording. The toolkit includes capabilities for training and evaluating custom audio separation models using labeled datasets and configuration files. It also features utilities for measuring model performance by comparing separation outputs against reference datasets. The system manages audio processing through spectral representati

mlx-audio is an audio processing toolkit built on Apple MLX that provides speech transcription, text-to-speech synthesis, voice cloning, and audio source separation using local models. It offers an OpenAI-compatible REST API and web interface for running audio generation and transcription tasks, enabling drop-in integration with existing tools that follow that endpoint structure. The toolkit supports text-prompted audio source separation, allowing specific sounds to be isolated from mixed recordings based on natural language descriptions. It also provides voice cloning from a short reference

This project is a deep learning toolkit designed for audio source separation and music information retrieval. It provides a framework for decomposing polyphonic audio signals into distinct components, such as vocals, drums, and bass, by processing raw waveforms through neural network architectures. The library enables users to train custom separation models or fine-tune existing ones to improve accuracy on specific audio datasets. It supports the entire model lifecycle, including the conversion of raw audio into structured, indexed formats to optimize data loading and training efficiency. Th

Linly-Dubbing is an automated video dubbing pipeline designed for multilingual video localization. It converts spoken content in videos into another language by coordinating speech-to-text transcription, text translation, and text-to-speech synthesis. The system distinguishes itself through AI-driven lip synchronization and animation, which aligns facial expressions and mouth movements to the synthesized voiceover. It also utilizes audio source separation to isolate vocals from background music and noise, allowing for clean voice replacement while preserving original background audio. The br

This project is a comprehensive toolkit for on-device speech recognition, synthesis, and audio processing, specifically engineered for Apple Silicon. It provides a framework for building real-time, full-duplex voice agents that operate entirely offline, leveraging native hardware acceleration to maintain performance and privacy. By utilizing optimized machine learning models, the library enables local execution of complex audio tasks without reliance on external cloud services. The library distinguishes itself through its specialized focus on local, high-performance voice interaction. It incl