Search Results for "audio processing"
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Showing 373 open source projects for "audio processing"
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OpenVINO AI Plugins for Audacity
A set of AI-enabled effects, generators, and analyzers for Audacity
A set of AI-enabled effects, generators, and analyzers for Audacity. These AI features run 100% locally on your PC, no internet connection is necessary. OpenVINO™ is used to run AI models on supported accelerators found on the user's system such as CPU, GPU, and NPU. -
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Step-Audio 2
Multi-modal large language model designed for audio understanding
Step-Audio2 is an advanced, end-to-end multimodal large language model designed for high-fidelity audio understanding and natural speech conversation: unlike many pipelines that separate speech recognition, processing, and synthesis, Step-Audio2 processes raw audio, reasons about semantic and paralinguistic content (like emotion, speaker characteristics, non-verbal cues), and can generate contextually appropriate responses — including potentially generating or transforming audio output. ... -
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Ultimate Vocal Remover (UVR5)
GUI for a Vocal Remover that uses Deep Neural Networks
This application uses state-of-the-art source separation models to remove vocals from audio files. UVR's core developers trained all of the models provided in this package (except for the Demucs v3 and v4 4-stem models). -
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StaxRip
Video encoding GUI for Windows
StaxRip is a powerful, open-source video and audio encoding GUI for Windows that orchestrates industry-standard console tools (such as x265, FFmpeg, mkvmerge) and frame-server systems (like AviSynth+ or VapourSynth) to allow users to transcode, mux, remux, or process media files with fine-grained control. It is not a “one-click” encoder; instead, it grants the user deep control over encoding settings, filtering, resizing, cropping, subtitles, audio processing, container formats, and more — making it a tool of choice for videophiles, enthusiasts, and anyone needing high-quality and customized media output. ... -
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AudioCraft
Audiocraft is a library for audio processing and generation
...It also contains training code and recipes, so researchers can fine-tune on custom data or explore new objectives without building infrastructure from scratch. Example notebooks, CLI tools, and audio utilities help with prompt design, conditioning on reference audio, and post-processing to produce ready-to-share outputs. -
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Faust
Functional programming language for signal processing
Faust (Functional Audio Stream) is a functional programming language for sound synthesis and audio processing with a strong focus on the design of synthesizers, musical instruments, audio effects, etc. Faust targets high-performance signal processing applications and audio plug-ins for a variety of platforms and standards. The core component of Faust is its compiler. -
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Librosa
Python library for audio and music analysis
Librosa is a powerful Python library for analyzing and processing audio and music signals. Built on top of NumPy, SciPy, and matplotlib, it provides a wide range of tools for feature extraction, time-series manipulation, audio display, and music information retrieval. Whether you're building machine learning models for audio classification or visualizing spectrograms, Librosa is a go-to library for researchers and developers working in audio signal processing. -
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Recorder
HTML5 js recording mp3 wav ogg webm amr format
...Provides multiple plug-in function support. Rich audio visualization, variable speed and pitch processing, speech recognition, audio stream playback, etc.; with powerful real-time processing support, it can be used in various web applications: from simple recording to complex real-time voice Recognition (ASR), and even audio-related games, are handled with ease. -
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SFBAudioEngine
A powerhouse of audio functionality for macOS, iOS, and tvOS
SFBAudioEngine is an advanced audio engine designed for macOS and iOS, focusing on high-quality playback, precise audio control, and support for a wide range of audio formats. Built for modern Apple platforms, it provides developers with a robust tool for integrating sophisticated audio functionalities into their applications. It emphasizes extensibility, performance, and clean API design. -
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Whisper
Robust Speech Recognition via Large-Scale Weak Supervision
OpenAI Whisper is a general-purpose speech recognition model. It is trained on a large dataset of diverse audio and is also a multitasking model that can perform multilingual speech recognition, speech translation, and language identification. A Transformer sequence-to-sequence model is trained on various speech processing tasks, including multilingual speech recognition, speech translation, spoken language identification, and voice activity detection. -
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mediasoup
Cutting Edge WebRTC Video Conferencing
mediasoup is a Node.js library that provides a cutting-edge WebRTC server capable of handling real-time communications with efficient media routing and processing. -
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TorchAudio
Data manipulation and transformation for audio signal processing
The aim of torchaudio is to apply PyTorch to the audio domain. By supporting PyTorch, torchaudio follows the same philosophy of providing strong GPU acceleration, having a focus on trainable features through the autograd system, and having consistent style (tensor names and dimension names). Therefore, it is primarily a machine learning library and not a general signal processing library. -
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MediaDevices
Go implementation of the MediaDevices API
mediadevices is a Go library developed by the Pion WebRTC team that enables real-time access to audio and video devices for building native Go applications involving media streaming and conferencing. It provides a cross-platform, unified API for capturing and manipulating media streams and is often used in combination with Pion WebRTC for peer-to-peer communications. Its support for device enumeration, media constraints, and frame processing makes it a powerful building block for custom voice and video solutions in Go. -
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Membrane Core
The core of Membrane Framework, multimedia processing framework
membrane_core is the foundation of the Membrane multimedia framework for Elixir, providing the abstractions and runtime needed to build real-time audio and video pipelines. It models media processing as a graph of lightweight, supervised OTP processes—elements connected by links—so work is isolated, fault-tolerant, and easy to scale or reconfigure at runtime. The core defines a clear lifecycle and callback API for elements, plus concepts like buffers, events, and capabilities/format negotiation to keep components interoperable and type-safe. ... -
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Overtone
Collaborative programmable music
Overtone is an open-source audio environment designed to explore new musical ideas from synthesis and sampling to instrument building, live coding and collaborative jamming. We combine the powerful SuperCollider audio engine, with Clojure, a state-of-the-art lisp, to create an intoxicating interactive sonic experience. Synchronize your visuals and noise with ease. -
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AudioKit
Swift audio synthesis, processing, & analysis platform
AudioKit is an entire audio development ecosystem of code repositories, packages, libraries, algorithms, applications, playgorunds, tests, and scripts, built and used by a community of audio programmers, app developers, engineers, researchers, scientists, musicians, gamers, and people new to programming. An important goal for AudioKit is to allow it to grow and be maintainable by a handful of volunteers. For this reason we have extensive tests that are run whenever changes are made to any... -
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SuperCollider
Audio server, programming language, and IDE for sound synthesis
SuperCollider is a platform for audio synthesis and algorithmic composition, used by musicians, artists, and researchers working with sound. It is free and open source software available for Windows, macOS, and Linux. scsynth, a real-time audio server, forms the core of the platform. It features 400+ unit generators (“UGens”) for analysis, synthesis, and processing. -
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Google AI Edge Gallery
A gallery that showcases on-device ML/GenAI use cases
...The project bundles runnable samples that show how to run TensorFlow Lite/Edge TPU models (and similar lightweight runtimes) on mobile and embedded platforms, demonstrating common tasks like image classification, object detection, audio recognition, and pose estimation. Each sample is intended to be both a learning aid and a practical starting point: code is organized to show model loading, pre/post-processing, performance measurement, and common optimization knobs (quantization, NNAPI/Delegate usage, and hardware accelerators). The repo also collects small, well-documented models and conversion scripts so developers can reproduce a pipeline from a full-size model down to a device-friendly artifact. -
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bfxr
Flash + AIR sound effects generator. Based on Sfxr.
...Its purpose is to enable users, especially game developers and sound designers, to quickly generate retro, 8-bit/“chiptune” style sound effects (“bleeps”, “booms”, “zaps”, etc.) without deep knowledge of audio signal processing. It offers an interactive GUI through which you can tweak many parameters (oscillators, envelopes, filters, etc.) to sculpt custom sound effects; you can preview in real time, export, and iterate. The project includes libraries, HTML templates, and both ActionScript and JavaScript code. It has been well-received (over a thousand stars), but as of 2025, it has been superseded by a newer version called bfxr2, which is a JavaScript reworking of the original. -
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Oboe
Oboe is a C++ library that makes it easy to build high-performance
oboe is a C++ library for building high-performance audio apps on Android, providing a unified, low-latency API over AAudio and OpenSL ES. It abstracts device and API-version differences so developers can focus on audio processing instead of platform quirks. The library emphasizes minimal latency and glitch-free playback/recording via tuned buffer strategies and callback-driven I/O. It supports features like floating-point audio, channel configuration, sample-rate negotiation, and stream sharing to match device capabilities. ... -
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abogen
Generate audiobooks from EPUBs, PDFs and text with captions
abogen is a tool designed to generate audiobooks (or speech narrations) from textual sources such as EPUBs, PDFs, or plain text, with synchronized captions. In other words, it automates the pipeline of reading a digital book (or document), converting its text into speech via a TTS engine, and packaging the result into an audiobook format — likely along with timestamped captions or subtitles that align with the spoken audio. This can be very useful for accessibility, content consumption on... -
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audioFlux
A library for audio and music analysis, feature extraction
A library for audio and music analysis, and feature extraction. Can be used for deep learning, pattern recognition, signal processing, bioinformatics, statistics, finance, etc. audioflux is a deep learning tool library for audio and music analysis, feature extraction. It supports dozens of time-frequency analysis transformation methods and hundreds of corresponding time-domain and frequency-domain feature combinations. -
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ffmpeg.wasm
FFmpeg for browser, powered by WebAssembly
ffmpeg.wasm is a pure WebAssembly (and JavaScript/TypeScript) port of FFmpeg that enables in-browser media recording, conversion, and streaming—letting developers perform video/audio processing entirely client-side without server uploads. Transpiled via Emscripten from FFmpeg and its codecs into WebAssembly. Supports both single-threaded and multi-threaded cores using web workers. Written in TypeScript for improved developer experience. -
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VibeVoice
Open-source multi-speaker long-form text-to-speech model
VibeVoice-1.5B is Microsoft’s frontier open-source text-to-speech (TTS) model designed for generating expressive, long-form, multi-speaker conversational audio such as podcasts. Unlike traditional TTS systems, it excels in scalability, speaker consistency, and natural turn-taking for up to 90 minutes of continuous speech with as many as four distinct speakers. A key innovation is its use of continuous acoustic and semantic speech tokenizers operating at an ultra-low frame rate of 7.5 Hz, enabling high audio fidelity with efficient processing of long sequences. ... -
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Competent Audio
Machine graph audio engine for computer games
.... - Mixers combine audio signals and optionally perform signal processing. - Sinks send audio signals to an output device. Stereo and mono sound output is supported via a slightly customized version of libsoundio 2.0. Audio clips can have arbitrary channel counts, and can be queued for streaming or dynamic music. CA contains a very simple embedded VM for running custom signal processors, allowing you to add custom DSP code (currently assembly language only) without compiling native code. ...
