Search Results for "audio processing"
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Showing 534 open source projects for "audio processing"
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Kimi-Audio
Audio foundation model excelling in audio understanding
Kimi-Audio is an ambitious open-source audio foundation model designed to unify a wide array of audio processing tasks — from speech recognition and audio understanding to generative conversation and sound event classification — within a single cohesive architecture. Instead of fragmenting work across specialized models, Kimi-Audio handles automatic speech recognition (ASR), audio question answering, automatic audio captioning, speech emotion recognition, and audio-to-text chat in one system, enabling developers to build rich, multimodal audio applications without stitching together disparate components. ... -
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Qwen Audio Agent
A realtime voice runtime that keeps Agents talking
...Interfaces include a browser UI, terminal UI, and a macOS desktop orb with an integrated local gateway. Local profile, memory, and task-state files preserve continuity across sessions while configured audio and backend services handle model 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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Fun Audio Chat
Large Audio Language Model built for natural interactions
Fun Audio Chat is an interactive voice-first conversational AI platform designed to let users engage in natural spoken dialogue with large language models in real time, turning speech into context-aware responses while maintaining a smooth back-and-forth experience. It combines speech recognition, audio processing, and AI generation so users can speak simply and receive spoken replies, enabling applications such as virtual assistants, voice bots, and hands-free chat interfaces. ... -
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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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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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iPlug 2
C++ Audio Plug-in Framework for desktop, mobile, xr and web
iPlug 2 is a cross-platform C++ framework for developing audio plug-ins and applications that can target multiple formats and environments from a single codebase. It abstracts both the audio processing layer and the graphical user interface, allowing developers to focus on signal processing and design while the framework handles platform-specific details. The framework supports a wide range of plug-in standards, including VST, Audio Units, AAX, and newer formats like CLAP, enabling compatibility with major digital audio workstations. ... -
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FFmpegFreeUI
3FUI is ffmpeg's light professional interactive shell on Windows
...It supports dozens of video, audio, and image encoders, including hardware-accelerated options, and allows custom parameter input for advanced use cases. The software also includes batch processing capabilities, real-time progress tracking, plugin extensibility, and integration with FFmpeg utilities like ffprobe and ffplay. -
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React Native Audio API
High-performance audio engine for react-native
React Native Audio API is a cross-platform audio library that brings a Web Audio API-style workflow to React Native. It is designed for apps that need real-time control over audio playback, recording, routing, effects, and signal processing. Developers can build consistent audio behavior across iOS, Android, and the web without rewriting separate logic for each platform. -
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FlyingMouse Format
A mouse-themed, offline Windows file converter
FlyingMouse Format is a desktop file converter built for offline use on Windows, with additional macOS builds available. It converts images, text, Office and WPS documents, spreadsheets, presentations, PDFs, audio, video, ZIP files, and several ebook formats. Bundled engines such as FFmpeg, LibreOffice, Poppler, and Tesseract handle media conversion, document processing, PDF work, and OCR locally. Batch jobs show per-file progress, results, and failure reasons. The app remembers target formats and save locations and can preview converted content. ... -
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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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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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pyAudioAnalysis
Python Audio Analysis Library: Feature Extraction, Classification
pyAudioAnalysis is an open-source Python library designed for audio signal analysis, machine learning, and music information retrieval tasks. The project provides a collection of tools that allow developers to extract meaningful features from audio files and use those features for classification, segmentation, and analysis. The library supports multiple audio processing workflows, including feature extraction from raw audio signals, training of machine learning models, and automatic audio segmentation. ... -
15
Shutter Encoder
A professional video compression tool accessible to all
Shutter Encoder is a cross-platform video and audio processing application designed to provide professional-grade encoding and conversion tools through an accessible graphical interface. Built primarily on FFmpeg, it offers a wide range of media operations including transcoding, compression, format conversion, and editing. The software supports numerous codecs and formats, enabling users to prepare media for broadcasting, streaming, or archiving. -
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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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quill macOS
Ultraminimalist macOS recording + transcription
...Its filesystem-based queue can resume unfinished jobs after relaunch and continue processing while a new meeting is recorded. Quill is distributed as a single Swift executable and supports launch-at-login installation, diagnostics, configurable output folders, and post-processing hooks. -
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ffmpeg-normalize
Audio Normalization for Python/ffmpeg
ffmpeg-normalize is a command-line utility designed to normalize audio levels in media files using FFmpeg, ensuring consistent volume across multiple tracks. It supports both EBU R128 loudness normalization and peak normalization methods, allowing users to choose the appropriate standard for their needs. The tool analyzes audio streams and applies adjustments to achieve target loudness levels without introducing distortion. It can process multiple files in batch mode, making it suitable for... -
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Tone.js
A Web Audio framework for making interactive music in the browser
Tone.js is an open source Web Audio framework for creating interactive music in the browser. It has advanced scheduling capabilities, synths, effects and more. Tone.js is built to be useful and familiar to both musicians and audio programmers. It has common DAW (digital audio workstation) features for those looking to schedule events and tinker with pre-built synths and effects. -
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ThinkDSP
Digital Signal Processing in Python, by Allen B. Downey
Think DSP is an educational Python project that teaches digital signal processing through executable examples rather than starting with heavy mathematical formalism. It accompanies Allen B. Downey’s book and organizes most lessons as Jupyter notebooks. Readers work directly with waves, spectra, harmonics, filtering, convolution, and other signal-processing concepts. Early exercises show how to decompose sounds, modify frequency components, and synthesize new audio. ... -
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JSNES
A JavaScript NES emulator
JSNES is a JavaScript-based emulator that replicates the functionality of the Nintendo Entertainment System (NES), enabling classic games to run directly in web browsers or Node.js environments. It implements the core components of NES hardware, including the CPU, graphics processing unit, and audio system, to deliver an accurate emulation experience. The project is designed as a library, allowing developers to embed emulation capabilities into web applications or custom interfaces. It includes support for rendering graphics via canvas, handling audio output, and processing input from keyboards or gamepads. ... -
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wlmedia
Next audio and video player SDK
wlmedia is a multimedia processing and playback project designed to provide tools for handling video and audio content in lightweight environments. It focuses on integrating FFmpeg capabilities into streamlined workflows for decoding, encoding, and playback. The project includes support for multiple media formats and emphasizes performance and efficiency in resource-constrained systems. -
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AutoSubSync
Automatic subtitle synchronization tool
AutoSubSync is a cross-platform desktop application designed to automatically synchronize subtitle files with video content using advanced alignment algorithms. It integrates tools like ffsubsync, autosubsync, and alass to analyze audio and match subtitle timing with high accuracy. The application supports both automatic synchronization and manual adjustment, allowing users to fine-tune results when needed. It provides a drag-and-drop interface that simplifies the process of loading video and subtitle files, making it accessible for non-technical users. AutoSubSync also includes batch processing capabilities, enabling users to handle entire media libraries efficiently. ... -
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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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Moonshine Voice
Fast and accurate automatic speech recognition (ASR) for edge devices
...The project is designed to enable real-time voice applications such as live transcription, voice commands, and embedded speech interfaces without requiring heavy cloud infrastructure. Its architecture emphasizes low latency and flexible input handling, allowing audio streams of varying durations rather than relying on fixed processing windows. Moonshine supports multiple platforms including mobile, desktop, and embedded systems, and provides example projects to accelerate integration into real-world products. The toolkit also includes specialized model variants, including monolingual options that improve accuracy for specific languages. ...
