Search Results for "audio processing" - Page 2
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Showing 68 open source projects for "audio processing"
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OmniPull
Just pull anything
OmniPull is a powerful, cross-platform download manager built with Python and PySide6. It provides a modern, intuitive interface for managing downloads with advanced features like multi-threading, queue management, and media extraction. -
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Kisekae UltraKiss
Kisekae UltraKiss is a full featured integrated development environmen
UltraKiss is a computer program that implements the Kisekae Set system, KiSS, a Japanese graphics system originally developed to facilitate costume changes on virtual dolls. UltraKiss was developed to help artists build their KiSS sets. It is a full featured viewer for all KiSS dolls, games, and visual applications. It is also a complete graphical development environment for creating KiSS applications. It fully implements the FKiSS event driven programming language up to and including... -
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Common Resource Grep - crgrep
Common Resource Grep
CRGREP searches for matching text in databases, various document formats, archives and other difficult to access resources. A command line tool for name and content text matching in database tables, plain files, MS Office documents, PDF, archives, MP3 audio, image meta-data, scanned documents, maven dependencies and web resources. CRGREP will search resources within resources of any arbitrary combination or depth, so text within a document within a zip archive, and so on. Here you... -
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slab3d
slab3d is a real-time virtual acoustic environment.
slab3d is a real-time virtual acoustic environment rendering system. It includes two rendering applications, libraries for spatial auditory display and audio signal processing development, and a set of tools for HRTF analysis and visualization. -
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Glicol
Graph-oriented live coding language and music/audio DSP library
Glicol is a graph-oriented live coding language and audio engine designed for real-time music creation and digital signal processing, written entirely in Rust. It introduces a unique paradigm where audio synthesis and sequencing are represented as interconnected nodes, allowing developers and musicians to construct complex sound pipelines through declarative code. The language is designed to be accessible to beginners while still offering powerful capabilities for advanced users, enabling both quick experimentation and precise control over audio generation. ... -
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VideoSrt
Windows-GUI
...Open source software tool that can recognize video speech and automatically generate subtitle SRT files. It is suitable for business scenarios that quickly and batch generate Chinese/English subtitles and text files for media (video/audio). Recognize video/audio speech to generate subtitle files (support Chinese-English translation, bilingual subtitles) Extract speech text from video/audio. Batch translation, filter processing/encoding SRT subtitle files. Using the Alibaba Cloud speech recognition interface, the accuracy is high, and the standard Mandarin/English recognition rate is over 95%. ... -
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SVoice (Speech Voice Separation)
We provide a PyTorch implementation of the paper Voice Separation
...The repository includes all necessary scripts for training, dataset preparation, distributed training, evaluation, and audio separation. -
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Beep
A little package that brings sound to any Go application
A little package that brings sound to any Go application. Suitable for playback and audio processing. Beep is built on top of its Streamer interface, which is like io.Reader, but for audio. It was one of the best design decisions I've ever made and it enabled all the rest of the features to naturally come together with not much code. Decode and play WAV, MP3, OGG, and FLAC. Encode and save WAV. Very simple API. Limiting the support to stereo (two channel) audio made it possible to simplify the architecture and the API. ... -
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CHOW Phaser
Phaser effect based loosely on the Schulte Compact Phasing 'A'
ChowPhaser is an open-source audio plugin that emulates the classic Schulte Compact Phasing 'A' effect. It offers a unique phasing effect with nonlinear feedback and modulation capabilities, suitable for various audio processing applications. -
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newlearnerspermit
C for programmable guitar amp
This project experiments with C language to discover and design a programmable guitar amplifier effects micro controller computer section of a tube and solid state designed guitar amp. This a collaboration electronics project with my son. The goal of this project is to use Arduino hardware with C language programming to manipulate the guitar signal by various effects of phaser, flanger, reverb, digital delay and echo, and tremelo. Also, the signals of two microphones, two guitars, and... -
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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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vinylsimtool
vinylsimtool: simulation of vinyl record playback
vinylsimtool: Simulation of audio artifacts of vinyl record playback systems, based on physical modelling. --- vinylsimtool: Simple command-line tool for WAV file processing. Copyright (c) 2017 by Klaus Michael Indlekofer. All rights reserved. --- Vinylsimtool-Plugin-VST: Real-time plug-in. Copyright (c) 2017-2019 by Klaus Michael Indlekofer and Sebastian Marcus Edelmann. -
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Faust : signal processing language
Faust is a programming language for realtime audio signal processing
[UPDATE] The project has been moved to GitHub (https://github.com/grame-cncm/faust). Do not use this repository anymore ! FAUST (Functional Audio Stream) is a functional programming language specifically designed for real-time signal processing and synthesis. FAUST targets high-performance signal processing applications and audio plug-ins for a variety of platforms and standards. The Faust compiler translates DSP specifications into very efficient C++ code. Thanks to the notion of architecture, FAUST programs can be easily deployed on a large variety of audio platforms and plugin formats (jack, alsa, ladspa, maxmsp, puredata, csound, supercollider, pure, vst, coreaudio) without any change to the FAUST code. -
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Accord.NET Framework
Machine learning, computer vision, statistics and computing for .NET
The Accord.NET Framework is a .NET machine learning framework combined with audio and image processing libraries completely written in C#. It is a complete framework for building production-grade computer vision, computer audition, signal processing and statistics applications even for commercial use. A comprehensive set of sample applications provide a fast start to get up and running quickly, and extensive documentation and a wiki help fill in the details. -
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CSCore
An advanced audio library, written in C#. Provides tons of features
An advanced audio library, written in C#. Provides tons of features. From playing/recording audio to decoding/encoding audio streams/files to processing audio data in real-time (e.g. applying custom effects during playback, creating visualizations). The possibilities are nearly unlimited. CSCore is a free .NET audio library which is completely written in C#. -
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WebDSP
A client-side signal processing library
WebDSP is a browser-based digital signal processing toolkit that leverages WebAssembly to perform high-performance audio and signal processing tasks directly within web applications. It is designed to bring real-time DSP capabilities to the browser, allowing developers to implement audio effects, filters, and transformations with low latency and high efficiency. The project provides a set of DSP primitives that can be composed into more complex processing pipelines, making it suitable for applications such as audio synthesis, analysis, and interactive sound design. ... -
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aeneas
Automagically synchronize audio and text (aka forced alignment)
aeneas is a Python/C library and a set of tools to automagically synchronize audio and text (aka forced alignment). aeneas automatically generates a synchronization map between a list of text fragments and an audio file containing the narration of the text. In computer science this task is known as (automatically computing a) forced alignment. -
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GTK+ IOStream
GTK+iostream, Data plots, ORB, Neural Networks, WSOLA
...You can also plot like so : Plot figure; figure.plot(x, y, x.size()); Plotting utilises GtkDataBox.sf.net ORB utilises www.zeroc.com Feed forward neural networks with different activation functions. Audio playback/recording and digital signal processing (DSP) - utilising Jack www.jackaudio.org -
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Distant Speech Recognition
Beamforming and Speech Recognition Toolkit
BTK contains C++ and Python libraries that implement speech processing and microphone array techniques such as speech feature extraction, speech enhancement, speaker tracking, beamforming, dereverberation and echo cancellation algorithms. The Millennium ASR provides C++ and python libraries for automatic speech recognition. The Millennium ASR implements a weighted finite state transducer (WFST) decoder, training and adaptation methods. These toolkits are meant for facilitating research and... -
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Icarus Scene Engine
3D .NET engine for Windows, Linux, Mac
Icarus Scene Engine is a cross-platform, CLS-compliant 3D framework, integrating open source APIs into a cohesive all-open-source, all .NET solution. For Windows, MacOSX & Linux. Uses OpenTK, OpenGL, OpenAL APIs for graphics and audio. Runs on Mono/.NET. Links to FFMpeg for streaming video, and FreeType as a temporary conversion mod to bring fonts into a native XML format. Binary builds of Icarus will be infrequent, roughly every 18 months or so, the GIT code updates regularly on a... -
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videoLat
Measure video delays
VideoLat helps you measure latency of audio and video processing pipelines such as conferencing tools. NOTE: development of videoLat has moved to github. See http://www.videolat.org for details. -
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EZAudio
An iOS and macOS audio visualization framework
An iOS and macOS audio visualization framework built upon Core Audio useful for anyone doing real-time, low-latency audio processing and visualizations. I've designed six audio components and two interface components to allow you to immediately get your hands dirty recording, playing, and visualizing audio data. These components simply plug into each other and build on top of the high-performance, low-latency AudioUnits API and give you an easy to use API written in Objective-C instead of pure C. ... -
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MARF is a general cross-platform framework with a collection of algorithms for audio (voice, speech, and sound) and natural language text analysis and recognition along with sample applications (identification, NLP, etc.) of its use, implemented in Java.
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JUCE is a C++ toolkit for building cross-platform applications on PC/Mac/Linux/iOS/Android. Particularly good for complex, customised GUIs and audio/midi processing, it also includes a vast range of classes to help with all your day-to-day programming tasks.
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Marsyas (Music Analysis, Retrieval and Synthesis for Audio Signals) is a framework for developing systems for audio processing. It provides an general architecture for connecting audio, soundfiles, signal processing blocks and machine learning. Source code at SF is outdated! Marsyas is now hosted at GitHub: https://github.com/marsyas/marsyas Downloads are now provided at Bintray: https://bintray.com/marsyas
