Search Results for "audio analysis and dsp"
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Showing 171 open source projects for "audio analysis and dsp"
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Our computer monitoring software allows employees, field contractors, and freelancers to manually clock in when they begin working on an assignment. The application will take screenshots randomly or at set intervals, which allows employers to observe the work process. The application only tracks activity when the employee is clocked in. No spying, only transparency.
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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. It can be provided to deep learning networks for training and is used... -
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Competent Audio
Machine graph audio engine for computer games
... 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. Errors are reported using well-documented negative return codes, and CA attempts to handle expected errors gracefully. -
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LAME (Lame Aint an MP3 Encoder)
A high quality MP3 encoder
LAME is an educational tool to be used for learning about MP3 encoding. The goal of the LAME project is to improve the psycho acoustics, quality and speed of MP3 encoding. Note: we provide source code only! -
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This project delivers Linux support (USB driver and Tools) for a Digital Signal Processor (DSP) cards of the Signal Ranger series, see http://www.softdb.com/product_dsp.shtml for details) connected to a Linux host via USB.
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It protects your internal resources such as behind-the-firewall applications, teams, and devices.
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Sneedacity
Audio Editor
... in Python, Perl, or any language that supports named pipes. Nyquist Very powerful built-in scripting language that may also be used to create plug-ins. Editing multi-track editing with sample accuracy and arbitrary sample rates. Accessibility for VI users. Analysis and visualization tools to analyze audio or other signal data. -
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The Pure Data (short Pd) project on SourceForge unifies the extensions (or externals) written for the Pure Data Graphical Computer Music System written by Miller Puckette with contributions from many others.
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xoscope is a digital oscilloscope using input from a sound card or EsounD and/or a ProbeScope/osziFOX and will soon support Bitscope hardware. Includes 8 signal displays, variable time scale, math, memory, measurements, and file save/load.
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Snd
Snd is a sound editor
Snd is a sound editor. -
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Linux Studio Plugins Project
Linux Studio Plugins Project
LSP (Linux Studio Plugins) is a collection of open-source plugins currently compatible with LADSPA, LV2 and LinuxVST formats. Standalone plugins for JACK are provided since version 1.0.8. Experimental support of ARMv7 added since version 1.1.4 Experimental support of AArch64 added since version 1.1.9 Decomposition of modules and new UI introduced in 1.2.0 Added CLAP support in 1.2.5 Added VST3 support in 1.2.15 The basic idea is to fill the lack of good and useful plugins under the... -
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SPTK is a suite of speech signal processing tools for UNIX environments, e.g., LPC analysis, PARCOR analysis, LSP analysis, PARCOR synthesis filter, LSP synthesis filter, vector quantization techniques, and other extended versions of them.
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Source code from the Research Institute for Signals, Systems and Computational Intelligence http://fich.unl.edu.ar/sinc
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TAPClean is a Commodore tape preservation / restoration tool. It will check, repair, and remaster Commodore 64 and VIC 20 TAP or DC2N DMP files (tape images).
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SpectrumViewer for Windows
Continuous display of audio spectrum from PC input or wave file
Reads audio from Windows audio interface or wave file and creates a continuous spectrum display. Command line support for generating sine, square, triangle and sawtooth wave files. Uses biquad IIR digital filters for frequency detection, plus digital lowpass filters for output extraction. Highly configurable. 60+ FPS with suitable hardware. Manual or automatic gain control. -
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GXSM
Scanning Probe Microscopy Controller and Data Visualization Software
GXSM -- Gnome X Scanning Microscopy: A multi-channel image and vector-probe data acquisition and visualization system designed for SPM techniques (STM,AFM..), but also SPA-LEED/LEED/LEEM data analysis. A plug-in interface allows any user add-on data-processing and special hardware and instrument support. Latest: NC-AFM and related explorative methods as SQDM can be configured. High-Speed external PAC-PLL hardware option with digital DSP link. Based on several hardware options it supports... -
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lbat
lmms projects builder and tagger
Wrapper atop "lmms render" - keep rendered tracks up to date with lmms projects. Creates EBU R128 compliat tracks. -
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The Sound Description Interchange Format (SDIF) is an established standard for the interchange of sound descriptions and analysis data. This project provides libraries, SDIF (in C) and Easdif (in C++), tools, and wrappers to read and write SDIF files.
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strobeTuner
Strobe Tuner for Windows
Musical instrument strobe tuner: a real-time emulation -
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mp3guessenc
What's inside your mp3 files REALLY ?
mp3guessenc is based upon the original project by Naoki Shibata. This command line utility was born to detect the encoder used for a mpeg layer III file but now it can do more and scan any mpeg audio file (any layer) and print a LOT of useful information. Latest modifications include fixes, new features and code optimizations. -
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Win32 dll to normalize wav pcm files using very well known replaygain algorithm, it can be used as the first pass prior to mp3 encode, notice that BENormalize that is available in the project http://sourceforge.net/projects/rsppmp3/ gives a lot better results than ReplayGain, give it a try
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fmedia
fast media player/recorder/converter
fmedia is a fast asynchronous media player/recorder/converter for Windows, Linux and FreeBSD. Its goal is to provide smooth playback even if input device is very slow and unresponsive. The architecture allows to extend the functionality of the application in any way: adding a new audio input/output format, a new DSP filter or even a new GUI. fmedia is very small and fast: it has low CPU & memory consumption making it ideal to listen to music or process audio files while running... -
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NanoDSP Open Source DSP
Audio Enhancer for Windows and Embedded Platform
NanoDSP is designed for embedded operation and aims for low CPU load. Main functions 1. Bass amplification using a quadratic curve Generates odd and even harmonics, and uses human auditory psychology missing fundamentals to create the illusion of bass enhancement. By using a quadratic curve for the bass component, distortion is created and harmonics are generated to amplify the bass. 2. Separation of bass and Mid, treble using moving average By using the moving average, the CPU... -
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cmp-wave
Compare a pair of Wave files or directories of similarly names files
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DRC generates digital room correction FIR filters to be used within HiFi systems in conjunction with real time convolution engines like BruteFIR.
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Sonic Flow are C/C++ libraries for dataflow-oriented audio signal processing. Advanced features: feedback, multirate and hierarchical networks; signal processing block library. High-quality chorus/flanger, compressor/expander, param.EQ etc. example code.
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rePhase
A loudspeaker phase linearization, EQ and filtering tool
rePhase project has moved to https://rephase.org, please go there to download rephase and to get info, support and updates. rePhase is a FIR generation tool for building fully linear-phase active crossovers with arbitrary slopes. It also let you manually compensate for the phase shifts of your loudspeakers and existing crossovers, be it active or passive. Once generated, the FIR can be applied using any hardware (openDRC, miniSHARC, Najda, ...) or software (foobar, convolver, JRiver, ...),...