Search Results for "audio signal processing"
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Showing 739 open source projects for "audio signal processing"
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Clock Signal
A latency-hating emulator of: the Acorn Electron, BBC Micro
Clock Signal also known as Clock Signal, is a highly sophisticated multi-system emulator designed with a strong emphasis on minimizing latency and maximizing authenticity in signal reproduction rather than relying on post-processing shortcuts. Its defining philosophy is to make emulation “invisible” to the user, meaning software can be launched directly without requiring manual configuration of machines, disks, or hardware profiles. -
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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. ... -
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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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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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Qwen2-Audio
Repo of Qwen2-Audio chat & pretrained large audio language model
Qwen2-Audio is a large audio-language model by Alibaba Cloud, part of the Qwen series. It is trained to accept various audio signal inputs (including speech, sounds, etc.) and perform both voice chat and audio analysis, producing textual responses. It supports two major modes: Voice Chat (interactive voice only input) and Audio Analysis (audio + text instructions), with both base and instruction-tuned models. -
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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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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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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. ... -
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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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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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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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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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SigLens
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Siglens is an open-source signal analysis toolkit designed for processing and visualizing time-series data, commonly used in scientific and engineering applications. -
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AERIS-10
Open-source, low-cost 10.5 GHz PLFM phased array RADAR system
AERIS-10 appears to be a specialized signal processing and radar analysis project focused on pulse linear frequency modulation techniques, which are commonly used in modern radar systems for high-resolution detection and ranging. The project likely implements algorithms for generating, analyzing, and visualizing chirp-based radar signals, enabling experimentation with time-frequency characteristics and signal reconstruction. -
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WhisperJAV
Uses Qwen3-ASR, local LLM, Whisper, TEN-VAD
WhisperJAV is an open-source speech transcription pipeline designed specifically for generating subtitles for Japanese adult video content. The project addresses challenges that standard speech recognition models face when transcribing this type of audio, which often includes low signal-to-noise ratios and large numbers of non-verbal vocalizations. Traditional automatic speech recognition systems can misinterpret these sounds as words, leading to inaccurate transcripts. WhisperJAV introduces a specialized pipeline that separates text generation from timestamp alignment, allowing the system to generate transcripts and then align them with audio using forced alignment techniques. ... -
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AudioMuse-AI
AudioMuse-AI is an Open Source Dockerized environment
AudioMuse-AI is an open-source system designed to automatically generate playlists and analyze music libraries using artificial intelligence and audio signal processing techniques. The platform runs locally in a Dockerized environment and performs detailed sonic analysis on audio files to understand characteristics such as tempo, mood, and acoustic similarity. By analyzing the underlying audio content rather than relying on external metadata services, the system can organize large personal music libraries and generate curated playlists for different moods or listening contexts. ... -
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EEGLAB
EEGLAB is an open source signal processing environment
...It incorporates powerful tools for data import, preprocessing, independent component analysis (ICA), time-frequency analysis, artifact rejection, and visualization—all within a GUI framework that also supports scripting and plugin extensions. EEGLAB is an open source signal processing environment for electrophysiological signals running on Matlab and Octave (command line only for Octave). This folder contains original Matlab functions from the EEGLAB (formerly ICA/EEG) Matlab toolbox, all released under the Gnu public license (see eeglablicence.txt). See the EEGLAB tutorial and reference paper (URLs given below) for more information. ... -
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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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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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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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DSPSR is a high-performance, object-oriented, digital signal processing library for radio pulsar astronomy. It implements an extensive range of algorithms and features, and can read data from most observatories, instruments, and file formats.
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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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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.
