Search Results for "audio processing"
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200 projects for "audio processing" with 1 filter applied:
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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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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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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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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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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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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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NovaSR
A lightning fast audio upsampler
...NovaSR is especially valuable for post-processing tasks in speech enhancement, TTS pipelines, and dataset restoration where low sampling rates degrade perceived audio clarity; the minimal model size also makes it suitable for edge and embedded use cases where memory is at a premium. Its performance can reach thousands of times realtime on modern GPUs, allowing massive audio batches to be processed with negligible compute overhead. -
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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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Claude Code Video Vision
Give Claude the ability to watch and understand videos
...It supports multiple backends for audio processing, including local and cloud-based options, enabling flexible deployment depending on privacy or performance requirements. -
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AI-Media2Doc
AI tool converting video/audio into structured documents instantly
AI-Media2Doc is a web-based application that uses large language models to convert video and audio content into structured, readable documents in a single workflow. It is designed to transform multimedia inputs into formats such as knowledge notes, summaries, mind maps, and social-style articles, making content easier to review and reuse. AI-Media2Doc emphasizes privacy by processing media locally in the browser using WebAssembly-based ffmpeg, ensuring that original video files are not uploaded externally. ... -
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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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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... -
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Hugging Face - Speech To Speech
Open speech-to-speech models and pipelines by Hugging Face toolkit AI
This project from Hugging Face focuses on enabling direct speech-to-speech processing using modern machine learning models. It provides tools and reference implementations that allow audio input to be transformed into audio output without requiring an intermediate text representation. Hugging Face - Speech To Speech builds on recent advances in speech modeling, combining components such as speech recognition, translation, and synthesis into unified pipelines. ... -
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ReClip
Download videos from almost any website
...Users can paste multiple URLs at once, select output formats such as MP4 or MP3, and choose quality settings before downloading. The system also includes features like automatic URL deduplication and batch processing to improve usability. -
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OmniTools
Self-hosted collection of powerful web-based tools for everyday tasks
...It’s designed to replace the random assortment of “free online tools” people use for quick tasks, while avoiding ads, tracking, and the need to upload sensitive files to unknown servers. A key design choice is that file processing happens entirely on the client side, meaning your data stays in your browser instead of being sent to the backend. The tool catalog spans both technical and non-technical needs, including image, video, audio, PDF, text, date/time, math, and data format utilities like JSON/CSV/XML helpers. It’s also packaged for straightforward self-hosting, with a lightweight Docker image and simple run commands, so it can be deployed quickly on a homelab or internal network. -
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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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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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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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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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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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AudioBookConverter
Improved AudioBookConverter based on freeipodsoftware release
AudioBookConverter is a lightweight desktop application designed to convert and organize audiobook files into optimized formats such as M4B for playback on modern devices. It allows users to combine multiple audio files into a single audiobook while preserving chapters and metadata for seamless listening. The software supports a wide range of input formats including MP3, FLAC, and AAC, and provides flexible output options for different devices. It includes intelligent artwork handling,... -
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LTX-Video
Official repository for LTX-Video
LTX-Video is a sophisticated multimedia processing framework from Lightricks designed to handle high-quality video editing, compositing, and transformation tasks with performance and scalability. It provides runtime components that efficiently decode, encode, and manipulate video streams, frame buffers, and audio tracks while exposing a rich API for building customized editing features like transitions, effects, color grading, and keyframe automation. -
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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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Verticals v3
Automated YouTube Shorts pipeline
...The pipeline emphasizes automation, allowing users to produce short-form content at scale with minimal manual intervention. It integrates FFmpeg and other media processing tools to handle video transformations, resizing, and encoding. The system also supports adding overlays, captions, and audio enhancements to improve engagement. Designed for creators and developers, it enables repeatable workflows for generating social media content efficiently. Its modular structure allows customization of each stage in the pipeline, making it adaptable to different content strategies. -
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Scriberr
Self-hosted AI audio transcription
Scriberr is a self-hosted AI-powered transcription platform designed to convert audio and video into highly accurate text while prioritizing privacy and local processing. Unlike cloud-based transcription services, Scriberr runs entirely on the user’s machine, ensuring that sensitive recordings are never sent to third-party servers and remain fully under user control. It leverages modern speech recognition models such as Whisper and other advanced architectures to deliver precise transcripts with word-level timing and speaker identification. ...
