Search Results for "audio"
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1880 projects for "audio" 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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NeuralNote
Audio Plugin for Audio to MIDI transcription using deep learning
NeuralNote is an open-source audio software tool designed to convert recorded audio into MIDI data using modern machine learning techniques. The software functions as an audio plugin that can be used inside digital audio workstations as well as a standalone application for music production and analysis. Its main purpose is to perform audio-to-MIDI transcription, allowing musicians to record a performance and automatically transform it into editable MIDI notes. ... -
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LTX-2.3
Official Python inference and LoRA trainer package
LTX-2.3 is an open-source multimodal artificial intelligence foundation model developed by Lightricks for generating synchronized video and audio from prompts or other inputs. Unlike most earlier video generation systems that only produced silent clips, LTX-2 combines video and audio generation in a unified architecture capable of producing coherent audiovisual scenes. The model uses a diffusion-transformer-based architecture designed to generate high-fidelity visual frames while simultaneously producing corresponding audio elements such as speech, music, ambient sound, or effects. ... -
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AudioCraft
Audiocraft is a library for audio processing and generation
AudioCraft is a PyTorch library for text-to-audio and text-to-music generation, packaging research models and tooling for training and inference. It includes MusicGen for music generation conditioned on text (and optionally melody) and AudioGen for text-conditioned sound effects and environmental audio. Both models operate over discrete audio tokens produced by a neural codec (EnCodec), which acts like a tokenizer for waveforms and enables efficient sequence modeling. ... -
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MusicPlayer2
Audio player that can play common audio formats
MusicPlayer2 is a simple music-player application (or prototype) implemented in — presumably — a web or desktop environment, intended to give users a clean, functional interface for managing and playing audio files. The project likely implements basic playlist management, playback controls (play, pause, skip), and possibly UI features to browse or organize music. Because many smaller music-player projects aim for simplicity, MusicPlayer2 may focus on providing a lightweight, minimal-dependency audio player compared to larger, heavy multimedia suites. ... -
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NovaSR
A lightning fast audio upsampler
NovaSR is an extremely lightweight and high-performance audio upsampling model that transforms low-quality 16 kHz audio into clearer, high-fidelity 48 kHz audio with remarkable speed and efficiency. At only about 50 KB in size, the model is orders of magnitude smaller than typical audio super-resolution networks, yet it achieves high quality and realtime performance thanks to its compact architecture and efficient convolutional design. -
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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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HLS.js
HLS.js is a JavaScript library that plays HLS in browsers
HLS.js is a JavaScript library that implements an HTTP Live Streaming client. It relies on HTML5 video and MediaSource Extensions for playback. It works by transmuxing MPEG-2 Transport Stream and AAC/MP3 streams into ISO BMFF (MP4) fragments. Transmuxing is performed asynchronously using a Web Worker when available in the browser. HLS.js also supports HLS + fmp4, as announced during WWDC2016. HLS.js works directly on top of a standard HTML<video> element. HLS.js is written in ECMAScript6... -
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HeartMuLa
A Family of Open Sourced Music Foundation Models
...The project also includes HeartCodec, a music codec optimized for high reconstruction fidelity, enabling efficient tokenization and reconstruction workflows that are critical for training and generation pipelines. For text extraction from audio, it provides HeartTranscriptor, a Whisper-based model tuned specifically for lyrics transcription, which helps bridge generated or recorded audio back into structured text. It also introduces HeartCLAP, which aligns audio and text into a shared embedding space. -
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SFML
Simple and Fast Multimedia Library
SFML provides a simple interface to the various components of your PC, to ease the development of games and multimedia applications. It is composed of five modules: system, window, graphics, audio and network. Discover their features more in detail in the tutorials and the API documentation. With SFML, your application can compile and run out of the box on the most common operating systems: Windows, Linux, macOS and soon Android & iOS. Pre-compiled SDKs for your favorite OS are available on the download page. SFML has official bindings for the C and .Net languages. ... -
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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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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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EeveeSpotify
A tweak to enhance Spotify experience
EeveeSpotifyReborn is an unofficial modification for the Spotify mobile application that alters client-side behavior to unlock premium-like features without requiring a paid subscription. It operates by injecting changes into the Spotify app, making it interpret the user account as having premium access and enabling functionalities that are normally restricted. The project was developed through reverse engineering techniques, including analyzing application behavior and intercepting requests... -
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SenseVoice
Multilingual speech recognition and audio understanding model
SenseVoice is a speech foundation model designed to perform multiple voice understanding tasks from audio input. It provides capabilities such as automatic speech recognition, spoken language identification, speech emotion recognition, and audio event detection within a single system. SenseVoice is trained on more than 400,000 hours of speech data and supports over 50 languages for multilingual recognition tasks. It is built to achieve high transcription accuracy while maintaining efficient inference performance. ... -
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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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OpenAI.fm
Code for openai.fm, a demo for the OpenAI Speech API
OpenAI.fm is an official interactive demo application built to showcase the OpenAI Speech API and its advanced text-to-speech capabilities, providing developers and creators with a hands-on web interface to convert text into high-quality, customizable audio using state-of-the-art TTS models. Developed using Next.js and the OpenAI Speech API, this demo illustrates how the latest neural voice models can produce natural, expressive speech with adjustable styles and voices, highlighting features like emotional range, tone, and real-time playback. Users can experiment with different input text and voice options directly in their browser, gaining a sense of how high-fidelity AI audio can be integrated into applications ranging from podcasts and narration to accessibility tools and interactive agents. ... -
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HunyuanVideo-Avatar
Tencent Hunyuan Multimodal diffusion transformer (MM-DiT) model
HunyuanVideo-Avatar is a multimodal diffusion transformer (MM-DiT) model by Tencent Hunyuan for animating static avatar images into dynamic, emotion-controllable, and multi-character dialogue videos, conditioned on audio. It addresses challenges of motion realism, identity consistency, and emotional alignment. Innovations include a character image injection module, an Audio Emotion Module for transferring emotion cues, and a Face-Aware Audio Adapter to isolate audio effects on faces, enabling multiple characters to be animated in a scene. Character image injection module for better consistency between training and inference conditioning. ... -
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AudioNotes
Extract audio and video content and organize it into a Markdown note
AudioNotes is an application (or proof-of-concept) that likely combines audio recording or playback with note-taking or annotation functionality — enabling users to record voice or audio and attach textual or timestamped notes, making it ideal for lectures, interviews, meetings, or personal memos. Such a tool offers a more expressive and flexible way to capture and revisit information: instead of just typed notes or raw audio, users get both audio context and structured notes. ... -
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Riffusion App
Stable diffusion for real-time music generation (web app)
Riffusion App Hobby is an open-source interactive web application that enables real-time music generation using stable diffusion models adapted for audio synthesis. Unlike traditional music generation tools, it treats audio as spectrogram images and applies diffusion techniques to generate continuous sound transitions, allowing users to create evolving musical loops and compositions. The application is built with modern web technologies including Next.js, React, and three.js, providing a responsive and visually engaging interface for experimentation. ... -
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Kaset
The missing YouTube Music macOS app
Kaset is a social audio platform framework that allows users to host, share, and interact with audio content in community-oriented spaces, combining elements of podcasting, voice rooms, and feedback-driven discovery. It provides an interface where creators can upload episodes, host live or scheduled voice sessions, and cultivate listener communities through comments, reactions, and follow systems. -
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Oboe
Oboe is a C++ library that makes it easy to build high-performance
...Example apps and utilities demonstrate best practices for thread priority, audio mixing, and underrun handling. By aligning with the native audio stack, oboe enables professional music apps, real-time communications, and games to achieve consistent audio performance across a wide range of devices. -
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S&box
s&box is a modern game engine, built on Valve's Source 2
...Built on a cutting-edge game engine, s&box allows creators to prototype, build, and share interactive game modes, tools, and environments using C#, JavaScript, and visual scripting, promoting accessible content creation for developers of varying skill levels. The platform emphasizes multiplayer and community experiences, giving creators direct control over networking, physics, rendering, and audio without needing to build those systems from scratch. With real-time recompilation and fast iteration loops, developers can see changes instantly, speeding up the creative process dramatically compared to traditional engines. The sbox ecosystem fosters collaboration, sharing, and extension through packages and community content, enabling a vibrant marketplace of user-created mods and assets. -
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WhisperLive
A nearly-live implementation of OpenAI's Whisper
WhisperLive is a “nearly live” implementation of OpenAI’s Whisper model focused on real-time transcription. It runs as a server–client system in which the server hosts a Whisper backend and clients stream audio to be transcribed with very low delay. The project supports multiple inference backends, including Faster-Whisper, NVIDIA TensorRT, and OpenVINO, allowing you to target GPUs and different CPU architectures efficiently. It can handle microphone input, pre-recorded audio files, and network streams such as RTSP and HLS, making it flexible for live events, monitoring, or accessibility workflows. ... -
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SoniTranslate
Synchronized Translation for Videos
SoniTranslate is a video translation and dubbing system that produces synchronized target-language audio tracks for existing video content. It provides a web UI built with Gradio, allowing users to upload a video, choose source and target languages, and then run a pipeline that handles transcription, translation and re-synthesis of speech. Under the hood, it uses advanced speech and diarization models to separate speakers, align audio with timecodes and respect subtitle timing, which lets the generated dub track stay in sync with the original video structure. ...
