Search Results for "audio linux" - Page 3
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3890 projects for "audio linux" with 1 filter applied:
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MongoDB Atlas runs apps anywhereMongoDB Atlas gives you the freedom to build and run modern applications anywhere—across AWS, Azure, and Google Cloud. With global availability in over 115 regions, Atlas lets you deploy close to your users, meet compliance needs, and scale with confidence across any geography.
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Train ML Models With SQL You Already KnowBuild and deploy ML models using familiar SQL. Automate data prep with built-in Gemini. Query 1 TB and store 10 GB free monthly.
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LiveAvatar
Streaming Real-time Audio-Driven Avatar Generation
LiveAvatar is an open-source research and implementation project that provides a unified framework for real-time, streaming, interactive avatar video generation driven by audio and other control signals. It implements techniques from state-of-the-art diffusion-based avatar modeling to support infinite-length continuous video generation with low latency, enabling interactive AI avatars that maintain continuity and realism over extended sessions. The project co-designs algorithms and system... -
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EasyVoice
Open source text-to-speech tool, supports extra-long text
easyVoice is an open-source text-to-speech platform aimed at turning long-form text and novels into high-quality audio, with a strong focus on usability and scalability. It provides a web interface where users can paste or upload large texts and generate speech and subtitles in a single workflow, even for works exceeding 100,000 characters. The system supports multi-role voice acting, letting users assign different neural voices to different characters or narrative roles and configure... -
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OuteTTS
Interface for OuteTTS models
OuteTTS is an interface library for running OuteTTS text-to-speech models across a range of backends, making it easier to deploy the same model on different hardware and runtimes. It provides a high-level Interface API that wraps model configuration, speaker handling, and audio generation so you can focus on integrating speech into your application rather than wiring up low-level engines. The project supports multiple backends including llama.cpp (Python bindings and server), Hugging Face... -
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Claude Code Video Vision
Give Claude the ability to watch and understand videos
Claude Video Vision is a plugin designed for Claude Code that enables large language models to process and understand video content by transforming it into multimodal inputs the model can reason over. Instead of attempting to directly interpret raw video streams, the system extracts key frames using tools like ffmpeg and processes audio through transcription engines, converting both visual and auditory signals into structured inputs for the model. The result is a perception layer that feeds... -
Go from Code to Production URL in SecondsSkip the Kubernetes configs. Cloud Run handles HTTPS, scaling, and infrastructure automatically. Two million requests free per month.
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VoxCPM
TTS for Context-Aware Speech Generation and True-to-Life Voice Cloning
VoxCPM is a tokenizer-free text-to-speech system that models speech in a continuous space, aiming for extremely realistic, context-aware synthesis and true-to-life zero-shot voice cloning. Instead of converting speech into discrete tokens, it uses an end-to-end diffusion-autoregressive architecture built on the MiniCPM-4 backbone, combining hierarchical language modeling, finite scalar quantization (FSQ), and local Diffusion Transformers. This design helps decouple semantic and acoustic... -
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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. It is designed to help... -
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whatsapp-web.js
WhatsApp library for NodeJS that connects through the browser app
A WhatsApp client library for NodeJS that connects through the WhatsApp Web browser app. Programmatically control WhatsApp whether you're running user or business accounts. It uses Puppeteer to run a real instance of Whatsapp Web to avoid getting blocked. Programmatically control WhatsApp whether you're running user or business accounts. Whatsapp-web.js connects to an official version of WhatsApp Web under the hood, reducing ban risks. The object-oriented approach makes it easy to get... -
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ffmpeg_develop_doc
2023, the latest audio and video learning materials, projects
ffmpeg_develop_doc is a curated repository that aggregates a comprehensive collection of learning resources related to FFmpeg and multimedia development. It includes command references, technical articles, academic papers, tutorials, and example projects covering audio and video processing concepts. The repository is structured as a knowledge base, offering materials on encoding, decoding, streaming protocols, and real-time media systems. It also contains interview preparation resources and... -
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TTS WebUI
A single Gradio + React WebUI with extensions for ACE-Step
TTS-WebUI is a unified Gradio + React web interface that brings together a large ecosystem of text-to-speech, voice conversion, and audio generation models under a single UI. It supports a wide range of models such as Bark, MusicGen, Tortoise, RVC, StyleTTS2, ParlerTTS, CosyVoice, XTTSv2, Stable Audio, SeamlessM4T, and many others, exposing them as interchangeable backends for speech and music synthesis. The project provides an installer that sets up Conda, Python environments, and all... -
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edge-tts
Use Microsoft Edge's online text-to-speech service from Python
edge-tts is a Python module and command-line tool that gives you direct access to Microsoft Edge’s online text-to-speech service without needing the Edge browser, Windows, or any API key. It wraps the same cloud voices used by Edge, exposing them through a simple CLI (edge-tts, edge-playback) and a Python API, so you can script high-quality speech generation in your own applications. The tool lets you list available voices, specify locale and voice name, and generate audio files in common... -
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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... -
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Markdownify MCP Server
Convert files and web content into clean, usable Markdown easily
Markdownify MCP is a Model Context Protocol server that converts many types of files and web content into clean Markdown. It supports formats such as PDFs, images, audio with transcription, DOCX, XLSX, and PPTX, along with web sources like YouTube transcripts, Bing results, and general webpages. Markdownify MCP is designed to simplify content extraction and make data easier to read, share, and reuse in structured workflows. Developers can install dependencies, build, and run the server... -
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TADA
Open Source Speech Language Model
TADA is an open-source speech-language modeling framework designed to unify spoken audio and text representations within a single generative architecture. The system focuses on aligning speech and text streams using a dual-alignment mechanism that synchronizes the acoustic signal with its textual representation. By modeling both modalities together, the framework allows developers to build systems capable of generating, understanding, and transforming speech and language simultaneously. This... -
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BlogWizard
Generate blog articles from video or audio
BlogWizard is a demo/utility project built on top of Groq’s LLM infrastructure that converts video or audio content into well-structured blog posts, enabling creators to repurpose multimedia content into text — useful for SEO, accessibility, or reaching audiences that prefer reading. The tool uses transcription (e.g. via Whisper) to extract text from audio/video, then runs an LLM-based generation pipeline to transform that content into coherent, readable blog-format posts — with sections,... -
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HY-World 1.5
A Systematic Framework for Interactive World Modeling
HY-WorldPlay is a Hunyuan AI project focusing on immersive multimodal content generation and interaction within virtual worlds or simulated environments. It aims to empower AI agents with the capability to both understand and generate multimedia content — including text, audio, image, and potentially 3D or game-world elements — enabling lifelike dialogue, environmental interpretations, and responsive world behavior. The platform targets use cases in digital entertainment, game worlds,... -
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Orpheus TTS
Towards Human-Sounding Speech
Orpheus TTS is a state-of-the-art open-source text-to-speech system built on a Llama-3B backbone, treating speech synthesis as a large language model problem instead of a traditional TTS pipeline. It is designed to produce human-like speech with natural intonation, emotion, and rhythm, targeting quality comparable to or better than many closed-source systems. The project ships both pretrained and finetuned English models, as well as a family of multilingual models released as a research... -
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PersonaPlex
PersonaPlex code
PersonaPlex is an open-source real-time conversational speech AI model that goes beyond traditional text chat by providing full-duplex speech-to-speech interaction, meaning it can listen and talk at the same time instead of waiting for you to finish speaking before responding. This architectural approach eliminates awkward pauses and makes conversations feel much more human-like, with natural behaviors such as overlapping speech, interruptions, and fluent turn-taking, traits that traditional... -
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HunyuanCustom
Multimodal-Driven Architecture for Customized Video Generation
HunyuanCustom is a multimodal video customization framework by Tencent Hunyuan, aimed at generating customized videos featuring particular subjects (people, characters) under flexible conditions, while maintaining subject/identity consistency. It supports conditioning via image, audio, video, and text, and can perform subject replacement in videos, generate avatars speaking given audio, or combine multiple subject images. The architecture builds on HunyuanVideo, with added modules for... -
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Descent 3
Descent 3 by Outrage Entertainment
...It provides the full C and C++ engine source code, including the historically significant “1.5” patch that was previously created by developers and later stabilized by fans. The codebase covers the game’s rendering, physics, audio, networking, tools, and editor components, allowing enthusiasts to build, run, and modify the classic 6-degrees-of-freedom space shooter on modern systems. To actually play the game, users must supply their own original game assets, following instructions in the repository’s usage documentation. The project uses CMake and related modern tooling for cross-platform builds, with support for Linux and Windows among other environments. ... -
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OmniTools
Self-hosted collection of powerful web-based tools for everyday tasks
OmniTools is a self-hosted web application that bundles a large collection of everyday utilities into a single clean interface you can run on your own infrastructure. 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... -
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MARS5
MARS5 speech model (TTS) from CAMB.AI
MARS5-TTS is CAMB.AI’s open-source English speech model designed for high-quality text-to-speech and voice emulation. It uses a two-stage architecture that combines an autoregressive (AR) model with a non-autoregressive (NAR) model, giving it both expressiveness and speed. The model is built to handle prosodically challenging content such as sports commentary, anime dialogue, and other high-energy or highly varied speech patterns with realistic rhythm and intonation. To control speaker... -
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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... -
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SDL
Simple DirectMedia Layer
SDL (Simple DirectMedia Layer) is a cross-platform multimedia development library designed to provide low-level access to hardware components such as graphics, audio, input devices, and system resources, making it a foundational tool for building games, emulators, and interactive applications. It abstracts platform-specific functionality into a consistent API, allowing developers to write code once and deploy it across multiple operating systems including Windows, macOS, Linux, iOS, and Android. ... -
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comfyui-mixlab-nodes
Workflow and speech recognition app
comfyui-mixlab-nodes is a large collection of custom nodes for ComfyUI that turns workflows into interactive apps and adds real-time multimedia, LLM, and TTS capabilities. It introduces a “Workflow-to-APP” concept, where a ComfyUI graph can be transformed into a Web App through an AppInfo node, complete with categories, batch prompts, and editable configurations. The project also brings Real-time Design features like screen capture and floating video nodes, enabling creative pipelines that... -
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projectM
Cross-platform Music Visualization Library
Cross-platform Music Visualization Library. Open-source and Milkdrop-compatible. Experience psychedelic and mesmerizing visuals by transforming music into equations that render a limitless array of user-contributed visualizations. projectM is an open-source project that reimplements the esteemed Winamp Milkdrop by Geiss in a more modern, cross-platform reusable library. Its purpose in life is to read an audio input and to produce mesmerizing visuals, detecting tempo, and rendering advanced...
