16 مستودعات
Captures signals from input devices and manages audio output through integrated hardware interfaces.
Distinct from Audio Input Selectors: None of the candidates provided a direct match for general input/audio management in an OS context.
Explore 16 awesome GitHub repositories matching operating systems & systems programming · Input and Audio Management. Refine with filters or upvote what's useful.
FFmpeg is a cross-platform multimedia framework designed for the recording, conversion, and streaming of audio and video content. It functions as a comprehensive toolkit that provides both a command-line utility for direct media manipulation and a collection of low-level libraries for integration into custom applications. At its core, the project utilizes a packet-based stream engine and a format-agnostic abstraction layer to handle diverse media standards, containers, and network protocols. The framework distinguishes itself through a modular, graph-based filter execution model that allows f
Ingests live audio and video data from hardware peripherals like cameras and microphones.
Redox is a POSIX-compliant, microkernel-based operating system written entirely in Rust. By utilizing a memory-safe language for the kernel and all system components, the project eliminates common vulnerabilities such as buffer overflows and use-after-free errors. Its architecture relies on a minimal kernel that manages only essential hardware and process isolation, delegating all other system services to unprivileged user-space processes. The system distinguishes itself through a modular design where hardware drivers and system services run as independent user-space daemons, allowing them to
Manages input device signals and audio output through integrated hardware interfaces.
Hammerspoon is a programmable automation engine for macOS that enables deep system-level control through a Lua scripting environment. By bridging high-level scripts with native Objective-C APIs, it allows users to interact with the operating system's accessibility tree, intercept hardware input streams, and manage the lifecycle of running applications. The project distinguishes itself through an event-driven architecture that registers asynchronous hooks for system notifications and hardware events. This allows for real-time automation, such as remapping keyboard and mouse inputs, managing wi
Manages and selects system audio input and output sources for routing control.
SFML is a cross-platform C++ multimedia library designed for building 2D games and interactive applications. It provides a unified interface for hardware-accelerated graphics, audio playback, and window management, allowing developers to create visual software that functions consistently across different operating systems. The library abstracts native windowing and graphics APIs, enabling the use of object-oriented primitives to render shapes, sprites, and text. It includes a dedicated audio engine that supports sound effects and music with controls for volume, pitch, and spatial positioning,
Manages audio playback including volume, pitch, and spatial positioning for immersive sound experiences.
EarTrumpet is a desktop utility for the Windows operating system that provides centralized control over system-wide audio. It functions as a taskbar-based interface for managing playback devices, monitoring active audio streams, and adjusting volume levels across the desktop environment. The application enables granular control by allowing users to adjust volume levels for individual running processes independently. It also supports multi-device audio routing, which permits the assignment of specific hardware output destinations to individual applications. These capabilities are facilitated t
Centralizes system sound settings and output device management to improve the Windows audio experience.
The Android NDK samples provide a comprehensive collection of code examples demonstrating how to integrate C and C++ native code into Android applications. This repository serves as a practical guide for developers utilizing the Android Native Development Kit to implement performance-critical application components that require direct hardware access and low-level system interaction. The project highlights the use of the Java Native Interface to bridge managed code with native modules, enabling cross-language function calls and efficient data exchange. It demonstrates how to manage native act
Configures recording parameters and hardware usage to reduce the delay between capturing and processing audio.
NoiseTorch is a cross-platform audio processor and real-time noise filter designed to suppress ambient sound from audio streams. It functions as a virtual microphone noise suppressor and routing tool, capturing system audio sources and directing filtered signals into virtual input or output devices. The application uses a recurrent neural network to distinguish between human speech and ambient noise. It provides a virtual denoising microphone that removes background noise from a selected input, alongside tools for filtering audio output streams. The system includes capabilities for audio dev
Enables the selection and configuration of specific microphones or monitor sources for the noise filter.
LMMS is a digital audio workstation and MIDI sequencer designed for composing, arranging, and mixing music. It functions as a comprehensive production environment that integrates a MIDI sequencer, a sample-based synthesizer, and an audio mixing console. The project distinguishes itself through a versatile synthesis engine that includes additive synthesis, wavetable generation, and emulations of vintage hardware such as NES audio and FM chips. It also serves as a VST plugin host, allowing for the integration of third-party virtual instruments and audio effects via a standardized interface. Be
Provides controls to adjust audio buffer sizes, balancing output delay against the risk of playback artifacts.
This project is a framework for developing multimodal AI agents that function as programmable participants in real-time communication rooms. It enables the construction of agents that can see, hear, and speak by integrating speech-to-text, large language models, and text-to-speech pipelines to facilitate low-latency, natural conversations. The system is distinguished by its advanced orchestration of real-time media and conversational flow, including support for full-duplex speech, preemptive response generation, and sophisticated interruption management. It further differentiates itself throu
Dynamically toggles audio input and output to support hybrid or text-only conversational sessions.
JUCE is a comprehensive C++ audio framework and digital signal processing library used to build cross-platform audio applications, audio plug-ins, and high-performance user interfaces. It serves as a development kit for creating audio processors compatible with industry-standard plugin formats for digital audio workstations, as well as a tool for MIDI and Open Sound Control communication between musical hardware and software. The framework is distinguished by its ability to maintain a single codebase for native desktop and mobile applications across multiple operating systems. It provides a f
Handles numeric, boolean, and selectable inputs for plugins with support for value smoothing and state persistence.
Deej هو جسر من الأجهزة إلى البرامج ومدير صوت نظام مكتوب بلغة Go يقوم بتعيين منزلقات Arduino المادية إلى مستويات الصوت على Windows وLinux. يعمل كخلاط صوت عبر المنصات يسمح بالتحكم في مستوى الصوت الرئيسي، ومستويات الميكروفون، وصوت التطبيقات الفردية من خلال واجهة أجهزة خارجية. يتميز المشروع بتوفير نظام تعيين قائم على التكوين يربط عمليات البرامج وأجهزة الصوت المادية بمنزلقات مادية محددة. يتضمن منطقاً لعكس اتجاه المنزلق لتصحيح الأسلاك المادية المعكوسة ويدعم إعادة تحميل التكوين في وقت التشغيل لتحديث التعيينات دون إعادة تشغيل الخدمة الخلفية. يغطي البرنامج قدرات إدارة صوت واسعة، بما في ذلك القدرة على تجميع تطبيقات برمجية متعددة تحت قناة تحكم واحدة وتعديل مستوى الصوت الجماعي لجميع التطبيقات غير المعينة. يتعامل مع اتصالات الأجهزة القائمة على التسلسل لنقل البيانات من وحدة تحكم إلى الكمبيوتر عبر اتصال USB.
Provides a backend service for managing system audio streams and device volumes via external hardware inputs.
هذا المشروع هو إضافة WebRTC لـ Flutter توفر إطار عمل للاتصالات في الوقت الفعلي لتنفيذ بث الصوت والفيديو والبيانات من نظير إلى نظير. يعمل كجهاز بث وسائط عابر للمنصات ووحدة تحكم وسائط للأجهزة، مما يسمح للتطبيقات بإدارة كاميرات وميكروفونات الأجهزة عبر منصات الجوال، وسطح المكتب، والويب. يتضمن إطار العمل واجهة قناة بيانات مخصصة من نظير إلى نظير لتبادل حزم بيانات عشوائية بزمن انتقال منخفض. ويضمن الخصوصية وسلامة البيانات من خلال التشفير من طرف إلى طرف لجميع عمليات نقل الصوت والفيديو والبيانات. تغطي القدرات الواسعة إدارة تدفق الوسائط ومعالجة إشارات الصوت، بما في ذلك إلغاء الصدى وقمع الضوضاء. كما يدعم النظام مشاركة شاشة الجهاز، وتسجيل تدفق الوسائط، وتوجيه الشبكة عبر بروتوكولات STUN وTURN لتسهيل الاتصالات المباشرة بين الأقران.
Applies echo cancellation, noise suppression, and automatic gain control to enhance voice quality in real-time streams.
Oboe is a native C++ library designed for building high-performance, low-latency audio applications on Android. It serves as a unified wrapper and native API for managing audio streams, sample rates, and hardware routing across different Android operating system versions. The library provides a consistent interface by automatically selecting the most efficient audio backend at runtime, switching between AAudio and OpenSL ES to ensure the lowest possible latency. It enables exclusive-mode hardware access to bypass the system mixer and utilizes a high-priority asynchronous pull model for audio
Optimizes sample rates and burst sizes to minimize audio delay across different hardware levels.
VDO.Ninja is a low-latency peer-to-peer media routing service and video streaming platform designed to integrate remote audio and video feeds into professional production workflows. It functions as a WebRTC broadcast integration tool and studio controller, allowing for the direct transmission of high-definition media between publishers and viewers with minimal delay. The platform distinguishes itself through extensive protocol bridging, converting between WebRTC, WHIP, WHEP, SRT, and RTMP to ensure compatibility across diverse network environments and professional studio software. It includes
Optimizes packet timing and bypasses processing pipelines to minimize audio capture and transmission delay.
nih-plug is an audio plugin SDK and development framework providing a set of tools and traits for processing audio and MIDI data in a real-time safe environment. It functions as a cross-format plugin wrapper, allowing a single implementation to be exported into multiple industry-standard audio plugin formats, including VST, CLAP, and LV2. The project includes a retained-mode GUI framework for creating interactive user interfaces and parameter controls for audio processors. It also provides a real-time audio library that utilizes hardware acceleration and asynchronous task management to mainta
Provides declarative management of numeric and boolean parameters with built-in value smoothing.
This project is a comprehensive toolkit for on-device speech recognition, synthesis, and audio processing, specifically engineered for Apple Silicon. It provides a framework for building real-time, full-duplex voice agents that operate entirely offline, leveraging native hardware acceleration to maintain performance and privacy. By utilizing optimized machine learning models, the library enables local execution of complex audio tasks without reliance on external cloud services. The library distinguishes itself through its specialized focus on local, high-performance voice interaction. It incl
Processes audio input using noise reduction and voice activity detection to improve clarity.