Search Results for "audio level meter"
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Showing 298 open source projects for "audio level meter"
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1
Step-Audio-EditX
LLM-based Reinforcement Learning audio edit model
Step-Audio-EditX is an open-source, 3 billion-parameter audio model from StepFun AI designed to make expressive and precise editing of speech and audio as easy as text editing. Rather than treating audio editing as low-level waveform manipulation, this model converts speech into a sequence of discrete “audio tokens” (via a dual-codebook tokenizer) — combining a linguistic token stream and a semantic (prosody/emotion/style) token stream — thereby abstracting audio editing into high-level token operations. ... -
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Audio Priority Bar
A native macOS menu bar app for managing audio device priorities
Audio Priority Bar is a lightweight macOS utility that gives users precise control over how audio output is prioritized across different apps and devices, filling a gap in the system audio stack that Apple doesn’t natively expose. Once installed, it places an always-accessible control in the menu bar that lets you assign priority levels to individual audio sources so that more important sounds (like alerts, calls, or music) can override or duck less important ones (like background noise or... -
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BlockTheSpot
Video, audio & banner adblock/skip for Spotify
...The project operates at a low level by modifying application components, which makes it effective but also susceptible to breaking when Spotify updates its software. It has gained popularity as a workaround for ad-supported tiers, though it requires manual installation and maintenance. -
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WhisperX
Automatic Speech Recognition with Word-level Timestamps
WhisperX is an advanced speech recognition system built on top of OpenAI’s Whisper model, designed to improve transcription accuracy and timing precision for long-form audio. It addresses key limitations of standard Whisper implementations by introducing voice activity detection and forced alignment techniques to produce word-level timestamps. The system enables batched inference, significantly increasing transcription speed while maintaining high accuracy. It is particularly effective for long recordings, where traditional approaches may suffer from drift, repetition, or misalignment. whisperx also supports speaker diarization, allowing identification of different speakers within a conversation. ... -
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5
JoyAI-Echo
Pushing the Frontier of Long Audio-Visual Generation
...JoyAI-Echo focuses on text-to-video and multi-shot long-video generation, while image-to-video support is not part of the current release scope. It is most useful for research and experimental video workflows that need synchronized audio, coherent characters, and editable story-level generation. -
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miniaudio
Audio playback and capture library written in C,
miniaudio is written in C with no dependencies except the standard library and should compile cleanly on all major compilers without the need to install any additional development packages. All major desktop and mobile platforms are supported. miniaudio gives you complete flexibility. With the low-level API, just initialize a connection to the device and send or receive raw audio data. The modular design of miniaudio allows you to use the low-level API without compromising your ability to make use of other features like the node graph and resource manager. miniaudio's node graph system gives you an easy way to set up advanced mixing and effect graphs. ... -
7
Boombox
Boombox is a simple streaming tool built on top of Membrane
Boombox is a high-level audio and video streaming tool built on top of the Membrane Framework. It is designed to make media transformations easier across common formats, containers, and streaming protocols. The project gives developers a simpler interface over lower-level multimedia pipelines, which can otherwise be complex to configure manually. Boombox is useful for Elixir projects that need to read, write, stream, transcode, or transform media without directly assembling every Membrane element. ... -
8
baru
A system monitor written in Rust and C
Baru gathers the information from /sys and /proc filesystems (filled by the kernel). Except for audio and network modules that use C libraries. There is no memory leak over time. All modules are threaded. Thanks to this design (as well as Rust and C), baru is lightweight and efficient. It can run at a high refresh rate with a minimal processor footprint. The audio module communicates with the PipeWire/PulseAudio server through client API to retrieve its data. Wireless and wired modules use... -
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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 Transformers, ExLlamaV2, VLLM and a JavaScript interface via Transformers.js, allowing it to run on CPUs, NVIDIA CUDA GPUs, AMD ROCm, Vulkan-capable GPUs, and Apple Metal. ... -
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10
Snapcast
Synchronous multiroom audio player
...Each client does continuous time synchronization with the server, so that the client is always aware of the local server time. Every received chunk is first decoded and added to the client's chunk-buffer. Knowing the server's time, the chunk is played out using a system dependend low level audio API (e.g. ALSA) at the appropriate time. -
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Libdragon
Open source library for N64 development
...It replaces the original proprietary development kits with a fully open toolchain based on GCC and standard C libraries, allowing a more accessible and flexible development experience. The library provides both low-level access to hardware and higher-level abstractions for graphics, audio, and input handling. It supports advanced features such as accelerated 2D rendering and efficient memory management tailored to the N64 architecture. Developers can test builds on emulators or real hardware using flash cartridges. The project is designed to modernize N64 development while maintaining compatibility with the console’s original capabilities. ... -
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HunyuanVideo-Foley
Multimodal Diffusion with Representation Alignment
HunyuanVideo-Foley is a multimodal diffusion model from Tencent Hunyuan for high-fidelity Foley (sound effects) audio generation synchronized to video scenes. It is designed to generate audio that matches both visual content and textual semantic cues, for use in video production, film, advertising, games, etc. The model architecture aligns audio, video, and text representations to produce realistic synchronized soundtracks. Produces high-quality 48 kHz audio output suitable for professional use. ... -
13
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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FluidSynth
Software synthesizer based on the SoundFont 2 specifications
FluidSynth is a real-time software synthesizer based on the SoundFont 2 specifications and has reached widespread distribution. FluidSynth itself does not have a graphical user interface, but due to its powerful API several applications utilize it and it has even found its way onto embedded systems and is used in some mobile apps. -
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EasyRPG Player
RPG Maker 2000/2003 and EasyRPG games interpreter
EasyRPG Player is a game interpreter to play RPG Maker 2000, 2003, and EasyRPG games. It uses the LCF parser library (liblcf) to read RPG Maker game data. EasyRPG Player is a program that allows to play games created with RPG Maker 2000 and 2003. It aims to be a free (as in freedom) cross-platform RPG Maker 2000/2003 interpreter. The main goal is to play all games created with them as the original game interpreter (RPG_RT) does. This project is being developed. FreeType2 for external font... -
16
DreamShell
Operating system for the Sega Dreamcast
...It has a dynamic loadable modular system and interface for creating applications with XML UI and both C/C++ and Lua scripts on. You can see examples in ready-made applications and modules, drivers for various devices, formats and interfaces. Examples for audio and video decoding, compression, packaging, binding, network, emulation, scripts and more. From hardcore low-level assembler to high-level applications. There is also large subproject is the ISO Loader, which contains emulation of BIOS system calls, CDDA playback, and VMU, also it can hooking interrupts for various SDKs and more. -
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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. ... -
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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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PJSIP
PJSIP project
PJSIP is a free and open-source multimedia communication library written in C language implementing standard-based protocols such as SIP, SDP, RTP, STUN, TURN, and ICE. It combines signaling protocol (SIP) with a rich multimedia framework and NAT traversal functionality into high-level API that is portable and suitable for almost any type of system ranging from desktops, and embedded systems, to mobile handsets. PJSIP is both compact and feature rich. It supports audio, video, presence, and instant messaging, and has extensive documentation. PJSIP is very portable. On mobile devices, it abstracts system-dependent features and in many cases is able to utilize the native multimedia capabilities of the device. -
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pyttsx3
Offline Text To Speech synthesis for python
...The library exposes a simple but flexible API for controlling voice selection, speaking rate, volume, and other synthesis parameters from Python code. It supports both a high-level speak convenience function and a lower-level engine object with event hooks, queuing, and saving output to audio files. The repository includes examples and documentation that show how to adjust properties dynamically, persist synthesized output, and integrate pyttsx3 into GUIs or background services. -
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FastRTC
The python library for real-time communication
FastRTC is a Python library designed to simplify real-time communication (RTC), especially for audio and video streaming applications. It abstracts away much of the complexity that typically comes with implementing WebRTC by providing a simple interface — e.g. a Stream class — that can be mounted within a web backend (for example a FastAPI application). This makes it particularly well suited for building real-time voice (or video) interfaces for applications such as AI assistants, live chat, or collaborative audio/video tools. ... -
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PyAV
Pythonic bindings for FFmpeg's libraries
...While powerful, it requires a solid understanding of FFmpeg concepts, as it prioritizes flexibility and control over abstraction. Overall, PyAV is a robust tool for developers building advanced video and audio processing systems in Python. -
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AudioKit
Swift audio synthesis, processing, & analysis platform
AudioKit is an entire audio development ecosystem of code repositories, packages, libraries, algorithms, applications, playgorunds, tests, and scripts, built and used by a community of audio programmers, app developers, engineers, researchers, scientists, musicians, gamers, and people new to programming. An important goal for AudioKit is to allow it to grow and be maintainable by a handful of volunteers. For this reason we have extensive tests that are run whenever changes are made to any... -
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Remotion
Make videos programmatically with React
Remotion is a cutting-edge library that lets developers create real videos programmatically using React components, transforming familiar UI paradigms into a flexible, code-driven video production workflow. Instead of traditional timeline editors, Remotion leverages HTML, CSS, and JavaScript to define video frames, animations, and transitions, which means developers can use states, props, loops, and component hierarchies to automate complex motion graphics. Because it integrates with the... -
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mediasoup
Cutting Edge WebRTC Video Conferencing
mediasoup is a Node.js library that provides a cutting-edge WebRTC server capable of handling real-time communications with efficient media routing and processing.
