Projects sharing features with DeepFilterNet

BackgroundMusic is a system-level macOS audio management utility that provides an application volume mixer for independent gain and level adjustments for every running application. It functions as a system audio router and pause controller to manage how sound is handled across the operating system. The project features a virtual audio driver that routes internal system sound to recording software and input devices. It also includes automatic music ducking, which monitors system audio activity to pause music playback when other applications produce sound and resume it once that audio ends. Th

EasyEffects is a real-time audio processor and system-wide effects manager designed for PipeWire audio streams. It functions as a comprehensive suite for applying filters, equalizers, and limiters to both input and output audio across the entire system. The project distinguishes itself through its use of deep learning for neural network noise suppression and voice isolation, as well as its ability to simulate physical acoustic environments using impulse-response convolution. It includes a sophisticated preset management system that allows users to associate specific audio configurations with

rnnoise is a real-time speech denoising library that uses a recurrent neural network to suppress background noise from live voice audio. It is implemented as a lightweight C library with a minimal API, designed for easy integration into audio applications that need low-latency noise reduction. The library employs a gated recurrent unit (GRU) architecture and frequency-domain feature extraction to capture temporal dependencies in speech, operating on short audio frames sequentially for streaming use. It also includes a training pipeline that allows users to train custom noise suppression model

This project is an RTMP media streaming SDK and a real-time communication framework designed for pushing and playing audio and video streams. It provides tools for interactive broadcasting, low-latency voice and video calls, and a cross-platform media player compatible with Windows, iOS, and Android. The toolkit enables interactive live broadcasting with support for multi-host interactions and the ability to push streams to distribution servers via CDN. It includes a cloud recording manager for capturing live sessions and saving them as files to cloud storage, along with a system for composit

SmarterStreaming is a cross-platform SDK for hardware-accelerated audio and video capture, encoding, and streaming. It provides a complete media pipeline for low-latency RTMP and RTSP streaming, and includes an embedded lightweight RTSP server that can serve live feeds directly from the source device to local network clients without requiring a separate server. The SDK also integrates with GB28181 surveillance platforms, enabling compliant device registration and streaming for standardized video monitoring systems. The project distinguishes itself through a set of integrated capabilities for

Pipecat is a framework and software development kit for building real-time multimodal AI agents and speech-to-speech systems. It utilizes a frame-based data pipeline to route audio, video, and text through a modular sequence of processors, enabling the orchestration of low-latency conversational AI. The project is distinguished by its ability to coordinate complex multimodal services, including speech-to-text, language models, and text-to-speech, within a single pipeline. It features semantic voice activity detection for natural turn-taking, state-machine conversation flows for dialogue manag

LiveKit is a comprehensive framework for building and orchestrating real-time, multimodal AI agents that interact with users through voice, video, and text. It provides a centralized, event-driven architecture to manage the entire lifecycle of automated participants, from initialization and session state management to graceful shutdown. By utilizing a selective forwarding unit, the platform efficiently routes media streams between participants and agents, ensuring low-latency communication and secure, token-based authentication for all connections. The platform distinguishes itself through it

This project is a comprehensive software entrepreneurship curriculum and solopreneurship business playbook designed for developers. It provides a strategic framework for building, validating, and monetizing side businesses using lean startup methodology and a systematic product development approach. The project distinguishes itself by offering specific guides for digital monetization and career anti-fragility, helping software engineers transition from employment to self-employment. It focuses on turning technical skills into scalable digital assets, paid communities, and independent software

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

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

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

This project is a collection of deep learning tools for image classification and audio tagging, providing a repository of pre-trained model weights and architectures. It serves as a Keras model zoo that enables the immediate use of established neural networks for inference and transfer learning. The library includes a music tagging framework that classifies audio recordings using convolutional recurrent neural networks and mel-spectrograms. For visual data, it provides implementations of architectures such as ResNet, VGG, and Xception, alongside a repository of weights trained on large datase

chineseocr_lite is a lightweight Chinese optical character recognition engine designed to detect text regions, analyze orientation, and convert Chinese characters from images into digital text. It supports both horizontal and vertical reading layouts and can be deployed as a web service for image uploads and result visualization. The system utilizes a multi-backend inference framework that supports ncnn, mnn, and tnn, allowing it to run across diverse hardware and platforms. It is specifically engineered for lightweight deployment on mobile and desktop environments through the use of small mo

Neuraltalk2 is a deep learning vision system designed for automatic image captioning. Built with PyTorch, it utilizes a hybrid architecture that combines a convolutional neural network encoder with a recurrent neural network decoder to generate textual descriptions from visual input. The project features a GPU-accelerated training pipeline capable of distributing workloads across multiple graphics processing units through multi-process distribution. It supports the generation of descriptions for both static image files and real-time video streams. The framework includes capabilities for enco

EZAudio is an audio library for Apple platforms that provides standardized interfaces for microphone capture, file playback, and hardware output. It functions as a low-latency audio processor and visualization framework designed to manipulate audio buffers and route signals with minimal delay. The project features a hardware-accelerated waveform renderer for drawing real-time audio amplitudes and rolling plots. It also includes a Fast Fourier Transform analyzer that converts time-domain audio samples into frequency-domain data for spectral analysis. The library covers a broad range of capabi

AudioKit is an audio framework for iOS, macOS, and tvOS that provides tools for digital audio synthesis, signal processing, and audio analysis. It functions as a synthesis engine for generating audio waveforms and textures, a processing library for modifying tonal characteristics, and a toolkit for extracting frequency and amplitude data from sonic signals. The framework utilizes a modular node architecture and graph-based signal routing to connect audio generators, processors, and outputs. It wraps low-level audio primitives in high-level classes to facilitate sound generation and modificati

ClearerVoice-Studio is a speech processing studio and framework designed for speech enhancement, audio super-resolution, and targeted voice extraction. It provides a suite of tools to remove background noise, increase the sampling rate of low-resolution recordings, and quantify audio clarity through objective quality evaluation metrics. The project features a target speaker extraction tool that isolates specific voices from mixed audio using acoustic, visual, or neural reference signals. It also includes capabilities for overlapping speech separation by capturing temporal patterns and long-ra

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

Metrolist is a music streaming application and library manager designed for high-fidelity audio and video playback. It functions as a collaborative audio player that enables real-time playback synchronization across multiple users through a request and approval system. The platform features an AI-driven lyrics translator that fetches time-synced lyrics and provides automated real-time translations. It also includes specialized integrations for Discord Rich Presence and a dedicated media client interface for Android Auto. The system manages music libraries with remote cloud storage synchroniz

This project is a comprehensive reference guide and directory of web browser capabilities. It serves as a technical map for accessing native operating system functions, hardware interfaces, and standard web APIs to bridge the gap between web applications and desktop or mobile environments. The resource provides detailed guidance on implementing Progressive Web App features, including offline caching, push notifications, and native installation prompts. It also catalogs methods for interacting with hardware peripherals via USB, Bluetooth, and NFC, as well as reading raw data from device sensor

This Python SDK provides a comprehensive toolkit for synthetic audio generation, voice cloning, and the development of conversational AI agents. It enables the creation of lifelike spoken audio from text, the replication of human voices through custom cloning, and the deployment of real-time voice agents capable of interacting with external large language models. The library distinguishes itself through deep integration of conversational AI capabilities, including the design of agent personas and the execution of real-time actions via APIs. It supports professional-grade audio production thro

This project is a comprehensive educational curriculum for learning data science and predictive modeling using the Python programming language. It provides structured instructional material and guides covering supervised learning, unsupervised learning, and neural network design. The curriculum focuses on building, training, and evaluating machine learning models. It includes specific guides for implementing linear regression, decision trees, and support vector machines for predictive analysis, as well as tutorials on designing convolutional and recurrent neural network architectures. The co

MIDI.js is a JavaScript library for playing MIDI files and triggering musical notes within web browsers. It functions as a web MIDI library and soundfont audio synthesizer, providing the core engines necessary to render instrument sounds for multiple simultaneous tracks. The project includes a specialized MIDI-to-visual sync engine that interpolates musical events into continuous animation loops to synchronize visuals with audio playback. It also provides a soundfont code generator to convert audio instrument files into base64 code for direct browser consumption. The library covers MIDI file

Audacity is a cross-platform digital audio workstation and multi-track audio editor. It serves as a comprehensive suite for capturing live audio input, refining sound files through splicing and effects, and mixing multi-track audio files using a non-destructive waveform interface. The project functions as a VST3 plugin host, providing a software environment to load and execute audio effects and virtual instruments for real-time signal processing. It also includes an audio spectrum analyzer for visualizing frequencies and waveforms to identify specific sonic characteristics. The software cove

Aubio is an audio analysis and digital signal processing library designed for music information retrieval. It provides a suite of tools for extracting musical features, estimating fundamental frequencies, and tracking rhythmic pulses in audio streams. The library specializes in the detection of pitch and beat, enabling the extraction of musical notes and the estimation of overall tempo. It also includes capabilities for automatic onset detection to identify the start of sonic events and the separation of audio signals into percussive transients and steady-state tonal components. The system c

RxFFmpeg is an Android multimedia framework and media transcoder based on FFmpeg. It provides a set of tools for video and audio editing, transcoding, and processing on Android devices. The framework integrates a video player component for rendering local files and network streams with zoom and rotation support. It also includes specialized libraries for Android video editing, such as cropping and splicing, and Android audio processing for mixing tracks and modifying voice pitch. The project covers broad media manipulation capabilities, including the conversion of images to video, the extrac