Search Results for "audio source separation" - Page 2
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Showing 1039 open source projects for "audio source separation"
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SoulSync
Automated Music Discovery and Collection Manager
SoulSync is an intelligent music discovery and automation platform designed to bridge streaming services with self-hosted media libraries, enabling users to automatically grow and maintain curated music collections. The system continuously monitors selected artists and detects new releases, then generates personalized playlists such as Release Radar and Discovery Weekly using its built-in recommendation logic. It can automatically download missing tracks from multiple sources including... -
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Podcastfy.ai
Transforming Multimodal Content into Captivating Multilingual Audio
Podcastfy is an open-source Python package that transforms multi-modal content (text, images) into engaging, multi-lingual audio conversations using GenAI. Input content includes websites, PDFs, youtube videos as well as images. Unlike UI-based tools focused primarily on note-taking or research synthesis (e.g. NotebookLM), Podcastfy focuses on the programmatic and bespoke generation of engaging, conversational transcripts and audio from a multitude of multi-modal sources enabling customization and scale. -
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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. NovaSR is especially valuable for post-processing tasks... -
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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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Bili23 Downloader
Cross platform GUI tool for downloading videos from Bilibili sites
Bili23-Downloader is an open source desktop application designed for downloading video content from the Bilibili platform. It provides a graphical interface that allows users to download various types of media including user-uploaded videos, series episodes, movies, and other hosted content. It focuses on ease of use with a zero-configuration setup, making it accessible to both beginners and experienced users. -
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ffmpeg-normalize
Audio Normalization for Python/ffmpeg
ffmpeg-normalize is a command-line utility designed to normalize audio levels in media files using FFmpeg, ensuring consistent volume across multiple tracks. It supports both EBU R128 loudness normalization and peak normalization methods, allowing users to choose the appropriate standard for their needs. The tool analyzes audio streams and applies adjustments to achieve target loudness levels without introducing distortion. It can process multiple files in batch mode, making it suitable for... -
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You-Get
Dumb downloader that scrapes the web
You-Get is a small command-line utility for downloading media (video, audio and images) from the Web when there are no other means to do so. It can download video and audio files from such popular web sites as YouTube, Twitter, Niconico, Vimeo, Flickr, Instagram and a whole lot more. You-Get is a great option for when you want to enjoy your favorite videos, audio or images from the internet without having to open any web browsers or get interrupted by ads. ... -
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VoxCPM2
Tokenizer-Free TTS for Multilingual Speech Generation
...The system is trained on massive multilingual datasets, enabling support for dozens of languages and dialects while maintaining high fidelity and realism in generated audio. VoxCPM stands out for its ability to perform voice cloning with minimal input, capturing not only the speaker’s timbre but also nuanced features such as rhythm, accent, and emotional delivery. It also introduces voice design capabilities, allowing users to generate entirely new voices from natural language descriptions without requiring reference audio. -
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TradingAgents
Chinese Financial Trading Framework Based on Multi-Agent LLM
TradingAgents-CN is a Chinese-enhanced, multi-agent LLM framework aimed at building financial analysis and trading-oriented workflows, with an emphasis on collaboration between specialized agents rather than a single monolithic prompt. It organizes market-related tasks into roles and stages so different agents can contribute research, reasoning, aggregation, and decision support in a structured pipeline. The project is oriented toward practical usage, including a stack that can be run in a... -
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LatentSync
Taming Stable Diffusion for Lip Sync
LatentSync is an open-source framework from ByteDance that produces high-quality lip-synchronization for video by using an audio-conditioned latent diffusion model, bypassing traditional intermediate motion representations. In effect, given a source video (with masked or reference frames) and an audio track, LatentSync directly generates frames whose lip motions and expressions align with the audio, producing convincing talking-head or animated lip-sync output. ... -
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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... -
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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... -
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Speakr
Speakr is a personal, self-hosted web application
...Behind the scenes, Speakr leverages modern TTS engines and streaming audio technologies to deliver smooth and responsive speech generation without noticeable delay. The project is built with extensibility in mind, enabling developers to add custom voices, integrate additional languages, and tailor the backend for different hardware or cloud environments. It also supports saving generated audio as downloadable files so users can reuse the speech outputs in other projects, presentations, or media content. -
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FFsubsync
Automagically synchronize subtitles with video
Language-agnostic automatic synchronization of subtitles with video, so that subtitles are aligned to the correct starting point within the video. First, make sure ffmpeg is installed. Make sure ffmpeg is on your path and can be referenced from the command line! Next, grab the script. It should work with both Python 2 and Python 3. There may be occasions where you have a correctly synchronized srt file in a language you are unfamiliar with, as well as an unsynchronized srt file in your... -
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VoiceStudio
VoiceStudio is the open-source, fully-local ElevenLabs alternative
VoiceStudio is an open-source, local-first platform for speech generation, voice cloning, transcription, dubbing, and audio production. It can clone voices from short reference recordings or design new voices from characteristics such as accent, pitch, age, and delivery style. Video dubbing combines transcription, translation, speaker preservation, speech synthesis, and final video export. -
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Crosvm
The Chrome OS Virtual Machine Monitor
crosvm (ChromeOS Virtual Machine Monitor) is a secure, lightweight virtual machine monitor built on top of the Linux KVM hypervisor. Developed for ChromeOS, it is designed to isolate and execute Linux and Android guests efficiently while maintaining strong security boundaries. Unlike general-purpose emulators like QEMU, crosvm avoids full hardware emulation and focuses on modern paravirtualized I/O using the virtio standard, reducing complexity and attack surface. Written in Rust, it... -
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Quod Libet
Music player and music library manager for Linux, Windows, and macOS
Quod Libet is a cross-platform audio/music management program. It provides many ways to view your local library, and supports streaming audio and feeds (podcasts, etc). It has extremely flexible metadata editing and searching capabilities. With over 90 plugins included, you can extend and integrate with almost anything, or write your own! Ex Falso is a bare-bones tag editor with the same editing interface as Quod Libet. Quod Libet is a GTK+-based audio player written in Python, using the... -
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Goutoujunshi
A Codex dating strategist who first captures emotions
Goutoujunshi is a Codex relationship-advice skill that combines emotional support, structured relationship analysis, and concrete next-step planning. It separates known facts, assumptions, and unknowns before evaluating attraction, reciprocity, risk, opportunity cost, and longer-term options. The skill can analyze user summaries, exported conversations, and chat screenshots while keeping track of who said what. Its knowledge base draws from relationship psychology, personality research,... -
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YuE
Open source AI model for generating full songs from lyrics prompts
YuE is an open source project that provides a foundation model designed for full-song music generation using artificial intelligence. It focuses on transforming text inputs such as lyrics and genre prompts into complete musical compositions that include both vocal and instrumental tracks. Unlike many shorter audio generators, the model is capable of producing songs that last several minutes while maintaining coherent musical structure and alignment with the provided lyrics. ... -
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Miso TTS
Miso TTS is an 8 billion, highly emotive text-to-speech model
Miso TTS is an advanced 8-billion-parameter text-to-speech model developed by Miso Labs for generating highly expressive and natural-sounding conversational speech. Built on an RVQ Transformer architecture inspired by Sesame CSM, it combines a powerful Llama-based backbone with an autoregressive audio decoder to produce high-quality audio from text. The model supports both standard speech synthesis and voice-conditioned generation using optional audio prompts for voice cloning. Miso TTS... -
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Text Generation Web UI
Oobabooga - The definitive Web UI for local AI, with powerful features
A gradio web UI for running Large Language Models like LLaMA, llama.cpp, GPT-J, Pythia, OPT, and GALACTICA. Dropdown menu for switching between models. Notebook mode that resembles OpenAI's playground. Chat mode for conversation and role playing. Instruct mode compatible with Alpaca and Open Assistant formats. Nice HTML output for GPT-4chan. Markdown output for GALACTICA, including LaTeX rendering. Custom chat characters. Advanced chat features (send images, get audio responses with TTS).... -
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Audiomentations
A Python library for audio data augmentation
A Python library for audio data augmentation. Inspired by albumentations. Useful for deep learning. Runs on CPU. Supports mono audio and multichannel audio. Can be integrated in training pipelines in e.g. Tensorflow/Keras or Pytorch. Has helped people get world-class results in Kaggle competitions. Is used by companies making next-generation audio products. Mix in another sound, e.g. a background noise. Useful if your original sound is clean and you want to simulate an environment where... -
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WanGP
AI video generator optimized for low VRAM and older GPUs use
Wan2GP is an open source AI video generation toolkit designed to make modern generative models accessible on consumer-grade hardware with limited GPU memory. It acts as a unified interface for running multiple video, image, and audio generation models, including Wan-based models as well as other systems like Hunyuan Video, Flux, and Qwen. A key focus of the project is reducing VRAM requirements, enabling some workflows to run on as little as 6 GB while still supporting older Nvidia and certain AMD GPUs. ... -
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ElevenLabs Python
The official Python SDK for the ElevenLabs API
elevenlabs-python is the official Python SDK for the ElevenLabs API, giving developers a convenient way to access ElevenLabs’ high-quality, lifelike voices. The library wraps the HTTP API into a typed Python client, so you can perform text-to-speech, streaming, voice cloning, voice management, and agents-related operations with simple method calls. It exposes ElevenLabs’ main models such as Eleven Multilingual v2, Eleven Flash v2.5, and Eleven Turbo v2.5, each targeting different trade-offs... -
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Pipecat
Framework for building real-time voice and multimodal AI agents
Pipecat is an open source Python framework designed for building real-time voice and multimodal conversational AI agents. It provides developers with tools to orchestrate complex pipelines that combine speech recognition, language models, audio processing, and speech synthesis into a cohesive conversational system. Pipecat focuses on low-latency interactions so voice conversations with AI feel natural and responsive during live use.
