Search Results for "dvd-audio" - Page 2
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Showing 1086 open source projects for "dvd-audio"
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1
OpenVoice
Instant voice cloning by MIT and MyShell. Audio foundation model
OpenVoice is a versatile instant voice cloning system that can replicate a speaker’s tone color from just a short audio clip and then generate speech in multiple languages. It is designed not only to match the timbre of the reference voice, but also to give granular control over style parameters such as emotion, accent, rhythm, pauses, and intonation. The model supports cross-lingual and even zero-shot cross-lingual voice cloning, so a speaker recorded in one language can be made to speak naturally in others. ... -
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Dia2
TTS model capable of streaming conversational audio in realtime
Dia2 is a streaming dialogue text-to-speech model created by Nari Labs for generating conversational audio in real time. It is designed to begin producing speech before receiving the entire input text, which makes it useful for interactive voice applications. The model supports audio conditioning, allowing generated speech to follow a reference voice or conversational style more naturally. Dia2 provides 1B and 2B model checkpoints along with inference code for research and experimentation. ... -
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Audiogen Codec
48khz stereo neural audio codec for general audio
...We found that training with EMA and adding a perceptual loss term with CLAP features improved performance. These codecs, being low compression, outperform Meta's EnCodec and DAC on general audio as validated from internal blind ELO games. We trained (relatively) very low compression codecs in the pursuit of solving a core issue regarding general music and audio generation, low acoustic quality, and audible artifacts, which hinder industry use for these models. Our hope is to encourage researchers to build hierarchical generative audio models that can efficiently use high sequence length representations without sacrificing semantic abilities. -
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HunyuanVideo-Avatar
Tencent Hunyuan Multimodal diffusion transformer (MM-DiT) model
HunyuanVideo-Avatar is a multimodal diffusion transformer (MM-DiT) model by Tencent Hunyuan for animating static avatar images into dynamic, emotion-controllable, and multi-character dialogue videos, conditioned on audio. It addresses challenges of motion realism, identity consistency, and emotional alignment. Innovations include a character image injection module, an Audio Emotion Module for transferring emotion cues, and a Face-Aware Audio Adapter to isolate audio effects on faces, enabling multiple characters to be animated in a scene. Character image injection module for better consistency between training and inference conditioning. ... -
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AudioNotes
Extract audio and video content and organize it into a Markdown note
AudioNotes is an application (or proof-of-concept) that likely combines audio recording or playback with note-taking or annotation functionality — enabling users to record voice or audio and attach textual or timestamped notes, making it ideal for lectures, interviews, meetings, or personal memos. Such a tool offers a more expressive and flexible way to capture and revisit information: instead of just typed notes or raw audio, users get both audio context and structured notes. ... -
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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. -
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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. -
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Basic Pitch
A lightweight audio-to-MIDI converter with pitch bend detection
...Provide a compatible audio file and a basic-pitch will generate a MIDI file, complete with pitch bends. The basic pitch is instrument-agnostic and supports polyphonic instruments, so you can freely enjoy transcription of all your favorite music, no matter what instrument is used. Basic pitch works best on one instrument at a time. -
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Bili23 Downloader
Cross platform GUI tool for downloading videos from Bilibili sites
...It can parse different types of links such as standard video pages, short links, and collection or activity pages to automatically retrieve downloadable media. It also allows users to choose video resolution, audio quality, and encoding format based on the available sources. Additional features include downloading subtitles, comments, metadata, and artwork associated with videos. -
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MOSS-TTS-Nano
MOSS-TTS-Nano is an open-source multilingual tiny speech generation
...The model operates efficiently on CPU-only systems, enabling deployment without specialized hardware. It supports multilingual voice cloning and produces high-fidelity audio with low latency. The system uses an autoregressive audio tokenization pipeline to generate natural-sounding speech. It is suitable for local applications, web services, and embedded systems. Overall, it brings advanced speech synthesis capabilities to lightweight and accessible environments. -
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Speakr
Speakr is a personal, self-hosted web application
...It provides a clean, user-friendly interface where users can input text, choose a voice style or language, and immediately hear the output, making it ideal for accessibility, content creation, and learning applications. 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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WavTokenizer
SOTA discrete acoustic codec models with 40/75 tokens per second
WavTokenizer is a state-of-the-art discrete acoustic codec designed specifically for audio language modeling, capable of compressing 24 kHz audio into just 40 or 75 tokens per second while preserving high perceptual quality. It is built to represent speech, music, and general audio with extremely low bitrate, making it ideal as a front-end for large audio language models like GPT-4o and similar architectures. The model uses a single-quantizer design together with temporal compression to achieve extreme compression without sacrificing reconstruction fidelity. ... -
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SoulSync
Automated Music Discovery and Collection Manager
...It can automatically download missing tracks from multiple sources including Soulseek, YouTube, and Beatport, then verify file accuracy through AcoustID fingerprinting to ensure the correct audio was obtained. The platform also enriches tracks with lyrics, album art, and MusicBrainz metadata while organizing files into clean, customizable folder structures. Once processed, the library can be synchronized with media servers like Plex, Jellyfin, or Navidrome so collections stay up to date without manual intervention. -
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ebook2audiobook
Generate audiobooks from e-books, voice cloning & 1107+ languages
ebook2audiobook is a tool to convert legally obtained eBooks (non-DRM) into fully narrated audiobooks, complete with chapters and metadata. It automates the pipeline: it reads the eBook file, splits it into appropriate segments (chapters, paragraphs), uses text-to-speech (TTS) models to synthesize audio, optionally applies voice cloning, and outputs a final audiobook — ideal for people who prefer listening over reading, or for accessibility purposes. The tool supports a wide array of underlying TTS backends (XTTSv2, Bark, VITS, Fairseq, Tacotron2, YourTTS and more), which gives flexibility depending on hardware availability, voice preference, and language. ... -
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Swing Music
Swing Music is a beautiful, self-hosted music player
Swing Music is a beautiful, self-hosted music player and streaming server that lets you bring your personal audio library online with a modern browser-based interface, giving you a private alternative to mainstream streaming services. Designed to be both elegant and powerful, the project scans your local music files (like MP3s or FLACs), organizes metadata, and streams them on-demand to any device with a browser or its Android client. It includes features like folder browsing, playlist creation, related artists/albums exploration, and intelligent metadata normalization so your collection feels clean and connected. ... -
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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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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 a specialized pipeline that separates text generation from timestamp alignment, allowing the system to generate transcripts and then align them with audio using forced alignment techniques. ... -
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AudioMuse-AI
AudioMuse-AI is an Open Source Dockerized environment
AudioMuse-AI is an open-source system designed to automatically generate playlists and analyze music libraries using artificial intelligence and audio signal processing techniques. The platform runs locally in a Dockerized environment and performs detailed sonic analysis on audio files to understand characteristics such as tempo, mood, and acoustic similarity. By analyzing the underlying audio content rather than relying on external metadata services, the system can organize large personal music libraries and generate curated playlists for different moods or listening contexts. ... -
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AutoSubSync
Automatic subtitle synchronization tool
AutoSubSync is a cross-platform desktop application designed to automatically synchronize subtitle files with video content using advanced alignment algorithms. It integrates tools like ffsubsync, autosubsync, and alass to analyze audio and match subtitle timing with high accuracy. The application supports both automatic synchronization and manual adjustment, allowing users to fine-tune results when needed. It provides a drag-and-drop interface that simplifies the process of loading video and subtitle files, making it accessible for non-technical users. AutoSubSync also includes batch processing capabilities, enabling users to handle entire media libraries efficiently. ... -
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WhisperLive
A nearly-live implementation of OpenAI's Whisper
WhisperLive is a “nearly live” implementation of OpenAI’s Whisper model focused on real-time transcription. It runs as a server–client system in which the server hosts a Whisper backend and clients stream audio to be transcribed with very low delay. The project supports multiple inference backends, including Faster-Whisper, NVIDIA TensorRT, and OpenVINO, allowing you to target GPUs and different CPU architectures efficiently. It can handle microphone input, pre-recorded audio files, and network streams such as RTSP and HLS, making it flexible for live events, monitoring, or accessibility workflows. ... -
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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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Moshi
A speech-text foundation model for real time dialogue
Moshi is a speech-text foundation model and full-duplex spoken dialogue framework. It uses Mimi, a state-of-the-art streaming neural audio codec. Mimi processes 24 kHz audio, down to a 12.5 Hz representation with a bandwidth of 1.1 kbps, in a fully streaming manner (latency of 80ms, the frame size), yet performs better than existing, non-streaming, codecs like SpeechTokenizer (50 Hz, 4kbps), or SemantiCodec (50 Hz, 1.3kbps). Moshi models two streams of audio: one corresponds to Moshi, and the other one to the user. ... -
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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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Voice-Pro
Comprehensive Gradio WebUI for audio processing
Voice-Pro is the best gradio WebUI for transcription, translation and text-to-speech. It can be easily installed with one click. Create a virtual environment using Miniconda, running completely separate from the Windows system (fully portable). Supports real-time transcription and translation, as well as batch mode. -
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FFsubsync
Automagically synchronize subtitles with video
...In this case, you can use the correctly synchronized srt file directly as a reference for synchronization, instead of using the video as the reference. ffsubsync uses the file extension to decide whether to perform voice activity detection on the audio or to directly extract speech from an srt file. ffsubsync usually finishes in 20 to 30 seconds, depending on the length of the video.
