Search Results for "audio codec"
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Showing 95 open source projects for "audio codec"
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Audiogen Codec
48khz stereo neural audio codec for general audio
AGC (Audiogen Codec) is a convolutional autoencoder based on the DAC architecture, which holds SOTA. 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. ... -
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Wrapper for VideoStation
Synology VideoStation and DLNA FFmpeg Wrapper with AAC, DTS, EAC3
...The project also includes options for configuring audio codec priorities and enabling multi-channel output such as 5.1 audio. It is compatible with modern Synology DSM versions and integrates directly into existing media server workflows. Overall, it enhances media server flexibility by removing codec restrictions and improving playback support. -
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HeartMuLa
A Family of Open Sourced Music Foundation Models
...At the center is HeartMuLa, a music language model that generates music conditioned on inputs like lyrics and tags, with multilingual support that broadens the range of lyric-driven use cases. The project also includes HeartCodec, a music codec optimized for high reconstruction fidelity, enabling efficient tokenization and reconstruction workflows that are critical for training and generation pipelines. For text extraction from audio, it provides HeartTranscriptor, a Whisper-based model tuned specifically for lyrics transcription, which helps bridge generated or recorded audio back into structured text. ... -
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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. -
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5
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. ... -
6
go2rtc
Ultimate camera streaming application
go2rtc is a lightweight, zero-dependency streaming server designed to unify and convert video streams across a wide range of protocols and devices, particularly in smart home and surveillance environments. Written in Go, it provides real-time streaming capabilities with extremely low latency by supporting protocols such as RTSP, WebRTC, RTMP, HTTP, and HomeKit, while also enabling seamless transcoding using FFmpeg when needed. The application can ingest streams from IP cameras, USB devices,... -
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WebRTC Camera
Home Assistant custom component for real-time viewing
...Supported inputs include RTSP, RTMP, HTTP streams, HomeKit cameras, USB cameras, ONVIF-style sources, and other go2rtc-compatible streams. It can output through WebRTC, RTSP, MSE, MP4, or MJPEG depending on browser, codec, and network conditions. The integration includes a custom Lovelace card, two-way audio workflows, snapshots to Telegram, stream sharing links, and Chromecast or Android TV casting. WebRTC Camera is useful for Home Assistant dashboards that need faster and more flexible live camera playback than standard camera cards. -
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rtmp-rtsp-stream-client-java
Library to stream in rtmp and rtsp for Android. All code in Java
Library for streaming in RTMP and RTSP. All code in Java. -
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ffmpeg-commander
A web-based GUI for quickly generating common FFmpeg command-line
ffmpeg-commander is a web-based graphical interface that simplifies the creation of FFmpeg commands for common video and audio encoding tasks. It provides a user-friendly environment where users can configure encoding options without needing to memorize complex command-line syntax. Built with modern web technologies, it generates FFmpeg commands dynamically based on user input. The tool focuses on common workflows such as format conversion, compression, and codec selection. ... -
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opencore-amr
Audio codecs extracted from Android Open Source Project
Library of OpenCORE Framework implementation of Adaptive Multi Rate Narrowband and Wideband (AMR-NB and AMR-WB) speech codec. Library of VisualOn implementation of Adaptive Multi Rate Wideband (AMR-WB) encoder and Advanced Audio Coding (AAC) encoder. Modified library of Fraunhofer AAC decoder and encoder. -
11
AudioCraft
Audiocraft is a library for audio processing and generation
AudioCraft is a PyTorch library for text-to-audio and text-to-music generation, packaging research models and tooling for training and inference. It includes MusicGen for music generation conditioned on text (and optionally melody) and AudioGen for text-conditioned sound effects and environmental audio. Both models operate over discrete audio tokens produced by a neural codec (EnCodec), which acts like a tokenizer for waveforms and enables efficient sequence modeling. ... -
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voxshare_gui
*VoxShare* is a simple Python-based push-to-talk multicast voice chat
VoxShare is a simple Python-based push-to-talk multicast voice chat application with a sleek modern GUI built using CustomTkinter. Provided as python source code or compiled standalone windows application (no need to install anything). -
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Apprentice Video
it's a video player, also works for music and pictures
This player stands on the giant shoulders of FFmpeg. Audio rendering is accomplished via portaudio v19. Video rendering is via OpenGL, using fragment programs when possible. User interface is implemented with Qt 4/5/6. ASS/SSA subtitle rendering is implemented with libass. This player provides several performance options to enable adequate video playback on slow hardware: * skip loop filter * skip non-reference frames * skip color converter * reduce playback speed to accommodate slow video decoding This player supports playback of HDR video on non-HDR displays: * colorspace transform to BT.709 colorspace via an auto-generated 3D LUT * tone mapping from HDR to SDR (BT. 709) This player supports playback of MPEG-TS files containing multiple programs, timeline anomalies, and codec changes. ... -
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Aeyae Remux
GOP-based video editor (remuxer)
...Aeyae Remux can be used to stitch together files... within reason. Since it is only a remuxer -- stitched files must be of compatible format. This means stitched files must have the same audio/video codecs, and codec properties (width, height, sample rate, etc...). That said... Aeyae Remux currently doesn't check or enforce this restriction, so beware -- garbage input will likely produce garbage output. Linux binaries are provided as AppImage, built on Ubuntu 14.04. 8 GB of RAM (but more is better) is highly recommended if working with source files longer than 2 hours. -
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Shutter Encoder
Free professional video converter Windows|Mac|Linux
Shutter Encoder is an video, audio and image converter based on FFmpeg and other great tools. It has been designed by video editors in order to be as accessible and efficient as possible. It's a swiss knife tool for any video editor. Link to website & downloads : https://www.shutterencoder.com - Without conversion: Cut without re-encoding, Replace audio, Rewrap, Conform, Merge, Extract, Subtitling, Video inserts - Sound conversions: WAV, AIFF, FLAC, ALAC, MP3, AAC, AC3, OPUS, OGG - Editing codecs: DNxHD, DNxHR, Apple ProRes, QT Animation, GoPro CineForm, Uncompressed YUV - Output codecs: H.264, H.265, VP8, VP9, AV1, OGV - Broadcast codecs: XDCAM HD422, AVC-Intra 100, XAVC, HAP - Old codecs: DV PAL, MJPEG, Xvid, WMV, MPEG - Archiving codec: FFV1 - Images creation: JPEG, Image - Burn & Rip: DVD, Blu-ray, DVD RIP - Analysis: Loudness & True Peak, Audio normalization, Cut detection, Black detection, Media, VMAF - Download: Web video -
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Virtualdub Batch Video DeShake v26.0204
Batch to compress [and deshake] all videos [or images] in folder
Installation: Execute "DeShakInst.BAT" VirtualDub2 44282; AviSynth+ 3.7.5 updated to C:\DVD DESHAK.BAT updated to C:\UT and added to PATH Usage: DESHAK task[s] [parameters] Tasks: tp1: deshake pass1 LOG generation for 2nd pass tp2: deshake pass2 and compress video and audio to MP3 tcomp: compress (no deshake) twav: extract WAV and/or uses external WAV audio Parameters (more in help): vEXT: video extension (ie: vmov), default: vAVI qN: h264 quality 1-9 (9=lossless), def: q3 (crf23) aN: mp3 quality 1-5, def: a3 (192k) * generates: ZZoriginalname.AVI * some settings at begining ie: vdPath Min Requirements: XP; Win7x64 for aviSynth video NoiseReduction Klite Mega Codec Pack (with LAME encoder) Other Utilities: LOG2CHAPS.BAT generate _OGG.txt chapters @ scene change VID2AUD.BAT extract Audios VID2MKV.BAT multiplex vid+aud+chapters VIDJOIN.BAT merges videos to MKV -
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ViaVoip
A portable peer to peer voice-chat/walkie-talkie.
ViaVoip is a simple Voice Over IP application that can be used when you need to talk, chat, or send files through the internet, but you can't or don't want to make use of any third party services. Its peer to peer design allows the two end points to connect directly to each other, without any central server nor account registration. It runs on Windows, Linux, Mac OS X and Android, and is portable, that is you don't need any setup, just get a copy and run it from any storage... -
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howler.js
Javascript audio library for the modern web
howler.js is an audio library for the modern web. It defaults to Web Audio API and falls back to HTML5 Audio. This makes working with audio in JavaScript easy and reliable across all platforms. Additional information, live demos and a user showcase are available at howlerjs.com. Single API for all audio needs, defaults to Web Audio API and falls back to HTML5 Audio, handles edge cases and bugs across environments, supports all codecs for full cross-browser support, automatic caching for... -
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VALL-E
PyTorch implementation of VALL-E (Zero-Shot Text-To-Speech)
We introduce a language modeling approach for text to speech synthesis (TTS). Specifically, we train a neural codec language model (called VALL-E) using discrete codes derived from an off-the-shelf neural audio codec model, and regard TTS as a conditional language modeling task rather than continuous signal regression as in previous work. During the pre-training stage, we scale up the TTS training data to 60K hours of English speech which is hundreds of times larger than existing systems. ... -
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Lyra
A Very Low-Bitrate Codec for Speech Compression
lyra is a neural audio codec designed to deliver intelligible, natural-sounding speech at extremely low bitrates, making real-time communication viable on constrained networks. It replaces hand-engineered codecs with learned models that capture speech characteristics more efficiently and reconstruct waveforms with a neural vocoder. The system targets mobile-class hardware, balancing latency and quality so it can run in real-time on phones. -
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EnCodec
State-of-the-art deep learning based audio codec
Encodec is a neural audio codec developed by Meta for high-fidelity, low-bitrate audio compression using end-to-end deep learning. Unlike traditional codecs (like MP3 or Opus), Encodec uses a learned quantizer and decoder to reconstruct complex waveforms with remarkable accuracy at bitrates as low as 1.5 kbps. It employs a convolutional encoder–decoder architecture trained with perceptual loss functions that optimize for human auditory quality rather than raw waveform distance. ... -
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This project is the lossless audio codec, Monkey's Audio, non-win32 platform port and includes some other plugins for music player of some other non-win32 platforms.
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AlphaPlayer
AlphaPlayer is a video animation engine
AlphaPlayer is positioned as a multimedia or media-player library or application under ByteDance, likely intended to provide video/audio playback functionality, streaming, or media rendering capabilities. It probably serves as a foundation for building media-heavy applications — offering features like playback control, streaming support, adaptive media handling, and possibly integration with custom codecs or streaming protocols. For developers building web, desktop, or mobile applications... -
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FFME
FFME: The Advanced WPF MediaElement (based on FFmpeg)
FFME: The Advanced WPF MediaElement (based on FFmpeg). FFME is an advanced and close drop-in replacement for Microsoft's WPF MediaElement Control. While the standard MediaElement uses DirectX (DirectShow) for media playback, FFME uses FFmpeg to read and decode audio and video. This means that for those of you who want to support stuff like HLS playback, or just don't want to go through the hassle of installing codecs on client machines, using FFME might just be the answer. Fast media seeking... -
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FXMovieManager
a filemanager written to help managing movie files and video clips
FxMovieManager is a filemanager written to help managing movie files and video clips. It includes a twin-panel filemanager, a playlist and act as a simple front-end to MPlayer. Both filemanager and playlist can display thumbnails of movie files. ChangeLog for v. 10.2.8: mpv now uses the ``edl://`` command in order to correctly play multiple files with start and duration times on the PlayList; italian translation on the configuration dialog has been improved.