Search Results for "audio decoder"
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Showing 55 open source projects for "audio decoder"
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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 generates Mimi audio codes and can leverage conversation history to create more contextually aware and realistic dialogue. Designed for local deployment, it offers watermarking by default to help promote responsible use of generated audio. ... -
2
h3-metal
MiniMax H3 inference engine for Mac computers
h3-metal is a native MiniMax-H3 inference engine for Apple Silicon focused on generating video and audio locally with Metal. It supports prompt-to-video and prompt-to-audio workflows as well as first-frame, last-frame, image, video, and audio references. An interactive terminal session keeps prompt conditioning, the diffusion transformer, and the video decoder in memory for faster repeated generations. Users can trade speed, quality, and memory through denoising steps, layer counts, reuse modes, token reduction, internal render size, and SSD streaming. ... -
3
Whisper
Robust Speech Recognition via Large-Scale Weak Supervision
...These tasks are jointly represented as a sequence of tokens to be predicted by the decoder, allowing a single model to replace many stages of a traditional speech-processing pipeline. The multitask training format uses a set of special tokens that serve as task specifiers or classification targets. -
4
TorchAudio
Data manipulation and transformation for audio signal processing
The aim of torchaudio is to apply PyTorch to the audio domain. By supporting PyTorch, torchaudio follows the same philosophy of providing strong GPU acceleration, having a focus on trainable features through the autograd system, and having consistent style (tensor names and dimension names). Therefore, it is primarily a machine learning library and not a general signal processing library. The benefits of PyTorch can be seen in torchaudio through having all the computations be through PyTorch... -
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5
IndexTTS2
Industrial-level controllable zero-shot text-to-speech system
...It builds on state-of-the-art models such as XTTS and other modern neural TTS backbones, improving them with a conformer-based speech conditional encoder and upgrading the decoder to a high-quality vocoder (BigVGAN2), leading to clearer and more natural audio output. The system supports zero-shot voice cloning — meaning it can mimic a target speaker’s voice from a short reference sample — making it versatile for multi-voice uses. Compared to many open-source TTS tools, IndexTTS emphasizes efficiency and controllability: it offers faster inference, simpler training pipelines, and controllable speech parameters (like duration, pitch, and prosody), which is critical for production use. -
6
XLD
A tool for transcoding lossless audio files
X Lossless Decoder(XLD) is a tool for Mac OS X that is able to decode/convert/play various 'lossless' audio files. The supported audio files can be split into some tracks with cue sheet when decoding. It works on Mac OS X 10.4 and later. -
7
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. -
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Multimodal
TorchMultimodal is a PyTorch library
...The library provides modular building blocks such as encoders, fusion modules, loss functions, and transformations that support combining modalities (vision, text, audio, etc.) in unified architectures. It includes a collection of ready model classes—like ALBEF, CLIP, BLIP-2, COCA, FLAVA, MDETR, and Omnivore—that serve as reference implementations you can adopt or adapt. The design emphasizes composability: you can mix and match encoder, fusion, and decoder components rather than starting from monolithic models. ... -
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CSM (Conversational Speech Model)
A Conversational Speech Generation Model
The CSM (Conversational Speech Model) is a speech generation model developed by Sesame AI that creates RVQ audio codes from text and audio inputs. It uses a Llama backbone and a smaller audio decoder to produce audio codes for realistic speech synthesis. The model has been fine-tuned for interactive voice demos and is hosted on platforms like Hugging Face for testing. CSM offers a flexible setup and is compatible with CUDA-enabled GPUs for efficient execution. -
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MPD
Music Player Daemon SACD/DVD-A ISO decoder plugins
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libTiMidity is a MIDI to WAVE converter library that uses Gravis Ultrasound-compatible patch files to generate digital audio data from General MIDI files. This library based on the TiMidity decoder from SDL_sound library.
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NÜWA - Pytorch
Implementation of NÜWA, attention network for text to video synthesis
Implementation of NÜWA, state of the art attention network for text-to-video synthesis, in Pytorch. It also contains an extension into video and audio generation, using a dual decoder approach. It seems as though a diffusion-based method has taken the new throne for SOTA. However, I will continue on with NUWA, extending it to use multi-headed codes + hierarchical causal transformer. I think that direction is untapped for improving on this line of work. In the paper, they also present a way to condition the video generation based on segmentation mask(s). ... -
13
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. -
14
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. ... -
15
Denoiser
Real Time Speech Enhancement in the Waveform Domain (Interspeech 2020)
Denoiser is a real-time speech enhancement model operating directly on raw waveforms, designed to clean noisy audio while running efficiently on CPU. It uses a causal encoder-decoder architecture with skip connections, optimized with losses defined both in the time domain and frequency domain to better suppress noise while preserving speech. Unlike models that operate on spectrograms alone, this design enables lower latency and coherent waveform output. -
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Vivid3D-Sharp
Modern 3D engine and IDE written using C# and C++.
Modern 3D engine and IDE written using C# and C++. Vivid features all the required functionality to write modern apps ang games. Including a fully working custom GL UI called "ResonanceUI". It features lighting and shadows(Unified) and can even embed and playback videos via FFMPEG, in the UI or full screen, with fully synced audio+picture. It offers two scripting solutions: C# and a custom language called VividScript, which is an OOP style language. The Editor is based on other... -
17
wav2letter++
Facebook AI research's automatic speech recognition toolkit
First, install Flashlight (using the 0.3 branch is required) with the ASR application. This repository includes recipes to reproduce the following research papers as well as pre-trained models. All results reproduction must use Flashlight <= 0.3.2 for exact reproducibility. At least one of LZMA, BZip2, or Z is required for LM compression with KenLM. It is highly recommended to build KenLM with position-independent code (-fPIC) enabled, to enable python compatibility. After installing, run... -
18
PlusV
An Open Source alternative to SBR (Spectral band replication)
...PlusV is not a compression scheme of its own, it is an extension that can be applied to existing audio formats. When combined with the MP3 technology, MP3+V files are fully compatible with existing MP3 files and decoders. To get full audio quality out of PlusV files, you just need a PlusV capable decoder, like a PlusV capable WinAmp plugin. -
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JuliusModels
Open source speech models for Julius in English and other languages.
Open source speech models for Julius speech decoder. Its aim is to give access a wider community of speech recognition enthusiasts to quality models, which they can use in their own projects on different OS platforms (Unix, Windows, etc...) All of the models are based on HTK modelling software and data sets available freely on the Internet. -
20
JSMpeg
MPEG1 Video Decoder in JavaScript
JSMpeg is a lightweight JavaScript MPEG1 video decoder and player designed for low-latency video streaming directly inside web browsers using HTML5 technologies. The project decodes MPEG video and MP2 audio entirely in JavaScript, allowing real-time playback without requiring plugins or native browser codec support. It is commonly used in applications involving webcam streaming, surveillance systems, remote monitoring, game streaming, and embedded video dashboards because of its low overhead and simplicity. jsmpeg supports WebSocket streaming pipelines, enabling live video feeds to be transmitted efficiently to browser clients with minimal latency. ... -
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FLAC is a free lossless compressed audio format which supports streaming and archival. The FLAC project maintains the format and provides a reference encoder/decoder and input plugins for several popular audio players.
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WS1281 decoder
Decoding of OOK signals transmitted by a weather sensor on 433,82 Mhz
WS1281DEC is a portable Win32 application that extract and display the data transmitted by WS1281 weather station external sensor, using the sound device of a PC computer and some additional hardware and software. Hardware requirements are RTL-SDR dongle (e4000 or R820T tuner ), external antenna, and a PC. As additional software, it is necessary to have SDR Sharp and Virtual Audio Cable. The whole system was tested on Windows 7 64 bit OS -
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simplest ffmpeg audio player
The simplest audio player based on FFmpeg
...simplest_ffmpeg_audio_decoder: Audio decoder. Uses libavcodec and libavformat. simplest_audio_play_sdl2: Example about using SDL2 play PCM data. -
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ArduinoRCLib
Library for Arduino based R/C equipment
A library containing functions for building Arduino based R/C equipment like transmitters, receivers and flight controllers. -
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Aurora.js
JavaScript audio decoding framework
Aurora.js is a JavaScript framework that simplifies audio decode pipeline implementation—covering source, demux, decode, and playback stages. It provides high-level APIs for inspecting and playing audio, supports plugins for decoders (MP3, AAC, FLAC), works in browser and Node.js (via Browserify), and abstracts browser audio APIs.
