Search Results for "audio waveform java"
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50 projects for "audio waveform java" with 2 filters applied:
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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... -
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WavTokenizer
SOTA discrete acoustic codec models with 40/75 tokens per second
...Its architecture incorporates a broader vector-quantization space, extended contextual windows, and improved attention networks, combined with multi-scale discriminators and inverse Fourier transform blocks to enhance waveform reconstruction. Extensive experiments show that WavTokenizer matches or surpasses previous neural codecs across speech, music, and general audio on both objective metrics and subjective listening tests. -
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II ElevenLabs UI
Component library and custom registry built on top of shadcn/ui
ElevenLabs UI is an open-source component library designed to accelerate the development of multimodal AI applications, particularly those involving voice agents and audio-based interactions. Built on top of modern frontend tooling such as React, Tailwind CSS, and shadcn/ui, it provides a collection of pre-built, customizable components that developers can easily integrate into their applications. The library includes specialized UI elements such as audio players, waveform visualizers, conversational interfaces, and interactive voice components, all tailored for building agent-driven experiences. ... -
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WhisperSpeech
An Open Source text-to-speech system built by inverting Whisper
WhisperSpeech is an open-source text-to-speech system created by “inverting” OpenAI’s Whisper, reusing its strengths as a semantic audio model to generate speech instead of only transcribing it. The project aims to be for speech what Stable Diffusion is for images: powerful, hackable, and safe for commercial use, with code under Apache-2.0/MIT and models trained only on properly licensed data. Its architecture follows a token-based, multi-stage pipeline inspired by AudioLM and SPEAR-TTS: Whisper is used to produce semantic tokens, EnCodec compresses the waveform into acoustic tokens, and Vocos reconstructs high-fidelity audio from those tokens. ... -
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Spring AI Alibaba Examples
Spring AI Alibaba examples for building and testing AI apps
Spring AI Alibaba Examples provides a collection of example projects that demonstrate how to use Spring AI and Spring AI Alibaba across different scenarios, from basic setups to more advanced AI applications. It is designed to help developers understand core concepts, explore practical implementations, and follow best practices when building AI-powered systems using the Spring ecosystem. Each module focuses on a specific use case such as chat, image processing, audio handling, graph... -
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Deep Chat
Customizable AI chat component for websites with API support
Deep Chat is a highly customizable web component designed to simplify the integration of AI-powered chat interfaces into websites. It allows developers to embed a fully functional chatbot using minimal setup, while still offering extensive control over behavior, appearance, and integrations. Deep Chat supports connections to a wide range of AI services as well as custom backends, enabling flexible deployment for different use cases. It is built as a framework-agnostic solution, meaning it... -
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eGuideDog project develops free software for the blind. Currently, we focus on WebSpeech, Ekho TTS and WebAnywhere.
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Parallel WaveGAN
Unofficial Parallel WaveGAN
Parallel WaveGAN is an unofficial PyTorch implementation of several state-of-the-art non-autoregressive neural vocoders, centered on Parallel WaveGAN but also including MelGAN, Multiband-MelGAN, HiFi-GAN, and StyleMelGAN. Its main goal is to provide a real-time neural vocoder that can turn mel spectrograms into high-quality speech audio efficiently. The repository is designed to work hand-in-hand with ESPnet-TTS and NVIDIA Tacotron2-style front ends, so you can build complete TTS or singing... -
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Common Resource Grep - crgrep
Common Resource Grep
CRGREP searches for matching text in databases, various document formats, archives and other difficult to access resources. A command line tool for name and content text matching in database tables, plain files, MS Office documents, PDF, archives, MP3 audio, image meta-data, scanned documents, maven dependencies and web resources. CRGREP will search resources within resources of any arbitrary combination or depth, so text within a document within a zip archive, and so on. Here you... -
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WaveRNN
WaveRNN Vocoder + TTS
WaveRNN is a PyTorch implementation of DeepMind’s WaveRNN vocoder, bundled with a Tacotron-style TTS front end to form a complete text-to-speech stack. As a vocoder, WaveRNN models raw audio with a compact recurrent neural network that can generate high-quality waveforms more efficiently than many traditional autoregressive models. The repository includes scripts and code for preprocessing datasets such as LJSpeech, training Tacotron to produce mel spectrograms, training WaveRNN on those... -
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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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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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HiFi-GAN
Generative Adversarial Networks for Efficient and High Fidelity Speech
...It introduces a generator architecture tailored to model the periodic structure of speech and a set of discriminators that focus on different scales and periods of the waveform to better capture naturalness. The model targets a sweet spot between sample quality and generation speed, outperforming many previous GAN vocoders while being far faster than typical autoregressive models. In experiments on LJSpeech, HiFi-GAN was shown to achieve mean opinion scores close to human recordings while synthesizing 22.05 kHz audio up to ~168× faster than real time on an NVIDIA V100 GPU. ... -
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Tacotron-2
DeepMind's Tacotron-2 Tensorflow implementation
...Separate log trees track mel-spectrograms, attention plots, evaluation audio, and vocoder outputs, so you can inspect how alignment and audio quality evolve over time. -
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jMIR
Music research software
jMIR is an open-source software suite implemented in Java for use in music information retrieval (MIR) research. It can be used to study music in the form of audio recordings, symbolic encodings and lyrical transcriptions, and can also mine cultural information from the Internet. It also includes tools for managing and profiling large music collections and for checking audio for production errors. jMIR includes software for extracting features, applying machine learning algorithms, applying heuristic error error checkers, mining metadata and analyzing metadata. -
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ILA - teachable voice assistant
ILA is a fully customizable and teachable voice assistant for Java
ILA stands for (kind of) intelligent, learning assistant and is a speech recognition system aka voice assistant very similar to Siri, Google Now and Cortana. ILA is fully customizable and you can teach her/him/it new things by yourself like executing system commands, opening web pages, programs and apps or just some basic conversation :-) ILA runs on Java und thus is compatible to Windows, Mac and Linux. It is designed to integrate with your home enviroment and for example build up your own,... -
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DC-TTS
TensorFlow Implementation of DC-TTS: yet another text-to-speech model
DC-TTS is a TensorFlow implementation of the DC-TTS architecture, a fully convolutional text-to-speech system designed to be efficiently trainable while producing natural speech. It follows the “Efficiently Trainable Text-to-Speech System Based on Deep Convolutional Networks with Guided Attention” paper, but the author adapts and extends the design to make it practical for real experiments. The model is split into two networks: Text2Mel, which maps text to mel-spectrograms, and SSRN... -
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MARF is a general cross-platform framework with a collection of algorithms for audio (voice, speech, and sound) and natural language text analysis and recognition along with sample applications (identification, NLP, etc.) of its use, implemented in Java.
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jaivox
Speech recognition application builder and library
Java library and tools to create open source speech recognition applications. Generates dialogs for conversational interfaces. Works with a popular open source speech recognition library. -
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Bermuda Text-to-Speech
This project includes basic NLP and DSP techniques for Text-to-Speech
See TTS demo at: http://rslp.racai.ro/index.php?page=tts This is an entirely written in JAVA project which includes a set of tools and methods designed to enable Multilingual Text-to-Speech (TTS) synthesis. We currently support English and Romanian but we will soon train more models and make them available for download. If you want to read more about our other NLP and TTS tools check out http://nlptools.racai.ro. -
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InproTK
An Incremental Spoken Dialogue Processing Toolkit
InproTK is an Incremental Spoken Dialogue Processing Toolkit, that is, a toolkit to help you build dialogue systems that listen and talk incrementally, allowing for advanced interactional behaviour. Please see our Wiki for more information: http://sourceforge.net/p/inprotk/wiki/ -
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HMM Speech Recognition in Java
HMM Speech Recognition in Java
HMM Speech Recognition in Java -
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A speech synthesis and recognition library that is cross-platform, accessible from Java and C++, and has a very small API. Uses CMU Sphinx4 and FreeTTS internally.
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High-order HMM in Java
A duration high-order hidden Markov model (DHO-HMM) in Java.
This project provides an implementation of duration high-order hidden Markov model (DHO-HMM) in Java. It is compactible with JDK 5 & 6. It was used in the author's research on speech recognition of Mandarin digits. There are some Chinese words in this project and I am afraid that I don't have enough time to translate to English recently. -
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Anthromorphic Scribe
Provides speech to text gui to sphinx4
It provides an interactive speech to text application that uses sphinx 4. With this you can use pre-recorded audio, record your own voice and convert incompatible audio/video to be compatible with sphinx 4. It currently supports U.S English by using hub4 acoustic and language model.
