Awesome GitHub RepositoriesSpeech Recognition

Whisper.cpp is a high-performance, local-first speech recognition engine designed to run large-scale machine learning models on consumer hardware. It functions as a portable library that converts audio into text, supporting both static file transcription and real-time stream processing. By utilizing a lightweight inference engine and weight quantization, the project minimizes memory and compute overhead, allowing for efficient execution without reliance on external cloud APIs or internet connectivity. The project distinguishes itself through a hardware-agnostic compute abstraction that offloa

This project is a comprehensive Chinese translation of a technical deep learning textbook, providing an educational resource on the theory and implementation of neural networks. It functions as a collaborative technical translation project designed to make complex academic AI literature accessible to non-English speakers. The project utilizes a community-driven translation model that integrates external suggestions and pull requests to refine linguistic accuracy and reduce bias. It employs standardized terminology mapping to ensure a uniform vocabulary throughout the translated content. To i

DeepSpeech este un framework open-source de tip speech-to-text și un motor de învățare automată conceput pentru a converti audio vorbit în text scris local pe un dispozitiv. Oferă recunoaștere vocală pe dispozitiv care funcționează fără a necesita o conexiune la internet către servere externe. Sistemul acceptă transcrierea vorbirii în timp real pe o varietate de platforme hardware, variind de la computere cu placă unică și dispozitive edge până la servere GPU. Acest lucru permite analiza și procesarea audio direct pe hardware-ul local.

AgenticSeek is a multi-agent orchestration system designed to decompose complex user objectives into granular, actionable tasks. By coordinating a team of specialized autonomous workers, the platform manages end-to-end workflows, ensuring that each component of a project is assigned to the most capable agent for execution. The system operates as a local-first runtime, executing all artificial intelligence models directly on user hardware to maintain data sovereignty and privacy. It integrates a browser automation engine for autonomous web research and interaction, alongside a sandboxed enviro

LiveKit is a comprehensive framework for building and orchestrating real-time, multimodal AI agents that interact with users through voice, video, and text. It provides a centralized, event-driven architecture to manage the entire lifecycle of automated participants, from initialization and session state management to graceful shutdown. By utilizing a selective forwarding unit, the platform efficiently routes media streams between participants and agents, ensuring low-latency communication and secure, token-based authentication for all connections. The platform distinguishes itself through it

Vosk is an offline speech-to-text engine and API that converts spoken audio into text locally on a device. It provides a cross-platform speech toolkit with language bindings for integrating voice recognition into server environments, Android, iOS, and Raspberry Pi. The project includes a speaker identification tool to distinguish between different voices and an acoustic model trainer for building custom neural network models. These training tools enable speech feature extraction and model accuracy evaluation to improve recognition for specialized domains. The system supports real-time audio

Omi is an open-source wearable AI platform that captures audio and screen data to provide real-time conversational assistance and memory. It integrates a wearable hardware development kit with a vector memory database and large language model capabilities to create a persistent digital record of user interactions. The platform is distinguished by its BLE audio streaming pipeline, which transmits raw audio from wearable hardware for real-time transcription and speaker identification. It utilizes a plugin-based agent tool framework that allows AI assistants to autonomously invoke custom functio

ESPnet is a comprehensive speech processing toolkit and PyTorch-based trainer designed for building end-to-end speech recognition, synthesis, and translation models. It provides a structured framework for developing automatic speech recognition systems using transducer and encoder-decoder architectures, alongside engines for text-to-speech synthesis and speech translation pipelines. The project distinguishes itself through a recipe-based workflow execution system that ensures experimental reproducibility by running standardized sequences of scripts for data preparation and model training. It

This repository is a comprehensive collection of reference implementations and sample libraries for the Universal Windows Platform. It provides practical examples of how to use Windows Runtime APIs to build cross-device applications, including detailed guidance on XAML-based declarative user interfaces and DirectX-integrated rendering. The project distinguishes itself by providing a wide array of hardware integration suites, covering low-level communication with USB, Serial, I2C, SPI, and GPIO peripherals. It includes specialized implementations for mixed reality holographic rendering, advanc

This project is an AI-powered IDE extension and LLM coding assistant that provides a conversational interface for generating, refactoring, and debugging code. It functions as an AI agent framework and a Model Context Protocol client, connecting AI models to external data sources and tools to automate complex development tasks. The system is distinguished by its use of autonomous AI agents capable of multi-step task execution, including the ability to read files, modify code, and run terminal commands iteratively. It supports recursive agent orchestration through subagent delegation and employ

This project is a framework for developing multimodal AI agents that function as programmable participants in real-time communication rooms. It enables the construction of agents that can see, hear, and speak by integrating speech-to-text, large language models, and text-to-speech pipelines to facilitate low-latency, natural conversations. The system is distinguished by its advanced orchestration of real-time media and conversational flow, including support for full-duplex speech, preemptive response generation, and sophisticated interruption management. It further differentiates itself throu

Moonshine is a complete on-device voice interface toolkit that provides speech recognition, text-to-speech synthesis, phonetic processing, speaker diarization, and intent recognition, all running locally on edge hardware without any cloud dependency. It executes quantized neural networks for speech and language tasks directly on the device, enabling fully offline conversational AI capabilities. The toolkit distinguishes itself by orchestrating multi-turn spoken exchanges through a conversational flow manager that maintains context across interactions and manages branching dialog flows. It inc

This project is a cross-platform messaging SDK and client development library used to build custom Telegram applications. It functions as a comprehensive framework that manages network encryption, local data storage, and API communication, providing a C-compatible JSON interface that allows integration with any programming language. The library distinguishes itself by providing a full database manager for encrypted local caching and synchronized state, alongside a dedicated bot framework for creating interactive bots with business account integration. It enables the implementation of speciali

This project is a Chinese automatic speech recognition framework and deep learning system designed to convert spoken Chinese audio into written text. It functions as a toolkit for training, evaluating, and deploying speech-to-text models, utilizing a specialized pinyin-to-text converter that transforms phonetic sequences into Chinese characters using a probability graph model. The system is distinguished by its deployment flexibility, offering a dockerized recognition server that provides transcription capabilities as a remote API. It supports high-performance streaming through a gRPC speech-

Annyang is a speech recognition library and web speech API wrapper that enables the integration of voice command interfaces into websites. It functions as a browser-based voice controller, mapping spoken phrases and regular expressions to specific JavaScript functions to trigger application actions. The library provides mechanisms for voice command mapping and simulation, allowing developers to associate spoken text with executable callbacks. It includes tools for command variable extraction using regular expression capture groups, which allows specific words from a spoken phrase to be passed

CTranslate2 is a C++ inference engine and runtime for Transformer models, designed to execute models on both CPU and GPU with optimizations for speed and memory efficiency. It functions as a model format converter, quantization tool, and REST API server, enabling deployment of neural machine translation, automatic speech recognition, and text generation models. The engine distinguishes itself through a suite of runtime optimizations including layer fusion, weight-matrix quantization, batch-by-length grouping, and a caching allocator that reuses GPU memory. It supports tensor-parallel model di

Rhubarb is an automated lip sync generator and phonetic speech analyzer that converts audio recordings into timed mouth-shape animation data. It identifies sounds and syllables within audio files to map them to specific visual mouth shapes, serving as an animation timing exporter for external character animation software. The tool utilizes a language-independent phonetic recognizer to process speech regardless of the spoken language. To increase accuracy, it supports dialogue-guided recognition by using external text files to guide the phonetic analysis of specific spoken scripts. The system

Python speech features is a signal processing toolkit and library for extracting standard speech recognition features from raw audio signals. It provides computational capabilities to calculate mel-frequency cepstral coefficients, raw and log filterbank energies, and spectral subband centroids for automated speech recognition systems and acoustic analysis. The library implements audio signal transformations including pre-emphasis filtering, overlapping windowed frame segmentation, Fast Fourier Transform spectral analysis, mel-filterbank projection with configurable filter weights and frequenc

Acest proiect este un toolkit cuprinzător pentru recunoașterea vocală on-device, sinteză și procesare audio, conceput special pentru Apple Silicon. Oferă un framework pentru construirea de agenți vocali full-duplex în timp real care operează complet offline, valorificând accelerarea hardware nativă pentru a menține performanța și confidențialitatea. Prin utilizarea modelelor de machine learning optimizate, biblioteca permite execuția locală a sarcinilor audio complexe fără dependență de servicii cloud externe. Biblioteca se distinge prin accentul său specializat pe interacțiunea vocală locală, de înaltă performanță. Include orchestrare sofisticată pentru pipeline-uri audio de streaming, permițând transcrierea în timp real, sinteza vocală și clonarea vocii cu latență scăzută. Sistemul este conceput pentru a gestiona conversații interactive, continue, având mecanisme încorporate pentru a preveni buclele de feedback audio și a gestiona sesiunile de streaming persistente. Dincolo de interacțiunea de bază, proiectul oferă o suită largă de capabilități de îmbunătățire și gestionare audio. Suportă procesarea avansată a semnalului, inclusiv separarea surselor, reducerea zgomotului și upsampling audio, alături de instrumente pentru diarizarea vorbitorilor și extracția de embedding-uri. Framework-ul oferă, de asemenea, utilitare extinse de gestionare a modelelor, cum ar fi controale de cuantizare, gestionarea memoriei și suport pentru încărcarea ponderilor de modele personalizate, asigurându-se că dezvoltatorii pot echilibra viteza de procesare și consumul de resurse pe hardware local. Proiectul include o interfață CLI pentru executarea sarcinilor audio și conversia ponderilor modelelor în formate optimizate. De asemenea, expune endpoint-uri HTTP și WebSocket pentru a facilita integrarea cu interfețele standard din industrie.