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 ist ein Open-Source-Spracherkennungs-Framework und eine Machine-Learning-Engine, die darauf ausgelegt ist, gesprochenes Audio lokal auf einem Gerät in geschriebenen Text umzuwandeln. Es bietet eine On-Device-Spracherkennung, die ohne Internetverbindung zu externen Servern funktioniert. Das System unterstützt die Echtzeit-Sprachtranskription auf einer Vielzahl von Hardwareplattformen, von Einplatinencomputern und Edge-Geräten bis hin zu GPU-Servern. Dies ermöglicht die Audioanalyse und -verarbeitung direkt auf der lokalen Hardware.

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

Dieses Projekt ist ein umfassendes Toolkit für On-Device-Spracherkennung, -Synthese und Audioverarbeitung, das speziell für Apple Silicon entwickelt wurde. Es bietet ein Framework für den Aufbau von Echtzeit-Voice-Agents mit Vollduplex-Funktionalität, die vollständig offline arbeiten und native Hardwarebeschleunigung nutzen, um Performance und Datenschutz zu wahren. Durch den Einsatz optimierter Machine-Learning-Modelle ermöglicht die Bibliothek die lokale Ausführung komplexer Audioaufgaben ohne Abhängigkeit von externen Cloud-Diensten. Die Bibliothek zeichnet sich durch ihren spezialisierten Fokus auf lokale, hochperformante Sprachinteraktion aus. Sie enthält eine ausgefeilte Orchestrierung für Streaming-Audio-Pipelines, die Echtzeit-Transkription, Sprachsynthese und Voice-Cloning mit geringer Latenz ermöglicht. Das System ist für die Handhabung kontinuierlicher, interaktiver Konversationen konzipiert und verfügt über integrierte Mechanismen zur Vermeidung von Audio-Feedback-Schleifen und zur Verwaltung persistenter Streaming-Sitzungen. Über die Kerninteraktion hinaus bietet das Projekt eine breite Palette an Audio-Enhancement- und Management-Funktionen. Es unterstützt fortgeschrittene Signalverarbeitung, einschließlich Quellentrennung, Rauschunterdrückung und Audio-Upsampling, neben Tools für Sprecher-Diarisierung und Embedding-Extraktion. Das Framework bietet zudem umfangreiche Modellmanagement-Utilities, wie z. B. Quantisierungskontrollen, Speicherverwaltung und Unterstützung für das Laden benutzerdefinierter Modellgewichte, um sicherzustellen, dass Entwickler Verarbeitungsgeschwindigkeit und Ressourcenverbrauch auf lokaler Hardware ausbalancieren können. Das Projekt enthält eine CLI für die Ausführung von Audioaufgaben und die Konvertierung von Modellgewichten in optimierte Formate. Es stellt zudem HTTP- und WebSocket-Endpunkte bereit, um die Integration mit Standard-Industrieschnittstellen zu erleichtern.