Open-source alternatives to Speech Recognition

Ecoute is a live transcription tool that provides real-time transcripts for both the user's microphone input (You) and the user's speakers output (Speaker) in a textbox.

EZAudio is an audio library for Apple platforms that provides standardized interfaces for microphone capture, file playback, and hardware output. It functions as a low-latency audio processor and visualization framework designed to manipulate audio buffers and route signals with minimal delay. The project features a hardware-accelerated waveform renderer for drawing real-time audio amplitudes and rolling plots. It also includes a Fast Fourier Transform analyzer that converts time-domain audio samples into frequency-domain data for spectral analysis. The library covers a broad range of capabi

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-

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

This project is a self-hosted meeting transcription and summarization tool that converts audio recordings into text transcripts and structured notes using large language models. It functions as an enterprise meeting documentation manager, allowing for the organization and editing of timestamped records. The system prioritizes data privacy through local-first processing and the ability to deploy on private infrastructure. It supports a provider-agnostic architecture, enabling users to connect to local AI engines, self-hosted servers, or cloud-based API endpoints for both transcription and summ

RealtimeSTT is a local speech-to-text engine and real-time automatic speech recognition server. It utilizes transformer-based recognition and omnilingual pipelines to convert live audio streams into text, providing a WebSocket-based streaming API for raw PCM audio transmission. The project is distinguished by a dual-backend transcription pipeline that uses a lightweight engine for immediate partial suggestions and a heavier model for final high-accuracy results. It includes a wake word detection system to trigger recording and employs a shared-resource inference model to distribute heavy spee

Casibase is an open-source platform that orchestrates multi-turn conversations with large language models and manages retrieval-augmented knowledge bases from a single interface. It provides a unified system for connecting to over 30 AI model providers, ingesting documents into vector embeddings for semantic search, and running autonomous agent loops that can drive a browser, search the web, execute commands, and integrate with external tools. The platform distinguishes itself by combining AI conversation management with infrastructure and application orchestration capabilities. It includes a

Sherpa-ncnn is an edge-based speech recognition and synthesis engine designed to run neural network models locally on mobile, embedded, and desktop hardware. It provides a cross-platform framework for offline speech-to-text transcription and text-to-speech synthesis, ensuring that all audio processing occurs on-device without requiring an internet connection or external cloud services. The project distinguishes itself through its use of the ncnn inference engine, which is optimized for low-latency execution on resource-constrained devices. It incorporates on-device model quantization to reduc

This project is a collection of Android application reference implementations and sample projects. It provides educational source code demonstrating standard development patterns for building functional mobile applications. The repository includes specific examples for Android UI design, data persistence, and hardware integration. It provides reference implementations for handling long-running background processes and system broadcasts, as well as demonstrations of building layouts and custom graphics. The codebase covers a broad range of capabilities including network programming for remote

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

This project provides a self-hosted server for automatic speech recognition, functioning as a containerized inference engine for the Whisper model. It exposes core transcription and translation capabilities through a standardized web interface, allowing for the integration of speech-to-text services into external applications. The service distinguishes itself by incorporating advanced audio analysis tools, including speaker diarization to attribute text to specific individuals and voice activity detection to filter non-speech segments. It supports automated language detection and provides out

GPAC is an open-source multimedia framework built around a pluggable filter graph pipeline, where modular processing units called filters connect into a directed graph to handle media workflows. At its core, the framework centers all media packaging and manipulation on the ISO Base Media File Format (ISOBMFF), with specialized tools for reading, writing, fragmenting, and encrypting MP4 and related containers. It also provides a declarative scene graph composition system for describing interactive multimedia scenes using MPEG-4 BIFS, X3D, SVG, or VRML syntax, alongside a hardware-accelerated re

Pydub is a Python audio manipulation library and digital audio processor used for editing, slicing, and converting audio files and segments. It serves as a programmatic wrapper for FFmpeg to import and export a wide variety of audio formats. The library functions as an audio signal generator capable of creating synthetic waveforms, such as sine waves and white noise. It also provides tools for digital signal processing, including the application of filters, fades, crossfades, and gain adjustments to sound signals. Its broader capabilities cover programmatic audio editing through concatenatio

Opus is a lossy audio compression standard and codec designed for high-quality speech and music transmission over the internet. It functions as a low-latency audio codec and network-resilient streamer, providing a framework for encoding and decoding digital audio. The project distinguishes itself through the support of multi-channel ambisonics for immersive three-dimensional spatial audio reproduction. It is specifically optimized for real-time interactive communication, utilizing dynamic bitrate adjustment and forward error correction to maintain audio quality on unstable networks. The syst

Audacity is a cross-platform digital audio workstation and multi-track audio editor. It serves as a comprehensive suite for capturing live audio input, refining sound files through splicing and effects, and mixing multi-track audio files using a non-destructive waveform interface. The project functions as a VST3 plugin host, providing a software environment to load and execute audio effects and virtual instruments for real-time signal processing. It also includes an audio spectrum analyzer for visualizing frequencies and waveforms to identify specific sonic characteristics. The software cove

MPD is a headless music server daemon that indexes audio libraries and streams music to local or remote outputs. It functions as a music library manager and network audio streamer, providing a remote audio control protocol that allows external clients to manage playback, playlists, and database queries. The system acts as a multiroom audio coordinator, synchronizing audio distribution across multiple networked clients and hardware devices. It supports a variety of remote management capabilities, including a dedicated control API and the ability to broadcast audio streams over network protocol

Geist is an open-source font family and typography collection designed for high legibility in technical interfaces. It consists of a series of web-optimized typefaces, including geometric sans-serif, monospaced, and pixel styles. The collection functions as a variable font library, utilizing coordinate interpolation to allow precise control over weight and style within a single font file. These fonts are built as OpenType typefaces, incorporating standardized layout tables to define advanced typographic behaviors such as kerning and ligatures. The project provides specific implementations fo

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

Vocode-python is an LLM voice AI framework and orchestration library used to build real-time conversational agents. It functions as a speech-to-speech pipeline that integrates large language models with speech-to-text and text-to-speech services to facilitate continuous spoken dialogue. The framework acts as a telephony integration gateway, connecting voice agents to phone lines and video conferencing platforms through automated calling infrastructure. It provides the means to deploy voice channels to external communication platforms and host telephony servers. The library manages the end-to

Screenpipe is a local screen and audio recorder that captures and indexes digital activity to create a searchable archive of computer usage. It functions as an AI context engine, providing a local database of visual and auditory history to ground large language models. The system serves as a Model Context Protocol server, delivering screen history and meeting transcriptions to external AI assistants. It utilizes an OCR screen search tool to extract text from visual data and a speech-to-text transcription tool for identifying speakers in system and microphone audio. The software includes capa

PocketSphinx is an offline speech recognition engine that converts raw audio from files or live microphone streams into written text without requiring a network connection. It functions as a speech-to-text library, a real-time transcription engine, and a voice command processor, capable of detecting and transcribing spoken commands from continuous audio streams with configurable acoustic and language models. The engine uses weighted finite-state transducers to represent acoustic, phonetic, and language models as a single search graph for efficient decoding. It employs fixed-point acoustic mod

Pipecat is a framework and software development kit for building real-time multimodal AI agents and speech-to-speech systems. It utilizes a frame-based data pipeline to route audio, video, and text through a modular sequence of processors, enabling the orchestration of low-latency conversational AI. The project is distinguished by its ability to coordinate complex multimodal services, including speech-to-text, language models, and text-to-speech, within a single pipeline. It features semantic voice activity detection for natural turn-taking, state-machine conversation flows for dialogue manag

Fonoster is a conversational AI framework and multi-tenant communications platform as a service. It serves as a programmable voice gateway and SIP telephony platform, enabling the creation of voice-based assistants and automated communication workflows using large language models. The project distinguishes itself through a vendor-agnostic speech integration engine that abstracts speech-to-text and text-to-speech providers. It features a multi-tenant architecture that isolates telephony resources and user identities into distinct organizational workspaces. The system covers a broad range of t

This project is a comprehensive framework for building AI-powered applications, providing a unified toolkit for orchestrating language models, autonomous agents, and interactive user interfaces. It serves as a central library for managing the entire lifecycle of AI interactions, from initial prompt generation and model provider abstraction to complex, multi-step reasoning and tool execution. The framework distinguishes itself through its deep integration with frontend development, specifically by enabling generative user interfaces that render dynamic components directly from model outputs. I

This project is a speech recognition and translation engine that utilizes a sequence-to-sequence transformer architecture to convert audio into text. It is built upon a weakly supervised learning framework, which leverages large-scale, unlabelled audio-transcript data to create generalized speech representations capable of performing simultaneous transcription, language identification, and translation. The system distinguishes itself through a unified multi-task modeling approach that shares token sequences across different objectives, allowing it to handle diverse languages and vocabularies

Wechaty is a cross-platform chatbot framework designed to build and manage automated messaging agents. It provides a unified programming interface that abstracts diverse instant messaging protocols, allowing developers to create bots that function consistently across multiple communication services. By utilizing a modular architecture, the framework enables the development of conversational agents capable of handling complex messaging workflows, contact management, and group room interactions. The framework distinguishes itself through a puppet-based protocol abstraction and a language-agnost