5 repositorios
Software implementations that translate compressed audio bitstreams back into raw waveforms.
Distinct from Audio Format Decoders: Candidates focus on resampling or specific sample extraction, not the core decoding process of a codec.
Explore 5 awesome GitHub repositories matching graphics & multimedia · Audio Codec Decoders. Refine with filters or upvote what's useful.
This project is a Python speech recognition library that serves as a unified interface for converting spoken audio into text. It functions as a bridge between Python applications and a variety of speech-to-text engines, providing a consistent way to interact with both local and cloud-based recognition services. The library distinguishes itself as a multi-engine transcription tool, wrapping diverse online APIs and offline recognition backends into a standardized format. This allows for interchangeable recognition engines and supports multilingual audio transcription through various language pa
Translates various audio file containers like WAV and FLAC into raw waveforms for transcription processing.
LAVFilters is an open-source media filter pack consisting of splitters and decoders designed for the DirectShow framework. It functions as a set of software components that convert compressed media data into raw formats for hardware or software rendering. The collection includes a media splitter to separate combined audio and video container formats into individual streams, as well as a specialized Blu-ray disc player filter used to identify and extract movie titles and playlists from disc structures. The project provides capabilities for audio and video stream decoding, media stream demuxin
Translates compressed audio bitstreams back into raw waveforms compatible with system hardware.
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
Translates compressed bitstreams back into raw audio for playback or further processing.
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
Parses various compressed and lossless audio formats into raw waveforms for playback.
Este proyecto es un toolkit integral para el reconocimiento de voz, síntesis y procesamiento de audio en el dispositivo, diseñado específicamente para Apple Silicon. Proporciona un framework para construir agentes de voz full-duplex en tiempo real que operan completamente offline, aprovechando la aceleración de hardware nativa para mantener el rendimiento y la privacidad. Al utilizar modelos de machine learning optimizados, la biblioteca permite la ejecución local de tareas de audio complejas sin depender de servicios externos en la nube. La biblioteca se distingue por su enfoque especializado en la interacción de voz local de alto rendimiento. Incluye una orquestación sofisticada para pipelines de audio en streaming, permitiendo la transcripción en tiempo real, síntesis de voz y clonación de voz con baja latencia. El sistema está diseñado para manejar conversaciones interactivas continuas, presentando mecanismos integrados para evitar bucles de retroalimentación de audio y gestionar sesiones de streaming persistentes. Más allá de la interacción central, el proyecto ofrece un amplio conjunto de capacidades de mejora y gestión de audio. Admite procesamiento de señales avanzado, incluyendo separación de fuentes, reducción de ruido y sobremuestreo de audio, junto con herramientas para diarización de hablantes y extracción de embeddings. El framework también proporciona amplias utilidades de gestión de modelos, como controles de cuantización, gestión de memoria y soporte para la carga de pesos de modelos personalizados, asegurando que los desarrolladores puedan equilibrar la velocidad de procesamiento y el consumo de recursos en hardware local. El proyecto incluye una interfaz de línea de comandos para ejecutar tareas de audio y convertir pesos de modelos en formatos optimizados. También expone endpoints HTTP y WebSocket para facilitar la integración con interfaces estándar de la industria.
Translates compressed audio bitstreams back into raw waveforms.