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53 repositorios

Awesome GitHub RepositoriesAudio Processing

Tools and models for analyzing, transforming, and synthesizing sound signals.

Distinguishing note: Focuses on audio-specific signal processing and machine learning tasks, distinct from general data processing.

Explore 53 awesome GitHub repositories matching artificial intelligence & ml · Audio Processing. Refine with filters or upvote what's useful.

Awesome Audio Processing GitHub Repositories

Encuentra los mejores repositorios con IA.Buscaremos los repositorios que mejor coincidan usando IA.
  • openbmb/voxcpmAvatar de OpenBMB

    OpenBMB/VoxCPM

    29,985Ver en GitHub↗

    VoxCPM is a multilingual speech synthesis system and text-to-speech inference server. It functions as an AI voice cloning tool and a synthetic voice designer, capable of generating natural speech across global languages and regional dialects using a GPU-accelerated audio generator. The project features a speech model fine-tuning framework that supports both full parameter updates and low-rank adaptation for customizing voice characteristics. It enables high-fidelity voice cloning from reference audio, including cross-lingual voice transfer and acoustic environment mimicry, as well as the crea

    Offers professional audio utilities for denoising reference clips and upsampling low-resolution samples to studio quality.

    Pythonaudiodeeplearningminicpm
    Ver en GitHub↗29,985
  • eugeneyan/applied-mlAvatar de eugeneyan

    eugeneyan/applied-ml

    29,783Ver en GitHub↗

    This project is a comprehensive, curated knowledge base designed to support the development and maintenance of production-grade machine learning systems. It serves as a centralized repository of industry-standard technical literature, engineering case studies, and research papers, providing a structured reference for practitioners navigating the complexities of modern data science and machine learning engineering. The resource distinguishes itself through a cross-domain approach that bridges the gap between academic research and practical implementation. By synthesizing proven industry archit

    Analyze and transform sound signals into digital representations for tasks like speech recognition, classification, or generative audio synthesis.

    applied-data-scienceapplied-machine-learningcomputer-vision
    Ver en GitHub↗29,783
  • openai/openai-agents-pythonAvatar de openai

    openai/openai-agents-python

    27,191Ver en GitHub↗

    This project is a Python framework for building autonomous, event-driven agent systems. It provides a unified runtime for orchestrating multi-agent workflows, managing persistent conversation state, and executing code within secure, isolated sandbox environments. The framework is designed to handle complex task delegation, allowing agents to invoke other agents as tools while maintaining context across multi-turn interactions. The framework distinguishes itself through its deep integration with the Model Context Protocol, enabling agents to connect to external data sources and remote services

    Processes static audio input buffers for voice pipeline ingestion and analysis.

    Pythonagentsaiframework
    Ver en GitHub↗27,191
  • guillaumekln/faster-whisperAvatar de guillaumekln

    guillaumekln/faster-whisper

    23,679Ver en GitHub↗

    faster-whisper is an automatic speech recognition framework and an optimized implementation of the Whisper speech-to-text engine. It functions as a CTranslate2 inference engine designed to convert spoken audio into written text. The project serves as a model quantization tool that transforms large audio model weights into lower precision formats. This process reduces memory usage and increases execution speed on hardware by utilizing integer quantized weights. The framework covers a broad range of capabilities including batch audio transcription for parallel processing and voice activity det

    Provides parallel processing of audio segments to maximize transcription throughput and reduce latency.

    Python
    Ver en GitHub↗23,679
  • anjok07/ultimatevocalremoverguiAvatar de Anjok07

    Anjok07/ultimatevocalremovergui

    23,673Ver en GitHub↗

    Ultimate Vocal Remover is a desktop application designed for AI-driven audio source separation. It utilizes deep learning models to isolate vocals, drums, and other individual instruments from mixed audio files, providing a utility for professional production and creative editing workflows. The software distinguishes itself by leveraging GPU-accelerated tensor computation to perform complex signal processing tasks, significantly reducing the time required for high-fidelity audio extraction. It incorporates a modular plugin architecture that integrates external utilities to support a wide rang

    Leverages GPU acceleration to perform complex audio source extraction and signal processing tasks.

    Pythonaudioinstrumentalkaraoke
    Ver en GitHub↗23,673
  • facebookresearch/audiocraftAvatar de facebookresearch

    facebookresearch/audiocraft

    23,379Ver en GitHub↗

    Audiocraft is a deep learning audio library and machine learning framework designed for training, fine-tuning, and evaluating generative models for music and sound effects. It functions as a text-to-music generative model and a neural audio codec, providing the tools necessary to compress audio signals into discrete representations and synthesize high-fidelity waveforms from textual descriptions. The framework is distinguished by its ability to combine multiple conditioning signals, allowing for the generation of audio based on text prompts, melodic excerpts, or style-based audio clips. It al

    Embeds invisible markers into audio signals to identify origin and protect content ownership.

    Jupyter Notebook
    Ver en GitHub↗23,379
  • resemble-ai/chatterboxAvatar de resemble-ai

    resemble-ai/chatterbox

    22,751Ver en GitHub↗

    Chatterbox is a comprehensive machine learning platform designed for multilingual speech synthesis and real-time audio generation. It functions as an engine that converts text into natural-sounding speech, capable of replicating specific human vocal characteristics and emotional expressions from short audio samples. The platform distinguishes itself through advanced control over the synthesis process, allowing for the manipulation of emotional intensity and the injection of non-verbal vocalizations such as laughter or coughing. It is engineered for low-latency performance, utilizing an optimi

    Embeds imperceptible digital signatures into generated audio to ensure reliable provenance tracking.

    Python
    Ver en GitHub↗22,751
  • microsoft/unilmAvatar de microsoft

    microsoft/unilm

    22,030Ver en GitHub↗

    This project is a comprehensive framework and toolkit for developing, optimizing, and deploying transformer-based models across multimodal, document intelligence, and natural language processing tasks. It provides a unified neural architecture that processes text, vision, audio, and document layout data through a shared set of weights, enabling researchers and developers to build foundational models that align cross-modal representations. The platform distinguishes itself through advanced training and inference strategies designed for large-scale deep learning. It incorporates specialized mec

    Applies fine-tuned acoustic models to categorize audio inputs into specific classes based on learned patterns.

    Pythonbeitbeit-3bitnet
    Ver en GitHub↗22,030
  • systran/faster-whisperAvatar de SYSTRAN

    SYSTRAN/faster-whisper

    21,043Ver en GitHub↗

    Faster-Whisper is a high-performance implementation of the Whisper speech-to-text model designed for efficient audio transcription. It provides an end-to-end processing pipeline that converts spoken audio into written text while maintaining lower memory consumption and faster execution speeds than standard implementations. The project achieves its performance through a specialized inference engine that utilizes optimized kernels and weight quantization to reduce computational complexity. It supports large-scale operations by grouping audio segments into dynamic batches and filtering out non-s

    Supports transcribing multiple audio segments or files simultaneously through a dedicated pipeline to increase throughput for large-scale tasks.

    Pythondeep-learninginferenceopenai
    Ver en GitHub↗21,043
  • harvard-edge/cs249r_bookAvatar de harvard-edge

    harvard-edge/cs249r_book

    20,217Ver en GitHub↗

    This project is a comprehensive educational framework designed to teach the design, deployment, and performance optimization of machine learning systems. It provides a structured curriculum that covers the full stack of artificial intelligence engineering, ranging from the construction of core framework components like tensors and automatic differentiation engines to the orchestration of large-scale distributed training clusters. The platform distinguishes itself through its integration of physics-grounded systems modeling and interactive simulation environments. Users can experiment with dis

    Implements on-device keyword spotting and voice command recognition for edge applications.

    JavaScriptartificial-intelligencecloud-mlcomputer-systems
    Ver en GitHub↗20,217
  • w-okada/voice-changerAvatar de w-okada

    w-okada/voice-changer

    19,729Ver en GitHub↗

    This software is a real-time voice changer that utilizes machine learning inference to transform live microphone input into target vocal characteristics. It functions as an artificial intelligence audio processing tool designed to modify vocal identity during active communication or live broadcasts. The application distinguishes itself by executing neural network models directly within the browser environment. It leverages web-based compute acceleration and dedicated audio threading to maintain low-latency performance, allowing users to switch between different voice profiles while processing

    Provides a utility for applying deep learning inference to microphone streams for low-latency voice conversion.

    Python
    Ver en GitHub↗19,729
  • livekit/livekitAvatar de livekit

    livekit/livekit

    19,358Ver en GitHub↗

    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

    Applies noise cancellation and signal conditioning to input audio to ensure high-quality voice recognition and interaction.

    Gogolangmedia-serversfu
    Ver en GitHub↗19,358
  • nvidia-nemo/nemoAvatar de NVIDIA-NeMo

    NVIDIA-NeMo/NeMo

    17,389Ver en GitHub↗

    NeMo is a comprehensive framework designed for the development, training, and deployment of large-scale conversational and generative artificial intelligence models. It provides an integrated platform for building multimodal systems, encompassing speech processing, language modeling, and reinforcement learning alignment. The framework is built to handle the entire lifecycle of AI development, from data curation and model pretraining to production-ready service deployment. The platform distinguishes itself through advanced distributed training capabilities, including tensor and pipeline parall

    Provides extensive tools for audio signal processing, including enhancement, restoration, and separation.

    Pythonasrdeeplearninggenerative-ai
    Ver en GitHub↗17,389
  • vercel/vercelAvatar de vercel

    vercel/vercel

    15,738Ver en GitHub↗

    Vercel is a cloud platform for building, deploying, and scaling web applications. It provides a unified infrastructure that automates the build process by detecting project frameworks and distributing static and dynamic content through a global content delivery network. The platform executes application logic using serverless functions that scale automatically based on real-time traffic demand. The platform distinguishes itself through a centralized AI gateway that proxies requests to multiple model providers, enabling standardized authentication, observability, and cost tracking. It supports

    Processes audio files to reduce background noise and improve sound clarity for interactive applications.

    TypeScriptclicloudcommand
    Ver en GitHub↗15,738
  • alibaba/mnnAvatar de alibaba

    alibaba/MNN

    14,242Ver en GitHub↗

    MNN is a high-performance inference engine and framework designed for on-device machine learning. It provides a comprehensive environment for executing, optimizing, and deploying neural network models directly on mobile and resource-constrained edge devices. The framework distinguishes itself through a robust model optimization toolkit that supports quantization, compression, and structural graph manipulation to minimize memory footprint and maximize execution speed. It features a modular architecture that abstracts hardware-specific backends, allowing models to run efficiently across diverse

    Extracts features and manipulates raw audio signals to prepare inputs for sound-based machine learning tasks.

    C++armconvolutiondeep-learning
    Ver en GitHub↗14,242
  • sidekiq/sidekiqAvatar de sidekiq

    sidekiq/sidekiq

    13,540Ver en GitHub↗

    Sidekiq is a background job processor and queue manager for Ruby that uses Redis to manage asynchronous tasks. It functions as a distributed task scheduler capable of handling periodic, delayed, and recurring jobs across a cluster of worker processes. The project features a job monitoring dashboard and administrative web interface for visualizing system state, tracking worker performance, and managing failed or dead jobs. It provides a distributed rate limiter to control execution frequency across multiple processes. The framework covers a broad range of operational capabilities, including j

    Allows administrators to locate specific jobs within the management interface using search criteria.

    Rubybackground-jobsjobsruby
    Ver en GitHub↗13,540
  • guofei9987/blind_watermarkAvatar de guofei9987

    guofei9987/blind_watermark

    13,405Ver en GitHub↗

    This is a blind image watermarking and steganography tool designed to embed and extract hidden data from images without requiring the original source file. It functions as a framework for concealing text or bit arrays within images using mathematical transforms to ensure the marks remain invisible to the viewer. The system is designed for robust watermark extraction, allowing hidden information to be recovered even after images have undergone rotations, cropping, resizing, noise injection, or brightness changes. It utilizes a blind extraction mechanism that retrieves data using a shared passw

    Retrieves hidden information from images that have been resized, cropped, or altered by noise and filters.

    Pythonblind-watermarkimage-processingwatermark
    Ver en GitHub↗13,405
  • chainlit/chainlitAvatar de Chainlit

    Chainlit/chainlit

    12,213Ver en GitHub↗

    Chainlit is a Python framework designed for building and deploying interactive, stateful conversational AI interfaces. It provides a backend-driven platform that connects language models and agent frameworks to a web-based chat frontend, managing the complexities of session state, message history, and real-time communication. The framework distinguishes itself by offering a component-based UI builder that allows developers to inject interactive widgets, rich media, and data visualizations directly into the chat stream. It supports the visualization of complex agent workflows, enabling users t

    Captures and processes audio segments from microphones for real-time voice interaction.

    Pythonchatgptlangchainllm
    Ver en GitHub↗12,213
  • speechbrain/speechbrainAvatar de speechbrain

    speechbrain/speechbrain

    11,624Ver en GitHub↗

    SpeechBrain is an all-in-one deep learning toolkit designed for speech and audio processing. Built as a modular library, it provides a structured environment for developing, training, and deploying neural network models across a wide range of tasks, including automatic speech recognition, speaker identification, and audio enhancement. The framework distinguishes itself through a configuration-driven approach that separates model architecture and training hyperparameters from application logic. By utilizing externalized configuration files and standardized recipes, it enables reproducible rese

    Provides a comprehensive toolkit for feature extraction, signal augmentation, and model inference across speech and audio tasks.

    Pythonasraudioaudio-processing
    Ver en GitHub↗11,624
  • wandb/wandbAvatar de wandb

    wandb/wandb

    10,844Ver en GitHub↗

    Wandb is a centralized platform for machine learning experiment tracking, model registry management, and workflow orchestration. It provides a comprehensive suite of tools for logging, visualizing, and versioning training metrics, model artifacts, and hyperparameter sweeps to ensure reproducibility across development cycles. The platform also functions as an observability tool for large language model applications, enabling the tracing of execution steps, token usage, and reasoning processes. The project distinguishes itself through its event-driven automation capabilities, which allow users

    Logs and visualizes audio files with associated metadata during machine learning experiments.

    Pythonaicollaborationdata-science
    Ver en GitHub↗10,844
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Explorar subetiquetas

  • Audio Watermarking3 sub-etiquetasTechniques for embedding imperceptible signatures into audio for provenance tracking. **Distinct from Audio Processing:** Distinct from general audio processing: focuses specifically on imperceptible digital signatures for provenance.
  • Batch Transcription1 sub-etiquetaAutomated workflows for transcribing multiple audio files or segments in parallel to increase throughput. **Distinct from Audio Processing:** Distinct from general audio processing: focuses on the batch transcription of audio files.
  • Deep Learning ProcessorsSoftware utilities leveraging GPU acceleration for complex audio extraction. **Distinct from Audio Processing:** Focuses on the processor utility aspect, distinct from general audio processing models.
  • High-Volume ProcessingSystems designed for transcribing or analyzing large batches of audio files simultaneously to maximize throughput. **Distinct from Audio Processing:** Distinct from general audio processing: focuses on the high-volume, pipeline-based transcription of large datasets.
  • Mel-Spectrogram Processing2 sub-etiquetasTechniques for transforming audio waveforms into mel-spectrograms for analysis by neural networks. **Distinct from Audio Processing:** Focuses specifically on the mel-frequency scaling transformation for CNN input, not general audio synthesis or transformation
  • Training Data Augmentation2 sub-etiquetasTechniques for diversifying audio datasets through signal manipulation to improve model robustness. **Distinct from Audio Processing:** Distinct from general Audio Processing: specifically targets the creation of diverse training samples via noise and filtering.