Key Features

Upscales 16kHz audio to crisp 48kHz output using a tiny 50kb model
Lightning-fast inference suitable for real-time applications
Preserves natural timbre and reduces common compression artifacts
Optimized for low-resource environments like mobile and edge devices
Supports batch processing for efficient handling of large audio files
Compatible with PyTorch and exportable to ONNX for broad deployment
Trained on diverse datasets covering speech, music, and ambient sounds
Open-source with easy-to-use inference scripts and examples

At the core of NovaSR's capabilities lies its sophisticated training on diverse datasets encompassing speech, music, and environmental sounds, enabling it to handle a wide spectrum of audio degradation scenarios with exceptional fidelity. The model employs advanced techniques like lightweight convolutional layers and residual connections to upscale frequencies while preserving temporal dynamics and minimizing artifacts such as phase distortion or unnatural reverb. Users benefit from its plug-and-play nature, supported by simple inference scripts in popular frameworks, allowing rapid deployment in Python environments or even browser-based applications via ONNX export, thus democratizing access to studio-quality audio enhancement.


NovaSR stands out not only for its minuscule footprint but also for its inference speed, processing audio streams in near real-time even on consumer-grade hardware, which opens doors to innovative uses in live streaming, voice assistants, and archival restoration projects. Its open-source availability fosters community contributions, with ongoing developments focusing on multilingual support, noise robustness, and hardware acceleration via platforms like CUDA or WebGPU. As audio demands grow in the era of immersive media and teleconferencing, NovaSR positions itself as an essential tool for developers and audio enthusiasts aiming to elevate sound quality without compromising on performance or portability.

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