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METHOD AND SYSTEM FOR REDUCTION OF QUANTIZATION-INDUCED BLOCK-DISCONTINUITIES AND GENERAL PURPOSE AUDIO CODEC

机译:减少量化引起的块不连续性和通用目的音频编解码器的方法和系统

摘要

A method and system for reduction of quantization-induced block-discontinuities arising from lossy compression and decompression of continuous signals, especially audio signals. One embodiment encompasses a general purpose, ultra-low latency, efficient audio codec algorithm. More particularly, the invention includes a method and apparatus for compression and decompression of audio signals using a novel boundary analysis and synthesis framework to substantially reduce quantization-induced frame or block discontinuity; a novel adaptive cosine packet transform (ACPT) as the transform of choice to effectively capture the input audio characteristics; a signal-residue classifier to separate the strong signal clusters from the noise and weak signal components (collectively called residue); an adaptive sparse vector quantization (ASVQ) algorithm for signal components; a stochastic noise model for the residue; and an associated rate control algorithm. The invention further includes corresponding computer program implementations of these and other algorithms.
机译:一种用于减少由连续信号尤其是音频信号的有损压缩和解压缩引起的量化引起的块不连续性的方法和系统。一个实施例包括一种通用的超低等待时间,有效的音频编解码器算法。更具体地说,本发明包括一种使用新颖的边界分析和合成框架对音频信号进行压缩和解压缩以实质上减少量化引起的帧或块不连续性的方法和装置。一种新颖的自适应余弦包变换(ACPT)作为有效转换输入音频特性的选择变换;信号残差分类器,用于将强信号簇与噪声和弱信号分量(统称为残差)分开;信号分量的自适应稀疏矢量量化(ASVQ)算法;残留物的随机噪声模型;以及相关的速率控制算法本发明还包括这些算法和其他算法的相应计算机程序实现。

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