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1. 发明申请

US20110282677A1 METHOD AND SYSTEM FOR REDUCTION OF QUANTIZATION-INDUCED BLOCK-DISCONTINUITIES AND GENERAL PURPOSE AUDIO CODEC 有权
标题翻译：用于减少量化诱导的块 - 不连续性和一般目的音频编解码器的方法和系统
公开(公告)号：US20110282677A1
公开(公告)日：2011-11-17
申请号：US13191496
申请日：2011-07-27
申请人： Shuwu WU , John Mantegna , Keren Perlmutter
发明人： Shuwu WU , John Mantegna , Keren Perlmutter
IPC分类号： G10L19/00
CPC分类号： G10L19/00 , G10L19/0212 , G10L19/022 , G10L19/028 , G10L19/038 , G10L2019/0012
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.
摘要翻译：用于减少由连续信号，特别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。一个实施例包括通用的，超低延迟的高效音频编解码算法。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来压缩和解压缩音频信号的方法和装置，以大幅度减少量化引起的帧或块不连续性; 一种新颖的自适应余弦分组变换（ACPT）作为选择的有效捕获输入音频特性的变换; 信号残差分类器将强信号簇与噪声和弱信号分量（统称为残差）分离开来; 用于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; 和相关联的速率控制算法。本发明还包括这些和其他算法的相应的计算机程序实现。

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