Parameter analysis for the generalized LZ compression of audio

D. Kirovski, Zeph Landau
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引用次数: 2

Abstract

Summary form only given. We introduced (Kirovski and Landau (2004)) a memory-based model of the source signal, which explores multimedia repetitiveness to improve upon compression rates achieved by classic memoryless or simple prediction-based audio compression algorithms such as MP3. The representation error is masked using a psycho-acoustic filter. The goal of the masking function is to set the error such that reconstruction of audible samples is exact whereas the reconstruction of inaudible samples is such that the absolute magnitude of the error is minimized. We compute the entropy of the quantized pointers to all blocks, the quantized pointers to the applied transforms, the quantized scalars used to create the linear combination of transformed blocks, and the error vector returned.
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音频广义LZ压缩的参数分析
只提供摘要形式。我们介绍了(Kirovski和Landau(2004))一种基于记忆的源信号模型,该模型探索了多媒体重复性,以提高经典的无记忆或简单的基于预测的音频压缩算法(如MP3)所达到的压缩率。使用心理-声学滤波器来掩盖表征误差。掩蔽函数的目标是设置误差,使可听样本的重建是精确的,而不可听样本的重建是使误差的绝对幅度最小化。我们计算指向所有块的量化指针的熵,指向应用变换的量化指针,用于创建转换块的线性组合的量化标量,以及返回的误差向量。
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