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A new quantization optimization algorithm for the MPEG advanced audio coder using a statistical subband model of the quantization noise

机译:使用量化噪声的统计子带模型的MPEG高级音频编码器的新量化优化算法

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In this paper, an improvement of the quantization optimization algorithm for the MPEG Advanced Audio Coder (AAC) is presented. This algorithm, given a bit-rate constraint, minimizes the perceived distortion generated by the signal compression. The distortion can be related to the quantization error level over frequency subbands through an auditory model. Thus, optimizing the quantization requires knowledge of the rate-distortion function for each subband. When this function can be modeled in a simple way, the algorithm can take a one-loop recursive structure. However, in the MPEG AAC, the rate-distortion function is hard to characterize, since AAC makes use of nonlinear quantizers and variable length entropy coders. As a result, the standard algorithm makes use of two nested loops with a local decoder, in order to measure the error level rather than predicting its value. We first describe a partial subband modeling of the rate-distortion function of interest in the MPEG AAC. Then, using a statistical approach, we find a relationship between the error level and the so-called quantization "scale-factor" and propose a new algorithm that is basically similar to a classical one loop "bit allocation" process. Finally, we describe the complete algorithm and show that it is more efficient than the standard one.
机译:本文提出了一种针对MPEG高级音频编码器(AAC)的量化优化算法的改进。在给定比特率约束的情况下,该算法可将信号压缩产生的感知失真降至最低。通过听觉模型,失真可以与频率子带上的量化误差水平有关。因此,优化量化需要了解每个子带的速率失真函数。如果可以通过简单的方式对该函数进行建模,则该算法可以采用单循环递归结构。但是,在MPEG AAC中,由于AAC利用非线性量化器和可变长度熵编码器,因此很难描述速率失真功能。结果,标准算法利用带有本地解码器的两个嵌套循环,以测量错误级别而不是预测其值。我们首先描述MPEG AAC中感兴趣的速率失真函数的部分子带建模。然后,使用一种统计方法,我们发现了错误级别和所谓的量化“比例因子”之间的关系,并提出了一种新算法,该算法与经典的单循环“位分配”过程基本相似。最后,我们描述了完整的算法,并表明它比标准算法更有效。

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