A systematic hybrid analog/digital audio coder

Proceedings of the 1999 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics. WASPAA'99 (Cat. No.99TH8452) Pub Date : 1999-10-17 DOI:10.1109/ASPAA.1999.810843

R. Barron, A. Oppenheim

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引用次数: 3

Abstract

This paper describes a signal coding solution for a hybrid channel that is the composition of two channels: a noisy analog channel through which a signal source is sent unprocessed and a secondary rate-constrained digital channel. The source is processed prior to transmission through the digital channel. Signal coding solutions for this hybrid channel are clearly applicable to the in-band on-channel (IBOC) digital audio broadcast (DAB) problem. We present the design of a perceptually-based subband audio coder, with complexity comparable to conventional coders, that exploits a signal at the receiver of the form y[n]=g[n]*x[n]+u[n], where x[n], g[n], and u[n] denote respectively the source, the impulse response of convolutional distortion, and additive Gaussian noise. Concepts from conventional subband coding, e.g. subband decomposition, quantization, bit allocation, and lossless signal coding, are tailored to exploit the analog signal at the receiver such that frequency-weighted mean-squared error is minimized.

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一种系统混合模拟/数字音频编码器

本文描述了一种混合信道的信号编码解决方案，混合信道由两个信道组成:一个是未经处理的信号源的噪声模拟信道，另一个是受速率限制的二级数字信道。在通过数字信道传输之前对源进行处理。这种混合信道的信号编码解决方案显然适用于带内信道(IBOC)数字音频广播(DAB)问题。我们设计了一种基于感知的子带音频编码器，其复杂性与传统编码器相当，该编码器利用形式为y[n]=g[n]*x[n]+u[n]的接收器信号，其中x[n]， g[n]和u[n]分别表示源，卷积失真的脉冲响应和加性高斯噪声。传统子带编码的概念，如子带分解、量化、位分配和无损信号编码，都是为了利用接收器上的模拟信号而量身定制的，从而使频率加权均方误差最小化。

本文章由计算机程序翻译，如有差异，请以英文原文为准。

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Proceedings of the 1999 IEEE Workshop on Applications of Signal Processing to Audio and Acoustics. WASPAA'99 (Cat. No.99TH8452)

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