Fast subband coder for telephone quality audio

H. Raittinen, K. Kaski
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引用次数: 0

Abstract

Summary form only given. A simple and fast audio signal compression method that uses subband filtering and quantization is presented. The method is suitable for compression of telephone quality audio signals. It can compress four CCITT 64 kbit/s PCM A- or /spl mu/-coded speech channels into one channel with sufficient sound quality for telephone use. A straightforward implementation of the compression and decompression methods have the following steps. First the incoming speech signal is converted from a /spl mu/ or A-law coded signal into 16 bit linear PCM signal and then divided into 16 bands of equal bandwidth by using the analysis filter bank. Then the sampling frequencies of the frequency channels are decreased by a factor of 16. After this decimation the subband samples are fed to a fixed quantizer. Finally the quantized subband values and the side information needed for decoding is packed into a data stream and sent to the receiver.
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快速子带编码器电话质量音频
只提供摘要形式。提出了一种基于子带滤波和量化的音频信号压缩方法。该方法适用于电话级音频信号的压缩。它可以将4个CCITT 64kbit /s PCM A或/spl mu/编码语音通道压缩成一个具有足够音质的电话通道。压缩和解压缩方法的简单实现有以下步骤。首先将输入的语音信号从a /spl mu/ or a -law编码信号转换为16位线性PCM信号,然后使用分析滤波器组将其划分为16个等带宽的频带。然后将频率通道的采样频率降低16倍。在抽取后,子带样本被送入固定的量化器。最后将量化后的子带值和解码所需的侧信息打包成数据流发送给接收端。
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