{"title":"Fast subband coder for telephone quality audio","authors":"H. Raittinen, K. Kaski","doi":"10.1109/DCC.1995.515581","DOIUrl":null,"url":null,"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.","PeriodicalId":107017,"journal":{"name":"Proceedings DCC '95 Data Compression Conference","volume":"62 1","pages":"0"},"PeriodicalIF":0.0000,"publicationDate":"1995-03-28","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":"0","resultStr":null,"platform":"Semanticscholar","paperid":null,"PeriodicalName":"Proceedings DCC '95 Data Compression Conference","FirstCategoryId":"1085","ListUrlMain":"https://doi.org/10.1109/DCC.1995.515581","RegionNum":0,"RegionCategory":null,"ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":null,"EPubDate":"","PubModel":"","JCR":"","JCRName":"","Score":null,"Total":0}
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.