{"title":"High quality audio encoding within 128 kbit/s","authors":"R. J. Beaton","doi":"10.1109/PACRIM.1989.48383","DOIUrl":null,"url":null,"abstract":"An algorithm to encode 15-KHz bandwidth audio at 128 kb/s has been developed which has important applications in ISDN, digital audio recording, and the broadcasting industry. The algorithm uses subband ADPCM coding combined with adaptive bit allocation over multiple subbands to achieve high audio quality at a net data rate of 4 b per sample. Subband coders are generally found to perform better than the equivalent full-band coder in subjective tests, even though the full-band coder may achieve a higher SNR rating. This is due to perceptual masking effects which cause the band-limited quantization noise below a certain threshold in each subband to be masked by the signal. Dynamic allocation of bits to subbands on a sample-by-sample basis, together with a simple multiplexing strategy, eliminates the need for side information and makes the algorithm robust in the presence of network bit errors.<<ETX>>","PeriodicalId":256287,"journal":{"name":"Conference Proceeding IEEE Pacific Rim Conference on Communications, Computers and Signal Processing","volume":"2 1","pages":"0"},"PeriodicalIF":0.0000,"publicationDate":"1989-06-01","publicationTypes":"Journal Article","fieldsOfStudy":null,"isOpenAccess":false,"openAccessPdf":"","citationCount":"1","resultStr":null,"platform":"Semanticscholar","paperid":null,"PeriodicalName":"Conference Proceeding IEEE Pacific Rim Conference on Communications, Computers and Signal Processing","FirstCategoryId":"1085","ListUrlMain":"https://doi.org/10.1109/PACRIM.1989.48383","RegionNum":0,"RegionCategory":null,"ArticlePicture":[],"TitleCN":null,"AbstractTextCN":null,"PMCID":null,"EPubDate":"","PubModel":"","JCR":"","JCRName":"","Score":null,"Total":0}
引用次数: 1
Abstract
An algorithm to encode 15-KHz bandwidth audio at 128 kb/s has been developed which has important applications in ISDN, digital audio recording, and the broadcasting industry. The algorithm uses subband ADPCM coding combined with adaptive bit allocation over multiple subbands to achieve high audio quality at a net data rate of 4 b per sample. Subband coders are generally found to perform better than the equivalent full-band coder in subjective tests, even though the full-band coder may achieve a higher SNR rating. This is due to perceptual masking effects which cause the band-limited quantization noise below a certain threshold in each subband to be masked by the signal. Dynamic allocation of bits to subbands on a sample-by-sample basis, together with a simple multiplexing strategy, eliminates the need for side information and makes the algorithm robust in the presence of network bit errors.<>