High-quality audio transform coding at 64 kbps

Author

Mahieux, Yannick ; Petit, Jean Pierre

Author_Institution

CNET, Lannion, France

Volume

42

Issue

11

fYear

1994

fDate

11/1/1994 12:00:00 AM

Firstpage

3010

Lastpage

3019

Abstract

This paper presents a transform coding algorithm devoted to high quality audio coding at a bit rate of 64 kbps per monophonic channel. It enables the transmission of a high quality stereo sound through the basic access (2B channels) of ISDN. Although a complete system including framing, synchronization and error correction has been developed, only the bit rate compression algorithm is described here. A detailed analysis of the signal processing techniques such as the time/frequency transformation, the pre-echo reduction by adaptive filtering, the fast algorithm computations, etc., is provided. The use of psychoacoustical properties is also precisely reported. Finally, some subjective evaluation results and one real time implementation of the coder using the ATT DSP32C digital signal processor are presented

Keywords

Hi-Fi equipment; audio signals; audio systems; data compression; digital signal processing chips; encoding; filtering and prediction theory; 2B channels; 64 kbit/s; ATT DSP32C digital signal processor; ISDN; adaptive filtering; audio transform coding algorithm; basic access channels; bit rate compression algorithm; coder; error correction; fast algorithm computations; framing; high quality audio coding; monophonic channel; pre-echo reduction; psychoacoustical properties; signal processing; stereo sound transmission; subjective evaluation results; synchronization; time/frequency transformation; Algorithm design and analysis; Audio coding; Bit rate; Compression algorithms; Error correction; Frequency synchronization; ISDN; Signal analysis; Signal processing algorithms; Transform coding;

fLanguage

English

Journal_Title

Communications, IEEE Transactions on

Publisher

ieee

ISSN

0090-6778

Type

jour

DOI

10.1109/26.328982

Filename

328982