Adaptive transform coding with an adaptive block size (ATC-ABS)
- 4 December 2002
- conference paper
- Published by Institute of Electrical and Electronics Engineers (IEEE)
- No. 15206149,p. 1093-1096
- https://doi.org/10.1109/icassp.1990.116122
Abstract
A coding technique is presented for high-quality audio signals based on adaptive transform coding (ATC). Adaptive block size selection by the proposed algorithm ensures an appropriate block size resulting in improved SNR (signal-to-noise ratio) for a wide variety of source signals. A feedback approach, based on SNR, and a feedforward approach, based on interblock differences in input time-domain samples, to adaptive block size assignment are proposed and evaluated. Computer simulation results show that average segmental SNR by the feedback approach is improved by as much as 4.8 dB over the conventional fixed-block-size ATC. The feedforward approach is realized with much-simplified hardware; nevertheless, its SNR degradation from that by the feedback approach is 1.6 dB, even in the worst case. Both approaches are successful in pre-echo suppression to a satisfactory level. Time-domain aliasing cancellation has the potential to increase the superiority of the new algorithm.Keywords
This publication has 6 references indexed in Scilit:
- Transform coding of audio signals using correlation between successive transform blocksPublished by Institute of Electrical and Electronics Engineers (IEEE) ,2003
- Perceptual transform coding of wideband stereo signalsPublished by Institute of Electrical and Electronics Engineers (IEEE) ,2003
- A robust 384 kbit/s stereo HiFi audio codec for ISDN applicationsPublished by Institute of Electrical and Electronics Engineers (IEEE) ,2003
- Analysis/Synthesis filter bank design based on time domain aliasing cancellationIEEE Transactions on Acoustics, Speech, and Signal Processing, 1986
- Approaches to adaptive transform speech coding at low bit ratesIEEE Transactions on Acoustics, Speech, and Signal Processing, 1979
- Minimum Mean-Squared-Error Quantization in Speech PCM and DPCM SystemsIEEE Transactions on Communications, 1972