Adaptive-block-length, adaptive-transforn, and adaptive-window t

Electrical audio signal processing systems and devices – One-way audio signal program distribution – Public address system

Patent

Rate now

  [ 0.00 ] – not rated yet Voters 0   Comments 0

Details

381 30, 395 267, 375240, G10L 500

Patent

active

053944739

DESCRIPTION:

BRIEF SUMMARY
DESCRIPTION

1. Technical Field
The invention relates in general to high-quality low bit-rate digital transform coding and decoding of information corresponding to audio signals such as music or voice signals. More particularly, the invention relates to signal analysis/synthesis in coding and decoding. The invention can optimize the trade off in transform coders between time resolution and frequency resolution by adaptively selecting the transform block length for each sampled audio segment, and/or can optimize coding gain by adaptively selecting the transform and/or by adaptively selecting the analysis window or the analysis/synthesis window pair.
The invention applies to all discrete orthogonal transforms. Transform orthogonality assures that exact signal reconstruction can be achieved by the forward/inverse transform pair. Hence for orthogonal transforms, this invention permits adaptive selection of the block length and/or adaptive selection of the transform without loss of information, i.e., in the absence of quantization errors, the original signal can be exactly recovered by the decoder portion of the invention.
The preferred embodiment of the invention, however, uses nonorthogonal transforms. In this preferred embodiment of the invention, a transform coder adapts the transform, and/or the analysis/synthesis window pair, and/or the block-length while retaining transform properties of complete aliasing cancellation in the absence of coefficient quantization errors and critical sampling.
2. Background Art
There is considerable interest among those in the field of signal processing to minimize the amount of information required to represent a signal without perceptible loss in signal quality. By reducing information requirements, signals impose lower information capacity requirements upon communication channels and storage media. With respect to digital coding techniques, minimal informational requirements are synonymous with minimal binary bit requirements.
Bit requirements for digital signals coded by techniques such as pulse code modulation (PCM) are proportional to the number of digitized signal samples and to the number of bits used to represent each digitized signal sample. The number of samples for a given segment of signal is determined by the sampling rate.
The minimum sampling rate is dictated by the Nyquist theorem. The Nyquist theorem holds that a signal may be accurately recovered from discrete samples when the interval between samples is no larger than one-half the period of the signal's highest frequency component. When the sampling rate is below this Nyquist rate, higher-frequency components are misrepresented as lower-frequency components. The lower-frequency component is an "alias" for the true component.
The number of bits used to represent each digitized signal sample determines the accuracy of the signal representation by the encoded signal samples. Lower bit rates generally mean that fewer bits are available to represent each sample, therefore lower bit rates imply greater quantizing inaccuracies or quantizing errors. In many applications, quantizing errors are manifested as quantizing noise, and if the errors are of sufficient magnitude, the noise will degrade the subjective quality of the coded signal.


Critical Bands and Psychoacoustic Masking

Some prior art techniques for coding audio signals intended for human hearing attempt to reduce information requirements without producing any audible degradation by exploiting psychoacoustic effects. The human ear displays frequency-analysis properties resembling those of highly asymmetrical tuned filters having variable center frequencies. The ability of the human ear to detect distinct tones generally increases as the difference in frequency between the tones increases, however, the ear's resolving ability remains substantially constant for frequency differences less than the bandwidth of the above mentioned filters. Thus, the frequency-resolving ability of the human ear varies according to the bandwidth of these filters throughout th

REFERENCES:
patent: 5109417 (1992-04-01), Fielder et al.
patent: 5115240 (1992-05-01), Fujiwara et al.
patent: 5142656 (1992-08-01), Fielder et al.
Speech Communication, 1988, pp. 125-149, "Review on Medium-Rate Coding" by Ulrich Heute.
Brigham, The Fast Fourier Transform, Prentice-Hall, Inc., 1974, pp. 166-169.
Oppenheim and Schafer, Digital Signal Processing, Prentice-Hall, Inc., 1975, pp. 307-314.
Zelinski, Noll, "Adaptive Transform Coding of Speech Signals," IEEE Trans. Acoust., Speech, and Signal Proc., vo. ASSP-25, Aug. 1977, pp. 299-309.
Harris, "On the Use of Windows for Harmonic Analysis with the Discrete Fourier Transform," Proc. IEEE, vol. 66, Jan., 1978, pp. 51-83.
Jayant and Noll, Digital Coding of Waveforms, Prentice-Hall, Inc., 1984, pp. 56-58, 554-556, 563-576.
Abdel-Fattah and Assal, "A study of the Different Orthogonal Transforms to Obtain an Optimum Speech Compression," Eurocon, Apr. 1986, pp. 647-652.
Krahe, "Bit-Rate Reduction Method for Digital Audio Signals Based on Psychoacoustic Masking Phenomena," Radio Engineering News, 1986, pp. 117-123.
Krahe, "New Source Coding Method for High Quality Digital Audio Signals," Lecture, NTG Meeting on Sound Broadcasting, Nov. 1985.
Princen and Bradley, "Analysis/Synthesis Filter Bank Design Based on Time Domain Aliasing Cancellation," IEEE, vol. ASSP-34, Oct. 1986, pp. 1153-1161.
Vaisey and Gersho, "Variable Block-Size Coding," ICASSP, Apr. 1987, pp. 1051-1054.
Princen, Johnson, Bradley, "Subband/Transform Coding Using Filter Bank Designs Based on Time Domain Aliasing Cancellation," ICASSP, Apr. 1987, pp. 2161-2164.
Johnson and Bradley, "Adaptive Transform Coding Incorporating Time Domain Aliasing Cancellation," Speech Communications., vol. 6, 1987, pp. 299-308.
Schroeder, Platte, Krahe, "`MSC `: Stereo Audio Coding with CD-Quality and 256 kBit/Sec," IEEE, vol. CE-33, Nov. 1987, pp. 512-519.
Audio Engineering Handbook, Benson ed., McGraw-Hill, 1988, pp. 1.40-1.42, 4.8-4.10.
Brandenburg, "High Quality Sound Coding at 2.5 Bit/Sample," AES Convention Preprint No. 2582, 84th Convention, Apr. 1988.
Lookabaugh, "Variable Rate and Adaptive Frequency Domain Vector Quantization of Speech," PhD Dissertation, Stanford University, Jun. 1988, pp. 166-182.
Brandenburg, Kapust, et. al., "Low Bit Rate Codecs for Audio Signals Implementation in Real Time," AES, 85th Convention, Nov. 1988.
Feiten, "Spectral Properties of Audio Signals and Masking with Aspect to Bit Data Reduction," AES, 86th Convention, Mar. 1989.
Edler, "Coding of Audio Signals with Overlapping Block Transform and Adaptive Window Functions," Frequenz, vol. 43, No. 9, 1989, pp. 252-256.
Sugiyama, Hazu, Iwadare, Nishitani, "Adaptive Transform Coding with an Adaptive Block Size (ATC-ABS)," ICASSP, Apr. 1990, pp. 1093-1096.

LandOfFree

Say what you really think

Search LandOfFree.com for the USA inventors and patents. Rate them and share your experience with other people.

Rating

Adaptive-block-length, adaptive-transforn, and adaptive-window t does not yet have a rating. At this time, there are no reviews or comments for this patent.

If you have personal experience with Adaptive-block-length, adaptive-transforn, and adaptive-window t, we encourage you to share that experience with our LandOfFree.com community. Your opinion is very important and Adaptive-block-length, adaptive-transforn, and adaptive-window t will most certainly appreciate the feedback.

Rate now

     

Profile ID: LFUS-PAI-O-854232

  Search
All data on this website is collected from public sources. Our data reflects the most accurate information available at the time of publication.