Low time-delay transform coder, decoder, and encoder/decoder for

Electrical audio signal processing systems and devices – Hearing aids – electrical – Specified casing or housing

Patent

Rate now

  [ 0.00 ] – not rated yet Voters 0   Comments 0

Details

381 30, 381 31, G01L 900

Patent

active

052221890

DESCRIPTION:

BRIEF SUMMARY
TECHNICAL FIELD

The invention relates in general to the high-quality low bit-rate digital signal processing of audio signals, such as music signals. More particularly, the invention relates to transform encoders and decoders for such signals, wherein the encoders and decoders have a short signal-propagation delay. Short delays are important in applications such as broadcast audio where a speaker must monitor his own voice. A delay in voice feedback causes serious speech disruption unless the delay is very short.


BACKGROUND ART



INTRODUCTION

Transform coding of high-quality signals in the prior art have used long signal sample block lengths to achieve low bit-rate coding without creating objectionable audible distortion. For example, a transform coder disclosed in EP 0 251 028 uses a block length of 1024 samples. Long block lengths have been necessary because shorter blocks degrade transform coder selectivity. Filter selectivity is critical because transform coders with sufficient filter bank selectivity can exploit psychoacoustic masking properties of human hearing to reduce bit-rate requirements without degrading the subjective quality of the coded signal.
Coders using long block lengths suffer from two problems: (1) audible distortion of signals with large transients caused by the temporal spreading of the transient's effects throughout the transform block, and (2) excessive propagation delay of the signal through the encoding and decoding process. In prior art coders, these processing delays are too great for applications such as broadcast audio where a speaker must monitor his own voice. A delay in voice feedback causes serious speech disruption unless the delay is kept very short.
The background art is discussed in more detail in the following Background Summary.


BACKGROUND SUMMARY

There is considerable interest among those in the field of signal processing to discover methods which minimize the amount of information required to represent adequately a given signal. By reducing required information, signals may be transmitted over communication channels with lower bandwidth, or stored in less space. With respect to digital techniques, minimal informational requirements are synonymous with minimal binary bit requirements.
Two factors limit the reduction of bit requirements:
(1) A signal of bandwidth W may be accurately represented by a series of samples taken at a frequency no less than 2.multidot.W. This is the Nyquist sampling rate. Therefore, a signal T seconds in length with a bandwidth W requires at least 2.multidot.W.multidot.T number of samples for accurate representation.
(2) Quantization of signal samples which may assume any of a continuous range of values introduces inaccuracies in the representation of the signal which are proportional to the quantizing step size or resolution. These inaccuracies are called quantization errors. These errors are inversely proportional to the number of bits available to represent the signal sample quantization.
If coding techniques are applied to the full bandwidth, all quantizing errors, which manifest themselves as noise, are spread uniformly across the bandwidth. Techniques which may be applied to selected portions of the spectrum can limit the spectral spread of quantizing noise. Two such techniques are subband coding and transform coding. By using these techniques, quantizing errors can be reduced in particular frequency bands where quantizing noise is especially objectionable by quantizing that band with a smaller step size.
Subband coding may be implemented by a bank of digital bandpass filters. Transform coding may be implemented by any of several time-domain to frequency-domain transforms which simulate a bank of digital bandpass filters. Although transforms are easier to implement and require less computational power and hardware than digital filters, they have less design flexibility in the sense that each bandpass filter "frequency bin" represented by a transform coefficient has a uniform bandwidth. By contrast, a bank of digital bandpass

REFERENCES:
patent: 4216354 (1980-08-01), Esteban et al.
patent: 4455649 (1984-06-01), Esteban et al.
patent: 4703480 (1987-10-01), Westall et al.
patent: 4790016 (1988-12-01), Mazor et al.
patent: 4914701 (1990-04-01), Zibman
patent: 5109417 (1992-04-01), Fielder et al.
patent: 5115240 (1992-04-01), Fujiwara et al.
D. Esteban, C. Galand, "32 KBPS CCITT Compatible Split Band Coding Scheme," IEEE Int. Conf. on Acoust., Speech, and Signal Proc., 1978, pp. 320-325.
Lee, "Effects of Delayed Speech Feedback," J. Acoust. Soc. Am., vol. 22, Nov., 1950, pp. 824-826.
Cooley, Tukey, "An Algorithm for the Machine Calculation of Complex Fourier Series," Math. Comput., vol. 19, 1965, pp. 297-301.
Parks, McClellan, "Chebyshev Approximation for Nonrecursive Digital Filters with Linear Phase," IEEE Trans., vol. CT-19, Mar. 1972, pp. 189-194.
Brigham, The Fast Fourier Transform, Englewood Cliffs, NJ: Prenctice-Hall, Inc., 1974, pp. 166-169.
Lee, Lipschutz, "Floating-Point Encoding for Transcription of High-Fidelity Audio Signals," J. Audio Eng. Soc., vol. 25, May, 1977, pp. 266-272.
Harris, "On the Use of Windows for Harmonic Analysis with the Discrete Fourier Transform," Proc. IEEE, vol. 66, Jan., 1978, pp. 51-83.
Tribolet, Crochiere, "Frequency Domain Coding of Speech," IEEE Trans. Acoust., Speech, and Signal Proc., vol. ASSP-27, Oct., 1979, pp. 512-530.
Crochiere, "A Weighted Overlap-Add Method of Short-Time Fourier Analysis/Synthesis," IEEE Trans., vol. ASSP-28, Feb., 1980, pp. 99-102.
S. Prakash, V. V. Rao, "Fixed-Point Error Analysis of Radix-4 FFT," Signal Processing, vol. 3, Apr., 1981, pp. 123-133.
Brandenburg, Schramm, "A 16 Bit Adaptive Transform Coder for Real-Time Processing of Sound Signals", Signal Processing II, 1983, pp. 359-362.
Smith, Digital Transmission Systems, New York, NY: Van Nostrand Reinhold Co., 1985, pp. 228-236.
Fielder, "Pre- and Postemphasis Techniques as Applied to Audio Recording Systems," J. Audio Eng. Soc., vol. 33, 1985, pp. 649-657.
Press, Flannery, Teukolsky, Vetterling, Numerical Recipes: The Art of Scientific Computing, New York: Cambridge University Press, 1986, pp. 254-259.
Peterson, Weldon, Error-Correcting Codes, Cambridge, Mass: The M.I.T. Press, 1986, pp. 269-309, 361-362.
Princen, Bradley, "Analysis/Synthesis Filter Bank Design Based on Time Domain Aliasing Cancellation," IEEE Trans., vol. ASSP-34, Oct., 1986, pp. 1153-1161.
Stoll, Theile, "New Digital Sound Transmission Methods--How is Sound Quality Assessed," Report, 14th Meeting of Audio Engineers, Munich, Nov., 1986.
Brandenburg, "OCF--A New Coding Algorithm for High Quality Sound Signals," IEEE Int. Conf. on Acoust., Speech, and Signal Proc., 1987, pp. 141-144.
Johnson, Bradley, "Adaptive Transform Coding Incorporating Time Domain Aliasing Cancellation," Speech Communications., vol. 6, 1987, pp. 299-308.
Fielder, "Evaluation of the Audible Distortion and Noise Produced by Digital Audio Converters," J. Audio Eng. Soc., vol. 35, Jul., 1987, pp. 517-534.
Audio Engineering Handbook, K. B. Benson ed., San Francisco: McGraw-Hill, 1988, pp. 1.40-1.42, 4.8-4.10.
Johnston, "Transform Coding of Audio Signals Using Perceptual Noise Criteria," IEEE J. on Selected Areas in Comm., vol. 6, Feb., 1988, pp. 314-323.
Brandenburg, Kapust, et al., "Real-Time Implementation of Low Complexity Transform Coding", AES Preprint 2581, 84th Convention, Paris, 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 Preprint 2707, 85th Convention, Nov., 1988.
Brandenburg, Seitzer, "OCF: Coding High Quality Audio with Data Rates of 64 kBit/Sec," AES Preprint 2723, 85th Convention, Los Angeles, Nov., 1988.
Feiten, "Spectral Properties of Audio Signals and Masking with Aspect to Bit Data Reduction," AES Preprint 2795, 86th Convention, Hamburg, Mar., 1989.
Edler, "Coding of Audio Signals with Overlappin

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

Low time-delay transform coder, decoder, and encoder/decoder for does not yet have a rating. At this time, there are no reviews or comments for this patent.

If you have personal experience with Low time-delay transform coder, decoder, and encoder/decoder for, we encourage you to share that experience with our LandOfFree.com community. Your opinion is very important and Low time-delay transform coder, decoder, and encoder/decoder for will most certainly appreciate the feedback.

Rate now

     

Profile ID: LFUS-PAI-O-1446028

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