Data processing: speech signal processing – linguistics – language – Audio signal bandwidth compression or expansion
Reexamination Certificate
1997-09-25
2001-07-31
Hudspeth, David R. (Department: 2741)
Data processing: speech signal processing, linguistics, language
Audio signal bandwidth compression or expansion
C704S219000, C341S051000, C341S065000, C341S143000
Reexamination Certificate
active
06269338
ABSTRACT:
FIELD OF THE INVENTION
The invention relates to the field of data compression and most closely relates to real time compression of a digitally sampled audio data stream.
BACKGROUND OF THE INVENTION
The invention relates to a data compression apparatus for data compressing an audio signal, to a data compression method, a transmitter including the data compression apparatus, a recording apparatus including the data compression apparatus, a record carrier having the data compressed audio signal recorded on it in a track of the record carrier, to a data expansion apparatus for data expanding a data compressed audio signal, to a data expansion method, a receiver including the data expansion apparatus and to a reproducing apparatus including the data expansion apparatus.
Data compression on an audio signal is well known in the art. Reference is made in this respect to EP-A 402,973, document D
1
herein list of related documents. The document describes a subband coder, in which an audio signal is A/D converted with a specific sampling frequency, such as 44.1 kHz, and the resulting samples in the form of eg. 24 bits wide words of the audio signal, are supplied to a subband splitter filter. The subband splitter filter splits the wideband digital audio signal into a plurality of relatively narrow band subband signals. Using a psycho acoustic model, a masked threshold is derived and blocks of samples of the subband signals are subsequently quantised with a specific number of bits per sample for each block of the subband signals, in response to the masked threshold, resulting in a significant data compression of the audio signal to be transmitted. The data compression carried out is based on ‘throwing away’ those components in the audio signal that are inaudible and is thus a lossy compression method. The data compression described in document D
1
is a rather intelligent data compression method and requires a substantial number of gates or instructions, when embodiment either in hardware or software respectively, so that it is expensive. Moreover, the subsequent expansion apparatus also requires a substantial number of gates or instructions, when implemented in hardware or software respectively.
Those skilled in the art are hereby referred to the following listed documents:
(D
2
) “A digital decimating filter for analog-to-digital conversion of hi-fi audio signals”, by J. J. van der Kam in Philips Tech. Rev. 42, no. 6/7, April. 1986, pp. 230-8
(D
3
) “A higher order topology for interpolative modulators for oversampling A/D converters”, by Kirk C. H. Chao et al in IEEE Trans. on Circuits and Systems, Vol. 37, no. 3, March. 1990, pp. 309-18
(D
4
) “A method for the construction of minimum-redundancy codes”, by D. A. Huffman in Proc. of the IRE, Vol. 40(10), September 1952.
(D
5
) “An introduction to arithmetic coding” by G. G. Langdon, IBM J. Res. Develop., Vol. 28(2), March 1984.
(D
6
) “A universal algorithm for sequential data compression” by J. Ziv et al, IEEE Trans. on Inform. Theory, Vol. IT-23, 1977.
The above citations and tho sin the list of related documents below are hereby incorporated in whole by reference.
SUMMARY OF THE INVENTION
The invention aims at providing a data compression apparatus for data compressing an audio signal such that it is more simple and that the corresponding expander apparatus can also be more simple and less expensive.
The data compression apparatus in accordance with the invention comprises
input apparatus for receiving the audio signal,
conversion apparatus for carrying out a conversion on the audio signal so as to obtain a 1-bit bitstream signal, the conversion apparatus comprising sigma-delta modulator apparatus,
lossless coding apparatus for carrying out a substantially lossless data compression step on the bitstream signal so as to obtain a data compressed bitstream signal, and
output apparatus for supplying the data compressed bitstream signal. More specifically, when the audio signal is an analog audio signal, the conversion apparatus is in the form of A/D conversion apparatus for carrying out a 1-bit A/D conversion on the analog audio signal so as to obtain the bitstream signal.
The invention is based on the following recognition. The audio signal can be applied in analog form or in digital form. When A/D converting, in accordance with the invention, an analog audio signal with a 1-bit A/D converter (also named: bitstream converter or sigma-delta modulator), the audio signal to be A/D converted is sampled with a frequency which is generally a multiplicity of the frequency of 44.1 kHz or 48 kHz. The output signal of the 1-bit A/D converter is a binary signal, named bitstream signal. When the audio signal is supplied in digital form, sampled at eg. 44.1 kHz, the samples being expressed in eg. 16 bits per sample, this digital audio signal is oversampled with a frequency which is again a multiplicity of this sampling frequency of 44.1 kHz (or 48 kHz), which results in the 1-bit bitstream signal.
Converting an audio signal into a 1-bit bitstream signal has a number of advantages. Bitstream conversion is a high quality encoding method, with the possibility of a high quality decoding or a low quality decoding with the further advantage of a simpler decoding circuit. Reference is made in this respect to the publications ‘A digital decimating filter for analog-to-digital conversion of hi-fi audio signals’, by J. J. van der Kam, document D
2
, and ‘A higher order topology for interpolative modulators for oversampling A/D converters’, by Kirk C. H. Chao et al, document D
3
.
1-bit D/A converters are used in CD players, as an example, to reconvert the bitstream audio signal into an analog audio signal. The audio signal recorded on a CD disk is however not a data compressed 1-bit bitstream signal.
It is well known in the art that the resulting bitstream signal of the 1-bit A/D converter is, roughly the, a random signal which has a ‘noisy-like’ frequency spectrum. Such types of signals are hard to data compress.
Surprisingly, however, it was established that using a lossless coder, such as a variable length coder in the form of a Huffman coder or an arithmetic coder, a significant data reduction could be obtained, in spite of the noisy character of the bitstream signal from the 1-bit A/D converter.
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Lerch et al., “A Monolithic Sigma-Delta A/D and D/A Converter with Filter for Broad-Band Speech Coding,” IEEE Journal of Solid-State Circuits, vol. 26, No. 12, Dec. 1991, pp. 1920 to 1927, Dec. 1991.*
Lars Risbo, “Improved Stability and Performance from Sigma-Delta Modulators using 1-bit Vector Quantization,” 1993 IEEE International Symposium on Circuits and Systems, May 1993, pp. 1365 to 1368.*
Kershaw et al., “Sigma-delta modulator for audio DSP,” IEE Colloquium on Audio DSP—Circuits and Systems, pp. 1/1 to 1/6, Dec. 1993.*
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“A Digital Decimating Filter for Analog-to-Digital Conversion of Hi-Fi Audio Signals”, By J.J. Van Der Kam in Philips Techn. Rev. 42, No. 6/7, Apr. 1986, pp. 230-238.
“A Higher Order Topology for Interpolative Modulators for Oversampling A/D Converters”, by Kirk C.H. Chao Et Al in IEEE Trans. on Circuits and Systems, vol. 37, No. 3, Mar. 1990, pp. 309-318.
“A Method for the Construction of Minimum-Redundancy Codes”, by D.A.
Bruekers Alphons A. M. L.
Oomen Arnoldus W. J.
Van Der Vleuten Renatus J.
Belk Michael E.
Hudspeth David R.
Lerner Martin
U.S. Philips Corporation
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