WO1994019791A1 - Filtre ameliore pour systemes de compression et decompression audio - Google Patents

Filtre ameliore pour systemes de compression et decompression audio Download PDF

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Publication number
WO1994019791A1
WO1994019791A1 PCT/US1994/002026 US9402026W WO9419791A1 WO 1994019791 A1 WO1994019791 A1 WO 1994019791A1 US 9402026 W US9402026 W US 9402026W WO 9419791 A1 WO9419791 A1 WO 9419791A1
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WO
WIPO (PCT)
Prior art keywords
sub
band
audio signal
filter
filter bank
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Ceased
Application number
PCT/US1994/002026
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English (en)
Inventor
Sriram Jayasimha
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Aware Inc
Original Assignee
Aware Inc
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Filing date
Publication date
Application filed by Aware Inc filed Critical Aware Inc
Priority to AU62499/94A priority Critical patent/AU6249994A/en
Publication of WO1994019791A1 publication Critical patent/WO1994019791A1/fr
Anticipated expiration legal-status Critical
Ceased legal-status Critical Current

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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10L—SPEECH ANALYSIS TECHNIQUES OR SPEECH SYNTHESIS; SPEECH RECOGNITION; SPEECH OR VOICE PROCESSING TECHNIQUES; SPEECH OR AUDIO CODING OR DECODING
    • G10L19/00—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis
    • G10L19/02—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders
    • G10L19/0204—Speech or audio signals analysis-synthesis techniques for redundancy reduction, e.g. in vocoders; Coding or decoding of speech or audio signals, using source filter models or psychoacoustic analysis using spectral analysis, e.g. transform vocoders or subband vocoders using subband decomposition
    • G10L19/0208—Subband vocoders
    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04B—TRANSMISSION
    • H04B1/00—Details of transmission systems, not covered by a single one of groups H04B3/00 - H04B13/00; Details of transmission systems not characterised by the medium used for transmission
    • H04B1/66—Details of transmission systems, not covered by a single one of groups H04B3/00 - H04B13/00; Details of transmission systems not characterised by the medium used for transmission for reducing bandwidth of signals; for improving efficiency of transmission
    • H04B1/667—Details of transmission systems, not covered by a single one of groups H04B3/00 - H04B13/00; Details of transmission systems not characterised by the medium used for transmission for reducing bandwidth of signals; for improving efficiency of transmission using a division in frequency subbands

Definitions

  • the present invention relates to audio compression and decompression systems, and more particularly, to an improved filter design which reduces the computational workload inherent in audio compression and decompression systems.
  • One class of audio compression systems divides the sound track into a series of segments. Over the time interval represented by each segment, the sound track is analyzed to determine the signal components in each of a plurality of frequency bands. The measured components are then replaced by approximations requiring fewer bits to represent, but which preserve features of the sound track that are important to a human listener. At the receiver, an approximation to the original sound track is generated by reversing the analysis process with the approximations in place of the original signal components.
  • the analysis and synthesis operations are normally carried out with the aid of perfect, or near perfect, reconstruction filter banks.
  • the systems in question include an analysis filter bank which generates a set of decimated sub-band outputs from a segment of the sound track. Each decimated sub-band output represents the signal in a predetermined frequency range.
  • the inverse operation is carried out by a synthesis filter bank which accepts a set of decimated sub-band ovtputs and generates therefrom a segment of audio sound track.
  • the synthesis and analysis filter banks are implemented on digital computers which may be general purpose computers or special computers designed to more efficiently carry out the operations.
  • the segment of audio sound track generated by the synthesis filter bank will match the original segment of audio sound track that was inputted to the analysis filter bank.
  • the differences between the reconstructed audio sound track and the original sound track can be made arbitrarily small.
  • the present invention is based on the observation that the method by which polyphase quadrature filter banks operate may be reorganized to allow the cosine modulation of the polyphase components to be accomplished by discrete cosine transformation of a set of modified polyphase components.
  • the most recently received W input digital audio signal values of an audio signal are stored in the apparatus. These signal values are used to generate M sub-band component signals, where W ⁇ M by generating M modified polyphase components from the stored input digital audio signal values and then forming differences of sums of the weighted values.
  • the sub-band component signals are then generated from the discrete cosine transform of the modified polyphase components.
  • Figure 1 is a block diagram of an audio compression system.
  • Figure 2 illustrates the relationship of two overlapping audio segments.
  • Figure 3(a) is a block diagram of a single filter constructed from a low- frequency bandpass filter and a mixer.
  • Figure 3(b) is a block diagram of a sub-band analysis filter for generating a set of M frequency components, Sj, from a W sample window.
  • Figure 4 is a block diagram of a sub-band analysis filter according to the present invention.
  • Figure 5 is a block diagram of a synthesis filter bank according to the present invention.
  • FIG. 1 is a block diagram of an audio compression system 10 using a conventional sub-band analysis system.
  • the audio compression system accepts an input signal 11 which is divided into a plurality of segments 19. Each segment is aralyzed by a filter bank 12 which provides the frequency components for the segment. Each frequency component is a time average of the amplitude of the signal in a corresponding frequency band. The time average is, in general, a weighted average.
  • the frequencies of the sub-bands are uniformly distributed between a minimum and maximum value which depend on the number of samples in each segment 19 and the rate at which samples are taken.
  • the input signal is preferably digital in nature; however, it will be apparent to those skilled in the art that an analog signal may be used by including an analog-to-digital converter prior to filter bank 12.
  • the component waveforms generated by filter bank 12 are replaced by digital approximations by quantizer 14.
  • the number of bits assigned to each amplitude is determined by a psycho-acoustic analyzer 16 which utilizes information about the auditory system to minimize the distortions introduced by the quantization.
  • the quantized frequency components are then further coded by coder 18 which makes use of the redundancy in the quantized components to further reduce the number of bits needed to represent the coded coefficients. Coder 18 does not introduce further errors into the frequency components. Coding algorithms are well known to those skilled in the signal compression arts, and hence, will not be discussed in more detail here.
  • the manner in which the input signal is divided into segments can effect the quality of the regenerated audio signal.
  • the signal is analyzed on segments that do not overlap.
  • This analysis is equivalent to employing a model in which the regenerated signal is produced by summing the signals of a number of harmonic oscillators whose amplitudes remain constant over the duration of the segment on which each amplitude was calculated.
  • this model is a poor approximation to an actual audio track.
  • the amplitudes of the various frequency components would be expected to change over the duration of the segments in question.
  • Models that do not take this change into account will have significantly greater distortions than models in which the amplitudes can change over the duration of the segment, since there will be abrupt changes in the amplitudes of the frequency components at each segment boundary.
  • One method for reducing the discontinuities in the frequency component amplitudes at the segment boundaries is to employ a sub-band analysis filter that utilizes overlapping segments to generate successive frequency component amplitudes. The relationship ofthe segments is shown in Figure 2 for a signal 301.
  • the sub-band analysis filter generates M frequency components for signal 301 for each M signal values. However, each frequency component is generated over a segment having a duration much greater than M. Each component is generated over a segment having a length of W sample values, where W>M. Typical segments are shown at 312 and 313. It should be noted that successive segments overlap by (W-M) samples.
  • the various frequency bands in a sub-band analysis filter bank preferably have the same shape but are shifted relative to one another. This arrangement guarantees that all frequency bands have the same aliasing properties.
  • Such a filter bank can be constructed from a single low frequency bandpass filter having the desired band shape.
  • the manner in which the various filter bands are constructed is most easily understood with reference to Figure 3(a) which is a block diagram of a single filter constructed from a low-frequency bandpass filter 377 and a mixer 376. Assume that the low-pass filter 377 has a center frequency of F c and that the desired center frequency of filter 350 is to be F.
  • the output of low-frequency bandpass filter 377 will be the amplitude ofthe audio signal in a band having a center frequency of F. Modulator 376 accomplishes this frequency shift.
  • a filter bank can then be constructed from a single prototype low- frequency bandpass filter by using different modulation frequencies to shift the incoming audio signal prior to analysis by the prototype filter. While such a filter bank can be constructed from analog circuit components, it is difficult to obtain filter performance ofthe type needed. Hence, the preferred embodiment ofthe present invention utilizes digital filter techniques.
  • a block diagram of a sub-band analysis filter 350 for generating a set of M frequency components, S[, from a W sample window is shown in Figure 3(b).
  • the M audio samples are clocked into a W-sample shift register 320 by controller 325.
  • the oldest M samples in shift register 320 are shifted out the end ofthe shift register and discarded.
  • the polyphase components are generated by a windowing operation followed by partial summation.
  • the windowing operation generates a W- component array Zj from the contents of shift register 320 by multiplying each entry in the shift register by a corresponding weight, i.e.,
  • the frequency components, Sj are obtained via the following matrix multiplication from the polyphase components
  • This operation is equivalent to passing the polyphase components through M finite impulse response filters of length 2M.
  • the cosine modulation ofthe polyphase components shown in Eq. (3a) may be replaced by other such modulation terms.
  • the form shown in Eq. (3a) leads to near-perfect reconstruction.
  • An alternative modulation scheme which allows for perfect reconstruction is as follows:
  • the method ofthe present invention utilizes an observation that if the above equations are re- written, the cosine modulation ofthe polyphase components may be carried out by performing a discrete cosine transform (DCT) on a vector constructed from the polyphase components.
  • DCT discrete cosine transform
  • Eq. (4a) will be recognized by those skilled in the art as the inverse DCT-II transform ofthe first M points ofthe anti-symmetric part ofthe 2M sequence, P' ⁇ .
  • the P' sequence is related to the original polyphase components by a rotation.
  • Fast implementations of requiring ofthe order of Mlog 2 M multiplies and adds for computing DCT's and their inverses are known to the art, and hence, will not be discussed in more detail here. Readers interested in more details of these implementations are referred to K.R. Rao and P. Yip, DISCRETE COSINE TRANSFORM, Algorithms, Advantages, Applications, Academic Press.
  • FIG. 4 A block diagram of a sub-band analysis filter 380 for generating a set of M frequency components, Sj, from a W sample window is shown in Figure 4.
  • the M audio samples are clocked into a W-sample shift register 382 by controller 381.
  • the oldest M samples in shift register 382 are shifted out the end ofthe shift register and discarded.
  • the contents ofthe shift register are then used to generate the M rotated polyphase components (P' ⁇ to P ⁇ M- l - k )- These are then transformed by a DCT transform generator 384.
  • Transform generator 384 may be constructed from a general purpose digital computer or from special purpose hardware. Dedicated hardware for carrying out DCT transformation is known to those skilled in the art, and hence, will not be discussed further here.
  • each polyphase component involves multiply and add operations.
  • a further subtract per pair of polyphase components is involved in computing the anti-symmetric part of the rotated polyphase component which is then transformed by the DCT generator.
  • this subtraction operation is integrated into the windowing operation to directly obtain the desired anti ⁇ symmetric part ofthe rotated polyphase components. This integration allows one to take advantage of pipelined multiply-add hardware in those embodiments in which the windowing operation is performed on special purpose hardware.
  • the subtraction operation shown in Eq. (5) is preferably carried out as part ofthe windowing operation that generates the polyphase components P ⁇ , because the efficiencies of pipelined multiply and add hardware can be utilized.
  • the original time domain audio samples may be recovered by first recovering the polyphase components and then performing an inverse ofthe windowing operation described above.
  • the polyphase components are recovered by using the inverse ofthe DCT operation used to generate the sub-band components to generate a set of components Q j . For example, if the sub-band components were generated using Eq. (4), then one first computes
  • the time domain samples x ⁇ may be generated from the polyphase components, P ⁇ , by the inverse ofthe windowing transform described above. However, a computationally more efficient method may be utilized.
  • a block diagram of a synthesis filter bank according to the present invention is shown in Figure 5 at 500. M frequency components are received and stored in a register 502. The contents of register 502 are then transformed utilizing the inverse ofthe DCT transform that was used to generate the frequency components as shown at 508. For example, if the analysis filter bank used the DCT of Eq.(4a), then inverse DCT generator would use the transform given in Eq. (6).
  • inverse DCT generator 508 utilizes one ofthe fast computational methods to perform the transformation, i.e., a method having a computational complexity of order Mlog 2 M.
  • the transformation can be performed in special purpose hardware or on a general purpose computer.
  • the output of generator 508 is then converted to the polyphase components by generator 510 according to Eqs. (7) and (8).
  • the resultant 2M polyphase components are then shifted into a 2W entry shift register 512 and the oldest 2M values in the shift register are shifted out and discarded.
  • the contents in the shift register are inputted to array generator 513 which builds a W value array 514 by iterating the following loop 8 times: take the first M samples from shift register 512, ignore the next 2M samples, then take the next M samples.
  • the UJ are the contents of array 514.
  • weighted values may be obtained directly from the sub-band components, Qj using a W entry shift register in place of shift register 512.
  • the M values of Q obtained from the inverse DCT generator are directly shifted sequentially into a W-entry shift register, and the oldest M values in the shift register are discarded. That is, the polyphase components are not regenerated.
  • Uj the contents ofthe shift register.
  • the weighted array, W j used in Eq. (9), for M even, may then be obtained by forming sums of products of u and the prototype filter coefficients:

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  • Engineering & Computer Science (AREA)
  • Signal Processing (AREA)
  • Physics & Mathematics (AREA)
  • Computer Networks & Wireless Communication (AREA)
  • Spectroscopy & Molecular Physics (AREA)
  • Computational Linguistics (AREA)
  • Health & Medical Sciences (AREA)
  • Audiology, Speech & Language Pathology (AREA)
  • Human Computer Interaction (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Compression, Expansion, Code Conversion, And Decoders (AREA)

Abstract

Méthode et appareil pour améliorer le rendement machine de bancs de filtrage audio de reconstitution parfaite. L'invention met en ÷uvre un banc de filtrage d'analyse (350) conçu pour générer (324) une pluralité de signaux de sous-bande à partir d'un signal audio d'entrée (320) en procédant à une transformation à fenêtrage modifié, suivie d'une variation discrète en cosinus. Etant donné que cette variation peut s'opérer avec un nombre de calculs rationnel, on obtient une économie nette de complexité calculatoire par rapport aux anciens systèmes de filtrage utilisant la modulation en cosinus d'une série de composants polyphasés obtenus par transformation du fenêtrage. Le banc de filtrage de synthèse qui recombine le signal de sous-bande pour régénérer le signal audio d'origine est mis en ÷uvre par variation discrète inverse en cosinus, ce qui permet de réduire considérablement le nombre de calculs.
PCT/US1994/002026 1993-02-18 1994-02-16 Filtre ameliore pour systemes de compression et decompression audio Ceased WO1994019791A1 (fr)

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Application Number Priority Date Filing Date Title
AU62499/94A AU6249994A (en) 1993-02-18 1994-02-16 Improved filter for use in audio compression and decompression systems

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
US1924193A 1993-02-18 1993-02-18
US08/019,241 1993-02-18

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WO1994019791A1 true WO1994019791A1 (fr) 1994-09-01

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Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO1996030894A1 (fr) * 1995-03-27 1996-10-03 Dolby Laboratories Licensing Corporation Procede et appareil destines a la mise en ×uvre efficace de bancs de filtrage a bande laterale unique fournissant des mesures precises de la phase et de la puissance spectrales
KR100460159B1 (ko) * 1996-03-19 2005-02-23 루센트 테크놀러지스 인크 오디오신호인코딩방법및장치

Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4393456A (en) * 1981-03-19 1983-07-12 Bell Telephone Laboratories, Incorporated Digital filter bank

Patent Citations (1)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
US4393456A (en) * 1981-03-19 1983-07-12 Bell Telephone Laboratories, Incorporated Digital filter bank

Cited By (2)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
WO1996030894A1 (fr) * 1995-03-27 1996-10-03 Dolby Laboratories Licensing Corporation Procede et appareil destines a la mise en ×uvre efficace de bancs de filtrage a bande laterale unique fournissant des mesures precises de la phase et de la puissance spectrales
KR100460159B1 (ko) * 1996-03-19 2005-02-23 루센트 테크놀러지스 인크 오디오신호인코딩방법및장치

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AU6249994A (en) 1994-09-14
IL108683A0 (en) 1994-05-30

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