AU640667B2 - Polyphonic coding - Google Patents

Abstract

A polyphonic (e.g. stereo) audioconferencing system, in which input left and right channels are time-aligned by variable delay stages (10a, 10b), controlled by a delay calculator (9) (e.g. by deriving the maximum cross-correlation value), and then summed in an adder (2) and subtracted in subtracter (3) to form sum and difference signals. The sum signal is transmitted in relatively high quality; the difference signal is reconstructed at the decoder by prediction from the sum signal using an adaptive filter (5). The decoder adaptive filter (5) is configured either by received filter coefficients or, using backwards adaptation, from a received residual signal produced by a corresponding adaptive filter (4) in the coder, or both. Preferably, the adaptive filter (4) is a lattice filter, employing a gradient algorithm for coefficient update. The complexity of the adaptive filter (4) is reduced by pre-whitening, in the encoder, both the sum and difference signals using corresponding whitening filters (14a, 14b) derived from the sum channel.

Description

OPI UATE U8/01/91 AOJP DATE 28/02/91 APPLN. ID 583/9 PCT NUMBER PCT/GB90/00928

PCT

INTERNATIONAL APPLICATION PUBLISHED UNDER THE PATENT COOPERATION TREATY (PCT) (51) International Patent Classification 5 (11) International Publication Number: WO 90/16136 H04S 1/00, H04M 3/56 Al H04H 5/00 (43) Intenational Publication Date: 27 December 1990 (27.12.90) (21) International Application Number: PCT/GB90/00928 (74) Agent: MUSKER, David, Charles; British Telecommunications public limited company Intellectual Property (22) International Filing Date: 15 June 1990 (15.06.90) Unit, 15! Gower Street, London WCIE 6BA (GB).

Priority datp: (81) Designated States: AT (European patent), AU, BE (Euro- 8913758.1 15 June 1989 (15.06.89) GB pean patent), CA, CH (European patent), DE (European patent)*, DK (European patent), ES (European patent), FI, FR (European patent), GB (European patent), (71) Applicant (for all designated States except US): BRITISH IT (European patent), JP, LU (European patent), NL TELECOMMUNICATIONS PUBLIC LIMITED (European patent), NO, SE (European patent), US.

COMPANY [GB/GB]; 81 Newgate Street, London ECIA 7AJ (GB).

Published (72) Inventors; and With international search report.

Inventors/Applicants (for US only) HOLT, Christopher, Ellis [GB/GB]; 3 Upper Melton Terrace, Melton, Woodbridge, Suffolk IP12 1QJ MUNDAY, Edward [GB/GB]; 1A3 Humber Doucy Lane, Ipswich, Suffolk IP4 3PA CHEETHAM, Barry, Michael, George [GB/GB]; 99 South Mossley Hill Road, Liverpool L19 9BQ (GB).

640667 (54) Title: POLYPHONIC CODING x xS 6 11a Ba ADAPTIVE FILTER 5 DELAY FILTER COEFFS CALCULATOR CONTROLLED 1 FILTER 11b b SO- DIFFERENCE SIGNAL x I i1 x D

RECONSTR

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ITON RCTOSTRUCTION xgO I b DATA D I 10b 3 I I LA DELAY ELAYD (57) Abstract A polyphonic stereo) audioconferencing system, in which input left and right channels are time-aligned by variable delay stages (10a, 10b), controlled by a delay calculator by deriving the maximum cross-correlation value), and then summed in an adder and subtracted in subtracter to form sum and difference signals. The sum signal is transmitted in relatively high quality; the difference signal is reconstructed at the decoder by prediction from the sum signal using an adaptive filter The decoder adaptive filter is configured either by received filter coefficients or, using backwards adaptation, from a received residual signal produced by a corresponding adaptive filter in the coder, or both. Preferably, the adaptive filter is a lattice filter, employing a gradient algorithm for coefficient update. The complexity of the adaptive filter is reduced by prewhitening, in the encoder, both the sum and difference signals using crresponding whitening filters (14a, 14b) derived from the sum channel.

See back of page WO 90/16136 PCT/GB90/00928 -1- POLYPHONIC CODING This invention relates to polyphonic coding techniques, particularly, but not exclusively, for coding speech signals.

It is well-known that polyphonic, specifically stereophonic, sound is more perceptually appealing than monophonic sound. Where several sound sources, say within a conference room, are to be transmitted to a second room, polyphonic sound allows a spatial reconstruction of the original sound field with an image of each sound source being perceived at an identifiable point corresponding to its position in the original conference room. This can eliminate confusion and misunderstandings during audio-conference discussions since each participant may be identified both by the sound is of his voice and by his perceived position within the conference room.

Inevitably, polyphonic transmissions require an increase in transmission capacity as compared with monophonic transmissions. The conventional approach of transmitting two independent channels, thus doubling the required transmission capacity, imposes an unnaceptably high cost penalty in many applications and is not possible in some cases because of the need to use existing channels with fixed transmission capacities.

In stereophonic two-channel polyphonic) systems, two microphones (hereinafter referred to as left and right microphones), at different positions, are used to pick up sound generated within a room (for example by a person or persons speaking). The signals picked up by the microphones are in general different. Each microphone signal (referred to hereinafter as xL(t) with Laplace WO 90/16136 PCT/GB90/00928 2 transform XL(s) and xR(t) with Laplace transform XR(s) respectively) may be considered to be the superposition of source signals processed by respective acoustic transfer functions. These transfer functions are strongly affected by the distances between the sound sources and each microphone and also by the acoustic properties of the room. Taking the case of a single source, e.g. a single person speaking at some fixed point within the room, the distances between the source and the left and right microphones give rise to different delays, and there will also be different degrees of attenuation.

In most practical environments such as conference rooms, the signal reaching each microphone may have travelled via many reflected paths from walls or ceilings) as well as directly, producing time spreading, frequency dependent colouration due to resonances and antiresonances, and perhaps discrete echos.

From the foregoing, in theory, the signal from one microphone may be formally related to that from the other by designating an interchannel transfer function H say; i.e. XL(s) H(s) where s is complex frequency parameter. This statement is based on an assumption of linearity and time-invariance for the effect of room acoustics on a sound signal as it travels from its source to a microphone. However, in the absence of knowledge as to the nature of H, this statement does no more than postulate a correlation between the two signals. Such a postulation seems inherently sensible, however, at least in the special case of a single sound source, and therefore one way of reducing the bit-rate needed to represent stereo signals should be to reduce the redundancy of one relative to the other (to reduce this correlation) prior to transmission and re-introduce it after reception.

3 In general, H(s) is not unique and can be signal and time dependent. However, when the source signals are white and uncorrelated, i. e. when their autocorrelation functions are zero except at t=0 and their crosscorrelation functions are zero for all t, H(s) will depend on factors not subject to rapid change, such as room acoustics and the positions of the micrc-hones and sound sources, rather than the nature of the source signals which may be rapidly changing.

To realise such a system in physical form, the fundamental problems of causality and stability must be overcome. Consider for a moment a si-ngle source signal which is delayed by d 1 seconds before reaching the left microphone and by dR seconds before reaching the right microphone (although the point to be made has more general implications). If the source is near to, say, the left microphone, then d L will be smaller than dr. The interchannel transfer function H(s) must delay xL(t) by the difference between the two delays, dR dL to produce the right channel xR(t). Since dR d, is positive, H(s) will be causal. If the signal source is now moved closer to the right microphone than to the left, dR d, becomes negative and H(s) becomes negative and H(s) becomes non-causal; in other words, there is no causal relationship between the right channel and the left channel, but rather the reverse so the right channel can no longer be predicted from the left channel, since a given event occurs first in the right channel. It will therefore be realised that a simple system in which one fixed channel is always transmitted and the other is reconstructed from it is impossible to realise in a direct sense.

According to a first aspect of the invention, there is provided polyphonic signal coding apparatus comprising: input means for receiving polyphonic input signals; 4 means connected to the input means for producing a first and at least one further channel from the input signals; means connected to the channel producing means and responsive to the first and further channels for periodically generating a plurality of filter coefficients which, if applied to a plural order predictor filter, would enable the prediction of the at least one further channel from the first channel thus filtered; and means for outputting data representing the first channel and further channel reconstruction data including data representing said filter coefficients to enable the reconstruction of the at least one further channel therefrom.

According to a second aspect of the invention, there is provided polyphonic signal coding apparatus comprising: input means for receiving polyphonic input signals; means connected to the input means for producing a first and at least one further ,hannel from the input signals; means connected to the channel producing means and responsive to the first and further channels for periodically generating a plurality of filter coefficients which, if applied to a plural order predictor filter, would enable the prediction of the at least one further channel from the first channel thus filtered; the generating means and including an adapted filter connected to receive the first channel and produce a predicted further channel therefrom and means for producing a residual signal representing the difference between the predicted further channel and the actual further channel; and means for outputting data representing the said first channel and further channel reconstruction data comprising data representing the said residual signal, to enable the reconstruction of the said further channel S therefrom.

4a In this embodiment, the prediction residual signal may be efficiently encoded to allow a backward adaption technique to be used at the decoder for deriving the prediction filter coef-'cients. The residual is also used as an error signal which is added to the prediction filter's output at the decoder to correct for inaccuracies in the prediction of the difference channel from the sum channel. This "residual only" embodiment is also useful where the left channel, say, is predicted from the right channel (without forming sxm and difference signals) provided suitable measures are taken to ensure causality to give high quality polyphonic reproduction. In a third embodiment, both are transmitted.

Preferably, the means for generating the filter coefficients is an adaptive filter, advantageously a lattice filter. This type of filter also gives advantages in non-sum and difference polyphonic systems.

zccording to a further aspect of the invention, there is provided a method of coding polyphonic input signals comprising: producing a sum signal representing the sum of such polyphonic input signals; producing at least one difference signal representing a difference between said polyphonic input signals analysing said sum and difference signals and generating therefrom a plurality of coefficients which, if applied to a multi-stage predictor filter, would enable the prediction of the at least one difference signal from the sum signal thus filtered; and outputting the said sum signal and data to enable the reconstruction of the said at least one difference signal therefrom.

WO 90/16136 PCT/GB90/00928 5 In preferred embodiments, variable delay means are disposed in at least one of the input signal paths, and controlled to time align the two signals prior to forming the sum and difference signals so that causal prediction filters of reasonable order can be used.

This aspect of the invention has several important advantages: The 'sum signal' is fully compatible with monophonic encoding and is unaffected by the polyphonic coding except for the introduction of an imperceptible delay. In the event of loss of stereo, monophonic back-up is thus available.

(ii) The sum signal may be transmitted by conventional low bit-rate coding techniques (eg. LPC) without modification.

(iii) The encoding technique for the difference signals can be varied to suit the application and the available transmission capacity between the above three embodiments. The type of residual signal and prediction coefficients can also be selected in varicus different ways, while still conforming to the basic encoding principle.

(iv) Overall, the apparatus encodes polyphonic signals with only a modest increase in bit-rate requirement as compared with monophonic transmission.

The encoding is digital and hence the performance of the apparatus will be predictable, not subject to ageing effects or component drift and easily mass-produced.

A method of calculating approximations to H(s) when the source signals are not white (which, of course, includes all speech or music signals) is proposed in a second aspect of the invention, using the idea of a 'prewhitening filter'.

6 According to a further aspect of the invention, there is provided polyphonic signal decoding apparatus comprising: means for receiving encoded data representing a sum signal, and difference signal reconstruction data; a configurable plural order predictor filter for receiving said difference signal reconstruction data and modifying its coefficients in accordance therewith, the filter being connected to receive the said sum signal and reconstruct therefrom an output difference signal; and means for adding the reconstructed difference signal to the sum signal, and for subtracting the reconstructed difference signal from the sum signal, so as to produce a polyphonic signal.

One embodiment of the invention seeks to provide the advantages of a digital system compatible with existing techniques and simplify the process of modelling (at the encoder) the required interchannel transfer function.

Broadly corresponding decoding apparatus is also provided according to the invention, as are systems including such encoding and decoding apparatus, particularly in a audioconferencing application, but also in a polyphonic recording application. Other aspects of the invention are as claimed and disclosed herein.

The words "prediction" and "predictor" in this specification include not only prediction of future data from past data, but also estimation of present data of a channel from past and present data of another channel.

The invention will now be illustrated, by way of example only, with reference to the accompanying drawings in which: Figure 1 illustrates generally an encoder according to a first aspect of the invention; WO 90/16136 PCT/GB90/00928 7 Figure 2 illustrates generally a corresponding decoder; Figure 3a illustrates an encoder according to a preferred embodiment of the invention; Figure 3b illustrates a corresponding decoder; Figures 4a and 4b show respectively a corresponding encoder and decoder according to a second aspect of the invention.

Figures 5a and 5b illustrate an encoder and a decoder according to a second aspect of the invention; Figure 6 illustrates part of an encoder according to a yet further embodiment of the invention.

The embodiments illustrated are restricted to 2 channels (stereo) for ease of presentation, but the invention may be generalised to any number of channels.

One possible way of removing the redundancy between two input signals (or predicting one from the other) would be to connect between the two channels an adaptive predictor filter whose slowly changing parameters are calculated by standard techniques (such as, for example, block cross-correlation analysis or sequential lattice adaptation). In an audioconferencing environment, the two signals will originate from sound sources within a room, and the acoustic transfer function between each source and each microphone will be characterised typically by weak poles (from room resonances) and strong zeros (due to absorption and destructive interference). An all-zero filter could therefore produce a reasonable approximation to the acoustic transfer function between a source and a microphone and such a filter could also be used to predict say the left microphone signal XL(t) from xR(t) when the source is close to the right microphone. However, if the source were now moved away from the right microphone and placed close to the left, the nature of the required WO 90/16136 PCr/GB90/00928 -8- 8 filter would be effectively inverted even when delays are introduced to guarantee causality. The filter must now model a transfer function with weak zeros and strong poles a difficult task for an all-zero filter. Other types of filter are not, in general, inherently stable. The net effect of this is to cause unequal degradation in the reconstructed channel when the source shifts from one microphone to the other. This further makes the simplistic prediction of one channel (say, the left) from the other (say, the right) hard to realise.

In a system according to the first aspect of the invention, better results have been obtained by forming a "sum signal" xs(t) XL(t) xR(t) and predicting either a difference signal xD(t) xL(t) xR(t) or simply xL(t) or xR(t) using an all-zero adaptive digital filter.

In practice, xR(t) and xL(t) (or xS(t) and xD(t) will be processed in sampled data form as the digital signals xR[n] and xL[n] or xs[n] and xD[n] and it will be more convenient to use the 'z-transform' transfer fuction H(z) rather than H(s).

Referring to Figure 1, in its essential form the invention comprises a pair of inputs la, lb for receiving a pair of speech signals, e.g. from left and right microphones. The signals at the inputs, xR(t) and xL(t), may be in digital form. It may be convenient at this point to pre-process the signals, e.g. by band limiting. Each signal is then supplied to an adder 2 and a subtractor 3, the output of the adder being the sum signal xS(t) xR(t) xL(t), and the output of the subtracter 3 being the difference signal xD(t) xR(t) xL(t) i.e. XD(t) H(s) XS(s). The sum and difference signals are then supplied to filter derivation stage 4, which derives the coefficients of a multi-stage WO 90/161136 PCr/GB9/00928 -9prediction filter which, when driven with the sum signal, will approximate the difference signal. The difference between the approximated difference signal and the actual difference signal, the prediction residual signal, will usually also be produced (although this is not invariably necessary). The sum signal is then encoded (preferably using LPC or sub-band coding), for transmission or storage, along with further data enabling reconstruction of the difference signal. The filter coefficients may be sent, or alternatively (as discussed further below), the re-Aldual signal may be transmitted, the difference channel being reconstituted by deriving the filter parameters at the receiver using a backwards adaptive process known in the art; or both may be transmitted.

Although it would be possible to calculate filter parameters directly (using LPC analysis techniques), one simple and effective way of providing the derivation stage 4 is to use an adaptive filter (for example, an adaptive transversal filter) receiving as input the sum channel and modelling the difference channel so as to reduce the prediction residual. Such' general techniques of filter adaptation are well-known in the art.

Our initial experiments with this structure have used a transversal FIR filter with coefficient update by an algorithm for minimising the mean square value of the residual, which is sir ple to implement. The filter coefficients change only slowly because the room accoustic (and hence the interchannel transfer function) is relatively stable.

Referring to Figure 2, in a corresponding receiver, the sum signal xS(t) is received together with either the filter parameters or the residual signal, or both, for the difference channel, and an adaptive filter corresponding to that for which the parameters were WO 90/16136 PCT/GB9O/00928 10 derived at the coder receives as input the sum signal and produces as output the reconstructed difference signal when configured either with the received parameters or with parameters derived by backwards adaptation from the received residual signal. Sum and difference signals are then both fed to an adder 6 and a subtracter 7, which.

produce as outputs respectively the reconstructed left and right channels at output nodes 8a and 8b.

Since a high-quality sum signal is sent, the encoder i0 is fully mono-compatible. In the event of loss of stereo information, monophonic back-up is thus available.

As discussed above, one component of the transfer functions HL and HR is a delay component relating to the direct distance between the signal source and each of the microphones, and there is a corresponding delay difference d. There is thus a strong cross-correlation between one channel and the other when delayed by d.

This method, however, requires considerably processing power.

An alternative method of delay estimation found in papers on sonar research is to use an adaptive filter.

The left channel input is delayed by half the filter length and the coefficients are updated using the LMS algorithm to minimise the mean-square error or the output. The transversal filter coefficients will, in theory, become the required cross-correlation coefficients. This may seem like unnecessary repetition of filter coefficient derivation were it not for the property of this delay estimator that the maximum value of the cross-correlation coefficient (at the position of the maximum filter coefficient) is obtained some time before the filter has converged. This method may be improved further because spatial information is also available from the relative amplitudes of the input channels; this could WO 90/16136 PCT/GB90/00928 11 be used to apply a weighting function to the filter coefficients to speed convergence.

Referring to Figure 3a, in a preferi embodiment of the invention, the complexity and length of the filter to be calculated is therefore reduced by calculating the required value of d in a delay calculator stage 9 (preferably employing one of the above methods), and then bringing the channels into time alignment by delaying one or other by d using, for example, a pair of variable.

delays 10a, 10b (although one fixed and one variable delay could be used) controlled by the delay calculator 9. With the major part of the speech information in the channels time aligned, the sum and difference signals are then formed.

Referring to Figure 3b, the delay length d is preferably transmitted to the decoder, so that after reconstructing the difference channel and subsequently the left and right channels, corresponding variable length delay stages lla, 1ib in one or other of the channels can restore the interchannl delay.

In the illustrated structure, the "sum" signal is thus no longer quite the true sum of xL(t) xR(t); because of the delay d it is xL(t) xR(t-d). It may therefore be preferred to locate the delays 10a, 10b (and, possibly, the delay calculator) downstream of the adder and subtractor 2 and 3; this gives, for practical purposes, the same benefits of reducing the necessary filter length., In practice, the delay is generally imperceptible; typically, up to 1.6 ms. Alternatively, a fixed delay, sufficiently long to guarantee causality, may be used, thus removing the need to encode the delay parameter.

In the first embodiment of the invention, as stated above, only the filter parameters are transmitted as WO 90/16136 PCT/GB90/00928 12 difference signal data. With 16 bits per coefficient, this meant that a transmission capacity of 5120 bits/sec is needed for the difference channel (plus 8 bits for the delay parameter). This is well within the capacity of a standard 64 kbit/sec transmission system used which allocates 48 kbits/sec to the sum channel (efficiently transmitted by an existing monophonic encoding technique) and offers 16 kbits/sec for other "overhead" data. This mode of the embodiment gives a good signal to noise ratio and the stereo imaje is present, although it is highly dependent on the arcuracy of the algorithm used to adapt the predictive filter. Inaccuracies tend to cause the stereo image to wander during the course of a conference particularly when the conversation is passed from one speaking person to another at some distance from the first.

Referring to Figure 4a, in a second embodiment of the invention, only the residual signal is transmitted as difference signal data. The sum signal is encoded (12a) using, for example, sub-band coding. It is also locally decoded (13a) to provide a signal equivalent to that at the decoder, for input to adaptive filter 4. The residual difference channel is also encoded (possibly including bandlimiting) by residual coder 12b, and a corresponding local decoder 13b provides the signal minimised to adapt filter 4. The advantage this creates is that inaccuracies in generating the parameters cause an increase in the dynamic range of the residual channel and a corresponding decrease in SNR, but with no loss in stereo image.

Referring to Figure 4b, at the decoder, the analysis filter parameters are recovered from the transmitted residual by using a backwards-adapting replica filter 5 of the adaptive filter 4 at the coder. Decoders 13c, 13d are identical to local decoders 13a, 13b and so the filter receives the same inputs, and thus produces the same parameters, as that of encoder filter 4.

WO 90/16136 PC~/GB90/00928 13 In a further embodiment (not shown), both filter parameters and residual signal are transmitted as side-information, overcoming many of the problems with the residual-only embodiment because the important stereo information in the first 2 kHz is preserved intact and the relative amplitude information at higher frequencies is largely retained by the filter parameters.

Both the above residual-only and hybrid residual plus pa:ameters) embodiments are preferably employed, as described, to predict the difference channel from the sum channel. However, it is found that the same advantages of retaining the stereo image (albeit with a decrease in SHR)' are found when the input channels are left and right, rather' than sum and difference, provided the problem of causality is overcome in some manner by inserting a relatively long fixed delay in one or other path). The scope of the invention therefore encompasses this also.

The parameter-only embodiment described above preferably uses a single adaptive filter 4 to remove redundancy between the sum and difference channels. An effect discovered during testing was a curious 'whispering' effect if the coefficients were not sent at a certain rate, which was far above what should have been necessary to describe changes in the acoustic environment. This was because the adaptive filter, in addition to modelling the room acoustic transfer function, was also trying to perform an LPC analysis of the speech.

This is solved in the second aspect of the invention by whitening the spectra of the input signals to the adaptive filter as shown in Figure 5, so as to reduce the rapidly-changing sppac component leaving principally the room acoustic component.

In the second aspect of the invention, thL adaptive filter 4 which models the acoustic transfer functions may WO 90/16136 PCT/GB90/00928 14 be the same as before (for example, a lattice filter of order 10). The sum channel is passed through a whitening filter 14a (which may be lattice or a simple transversal structure).

The master whitening filter 14a receives the sum channel and adapts to derive an approximate spectral inverse filter to the sum signal (or, at least, the speech components thereof) by minimising its own output. The output of the filter 14a is therefore substantially white. The parameters derived by the master filter 14a are supplied to the slave whitening filter 14b, which is connected to receive and filter the difference signal.

The output of the slave whitening filter 14b is therefore the difference signal filtered by the inverse of the sum signal, which substantially removes common signal components, reducing the correlation between the two and leaving the output of 14b as consisting primarily of the acoustic response of the room. It thus reduces the dynamic range of the residual considerably.

The effect is to whiten the sum channel and to partially whiten the difference channel without affecting the sjecttal differences between them as a result of room acoustics, so that the derived coefficients of adaptive filter 4 are model parameters of the room acoustics.

In one embodiment, the coefficients only are transmitted and the decoder is simply that of Figure 2 (needing no further filters). In this embodiment, of course, residual encoder 12b and decoder 13b are omitted.

An adaptive filter will generally not be long enough to filter out long-term information, such as pitch information in speech, so the sum channel will not be completely "white". However, if a long-term predictor (known in LPC coding) is additionally employed in filters 14a and 14b, then filter 4 could, in principle, be WO 90/16136 PCT/GB90/00928 15 connected to filter the difference channel alone, and thus to model the inverse of the room acoustic.

Since this second aspect of the invention reduces the dynamic range of the residual, it is particularly advantageous to employ this whitening scheme with the residual-only transmission described above. In this case, prior to backwards adaptation at the decoder, it is necessary to filter the residual using the-inverse of the whitening filter, or to filter the sum channel using the 1o whitening filter. Either filter can be derived from the sum channel information which is transmitted.

Referring to Figure 5b, in residual-only transmission, an adaptive whitening filter 24a (identical to 14a at the encoder) receives the (decoded) sum channel and adapts to whiten its output. A slave filter 24b (identical to 14b at the encoder) receives the coefficients of 24a, Using the whitened sum channel as its input, and adapting from the (decoded) residual by backwards adaptation, adaptive filter 5 regenerates a filtered signal which is added to the (decoded) residual and the sum is filtered by slave filter 24b to yield the difference channel. The sum and difference channels are then processed 7 not shown) to yield the original left and right channels.

In a further embodiment (not shown), both residual and coefficients are transmitted.

Although this pre-whitening aspect of the invention has been described in relation to the preferred embodiment of the invention using sum and difference channels, it is also applicable where the two channels are 'left' and 'right' channels.

For a typical audioconferencing application, the residual will have a bandwidth of 8 kHz and must be quantised and transmitted using spare channel capacity of about 16 kbit/s. The whitened residual will be, in WO 90/16136 PCT/GB90/00928 16 principle, small in mean square value, but will not be optimally whitened since the copy pre-whitening filter 14b through which the residual passes has coefficients derived to whiten the sum channel and not necessarily the difference channel. Typically, the dynamic range of the filtered signal is reduced by 12dB over the unfiltered difference channel. O. ne approach to this residual quantisation problem is to reduce the bandwidth of the residual signal. This allows downsampling to a lower rate, with a consequential increase in bits per sample.

It is well known that most of the spatial information in a stereo signal is contained within the 0-2 kHz band, and therefore reducing the residual bandwidth from 8 kHz to a value in excess of 2 kHz does not affect the perceived stereo image appreciably. Results have shown that reducing the residual bandwidth to 4 kHz (and taking the upper 4 kHz band to be identical to that of the sum channel) produces good quality stereophonic speech when the reduced bandwidth residual is sub-band coded using a standard technique.

Experiments with various adaptive filters for the filter 4 (and, where applicable, 12) showed that a standard transversal FIR filter was slow to converge.

A faster performance can be obtained by using a lattice structure, with coefficient update using a gradient algorithm based on Burg's method, as shown in Figure 7.

The structure uses a lattice filter 14a to pre-whiten the spectrum of the primary input. The decorrelated backwards residual outputs are then used as inputs to a simple linear combiner which attempts to model the input spectrum of the secondary input. Although the modelling process is the same as with the simple transversal FIR filter, the effect of the lattice filter is to point the error vector in the direction of the optimum LMS residual WO 90/16136 PCT/GB90/00928 17 solution. This speeds convergence considerably.

A lattice filter of order 20 is found effective in practice.

The lattice filter structure is particularly useful as described above, but could also be used in a system in which, instead of forming sum and difference signals, a (suitably delayed) left channel is predicted from the right channel.

Although the embodiments described show a stereophonic system, it will be appreciated that with, for example, quadrophonic systems, the invention is implemented by forming a sum signal and 3 difference signals, and predicting each from- the sum signal as above.

Whilst the invention has been described as applied to a low bit-rate transmission system, e.g. for teleconferencing, it is also useful for example for digital storage of music on well known digital record carriers such as Compact Discs, by providing a formatting means for arranging the data in a format suitable for such record carriers.

Conveniently, much or all of the signal processing involved is realised in a single suitably programmed digital signal processing (dsp) chip package; two channel packages are also commercially available. Software to implement adaptive filters, LPC- analysis and cross-correlations are well known.

Claims (14)

1. Polyphonic signal coding apparatus comprising: input means for receiving polyphonic input signals; means connected to the input means for producing a first and at least one further channel from the input signals; means connected to the channel producing means and responsive to the first and further channels for periodically generating a plurality of filter coefficients which, if applied to a plural order predictor filter, would enable the prediction of the at least one further channel from the first channel thus filtered; and means for outputting data representing the first channel and further channel reconstruction data including data representing said fi-..er coefficients to enable the reconstruction of the at least one further channel therefrom.

2. Polyphonic signal coding apparatus comprising: input means for receiving polyphonic input signals; r.eans connected to the input means for producing a first and at least one further channel from the input signals; means connected to the channel producing means and responsive to the first and further channels for periodically generating a plurality of filter coefficients which, if applied to a plural order predictor filter, would enable the prediction of the at least one further channel from the first channel thus filtered; the generating means including an adaptive filter connected to receive the first channel and produce a predicted further channel therefrom and means for producing a residual signal representing the difference between the predicted further channel and the actual further channel; and means for outputting data representing the said. first channel and further' channel reconstruction data z-" 19 comprising data representing the said residual signal, to enable the recoiistruction of the said further channel therefrom.

3. Apparatus according to claim 2 wherein the further channel reconstruction data also comprises the said filter coefficients.

4. Apparatus according to claim 2 in which the adaptive filter is controlled only by the said residual signal and the said further channel reconstruction data consists of the said residual signal.

Apparatus according to any one of the preceding claims further comprising: means for filtering the first and at least one further channel in accordance with a filter approximating the spectral inverse of the first channel to produce respective filtered channels, the first said filtered channel thereby being substantially spectrally whitened; the generating means being connected to receive the filtered channels.

6. Apparatus according to claim 5, wherein said filtering means comprises an adaptive, master, filter arranged to filter the first channel so as to produce a whitened output, and a slave filter arranged to filter said at least one further channel, the slave filter being configured so as to have an equivalent response to the adaptive filter of the filtering means.

7. Apparatus according to any one of claims 1 to 6, wherein the first channel is a sum channel representing the sum of the polyphonic input signals and the further channels represent the differences therebetween.

8. Apparatus according to any preceding claim, in which the input means includes variable delay means for delaying at least one of the input signals, and means for controlling the differential delay applied to the signals so as to increase correlation upstream of the generating means, the output means being arranged also to output data representing the said differential delay.

9. Polyphonic signal decoding apparatus comprising: SA4" AXIT~ 20 means for receiving encoded data representing a sum signal, and difference signal reconstruction data; a configurable plural order predic.or filter for receiving said difference signal reconstruction data and modifying its coefficients in accordance therewith, the filter being connected to receive the said sum signal and reconstruct therefrom an output difference signal; and means for adding the reconstructed difference signal to the sum signal, and for subtracting the reconstructed difference signal from the sum signal, so as to produce a polyphonic signal.

Apparatus as claimed, in claim 9, in which the difference signal reconstruction data comprises residual signal data and the apparatus includes means to add the residual signal data to the'output of the filter to form the reconstructed difference signal.

11. Apparatus as claimed in claim 10 in which the predictor filter is connected to receive the residual signal data and to modify its coefficients in accordance therewith.

12. A method of coding polyphonic input signals comprising: producing a sum signal representing the sum of such polyphonic signals; producing at least one difference signal representing a difference between said polyphonic input signals; analysing said sum and difference signals and generating therefrom a plurality of coefficients which, if applied to a multi-stage predictor filter, would enable the prediction of the at least one difference signal from the sum signal thus filtered; and outputting the said sum signal a.id data enabling the reconstruction of the said at least one difference signal therefrom.

13. Signal coding apparatus substantially as herein described, with reference to any of the accompanying figures 1, 3a, 4, 5 or 6. 21

14. Signal decoding apparatus substantially as herein described, with reference to any of the accompanying figures 2, 3b, 4 or DATED this 29th day of June 1993 BRITISH TELECOM~MUNICATIONS Public limited company Atforney. FPEMEf IFEATHSATE F'low Insijft3 df Pale-t Af-urnvs of Australia of SHELSTONIN PA-1-EJlS ,a-