Title of Invention	AN APPARATUS FOR ENCODING A STEREO SIGNAL TO OBTAIN A MONO OUTPUT SIGNAL AND A STEREO PARAMETER SET
Abstract	The invention relates to an apparatus for encoding a stereo signal to obtain a mono output signal and a stereo parameter set, comprising means for calculating the mono signal by combining a left and a right channel of the stereo signals; means (403) for generating a first stereo parameter set using a portion of the left channel and a portion of the right channel, the portion starting at a first time border; means (401,402) for determining a validity of the first stereo parameter set for subsequent portions of the left channel and the right channel, wherein the means for determining is operative to generate second time border, and activate the means for generating, when it is determined that the stereo parameter set is not valid anymore so that a second stereo parameter set for portions of the left and right signals starting at the second time border is generated; and means for outputting the mono signal and the first stereo parameter set and the first time border associated with the first parameter set, and the second stereo parameter set and the second time border associated with the second stereo parameter set.

Title of Invention

AN APPARATUS FOR ENCODING A STEREO SIGNAL TO OBTAIN A MONO OUTPUT SIGNAL AND A STEREO PARAMETER SET

Abstract

The invention relates to an apparatus for encoding a stereo signal to obtain a mono output signal and a stereo parameter set, comprising means for calculating the mono signal by combining a left and a right channel of the stereo signals; means (403) for generating a first stereo parameter set using a portion of the left channel and a portion of the right channel, the portion starting at a first time border; means (401,402) for determining a validity of the first stereo parameter set for subsequent portions of the left channel and the right channel, wherein the means for determining is operative to generate second time border, and activate the means for generating, when it is determined that the stereo parameter set is not valid anymore so that a second stereo parameter set for portions of the left and right signals starting at the second time border is generated; and means for outputting the mono signal and the first stereo parameter set and the first time border associated with the first parameter set, and the second stereo parameter set and the second time border associated with the second stereo parameter set.

Full Text	ADVANCED PROCESSING BASED ON A COMPLEX-EXPONENTIAL-MODULATED FILTERBANK AND ADAPTIVE TIME SIGNALLING METHODS TECHNICAL FIELD The present invention relates to audio source coding systems but the same methods could also be applied in many other tech- nical fields. Different techniques that are useful for audio coding systems using parametric representations of stereo prop- erties are introduced. EACKGROUND OF THE INVENTION AND PRIOR ART The present invention relates to parametric coding of the ste- reo image of an audio signal. Typical parameters used for de- scribing stereo image properties are inter-channel intensity difference (IID), inter-channel time difference (ITD), and in- ter-channel coherence (IC). In order to re-construct the stereo image based on these parameters, a method is required that can re-construct the correct level of correlation between the two channels, according to the IC parameter. This is accomplished by a de-correlation method. There are a couple of methods available for creation of decor- related signals. Ideally,, a linear time invariant (LTI) func- tion with all-pass frequency response is desired. One obvious method for achieving this is by using a constant delay. How- ever, using a delay, or any other LTI all-pass functions, will result in non-all-pass response after adding the non-processed signal. In the case of a delay, the result will be a typical comb-filter. The comb-filter often gives an undesirable "me- CROSS REFERENCE The instant Indian Patent Application constitutes a divisional application divided out from Indian Patent Application no. 2162/KOLNP/05 of 28.10.2005 which is now been granted of Indian Patent No. 242206. FIELD OF THE INVENTION The present invention relates to audio source coding systems but the same methods could also be applied in many other technical fields. Different techniques that are useful for audio coding systems using parametric representations of stereo properties are introduced. BACKGROUND OF THE INVENTION The present invention relates to parametric coding of the stereo image of an audio signal. Typical parameters used for describing stereo image properties are inter-channel intensity difference (IID), inter-channel time difference (ITD), and inter-channel coherence (IC). In order to re-construct the stereo image based on these parameters, a method is required that can re-construct the correct level of correlation between the two channels, according to the IC parameter. This is accomplished by a de-correlation method. There are a couple of methods available for creation of decorrelated signals. Ideally, a linear time invariant (LTI) function with all-pass frequency response is desired. One obvious method for achieving this is by using a constant delay. However, using a delay, or any other LTI all-pass functions, will result in non-all- pass response after adding the non-processed signal. In the case of a delay, the result will be a typical comb-filter. The comb-filter often gives an undesirable "me- tallic" sound that, even if the stereo widening effect can be efficient, reduces much naturalness of the original. Frequency domain methods for generating a de-correlated signal by adding a random sequence to the IID values along the fre- quency axis, where different sequences are used for the differ- ent audio channels, are also known from prior art. One problem with frequency domain decorrelation by the random sequence modifications is the introduction of pre-echoes. Subjective tests have shown that for non-stationary signals, pre-echoes are by far more annoying than post-echoes, which is also well supported by established psycho acoustical principles. This problem could be reduced by dynamically adapting transform sizes to the signal characteristics in terms of transient con- tent. However, switching transform sizes is always a hard (i.e., binary) decision that affects the full signal bandwidth and that can be difficult to accomplish in a robust manner. United States patent application publication US 2003/0219130 Al discloses a coherence-based audio coding and synthesis. In par- ticular, an auditory scene is synthesized from a mono audio signal by modifying, for each critical band, an auditory scene parameter such as an inter-aural level difference (ILD) and/or an inter-aural time difference (ITD) for each subband within the critical band, where the modification is based on an aver- age estimated coherence for the critical band. The coherence- based modification produces auditory scenes having object widths, which more accurately match the widths of the objects in the original input auditory scene. Stereo parameters are the well-known BCC parameters, wherein BCC stands for binaural cue coding. When generating two different decorrelated output chan- nels, frequency coefficients as obtained by a discrete Fourier- transform are grouped together in a single critical band. Based on the inter-channel coherence measure, weighting factors are multiplied by a pseudo-random sequence which is preferably chosen such that the variance is approximately constant for all critical bands, and the average is "0" within each critical band. The same sequence is applied to the spectral coefficients of each different frame. Other examples of prior art are given in the patent applications WO 91 20167 A and WO 03 007656 A. SUMMARY OF THE INVENTION It is the object of the present invention to provide a decoding concept for parametrically encoded multi-channel signals or an encoding concept for generating such signals which result in a good audio quality and a good coding efficiency. This object is achieved by an apparatus for generating a decorrelation signal in accordance with claim 1, a multi-channel de-coder in accordance with claim 13, a method of generating a decorrelation signal in accordance with claim 20, a method of multi-channel decoding in accordance with claim 21, an apparatus for encoding a stereo signal in accordance with claim 22 or a method of encoding a stereo signal in accordance with claim 26 or a computer program in accordance with claim 27. The present invention is based on the finding that, on the decoding side, a good decorrelation signal for generating a first and a second channel of a multi- channel signal based on the input mono signal is obtained, when a reverberation filter is used, which introduces an integer or preferably a fractional delay into the input signal. Importantly, this reverberation filter is not applied to the whole input signal. Instead, several reverberation filters are applied to several subbands of the original input signal, i.e., the mono signal so that the reverberation filtering using the reverberation filters is not applied in a time domain or in the frequency domain, i.e., in the domain which is reached, when a Fourier transform is ap- plied. Inventively, the reverberation filtering using rever- beration filters for the subbands is individually performed in the subband domain. A subband signal includes a sequence of at least two subband samples, the sequence of the subband samples representing a bandwidth of the subband signal, which is smaller than the bandwidth of the input signal. Naturally, the frequency band- width of a subband signal is higher than a frequency bandwidth attributed to a frequency coefficient obtained by Fourier transform. The subband signals are preferably generated by means of a filterbank having for example 32 or 64 filterbank channels, while an FFT would have, for the same example, 1.024 or 2.048 frequency coefficients, i.e., frequency channels. The subband signals can be subband signals obtained by subband- filtering a block of samples of the input signal. Alterna- tively, the subband filterbank can also be applied continuously without a block wise processing. For the present invention, however, block wise processing is preferred. Since the reverberation filtering is not applied to the whole signal, but is applied subband-wise, a "metallic" sound caused by comb-filtering is avoided. In cases, in which a sample period between two subsequent sub- band . samples of the subband is too large for a good sound im- pression at the, decoder end, it is preferred to use fractional delays in a reverberation filter such as a delay between 0.1 and 0.9 and preferably 0.2 to 0.8 of the sampling period of the subband signal. It is noted that in case of critical sampling, and when 64 subband signals are generated using a filterbank having 64 filterbank channels, the sampling period in a subband signal is 64 times larger than the sampling period of the original input signal. It is to be noted here that the delays are an integral part of the filtering process used in the reverberation device. The output signal constitutes of a multitude of delayed versions of the input signal. It is preferred to delay signals by fractions of the subband sampling period, in order to achieve a good re- verberation device in the subband domain. In preferred embodiments of the present invention, the delay, and preferably the fractional delay introduced by each rever- beration filter in each subband is equal for all subbands. Nev- ertheless, the filter coefficients are different for each sub- bands. It is preferred to use IIR filters. Depending on the ac- tual situation, fractional delay and the filter coefficients for the different filters can be determined empirically using listening tests. The subbands filtered by the set of reverberation filters con- stitute a decorrelation signal which is to be mixed with the original input signal, i.e., the mono signal to obtain a de- coded left channel and decoded right channel. This mixing of a decorrelation signal with the original signal is performed based on an inter-channel coherence parameter transmitted to- gether with the parametrically encoded signal. To obtain dif- ferent left and right channels, i.e., different first and sec- ond channels, mixing of the decorrelation signal with a mono signal to obtain the first output channel is different from mixing the decorrelation signal with the mono signal to obtain the second output channel.. To obtain higher efficiency on the encoding side, multi-channel encoding is performed using an adaptive determination of the stereo parameter set. To this end, an encoder includes, in ad- dition to a means for calculating the mono signal and in addi- tion to a means for generating a stereo parameter set, a means for determining a validity of stereo parameter sets for subse- quent portions of the left an right channels. Preferably, the means for determining is operative to activate the means for generating, when it is determined that the stereo parameter set is not valid anymore so that a second stereo parameter set is calculated for portions of the left and right channels starting at a second time border. This second time border is also deter- mined by the means for determining a validity. The encoded output signal then includes the mono signal, a first stereo parameter set and a first time border associated with the first parameter set and the second stereo parameter set and the second time border associated with the second ste- reo parameter set. On the decoding side, the decoder will use a valid stereo parameter set until a new time border is reached. When this new time border is reached, the decoding operations are performed using the new stereo parameter set. Compared to prior art methods, which did a block wise process- ing and, therefore, a block wise determination of stereo pa- rameter sets, the inventive adaptive determination of stereo parameter sets for different encoder-side determined time bor- ders provides a high coding efficiency on the one hand end and a high coding quality on the other hand. This is due to the fact that for relatively stationary signals, the same stereo parameter set can be used for many blocks of the samples of the mono signal without introducing audible errors. On the other hand, when non-stationary signals are concerned, the inventive adaptive stereo parameter determination provides an improved time resolution so that each signal portion has its optimum stereo parameter set. The present invention provides a solution to the prior art problems by using a reverberation unit as a de-correlator im- plemented with fractional delay lines in a filterbank, and us- ing adaptive level adjustment of the de-correlated reverberated signal. Subsequently, several aspects of the present invention are out- lined. One aspect of the invention is a method for delaying a signal by: filtering a real-valued time domain signal through the analysis part of complex filterbank; modifying the complex- valued subband signals obtained from the filtering; and fil- tering the modified complex-valued subband signals through the synthesis part of the filterbank; and taking the real part of the complex-valued time domain output signal, where the output signal is the sum of the signals obtained from the synthesis filtering. Another aspect of the invention is a method for modifying the complex valued subband signals by filtering each complex-valued subband signal with a complex valued finite impulse response filter where the finite impulse response filter for subband number n is given by a discrete time Fourier transform of the form where the parame- ter T=T/L, and where the synthesis filter bank has L subbands and the desired delay is T measured in output signal sample units. Another aspect of the invention is a method for modifying the complex valued subband signals by filtering where the filter Gr(ω) approximately satisfies where is the discrete time Fourier transform of the sequence and p(l)is the prototype filter of said complex filterbank and A is an appropriate real normalization factor. Another aspect of the invention is a method for modifying the complex valued subband signals by filtering where the filter satisfies such that even indexed impulse response samples are real valued and odd indexed impulse re- sponse samples are purely imaginary valued. Another aspect of the invention is a method for coding of ste- reo properties of an input signal, by at an encoder, calculate time grid parameters describing the location in time for each stereo parameter set, where the number of stereo parameter sets are arbitrary, and at a decoder, applying parametric stereo synthesis according to that time grid. Another aspect of the invention is a method for coding of ste- reo properties of an input signal, where the time localisation for the first stereo parameter set is, in the case of where a time cue for the stereo parameter set coincides with the begin- ning of a frame, signalled explicitly instead of signalling the time pointer. Another aspect of the invention is a method for generation of stereo decorrelation for parametric stereo reconstruction, by at a decoder, applying an artificial reverberation process to synthesise the side signal. Another aspect of the invention is a method for generation of stereo decorrelation for parametric stereo reconstruction by, at the decoder, the reverberation process is made within a com- plex modulated filterbank using phase delay adjustment in each filter bank channel. Another aspect of the invention is a method for generation of stereo decorrelation for parametric stereo reconstruction by, at the decoder, the reverberation process utilises a detector designed for finding signals where the reverberation tail could be unwanted and let the reverberation tail be attenuated or re- moved. BRIEF DESCRIPTION OF THE ACCOMPANYlNG DRAWINGS The present invention will now be described by way of illus- trative examples, not limiting the scope or spirit of the in- vention, with reference to the accompanying drawings, in which: Fig. 1 illustrates a block diagram of the inventive appara- tus; Fig. 2 illustrates a block diagram of the means for generat- ing a de-correlated signal; Fig. 3 illustrates the analysis of a single channel and the synthesis of the stereo channel pair based on the re- constructed stereo subband-signals according to the present invention; Fig. 4 illustrates a block diagram of the division of the parametric stereo parameters sets into time segments, based on the signal characteristic; and Fig. 5 illustrates an example of the division of the para- metric stereo parameters sets into time segments, based on the signal characteristic. DESCRIPTION OF PREFERRED EMBODIMENTS The below-described embodiments are merely illustrative for the principles of the present invention for parametric stereo coding. It is understood that modifications and variations of the arrangements and the details described herein will be ap- parent to others skilled in the art. It is the intent, there- fore, to be limited only by the scope of the impending patent claims and not by the specific details presented by way of de- scription and explanation of the embodiments herein. Delaying a signal by a fraction of a sample can be achieved by several prior art interpolation methods. However, special cases arises when the original signal is available as oversampled complex valued samples. Performing fractional delay in the qmf bank by only applying phase delay by a factor for, each qmf channel corresponding to a constant time delay, results in se- vere artefacts. This can efficiently be avoided by using a compensation filter according to a novel approach allowing high quality approxima- tions to arbitrary delays in any complex-exponential-modulated filterbank. A detailed description follows below. A continuous time model For ease of computations a complex exponential modulated L- band filterbank will be modelled here by a continuous time win- dowed transform using the synthesis waveforms where n,kare integers with > 0 and θ is a fixed phase term. Re- sults for discrete-time signals are obtain by suitable sampling of the t-variable with spacing 1/L . It is assumed that the real valued window v(t) is chosen such that for real valued signals x(t) it holds to very high precision that (3) where * denotes complex conjugation. It is also assumed that v(t)is essentially band limited to the frequency interval Consider the modification of each frequency band n by filtering the discrete time analysis samples cn(k) with a filter with impulse response Then the modified synthesis can be computed in the frequency domain to be (6) where denotes Fourier transforms of f(t) and Here, is the discrete time Fourier trans- form of the filter applied in frequency band n for n > 0 and Observe here that the special case Hn (ω) =1 leads to H(ω)=1 in (7) due to the special design of the window v(t) . Another case of interest is which gives , so that The proposed solution In order to achieve a delay of size r, such that y(t) = x(t-r), the problem is to design filters Hn(ω) for n > 0 such that (9) where H(ω) is given by (7) and (8) . The particular solution proposed here is to apply the filters (10) Here implies consistency with (8) for all n. Insertion of (10) into the right hand side of (7) yields (11) where with . Elementary computations show that is the discrete time Fourier transform of Very good approximations to the perfect delay can be obtained by solving the linear system in the least squares sense with a FIR filter In terms of filter coefficients, the equation (13) can be written where for K = 0and for k=0. In the case of a discrete time L-band filter bank with proto- type filter p(k), the obtained delay in sample units is Lr and the computation (12) is replaced by where T is the integer closest to Lr . Here p(k) is extended by zeros outside its support. For a finite length prototype fil- ter, only finitely many vr(k) are different from zero, and (14) is system of linear equations. The number of unknowns gr{k) is typically chosen to be a small number. For good QMF filter bank designs, 3-4 taps already give very good delay performance. Moreover, the dependence of the filter taps on the delay parameter r can often be modelled successfully by low order polynomials. Signalling adaptive time grid for stereo parameters Parametric stereo systems always leads to compromises in terms of limited time or frequency resolution in order to minimise conveyed data. It is however well known from psycho-acoustics that some spatial cues can be more important than others, which leads to the possibility to discard the less important cues. Hence, the time resolution does not have to be constant. Great gain in bitrate could be achieved by letting the time grid syn- chronise with the spatial cues. It can easily be done by send- ing. a variable number of parameter sets for each data frame that corresponds to a time segment of fixed size. In order to synchronise the parameter sets with corresponding spatial cues, additional time grid data describing the location in time for each parameter set has to be sent. The resolution of those time pointers could be chosen to be quite low to keep the total amount of data minimised. A special case where a time cue for a parameter set coincides with the beginning of a frame could be signalled explicitly to avoid sending that time pointer. Fig. 4 illustrates an inventive apparatus for performing pa- rameter analysis for time segments having variable and signal dependant time borders. The inventive apparatus includes means 401 for dividing the input signal into one or several time seg- ments. The time borders that separate the time segments are provided by means 402. Means 402 uses a detector specially de- signed for extracting spatial cues that is relevant for decid- ing where to set the time borders. Means 401 outputs all the input signal divided into one or several time segments. This output is input to means 403 for separate parameter analysis for each time segment. Means 403 outputs one parameter set per time segment' being analysed. Fig. 5 illustrates an example of how the time grid generator can perform for a hypothetical input signal. In this example one parameter set per data frame is used if no other time bor- der information is present. Hence, when no other time border information is present, the inherent time borders of the data frame is used. The in Fig. 5 depicted time borders are the out- put from means 402 in Fig. 4. The in Fig. 5 depicted time seg- ments are provided by means 401 in Fig. 4. The apparatus for encoding a stereo signal to obtain a mono output signal and the stereo parameter set includes the means for calculating the mono signal by combining a left and a right channel of the stereo signals by weighted addition. Addition- ally, a means 403 are generating a first stereo parameter set using a portion of the left channel and a portion of the right channel, the portions starting at a first time border is con- nected to the means for determining the validity of the first stereo parameter set for subsequent portions of the left chan- nel and the right channel. The means for determining is collectively formed by the means 402 and 401 in Fig. 1. Particularly, the means for determining is operative to gener- ate a second time border and to activate the means for generat- ing, when it is determined that this first stereo parameter set is not valid anymore so that a second stereo parameter set for portions of the left and right channels starting at the second time border is generated. Not shown in Fig. 4 are means for outputting the mono signal, the first stereo parameter set and the first time border asso- ciated with the first stereo parameter set and the second ste- reo parameter set and the second time border associated with the second stereo parameter set as the parametrically encoded stereo signal. The means for determining a validity of a stereo parameter set can include a transient detector, since the prob- ability is high that, after a transient, a new stereo parameter has to be generated, since a signal has changed its shape sig- nificantly. Alternatively, the means for determining a validity can include an analysis-by-synthesis device, which is adapted for decoding the mono signal and the stereo parameter set to obtain a decoded left and a decoded right channel, to compare the decoded left channel and the decoded right channel to the left channel and to the right channel, and to activate the means for generating, when the decoded left channel and the de- coded right channel are different from the left channel and the right channel by more than the predetermined threshold. Data frame 1: The time segment corresponding to parameter set 1 starts at the beginning of data frame 1 since no other time border information is present in this data frame. Data frame 2: Two time borders are present in this data frame. The time segment corresponding to parameter set 2 starts at the first time border in this data frame. The time segment corre- sponding to parameter set 3 starts at the second time border in this data frame. Data frame 3: One time border is present in this data frame. The time segment corresponding to parameter set 4 starts at the time border in this data frame. Data frame 4: One time border is present in this data frame. Thi.s time border coincides with the start border of the data frame 4 and does not have to be signalled since this is handled by the default case. Hence, this time border signal can be re- moved. The time segment corresponding to parameter set 5 starts at the beginning of data frame 4, even without signalling this time border. Using artificial reverberation as decorrelation method for pa- rametric stereo reconstruction One vital part of making the stereo synthesis in a parametric stereo system is to decrease the coherence between the left and right channel in order to create wideness of the stereo image. It can be done by adding a filtered version of the original mono signal to the side signal, where the side and mono signal is defined by: mono = (left + right) / 2, and side = (left - right) / 2, respectively. In order to not change the timbre too much, the filter in ques- tion should preferably be of all-pass character. One successful approach is to use similar all-pass filters used for artificial reverberation processes. Artificial reverberation algorithms usually requires high time resolution to give an impulse re- sponse that is satisfactory diffuse in time. There are great advantages in basing an artificial reverberation algorithm on a complex filter bank such as the complex qmf bank. The filter- bank provides excellent possibilities to let the reverberation properties be frequency selective in terms of for example re- verberation equalisation, decay time, density and timbre. How- ever, the filter bank implementations usually exchanges time resolution for higher frequency resolution which normally makes it hard to implement a reverberation process that is smooth enough in time. To deal with this problem a novel method would be to use a fractional delay approximation by only applying phase delay by a factor for, each qmf channel corresponding to a constant time delay. This primitive fractional delay method introduces severe time smearing that fortunately is very much desired in this case. The time smearing contributes to the time diffusion which is highly desirable for reverberation algo- rithms and gets bigger as the phase delay approaches pi/2 or - pi/2. Artificial reverberation processes are for natural reasons processes with an infinite impulse response, and offers natural exponential decays. In [PCT/SE02/01372] it is pointed out that if a reverberation unit is used for generating a stereo signal, the reverberation decay might sometimes be unwanted after the very end of a sound. These unwanted reverb-tails can however easily be attenuated or completely removed by just altering the gain of the reverb signal. A detector designed for finding sound endings can be used for that purpose. If the reverbera- tion unit generates artefacts at some specific signals e.g., transients, a detector for those signals can also be used for attenuating the same. Fig. 1 illustrates an inventive apparatus for the de- correlation method of signals as used in a parametric stereo system. The inventive apparatus includes means 101 for provid- ing a plurality of subband signals. The providing means can be a complex QMF filterbank, where every signal is associated with a subband index. The subband signals output by the means 101 from Fig 1. are in- put into a means 102 for providing a de-correlated signal 102, and into a means 103 and 106 for modifying the subband signal. The output from 102 is input into a means 104 and 105 for modi- fying the of the signal, and the output of 103, 104, 105 and 106 are input into a means for adding, 107 and 108, the subband signals. In the presently described embodiment of the invention, the means for modifying 103,104, 105 and 106, the subband signals, adjusts the level of the de-correlated signal and the un- processed signal being the output of 101, by multiplying the subband signal with a gain factor, so that every sum of every pair results in a signal with the amount of de-correlated sig- nal given by the control parameters. It should be noted that the gain factors used in the means for modifying, 103 - 106, are not limited to a positive value. It can also be a negative value. The output from the means for adding subband signals 107 and 108, is input to the means for providing a time-domain signal 109 and 110. The output from 109 corresponds to the left chan- nel of the re-constructed stereo signal, and the output from 110 corresponds to the right channel of the re-constructed ste- reo signal. In the here described embodiment the same de- correlator is used for both output channels, while the means for adding the de-correlated signal with the un-processed sig- nal are separate for the two output channels. The presently de- scribed embodiment thereby ensures that the two output signals can be identical as well as completely de-correlated, dependent on the control data provided to the means for adjusting the levels of the signals, and the control data provided to the means for adding the signals. In Fig. 2 a block diagram of the means for providing a de- correlated signal is displayed. The input subband signal is in- put to the means for filtering a subband signal 201. In the presently described embodiment of the present invention the filtering step is a reverberation unit incorporating all-pass filtering. The filter coefficients used are given by the means for providing filter coefficients 202. The subband index of the currently processed subband signal is input to 202. In one em- bodiment of the present invention different filter coefficients are calculated based on the subband index provided to 202. The filtering step in 201, relies on delayed samples of the input subband signal as well as delayed samples of intermediate sig- nals in the filtering procedure. It is an essential feature of the present invention that means for providing integer subband sample delay and fractional sub- band sample delay are provided by 203. The output of 201 is in- put to a means for adjusting the level of the subband signal 204, and also to a means for estimating signal characteristics of the subband signal 205. In a preferred embodiment of the present invention the characteristics estimated is the tran- sient behaviour of the subband signal. In this embodiment a de- tected transient is signalled to the means for adjusting the level of a subband signal 204, so that the level of the signal is reduced during transient passages. The output from 204 is the de-correlated signal input to 104 and 105 of Fig. 1. In Fig. 3 the single analysis filterbank and the two synthesis filterbanks are shown. The analysis filterbank 301,. operates on the mono input signal, while the synthesis filterbanks 302 and 303 operate on the re-constructed stereo signals. Fig. 1, therefore, shows the inventive apparatus for generating a decorrelation signal which is indicated by reference 102. As it is shown in Fig. 1 or 3, this apparatus includes means for providing a plurality of subband signals, wherein a subband signal includes the sequence of at least two subband samples, the sequence of the subband samples representing a bandwidth of the subband signal which is smaller than a bandwidth of the in- put signal. Each subband signal is input into the means 201 for filtering. Each means 201 for filtering includes a reverbera- tion filter so that a plurality of reverberated subband signals are obtained, wherein the plurality of reverberated subband signals together represent the decorrelation signal. Prefera- bly, as it is shown in Fig. 2, there can be a subband-wise postprocessing of reverberated subband signals which is per- formed by block 204, which is controlled by block 205. Each reverberation filter is set to a certain delay, and pref- erably a fractional delay, and each reverberation filter has several filter coefficients, which, as it is shown in Fig. 2, "depend on the subband index. This means that it is preferred to use the same delay for each subband but to use different sets of filter coefficients for the different subbands. This is sym- bolized by means 203 and 202 in Fig. 2, although it is to be mentioned here that delays and filter coefficients are prefera- bly fixedly determined when shipping a decorrelation device, wherein the delays and filter coefficients may be determined empirically using listening tests etc. A multi-channel decoder is shown by Fig. 1 and includes the in- ventive apparatus for generating the correlation signal, which is termed 102 in Fig. 1. The multi-channel decoder shown in Fig. 1 is for decoding a mono signal and an associated inter- channel coherence measure, the inter-channel coherence measure representing a coherence between a plurality of original chan- nels, wherein the mono signal is derived from the plurality of original channels. Block 102 in Fig. 1 constitutes a generator for generating a decorrelation signal for the mono signal. Blocks 103, 104, 105, 106 and 107 and 108 constitute a mixer for mixing the mono signal and the decorrelation signal in accordance with the first mixing mode to obtain a first decoded output signal and in accordance with the second mixing mode to obtain a second decoded output signal, wherein the mixer is op- erative to determine the first mixing mode and the second mix- ing mode based on the inter-channel coherence measure transmit- ted as a side information to the mono signal. The mixer is preferably operative to mix in a subband domain based on separate inter-channel coherence measures for differ- ent subbands. In this case, the multi-channel decoder further comprises means 109 and 110 for converting the first and second decoded output signals from the subband domain in a time domain to obtain a first decoded output signal and. a second decoded output signal in the time domain. Therefore, the inventive means 102 for generating a decorrelation signal and the inven- tive multi-channel decoder as shown in Fig. 1 operate in the subband domain and perform, as the very last step, a subband domain to time domain conversion. Depending on the actual situation, the inventive device can be implemented in hardware or in software or in a firmware includ- ing hardware constituents and software constituents. When im- plemented in software partially or fully, the invention also is a computer program having a computer-readable code for carrying out the inventive methods when running on a computer. We Claim: 1. Apparatus for encoding a stereo signal to obtain a mono output signal and a stereo parameter set, comprising: means for calculating the mono signal by combining a left and a right channel of the stereo signals; means (403) for generating a first stereo parameter set using a portion of the left channel and a portion of the right channel, the portion starting at a first time border; means (401,402) for determining a validity of the first stereo parameter set for subsequent portions of the left channel and the right channel, wherein the means for determining is operative to: generate second time border, and activate the means for generating, when it is determined that the stereo parameter set is not valid anymore so that a second stereo parameter set for portions of the left and right signals starting at the second time border is generated; and means for outputting the mono signal and the first stereo parameter set and the first time border associated with the first parameter set, and the second stereo parameter set and the second time border associated with the second stereo parameter set. 2. Apparatus as claimed in claim 1, wherein the means for generating is operative to calculate, as the stereo parameter set, an inter channel time difference parameter, an inter-channel level difference parameter, and/or an inter-channel coherence parameter. 3. Apparatus as claimed in claim 1 or 2, with the means for determining includes the transient detector, which is arranged for activating the means for generating, when a transient is detected, and to generate a time instant of the transient as the second time border. 4. Apparatus as claimed in any of claims 1 to 3, in which the means for determining is an analysis-by-synthesis device, which is adapted for: decoding the mono signal and the stereo parameter set to obtain a decoded left channel and a decoded right channel; comparing the decoded left channel and the decoded right channel to the left channel and the right channel; and activating the means for generating, when the decoded left channel and the decoded right channel are different from the left channel and the right channel by more than a predetermined threshold. ABSTRACT TITLE: "An Apparatus for encoding a stereo signal to obtain a mono output signal and a stereo parameter set" The invention relates to an apparatus for encoding a stereo signal to obtain a mono output signal and a stereo parameter set, comprising means for calculating the mono signal by combining a left and a right channel of the stereo signals; means (403) for generating a first stereo parameter set using a portion of the left channel and a portion of the right channel, the portion starting at a first time border; means (401,402) for determining a validity of the first stereo parameter set for subsequent portions of the left channel and the right channel, wherein the means for determining is operative to generate second time border, and activate the means for generating, when it is determined that the stereo parameter set is not valid anymore so that a second stereo parameter set for portions of the left and right signals starting at the second time border is generated; and means for outputting the mono signal and the first stereo parameter set and the first time border associated with the first parameter set, and the second stereo parameter set and the second time border associated with the second stereo parameter set.

Full Text

ADVANCED PROCESSING BASED ON A COMPLEX-EXPONENTIAL-MODULATED
FILTERBANK AND ADAPTIVE TIME SIGNALLING METHODS
TECHNICAL FIELD
The present invention relates to audio source coding systems
but the same methods could also be applied in many other tech-
nical fields. Different techniques that are useful for audio
coding systems using parametric representations of stereo prop-
erties are introduced.
EACKGROUND OF THE INVENTION AND PRIOR ART
The present invention relates to parametric coding of the ste-
reo image of an audio signal. Typical parameters used for de-
scribing stereo image properties are inter-channel intensity
difference (IID), inter-channel time difference (ITD), and in-
ter-channel coherence (IC). In order to re-construct the stereo
image based on these parameters, a method is required that can
re-construct the correct level of correlation between the two
channels, according to the IC parameter. This is accomplished
by a de-correlation method.
There are a couple of methods available for creation of decor-
related signals. Ideally,, a linear time invariant (LTI) func-
tion with all-pass frequency response is desired. One obvious
method for achieving this is by using a constant delay. How-
ever, using a delay, or any other LTI all-pass functions, will
result in non-all-pass response after adding the non-processed signal. In the case of a delay, the result will be a typical
comb-filter. The comb-filter often gives an undesirable "me-

CROSS REFERENCE
The instant Indian Patent Application constitutes a divisional application divided
out from Indian Patent Application no. 2162/KOLNP/05 of 28.10.2005 which is
now been granted of Indian Patent No. 242206.
FIELD OF THE INVENTION
The present invention relates to audio source coding systems but the same
methods could also be applied in many other technical fields. Different
techniques that are useful for audio coding systems using parametric
representations of stereo properties are introduced.
BACKGROUND OF THE INVENTION
The present invention relates to parametric coding of the stereo image of an
audio signal. Typical parameters used for describing stereo image properties are
inter-channel intensity difference (IID), inter-channel time difference (ITD), and
inter-channel coherence (IC). In order to re-construct the stereo image based on
these parameters, a method is required that can re-construct the correct level of
correlation between the two channels, according to the IC parameter. This is
accomplished by a de-correlation method.

There are a couple of methods available for creation of decorrelated signals.
Ideally, a linear time invariant (LTI) function with all-pass frequency response is
desired. One obvious method for achieving this is by using a constant delay.
However, using a delay, or any other LTI all-pass functions, will result in non-all-
pass response after adding the non-processed signal. In the case of a delay, the
result will be a typical comb-filter. The comb-filter often gives an undesirable
"me-

tallic" sound that, even if the stereo widening effect can be
efficient, reduces much naturalness of the original.
Frequency domain methods for generating a de-correlated signal
by adding a random sequence to the IID values along the fre-
quency axis, where different sequences are used for the differ-
ent audio channels, are also known from prior art. One problem
with frequency domain decorrelation by the random sequence
modifications is the introduction of pre-echoes. Subjective
tests have shown that for non-stationary signals, pre-echoes
are by far more annoying than post-echoes, which is also well
supported by established psycho acoustical principles. This
problem could be reduced by dynamically adapting transform
sizes to the signal characteristics in terms of transient con-
tent. However, switching transform sizes is always a hard
(i.e., binary) decision that affects the full signal bandwidth
and that can be difficult to accomplish in a robust manner.
United States patent application publication US 2003/0219130 Al
discloses a coherence-based audio coding and synthesis. In par-
ticular, an auditory scene is synthesized from a mono audio
signal by modifying, for each critical band, an auditory scene
parameter such as an inter-aural level difference (ILD) and/or
an inter-aural time difference (ITD) for each subband within
the critical band, where the modification is based on an aver-
age estimated coherence for the critical band. The coherence-
based modification produces auditory scenes having object
widths, which more accurately match the widths of the objects
in the original input auditory scene. Stereo parameters are the
well-known BCC parameters, wherein BCC stands for binaural cue
coding. When generating two different decorrelated output chan-
nels, frequency coefficients as obtained by a discrete Fourier-
transform are grouped together in a single critical band. Based
on the inter-channel coherence measure, weighting factors are

multiplied by a pseudo-random sequence which is preferably chosen such that
the variance is approximately constant for all critical bands, and the average is
"0" within each critical band. The same sequence is applied to the spectral
coefficients of each different frame.
Other examples of prior art are given in the patent applications WO 91 20167 A
and WO 03 007656 A.
SUMMARY OF THE INVENTION
It is the object of the present invention to provide a decoding concept for
parametrically encoded multi-channel signals or an encoding concept for
generating such signals which result in a good audio quality and a good coding
efficiency.
This object is achieved by an apparatus for generating a decorrelation signal in
accordance with claim 1, a multi-channel de-coder in accordance with claim 13, a
method of generating a decorrelation signal in accordance with claim 20, a
method of multi-channel decoding in accordance with claim 21, an apparatus for
encoding a stereo signal in accordance with claim 22 or a method of encoding a
stereo signal in accordance with claim 26 or a computer program in accordance
with claim 27.

The present invention is based on the finding that, on the decoding side, a good
decorrelation signal for generating a first and a second channel of a multi-
channel signal based on the input mono signal is obtained, when a reverberation
filter is used, which introduces an integer or preferably a fractional delay into the
input signal. Importantly, this reverberation filter is not applied to the whole
input signal. Instead, several reverberation filters are applied to several
subbands of the original input signal, i.e., the mono signal so that the
reverberation filtering using the reverberation filters is not applied in a time
domain or in the frequency domain, i.e., in

the domain which is reached, when a Fourier transform is ap- plied. Inventively, the reverberation filtering using rever-
beration filters for the subbands is individually performed in
the subband domain.
A subband signal includes a sequence of at least two subband
samples, the sequence of the subband samples representing a
bandwidth of the subband signal, which is smaller than the
bandwidth of the input signal. Naturally, the frequency band-
width of a subband signal is higher than a frequency bandwidth
attributed to a frequency coefficient obtained by Fourier
transform. The subband signals are preferably generated by
means of a filterbank having for example 32 or 64 filterbank
channels, while an FFT would have, for the same example, 1.024
or 2.048 frequency coefficients, i.e., frequency channels.
The subband signals can be subband signals obtained by subband-
filtering a block of samples of the input signal. Alterna-
tively, the subband filterbank can also be applied continuously
without a block wise processing. For the present invention,
however, block wise processing is preferred.
Since the reverberation filtering is not applied to the whole
signal, but is applied subband-wise, a "metallic" sound caused
by comb-filtering is avoided.
In cases, in which a sample period between two subsequent sub-
band . samples of the subband is too large for a good sound im-
pression at the, decoder end, it is preferred to use fractional
delays in a reverberation filter such as a delay between 0.1
and 0.9 and preferably 0.2 to 0.8 of the sampling period of the
subband signal. It is noted that in case of critical sampling,
and when 64 subband signals are generated using a filterbank
having 64 filterbank channels, the sampling period in a subband

signal is 64 times larger than the sampling period of the original input signal.
It is to be noted here that the delays are an integral part of
the filtering process used in the reverberation device. The
output signal constitutes of a multitude of delayed versions of
the input signal. It is preferred to delay signals by fractions
of the subband sampling period, in order to achieve a good re-
verberation device in the subband domain.
In preferred embodiments of the present invention, the delay,
and preferably the fractional delay introduced by each rever-
beration filter in each subband is equal for all subbands. Nev-
ertheless, the filter coefficients are different for each sub-
bands. It is preferred to use IIR filters. Depending on the ac-
tual situation, fractional delay and the filter coefficients
for the different filters can be determined empirically using
listening tests.
The subbands filtered by the set of reverberation filters con-
stitute a decorrelation signal which is to be mixed with the
original input signal, i.e., the mono signal to obtain a de-
coded left channel and decoded right channel. This mixing of a
decorrelation signal with the original signal is performed
based on an inter-channel coherence parameter transmitted to-
gether with the parametrically encoded signal. To obtain dif-
ferent left and right channels, i.e., different first and sec-
ond channels, mixing of the decorrelation signal with a mono
signal to obtain the first output channel is different from
mixing the decorrelation signal with the mono signal to obtain
the second output channel..
To obtain higher efficiency on the encoding side, multi-channel
encoding is performed using an adaptive determination of the

stereo parameter set. To this end, an encoder includes, in ad-
dition to a means for calculating the mono signal and in addi-
tion to a means for generating a stereo parameter set, a means
for determining a validity of stereo parameter sets for subse-
quent portions of the left an right channels. Preferably, the
means for determining is operative to activate the means for
generating, when it is determined that the stereo parameter set
is not valid anymore so that a second stereo parameter set is
calculated for portions of the left and right channels starting
at a second time border. This second time border is also deter-
mined by the means for determining a validity.
The encoded output signal then includes the mono signal, a
first stereo parameter set and a first time border associated
with the first parameter set and the second stereo parameter
set and the second time border associated with the second ste-
reo parameter set. On the decoding side, the decoder will use a
valid stereo parameter set until a new time border is reached.
When this new time border is reached, the decoding operations
are performed using the new stereo parameter set.
Compared to prior art methods, which did a block wise process-
ing and, therefore, a block wise determination of stereo pa-
rameter sets, the inventive adaptive determination of stereo
parameter sets for different encoder-side determined time bor-
ders provides a high coding efficiency on the one hand end and
a high coding quality on the other hand. This is due to the
fact that for relatively stationary signals, the same stereo
parameter set can be used for many blocks of the samples of the
mono signal without introducing audible errors. On the other
hand, when non-stationary signals are concerned, the inventive
adaptive stereo parameter determination provides an improved
time resolution so that each signal portion has its optimum
stereo parameter set.

The present invention provides a solution to the prior art
problems by using a reverberation unit as a de-correlator im-
plemented with fractional delay lines in a filterbank, and us-
ing adaptive level adjustment of the de-correlated reverberated
signal.
Subsequently, several aspects of the present invention are out-
lined.
One aspect of the invention is a method for delaying a signal
by: filtering a real-valued time domain signal through the
analysis part of complex filterbank; modifying the complex-
valued subband signals obtained from the filtering; and fil-
tering the modified complex-valued subband signals through the
synthesis part of the filterbank; and taking the real part of
the complex-valued time domain output signal, where the output
signal is the sum of the signals obtained from the synthesis filtering.
Another aspect of the invention is a method for modifying the
complex valued subband signals by filtering each complex-valued
subband signal with a complex valued finite impulse response
filter where the finite impulse response filter for subband
number n is given by a discrete time Fourier transform of the
form where the parame-
ter T=T/L, and where the synthesis filter bank has L subbands
and the desired delay is T measured in output signal sample
units.
Another aspect of the invention is a method for modifying the
complex valued subband signals by filtering where the filter
Gr(ω) approximately satisfies where

is the discrete time Fourier transform of the sequence
and p(l)is the prototype filter of said
complex filterbank and A is an appropriate real normalization
factor.
Another aspect of the invention is a method for modifying the
complex valued subband signals by filtering where the filter
satisfies such that even indexed impulse
response samples are real valued and odd indexed impulse re-
sponse samples are purely imaginary valued.
Another aspect of the invention is a method for coding of ste-
reo properties of an input signal, by at an encoder, calculate
time grid parameters describing the location in time for each
stereo parameter set, where the number of stereo parameter sets
are arbitrary, and at a decoder, applying parametric stereo
synthesis according to that time grid.
Another aspect of the invention is a method for coding of ste-
reo properties of an input signal, where the time localisation
for the first stereo parameter set is, in the case of where a
time cue for the stereo parameter set coincides with the begin-
ning of a frame, signalled explicitly instead of signalling the
time pointer.
Another aspect of the invention is a method for generation of
stereo decorrelation for parametric stereo reconstruction, by
at a decoder, applying an artificial reverberation process to
synthesise the side signal.
Another aspect of the invention is a method for generation of
stereo decorrelation for parametric stereo reconstruction by,
at the decoder, the reverberation process is made within a com-

plex modulated filterbank using phase delay adjustment in each filter bank channel.
Another aspect of the invention is a method for generation of
stereo decorrelation for parametric stereo reconstruction by,
at the decoder, the reverberation process utilises a detector
designed for finding signals where the reverberation tail could
be unwanted and let the reverberation tail be attenuated or re-
moved.

BRIEF DESCRIPTION OF THE ACCOMPANYlNG DRAWINGS
The present invention will now be described by way of illus-
trative examples, not limiting the scope or spirit of the in-
vention, with reference to the accompanying drawings, in
which:
Fig. 1 illustrates a block diagram of the inventive appara-
tus;
Fig. 2 illustrates a block diagram of the means for generat-
ing a de-correlated signal;
Fig. 3 illustrates the analysis of a single channel and the
synthesis of the stereo channel pair based on the re-
constructed stereo subband-signals according to the
present invention;
Fig. 4 illustrates a block diagram of the division of the
parametric stereo parameters sets into time segments,
based on the signal characteristic; and

Fig. 5 illustrates an example of the division of the para-
metric stereo parameters sets into time segments,
based on the signal characteristic.
DESCRIPTION OF PREFERRED EMBODIMENTS
The below-described embodiments are merely illustrative for
the principles of the present invention for parametric stereo
coding. It is understood that modifications and variations of
the arrangements and the details described herein will be ap-
parent to others skilled in the art. It is the intent, there-
fore, to be limited only by the scope of the impending patent
claims and not by the specific details presented by way of de-
scription and explanation of the embodiments herein.
Delaying a signal by a fraction of a sample can be achieved by
several prior art interpolation methods. However, special cases
arises when the original signal is available as oversampled
complex valued samples. Performing fractional delay in the qmf
bank by only applying phase delay by a factor for, each qmf
channel corresponding to a constant time delay, results in se-
vere artefacts.
This can efficiently be avoided by using a compensation filter
according to a novel approach allowing high quality approxima-
tions to arbitrary delays in any complex-exponential-modulated
filterbank. A detailed description follows below.
A continuous time model

For ease of computations a complex exponential modulated L-
band filterbank will be modelled here by a continuous time win-
dowed transform using the synthesis waveforms

where n,kare integers with > 0 and θ is a fixed phase term. Re-
sults for discrete-time signals are obtain by suitable sampling
of the t-variable with spacing 1/L . It is assumed that the real
valued window v(t) is chosen such that for real valued signals
x(t) it holds to very high precision that

(3)
where * denotes complex conjugation. It is also assumed that
v(t)is essentially band limited to the frequency interval
Consider the modification of each frequency band n by
filtering the discrete time analysis samples cn(k) with a filter
with impulse response

Then the modified synthesis

can be computed in the frequency domain to be

(6)
where denotes Fourier transforms of f(t) and

Here, is the discrete time Fourier trans-
form of the filter applied in frequency band n for n > 0 and

Observe here that the special case Hn (ω) =1 leads to H(ω)=1 in
(7) due to the special design of the window v(t) . Another case
of interest is which gives , so that

The proposed solution
In order to achieve a delay of size r, such that y(t) = x(t-r),
the problem is to design filters Hn(ω) for n > 0 such that

(9)
where H(ω) is given by (7) and (8) . The particular solution
proposed here is to apply the filters

(10)
Here implies consistency with (8) for all n.
Insertion of (10) into the right hand side of (7) yields

(11)
where with . Elementary
computations show that is the discrete time Fourier
transform of

Very good approximations to the perfect delay can be obtained
by solving the linear system

in the least squares sense with a FIR filter
In terms of filter coefficients, the
equation (13) can be written

where for K = 0and for k=0.
In the case of a discrete time L-band filter bank with proto-
type filter p(k), the obtained delay in sample units is Lr and
the computation (12) is replaced by

where T is the integer closest to Lr . Here p(k) is extended by
zeros outside its support. For a finite length prototype fil-
ter, only finitely many vr(k) are different from zero, and (14)
is system of linear equations. The number of unknowns gr{k) is
typically chosen to be a small number. For good QMF filter bank
designs, 3-4 taps already give very good delay performance.
Moreover, the dependence of the filter taps on the delay
parameter r can often be modelled successfully by low order
polynomials.
Signalling adaptive time grid for stereo parameters
Parametric stereo systems always leads to compromises in terms
of limited time or frequency resolution in order to minimise
conveyed data. It is however well known from psycho-acoustics
that some spatial cues can be more important than others, which
leads to the possibility to discard the less important cues.
Hence, the time resolution does not have to be constant. Great
gain in bitrate could be achieved by letting the time grid syn-

chronise with the spatial cues. It can easily be done by send-
ing. a variable number of parameter sets for each data frame
that corresponds to a time segment of fixed size. In order to
synchronise the parameter sets with corresponding spatial cues,
additional time grid data describing the location in time for
each parameter set has to be sent. The resolution of those time
pointers could be chosen to be quite low to keep the total
amount of data minimised. A special case where a time cue for a
parameter set coincides with the beginning of a frame could be
signalled explicitly to avoid sending that time pointer.
Fig. 4 illustrates an inventive apparatus for performing pa-
rameter analysis for time segments having variable and signal
dependant time borders. The inventive apparatus includes means
401 for dividing the input signal into one or several time seg-
ments. The time borders that separate the time segments are
provided by means 402. Means 402 uses a detector specially de-
signed for extracting spatial cues that is relevant for decid-
ing where to set the time borders. Means 401 outputs all the
input signal divided into one or several time segments. This
output is input to means 403 for separate parameter analysis
for each time segment. Means 403 outputs one parameter set per
time segment' being analysed.
Fig. 5 illustrates an example of how the time grid generator
can perform for a hypothetical input signal. In this example
one parameter set per data frame is used if no other time bor-
der information is present. Hence, when no other time border
information is present, the inherent time borders of the data
frame is used. The in Fig. 5 depicted time borders are the out-
put from means 402 in Fig. 4. The in Fig. 5 depicted time seg-
ments are provided by means 401 in Fig. 4.

The apparatus for encoding a stereo signal to obtain a mono
output signal and the stereo parameter set includes the means
for calculating the mono signal by combining a left and a right
channel of the stereo signals by weighted addition. Addition-
ally, a means 403 are generating a first stereo parameter set
using a portion of the left channel and a portion of the right
channel, the portions starting at a first time border is con-
nected to the means for determining the validity of the first
stereo parameter set for subsequent portions of the left chan-
nel and the right channel.
The means for determining is collectively formed by the means 402 and 401 in Fig. 1.
Particularly, the means for determining is operative to gener-
ate a second time border and to activate the means for generat-
ing, when it is determined that this first stereo parameter set
is not valid anymore so that a second stereo parameter set for
portions of the left and right channels starting at the second
time border is generated.
Not shown in Fig. 4 are means for outputting the mono signal,
the first stereo parameter set and the first time border asso-
ciated with the first stereo parameter set and the second ste-
reo parameter set and the second time border associated with
the second stereo parameter set as the parametrically encoded
stereo signal. The means for determining a validity of a stereo
parameter set can include a transient detector, since the prob-
ability is high that, after a transient, a new stereo parameter
has to be generated, since a signal has changed its shape sig-
nificantly. Alternatively, the means for determining a validity
can include an analysis-by-synthesis device, which is adapted
for decoding the mono signal and the stereo parameter set to
obtain a decoded left and a decoded right channel, to compare

the decoded left channel and the decoded right channel to the left channel and to the right channel, and to activate the
means for generating, when the decoded left channel and the de-
coded right channel are different from the left channel and the
right channel by more than the predetermined threshold.
Data frame 1: The time segment corresponding to parameter set 1
starts at the beginning of data frame 1 since no other time
border information is present in this data frame.
Data frame 2: Two time borders are present in this data frame.
The time segment corresponding to parameter set 2 starts at the
first time border in this data frame. The time segment corre-
sponding to parameter set 3 starts at the second time border in
this data frame.
Data frame 3: One time border is present in this data frame.
The time segment corresponding to parameter set 4 starts at the
time border in this data frame.
Data frame 4: One time border is present in this data frame.
Thi.s time border coincides with the start border of the data
frame 4 and does not have to be signalled since this is handled
by the default case. Hence, this time border signal can be re-
moved. The time segment corresponding to parameter set 5 starts
at the beginning of data frame 4, even without signalling this
time border.
Using artificial reverberation as decorrelation method for pa-
rametric stereo reconstruction
One vital part of making the stereo synthesis in a parametric
stereo system is to decrease the coherence between the left and

right channel in order to create wideness of the stereo image.
It can be done by adding a filtered version of the original
mono signal to the side signal, where the side and mono signal
is defined by:
mono = (left + right) / 2, and
side = (left - right) / 2, respectively.
In order to not change the timbre too much, the filter in ques-
tion should preferably be of all-pass character. One successful
approach is to use similar all-pass filters used for artificial
reverberation processes. Artificial reverberation algorithms
usually requires high time resolution to give an impulse re-
sponse that is satisfactory diffuse in time. There are great
advantages in basing an artificial reverberation algorithm on a
complex filter bank such as the complex qmf bank. The filter-
bank provides excellent possibilities to let the reverberation
properties be frequency selective in terms of for example re-
verberation equalisation, decay time, density and timbre. How-
ever, the filter bank implementations usually exchanges time
resolution for higher frequency resolution which normally makes
it hard to implement a reverberation process that is smooth
enough in time. To deal with this problem a novel method would
be to use a fractional delay approximation by only applying
phase delay by a factor for, each qmf channel corresponding to
a constant time delay. This primitive fractional delay method
introduces severe time smearing that fortunately is very much
desired in this case. The time smearing contributes to the time
diffusion which is highly desirable for reverberation algo-
rithms and gets bigger as the phase delay approaches pi/2 or -
pi/2.
Artificial reverberation processes are for natural reasons
processes with an infinite impulse response, and offers natural
exponential decays. In [PCT/SE02/01372] it is pointed out that

if a reverberation unit is used for generating a stereo signal, the reverberation decay might sometimes be unwanted after the
very end of a sound. These unwanted reverb-tails can however
easily be attenuated or completely removed by just altering the
gain of the reverb signal. A detector designed for finding
sound endings can be used for that purpose. If the reverbera-
tion unit generates artefacts at some specific signals e.g.,
transients, a detector for those signals can also be used for
attenuating the same.
Fig. 1 illustrates an inventive apparatus for the de-
correlation method of signals as used in a parametric stereo
system. The inventive apparatus includes means 101 for provid-
ing a plurality of subband signals. The providing means can be
a complex QMF filterbank, where every signal is associated with
a subband index.
The subband signals output by the means 101 from Fig 1. are in-
put into a means 102 for providing a de-correlated signal 102,
and into a means 103 and 106 for modifying the subband signal.
The output from 102 is input into a means 104 and 105 for modi-
fying the of the signal, and the output of 103, 104, 105 and
106 are input into a means for adding, 107 and 108, the subband
signals.
In the presently described embodiment of the invention, the
means for modifying 103,104, 105 and 106, the subband signals,
adjusts the level of the de-correlated signal and the un-
processed signal being the output of 101, by multiplying the
subband signal with a gain factor, so that every sum of every
pair results in a signal with the amount of de-correlated sig-
nal given by the control parameters. It should be noted that
the gain factors used in the means for modifying, 103 - 106,

are not limited to a positive value. It can also be a negative
value.
The output from the means for adding subband signals 107 and
108, is input to the means for providing a time-domain signal
109 and 110. The output from 109 corresponds to the left chan-
nel of the re-constructed stereo signal, and the output from
110 corresponds to the right channel of the re-constructed ste-
reo signal. In the here described embodiment the same de-
correlator is used for both output channels, while the means
for adding the de-correlated signal with the un-processed sig-
nal are separate for the two output channels. The presently de-
scribed embodiment thereby ensures that the two output signals
can be identical as well as completely de-correlated, dependent
on the control data provided to the means for adjusting the
levels of the signals, and the control data provided to the
means for adding the signals.
In Fig. 2 a block diagram of the means for providing a de-
correlated signal is displayed. The input subband signal is in-
put to the means for filtering a subband signal 201. In the
presently described embodiment of the present invention the
filtering step is a reverberation unit incorporating all-pass
filtering. The filter coefficients used are given by the means
for providing filter coefficients 202. The subband index of the
currently processed subband signal is input to 202. In one em-
bodiment of the present invention different filter coefficients
are calculated based on the subband index provided to 202. The
filtering step in 201, relies on delayed samples of the input
subband signal as well as delayed samples of intermediate sig-
nals in the filtering procedure.
It is an essential feature of the present invention that means
for providing integer subband sample delay and fractional sub-

band sample delay are provided by 203. The output of 201 is in-
put to a means for adjusting the level of the subband signal
204, and also to a means for estimating signal characteristics
of the subband signal 205. In a preferred embodiment of the
present invention the characteristics estimated is the tran-
sient behaviour of the subband signal. In this embodiment a de-
tected transient is signalled to the means for adjusting the
level of a subband signal 204, so that the level of the signal
is reduced during transient passages. The output from 204 is
the de-correlated signal input to 104 and 105 of Fig. 1.
In Fig. 3 the single analysis filterbank and the two synthesis
filterbanks are shown. The analysis filterbank 301,. operates on
the mono input signal, while the synthesis filterbanks 302 and
303 operate on the re-constructed stereo signals.
Fig. 1, therefore, shows the inventive apparatus for generating
a decorrelation signal which is indicated by reference 102. As
it is shown in Fig. 1 or 3, this apparatus includes means for
providing a plurality of subband signals, wherein a subband
signal includes the sequence of at least two subband samples,
the sequence of the subband samples representing a bandwidth of
the subband signal which is smaller than a bandwidth of the in-
put signal. Each subband signal is input into the means 201 for
filtering. Each means 201 for filtering includes a reverbera-
tion filter so that a plurality of reverberated subband signals
are obtained, wherein the plurality of reverberated subband
signals together represent the decorrelation signal. Prefera-
bly, as it is shown in Fig. 2, there can be a subband-wise
postprocessing of reverberated subband signals which is per-
formed by block 204, which is controlled by block 205.
Each reverberation filter is set to a certain delay, and pref-
erably a fractional delay, and each reverberation filter has

several filter coefficients, which, as it is shown in Fig. 2,
"depend on the subband index. This means that it is preferred to
use the same delay for each subband but to use different sets
of filter coefficients for the different subbands. This is sym-
bolized by means 203 and 202 in Fig. 2, although it is to be
mentioned here that delays and filter coefficients are prefera-
bly fixedly determined when shipping a decorrelation device,
wherein the delays and filter coefficients may be determined
empirically using listening tests etc.
A multi-channel decoder is shown by Fig. 1 and includes the in-
ventive apparatus for generating the correlation signal, which
is termed 102 in Fig. 1. The multi-channel decoder shown in
Fig. 1 is for decoding a mono signal and an associated inter-
channel coherence measure, the inter-channel coherence measure
representing a coherence between a plurality of original chan-
nels, wherein the mono signal is derived from the plurality of
original channels. Block 102 in Fig. 1 constitutes a generator
for generating a decorrelation signal for the mono signal.
Blocks 103, 104, 105, 106 and 107 and 108 constitute a mixer
for mixing the mono signal and the decorrelation signal in
accordance with the first mixing mode to obtain a first decoded
output signal and in accordance with the second mixing mode to
obtain a second decoded output signal, wherein the mixer is op-
erative to determine the first mixing mode and the second mix-
ing mode based on the inter-channel coherence measure transmit-
ted as a side information to the mono signal.
The mixer is preferably operative to mix in a subband domain
based on separate inter-channel coherence measures for differ-
ent subbands. In this case, the multi-channel decoder further
comprises means 109 and 110 for converting the first and second
decoded output signals from the subband domain in a time domain
to obtain a first decoded output signal and. a second decoded

output signal in the time domain. Therefore, the inventive means 102 for generating a decorrelation signal and the inven-
tive multi-channel decoder as shown in Fig. 1 operate in the
subband domain and perform, as the very last step, a subband
domain to time domain conversion.
Depending on the actual situation, the inventive device can be
implemented in hardware or in software or in a firmware includ-
ing hardware constituents and software constituents. When im-
plemented in software partially or fully, the invention also is
a computer program having a computer-readable code for carrying
out the inventive methods when running on a computer.

We Claim:
1. Apparatus for encoding a stereo signal to obtain a mono output signal and
a stereo parameter set, comprising:
means for calculating the mono signal by combining a left and a right channel
of the stereo signals;
means (403) for generating a first stereo parameter set using a portion of the
left channel and a portion of the right channel, the portion starting at a first
time border;
means (401,402) for determining a validity of the first stereo parameter set
for subsequent portions of the left channel and the right channel, wherein the
means for determining is operative to:
generate second time border, and
activate the means for generating, when it is determined that the stereo
parameter set is not valid anymore so that a second stereo parameter set for
portions of the left and right signals starting at the second time border is
generated; and
means for outputting the mono signal and the first stereo parameter set and
the first time border associated with the first parameter set, and the second
stereo parameter set and the second time border associated with the second
stereo parameter set.

2. Apparatus as claimed in claim 1, wherein the means for generating is
operative to calculate, as the stereo parameter set, an inter channel time
difference parameter, an inter-channel level difference parameter, and/or
an inter-channel coherence parameter.
3. Apparatus as claimed in claim 1 or 2, with the means for determining
includes the transient detector, which is arranged for activating the means
for generating, when a transient is detected, and to generate a time
instant of the transient as the second time border.
4. Apparatus as claimed in any of claims 1 to 3, in which the means for
determining is an analysis-by-synthesis device, which is adapted for:
decoding the mono signal and the stereo parameter set to obtain a
decoded left channel and a decoded right channel;
comparing the decoded left channel and the decoded right channel to the
left channel and the right channel; and

activating the means for generating, when the decoded left channel and
the decoded right channel are different from the left channel and the right
channel by more than a predetermined threshold.

ABSTRACT

TITLE: "An Apparatus for encoding a stereo signal to obtain a mono output
signal and a stereo parameter set"
The invention relates to an apparatus for encoding a stereo signal to obtain a
mono output signal and a stereo parameter set, comprising means for calculating
the mono signal by combining a left and a right channel of the stereo signals;
means (403) for generating a first stereo parameter set using a portion of the
left channel and a portion of the right channel, the portion starting at a first time
border; means (401,402) for determining a validity of the first stereo parameter
set for subsequent portions of the left channel and the right channel, wherein
the means for determining is operative to generate second time border, and
activate the means for generating, when it is determined that the stereo
parameter set is not valid anymore so that a second stereo parameter set for
portions of the left and right signals starting at the second time border is
generated; and means for outputting the mono signal and the first stereo
parameter set and the first time border associated with the first parameter set,
and the second stereo parameter set and the second time border associated with
the second stereo parameter set.

AN APPARATUS FOR ENCODING A STEREO SIGNAL TO OBTAIN A MONO OUTPUT SIGNAL AND A STEREO PARAMETER SET

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