MX2008009527A - Efficient filtering with a complex modulated filterbank - Google Patents

Abstract

A filter apparatus for filtering a time domain input signal to obtain a time domain output signal, which is a representation of the time domain input signal filtered using a filter characteristic having an non-uniform amplitude/frequency characteristic, comprises a complex analysis filter bank (101) for generating a plurality of complex subband signals from the time domain input signals, a plurality of intermediate filters, wherein at least one of the intermediate filters (102) of the plurality of the intermediate filters has a non-uniform amplitude/frequency characteristic, wherein the plurality of intermediate filters have a shorter impulse response compared to an impulse response of a filter having the filter characteristic, and wherein the non-uniform amplitude/frequency characteristics of the plurality of intermediate filters together represent the non-uniform filter characteristic, and a complex synthesis filter bank (103) for synthesizing the output of the intermediate filters to obtain the time domain output signal.

Description

EFFICIENT FILTRATION WITH A COMPLEX MODULATED FILTER BANK TECHNICAL FIELD The present invention. refers to a filter apparatus and a method for filtering a time domain input signal, a filter generator and a method for generating an intermediate filter definition signal, especially for the field of coding, decoding, manipulation and filtering of audio signals, eg in the field of HRTF (head-related transfer function). BACKGROUND OF THE INVENTION It has been shown in [P. Ekstrand, "Bandwidth extension of audio sign by spectral band replication" ("Bandwidth Extension of Audio Signals Through Spectral Band Replication"), Proc. Isthe IEEE Benelux orkshop in Model based Processing and Coding of Audio (MPCA-2002), pp. 53-58, Leuven, Belgium, 2002], that a complex-exponential modulated filter bank is an excellent tool for spectral adjustment of the envelope of audio signals. An application of this feature is audio coding based on Spectral Band Replication (SBR). Other successful applications of a bank of complex filters include selective frequency and spatialization approach for parametric stereo, see [E. Schuijers, J. Breebart, H. Purnhagen, J. Engdegard: "Low complexity parametric stereo coding" ("Low complexity parametric stereo coding"), Proc. 116th AES convention, 2004, document 6073] and parametric multichannel coding, see [J. Herré et al .: "The reference model architecture for MPEG spatial audio coding" ("The reference model architecture for MPEG spatial audio coding"), Proc. 118th AES Convention, 2005, document 6447]. In those applications the frequency resolution of the complex filter bank is further improved at low frequencies by means of sub-subband filtering. The hybrid hybrid filter bank thus achieves a frequency resolution that allows the processing of spatial indications at a spectral resolution that closely follows the spectral resolution of the binaural auditory system. However, in some applications, the resolution of the filter bank is still insufficient, in the sense that the simple gain modifications in each subband are not enough to really model the action of a given filter. For multichannel audio binaural reproduction by filtering related to HRTF (head-related transfer function), the intricate phase characteristics of the filters are important for the perceived audio quality. Of course it is possible to apply fast convolution methods based on the DFT (Discrete Fourier Transform) as a post-process to the multichannel reproduction, but if the reproduction device already contains the signals in the subband domain of the modulated filter bank exponential complex, there are significant advantages in terms of computational complexity and algorithmic integration to perform filtering derived from HRTF in the subband domain, which will be delineated in more detail later. Since the HRTF 's are different for each individual and the derived filters depend on the virtual source and / or listener positions that can, for example, be changed by control signals, user interfaces or by other signals of description, it is also It is important to be able to efficiently convert a given HRTF-related filter into subband domain filters. It is therefore the aim of the present invention to provide a filter apparatus for filtering a time domain input signal, a method for filtering a time domain input signal, a filter generator or a method for providing a signal definition of intermediate filter, allowing a more efficient or more flexible manipulation of a time domain input signal with better quality. This objective is achieved by a filter apparatus of According to claim 1, by a method for filtering a time domain input signal according to claim 41, a filter generator according to claim 25, a method for providing an intermediate filter definition according to the claim 42, a system according to claim 40 or a computer program according to claim 43. SUMMARY OF THE INVENTION One embodiment of the present invention relates to a filter apparatus for filtering a digital domain input signal. time to obtain a time domain output signal, which is a representation of the filtered time domain input signal using a filter characteristic having a non-uniform amplitude / frequency characteristic comprising a complex analysis filter bank to generate a plurality of complex subband signals of the time domain input signal, a plural intermediate filter, wherein an intermediate filter is provided for each complex subband signal, wherein at least one of the intermediate filters of the plurality of intermediate filters has a non-uniform amplitude / frequency characteristic, wherein the plurality of filters intermediates has a shorter response to the impulse compared to a response to the impulse of a filter having the filter characteristic, and wherein the the non-uniform amplitude / frequency characteristic of the plurality of intermediate filters together represent the non-uniform filter characteristic, and a complex synthesis filter bank to synthesize the output of the intermediate filters to obtain the time domain output signal. As a second aspect, a further embodiment of the present invention is a filter generator for providing an intermediate filter definition signal comprising a complex modulated filter bank for filtering an impulse response signal indicative of an amplitude filter characteristic. / frequency in a time domain to obtain a plurality of complex valued subband signals as the intermediate filter definition signal, wherein each complex valued subband signal of the complex modulated filter bank corresponds to an impulse response for an intermediate filter for a subband signal, wherein at least one of the complex valued subband signals comprises at least two different non-dissipated values, and wherein each complex valued subband signal is shorter than the impulse response signal. The embodiments of the first aspect of the present invention are based on the discovery that a more efficient and / or more flexible filtering (or manipulation) of a signal Time domain input can be achieved in the subband domain, which is sometimes also referred to as QMF domain (quadrature mirror filter), with better quality compared to other handling schemes. The gain with respect to efficiency, especially computational efficiency, is a consequence of the shorter impulse responses of the intermediate filters compared to the impulse response of a filter that has the non-uniform filter characteristic in the time domain and the fact that the subband signals can be processed independently of one another. Due to the shorter impulse responses, one mode of a filter apparatus can process each of the complex subband signals emitted by the complex analysis filter bank individually. Therefore, the filtering can be carried out in parallel, which accelerates the processing of the time domain input signal dramatically compared to the manipulation of the time domain input signal directly due to the responses to the impulse shorter. The modalities according to the first aspect of the present invention are especially favorable when balancing computational efficiency on the one hand and quality on the other side. Although a direct processing of the domain input signal can be achieved time in the time domain by a convolution with the impulse response of a filter that has the characteristic of non-uniform amplitude / frequency, which usually leads to a very good quality, the convolution requires a high computational effort due to the length of the response to the impulse of the filter in the time domain. On the other hand, the transformation of an audio signal in the frequency domain when performing a Fourier Transform represents the tremendous disadvantage that other manipulations, which are necessary in modern acoustic systems, can not efficiently perform in the Fourier domain with a high quality Therefore, by employing a plurality of intermediate filters, each having a shorter pulse response compared to a pulse response of a filter having the filter characteristic of a corresponding filter in the time domain, of which at less one has an impulse response with at least two values without dissipation represents a highly favorable compromise between computational efficiency on the one hand and quality on the other side. As a consequence, the embodiments of the inventive filter apparatuses represent an excellent compromise between a direct processing of the time domain input signal, for example, by means of the convolution of the time domain input signal with the response to the longest impulse indicative of the non-uniform filter characteristic, which leads to an enormous computational effort, and employs a Fourier transform, which leads to more problems in the further course of signal processing. The advantages of the embodiments of the first aspect of the present invention are developed especially in the context of FIR filters (response to the final impulse), since each of the intermediate filters of the plurality of intermediate filters has a significantly shorter impulse response. compared to the impulse response of the FIR filter in the time domain. Therefore, by processing in parallel the different subband signals emitted by the complex analysis filter bank, the computational efficiency can be drastically improved. This aspect is especially important in the field of filters that have long impulse responses. A field of application, in which filters with very long impulse responses occur frequently, are applications related to HRTF (HRTF = head-related transfer function), such as, for example, sub-mixing multiple-channel audio signals for feedback to hearing aids, other talking systems related to the head or stereo sound systems. In many concrete applications the efficiency computational is further increased, since the audio signals are already present in the subband (complex) or QMF domain. Therefore, in many concrete implementations, the complex analysis filter bank and the complex synthesis filter bank for generating the plurality of complex subband signals of the time domain input signal and for synthesizing the output signal of the Time domain is already present. With respect to the second aspect, the embodiments of the present invention are based on the discovery that a more flexible and more efficient filtering of the time domain input signal with a better quality can be achieved by providing an intermediate filter definition signal , which can, for example, be provided in a filter apparatus according to the first aspect to define its intermediate filters. A significant advantage of the embodiments according to the second aspect of the present invention is that an intermediate filter definition signal for a set of intermediate filters is obtained by providing an inventive filter generator mode with a signal defining the filter, such as an impulse response signal indicative of an amplitude / frequency filter characteristic of a time domain filter or other filter definition signals. Therefore, one mode of a generator The filter provides a filter definition signal for a set of intermediate filters for the same filtering effectively as a filter in the time domain defined by the filter definition signal with virtually no aliasing effects. As a consequence, the embodiments of an inventive filter generator allow virtually free performance of aliases of an arbitrary filter in the subband domain. By using an inventive filter generator mode, arbitrary filter characteristics can be transferred from the time domain to the subband signal domain, such as virtually free aliasing, low-pass filter characteristics, high-pass filter characteristics, filter characteristics bandpass, band rejection filter characteristics, resonance filter characteristics, notch filter characteristics or more complex filter characteristics. Among the more complex filter characteristics, it is important to mention a combination of several characteristics as well as filter characteristics related to HRTF. Especially in the context of HRTF-related applications in the field of multichannel audio systems and other high-quality applications, it is important to note that the inventive filter generator modalities allow a real action modeling to be modeled. filter given in the time domain in the subband domain. The virtually free aliasing performance, which is especially important in applications related to HRTF, is made possible since the phase characteristics of a filter in the time domain are (almost) perfectly transferred to the subband domain. The examples illustrating this will be delineated in the additional course of the present application. Among the advantages of the modalities of the second aspect of the present invention is especially the significant gain with respect to the computational efficiency that can be achieved. The complex modulated filter banks of the inventive filter generator embodiments produce a plurality of complex valued subband signals as the intermediate filter definition signal, wherein each complex valued subband signal is shorter than the impulse response signal. indicative of the amplitude / frequency filter characteristic in the time domain. The filter generator, therefore, produces an intermediate filter definition signal comprising the output of the complex modulated filter bank with its plurality of complex valued subband short signals, which not only allows fast, efficient and parallel computing with respect to the filtering of a time domain input signal to obtain a time domain output signal in the framework of a filter apparatus mode, but also allows a fast, efficient and parallel computation of the intermediate filter definition signal by itself. Compared to a direct application of the impulse response signal indicative of the amplitude / frequency filter characteristic in the time domain by convolving the impulse response signal with the time domain input signal, the application of a The embodiment of an inventive filter generator according to the second aspect of the present invention allows a simplified, faster and more efficient computation, which leads to a result audibly indistinguishable in comparison with the more complex convolution method. In addition, one embodiment of the inventive filter generator also offers the advantage of significantly improved flexibility with respect to the possible filter characteristics applied in the subband domain. Since the arbitrary filter characteristics can be transferred from the time domain to the subband domain by a mode of an inventive filter generator, enormous flexibility is introduced into the processing and manipulation of the audio signal. For example, one embodiment of an inventive filter generator is capable of providing a filter definition signal intermediate that corresponds to an individually altered filter characteristic of a filter related to HRTF. In the field of HRTF this offers the opportunity to individually modify the HRTF filters according to the needs and auditory capacities of an individual. In addition, the position of the source as well as the position of the listener can be adapted with respect to each other and with respect to an environment (simulated or calculated) (e.g., a concert hall, an open space, or stadium). This offers the great advantage of providing a listener with great flexibility with respect to acoustic conditions. One mode of the inventive filter generator, therefore, provides the possibility of virtually changing from a stadium to a concert hall or an open field, without employing the need to transfer the audio signals between the time domain, the domain of subband and / or the frequency domain. By employing one embodiment of an inventive filter generator all these manipulations of the audio signal can be performed within the subband domain with a very high quality, which is perceptually indistinguishable from a signal processing in the time domain, but which offers a huge improvement in computational efficiency. This flexibility is not limited only to the change from one environment to another, e.g. change from a stadium to a concert hall and visa versa. One mode of a generator The inventive filter offers the possibility of altering the filter characteristics of the plurality of the intermediate filters in a quasi-continuous manner. An application in the field of HRTF is an application of a mode of the filter generator and / or filter apparatus in a tracking application, in which, for example, the position of the listener with respect to different audio sources varies in a quasi-continuous way. Possible applications include, for example, simulations and computer games with a very high quality. Another advantage of one embodiment of a filter generator is that the application of one mode of a filter generator is more efficient with respect to memory usage, since an impulse response signal provided to the complex modulated filter bank of the generator The filter is typically a real valued signal, while the intermediate filter definition signal is a complex valued signal of approximately the same total length. As a consequence, the storage of the impulse response signals compared to the intermediate filter definition signals (or the filter connections of the intermediate filters) saves memory, more or less, by an order of 2. Due to the possibility of a fast and efficient parallel computing, especially in the field of memory-sensitive applications comprising a larger space of parameter with respect to the possible impulse response signals, this represents a significant advantage. In one embodiment of an inventive filter generator the filter generator is provided with a filter definition signal, which may comprise for example the filter connections of a digital filter in the time domain or by a transfer function in the domain of frequency, which may comprise the amplitude / frequency characteristic and / or the phase / frequency characteristic of a filter. In these cases, one embodiment of the filter generator further comprises an impulse response signal generator, which provides the appropriate impulse response signal indicative of the resultant amplitude / frequency filter characteristic in the time domain to the filter bank Modulated complex filter generator. Therefore, the inclusion of an impulse response signal generator in some embodiments of an inventive filter generator offers even more flexibility with respect to providing the intermediate filter definition signal, since not only the impulse response signals in the form of discrete time signals can be provided to a filter generator mode but also the filter connections or the frequency domain description of a time domain filter can be transferred to the subband domain by a mode appropriate for a filter generator. BRIEF DESCRIPTION OF THE DRAWINGS The present invention will now be described by way of illustrative examples, which do not limit the scope or spirit of the invention, with reference to the accompanying drawings, in which: Figure illustrates the processing of an audio signal digital by filtering by subband in a system comprising a filter generator and a filter apparatus; Figure Ib illustrates a possible solution for a complex analysis bank; The Figure illustrates a possible solution for a bank of complex synthesis filters; Figure Id illustrates a possible additional solution for a complex synthesis filter bank; The Figure illustrates an interaction of a mode of a filter generator with a plurality of intermediate filters of a mode of a filter apparatus; Figure 2 illustrates the processing of a digital audio signal by means of filtering directly; Figure 3 illustrates a preferred embodiment of a system with a filter converter; Figure 4 illustrates a response given to the filter pulse; Figure 5 illustrates an impulse response obtained by complex gain adjustment of the subbands; Figure 6 illustrates the magnitude response of a given filter; Figure 7 illustrates the magnitude response of a filter obtained by the complex gain adjustment of the subbands; Figure 8 compares the performance of the present invention with the complex gain adjustment of the subbands; Figure 9 illustrates a preferred embodiment of a filter apparatus comprising an optional embodiment of a filter generator and additional components; Figure 10 illustrates a filter characteristic together with several frequency bands for different subbands; and Figure 11 illustrates a preferred embodiment of a filter generator. DESCRIPTION OF THE PREFERRED MODALITIES The embodiments described below are merely illustrative for the principles of the present invention of efficient filtration with a complex modulated filter bank. It should be understood that the modifications and variations of the facilities and the details described herein will be apparent to other experts in the field. matter. Therefore, it is the attempt to limit itself only by the scope of impending patent claims and not by the specific details presented by way of description and explanation of the modalities therein. Next, objects with the same functional or similar properties are denoted by the same reference signs. Unless explicitly noted otherwise, the description with respect to objects with similar or similar functional properties may be interchanged with each other. The Figure illustrates in the form of a system comprising embodiments of both a filter apparatus and a filter generator processing a digital audio signal by means of subband filtering in accordance with the present invention. This signal path, for example, may represent a part of a spatial audio playback system where the input is a received audio channel and the output is a component of a signal to be reproduced in the right ear. The input signal (Digital audio signal or time domain input signal) is analyzed by the complex analysis bank 101 by means of filtering with a set of L analysis filters followed by the subsampling of an L factor, where L is a positive integer, preferably larger than 1. Typically the L factor is a power of 2, preferably L = 64. The analysis filters are usually obtained by a complex modulation of a prototype filter p. { v), where v is a positive integer that indicates an index in a data set or an index of a value in a signal not subsampled by the factor L. The output of the filter bank consists of L signals of subband that are processed by a subband filtering 102. This subband filtering consists of a combination of manipulations such as subband gain adjustment according to received control data and application of finite response filters to the pulse applied separately in each subband. The filter connections of the subband filters are obtained from a filter converter (inventive) 104 as a mode of a filter generator that takes as input a filter described by filter connections directly, a description of the frequency domain or an impulse response (signal). Complex synthesis bank 103 reconstructs an output signal by means of up sampling by a factor L, filtering by L synthesis filters, summing all the results, and extracting the real part. The sum of all the results and the extraction of the real part can also be changed with respect to their order, as will be delineated more closely with respect to Figures 1 and Id. Figure Ib shows a complex analysis bank 101 in more detail. The complex analysis bank 101 comprises a plurality of L intermediate analysis filters 120 for each subband to be issued by the complex analysis bank 101. To be more precise, each of the L intermediate analysis filters 120 is connected in parallel to a node 130 to which the time domain input signal to be processed is provided. Each of the intermediate analysis filters 120 is adapted to filter the input signal from the complex analysis bank 101 with respect to a center frequency of each subband. According to the center frequencies of the different subbands, each subband is marked by a subband index or index n, where n is a non-negative integer, typically in the range of 0 to L-1. The intermediate analysis filters 120 of the complex analysis bank 101 can be derived from a prototype filter p (v) by a complex modulation according to the subband index n of the subband to which the intermediate analysis filter 120 is applied. below, more details that refer to the complex modulation of a prototype filter. Either directly by the intermediate analysis filters 120 or by an optional sub-sampler 140 (denoted by the dotted line in Figure Ib) the sampling frequency of the signal emitted by the intermediate analysis filter bank 120 is reduced by a factor L. As mentioned above, the sub-samplers 140 supplied to each subband signal issued by the corresponding intermediate analysis filters 120 are optional since, depending on the particular implementation, the sub-sampling can also be carried out in the filter framework. intermediate analyzes 120. In principle, the sub-sampling of the signal issued by the intermediate analysis filters 120 is not required. However, the presence of the explicit or implicit sub-samplers 140 is a preferred option since the amount of data provided by the bank Complex analysis 101 would be raised alternatively by a factor of L, which leads to significant redundancy of data. Figure 1c illustrates a possible solution for a complex synthesis bank 103. The complex synthesis bank 103 comprises L intermediate synthesis filters to which the L subband signals of the subband filtering 102 are provided. Depending on the particular implementation of the complex synthesis bank 103 prior to filtering in the framework of the intermediate synthesis filters 150, the subband signals are sampled upwardly by L rising sampler 160, which reconstructs the sampled frequency of the subband signals by increasing the frequency sampling by a factor of L. In other words, the ascending sampler 160 optionally reconstructs or reshapes the subband signals provided to the upward sampler 160 such that the information contained in each of the subband signals is retained while the sampling frequency is increased by a factor of L. However, as already explained in the context of Figure Ib, ascending samplers 160 are optional components, since ascending sampling can also be carried out in the framework of intermediate synthesis filters 150. Therefore, the ascending sampling stage of the subband signals. carried out by the ascending sampler 160 may be processed simultaneously in the framework of the intermediate synthesis filters 150. However, if the sub-samplers 190 are not implemented either explicitly or implicitly, the ascending samplers 160 do not have to be implemented explicitly or implicitly. . The intermediate synthesis filters 150 are connected through an output to an adder 170 which adds the filtered subband signals emitted by the intermediate synthesis filters 150. Adder 170 is further connected to a real part extractor 180, the which extracts or forms a real valued signal or in its place a time domain (real valued) output signal based on the complex valued signal provided by the adder 170. The real part extractor 180 can perform this task, example, by extracting the real part of a complex valued signal provided by the adder 170, when calculating the absolute value of the complex valued signal provided by the adder 170 or by another method that forms an actual valued output signal based on a signal Input valued complex. In the case of the system shown in Figure la, the signal emitted by the real part extractor 180 is the time domain output signal emitted by the inventive filter apparatus mode. The second possible solution for a complex synthesis bank 103 shown in Figure Id differs from the first possible solution shown in the Figure, which concerns only the real parts extractor 180 and the adder 170. To be more precise, the outputs of the intermediate synthesis filters 150 are connected separately from each subband to a real part extractor 180 which extracts or forms a real valued signal based on the complex valued signal emitted by the intermediate synthesis filters 150. The real part extractor 180 is then connects to the adder 170, which adds the L actual valued signals derived from the L subband signals filtered to form the actual valued output signal provided by the addend 170, which in the case of the system shown in Figure 1 is the signal of the time domain exit. Figure shows the subband filtering 102 and its interaction with the filter converter 104 in more detail. The subband filtering 102 comprises a plurality of intermediate filters 190, wherein an intermediate filter 190 is provided for each complex valved subband signal provided to the subband filtering 102. Therefore, subband filtering 102 comprises L intermediate filters 190 The filter converter 104 is connected to each of the intermediate filters 190. As a consequence, the filter converter 104 is capable of providing the filter connections for each of the intermediate filters 190 of the subband filtering 102. More details concerning the filtration made by the intermediate filters 190 will be explained in the additional course of the application. Therefore, the filter connections provided to the different intermediate filters 190 and emitted by the filter converter 104 form the intermediate filter definition signal. Furthermore, it should be noted that the modalities, solutions and implementations may comprise additional and / or optional delays to delay any of the signals or a subset of signals, which have been omitted in the Figure a for simplicity. Also in Figures 2 to 11 the optional delays have been issued for simplicity. However, delays or delays can be understood in the elements shown (e.g. filters) or added as optional elements in all modalities depending on their specific implementation. Figure 2 illustrates the processing of a digital audio signal by means of direct filtering 201. If the same filter is given as input to the filter converter 104 of Figure 1 and direct filtration 201, a design objective for the filter converter 104 is that the digital audio output of 103 must be perceptually (or aurally) indistinguishable from the digital audio output of the right filtering 201, if the digital audio inputs to the complex analysis bank 101 and the filtering direct 201 are identical and the processing in direct filtration 102 consists of pure fixed subband filtering. In the embodiment of the system shown in the Figures, the figure of the filter input to the filter converter 104 is given as a filter definition signal, which, for example, may comprise the filter connections of a time domain filter. corresponding, a description-of the frequency domain (amplitude / frequency characteristic and / or phase / frequency characteristic) or an impulse response signal of the appropriate filter. In the case of direct filtering 201 the same filter definition signal, in principle, can be used. Depending on the concrete implementation and the filter definition signal, filtering can be carried out by direct application of the filter connections in the framework of a digital filter, by a discrete Fourier transform together with a transfer function or other description of the frequency domain or by means of of convolution with the impulse response signal. Figure 3 illustrates a preferred embodiment of a filter converter 104 according to the present invention as a mode of a filter generator. It is assumed that the filter is given by its impulse response. Seeing this response to the impulse as a discrete time signal, it is analyzed by a bank (of filters) of complex analysis of L-band 301. The resulting subband signal outputs are then exactly the filter impulse responses to be applied separately in each subband in the subband filtering 102. In the preferred embodiment shown in Figure 3, the filter definition signal provided to the filter converter 104 and its complex analysis bank or complex analysis filter bank 301 is the signal of impulse response indicative of the amplitude / frequency characteristic of a filter, which is about to be transferred to the subband domain. Therefore, the output of the complex analysis bank (of filters) 301 from each of the L subbands represents the impulse response of the intermediate filters included in the subband filtering 102. The complex analysis bank 301, in principle, is derived from the analysis bank 101 but has a different prototype filter and a slightly different modulation structure, the details of which will be delineated in the following description. The same fast algorithms that are used for an implementation of the complex analysis bank 101 can be reused by the complex analysis bank 301, leading to a very fast and very efficient conversion process. In addition, the length of the prototype filter q (v) can be designed to be only a fraction of the length of the prototype filter p (v). Due to sub-sampling by a factor L, the length of subband filters is also a factor L smaller than the sum of the lengths of the given time domain filter and the prototype filter q (v). The computational effort is reduced in this way compared to direct filtering 201 by approximately one LIA factor. The compensation factor of 4 is due to the actual filtration replacement with complex filtration. Another compensation is the computational cost of complex analysis and synthesis banks 101 and 103. For efficient implementations this cost is comparable to the cost of FIR filters that are preferably short, and therefore insignificant, as outlined above. In addition, this Compensation for the reduction in computational cost does not exist for systems that already employ these two filter banks 101 and 103. Figure 4 illustrates an example of a given response to the filter 400 pulse. It consists of 192 (= 64-3) connections without zero. In other words, the response to pulse 400 shown in Figure 4 comprises 192 values without dissipation. In the present application, a non-dissipating key or value is a key or a value that is ideally not equal to zero. However, due to the implementation limits in the framework of this application a non-dissipating value or key is a real valued complex or valued key with an absolute value that is larger than a predetermined threshold, e.g. 10"s or 2" s, where s is a positive integer depending on the requirements of a specific implementation. In digital systems this threshold is preferably defined in the binary system (base 2), where the whole number s has a predetermined value depending on the implementation specifications. Typically, the value s is 4, 5, 6, 7, 8, 10, 12, 14, 16 or 32. The impulse response 400 of the system of Figure 1 is indistinguishable from this response to the given impulse in the resolution of the image, in a case where it applies a bank of band filters L = 64 with a prototype filter of length 640 (= 64-10) and a prototype filter of length 192 (= 64 · 3) is used for the filter converter 104 of Figure 3. The filters of corresponding intermediate sub-bands have only 5 (= 3 + 3-1) connections each, as will be explained later. Figure 5 illustrates the impulse response 410 of the system of Figure 1 with a bank of band filters 64, in a special case corresponding to a prior art use for envelope wrapping and equalization. In this case, the subband filters or preferably the intermediate filters 190 are all of one key only, so that a constant complex gain is applied to each subband. For each subband, the corresponding gain is chosen to be equal to the complex frequency response of the filter of Figure 4 evaluated at the center frequency of the particular subband. As can be seen from the result, there are severe pre-echo artifacts and there will be a significant perceptual difference between the application of this filter response compared to the response to the target pulse 400 of Figure 4. Figure 6 illustrates the magnitude response 420 of the filter of Figure 4. The frequency scale of Figure 6 is adjusted to the resolution of a 64-band filter bank (L = 64).

Figure 7 illustrates the magnitude response 430 of the filter underlying the impulse response 410 shown in Figure 5. As can be seen, the use of only one gain per subband results in a poor approximation to the desired frequency response. The main reason for this is the rapid variation of the target phase spectrum. In fact, this method of the prior art is better suited in the modeling of linear phase responses. Figure 8 finally compares the performance of an embodiment of the present invention and the prior art method of complex gain adjustment of the subbands. The dotted curve is a design of the target magnitude response 420 of Figure 6. The dashed curve 440 is the magnitude response of the difference between the complex frequency responses of the objective filter and its approach by the prior art method. The solid curve 450 is the magnitude response of the difference between the complex frequency responses of the objective filter and its approximation by the method taught by the present invention with the parameters as discussed during the description of Figure 4. As can be seen, the error of the prior art method is small only at the 64 midpoints of the filter bank sub-bands while the inventive method leads to an approach quality in the 50 dB range. It should be noted that this is also the level of performance that is measured when the output of the inventive system is compared with the output of the reference system for an arbitrary input signal. As the comparison of the two curves 440 and 450 in Figure 8 shows, one embodiment of an inventive filter apparatus i, one embodiment of a filter generator and one system comprising both modalities offers a significant advantage concerning the quality of the filter. the manipulation of an input signal. The significant difference concerning the filtering quality (or manipulation) of the input signal underlined above is a consequence of the fact that at least one of the intermediate filters 190 has an impulse response with two or more values without dissipation. In other words, at least one of the intermediate filters 190 comprises at least two filter connections without dissipation. In addition, it is important to note that the number of subbands L processed by one mode of a filter apparatus is larger or at least equal to 2. However, the number of subbands L is significantly smaller than the number of bands of frequency required for a comparable quality in the case of filtering based on the Fourier transform combined with a filter mainly described by an amplitude / frequency characteristic and / or a phase / frequency characteristic as the filter transfer function.

Due to the fact that the impulse response of the intermediate filters 190 is significantly shorter than the impulse response of the underlying filter property in the time domain, the computations with respect to each subband can be carried out significantly faster. In addition, since the different subband signals can be processed independently, both a filter apparatus mode as well as a filter generator 104 mode can process the respective input signals highly efficiently in a fast and parallel manner. Therefore, the processing of both a digital audio input and an input signal as well as an impulse response indicative of a filter characteristic can be carried out highly efficiently in a parallel manner. As outlined above, one embodiment of an inventive filter apparatus as well as one embodiment of an inventive filter generator combine the advantages of both direct processing of audio signals in the time domain leading to a very high quality as well as the use of a combination of a Fourier transform together with a transfer function in the frequency domain offering a high efficiency since each frequency band is only multiplied with a key (valued real or complex) in the filtering process of the signal.

On the other hand, the disadvantages of both, purely processing the input signals in the time domain, which leads to a huge computational effort, such as that of a Fourier transform, can be significantly reduced and suppressed at a level that the output of a modality of a filter apparatus is perceptually indistinguishable from the quality of a direct processing in the time domain. These two advantages offer greater flexibility to filter digital signals with variable filtering characteristics. This is especially important in the field of HRTF, since filters related to HRTF usually have a very long impulse response. Therefore, one embodiment of an inventive filter apparatus comprising a complex analysis filter bank 101, a plurality of intermediate filters 190 in subband filtering 102 and a complex synthesis filter bank 103 offers especially in the field of Applications related to HRTF significant computational advantages due to the possible parallel processing of subband signals. The modalities of a filter generator and system modalities comprising both a filter apparatus and a filter generator also offer the advantage that the filters can be easily adapted to specific environments, parameters or other specific needs of the filter. application by hand. Especially, in terms of HRTF-related applications, one embodiment of such a system can be used in tracking applications, in which various sources of sounds and noises as well as the position of the listener vary with time. Such a modality of a system comprising a filter apparatus and a filter generator therefore offers a highly efficient and flexible way of presenting an audio impression of a three-dimensional installation of sound sources with respect to a variable position and orientation of the sound. a hypothetical listener through hearing aids or other sound systems related to the head (stereo sound systems). As this last example illustrates, one embodiment of an inventive filter apparatus together with an inventive filter generator offers not only a highly efficient audio handling system with excellent quality but also a very flexible way of introducing audio impressions into alteration. in an efficient way. Banks of complex modulated filters Next, let be the Fourier transform of the discrete time of a discrete time signal z (v). As before, v is an integer that indicates an index or a time index of a time signal, while? = 2 n f is the associated circular frequency with the frequency f, n is the circular number (n = 3.1415926 ...) and i = j = V- is the imaginary unit. The filter bank of L - exponentially modulated, complex band is defined by a real valued prototype filter p (v) of finite length. For the computations below it will be assumed by extension with zeros that the prototype filter is defined for all integers v. Given a real valued discrete time signal x. { v) the bank of analysis filters 101 applies, as already explained, the complex modulated prototype filters followed by sub-sampling by an L-factor to emit the sub-band signals, \ c "(k) =? x (Y + kL) p (y) exp -i (n +) (y + T) (1) for each subband index n = 0,1, ..., Z, -1, and integer time index k. The time index k differs from the time index v with respect to the fact that k refers to the sub-sampled signals, while the integer v indicates the signals with the full sample frequency. Given the complex valued subband signals dn (k), the synthesis filter bank 103 applies filtering followed by up sampling by a factor of L and a real value extraction to output the actual valued signals, as already explained, to obtain the output signal .y (v /) = Re (2) In equations (1) and (2) T and? represent phase factors (constants) to filter the real valued discrete time signal x (v) to the complex valued subband signal and to reconstruct the real valued output samples and (v) complex valued subband signals dn (k) . It is well known that a prototype filter and fixed phase factors T and? they can be chosen to give the perfect reconstruction, ^ (v) = x (v), in the case where dn (k) = cn. { k), that is, when the subband signals are not altered. In practice, the perfect reconstruction property will be maintained until a delay (and / or a signal change), but in the computations that follow, this detail will be ignored by allowing the use of an acausal prototype filter. The present invention is applicable to the pseudo QMF type of the design as taught by PCT / SE02 / 00626"Aliasing reduction using complex exponential modulated filter banks". Here the prototype filter is symmetric p (-v) = p (v), and its Fourier transform of discrete time?. { ?) essentially dissipates outside the interval | ß > | = 7t /? .. The perfect reconstruction is also replaced by an almost perfect reconstruction property. For the derivation that follows it will be assumed for simplicity that both the perfect reconstruction is maintained like that? (?) = 0 for p / L . In addition, it is assumed that phase factors satisfy the condition that? -? is equal to a whole number multiple of AL. In a critically sampled filter bank, alteration of subband signals before synthesis usually leads to the introduction of aliasing artifacts. This is overcome here due to the fact that an oversampling by a factor two is introduced when using complex valued signals. Although the total sampling rate of the subband samples is identical to the sampling rate of the discrete time input signal, the input signal is valued in real and the subband samples are valued complex. As will be delineated below, the absence of aliases opens the door for time-invariant time-signal processing. Subband filtering in a complex modulated filter bank o Consider the subband filtering modification 102 of each subband signal obtained by filtering the analysis samples cn (k) from the complex analysis bank 101 with a filter with impulse response g " (k) before the synthesis (2) performed by the complex synthesis (filter) bank 103 The elementary computations show that given the assumptions in the frequency response of the prototype filter, the resulting effect on the reconstructed time signal is due to a discrete time filtering.

? (?) = G (a >) X (< a), (4) where Here, Gn (?) '^; gn (k) exp (-iko) is the Fourier transform of the discrete time of the filter applied in subband n for n = 0 and GB (fi>) = G _, _ n (- < ø) * for »< 0. (6) where * denotes complex conjugation. Note here that the special case G "(«) = l leads to G (ca) = l in (5) due to the special design assumed of the prototype p (v), which implies Another case of interest is Gn (?) = Exp (-z¿y) that leads to G (& >) = exp (-j'Z, < w), so that y. { v) = x (v - L).

Approximation of a filter response given by the subband filtering Leave that? { ?) is a given filter (e.g. transfer function) with response to the real valued impulse h (y). This data is considered as input to the filter converter 104. In view of (5) and (7), a trivial choice is given for the subband filters which results in the desired response G (co) = H (co) by Gs (< 3J) = H (ü &ft; f L): for + 1/2) I = p, (8) The disadvantage of this formula is that although ??) is a smooth function of? , the segment in periods of it defined by (8) will show jumps and the impulse response of the subband filters will be unnecessarily long. The use of the prior art of the complex pseudo QMF bank for equalization or wrapping adjustment consists of applying a single gain gn on each subband, which results in the transfer function with the extension gn - -g *, _ "for« < 0 defined according to (6). In view of (7), one achieves = gx, for «= 0.1 ¿-1 (10) and the transfer function is interpolated between those frequencies. For objective filter responses. { ?) that vary slowly as a function of frequency? , a first filter approach method is therefore obtained by choosing An example of the quality resulting from this procedure is given in Figures 5 and 7. In accordance with one embodiment of the present invention a filter generator or a filter converter 104 is used to teach how to convert the filter (defined by its response). pulse) h (v) in intermediate subband filters 190 by means of the second analysis filter bank 301 employing the actual prototype valued filter 8Ák) = (12) In terms of Fourier transforms this reads ? + 2p? ? + 2p? p Q (13) 1 = 0 The advantage of this procedure is that any given filter h (y) can be efficiently transformed into intermediate subband filter responses. If q (v) has KQ-L connections, a time domain filter h. { v) of connections KH-L becomes subband domain filters (12) with connections KH + KQ- \, where KH and KQ are positive integers. With respect to the exemplary numbers given in the context of the description in Figure 4, KH and KQ are equal to 3 and with a prototype filter length and an impulse response corresponding to a length of 3 64 = 192 (L = 64) each. Therefore, each intermediate subband filter 190 has an impulse response length of only 3 + 3 - 1 = 5 connections each. Design of the prototype filter for the filter converter The insert (13) in (5) produces 0 (?) = + (14) Therefore, the condition for keeping Gco) =? { ?) is that where £ > [/] = 1 for / = 0 and d [1] = 0 for l? 0. A simple solution to (15) is given by the brick wall filter This prototype filter corresponds to the choice (8) and has the disadvantage of having a slowly decaying and infinite response to the impulse q (v). Instead, the present invention teaches to solve (15) approximately (e.g. in the sense of least squares) with a finite response filter to the pulse q (v). The time domain equivalent of (15) is the system of linear equations for n = 0, l, ..., L- \ and for all integers k, ? p2 (n + vL-2kL) q (n + vL) = ^ -S [k] r (16) where P2 (v =? P (l) p (l + v) (17) / = - 8 is the autocorrelation of p (v). For any given support length, the system of linear equations (16) can be solved in the least squares direction for a prototype filter q. { v). It is desirable to use a support significantly shorter than that of the prototype filter of the original filter bank p (v), and in that case the system linear (16) is overdetermined. A given quality of approximation can also be exchanged for other desirable properties through union optimization. An example of such a property is a low pass type of frequency response Q (a >). The determination of a multi-slot QMF representation (subband domain) of the HRTF filters is described below. The time domain filter conversion in the complex QMF subband domain is performed by an FIR filter in the filter converter 104 of Figure la. To be more precise, the following description underlines a method for implementing a given FIR filter h (v) of length Nh in the complex QMF subband domain. The principle of the operation is illustrated in Figure la in the case of a system which also comprises an embodiment of an inventive filter apparatus. The subband filtering itself is carried out by a set of or a plurality of intermediate filters 190 within the subband filtering 102. To be more precise, the subband filtering consists of the separate application of an intermediate FIR filter valued complex gn (l) for each sub-band QMF with an index n -0.1, ..., 63. In other words, in the following description special reference will be made to the modalities with L = 64 different subband signals. However, this number of Specific subband signals are not essential and the appropriate equations will also be given in a more general form. One of the key components of the system shown in Figure la is the filter converter 104, which converts the given time domain FIR filter h (v) into the complex subband domain gn (l). The filter converter 104 comprises a complex analysis bank 301 similar to the analysis bank QMF 101. The prototype filter of the complex analysis filter bank 301 of the filter converter 104 q (v) of length 192 (= 3 · 64) for the specific case of L = 64 Subband signals are created by solving in the least squares direction the overdetermined system of equation (16). The filter coefficients q (v) or preferably the relationships they meet will be described in more detail for the case of L = 64 subband signals later. To be more exact in terms of mathematical description, an extension with zeros in the time domain FIR filter is defined by Intermediate subband domain filters results are based on equation (12) and can be expressed in the general case as (19) where lo and o are delays, 1 is an integer that indicates an index of the filter connections and Nq (= NQ) is the length of the impulse response of the prototype filter g (v). It should be noted, that in the framework of the present application under an equation. which is based on an equation, is understood as an introduction of additional delay factors (see 10 and o), additional coefficients and an introduction of a window function or another simple function. In the case L '= 64, the expression for the subband domain filters or intermediate filters 190 becomes These subdomain filters have a length Lq - Kh + 2, where and Nh is the length of the impulse response h (v) of the filter characteristics to be transferred to the subband domain. In this case, the integer n = 0, 1, 63 is again the index of a subband and 1 = 0, 1, (Kh + 1) is an integer that indicates the connections of the resulting intermediate filters 190. Extra addition of (-2) in equation (20) compared to equation (12) is there, because equation (12) is developed without considering the lowering of filters. Real implementations will always cause delays. Therefore, depending on the particular implementation, additional delays or delays may be implemented in the modes shown in Figures a to 1 and Figures 2 to 11, which have been omitted for simplicity in the aforementioned Figures. As outlined above, in many cases the system of linear equations (16) is overdetermined. However, it can be solved or approximated in the sense of least squares with respect to the prototype filter coefficients g (v). The resolution of the system of linear equations (16) in the direction of least squares, leads to the filter connections of the prototype filter qr (v) to meet the following relationships for integers v from 0 to 191: -0.204 < q [0] < -0.202 -0 .199 < q [l] < -0. 197 -0 .194 < q [2] < -0. 192 -0 .189 < q [3] < -0. 187 -0 .183 < q [4] < -0. 181 -0 .178 < q [5] < -0. 176 -0 .172 < q [6] < -0. 170 -0 .166 < q [7] < -0. 164 -0 .160 < q [8] < -0. 158 -0 .154 < q [9] < -0. 152 -0. 148 < q [10] < -0 .146 -0. 142 < q [H] < -0 .140 -0. 135 < q [12] < -0 .133 -0. 129 < q [13] < -0 .127 -0. 122 < q [14] < -0 .120 -0. 116 < q [15] < -0 .114 -0. 109 < q [16] < -0 .107 -0. 102 < q [17] < -0 .100 -0. 096 < q [18] < -0 .094 -0. 089 < q [19] < -0 .087 -0. 082 < q [20] < -0 .080 -0. 075 < q [21] < -0 .073 -0. 068 < q [22] < -0 .066 -0. 061 < q [23] < -0 .059 -0.054 < q [24] < - 0.052 -0. 046 < q [25] < - 0.044 -0. 039 < q [26] < - 0.037 -0. 032 < q [27] < - 0.030 -0. 024 < q [28] < - 0.022 -0. 017 < q [29] < - 0.015 -0. 009 < q [30] < - 0.007 -0 .002 < q [31] < 0.000 0. 006 < q [32] < 0,008 0. 014 < q [33] < 0 .016 0. 021 < q [34] < 0 .023 0. 029 < q [35] < 0 .031 0. 037 < q [36] < 0 .039 0. 045 < q [37] < 0 .047 0. 054 < q [38] < 0 .056 0. 062 < q [39] < 0 .064 0. 070 < q [40] < 0 .072 0. 079 < q [41] < 0 .081 0. 087 < q [42] < 0 .089 0. 096 < q [43] < 0 .098 0. 105 < q [44] < 0 .107 0.; q [128 < 0.292 0.278 < q [129 < 0.280 0.266 < q [130 < 0.268 0.254 < q [131 < 0.256 0.243 < q [132 < 0.245 0.232 < q [133 < 0.234 0.221 < q [134 < 0.223 0.210 < q [135 < 0.212 0.200 < q [136 < 0.202 0.190 < q [137 < 0.192 0.180 < q [138 < 0.182 0.170 < q [139 < 0.172 0.160 < q [140 < 0.162 0.150 < q [141 < 0.152 0.141 < q [142 < 0.143 0.132 < q [143 < 0.134 0.122 < q [144 < 0.124 0.113 < q [145 < 0.115 0.105 < q [146 < 0.107 0.096 < q [147 < 0.098 0.087 < q [148 < 0.089 0. 079 < q [149] < 0.081 0. 070 < q [150] < 0. 072 0. 062 < q [151] < 0. 064 0. 054 < q [152] < 0. 056 0. 045 < q [153] < 0. 047 0. 037 < q [154] < 0. 039 0. 029 < q [155] < 0. 031 0. 021 < q [156] < 0. 023 0. 014 < q [157] < 0. 016 0. 006 < q [158] < 0. 008 -0 .002 < q [159] < 0 000 -0. 009 < q [160] < -0 .007 -0. 017 < q [161] < -0 .015 -0. 024 < q [162] < -0 .022 -0. 032 < q [163] < -0 .030 -0. 039 < q [164] < -0 .037 -0. 046 < q [165] < -0 .044 -0. 054 < q [166] < -0 .052 -0. 061 < q [167] < -0 .059 -0. 068 < q [168] < -0 .066 -0. 075 < q [169] < -0 .073 -0. 082 < q [170] < -0 .080 -0. 089 < q [171] < -0 .087 -0. 096 < q [172] < -0 .094 -0. 102 < q [173] < -0 .100 -0.109 < q; i74] < -0.107 -0. 116 < 175] < -0. 114 -0. 122 < q [176] < -0. 120 -0. 129 < q; i77] < -0. 127 -0. 135 < q 178] < -0. 133 -0. 142 < q; i79] < -0. 140 -0. 148 < q; i80] < -0. 146 -0. 154 < q: i8i] < -0. 152 -0. 160 < q [182] < -0. 158 -0. 166 < q; i83] < -0. 164 -0. 172 < q: i84] < -0. 170 -0. 178 < q L 185] < -0. 176 -0. 183 < q [186] < -0. 181 -0. 189 < q [187] < -0. 187 -0. 194 < q [188] < -0. 192 -0. 199 < q [189] < -0. 197 -0. 204 < q [190] < -0. 202 -0. 209 < q [191] < -0. 207 To be more precise, the filter coefficients q (v) obey the following relationships: -0.20294 < q [0] < -0.20292 -0.19804 < q [l] < -0.19802 -0.19295 < q [2] < -0.19293 -0.18768 < q [3] < -0.18766 -0.18226 < q [4] < -0.18224 -0.17668 < q [5] < -0.17666 -0.17097 < q [6] < -0.17095 -0.16514 < q [7] < -0.16512 - . 5 -0.15919 < q '[8] < -0.15917 -0.15313 < q [9] < -0.15311 -0.14697 < q [10] < -0.14695 -0.14071 < q [H] < -0.14069 -0.13437 < q [12] < -0.13435 -. 10 -0.12794 < q [13] < -0.12792 -0.12144 < q [14] < -0.12142 -0.11486 < q [15] < -0.11484 -0.10821 < q [16] < -0.10819 -0.10149 < q [17] < -0.10147 fifteen - . 15 -0.09471 < q [18] < -0.09469 -0.08786 < qtl9] < -0.08784 -0.08095 < q [20] < -0.08093 -0.07397 < q [21] < -0.07395 -0.06694 < q [22] < -0.06692 twenty - . 20 -0.05984 < q [23] < -0.05982 -0.05269 < q [24] < -0.05267 -0.04547 < q [25] < -0.04545 -0.03819 < q [26] < -0.03817 -0.03085 < q [27] < -0.03083 -. 25 -0.02345 < q [28] < -0.02343 -0.01598 < q 29] < -0.01596 -0. 00845 < q 30] < -0.00843 -0. 00084 < 31] < -0.00082 0. 00683 < q 32] < 0.00685 0. 01458 < q 33] < 0.01460 0. 02240 < 34] < 0.02242 0. 03030 < 35] < 0.03032 0. 03828 < 36] < 0.03830 0. 0.95758 < q [73] < 0.95760 0.96190 < q [74] < 0.96192 0.96593 < q [75] < 0.96595 0.96968 < q [76] < 0.96970 0.97317 < q [77] < 0.97319 0.97641 < q [78] < 0.97643 0. 97940 < q [79] < 0.97942 .98217 < q [80] < 0. 98219 0 .98472 < q [81] < 0. 98474 0 .98706 < q [82] < 0. 98708 0 .98919 < q [83] < 0. 98921 0 .99113 < q [84] < 0. 99115 0 .99288 < q [85] < 0. 99290 0 .99444 < q [86] < 0.99446 0 .99583 < q [87] < 0. 99585 0 .99704 < q [88] < 0.99706 0 .99809 < q [89], < 0. 99811 0 .99896 < q [90] < 0. 99898 0 .99967 < q [91] < 0. 99969 1 .00023 < q [92] < 1. 00025 1 .00062 < q [93] < 1. 00064 1 .00086 < q [94] < 1. 00088 1 .00093 < q [95] < 1. 00095 1 .00086 < q [96] < 1. 00088 1 .00062 < q [97] < 1. 00064 1 .00023 < q [98] < 1. 00025 0.99967 < q [99] < 0. 99969 0. 99896 < q [100] < 0 .99898 0.99809 < q [101] < 0.99811 0.99704 < q [102] < 0 .99706 0. 99583 < q [103] < 0 .99585 0. 99444 < q [104] < 0.99446 0. 99288 < q [105] < 0.99290 0. 99113 < q [106] < 0.99115 0. 98919 < q [107] < 0.98921 0. 98706 < q [108] < 0.98708 0. 98472 < q [109] < 0.98474 0. 98217 < q [110] < 0.98219 0. 97940 < q [111] < 0.97942 0. 97641 < q; ii2] < 0.97643 0. 97317 < q; ii3] < 0.97319 0. 96968 < q "114] < 0.96970 0. 96593 < q 115] < 0.96595 0. 96190 < q 116] < 0.96192 0. 95758 < q 117] < 0.95760 0. 95295 < q 118] < 0.95297 0. 94800 < q 119] < 0.94802 0. 94270 < q 120] < 0.94272 0. 93705 < q 121] < 0.93707 0. 93101 < q 122] < 0.93103 0. 92457 < q 123] < 0.92459 0. 91769 < q 124] < 0.91771 0. 91035 < q 125] < 0.91037 0. 90252 < q 126] < 0.90254 0.; q [185] < -0. 17666 -0. 18226 < q [186] < -0. 18224 -0. 18768 < q [187] < -0. 18766 -0. 19295 < q [188] < -0. 19293 -0. 19804 < q [189] < -0. 19802 -0. 20294 < q [190] < -0. 20292 -0. 20764 < q [191] < -0. 20762 Even more exactly, the filter coefficients g (v) can be expressed by the following equations for the integer v in the range between 0 and 191, where according to the requirements and specifications of special implementations, the prototype filter coefficients they can deviate from the following equations either individually or from the maximum absolute value typically by 10%, 5% or 2% and preferably by 1% or 0.1%: q [0] 0.2029343380 q [l] 0.1980331588 q [2] 0.1929411519 q [3] 0.1876744222 q [4] = - .1822474011 q [5] = -0. 1766730202 q [6] = -0. 1709628636 q [7] = -0. 1651273005 q [8] = -0. 1591756024 q [9] = -0. 1531160455 q [10] = -0 .1469560005 q [H] = -0 .1407020132 q [12] = -0 .1343598738 q [13] = -0 .1279346790 q [14] = -0 .1214308876 q [15] -0 .1148523686 q [16] = -0 .1082024454 q [17] = -0 .1014839341 q [18] = -0 .0946991783 q [19] = -0 .0878500799 q [20] = -0 .0809381268 q [21] -0 .0739644174 q [22] = -0 .0669296831 q [23] = -0 .0598343081 q [24] = -0 .0526783466 q [25] = -0 .0454615388 q [26] = -0 .0381833249 q [27] = -0 .0308428572 q [28] -0 .0234390115 q [29] = -0.0159703957 q [30] = -0 .0084353584 q [31] = -0 .0008319956 q [32] = 0. 0068418435 q [33] = 0. 0145885527 q [34] = 0. 0224107648 q [35] = 0. 0303113495 q [36] = 0. 0382934126 q [37] = 0. 0463602959 q [38] = 0. 0545155789 q [39] - 0. 0627630810 q [40] = 0. 0711068657 q [41 ] = 0. 0795512453 q [42] = 0. 0881007879 q [43] = 0. 0967603259 q [44] = 0. 1055349658 q [45] = 0. 1144301000 q [46] = 0. 1234514222 q [47] 0 1326049434 q [48] = 0. 1418970123 q [49] = 0. 1513343370 q [50] = 0. 1609240126 q [51] = 0. 1706735517 q [52] = 0. 1805909194 q [53] = 0. 1906845753 q [54] 0., 2009635191 q [55] = 0. 2114373458 q [56] = 0. 2221163080 q [57] = 0. 2330113868 q [58] = 0. 2441343742 q [59] = 0. 2554979664 q [ 60] = 0. 2671158700 q [61] = 0. 2790029236 q [62] = 0. 2911752349 q [63] = 0. 3036503350 q [64] 0. 9025275713 q [65] = 0. 9103585196 q [66] = 0. 9176977825 q [67] = 0. 9245760683 q [68] = 0. 9310214581 q [69] = 0. 9370596739 q [70] = 0. 9427143143 q [71] = 0. 9480070606 q [72] = 0. 9529578566 q [73] - 0. 9575850672 q [74] = 0. 9619056158 q [75] = 0. 9659351065 q [76] = 0. 9696879297 q [77] = 0. 9731773547 q [78] = 0. 9764156119 q [79] = 0..9794139640 q [80] = 0. .9821827692 q [81] = 0., 9847315377 q [82] = 0. 9870689790 q [83] = 0. 9892030462 q [84] = 0. 9911409728 q [85] = 0. 9928893067 q [86] = 0. 9944539395 q [87] = 0. 9958401318 q [88] = 0. 9970525352 q [89] = 0. 9980952118 q [90] = 0. 9989716504 q [91] = 0. 9996847806 q [92] = 1. 0002369837 q [93] = 1. 0006301028 q [94] = 1. 0008654482 q [95] = 1. 0009438063 q [96] = 1. 0008654482 q [97 ] 1. 0006301028 q [98] = 1. 0002369837 q [99] = 0. 9996847806 q [100] = 0, .9989716504 q [101] = 0, .9980952118 q [102] = 0, .9970525352 q [103 ] = 0, .9958401318 0. 9944539395 0.9928893067 0.9911409728 0.9892030462 0.9870689790 0.9847315377 0.9821827692 0.9794139640 0.9764156119 0.9731773547 0.9696879297 0.9659351065 0.9619056158 0.9575850672 0.9529578566 0.9480070606 0.9427143143 0.9370596739 0. 310214581 0.9245760683 0.9176977825 0.9103585196 0.9025275713 0.8941712974 0.2911752349 0. 2790029236 0.2671158700 0.2554979664 0.2441343742 0.2330113868 0.2221163080 0.2114373458 0.2009635191 0.1906845753 0.1805909194 0.1706735517 0.1609240126 0.1513343370 0.1418970123 0.1326049434 0.1234514222 0.1144301000 0.1055349658 0.0967603259 0.0881007879 0.0795512453 0.0711068657 0.0627630810 0.0545155789 0.0463602959 q [154] 0.0382934126 q [155] 0.0303113495 q [156] 0.0224107648 q [157] 0.0145885527 q [158] 0.0068418435 q [159] -0.0008319956 q [160] -0.0084353584 q [161] -0.0159703957 q [162] -0.0234390115 q [163] -0.0308428572 q [164] -0.0381833249 q [165] -0.0454615388 q [166] -0.0526783466 q [167] -0.0598343081 q [168] -0.0669296831 q [169] -0.0739644174 q [170] -0.0809381268 q [171] -0.0878500799 q [172] -0.0946991783 q [173] -0.1014839341 q [174] -0.1082024454 q [175] -0.1148523686 q [176] -0.1214308876 q [177] -0.1279346790 q [178] -0.1343598738 q [179] = -0.1407020132 q [180] = -0. 1469560005 q [181] = -0. 1531160455 q [182] = -0. 1591756024 q [183] = -0. 1651273005 q [184] = -0. 1709628636 q [185] = -0. 1766730202 q [186] = -0. 1822474011 q [187] = -0. 1876744222 q [188] = -0. 1929411519 q [189] = -0. 1980331588 q [190] = -0. 2029343380 q [191] = -0. 2076267137 Therefore, the present invention relates to the application of an arbitrary filter to a signal that is available in the transform domain of an exponential complex modulated filter bank, when this filter bank is designed to give virtually free performance of aliases of operations such as equalization, spectral envelope adjustment, selective frequency focus, or selective specialization of audio signal frequency. The present invention allows efficiently transforming a finite impulse response (FIR) filter given in the time domain into a set of shorter FIR filters, to be applied with a filter for each subband of the filter bank. The present invention also teaches how to convert a given discrete time domain filter into a set of subband domain filters. The result is that any given filter can be implemented with a high degree of accuracy in the subband domain of an exponential complex modulated filter bank. In a preferred embodiment, the filter converter consists of a second bank of exponential complex modulated analysis filters. For the special case of filters that implement a pure delay, the methods of the present invention coincide with those of PCT / EP2004 / 004607"Advanced processing based on a complex-exponential modulated filterbank and adaptive time framing" ("Advanced processing based on a complex-exponential modulated filter bank and adaptive time structuring "). In addition, the present invention comprises the following features: A method for obtaining a high quality approximation to the filtering of a discrete time input signal with a given filter, comprising the steps of analyzing the input signal with a filter bank of subsampled complex analysis to obtain a multitude of subband signals, filtering each subband signal with a subband filter, wherein the plurality of subband filters is obtained from the given filter by means of a filter converter, - synthesizing an output signal from the filtered subband signals with a filter bank of Sub-sampled complex synthesis. A method according to the previous one in which the filter converter consists of a bank of sub-sampled complex analysis filters. An apparatus for performing a method for obtaining a high-quality approach to the filtering of a discrete time input signal with a given filter, the method comprises the steps of analyzing the input signal with a sub-sampled complex analysis filter bank. to obtain a multitude of subband signals, filter each subband signal with a subband filter, wherein the multitude of subband filters is obtained from the given filter by means of a filter converter, synthesizing an output signal from the subband signals. subband filtered with a bank of sub-sampled complex synthesis filters. - A computer program that has instructions for performing, when running on a computer, a method for obtaining a high quality approximation to the filtering of a discrete time input signal with a given filter, the method comprises the steps of analyzing the input signal with a bank of complex subsampling analysis filters to obtain a multitude of subband signals, filter each subband signal with a subband filter, where the multitude of subband filters is obtained from the given filter by means of a filter converter, synthesizing a signal from output of filtered subband signals with a subsampling complex synthesis filter bank. Adaptation for real-cosine-modulated filter banks While the previous derivation is based on complex modulated filter banks, a note can be made here for the real, critically sampled representation obtained by a defined cosine-modulated filter bank by taking the real part of the filter. the subband samples (1) for an appropriate phase factor T. In this case, it is no longer feasible to use the band subband filtering method (3) to obtain a good approximation to a given filter. However, due to the assumptions made in the response of the prototype filter, a generalization to a Multi-band filter type 4, (*) =? ? S; (+, (* -, (22) it will be applicable, (with obvious modifications for the sub-bands, first and last). Due to critical sampling there is much less freedom in the construction of the filter mask g "r (l) | One has to do the following, which is obvious to those skilled in the art. For each m - 0, 1, ..., Z-1, use the elementary subband signal dn (k) = d [? -m] S [k] as input to the real synthesis bank, and filter the resulting output and (v) with the filter h (v) to obtain the filtered synthesis waveform z (v). Now use this filtered waveform as input to the real analysis bank. The resulting subband signal carries the coefficients of the g "r masks. { l) for n + r = m. Any reduction in the work necessary for the filter is obtained by observing that the three cases ?? = 3? + e for £ = 0,1,2 can be processed in parallel when feeding the first synthesis bank with all the corresponding elementary subband signals for each case. In this way the actual valued filter converter comprises three operations of the real analysis bank and three of real synthesis. This parallel computation represents an implementation cut for the actual valued filter converter for the case of a QMF band with good lateral jump suppression.

Figure 9 illustrates an embodiment of an inventive filter apparatus for filtering a time domain input signal from an inventive filter apparatus to obtain a time domain output signal. As already mentioned in the context of Figure la, the filter apparatus of Figure 9 comprises a bank of complex analysis filters 101, a subband filtering 102 and a complex synthesis filter bank 103, which emits the signal of time domain output. Although Figure 1 shows a system comprising one embodiment of an inventive filter apparatus in conjunction with one embodiment of a filter generator 104, the filter apparatus shown in Figure 9 comprises as an option only one filter converter 104, which provides filtering of subband 102 with the intermediate filter definition signal, for example in the form of filter connections or pulse response for each of the intermediate filters 190 of subband filtering 102. The filter apparatus shown in FIG. Figure 9 comprises additional optional components, which can provide subband filtering 102 with filter connections for the plurality of intermediate filters 190 of subband filtering 102. As an example, filter connections can also be taken from a base optional data 500, which connects to the subband filtering 102. In one embodiment, the database 500 comprises the complex valued filter connections of the intermediate filters 190. The database can be implemented as a memory system, for example in the form of a system of non-volatile memory or volatile memory system depending on the concrete implementation. Therefore, the memory solutions for the database 500 comprise ROM (ROM = read-only memory), RAM (RAM = random access memory), flash memory, magnetic memory, optical memory or other system memories. Depending on the particular implementation, a processor or a CPU (CPU = central processing unit) 510 can access the database and provide the filter connections to the subband filtering 102 or it can also access the database to provide the filter connections corresponding to the intermediate filters of the subband filtering 102. Therefore, such an embodiment comprises a database 500 from which the filter connections for subband filtering 102 can be taken. In a further embodiment of an inventive filter apparatus, which is also represented as an option in Figure 9, CPU 510 is able to calculate the filter connections online. In such a modality, CPU 510 accesses the database 500 according to a set of parameters provided by the user and / or according to a set of parameters, which are based on additional circumstances, reads one or more sets of filter connections for the intermediate filters of the filtration of subband 102 and calculates, optionally accompanied by the interpolation scheme or other estimation scheme, the desired intermediate filter connections and provides them to subband filtering 102. In a further embodiment, CPU 510 or other processor or computer system provides the filter connections of the intermediate filters 190 to the subband filtering 102 without accessing a database 500. In such an embodiment, CPU 510 or another processor calculates the filter connections and provides them to the subband filtering 102. The examples for such modality they will be explained more closely with respect to Figure 10. In an additional embodiment shown in Figure 9, C PU 510 accesses an additional database 520, reads one or more filter definition signals (e.g. in the form of impulse response signals corresponding to the time domain filter characteristic), calculates an effective filter definition signal, for example an appropriate impulse response, and provides the results of this computation to the filter converter 104. In this embodiment, the filter converter 104 then provides the subband filtering 102 with the appropriate filter connections for the intermediate filters 190. Therefore, in this embodiment, the filter converter 104 generates the effective subband filters or intermediate filters applied to each individual subband filter of each signal individual subband within the subband filtering 102 which leads to an audibly indistinguishable filtering effect of a corresponding filter applied to the time domain input signal (input signal). As a consequence, this mode is also capable of calculating the filter connections online through the filter converter 104. An example can be, for example, a device, which calculates the connections of the intermediate filters 190 of the subband filtering. 102 according to a set of parameters for example provided by the user, where the parameter base is too long, that an effective predetermination of the filter connections, optionally accompanied by some kind of interpolation scheme, would not lead to the results desired. A more specific application comes for example from the field of dynamic choice of HRTF filters in a domain to become the subband or QMF domain. As mentioned before, this is, for example, relevant in applications that include a tracker in which the database 520 is a HRTF database comprising responses to the time pulse of the HRTF filters. Since the HRTF filters usually have very long impulse responses, the use of such a scheme is especially interesting, since the connections for the intermediate filters 190 or the QMF connections are complex. The storage of the database in this domain would approximately double the memory requirements compared to the memory requirement of storing impulse responses in the time domain. However, the advantage of the reduced memory requirement can also be employed without having a CPU 510 calculating the impulse response provided to the filter converter 504. Instead, the database 520 can simply be instructed to output the corresponding definition signal. , which may be an impulse response in the time domain to the filter converter 104. In Figure 10, an amplitude / frequency characteristic 550 in the frequency domain is illustrated. In some applications, as explained above, the filter coefficients or filter connections are the intermediate filters 190 of the subband filtering 102 can be stored in the database as the database 500 of Figure 9. Alternatively or additionally, for some applications, the filter connections of the intermediate filters can also be calculated by CPU 510 of Figure 9. In the case of a special effect filtering or a signal processing of inferior quality, in which the effects of aliasing may become tolerable (at least to some degree), the filter connections of the intermediate filters 190 after the filtering of subband 102 can be estimated without a filter converter 104 or another mode of a filter generator. Possible applications especially include voice transmission over low quality lines, such as telephones or small band radio communications. Therefore, in these applications a determination of the filter connections corresponding to the transfer function 550 of FIG. 10 or another amplitude / frequency characteristic can be carried out in several sub-bands 560 with different subband frequencies without using a frequency converter. inventive filter. Figure 11 shows an embodiment of an inventive filter converter 104. As previously outlined in the context of Figure 3, the filter converter 104 comprises a complex analysis filter bank 301 to which an impulse response signal can be supplied. (actual value) indicative of an amplitude / frequency filter characteristic through an input 104a and through an optional switch 600. As outlined above, the complex analysis filter bank 301 converts the impulse response signal into a plurality of signals subband values and the intermediate filter definition signal emitted at an output 104b of the filter converter. As indicated in Figure la and Figure 9, the output 104b of the filter converter 104 can be connected to a subband filtering 102. As already mentioned above, each of the complex valued subband signals of the complex modulated filter bank 301 corresponds to an impulse response for one of the intermediate filters 190 for a subband signal in the subband filtering 102 shown in Figure la and 9. ' Typically, complex valued subband signals are significantly shorter than the impulse response signal of the filter characteristic provided at input 104a in the time domain. In addition, typically at least one of the complex valued subband signals emitted at the output 104b comprises at least two different non-dissipated values. Especially the latter characteristic distinguishes the output of the filter converter 104 from a simple gain adjustment in the filtering framework using a direct Fourier transform procedure. If, however, the filter converter 104 is not provided with an impulse response signal indicative of an amplitude / frequency filter characteristic, but a filter definition signal, comprising at least one of an amplitude / frequency filter characteristic, a phase / frequency filter characteristic or filter connections in the time domain or another domain of a filter, the filter converter 104 comprises a pulse response generator 610 for converting the filter definition signal in the impulse response signal, which is then provided through the optional switch 600 to the complex analysis filter bank 301. In a particular implementation, the impulse response generator 610 for example can calculate the impulse response signal provided to the complex analysis filter bank 301 by superposition of real valued oscillations (Fourier synthesis), wherein the amplitude characteristics and phase characteristics of the proposed filter to be transferred to the complex subband domain are considered as defined by the definition signal provided to the input 104c. In other words, if at least one of an amplitude / frequency characteristic and a phase / frequency characteristic is applied to the impulse response generator 610, an impulse response signal may be computerized by the impulse response generator 610 by assumption. of oscillations (harmonics) that consider the amplitude and phase relationships as defined by the filter definition signal. Applications of both modalities are possible of the filter apparatus and the filter generator and especially in the field of high quality audio coding and decoding. Recent developments in audio coding have provided means to obtain multi-channel signal printing on stereo headphones. This is commonly done by submixing a multi-channel signal to the stereo using the original multi-channel signal and HRTF filters. It has been shown in the prior art that the parametric multi-channel audio decoder can be combined with a binaural submix algorithm that makes it possible to interpret a multi-channel signal above the headphones without the need to first recreate the multi-channel signal of the transmitted submix signal, and subsequently submix it by means of the HRTF filters. However, this requires that the parameters to recreate the multi-channel signal (e.g., IID, CLD parameters) are combined with the HRTF filters, which in turn requires a parameterization of the HRTF filters. This requirement of a parametrization of the HRTF filters imposes high limitation on the system, since the HRTF filters can be long and thus very hard to 'model correctly with a parametric approach. This limitation makes it impossible to use long HRTF filters for combined binaural and multi-channel parametric sub-mix decoders. The crucial algorithmic component required to obtain an appropriate combination of multi-channel parameters and HRTF filters is to have access to a representation of the HRTF filters given in the subband domain assumed by the spatial parameters. This is exactly what is offered by the embodiments of the present invention. Once this representation is available, the HRTF filters can be combined into 2N filters as a function of the parametric multi-channel representation. This gives a significant advantage in terms of computational complexity over the method that first recreates the ^ channels and then applies 2 filtering operations. An example of a different application of the method employed by the embodiments of the present invention is the efficient compensation of non-perfect audio reproduction devices for audio content encoded in the MPEG HE-AAC format [ISO / IEC 14496-3: 2001 / AMD1: 2003]. Such advanced filtering steps, possibly including cancellation of cross talk, can be applied directly in the subband domain before the time domain synthesis. Other developments in audio coding have made methods available for recreating a multi-channel representation of an audio signal based on a stereo (or mono) signal and corresponding control data. These methods they differ substantially from older matrix-based solution such as Dolby® Prologic, since the additional control data is transmitted to control the recreation, also referred to as up-mixing, of the surround channels based on the transmitted mono or stereo channels. Therefore, such a parametric multi-channel audio decoder, e.g. MPEG Surround reconstructs N channels based on M transmitted channels, where N > M r y] _the additional control data. The additional control data represents a significantly lower data rate than that required for the transmission of all N channels, making the coding very efficient while at the same time ensuring compatibility with both M channel devices and N channel devices. [J. Breebaart et al. "MPEG spatial audio coding / MPEG Surround: overview and current status", Proc. 119th AES convention, New York, USA, October 2005, Preprint 6447]. These parametric surround encoding methods usually include a parametrization of the surround signal based on the Channel Level Difference (CLD) and inter-channel cross-correlation coherence (ICC). These parameters describe energy ratios and correlation between channel pairs in the up-mixing process. The additional Channel Prediction Coefficients (CPC) also they are used in the prior art to predict the exit or intermediate channels during the up-mixing process. Depending on certain implementation requirements of the inventive methods, the inventive methods can be implemented in hardware or software. The implementation can be carried out using a digital storage medium, in particular a disk, CD or DVD having an electronically readable control signal stop therein, which cooperates with a programmable computer system in such a way as to perform a mode of inventive methods. Generally, one embodiment of the present invention is, therefore, a product of the computer program with a program code stored in a machine readable carrier, the program code being operative to perform the inventive methods when the product of the program of computer runs on a computer or a processor. In other words, the modalities of the inventive methods are, therefore, a computer program having a program code to perform at least one of the inventive methods when the computer program runs from a computer. Although the foregoing has been particularly shown and described with reference to the particular modalities thereof, it will be understood by those experts in the field that Various other changes in form and details can be made without departing from the spiritual scope of it. It should be understood that various changes can be made to adapt to different modalities without departing from the broader concept described herein and to be understood by the following claims.

Claims (23)

1. CLAIMS 1. A filter apparatus for filtering a time domain input signal to obtain a time domain output signal, which is a representation of the time domain input signal filtered using a filter characteristic having a non-uniform amplitude / frequency characteristic, comprising: a complex analysis filter bank for generating L complex subband signals of the time domain input signal; a plurality of intermediate filters, each intermediate filter having a finite response to the pulse comprising (KH + KQ-1) filter connections, wherein an intermediate filter is provided for each complex subband signal; a complex synthesis filter bank for synthesizing the output of the intermediate filters to obtain the time domain output signal, a filter connection generator comprising a complex modulated filter bank based on a prototype filter comprising KQ · L connections for filtering a finite response signal to the impulse indicative of the amplitude / frequency filter characteristic in the time domain and comprising KH | L filter connections to obtain L complex valued subband signals as a intermediate filter definition signal, wherein each complex valued subband signal of the complex modulated filter bank of the filter connection generator corresponds to an impulse response for an intermediate filter comprising (KH + KQ-1) filter connections; wherein at least one of the complex valued subband signals of the complex modulated filter bank of the filter connection generator comprises at least two different non-dissipated values; wherein each complex valued subband signal from the modulated filter bank of the filter connection generator comprising (KH + KQ-1) filter connections is shorter than the impulse response signal comprising KH · L filter connections provided to the filter connection generator; wherein the plurality of intermediate filters is operative to receive the intermediate filter definition signal of the filter connection generator; wherein each intermediate filter of the plurality of intermediate filters is operative to have an impulse response that depends on the intermediate filter definition signal; wherein at least one of the intermediate filters of the plurality of the intermediate filters has a non-uniform amplitude / frequency characteristic; wherein the non-uniform amplitude / frequency characteristics of the plurality of intermediate filters together represent the non-uniform filter characteristic; and where L, KQ and KH are positive integers. The filter apparatus according to claim 1, wherein at least one of the intermediate filters has a low pass filter characteristic, a high pass filter characteristic, a bandpass filter characteristic, a characteristic of band rejection filter or a notch filter feature. 3. The filter apparatus according to any of the preceding claims, wherein the intermediate filters of the plurality of intermediate filters are finite impulse response filters. The filter apparatus according to any of the preceding claims, wherein the plurality of intermediate filters is operative to receive the intermediate filter definition signal of a database or a processor. 5. The filter apparatus according to any of the preceding claims, wherein the complex analysis filter bank is operative to emit L complex subband signals, wherein the plurality of intermediate filters comprises L intermediate filters, wherein the complex synthesis filter bank is operative to synthesize the output of the intermediate L filters, and wherein L is a positive integer greater than 1. 6. complex filter synthesis according to claim 5, wherein the bank of complex analysis filters, the plurality of intermediate filters and the complex synthesis filter bank is operative to have L = 6. The filter apparatus according to any of claims 5 or 6, wherein the plurality of intermediate filters is operative to filter the complex subband signals based on the equation 4, (*) =? Ar "('(* - /) O) where n is an integer in the range of 0 to (L-1) which indicates an index of the subband signals, where L and k are integers, where dn. { k) is the output of the intermediate filter of the subband signal with the index n, where cn (k) is the subband signal with the index n, and where gn (I) is the impulse response of the intermediate filter for the subband signal with the index n. The filter apparatus according to any of claims 5 to 7, wherein the intermediate filter with an index n has an impulse response g ". { k), which is based on the equation i "\ 8Ak) =? * (v + A £)? (v) exp -Iy (/! +) V (12) where n is an integer in the range of 0 to (L-1) that indicates the index of the subband signal, where k and v are integers, where h. { v) is the response of a filter that has the filter characteristic, where n = 3.1415926 ... is the circular number, where i = V-T is the complex unit, and where q. { u) are filter connections of a real valued prototype filter. The filter apparatus according to any of claims 5 to 8, wherein at least one of the intermediate filters with an index n has a response to the impulse gn [k), which is based on the equation gn (l) = (20) where where Nh is the length of the impulse response h (?) of a filter that has the filter characteristic, where n = 3.1415926 ... is the circular number, where i = VT is the complex unit, and in where q. { u) are filter connections of a real valued prototype filter. 10. The filter apparatus according to any of claims 8 or 9, wherein the intermediate filters are adapted so that the prototype filter connections q (u) satisfy the integers? from 0 to 191 of the relations: -0.204 < q [0] < -0.202 -0.199 < q [l] < -0.197 -0.194 < q [2] < -0.192 -0.189 < q [3] < -0.187 -0.183 < q [4] < -0.181 -0.178 < q [5]. < -0.176 -0.172 < q [6] < -0.170 -0.166 < q [7] < -0.164 -0.160 < q [8] < -0.158 -0.154 < q [9] < -0.152 -0.148 < q [10] < -0.146 -0.142 < q [H] < -0.140 -0.135 < q [12] < -0.133 -0.129 < q [13] < -0.127 -0.122 < q [14] < -0.120 -0.116 < q [15] < -0.114 -0.109 < q [16] < -0.107 -0.102 < q [17] < -0.100 -0.096 < q [18] < -0.094 -0.089 < q [19] < -0.087 0. 082 < q [20] < -0.080 0.075 < q [21] < -0.073 0.068 < q [22] < -0.066 0.061 < q [23] < -0.059 0.054 < q [24] < -0.052 0.046 < q [25] < -0.044 0.039 < q [26] < -0.037 0.032 < q [27] < -0.030 0.024 < q [28] < -0.022 0.017 < q [29] < -0.015 -0.009 < q [30] < -0.007 -0.002 < q [31] < 0.000 0.006 < q [32] < 0.008 0.014 < q [33] < 0.016 0.021 < q [34] = 0.023 0.029 < q [35] < 0.031 0.037 < q [36] < 0.039 0.045 < q [37] < 0.047 0.054 < q [38] < 0.056 0.062 < q [39] < 0.064 0.070 < q [40] < 0.072 0.079 < q [41] < 0.081 0.087 < q [42] < 0.089 0.096 < q [43] < 0.098 0.105 < q [44] < 0.107 0. 113 < q [45] < 0.115 0. 122 < q [46] < 0.124 0. 132 < q [47] < 0.134 0. 141 < q [48] < 0.143 0. 150 < q [49] < 0.152 0. 160 < q [50] < 0.162 0. 170 < q [51] < 0.172 0. 180 < q [52] < 0.182 0. 190 < q [53] < 0.192 0. 200 < q [54] < 0.202 0. 210 < q [55] < 0.212 0. 10 0.978 = q [79] < 0.980 0.981 = q [80] < 0.983 0.984 = q [81] < 0.986 0.986 = q [82] < 0.988 0.988 = q [83] < 0.990 15 0.990 = q [84] < 0.992 0.992 = q [85] = 0.994 0.993 = q [86] < 0.995 0.995 = q [87] < 0.997 0.996 = q [88] < 0.998 20 0.997 = q [89] = 0.999 0.998 = q [90] < 1,000 0.999 = q [91] < 1.001 0.999 = q [92] < 1.001 1,000 = q [93] < 1.002 25 1,000 = q [94] < 1.002 1. 000 < q [95 ^ < 1.002 1,000 < q [96] < 1.002 1,000 < q [97] = 1.002 .999 < q [98] < 1.001 .999 < q [99] < 1.001 0. 998 < q [100 < 1,000 0. 997 < q [101 < 0.999 0. 996 < q [102 < 0.998 0. 995 < q [103 < 0.997 0. 993 < q [104 < 0.995 0. 992 < q [105 < 0.994 0. 990 < q [106 < 0.992 0. 988 < q [107 < 0.990 0. 986 < q [108 < 0.988 0. 984 < q [109 < 0.986 0. 981 < q [110 < 0.983 0. 978 < qtlll < 0.980 0. 975 < q [H2 < 0.977 0. 972 < q [H3 < 0.974 0. 969 < q [H4 < 0.971 0. 965 < q [H5 < 0.967 0. #; -0.17666 -0 .17097 < q [6] < -0.17095 -0 .16514 < q [7] < -0.16512 -0 .15919. < q [8] < -0.15917 -0 .15313 < q [9] < -0.15311 -0. 14697 < q [10] < -0.14695 -0. 14071 < q [H] < -0.14069 -0. 13437 < q [12] < -0.13435 -0. 12794 < q [13] < -0.12792 -0. 12144 < q [14] < -0.12142 -0. 11486 < q [15] < -0.11484 -0. 10821 < q [16] < -0.10819 -0. 10149 < q [17] < -0.10147 -0. 09471 < q [18] < -0.09469 -0. 08786 < q [19] < -0.08784 -0. 08095 < q [20] < -0.08093 -0. 07397 < q [21] < -0.07395 -0. 06694 < q [22] < -0.06692 -0.05984 < 23] < -0.05982 -0. 05269 < q 24] < -0.05267 -0. 04547 < q 25] < -0.04545 -0. 03819 < 26] < -0.03817 -0. 03085 < 27] < -0.03083 -0. 02345 < q 28] < -0.02343 -0. 01598 < q 29] < -0.01596 -0. 00845 < q 30] < -0.00843 -0. 00084 < 31] < -0.00082 0. 00683 < q 32] < 0.00685 0. 01458 < q "33] < 0.01460 0. 02240 < 34] < 0.02242 0. 03030 < q "35] < 0.03032 0. 03828 < 36] < 0.03830 0. 04635 < q "37] < 0.04637 0. 05451 < q! 38] < 0.05453 0. 06275 < q [39] < 0.06277 0. 07110 < q; 40] < 0.07112 0. 07954 < q [41] < 0.07956 0. 08809 < q [42] < 0.08811 0. 0. 19067 < q [137] < 0. 19069 0. 18058 < q "138] < 0. 18060 0. 17066 < q 139] < 0. 17068 0. 16091 < 140] < 0. 16093 0. 15132 < q 141] < 0. 15134 0. 14189 < q 142] < 0. 14191 0. 13259 < q 143] < 0. 13261 0. 12344 < q 144] < 0. 12346 0. 11442 < q 145] < 0. 11444 0. 10552 < q 146] < 0. 10554 0. 09675 < 147] < 0. 09677 0. 08809 < q [148] < 0.08811 0. 07954 < q [149] < 0.07956 0. 07110 < q [150] < 0.07112 0. 06275 < q [151] < 0.06277 0. 05451 < q; i52] < 0.05453 0. 04635 < q [153] < 0.04637 0. 03828 < q; i54] = 0.03830 0. 03030 < q [155] < 0.03032 0. 02240 < q [156] < 0.02242 0. 01458 < q [157] < 0.01460 0. 00683 < q; 158] < 0.00685 -0. 00084 < q; i59] < -0.00082 -0. 00845 < 160] < -0.00843 -0. 01598 < q [161] < -0.01596 -0. 02345 < q [162] < -0.02343 -0. 03085 < q; i63] < -0.03083 -0. 03819 < q .164] < -0.03817 -0. 04547 < 165] < -0.04545 -0. 05269 < 166] < -0.05267 -0. 05984 < q .167] < -0.05982 -0. 06694 < q [168] < -0.06692 -0. 07397 < q [169] < -0.07395 -0. 08095 < q "170] < -0.08093 -0. 08786 < q 171] < -0.08784 -0. 09471 < q 172] < -0.09469 -0.10149 < q [173] < -0.10147 -0. 10821 < q; 174] < -0. 10819 -0. 11486 < q; i75] < -0. 11484 -0. 12144 < "176" < -0.12142 -0.12794 < q77; < -0.12792 -0.1347 < q; i78 < -0.13435 -0.14071 < q; < -0. i79] < -0.14069 -0.114697 < q080] < -0.1 14695 -0 .15313 < q181] < -0 .15 15311 -0. 15919 < q 182] < -l. 0. 15917 -0. 16514 <183> <-0. 16512 -0.1707 <184> <-0.17095 -0.17668 <185 <<-0.17666 -0 18226 < q: i86] < -0. 18224 -0. 18768 < q 187] < -0. 18766 -0. 19295 < q? 88 < -0. 19293 -0. 19804 < 189] < -0. 19802 -0.20294 < q190] < -0.20292 -0.20764 < q191] < -0.20762 12. The filter apparatus according to any of claims 8 to 11, wherein the intermediate filters are adapted, so that the actual valuated prototype filter coefficients q. {v) for the integer v in the range of 0 to 191 are given by q [0] = -0.2029343380 0. 1980331588 0.1929411519 0.1876744222 0.1822474011 0.1766730202 0.1709628636 0.1651273005 0.1591756024 0.1531160455 -0.1469560005 -0.1407020132 -0.1343598738 -0.1279346790 -0.1214308876 -0.1148523686 -0.1082024454 -0.1014839341 -0.0946991783 -0.0878500799 -0.0809381268 -0.0739644174 |0.0669296831 -0.0598343081 -0.0526783466 -0.0454615388 q [26] = -0.0381833249 q [27] = -0 .0308428572 q [28] = -0 .0234390115 q [29] -0 .0159703957 q [30] = -0 .0084353584 q [31] = -0. 0008319956 q [32] = 0. 0068418435 q [33] = 0. 0145885527 q [34] = 0. 0224107648 q [35] = 0. 0303113495 q [36] = 0. 038293 0. 9980952118 0.9970525352 0.9958401318 0.9944539395 0.9928893067 0.9911409728 0.9892030462 0.9870689790 0.9847315377 0.9821827692 0.9794139640 0.9764156119 0.9731773547 0.9696879297 0.9659351065 0.9619056158 0.9575850672 0.9529578566 0.9480070606 0.9427143143 0.9370596739 0.9310214581 0.9245760683 0.9176977825 0.9103585196 0. 9025275713 0.8941712974 0.2911752349 0.2790029236 0.2671158700 0.2554979664 0.2441343742 0.2330113868 0.2221163080 0.2114373458 0.2009635191 0.1906845753 0.1805909194 0.1706735517 0.1609240126 0.1513343370 0.1418970123 0.1326049434 0.1234514222 0.1144301000 0.1055349658 0.0967603259 0.0881007879 0.0795512453 0.0711068657 q [151] = 0..0627630810 q [152] = 0. .0545155789 q [153] = 0. .0463602959 q [154] = 0. 0382934126 q [155] = 0. 0303113495 q [156] = 0. 0224107648 q [157] = 0. 0145885527 q [158] = 0. 0068418435 q [159] = -0 .0008319956 q [160] = -0 .0084353584 q [161] = -0 .0159703957 q [162] = - 0 .0234390115 q [163] = -0 .0308428572 q [164] = -0 .0381833249 q [165] = -0 .0454615388 q [166] = -0 .0526783466 q [167] - -0 .0598343081 q [ 168] = -0 .0669296831 q [169] = -0 .0739644174 q [170] = -0 .0809381268 q [171] = -0 .0878500799 q [172] = -0 .0946991783 q [173] = -0 .1014839341 q [174] = -0 .1082024454 q [175] = -0 .1148523686 q [176] -0.1214308876 q [177] = -0.1279346790 q [178] = -0.1343598738 q [179] = -0.1407020132 q [180] = -0.1469560005 q [181] = -0.1531160455 q [182] = -0.1591756024 q [ 183] = -0.1651273005 q [184] = -0.1709628636 q [185] = -0.1766730202 q [186] = -0.1822474011 q [187] = -0.1876744222 q [188] = -0.1929411519 q [189] = -0.1980331588 q [190 ] = -0.2029343380 q [191] = -0.2076267137 13. The filter apparatus according to any of the preceding, wherein the filter characteristic is based on an HRTF filter characteristic. The filter apparatus according to any of the preceding claims, wherein the complex analysis filter bank comprises a sub-sampler for each subband signal issued by the complex analysis filter bank. 15. The filter apparatus according to the claim 14, wherein the complex analysis filter bank is adapted to emit L complex subband signals, wherein L is a positive integer greater than 1, and wherein each sub-sampler is adapted to sub-sample the sub-band signals by a factor of L. 16. The filter apparatus according to any of the preceding claims, wherein the complex analysis filter bank comprises a complex modulated filter for each complex subband signal based on a prototype filter. 17. The filter apparatus according to any of the preceding claims, wherein the complex synthesis filter bank comprises an ascending sampler for each of the subband signals. 18. The filter apparatus according to claim 17, wherein the complex synthesis filter bank is operative to synthesize L signals from the intermediate filters to obtain the time domain output signal, where L is a "number". positive integer greater than 1, wherein the complex synthesis filter bank comprises L ascending sampler and wherein each ascending sampler is adapted to sample upward the output of the intermediate filters by a factor of L. 19. The filter apparatus according to anyone of the preceding claims, wherein the complex synthesis filter bank comprises for each subband signal an intermediate synthesis filter, wherein the complex synthesis filter bank comprises a real part extractor for each signal emitted by intermediate synthesis filters. , and wherein the complex synthesis filter bank further comprises an adder to add the output of each real part extractor to obtain the time domain output signal. 20. The filter apparatus according to any of claims 1 to 18, wherein the complex synthesis filter bank comprises an intermediate synthesis filter for each of the subband signals emitted by the intermediate filters., wherein the complex synthesis filter bank further comprises an adder to sum the outputs of each intermediate synthesis filter and wherein the complex synthesis filter bank further comprises a real part extractor to extract a real valued signal as the signal output of the time domain of the output of the adder. The filter apparatus according to any of the preceding claims, wherein the filter apparatus further comprises a gain adjuster for at least one subband signal or for at least one signal emitted by an intermediate filter for adjusting the gain. 22. The filtration apparatus according to any of the preceding claims, wherein the filtering apparatus further comprises an additional intermediate filter for filtering at least one of the complex valued subband signals or for filtering at least one of the signals emitted by one of the the intermediate filters. 23. The filter connection generator for providing an intermediate filter definition signal comprising filter connections for intermediate subband filters based on an impulse response signal indicative of an amplitude / frequency filter characteristic in a frequency domain. time, comprising: a complex modulated filter bank for filtering the impulse response signal to obtain 64 complex valued subband signals as the intermediate filter definition signal, wherein the complex modulated filter bank is adapted to provide signals of sub-ensemble valuadas complex that have values gn (k) based on the equation (twenty) where (18) where Nh is the length of the impulse response h (?) Of a filter that has the filter characteristic, where n = 3. 1415926 ... is the circular number, where it is the complex unit, and where q. { v) are filter connections of a real valued prototype filter; wherein each complex valued subband signal of the complex modulated filter bank corresponds to an impulse response for an intermediate filter for a subband signal; wherein at least one of the complex valued subband signals comprises at least two different non-dissipated values; and wherein each complex valued subband signal comprises (Kh + 2) filter connections; where Kh is given by where the prototype filter connections q. { v) they satisfy the integers? from 0 to 191 of the relations: -0.204 < q [0] < -0.202 -0.199 < q [l] < -0.197 -0.194 < q [2] < -0.192 -0.189 < q [3] < -0.187 -0.183 < q [4] < -0.181 -0.178 < q [5] < -0.176 -0.172 < q [6] < -0.170 -0.166 < q [7] < -0.164 -0.160 < q [8] < -0.158 5 - . 5 -0.154 < q [9] < -0.152 -0.148 < q [10] < -0.146 -0.142 < q [ll] < -0.140 -0.135 < q [12] < -0.133 -0.129 < q [13] < -0.127 10 -. 10 -0.122 < q [14] < -0.120 -0.116 < q [15] < -0.114 -0.109 < q [16] < -0.107 -0.102 < q [17] < -0.100 -0.096 < q [18] < -0.094 fifteen - . 15 -0.089 < q [19] < -0.087 -0.082 < q [20] < -0.080 -0.075 < q [21] < -0.073 -0.068 < q [22] < -0.066 -0.061 < q [23] < -0.059 twenty - . 20 -0.054 < q [24] < -0.052 -0.046 < q [25] < -0.044 -0.039 < q [26] < -0.037 -0.032 < q [27] < -0.030 -0.024 < q [28] < -0.022 25 -. 25 -0.017 < q [29] < -0.015 -0.009 < q [30] < -0.007 -0 .002 < q [31] < 0.000 0. 006 < q [32] < 0.008 0. 014 < q [33] < 0.016 0. 021 < q [34] < 0.023 0. 029 < q [35] < 0.031 0. 037 < q [36] < 0.039 0. 045 < q [37] < 0.047 0. 054 < q [38] < 0.056 0. 062 < q [39] < 0.064 0. 070 < q [40] < 0.072 0. 079 < q [41] < 0.081 0. 087 < q [42] < 0.089 0. 096 < q [43] < 0.098 0. 105 < q [44] < 0.107 0. 113 < q '45] < 0.115 0. 122 < 46] < 0.124 0. # 15 0.936 = q [69] < 0.938 0.942 = q [70] < 0.944 0.947 = q [71] < 0.949 0.952 = q [72] < 0.954 0.957 = q [73] < 0.959 20 0.961 = q [74] < 0.963 0.965 = q [75] < 0.967 0.969 = q [76] < 0.971 0.972 = q [77] < 0.974 0.975 = q [78] < 0.977 25 0.978 = q [79] < 0.980 0. 981 < q [80] < 0.983 0 .984 < q [81] < 0.986 0 .986 < q [82] < 0.988 .988 < q [83] < 0.990 0 .990 < q [84] < 0.992 .992 < q [85] < 0.994 .993 < q [86] < 0.995 .995 < q [87] < 0.997 .996 < q [88] < 0.998 .997 < q [89] < 0.999 .998 < q [90] < 1,000 .999 < q [91] < 1.001 .999 < q [92] < 1.001 1,000 < q [93] < 1.002 1,000 < q [94] < 1.002 1,000 < q [95] < 1.002 1,000 < q [96] < 1.002 1,000 < q [97] < 1.002 .999 < q [98] < 1.001 .999 < q [99] < 1.001 0. 998 < q [100] < 1,000 0. 997 < q [101] < 0.999 0. 996 < q [102] < 0.998 0. 995 < q [103] < 0.997 0. '993 < q [104] < 0.995 0. 992 < q [105] < 0.994 0. 990 < q [106] < 0.992 0. 988 < q [107] < 0.990 0. 986 < q [108] < 0.988 0. 984 < q [109] < 0.986 0. 981 < q [110] < 0.983 0. 978 < q [111] < 0.980 0. 975 < q [112] < 0.977 0. 972 < q [113] < 0.974 0. 969 < q; ii4] < 0.971 0., wherein the complex modulated filter bank is adapted to filter an impulse response signal of a non-uniform amplitude / frequency filter characteristic. 26. The filter generator according to any of claims 23 to 25, wherein the complex modulated filter bank is operative to filter the impulse response signal, and wherein the impulse response signal is based on an impulse response related to HRTF. 27. The filter generator according to any of claims 23 to 26, wherein the complex modulated filter bank is adapted to emit L complex valued subband signals, where L is a positive integer greater than 1. 28. The filter generator according to claim 27, wherein the complex modulated filter bank is adapted to provide the L sub-sampled complex valued subband signals by a factor L. 29. The filter generator according to any of the claims 27 to 28, where the complex modulated filter bank is adapted to emit L = 64 complex valued subband signals. 30. The filter generator according to any of claims 23 to 29, wherein the complex modulated filter bank is adapted to provide complex valued subband signals having gn (k) values based on the equation where n is an integer in the range of 0 to (L-1) that indicates an index of the complex valved subband signal, where k and v are integers, where h (v) is the response of a filter that has the filter characteristic, where n = 3.1415926 ... is the circular number, where i = VT is the complex unit, and where q. { v) are filter connections of a real valued prototype filter. 31. The filter generator according to any of claims 23 to 30, wherein the complex modulated filter bank is adapted such that the prototype filter q. { u) satisfies the whole numbers? from 0 to 191 of the relations: -0 .20294 < q [0] < -0. 20292 -0 .19804 < q [l] < -0. 19802 -0 .19295 < q [2] < -0. 19293 -0 .18768 < q [3] < -0. 18766 -0 .18226 < q [4] < -0. 18224 -0 .17668 < q [5] < -0. 17666 -0 .17097 < q [6] < -0. 17095 -0 .16514 < q [7] < -0. 16512 -0 .15919 < q [8] < -0. 15917 -0 .15313 < q [9] < -0. 15311 -0. 14697 < qtlO] < -0 .14695 -0. 14071 < q [H] < -0 .14069 -0. 13437 < q [12] < -0 .13435 -0. 12794 < q [13] < -0 .12792 • 0.12144 = q [14 = -0.12142 -0.11486 = [15 < -0.11484 • 0.10821 = q [16 < -0.10819 -0.10149 = q [17 < -0.10147 -0.09471 = q [18 < -0.09469 -0.08786 = q [19 < -0.08784 -0.08095 = q [20 < -0.08093 -0.07397 = q [21 < -0.07395 -0.06694 = q [22 < -0.06692 10 -. 10 -0.05984 = q [23 < -0.05982 -0.05269 = q [24 < -0.05267 -0.04547 = q [25 < -0.04545 -0.03819 = q [26 < -0.03817 -0.03085 = q [27 < -0.03083 fifteen - . 15 -0.02345 = q [28 < -0.02343 -0.01598 = q [29 < -0.01596 -0.00845 = q [30 < -0.00843 -0.00084 = q [31 < -0.00082 0.00683 < q [32 < 0.00685 20 0.01458 < q [33 < 0.01460 0.02240 = q [34 < 0.02242 0.03030 = q [35 < 0.03032 0.03828 = q [36 < 0.03830 0.04635 = q [37 < 0.04637 25 0.05451 = q [38 < 0.05453 0. 06275 < q [39] < 0.06277 0. 07110 < q [40] < 0.07112 0. 07954 < q [41] < 0.07956 0. 08809 < q [42] < 0.08811 0. 09675 < q [43] < 0.09677 0. 10552 < q [44] < 0.10554 0. 11442 < q [45] < 0.11444 0. 12344 < q [46] < 0.12346 0. 13259 < q [47] < 0.13261 0. 14189 < q [48] < 0.14191 0. 15132 < q [49] < 0.15134 0. 16091 < q [50] < 0.16093 0. 17066 < q .51] < 0.17068 0. #; 0.91771 0.92457 < q [67 < 0.92459 0.93101 < q [68 < 0.93103 0.93705 < q [69 < 0.93707 0.94270 < q [70 < 0.94272 0.94800 < q [71 < 0.94802 0.95295 < q [72 < 0.95297 0.95758 < q [73 < 0.95760 0.96190 < q [74 < 0.96192 0.96593 < q [75 < 0.96595 0.96968 < q [76 < 0.96970 0.97317 < q [77 < 0.97319 0.97641 < q [78 = 0.97643 0.97940 < q [79 < 0.97942 0.98217 < q [80 < 0.98219 0.98472 < q [81 < 0.98474 0.98706 < q [82 < 0.98708 0.98919 < q [83 < 0.98921 0.99113 < q [84 < 0.99115 0.99288 < q [85 < 0.99290 0.99444 < q [86 < 0.99446 0.99583 < q [87 < 0.99585 0.99704 < q [88 < 0.99706 0. 99809 < q [89] < 0.99811 0.99896 < q [90] < 0. 99898 0 .99967 < q [91] < 0. 99969 1 .00023 < q [92] < 1. 00025 1 .00062 < q [93] < 1. 00064 1 .00086 < q [94] < 1. 00088 1 .00093 < q [95] < 1. 00095 1 .00086 < q [96] < 1. 00088 1 .00062 < q [97] < 1. 00064 1 .00023 < q [98] < 1. 00025 0 .99967 < q [99] < 0. 99969 0. 99896 < q [100] < 0 .99898 0. 99809 < q [101] < 0 .99811 0. 99704 < q [102] < 0 .99706 0. 99583 < q [103] < 0 .99585 0. 99444 < q [104] < 0 .99446 0. 99288 < q [105] < 0 .99290 0. 99113 < q [106] < 0 .99115 0. 98919 < q [107] < 0,99821 0. 98706 < q [108] < 0 .98708 0. 98472 < q [109] < 0 .98474 0. 98217 < q [H0] < 0 .98219 0. 97940 < q [Hl] < 0 .97942 0. 97641 < q [H2] < 0 .97643 0. 97317 < q [113] < 0 .97319 0. 96968 < q [114 < 0.96970 0.96593 < q [115 < 0.96595 0.96190 < q [116 < 0.96192 0.95758 < q [117 < 0.95760 0.95295 < q [118 < 0.95297 0.94800 < q [119 < 0.94802 0.94270 < q [120 < 0.94272 0.93705 < q [121 < 0.93707 0.93101 < q [122 < 0.93103 0.92457 < q [123 < 0.92459 0.91769 < q [124 < 0.91771 0.91035 < q [125 < 0.91037 0.90252 < q [126 < 0. 0254 0.89416 < q [127 < 0.89418 0.29117 < q [128 < 0.29119 0.27899 < q [129 < 0.27901 0.26711 < q [130 < 0.26713 0.25549 < q [131 < 0.25551 0.24412 < q [132 < 0.24414 0.23300 < q [133 < 0.23302 0.22211 < q [13 < 0.22213 0.21143 < q [135 < 0.21145 0.20095 < q [136 < 0.20097 0.19067 < q [137; < 0.19069 0.18058 < q [138; < 0.18060 0. 17066 < ql 139] < 0.17068 0. 16091 < q (140) < 0.16093 0. 15132 < q 141] < 0.15134 0. 14189 < q 142] < 0.14191 0. 13259 < q 143] < 0.13261 0. # -0.1014839341 -0.0946991783 -0.0878500799 -0.0809381268 -0.0739644174 -0.0669296831 -0.0598343081 -0.0526783466 -0.0454615388 -0.0381833249 -0.0308428572 -0.0234390115 -0.0159703957 -0.0084353584 -0.0008319956 0.0068418435 0.0145885527 0.0224107648 0.0303113495 0.0382934126 0.0463602959 0.0545155789 0.0627630810 0.0711068657 0.0795512453 q [42] = 0..0881007879 q [43] = 0. .0967603259 q [44] = 0.. 1055349658 q [45] = 0., 1144301000 q [46] = 0. 1234514222 q [47] = 0 1326049434 q [48] = 0. 1418970123 q [49] = 0. 1513343370 q [50] = 0. 1609240126 q [51] = 0. 1706735517 q [52] = 0. 1805909194 q [53] = 0. 1906845753 q [54] = 0. 2009635191 q [55] = 0. 2114373458 q [56] = 0. 2221163080 q [57] = 0. 2330113868 q [58] = 0. 2441343742 q [59] = 0. 2554979664 q [ 60] = 0. 2671158700 q [61] = 0. 2790029236 q [62] = 0. 2911752349 q [63] = 0. 3036503350 q [64] = 0. 9025275713 q [65] = 0. 9103585196 q [66] = 0. 9176977825 0. 9245760683 0.9310214581 0.9370596739 0.9427143143 0.9480070606 0.9529578566 0.9575850672 0.9619056158 0.9659351065 0.9696879297 0.9731773547 0.9764156119 0.9794139640 0.9821827692 0.9847315377 0.9870689790 0.9892030462 0.9911409728 0.9928893067 0.9944539395 0.9958401318 0.9970525352 0.9980952118 0.9989716504 0.9996847806 q [92] = .0002369837 q [93] = 1. 0006301028 q [94] = 1. 0008654482 q [95] = 1. 0009438063 q [96] = 1. 0008654482 q [97] = 1. 0006301028 q [98 ] = 1. 0002369837 q [99] = 0. 9996847806 q [100] = 0 .9989716504 q [101] = 0 .9980952118 q [102] = 0 .9970525352 q [103] = 0 .9958401318 q [104] = 0 .9944539395 q [105] = 0 .9928893067 q [106] = 0 .9911409728 q [107] = 0 .9892030462 q [108] = 0 .9870689790 q [109] = 0 .9847315377 qtHO] = 0 .9821827692 q [lll] = 0 .9794139640 q [112] = 0 .9764156119 q [H3] = 0 .9731773547 q [H4] = 0 .9696879297 q [115] = 0 .9659351065 q [116] 0 .9619056158 q [117] 0.9575850672 q [118] 0.9529578566 q [119] 0.9480070606 q [120] 0.9427143143 q [121] 0.9370596739 q [122] 0.9310214581 q [123] 0.9245760683 q [124] 0.9176977825 q [125] 0.9103585196 q [126] 0.9025275713 q [127] 0.8941.712974 q [128] 0.2911752349 q [129] 0.2790029236 q; i30] 0.2671158700 q .131] 0.2554979664 q 132] 0.2441343742 q 133] 0.2330113868 q? 34] 0.2221163080 q "135] 0.2114373458 q 136] 0.2009635191 q 137] 0.1906845753 q 138] 0.1805909194 q 139] 0.1706735517 q 140] 0.1609240126 q 141] 0.1513343370 O .1418970123 0.1326049434 0.1234514222 0.1144301000 0.1055349658 0.0967603259 0.0881007879 0.0795512453 0.0711068657 0.0627630810 0.0545155789 0.0463602959 0.0382934126 0.0303113495 0.0224107648 0.0145885527 0.0068418435 -0.0008319956 -0.0084353584 -0.0159703957 -0.0234390115 -0.0308428572 -0.0381833249 -0.0454615388 -0.0526783466 0. 0598343081 0.0669296831 0.0739644174 0.0809381268 0.0878500799 0.0946991783 0.1014839341 0.1082024454 0.1148523686 0.1214308876 0.1279346790 0.1343598738 0.1407020132 0.1469560005 0.1531160455 0.1591756024 0.1651273005 0.1709628636 0.1766730202 0.1822474011 0.1876744222 0.1929411519 0.1980331588 0.2029343380 0.2076267137 33. . The filter generator according to any of claims 23 to 32, wherein the complex modulated filter bank further comprises a gain adjuster for adjusting at least one complex valued subband signal with respect to its value before emitting the signal from valuated subband adjusted by gain as the intermediate filter definition signal. 34. The filter generator according to any of claims 23 to 33, wherein the complex modulated filter bank further comprises a pulse response generator for generating the impulse response signal based on a filter definition signal. provided to the filter generator, wherein the impulse response signal emitted by the impulse response generator is provided to the complex modulated filter bank. 35. The filter generator according to claim 34, wherein the pulse response generator is adapted to generate the impulse response signal based on at least one of an amplitude / frequency filter characteristic, a characteristic of phase / frequency filter and a signal comprising a set of filter connections indicative of the amplitude / frequency filter characteristic in the time domain as a filter definition signal. 36. A method for filtering an input signal from time domain for obtaining a time domain output signal, which is a representation of the filtered time domain input signal using a filter characteristic having a non-uniform amplitude / frequency characteristic, comprising the steps of: filter a finite response signal to the pulse comprising KH | L filter connections and that is indicative of the filter characteristic of the non-uniform amplitude / frequency characteristic based on a prototype filter comprising KQ · L connections to obtain L signals of subband complex valuates as an intermediate filter definition signal, where each complex valued subband signal of the intermediate filter definition signal corresponds to a response to the filter pulse for a subband comprising (KH + KQ - 1) connections of filter; wherein at least one of the complex valued subband signals of the intermediate filter definition signal comprises at least two different non-dissipated values; and wherein at least one of the complex valued subband signals of the intermediate filter definition signal corresponds to a non-uniform amplitude / frequency characteristic; analyze the time domain input signal to obtain L complex subband signals; filtering each of the complex subband signals analyzed, wherein at least one of the complex subband signals is filtered using a non-uniform amplitude / frequency characteristic, wherein each of the complex subband signals is filtered based on a filter pulse response of the fylight definition signal where the responses to the filter pulse of the filter definition signal comprising (KH + KQ-1) filter connections are each shorter than the impulse response of a filter. filter having the filter characteristic comprising KH | L connections; and wherein the non-uniform amplitude / frequency characteristic of the impulse responses used to filter the plurality of subband signals together represent the non-uniform filter characteristic; and synthesizing from the filtering output of the complex subband signals analyzed the time domain output signal, where L, KQ and KH are positive integers. 37. The method for providing an intermediate filter definition signal comprising filter connections for intermediate subband filters based on a impulse response signal indicative of an amplitude / frequency filter characteristic in a time domain, comprising the steps of: filtering the impulse response signal indicative of the amplitude / frequency filter characteristic in a time domain for obtain 64 complex valued subband signals as the intermediate filter definition signal, where each of the complex valued subband signals comprises gn () values based on the equation *. (/) =? ? ("+ 64 · (/ - 2)) · * (?). (-95) (20) where where Nh is the length of the impulse response h (v) of a filter that has the filter characteristic, where n = 3.1415926 ... is the circular number, where i =? it is the complex unit, and where q. { u) are filter connections of a real valued prototype filter; where each complex valued subband signal corresponds to an impulse response for an intermediate filter for subband signal; wherein at least one of the complex valued subband signals comprises at least two different non-dissipated values; and wherein each complex valued subband signal comprises (Kh + 2) filter connections; where Kh is given by where the prototype filter connections q. { or) they satisfy the whole numbers? from 0 to 191 of the relations: -0.204 < q [0] < -0.202 -0.199 < q [l] < -0.197 -0.194 < q [2] < -0.192 -0.189 < q [3] < -0.187 -0.183 < q [4] < -0.181 -0.178 < q [5] < -0.176 -0.172 < q [6] < -0.170 -0.166 < q [7] < -0.164 -0.160 < q [8] < -0.158 -0.154 < q [9] < -0.152 -0.148 < q [10] < -0.146 -0.142 < q [H] < -0.140 -0.135 < q [12] < -0.133 -0. 129 < q [13] < -0.127 -0. 122 < q [14] < -0.120 -0. 116 < q [15] < -0.114 -0. 109 < q [16] < -0.107 -0. 102 < q [17] < -0.100 -0. 096 < q [18] < -0.094 -0. 089 < q; i9] = -0.087 -0. 082 < q [20] < -0.080 -0. 075 < q [21] < -0.073 -0. 068 < q [22] < -0.066 -0. 061 < q "23] &-0.059 -0. 054 < q "24] &-0.052 -0. 046 < q .25] < -0.044 -0. 039 < 26] < -0.037 -0. 032 < 27] < -0.030 -0. 024 < q 28] < -0.022 -0. 017 < q 29] < -0.015 -0. 009 < q 30] < -0.007 -0 .002 < q [31] < 0.000 0. 006 < q 32] < 0.008 0. 014 < q 33] < 0.016 0. 021 < 34] < 0.023 0. 029 < 35] < 0.031 0. 037 < 36] < 0.039 0. 045 < q [37] < 0.047 0. 054 < q [38] < 0.056 0. 062 < q [39] < 0.064 0. 070 < q [40] < 0.072 0. 079 < q [41] < 0.081 0. 087 < q [42] < 0.089 0. 096 < q [43] < 0.098 0. # 0. 190 < q [137] = 0.192 0. 180 < q [138] < 0.182 0. 170 < q [139] < 0.172 0. 160 < q [140] < 0.162 0. 150 < q [141] < 0.152 0. 141 < q [142] < 0.143 0. 132 < q [143] < 0.134 0. 122 < q [144] < 0.124 0. 113 < q [145] < 0.115 0. 105 < q [146] < 0.107 0. 096 < q [147] < 0.098 0. 087 < q [148] < 0.089 0. 079 < q [149] < 0.081 0. 070 < q [150] < 0.072 0. 062 < q [151] < 0.064 0. 054 < q [152] < 0.056 0. 045 < q [153] < 0.047 0. 037 < q [154] < 0.039 0. 029 < q [155] < 0.031 0. 021 < q [156] < 0.023 0. 014 < q [157] < 0.016 0. 006 < q [158] < 0.008 -0 .002 < q [159] < 0.000 -0. 009 < q [160] < -0.007 -0. 017 < q [161] < -0.015 -0.024 < q 162] < -0 022 -0.032 < q 163] < -0, 030 -0.039 < q 164] < -0, 037 -0.046 < 165] < -0, 044 -0.054 < 166] < -0, 052 -0.061 < q 167] < -0, 059 -0.068 < q 168] < -0.066 -0.075 < q 169] < -0.073 -0.082 < 170] < -0.080 10 -. 10 -0.089 < q 171] < -0.087 -0.096 < q 172] < -0.094 -0.102 < q 173] < -0.100 -0.109 < q 174] < -0.107 -0.116 < 175] < -0.114 fifteen - . 15 -0.122 < 176] < -0.120 -0.129 = q 177] < -0.127 -0.135 < q 178] < -0.133 -0.142 < 179] < -0.140 -0.148 < 180] < -0.146 twenty - . 20 -0.154 < q 181] < -0.152 -0.160 < q 182] < -0.158 -0.166 < q 183] < -0.164 -0.172 < 184] < -0.170 -0.178 < q 185] < -0.176 25 -. 25 -0.183 < 186] < -0.181 -0.189 < q [187] < -0.187 -0.194 < q [188] < -0.192 -0.199 < q [189] < -0.197 -0.204 < q [190] < -0.202 -0.209 < q [191] < -0.207. 38. A computer program for performing, when running on a computer, a method according to one of the methods of claims 36 or 37.