Title of Invention  APPARATUSES AND METHODS FOR DERIVING AND RECONSTRUCTING AN AUDIO SIGNAL 

Abstract  The invention relates to an encoder for deriving a representation of an audio signal having a first frame, a second frame following the first frame, and a third frame following the second frame, the encoder comprising a warp estimator for estimating first warp information for the first and the second frame and for estimating second warp information for the second frame and the third frame, the warp information describing a pitch information of the audio signal; a spectral analyzer for deriving first spectral coefficients for the first and the second frame using the first warp information and for deriving second spectral coefficients for the second and the third frame using the second warp information; and an output interface for outputting the representation of the audio signal including the first and the second spectral coefficients. 
Full Text  Time Warped Modified Transform Coding of Audio Signals Field of the Invention The present invention relates to audio source coding systems and in particular to audio coding schemes using blockbased transforms. Background of the Invention and Prior Art Several ways are known in the art to encode audio and video content. Generally, of course, the aim is to encode the content in a bitsaving manner without degrading the recon¬struction quality of the signal. Recently, new approaches to encode audio and video content have been developed, amongst which transformbased perceptual audio coding achieves the largest coding gain for stationary signals, that is when large transform sizes, can be applied. (See for example T. Painter and A. Spanias: "Perceptual coding of digital audio", Proceedings of the IEEE, Vol. 88, No. 4, April 2000, pages 451513). Stationary parts of audio are often well modelled by a fixed finite number of stationary sinusoids. Once the transform size is large enough to resolve those components, a fixed number of bits is required for a given distortion target. By further increasing the transform size, larger and larger segments of the audio signal will be described without increasing the bit demand. For nonstationary signals, however, it becomes necessary to reduce the transform size and thus the coding gain will decrease rapidly. To overcome this problem, for abrupt changes and transient events, transform size switching can be applied without significantly increasing the mean coding cost. That is, when a transient event is detected, the block size (frame size) of the samples to be encoded together is decreased. For more persistently transient signals, the bit rate will of course increase dramatically. A particular interesting example for persistent transient behaviour is the pitch variation of locally harmonic signals, which is encountered mainly in the voiced parts of speech and singing, but can also originate from the vibratos and glissandos of some musical instruments. Having a harmonic signal, i.e. a signal having signal peaks distrib¬uted with equal spacing along the time axis, the term pitch describes the inverse of the time between adjacent peaks of the signal. Such a signal therefore has a perfect harmonic spectrum, consisting of a base frequency equal to the pitch and higher order harmonics. In more general terms, pitch can be defined as the inverse of the time between two neighbouring corresponding signal portions within a locally harmonic signal. However, if the pitch and thus the base frequency varies with time, as it is the case in voiced sounds, the spectrum will become more and more complex and thus more inefficient to encode. A parameter closely related to the pitch of a signal is the warp of the signal. Assuming that the signal at time t has pitch equal to p(t) and that this pitch value varies smoothly over time, the warp of the signal at time t is defined by the logarithmic derivative For a harmonic signal, this definition of warp is insensitive to the particular choice of the harmonic component and systematic errors in terms of multiples or fractions of the pitch. The warp measures a change of frequency in the loga¬rithmic domain. The natural unit for warp is Hertz [Hz], but in musical terms, a signal with constant warp a(t) = aQis a sweep with a sweep rate of a0/log2 octaves per second [oct/s]. Speech signals exhibit warps of up to 10 oct/s and mean warp around 2 oct/s. As typical frame length (block length) of transform coders are so big, that the relative pitch change is significant within the frame, warps or pitch variations of that size lead to a scrambling of the frequency analysis of those coders. As, for a required constant bit rate, this can only be overcome by increasing the coarseness of quantization, this effect leads to the introduction of quantization noise, which is often perceived as reverberation. One possible technique to overcome this problem is time warping. The concept of timewarped coding is best explained by imagining a tape recorder with variable speed. When recording the audio signal, the speed is adjusted dynamically so as to achieve constant pitch over all voiced segments. The resulting locally stationary audio signal is encoded together with the applied tape speed changes. In the decoder, playback is then performed with the opposite speed changes. However, applying the simple time warping as described above has some significant drawbacks. First of all, the absolute tape speed ends up being uncontrollable, leading to a violation of duration of the entire encoded signal and bandwidth limitations. For reconstruction, addi¬tional side information on the tape speed (or equivalently on the signal pitch) has to be transmitted, introducing a substantial bitrate overhead, especially at low bitrates. The common approach of prior art methods to overcome the problem of uncontrollable duration of timewarped signals is to process consecutive nonoverlapping segments, i.e. individual frames, of the signal independently by a time warp, such that the duration of each segment is preserved. This approach is for example described in Yang et. al. "Pitch synchronous modulated lapped transform of the linear prediction residual of speech", Proceedings of ICSP '98, pages 591594. A great disadvantage of such a proceeding is that although the processed signal is stationary within segments, the pitch will exhibit jumps at each segment boundary. Those jumps will evidently lead to a loss of cod¬ing efficiency of the subsequent audio coder and audible discontinuities are introduced in the decoded signal. Time warping is also implemented in several other coding schemes. For example, US2002/0120445 describes a scheme, in which signal segments are subject to slight modifications in duration prior to blockbased transform coding. This is to avoid large signal components at the boundary of the blocks, accepting slight variations in duration of the single segments. Another technique making use of time warping is described in US6,169,970, where time warping is applied in order to boost the performance of the longterm predictor of a speech encoder. Along the same lines, in US 2005/0131681, a preprocessing unit for CELP coding of speech signals is described which applies a piecewise linear warp between nonoverlapping intervals, each containing one whitened pitch pulse. Finally, it is described in (R.J. Sluijter and A.J.E.M. Janssen, "A time warper for speech signals" IEEE workshop on Speech Coding'99, June 1999, pages 150152) how to improve on speech pitch estimation by application of a quadratic time warping function to a speech frame. Summarizing, prior art warping techniques share the problems of introducing discontinuities at frame borders and of requiring a significant amount of additional bit rate for the transmission of the parameters describing the pitch variation of the signal. Summary of the Invention It is the object of this invention to provide a concept for a more efficient coding of audio signals using time warp¬ing . In accordance with a first aspect of the present invention, this object is achieved by an encoder for deriving a repre¬sentation of an audio signal having a first frame, a second frame following the first frame, and a third frame follow¬ing the second frame, the encoder comprising: a warp esti¬mator for estimating first warp information for the first and the second frame and for estimating second warp infor¬mation for the second frame and the third frame, the warp information describing a pitch of the audio signal; a spec¬tral analyzer for deriving first spectral coefficients for the first and the second frame using the first warp infor¬mation and for deriving second spectral coefficients for the second and the third frame using the second warp infor¬mation; and an output interface for outputting the repre¬sentation of the audio signal including the first and the second spectral coefficients. In accordance with a second aspect of the present invention, this object is achieved by a decoder for reconstruct¬ing an audio signal having a first frame, a second frame following the first frame and a third frame following the second frame, using first warp information, the first warp information describing a pitch of the audio signal for the first and the second frame, second warp information, the second warp information describing a pitch of the audio signal for the second and the third frame, first spectral coefficients for the first and the second frame and second spectral coefficients for the second and the third frame, the decoder comprising: a spectral value processor for de¬riving a first combined frame using the first spectral co¬efficients and the first warp information, the first com¬bined frame having information on the first and on the sec ond frame; and for deriving a second combined frame using the second spectral coefficients and the second warp infor¬mation, the second combined frame having information on the second and the third frame; and a synthesizer for recon¬structing the second frame using the first combined frame and the second combined frame. In accordance with a third aspect of the present invention, this object is achieved by method of deriving a representa¬tion of an audio signal having a first frame, a second frame following the first frame, and a third frame follow¬ing the second frame, the method comprising: estimating first warp information for the first and the second frame and for estimating second warp information for the second frame and the third frame, the warp information describing a pitch of the audio signal; deriving first spectral coef¬ficients for the first and the second frame using the first warp information and for deriving second spectral coeffi¬cients for the second and the third frame using the second warp information; and outputting the representation of the audio signal including the first and the second spectral coefficients. In accordance with a fourth aspect of the present invention, this object is achieved by a method of reconstructing an audio signal having a first frame, a second frame fol¬lowing the first frame and a third frame following the sec¬ond frame, using first warp information, the first warp in¬formation describing a pitch of the audio signal for the first and the second frame, second warp information, the second warp information describing a pitch of the audio signal for the second and the third frame, first spectral coefficients for the first and the second frame and second spectral coefficients for the second and the third frame, the method comprising: deriving a first combined frame using the first spectral coefficients and the first warp information, the first combined frame having information on the first and on the second frame; and deriving a second combined frame using the second spectral coefficients and the second warp information, the second combined frame hav¬ing information on the second and the third frame; and re¬constructing the second frame using the first combined frame and the second combined frame. In accordance with a fifth aspect of the present invention, this object is achieved by a representation of an audio signal having a first frame, a second frame following the first frame and a third frame following the second frame, the representation comprising first spectral coefficients for the first and the second frame, the first spectral co¬efficients describing the spectral composition of a warped representation of the first and the second frame; and sec¬ond spectral coefficients describing a spectral composition of a warped representation of the second and the third frame. In accordance with a sixth aspect of the present invention, this is achieved by a computer program having a program code for performing, when running on a computer, any of the above methods. The present invention is based on the finding that a spectral representation of an audio signal having consecutive audio frames can be derived more efficiently when a common time warp is estimated for any two neighbouring frames, such that a following block transform can additionally use the warp information. Thus, window functions required for successful application of an overlap and add procedure during reconstruction can be derived and applied, already anticipating the re¬sampling of the signal due to the time warping. Therefore, the increased efficiency of blockbased transform coding of timewarped signals can be used without introducing audible discontinuities. The present invention thus offers an attractive solution to the prior art problems. On the one hand, the problem re¬lated to the segmentation of the audio signal is overcome by a particular overlap and add technique, that integrates the timewarp operations with the window operation and in¬troduces a time offset of the block transform. The result¬ing continuous time transforms have perfect reconstruction capability and their discrete time counterparts are only limited by the quality of the applied resampling technique of the decoder during reconstruction. This property results in a high bit rate convergence of the resulting audio cod¬ing scheme. It is principally possible to achieve lossless transmission of the signal by decreasing the coarseness of the quantization, that is by increasing the transmission bit rate. This can, for example, not be achieved with purely parametric coding methods. A further advantage of the present invention is a strong decrease of the bit rate demand of the additional information required to be transmitted for reversing the time warping. This is achieved by transmitting warp parameter side information rather than pitch side information. This has the further advantage that the present invention exhib¬its only a mild degree of parameter dependency as opposed to the critical dependence on correct pitch detection for many pitchparameter based audio coding methods. This is since pitch parameter transmission requires the detection of the fundamental frequency of a locally harmonic signal, which is not always easily achievable. The scheme of the present invention is therefore highly robust, as evidently detection of a higher harmonic does not falsify the warp parameter to be transmitted, given the definition of the warp parameter above. In one embodiment of the present invention, an encoding scheme is applied to encode an audio signal arranged in consecutive frames, and in particular a first, a second, and a third frame following each other. The full informa tion on the signal of the second frame is provided by a spectral representation of a combination of the first and the second frame, a warp parameter sequence for the first and the second frame as well as by a spectral representation of a combination of the second and the third frame and a warp parameter sequence for the second and the third frame. Using the inventive concept of time warping allows for an overlap and add reconstruction of the signal without having to introduce rapid pitch variations at the frame borders and the resulting introduction of additional audi¬ble discontinuities. In a further embodiment of the present invention, the warp parameter sequence is derived using wellknown pitchtracking algorithms, enabling the use of those wellknown algorithms and thus an easy implementation of the present invention into already existing coding schemes. In a further embodiment of the present invention, the warp¬ing is implemented such that the pitch of the audio signal within the frames is as constant as possible, when the au¬dio signal is time warped as indicated by the warp parame¬ters . In a further embodiment of the present invention, the bit rate is even further decreased at the cost of higher compu¬tational complexity during encoding when the warp parameter sequence is chosen such that the size of an encoded repre¬sentation of the spectral coefficients is minimized. In a further embodiment of the present invention, the inventive encoding and decoding is decomposed into the appli¬cation of a window function (windowing), a resampling and a block transform. The decomposition has the great advantage that, especially for the transform, already existing soft¬ware and hardware implementations may be used to effi¬ciently implement the inventive coding concept. At the de coder side, a further independent step of overlapping and adding is introduced to reconstruct the signal. In an alternative embodiment of an inventive decoder, addi¬tional spectral weighting is applied to the spectral coef¬ficients of the signal prior to transformation into the time domain. Doing so has the advantage of further decreas¬ing the computational complexity on the decoder side, as the computational complexity of the resampling of the sig¬nal can thus be decreased. The term "pitch" is to be interpreted in a general sense. This term also covers a pitch variation in connection with places that concern the warp information. There can be a situation, in which the warp information does not give access to absolute pitch, but to relative or normalized pitch information. So given a warp information one may arrive at a description of the pitch of the signal, when one accepts to get a correct pitch curve shape without values on the yaxis. Brief Description of the Drawings Preferred embodiments of the present invention are subsequently described by referring to the enclosed drawings, wherein: Fig. 1 shows an example of inventive warp maps; Figs. 22b show the application of an inventive warpdependent window; Figs. 3a, 3b show an example for inventive resampling; Figs. 4a, 4b show an example for inventive signal synthesis on the decoder side; Figs. 5a, 5b show an example for inventive windowing on the decoder side; Figs. 6a, 6b show an example for inventive time warping on the decoder side; Fig. 7 shows an example for an inventive overlap and add procedure on the decoder side; Fig. 8 shows an example of an inventive audio encoder; Fig. 9 shows an example of an inventive audio decoder; Fig. 10 shows a further example of an inventive decoder; Fig. 11 shows an example for a backwardcompatible implementation of the inventive concepts; Fig. 12 shows a block diagram for an implementation of the inventive encoding; Fig. 13 shows a block diagram for an example of inventive decoding; Fig. 14 shows a block diagram of a further embodiment of inventive decoding; Figs. 15a,15b show an illustration of achievable coding efficiency implementing the inventive concept. Detailed Description of Preferred Embodiments The embodiments described below are merely illustrative for the principles of the present invention for time warped transform coding of audio signals. It is understood that modifications and variations of the arrangements and the details described herein will be apparent to others skilled in the art. It is the intent, therefore, to be limited only by the scope of the impending patent claims and not by the specific details presented by way of description and expla¬nation of the embodiments herein. In the following, basic ideas and concepts of warping and block transforms are shortly reviewed to motivate the in¬ ventive concept, which will be discussed in more detail be¬ low, making reference to the enclosed figures. Generally, the specifics of the timewarped transform are easiest to derive in the domain of continuoustime signals. The following paragraphs describe the general theory, which will then be subsequently specialized and converted to its inventive application to discretetime signals. The main step in this conversion is to replace the change of coordi¬nates performed on continuoustime signals with nonuniform resampling of discretetime signals in such a way that the mean sample density is preserved, i.e. that the duration of the audio signal is not altered. Let s = ѱ(t) describe a change of time coordinate described by a continuously differentiable strictly increasing func¬tion ѱ, mapping the / axis interval / onto the s axis interval J . ѱ (t) is therefore a function that can be used to transform the timeaxis of a timedependent quantity, which is equivalent to a resampling in the time discrete case. It should be noted that in the following discussion, the taxis interval I is an interval in the normal timedomain and the xaxis interval J is an interval in the warped time domain. Given an orthonormal basis {vα} for signals of finite energy on the interval J, one obtains an orthonormal basis {ᶙα}for signals of finite energy on the interval / by the rule Given an infinite time interval I, local specification of time warp can be achieved by segmenting I and then constructing yr by gluing together rescaled pieces of normalized warp maps. A normalized warp map is a continuously differentiable and strictly increasing function which maps the unit interval [0,1] onto itself. Starting from a sequence of segmentation points t = tk where tk+1>tk, and a corresponding sequence of normalized warp maps ѱk, one constructs where dk = sk+l sk and the sequence dk is adjusted such that ѱ/(t) becomes continuously dif ferentiable. This defines ѱ(t) from the sequence of normalized warp maps y^up to an affine change of scale of the type Aѱ(t) + B . Let \{vkn} ke an orthonormal basis for signals of finite en¬ergy on the interval J , adapted to the segmentation sk=ѱ(tt), in the sense that there is an integer K, the overlap factor , such thatvkn(s) = Oif s The present invention focuses on cases K ≥ 2, since the case K = 1 corresponds to the prior art methods without overlap. It should be noted that not many constructions are presently known for K ≥ 3. A particular example for the in¬ventive concept will be developed for the case K = 2 below, including local trigonometric bases that are also used in modified discrete cosine transforms (MDCT) and other dis¬crete time lapped transforms. Let the construction of {vk„) from the segmentation be lo¬cal, in the sense that there is an integer p , such that vkn(s) does not depend on s, for / is a timewarped orthonormal basis for signals of finite energy on the interval /, which is well defined from the segmentation points tk and the sequence of normalized warp maps ѱk, independent of the initialization of the parame¬ter sequences sk and dk in (2) . It is adapted to the given segmentation in the sense that ukn(t) = 0 if t for a sequence J]k>0. Then defines a biorthogonal pair if of Riesz bases for the space of signals of finite energy on the interval / . Thus, /*„(/) as well as gk Based on the general considerations above, an example for the inventive concept will be derived in the subsequent paragraphs for the case of uniform segmentation tk=k and overlap factor K2, by using a local cosine basis adapted to the resulting segmentation on the saxis. It should be noted that the modifications necessary to deal with nonuniform segmentations are obvious such that the inventive concept is as well applicable to such nonuniform segmentations. As for example proposed by M.W. Wickerhauser, "Adapted wavelet analysis from theory to software", A. K. Peters, 1994, Chapter 4, a starting point for build¬ing a local cosine basis is a rising cutoff function p such that p(r) = 0 for r\, and Pirf + p(rf = 1 in the active region l with cutoff midpoints ck =(sk +sk+l)/2 and cutoff radii £k ~ \sk+\ ~sk)12 . This corresponds to the middle point construction of Wickerhauser. With lk  ckfi ck sk + sM , an orthornormal basis results from where the frequency index n= 0,1,2,K It is easy to verify that this construction obeys the condition of locality with p=0 and affine invariance described above. The resulting warped basis (3) on the taxis can in this case be rewrit¬ten in the form for k≤t ≤k + 2, where Øk is defined by gluing together ѱk and ѱk+i to form a continuously differentiable map of the in¬terval [0,2] onto itself, This is obtained by putting The construction of ѱk is illustrated in Fig. 1, showing the normalized time on the xaxis and the warped time on the yaxis. Fig. 1 shall be particularly discussed for the case k = 0, that is for building Ø,(/) and therefore deriv¬ing a warp function for a first frame 10, lasting from nor¬malized time 0 to normalized time 1 and for a second frame 12 lasting from normalized time 1 to normalized time 2. It is furthermore assumed that first frame 10 has a warp function 14 and second frame 12 has a warp func¬tion 16, derived with the aim of achieving equal pitch within the individual frames, when the time axis is trans¬formed as indicated by warp functions 14 and 16. It should be noted that warp function 14 corresponds to f0 . and warp function 16 corresponds to y/t . According to equation 9, a combined warp function $,(/) 18 is constructed by gluing to¬gether the warp maps 14 and 16 to form a continuously differentiable map of the interval [0,2] onto itself. As a re¬sult, the point (1,1) is transformed into (l,a), wherein a corresponds to 1mk in equation 9. As the inventive concept is directed to the application of time warping in an overlap and add scenario, the example of building the next combined warped function for frame 12 and the following frame 20 is also given in Fig. 1. It should be noted that following the overlap and add principle, for full reconstruction of frame 12, knowledge on both warp functions 18 and 22 is required. It should be further noted that gluing together two independently derived warp functions is not necessarily the only way of deriving a suitable combined warp function (j). (18, 22) as (> may very well be also derived by directly fitting a suitable warp function to two consecutive frames. It is preferred to have affine consistence of the two warp functions on the overlap of their definition domains. According to equation 6, the window function in equation 8 is defined by which increases from zero to one in the interval [0,2mk] and decreases from one to zero in the interval [2wA,2]. A biorthogonal version of (8) can also be derived if there are constants 0 for al k. Choosing ɳk=lk in (4) leads to the specializa¬tion of (5) to Thus, for the continuous time case, synthesis and analysis functions (equation 12) are derived, being dependent on the combined warped function. This dependency allows for time warping within an overlap and add scenario without loss of information on the original signal, i.e. allowing for a perfect reconstruction of the signal. It may be noted that for implementation purposes, the operations performed within equation 12 can be decomposed into a sequence of consecutive individual process steps. A particularly attractive way of doing so is to first perform a windowing of the signal, followed by a resampling of the windowed signal and finally by a transformation. As usually, audio signals are stored and transmitted digitally as discrete sample values sampled with a given sample frequency, the given example for the implementation of the inventive concept shall in the following be further devel¬oped for the application in the discrete case. The timewarped modified discrete cosine transform (TWMDCT) can be obtained from a timewarped local cosine basis by discretizing analysis integrals and synthesis waveforms. The following description is based on the biorthogonal ba¬sis (see equ. 12) . The changes required to deal with the orthogonal case (8) consist of an additional time domain weighting by the Jacobian factor . In the special case where no warp is applied, both constructions reduce to the ordinary MDCT. Let L be the transform size and assume that the signal x(t)to be analyzed is band limited by qnL (rad/s) for some q The analysis coefficients are given by Defining the windowed signal portion xk(r) = x(r + k)bk ( A particularly attractive way of discretizing this integral taught by the current invention is to choose the sample points where v is integer valued. Assum ing mild warp and the band limitation described above, this gives the approximation where The summation interval in (15) is defined by 0≤rv As it is well known that the sum (equation 15) can be computed by elementary folding operations followed by a DCT of type IV, it may be appropriate to decompose the operations of equation 15 into a series of subsequent operations and transformations to make use of already existing efficient hardware and software implementations, particularly of DCT (discrete cosine transform). According to the discretized integral, a given discrete time signal can be interpreted as the equidistant samples at sampling periods 1/L of x(t). A first step of windowing would thus lead to: for p = 0,1,2,K ,2L1 . Prior to the block transformation as described by equation 15 (introducing an additional offset depending on mk ), a resampling is required, mapping The resampling operation can be performed by any suitable method for nonequidistant resampling. Summarizing, the inventive timewarped MDCD can be decomposed into a windowing operation, a resampling and a blocktransform. The individual steps shall in the following be shortly described referencing Figs. 2 to 3b. Figs. 2 to 3b show the steps of time warped MDCT encoding considering only two windowed signal blocks of a synthetically generated pitched signal. Each individual frame comprises 102 4 samples such that each of two considered combined frames 2 4 and 26(original frames 30 and 32 and original frames 32 and 34) consists of 2 04 8 samples such that the two windowed combined frames have an overlap of 1024 samples. Figs. 2 to 2b show at the xaxis the normalized time of 3 frames to be processed. First frame 30 ranges from 0 to 1, second frame 32 ranges from 1 to 2, and 3 frame ranges from 2 to 3 on the time axis. Thus, in the normalized time domain, each time unit corresponds to one complete frame having 1024 signal samples. The normalized analysis windows span the normalized time intervals [0,2] and [1,3] . The aim of the following considerations is to recover the middle frame 32 of the signal. As the reconstruction of the outer signal frames (30, 34) requires data from adjacent windowed signal segments, this reconstruction is not to be considered here. It may be noted that the combined warp maps shown in Fig. 1 are warp maps derived from the signal of Fig. 2, illustrating the inventive combination of three subsequent normalized warp maps (dotted curves) into two overlapping warp maps (solid curves). As explained above, inventive combined1 warp maps 19 and 11 are derived for the signal analysis. Furthermore, it may be noted that due to the affine invariance of warping, this curve represents a warped map with the same warp as in the original two seg¬ments . Fig. 2 illustrates the original signal by a solid graph, its stylized pulsetrain has a pitch that grows linearly with time, hence, it has positive and decreasing warp con sidering that warp is defined to be the logarithmic deriva¬tive of the pitch. In Fig. 2, the inventive analysis win¬dows as derived using equation 17 are superimposed as dot¬ted curves. It should be noted that the deviation from standard symmetric windows (as for example in MDCT) is largest where the warp is largest that is, in the first segment [0,1]. The mathematical definition of the windows alone is given by resampling the windows of equation 11, resampling implemented as expressed by the second factor of the right hand side of equation 17. Figs. 2a and 2b illustrate the result of the inventive win¬dowing, applying the windows of Fig. 2 to the individual signal segments. Figs. 3a and 3b illustrate the result of the warp parameter dependent resampling of the windowed signal blocks of Figs. 2a and 2b, the resampling performed as indicated by the warp maps given by the solid curves of Fig. 1. Normal¬ized time interval [0,1] is mapped to the warped time in¬terval [0,a], being equivalent to a compression of the left half of the windowed signal block. Consequently, an expan¬sion of the right half of the windowed signal block is per¬formed, mapping the internal [1,2] to [a,2]. Since the warp map is derived from the signal with the aim of deriving the warped signal with constant pitch, the result of the warp¬ing (resampling according to equation 18) is a windowed signal block having constant pitch. It should be noted that a mismatch between the warped map and the signal would lead to a signal block with still varying pitch at this point, which would not disturb the final reconstruction. The offset of the following block transform is marked by circles such that the interval [m, m+1] corresponds to the discrete samples v = 1,0,...Ll with L = 1024 in formula 15. This does equivalently mean that the modulating wave forms of the block transform share a point of even symmetry at m and a point of odd symmetry at m+1. It is furthermore important to note that a equals 2m such that m is the mid point between 0 and a and m+1 is the mid point between a and 2. Summarizing, Figs. 3a and 3b describe the situation after the inventive resampling described by equa¬tion 18 which is, of course, depending on the warp parame¬ters . The timewarped transform domain samples of the signals of Figs. 3a and 3b are then quantized and coded and may be transmitted together with warp side information describing normalized warp maps y/k to a decoder. As quantization is a commonly known technique, quantization using a specific quantization rule is not illustrated in the following figures, focusing on the reconstruction of the signal on the decoder side. In one embodiment of the present invention, the decoder re¬ceives the warp map sequence together with decoded timewarped transform domain samples dkn, where dkn = 0 for n > L can be assumed due to the assumed band limitation of the signal. As on the encoder side, the starting point for achieving discrete time synthesis shall be to consider con¬tinuous time reconstruction using the synthesis waveforms of equation 12: where and with Equation (19) is the usual overlap and ad procedure of a windowed transform synthesis. As in the analysis stage, it is advantageous to sample equ. (21) at the points giving rise to which is easily computed by the following steps: First, a DCT of type IV followed by extension in 2L into samples depending on the offset parameter mk according to the rule 0≤rv gives the signal segment yk at equidistant sample points ready for the overlap and add operation described in formula (19). The resampling method can again be chosen quite freely and does not have to be the same as in the encoder. In one embodiment of the present invention spline interpolation based methods are used, where the order of the spline func¬tions can be adjusted as a function of a band limitation parameter q so as to achieve a compromise between the computational complexity and the quality of reconstruction. A common value of parameter q is q = 1/3, a case in which quadratic splines will often suffice. The decoding shall in the following be illustrated by Figs. 4a to 7 for the signal shown in Figs. 3a and 3b. It shall again be emphasized that the block transform and the transmission of the transform parameters is not described here, as this is a technique commonly known. As a start for the decoding process, Figs. 4a and 4b show a configuration, where the reverse block transform has already been per¬formed, resulting in the signals shown in Figs. 4a and 4b. One important feature of the inverse block transform is the addition of signal components not present in the original signal of Figs. 3a and 3b, which is due to the symmetry properties of the synthesis functions already explained above. In particular, the synthesis function has even sym¬metry with respect to m and odd symmetry with respect to m+1. Therefore, in the interval [0,a], positive signal com¬ponents are added in the reverse block transform whereas in the interval [a,2], negative signal components are added. Additionally, the inventive window function used for the synthesis windowing operation is superimposed as a dotted curve in Figs. 4a and 4b. The mathematical definition of this synthesis window in the warped time domain is given by equation 11. Figs. 5a and 5b show the signal, still in the warped time domain, after ap¬plication of the inventive windowing. Figs. 6a and 6b finally show the result of the warp parame¬terdependent resampling of the signals of Figs. 5a and 5b. Finally, Fig. 7 shows the result of the overlapandadd op¬eration, being the final step in the synthesis of the sig¬nal, (see equation 19). The overlapandadd operation is a superposition of the waveforms of Figs. 6a and 6b. As al¬ready mentioned above, the only frame to be fully reconstructed is the middle frame32, and, a comparison with the original situation of Fig. 2 shows that the middle frame 32 is reconstructed with high fidelity. The precise cancella¬tion of the disturbing addition signal components intro¬duced during the inverse block transform is only possible since it is a crucial property of the present invention that the two combined warped maps 14 and 22 in Fig. 1 dif fer only by an affine map within the overlapping normalized time interval [1,2]. A consequence of this is that there is a correspondence between signal portions and windows on the warped time segments [a,2] and [l,b]. When considering Figs. 4a and 4b, a linear stretching of segments [l,b] into [a, 2] will therefore make the signal graphs and window halves describe the well known principle of time domain aliasing cancellation of standard MDCT. The signal, already being aliascancelled, can then simply be mapped onto the normalized time interval [1,2] by a common inverse warp map. It may be noted that, according to a further embodiment of the present invention, additional reduction of computational complexity can be achieved by application of a prefiltering step in the frequency domain. This can be implemented by simple preweighting of the transmitted sample values dkn. Such a prefiltering is for example described in M. Unser, A. Aldroubi, and M. Eden, "Bspline signal processing part IIefficient design and applications". A implementation requires Bspline resampling to be applied to the output of the inverse block transform prior to the windowing operation. Within this embodiment, the resampling operates on a signal as derived by equation 22 having modi¬fied dkn . The application of the window function bk{rv) is also not performed. Therefore, at each end of the signal segment, the resampling must take care of the edge condi¬tions in terms of periodicities and symmetries induced by the choice of the block transform. The required windowing is then performed after the resampling using the window Summarizing, according to a first embodiment of an inventive decoder, inverse timewarped MDCT comprises, when decomposed into individual steps: • Inverse transform • Windowing • Resampling • Overlap and add. According to a second embodiment of the present invention inverse timewarped MDCT comprises: • Spectral weighting • inverse transform • Resampling • Windowing • Overlap and add. It may be noted that in a case when no warp is applied, that is the case where all normalized warp maps are triv¬ial, (ѱk(t) = t), the embodiment of the present invention as detailed above coincides exactly with usual MDCT. Further embodiments of the present invention incorporating the abovementioned features shall now be described referencing Figs. 8 to 15. Fig. 8 shows an example of an inventive audio encoder receiving a digital audio signal 100 as input and generating a bit stream to be transmitted to a decoder incorporating the inventive timewarped transform coding concept. The digital audio input signal 100 can either be a natural audio signal or a preprocessed audio signal, where for instance the preprocessing could be a whitening operation to whiten the spectrum o£ the input signal. The inventive encoder incorporates a warp parameter extractor 101, a warp transformer 102, a perceptual model calculator 103, a warp coder 104, an encoder 105, and a multiplexer 106. The warp parameter extractor 101 estimates a warp parameter se¬quence, which is input into the warp transformer 102 and into the warp coder 104. The warp transformer 102 derives a time warped spectral representation of the digital audio input signal 100. The timewarped spectral representation is input into the encoder 105 for quantization and possible other coding , as for example differential coding. The en¬coder 105 is additionally controlled by the perceptual model calculator 103. Such, for example, the coarseness of quantization may be increased when signal components are to be encoded that are mainly masked by other signal compo¬nents. The warp coder 104 encodes the warp parameter se¬quence to reduce its size during transmission within the bit stream. This could for example comprise quantization of the parameters or, for example, differential encoding or entropycoding techniques as well as arithmetic coding schemes. The multiplexer 106 receives the encoded warp parameter se¬quence from the warp coder 104 and an encoded timewarped spectral representation of the digital audio input sig¬nal 100 to multiplex both data into the bit stream output by the encoder. Fig. 9 illustrates an example of a timewarped transform decoder receiving a compatible bit stream 200 for deriving a reconstructed audio signal as output. The decoder comprises a demultiplexer 201, a warp decoder 202, a de¬coder 203, and an inverse warp transformer 204. The demultiplexer demultiplexes the bit stream into the encoded warp parameter sequence, which is input into the warp decoder 202. The demultiplexer further demultiplexes the encoded representation of the timewarped spectral representation of the audio signal, which is input into the de coder 203 being the inverse of the corresponding en¬coder 105 of the audio encoder of Fig. 8. Warp decoder 202 derives a reconstruction of the warp parameter sequence and decoder 203 derives a timewarped spectral representation of the original audio signal. The representation of the warp parameter sequence as well as the timewarped spectral representation are input into the inverse warp trans¬former 204 that derives a digital audio output signal im plementing the inventive concept of timewarped overlapped transform coding of audio signals. Fig. 10 shows a further embodiment of a timewarped transform decoder in which the warp parameter sequence is derived in the decoder itself. The alternative embodiment shown in Fig. 10 comprises a decoder 203, a warp estima¬tor 301, and an inverse warp transformer 204. The de¬coder 203 and the inverse warp transformer 204 share the same functionalities as the corresponding devices of the previous embodiment and therefore the description of these devices within different embodiments is fully interchangeable. Warp estimator 301 derives the actual warp of the timewarped spectral representation output by decoder 203 by combining earlier frequency domain pitch estimates with a current frequency domain pitch estimate. Thus, the warp parameters sequence is signalled implicitly, which has the great advantage that further bit rate can be saved since no additional warp parameter information has to be transmitted in the bit stream input into the decoder. However, the im¬plicit signalling of warped data is limited by the time resolution of the transform. Fig. 11 illustrates the backwards compatibility of the inventive concept, when prior art decoders not capable of the inventive concept of timewarped decoding are used. Such a decoder would neglect the additional warp parameter infor¬mation, thus decoding the bit stream into a frequency do¬main signal fed into an inverse transformer 401 not imple¬menting any warping. Since the frequency analysis performed by timewarped transformation in inventive encoders is well aligned with the transform that does not include any time warping, a decoder ignoring warp data will still produce a meaningful audio output. This is done at the cost of degraded audio quality due to the time warping, which is not reversed within prior art decoders. Fig. 12 shows a block diagram of the inventive method of timewarped transformation. The inventive timewarp transforming comprises windowing 501, resampling 502, and a block transformation 503. First, the input signal is windowed with an overlapping window sequence depending on the warp parameter sequence serving as additional input to each of the individual encoding steps 501 to 503. Each windowed input signal segment is subsequently resampled in the re¬sampling step 502, wherein the resampling is performed as indicated by the warp parameter sequence. Within the block transformation step 503, a block transform is derived typically using a wellknown discrete trigono¬metric transform. The transform is thus performed on the windowed and resampled signal segment. It is to be noted that the block transform does also depend on an offset value, which is derived from the warp parameter sequence. Thus, the output consists of a sequence of transform domain frames. Fig. 13 shows a flow chart of an inverse timewarped trans¬form method. The method comprises the steps of inverse block transformation 601, windowing 602, resampling 603, and overlapping and adding 604. Each frame of a transform domain signal is converted into a time domain signal by the inverse block transformation 601. Corresponding to the en¬coding step, the block transform depends on an offset value derived from the received parameter sequence serving as ad¬ditional input to the inverse block transforming 601, the windowing 602, and the resampling 603. Thg signal segment derived by the block transform 601 is subsequently windowed in the windowing step 602 and resampled in the resam¬pling 603 using the warped parameter sequence. Finally, in overlapping and adding 604, the windowed and resampled seg¬ment is added to the previously inversely transformed seg¬ments in an usual overlap and add operation, resulting in a reconstruction of the time domain output signal. Fig. 14 shows an alternative embodiment of an inventive in¬verse timewarp transformer, which is implemented to addi¬tionally reduce the computational complexity. The decoder partly shares the same functionalities with the decoder of Fig. 13.Therefore the description of the same functional blocks in both embodiments are fully interchangeable. The alternative embodiment differs from the embodiment of Fig. 13 in that it implements a spectral preweighting 701 before the inverse block transformation 601. This fixed spectral preweighting is equivalent to a time domain fil¬tering with periodicities and symmetries induced by the choice of the block transform. Such a filtering operation is part of certain spline based resampling methods, allow¬ing for a reduction of the computational complexity of sub¬sequent modified resampling 702. Such resampling is now to be performed in a signal domain with periodicities and sym¬metries induced by the choice of the block transform. Therefore, a modified windowing step 703 is performed after resampling 702. Finally, in overlapping and adding 604 the windowed and resampled segment is added to the previously inverse transformed segment in an usual overlap and add procedure giving the reconstructed time domain output sig¬nal . Figs. 15a and 15b show the strength of the inventive concept of timewarped coding, showing spectral representations of the same signal with and without time warping applied. Fig. 15a illustrates a frame of spectral lines originating from a modified discrete cosine transform of transform size 1024 of a male speech signal segment sampled at 16 kHz. The resulting frequency resolution is 7.8 Hz and only the first 600 lines are plotted for this illustration, corresponding a bandwidth of 4.7 kHz. As can be seen from the fundamental frequency and the plot, the segment is a voiced sound with a mean pitch of approximately, 155 Hz. As can be furthermore seen from Fig. 15a, the. few first harmonics of the pitchfrequency are clearly distinguishable, but towards high frequencies, the analysis becomes increas ingly dense and scrambled. This is due to the variation of the pitch within the length of the signal segment to be analyzed. Therefore, the coding of the mid to high frequency ranges requires a substantial amount of bits in order to not introduce audible artefacts upon decoding. Conversely, when fixing the bit rate, substantial amount of distortion will inevitably result from the demand of increasing the coarseness of quantization. Fig. 15b illustrates a frame of spectral lines originating from a timewarped modified discrete cosine transform according to the present invention. Obviously, the same original male audio signal has been used as in Fig. 15a. The transform parameters are the same as for Fig. 15a, but the use of a timewarped transform adapted to the signal has the visible dramatic effect on the spectral representation. The sparse and organized character of the signal in the timewarped transform domain yields a coding with much bet¬ter rate distortion performance, even when the cost of cod¬ing the additional warp data is taken into account. As already mentioned, transmission of warp parameters instead of transmission of pitch or speed information has the great advantage of decreasing the additional required bit rate dramatically. Therefore, in the following paragraphs, several inventive schemes of transmitting the required warp parameter information are detailed. For a signal with warp alt) at time i, the optimal choice of normalized warp map sequence yk for the local cosine bases (see (8), (12) is obtained by solving However, the amount of information required to describe this warp map sequence is too large and the definition and measurement of pointwise values of a(t) is difficult. For practical purposes, a warp update interval At is decided upon and each warp map y/k is described by N = l/At parame¬ters. A Warp update interval of around 1020 ms is typi¬cally sufficient for speech signals. Similarly to the con¬struction in (9) of >k from y/k and y/k+l, a continuously differentiable normalized warp map can be pieced together by N normalized warp maps via suitable affine rescaling operations. Prototype examples of normalized warp maps in¬clude where a is a warp parameter. Defining the warp of a map h(t) by h"Ih', all three maps achieve warp equal to a at / = l/2. The exponential map has constant warp in the whole interval 0≤t≤l, and for small values of a, the other two maps exhibit very small deviation from this constant value. For a given warp map applied in the decoder for the resam¬pling (23) , its inverse required in the encoder for the re¬sampling (equ. 18). A principal part of the effort for in¬version originates from the inversion of the normalized warp maps. The inversion of a quadratic map requires square root operations, the inversion of an exponential map re¬quires a logarithm, and the inverse of the rational Moebius map is a Moebius map with negated warp parameter. Since ex¬ponential functions and divisions are comparably expensive, a focus on maximum ease of computation in the decoder leads to the preferred choice of a piecewise quadratic warp map sequence ѱk . The normalized warp map ѱk is then fully defined by N warp parameters ak(0),ak(1),K ,ak(Nl) by the requirements that it • is a normalized warp map; • is pieced together by rescaled copies of one of the smooth prototype warp maps (25); • is continuously differentiable; • satisfies The present invention teaches that the warp parameters can be linearly quantized, typically to a step size of around 0.5 Hz. The resulting integer values are then coded. Alter¬natively, the derivative ѱk can be interpreted as a nor¬malized pitch curve where the values are quantized to a fixed step size, typically 0.005. In this case the resulting integer values are further differ¬ence coded, sequentially or in a hierarchical manner. In both cases, the resulting side information bitrate is typi¬cally a few hundred bits per second which is only a frac¬tion of the rate required to describe pitch data in a speech codec. An encoder with large computational resources can determine the warp data sequence that optimally reduces the coding cost or maximizes a measure of sparsity of spectral lines. A less expensive procedure is to use well known methods for pitch tracking resulting in a measured pitch function p(t) 4 and approximating the pitch curve with a piecewise linear function p0(t) in those intervals where the pitch track ex¬ist and does not exhibit large jumps in the pitch values. The estimated warp sequence is then given by inside the pitch tracking intervals. Outside those intervals the warp is set to zero. Note that a systematic error in the pitch estimates such as pitch period doubling has very little effect on warp estimates. As illustrated in Fig. 10, in an alternative embodiment of the present invention, the warped parameter sequence may be derived from the decoded transform domain data by a warp estimator . The principle is to compute a frequency domain pitch estimate for each frame of transform data or from pitches of subsequent decoded signal blocks. The warp in¬formation is then derived from a formula similar to formula 28. The application of the inventive concept has mainly been described by applying the inventive time warping in a single audio channel scenario. The inventive concept is of course by no way limited to the use within such a monophonic scenario. It may be furthermore extremely advantageous to use the high coding gain achievable by the inventive concept within multichannel coding applications, where the single or the multiple channel has to be transmitted may be coded using the inventive concept. Furthermore, warping could generally be defined as a trans¬formation of the xaxis of an arbitrary function depending on x. Therefore, the inventive concept may also be applied to scenarios where functions or representation of signals are warped that do not explicitly depend on time. For exam ple, warping of a frequency representation of a signal may also be implemented. Furthermore, the inventive concept can also be advantageously applied to signals that are segmented with arbitrary segment length and not with equal length as described in the preceding paragraphs. The use of the base functions and the discretization pre¬sented in the preceding paragraphs is furthermore to be un¬derstood as one advantageous example of applying the inven¬tive concept. For other applications, different base func¬tions as well as different discretizations may also be used. Depending on certain implementation requirements of the inventive methods, the inventive methods can be imple¬mented in hardware or in software. The implementation can be performed using a digital storage medium, in particular a disk, DVD or a CD having electronically readable control signals stored thereon, which cooperate with a programmable computer system such that the inventive methods are per¬formed. Generally, the present invention is, therefore, a computer program product with a program code stored on a machinereadable carrier, the program code being operative for performing the inventive methods when the computer program product runs on a computer. In other words, the inventive methods are, therefore, a computer program having a program code for performing at least one of the inventive methods when the computer program runs on a computer. While the foregoing has been particularly Shown and described with reference to particular embodiments thereof, it will be understood by those skilled in the art that various other changes in the form and details may be made without departing from the spirit and scope thereof. It is to be understood that various changes may be made in adapt¬ing to different embodiments without departing from the broader concepts disclosed herein and comprehended by the claims that follow. WE CLAIM: 1. Encoder for deriving a representation of an audio signal having a first frame, a second frame following the first frame, and a third frame following the second frame, the encoder comprising: a warp estimator for estimating first warp information for the first and the second frame and for estimating second warp information for the second frame and the third frame, the warp information describing a pitch information of the audio signal; a spectral analyzer for deriving first spectral coefficients for the first and the second frame using the first warp information and for deriving second spectral coefficients for the second and the third frame using the second warp information; and an output interface for outputting the representation of the audio signal including the first and the second spectral coefficients. 2 . Encoder as claimed in claim 1, wherein the warp estimator is operative to estimate the warp information such that a pitch within a warped representation of frames,the warped representation derived from frames transforming the time axis of the audio signal within the frames as indicated by the warp information, is more constant than a pitch within the frames. 3. Encoder as claimed in claim 1, wherein the warp estimator is operative to estimate the warp information using information on the variation of the pitch within the frames. 4. Encoder as claimed in claim 3, wherein the warp estimator is operative to estimate the warp information such that the information on the variation of the pitch is used only when the pitch variation is lower than a predetermined maximum pitch variation. 5. Encoder as claimed in with claim 1, wherein the warp estimator is operative to estimate the warp information such that a spectral representation of a warped representation of a frame, the warped representation derived from frames transforming the time axis of the audio signal within the frames as indicated by the warp information, is more sparsely populated than a spectral representation of the frame. 6. Encoder as claimed in claim 1, wherein the warp estimator is operative to estimate the warp information such that a number of bits consumed by an encoded representation of spectral coefficients of a warped representation of frames, the warped representation derived from frames transforming the time axis of the audio signal within the frames as indicated by the warp information, is lower than an encoded representation of spectral coefficients of the frames when both representations are derived using the same encoding rule. 7. Encoder as claimed in claim 1, wherein the encoder is adapted to derive a representation of an audio signal given by a sequence of discrete sample values. 8. Encoder as claimed in claim 1, wherein the warp estimator is operative to estimate the warp information such that a warped representation of frames, the warped representation derived from frames transforming the time axis of the audio signal within the frames as indicated by the warp information, describes the same length of the audio signal as the corresponding frames. 9. Encoder as claimed in claim 1, wherein the warp estimator is operative to estimate the warp information such that first intermediate warp information of a first corresponding frame and second intermediate warp information of a second corresponding frame are combined using a combination rule. 10. Encoder as claimed in claim 9, wherein the combination rule is such that rescaled warp parameter sequences of the first intermediate warp information are concatenated with rescaled warp parameter sequences of the second intermediate warp information. 11. Encoder as claimed in claim 10, wherein the combination rule is such that the resulting warp information comprises a continuously differentiable warp parameter sequence. 12. Encoder as claimed in claim 1, wherein the warp estimator is operative to estimate the warp information such that the warp information comprises an increasing sequence of warp parameters. 13. Encoder as claimed in claim 1, wherein the warp estimator is operative to estimate the warp information such that the warp information describes a continuously differentiable resampling rule mapping the interval [0,2] onto itself. 14. Encoder as claimed in claim 1, wherein the spectral analyzer is adapted to derive the spectral coefficients using cosine basis depending on the warp information. 15. Encoder as claimed in claim 1, wherein the spectral analyzer is adapted to derive the spectral coefficients using a weighted representation of two frames. 16. Encoder as claimed in claim 15, wherein the spectral analyzer is operative to derive the spectral coefficients such that the weighted representation of two frames is obtained by applying a window function to the two frames, and wherein the window function depends on the warp information . 17. Encoder as claimed in claim 1, wherein the spectral analyzer is adapted to derive the spectral coefficients using a resampled representation of the frames. 18. Encoder as claimed in claim 17, wherein the spectral analyzer is further adapted to derive the resampled representation transforming the time axis of the frames as indicated by the warp information. 19. Encoder as claimed in claim 1, wherein the warp information derived describes a pitch variation of the audio signal normalized to the pitch of the audio signal. 20. Encoder as claimed in claim 1, wherein the warp estimator is operative to estimate the warp information such that the warp information comprises a sequence of warp parameters, wherein each warp parameter describes a finite length interval of the audio signal. 21. Encoder as claimed in claim 1, wherein the output interface is operative to have the warp information . 22. Encoder as claimed in claim 1, wherein the output interface is operative to have a quantized representation of the warp information. 23.Decoder for reconstructing an audio signal having a first frame, a second frame following the first frame and a third frame following the second frame, using first warp information, the first warp information describing a pitch information of the audio signal for the first and the second frame, second warp information, the second warp information describing a pitch information of the audio signal for the second and the third frame, first spectral coefficients for the first and the second frame and second spectral coefficients for the second and the third frame, the decoder comprising: a spectral value processor for deriving a first combined frame using the first spectral coefficients and the first warp information, the first combined frame having information on the first and on the second frame; and for deriving a second combined frame using the second spectral coefficients and the second warp information, the second combined frame having information on the second and the third frame; and a synthesizer for reconstructing the second frame using the first combined frame and the second combined frame. 24. Decoder as claimed in claim 23, wherein the spectral value processor is operative to use cosine base functions for deriving the combined frames, the cosine base functions depending on the warp information. 25. Decoder as claimed in claim 24, wherein the spectral value processor is operative to use such cosine base functions, that using the cosine base functions on the spectral coefficients yields a timewarped unweighted representation of a combined frame. 26. Decoder as claimed in claim 23, wherein the spectral value processor is operative to use a window function for applying weights to sample values of the combined frames, the window function depending on the warp information . 27. Decoder as claimed in claim 25, wherein the spectral value processor is operative to use a window function that, when applied to the timewarped unweighted representation of a combined frame, yields a timewarped representation of a combined frame. 28. Decoder as claimed in claim 23, wherein the spectral value processor is operative to use warp information for deriving a combined frame by transforming the time axis of representations of combined frames as indicated by the warp information. 29. Decoder as claimed in claim 23, wherein the synthesizer is operative to reconstruct the second frame adding the first combined frame and the second combined frame. 30. Decoder as claimed in claim 23, wherein the decoder is adapted to reconstruct an audio signal represented by a sequence of discrete sample values. 31. Decoder as claimed in claim 23, comprising a warp estimator for deriving the first and the second warp information from the first and the second spectral coefficients. 32. Decoder as claimed in claim 23, wherein the spectral value processor is operative to perform a weighting of the spectral coefficients, applying predetermined weighting factors to the spectral coefficients. 33. Method of deriving a representation of an audio signal having a first frame, a second frame following the first frame, and a third frame following the second frame, the method comprising: estimating first warp information for the first and the second frame and for estimating second warp information for the second frame and the third frame, the warp information describing a pitch information of the audio signal; deriving first spectral coefficients for the first and the second frame using the first warp information and for deriving second spectral coefficients for the second and the third frame using the second warp information; and outputting the representation of the audio signal having the first and the second spectral coefficients. 34. Method of reconstructing an audio signal having a first frame, a second frame following the first frame and a third frame following the second frame, using first warp information, the first warp information describing a pitch information of the audio signal for the first and the second frame, second warp information, the second warp information describing a pitch information of the audio signal for the second and the third frame, first spectral coefficients for the first and the second frame and second spectral coefficients for the second and the third frame, the method comprising: deriving a first combined frame using the first spectral coefficients and the first warp information, the first combined frame having information on the first and on the second frame; and deriving a second combined frame using the second spectral coefficients and the second warp information, the second combined frame having information on the second and the third frame; and reconstructing the second frame using the first combined frame and the second combined frame. ABSTRACT TITLE: APPARATUSES AND METHODS FOR DERIVING AND RECONSTRUCTING AN AUDIO SIGNAL The invention relates to an encoder for deriving a representation of an audio signal having a first frame, a second frame following the first frame, and a third frame following the second frame, the encoder comprising a warp estimator for estimating first warp information for the first and the second frame and for estimating second warp information for the second frame and the third frame, the warp information describing a pitch information of the audio signal; a spectral analyzer for deriving first spectral coefficients for the first and the second frame using the first warp information and for deriving second spectral coefficients for the second and the third frame using the second warp information; and an output interface for outputting the representation of the audio signal including the first and the second spectral coefficients. 

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1702KOLNP2008(08052013)ABSTRACT.pdf
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1702KOLNP2008INTERNATIONAL EXM REPORT.pdf
1702KOLNP2008INTERNATIONAL PUBLICATION.pdf
1702KOLNP2008INTERNATIONAL SEARCH REPORT & OTHERS.pdf
1702KOLNP2008INTERNATIONAL SEARCH REPORT 1.1.pdf
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Patent Number  258911  

Indian Patent Application Number  1702/KOLNP/2008  
PG Journal Number  08/2014  
Publication Date  21Feb2014  
Grant Date  14Feb2014  
Date of Filing  28Apr2008  
Name of Patentee  DOLBY INTERNATIONAL AB  
Applicant Address  APOLLO BUILDING, 3E HERIKERBERGWEG 135, 1101 CN, AMSTERDAM ZUIDOOST, NETHERLANDS  
Inventors:


PCT International Classification Number  G10L 19/02  
PCT International Application Number  PCT/EP2006/010246  
PCT International Filing date  20061024  
PCT Conventions:
