Patent Abstract:
Methods and units are shown for supporting a switching from a first coding scheme to a Modified Discrete Cosine Transform (MDCT) based coding scheme calculating a forward or inverse MDCT with a window (h(n)) of a first type for a respective coding frame, which satisfies constraints of perfect reconstruction. To avoid discontinuities during the switching, it is proposed that for a transient frame immediately after a switching, a sequence of windows (h 0 (n),h 1 (n),h 2 (n)) is provided for the forward and the inverse MDCTs. The windows of the window sequence are shorter than windows of the first type. The window sequence splits the spectrum of a respective first coding frame into nearly uncorrelated spectral components when used as basis for forward MDCTs, and the second half of the last window (h 2 (n)) of the sequence of windows is identical to the second half of a window of the first type.

Full Description:
CROSS-REFERENCE TO RELATED APPLICATION 
     This is the U.S. National Stage of International Application No. PCT/IB2003/000884 filed Mar. 11, 2003 and published in English on Sep. 23, 2004 under International Publication No. WO 2004/082288 A1. 
     FIELD OF THE INVENTION 
     The invention relates to a hybrid coding system. The invention relates more specifically to methods for supporting a switching from a first coding scheme to a second coding scheme at an encoding end and a decoding end of a hybrid coding system, the second coding scheme being a Modified Discrete Cosine Transform based coding scheme. The invention relates equally to a corresponding hybrid encoder, to a transform encoder for such a hybrid encoder, to a corresponding hybrid decoder, to a transform decoder for such a hybrid decoder, and to a corresponding hybrid coding system. 
     BACKGROUND OF THE INVENTION 
     Coding systems are known from the state of the art. They can be used for instance for coding audio or video signals for transmission or storage. 
       FIG. 1  shows the basic structure of an audio coding system, which is employed for transmission of audio signals. The audio coding system comprises an encoder  10  at a transmitting side and a decoder  11  at a receiving side. An audio signal that is to be transmitted is provided to the encoder  10 . The encoder is responsible for adapting the incoming audio data rate to a bitrate level at which the bandwidth conditions in the transmission channel are not violated. Ideally, the encoder  10  discards only irrelevant information from the audio signal in this encoding process. The encoded audio signal is then transmitted by the transmitting side of the audio coding system and received at the receiving side of the audio coding system. The decoder  11  at the receiving side reverses the encoding process to obtain a decoded audio signal with little or no audible degradation. 
     Alternatively, the audio coding system of  FIG. 1  could be employed for archiving audio data. In that case, the encoded audio data provided by the encoder  10  is stored in some storage unit, and the decoder  11  decodes audio data retrieved from this storage unit. In this alternative, it is the target that the encoder achieves a bitrate which is as low as possible, in order to save storage space. 
     Depending on the available bitrate, different coding schemes can be applied to an audio or video signal, the term coding being employed for both, encoding and decoding. 
     Speech signals have traditionally been coded at low bitrates and sampling rates, since very powerful speech production models exist for speech waveforms, e.g. Linear Prediction (LP) coding models. A good example of a speech coder is an Adaptive Multi-Rate Wideband (AMR-WB) coder. Music signals, on the other hand, have traditionally been coded at relatively high bitrates and sampling rates due to different user expectations. For coding music signals, typically transformation techniques and principles of psychoacoustics are applied. Good examples of music coders are, for example, generic Moving Picture Expert Group (MPEG) Layer III (MP3) and Advanced Audio Coding (AAC) audio coders. Such coders usually employ a Modified Discrete Cosine Transform (MDCT) for transforming received excitation signals into the frequency domain. 
     In recent years, it has been an aim to develop coding systems which can handle both, speech and music, at competitive bitrates and qualities, e.g. with 20 to 48 kbps and 16 Hz to 24 kHz. It is well-known, however, that speech coders handle music segments quite poorly, whereas generic audio coders are not able to handle speech at low bitrates. Therefore, a combination of two different coding schemes might provide a solution for filling-in the gap between low bitrate speech coders and high bitrate, high quality generic audio coders. The combination of a speech coder and a transform coder is commonly known as hybrid audio coder. A mode switching decision indicating which coder should be used for the current frame is made on a frame-by-frame basis. 
     In a hybrid coder, it is one of the main challenges to achieve a smooth transition between two enabled coding schemes. Abrupt changes at the frame boundaries when switching from one coder to another should be minimized, since any discontinuity will result in audible degradation at the output signal. 
     A smooth transition is particularly difficult to achieve when switching from a first coder, e.g. a speech coder, to an MDCT based coder. 
     MDCT based encoders apply an MDCT to coding frames which overlap by 50% to obtain the spectral representation of the excitation signal. For illustration,  FIG. 2  shows four MDCT windows over time samples of an input signal, each MDCT window being associated to another one of consecutive, overlapping coding frames. As can be seen, the overlapping portion of the windows of two consecutive coding frames n, n+1 corresponds to half of the length of a coding frame. 
       FIG. 3  illustrates how discontinuities are caused when switching from an AMR-WB speech coder to an MDCT coder. Each frame of a signal can be encoded either by an AMR-WB encoder or by an MDCT transform encoder. At the decoder, first an inverse MDCT (IMDCT) is applied to all frames which were encoded by the MDCT based transform encoder, and then the original signal is reconstructed by adding the first half of a current frame to the latter half of the preceding frame. In case a first frame n was encoded by the AMR-WB encoder and the following frame n+1 by the MDCT based transform encoder, discontinuities will be present at the decoder side at frame n+1, since the overlap component from the preceding frame n is missing. 
     The overlap component is important for the reconstruction, since it contains the original windowed signal and in addition the time aliased version of the windowed signal. 
     As described by Y. Wang, M. Vilermo, et. al. in “Restructured audio encoder for improved computational efficiency”, 108th AES Convention, Paris 2000, Preprint 5103, the MDCT works such that a signal sequence of 2N samples contains the following components: Between 0 and N−1 time samples of the original windowed signal plus the mirrored and inverted original windowed signal; between N and 2N−1 time samples of the original windowed signal plus the mirrored original windowed signal. The mirrored components are time aliases and will be canceled in the overlap-add operation. 
     In case the overlap component from the preceding frame is missing, the alias term cannot be canceled from the current frame n+1. This will result in audible degradation at the output signal. 
     In document “High-level description for the ITU-T wideband (7 kHz) ATCELP speech coding algorithm of Deutche Telekom, Aachen University of Technology (RWTH) and France Telekom (CNET)”, ITU-T SQ16 delayed contribution D.130, February 1998, by Deutsche Telekom and France Telekom, it is, proposed to use a special transition window and an extrapolation when switching from a Code Excited Linear Prediction (CELP) coder to an Adaptive Transform Coder (ATC). The transition window enables the ATC to decode the last samples of a frame. The first samples are obtained by extrapolating the samples from the previous frames via an LP-filter. Such an extrapolation, however, might introduce discontinuities and artifacts especially in the case where the frame boundaries are at the onset of a transient signal segment. 
     SUMMARY OF THE INVENTION 
     It is an object of the invention to support a smooth transition between two coding schemes. It is in particular an object of the invention to support a smooth transition from a first coding scheme to a second coding scheme which constitutes an MDCT coding scheme. 
     For the encoding end of a hybrid coding system, a first method for supporting a switching from a first coding scheme to a second coding scheme is proposed. Both coding schemes code input signals on a frame-by-frame basis. The second coding scheme is a Modified Discrete Cosine Transform based coding scheme calculating at the encoding end a Modified Discrete Cosine Transform with a window of a first type for a respective coding frame, a window of the first type satisfying constraints of perfect reconstruction. The proposed first method comprises providing for each first coding frame, which is to be encoded based on the second coding scheme after a preceding coding frame has been encoded based on the first coding scheme, a sequence of windows. The window sequence splits the spectrum of a respective first coding frame into nearly uncorrelated spectral components when used as basis for forward Modified Discrete Cosine Transforms. Further, the second half of the last window of the sequence of windows is identical to the second half of a window of the first type. The proposed first method moreover comprises calculating for a respective first coding frame a forward Modified Discrete Cosine Transform with each window of the window sequence and providing the resulting samples as encoded samples of the respective first coding frame. 
     In addition, a hybrid encoder and a transform encoder component for a hybrid encoder are proposed, which comprise means for realizing the first proposed method. 
     For the decoding end of a hybrid coding system, a second method for supporting a switching from a first coding scheme to a second coding scheme is proposed. Both coding schemes code input signals on a frame-by-frame basis. The second coding scheme is a Modified Discrete Cosine Transform based coding scheme calculating at the decoding end an Inverse Modified Discrete Cosine Transform with a window of a first type for a respective coding frame and overlap-adding the resulting samples with samples resulting for a preceding coding frame to obtain a reconstructed signal. A window of the first type satisfies constraints of perfect reconstruction. The proposed second method comprises providing for each first coding frame, which is to be decoded based on the second coding scheme after a preceding coding frame has been decoded based on the first coding scheme, a sequence of windows. The window sequence would split the spectrum of a coding frame into nearly uncorrelated spectral components when used as basis for forward Modified Discrete Cosine Transforms, and the second half of the last window of the sequence of windows is identical to the second half of a window of the first type. The proposed second method moreover comprises calculating for a respective first coding frame an Inverse Modified Discrete Cosine Transform with each window of the window sequence and providing the first half of the resulting samples as reconstructed frame samples without overlap adding. 
     In addition, a hybrid decoder and a transform decoder component for a hybrid decoder are proposed, which comprise means for realizing the second proposed method. 
     Finally, a hybrid coding system is proposed, which comprises as well the proposed hybrid encoder as the proposed hybrid decoder. 
     The invention proceeds from the consideration that forward MDCTs using a window sequence instead of a single window for a respective transition coding frame can be employed at an encoding end for splitting the source spectrum into nearly uncorrelated spectral components. The same window sequence can then be used for inverse MDCTs at a decoding end. As a result, no overlap component from a preceding coding frame which is coded by some other coding scheme will be needed for a reconstruction of the transition frame. At the same time, the window sequence can satisfy the constraints of perfect reconstruction, if the second half of the window sequence is identical to the second half of the single windows employed for all other coding frames. 
     It is an advantage of the invention that it allows a smooth transition from a first coding scheme to an MDCT based coding scheme. 
     It is further an advantage of the invention that it does not require extrapolations during codec switching. 
     It is further an advantage of the invention that since a special MDCT window sequence takes care of the switching, also the overall operation of the coding system can be simplified. 
     Preferred embodiments of the invention become apparent from the dependent claims. 
     In an advantageous embodiment of the invention as well for the encoding end as for the decoding end, the shape of the windows of the first type is determined by a function, in which one parameter is the number of samples per coding frame. In the first half of a respective first coding frame at least one subframe is defined, to which a respective window of a second type is assigned by the window sequence, the shape of a window of the second type being determined by the same function as the shape of a window of the first type, in which function the parameter representing the number of samples per coding frame is substituted by a parameter representing the number of samples per subframe. It is understood that also a different offset is selected, since the window of the second type has to start off at a different position in the coding frame. In case more than one subframe is defined, the at least one subframe constitutes preferably a sequence of subframes overlapping by 50%. A window associated to the at least one subframe is overlapped respectively by one half by a preceding window and a subsequent window of the sequence of windows, the preceding window and the subsequent window having at least for the samples in the at least one subframe a shape corresponding to the shape of the window of the second type. The sum of the values of the windows of the window sequence is equal to ‘one’ for each sample of the coding frame which lies within the first half of the coding frame and outside of the at least one subframe. Finally, the values of the windows of the window sequence are equal to ‘zero’ for each sample which lies outside of the first coding frame. 
     While the second coding scheme has to be an MDCT coding scheme, the first coding scheme can be an AMR-WB coding scheme or any other coding scheme. The domain of the signal which is provided to the MDCT based coder can be the LP domain, the time domain or some other signal domain. 
     Further, the window of the first type can be a sine based window, but equally of any other window, as long as it satisfies the constraints of perfect reconstruction. 
     The invention can be employed for audio coding, e.g. for speech coding by the first coding scheme and music coding by the MDCT coding scheme. Moreover, it can be used in video coding to switch between different coding schemes. In video coding, the invention should be applied in a two-dimensional manner, in which first the rows are coded and then the columns, or vice versa. 
     The invention can be employed in particular for storage purposes and/or for transmissions, e.g. to and from mobile terminals. 
     The invention can further be implemented either in software or using a dedicated hardware solution. Since the invention is part of a hybrid coding system, it is preferably implemented in the same way as the overall hybrid coding system. 
    
    
     
       BRIEF DESCRIPTION OF THE FIGURES 
       Other objects and features of the present invention will become apparent from the following detailed description of an exemplary embodiment of the invention considered in conjunction with the accompanying drawings. 
         FIG. 1  is a block diagram presenting the general structure of a coding system; 
         FIG. 2  illustrates the functioning of an MDCT coder; 
         FIG. 3  illustrates a problem resulting in a hybrid coding system employing an MDCT coding scheme; 
         FIG. 4  is a high level block diagram of a hybrid coding system in which an embodiment of the invention can be implemented; 
         FIG. 5  illustrates a window sequence employed in the embodiment of the invention. 
     
    
    
     DETAILED DESCRIPTION OF THE INVENTION 
       FIGS. 1 to 3  have already been described above. 
       FIG. 4  presents the general structure of a hybrid audio coding system, in which the invention can be implemented. The hybrid audio coding system can be employed for transmitting speech signals with a low bitrate and music signals with a high bitrate. 
     The hybrid audio coding system of  FIG. 4  comprises to this end a hybrid encoder  40  and a hybrid decoder  41 . The hybrid encoder  40  encodes audio signals and transmits them to the hybrid decoder  41 , while the hybrid decoder  41  receives the encoded signals, decodes them and makes them available again as audio signals. Alternatively, the encoded audio signals could also be provided by the hybrid encoder  40  for storage in a storing unit, from which they could then be retrieved again by the hybrid decoder  41 . 
     The hybrid encoder  40  comprises an LP analysis portion  401 , which is connected to an AMR-WB encoder  402 , to a transform encoder  403  and to a mode switch  404 . The mode switch  404  is also connected to the AMR-WB encoder  402  and the transform encoder  403 . The AMR-WB encoder  402 , the transform encoder  403  and the mode switch  404  are further connected to an AMR-WB+ (Adaptive Multi-Rate Wideband extension for high audio quality) bitstream multiplexer (MUX)  405 . 
     The hybrid encoder  40  comprises an LP (linear prediction) analysis portion  401 , which is connected to an AMR-WB encoder  402 , to a transform encoder  403  and to a mode switch  404 . The mode switch  404  is also connected to the AMR-WB encoder  402  and the transform encoder  403 . The AMR-WB encoder  402 , the transform encoder  403  and the mode switch  404  are further connected to an AMR-WB+ (Adaptive Multi-Rate Wideband extension for high audio quality) bitstream multiplexer (MUX)  405 . 
     When an audio signal is to be transmitted, it is first input to the LP analysis portion  401  of the hybrid encoder  40 . The LP analysis portion  401  performs an LP analysis on the input signal and quantizes the resulting LP parameters. The LP analysis is described in detail in the technical specification 3 GPP TS 26.190, “AMR Wideband speech codec; Transcoding functions”, Release 5, version 5.1.0 (2001-12), as first step of an AMR-WB encoding process. The quantized LP parameters are used for obtaining an excitation signal which is forwarded to the AMR-WB encoder component  402  and to the transform encoder component  403 . The quantized LP parameters are provided in addition to the mode switch  404 . 
     Based on the received LP parameters, the mode switch  404  determines in a known manner on a frame-by-frame basis which encoder component  402 ,  403  should be used for encoding the current frame. The mode switch  404  informs the encoder components  402 ,  403  on the respective selection and provides in addition a corresponding indication in the form of a bitstream to the AMR-WB+ bitstream multiplexer (MUX)  405 . 
     The AMR-WB encoder component  402  is selected by the mode switch  404  for encoding excitation signals resulting apparently from speech signals. Whenever the AMR-WB encoder component  402  receives from the mode switch  404  an indication that it has been selected for encoding the current signal frame, the AMR-WB encoder component  402  applies an AMR-WB encoding process to received excitation signals. Such an AMR-WB encoding process is described in detail in the above mentioned specification 3 GPP TS 26.190. Only an LP analysis, which forms in specification 3 GPP TS 26.190 part of the AMR-WB encoding process, has already been carried out separately in the LP analysis portion  401 . The AMR-WB encoder component  402  provides the resulting bitstream to the AMR-WB+ bitstream MUX  405 . 
     The transform encoder component  403  is selected by the mode switch  404  for encoding excitation signals resulting apparently from other audio signals than speech signals, in particular music signals. Whenever the transform encoder component  403  receives from the mode switch  404  an indication that it has been selected for encoding the current signal frame, the transform encoder component  403  employs a known MDCT with 50% window overlapping, as shown in  FIG. 2 , to obtain a spectral representation of the excitation signal. The known MDCT is modified, however, for the transitions from the AMR-WB coding scheme to the MDCT coding scheme, as will be described in more detail further below. The obtained spectral components are quantized, and the resulting bitstream is equally provided to the AMR-WB+ bitstream MUX  405 . 
     The AMR-WB+ bitstream MUX  405  multiplexes the received bitstreams to a single bitstream and provides them for transmission. 
     At the decoder side of the hybrid audio coding system, reverse operations are performed. 
     The AMR-WB+ bitstream DEMUX  415  of the hybrid decoder  41  receives a bitstream transmitted by the hybrid encoder  40  and demultiplexes this bitstream into a first bitstream, which is provided to the AMR-WB decoder component  412 , a second bitstream, which is provided to the transform decoder component  413 , and a third bitstream, which is provided to the mode switch  414 . 
     Based on the indication in the received bitstream, the mode switch  411  selects on a frame-by-frame basis the decoder component  412 ,  413  which is to carry out the decoding of a particular frame and informs the respective decoder component  412 ,  413  by a corresponding signal. 
     The AMR-WB decoding process which is performed by the AMR-WB decoder component  412  when selected is described in detail in the above mentioned specification 3 GPP TS 26.190. An LP synthesis, which is described in specification 3 GPP TS 26.190 as part of the AMR-WB decoding process, follows separately in the LP synthesis portion  411 , to which the AMR-WB decoder component  412  provides the LP parameters resulting in the decoding. 
     The transform decoder component  413  applies a known IMDCT when selected. The known IMDCT is modified, however, for the transitions from the AMR-WB coding scheme to the MDCT decoding scheme, as will be described in more detail further below. The transform decoder component  413  provides the LP parameters resulting in the decoding equally to the LP synthesis portion  411 . 
     The LP synthesis portion  411 , finally, performs an LP synthesis as described in detail in the above mentioned specification 3GPP TS 26.190 as the last processing step of an AMR-WB decoding process. The resulting restored audio signal is then provided for further use. 
     This AMR-WB extended coder framework is also referred to as AMR-WB+. 
     A known MDCT based encoding and a known IMDCT based decoding are described in detail for example by J. P. Princen and A. B. Bradley in “Analysis/synthesis filter bank design based on time domain aliasing cancellation”, IEEE Trans. Acoustics, Speech, and Signal Processing, 1986, Vol. ASSP-34, No. 5, October 1986, pp. 1153-1161, and by S. Shlien in “The modulated lapped transform, its time-varying forms, and its applications to audio coding standards”, IEEE Trans. Speech, and Audio Processing, Vol. 5, No. 4, July 1997, pp. 359-366. 
     The analytical expression for the regular forward MDCT of a k th  coding frame is given by the equation: 
                         X   k     ⁡     (   m   )       =       1   N     ·       ∑     i   =   0       N   -   1       ⁢       f   ⁡     (   i   )       ·       x   k     ⁡     (   i   )       ·     cos   ⁡     (       π   N     ⁢     (       2   ⁢   i     +   1   +     N   2       )     ⁢     (       2   ⁢   m     +   1     )       )               ,     
     ⁢     m   =   0     ,   …   ⁢           ,       N   /   2     -   1     ,           (   1   )               
where N is the length of the signal segment, i.e. the number of samples per frame, where f(i) defines the analysis window and where x k (i) are the samples of the excitation signal provided by the LP analysis portion  401  to the transform encoder component  403 .
 
     The analytical expression for the regular inverse MDCT for the k th  coding frame is given by the equation: 
                         q   k     ⁡     (   m   )       =       ∑     i   =   0         N   /   2     -   1       ⁢       h   ⁡     (   m   )       ·       X   k     ⁡     (   i   )       ·     cos   ⁡     (       π   N     ⁢     (       2   ⁢   m     +   1   +     N   2       )     ⁢     (       2   ⁢   i     +   1     )       )             ,     
     ⁢     m   =   0     ,   …   ⁢           ,     N   -   1     ,           (   2   )               
where N is again the length of the signal segment and where h(m) defines the synthesis window.
 
     The reconstructed k th  frame can be retrieved by an overlap-add according to the equation: 
                           x   ~     k     ⁡     (   m   )       =         q     k   -   1       ⁡     (     m   +     N   2       )       +       q   k     ⁡     (   m   )           ,     m   =   0     ,   …   ⁢           ,       N   /   2     -   1     ,           (   3   )               
where {tilde over (x)} k (m) constitute the samples which are provided by the transform decoder component  413  to the LP synthesis portion  411 .
 
     The analysis and synthesis windows f(n) and h(n) satisfy the following constraints of perfect reconstruction:
 
 f ( n )= h ( n ),  n= 0 , . . . , N/ 2−1
 
 h ( N− 1 −n )= h ( n )
 
 h   2 ( n )+ h   2 ( n+N/ 2)=1  (4)
 
     Perfect reconstruction ensures that any aliasing error introduced at the decimation stage is canceled during the reconstruction. In practice, perfect reconstruction cannot be maintained since the spectral values are quantized. Therefore, the filters should be designed in a way that the aliasing error is minimized. This goal can be achieved with filters having a sharp transition band and high stop-band attenuation. 
     A window which is frequently employed for the MDCT and the IMDCT is the sine window, since it satisfies the constraints of equation (3) and minimizes the aliasing error: 
     
       
         
           
             
               
                 
                   
                     
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     The transform encoder component  403  and the transform decoder component  413  of the hybrid audio coding system of  FIG. 4  employ the above equations (1), (2), (3) and (5) for all frames but those following immediately after a frame that was coded by AMR-WB. 
     For these transition frames, a special window sequence is defined, which satisfies the constraints for the analysis and synthesis windows and which achieves at the same time a smooth transition between AMR-WB and the MDCT based transform codec. 
     The definition of this window sequence will now be presented with reference to  FIG. 5 .  FIG. 5  is a diagram depicting an exemplary window sequence over samples in the time domain, a sample numbered ‘0’ representing the first sample of the current coding frame. It is to be noted that the representation of the samples is not linear. 
     The length of the frame in samples present in the MDCT domain is denoted as frameLen. The length of the frame in the time domain is 2*frameLen, i.e. N=2*frameLen. In the example of  FIG. 5 , there are 256 samples per frame in the MDCT domain, i.e. frameLen=256, and thus 512 samples per coding frame in the time domain. Two consecutive coding frames are overlapping by 256 samples in the time domain. 
     First, a subframe length is determined, which subframe length is denoted as frameLenS. The subframe length has to satisfy the following conditions: 
     
       
         
           
             
               
                 
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     That is, the value frameLen is to be an entire multiple of the value frameLenS, and the value frameLenS is to constitute an even number. For the example of  FIG. 5 , frameLenS is defined to be equal to 64, which satisfies the above conditions (6). 
     Next, a first offset zeroOffset, a number of short windows numShortWins and a second offset winOffset are defined as helper parameters and calculated according to the following equations:
 
zeroOffset=(frameLen−frameLenS)/2  (7)
 
numShortWins=└zeroOffset/frameLenS┘
 
if(zeroOffset mod 2≠0)
 
numShortWins=numShortWins+1  (8)
 
winOffset=zeroOffset+frameLenS  (9)
 
where the expression └x┘ in equation (8) indicates the largest integer smaller than x. The number of short windows numShortWins has to be even according to equation (8).
 
     For the example of  FIG. 5 , zeroOffset is calculated to be 96, numShortWins is calculated to be 2 and winOffset is calculated to be 160. 
     The defined parameter values are all stored fixedly in the transform encoder component  403 . 
     Based on the stored parameter values, the transform encoder component  403  calculates numShortWins forward MDCTs of a length of frameLenS and one forward MDCT of a length of frameLen for the current transition coding frame. Each MDCT is calculated according to above equation (1), in which the window f(n)=h(n) is substituted by new windows h 0 (n), h 1 (n) and h 2 (n), respectively. 
     The first MDCT window h 0 (n) has a shape according to the following equation: 
     
       
         
           
             
               
                 
                   
                     
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     In the example of  FIG. 5 , the first window h 0 (n) is equal to zero for samples −32 to −1, i.e. for all samples preceding the samples of the current coding frame. For the following samples 0 to 31, the first window h 0 (n) is equal to one. For the samples 32 to 95, it has a sine shape. Thus, the first window h 0 (n) is positioned within the coding frame so that it starts from time instant −32, while time instant 0 is the start of the coding frame. In equation (10), the first time sample from the coding frame is therefore multiplied with h 0 (32), the second sample with h 0 (33) etc. Since the values of h 0 (0) to h 0 (31) are all equal to zero, the time samples that correspond to time instants −31 to −1 are not needed. Whatever value they may have, the results of the multiplication would always be equal to zero. 
     The next numShortWins−1 MDCTs are calculated by the transform encoder component  403  based on the following window shape: 
     
       
         
           
             
               
                 
                   
                     
                       
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                       frameLenS 
                     
                   
                 
               
               
                 
                   ( 
                   11 
                   ) 
                 
               
             
           
         
       
     
     This equation thus corresponds to equation (5), in which N was substituted by 2*frameLenS. In the example of  FIG. 5 , there is a single window following equation (11), and this window h 1 (n) is positioned within the coding frame so that it starts from time instant 32 and ends with time instant 159. 
     Finally, the transform encoder component  403  calculates the MDCT of the length frameLen using the following window shape: 
     
       
         
           
             
               
                 
                   
                     
                       h 
                       2 
                     
                     ⁡ 
                     
                       ( 
                       n 
                       ) 
                     
                   
                   = 
                   
                     { 
                     
                       
                         
                           0 
                         
                         
                           
                             0 
                             ≤ 
                             n 
                             &lt; 
                             zeroOffset 
                           
                         
                       
                       
                         
                           
                             sin 
                             ⁡ 
                             
                               ( 
                               
                                 
                                   π 
                                   · 
                                   
                                     ( 
                                     
                                       n 
                                       - 
                                       zeroOffset 
                                       + 
                                       0.5 
                                     
                                     ) 
                                   
                                 
                                 
                                   2 
                                   · 
                                   frameLenS 
                                 
                               
                               ) 
                             
                           
                         
                         
                           
                             zeroOffset 
                             ≤ 
                             n 
                             &lt; 
                             winOffset 
                           
                         
                       
                       
                         
                           1 
                         
                         
                           
                             winOffset 
                             ≤ 
                             n 
                             &lt; 
                             frameLen 
                           
                         
                       
                       
                         
                           
                             sin 
                             ⁡ 
                             
                               ( 
                               
                                 
                                   π 
                                   · 
                                   
                                     ( 
                                     
                                       n 
                                       + 
                                       0.5 
                                     
                                     ) 
                                   
                                 
                                 
                                   2 
                                   · 
                                   frameLen 
                                 
                               
                               ) 
                             
                           
                         
                         
                           
                             frameLen 
                             ≤ 
                             n 
                             &lt; 
                             
                               2 
                               · 
                               frameLen 
                             
                           
                         
                       
                     
                   
                 
               
               
                 
                   ( 
                   12 
                   ) 
                 
               
             
           
         
       
     
     In the example of  FIG. 5 , the last window h 2 (n) is equal to zero for samples 0 to 95, it has a modified sine shape like the first half of window h 1 (n) for samples 96 to 159, and it is equal to one for samples 160 to 259. The last part of the window from samples 259 to 511 is equal to the window employed for all other frames than the transition frames. Thus, this window h 2 (n) is positioned to cover exactly the entire coding frame. 
     The last window h(n) indicated in  FIG. 5  belongs already to the subsequent coding frame, which is overlapping by 256 samples with the current transition coding frame. 
     On the whole, the described determination of the window sequence allows a variable length windowing scheme, which depends on the frame length frameLen and on the selected length of the subframesframeLenS. 
     The application of the described window sequence to a received coding frame results in frameLen+numShortWins*frameLenS spectral samples, i.e. in the example of  FIG. 5  in  384  spectral samples. The spectral samples are then quantized by the transform encoder component  403  and provided as a bitstream to the AMR-WB+ bitstream MUX  405  of the encoder  40 . 
     At the receiver side the same window sequence is applied by the transform decoder component  413  of the hybrid decoder  41  for calculating separate IMDCTs according to the above equation (2) to obtain the reconstructed output signal for that frame. No knowledge is required about an overlap component from the previous frame. 
     The above presented special window sequence is valid only for the duration of a current frame, in case the previous frame was coded with the AMR-WB coder  402 ,  412  and in case the current frame is coded with the transform coder  403 ,  413 . The special window sequence is not applied for the following frame anymore, regardless of whether the next frame is coded by the AMR-WB coder  402 ,  412  or the transform coder  403 ,  413 . If the next frame is coded by the transform coder  403 ,  413 , the conventional window sequence is used. 
     It is to be noted that the described embodiment constitutes only one of a variety of possible embodiments of the invention.

Technology Classification (CPC): 6