Abstract:
A speech coding system that employs target signal reference shifting in code-excited linear prediction speech coding. The speech coding system performs modification of a target signal that is used to perform speech coding of a speech signal. The modified target signal that is generated from a preliminary target signal is then used to calculate an adaptive codebook gain that is used to perform speech coding of the speech signal. The speech coding performed in accordance with the present invention provides for a substantially reduced bit-rate of operation when compared to conventional speech coding methods that inherently require a significant amount of bandwidth to encode a fractional pitch lag delay during pitch prediction that is performed within conventional code-excited linear prediction speech coding systems. The speech coding system of the present invention nevertheless provides for speech coding wherein a reproduced speech signal, generated from the encoded speech signal, is substantially perceptually indistinguishable from the original speech signal. In certain embodiments of the invention, the invention provides for an alternative speech coding method that is invoked at times within the speech coding system when the conservation of bandwidth is more desirable than maintaining a high level of complexity. This instance arises frequently in relatively low bit-rate speech coding applications. The present invention is ideally operable within such low bit-rate speech coding applications.

Description:
BACKGROUND 
     1. Technical Field 
     The present invention relates generally to speech coding; and, more particularly, it relates to target signal reference shifting within speech coding. 
     2. Related Art 
     Conventional speech coding systems tend to require relatively significant amounts of bandwidth to encode speech signals. Using conventional code-excited linear prediction techniques, waveform matching between a reference signal, an input speech signal, and a re-synthesized speech signal are all used as error criteria to perform speech coding of the speech signal. To provide a high perceptual quality of the re-synthesized speech signal, the relatively significant amounts of bandwidth are required within conventional speech coding systems. Specifically, to perform good matching and thereby providing a high perceptual quality of the re-synthesized speech signal, a high bit-rate is used to encode the fractional pitch lag delay during the calculation of pitch prediction. This use of relatively significant amounts of bandwidth, as necessitated to provide this high perceptual quality, are inherently costly and wasteful to low bitrate applications. This highly consumptive use of the available bandwidth is very undesirable for low bit-rate applications. The present art does not provide an adequate solution to encode the fractional pitch lag delay during the calculation of pitch prediction within conventional speech coding systems. 
     As speech coding systems continue to move toward lower bit-rate applications, the traditional solution of dedicating a high amount of bandwidth to the coding of the fractional pitch lag delay will prove to be one of the limiting factors, especially of those speech coding systems employing code-excited linear prediction speech coding. The inherent speech coding performed within the code-excited linear prediction speech coding method does not afford a good opportunity to reduce the bandwidth dedicated to coding the fractional pitch lag delay while still maintaining a high perceptual quality of reproduced speech, i.e., high perceptual quality of the re-synthesized speech signal. 
     Traditional methods of speech coding that use a target signal (T g ) to find an adaptive codebook gain (g p ) within code-excited linear prediction speech coding commonly calculate the target signal (T g ) by matching old frame of the speech signal to a new or current frame of the speech signal. This matching gives an adaptive codebook contribution (C p ) and subsequently the contribution provided by a speech synthesis filter (H) with it as shown by the following relation 
     
       
         
           C 
           p 
           →C 
           p 
           H 
         
       
     
     Subsequently, using the calculated target signal (T g ) and the combined contribution of the contribution (C p ) and the speech synthesis filter (H), namely C p H. then the adaptive codebook gain (g p ) is uniquely solved by the following relation. 
     
       
           g   p ←Min( T   g   −g   p   C   p   H ) 2   
       
     
     Further limitations and disadvantages of conventional and traditional systems will become apparent to one of skill in the art through comparison of such systems with the present invention as set forth in the remainder of the present application with reference to the drawings. 
     SUMMARY OF THE INVENTION 
     Various aspects of the present invention can be found in a code-excited linear prediction speech coding system that performs target signal reference shifting during encoding of a speech signal. The code-excited linear prediction speech coding system itself contains, among other things, a speech synthesis filter and the speech synthesis filter contains a linear prediction coding synthesis filter and a perceptual weighting filter. The speech synthesis filter generates a target signal during encoding of the speech signal using the linear prediction coding synthesis filter and the perceptual weighting filter. In addition, the code-excited linear prediction speech coding system generates a modified target signal using the target signal that is generated during the encoding of the speech signal, and the code-excited linear prediction speech coding system generates an encoded speech signal during the encoding of the speech signal. Also, the code-excited linear prediction speech coding system is operable to decode the encoded speech signal to generate a reproduced speech signal, the reproduced speech signal is substantially perceptually indistinguishable from the speech signal prior to the encoding of the speech signal. 
     In certain embodiments of the invention, the code-excited linear prediction speech coding system is found within a speech codec. In some instances, the speech codec contains, among other things, an encoder circuitry and a decoder circuitry, and the modified target signal is generated within the encoder circuitry. If desired, the encoding of the speech signal is performed on a frame basis. Alternatively, the encoding of the speech signal is performed on a sub-frame basis. Within speech coder applications, the reproduced speech signal is generated using the modified target signal. In addition, the code-excited linear prediction speech coding system is operable within a speech signal processor. The code-excited linear prediction speech coding system is operable within a substantially low bit-rate speech coding system. 
     Other aspects of the present invention can be found in a speech coding system that performs target signal reference shifting of a speech signal. The speech coding system contains, among other things, a target signal calculation circuitry that generates a target signal and an adaptive codebook gain calculation circuitry that generates an adaptive codebook gain. The target signal corresponds to at least one portion of the speech signal, and the adaptive codebook gain is generated using the modified target signal. 
     Similar to the aspects of the invention can be found in the code-excited linear prediction speech coding system described above, the speech coding system of this particular embodiment of the invention is found with in a speech codec in certain embodiments of the invention. When the speech codec contains encoder circuitry, the speech coding system is contained within the encoder circuitry. Also, the speech coding system is operable within a speech signal processor. 
     In other embodiments of the invention, the speech coding system contains a speech synthesis filter. The speech synthesis filter contains a linear prediction coding synthesis filter and a perceptual weighting filter. If desired, the at least one portion of the speech signal that is used to encode the speech signal is extracted from the speech signal on a frame basis. Alternatively, the at least one portion of the speech signal that is used to encode the speech signal is extracted from the speech signal on a sub-frame basis. The speech coding system is operable within a substantially low bit-rate speech coding system. 
     Other aspects of the present invention can be found in a method that is used to perform target signal reference shifting on a speech signal. The method includes, among other things, calculating a target signal, modifying the target signal to generate a modified target signal, and calculating an adaptive codebook gain using the modified target signal. The target signal corresponds to at least one portion of the speech signal. 
     In certain embodiments of the invention, the method is performed on the speech signal on a frame basis; alternatively, the method is performed on a sub-frame basis. The generation of the modified target signal includes maximizing a correlation between the target signal and a product of an adaptive codebook contribution and a speech synthesis filter contribution. If further desired, the correlation is normalized during its calculation. The method is operable within speech coding system that operate using code-excited linear prediction. 
     Other aspects, advantages and novel features of the present invention will become apparent from the following detailed description of the invention when considered in conjunction with the accompanying drawings. 
    
    
     BRIEF DESCRIPTION OF THE DRAWINGS 
     FIG. 1 is a system diagram illustrating one embodiment of a speech coding system built in accordance with the present invention. 
     FIG. 2 is a system diagram illustrating another embodiment of a speech coding system built in accordance with the present invention. 
     FIG. 3 is a system diagram illustrating an embodiment of a speech signal processing system built in accordance with the present invention. 
     FIG. 4 is a system diagram illustrating an embodiment of a speech codec built in accordance with the present invention that communicates using a communication link. 
     FIG. 5 is a system diagram illustrating an embodiment of a speech codec that is a specific embodiment of the speech codec illustrated above in FIG.  4 . 
     FIG. 6 is a functional block diagram illustrating a speech coding method performed in accordance with the present invention. 
     FIG. 7 is a functional block diagram illustrating a speech coding method that is a specific embodiment of the speech coding method of FIG.  6 . 
    
    
     DETAILED DESCRIPTION OF THE INVENTION 
     FIG. 1 is a system diagram illustrating one embodiment of a speech coding system  100  built in accordance with the present invention. A speech signal is input into the speech coding system  100  as shown by the reference numeral  110 . The speech signal is partitioned into a number of frames. If desired, each of the frames of the speech signal is further partitioned into a number of sub-frames. A given frame or sub-frame of the given frame is shown by the iteration ‘i’ associated with the reference numeral  114 . For the given frame or sub-frame, a particular excitation vector (C c(i) )  116  is selected from among a fixed codebook (C c )  112 . The selected excitation vector (C c(i) )  116 , chosen from among all of the excitation vectors contained within the fixed codebook (C c )  112  for the given frame or sub-frame of the speech signal, is scaled using a fixed gain (g c )  118 . After having undergone any required scaling (either amplification or reduction) by the fixed gain (g c )  118 , the now-scaled selected excitation vector (C c(i) )  116  is fed into a summing node  120 . An excitation signal  122  is fed into the signal path of the now-scaled selected excitation vector (C c(i) )  116  after the summing node  120 . A feedback path is provided wherein pitch prediction is performed in the block  124  as shown by z −LAG . 
     The output of this signal path, after having undergone the pitch prediction is performed in the block  124  as shown by z −LAG , is then scaled using an adaptive codebook gain (g p )  126 . After having undergone any required scaling (either amplification or reduction) by the adaptive codebook gain (g p )  126 , this signal path is then fed into the summing node  120 . The output of the summing node  120 , is fed into a linear prediction coding (LPC) synthesis filter (1/A(z))  128 . The output of the linear prediction coding (LPC) synthesis filter (1/A(z))  128  and the input signal  110  are both fed into another summing node  130  wherein their combined output is fed to a perceptual weighting filter W(z)  134 . A coding error  132  is also fed into the signal path that is the output of the summing node  130 , prior to the entrance of the signal path to the perceptual weighting filter W(z)  134 . After the signal path has undergone any processing required by the perceptual weighting filter W(z)  134 , a weighted error  136  is generated. 
     From certain perspectives, the target signal reference shifting performed in accordance with the present invention is performed in either one of the perceptual weighting filter W(z)  134  or the linear prediction coding (LPC) synthesis filter (1/A(z))  128 . The combination of both the linear prediction coding (LPC) synthesis filter (1/A(z))  128  and the perceptual weighting filter W(z)  134  comprise the target signal reference shifting in other embodiments of the invention. The combination of both the linear prediction coding (LPC) synthesis filter (1/A(z))  128  and the perceptual weighting filter W(z)  134  constitute a speech synthesis filter (H) in code-excited linear prediction speech coding. It is within this synthesis filter (H) that the target signal reference shifting, performed in accordance with the present invention, provides for, among other things, the ability to reduce number of bits required to encode a speech signal and specifically the fractional pitch lag delay that is calculated during pitch prediction of the speech coding of the speech signal. 
     FIG. 2 is a system diagram illustrating another embodiment of a speech coding system  200  built in accordance with the present invention. From certain perspectives, the speech coding system  200  is a specific embodiment of the speech coding system  100  illustrated above in the FIG.  1 . While there are many similarities between the speech coding system  200  and the speech coding system  100 , it is reiterated that the speech coding system  200  is one specific embodiment of the speech coding system  100 , and that the speech coding system  100  includes not only the speech coding system  200 , but additional embodiments of speech coding systems as well. 
     A speech signal is input into the speech coding system  200  as shown by the reference numeral  210 . The speech signal is partitioned into a number of frames. If desired, each of the frames of the speech signal is further partitioned into a number of sub-frames. A given frame or sub-frame of the given frame is shown by the iteration ‘i’ associated with the reference numeral  214 . For the given frame or sub-frame, a particular excitation vector (C c(i) )  216  is selected from among a fixed codebook (C c )  212 . The selected excitation vector (C c(i) )  216 , chosen from among all of the excitation vectors contained within the fixed codebook (C c )  212  for the given frame or sub-frame of the speech signal, is scaled using a fixed gain (g c )  218 . After having undergone any required scaling (either amplification or reduction) by the fixed gain (g c )  218 , the now-scaled selected excitation vector (C c(i) )  216  is fed into a summing node  220 . An excitation signal  222  is fed into the signal path of the now-scaled selected excitation vector (C c(i) )  216  after the summing node  220 . A feedback path is provided wherein pitch prediction is performed in the block  224  as shown by z −LAG . 
     The output of this signal path, after having undergone the pitch prediction is performed in the block  224  as shown by z −LAG , is then scaled using an adaptive codebook gain (g p )  226 . After having undergone any required scaling (either amplification or reduction) by the adaptive codebook gain (g p )  226 , this signal path is then fed into the summing node  220 . The output of the summing node  220 , is fed into a synthesis filter (H(z))  229 . The synthesis filter (H(z))  229  itself contains, among other things, a linear prediction coding (LPC) synthesis filter (1/A(z))  228  and a perceptual weighting filter W(z)  234 . The output from the synthesis filter (H(z))  229  is fed to a summing node  230 . 
     In another signal path of the speech coding system  200 , the input speech signal  210  is fed into a perceptual weighting filter W(z)  234 . In addition, depending upon the particular frame or sub-frame of the speech signal that is being processed by the speech coding system  200  at the given time, as shown by the iteration ‘a i ’  210   a , linear prediction coding (LPC) analysis  210   b  is performed, and the parameters derived during the linear prediction coding (LPC) analysis  210   b  are also fed into the perceptual weighting filter W(z)  234 . The output of the perceptual weighting filter W(z)  234 , within this signal path, is fed into a summing mode  231 . 
     In addition, the output of a ringing filter  229   a  is also fed into the summing mode  231 . The ringing filter  229   a  is a ringing filter that contains memories from a previous sub-frame of the speech signal during its processing within the speech coding system  200 . The ringing filter  229   a  itself contains, among other things, a linear prediction coding (LPC) synthesis filter (1/A(z))  228  and a perceptual weighting filter W(z)  234 . Zero input is provided into the ringing filter  229   a , as its output is generated only from the ringing effect from memories from the previous sub-frame. If desired, the memories of multiple previous sub-frames are used within the ringing filter  229   a  in certain embodiments of the invention. That is to say, the memories from a single previous sub-frame are not used, but the memories from a predetermined number of previous sub-frames of the speech signal. Alternatively, the ringing effect of the ringing filter  229   a , with its zero input, is generated using multiple previous frames of the speech signal, and not simply previous sub frames. Varying numbers of previous portions of the speech signal are used to the ringing effect of the ringing filter  229   a  in other embodiments of the invention without departing from the scope and spirit of the speech coding system  200  illustrated in the FIG.  2 . 
     From certain perspectives, borrowing upon the linear transformation performed within the speech coding system  200 , the perceptual weighting filter W(z)  234 , the perceptual weighting filter W(z)  234  contained within the ringing filter  229   a , and the perceptual weighting filter W(z)  234  contained within the synthesis filter (H(z))  229  having zero memory are all a single perceptual weighting filter W(z). That is to say, each of the individual components of the perceptual weighting filter W(z), shown in the various portions of the speech coding system  200 , are all contained within a single integrated perceptual weighting filter W(z) within the speech coding system  200 . From this perspective and for illustrative purposes, the perceptual weighting filter W(z) is shown as being translated into each of the various components described above. However, each of the illustrated portions of the perceptual weighting filter W(z) could also be located on the other side of the summing nodes  230  and  231  without altering the performance of the speech coding system  200 . Again 
     After the signal paths of the ringing filter  229   a  and that of the perceptual weighting filter W(z)  234  are combined within the summing node  231 , their combined output is fed into the summing node  230 . In the interim, before the output of the summing node  231  is fed into the summing node  230 , a target signal (T g )  233  is added to the signal path. Subsequently, the output of the summing node  230  is combined with a coding error  232  that is also fed into the signal path that is the output of the summing node  230 . Finally, a weighted error  236  is generated by the speech coding system  200 . 
     FIG. 3 is a system diagram illustrating an embodiment of a speech signal processing system  300  built in accordance with the present invention. The speech signal processor  310  receives an unprocessed speech signal  320  and produces a processed speech signal  330 . 
     In certain embodiments of the invention, the speech signal processor  310  is processing circuitry that performs the loading of the unprocessed speech signal  320  into a memory from which selected portions of the unprocessed speech signal  320  are processed in various manners including a sequential manner. The processing circuitry possesses insufficient processing capability to handle the entirety of the unprocessed speech signal  320  at a single, given time. The processing circuitry may employ any method known in the art that transfers data from a memory for processing and returns the processed speech signal  330  to the memory. In other embodiments of the invention, the speech signal processor  310  is a system that converts a speech signal into encoded speech data. The encoded speech data is then used to generate a reproduced speech signal that is substantially perceptually indistinguishable from the speech signal using speech reproduction circuitry. In other embodiments of the invention, the speech signal processor  310  is a system that converts encoded speech data, represented as the unprocessed speech signal  320 , into decoded and reproduced speech data, represented as the processed speech signal  330 . In other embodiments of the invention, the speech signal processor  310  converts encoded speech data that is already in a form suitable for generating a reproduced speech signal that is substantially perceptually indistinguishable from the speech signal, yet additional processing is performed to improve the perceptual quality of the encoded speech data for reproduction. 
     The speech signal processing system  300  is, in some embodiments, the speech coding system  100 , or, alternatively, the speech coding system  200  as described in the FIGS. 1 and 2, respectively. The speech signal processor  310  operates to convert the unprocessed speech signal  320  into the processed speech signal  330 . The conversion performed by the speech signal processor  310  is viewed, in various embodiments of the invention, as taking place at any interface wherein data must be converted from one form to another, i.e. from speech data to coded speech data, from coded data to a reproduced speech signal, etc. 
     FIG. 4 is a system diagram illustrating an embodiment of a speech codec  400  built in accordance with the present invention that communicates across a communication link  410 . A speech signal  420  is input into an encoder circuitry  440  in which it is coded for data transmission via the communication link  410  to a decoder circuitry  450 . The decoder processing circuit  450  converts the coded data to generate a reproduced speech signal  430  that is substantially perceptually indistinguishable from the speech signal  420 . 
     In certain embodiments of the invention, the decoder circuitry  450  includes speech reproduction circuitry. Similarly, the encoder circuitry  440  includes selection circuitry that is operable to select from a plurality of coding modes. The communication link  410  is either a wireless or a wireline communication link without departing from the scope and spirit of the invention. Also, the communication link  410  is a network capable of handling the transmission of speech signals in other embodiments of the invention. Examples of such networks include, but are not limited to, internet and intra-net networks capable of handling such transmission. If desired, the encoder circuitry  440  identifies at least one perceptual characteristic of the speech signal and selects an appropriate speech signal coding scheme depending on the at least one perceptual characteristic. The speech codec  400  is, in one embodiment, a multi-rate speech codec that performs speech coding on the speech signal  420  using the encoder circuitry  440  and the decoder circuitry  450 . The speech codec  400  is operable to employ code-excited linear prediction speech coding as well as a modified form of code-excited linear prediction speech coding capable of performing target signal reference shifting in accordance with the present invention. 
     FIG. 5 is a system diagram illustrating an embodiment of a speech codec  500  that is a specific embodiment of the speech codec  400  illustrated above in FIG.  4 . The speech codec  500  communicates across a communication link  510 . A speech signal  520  is input into an encoder circuitry  540  in which it is coded for data transmission via the communication link  510  to a decoder circuitry  550 . The decoder processing circuit  550  converts the coded data to generate a reproduced speech signal  530  that is substantially perceptually indistinguishable from the speech signal  520 . 
     In the specific embodiment of the speech codec  500  illustrated in the FIG. 5, the encoder circuitry  540  contains, among other things, a reference shifting circuitry  542  that is used to perform modification of a target signal (T g ) that is generated during speech coding performed within the encoder circuitry  542 . The target signal (T g ) itself is calculated using a target signal (T g ) calculation circuitry  542   a  that is located within the reference shifting circuitry  542 . The target signal (T g  calculation circuitry  542   a  provides the calculated target signal (T g ) to a target signal (T g ) modification circuitry  542   aa . It is within the target signal (T g ) modification circuitry  542   aa  that the target signal reference shifting is performed in accordance with the present invention. In addition to calculating a modified target signal (T g ) is using the target signal (T g ) modification circuitry  542   aa , the reference shifting circuitry  542  employs an adaptive codebook gain (g p ) calculation circuitry  542   b  to calculate an adaptive codebook gain (g p ) that is used to perform speech coding in accordance with the present invention. In certain embodiments of the invention, the modified target signal (T g ) is used to perform the calculation of the adaptive codebook gain (g p ). That is to say, the modified target signal (T g ) is the ultimate target signal (T g ) that is used to select the adaptive codebook gain (g p ) during speech coding of a speech signal in accordance with speech coding performed using the speech codec  500  illustrated in the FIG.  5 . 
     In certain embodiments of the invention, the decoder circuitry  550  includes speech reproduction circuitry. Similarly, the encoder circuitry  540  includes selection circuitry that is operable to select from a plurality of coding modes. The communication link  510  is either a wireless or a wireline communication link without departing from the scope and spirit of the invention. Also, the communication link  510  is a network capable of handling the transmission of speech signals in other embodiments of the invention. Examples of such networks include, but are not limited to, internet and intra-net networks capable of handling such transmission. If desired, the encoder circuitry  540  identifies at least one perceptual characteristic of the speech signal and selects an appropriate speech signal coding scheme depending on the at least one perceptual characteristic. The speech codec  500  is, in one embodiment, a multi-rate speech codec that performs speech coding on the speech signal  520  using the encoder circuitry  540  and the decoder circuitry  550 . The speech codec  500  is operable to employ code-excited linear prediction speech coding as well as a modified form of code-excited linear prediction speech coding capable of performing target signal reference shifting in accordance with the present invention. 
     FIG. 6 is a functional block diagram illustrating a speech coding method  600  performed in accordance with the present invention. In a block  610 , a target signal (T g ) is calculated. Subsequently, in a block  620 , the target signal (T g ) that is calculated in the block  610  is modified to attain a modified target signal (T g ′). After the target signal (T g ) has been modified to achieve the modified target signal (T g ′) in the block  620 , an adaptive codebook gain (g p ) is calculated in a block  630  using the modified target signal (T g ′) that is calculated in the block  620 . 
     The speech coding method  600  performs target signal reference shifting in accordance with the present invention by modifying the target signal (T g ) calculated in the block  610  to generate the modified target signal (T g ′) calculated in the block  620 . The speech coding method  600  provides for a way to decrease the bit-rate necessitated for coding the fractional pitch lag delay required during the calculation of pitch prediction integrated circuit code-excited linear prediction speech coding systems. In certain embodiments of the invention, the modified target signal (T g ′) calculated in the block  620  does not provide any substantially perceptually distinguishable difference from the target signal (T g ) calculated in the block  610 . 
     FIG. 7 is a functional block diagram illustrating a speech coding method  700  that is a specific embodiment of the speech coding method  600  as shown above in FIG.  6 . In a block  710 , a target signal (T g ) is calculated for either a frame or a sub-frame. As a speech signal is provided to be coded using the method  700 , the speech signal is partitioned into a number of frames. The frames of the speech signal are further partitioned into a number of sub-frames. The calculation of the target signal (T g ) is performed either on a frame of the speech signal or on a sub-frame of a frame of the speech signal without departing from the scope of the present invention. 
     Subsequently, in a block  720 , for a given pitch lag (LAG), an adaptive codebook excitation (C p ) is filtered and a speech synthesis filter (H) is defined. The combination of both the generation of the adaptive codebook excitation (C p ) and the speech synthesis filter (H) provides for the product of (C p H) as required in accordance with code-excited linear prediction speech coding. Then, in a block  730 , the target signal (T g ) calculated in the block  710  to generate the modified target signal (T g ′). In the embodiment shown in the speech coding method  700  of FIG. 7, the modified target signal (T g ′) is generated by finding the value of target signal (T g ) that maximizes the correlation of the dot product of the target signal (T g ) found originally in the block  710  and the product (C p H) as found above in the block  720 . The maximization of the dot product between the target signal (T g ) and the product (C p H) is shown as Max[(T g ·C p H) 2 ], or alternatively as the maximization of the normalized dot product between the target signal (T g ) and the product (C p H) that is shown as Max[(T g ·C p H) 2 /∥C p H∥ 2 ] in the block  730 . For clarity, the calculation of the maximization of the dot product between the target signal (T g ) and the product (C p H) is shown below. 
       T   g ′←Max{( T   g   ·C   p   H ) 2 } 
     From this, the product of an adaptive codebook contribution (C p ) and subsequently the contribution provided by a speech synthesis filter (H), and the product of those two elements, namely, C p H is then defined. Alternatively, if the maximization of the normalized dot product between the target signal (T g ) and the product (C p H) is desired, it is shown below. 
     
       
           T   g ′←Max ( T   g   ·C   p   H ) 2   
       
     
     
       
         ∥ C   p   H∥   2   
       
     
     For each of the above situations, the target signal (T g ) is shown on the right hand side of the relation, and the modified target signal (T g ′) is provided on the left hand side of the relation. 
     Finally, in the block  740 , an adaptive codebook gain (g p ) is calculated using the modified target signal (T g ′) that is calculated in the block  730 . Specifically, the adaptive codebook gain (g p ) calculated in the block  740  is found by finding the adaptive codebook gain (g p ) that minimizes the equation of Min[(T g ′−g p C p H) 2 ]. Once the modified target signal (T g ′) is found in the block  730 , that modified target signal (T g ′) is used to find the specific adaptive codebook gain (g p ) in the block  740  for the speech coding method  700 . 
     Lastly, and using the modified target signal (T g ′), it is possible to solve for the adaptive codebook gain (g p ) as shown below. 
     
       
           g   p ←Min [( T   g   ′−g   p   C   p   H)   2 ] 
       
     
     In view of the above detailed description of the present invention and associated drawings, other modifications and variations will now become apparent to those skilled in the art. It should also be apparent that such other modifications and variations may be effected without departing from the spirit and scope of the present invention.