Patent Abstract:
A method and apparatus for reducing the complexity of linear prediction analysis-by-synthesis (LPAS) speech coders. The speech coder includes a multi-tap pitch predictor having various parameters and utilizing an adaptive codebook subdivided into at least a first vector codebook and a second vector codebook. The pitch predictor removes certain redundancies in a subject speech signal and vector quantizes the pitch predictor parameters. Further included is a source excitation (fixed) codebook that indicates pulses in the subject speech signal by deriving corresponding vector values. Serial optimization of the adaptive codebook first and then the fixed codebook produces a low complexity LPAS speech coder of the present invention.

Full Description:
RELATED APPLICATIONS 
   This application is a Divisional of U.S. application Ser. No. 09/991,763, filed on Nov. 21, 2001 now U.S. Pat. No. 6,865,530, which is a Continuation of U.S. application Ser. No. 09/455,063, filed on Dec. 6, 1999, now U.S. Pat. No. 6,393,390, which is a Continuation of U.S. application Ser. No. 09/130,688, filed Aug. 6, 1998, now U.S. Pat. No. 6,014,618, the entire contents of which are incorporated herein by reference. 

   FIELD OF INVENTION 
   The present invention relates to the improved method and system for digital encoding of speech signals, more particularly to Linear Predictive Analysis-by-Synthesis (LPAS) based speech coding. 
   BACKGROUND OF THE INVENTION 
   LPAS coders have given new dimension to medium-bit rate (8–16 Kbps) and low-bit rate (2–8 Kbps) speech coding research. Various forms of LPAS coders are being used in applications like secure telephones, cellular phones, answering machines, voice mail, digital memo recorders, etc. The reason is that LPAS coders exhibit good speech quality at low bit rates. LPAS coders are based on a speech production model  39  (illustrated in  FIG. 1 ) and fall into a category between waveform coders and parametric coders (Vocoder); hence they are referred to as hybrid coders. 
   Referring to  FIG. 1 , the speech production model  39  parallels basic human speech activity and starts with the excitation source  41  (i.e., the breathing of air in the lungs). Next the working amount of air is vibrated through a vocal chord  43 . Lastly, the resulting pulsed vibrations travel through the vocal tract  45  (from vocal chords to voice box) and produce audible sound waves, i.e., speech  47 . 
   Correspondingly, there are three major components in LPAS coders. These are (i) a short-term synthesis filter  49 , (ii) a long-term synthesis filter  51 , and (iii) an excitation codebook  53 . The short-term synthesis filter includes a short-term predictor in its feed-back loop. The short-term synthesis filter  49  models the short-term spectrum of a subject speech signal at the vocal tract stage  45 . The short-term predictor of  49  is used for removing the near-sample redundancies (due to the resonance produced by the vocal tract  45 ) from the speech signal. The long-term synthesis filter  51  employs an adaptive codebook  55  or pitch predictor in its feedback loop. The pitch predictor  55  is used for removing far-sample redundancies (due to pitch periodicity produced by a vibrating vocal chord  43 ) in the speech signal. The source excitation  41  is modeled by a so-called “fixed codebook” (the excitation code book)  53 . 
   In turn, the parameter set of a conventional LPAS based coder consists of short-term parameters (short-term predictor), long-term parameters and fixed codebook  53  parameters. Typically short-term parameters are estimated using standard 10–12th order LPC (Linear predictive coding) analysis. 
   The foregoing parameter sets are encoded into a bit-stream for transmission or storage. Usually, short-term parameters are updated on a frame-by-frame basis (every 20–30 msec or 160–240 samples) and long-term and fixed codebook parameters are updated on a subframe basis (every 5–7.5 msec or 40–60 samples). Ultimately, a decoder (not shown) receives the encoded parameter sets, appropriately decodes them and digitally reproduces the subject speech signal (audible speech)  47 . 
   Most of the state-of-the art LPAS coders differ in fixed codebook  53  implementation and pitch predictor or adaptive codebook implementation  55 . Examples of LPAS coders are Code Excited Linear Predictive (CELP) coder, Multi-Pulse Excited Linear Predictive (MPLPC) coder, Regular Pulse Linear Predictive (RPLPC) coder, Algebraic CELP (ACELP) coder, etc. Further, the parameters of the pitch predictor or adaptive codebook  55  and fixed codebook  53  are typically optimized in a closed-loop using an analysis-by-synthesis method with perceptually-weighted minimum (mean squared) error criterion. See Manfred R. Schroeder and B. S. Atal, “Code-Excited Linear Prediction (CELP): High Quality Speech at Very Low Bit Rates,”  IEEE Proceedings of the International Conference on Acoustics, Speech and Signal Processing , Tampa, Fla., pp. 937–940, 1985. 
   The major attributes of speech-coders are: 
   1. Speech Quality 
   2. Bit-rate 
   3. Time and Space complexity 
   4. Delay 
   Due to the closed-loop parameter optimization of the pitch-predictor  55  and fixed codebook  53 , the complexity of the LPAS coder is enormously high as compared to a waveform coder. The LPAS coder produces considerably good speech quality around 8–16 kbps. Further improvement in the speech quality of LPAS based coders can be obtained by using sophisticated algorithms, one of which is the multi-tap pitch predictor (MTPP). Increasing the number of taps in the pitch predictor increases the prediction gain, hence improving the coding efficiency. On the other hand, estimating and quantizing MTPP parameters increases the computational complexity and memory requirements of the coder. 
   Another very computationally expensive algorithm in an LPAS based coder is the fixed codebook search. This is due to the analysis-by-synthesis based parameter optimization procedure. 
   Today, speech coders are often implemented on Digital Signal Processors (DSP). The cost of a DSP is governed by the utilization of processor resources (MIPS/RAM/ROM) required by the speech coder. 
   SUMMARY OF THE INVENTION 
   One object of the present invention is to provide a method for reducing the computational complexity and memory requirements (MIPS/RAM/ROM) of an LPAS coder while maintaining the speech quality. This reduction in complexity allows a high quality LPAS coder to run in real-time on an inexpensive general purpose fixed point DSP or other similar digital processor. 
   Accordingly, the present invention method provides (i) an LPAS speech encoder reduced in computational complexity and memory requirements, and (ii) a method for reducing the computational complexity and memory requirements of an LPAS speech encoder, and in particular a multi-tap pitch predictor and the source excitation codebook in such an encoder. The invention employs fast structured product code vector quantization (PCVQ) for quantizing the parameters of the multi-tap pitch predictor within the analysis-by-synthesis search loop. The present invention also provides a fast procedure for searching the best code-vector in the fixed-code book. To achieve this, the fixed codebook is preferably formed of ternary values (1,−1,0). 
   In a preferred embodiment, the multi-tap pitch predictor has a first vector codebook and a second (or more) vector codebook. The invention method sequentially searches the first and second vector codebooks. 
   Further, the invention includes forming the source excitation codebook by using non-contiguous positions for each pulse. 

   
     BRIEF DESCRIPTION OF THE DRAWINGS 
     The foregoing and other objects, features and advantages of the invention will be apparent from the following more particular description of preferred embodiments of the invention, as illustrated in the accompanying drawings in which like reference characters refer to the same parts throughout the different views. The drawings are not necessarily to scale, emphasis instead being placed upon illustrating the principles of the invention. 
       FIG. 1  is a schematic illustration of the speech production model on which LPAS coders are based. 
       FIGS. 2   a  and  2   b  are block diagrams of an LPAS speech coder with closed loop optimization. 
       FIG. 3  is a block diagram of an LPAS speech encoder embodying the present invention. 
       FIG. 4  is a schematic diagram of a multi-tap pitch predictor with so-called conventional vector quantization. 
       FIG. 5  is a schematic illustration of a multi-tap pitch predictor with product code vector quantized parameters of the present invention. 
       FIGS. 6 and 7  are schematic diagrams illustrating fixed codebook vectors of the present invention, formed of blocks corresponding to pulses of the target speech signal. 
   

   DETAILED DESCRIPTION OF THE INVENTION 
   Generally illustrated in  FIG. 2   a  is an LPAS coder with closed loop optimization. Typically, the fixed codebook  61  holds over 1024 parameter values, while the adaptive codebook  65  holds just over 128 or so values. Different combinations of those values are adjusted by a term 
           1     A   ⁡     (   z   )             
(i.e., the short term synthesis filter  63 ) to produce synthesized signal  69 . The resulting synthesized signal  69  is compared to (i.e., subtracted from) the original speech signal  71  to produce an error signal. This error term is adjusted through perceptual weighting filter  62 , i.e.,
 
               A   ⁡     (   z   )         A   ⁡     (     z   /   γ     )         ,         
and fed back into the decision making process for choosing values from the fixed codebook  61  and the adaptive codebook  65 .
 
   Another way to state the closed loop error adjustment of  FIG. 2   a  is shown in  FIG. 2   b . Different combinations of adaptive codebook  65  and fixed codebook  61  are adjusted by weighted synthesis filter  64  to produce weighted synthesis speech signal  68 . The original speech signal is adjusted by perceptual weighted filter  62  to produce weighted speech signal  70 . The weighted synthesis signal  68  is compared to weighted speech signal  70  to produce an error signal. This error signal is fed back into the decision making process for choosing values from the fixed codebook  61  and adaptive codebook  65 . 
   In order to minimize the error, each of the possible combinations of the fixed codebook  61  and adaptive codebook  65  values is considered. Where, in the preferred embodiment, the fixed codebook  61  holds values in the range 0 through 1024, and the adaptive codebook  65  values range from 20 to about 146, such error minimization is a very computationally complex problem. Thus, Applicants reduce the complexity and simplify the problem by sequentially optimizing the fixed codebook  61  and adaptive codebook  65  as illustrated in  FIG. 3 . 
   In particular, Applicants minimize the error and optimize the adaptive codebook working value first, and then, treating the resulting codebook value as a constant, minimize the error and optimize the fixed codebook value. This is illustrated in  FIG. 3  as two stages  77 , 79  of processing. In a first (upper) stage  77 , there is a closed loop optimization of the adaptive codebook  11 . The value output from the adaptive codebook  11  is multiplied by the weighted synthesis filter  17  and produces a first working synthesized signal  21 . The error between this working synthesized signal  21  and the weighted original speech signal S tv  is determined. The determined error is subsequently minimized via a feedback loop  37  adjusting the adaptive codebook  11  output. Once the error has been minimized and an optimum adaptive contribution is estimated, the first processing stage  77  outputs an adjusted target speech signal S′ tv . 
   The second processing stage  79  uses the new/adjusted target speech signal S′ tv , for estimating the optimum fixed codebook  27  contribution. 
   In the preferred embodiment, multi-tap pitch predictor coding is employed to efficiently search the adaptive codebook  11 , as illustrated in  FIGS. 4 and 5 . In that case, the goal of processing stage  77  ( FIG. 3 ) becomes the task of finding the optimum adaptive codebook  11  contribution. 
   Multi-tap Pitch Predictor (MTPP) Coding: 
   The general transfer function of the MTPP with delay M and predictor coefficient&#39;s g k  is given as 
             P   ⁡     (   z   )       =     1   -       ∑     k   =   0       p   -   1       ⁢       g   k     ⁢           ⁢     z     -     (     M   -     [     p   /   2     ]     +   k     )                     
For a single-tap pitch predictor p=1. The speech quality, complexity and bit-rate are a function of p. Higher values of p result in higher complexity, bit rate, and better speech quality. Single-tap or three-tap pitch predictors are widely used in LPAS coder design. Higher-tap (p&gt;3) pitch predictors give better performance at the cost of increased complexity and bit-rate.
 
   The bit-rate requirement for higher-tap pitch predictors can be reduced by delta-pitch coding and vector quantizing the predictor coefficients. Although use of vector quantization adds more complexity in the pitch predictor coding, the vector quantization (VQ) of the multiple coefficients g k  of the MTPP is necessary to reduce the bits required in encoding the coefficients. One such vector quantization is disclosed in D. Veeneman &amp; B. Mazor, “Efficient Multi-Tap Pitch Predictor for Stochastic Coding,”  Speech and Audio Coding for Wireless and Network Applications , Kluwner Academic Publisher, Boston, Mass., pp. 225–229. 
   In addition, by integrating the VQ search process in the closed-loop optimization process  37  of  FIG. 3  (as indicated by  37   a  in  FIG. 4 ), the performance of the VQ is improved. Hence perceptually weighted mean squared error criterion is used as the distortion measure in the VQ search procedure. One example of such weighted mean square error criterion is found in J. H. Chen, “Toll-Quality 16 kbps CELP Speech Coding with Very Low Complexity,”  Proceedings of the International Conference on Acoustics, Speech and Signal Processing , pp. 9–12, 1995. Others are suitable. Moreover, for better coding efficiency, the lag M and coefficient&#39;s gk are jointly optimized. The following explains the procedure for the case of a 5-tap pitch predictor  15  as illustrated in  FIG. 4 . The method of  FIG. 4  is referred to as “Conventional VQ”. 
   Let r(n) be the contribution from the adaptive codebook  11  or pitch predictor  13 , and let s tv (n) be the target vector and h(n) be the impulse response of the weighted synthesis filter  17 . The error e(n) between the synthesized signal  21  and target, assuming zero contribution from a stochastic codebook  11  and 5-tap pitch predictor  13 , is given as 
             e   ⁡     (   n   )       =         s   tv     ⁡     (   n   )       -       ∑     j   =   0       j   =   n       ⁢       h   ⁡     (     n   -   j     )       ⁢           ⁢       ∑     k   =   0       k   =   4       ⁢       g   k     ⁢     r   ⁡     (     n   -     (     M   -   2   +   k     )       )                       
In matrix notation with vector length equal to subframe length, the equation becomes
   e=s   tv   −g   0   Hr   0   −g   1   Hr   1   −g   2   Hr   2   −g   3   Hr   3   −g   4   Hr   4   
where H is impulse response matrix of weighted synthesis filter  17 . The total mean squared error is given by
   E=e   T   e=s    tv   T   s   tv −2 g   0   s   tv   T   Hr   0 −2 g   1   s   tv   T   Hr   1 −2 g   2   s   tv   T   Hr   2 −2 g   3   s   tv   T   Hr   3 −2 g   4 S tv   T   Hr   4   +g   0   2   r   0   T   H   T   Hr   0   h   +g   1   2   r   1   T   H   T   Hr   1   h   +g   2   2   r   T   H   T   Hr   2   h   +g   3   2   r   3   T   H   T   Hr   3   h    +g   4   2   r   4   T   H   T   Hr   4   h +2 g   0   g   1   r   0   T   H   T   Hr   1   h +2 g   0   g   2   r   0   T   H   T   Hr   2   h +2 g   0   g   3   r   0   T   H   T   Hr   3   h  +2 g   0   g   4   r   0   T   H   T   Hr   4   h +2 g   1   g   2   r   1   T   H   T   Hr   2   h +2 g   1   g   3   r   1   T   H   T   Hr   3   h +2 g   1   g   4   r   1   T   h   T   Hr   4   h  +2 g   2   g   3   r   2   T   H   T   Hr   3   h +2 g   2   g   4   r   2   T   H   T   Hr   4   h +2 g   3   g   4   r   3   T   H   T   Hr   4   h    Let  g=[g   0   ,g   1   ,g   2   ,g   3   ,g   4 , −0.5 g   0   2 , −0.5  g   1   2 , −0.5 g   2   2 , −0.5 g   3   2 , 0.5 g   4   2   , −g   0   g   1   , −g   0   g   2   , −g   0   g   3   , −g   0   g   4   , −g   1   g   2   , −g   1   g   3   , −g   1   g   4   , −g   2   g   3   , −g   2   g   4   , −g   3   g   4   ]   Let c M   =[s   tv   T Hr 0   , s   tv   T Hr 1   , s   tv   T   Hr   2   , s   tv   T   Hr   3   , s   tv   T   Hr   4   , r   0   T   H   T   Hr   0   h   , r   1   T   H   T   Hr   1   h   , r   2   T   H   T   Hr   2   h   , r   3   T   H   T   Hr   3   h   , r   4   T   H   T   Hr   4   h   , r   0   T   H   T   Hr   1   h   , r   0   T   H   T   Hr   2   h   , r   0   T   H   T   Hr   3   h   , r   0   T   H   T   Hr   4   h   , r   1   T   H   T   Hr   2   h   , r   1   T   H   T   Hr   3   h   , r   1   T   H   T   Hr   4   h   , r   2   T   H   T   Hr   3   h   , r   2   T   H   T   Hr   4   h   , r   3   T   H   T   Hr   4   h   ]   E=e T   e=s   tv   T   s   tv −2 c   M   T   g    
   The g vector may come from a stored codebook  29  of size N and dimension  20  (in the case of a 5-tap predictor). For each entry (vector record) of the codebook  29 , the first five elements of the codebook entry (record) correspond to five predictor coefficients and the remaining 15 elements are stored accordingly based on the first five elements, to expedite the search procedure. The dimension of the g vector is T+(T*(T−1)/2), where T is the number of taps. Hence the search for the best vector from the codebook  29  may be described by the following equation as a function of M and index i.
 
 E ( M,i )= e   T   e=s   tv   T   s   tv −2 c   M   T   g   i 
 
where M olp −1≦M≦M olp −2, and i=0 . . . N.
 
   Minimizing E(M,i) is equivalent to maximizing c M   T g i , the inner product of two 20 dimensional vectors. The best combination (M,i) which maximize c M   T g i  is the optimum index and pitch value. Mathematically,  (M,i)  max{c M   T g i } 
   where M olp −1≦M≦M olp −2, and i=0 . . . N. 
   For an 8-bit VQ, the complexity reduction is a trade-off between computational complexity and memory (storage) requirement. See the inner 2 columns in Table 2. Both sets of numbers in the first three rows/VQ methods are high for LPAS coders in low cost applications such as digital answering machines. 
   The storage space problem is solved by Product Code VQ (PCVQ) design of S. Wang, E. Paksoy and A. Gersho, “Product Code Vector Quantization of LPC Parameters,”  Speech and Audio Coding for Wireless and Network Applications , Kluwner Academic Publisher, Boston, Mass. A copy of this reference is attached and incorporated herein by reference for purposes of disclosing the overall product code vector quantization (PCVQ) technique. Wang et al used the PCVQ technique to quantize the Linear Predictive Coding (LPC) parameters of the short term synthesis filter in LPAS coders. Applicants in the present invention apply the PCVQ technique to quantize the pitch predictor (adaptive codebook)  55  parameters in the long term synthesis filter  51  ( FIG. 1 ) in LPAS coders. Briefly, the g vector is divided into two subvectors g 1  and g 2 . The elements of g 1  and g 2  come from two separate codebooks C 1  and C 2 . Each possible combination of g 1  and g 2  to make g is searched in analysis-by-synthesis fashion, for optimum performance.  FIG. 5  is a graphical illustration of this method. 
   In particular, codebooks C 1  and C 2  are depicted at  31  and  33 , respectively in  FIG. 5 . Codebook C 1  (at  31 ) provides subvector g i  while codebook C 2  (at  33 ) provides subvector g j . Further, codebook C 2  (at  33 ) contains elements corresponding to g 0  and g 4 , while codebook C 1  (at  31 ) contains elements corresponding to g 1 , g 2  and g 3 . Each possible combination of subvectors g j  and g i  to make a combined g vector for the pitch predictor  35  is considered (searched) for optimum performance. The VQ search process is integrated in the closed loop optimization  37  ( FIG. 3 ) as indicated by  37   b  in  FIG. 5 . As such, lag M and coefficients g i  and g j  are jointly optimized. Preferably, a perceptually weighted mean square error criterion is used as the distortion measure in the VQ search procedure. Hence the best combination of subvectors g i  and g j  from codebooks C 1  and C 2  may be described as a function of M and indices i,j as the best combination of (M,i,j) which maximizes C M   T g ij  (the optimum indices and pitch values as further discussed below). 
   Specifically, g ij =g 1   i +g 2   j +g 12   ij 
 
max{C M   T g ij }(M,ij)
 
where M olp −1≦M≦M olp −2, i=0 . . . N 1 , and j=0 . . . N 2 . T is the number of taps. N=N 1 *N 2 . N 1  and N 2  are, respectively, the size of codebooks C 1  and C 2 .
 
   Where C 1  contains elements corresponding to g 1 , g 2 , g 3 , then g 1   i  is a 9-dimensional vector as follows.
 
 g 1 i =[0 ,g   1i   ,g   2i   ,g   3i ,0,0,−0.5g 1i   2 ,0.5 g   2i   2 ,−0.5 g   3i   2 , 0,0,0,0,0 ,−g   1i   g   2i   ,−g   1i   g   3i ,0 ,−g   2i   g   3i ,0,0]
 
Let the size of C 1  codebook be N 1 =32. The storage requirement for codebook C 1  is S 1 =9*32=288 words.
 
   Where C 2  contains elements corresponding to g 0 ,g 4 , then g 2   j  is a 5 dimensional vector as shown in the following equation.
 
 g 2 j   =[g   0j 0,0,0, g 4j ,−0.5 g   0j   2 ,0,0,0,−0.5 g   4j   2 0,0,0 ,−g   0j   g   4j ,0,0,0,0,0,0]
 
Let the size of C 2  codebook be N 2 =8. The storage requirement for codebook C 2  is S 2 =5*8=40 words.
 
   Thus, the total storage space for both of the codebooks=288+40=328 words. This method also requires 6*4*256=6144 multiplications for generating the rest of the elements of g 12   ij  which are not stored, where
 
 g 12 ij =[0,0,0,0,0,0,0,0,0,0, −g 0j   g   1i   ,−g   0j   g   2i   ,−g   0j   g   3i ,0,0,0, −g   1i   g   4j ,0 ,−g   3i   g   4j ]
 
   Hence a savings of about 4800 words is obtained by computing 6144 multiplication&#39;s per subframe (as compared to the Fast D-dimension VQ method in Table 2). The performance of PCVQ is improved by designing the multiple C 2  codebook based on the vector space of the C 1  codebook. A slight increase in storage space and complexity is required with that improvement. The overall method is referred to in the Tables as “Full Search PCVQ”. 
   Applicants have discovered that further savings in computational complexity and storage requirement is achieved by sequentially selecting the indices of C 1  and C 2 , such that the search is performed in two stages. For further details see J. Patel, “Low Complexity VQ for Multi-tap Pitch Predictor Coding,” in  IEEE Proceedings of the International Conference on Acoustics, Speech and Signal Processing , pp. 763–766, 1997, herein incorporated by reference (copy attached). 
   Specifically, 
   Stage 1: For all candidates of M, the best index i=I[M] from codebook C 1  is determined using the perceptually weighted mean square error distortion criterion previously mentioned. 
   For M olp −1≦M≦M olp −2 
                         I   [     i     ⁢   M     ]     =     max   ⁢     {       c   M   T     ⁢           ⁢     g1   i       }                           i   =     0   ⁢           ⁢   …   ⁢           ⁢   N1                 
Stage 2: The best combination M, I[M] and index j from codebook C 2  is selected using the same distortion criterion as in Stage 1 above.
   g   I[M]j   =g 1     I[M]   =g 2     j   =g 12     I[M]j   max {c M   T g I [M] J }(M,I[M]j) 
where M olp −1≦M≦M olp −2, and j=0 . . . N 2 .
 
   This (the invention) method is referred to as “Sequential PCVQ”. In this method c M   T g is evaluated (32*4)+(8*4)=160 times while in “Full Search PCVQ”, c M   T g is evaluated 1024 times. This savings in scalar product (c M   T g) computations may be utilized in computing the last 15 elements of g when required. The storage requirement for this invention method is only 112 words. 
   Comparisons: 
   A comparison is made among all the different vector quantization techniques described above. The total multiplication and storage space are used in the comparison. 
   Let T=Taps of pitch predictor=T 1 +T 2 ,
     D=Length of g vector=T+T x ,   T x =Length of extra vector=T(T+1)/2   N=size of g vector VQ,   D 1 =Length of g 1  vector=T 1 +T 1   x ,   T 1   x =T 1 (T 1 + 1 )/2,   N 1 =size of g 1  vector VQ,   D 2 =Length of g 2  vector=T 2 +T 2   x ,   T 2   x =T 2 (T 2 +1)/2,   N 2 =size of g 2  vector VQ,   D 12 =size of g 12  vector=T x −T 1   x −T 2   x ,   R=Pitch search range,   N=N 1 *N 2 .   

   
     
       
             
           
             
             
             
           
         
             
               TABLE 1 
             
           
           
             
                 
             
             
               Complexity of MTPP 
             
           
        
         
             
                 
               Total 
               Storage 
             
             
               VQ Method 
               Multiplication 
               Requirement 
             
             
                 
             
             
               Fast D-dimension 
               N*R*D 
               N*D 
             
             
               conventional VQ 
             
             
               Low Memory D- 
               N*R*(D + T x )  
               N*T 
             
             
               dimension 
             
             
               conventional VQ 
             
             
               Full Search Product 
               N*R*(D + D12) 
               (N1*D1) + (N2*D2) 
             
             
               Code VQ 
             
             
               Sequential Search Product Code 
               N1*R*(D1 + T1 X ) + 
               (N1*T1) + (N2*T2) 
             
             
               VQ 
               N2*R*(D2 + T2 x )  
             
             
                 
             
           
        
       
     
   
   For the 5-tap pitch predictor case,
     T=5,N=256, T 1 =3, T 2 =2,N 1 =32,N 2 =8,R=4,   D=20, D 1 =9, D 2 =5, D 12 =6, T x =15, T 1   x =6, T 2   x =3.   

   All four of the methods were used in a CELP coder. The rightmost column of Table 2 shows the segmental signal-to-noise ratio (SNR) comparison of speech produced by each VQ method. 
   
     
       
             
           
             
             
             
             
           
         
             
               TABLE 2 
             
           
           
             
                 
             
             
               5-Tap Pitch Predictor Complexity and Performance 
             
           
        
         
             
                 
               Total 
               Storage 
               Seg. SNR 
             
             
               VQ Method 
               Multiplication 
               Space in Words 
               dB 
             
             
                 
             
             
               Fast D-dimension VQ 
               20480 
               5120 
               6.83 
             
             
               Low Memory D- 
               20480 + 15360 
               1280 
               6.83 
             
             
               dimension VQ 
             
             
               Full Search Product 
               20480 + 6144  
               288 + 40 
               6.72 
             
             
               Code VQ 
             
             
               Sequential Search 
               1920 + 256 + 6144 
                96 + 16 
               6.59 
             
             
               Product Code VQ 
             
             
                 
             
           
        
       
     
   
   Referring back to  FIG. 3 , after optimizing the adaptive codebook  11  search according to the foregoing VQ techniques illustrated in  FIG. 5 , first processing stage  77  is completed and the second processing stage  79  follows. In the second processing stage  79 , the fixed codebook  27  search is performed. Search time and complexity is dependent on the design of the fixed codebook  27 . To process each value in the fixed codebook  27  would be costly in time and computational complexity. Thus the present invention provides a fixed codebook that holds or stores ternary vectors (−1,0,1) i.e., vectors formed of the possible permutations of 1,0,−1, as illustrated in  FIGS. 6 and 7  and discussed next. 
   In the preferred embodiment, for each subframe, target speech signal S′ tv  is backward filtered  18  through the synthesis filter ( FIG. 3 ) to produce working speech signal S bf  as follows. 
                     S   bf     ⁡     (   j   )       =       ∑     n   =   j       n   =     NSF   -   1         ⁢         S   tv   ′     ⁡     (   n   )       ⁢           ⁢   h   ⁢           ⁢     (     n   -   j     )                             0   ≤   j   ≤     NSF   -   1                 
where, NSF is the sub-frame size and
 
   
     
       
         
           
             h 
             ⁡ 
             
               ( 
               n 
               ) 
             
           
           = 
           
             
               1 
               
                 A 
                 ⁡ 
                 
                   ( 
                   
                     z 
                     / 
                     γ 
                   
                   ) 
                 
               
             
             . 
           
         
       
     
   
   Next, the working speech signal S bf  is partitioned into N p  blocks Blk 1 , Blk 2  . . . Blk N p  (overlapping or non-overlapping, see  FIG. 6 ). The best fixed codebook contribution (excitation vector v) is derived from the working speech signal S bf . Each corresponding block in the excitation vector v(n) has a single or no pulse. The position P n  and sign S n  of the peak sample (i.e., corresponding pulse) for each block Blk 1 , . . . Blk N p  is determined. Sign is indicated using +1 for positive, −1 for negative, and 0. 
   Further, let S bf max be the maximum absolute sample in working speech signal S bf . Each pulse is tested for validity by comparing the pulse to the maximum pulse magnitude (absolute value thereof) in the working speech signal S bf . In the preferred embodiment, if the signed pulse of a subject block is less than about half the maximum pulse magnitude, then there is no valid pulse for that block. Thus, sign S n  for that block is assigned the value 0. 
                                                                                             That is,                For n = 1 to N p                  If S bf (P n )*S n  &lt; μ*S bf max                S n  = 0                EndIf                EndFor                        
The typical range for μ is 0.4–0.6.
 
   The foregoing pulse positions P n  and signs S n  of the corresponding pulses for the blocks Blk ( FIG. 6 ) of a fixed codebook vector, form position vector P n  and sign vector S n  respectively. In the preferred embodiment, only certain positions in working speech signal S bf  are considered, in order to find a peak/subject pulse in each block Blk. It is the sign vector S n  with elements adjusted to reflect validity of pulses of the blocks Blk of a codebook vector which ultimately defines the codebook vector for the present invention optimized fixed codebook  27  ( FIG. 3 ) contribution. 
   In the example illustrated in  FIG. 7 , the working speech signal (or subframe vector) S bf (n) is partitioned into four non-overlapping blocks  83   a ,  83   b ,  83   c  and  83   d . Blocks  75   a ,  75   b ,  75   c ,  75   d  of a codebook vector  81  correspond to blocks  83   a ,  83   b ,  83   c ,  83   d  of working speech signal S bf  (i.e., backward filtered target signal S′ tv ). The pulse or sample peak of block  83   a  is at position  2 , for example, where only positions  0 ,  2 ,  4 ,  6 ,  8 ,  10  and  12  are considered. Thus, P 1 =2 for the first block  75   a . Corresponding sign of the subject pulse is positive; so S 1 =1. Block  83   b  has a sample peak (corresponding negative pulse) at say for example position  18 , where positions  14 ,  16 ,  18 ,  20 ,  22 ,  24  and  26  are considered. So the corresponding block  75   b  (the second block of codebook vector  81 ) has P 2 =18 and sign S 2 =−1. Likewise, block  83   c  (correlated to third codebook vector block  75   c ) has a sample positive peak/pulse at position  32 , for example, where only every other position is considered in that block  83   c . Thus, P 3=32  and S 3 =1. It is noted that this block  83   c  also contains S bf max, the working speech signal pulse with maximum magnitude, i.e., absolute value, but at a position not considered for purposes of setting P n . 
   Lastly, block  83   d  and corresponding block  75   d  have a sample positive peak/pulse at position  46  for example. In that block  83   d , only even positions between  42  and  52  are considered. As such, P 4 =46 and S 4 =1. 
   The foregoing sample peaks (including position and sign) are further illustrated in the graph line  87 , just below the waveform illustration of working speech signal S bf  in  FIG. 7 . In that graph line  87 , a single vertical scaled arrow indication per block  83 , 75  is illustrated. That is, for corresponding block  83   a  and block  75   a , there is a positive vertical arrow  85   a  close to maximum height (e.g., 2.5) at the position labeled  2 . The height or length of the arrow is indicative of magnitude (=2.5) of the corresponding pulse/sample peak. 
   For block  83   b  and corresponding block  75   b , there is a graphical negative directed arrow  85   b  at position  18 . The magnitude (i.e., length=2) of the arrow  85   b  is similar to that of arrow  85   a  but is in the negative (downward) direction as dictated by the subject block  83   b  pulse. 
   For block  83   c  and corresponding block  75   c , there is graphically shown along graph line  87  an arrow  85   c  at position  32 . The length (=2.5) of the arrow is a function of the magnitude (=2.5) of the corresponding sample peak/pulse. The positive (upward) direction of arrow  85   c  is indicative of the corresponding positive sample peak/pulse. 
   Lastly, there is illustrated a short (length=0.5) positive (upward) directed arrow  85   d  at position  46 . This arrow  85   d  corresponds to and is indicative of the sample peak (pulse) of block  83   d /codebook vector block  75   d.    
   Each of the noted positions are further shown to be the elements of position vector P n  below graph line  87  in  FIG. 7 . That is, P n ={2,18,32,46}. Similarly, sign vector S n  is initially formed of (i) a first element (=1) indicative of the positive direction of arrow  85   a  (and hence corresponding pulse in block  83   a ), (ii) a second element (=−1) indicative of the negative direction of arrow  85   b  (and hence corresponding pulse in block  83   b ), (iii) a third element (=1) indicative of the positive direction of arrow  85   c  (and hence corresponding pulse of block  83   c ), and (iv) a fourth element (=1) indicative of the positive direction of arrow  85   d  (and hence corresponding pulse of block  83   d ). However, upon validating each pulse, the fourth element of sign vector S n  becomes 0 as follows. 
   Applying the above detailed validity routine/procedure obtains:
     S bf (P 1 )*S 1 =S bf (position  2 )*(+1)=2.5 which is &gt;μS bf max;   S bf (P 2 )*S 2 =S bf (position  18 )*(−1)=−2*(−1)=2 which is &gt;μS bf max;   S bf (P 3 )*S 3 =S bf (position  32 )*(+1)=2.5 which is &gt;μS bf max; and   S bf (P 4 )*S 4 =S bf (position  46 )*(+1)=0.5 which is &lt;μS bf max,
 
where 0.4≦μ≦0.6 and S bf max=/S bf (position  31 )/=3. Thus the last comparison, i.e., S 4  compared to S bf max, determines S 4  to be an invalid pulse where 0.5&lt;μS bf max. So S 4  is assigned a zero value in sign vector S n , resulting in the S n  vector illustrated near the bottom of  FIG. 7 .
   

   The fixed codebook contribution or vector  81  (referred to as the excitation vector v(n)) is then constructed as follows: 
                                                                                   For n = 0 to NSF−1                If n = =P n                  v(n) = S n                  EndIf                EndFor                        
Thus, in the example of  FIG. 7 , codebook vector  81 , i.e., excitation vector v(n), has three non-zero elements. Namely, v(2)=1; v(18)=−1; v(32)=1, as illustrated in the bottom graph line of  FIG. 7 .
 
   The consideration of only certain block  83  positions to determine sample peak and hence pulse per given block  75 , and ultimately excitation vector  81  v(n) values, decreases complexity with substantially minimal loss in speech quality. As such, second processing phase  79  is optimized as desired. 
   EXAMPLE 
   The following example uses the above described fast, fixed codebook search for creating and searching a 16-bit codebook with subframe size of 56 samples. The excitation vector consists of four blocks. In each block, a pulse can take any of seven possible positions. Therefore, 3 bits are required to encode pulse positions. The sign of each pulse is encoded with 1 bit. The eighth index in the pulse position is utilized to indicate the existence of a pulse in the block. A total of 16 bits are thus required to encode four pulses (i.e., the pulses of the four excitation vector blocks). 
   By using the above described procedure, the pulse position and signs of the pulses in the subject blocks are obtained as follows. Table 3 further summarizes and illustrates the example 16-bit excitation codebook. 
                 p1   ⁢     =   j     ⁢     max   ⁢           ⁢     {     abs   ⁡     (       s   bf     ⁡     (   j   )       )       }                             j   =   0     ,   2   ,   4   ,   6   ,   8   ,   10   ,   12                 v   ⁡     (   p1   )       =       s   bf     ⁡     (   p1   )                                           p2   =       max   j     ⁢     {     abs   ⁡     (       s   bf     ⁡     (   j   )       )       }                             j   =   14     ,   16   ,   18   ,   20   ,   22   ,   24   ,   26                 v   ⁡     (   p2   )       =       s   bf     ⁡     (   p2   )                                           p3   =       max   j     ⁢     {     abs   ⁡     (       s   bf     ⁡     (   j   )       )       }                             j   =   28     ,   30   ,   32   ,   34   ,   36   ,   38   ,   40                 v   ⁡     (   p3   )       =       s   bf     ⁡     (   p3   )                                           p4   =       max   j     ⁢     {     abs   ⁡     (       s   bf     ⁡     (   j   )       )       }                             j   =   42     ,   44   ,   46   ,   48   ,   50   ,   52   ,   54                 v   ⁡     (   p4   )       =       s   bf     ⁡     (   p4   )                                           
where abs(s) is the absolute value of the pulse magnitude of a block sample in S bf .
 
   
     
       
             
             
           
             
             
           
             
             
           
             
             
           
             
             
           
         
             
                 
                 
             
           
           
             
                 
               MaxAbs = max(abs(v(i))) 
             
           
        
         
             
                 
               where i = p1, p2, p3, p4; and 
             
           
        
         
             
                 
               v(i) = 0  if v(i) &lt;0.5 *MaxAbs, or 
             
           
        
         
             
                 
               sign (v(i)) otherwise 
             
           
        
         
             
                 
               for i = p1, p2, p3, p4. 
             
             
                 
                 
             
           
        
       
     
   
   Let v(n) be the pulse excitation and v h (n) be the filtered excitation ( FIG. 3 ), then prediction gain G is calculated as 
   
     
       
             
           
             
             
             
             
           
         
             
               TABLE 3 
             
           
           
             
                 
             
             
               16-bit fixed excitation codebook 
             
             
               
                 
                   
                     
                       G 
                       = 
                       
                         
                           
                             ∑ 
                             
                               n 
                               = 
                               0 
                             
                             
                               n 
                               = 
                               
                                 NSF 
                                 - 
                                 1 
                               
                             
                           
                           ⁢ 
                           
                             
                               
                                 S 
                                 tv 
                                 ′ 
                               
                               ⁡ 
                               
                                 ( 
                                 n 
                                 ) 
                               
                             
                             ⁢ 
                             
                               
                                 v 
                                 h 
                               
                               ⁡ 
                               
                                 ( 
                                 n 
                                 ) 
                               
                             
                           
                         
                         
                           
                             ∑ 
                             
                               n 
                               = 
                               0 
                             
                             
                               n 
                               = 
                               
                                 NSF 
                                 - 
                                 1 
                               
                             
                           
                           ⁢ 
                           
                             
                               
                                 V 
                                 h 
                               
                               ⁡ 
                               
                                 ( 
                                 n 
                                 ) 
                               
                             
                             ⁢ 
                             
                               
                                 v 
                                 h 
                               
                               ⁡ 
                               
                                 ( 
                                 n 
                                 ) 
                               
                             
                           
                         
                       
                     
                   
                 
               
             
             
                 
             
           
        
         
             
                 
                 
               Bits 
               Bits 
             
             
               Block 
               Pulse Position 
               Sign 
               Position 
             
             
                 
             
             
               1 
               0, 2, 4, 6, 8, 10, 12 
               1 
               3 
             
             
               2 
               14, 16, 18, 20, 
               1 
               3 
             
             
                 
               22, 24, 26 
             
             
               3 
               28, 30, 32, 34, 
               1 
               3 
             
             
                 
               36, 38, 40 
             
             
               4 
               42, 44, 46, 48, 
               1 
               3 
             
             
                 
               50, 52, 54 
             
             
                 
             
           
        
       
     
   
   EQUIVALENTS  
   While this invention has been particularly shown and described with references to preferred embodiments thereof, it will be understood by those skilled in the art that various changes in form and details may be made therein without departing from the spirit and scope of the invention as defined by the appended claims. Those skilled in the art will recognize or be able to ascertain using no more than routine experimentation, many equivalents to the specific embodiments of the invention described specifically herein. Such equivalents are intended to be encompassed in the scope of the claims. 
   For example, the foregoing describes the application of Product Code Vector Quantization to the pitch predictor parameters. It is understood that other similar vector quantization may be applied to the pitch predictor parameters and achieve similar savings in computational complexity and/or memory storage space. 
   Further a 5-tap pitch predictor is employed in the preferred embodiment. However, other multi-tap (&gt;2) pitch predictors may similarly benefit from the vector quantization disclosed above. Additionally, any number of working codebooks  31 , 33  ( FIG. 5 ) for providing subvectors g i , g j  . . . may be utilized in light of the discussion of  FIG. 5 . The above discussion of two codebooks  31 , 33  is for purposes of illustration and not limitation of the present invention. 
   In the foregoing discussion of  FIG. 7 , every even numbered position was considered for purposes of defining pulse positions P n  in corresponding blocks  83 . Every third or every odd position or a combination of different positions for different blocks  83  and/or different subframes S bf  and the like may similarly be utilized. Reduction of complexity and bit rate is a function of reduction in number of positions considered. There is a tradeoff however with final quality. Thus, Applicants have disclosed consideration of every other position to achieve both low complexity and high quality at a desired bit-rate. Other combinations of reduced number of positions considered for low complexity but without degradation of quality are now in the purview of one skilled in the art. 
   Likewise, the second processing phase  79  (optimization of the fixed codebook search  27 ,  FIG. 3 ) may be employed singularly (without the vector quantization of the pitch predictor parameters in the first processing phase  77 ), as well as in combination as described above.

Technology Classification (CPC): 6