# Patent

## Title
Speech feature extraction system

## Abstract

The present invention provides a speech feature extraction system suitable for use in a speech recognition system or other voice processing system that extracts features related to the frequency and amplitude characteristics of an input speech signal using a plurality of complex band pass filters and processing the outputs of adjacent band pass filters. The band pass filters can be arranged according to linear, logarithmic or mel-scales, or a combination thereof.

## Background
REFERENCE TO RELATED APPLICATIONS
This application is a continuation-in-part of U.S. patent application Ser. No. 09/882,744, filed Jun. 15, 2001, now U.S. Pat. No. 6,493,668, the entirety of which is incorporated herein by reference.


BACKGROUND OF THE INVENTION
This invention relates to a speech feature extraction system for use in speech recognition, voice identification or voice authentication systems. More specifically, this invention relates to a speech feature extraction system that can be used to create a speech recognition system or other speech processing system with a reduced error rate.
Generally, a speech recognition system is an apparatus that attempts to identify spoken words by analyzing the speaker's voice signal. Speech is converted into an electronic form from which features are extracted. The system then attempts to match a sequence of features to previously stored sequence of models associated with known speech units. When a sequence of features corresponds to a sequence of models in accordance with specified rules, the corresponding words are deemed to be recognized by the speech recognition system.
However, background sounds such as radios, car noise, or other nearby speakers can make it difficult to extract useful features from the speech. In addition, a change in the ambient conditions such as the use of a different microphone, telephone handset or telephone line can interfere with system performance. Also, a speaker's distance from the microphone, differences between speakers, changes in speaker intonation or emphasis, and even a speaker's health can adversely impact system performance. For a further description of some of these problems, see Richard A. Quinnell, “Speech Recognition: No Longer a Dream, But Still a Challenge,” EDN Magazine, Jan. 19, 1995, p. 41–46.
In most speech recognition systems, the speech features are extracted by cepstral analysis, which generally involves measuring the energy in specific frequency bands. The product of that analysis reflects the amplitude of the signal in those bands. Analysis of these amplitude changes over successive time periods can be modeled as an amplitude modulated signal.
Whereas the human ear is a sensitive to frequency modulation as well as amplitude modulation in received speech signals, this frequency modulated content is only partially reflected in systems that perform cepstral analysis.
Accordingly, it would be desirable to provide a speech feature extraction system capable of capturing the frequency modulation characteristics of speech, as well as previously known amplitude modulation characteristics.
It also would be desirable to provide speech recognition and other speech processing systems that incorporate feature extraction systems that provide information on frequency modulation characteristics of the input speech signal.
SUMMARY OF THE INVENTION
In view of the foregoing, it is an object of the present invention to provide a speech feature extraction system capable of capturing the frequency modulation characteristics of speech, as well as previously known amplitude modulation characteristics.
It also is an object of this invention to provide speech recognition and other speech processing systems that incorporate feature extraction systems that provide information on frequency modulation characteristics of the input speech signal.
The present invention provides a speech feature extraction system that reflects frequency modulation characteristics of speech as well as amplitude characteristics. This is done by a feature extraction stage which, in one embodiment, includes a plurality of complex band pass filters arranged in adjacent frequency bands according to a linear frequency scale (“linear scale”). The plurality of complex band pass filters are divided into pairs. A pair includes two complex band pass filters in adjacent frequency bands. For every pair, the output of the filter in the higher frequency band (“primary frequency”) is multiplied by the conjugate of the output of the filter in the lower frequency band (“secondary filter”). The resulting signal is low pass filtered.
In another embodiment, the feature extraction phase includes a plurality of complex band pass filters arranged according to a logarithmic (or exponential) frequency scale (“log scale”). The primary filters of the filter pairs are centered at various frequencies along the log scale. The secondary filter corresponding to the primary filter of each pair is centered at a predetermined frequency below the primary filter. For every pair, the output of the primary filter is multiplied by the conjugate of the output of the secondary filter. The resulting signal is low pass filtered.
In yet another embodiment, the plurality of band pass filters are arranged according to a mel-scale. The primary filters of the filter pairs are centered at various frequencies along the mel-scale. The secondary filter corresponding to the primary filter of each pair is centered at a predetermined frequency below the primary filter. For every pair, the output of the primary filter is multiplied by the conjugate of the output of the secondary filter. The resulting signal is low pass filtered.
In still another embodiment, the plurality of band pass filters are arranged according to a combination of the linear and log scale embodiments mentioned above. A portion of the pairs of the band pass filters are arranged in adjacent frequency bands according to a linear scale. For each of these pairs, the output of the primary filter is multiplied by the conjugate of the output of the secondary filter. The resulting signal is low pass filtered.
The primary filters of the remaining pairs of band pass filters are centered at various frequencies along the log scale and the secondary filters corresponding to the primary filters are centered a predetermined frequency below the primary filters. For every pair, the output of the primary filter is multiplied by the conjugate of the output of the secondary filter. The-resulting signal is low pass filtered.
For the embodiments described above, each of the low pass filter outputs is processed to compute two components: a FM component that is substantially sensitive to the frequency of the signal passed by the adjacent band pass filters from which the low pass filter output was generated, and an AM component that is substantially sensitive to the amplitude of the signal passed by the adjacent band pass filters. The FM component reflects the difference in the phase of the outputs of the adjacent band pass filters used to generate the lowpass filter output.
The AM and FM components are then processed using known feature enhancement techniques, such as discrete cosine transform, mel-scale translation, mean normalization, delta and acceleration analysis, linear discriminant analysis and principal component analysis, to generate speech features suitable for statistical processing or other recognition or identification methods. In an alternative embodiment, the plurality of complex band pass filters can be implemented using a Fast Fourier Transform (FFT) of the speech signal or other digital signal processing (DSP) techniques.
In addition, the methods and apparatus of the present invention may be used in addition to performing cepstral analysis in a speech recognition system.



BRIEF DESCRIPTION OF THE DRAWINGS
The above and other objects and advantages of the present invention will be apparent upon consideration of the following detailed description, taken in conjunction with the accompanying drawings, in which like reference characters refer to like parts throughout, and in which:
FIG. 1 is a block diagram of an illustrative speech recognition system incorporating the speech feature extraction system of the present invention;
FIG. 2 is a detailed block diagram of the speech recognition system of FIG. 1; and
FIG. 3 is a detailed block diagram of a band pass filter suitable for implementing the feature extraction system of the present invention; and
FIG. 4 is a detailed block diagram of an alternative embodiment of a speech recognition including an alternative speech feature extraction system of the present invention; and
FIG. 5 illustrates is a graph showing band pass filter frequencies spaced according to a linear frequency scale; and
FIG. 6 is a graph showing pairs of band pass filters spaced according to a logarithmic frequency scale; and
FIG. 7 is a graph showing band pass filter frequency pairs spaced according to a mel-scale; and
FIG. 8 is a graph showing band pass filter frequencies spaced according to a combination of linear and logarithmic frequency scales.



DETAILED DESCRIPTION OF THE INVENTION
Referring to FIG. 1, a generalized depiction of illustrative speech recognition system 5 is described that incorporates the speech extraction system of the present invention. As will be apparent to one of ordinary skill in the art, the speech feature extraction system of the present invention also may be used in speaker identification, authentication and other voice processing systems.
System 5 illustratively includes four stages:

    
    
        pre-filtering stage 10, feature extraction stage 12, statistical processing stage 14, and energy stage 16.
    
    


Pre-filtering stage 10, statistical processing stage 14 and energy stage 16 employ speech processing techniques known in the art and do not form part of the present invention. Feature extraction stage 12 incorporates the speech feature extraction system of the present invention, and further includes feature enhancement techniques which are known in the art, as described hereinafter.
Audio speech signal is converted into an electrical signal by a microphone, telephone receiver or other device, and provided as an input speech signal to system 5. In a preferred embodiment of the present invention, the electrical signal is sampled or digitized to provide a digital signal (IN) representative of the audio speech. Pre-filtering stage 10 amplifies the high frequency components of audio signal IN, and the prefiltered signal is then provided to feature extraction stage 12.
Feature extraction stage 12 processes pre-filtered signal X to generate a sequence of feature vectors related to characteristics of input signal IN that may be useful for speech recognition. The output of feature extraction stage 12 is used by statistical processing stage 14 which compares the sequence of feature vectors to predefined statistical models to identify words or other speech units in the input signal IN. The feature vectors are compared to the models using known techniques, such as the Hidden Markov Model (HMM) described in Jelinek, “Statistical Methods for Speech Recognition,” The MIT Press, 1997, pp. 15–37. The output of statistical processing stage 14 is the recognized word, or other suitable output depending upon the specific application.
Statistical processing at stage 14 may be performed locally, or at a remote location relative to where the processing of stages 10, 12, and 16 are performed. For example, the sequence of feature vectors may be transmitted to a remote server for statistical processing.
The illustrative speech recognition system of FIG. 1 preferably also includes energy stage 16 which provides an output signal indicative of the total energy in a frame of input signal IN. Statistical processing stage 14 may use this total energy information to provide improved recognition of speech contained in the input signal.
Referring now to FIG. 2, pre-filtering stage 10 and feature extraction stage 12 are described in greater detail. Pre-filtering stage 10 is a high pass filter that amplifies high frequency components of the input signal. Pre-filtering stage 10 comprises one-sample delay element 21, multiplier 23 and adder 24. Multiplier 23 multiplies the one-sample delayed signal by constant Kf, which typically has a value of −0.97. The output of pre-filtering stage 10, X, is input at the sampling rate into a bank of band pass filters 301, 302, . . . 30n.
In one embodiment, the band pass filters 301, 302, . . . 30n are positioned in adjacent frequency bands. The spacing of the band pass filters 301, 302 . . . 30n, is done according to a linear frequency scale (“linear scale”) 68 as shown in graph 72 of FIG. 5. The term “linear frequency scale” is used in this specification in accordance with its ordinary and accustomed meaning, i.e., the actual frequency divisions are uniformly spaced. The plurality of complex band pass filters 301, 302, . . . 30n are divided into pairs P1-2. A pair (P1 or P2) includes two complex band pass filters (301-2 or 303-4) respectively in adjacent frequency bands. For every pair (P1 or P2), the output of the filter in the higher frequency band (302 or 304) (referred to hereinafter as the “primary filter”) is multiplied by the conjugate of the output of the filter in the lower frequency band (301 or 303) (referred to hereinafter as the “secondary filter”). The resulting signal is low pass filtered.
The number of band pass filters 301, 302 . . . 30n and width of the frequency bands preferably are selected according to the application for the speech processing system. For example, a system useful in telephony applications preferably will employ about forty band pass filters 301, 302, . . . 30n having center frequencies approximately 100 Hz apart. For example, filter 301 may have a center frequency of 50 Hz, filter 302 may have a center frequency of 150 Hz, filter 303 may have a center frequency of 250 Hz, and so on, so that the center frequency of filter 3040 is 3950 Hz. The bandwidth of each filter may be several hundred Hertz.
In another embodiment, as illustrated in the graph 70 of FIG. 6, the band pass filters 301, 302 . . . 30108 are arranged according to a non-linear frequency scale such as a logarithmic (or exponential) frequency scale 74 (“log scale”). The term logarithmic frequency scale is used in this specification according to its ordinary and accustomed meaning.
Empirical evidence suggests that using log scale 74 of FIG. 6 instead of linear scale 68 of FIG. 5 improves voice recognition performance. That is so because the human ear resolves frequencies non-linearly across the audio spectrum. Another advantage using log scale 74 instead of linear scale 68 is that log scale 74 can cover a wider range of frequency spectrum without using additional band pass filters 301, 302 . . . 30n.
Pairs P1-54 of band pass filters 301-108 are spaced according to log scale 74. Pair P1 includes filters 301, and 302, pair P10 includes filters 3019 and 3020, and pair P54 includes filters 30107 and 30108. In this arrangement, filters 302, 3020 and 30108 are the primary filters and filters 301, 3019 and 30107 are the secondary filters.
In one preferred embodiment, primary filters 302, 3020 . . . 30108 are centered at various frequencies along log scale 74, while secondary filters 301, 303 . . . 30107 are centered 100 hertz (Hz) below corresponding primary filters 302, 304 . . . 30108 respectively. An exemplary MATLAB code to generate graph 70 of FIG. 6 is shown below.













v=(2.{circumflex over ( )}([26.715:0.25:40]/3.345));
% Generate center frequencies for bandpass filter pairs


f(2:2:2*length(v))=v+50;
% Primary filters center frequencies


f(1:2:2*length(v))=v−50;
% Secondary filters center frequencies


semilogy([v′+50 v′−50],′.′);grid
% Plot center frequencies on a logarithmic scale′









In another embodiment, the center frequencies of primary filters 302, 3020 . . . 30108 and secondary filters 301, 303 . . . 30107 may be placed using separate and independent algorithms, while ensuring that secondary filters 301, 303 . . . 30107 are centered 100 hertz (Hz) below their corresponding primary filters 302, 304 . . . 30108, respectively.
In one embodiment, band pass filters 301-108 are of triangular shape. In other embodiments, band pass filters 301-108 can be of various shapes depending on the requirements of the particular voice recognition systems.
Log scale 74 is shown to range from 0–4000 Hz. For every pair P1-54, the output of the primary filter 302, 304 . . . 30108 is multiplied by the conjugate of the output of secondary filter 301, 303 . . . 30107. The resulting signal is low pass filtered.
The pairs P1-54 are arranged such that the lower frequencies include a higher concentration of pairs P1-54 than the higher frequencies. For example, there are 7 pairs (P16-22) in the frequency range of 500–1000 Hz while there are only 3 pairs (P49-51) in the frequency range of 3000–3500 Hz. Thus, although there is over sampling at the lower frequencies, this embodiment also performs at least some sampling at the higher frequencies. The concentration of pairs P1-54 along log scale 74 can be varied depending on the needs of a particular voice recognition system.
As will be apparent to one of ordinary skill in the art of digital signal processing design, the band pass filters of the preceding embodiments may be implemented using any of a number of software or hardware techniques. For example, the plurality of complex filters may be implemented using a Fast Fourier Transform (FFT), Chirp-Z transform, other frequency domain analysis techniques.
In an alternative embodiment, depicted in FIG. 7, band pass filters 301, 302, . . . 30n are arranged according to a non-linear frequency scale, such as mel-scale 80. Mel-scale 80 is well known in the art of voice recognition systems and is typically defined by the equation, 



  
    Mel
    ⁡
    
      (
      f
      )
    
  
  =
  
    2595
    ⁢
    
      
    
    ⁢
    
      
        log
        10
      
      ⁡
      
        (
        
          1
          +
          
            f
            700
          
        
        )
      
    
  




where f represents frequency according to the linear scale 68 and Mel(f) represents its corresponding mel-scale 80 frequency.

FIG. 7 illustrates one embodiment of a graph 84 showing band pass filters 301-9 spaced according to mel-scale 80. The center frequencies (CF1-9) are the Mel(f) values calculated by using the above equation. Typically, filters 301-9 are spread over the whole frequency range from zero up to the Nyquist frequency. In one embodiment, filters 301-9 have the same band width. In another embodiment, filters 301-9 may have different bandwidths.
In yet another embodiment, depicted in FIG. 8, the band pass filters 301-8 are spaced according to a combination of linear 68 and non-linear 74 frequency scales. Band pass filters 301-4 (P1-2) are arranged in adjacent frequency bands according to linear frequency 68.
Primary filters 306 and 308 are centered along log scale 74. Secondary filters 305 and 307 are centered at frequencies of 100 Hz below the center frequencies of 306 and 308 respectively. For each of these pairs (P1 or P2), the output of the primary filter (306 or 308) is multiplied by the conjugate of the output of the secondary filter (305 and 307), respectively and the resulting signal is low pass filtered.
Referring again to FIG. 2, blocks 401-20 provide the complex conjugate of the output signal of band pass filter 301, 303, . . . 30n−1. Multiplier blocks 421-20 multiply the complex conjugates by the outputs of an adjacent higher frequency band pass filter 302, 304, 306, . . . 3040 to provide output signals Z1-20. Output signals Z1-20 then are passed through a series of low pass filters 441-20. The outputs of the low pass filters typically are generated only at the feature frame rate. For example, at a input speech sampling rate of 8 kHz, the output of the low pass filters is only computed at a feature frame rate of once every 10 msec.
Each output of low pass filters 441-20 is a complex signal having real component R and imaginary component I. Blocks 461-20 process the real and imaginary components of the low pass filter outputs to provide output signals A1-20 and F1-20 as shown in equations (1) and (2): 



  
    
      
        
          A
          i
        
        =
        
          log
          ⁢
          
            
          
          ⁢
          
            R
            i
          
        
      
    
    
      
        (
        1
        )
      
    
  
  
    
      
        
          f
          i
        
        =
        
          
            I
            i
          
          
            
              
                R
                i
                2
              
              +
              
                I
                i
                2
              
            
          
        
      
    
    
      
        (
        2
        )
      
    
  




wherein Ri and Ii are the real and imaginary components of the corresponding low pass filter output. Output signals Ai are a function of the amplitude of the low pass filter output and signals Fi are a function of the frequency of the signal passed by the adjacent band pass filters from which the low pass filter output was generated. By computing two sets of signals that are indicative of the amplitude and frequency of the input signal, the speech recognition system incorporating the speech feature extraction system of the present invention is expected to provide reduced error rate.

The amplitude and frequency signals A1-20 and F1-20 then are processed using conventional feature enhancement techniques in feature enhancement component 12b, using, for example, discrete cosine transform, mel-scale translation, mean normalization, delta and acceleration analysis, linear discriminant analysis and principal component analysis techniques that are per se known in the art. A preferred embodiment of a speech recognition system of the present invention incorporating the speech extraction system of the present invention employs a discrete cosine transform and delta features technique, as described hereinafter.
Still referring to FIG. 2, feature enhancement component 12b receives output signals A1-20 and F1-20, and processes those signals using discrete cosine transform (DCT) blocks 50 and 54, respectively. DCTs 50 and 54 attempt to diagonalize the co-variance matrix of signals A1-20 and F1-20. This helps to uncorrelate the features in output signals B0-19 of DCT 50 and output signals C0-19 of DCT 54. Each set of output signals B0-19 and C0-19 then are input into statistical processing stage 14. The function performed by DCT 50 on input signals A1-20 to provide output signals B0-19 is shown by equation (3), and the function performed by DCT 54 on input signals F1-20 to provide output signals C0-19 is shown by equation (4). 



  
    
      
        
          B
          r
        
        =
        
          
            D
            ⁡
            
              (
              r
              )
            
          
          ⁢
          
            
              ∑
              
                n
                =
                0
              
              
                N
                -
                1
              
            
            ⁢
            
              
                
                  A
                  
                    n
                    +
                    1
                  
                
                ·
                cos
              
              ⁢
              
                
              
              ⁢
              
                
                  
                    (
                    
                      
                        2
                        ⁢
                        n
                      
                      +
                      1
                    
                    )
                  
                  ⁢
                  π
                  ⁢
                  
                    
                  
                  ⁢
                  r
                
                
                  2
                  ⁢
                  N
                
              
            
          
        
      
    
    
      
        (
        3
        )
      
    
  
  
    
      
        
          C
          r
        
        =
        
          
            D
            ⁡
            
              (
              r
              )
            
          
          ⁢
          
            
              ∑
              
                n
                =
                0
              
              
                N
                -
                1
              
            
            ⁢
            
              
                
                  F
                  
                    n
                    +
                    1
                  
                
                ·
                cos
              
              ⁢
              
                
              
              ⁢
              
                
                  
                    (
                    
                      
                        2
                        ⁢
                        n
                      
                      +
                      1
                    
                    )
                  
                  ⁢
                  π
                  ⁢
                  
                    
                  
                  ⁢
                  r
                
                
                  2
                  ⁢
                  N
                
              
            
          
        
      
    
    
      
        (
        4
        )
      
    
  




In equations (3) and (4), N equals the length of the input signal vectors A and F (e.g., N=20 in FIG. 2), n is an index from 0 to N−1 (e.g., n=0 to 19 in the embodiment of FIG. 2), and r is the index of output signals B and C (e.g., r=0 to 19 in the embodiment of FIG. 2). Thus, for each vector output signal Br, each vector of input signals A1-20 are multiplied by a cosine function and D(r) and summed together as shown in equation (3). For each vector output signal Cr, each vector of input signals S1-20 are multiplied by a cosine function and D(r) and summed together as shown in equation (4). D(r) are coefficients that are given by the following equations: 



  
    
      
        
          D
          ⁡
          
            (
            r
            )
          
        
        =
        
          
            
              1
              
                N
              
            
            ⁢
            
              
            
            ⁢
            
              for
            
            ⁢
            
              
            
            ⁢
            r
          
          =
          0
        
      
    
    
      
        (
        5
        )
      
    
  
  
    
      
        
          D
          ⁡
          
            (
            r
            )
          
        
        =
        
          
            
              
                2
              
              
                N
              
            
            ⁢
            
              
            
            ⁢
            
              for
            
            ⁢
            
              
            
            ⁢
            r
          
          >
          0
        
      
    
    
      
        (
        6
        )
      
    
  




Output signals B0-19 and C0-19 also are input into delta blocks 52 and 56, respectively. Each of delta blocks 52 and 56 takes the difference between measurements of feature vector values between consecutive feature frames and this difference may be used to enhance speech recognition performance. Several difference formulas may be used by delta blocks 52 and 56, as are known in the art. For example, delta blocks 52 and 56 may take the difference between two consecutive feature frames. The output signals of delta blocks 52 and 56 are input into statistical processing stage 14.
Energy stage 16 of FIG. 2 is a previously known technique for computing the logarithm of the total energy (represented by E) of each frame of input speech signal IN, according to the following equation: 



  
    
      
        
          E
          ⁡
          
            (
            
              n
              ⁢
              
                
              
              ⁢
              T
            
            )
          
        
        =
        
          log
          (
          
            
              
                ∑
                
                  i
                  =
                  0
                
                
                  K
                  -
                  1
                
              
              ⁢
              
                IN
                
                  
                    (
                    
                      n
                      -
                      i
                    
                    )
                  
                  ⁢
                  T
                
                2
              
            
            K
          
          )
        
      
    
    
      
        (
        7
        )
      
    
  




Equation 7 shows that energy block 16 takes the sum of the squares of the values of the input signal IN during the previous K sampling intervals (e.g., K=220, T= 1/8000 seconds), divides the sum by K, and takes the logarithm of the final result. Energy block 16 performs this calculation every frame (e.g., 10 msec), and provides the result as an input to statistical processing block 14.
Referring now to FIG. 3, illustrative complex bandpass filter 30′ suitable for use in the feature extraction system of the present invention is described. Filter 30′ comprises adder 31, multiplier 32 and one-sample delay element 33. Multiplier 32 multiples the one-sample delayed output Y by complex coefficient G and the resultant is added to the input signal X to generate an output signal Y.
An alternative embodiment of the feature extraction system of the present invention is described with respect to FIG. 4. The embodiment of FIG. 4 is similar to the embodiment of FIG. 2 and includes pre-filtering stage 10, statistical processing stage 14, and energy stage 16 the operate substantially as described above. However, the embodiment of FIG. 4 differs from the previously described embodiment in that feature extraction stage 12′ includes additional circuitry within feature extraction system 12a, so that the feature vectors include additional information.
For example, feature extraction stage 12a′ includes a bank of 41 band pass filters 301-41 and conjugate blocks 401-40. The output of each band pass filter is combined with the conjugate of the output of a lower adjacent band bass filter by multipliers 421-40. Low pass filters 441-40, and computation blocks 461-40 compute vectors A and F as described above, except that the vectors have a length of forty elements instead of twenty. DCTs 50 and 54, and delta blocks 52 and 56 of feature enhancement component 12b′ each accept the forty element input vectors and output forty element vectors to statistical processing block 14. It is understood that the arrangement illustrated in FIG. 4 is not applicable if the band pass filters 301-41 arranged according to a non-linear frequency scale such as a log scale or a mel-scale.
The present invention includes feature extraction stages which may include any number of band pass filters 30, depending upon the intended voice processing application, and corresponding numbers of conjugate blocks 40, multipliers 42, low pass filters 44 and blocks 46 to provide output signals A and F for each low pass filter. In addition, signals A and F may be combined in a weighted fashion or only part of the signals may be used. For example, it may be advantageous to use only the amplitude signals in one frequency domain, and a combination of the amplitude and frequency signals in another.
While preferred illustrative embodiments of the invention are described above, it will be apparent to one skilled in the art that various changes and modifications may be made therein without therein without departing from the invention, and it is intended in the appended claims to cover all such changes and modifications which fall within the true spirit and scope of the invention.

## Claims

1. Apparatus for use in a speech processing system for extracting features from an input speech signal having frequency and amplitude characteristics, the apparatus comprising:
first and second band pass filters adapted to receive the input speech signal and providing respectively, first and second signals;
a conjugate circuit coupled to the second band pass filter and providing a third signal that is the conjugate of the second signal;
a multiplier coupled to the first band pass filter and to the conjugate circuit and providing a fourth signal that is the product of the first and third signals; and
filter means coupled to the multiplier for providing a fifth signal related to the frequency characteristics of the input signal, and a sixth signal related to amplitude characteristics of the input signal.

2. The apparatus of claim 1 wherein the filter means includes a low pass filter.

3. The apparatus of claim 1 further comprising a high pass filter coupled between the input signal and the first and second band pass filters.

4. The apparatus of claim 2 wherein the low pass filter provides a signal having a real part, R, and an imaginary part, I, and the filter means includes circuitry for calculating:
A=log R and
F=I/Sqrt(R2+I2).

5. The apparatus of claim 1 wherein the first and second band pass filters each comprises a delay element, a complex multiplier and an adder.

6. The apparatus of claim 5 wherein the adder receives the input speech signals and outputs an output signal, the adder adding the input speech signal and a signal comprising a delayed sample of the output signal multiplied by the complex coefficient.

7. The apparatus of claim 4 further comprising feature enhancement circuitry.

8. The apparatus of claim 7 further comprising circuitry for statistically processing an output of the feature enhancement circuitry to compare that output to a plurality of predetermined models.

9. The apparatus of claim 7 wherein the feature enhancement circuitry comprises circuitry for computing a discrete cosine transform of A and F.

10. The apparatus of claim 7 wherein the feature enhancement circuitry further comprises circuitry for calculating a difference between successive discrete cosine transforms of A and for calculating a difference between successive discrete cosine transforms of F.

11. The apparatus of claim 1 further comprising:
a transducer for converting sound into an electrical signal; and
a sampler for converting the electrical signal into a digital signal, wherein the input speech signal comprises the digital signal.

12. The apparatus of claim 1 further comprising a digital processor, wherein the first and second band pass filters, the conjugate circuit, the multiplier, and the filter means comprise algorithms adapted to be executed on the digital processor.

13. The apparatus of claim 1, wherein a frequency of the first bandpass filter is selected according to a logarithmic scale and a frequency of the second bandpass filter is selected at a predetermined frequency below the frequency of the first bandpass filter.

14. The apparatus of claim 13, wherein a center frequency of the second filter is one hundred hertz less than a center frequency of the first filter.

15. The apparatus of claim 13, wherein the bands of the first and second filters overlap.

16. The apparatus of claim 13, wherein the bandwidths of the first and second filters are the same.

17. The apparatus of claim 1, wherein a frequency of the first bandpass filter is selected according to a mel-scale and a frequency of the second bandpass filter is selected at a predetermined frequency below the frequency of the first bandpass filter.

18. The apparatus of claim 17, wherein a center frequency of the second filter is one hundred hertz less than a center frequency of the first filter.

19. The apparatus of claim 17, wherein the bands of the first and second filters overlap.

20. The apparatus of claim 17, wherein the mel-scale is defined by the equation 



  
    Mel
    ⁡
    
      (
      f
      )
    
  
  =
  
    2595
    ⁢
    
      
    
    ⁢
    
      
        
          log
          10
        
        ⁡
        
          (
          
            1
            +
            
              f
              700
            
          
          )
        
      
      .

21. The apparatus of claim 17, wherein the bandwidths of the first and second filters are the same.

22. The apparatus of claim 1, wherein the first and second band pass filters are implemented using a Fast Fourier Transform.

23. A method for extracting features from an input speech signal for use in a speech processing device, the method comprising:
separating the input signal into a first signal in a first frequency band and a second signal in a second frequency band;
providing a conjugate of the first signal;
multiplying the conjugate of the first signal with the second signal to provide a third signal; and
processing the third signal to generate a frequency component related to frequency features in the input speech signal and an amplitude component related to amplitude features in the input speech signal.

24. The method of claim 23 further comprising high pass filtering the input speech signal prior to processing the input signal to generate the first and second signals.

25. The method, of claim 23 wherein processing the third signal further comprises low pass filtering the third signal.

26. The method of claim 25 wherein low pass filtering the third signal provides a signal having a real part, R, and an imaginary part, I, and processing the third signal includes calculating:
A=log R and
F=I/Sqrt(R2+I2).

27. The method of claim 26 further comprising processing the third signal using feature enhancement techniques to generate a series of feature vectors.

28. The method of claim 27 further comprising statistically processing the series of feature vectors by comparing the series of feature vectors to a plurality of predetermined models.

29. The method of claim 27 further comprising calculating a discrete cosine transform of A and F.

30. The method of claim 29 further comprising calculating a difference between successive discrete cosine transforms of A and between successive discrete cosine transforms of S.

31. The method of claim 30 further comprising statistically processing the differences between successive discrete cosine transforms to compare the differences to a plurality of predetermined patterns.

32. The method of claim 23 further comprising:
converting a sound into an electrical signal; and
sampling the electrical signal to provide a digital signal, wherein the input speech signal comprises the digital signal.

33. A method of claim 23, comprising selecting a frequency of the first frequency band according to a logarithmic scale and selecting a frequency of the second frequency band at a predetermined frequency below the frequency of the first frequency band.

34. The method of claim 33, wherein a center frequency of the second frequency band is one hundred hertz less than a center frequency of the first frequency band.

35. The method of claim 33, comprising overlapping the first and second frequency bands.

36. The method of claim 23, comprising selecting a frequency of the first frequency band according to a mel-scale and selecting the frequency of the second frequency band at a predetermined frequency below the frequency of the first frequency band.

37. The apparatus of claim 36, wherein a center frequency of the second frequency band is one hundred hertz less than a center frequency of the first frequency band.

38. The apparatus of claim 36, comprising overlapping the first and second frequency bands.