EP0713208B1

EP0713208B1 - Pitch lag estimation system

Info

Publication number: EP0713208B1
Application number: EP95118142A
Authority: EP
Inventors: Huan-Yu Su; Tom Hong Li
Original assignee: Conexant Systems LLC
Current assignee: Conexant Systems LLC
Priority date: 1994-11-21
Filing date: 1995-11-17
Publication date: 2002-02-20
Anticipated expiration: 2015-11-17
Also published as: EP0713208A2; DE69525508D1; JPH08211895A; EP0713208A3; DE69525508T2

Description

Signal modeling and parameter estimation play increasingly important roles in data compression, decompression, and coding. To model basic speech sounds, speech signals must be sampled as a discrete waveform to be digitally processed. In one type of signal coding technique, called linear prediction coding (LPC), the signal value at any particular time index is modeled as a linear function of previous values. A subsequent signal is thus linearly predicted according to an earlier value. As a result, efficient signal representations can be determined by estimating and applying certain prediction parameters to represent the signal. Presently, LPC techniques are being used for speech coding involving code excited linear prediction (CELP).
It is recognized that pitch information is a reliable indicator and representative of sounds for coding purposes. Pitch describes a key feature or parameter of a speaker's voice. Because human speech is generally not easily mathematically quantifiable, speech estimation models which can effectively estimate the speech pitch data provide for more accurate and precise coded and decoded speech. In current speech coding models, however, such as certain CELP (e.g., vector sum excited linear prediction (VSELP), multi-pulse, regular pulse, algebraic CELP, etc.) and MBE coder/decoders ("codecs"), pitch estimation is often difficult due to the need for high precision and low complexity of the pitch estimation algorithm.
Several pitch lag estimation schemes are used in conjunction with the above-mentioned codecs: a time domain approach, frequency domain approach, and cepstrum domain approach. The precision of the pitch estimation has a direct impact on the speech quality due to the close relationship between pitch lag and speech reproduction. In CELP coders, speech generation is based on predictions -- long-term pitch prediction and short-term linear prediction. Figure 1 shows a speech regeneration block diagram of a typical CELP coder.
To compress speech data, it is desirable to extract only essential information to avoid transmitting redundancies. Speech can be grouped into short blocks, where representative parameters can be identified in all of the blocks. As indicated in Figure 1, to generate good quality speech, a CELP speech coder must extract LPC parameters 110, pitch lag parameters 112 (including lag and its coefficient), and an optimal innovation code vector 114 with its gain parameter 116 from the input speech to be coded. The coder quantizes the LPC parameters by implementing appropriate coding schemes. The indices of quantization of each parameter comprise the information to be stored or transmitted to the speech decoder. In CELP codecs, determination of pitch prediction parameters (pitch lag and pitch coefficients) is performed in the time domain, while in MBE codecs, pitch parameters are estimated in the frequency domain.
After LPC analysis, the CELP encoder determines an appropriate LPC filter 110 for the current speech coding frame (usually taken about 10-40 ms). The LPC filter is represented by the equation: A(z) = 1-a1z-1-a2z-2- ... -anpz-np or the nth sample can be predicted by
where np is the LPC prediction order (usually approximately 10), y(n) is sampled speech data, and n represents the time index. The LPC equations above describe the estimation of the current sample according to the linear combination of the past samples. A perceptual weighting filter based on the LPC filter which models the sensitivity of the human ear is then defined by W(z) = A(z/γ1) A(z/γ2) where 0 < γ2 < γ1≤ 1
To extract the desired pitch parameters, the pitch parameters which minimize the following weighted coding error energy must be calculated for each coding subframe, where one coding frame may be divided into several coding subframes for analysis and coding: d = ||T - βPLagH - αCiH||2 where T is the target signal which represents the perceptually filtered input signal, and H is the impulse response matrix of the filter W(z)/A(z). P_Lag is the pitch prediction contribution having pitch lag "Lag" and prediction coefficient β which is uniquely defined for a given lag, and C, is the codebook contribution associated with index i in the codebook and its corresponding gain α. Typically, the pitch of human speech varies from 2 ms - 20 ms. Thus, when the speech is sampled at an 8 KHz sampling rate, the pitch lag corresponds roughly to 20 - 147 samples. In addition, i takes values between 0 and Nc-1, where Nc is the size of the innovation codebook.
A one-tap pitch predictor and one innovation codebook are assumed. Typically, however, the general form of the pitch predictor is a multi-tap scheme, and the general form of the innovation codebook is a multi-level vector quantization, or utilizes multiple innovation codebooks. More particularly, in speech coding, one-tap pitch predictor indicates that the current speech sample can be predicted by a past speech sample, while the multi-tap predictor means that the current speech sample can be predicted by multiple past speech samples.
Due to complexity concerns, sub-optimal approaches have been used in speech coding schemes. For example, pitch lag estimation may be performed by simply evaluating the possible lag values in the range between L₁ and L₂ samples to cover 2.5 ms - 18.5 ms. Consequently, the estimated pitch lag value is determined by maximizing the following:
Even though this time domain approach may enable the determination of the real pitch lag, for female speech having a high pitch frequency, the pitch lag found by Eqn. (1) may not be the real lag, but a multiple of the real lag. To avoid this estimation error, additional processes are necessary to correct the estimation error (e.g., lag smoothing) at the cost of undesirable complexity.
However, such excess complexity is a significant drawback of using the time domain approach. For example, the time domain approach requires at least 3 million operations per second (MOPs) to determine the lag using integer lag only. Moreover, if pitch lag smoothing and a fractional pitch lag are used, the complexity is more likely approximately 4 MOPs. In practice, approximately 6 million digital signal processing machine instructions per second (DSP MIPs) are required to implement full range pitch lag estimation with acceptable precision. Thus, it is generally accepted that pitch estimation requires 4-6 DSP MIPs. Although there exist other approaches which can reduce the complexity of pitch estimation, such approaches often sacrifice quality.
In MBE coders, an important member in the class of sinusoidal coders, coding parameters are extracted and quantized in the frequency domain. The MBE speech model is shown in Figures 2-4. In the MBE voice encoder/decoder ("vocoder"), described in Figures 2 and 3, the fundamental frequency (or pitch lag) 210, voiced/unvoiced decision 212, and spectral envelope 214 are extracted from the input speech in the frequency domain. The parameters are then quantized and encoded into a bit stream which can be stored or transmitted.
In the MBE vocoder, to achieve high speech quality, the fundamental frequency must be estimated with high precision. The estimation of the fundamental frequency is performed in two stages. First, an initial pitch lag is searched within the range of 21 samples to 114 samples to cover 2.6 - 14.25 ms at the sampling rate of 8000 Hz by minimizing a weighted mean square error equation 310 (Figure 3) between the input speech 216 and the synthesized speech 218 in the frequency domain. The mean square error between the original speech and the synthesized speech is given by the equation:
where S(ω) is the original speech spectrum, S and(ω) is the synthesized speech spectrum, and G(ω) is a frequency-dependent weighting function. As shown in Figure 4, a pitch tracking algorithm 410 is used to update the initial pitch lag estimate 412 by using the pitch information of neighboring frames.
The motivation for using this approach is based upon the assumption that the fundamental frequency should not change abruptly between neighboring frames. The pitch estimates of the two past and two future neighbor frames are used for the pitch tracking. The mean-square error (including two past and future frames) is then minimized to find a new pitch lag value for the current frame. After tracking the initial pitch lag, a pitch lag multiple checking scheme 414 is applied to eliminate the multiple pitch lag, thus smoothing the pitch lag.
Referring to Figure 4, in the second stage of the fundamental frequency estimation, pitch lag refinement 416 is employed to increase the precision of the pitch estimate. The candidate pitch lag values are formed based on the initial pitch lag estimate (i.e., the new candidate pitch lag values are formed by adding or subtracting some fractional number from the initial pitch lag estimate). Accordingly, a refined pitch lag estimate 418 can be determined among the candidate pitch lags by minimizing the mean square error function.
However, there are certain drawbacks to frequency domain pitch estimation. First, the complexity is very high. Second, the pitch lag must be searched within the range of 20 and 114 samples covering only 2.5 - 14.25 ms to limit the window size to 256 samples to accommodate a 256-point FFT. However, for very low pitch frequency talkers, or for speech having a pitch lag beyond 14.25 ms, it is impossible to gather a sufficient number of samples within a 256-sample window. Moreover, only an averaged pitch lag is estimated over a speech frame.
Using cepstrum domain pitch lag estimation (Figure 5), which was proposed by A.M. Noll in 1967, other modified methods were proposed. In cepstrum domain pitch lag estimation, approximately 37 ms of speech are sampled 510 so that at least two periods of the maximum possible pitch lag (e.g., 18.5 ms) are covered. A 512-point FFT is then applied to the windowed speech frame (at block 512) to obtain the frequency spectrum. taking the logarithm 514 amplitude of the frequency spectrum, another 512-point inverse FFT 516 is applied to get the cepstrum. A weighting function 518 is applied to the cepstrum, and the peak of the cepstrum is detected 520 to determine the pitch lag. A tracking algorithm 522 is then implemented to eliminate any pitch multiples.
Several drawbacks of the cepstrum pitch detection method can be observed however. For example, the computational requirement is high. To cover the pitch range between 20 and 147 samples at an 8 KHz sampling rate, the 512-point FFT must be performed twice. The precision of the estimate is inadequate since the cepstrum pitch estimate will provide only the estimate of an averaged pitch lag over the analysis frame. However, for low bit rate speech coding, it is critical for the pitch lag value to be estimated over a shorter time period. As a result, the cepstrum pitch estimate is almost never used today for high-quality low bit rate speech coding. Thus, because of the limitations of each approach mentioned before, a means for efficient pitch lag estimation is desired to meet the needs of high-quality low bit rate speech coding.
Attention is drawn to the document entitled "Application of a Digital Inverse Filter for Automatic Formant and F0 Analysis", IEEE Tr. on Audio and Electroacoustics, Vol. AU-21, No. 3, June 1973, pp. 154-160, which discloses a new algorithm based upon digital inverse filter formulation for automatically determining a voiced-unvoiced decision, and the first three formant frequencies as a function of time.
Furthermore, attention is drawn to the European Patent Application EP-A-415163, which discloses a method and apparatus for determining the lag of a long term filter in a code excited linear prediction speech coder. An open loop lag is first determined using an autocorrelation function. The open loop lag is then utilized to generate a limited range over which a closed loop search is performed. The range for appropriate values includes lags that are harmonically related to open loop lag as well as adjacent lags.
In accordance with the present invention, a speech coding apparatus for reproducing and coding input speech, as set forth in claim 1, and a speech coding method for reproducing and coding input speech, as set forth in claim 11, is provided. Preferred embodiments of the invention are disclosed in the dependent claims.
Accordingly, it is an object of the present invention to provide a pitch estimation system incorporating multi-resolution analysis for speech coding, requiring minimal complexity and greater precision. In particular embodiments, the present invention is directed to a device and method of speech coding using CELP techniques, as well as a variety of other speech coding and recognition systems. Consequently, better results are provided with fewer computational resources, while maintaining the necessary high precision.
These and other objects are accomplished, according to an embodiment of the invention, by a pitch lag estimation scheme which quickly and efficiently enables the accurate reproduction and regeneration of speech. The pitch lag is extracted for a given speech frame and then refined for each subframe. After a minimum number of speech samples have been obtained by sampling speech directly, a Discrete Fourier Transform (DFT) is applied, and the resultant amplitude is squared. A second DFT is then performed. Accordingly, an accurate initial pitch lag for the speech samples within the frame can be determined between the possible minimum value of 20 samples and the maximum lag value of 147 samples at the 8 KHz sampling rate. After obtaining the initial pitch lag estimate, time domain refinement must be performed for each subframe to further improve the estimation precision.
Figure 1 is a block diagram of a CELP speech model.
Figure 2 is a block diagram of an MBE speech model.
Figure 3 is a block diagram of an MBE encoder.
Figure 4 is a block diagram of pitch lag estimation in an MBE vocoder.
Figure 5 is block diagram of a cepstrum-based pitch lag detection scheme.
Figure 6 is an operational flow diagram of pitch lag estimation according to an embodiment of the present invention.
Figure 7 is a flow diagram of pitch lag estimation according to another embodiment of the present invention.
Figure 8 is a diagrammatic view of speech coding according to the embodiment of Figure 6.
A pitch lag estimation scheme in accordance with a preferred embodiment of the present invention is indicated generally in Figures 6, 7, and 8. First, N speech samples {x(n), n = 0, 1, ... , N-1} are gathered. (Step 602 of Figure 6) For example, N may equal 320 speech samples to accommodate a typical 40 ms speech window at an 8000 Hz sampling rate. The value of N is determined by the roughly estimated speech period, wherein at least two periods are generally required to generate the speech spectrum. Thus, N must be greater than twice the maximum possible pitch lag, where { x(n), n = 0, 1, ... , N-1). In addition, a Hamming window 604 or other window which covers at least two pitch periods is preferably implemented.
An N-point DFT is applied in step 606 over {x(n), n = 0, 1, ... , N-1} to get amplitude {Y(f), f = 0,1, ... , N-1}, where
Y(f) is then squared in step 608 according to: G(f)=|Y(f)|2 for f = 0, 1, ..., N-1 A second N-point DFT is applied to G(f) in Step 610 to obtain
It will be recognized that, according to embodiments of the present invention, C(n) is unlike the conventional cepstrum transformation in which the logarithm of G(f) is used in Eqn. (4) rather than the function G(f). This difference is generally attributable to complexity concerns. It is desirable to reduce the complexity by eliminating the logarithmic function, which otherwise requires substantially greater computational resources. In addition, upon comparison of pitch lag estimation schemes using cepstrum or the C(n) function, varying results have been obtained only for unvoiced or transition segments of the speech. For example, for unvoiced or transition speech, the definition of pitch is unclear. It has been said that there is no pitch in transition speech, while others say that some prediction can always be designated to minimize the error.
Accordingly, once C(n) is determined (step 610), the pitch lag for the given speech frame can be found in step 614 by solving the following:
where arg [•] determines the variable n which satisfies the inside optimization function, L₁ and L₂ are defined as the minimum and maximum possible pitch lag, respectively. For speech coding convenience, it is desirable for the difference between L₂ and L₁ to be a power of 2 for the binary representation. In preferred embodiments, L₁ and L₂ take values of 20 and 147, respectively, to cover the typical human speech pitch lag range of 2.5 to 18.375 ms, where the distance between L₁ and L₂ is a power of 2. W(i) is a weighting function, and 2M+1 represents the window size. Preferably, {W(i) = 1, i = 0, 1,..., 2M}, and M = 1.
Although the resultant pitch lag is an averaged value, it has been found to be reliable and accurate. The averaging effect is due to the relatively large analysis window size; for a lag of 147 samples, the window size should be at least twice as large as the lag value. Undesirably, however, with such a large window, signals from some speakers, such as female talkers who typically display a small pitch lag, may contain 4-10 pitch periods. If there is a change in the pitch lag, the proposed pitch lag estimation only produces an averaged pitch lag. As a result, the use of such an averaged pitch lag in speech coding could cause severe degradation in speech estimation and regeneration.
Due to relatively quick changes of pitch information in speech, most speech coding systems based on the CELP model evaluate and transmit the pitch lag once per subframe. Thus, in CELP type speech coding in which one speech frame is divided into several speech subframes which are typically 2-10 ms long (16-80 samples), pitch information is updated in each of the subframes. Accordingly, correct pitch lag values are needed only for the subframes. The pitch lag estimated according to the above scheme, however, does not have sufficient precision for accurate speech coding due to the averaging effect.
Thus, in particular embodiments of the present invention, to improve the estimation precision, a refined search based on the initial pitch lag estimate is performed in the time domain (Step 618). A simple autocorrelation method is performed around the averaged Lag value for the particular coding period, or subframe:
where arg [•] determines the variable n which satisfies the inside optimization function, k denotes the first sample of the subframe, l represents the refine window size and m is a searching range. To determine an accurate pitch lag value, the refine window size should be at least one pitch period. The window, however, should not be too large to avoid the effects of averaging. For example, preferably, l = Lag + 10, and m = 5. Thus, according to the time domain refinement of Eqn. 6, a more precise pitch lag can be estimated and applied to the coding of the subframe.
In operation, although the Fast Fourier Transform (FFT) is sometimes more computationally efficient than the general DFT, the drawback of using an FFT is that the window size must be power of 2. For example, it has been shown that the maximum pitch lag of 147 samples is not a power of 2. To include the maximum pitch lag, a window size of 512 samples is necessary. However, this results in a poor pitch lag estimation for female voices due to the averaging effect, discussed above, and the large amount of computation required. If a window size of 256 samples is used, the averaging effect is reduced and the complexity is less. However, to use such a window, a pitch lag larger than 128 samples in the speech cannot be accommodated.
To overcome some of these problems, an alternative preferred embodiment of the present invention utilizes a 256-point FFT to reduce the complexity, and employ a modified signal to estimate the pitch lag. The modification of the signal is a down sampling process. Referring to Figure 7, N speech samples are gathered (Step 702), with N being greater than twice the maximum pitch lag, {x(n), n = 0, 1, ... , N-1}. The N speech samples are then down-sampled into 256 new analysis samples (Step 704) using linear interpolation, according to: y(i) = x([i • λ])+{x([i • λ]+1)-x([i • λ])}(i • λ-[i • λ]) for i = 0, 1, ... , 255 where λ=N/256, and the values within the brackets, i.e., [i • λ], denote the largest integer value not greater than i • λ. A Hamming window, or other window, is then applied to the interpolated data in step 705.
In step 706, the pitch lag estimation is performed over y(i) using a 256-point FFT to generate the amplitude Y(f). Steps 708-710 are then carried out similarly to those described with regard to Figure 6. In addition, however, G(f) is filtered (step 709) to reduce the high frequency components of G(f) which are not useful for pitch detection. Once the lag of y(i), i.e., Lag_y, is found (step 714) according to Eqn. (5), it is rescaled in step 716 to determine the pitch lag estimate: Lag = Lagy • λ
In summary, the above procedure to find an initial pitch estimate for the coding frame is as follows:
(1) subdividing the standard 40 ms coding frame into pitch subframes 802 and 804, each pitch subframe being approximately 20 ms long;
(2) taking N = 320 speech samples such that the pitch analysis window 806 is positioned at the center of the last subframe, and find the lag for that subframe using the proposed algorithm; and
(3) determining initial pitch lag values for the pitch subframes.
More particularly, time domain refinement is performed over the original speech samples. For example, first, the 40 ms coding frame is divided into eight 5 ms coding subframes 808, as shown in Figure 8. Initial pitch lag estimates lag₁ and lag₂ are the lag estimates for the last coding subframe of each pitch subframe in the current coding frame. Lag₀ is the refined lag estimate of the second pitch subframe in the previous coding frame. The relationship among lag₁, lag₂, and lag₀ is shown in Figure 8.
The initial pitch lags lag₁ and lag₂ are refined first to improve their precision (step 718 in Figure 7) according to:
where N₁ is the index of the starting sample in the pitch subframe for its pitch lag₁. Preferably, M is selected to be 10, L is lag_i + 10, and i indicates the index of the pitch subframe.
Once the refinement of initial pitch lags is finished, the pitch lags of the coding subframes can be determined. The pitch lags of the coding subframes are estimated by linearly interpolating lag₁, lag₂, and lag₀. The precision of the pitch lag estimates of the coding subframes is improved by refining the interpolated pitch lag of each coding subframe according to the following procedure. If {lag₁(i), i = 0, 1, ..., 7} represents the interpolated pitch lags of coding subframes based on the refined initial pitch estimates lag₁, lag₂, and lag₀, lag_I(i) is determined by:
Because the precision of the pitch lag estimates given by linear interpolation is not sufficient, further improvement may be required. For the given pitch lag estimates {lag_I(i), i = 0, 1, ..., 7}, each lag_I(i) is further refined (step 722) by:
where Ni is the index of the starting sample in the coding subframe for pitch lag(i). In the example, M is chosen to be 3, and L equals 40.
Furthermore, the linear interpolation of pitch lag is critical in unvoiced segments of speech. The pitch lag found by any analysis method tends to be randomly distributed for unvoiced speech. However, due to the relatively large pitch subframe size, if the lag for each subframe is too close to the initially determined subframe lag (found in step (2) above), an artificial periodicity that originally was not in the speech is added. In addition, linear interpolation provides a simple solution to problems associated with poor quality unvoiced speech. Moreover, since the subframe lag tends to be random, once interpolated, the lag for each subframe is also very randomly distributed, which guarantees voice quality.

Claims

A speech coding apparatus for reproducing and coding input speech, the speech coding apparatus operable with linear prediction coding (LPC) parameters and an innovation codebook representing a plurality of vectors which are referenced to excite speech reproduction to generate speech, the speech coding apparatus comprising:

speech input means (602) for receiving the input speech;

a computer for processing the input speech, wherein the computer includes:

means for segregating a current coding frame within the input speech,

means for dividing the coding frame into plural pitch subframes (802, 804),

means for defining a pitch analysis window (806) having N speech samples, the pitch analysis window extending across the pitch subframes (802, 804),

means for estimating an initial pitch lag value (714) for each pitch subframe (802, 804),

means for dividing each pitch subframe (802, 804) into multiple coding subframe (808), wherein the initial pitch lag estimate for each pitch subframe (802, 804) represents the lag estimate for the last coding subframe (808) of each pitch subframe (802, 804) in the current coding frame, and

means for linearly interpolating (720) the estimated pitch lag values (714) between the pitch subframes (802, 804) to determine a pitch lag estimate for each coding subframe (808), and

means for refining (722) the linearly interpolated lag values (720) of each coding subframe; and

speech output means for outputting speech reproduced according to the refined pitch lag values (722).
The apparatus of claim 1, further comprising sampling means which sample the input speech at a sampling rate R, wherein the N speech samples are determined according to the equation N = R * X, and wherein X is a downsampling value for representation by fewer samples.
The apparatus of claim 2, wherein X = 25 ms, R = 8000 Hz, and N = 320 samples.
The apparatus of claim 1, wherein each coding frame has a length of. approximately 40 ms.
A system as in claim 1 further system comprising:

means for applying a first discrete Fourier transform (606) (DFT) to the samples, the first DFT having an associated amplitude;

means for squaring the amplitude (608) of the first DFT (606);

means for applying a second DFT (610) over the squared amplitude (608);
The system of claim 5, wherein the initial pitch lag value has an associated prediction error, and the means for refining the initial pitch lag value, minimize the associated prediction error.
The system of claim 5, further comprising:

means for estimating initial pitch lag estimates lag₁ and lag₂ (714) hich represent the lag estimates, respectively, for the last coding subframe (808) of each pitch subframe in the current coding frame;

means for refining (718) the pitch lag estimate lag₀ of the second pitch subframe in the previous coding frame;

means for linearly interpolating (720) lag₁, lag₂, and lag₀ to estimate pitch lag values (714) of the coding subframes (808).
The system of claim 1, further comprising means for downsampling (704) the speech samples to a downsampling value for approximate representation by fewer samples.
The system of claim 8, wherein the initial pitch lag value is scaled (716) according to the equation: Lag_scaled = Number speech samples/Downsampling value.
The system of claim 5, wherein the means for refining the initial pitch lag value comprises autocorrelation.
A speech coding method for reproducing and coding input speech, the speech coding apparatus operable with linear prediction coding (LPC) parameters and an innovation codebook representing pseudo-random signals which form a plurality of vectors which are referenced to excite speech reproduction to generate speech, the speech coding method comprising the steps of:

receiving (602) and processing the input speech;

processing the input speech, wherein the step of processing includes:

determining a speech coding frame within the input speech,

subdividing the coding frame into plural pitch subframes (802, 804),

defining a pitch analysis window (806) having N speech samples, the pitch analysis window extending across the pitch subframes (802, 804),

roughly estimating an initial pitch lag value (714) for each pitch subframe (802, 804),

dividing each pitch subframe (802, 804) into multiple coding subframes (808) such that the initial pitch lag estimate for each pitch subframe (802, 804) represents the lag estimate for the last coding subframe (808) of each pitch subframe (802, 804), and

linearly interpolating (720) the estimated pitch lag values (714) between the pitch subframes (802, 804) for determining a pitch lag estimate for each coding subframe (808), and

refining (722) the linearly interpolated lag values (720); and

outputting speech reproduced according to the refined pitch lag values (722).
The method of claim 11, further comprising the steps of sampling the input speech at a sampling rate R, such that the N speech samples are determined according to the equation N = R * X, wherein X is a downsampling value for repesentation by fewer samples.
A method of claim 11 further comprising the steps of:

applying a first discrete Fourier transform (606) (DFT) to the samples, the first DFT having an associated amplitude;

squaring the amplitude (608) of the first DFT (606);

applying a second DFT (610) over the squared amplitude (608) of the first DFT (606);

wherein the initial pitch lag value has an associated prediction error; and

the step of refining the initial pitch lag value uses autocorrelation to minimize the associated prediction error.
The method of claim 13, further comprising the steps:

estimating initial pitch lag estimates lag₁ and lag₂ (714) which represent the lag estimates, respectively, for the last coding subframe (808) of each pitch subframe (802, 804) in the current coding frame;

refining (718) the pitch lag estimate lag₀ of the second pitch subframe in the previous coding frame;

linearly interpolating (720) lag₁, lag₂, and lag₀ to estimate pitch lag values (714) of the coding subframes (808).
The method of claim 11, further comprising the step of downsampling (704) the speech samples to a downsampling value for approximate representation by fewer samples.
The method of claim 11, further comprising the step of scaling (716) the initial pitch lag value according to the equation: Lag_scaled = Number speech samples/Downsampling value.