The present invention relates to a method and device for searching a fixed codebook having an algebraic structure. The codebook searching method and device according to the invention can be used in a technique for encoding and decoding sound signals (including speech and audio signals).
The demand for efficient digital wideband speech/audio encoding techniques with a good subjective quality/bit rate trade-off is increasing for numerous applications such as audio/video teleconferencing, multimedia, and wireless applications, as well as Internet and packet network applications. Until recently, telephone bandwidths filtered in the range of 200-3400 Hz were mainly used in speech coding applications. However, there is an increasing demand for wideband speech applications in order to increase the intelligibility and naturalness of the speech signals. A bandwidth in the range 50-7000 Hz was found sufficient for delivering a face-to-face speech quality. For audio signals, this range gives an acceptable audio quality, but is still lower than the CD (Compact Disk) quality which operates in the range 20-20000 Hz.
A speech encoder converts a speech signal into a digital bit stream which is transmitted over a communication channel (or stored in a storage medium). The speech signal is digitized (sampled and quantized with usually 16-bits per sample) and the speech encoder has the role of representing these digital samples with a smaller number of bits while maintaining a good subjective speech quality. The speech decoder or synthesizer operates on the transmitted or stored bit stream and converts it back to a sound signal.
One of the best prior art techniques capable of achieving a good quality/bit rate trade-off is the so-called CELP (Code Excited Linear Prediction) technique. According to this technique, the sampled speech signal is processed in successive blocks of L samples usually called frames where L is some predetermined number (corresponding to 10-30 ms of speech). In CELP, an LP (Linear Prediction) synthesis filter is computed and transmitted every frame. The L-sample frame is then divided into smaller blocks called subframes of N samples, where L=kN and k is the number of subframes in a frame (N usually corresponds to 4-10 ms of speech). An excitation signal is determined in each subframe, which usually consists of two components: one from the past excitation (also called pitch contribution or adaptive codebook) and the other from an innovative codebook (also called fixed codebook). This excitation signal is transmitted and used at the decoder as the input of the LP synthesis filter in order to obtain the synthesized speech.
To synthesize speech according to the CELP technique, each block of N samples is synthesized by filtering an appropriate codevector from the innovative codebook through time-varying filters modeling the spectral characteristics of the speech signal. These filters consist of a pitch synthesis filter (usually implemented as an adaptive codebook containing the past excitation signal) and an LP synthesis filter. At the encoder end, the synthesis output is computed for all, or a subset, of the codevectors from the innovative codebook (codebook search). The retained innovative codevector is the one producing the synthesis output closest to the original speech signal according to a perceptually weighted distortion measure. This perceptual weighting is performed using a so-called perceptual weighting filter, which is usually derived from the LP synthesis filter.
In the CELP context, an innovative codebook is an indexed set of N-sample-long sequences which will be referred to as N-dimensional codevectors. Each codebook sequence is indexed by an integer k ranging from 0 to Mc−1 where Mc represents the size of the innovative codebook often expressed as a number of bits b, where Mc=2b.
A codebook can be stored in a physical memory, e.g. a look-up table (stochastic codebook), or can refer to a mechanism for relating the index to a corresponding codevector, e.g. a formula (algebraic codebook).
A drawback of the first type of codebooks, the stochastic codebooks, is that they often involve substantial physical storage. They are stochastic, i.e. random in the sense that the path from the index to the associated codevector involves look-up tables which are the result of randomly generated numbers or statistical techniques applied to large speech training sets. The size of stochastic codebooks tends to be limited by storage and/or search complexity.
The second type of codebooks are the algebraic codebooks. By contrast with the stochastic codebooks, algebraic codebooks are not random and require no substantial storage. An algebraic codebook is a set of indexed codevectors of which the amplitudes and positions of the pulses of the kth codevector can be derived from a corresponding index k through a rule requiring no, or minimal, physical storage. Therefore, the size of algebraic codebooks is not limited by storage requirements. Algebraic codebooks can also be designed for efficient search.
The CELP model has been very successful in encoding telephone band sound signals, and several CELP-based standards exist in a wide range of applications, especially in digital cellular applications. In the telephone band, the sound signal is band-limited to 200-3400 Hz and sampled at 8000 samples/sec. In wideband speech/audio applications, the sound signal is band-limited to 50-7000 Hz and sampled at 16000 samples/sec.
An important issue that arises in coding wideband signals is the need to use very large excitation codebooks. Therefore, efficient codebook structures that require minimal storage and can be rapidly searched become very important. Algebraic codebooks have been known for their efficiency and are now widely used in various speech coding standards. Algebraic codebooks with larger number of bits can be searched efficiently using non-exhaustive search methods. Examples are the nested-loop search [4], the depth-first tree search [5] that searches pulses in subsets of pulses, and the global pulse replacement [6]. A simple search was used in ITU-T Recommendation G.723.1 [7] similar to the multipulse sequential search [3]. In Reference [7], the excitation consists of several signed pulses in a frame (no track structure as in ACELP) with a fixed gain for all pulses. The pulses are sequentially searched by updating the so-called backward filtered target signal d(n) and placing the new pulse at the absolute maximum of the signal d(n). The search is repeated for several gain values but the gain is assumed constant during each iteration.
More specifically, according to the present invention, there is provided a method of searching an algebraic codebook during encoding of a sound signal, wherein the algebraic codebook comprises a set of codevectors formed of a number of pulse positions and a number of pulses each having a sign and distributed over the pulse positions. The algebraic codebook searching method comprises: calculating a reference signal for use in searching the algebraic codebook; in a first stage, (a) determining, in relation with the reference signal and among the number of pulse positions, a position of a first pulse; in each of a number of stages subsequent to the first stage, (a) recomputing an algebraic codebook gain, (b) updating the reference signal using the recomputed algebraic codebook gain and (c) determining, in relation with the updated reference signal and among the number of pulse positions, a position of another pulse; and computing a codevector of the algebraic codebook using the signs and positions of the pulses determined in the first and subsequent stages, wherein a number of the first and subsequent stages corresponds to the number of pulses in the codevectors of the algebraic codebook.
The present invention also relates to a device for searching an algebraic codebook during encoding of a sound signal, wherein the algebraic codebook comprises a set of codevectors formed of a number of pulse positions and a number of pulses each having a sign and distributed over the pulse positions, and wherein the algebraic codebook searching device comprises: means for calculating a reference signal for use in searching the algebraic codebook; means for determining, in a first stage, a position of a first pulse in relation with the reference signal and among the number of pulse positions; means for recomputing an algebraic codebook gain in each of a number of stages subsequent to the first stage, means for updating, in each of the subsequent stages, the reference signal using the recomputed algebraic codebook gain and means for determining, in each of the subsequent stages, a position of another pulse in relation with the updated reference signal and among the number of pulse positions; and means for computing a codevector of the algebraic codebook using the signs and positions of the pulses determined in the first and subsequent stages, wherein a number of the first and subsequent stages corresponds to the number of pulses in the codevectors of the algebraic codebook.
The present invention further relates to a device for searching an algebraic codebook during encoding of a sound signal, wherein the algebraic codebook comprises a set of codevectors formed of a number of pulse positions and a number of pulses each having a sign and distributed over the pulse positions, and wherein the algebraic codebook searching device comprises: a first calculator of a reference signal for use in searching the algebraic codebook; a second calculator for determining, in a first stage, a position of a first pulse in relation with the reference signal and among the number of pulse positions; a third calculator for recomputing an algebraic codebook gain in each of a number of stages subsequent to the first stage, a fourth calculator for updating, in each of the subsequent stages, the reference signal using the recomputed algebraic codebook gain and a fifth calculator for determining, in each of the subsequent stages, a position of another pulse in relation with the updated reference signal and among the number of pulse positions; and a sixth calculator of a codevector of the algebraic codebook using the signs and positions of the pulses determined in the first and subsequent stages, wherein a number of the first and subsequent stages corresponds to the number of pulses in the codevectors of the algebraic codebook.
The foregoing and other objects, advantages and features of the present invention will become more apparent upon reading of the following non restrictive description of illustrative embodiments thereof, given by way of example only with reference to the accompanying drawings.
In the appended drawings:
The non-restrictive illustrative embodiment of the present invention is concerned with a method and device for fast codebook search in CELP-based encoders. The codebook searching method and device can be used with any sound signals, including speech and audio signals. The codebook searching method and device can also be applied to narrowband, wideband, or full band signals sampled at any rate.
Still referring to
As illustrated in
Presently, the most widespread speech coding techniques are based on Linear Prediction (LP), in particular CELP. In LP-based coding, the sound signal 230 is synthesized by filtering an excitation 214 through a LP synthesis filter 216 having a transfer function. In CELP, the excitation 214 is typically composed of two parts: a first-stage, adaptive-codebook contribution 222 selected from an adaptive codebook 218 and amplified by an adaptive-codebook gain gp 226 and a second-stage, fixed-codebook contribution 224 selected from a fixed codebook 220 and amplified by a fixed-codebook gain gc 228. Generally speaking, the adaptive codebook contribution 222 models the periodic part of the excitation and the fixed codebook contribution 224 is added to model the evolution of the sound signal.
The sound signal is processed by frames of typically 20 ms and the LP filter coefficients are transmitted once per frame. In CELP, the frame is further divided in several subframes to encode the excitation. The subframe length is typically 5 ms.
The main principle behind CELP is called Analysis-by-Synthesis where possible decoder outputs are tried (synthesized) already during the coding process and then compared to the original sound signal. The search minimizes the mean-squared error 232 between the input speech signal s(n) 211 and the synthesized speech s′(n) 230 in a perceptually weighted domain, where discrete time index n=0, 1, . . . , N−1, and N is the length of the subframe. The perceptual weighting filter 233 exploits the frequency masking effect and typically is derived from the LP filter A(z). An example of the perceptual weighting filter 233 is given in Equation (1):
where the factors γ1 and γ2 control the amount of perceptual weighting and where 0<γ2<γ1≦1. The traditional perceptual weighting filter of Equation (1) works well for NB (narrowband, bandwidth of 200-3400 Hz) signals. An example of the perceptual weighting filter for WB (wideband, bandwidth of 50-7000 Hz) signals can be found in Reference [2].
Since the memory of the LP synthesis filter 1/A(z) and the weighting filter W(z) is independent of the searched codevectors, this memory can be subtracted from the input speech signal s(n) prior to the fixed codebook search. Filtering of the candidate codevectors can then be done by means of a convolution with the impulse response of the cascade of the filters 1/A(z) and W(z), represented by H(z) in
The bit stream transmitted from the encoder 210 to the decoder 212 contains typically the following parameters: the quantized parameters of the LP synthesis filter A(z), the adaptive and fixed codebook indices and the gains gp and gc of the adaptive and the fixed codebooks. The block diagram of the encoder 210 and the decoder 212 containing the described parameters is shown in
Adaptive Codebook Search
The adaptive codebook search in CELP-based codecs will be only briefly described in the following paragraph since such adaptive codebook search is believed to be otherwise well known to those of ordinary skill in the art.
The adaptive codebook search in CELP-based codecs is performed in a weighted speech domain to determine the delay (pitch period) t and the pitch gain (or adaptive codebook gain) gp, and to construct the adaptive codebook contribution of the excitation. The pitch period t is strongly dependent on the particular speaker and its accurate determination critically influences the quality of the synthesized speech.
In recent CELP codecs, a three-stage procedure is used to determine the pitch period t. In the first stage, an estimate Top of the open-loop pitch period is computed for each frame. The open-loop pitch period is typically searched using the weighted sound signal sw(n) and normalized correlation computation; the weighted sound signal sw(n) is calculated as shown in
where x1(n) is the target signal and y1(n) is the filtered adaptive codevector. As shown in
y1(n)=v(n)*h(n) (3)
The filter H(z) 238 is formed by the cascade of the LP synthesis filter 1/A(z) and the perceptual weighting filter W(z). The target signal x1(n) corresponds to the perceptually weighted input speech signal sw(n) after subtracting the zero-input response of the filter H(z) (see subtractor 236).
The pitch gain gp 240 is found by minimizing the mean-squared error between the signals x1(n) and y1(n), and given by the following relation:
The pitch gain gp is usually bounded by 0≦gp≦1.2. In most CELP implementations, the pitch gain gp is quantized with the fixed codebook gain once the innovative codevector is found.
The adaptive codebook contribution 250 is calculated by multiplying the filtered adaptive codevector y1(n) by the pitch gain gp.
Fixed Codebook Search
The objective of searching the fixed (innovative) codebook (FCB) contribution in CELP-based codecs is to minimize the residual error after the use of the adaptive codebook. The residual error is given by the following relation (see subtractor 256 of
where gc is the fixed codebook gain, and y2(k)(n) is the filtered innovative codevector. k is the fixed codebook index and the filtered innovative codevector y2(k)(n) is the codevector ck(n) from the fixed codebook 244 at index k convolved with the impulse response h(n) of the weighted synthesis filter H(z) 246.
The fixed codebook contribution 252 is calculated by multiplying the filtered innovative codevector y2(k)(n) by the fixed codebook gain gc 248.
The algebraic fixed codebook target signal x2(n) is computed by subtracting the adaptive codebook contribution 250 from the adaptive codebook target signal x1(n) (see subtractor 254):
x2(n)=x1(n)−gpy1(n). (6)
Minimizing E from Equation (5) results in the optimum fixed codebook gain gc:
and the minimum error from Equation (5) then results in:
Thus, the search is performed by maximizing the term:
The fixed codebook can be implemented in several ways. One of the most frequent implementations consists of using an algebraic codebook [1] in which a set of pulses is placed in each subframe. The efficiency of such an algebraic codebook depends on the number of pulses, their signs, positions and amplitudes. Since large codebooks are used to guarantee a high subjective quality of the coding, an efficient codebook search is also implemented.
In Algebraic CELP (ACELP (Algebraic Code Excited Linear Prediction)) codecs, the algebraic fixed codebook vector (hereinafter denoted as fixed codevector) ck(n) contains M unit pulses with respective signs sj and positions mj, and is thus given by the following relation:
where sj=±1 and δ(n)=1 for n=0, and δ(n)=0 for n≠0. The fixed codevector after filtering through the filter 246 can be then expressed in the form:
In general, the number of pulses M is limited by the bit rate availability. The fixed codebook index (or codeword) k represents the pulse positions and signs in each subframe. Thus no codebook storage is needed, since the selected codevector can be reconstructed at the decoder through the information contained in the index k itself without lookup tables. Unlike the multi-pulse approach [3], the algebraic fixed codebook gain g, is the same for all the pulses.
Let us denote ck the algebraic codevector at the codebook index k, and y2(k) the corresponding codevector filtered through the filter H(z) 246 (
Where T denotes vector transpose and H is the lower triangular Toeplitz convolution matrix with diagonal h(0) and lower diagonals h(1), . . . , h(N−1):
Vector d=HTx2 is the correlation between x2(n) and h(n), also known as the backward filtered target vector (since it can be computed using time-reversed filtering of x2(n) through the weighted synthesis filter:
and matrix Φ=HTH is the matrix of correlations of h(n). Both d and Φ are usually computed prior to the codebook search. If the algebraic codebook contains only a few non-zero pulses, the computation of the maximization criterion for all possible indexes k is very fast [1].
Algebraic codebooks with larger number of bits can be searched efficiently using non-exhaustive search methods. Examples are the nested-loop search [4], the depth-first tree search [5] that searches pulses in subsets of pulses, and the global pulse replacement [6]. A simple search was used in ITU-T Recommendation G.723.1 [7] similar to the multipulse sequential search [3]. In Reference [7], the excitation consists of several signed pulses in a frame (no track structure as in ACELP) with a fixed gain for all pulses. The pulses are sequentially searched by updating the backward filtered target vector d(n) and placing the new pulse at the absolute maximum of d(n). The search is repeated for several gain values but the gain is assumed constant during each iteration. The embodiment of the present invention disclosed in this specification is concerned with a method and device for searching an algebraic codebook wherein the frame can be divided into interleaved tracks of pulse positions and where several pulses are placed in each track. The disclosed codebook searching method and device implement the use of a sequential search of the pulses by maximizing a certain criterion based on a maximum likelihood signal. The fixed codebook gain is then recomputed at each stage. Several iterations can be used by changing the order of the searched tracks.
Several non-restrictive embodiments of the codebook searching method and device will be disclosed in the following description to illustrate the present invention.
Algebraic Fixed Codebook Structure
The codebook structure can be based on an interleaved single-pulse permutation (ISPP) design. In this structure, the pulse positions are divided into several tracks of interleaved positions. For example, a 64-position codevector that is divided into 4 tracks T0, T1, T2 and T3 of interleaved positions results in 16 positions in each track as shown in Table I below. This structure will be used in the following examples.
If a single signed pulse is placed in each track (M=4), the pulse position is encoded with 4 bits and its sign is encoded with 1 bit, resulting in a 20-bit codebook. If two signed pulses are placed in each track, the two pulse positions are encoded with 8 bits and their corresponding signs can be encoded with only 1 bit by exploiting pulse ordering; therefore a total of 4×(4+4+1)=36 bits are required to specify the pulse positions and signs for this particular algebraic codebook structure. Other codebook structures can be designed, for example, by placing 3, 4, 5 or 6 pulses in each track T0, T1, T2 and T3. The encoding of the pulses in each track is described in Reference [8].
Another example of codebook structure comprises a 64-position codevector divided into 2 tracks T0 and T1 of interleaved positions resulting in 32 positions in each track as shown in Table II. If a single signed pulse is placed in each track, the pulse position is encoded with 5 bits and its sign is encoded with 1 bit, resulting in a 12-bit codebook. Again, other codebook structures can be designed by placing more pulses in each track, or by fixing the signs of some pulses.
Other combinations of number of tracks and number of pulses per track can be used; the above 12-bit and 20-bit codebooks have been shown in detail because they are used in the ITU-T Recommendation G.718 codec implementation framework that will be summarized herein below.
As already stated, in the 20-bit codebook with the structure as described in Table I each pulse position in one track is encoded with 4 bits and the sign of the pulse is encoded with 1 bit. The position index is given by the pulse position in the subframe divided by the number of tracks (integer division). The division remainder gives the track index. For example, a pulse at position 31 has a position index of 31/4=7 and it belongs to the track with index 3 (fourth track). In this illustrative embodiment, the sign index is set to 0 for positive signs and 1 for negative signs. The index of the signed pulse is thus given by the following relation:
Im=m+s×2P. (15)
where m is the position index, s is the sign index, and P=4 is the number of bits per track.
The Autocorrelation Approach
A common approach to simplify the FCB (Fixed Codebook) search procedure is to use the autocorrelation method [9]. In accordance with this approach, the matrix of correlations Φ from Equation (12) with elements:
is reduced to a Toeplitz form by modifying the summation limits in Equation (16) so that φ(i, j)=α(|i−j|), where:
The autocorrelation approach results from modifying the N×N convolution matrix of Equation (13) into a (2N−1)×N matrix of the form:
The convolution Hck using this matrix results into a 2N−1 long codevector obtained when convolving two segments each of length N. In the covariance approach only the first N samples of the convolution are considered and any samples beyond this subframe limit are not taken into consideration. This approach can be used in the technique according to the invention.
Using the autocorrelation approach means that the mean-squared weighted error is minimized over 2N−1 samples. This requires computing the target signal x2(n) over 2N−1 samples by inputting zero-value samples after the N sound samples into the weighted synthesis filter H(z) 246. Consequently, the computation of the signal x2(n) given by d=HTx2 will be modified to take into account the new matrix dimensions. As an approximation, the computation of the signals x2(n) and d(n) can be performed as in the conventional approach, but the computation of the energy of the filtered fixed codevector y2(k)(n) can be performed using the autocorrelation approach.
From Equations (10)-(12), it can be shown that for an algebraic fixed codebook with M pulses, the criterion to be maximized can be written as:
Using the autocorrelation approach, this can be expressed as:
From Equation (7), the algebraic codebook gain can be expressed as:
and in case of the autocorrelation approach:
The autocorrelation approach has been used in sequential multipulse search [3] since, for a single pulse, the search criterion reduces to placing the pulse at the absolute maximum of d(n).
Fast Algebraic Fixed Codebook Search
The method and device for conducting a fast algebraic codebook search in, for example, a fixed codebook will now be described. The general idea behind the method and device for conducting a fast algebraic codebook search is to search pulses sequentially in several iterations. In the following non-restrictive illustrative embodiments, the autocorrelation approach will be used. However the more usual covariance approach [8] can be used as well. The fundamental principle of the method and device resides in updating the fixed codebook gain gc and the backward filtered target vector d(n) after each new pulse is determined. The basic search can be summarized by the following steps.
The following description explains the use of the method and device for conducting a fast algebraic codebook search in fixed codebooks that consist of several tracks of interleaved positions, where M is the number of pulses, L the number of tracks and N the subframe length. First a description of the specific situation where M=L=4 will be given. The procedure will be then generalized for M pulses (when still M=L) and further extended for the case where M≠L.
Generic Procedure for the Disclosed Search Method and Device
An example of implementation of the method and device for conducting a fast algebraic codebook search, for searching a fixed codebook with 4 tracks of pulse positions and one pulse per track will now be described.
The FCB search procedure starts with computing the backward filtered target vector d(n) (in this embodiment a reference signal used for searching the algebraic fixed codebook) defined by Equation (14) and the vector α(k) defined by Equation (17) (or the matrix φ(i, j) defined by Equation (16)). In the following description, the index i represents the position of a pulse in a track (see Table I or Table II), and the index n represents the number of a sample in a subframe, wherein n=0, . . . , N−1.
In the first iteration, m0 designates the pulse position determined in track T0, m1 the pulse position determined in track T1, m2 the pulse position determined in track T2 and m3 the pulse position determined in track T3.
For a single pulse, the criterion in Equation (19) is reduced to:
and in case of the autocorrelation approach, Equation (20) is reduced to:
As can be seen from Equation (24), the position of the first pulse is found as the index of the maximum absolute value of the backward filtered target vector d(i) for iεT0, i.e.:
m0=index(max(|d(i)|)) (25)
and its sign is given by the sign of d(m0), i.e.:
s0=sgn(d(m0)). (26)
From Equation (22), the gain of the first pulse is given by the relation:
or in the case of the autocorrelation approach by the relation:
In the second stage (second pulse search), the target signal is updated by subtracting the first pulse contribution from the target signal x2(n) as follows:
x2(1)(n)=x2(n)−gc(0)y2(0)(n). (29)
The upper index in brackets used above is from the range [0, . . . , M−1] and corresponds to the searched pulse number j. Note that the codebook index k is omitted for the sake of simplicity and clarity to describe the signal y2(k)(n).
Using Equation (11), the Equation (29) can be written as:
x2(1)(n)=x2(n)−gc(0)s0h(n−m0). (30)
To find the second pulse position and gain, the backward filtered target vector d(i) for iεT1 is updated as follows:
In case of the autocorrelation approach, the backward filtered target vector d(n) is updated as follows:
d(1)(i)=d(i)−s0gc(0)α(|i−m0|) (32)
Similar to Equations (25) and (26), the position and sign of the second pulse are found for iεT1 using the following relations:
m1=index(max(|d(1)(i)|)), (33)
s1=sgn(d(1)(m1)). (34)
The third stage is performed in the same manner as the second stage. The only difference is that we take into account both first and second pulse contributions to find the position and sign of the third pulse.
From Equation (21), the gain gc after two pulses is recomputed using the following relation:
and from Equation (22) for the autocorrelation approach:
The update of the target signal is made using the following relation:
x2(2)(n)=x2(n)−gc(1)y2(1)(n)=x2(n)−gc(1)s0h(n−m0)−gc(1)s1h(n−m1) (37)
and the update of the vector d(i) for iεT2 is made using the following relation:
and using the autocorrelation approach by the following relation:
d(2)(i)=d(i)−s0gc(1)α(|i−m0|)−s1gc(1)α(|i−m1|). (39)
Similar to Equations (25) and (26), the position and the sign of the third pulse are found for iεT2 as follows:
m2=index(max(|d(2)(i)|)), (40)
s2=sgn(d(2)(m2)). (41)
Similarly, in the fourth stage, using the autocorrelation approach, the update of the backward filtered target vector d(n) is made for iεT3 as follows:
d(3)(i)=d(i)−s0gc(2)α(|i−m0|)−s1gc(2)α(|i−m1|)−s2gc(2)α(|i−m2|), (42)
where the fixed codebook gain gc(2) for the third pulse is given by:
and the position and sign of the fourth pulse are found for iεT3 using the following relations:
m3=index(max(|d(3)(i)|)), (44)
s3=sgn(d(3)(m3)). (45)
Using the above procedure, the positions and signs of all 4 pulses are found.
The above procedure is repeated L=4 times by starting each iteration at a different track. For example, in the second iteration, pulse position m0 is assigned to track T1, pulse position m1 is assigned to track T2, pulse position m2 is assigned to track T3, and pulse position m3 is assigned to track T0. Finally, the selected pulse positions and signs of the iteration that minimizes the mean-squared weighted error are chosen to form the final fixed codevector and filtered fixed codevector. More specifically, after all the iterations, the best set of pulse positions and signs are chosen as the those that maximize the following criteria:
where y2(k) (n) is given by Equation (11) for an optimal codebook index k.
This procedure can be easily extended to more than 4 pulses and for different methods of performing the iterations. Also this procedure can be extended to the case where several pulses are placed in each track of pulse positions.
For the case of 4 pulses in 4 tracks, the procedure can be summarized as below using the following assumptions. The pulses are searched sequentially and the backward filtered target vector d(n) (in this embodiment a reference signal used for searching the algebraic fixed codebook) is updated at each stage. The number of stages is equal to the number of pulses M. The number of iterations is equal to the number of tracks L. The autocorrelation approach is used.
Procedure for Searching M Pulses in M Tracks
The method and device for conducting a fast algebraic codebook search as described in above can be further generalized for M pulses as follows. In this example, the number of tracks is equal to the number of pulses to search, that is M=L.
The procedure can be summarized by the following operations:
Procedure for Searching M Pulses in L Tracks
The above procedure can be further extended for a situation where a number of M pulses is searched in a number of L tracks, M being an integer multiple of L. In this example, there are several pulses per track. This situation also covers the case when only one track is used (i.e. the general case when the ISPP approach is not used).
The pulses in the same track are searched sequentially using Equations (47) to (60). The pulses in a track are searched for all the positions of the track. There could be some situations when two or more pulses occupy the same position. If these pulses have the same signs, they add and strengthen the codebook contribution at this position. The case where the pulses have opposite signs is not allowed.
The sequential search of multiple pulses per track is sensitive to the search pulse order. There are two basic sequential search approaches that can be used. The first one supposes that all the pulses in one track are searched before searching the other tracks. The second approach supposes that the first pulse is searched in track T0, the second pulse in track T1, etc. If needed, the pulses are searched again in the following tracks up to track TL-1, one pulse per track, etc. An example of these two approaches is shown in Table III. As experimentally observed the second approach achieves better results and is therefore used in the following example of implementation. If more complexity can be afforded, both approaches can be used however resulting in more iterations.
Yet another approach can be based on some criterion to select the track the next pulse is searched in. Such criterion can be, for example, the absolute maximum of the backward filtered target vector d(n) or its update. The criterion can be used only to select tracks where all the pulses have not yet been assigned.
Search within a Reference Signal
To further improve the efficiency of the search procedure, the amplitude and sign of the pulses can be determined on the basis of a reference signal b(n). In the signal-selected pulse amplitude approach used for example in AMR-WB [8], the sign of a pulse at position n is set equal to the sign of the reference signal at that position. Also, the reference signal b(n) can be used to set the positions of some pulses in case of very large algebraic codebooks. The application of the signal-selected pulse amplitude approach in the presented procedure will be discussed later. In the present non-restrictive, illustrative embodiment, the reference signal b(n) is defined as a combination of the backward filtered target vector d(n) and the ideal excitation signal r(n).
The reference signal can be expressed as follows:
which is a weighted sum of the normalized backward filtered target vector d(n) and the ideal excitation signal r(n). Ed=dTd is the energy of the backward filtered target vector, and Er=rTr is the energy of the ideal excitation signal. The value of δ is closer to 1 for small number of pulses and closer to zero for large number of pulses. The reference signal can be also expressed as follows:
where the scaling factor β=δ/(1−δ). In typical implementations, β=4 for 2 pulses (δ=0.8), β=2 for 4 pulses (δ=0.66), and β=1 for 8 pulses (δ=0.5).
The ideal excitation signal r(n) is obtained by filtering the target signal x2(n) through the inverse of the weighted synthesis filter H(z) with zero states. This can be also done by first filtering the target signal x1(n) through the inverse of the filter H(z) with zero states giving r0(n). The signal r0(n) is then updated by subtracting the selected adaptive vector contribution, i.e. r(n)=r0(n)−gpv(n) for n=0, . . . , N−1.
The signal r0(n), or a part of this signal, can be approximated by the LP residual signal to save complexity. In the present exemplary implementation, the signal r0(n) is computed by filtering of the target signal x1(n) through the inverse of the filter H(z) only in the first half of the subframe. The LP residual signal is used in the second half of the subframe. This LP residual signal is calculated using the following relation:
where âk are quantized LP filter coefficients and s(n) is the input speech signal.
As mentioned herein above, the scaling factor β in Equation (62) controls the dependence of the reference signal b(n) on the backward filtered target vector d(n) and is generally lowered as the number of pulses increases. This approach makes an intelligent guess on the potential positions to be considered. The reference signal b(n) defined by Equation (62) is used for determining the pulse positions.
The procedure for searching pulses using the reference signal b(n) can be summarized with the following operation in connection with
The value of the scaling factor β used in the previous equations is constant for all stages. However its value can be changed according to the stage of the search making the value of the scaling factor adaptive. The idea is to increase its value for later stages. This will emphasize the contribution of the updated backward filtered target vector d(n) in the reference signal b(n) for higher stages where the number of pulses left to be determined reduces. In fact, the reference signal b(n) can be in higher stages approximated by the updated backward filtered target vector d(n) only and the procedure from the previous section can be used in higher stages. An example is described further by Equations (87) and (88). The adaptive scaling factor is symbolized in
Preselection of Signs
To further simplify the search, the signal-selected pulse amplitude method described in Reference [10] can be used. Then, the sign of the pulse at a certain position is set equal to the sign of the reference signal b(n) from Equation (62) at that position. For that purpose, a vector zb(n) containing the signs of the original reference signal b(n) is constructed. The vector zb(n) is computed at the beginning of the codebook search process, i.e. prior to entering the iteration loop. In this manner, the signs of the pulses which are searched are pre-selected and Equations (64) and (65) are changed for the following equations:
m0=index(max(zb(n)·b(n))), (72)
s0=zb(m0) (73)
For the other stages the same principle is used and the position and sign of the pulse for j=1 to M−1 are determined using the following relations:
mj=index(max(zb(n)·b(j)(n))), (74)
sj=zb(mj). (75)
The same principle of sign pre-selection can also be used in relation to a search using the backward filtered target vector d(n) where the vector zb(n) contains the signs of the original backward filtered target vector d(n).
Track Order Determination
As indicated in the foregoing description, the search procedure searches pulses sequentially track by track. The order of the tracks can be chosen sequentially in accordance with the track number, i.e. for the 20-bit algebraic fixed codebook the first iteration searches tracks in the order T0-T1-T2-T3, the second iteration in the order T1-T2-T3-T0, etc. However the sequential order of tracks is not optimal and another order of tracks could be advantageous. One possible solution is to order the tracks in accordance with the absolute maximum of the reference signal b(n) in the respective track.
As an example of track ordering, let us suppose a 20-bit algebraic fixed codebook. Further, bT0max is defined as the absolute maximum value of the reference signal b(n) in track T0, bT1max as the absolute maximum value of b(n) in track T1, bT2max as the absolute maximum value of b(n) in track T2 and bT3max as the absolute maximum value of b(n) in track T3. Prior to entering the iteration loop in the search procedure the absolute maximum values of b(n) of the respective tracks are arranged in descending order. Let it be bT1max>bT3max>bT2max>bT0max in the above example. Then the first iteration searches the tracks in the order T0-T1-T3-T2, the second iteration in the order T1-T3-T2-T0, the third iteration in the order T2-T1-T3-T0, and the fourth iteration in the order T3-T1-T2-T0.
The above example track order determination helps to find a more accurate estimate of the potential position of a pulse. This track order determination is implemented in the ITU-T Recommendation G.718 codec. In the case the search is conducted using the backward filtered target vector d(n), the same principle can be used to arrange the track order.
Summary of the Search Procedure
The fast algebraic codebook search method and device can be summarized as follows with reference to
Implementation of the Fast Codebook Search in G.718 Codec
The fast algebraic fixed codebook searching method and device described above was implemented and tested with the ITU-T Recommendation G.718 (previously known as G.EV-VBR) codec baseline that has been recently standardized. The implementation of the fast algebraic fixed codebook search in the G.718 codec correspond to the implementation described above with reference to
The coding of the first layer L1 takes advantage of a signal classification based encoding. Four distinct signal classes are considered in the ITU-T Recommendation G.718 codec for different coding of each frame: Unvoiced coding, Voiced coding, Transition coding, and Generic coding. The algebraic FCB search in L1 employs 20-bit and 12-bit codebooks. Their use in different subframes depends on the coding mode. The FCB search in layer L2 employs the 20-bit codebook in two subframes and the 12-bit codebook in the other two subframes in Generic and Voiced coding frame and the 20-bit codebook in three subframes and the 12-bit codebook in one subframe in Transition and Unvoiced coding frame. The FCB search in G.722.2 option employs 36-bit codebooks in all four subframes. The configuration of these codebooks is summarized in Table IV.
The value of scaling factor β can be set as a constant (same for all stages) as follows:
Nevertheless, as mentioned above, the value of the scaling factor β can be different for every stage. In an example of implementation, it was found that the optimum values of the scaling factor β were the following for a 20-bit algebraic fixed codebook:
and for a 12-bit codebook:
The value β=∞ means that the updated reference signal b(n) is equal to the updated backward filtered target vector d(n) in this stage.
The criterion of Equation (12) can be used in the codec as described above. However to avoid division when comparing between two candidate values, the criterion is implemented using multiplications only, for details see for example Reference [8].
Fast Codebook Search Performance
The performance of the fast algebraic fixed codebook searching method and device described above was tested in the G.718 codec where the original FCB search [8] was replaced by the above described one. The objective was to achieve similar synthesized speech quality with a decrease of complexity.
Tables V to X summarize the new fast FCB search performance measured using segmental signal-to-noise ratio (segmental SNR) values, In the tables, ‘FCB 1’ stands for the technique presented in Reference [8], ‘FCB 2’ for the technique presented in Reference [6], and the technique presented in this report is called ‘new FCB’. A database of clean speech sentences at nominal level comprising both male and female English speakers was used as a speech material. The length of the database was about 456 seconds. The performance of the method within the G.718 codec was evaluated in layers where algebraic fixed codebook search is used, i.e. for layers L1, L2 and the G.722.2-option core layer. This resulted in 3 groups of tests: 8 kbps tests (only layer L1), 12 kbps tests (layers L1 and L2 are used), and G.722.2-option tests for 12.65 kbps. The above described technique was implemented both in 12-bit FCB and 20-bit FCB using algorithms described above. For the G.722.2 option the above described technique was implemented in the 36-bit FCB.
The complexity of the FCB search and the total G.718 encoder complexity are summarized in Table VII and Table IX. The complexity is given in wMOPS (weighted Million Operations Per Second) for the worst case.
As can be seen from Tables V-VII, the presented algorithm reduces computational requirements significantly, but for a cost of a little segmental SNR decrease compared to technique presented in Reference [8]. Therefore it was decided to use the proposed algorithm only in the second layer (L2) in G.718 where the SNR drop is insignificant. The Recommendation G.718 thus employs the fast algebraic fixed codebook search in layer 2. The implementation corresponds to the implementation described above with reference to
The performance was also tested in ITU-T Recommendation G.729.1 codec [6] at 8 kbps where the original FCB search [6] was replaced by the fast algebraic fixed codebook searching method and device described hereinabove. The G.729.1 codec uses 4 subframes of 40 samples. The position of the pulses m0, m1 and m2 are encoded with 3 bits each, while position of the pulse m3 is encoded with 4 bits. The sign of each pulse sign is encoded with 1 bit. This gives a total of 17 bits for the 4 pulses.
Although the present invention has been described in the foregoing specification in relation to non-restrictive illustrative embodiments thereof, these embodiments can be modified at will within the scope of the appended claims without departing from the spirit and nature of the present invention.
Filing Document | Filing Date | Country | Kind | 371c Date |
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PCT/CA2008/001620 | 9/11/2008 | WO | 00 | 6/11/2010 |
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WO2009/033288 | 3/19/2009 | WO | A |
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