Method and device for frequency-selective pitch enhancement of synthesized speech

In a method and device for post-processing a decoded sound signal in view of enhancing a perceived quality of this decoded sound signal, the decoded sound signal is divided into a plurality of frequency sub-band signals, and post-processing is applied to at least one of the frequency sub-band signal. After post-processing of this at least one frequency sub-band signal, the frequency sub-band signals may be added to produce an output post-processed decoded sound signal. In this manner, the post-processing can be localized to a desired sub-band or sub-bands with leaving other sub-bands virtually unaltered.

CROSS-REFERENCE TO RELATED APPLICATIONS

This application is the national phase of International (PCT) Patent Application Serial No. PCT/CA03/00828, filed May 30, 2003, published under PCT Article 21(2) in English, which claims priority to and the benefit of Canadian Patent Application No. 2,388,352, filed May 31, 2002, the disclosures of which are incorporated herein by reference.

BACKGROUND OF THE INVENTION

1. Field of the Invention

The present invention relates to a method and device for post-processing a decoded sound signal in view of enhancing a perceived quality of this decoded sound signal.

This post-processing method and device can be applied, in particular but not exclusively, to digital encoding of sound (including speech) signals. For example, this post-processing method and device can also be applied to the more general case of signal enhancement where the noise source can be from any medium or system, not necessarily related to encoding or quantization noise.

2. Brief Description of the Current Technology

Speech encoders are widely used in digital communication systems to efficiently transmit and/or store speech signals. In digital systems, the analog input speech signal is first sampled at an appropriate sampling rate, and the successive speech samples are further processed in the digital domain. In particular, a speech encoder receives the speech samples as an input, and generates a compressed output bit stream to be transmitted through a channel or stored on an appropriate storage medium. At the receiver, a speech decoder receives the bit stream as an input, and produces an output reconstructed speech signal.

To be useful, a speech encoder must produce a compressed bit stream with a bit rate lower than the bit rate of the digital, sampled input speech signal. State-of-the-art speech encoders typically achieve a compression ratio of at least 16 to 1 and still enable the decoding of high quality speech. Many of these state-of-the-art speech encoders are based on the CELP (Code-Excited Linear Predictive) model, with different variants depending on the algorithm.

In CELP encoding, the digital speech signal is processed in successive blocks of speech samples called frames. For each frame, the encoder extracts from the digital speech samples a number of parameters that are digitally encoded, and then transmitted and/or stored. The decoder is designed to process the received parameters to reconstruct, or synthesize the given frame of speech signal. Typically, the following parameters are extracted from the digital speech samples by a CELP encoder:Linear Prediction Coefficients (LP coefficients), transmitted in a transformed domain such as the Line Spectral Frequencies (LSF) or Immitance Spectral Frequencies (ISF);Pitch parameters, including a pitch delay (or lag) and a pitch gain; andInnovative excitation parameters (fixed codebook index and gain).
The pitch parameters and the innovative excitation parameters together describe what is called the excitation signal. This excitation signal is supplied as an input to a Linear Prediction (LP) filter described by the LP coefficients. The LP filter can be viewed as a model of the vocal tract, whereas the excitation signal can be viewed as the output of the glottis. The LP or LSF coefficients are typically calculated and transmitted every frame, whereas the pitch and innovative excitation parameters are calculated and transmitted several times per frame. More specifically, each frame is divided into several signal blocks called subframes, and the pitch parameters and the innovative excitation parameters are calculated and transmitted every subframe. A frame typically has a duration of 10 to 30 milliseconds, whereas a subframe typically has a duration of 5 milliseconds.

Several speech encoding standards are based on the Algebraic CELP (ACELP) model, and more precisely on the ACELP algorithm. One of the main features of ACELP is the use of algebraic codebooks to encode the innovative excitation at each subframe. An algebraic codebook divides a subframe in a set of tracks of interleaved pulse positions. Only a few non-zero-amplitude pulses per track are allowed, and each non-zero-amplitude pulse is restricted to the positions of the corresponding track. The encoder uses fast search algorithms to find the optimal pulse positions and amplitudes for the pulses of each subframe. A description of the ACELP algorithm can be found in the article ofR. SALAMIet al.,“Design and description of CS-ACELP: a toll quality8kb/s speech coder” IEEE Trans. on Speech and Audio Proc.,Vol. 6, No. 2, pp. 116-130, March 1998, herein incorporated be reference, and which describes the ITU-T G.729 CS-ACELP narrowband speech encoding algorithm at 8 kbits/second. It should be noted that there are several variations of the ACELP innovation codebook search, depending on the standard of concern. The present invention is not dependent on these variations, since it only applies to post-processing of the decoded (synthesized) speech signal.

A recent standard based on the ACELP algorithm is the ETSI/3GPP AMR-WB speech encoding algorithm, which was also adopted by the ITU-T (Telecommunication Standardization Sector of ITU (International Telecommunication Union)) as recommendation G.722.2 . [ITU-T Recommendation G.722.2 “Wideband coding of speech at around16kbit/s using Adaptive Multi-Rate Wideband(AMR-WB)” Geneva, 2002], [3GPP TS 26.190, “AMR Wideband Speech Codec: Transcoding Functions,”3GPP Technical Specification]. The AMR-WB is a multi-rate algorithm designed to operate at nine different bit rates between 6.6 and 23.85 kbits/second. Those of ordinary skill in the art know that the quality of the decoded speech generally increases with the bit rate. The AMR-WB has been designed to allow cellular communication systems to reduce the bit rate of the speech encoder in the case of bad channel conditions; the bits are converted to channel encoding bits to increase the protection of the transmitted bits. In this manner, the overall quality of the transmitted bits can be kept higher than in the case where the speech encoder operates at a single fixed bit rate.

FIG. 7is a schematic block diagram showing the principle of the AMR-WB decoder. More specifically,FIG. 7is a high-level representation of the decoder, emphasizing the fact that the received bitstream encodes the speech signal only up to 6.4 kHz (12.8 kHz sampling frequency), and the frequencies higher than 6.4 kHz are synthesized at the decoder from the lower-band parameters. This implies that, in the encoder, the original wideband, 16 kHz-sampled speech signal was first down-sampled to the 12.8 kHz sampling frequency, using multi-rate conversion techniques well known to those of ordinary skill in the art. The parameter decoder701and the speech decoder702ofFIG. 7are analogous to the parameter decoder106and the source decoder107ofFIG. 1. The received bitstream709is first decoded by the parameter decoder701to recover parameters710supplied to the speech decoder702to resynthesize the speech signal. In the specific case of the AMR-WB decoder, these parameters are:ISF coefficients for every frame of 20 milliseconds;An integer pitch delay T0, a fractional pitch value T0_frac around T0, and a pitch gain for every 5 millisecond subframe; andAn algebraic codebook shape (pulse positions and signs) and gain for every 5 millisecond subframe.
From the parameters710, the speech decoder702is designed to synthesize a given frame of speech signal for the frequencies equal to and lower than 6.4 kHz, and thereby produce a low-band synthesized speech signal712at the 12.8 kHz sampling frequency. To recover the full-band signal corresponding to the 16 kHz sampling frequency, the AMR-WB decoder comprises a high-band resynthesis processor707responsive to the decoded parameters710from the parameter decoder701to resynthesize a high-band signal711at the sampling frequency of 16 kHz. The details of the high-band signal resynthesis processor707can be found in the following publications which are herein incorporated by reference:ITU-T Recommendation G.722.2“Wideband coding of speech at around16kbit/s using Adaptive Multi-Rate Wideband(AMR-WB)”, Geneva, 2002; and3GPP TS26.190,“AMR Wideband Speech Codec: Transcoding Functions,”3GPP Technical Specification.
The output of the high-band resynthesis processor707, referred to as the high-band signal711ofFIG. 7, is a signal at the 16 kHz sampling frequency, having an energy concentrated above 6.4 kHz. The processor708sums the high-band signal711to a 16-kHz up-sampled low-band speech signal713to form the complete decoded speech signal714of the AMR-WB decoder at the 16 kHz sampling frequency.
2.2 Need for Post-Processing

Whenever a speech encoder is used in a communication system, the synthesized or decoded speech signal is never identical to the original speech signal even in the absence of transmission errors. The higher the compression ratio, the higher the distortion introduced by the encoder. This distortion can be made subjectively small using different approaches. A first approach is to condition the signal at the encoder to better describe, or encode, subjectively relevant information in the speech signal. The use of a formant weighting filter, often represented as W(z), is a widely used example of this first approach [B. Kleijn and K. Paliwal editors, <<Speech Coding and Synthesis, >> Elsevier, 1995]. This filter W(z) is typically made adaptive, and is computed in such a way that it reduces the signal energy near the spectral formants, thereby increasing the relative energy of lower energy bands. The encoder can then better quantize lower energy bands, which would otherwise be masked by encoding noise, increasing the perceived distortion. Another example of signal conditioning at the encoder is the so-called pitch sharpening filter which enhances the harmonic structure of the excitation signal at the encoder. Pitch sharpening aims at ensuring that the inter-harmonic noise level is kept low enough in the perceptual sense.

A second approach to minimize the perceived distortion introduced by a speech encoder is to apply a so-called post-processing algorithm. Post-processing is applied at the decoder, as shown inFIG. 1. InFIG. 1, the speech encoder101and the speech decoder105are broken down in two modules. In the case of the speech encoder101, a source encoder102produces a series of speech encoding parameters109to be transmitted or stored. These parameters109are then binary encoded by the parameter encoder103using a specific encoding method, depending on the speech encoding algorithm and on the parameters to encode. The encoded speech signal (binary encoded parameters)110is then transmitted to the decoder through a communication channel104. At the decoder, the received bit stream111is first analysed by a parameter decoder106to decode the received, encoded sound signal encoding parameters, which are then used by the source decoder107to generate the synthesized speech signal112. The aim of post-processing (see post-processor108ofFIG. 1) is to enhance the perceptually relevant information in the synthesized speech signal, or equivalently to reduce or remove the perceptually annoying information. Two commonly used forms of post-processing are formant post-processing and pitch post-processing. In the first case, the formant structure of the synthesized speech signal is amplified by the use of an adaptive filter with a frequency response correlated to the speech formants. The spectral peaks of the synthesized speech signal are then accentuated at the expense of spectral valleys whose relative energy becomes smaller. In the case of pitch post-processing, an adaptive filter is also applied to the synthesized speech signal. However in this case, the filter's frequency response is correlated to the fine spectral structure, namely the harmonics. A pitch post-filter then accentuates the harmonics at the expense of inter-harmonic energy which becomes relatively smaller. Note that the frequency response of a pitch post-filter typically covers the whole frequency range. The impact is that a harmonic structure is imposed on the post-processed speech even in frequency bands that did not exhibit a harmonic structure in the decoded speech. This is not a perceptually optimal approach for wideband speech (speech sampled at 16 kHz), which rarely exhibits a periodic structure on the whole frequency range.

SUMMARY OF THE INVENTION

The present invention relates to a method for post-processing a decoded sound signal in view of enhancing a perceived quality of this decoded sound signal, comprising dividing the decoded sound signal into a plurality of frequency sub-band signals, and applying post-processing to at least one of the frequency sub-band signals, but not all the frequency sub-band signals.

The present invention is also concerned with a device for post-processing a decoded sound signal in view of enhancing a perceived quality of this decoded sound signal, comprising means for dividing the decoded sound signal into a plurality of frequency sub-band signals, and means for post-processing at least one of the frequency sub-band signals, but not all the frequency sub-band signals.

According to an illustrative embodiment, after post-processing of the above mentioned at least one frequency sub-band signal, the frequency sub-band signals are summed to produce an output post-processed decoded sound signal.

Accordingly, the post-processing method and device make it possible to localize the post-processing in the desired sub-band(s) and to leave other sub-bands virtually unaltered.

The present invention further relates to a sound signal decoder comprising an input for receiving an encoded sound signal, a parameter decoder supplied with the encoded sound signal for decoding sound signal encoding parameters, a sound signal decoder supplied with the decoded sound signal encoding parameters for producing a decoded sound signal, and a post processing device as described above for post-processing the decoded sound signal in view of enhancing a perceived quality of this decoded sound signal.

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.

DETAILED DESCRIPTION OF THE ILLUSTRATIVE EMBODIMENTS

FIG. 2is a schematic block diagram illustrating the general principle of an illustrative embodiment of the present invention.

InFIG. 1, the input signal (signal on which post-processing is applied) is the decoded (synthesized) speech signal112produced by the speech decoder105(FIG. 1) at the receiver of a communications system (output of the source decoder107ofFIG. 1). The aim is to produce a post-processed decoded speech signal at the output113of the post-processor108ofFIG. 1(which is also the output of processor203ofFIG. 2) with enhanced perceived quality. This is achieved by first applying at least one, and possibly more than one, adaptive filtering operation to the input signal.112(see adaptive filters201a,201b,. . . ,201N). These adaptive filters will be described in the following description. It should be pointed out here that some of the adaptive filters201ato201N can be trivial functions whenever required, for example with the output equal to the input. The output204a,204b,. . . ,204N of each adaptive filter201a,201b,. . . ,201N is then band-pass filtered through a sub-band filter202a,202b,. . . ,202N, respectively, and the post-processed decoded speech signal113is obtained by adding through a processor203the respective resulting outputs205a,205b,. . . ,205N of sub-band filters202a,202b,. . . ,202N.

In one illustrative embodiment, a two-band decomposition is used and adaptive filtering is applied only to the lower band. This results in a total post-processing that is mostly targeted at frequencies near the first harmonics of the synthesized speech signal.

FIG. 3is a schematic block diagram of a two-band pitch enhancer, which constitutes a special case of the illustrative embodiment ofFIG. 2. More specifically,FIG. 3shows the basic functions of a two-band post-processor (see post-processor108ofFIG. 1). According to this illustrative embodiment, only pitch enhancement is considered as post-processing although other types of post-processing could be contemplated. InFIG. 3, the decoded speech signal (assumed to be the output112of the source decoder107ofFIG. 1) is supplied through a pair of sub-branches308and309.

In the higher branch308, the decoded speech signal112is filtered by a high-pass filter301to produce the higher band signal310(sH). In this specific example, no adaptive filter is used in the higher branch. In the lower branch309, the decoded speech signal112is first processed through an adaptive filter307comprising an optional low-pass filter302, a pitch tracking module303, and a pitch enhancer304, and then filtered through a low-pass filter305to obtain the lower band, post processed signal311(sLEF). The post-processed decoded speech signal113is obtained by adding through an adder306the lower311and higher312band post-processed signals from the output of the low-pass filter305and high-pass filter301, respectively. It should be pointed out that the low-pass305and high-pass301filters could be of many different types, for example Infinite Impulse Response (UR) or Finite Impulse Response (FIR). In this illustrative embodiment, linear phase FIR filters are used.

Therefore, the adaptive filter307ofFIG. 3is composed of two, and possibly three processors, the optional low-pass filter302similar to low-pass filter305, the pitch tracking module303and the pitch enhancer304.

The low-pass filter302can be omitted, but it is included to allow viewing of the post-processing ofFIG. 3as a two-band decomposition followed by specific filtering in each sub-band. After optional low-pass filtering (filter302) of the decoded speech signal112in the lower-band, the resulting signal sLis processed through the pitch enhancer304. The object of the pitch enhancer304is to reduce the inter-harmonic noise in the decoded speech signal. In the present illustrative embodiment, the pitch enhancer304is achieved by a time-varying linear filter described by the following equation:

y⁡(n)=(1-α2)⁢x⁡[n]+α4⁢{x⁡[n-T]+x⁡[n+T]}(1)
where α is a coefficient that controls the inter-harmonic attenuation, T is the pitch period of the input signal x[n], and y[n] is the output signal of the pitch enhancer. A more general equation could also be used where the filter taps at n−T and n+T could be at different delays (for example n−T1 and n+T2). Parameters T and a vary with time and are given by the pitch tracking module303. With a value of α=1, the gain of the filter described by Equation (1) is exactly 0 at frequencies 1/(2T),3/(2T), 5/(2T), etc, i.e. at the mid-point between the harmonic frequencies 1/T, 3/T, 5/T, etc. When α approaches 0, the attenuation between the harmonics produced by the filter of Equation (1) reduces. With a value of α=0, the filter output is equal to its input.FIG. 8shows the frequency response (in dB) of the filter described by Equation (1) for the values α=0.8 and 1, when the pitch delay is (arbitrarily) set at a value T=10 samples. The value of α can be computed using several approaches. For example, the normalized pitch correlation, which is well-known by those of ordinary skill in the art, can be used to control the coefficient α: the higher the normalized pitch correlation (the closer to 1 it is), the higher the value of α. A periodic signal x[n] with a period of T=10 samples would have harmonics at the maxima of the frequency responses ofFIG. 8, i.e. at normalized frequencies 0.2, 0.4, etc. It is easy to understand fromFIG. 8that the pitch enhancer of Equation (1) would attenuate the signal energy only between its harmonics, and that the harmonic components would not be altered by the filter.FIG. 8also shows that varying parameter α enables control of the amount of inter-harmonic attenuation provided by the filter of Equation (1). Note that the frequency response of the filter of Equation (1), shown inFIG. 8, extends to all frequencies of the spectrum.

Since the pitch period of a speech signal varies in time, the pitch value T of the pitch enhancer304has to vary accordingly. The pitch tracking module303is responsible for providing the proper pitch value T to the pitch enhancer304, for every frame of the decoded speech signal that has to be processed. For that purpose, the pitch tracking module303receives as input not only the decoded speech samples but also the decoded parameters114from the parameter decoder106ofFIG. 1.

Since a typical speech encoder extracts, for every speech subframe, a pitch delay which we call T0and possibly a fractional value T0—fracused to interpolate the adaptive codebook contribution to fractional sample resolution, the pitch tracking module303can then use this decoded pitch delay to focus the pitch tracking at the decoder. One possibility is to use T0and T0—fracdirectly in the pitch enhancer304, exploiting the fact that the encoder has already performed pitch tracking. Another possibility, used in this illustrative embodiment, is to recalculate the pitch tracking at the decoder focussing on values around, and multiples or submultiples of, the decoded pitch value T0. The pitch tracking module303then provides a pitch delay T to the pitch enhancer304, which uses this value of T in Equation (1) for the present frame of decoded speech signal. The output is signal sLE.

Pitch enhanced signal sLEis then low-pass filtered through filter305to isolate the low frequencies of the pitch enhanced signal sLE, and to remove the high-frequency components that arise when the pitch enhancer filter of Equation (1) is varied in time, according to the pitch delay T, at the decoded speech frame boundaries. This produces the lower band post-processed signal sLEF, which can now be added to the higher band signal sHin the adder306. The result is the post-processed decoded speech signal113, with reduced inter-harmonic noise in the lower band. The frequency band where pitch enhancement will be applied depends on the cut-off frequency of the low-pass filter305(and optionally in low-pass filter302).

FIGS. 6aand6bshow an example signal spectrum illustrating the effect of the post-processing described inFIG. 3.FIG. 6ais the spectrum of the input signal112of the post-processor108ofFIG. 1(decoded speech signal112inFIG. 3). In this illustrative example, the input signal is composed of 20 harmonics, with fundamental frequency f0=373 Hz chosen arbitrarily, with <<noisy>> components added at frequencies f0/2, 3f0/2 and 5f0/2. These three noisy components can be seen between the low-frequency harmonics inFIG. 6a.The sampling frequency is assumed to be 16 kHz in this example. The two-band pitch enhancer shown inFIG. 3and described above is then applied to the signal ofFIG. 6a.With a sampling frequency of 16 kHz and a periodic signal of fundamental frequency equal to 373 Hz as inFIG. 6a,the pitch tracking module303should find a period of T=16000/373 ≈43 samples. This is the value that was used for the pitch enhancer filter of Equation (1), applied to the pitch enhancer304ofFIG. 3. A value of α=0.5 was also used. The low-pass305and high-pass301filters are symmetric, linear phase FIR filters with 31 taps. The cut-off frequency for this example is chosen as 2000 Hz. These specific values are given only as an illustrative example.

The post-processed decoded speech signal113at the output of the adder306has a spectrum shown inFIG. 6b.It can be seen that the three inter-harmonic sinusoids inFIG. 6ahave been completely removed, while the harmonics of the signal have been practically unaltered. Also it is noted that the effect of the pitch enhancer diminishes as the frequency approaches the low-pass filter cut-off frequency (2000 Hz in this example). Hence, only the lower band is affected by the post-processing. This is a key feature of this illustrative embodiment of the present invention. By varying the cut-off frequencies of the optional low-pass filter302, low-pass filter305and high-pass filter301, it is possible to control up to which frequency pitch enhancement is applied.

Application to the AMR-WB Speech Decoder

The present invention can be applied to any speech signal synthesized by a speech decoder, or even to any speech signal corrupted by inter-harmonic noise that needs to be reduced. This section will show a specific, exemplary implementation of the present invention to an AMR-WB decoded speech signal. The post-processing is applied to the low-band synthesized speech signal712ofFIG. 7, i.e. to the output of the speech decoder702, which produces a synthesized speech at a sampling frequency of 12.8 kHz.

FIG. 4shows the block diagram of a pitch post-processor when the input signal is the AMR-WB low-band synthesized speech signal at the sampling frequency of 12.8 kHz. More precisely, the post-processor presented inFIG. 4replaces the up-sampling unit703, which comprises processors704,705and706. The pitch post-processor ofFIG. 4could also be applied to the 16 kHz up-sampled synthesized speech signal, but applying it prior to up-sampling reduces the number of filtering operations at the decoder, and thus reduces complexity.

The input signal (AMR-WB low-band synthesized speech (12.8 kHz)) ofFIG. 4is designated as signal s. In this specific example, signal s is the AMR-WB low-band synthesized speech signal at the sampling frequency of 12.8 kHz (output of processor702). The pitch post-processor ofFIG. 4comprises a pitch tracking module401to determine, for every 5 millisecond subframe, the pitch delay T using the received, decoded parameters114(FIG. 1) and the synthesized speech signal s. The decoded parameters used by the pitch tracking module are T0, the integer pitch value for the subframe, and T0—frac, the fractional pitch value for subsample resolution. The pitch delay T calculated in the pitch tracking module401will be used in the next steps for pitch enhancement. It would be possible to use directly the received, decoded pitch parameters T0and T0—fracto form the delay T used by the pitch enhancer in the pitch filter402. However, the pitch tracking module401is capable of correcting pitch multiples or submultiples, which could have a harmful effect on the pitch enhancement.

An illustrative embodiment of pitch tracking algorithm for the module401is the following (the specific thresholds and pitch tracked values are given only by way of example):First, the decoded pitch information (pitch delay T0) is compared to a stored value of the decoded pitch delay T_prev of the previous frame. T_prev may have been modified by some of the following steps according to the pitch tracking algorithm. For example, if T0<1.16*T_prev then go to case 1 below, else if T0>1.16*T_prev, then set T_temp=T0and go to case 2 below.Case 1: First, calculate the cross-correlation C2 (cross-product) between the last synthesized subframe and the synthesis signal starting at T0/2 samples before the beginning of the last subframe (look at correlation at half the decoded pitch value).Then, calculate the cross-correlation C3 (cross-product) between the last synthesized subframe and the synthesis signal starting at T0/3 samples before the beginning of the last subframe (look at correlation at one-third the decoded pitch value).Then, select the maximum value between C2 and C3 and calculate the normalized correlation Cn (normalized version of C2 or C3) at the corresponding sub-multiple of T0(at T0/2 if C2>C3 and at T0/3 if C3>C2). Call T_new the pitch sub-multiple corresponding to the highest normalized correlation.If Cn>0.95 (strong normalized correlation) the new pitch period is T_new (instead of T0). Output the value T =T_new from the pitch tracking module401. Save T_prev=T for next subframe pitch tracking and exit the pitch tracking module401.If 0.7<Cn<0.95, then save T_temp=T0/2 or T0/3 (according to C2 or C3 above) for comparisons in case 2 below. Otherwise, if Cn<0.7 save T_temp=T0.Case 2: Calculate all possible values of the ratio Tn=[T_temp/n]where [x] means the integer part of x and n=1,2,3, etc. is an integer.Calculate all cross correlations Cn at the pitch delay submultiples Tn. Retain Cn_max as the maximum cross correlation among all Cn. If n>1 and Cn>0.8, output Tn as the pitch period output T of the pitch tracking unit401. Otherwise, output T1=T temp. Here, the value of T_temp will depend on the calculations in Case 1 above.

It should be noted that the above example of pitch tracking module401is given for the purpose of illustration only. Any other pitch tracking method or device could be implemented in module401(or303and502) to ensure a better pitch tracking at the decoder.

Therefore, the output of the pitch tracking module is the period T to be used in the pitch filter402which, in this preferred embodiment, is described by the filter of Equation (1). Again, a value of α=0 implies no filtering (output of the pitch filter402is equal to its input), and a value of α=1 corresponds to the highest amount of pitch enhancement.

Once the enhanced signal SE(FIG. 4) is determined, it is combined with the input signal s such that, as inFIG. 3, only the lower band is subjected to pitch enhancement. InFIG. 4, a modified approach is used compared toFIG. 3. Since the pitch post-processor ofFIG. 4replaces the up-sampling unit703inFIG. 7, the sub-band filters301and305ofFIG. 3are combined with the interpolation filter705ofFIG. 7to minimize the number of filtering operations, and the filtering delay. More specifically, filters404and407ofFIG. 4act both as band-pass filters (to separate the frequency bands) and as interpolation filters (for up-sampling from 12.8 to 16 kHz). These filters404and407could be further designed such that the band-pass filter407has relaxed constraints in its low-frequency stop band (i.e. it does not have to completely attenuate the signal at low frequencies). This could be achieved by using design constraints similar to those shown inFIG. 9.FIG. 9ais an example of frequency response for the low-pass filter404. It should be noted that the DC (Direct Current) gain of this filter is 5 (instead of 1) since this filter also acts as interpolation filter, with a 5/4 interpolation ratio which implies that the filter gain must be 5 at 0 Hz. Then,FIG. 9bshows the frequency response of the band-pass filter407making this filter407complementary, in the low band, to the low-pass filter404. In this example, the filter407is a band-pass filter, not a high-pass filter such as filter301, since it must act both as high-pass filter (such as filter301) and low-pass filter (such as interpolation filter705). Referring again toFIG. 9, we see that the low-pass and band-pass filters404and407are complementary when considered in parallel, as inFIG. 4. Their combined frequency response (when used in parallel) is shown inFIG. 9c.

For completeness, the tables of filter coefficients used in this illustrative embodiment of the filters404and407are given below. Of course, these tables of filter coefficients are given by way of example only. It should be understood that these filters can be replaced without modifying the scope, spirit and nature of the present invention.

TABLE 1Low-pass coefficients of filter 404hlp[0]0.04375000000000hlp[1]0.04371500000000hlp[2]0.04361200000000hlp[3]0.04344000000000hlp[4]0.04320000000000hlp[5]0.04289300000000hlp[6]0.04252100000000hlp[7]0.04208300000000hlp[8]0.04158200000000hlp[9]0.04102000000000hlp[10]0.04039900000000hlp[11]0.03972100000000hlp[12]0.03898800000000hlp[13]0.03820200000000hlp[14]0.03736700000000hlp[15]0.03648600000000hlp[16]0.03556100000000hlp[17]0.03459600000000hlp[18]0.03359400000000hlp[19]0.03255800000000hlp[20]0.03149200000000hlp[21]0.03039900000000hlp[22]0.02928400000000hlp[23]0.02814900000000hlp[24]0.02699900000000hlp[25]0.02583700000000hlp[26]0.02466700000000hlp[27]0.02349300000000hlp[28]0.02231800000000hlp[29]0.02114600000000hlp[30]0.01998000000000hlp[31]0.01882400000000hlp[32]0.01768200000000hlp[33]0.01655700000000hlp[34]0.01545100000000hlp[35]0.01436900000000hlp[36]0.01331200000000hlp[37]0.01228400000000hlp[38]0.01128600000000hlp[39]0.01032300000000hlp[40]0.00939500000000hlp[41]0.00850500000000hlp[42]0.00765500000000hlp[43]0.00684600000000hlp[44]0.00608100000000hlp[45]0.00535900000000hlp[46]0.00468200000000hlp[47]0.00405100000000hlp[48]0.00346700000000hlp[49]0.00292900000000hlp[50]0.00243900000000hlp[51]0.00199500000000hlp[52]0.00159900000000hlp[53]0.00124800000000hlp[54]0.00094400000000hlp[55]0.00068400000000hlp[56]0.00046800000000hlp[57]0.00029500000000hlp[58]0.00016300000000hlp[59]0.00007100000000hlp[60]0.00001800000000

TABLE 2Band-pass coefficients of filter 407hbp[0]0.95625000000000hbp[1]0.89115400000000hbp[2]0.71120900000000hbp[3]0.45810600000000hbp[4]0.18819900000000hbp[5]−0.04289300000000hbp[6]−0.19474300000000hbp[7]−0.25136900000000hbp[8]−0.22287200000000hbp[9]−0.13948000000000hbp[10]−0.04039900000000hbp[11]0.03868100000000hbp[12]0.07548400000000hbp[13]0.06566500000000hbp[14]0.02113800000000hbp[15]−0.03648600000000hbp[16]−0.08465300000000hbp[17]−0.10763400000000hbp[18]−0.10087600000000hbp[19]−0.07091900000000hbp[20]−0.03149200000000hbp[21]0.00234200000000hbp[22]0.01970000000000hbp[23]0.01715300000000hbp[24]−0.00110700000000hbp[25]−0.02583700000000hbp[26]−0.04678900000000hbp[27]−0.05654900000000hbp[28]−0.05281800000000hbp[29]−0.03851900000000hbp[30]−0.01998000000000hbp[31]−0.00412400000000hbp[32]0.00414300000000hbp[33]0.00343300000000hbp[34]−0.00416100000000hbp[35]−0.01436900000000hbp[36]−0.02267300000000hbp[37]−0.02601800000000hbp[38]−0.02370000000000hbp[39]−0.01723200000000hbp[40]−0.00939500000000hbp[41]−0.00297000000000hbp[42]0.00030500000000hbp[43]0.00019000000000hbp[44]−0.00226000000000hbp[45]−0.00535900000000hbp[46]−0.00756800000000hbp[47]−0.00805800000000hbp[48]−0.00687000000000hbp[49]−0.00469500000000hbp[50]−0.00243900000000hbp[51]−0.00080600000000hbp[52]−0.00006300000000hbp[53]−0.00005300000000hbp[54]−0.00038700000000hbp[55]−0.00068400000000hbp[56]−0.00074400000000hbp[57]−0.00057600000000hbp[58]−0.00031900000000hbp[59]−0.00011300000000hbp[60]−0.00001800000000

The output of the pitch filter402ofFIG. 4is called SE.To be recombined with the signal of the upper branch, it is first up-sampled by processor403, low-pass filter404and processor405, and added through an adder409to the up-sampled upper branch signal410. The up-sampling operation in the upper branch is performed by processor406, band-pass filter407and processor408.

Alternate Implementation of the Proposed Pitch Enhancer

FIG. 5shows an alternative implementation of a two-band pitch enhancer according to an illustrative embodiment of the present invention. It should be noted that the upper branch ofFIG. 5does not process the input signal at all. This means that, in this particular case, the filters in the upper branch ofFIG. 2(adaptive filters201aand201b) have trivial input-output characteristics (output is equal to input). In the lower branch, the input signal (signal to be enhanced) is processed first through an optional low-pass filter501, then through a linear filter called inter-harmonic filter503, defined by the following equation:

y⁡[n]=12⁢x⁡[n]-14⁢{x⁡[n-T]+x⁡[n+T]}(2)
It should be noted that the negative sign in front of the second term on the right hand side, compared to Equation (1). It should also be noted that the enhancement factor α is not included in Equation (2), but rather it is introduced by means of an adaptive gain by the processor504ofFIG. 5. The inter-harmonic filter503, described by Equation (2), has a frequency response such that it completely removes the harmonics of a periodic signal having a period of T samples, and such that a sinusoid at a frequency exactly between the harmonics passes through the filter unchanged in amplitude but with a phase reversal of exactly 180 degrees (same as sign inversion). For example,FIG. 10shows the frequency response of the filter described by Equation (2) when the period is (arbitrarily) chosen at T=10 samples. A periodic signal with period T=10 samples would present harmonics at normalized frequencies 0.2, 0.4, 0.6, etc., andFIG. 10shows that the filter of Equation (2), with T=10 samples, would completely remove these harmonics. On the other hand, the frequencies at the exact mid-point between the harmonics would appear at the output of the filter with the same amplitude but with a 180° phase shift. This is the reason why the filter described by Equation (2) and used as filter503is called inter-harmonic filter.

The pitch value T for use in the inter-harmonic filter503is obtained adaptively by the pitch tracking module502. Pitch tracking module502operates on the decoded speech signal and the decoded parameters, similarly to the previously disclosed methods as shown inFIGS. 3 and 4.

Then, the output507of the inter-harmonic filter503is a signal formed essentially of the inter-harmonic portion of the input decoded signal112, with 180° phase shift at mid-point between the signal harmonics. Then, the output507of the inter-harmonic filter503is multiplied by a gain α (processor504) and subsequently low-pass filtered (filter505) to obtain the low frequency band modification that is applied to the input decoded speech signal112ofFIG. 5, to obtain the post-processed decoded signal (enhanced signal)509. The coefficient α in processor504controls the amount of pitch or inter-harmonic enhancement. The closer to 1 is α, the higher the enhancement is. When α is equal to 0, no enhancement is obtained, i.e. the output of adder506is exactly equal to the input signal (decoded speech inFIG. 5). The value of α can be computed using several approaches. For example, the normalized pitch correlation, which is well known to those of ordinary skill in the art, can be used to control coefficient α: the higher the normalized pitch correlation (the closer to 1 it is), the higher the value of α.

The final post-processed decoded speech signal509is obtained by adding through an adder506the output of low-pass filter505to the input signal (decoded speech signal112ofFIG. 5). Depending on the cut-off frequency of the low-pass filter505, the impact of this post-processing will be limited to the low frequencies of the input signal112, up to a given frequency. The higher frequencies will be effectively unaffected by the post-processing.

One-Band Alternative Using an Adaptive High-Pass Filter

One last alternative for implementing sub-band post-processing for enhancing the synthesis signal at low frequencies is to use an adaptive high-pass filter, whose cut-off frequency is varied according to the input signal pitch value. Specifically, and without referring to any drawing, the low frequency enhancement using this illustrative embodiment would be performed, at each input signal frame, according to the following steps:1. Determine the input signal pitch value (signal period) using the input signal and possibly the decoded parameters (output of speech decoder105) if post-processing a decoded speech signal; this is a similar operation as the pitch tracking operation of modules303,401and502.2. Calculate the coefficients of a high-pass filter such that the cut-off frequency is below, but close to, the fundamental frequency of the input signal; alternatively, interpolate between pre-calculated, stored high-pass filters of known cut-off frequencies (the interpolation can be done in the filtertaps domain, or in the pole-zero domain, or in some other transformed domain such as the LSF (Line Spectral Frequencies) of ISF (Immitance Spectral Frequencies) domain).3. Filter the input signal frame with the calculated high-pass filter, to obtain the post-processed signal for that frame.

It should be pointed out that the present illustrative embodiment of the present invention is equivalent to using only one processing branch inFIG. 2, and to define the adaptive filter of that branch as a pitch-controlled high-pass filter. The post-processing achieved with this approach will only affect the frequency range below the first harmonic and not the inter-harmonic energy above the first harmonic.

Although the present invention has been described in the foregoing description with reference to 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. For example, although the illustrative embodiments have been described in relation to a decoded speech signal, those of ordinary skill in the art will appreciate that the concepts of the present invention can be applied to other types of decoded signals, in particular but not exclusively to other types of decoded sound signals.