JPH0449959B2 - - Google Patents

Description

[Detailed description of the invention] 〔Technical field〕 The present invention relates to a residual excitation type vocoder.

[Prior art]

The input audio signal is converted to LPC (Linear Prediction).
Coefficient, linear prediction coefficient) obtained by analyzing
Along with the LPC coefficients, information regarding the low frequency components of a preset region of the residual waveform as waveform information is transmitted from the analysis side to the synthesis side, and the synthesis side uses the input low frequency components to restore the high frequency components. Synthesize the residual waveform along with the frequency component, and
RELP (Residual Exciting Linear) is a residual excitation type vocoder that operates a synthesis filter using LPC coefficients and reproduces high-quality input audio signals.
It is also known by the abbreviation ``Prediction Vocoder''.

Compared to a conventional general vocoder, which sends a modeled residual waveform from the analysis side to the synthesis side together with the LPC coefficients, RELP sends a part of the low frequency components of the residual waveform through modeling. Basically, the playback quality is excellent in that part of the waveform is transmitted from the analysis side to the synthesis side without any noise, and it is comparable to conventional vocoders with a speed of about 4.8Kb/s (kilobits per second). CODEC (COder
It has recently become widely used as a DECorder (codec).

However, in conventional RELP of this type, the residual waveform information supplied from the analysis side to the synthesis side is limited to low frequency components in a preset frequency domain due to the available data bit rate. The basic method for high frequency components is to apply this low frequency component to a nonlinear circuit on the synthesis side and simply add the resulting harmonic components, ignoring phase information. It is well known that this method has the disadvantage that reproduction distortion, so-called RELP sound, occurs in which the waveform of the reproduced audio differs from the input audio signal due to the inability to obtain reproducibility.

In order to improve such RELP sound, residual excitation type vocoders equipped with high frequency reproduction means on the analysis side have recently been used. This is because the analysis side estimates the tap coefficient of the transversal high-frequency reproduction filter used to reproduce the residual waveform on the synthesis side, and then supplies it to the synthesis side. For a transversal filter with the same configuration as the reproduction filter, set the impulse response characteristics including the phase conditions for the low frequency and high frequency to be synthesized, that is, set the tap coefficient, and then supply this to the synthesis side. It is something.

However, in conventional residual excitation vocoders of this type, the high frequency components generated on the synthesis side are generated by applying the low frequency components supplied from the analysis side to a non-linear circuit. Since harmonic components are used, it is often difficult to obtain the desired high frequency reproduction.

[Purpose of the invention]

The purpose of the present invention is to eliminate the above-mentioned drawbacks, and to use the down-sampled value of the low frequency component of the residual waveform as a reference sampling conversion sample sequence through interpolation of zero-level samples, and to utilize a nonlinear circuit. It is an object of the present invention to provide a residual excitation type vocoder that can stably and reliably achieve a significant improvement in RELP sound by analyzing and synthesizing an input audio signal using a means for estimating the tap coefficient of a high-frequency reproduction filter. .

[Structure of the invention]

The residual excitation type vocoder of the present invention includes a first means for determining a reference sample sequence of a residual waveform of an input audio signal corresponding to a preset reference sampling frequency; a second means for determining a down-sampling sequence of low frequency components of the residual wave type corresponding to the repeated down-sampling frequency; and a second means for determining a zero-level interpolation point in the down-sampling sequence at the timing of the reference sampling frequency. a transversal type that should be provided on the synthesis side based on the reference sampling conversion sample sequence of the low frequency component of the residual waveform that is formed while setting and the reference sample sequence of the residual waveform and without going through a nonlinear circuit; and third means for estimating coefficients of the high frequency reproduction filter.

〔Example〕

Next, the present invention will be explained in detail with reference to the drawings.
FIG. 1A is a block diagram showing the structure of the analysis side of a first embodiment of the residual excitation type vocoder according to the present invention, and FIG. 1B is a block diagram showing the structure of the synthesis side.

The analysis side shown in FIG. 1A is an A/D converter 1,
LPC analyzer 2, quantizer/decoder 3, LPC inverse filter 4, LPF (Low Pass Filter) 5, down sampler 6, quantizer/decoder 7, filter coefficient estimator (1) 8 , quantizer 9, power calculator 10, quantizer 11 and multiplexer 12
The synthesis side shown in FIG.
4, a multiplier 15, an LPC synthesis filter 16, and a D/A converter 17.

The audio signal input from the input terminal 101 is
After performing low-pass filtering to cut out components of a preset high frequency, in this example, 3.4KHz or higher, which is supplied to the D converter 1, sampling is performed at a preset reference sampling frequency of 8KHz, and a predetermined number of bits is sampled. It is converted into a digital quantity and sent as quantized input audio data to an LPC analyzer 2 and an LPC inverse filter 4. The 8KHz sampling frequency matches the highest frequency of the input audio signal.
This was determined by considering the Nyquist rate under the conditions set at 3.4KHz. The LPC analyzer 2 sequentially performs window processing in which the input quantized input audio data is multiplied by a preset window function for each basic frame period, and then measures the autocorrelation for each basic frame. LPC analysis is performed to extract LPC coefficients of a predetermined order using the well-known Auto Correlation method.

In this way, the output from LPC analyzer 2 is
The LPC coefficients are quantized by the quantizer/decoder 3 and supplied to the multiplexer 12, and this quantized LPC coefficient data is once decoded and supplied to the LPC inverse filter 4.

The LPC inverse filter 4 is a filter whose impulse response characteristic is opposite to that of the LPC synthesis filter 16 on the synthesis side shown in FIG. The quantizer/decoder 3 receives the decoded LPC coefficient data, and then the A/D
Only data related to the residual waveform of the quantized input audio data received from the output of the converter 1 is extracted and supplied to the LPF 5, filter coefficient estimator (1) 8, and power calculator 10.

LPF5 is a low-pass filter with high-frequency cutoff characteristics that is set taking into consideration the data bit rate of the vocoder and other conditions.
1 KHz of the residual waveform received from is set as a high cutoff frequency, and low frequency components of 1 KHz or less of the residual waveform are supplied to the down sampler 6.

The down sampler 6 samples the input low frequency components of 1KHz or less at a preset rate of the reference sampling frequency, in this example, the downsampling frequency is 2KHz, which is down to 1/4, and quantizes/decodes this. It is sent to the converter 7.

The quantizer/decoder 7 quantizes the input residual waveform low frequency component and sends it to the multiplexer 12, and also sends the decoded data to the filter coefficient estimator (1) 8. supply The reason why the downsampled data supplied to the filter coefficient estimator (1) 8 is the data that has been decoded by the quantizer/decoder 7 is to avoid the influence of quantization errors in the high frequency reproduction filter 14 on the synthesis side. This is to make the influence of the quantization error on the filter coefficient estimator (1) 8 almost the same.

Incidentally, in this embodiment, the quantization and decoding processing of the residual waveform low frequency component in the quantization/decoder 7 is performed by normalizing the power data supplied from the power calculator 10, which will be described later. The quantization process efficiently quantizes the low frequency components of the residual waveform.

Filter coefficient estimator (1) 8 is LPC inverse filter 4
A sample sequence of the residual waveform with a reference sampling frequency of 8KHz is input from the quantizer/decoder 7, and a downsample sequence of the residual waveform low frequency component with a downsampling frequency of 2LHz is input from the quantizer/decoder 7. In this way, the tap coefficient of the high frequency reproduction filter to be provided on the synthesis side is estimated.

FIG. 2a is a residual waveform reference sample diagram showing an example of the residual waveform obtained by sampling the residual waveform of the input audio signal at the reference sampling frequency in the first embodiment of the present invention shown in FIGS. 1A and B; Figure 2b
is a down-sampling sample diagram showing an example of down-sampling in which the low-frequency components of the residual waveform are sampled at the down-sampling frequency, and Figure 2c is a down-sampling sample of the low-frequency components of the residual waveform shown in Figure 2b. On the other hand, it is a residual waveform low frequency component reference sampling conversion sample sequence that has undergone reference sampling conversion corresponding to the reference sampling frequency.
The embodiments shown in FIGS. 1A and 1B will be described below with reference to FIGS. 2A, 2B, and 2C.

The residual waveform sample sequence shown in FIG. 2a is the 8KHz sampling output from the LPC inverse filter 4,
The residual waveform low frequency component down-sampled sample shown in Figure 2b is the 2KHz sampling output from the quantizer/decoder 7, which is further converted into a sample with a reference sampling frequency of 8KHz as the second sample.
a ₁ , a ₂ , a ₃ , a ₄ , a ₅ , a ₆ and b1 to b5 shown in figure c
This is the sample group. a1 to a6 of these sample groups
corresponds to the sample shown in Figure 2 b, and b1 to b5 are interpolated between samples a1 to a6, and the overall result is
These are zero-level sampling points arranged at equal intervals in order to convert the standard sampling frequency to 8KHz, and in this way, the low-frequency components of the residual waveform can be down-sampled with an image similar to that of a sample using the standard sampling frequency of 8KHz. Standard sampling conversion of samples can be achieved.

The filter coefficient estimator (1)8 is thus 2K
A downsampling sampling sequence with a downsampling frequency of Hz is converted into a sequence of sampling images with a reference sampling frequency of 8KHz, and this is supplied as an input to a built-in filter similar to the transversal filter as a high-frequency reproduction filter that should be provided on the synthesis side. do. The tap coefficients of this transversal filter are determined using the learning identification method or the A-
It is controlled and determined for each analysis frame based on a method such as the b-S (Analysis-by-Synthesis) method.

That is, in the filter coefficient estimator (1) 8,
The sampling sequence of the low frequency component of the residual waveform of 2KHz sampling shown in Figure 2b is input.This is converted into the 8KHz sampling image sampling sequence shown in Figure 2b, which is applied to the built-in transversal filter, and is used as audio material, etc. The error output obtained by comparing the output of the transversal filter to which the initial value of the tap coefficient set in advance is given and the 8KHz sampling sequence of the residual waveform as shown in Figure 2a, which is input separately, is obtained. obtain. In the case of this embodiment, the power of this error output is calculated, and the filter coefficients are estimated so that the error power can be minimized while controlling the transversal filter coefficients for each analysis frame in accordance with the magnitude of the error power. is carried out using methods such as the learning identification method and the A-b-S method.

Figure 3 shows the filter coefficient estimator (1) shown in Figure 1A.
FIG. 8 is a block diagram showing in detail the configuration of FIG.

The filter coefficient estimator (1) 8 includes a transversal filter 81, a subtracter 82, and a power calculator 83, which have basically the same configuration as the high-frequency reproduction filter to be provided on the synthesis side. Out of the output of the quantizer/decoder 7, the down-sampled residual waveform low frequency component as shown in Figure 2b is input, and while receiving the 8KHz sampling frequency, the filter coefficients are estimated as follows. Let's do it.

That is, the transversal filter 81 is
When the residual waveform low frequency component down sample as shown in Figure 2b is input, the second
Convert the residual waveform low frequency component into a reference sampling conversion sampling sequence as shown in Figure c, send the transversal filter output based on the initial setting tap coefficient to the subtracter 82 via the output line 811, and output it to the LPC inverse filter 4. The difference from the residual waveform is taken, and this error output is supplied to a power calculator 83 via an output line 821, and its power is calculated.

The filter coefficient estimator (1) 8 controls the tap coefficient of the transversal filter 81 based on the error power calculated in this way and outputs the tap coefficient when the error power is minimized as an estimated filter coefficient. The quantized data is converted into quantized data by the quantizer 9 and then supplied to the multiplexer 12.

In this way, the 8KHz sampling image sampling sequence of down-sampling of the low frequency component of the residual waveform is created, and the coefficients of the filter can be estimated without using a nonlinear circuit based on the learning identification method or the A-b-S method. It becomes possible.

The power calculator 10 inputs the residual waveform, calculates the short-term average audio power for each analysis frame, and supplies this to the quantizer 11 and the quantizer/decoder 7.

This short-time average audio power is used to set the level of the audio to be reproduced when the input audio signal is reproduced by the LPC synthesis filter 16 on the synthesis side. This data transmission is not necessary when the value is approximately constant over time.

The multiplexer 12 multiplexes each of the quantized data thus supplied using a preset method and sends the multiplexed data to the combining side via a transmission path.

On the synthesis side in Figure 1B, each quantized data that has been multiplexed and supplied from the analysis side is demultiplexed and demultiplexed by the demultiplexer/decoder 13, and further decoded, as shown in Figure 2B. The downsampled sampled data of the low frequency component of the residual waveform is supplied to the high frequency reproduction filter 14 via an output line 1301, and the estimated filter coefficient data is also supplied to the high frequency reproduction filter 14 via an output line 1302. In addition, the short-time average audio power data is input to the input line 1303.
The LPC coefficient data is supplied to the multiplier 16 via an input line 1304, and the LPC coefficient data is supplied to the LPC synthesis filter 16 via an input line 1304.

The high frequency reproduction filter 14 has a built-in transversal filter that uses the estimated filter coefficients as tap coefficients, and converts the input downsampling sample into an 8KHz sampling image sample as shown in Figure 2c. As a filter input, residual waveform data of 8KHz sampling as shown in FIG. 2a is reproduced. This residual waveform data is supplied to the multiplier 15 and input line 1303
The level of the reproduced residual waveform data is corrected through multiplication with the short-term average voice power inputted via the LPC synthesis filter 16.

The LPC synthesis filter 16 is connected to the input line 130.
The input audio signal is digitally reproduced using the LPC coefficients inputted through 4 as filter coefficients, and the reproduced residual waveform data as driving sound source information, and this is used as a D/A.
The signal is supplied to the converter 17, which converts it into an analog quantity, and further outputs only a predetermined low frequency component to the output terminal 1701 using the built-in LPF.

FIG. 4 is a block diagram showing the configuration of the analysis side in the second embodiment of the present invention. The configuration of the analysis side shown in FIG. 4 differs from FIGS. 1A and 3 only in the contents of the filter coefficient estimator (2) 18, and the other components with the same symbols are exactly the same, so we will not be concerned about these. Detailed explanation will be omitted.

Further, since the composition side in the second embodiment has the same configuration as the composition side in the first embodiment shown in FIG. 1B, detailed explanation thereof will also be omitted.

The filter coefficient estimator (2) 18 in the second embodiment shown in FIG. 4 is similar to the filter coefficient estimator (1) 8 shown in FIG. This method attempts to estimate the filter coefficients of a transversal filter that has an impulse response that minimizes the difference from the residual waveform of the input audio signal. The difference is that the filter coefficients are estimated by a so-called forward filter coefficient setting method, which attempts to determine a transversal filter.

Figure 5 shows the filter coefficient estimator (2) 1 shown in Figure 4.
FIG. 8 is a block diagram showing in detail the configuration of FIG.

The filter coefficient estimator (2) 18 in FIG. 5 includes a cross-correlation circuit 181, an autocorrelation circuit 182, and a maximum value search circuit 183.

When the residual waveform low frequency component downsampling as shown in FIG. This down-sampling by frequency is converted to an apparent 8KHz sampling sequence while interpolating zero-level samples as shown in Figure 2c, and then the output from the LPC inverse filter 4 as shown in Figure 2a is Calculate the cross-correlation coefficient with the residual waveform of 8KHz sampling and use this as maximum value search circuit 1
83.

Further, the autocorrelation circuit 182 generates a sampling sequence as shown in FIG. Calculate the autocorrelation coefficient sequence. An 8KHz sampling frequency is used to calculate these cross-correlation coefficient sequences and autocorrelation coefficient sequences.

Now, the cross-correlation coefficient sequence supplied to the maximum value search circuit 183 searches for its maximum value within a preset limited delay time. In this case, the tap position corresponding to the delay time is, for example, tap γ, and the mutual correlation at this tap γ is taken as a number φγ, and the autocorrelation coefficient at zero delay of the autocorrelation coefficient sequence obtained in the autocorrelation circuit 182 is If ψο, then the value of φγ divided by ψο is Cγ = φγ/ψο
is the tap coefficient at tap γ of the transversal filter.

Next, an operation is performed to subtract the autocorrelation coefficient sequence multiplied by Cγ from the cross-correlation coefficient sequence described above so that the tap position γ coincides with the position of the autocorrelation value ψο,
While repeating such calculations, Cγ is successively obtained, all filter coefficients of the transversal filter are determined, and these are output as estimated filter coefficients. In this way, the correction message shown in FIG.
This is the data required to perform subsequent calculations and outputs.

In this way, forward coefficient settings can be performed to obtain the maximum impulse response based on continuous changes in the coefficients of the transversal filter, and the analysis and synthesis processing of the residual excitation type vocoder, which significantly suppresses RELP sound, can be performed stably and reliably. be able to.

FIGS. 6A and 6B are block diagrams showing the configurations of analysis side A and synthesis side B, respectively, in a third embodiment of the present invention.

In FIG. 6A, filter coefficient estimator (3) 19
However, in FIG. 6B, the coefficient regenerator (1) 20 and the high frequency regeneration filter 21 are the same as those in FIG. 1A, respectively.
The only difference from or addition to that in B is that everything else is the same, so a detailed explanation regarding these will be omitted.

The filter coefficient estimator (3) 19 shown in FIG. 6A is
In addition to the filter coefficient estimator (1) 8 shown in FIG. The signal is converted to zero phase and then output.

FIG. 7A is a high-frequency reproduction filter prediction coefficient characteristic diagram formed on the analysis side of the third embodiment shown in FIG. 6A, and FIG. 7B is a zero-phase prediction coefficient characteristic diagram of FIG. 7A. . The embodiment shown in FIGS. 6A and 6B will be described below with reference to FIGS. 7A and 7B.

The filter coefficient estimator (3) 19 first generates estimated filter coefficients as shown in FIG. 7A in substantially the same manner as the filter coefficient estimator (1) 8 shown in FIG. 1A. Next, the estimated filter coefficients are converted from a group of coefficients as a complex spectrum by a transversal filter to a group of coefficients shown in FIG. 7B as a power spectrum by a phase linear filter having substantially the same impulse response characteristics. Although the coefficient group formed by this conversion loses phase information, the transversal filter coefficients in FIG. It will be done. In this way, if the energy center position remains unchanged, even if phase information is lost, there is practically no problem as a coefficient of a high frequency reproduction filter.

Now, the phase linear filter coefficient group shown in Figure 7B has left-right symmetry, so even if only approximately 1/2 of the total is sent to the synthesis side as estimated coefficients from either the left or right of the energy center position, the whole can be easily reproduced. It is possible to do so. For this reason, from the filter coefficient estimator (3) 19, only approximately half of the coefficient group of the entire coefficient group is sent to the quantizer 9, where it is subjected to predetermined quantization and supplied to the multiplexer 12. The quantized data is multiplexed with other quantized data and transmitted to the combining side in FIG. 6B.

On the synthesis side shown in FIG.
The decoded estimated coefficient data is supplied to the coefficient regenerator (1) 20. The estimated coefficient data thus supplied is approximately 1/2 including the center tap position of the coefficient group shown in FIG. 6B. The coefficient regenerator (1) 20 regenerates the symmetrization coefficient group shown in FIG. The reproduction input audio signal is outputted to the output terminal 1701 in substantially the same manner as described with reference to FIG. becomes.

FIGS. 8A and 8B are block diagrams showing the configurations of analysis side A and synthesis side B in a fourth embodiment of the present invention.

Figure 8A is the same as Figure 1A and B except for the filter coefficient estimator (4) 22, and Figure 8B is the same as Figure 1A and B except for the coefficient regenerator (2) 23 and the high frequency reproduction filter 24, Detailed explanation will be omitted.

The fourth embodiment shown in FIGS. 8A and 8B is as shown in FIG. 1A.
The tap coefficient group of the high-frequency reproducing filter obtained in the same manner as the filter coefficient estimator (1) 8 shown in Fig. 8 is an estimated coefficient sequence of a triangular envelope array having almost the same power spectrum and the energy center position unchanged. This method aims to reduce the transmission data rate by converting the equation so that be.

FIG. 9A is a high frequency reproduction filter prediction coefficient characteristic diagram formed on the analysis side of the fourth embodiment shown in FIG. 8A, and shows basically the same contents as FIG. FIG. 9B is a triangular envelope distribution transformation characteristic diagram obtained while keeping the estimated coefficient group of FIG. 9A in substantially the same power spectrum.

In FIG. 9B, the coefficient group in FIG. 9A is transformed so that its approximate envelope becomes a desired triangular shape. This transformation is realized by Fourier transforming the coefficient group shown in FIG. 9A, and adjusting the phase of the Fourier transformed value in FIG. 9A so as to match the phase shift of the desired triangular coefficient.

The motivation for making the envelope of the transformation coefficient group triangular is based on the fact that the vocal fold vibration waveform generally takes a triangular waveform, and this kind of transformation causes loss of phase information. However, due to this envelope shape and the characteristic of the coefficient array in which the energy center position remains unchanged, there is almost no effect on the quality of the reproduced audio by synthesis.

In this way, after being converted into a coefficient series that shows a triangular envelope distribution and has almost the same power spectrum, only the difference component from the desired triangular shape is output from the filter coefficient estimator (4) 22, and quantized. It is transmitted to the synthesis side via the device 9.

On the synthesis side shown in FIG.
23, thereby adding the desired triangular shape subtracted on the analysis side to reproduce the coefficient group, and supplying these to the high frequency reproduction filter 24.

The high-frequency reproduction filter 24 uses these coefficient groups as its tap coefficients, and inputs the residual waveform low-frequency component that is converted into an 8KHz sampling image via the output line 1302, and generates the total residual waveform including the high-frequency component. Play.

The synthesis side then performs the same playback operation as shown in Figure 1B to play back the input audio signal, thus basically removing the RELP sound and also changing the transmission data bits of the tap coefficient of the high-frequency playback filter. Analysis and synthesis processing can be performed at significantly reduced rates.

In the above explanation, the power calculators 10 are provided in each of FIGS. 1A, 4, 6A, and 8A, and the first
Although multipliers 15 were provided in each of FIG. B, FIG. 6B, and FIG. If absolute level quantization and decoding processing are performed without performing quantization processing, the power calculator 10 and multiplier 15 are unnecessary.

〔Effect of the invention〕

As explained above, according to the present invention, the down-sampled value of the low-frequency component of the residual waveform is used as the reference sampling-converted sample sequence through interpolation of the zero-level sample, and without using a nonlinear circuit. By providing a means for estimating the tap coefficient of the high-frequency reproduction filter and analyzing and synthesizing the input audio signal, it is possible to realize a residual excitation type vocoder that can significantly improve the RELP sound in an extremely stable and reliable manner. be.

[Brief explanation of drawings]

FIG. 1A is a block diagram showing the configuration on the analysis side in the first embodiment of the present invention, FIG. 1B is a block diagram showing the configuration on the synthesis side, and FIG. A residual waveform standard sample diagram showing an example of a residual waveform obtained by sampling the residual waveform of an input audio signal at a standard sampling frequency in the first embodiment of the invention, FIG. 2b shows a low frequency component of the residual waveform. A down-sampling sample diagram showing an example of down-sampling sampled at the down-sampling frequency, Fig. 2 c is a reference sampling conversion corresponding to the reference sampling frequency for the down-sampling sample of the residual waveform low frequency component shown in Fig. 2 b. FIG. 3 shows the reference sampling conversion sample sequence of the low frequency component of the residual waveform obtained by performing the following steps.
4 is a block diagram showing the configuration of the analysis side in the second embodiment of the present invention, and FIG. 5 is a block diagram showing the configuration of the filter in the second embodiment of the present invention shown in FIG. A block diagram showing in detail the configuration of the coefficient estimator (2) 18, FIG. 6A is a block diagram showing the configuration of the analysis side in the third embodiment of the present invention,
Figure 6B is a block diagram showing the configuration of the synthesis side, Figure 7
Figures A and B are a high frequency reproduction filter prediction coefficient characteristic diagram A formed on the analysis side of the third embodiment of the present invention shown in Figure 6A, its zero-phase prediction coefficient characteristic diagram B, Figure 8A, B is a block diagram showing the configuration of the analysis side A and the synthesis side B in the fourth embodiment of the present invention, and FIGS. 9A and B are the high regions formed on the analysis side in the fourth embodiment of FIG. They are a reproduction filter prediction coefficient characteristic diagram A and a triangular envelope distribution transformation characteristic diagram B based on the same power spectrum. 1...A/D converter, 2...LPC analyzer,
3...Quantizer/decoder, 4...LPC inverse filter, 5...LPF, 6...Down sampler, 7...
...quantizer/decoder, 8...filter coefficient estimator
(1), 9...Quantizer, 10...Power calculator, 11
... Quantizer, 12 ... Multiplexer, 13 ... Demultiplexer/decoder, 14 ... High frequency reproduction filter, 1
5... Multiplier, 16... LPC synthesis filter, 1
7...D/A converter, 18...Filter coefficient estimator (2), 19...Filter coefficient estimator (3), 20
... Coefficient regenerator (1), 21 ... High frequency regeneration filter,
22... Filter coefficient estimator (4), 23... Coefficient regenerator (2), 24... High frequency reproduction filter, 81... Transversal filter, 82... Subtractor, 83
...Power calculator, 181...Cross correlation circuit, 18
2...Autocorrelation circuit, 183...Maximum value search circuit.

Claims (1)

[Claims] 1. A first means for determining a reference sample sequence of a residual waveform of an input audio signal corresponding to a preset reference sampling frequency; a second means for obtaining a down-sampled sequence of low-frequency components of the residual waveform corresponding to a sampling frequency; a transversal type high frequency reproduction filter to be provided on the synthesis side based on the reference sampling conversion sample sequence of the low frequency component of the residual waveform and the reference sample sequence of the residual waveform and without going through a nonlinear circuit; A residual excitation type vocoder comprising: third means for estimating coefficients of; 2. The estimation of the coefficients of the high frequency reproduction filter performed by the third means minimizes the difference between the output of the high frequency reproduction filter to be provided on the synthesis side and the residual waveform of the input audio signal. The residual excitation type vocoder according to claim 1, characterized in that the residual excitation type vocoder is based on a forward coefficient setting method in which coefficients are determined as coefficients when obtaining an optimum impulse response while continuously changing the coefficients. 3. The residual excitation type according to claim 1, characterized in that the estimated coefficients of the high frequency reproduction filter by the third means are converted and output as phase linear filter coefficients having an approximately equivalent impulse response. vocoder. 4. A patent characterized in that the estimated coefficients of the high-frequency reproduction filter according to the third means are converted into estimated coefficients of a triangular envelope array having almost the same power spectrum as the third means and whose energy center position is unchanged and output. A residual excitation type vocoder according to claim 1.