JPH01314300A - Voice coding and decoding system and device thereof - Google Patents

Abstract

PURPOSE:To improve sound quality and to drastically decrease the quantity of computation by obtaining multipulses without using pitch prediction for the pulses which are not strongly correlated with each of pitches. CONSTITUTION:The sound source signal of the voice signal of a frame section is indicated by using the 1st multipulses obtd. by pitch prediction and the 2nd multipulses obtd. by without pitch prediction. Further, the frame is previously divided to the sub-frames corresponding to the pitch periods and the multipulses are obtd. by the pitch prediction only with the one section of the sub-frames in order to utilize the quasiperiodicity of every pitch of the multipulse sound source and to decrease the computation quantity. The signal is reproduced by the multipulses to remove the influence thereof; thereafter, the multipulses are obtd. without the p[itch prediction in said frame. Namely, a 1st pulse decoder 720 decodes the amplitude and position of the pitch predicted multipulses and a 2nd pulse decoder 725 decodes the amplitude and position of the 2nd multipulses.

Description

DETAILED DESCRIPTION OF THE INVENTION (Field of Industrial Application) The present invention relates to an audio encoding/decoding method and apparatus for efficiently encoding and decoding audio signals at a low bit rate.

(Prior Art) As a method for transmitting audio signals at a low bit rate, for example, about 16 Kb/s or less, a multipulse encoding method is known. These represent the sound source signal as a combination of multiple pulses (multipulse), represent the characteristics of the vocal tract with a digital filter, and transmit the information on the sound source pulse and the filter coefficients after determining them for each fixed time interval (frame). ing. For details of this method, see for example Araseki,
“M” by Ozawa, Ono, Ochiai
ulti-pulse Excited 5peech
Coder Ba5ed on Maximum Cr
oss-correlation 5earch Al
gorithm”.

(GLOBECOM 83. IEEE Global
Tele-communication.

It is described in lecture number 23.3.1983X document 1). This method can output a good audio signal after decoding by separately expressing the vocal tract information and the sound source signal, and by using a combination of multiple pulse trains (multipulse) as a means of expressing the sound source signal. . The basic concept of finding a pulse train representing a sound source signal will be explained using FIG. 4. An audio signal divided into frames is manually inputted from an input terminal 800 in the figure. Spectral parameters determined from the audio signal of the current frame are input to the synthesis filter 820. An initial multi-pulse is generated in the sound source calculation circuit 810, and by inputting it to the synthesis filter 820, a synthesized speech waveform is obtained as an output. A subtracter 840 subtracts the synthesized speech waveform from the human input signal. This result is input to the weighting circuit 850, and the weighting in the current frame obtains the error power. And this weighting is done so that the error power is minimized.
In the sound source generation circuit 810, the amplitude and position of a specified number of multipulses are determined.

(Problem to be solved by the invention) However, with this conventional method, the sound quality was good when the bit rate was high enough and the number of sound source pulses was sufficient, but as the bit rate was lowered, the sound quality deteriorated. There was a problem.

In order to improve this problem, a pitch prediction multi-pulse method has been proposed that utilizes the pitch-wise quasi-periodicity (pitch correlation) of a multi-pulse sound source. The details of this method are detailed in, for example, Japanese Patent Application No. 58-139022 (Reference 2), so the explanation will be omitted here. However, although the pitch-wise quasi-periodicity of a multipulse sound source is considered to be large for large-amplitude pulses, such periodicity does not exist for all pulses, and small-amplitude pulses have pitch-wise periodicity. is considered to be small. In the pitch prediction multi-pulse method of Document 2, all pulses are determined by pitch prediction assuming periodicity for each pitch for a predetermined number of pulses within a frame, so all pulses are determined by pitch prediction. The problem with pulses is that pitch prediction actually worsens the characteristics. This is particularly noticeable in transition sections and transitional parts between vowels, and there is a problem in that the sound quality deteriorates in such parts.

Furthermore, in the method of Document 2, since pitch information is included in the impulse response, an impulse response with a very long time length (for example, 20-32 m5) is required, and all pulses of a predetermined number are used for pitch prediction. Therefore, the amount of calculation required to search for pulses is extremely large, and even with current device technology, it is quite difficult to realize a compact device.

The purpose of the present invention is to provide an audio encoding/decoding method and its device that can reproduce better audio than ever before even when the bit rate is high or lower, and that can be realized with a small amount of calculation. It's about doing.

(Means for Solving the Problems) The audio encoding/decoding method of the present invention inputs a discrete audio signal on the transmitting side, and extracts a spectral parameter representing a spectral envelope and a pitch representing a pitch from the audio signal for each frame. the audio signal of the frame is divided into small sections according to the pitch parameter, and the sound source of the audio signal of the frame is divided into one section of the small sections using the spectrum parameter and the pitch parameter. and a second multipulse obtained in the frame using the spectrum parameter after removing the influence of the multipulse, and on the receiving side, the first multipulse and the pitch parameter are expressed as The second multi-pulse is used to restore the sound source signal, and the spectral parameter is used to obtain a synthesized speech signal.

The audio encoding device of the present invention includes a parameter calculation circuit that extracts and encodes a spectral parameter representing a spectral envelope and a pitch parameter for each frame from a manually generated discrete audio signal, and converts the audio signal of the frame into the pitch parameter. a dividing circuit that divides into corresponding sub-intervals, and a first multi-pulse for one of the sub-intervals using the spectrum parameter and the pitch parameter, and encodes the first multi-pulse. an excitation calculation circuit that calculates and encodes a second multi-pulse using the spectral parameters in the frame after removing the influence of the pulse; and a combination of the output code of the parameter calculation circuit and the output code of the excitation calculation circuit. It is characterized by outputting

The audio decoding device of the present invention includes a demultiplexer circuit that separates and decodes a code representing a spectral parameter, a code representing a pitch parameter, a code representing a first multipulse, and a code representing a second multipulse. , divides a frame section into subsections according to the decoded pitch parameter, generates the decoded first multi-pulse for one of the subsections, reproduces the pitch using the pitch parameter, and reproduces the pitch using the pitch parameter. a sound source restoration circuit that generates and adds the decoded second multi-pulse to the reproduced signal to restore the sound source signal; and a synthesized sound signal is obtained using the restored sound source signal and the decoded spectral parameter. It is characterized by having a synthesis filter circuit that outputs.

(Operation) The audio encoding/decoding method and its device according to the present invention provide a multi-pulse (first multi-pulse) obtained by pitch prediction and a multi-pulse obtained without pitch prediction for the sound source signal of an audio signal in a frame section. It is characterized by being expressed using a multi-pulse (second multi-pulse). Furthermore, in order to very efficiently utilize the quasi-synchrony of each pitch of the multipulse sound source and to greatly reduce the amount of calculation, in order to calculate the first multipulse, the frame is preliminarily divided into small sections according to the pitch period. (subframes), and obtain multipulses by pitch prediction for only one section of the subframes. After reproducing the signal using the multi-pulse and removing its influence, the multi-pulse is obtained in the frame using the same method as in Document 1 without pitch prediction.

The basic idea of this method will be explained below using FIG. 2. FIG. 2 is a block diagram showing the operation of the present invention.

An audio signal is input manually from the input terminal 100, and the audio signal is divided into frames having a predetermined time length (for example, 20 m5). LPG analysis unit 150 from the frame audio signal
calculates LPG coefficients of a predetermined order representing the spectral envelope by well-known LPC analysis. In addition to linear prediction coefficients, LPG coefficients include LSP, formant, and LPC.
Other good parameters such as cepstrum can also be used. Furthermore, analysis methods other than LPC, such as cepstrum, PSE, and ARMA methods, can also be used. The following explanation assumes that linear prediction coefficients are used.

The pitch calculation circuit 200 calculates a pitch period M and a pitch coefficient (gain) b from the audio signal of the frame. The well-known autocorrelation method can be used for this. In addition, the pitch coefficient b (gain) can be calculated by using the value of the autocorrelation coefficient at the time delay M in the autocorrelation described above, or by dividing the audio signal into small sections (subframes) for each pitch period M, and each It is also possible to use the slope value when the rms value of the audio signal or prediction residual signal in a subframe is approximated by a linear regression line.

Regarding the latter method, “2.4Kb
pspitch prediction multi-
pulse 5peech coding” (proc
.

IEEE ICASSP88. S4.9.1988)
You can refer to the paper titled (Reference 3).

The operations of the pitch prediction multipulse calculation section 250 and the multipulse calculation section 270 will be explained with reference to FIG. Third
Figure (a) represents the audio signal of a frame. Here, as an example, the frame length is 20 m5. The pitch prediction multipulse calculation unit 250 first divides the frame into small sections (subframes) using the pitch period M, as shown in (b).
Divide into. Here, the subframe length is the same as the pitch period M. Next, using the same method as in Document 2, the impulse response hw of the cascade-connected filter consisting of the pitch recovery filter and the perceptually weighted spectrum envelope synthesis filter is
Find (n). Here, the spectral envelope synthesis filter,
The transfer characteristics of the pitch recovery filter are expressed by equations (1) and (2), respectively.

H, (z) = 1/(1-Σa, z-1)
(1) Hμ) = 1/(1-bz-M)
(2) Here, the order of the pitch recovery filter is 1
It is said that

The impulse response of equations (1) and (2) is, (n), h
When p(n) is the impulse response of the perceptual weighting filter, w(n) is the impulse response hw(n), which is expressed by the following equation.

hw(n)=h, (n) order, (n)*w(n)
(3) Also, the impulse response hws(n) of the spectral envelope synthesis filter subjected to perceptual weighting is h
ws(n) = h, (n) umbrella w(n)
(4) Here, the symbol "-'° represents convolution.

Next, by the same method as in Document 2, we calculate the autocorrelation number R,,(m) of the impulse response hW(n), the cross-correlation function Φhx(m) between the perceptually weighted speech signal and the impulse response hW(n). ). Next, for only one predetermined section of the subframe (for example, section ■ in FIG. 3(b)), the amplitude gi of a predetermined number L (here, 4) of multipulses is , position m1
is determined by pitch prediction. FIG. 3(C) shows the obtained multipulse. Next, the obtained multipulse is passed through a pitch recovery filter defined by equation (2) as shown in Fig. 3(d).
Play the pulse in other subframes like so: Next, a spectral envelope synthesis filter defined by equation (1) is driven using this reproduction pulse to obtain a reproduction signal x'(n).

Subtractor 260 subtracts x'(n) from audio signal x(n) to obtain e(n). Next, the multipulse calculation section 270
is the cross-correlation function between the signal obtained by applying auditory weight to e(n) and the weighted impulse response of the spectral envelope synthesis filter using the same method as in Document 1, and the weighted impulse response of the spectrum envelope synthesis filter. Using the autocorrelation function of the wall impulse response, a predetermined number Q of multipulses within a frame is determined. This is shown in FIG. 3(e). In the diagram, Q
is set as 7.

The transmission information on the transmitting side includes a spectral parameter representing the spectral envelope, the pitch period M of the pitch recovery filter, the gain b, the amplitude and position of L pitch prediction multipulses,
These are the amplitudes and positions of Q multipulses.

(Embodiment) In FIG. 1 showing an embodiment of the present invention, an input terminal 50
A discrete audio signal x(n) from 0 is input. The spectral and pitch parameter calculation circuit 520 calculates the spectral parameters ai of the spectral envelope synthesis filter representing the spectral envelope of the audio signal in the divided frame sections (for example, 20 m5) using the well-known LPC analysis method. Further, the coefficients of the pitch recovery filter and the pitch period M are determined by the well-known autocorrelation method or the method shown in the above-mentioned document 3.

A quantizer 525 performs quantization on the obtained spectrum parameters, coefficients, and pitch period.

The quantization method is described in Japanese Patent Application No. 59-272435 (
Scalar quantization or vector quantization as shown in Reference 4) may be performed. For a specific method of vector quantization, see Makhou1, for example.
'Vector quantization by Mr.
in 5peech coding” (Proc, I
You can refer to papers such as EEE, pp. 1551-1588° 1985X Reference 5). The dequantizer 530 dequantizes and outputs the quantized result.

A subtracter 535 subtracts the influence signal from the audio signal of the frame and outputs the result. Weighting circuit 540 perceptually weights the audio signal using the dequantized spectral parameters. Regarding the weighting method, reference can be made to the weighting circuit 200 in Document 2.

The impulse response calculation circuit 550 calculates the impulse response hw, (n
) and the weighted impulse response hw(n) of the filter consisting of the cascade connection of equations (1) and (2) above are calculated using aI, pitch period M', and coefficient b'. For a specific method, refer to the impulse response calculation circuit in Document 2.

The autocorrelation function calculation circuit 560 calculates two types of autocorrelation functions for the two types of impulse responses, and outputs them to the sound source pulse calculation circuit 580 and the pulse calculation circuit 586, respectively. For the method of calculating the autocorrelation function, reference can be made to the autocorrelation function calculation circuit 180 in Document 2 and Document 4.

A cross-correlation function calculation circuit 570 calculates a cross-correlation function Φxh(m) between the perceptually weighted signal and the impulse response hw(n).

The sound source pulse calculation circuit 580 first divides the frame into subframe sections as shown in FIG. 3(b) using the pitch period M' obtained by dequantizing the frame. Then, for one predetermined subframe section (for example, subframe ■ in FIG. 3(b)), Φ, , (m) and R,h(m)
The amplitude g, and the position m of the L multi-pulse trains are calculated using
, find. Regarding the pulse train calculation method, reference can be made to the sound source pulse calculation circuit in Document 2.

A quantizer 585 quantizes the amplitude and position of the multi-pulse train and outputs a code. The specific method is described in the above document 4.
etc. can be referred to. This output is further dequantized and passed through a pinch reconstruction filter 605 and a spectral envelope synthesis filter 610 to obtain a synthesized speech signal x'(n) based on the pitch prediction multipulse.

The subtracter 615 subtracts the synthesized speech signal x'(n) from the speech signal x(n) to obtain a residual signal e(n).
get.

A weighting circuit 600 performs perceptual weighting on the residual signal.

A cross-correlation function calculation circuit 603 calculates a cross-correlation function between the output of the weighting circuit 600 and the impulse response hws(n).

The pulse calculation circuit 586 uses the cross-correlation function and the autocorrelation function of the impulse response hw8(n) to find the amplitude and position of a predetermined number of multi-pulses.

The quantizer 620 quantizes and outputs the amplitude and position of the multi-pulse, and also dequantizes and outputs them to the synthesis filter 625.

A synthesis filter 625 synthesizes and outputs the residual signals.

The adder 627 includes the synthesis filter 625 and the synthesis filter 61.
The reproduced signal of the frame is obtained by adding the outputs of 0, and the influence signal for the next frame is also determined and output. For this specific method, reference can be made to the above-mentioned document 4.

The multiplexer 635 combines and outputs the code representing the amplitude and position of the multi-pulse train output from the quantizers 585 and 620, and the code representing the spectrum parameter, pitch period, and coefficient that are the output from the parameter quantizer 525.

On the other hand, on the receiving side, the demultiplexer 710 outputs the amplitude of the pitch prediction multipulse (first multipulse), the code representing the position, the amplitude of the multipulse (second multipulse), the code representing the position, the spectrum parameter, Separate and output codes representing pitch periods and coefficients.

The first pulse decoder 720 decodes the amplitude and position of the pitch prediction multi-pulse. A second pulse decoder 725 decodes the amplitude and position of the second multi-pulse. The spectral and pitch parameter decoder 750 operates in the same manner as the inverse quantizer 530 on the transmitting side, and decodes and outputs the spectral parameter, pitch period M', and coefficient b'.

The first pulse generator 720 generates and outputs a sound source signal based on the first multi-pulse (pitch prediction multi-pulse) for a frame length. The second pulse generator 725 generates a sound source signal based on the second multi-pulse for a frame length.

The pitch recovery filter 735 performs the same operation as the pitch recovery filter 605 on the transmission side, and the pulse generator 730
, the decoded pitch period M' and the coefficient b' are input, and a sound source signal in which the pitch is reproduced in a frame is obtained and output to an adder 740.

The adder 740 combines the sound source signal and the second pulse generator 72.
7 is added to obtain a frame drive sound source signal, and the synthesis filter circuit 760 is driven.

The synthesis filter circuit 760 uses the driving sound source signal and the decoded spectral parameters to obtain and output a synthesized speech waveform for each frame.

The configuration described above is only one embodiment of the present invention, and various modifications are possible.

As a method for calculating multi-pulses, in addition to the method shown in Document 1, various well-known methods can be used.

For example, “A 5tu” by Ozawa et al.
dy on Pulse 5earch Algori
thms for Multi-pulse 5pee
ch Coder Realization” (IEE
E JSAC, pp.

133-141.1986X document 6).

Furthermore, as a method for calculating the pitch period, in addition to the method shown in the above-mentioned embodiment, for example, as shown in the following equation (3), the past sound source signal v(n), a pitch reproduction filter, a spectral envelope synthesis filter, etc. It is also possible to search for a position M that minimizes the error power E between the reproduced signal and the input audio signal x(n) of the current subframe, and find the coefficient S at that time.

E=Σ[(x(n)-b-v(n-M)-h,(n))
−w(n, )]2(5) Here, hs(n) represents the impulse response of the spectrum synthesis filter, and w(n) represents the impulse response of the perceptual weighting circuit.

Furthermore, a linear deviation I may be allowed in the pitch period M of the subframe. For a specific method, reference can be made to the above-mentioned document 3 and the like. However, in this case, in addition to the period M, it is also necessary to transmit the above-mentioned I as pitch information.

Further, if the transmitting side synthesis filter 610 is configured to reproduce the weighted signal and subtract it from the weighting circuit 540, the weighting circuit 600 can be omitted.

Furthermore, it is also possible to omit the subtraction of the influence signal on the transmitting side. With such a configuration, the synthesis filter 625 and the adder 627 become unnecessary.

(Effects of the Invention) According to the present invention, pulses with strong periodicity for each pitch can be expressed very efficiently by determining pulses in one subframe section by pitch prediction, and pulses with strong periodicity for each pitch can be expressed very efficiently. For pulses that do not exist, multi-pulses are obtained without using pitch prediction, so compared to the conventional method of calculating all pulses using pitch prediction, the sound quality is improved in areas where periodicity is weak, such as vowel transition parts and transient parts. The effect is that it can be greatly improved. Furthermore, since only some multipulses are determined by pitch prediction,
It is possible to significantly reduce the amount of calculation required to search for pitch prediction multi-pulses, and has the great effect of significantly reducing the amount of calculation compared to conventional methods.

[Brief explanation of the drawing]

FIG. 1 is a block diagram showing the configuration of an embodiment of a speech encoding/decoding method and apparatus according to the present invention, and FIG. 2 is a block diagram showing the operation of the present invention. FIG. 3 is a block diagram showing an example of pulse train search. FIG. 4 is a block diagram showing an example of a conventional system. In the figure, 150...LPG analysis unit, 200...Pitch calculation unit, 250...Pitch prediction multipulse calculation unit, 270-1...Multipulse calculation unit, 520...
Spectral/Pichi parameter calculation circuit, 525... Parameter quantizer, 530... Inverse quantizer, 535.
260.615.840...Subtractor, 540.600
．． 850... Weighting circuit, 550... Impulse response calculation circuit, 560... Autocorrelation function calculation circuit, 5
70.603... Cross-correlation function calculation circuit, 585.6
20...Quantizer, 627.740...Adder, 5
86...Pulse calculation circuit, 605.735...Pitch reproduction filter, 610.625.760.820...
-Synthesis filter, 635...Multiplexer, 710
... Demultiplexer, 720... First pulse decoder, 725... Second pulse decoder, 750...
Spectral pitch parameter decoder, 730...first pulse generator, 727...second pulse generator.

Claims (3)

[Claims] (1) On the transmitting side, a discrete audio signal is input, and a spectral parameter representing a spectral envelope and a pitch parameter representing a pitch are extracted from the audio signal for each frame,
The audio signal of the frame is divided into small sections according to the pitch parameter, and the sound source of the audio signal of the frame is determined for one of the small sections using the spectrum parameter and the pitch parameter. It is expressed as a multi-pulse and a second multi-pulse obtained in the frame using the spectral parameter after removing the influence of the multi-pulse, and on the receiving side, the first multi-pulse, the pitch parameter and the second multi-pulse are expressed.
A speech encoding/decoding method characterized in that a sound source signal is restored using the multipulses of the above, and a synthesized speech signal is obtained using the spectral parameters. (2) A parameter calculation circuit that extracts and encodes a spectral parameter representing a spectral envelope and a pitch parameter representing a pitch from an input discrete audio signal for each frame; a dividing circuit that divides into sections, a first multipulse for one section of the small sections using the spectral parameter and the pitch parameter, and an effect due to the encoded first multipulse; an excitation calculation circuit that calculates and encodes a second multipulse using the spectral parameters in the frame after removing the spectral parameters; and outputs a combination of the output code of the parameter calculation circuit and the output code of the excitation calculation circuit. A speech encoding device characterized by: (3) A code representing the spectral parameter, a code representing the pitch parameter, a code representing the first multipulse, and a code representing the second multipulse.
a demultiplexer circuit that separates and decodes the code representing the multi-pulse of
The decoded first multi-pulse is generated for each signal, the pitch is reproduced using the pitch parameter, and the decoded second multi-pulse is generated and added to the pitch-reproduced signal to generate the sound source signal. A speech decoding device comprising: a sound source restoration circuit for restoring; and a synthesis filter circuit for obtaining and outputting a synthetic speech signal using the restored sound source signal and the decoded spectrum parameter.