JP3362471B2 - Audio signal encoding method and decoding method - Google Patents

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention divides an input speech signal into blocks and performs an encoding process in units of the divided blocks. On the synthesis side, a periodic fundamental wave such as a sine wave and its harmonics are generated. The present invention relates to an audio signal encoding method and an audio signal decoding method for synthesizing an audio signal by using.

[0002]

2. Description of the Related Art Various coding methods are known in which signal compression is performed by utilizing the statistical properties of audio signals (including voice signals and acoustic signals) in the time domain and frequency domain and human auditory characteristics. ing. As this encoding method,
Broadly speaking, time domain coding, frequency domain coding,
Examples include analysis and synthesis coding.

As an example of high-efficiency encoding of a voice signal or the like, M
BE (Multiband Excitation) coding, SBE (Singleband Excitation) coding, Harmonic coding, SBC
(Sub-band Coding), LPC (Linear
Predictive Coding: Linear predictive coding) or DC
T (Discrete Cosine Transform), MDCT (Modified DC)
In T), FFT (Fast Fourier Transform), etc., when quantizing various information data such as spectrum amplitude and its parameters (LSP parameter, α parameter, k parameter, etc.), conventionally, scalar quantization is performed. There are many.

In the voice analysis / synthesis system such as the PARCOR method, since the excitation source is switched at each block (frame) on the time axis, voiced sound and unvoiced sound cannot be mixed in the same frame. , As a result, high quality voice was not obtained.

On the other hand, in the above MBE coding, for each voice in one block (frame), each harmonic (harmonic) of the frequency spectrum or each band (band) in which a few harmonics are grouped together. Since voiced / unvoiced sound discrimination (V / UV discrimination) is performed for each band divided for each band or for each band divided by a fixed bandwidth (for example, 300 to 400 Hz), sound quality Is recognized. V / U for each band
V discrimination mainly looks at how strongly the spectrum in the band has a harmonic structure.

The pitch information, the V / UV discrimination information for each band, and the amplitude information obtained on the analysis side of this MBE encoding are sent to the synthesis side, and the sine wave (or cosine wave) is sent on the synthesis side based on these information. Etc.) The original speech waveform is synthesized by processing signals and noise signals. When transmitting information, the phase information of each harmonic component is not transmitted because the amount of data transmission (bit rate) becomes very high.

[0007]

By the way, when synthesizing an original speech waveform using a sine wave, a cosine wave, etc. on the synthesizer side,
Since the phase information about each harmonic component is not sent in particular, the phase is determined by a method such as the so-called minimum phase transition method or zero phase method.

However, in the voice signal waveform which is phase-determined and synthesized by these methods, when the original voice signal waveform also has a peak at the intermediate position (1/2 pitch position) of the peaks of the pitch period. The intermediate peak may not appear in the composite waveform. Such a synthesized sound has a stronger offensive sensation than the original sound. As one of the causes of this harshness, it is considered that the energy of the intermediate peak is added to the peak of the pitch cycle and the peak level rises.

The present invention has been made in view of the above circumstances, and it is possible to favorably perform transmission and reproduction of an audio signal waveform having a peak at an intermediate position of each peak of the pitch period. It is an object of the present invention to provide a method for encoding and decoding a simple voice signal.

[0010]

A method of encoding a voice signal according to the present invention comprises a step of dividing an input voice signal into blocks on a time axis, and detecting a basic pitch period in the divided blocks. A step of detecting a pitch peak position which is a peak position of the basic pitch cycle including a maximum peak position in the block, and an intermediate position in the block which is ½ pitch cycle away from the pitch peak position. A step of detecting an intermediate peak in the vicinity and obtaining a statistic e ₂ of the value of the intermediate peak; and detecting a quarter pitch peak near the intermediate position between each pitch peak position and the position of the intermediate peak in the block. Then, the step of obtaining the statistic e ₃ of the value of the ¼ pitch peak and the above-mentioned statistic e ₂ and e
The problem described above is solved by having a step of obtaining phase information based on the magnitude relationship of ₃ .

More specifically, a step of dividing the input audio signal into blocks on the time axis, a step of detecting a basic pitch period in the divided blocks, and a detection of a maximum peak position in the block. And a step of detecting one or more (or all of) pitch peak positions in the block based on the maximum peak position and the basic pitch period, and a plurality of pitches determined in the block. A step of detecting an intermediate peak which is a peak near the intermediate position of the peak position and obtaining a statistic (for example, an average value or a cumulative addition value) e ₂ of the values of these intermediate peaks;
The 1/4 pitch peaks near the intermediate position between the respective pitch peak positions and the corresponding intermediate peak positions are detected, and the statistic e ₃ of the values of these 1/4 pitch peaks is detected.
And a step of obtaining phase information based on the magnitude relation between the statistics e ₂ and e ₃ .

The phase information may be 1 bit or 2 bits or more. The phase information may not be sent when the statistics e ₂ and e ₃ are small. Also,
A statistic e ₁ of the values of the plurality of pitch peaks in the block is detected, and a ratio e _{2 to} each statistic e ₂ or e ₃ is detected.
It is also possible to obtain the phase information according to the magnitude relationship between / e ₁ and e ₃ / e ₁ .

Next, a voice signal decoding method according to the present invention is a voice signal decoding method for synthesizing a voice signal using a periodic fundamental wave and its harmonic components, wherein the peak of the fundamental pitch period is used. The phase information obtained based on the peak at the intermediate position and the peak at the ¼ pitch position is supplied,
The above problem is solved by changing the terminal phase of the block according to the phase information.

Here, when synthesizing the fundamental wave and its harmonic components, it is conceivable to shift only the termination phase of the odd harmonics with respect to the reference phase according to the phase information.
Specifically, the termination phase of the odd harmonics may be shifted with respect to the reference phase. When the phase information is 1 bit, it is shifted by π / 4, and when it is 2 bits, it is π / 8 according to the phase information. It is preferable to shift by an integer multiple of.

[0015]

[Effect] Even in the case of an audio signal in which a peak also occurs at the middle position of the peak of the pitch period, the synthesis processing is performed to reproduce the middle peak by the phase information,
It is possible to suppress the unpleasant sensation that has occurred in the conventional synthesis processing and obtain a good reproduced sound (synthesized sound).

[0016]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENTS An embodiment of a voice signal encoding method and a voice decoding method according to the present invention will be described below. This audio signal encoding method and decoding method includes MBE (Mu
ltibandExcitation: Multiband excitation (encoding), etc., the signal for each block is converted on the frequency axis, divided into multiple bands, and V (voiced sound) or UV (unvoiced sound) for each band.
It is possible to use a voice signal analysis / synthesis method for determining whether or not.

That is, as a general audio signal encoding method and decoding method to which the present invention is applied, an audio signal is divided into blocks every fixed number of samples (for example, 256 samples), and orthogonal transformation such as FFT is performed. While converting to spectrum data on the frequency axis, the pitch of the voice in the block is extracted, the spectrum on the frequency axis is band-divided at intervals according to this pitch, and V (voiced sound) for each divided band. / UV (unvoiced sound) is discriminated. The V / UV discrimination information is encoded and transmitted together with the pitch information and the spectrum amplitude data, and the synthesizing (decoding) side performs the audio signal waveform synthesizing process based on these information.

Here, assuming a voice analysis / synthesis system such as an MBE vocoder, the sampling frequency fs for the voice signal input on the time axis is usually 8 kHz.
The total bandwidth is 3.4 kHz (the effective bandwidth is 200-
3400Hz), the pitch lag from the higher female voice to the lower male voice (the number of samples corresponding to the pitch period)
Is about 20 to 147. Therefore, the pitch frequency is 8000/147 ≒ 54 (Hz) to 8000/20 = 400 (Hz)
It will fluctuate within the range. Therefore, about 8 to 63 pitch pulses (harmonics) stand up to 3.4 kHz on the frequency axis.

In this way, considering that the number of divided bands (the number of bands) for each block (frame) changes between about 8 and 63 when the bands are divided at intervals according to the pitch, It is preferable to reduce or reduce the number of divided bands to a fixed number (for example, about 12).

Further, in the embodiment of the present invention, it is detected whether or not there is a certain amount of peak at the intermediate position (1/2 pitch period position) of each peak of the pitch period in the time waveform of the input voice signal. However, a phase code, which is 1-bit phase information depending on the presence or absence of the intermediate peak, is sent. On the synthesizing side, the terminal phase of each block (frame) of the fundamental wave and the harmonic wave used for synthesizing the waveform is changed according to this phase code. Specifically, when there is an intermediate peak, the odd harmonics are shifted by π / 4 with respect to the reference phase, and when there is no intermediate peak,
The same phase determination method as the conventional zero phase and minimum phase transition is used.

Next, an MBE (Multib) which is a kind of an audio signal analyzing / synthesizing device (so-called vocoder) to which the above audio signal encoding method and audio signal decoding method can be applied.
andExcitation: Multiband Excitation) A specific example of a vocoder will be described with reference to the drawings.

The MBE vocoder described below is a DW G
riffin and JS Lim, "MultibandExcitation Vocode
r, "IEEE Trans.Acoustics, Speech, and Signal Process
ing, vol.36, No.8, pp.1223-1235, Aug. 1988, the conventional PARCOR (PARtial au
In a to-CORrelation (partial autocorrelation) vocoder or the like, a voiced sound section and an unvoiced sound section were switched for each block or frame when modeling a voice, whereas in the MBE vocoder, at the same time (same block or frame). The model is based on the assumption that there are voiced and unvoiced intervals in the frequency domain of (in).

FIG. 1 is a block diagram showing the overall schematic configuration of an embodiment in which the audio signal encoding method according to the present invention is applied to the encoder of the MBE vocoder. In FIG. 1, an audio signal is supplied to an input terminal 11, and the input audio signal is sent to a filter 12 such as an HPF (high-pass filter) to be a so-called DC signal.
Removal of (DC) offset and band limitation (for example, 200
At least low frequency component (2)
(Less than 00 Hz) is removed. The signal obtained through the filter 12 is sent to the pitch extraction unit 13 and the windowing processing unit 14, respectively. In the pitch extraction unit 13, the input voice signal data has a predetermined sample number N (for example, N = 25
6) It is divided into blocks in units (or is cut out by a rectangular window), and pitch extraction is performed on the voice signal in this block. Such a cut-out block (256 samples) is converted into, for example, L as shown in FIG.
Samples (eg, L = 160) are moved in the time axis direction at frame intervals, and the overlap between blocks is NL samples (eg, 96 samples).
In addition, the windowing processing unit 14 applies a predetermined window function, for example, a Hamming window, to one block of N samples, and sequentially moves the windowed block in the time axis direction at intervals of one frame of L samples.

When this windowing process is expressed by a mathematical expression, x _w (k, q) = x (q) w (kL-q) (1) In this equation (1), k is a block number,
q represents the time index (sample number) of the data, and the q-th data x (q) of the input signal before processing is windowed by the window function (w (kL-q)) of the kth block. It is shown that the data x _w (k, q) can be obtained by doing so. The window function w _r (r) in the case of the rectangular window as shown in A of FIG. 2 in the pitch extraction unit 13 is w _r (r) = 1 0 ≦ r <N (2) = 0 r < 0, N ≦ r Further, the window function w _h in the case of Hamming window as shown in B of FIG. 2 in the windowing processing section 14 (r) _{is, w h (r) = 0.54} - 0.46 cos (2πr / (N-1)) 0 ≦ r <N (3) = 0 r <0, N ≦ r. Such a window function w _r (r) or w (1) when using the _h (r) formula of the window function w (r) (= w (KL-
q)), the zero-zero interval is 0 ≦ kL−q <N, and is modified as follows: kL−N <q ≦ kL Therefore, for example, in the case of the above rectangular window, the window function w _r (kL−q) = 1
This is when kL−N <q ≦ kL, as shown in FIG. In addition, the above formulas (1) to (3) have a length N
It shows that the window of (= 256) samples advances by L (= 160) samples. Below, above (2)
N points (0≤r
The non-zero sample sequences of <N) are respectively x _wr (k, r) and x _wh
We will denote it as (k, r).

In the windowing processing section 14, as shown in FIG. 4, 179 is applied to the sample sequence x _wh (k, r) of one block of 256 samples to which the Hamming window of the equation (3) is applied.
Two samples of 0 data are added (so-called zero padding) to form 2048 samples, and the orthogonal transform unit 15 performs orthogonal transform such as FFT (Fast Fourier Transform) on the time-axis data sequence of 2048 samples. Processing is performed.

Alternatively, there is also a method of reducing the amount of processing by performing FFT with 256 points as it is without zero padding.

In the pitch extraction unit 13, pitch extraction is performed based on the sample string of x _wr (k, r) (one block N samples). The pitch extraction method is known to use periodicity of time waveform, periodic frequency structure of spectrum, autocorrelation function, etc. In this embodiment, the autocorrelation method of center clip waveform is adopted. is doing. For the center clip level in the block at this time,
One clip level may be set for each block, but when the peak level of the signal of each part (each sub-block) obtained by subdividing the block is detected, and the difference in peak level between these sub-blocks is large, In addition, the clip level is changed stepwise or continuously within the block. The pitch period is determined based on the peak position of the autocorrelation data of this center clip waveform.
At this time, a plurality of peaks are obtained from the autocorrelation data belonging to the current frame (the autocorrelation is obtained for the data of N samples of one block), and the maximum peak of the plurality of peaks is equal to or larger than a predetermined threshold value. In the case of, the maximum peak position is set as the pitch cycle, and in other cases, the pitch is within the pitch range that satisfies a predetermined relationship with the pitch other than the current frame, for example, the pitch of the previous frame and the pitch of the previous frame. As ± 2
The peak within the range of 0% is obtained, and the pitch of the current frame is determined based on this peak position. In this pitch extraction unit 13, a relatively rough pitch search is performed by an open loop, and the extracted pitch data is sent to a high precision (fine) pitch search unit 16 and a closed loop high precision pitch search (pitch) is performed. Fine search) is performed.

A method of obtaining the pitch from the autocorrelation of the residual waveform obtained by LPC analysis of the input waveform instead of the center clip waveform may be used.

The high-precision (fine) pitch search section 16 includes coarse (rough) pitch data of integer (integer) values extracted by the pitch extraction section 13 and, for example, FFT-processed on the frequency axis by the orthogonal transformation section 15. And the data of. In this high-precision pitch search unit 16, with the coarse pitch data value as the center, ± several samples are shaken in increments of 0.2 to 0.5 to give an optimum decimal point (floating).
To the value of the fine pitch data of. As a fine search method at this time, so-called synthesis analysis
Using the (Analysis by Synthesis) method, the pitch is selected so that the synthesized power spectrum is closest to the power spectrum of the original sound.

The fine search of the pitch will be described. First, in the MBE vocoder, the FF
Let S (j) as spectrum data on the frequency axis orthogonally transformed by T etc. be expressed as S (j) = H (j) | E (j) | 0 <j <J (4) It is assumed that the model. Here, J corresponds to ω _s / 4π = f _s / 2, and the sampling frequency f
_{When s} = ω _s / 2π is, for example, 8 kHz, it corresponds to 4 kHz. In the equation (4), when the spectrum data S (j) on the frequency axis has a waveform as shown in FIG.
(j) indicates the spectrum envelope (envelope) of the original spectrum data S (j) as shown in B of FIG.
(j) shows the spectrum of the excitation signal (excitation) which is periodic at the same level as shown in C of FIG. That is, the FFT spectrum S (j) is the spectrum envelope H (j) and the power spectrum | E of the excitation signal.
(j) | is modeled as |.

Power spectrum of the above excitation signal | E (j)
| Is a spectral waveform corresponding to the waveform of one band (band) for each band on the frequency axis in consideration of the periodicity (pitch structure) of the waveform on the frequency axis determined according to the pitch. It is formed by arranging it repeatedly. The waveform for one band is obtained by FFT by regarding a waveform obtained by adding (filling with 0) 1792 samples of 0 data to a Hamming window function of 256 samples as shown in FIG. 4 as a time axis signal. It can be formed by cutting out an impulse waveform having a certain bandwidth on the frequency axis according to the pitch.

Next, for each band divided according to the above pitch, a value (a kind of amplitude) | A that represents the above H (j) (minimizes the error for each band) | A _m
Ask for |. Here, for example, when the lower limit point and the upper limit point of the m-th band (band of the m-th harmonic) are a _m and b _m , respectively, the error ε _m of the m-th band is

[0033]

[Equation 1]

It can be represented by | A _m | that minimizes this error ε _m is

[0035]

[Equation 2]

Thus, when | A _m | in the equation (6),
Minimize the error ε _m .

Such an amplitude | A _m | is obtained for each band, and the obtained amplitude | A _m | is used to obtain the error ε _m for each band defined by the above equation (5). Next, the sum Σε of all the bands of the error ε _m for each band as described above.
_{Find m} . Further, such an error sum value Σε _m of all bands is obtained for some slightly different pitches, and a pitch that minimizes the error sum value Σε _m is obtained.

That is, with the rough pitch obtained by the pitch extraction unit 13 as the center, several types are prepared up and down in steps of, for example, 0.25. The error sum value Σε _m is obtained for each of these plural kinds of slightly different pitches. In this case, if the pitch is determined, the bandwidth is determined, and from the above formula (6), the power spectrum | S (j) |
(j) | can be used to find the error ε _m in the above equation (5), and the total sum Σε _m of all the bands can be found. This error sum total value Σε _m is obtained for each pitch, and the pitch corresponding to the minimum error sum value is determined as the optimum pitch. As described above, the high-precision pitch search unit obtains the optimum fine (for example, 0.25 step) pitch, and the amplitude | A _m | corresponding to this optimum pitch is determined. The calculation of the amplitude value at this time is performed in the voiced sound amplitude evaluation unit 18V.

In the above description of the pitch fine search, in order to simplify the explanation, all bands are voiced (Vo
Assuming the case of iced), MBE as described above
In the vocoder, unvoiced sound (Un
Since a model in which a voiced) region exists is used, it is necessary to distinguish voiced sound / unvoiced sound for each band.

Optimal pitch from the high precision pitch search section 16 and amplitude from amplitude evaluation section (voiced sound) 18V | A
The data of _m | is sent to the voiced sound / unvoiced sound discrimination unit 17,
The voiced sound / unvoiced sound is discriminated for each band. NSR (noise to signal ratio) is used for this determination. That is, the NSR that is the NSR of the m-th band
_m is

[0041]

[Equation 3]

This NSR _m is a predetermined threshold value Th ₁
When larger than (for example, Th ₁ = 0.2) (that is, when the error is large), | A _m || E (j) in that band
The approximation of | S (j) | by | is not good (the above excitation signal |
E (j) | is unsuitable as a base) and the band is determined to be UV (Unvoiced). In other cases, it can be determined that the approximation is performed to a certain extent, and the band is determined to be V (Voiced, voiced sound).

By the way, as described above, the number of bands divided by the fundamental pitch frequency (the number of harmonics) is
The number of V / UV flags for each band similarly varies because the voice varies in the range of about 8 to 63 depending on the pitch (the size of the pitch). Therefore, in this embodiment, the V / UV discrimination results are collected (or degenerated) for each fixed number of bands divided by a fixed frequency band.
I am trying. Specifically, a predetermined band (for example, 0 to 4000 Hz) including a voice band is divided into N _B (for example, 12) bands, and according to the NSR value in each band,
For example, the weighted average value is discriminated by a predetermined threshold Th ₂ ,
The V / UV of the band is judged.

Next, in the unvoiced sound amplitude evaluation unit 18U, the frequency axis data from the orthogonal transformation unit 15, the fine pitch data from the pitch search unit 16, and the voiced sound amplitude evaluation unit 1 are included.
Data of amplitude | A _m | from 8 V and V / UV (voiced sound / unvoiced sound) discrimination data from the voiced sound / unvoiced sound discrimination unit 17 are supplied. This amplitude evaluation section (unvoiced sound) 1
In 8U, the amplitude is obtained again (amplitude re-evaluation is performed) with respect to the band that is determined to be unvoiced sound (UV) by the voiced sound / unvoiced sound determination unit 17. The amplitude | A _m | _UV for this UV band is

[0045]

[Equation 4]

It is calculated by

The data from the amplitude evaluation unit (unvoiced sound) 18U is sent to the data number conversion (a kind of sampling rate conversion) unit 19. The data number conversion unit 19 has a different number of divided bands on the frequency axis according to the pitch,
This is for keeping the number constant, considering that the number of data (especially the number of amplitude data) is different. That is, for example, when the effective band is set to 3400 kHz, the effective band is divided into 8 bands to 63 bands according to the pitch, and the amplitude | A _m | (UV The number of data m _MX +1 including the band amplitude | A _m | _UV also changes from 8 to 63.
Therefore, in the data number conversion unit 19, this variable number m _MX +
One amplitude data is converted into a fixed number M (for example, 44) of data.

Here, in the present embodiment, for example, for amplitude data of one block of the effective band on the frequency axis,
After adding dummy data that interpolates the values from the last data in the block to the first data in the block to increase the number of data to N _F , the bandwidth-limited type _OS times (for example, 8 times) The amplitude data of the number of O _S times is obtained by performing the oversampling of the number of (A), and the number of the amplitude data of the number of O _S times ((m _MX +1) × O _S number) is linearly interpolated to obtain a larger number of N _M ( For example, the number of data is expanded to 2048, and the N _M pieces of data are thinned out to obtain the fixed number M (for example, 4 pieces).
4) data.

The data from the data number conversion unit 19 (the above-mentioned fixed number M of amplitude data) is the vector quantization unit 20.
To a vector, and a predetermined number of pieces of data are combined into a vector, and vector quantization is performed. The quantized output data from the vector quantizer 20 (main part thereof) is the high-precision (fine) pitch data from the high-precision pitch search unit 16 and the voiced / unvoiced sound from the voiced / unvoiced sound discrimination unit 17. The (V / UV) discrimination data and the phase code data described later are sent to the encoding unit 21 and encoded.

Each of these data is obtained by processing the data in the block of N samples (for example, 256 samples), but the block is the frame of L samples on the time axis. , The data to be transmitted is obtained in the frame unit. That is, the pitch data, the V / UV discrimination data, the amplitude data, and the phase code data are updated at the frame cycle. Further, regarding the V / UV discrimination data from the voiced sound / unvoiced sound discrimination section 17,
As mentioned above, if necessary, it is reduced (degenerated) to about 12 bands, and voiced sound (V) at one place or less in all bands.
A V / UV discrimination data pattern in which a region and an unvoiced sound (UV) region are separated and V (voiced sound) on the low frequency side is expanded to the high frequency side when a predetermined condition is satisfied. is there.

In the encoding unit 21, for example, CR
C addition and rate 1/2 convolutional code addition processing is performed. That is, C is the important data of the phase code data, the pitch data, the voiced / unvoiced (V / UV) discrimination data, and the quantized output data.
After RC error correction coding is performed, convolutional coding is performed. The encoded output data from the encoding unit 21 is sent to the frame interleaving unit 22, is subjected to interleave processing together with some (for example, low importance) data from the vector quantization unit 20, and is extracted from the output terminal 23. Transmission (or recording / playback) to the synthesis side (decoding side)
To be done.

Next, on the time waveform in the block of the input audio signal, phase code data indicating whether or not there is a peak (intermediate peak) of a certain size at the intermediate position of each peak of the basic pitch period. Will be described. In the following description, a signal having no intermediate peak will be referred to as a type I audio signal, and a signal having an intermediate peak will be referred to as a type II audio signal.

The output signal from the filter 12 is sent to the peak search section 26, and the peak having the largest value (maximum peak) in the block (for example, 256 samples) is detected.

In the pitch peak position detecting section 27, the position of the maximum peak in the block (position on the time axis) and the basic pitch information from the high precision pitch searching section 16 (or the pitch information from the pitch extracting section 13). However, the peaks (pitch peaks) that repeat every pitch period p are detected. This is the maximum sample value m1 in the vicinity (a few samples) of a position distant from the maximum peak position by an integer multiple of the basic pitch period p.
Should be detected. Generally, a plurality of sample values m1 are obtained in one block, and these sample values m1 ₁ , m1 ₂ , ...
・ ・

In the next local maximum detection unit 28, in the block, in the vicinity of the intermediate position of each pitch peak position (positions apart from each pitch peak position by 1/2 pitch period), and further each pitch peak. A peak (so-called maximum) in the vicinity of a position intermediate between the position and each of the intermediate positions (positions separated from each pitch peak position by ¼ pitch cycle) is detected.

Here, the above pitch peaks m1 ₁ and m1
₂ , the intermediate peaks which are peaks in the vicinity of the positions ½ pitch period (p / 2) apart from the respective positions t _p1 , t _p2 , ... Of m2 ₁ , m2 ₂ , ..., and the 1/4 pitch peaks, which are peaks in the vicinity of positions separated by 1/4 pitch period (p / 4), are m3 ₁ ,
m3 ₂ , ... An example of the relationship between these peaks and the input signal waveform is shown in FIG.

Here, the pitch peaks m1 ₁ , m1 ₂ ,
As the statistic amount of ..., the sum e of the set of pitch peaks m1 _k satisfying a condition larger than a predetermined threshold Th _m
Ask for ₁ . That is, the sum total value of m1 _k of the set of k satisfying m1 _k > Th _m is e ₁ . Similarly, for the intermediate peaks m2 ₁ , m2 ₂ , ..., The sum value e ₂ of the intermediate peaks m2 _k corresponding to the set of k satisfying the above conditions is obtained, and the quarter pitch peaks m3 ₁ , m3 ₂ ,. for even ..., obtaining the sum value e ₃ 1/4 pitch peak m @ 2 _k corresponding to the set of k that satisfies the above conditions. As the statistic, an average value, an RMS value or the like can be used in addition to the above sum value.

The next classification unit 29 classifies the type I and the type II using the statistics (sum values) e ₁ , e ₂ and e ₃ . That is, the ratio r ₁ of the total sum value e _{1 of the} pitch peaks to the total sum values e ₂ and e ₃ ,
As r ₂ , r ₁ = e ₂ / e ₁ and r ₂ = e ₃ / e ₁ are used. For example, (c1) r ₁ > 0.4 and r ₁ > r ₂ × 1.5 (c2) r ₁ > 0.5 and r ₁ > r ₂ × 1.3 (c3) For three conditions (c1), (c2), and (c3) of r ₁ <1.5 and r ₂ <1.5, ((c1 ) Or (c2)) and (c3) are satisfied, it is determined that the above-mentioned type II classification, that is, the above intermediate peak is significantly present.

The above condition (c3) is for excluding cases where there are many noises and errors. The above conditions (c1), (c
2) is for detecting the presence of a strong 1/2 pitch period peak (intermediate peak). The signal of the type II waveform is different from the signal of the type I waveform having no or a small intermediate peak.

The classifying unit 29 outputs 1-bit phase code data which is 0 for the type I and 1 for the type II, and is sent to the encoding unit 21. Note that the operation of the encoding unit 21 and thereafter is the same as described above, and thus the description thereof is omitted.

Next, FIG. 7 shows a schematic configuration of a synthesis side (decoding side) for synthesizing an audio signal based on the above-mentioned respective data obtained by being transmitted (or recorded / reproduced).
Will be described with reference to.

In FIG. 7, a data signal at the input terminal 31 is substantially equal to the data signal taken out from the output terminal 23 on the encoder side shown in FIG. 1 (ignoring signal deterioration due to transmission and recording / reproduction). Is supplied. The data from the input terminal 31 is sent to the frame deinterleave unit 32, subjected to deinterleave processing which is the reverse processing of the interleave processing of FIG. 1 above, and subjected to CRC and convolutional coding on the main portion (encoder side). In the portion, generally, the data portion having high importance) is subjected to decoding processing by the decoding unit 33, and the remaining portion (data of low importance not subjected to the encoding processing) is directly subjected to bad frame mask processing in the bad frame mask processing unit 34. To the section 34. In the decoding unit 33, for example, so-called Viterbi decoding processing or error detection processing using a CRC check code is performed. Bad frame mask processing unit 34
Performs processing such as obtaining a parameter of a frame with many errors by interpolation, and also performs the above pitch data, voiced sound /
Unvoiced sound (V / UV) data and vector-quantized amplitude data are separated and extracted. Further, phase code data indicating the presence of a peak (intermediate peak) at the intermediate pitch position on the signal waveform or the type I or II is extracted from the bad frame mask processing unit 34.

The vector-quantized amplitude data from the bad frame mask processing unit 34 is sent to the inverse vector quantization unit 35 and inversely quantized, and the data number inverse conversion unit 3
It is sent to 6 and is inversely transformed. This data number inverse conversion unit 36
Then, the inverse conversion, which is in contrast to the above-described data number conversion unit 19 of FIG.
And the unvoiced sound synthesizer 38. The pitch data from the mask processing unit 34 is sent to the voiced sound synthesis unit 37 and the unvoiced sound synthesis unit 38. The V / UV discrimination data from the mask processing section 34 is also sent to the voiced sound synthesis section 37 and the unvoiced sound synthesis section 38. Furthermore, the voiced sound synthesis unit 37
The phase code data from the mask processing unit 34 is also sent.

In the voiced sound synthesizer 37, for example, cosine
The voiced sound waveform on the time axis is synthesized by wave synthesis, and the unvoiced sound synthesis unit 38 synthesizes the unvoiced sound waveform on the time axis by filtering white noise, for example, with a bandpass filter, and synthesizes each voiced sound waveform and unvoiced sound. The waveform and the waveform are added and synthesized by the adder 41 and output from the output terminal 42. In this case, the amplitude data, the pitch data, and the V / UV discrimination data are updated and given for each frame (L sample, for example, 160 samples) at the time of the analysis, but the continuity between the frames is improved (smoothed). To set each value of the above amplitude data and pitch data to 1
For example, each data value at the center position in the frame,
Each data value up to the center position of the next frame (one frame at the time of composition) is obtained by interpolation. That is, in one frame at the time of composition (for example, from the center of the above analysis frame to the center of the next analysis frame), each data value at the leading end sample point and each data value at the end (leading edge of the next composition frame) sample point Are given, and each data value between these sample points is obtained by interpolation.

Further, according to the V / UV discrimination data, the voiced sound (V) region and the unvoiced sound (U
V) area, and according to this division,
V / UV discrimination data for each band can be obtained.
Here, a fixed number (for example, 12
When the number of bands is reduced (degenerated), the number of bands is solved (restored) to obtain a variable number of bands at intervals according to the original pitch.

The synthesis processing in the voiced sound synthesis unit 37 will be described in detail below. First, a case will be described in which the phase code data is 0, that is, the type I audio signal without the intermediate peak (or small) is synthesized.

The voiced sound corresponding to the above-mentioned one composite frame (L samples, for example, 160 samples) on the time axis in the m-th band (band of the m-th harmonic) determined to be V (voiced sound) is V _m (n ), V _m (n) = A _m (n) cos (θ _m (n)) 0 ≦ n <L (9) by using the time index (sample number) n in this combined frame ) It can be expressed as. The final voiced sound V (n) is synthesized by adding (ΣV _m (n)) the voiced sounds of all bands that are determined to be V (voiced sound) of all bands.

A _m (n) in the equation (9) is the amplitude of the m-th harmonic wave interpolated from the beginning to the end of the composite frame. The simplest way is to linearly interpolate the value of the m-th harmonic of the amplitude data updated in frame units.
That is, the m-th frame at the tip (n = 0) of the composite frame
The amplitude value of the harmonic is A _0m , the end of the composite frame (n =
L: the amplitude value of the m-th harmonic at the next composite frame) is A _Lm, and the following equation is used: A _m (n) = (Ln) A _0m / L + nA _Lm / L (10) It suffices to calculate A _m (n).

Next, the phase θ _m (n) in the above equation (9) is calculated by θ _m (n) = mω _O1 n + n ² m (ω _L1 −ω ₀₁ ) / 2L + φ _{0 m} + Δω n (11) You can ask. In the equation (11), φ _0m represents the phase of the m-th harmonic (frame initial phase) at the tip (n = 0) of the composite frame, and ω ₀₁ is at the tip of the composite frame (n = 0). The fundamental angular frequency, ω _L1, indicates the fundamental angular frequency at the end of the composite frame (n = L: leading end of the next composite frame). Δω in the above equation (11) is
Set a minimum Δω such that the phase φ _{Lm at} n = L is equal to θ _m (L).

Hereinafter, how to obtain the amplitude A _m (n) and the phase θ _m (n) according to the V / UV discrimination result when n = 0 and n = L in an arbitrary m-th band will be described. To do.

When the m-th band is V (voiced sound) for both n = 0 and n = L, the amplitude A _m (n) is the amplitude value transmitted by the above equation (10). The amplitude A _m (n) may be calculated by linearly interpolating A _0m and A _Lm . Phase θ _m (n)
From n 0, θ _m (0) = φ _{0 m} to n = L, θ _m (L) is φ
Set Δω to be _Lm .

Next, when n = 0, V (voiced sound) and n =
Amplitude A _m (n) when UV (unvoiced sound) when L
Is linearly interpolated so that 0 A _m (L) from the transmission amplitude value A _{0 m} of A _m (0). The transmission amplitude value A _{Lm when} n = L is the amplitude value of unvoiced sound and is used in unvoiced sound synthesis described later. The phase θ _m (n) is θ _m (0) = φ _{0 m} , and Δω =
Set to 0.

Further, when n = 0, UV (unvoiced sound)
When n = L, when V (voiced sound) is used, the amplitude A _m
(n) is linearly interpolated so that the amplitude A _m (0) at n = 0 is 0 and the transmitted amplitude value A _Lm is obtained at n = L. Phase θ
_{For m} (n), using the phase value φ _Lm at the end of the frame as the phase θ _m (0) at n = 0, θ _m (0) = φ _Lm −m (ω _O1 + ω _L1 ) L / 2 ... (12) and Δω = 0.

When both n = 0 and n = L are V (voiced sound), Δω is set so that θ _m (L) becomes φ _Lm.
A method of setting will be described. In the above formula (11), n
= L, then θ _m (L) = mω _O1 L + L ² m (ω _L1 − ω ₀₁ ) / 2L + φ _0m + ΔωL = m (ω _O1 + ω _L1 ) L / 2 + φ _0m + ΔωL = φ _Lm . Then, Δω becomes Δω = (mod2π ((φ _Lm −φ _0m ) −mL (ω _O1 + ω _L1 ) / 2) / L (13). In this equation (13), mod 2π (x) Is the principal value of x
It is a function that returns a value between π and + π. For example, x = 1.3
mod2 π (x) = -0.7π when π, mod2 when x = 2.3π
When π (x) = 0.3π and x = -1.3π, mod2π (x) = 0.7
π, and so on.

In the above voiced sound synthesis, when there is no peak (intermediate peak) in the vicinity of the intermediate position of the peak (pitch peak) of the pitch period of the signal waveform, or if it is small,
That is, this is an example of the case of type I (phase code is 0), but in the case where the intermediate peak exists, that is, in the case of type II (phase code is 1), the phase value at the end of the frame (block) Processing is performed to shift φ _Lm .

That is, a specific example of this phase adjustment operation or phase operation processing will be described with reference to FIG.

In the first step 51 of this processing, the transmitted phase code is discriminated and 1 (type I
I, with intermediate peak), proceed to step 52, 0
If it is (type I), the process proceeds to step 55. In step 52, it is determined whether the current phase flag (contents of the phase code sent in the previous frame) is 0 or 1, and when it is 0, the process proceeds to step 53, and when it is 1, the process returns to step 51. In step 53, the phase adjustment operation is performed, and the process proceeds to the next step 54 to set the phase flag to 1. That is, when the phase code changes from 0 to 1, the phase adjustment operation is performed in step 53.

The phase adjusting operation includes an operation of shifting the frame (block) end phase with respect to the reference phase, and more specifically, the phase value φ _{Lm of the} frame end.
Is an odd harmonic (m = 1, 3, 5, ...) Only π / 4
It just shifts. φ _Lm = φ _Lm + π / 4 (18) where m = 1, 3, 5, ...

As described above, by advancing the odd harmonics by π / 4, it is possible to perform signal synthesis that effectively adds an intermediate peak between pitch peaks.

In step 55, the phase flag is discriminated. When it is 0, the process returns to step 51, and when it is 1, the process proceeds to step 56. In step 56, the original normal speech synthesis process, that is, the method using the zero phase or the minimum phase transition is restored. Specifically, (1
_.Phi.Lm = _.phi.Lm- .pi. / 4 (19), where m = 1, 3, 5, ...

After the phase restoring operation is performed in step 56, the process proceeds to step 57 and the phase flag is set to 0. In this way, when the phase code changes from 1 to 0, the phase restoring process for returning the phase to the original state is performed in step 56.

The unvoiced sound synthesizing process in the unvoiced sound synthesizing section 38 will be described below. The white noise signal waveform on the time axis from the white noise generation unit 43 is sent to the windowing processing unit 44, and windowed by a suitable window function (for example, Hamming window) with a predetermined length (for example, 256 samples), STF
A power spectrum on the frequency axis of white noise is obtained by performing STFT (Short Term Fourier Transform) processing by the T processing unit 45. This STFT processing unit 4
The power spectrum from No. 5 is sent to the band amplitude processing unit 46, the above UV (unvoiced sound) band is multiplied by the above amplitude | A _m | _UV, and the amplitude of the other V (voiced sound) band is set to 0. To The band amplitude processing section 46 is supplied with the amplitude data, the pitch data, and the V / UV discrimination data.

The output from the band amplitude processing unit 46 is IS
The signal is sent to the TFT processing unit 47, and the phase is converted into a signal on the time axis by performing inverse STFT processing using the phase of the original white noise. The output from the ISTFT processing unit 47 is sent to the overlap adding unit 48, and the overlap and addition are repeated while appropriate weighting (so that the original continuous noise waveform can be restored) on the time axis,
Synthesize continuous time axis waveforms. The output signal from the overlap adder 48 is sent to the adder 41.

As described above, the signals of the voiced sound portion and the unvoiced sound portion which have been synthesized by the respective synthesis units 37 and 38 and have been returned to the time axis are added by the addition unit 41 at an appropriate fixed mixing ratio, The reproduced audio signal is taken out from the output terminal 42.

The above is the embodiment in which the phase information is 1 bit, but the amplitude information (information of the peak intensity) may be sent with 2 bits or more. In this case, the phase information is represented according to the ratio e ₂ / e ₁ of the total sum value e ₂ of the intermediate peaks to the total sum value e ₁ of the pitch peaks described above, and the phase of the odd harmonics on the synthesis side (decoding side). The strength of the intermediate peak can be made variable by adjusting the strength of.

An embodiment using 2 bits as the phase information (phase code data) will be described below.

As a specific example of the 2-bit phase code PC, the phase code data is determined according to the values of α and β under the following conditions. That is, (e ₂ / e ₁ ≧ α) and (e ₂ / e ₁ <β) and (e ₂
> E ₃ × 1.3), the phase code PC for the values of α and β and the odd harmonics (m = 1, 3, 5, ...) Of the frame end phase value φ _{Lm on} the combining side Phase shift amount δ
When α = 0.25, β = 0.5, PC = 1, δ = π / 8 α = 0.5, β = 0.75, PC = 2, δ = π / 4 α = 0.75, β = 1.5, PC = 3 and δ = 3π / 8. When these conditions are not met, the phase code PC = 0 and the frame end phase is not shifted (or δ = 0).

On the voice synthesizing side (decoding side) where such a 2-bit phase code PC is transmitted, a frame end phase φ _Lm (where m = 1, 3, 5, ...
The phase error (phase adjustment value) ψ is added to the odd harmonics of. This phase difference ψ can be obtained by the configuration as shown in FIG. 9, for example.
In FIG. 9, the phase code PC is sent to the adder 62 via the input terminal 61, and the addition output is sent to the delay element (for one frame delay) 63, and becomes the phase flag PF to the adder 62. Sent as a subtraction input. Further, the output from the delay element 63 is multiplied by π / 8 by the π / 8 multiplier 64, and becomes the phase error component ψ and is taken out from the output terminal 65. Therefore, the phase error component (phase adjustment value) ψ is obtained as ψ = (PC-PF) × (π / 8), and this phase error component is _{calculated as} the frame termination phase φ _Lm (m = 1) of the odd harmonics. 3, 5, ...), and the phase adjustment operation or the phase operation processing is performed as φ _Lm = φ _Lm + φ.

The present invention is not limited to the above-mentioned embodiment, and for example, regarding the configuration of the voice analysis side (encoding side) of FIG. 1 and the configuration of the voice synthesis side (decoding side) of FIG. Although each unit is described as hardware, it can be realized by a software program using a so-called DSP (digital signal processor) or the like. Further, it is sufficient to collectively (degenerate) a band for each harmonic (harmonic) into a certain number of bands, and the number of degenerate bands is not limited to 12 bands. Further, the application of the present invention is not limited to the above-mentioned multi-band excitation speech analysis / synthesis method, and can be easily applied to various speech analysis / synthesis methods using sine wave synthesis. It can be applied not only to transmission and recording / reproduction, but also to various applications such as pitch conversion, speed conversion, noise suppression, and the like.

[0090]

As is apparent from the above description, according to the audio signal encoding method of the present invention, the input audio signal is divided into blocks on the time axis to detect the basic pitch period. , A pitch peak position which is a peak position of the basic pitch period including the maximum peak position in the block is detected, and an intermediate position in the vicinity of an intermediate position ½ pitch period away from the pitch peak position in the block is detected. Based on the peak and the ¼ pitch peak near the intermediate position between each pitch peak position and the position of the intermediate peak,
Since it is determined whether or not the intermediate peak has a certain size or more and the phase information is obtained and transmitted, the reproduction sound quality on the combining side can be improved.

Further, according to the audio signal decoding method of the present invention, in the audio signal decoding method of synthesizing an audio signal using a periodic fundamental wave and its harmonic components, Since the end phase of the block is changed according to the phase information obtained according to the existence of the intermediate peak near the intermediate position of the peak, the voice in which the peak also occurs at the intermediate position of the peak of the pitch period Even in the case of a signal, synthesis processing is performed to reproduce this intermediate peak by the phase information, and it is possible to suppress annoying sensation that has occurred in conventional synthesis processing and obtain a good reproduced sound (synthesized sound). .

Furthermore, if the phase information is 2 bits or more,
By variably controlling the intensity of the intermediate peak on the synthesis side,
It is possible to reproduce fine intermediate peaks, and to improve the quality of the reproduced sound.

[Brief description of drawings]

FIG. 1 is a functional block diagram showing a schematic configuration of an analysis side (encoding side) of a speech signal analysis / synthesis coding apparatus as a specific example of an apparatus to which a speech signal coding method according to the present invention is applied. .

FIG. 2 is a diagram for explaining a windowing process.

FIG. 3 is a diagram for explaining a relationship between windowing processing and a window function.

FIG. 4 is a diagram showing time axis data as an object of orthogonal transform (FFT) processing.

FIG. 5 is a diagram showing spectrum data on a frequency axis, a spectrum envelope (envelope), and a power spectrum of an excitation signal.

FIG. 6 is a diagram for explaining an intermediate peak in the vicinity of an intermediate position of a peak of a pitch period in an input audio signal.

FIG. 7 is a functional block diagram showing a schematic configuration of a synthesis side (decoding side) of a speech signal analysis / synthesis coding apparatus as a specific example of an apparatus to which a speech signal decoding method according to the present invention is applied. .

FIG. 8 is a flowchart for explaining a phase adjustment operation when synthesizing an audio signal based on phase code information.

FIG. 9 is a diagram for explaining a phase adjustment operation when synthesizing an audio signal based on 2-bit phase code information.

[Explanation of symbols]

13 Pitch extraction unit 14 Windowing processing unit 15 Orthogonal transform (FFT) unit 16 High precision (fine) pitch search unit 17 Voiced sound / unvoiced sound (V / UV) discrimination unit 18V: Voiced sound amplitude evaluation unit 18U: Unvoiced sound amplitude evaluation unit 19: Data number conversion (data rate conversion) unit 20. Vector quantizer 26. Peak search unit 27. Pitch peak position detector 28. Local maximum detector 29. Sorting unit 37. .... Voiced sound synthesizer 38 ... Unvoiced sound synthesizer

Claims (9)

(57) [Claims] 1. A step of partitioning an input audio signal in block units on a time axis, a step of detecting a basic pitch cycle in the partitioned block, and a basic pitch cycle including a maximum peak position in the block. The step of detecting the pitch peak position which is the peak position of
A step of detecting an intermediate peak in the vicinity of an intermediate position separated by a pitch cycle and obtaining a statistic e ₂ of the value of the intermediate peak; and a step of detecting the intermediate value of each pitch peak position and the intermediate peak position in the block. The method includes a step of detecting a quarter pitch peak and obtaining a statistic e ₃ of the value of the quarter pitch peak, and a step of obtaining phase information based on the magnitude relation between the above statistics e ₂ and e _3. An audio signal encoding method characterized by the above. 2. The audio signal encoding method according to claim 1, wherein the phase information is 1 bit. 3. The method of encoding a voice signal according to claim 1, wherein phase information of 2 bits or more is calculated in consideration of peak intensity based on the magnitude relation between the statistics e ₂ and e ₃ . 4. The audio signal encoding method according to claim 1, wherein the phase information is not transmitted when the statistics e ₂ and e ₃ are small. 5. The statistic e ₁ of the values of the plurality of pitch peaks in the block is detected, and the statistic e ₂ and e ₃ are detected.
Ratio _{e 2 / e 1, e 3} / e coding method according to claim 1, wherein the sound signal and obtains the phase information corresponding to the magnitude of the _first. 6. A method for decoding an audio signal, which synthesizes an audio signal using a periodic fundamental wave and its harmonic components, the method being determined according to the presence of an intermediate peak near the intermediate position of the peak of the fundamental pitch period. A method for decoding an audio signal, characterized in that the terminal phase of the block is changed in accordance with the supplied phase information. 7. The method according to claim 6, wherein, when synthesizing the fundamental wave and its harmonic component, only the termination phase of the odd-order harmonic wave is shifted from the reference phase in accordance with the phase information. Method for decoding the audio signal of. 8. The voice according to claim 7, wherein 1 bit is used as the phase information, and the termination phase of the odd-order harmonic is shifted by π / 4 with respect to the reference phase according to the phase information. Signal decoding method. 9. The method according to claim 7, wherein 2 bits are used as the phase information, and the termination phase of the odd-order harmonic is shifted by an integral multiple of π / 8 with respect to the reference phase according to the phase information. A method for decoding an audio signal as described.