JPH07221649A - INFORMATION ENCODING METHOD AND DEVICE, INFORMATION DECODING METHOD AND DEVICE, INFORMATION RECORDING MEDIUM, AND INFORMATION TRANSMISSION METHOD - Google Patents

Abstract

(57) [Abstract] [Purpose] By performing gain control according to the degree of amplitude change in the attack portion, more efficient encoding and decoding with higher sound quality are performed, and pre-echo generation is prevented. [Structure] (A) A gain control function G1 having a relatively small gain control amount shown in (B) is applied to a signal waveform SW1 having a level immediately above the attack part above a certain level, and the level immediately before the attack part is extremely high. A gain control function G2 having a relatively large gain control amount is applied to the low signal waveform SW2 to perform gain control and gain control correction processing. In this way, by changing the gain control amount according to the degree of amplitude change in the attack portion of the signal waveform, the occurrence of pre-echo and the decrease in efficiency due to the energy diffusion in the frequency domain are avoided.

Description

Detailed Description of the Invention

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an information coding method for coding input digital data by so-called high efficiency coding, and transmitting, recording, reproducing and decoding the input digital data to obtain a reproduced signal. The present invention relates to an apparatus, an information decoding method and apparatus, an information recording medium, and an information transmission method.

[0002]

2. Description of the Related Art Conventionally, various techniques for high-efficiency coding of signals such as audio and voice have been known. For example, audio signals on the time axis are not divided into blocks and divided into a plurality of frequency bands. Band division coding (sub-band coding: SBC), which is a non-blocking frequency band division method that performs encoding by using a plurality of frequency bands by converting a signal on the time axis into a signal on the frequency axis (spectral conversion). There is a blocking frequency band division method in which each frequency band is divided into, and so-called transform coding and the like. Further, a method of high efficiency coding in which the above band division coding and transform coding are combined is also considered, and in this case, for example, after performing band division by the band division coding, A signal for each band is spectrum-converted into a signal on the frequency axis, and each spectrum-converted band is encoded.

Here, as a band division filter used in the above band division coding, for example, QM
There is an F filter, and this QMF filter is based on the document "Digital Coding of Speech in.
Subbands "(Digital coding of speech in subbands,
RE Crochiere, Bell Syst.Tech. J., Vol.55, No.819
76).

In addition, the document "Polyphase Quadrature Filters-New Band Division Coding Technology" (Po
lyphase Quadrature filters -A new subband coding t
echnique, Joseph H. Rothweiler, ICASSP 83 BOSTON)
Describes an equal bandwidth filter partitioning technique.

Further, as the above-mentioned spectrum conversion,
For example, an input audio signal is divided into blocks in a predetermined unit time (frame), and a discrete Fourier transform (DFT), a cosine transform (DCT), a modified DCT transform (MDCT), etc. are performed for each block to set a time axis as a frequency axis. There is a spectral transformation that transforms. Regarding the MDCT, refer to the document “Time Domain Aliasing.
Subband / Transform Coding with Cancellation-Based Filter Bank Design "
Using Filter Bank Designs Based on Time Domain Ali
asing Cancellation, JPPrincen, ABBradley, Uni
v. of Surrey, Royal Melbourne Inst.ofTech. ICASSP
1987).

As described above, by quantizing the signal divided into each band by the filter and the spectrum conversion, the band in which the quantization noise is generated can be controlled,
By utilizing the properties such as the masking effect, it is possible to perform more efficient audio coding. Further, if the normalization is performed for each band, for example, with the maximum value of the absolute value of the signal component in that band before the quantization is performed here, more efficient encoding can be performed.

Here, as the frequency division width in the case of quantizing each frequency component divided into frequency bands, for example, a bandwidth considering human auditory characteristics is often used. That is, an audio signal may be divided into a plurality of bands (for example, 25 bands) with a bandwidth generally called a critical band in which the bandwidth increases as the frequency band increases. Also, when encoding the data for each band at this time, a predetermined bit allocation for each band, or
Coding by adaptive bit allocation (bit allocation) is performed for each band. For example, when the coefficient data obtained by the MDCT processing is encoded by the bit allocation, the MD of each block is
The MDCT coefficient data for each band obtained by the CT process is encoded with the adaptive allocation bit number. The following two methods are known as bit allocation methods.

For example, the document "Adaptive Transform Coding of Speech Signals,"
IEEE Transactions of Accoustics, Speech, and Signa
l Processing, vol.ASSP-25, No.4, August 1977), bit allocation is performed based on the signal size of each band. In this method, the quantization noise spectrum becomes flat and the noise energy becomes the minimum, but the actual noise feeling is not optimal because the masking effect is not used auditorily.

In addition, for example, the document "Critical band encoder-
Digital Encoding of Perceptual Requirements of the Auditory System "(The critical band coder --digital encodi
ng of the perceptual requirements of the auditory
system, MAKransner MIT, ICASSP 1980) describes a method of performing fixed bit allocation by using the auditory masking to obtain the required signal-to-noise ratio for each band. However, in this method, even when the characteristic is measured with a sine wave input, the characteristic value is not so good because the bit allocation is fixed.

In order to solve these problems, all bits that can be used for bit allocation have a fixed bit allocation pattern predetermined for each small block and a bit allocation depending on the signal size of each block. A high-efficiency coding apparatus has been proposed which is used in a divided manner, and the division ratio depends on a signal related to an input signal, and the smoother the spectrum of the signal, the larger the division ratio into the fixed bit allocation pattern. ing.

According to this method, when energy is concentrated on a specific spectrum such as a sine wave input, a large number of bits are allocated to a block including the spectrum, thereby significantly improving the overall signal-to-noise characteristic. can do. In general, human hearing is extremely sensitive to a signal having a steep spectrum component. Therefore, improving the signal-to-noise characteristic by using such a method does not only improve the numerical value in measurement. Not
It is effective for improving the sound quality in terms of hearing.

Many other methods have been proposed for the bit allocation method. Further, if the model relating to hearing is further refined and the performance of the coding apparatus is improved, a more efficient coding can be achieved auditorily. It will be possible.

By the way, when the above-described DFT or DCT is used as a method for converting a waveform signal into a spectrum, M pieces of independent real number data are obtained by performing conversion with a time block consisting of M pieces of samples. In order to reduce the connection distortion between time blocks, normally, since the overlapped by sample blocks and _one M each neighboring, on average, in DFT or DCT (M-M ₁₎ with respect to samples M pieces of real number data are quantized and coded.

On the other hand, when the above-mentioned MDCT is used as a method for converting into a spectrum, from 2M samples which are overlapped by N times on both adjacent times,
Since M independent real number data can be obtained, the average is MD
In CT, M pieces of real number data are quantized and coded for M pieces of samples. In the decoding device, the waveform signal can be reconstructed by adding the waveform elements obtained by performing the inverse transformation in each block from the code obtained by using the MDCT while interfering with each other. it can.

Generally, by lengthening the time block for conversion, the frequency resolution of the spectrum is increased and the energy is concentrated on a specific spectral component. Therefore, DFT and DFT can be performed by using the MDCT in which the number of spectral signals obtained does not increase with respect to the number of original time samples, by performing the conversion with a long block length by overlapping the adjacent blocks by half. It is possible to perform encoding more efficiently than when using DCT. In addition, it is also possible to reduce the inter-block distortion of the waveform signal by allowing adjacent blocks to have a sufficiently long overlap.

As described above, when a method of decomposing a signal into frequency components and quantizing and coding the frequency components is used, a waveform signal obtained by decoding and combining the frequency components is also quantized. Although noise is generated, if the original signal component changes abruptly, the quantization noise on the waveform signal becomes large even in the part where the original signal waveform is not large, and this quantization noise is Since it is not hidden by masking, it becomes a hearing problem. The quantization noise generated in this way at the attack portion where the sound abruptly increases is called a pre-echo.

In particular, when the spectrum conversion is used to decompose into a large number of frequency components, the time resolution becomes poor,
Pre-echo occurs over a long period.

Here, the operation principle of generation of pre-echo when spectrum conversion is used at the time of band division is as follows.
This will be described with reference to FIG.

When quantization noise QN is added to a spectrum signal obtained by subjecting the input waveform signal SW to forward spectrum conversion using the window function or window function shown in FIG. 7A, this quantization noise is added. When the spectrum signal is subjected to the inverse spectrum conversion and is returned to the waveform signal on the time axis again, the quantization noise spreads over the entire conversion block. Here, when the input signal waveform suddenly increases in the middle of the conversion block as shown in (B), the quantization noise QN is reduced to the signal waveform S in the section where the original signal waveform is small.
Since it becomes large with respect to W, simultaneous masking does not work, and it becomes an auditory obstacle as a pre-echo.

Here, if the conversion length of the spectrum conversion is shortened, the generation period of the above-mentioned quantization noise is also shortened, but if so, the frequency resolution is deteriorated and the coding efficiency in the quasi-stationary part is deteriorated. As a means for solving such a problem, a method has been proposed in which the conversion length is shortened at the expense of the frequency resolution only in the portion where the signal waveform changes abruptly.

FIG. 8 is a view for explaining an example of a conventional technique devised to reduce the hearing loss caused by such pre-echo. In general, for a quasi-stationary signal waveform, a longer transform block length concentrates energy on a specific spectral coefficient, resulting in higher coding efficiency, but the sound volume changes rapidly. In the part, if the conversion block length is long, the above-mentioned pre-echo becomes a problem.

Therefore, a portion where the loudness of the sound changes rapidly, for example, the input signal waveform SW as shown in FIG. 8B.
In the place where the amplitude of becomes sharply large, as shown in FIG. 8A, a short conversion window function or a short conversion window function for shortening the conversion block length is applied,
As a result, if the generation period of the pre-echo is sufficiently shortened, so-called reverse masking by the original signal is effective, and the hearing loss is eliminated. In the method of FIG. 8, this is used to selectively switch the conversion block length according to the property of each part of the signal waveform.

When this method is used, sufficient frequency resolution is ensured in the quasi-stationary portion, and the pre-echo in the attack portion has a sufficiently short generation period and is concealed by so-called reverse masking, so that efficient coding is performed. It will be possible.

However, in the method of making the conversion length variable as described above, it is necessary to provide the encoding device and the encoding device with the conversion means corresponding to the conversion of different lengths. Furthermore, in this method, since the number of spectral components obtained by the conversion is proportional to the length of the conversion length, the frequency band corresponding to each spectral component also differs depending on the conversion length, and a plurality of spectra are collected, for example, for each critical bandwidth. However, the number of spectra included in each critical band also differs, and the encoding and decoding processes become complicated. As described above, the method of making the conversion length variable has a drawback that both the encoding device and the decoding device are complicated.

By the way, as a method for solving the above-mentioned problem of pre-echo while keeping the conversion block length constant, Japanese Patent Laid-Open No. 132322/1993 performs adaptive gain control on an input waveform signal. After that, DFT and D
A method of converting a spectrum signal into a spectrum signal by using CT and performing encoding is described. Here, the gain control is to increase the gain (amplify the amplitude) at a low power level.

In this method, the encoding device performs gain control in which the gain is drastically reduced in the attack portion before conversion into a spectrum signal, and the gain is increased again in the portions other than the attack portion according to the attenuation. The gain control is carried out so that the decoding device outputs the signal obtained by performing the inverse gain control for correcting the gain control on the signal waveform obtained by the inverse spectrum conversion. In this way, the quantization noise in the small amplitude signal portion where the masking level becomes low is suppressed. Moreover, since the conversion length can be made constant at all times, the configurations of the encoding device and the decoding device can be simplified.

However, with this method, it is necessary to perform gain control even when the signal is attenuated. In general, performing gain control distorts the original signal waveform, so
When converted to a spectrum, the energy distribution is dispersed, making it difficult to perform efficient coding. Especially when the signal is attenuated, the forward masking that masks the sound generated after the previous sound works effectively, so it is important to lower the noise level itself rather than temporally controlling the generation of quantization noise. Is. Further, it is not preferable to always perform the gain control process from the viewpoint of the amount of calculation processing.

Another method for preventing pre-echo while keeping the conversion block length constant is, for example, Japanese Patent Laid-Open No. 61-2.
Techniques such as those disclosed in JP-A-01526 and JP-A-63-7023 are known. In these publications, in an encoding device, an input signal waveform is cut out for each time block, a window is applied, an attack portion is detected, and a small amplitude waveform immediately before the attack portion is amplified, and then DFT or DC is applied.
A spectrum signal is converted into a spectrum signal by using T and encoded, and in the decoding device, an inverse DFT is applied to the restored spectrum signal.
(Inverse DFT: IDFT) and inverse DCT (Inverse DFT)
There has been proposed a method for preventing pre-echo by performing an inverse transformation such as CT: IDCT) and then performing a process of correcting that the signal immediately before the attack portion is amplified by the encoding device. Also in this case, the conversion length can be made constant and the configurations of the encoding device and the decoding device can be simplified.

FIG. 9 shows the above-mentioned Japanese Patent Laid-Open No. 61-2015.
No. 26 and Japanese Patent Laid-Open No. 63-7023 describe the principle of operation for encoding / decoding using the windowing processing technique. FIGS. 10 and 11 use this technique. 9 shows a processing flow of the encoding device and the decoding device.

The input terminal 400 shown in FIG.
(A) of FIG. 9 is input, and in the window circuit 401, the time waveform signal is cut out by setting a time window that is sequentially continuous in time and overlaps with each other. The window function shown in B) (the characteristic curve referred to in Japanese Patent Laid-Open No. 61-201526) is multiplied. The attack portion detection circuit 402 detects a portion (attack portion) where the amplitude of the input signal suddenly increases. In the gain control circuit 403, if an attack portion is detected, processing is performed so as to amplify the minute amplitude portion, and if no attack portion is detected, amplification processing is not performed. The output from the gain control circuit 403 is sent to the forward spectrum conversion circuit 404 and DFT, D
It is converted into a spectrum signal by CT or the like. The spectrum signal thus obtained is normalized and quantized by the normalization / quantization circuit 405, coded by the coding circuit 406, and taken out from the output terminal 407 as a code string.

Further, in the decoding apparatus shown in FIG. 11, for the code string signal supplied to the input terminal 410,
The decoding circuit 411 performs the inverse decoding of the encoding in the encoding circuit 406, and the inverse normalization / inverse quantization circuit 412.
Sent to. The output from the inverse normalization / inverse quantization circuit 412 is output from the inverse spectrum conversion circuit 413 to IDFT or IDCT.
After being inversely transformed into the time domain by means such as the above, it is sent to the gain control correction circuit 414, and processing for correcting the gain control processing performed by the encoding device is performed. The output from the gain control correction circuit 414 is sent to the adjacent block synthesis circuit 415 to be synthesized with the adjacent block and is taken out via the output terminal 416.

In this method, after the window function is applied as described above, the attack portion detection processing is performed on the deformed waveform signal, so that the large amplitude portion is also relaxed at both ends of the block. For example, as shown in FIG. 9, an attack part may not be detected in the block BL1 and an attack part may be detected only in the next block BL2. However, the DFT or DCT is used as the spectrum conversion. In this case, if the inverse spectrum transform is applied to the spectrum obtained by applying the forward spectrum transform, the original time series block is completely restored.
If the decoding device performs a gain control correction process for each block, no problem occurs.

[0033]

However, in the method described in the embodiment of the method utilizing the gain control, the gain control amount in the attack portion is fixed and is determined depending on whether or not the attack portion is detected. It was decided whether or not to perform gain control of the value of. In the pre-echo, the quantization noise generated in the frequency domain is converted into the time domain.Therefore, if the compression rate of coding increases and the quantization noise in the frequency domain increases, the sound quality deterioration due to the pre-echo increases. However, performing the gain control more than necessary spreads the energy distribution in the frequency domain and is not desirable for efficient compression. Therefore, in the conventional method of selectively selecting whether or not to perform the gain control of the predetermined value depending on the presence or absence of the detection of the attack portion, the deterioration of the sound quality is prevented especially when the compression rate is high. Was difficult.

The present invention has been made in view of the above circumstances, and enables gain control in accordance with the degree of amplitude change of the attack portion, which enables more efficient encoding and decoding with higher sound quality. It can be recorded or transmitted, the configuration is simple,
An object of the present invention is to provide an information encoding method and device, an information decoding method and device, an information recording medium, and an information transmitting method, which can effectively prevent pre-echo.

[0035]

In order to solve the above problems, the present invention is effective even when the compression ratio is high by changing the gain control amount of the attack portion according to the level change of the waveform signal. The purpose is to prevent pre-echo.

That is, the information coding method according to the present invention is such that a frequency component decomposition process for decomposing an input signal into frequency components, a gain control process for the input waveform signal to the frequency component decomposition process, and the frequency component decomposition process. The output information and the control information of the gain control described above are encoded, and the gain control amount of the gain control process in the portion where the waveform signal suddenly increases is selectively determined from a plurality of sizes. It is a thing.

Further, the information coding apparatus according to the present invention comprises a frequency component decomposition processing means for decomposing an input signal into frequency components, and a gain control processing for performing gain control processing of the input waveform signal to the frequency component decomposition processing means. And a coding means for coding the output information from the frequency component decomposition processing means and the control information for the gain control, the gain control of the gain control processing in a portion where the waveform signal suddenly increases. The quantity is selectively determined from a plurality of sizes.

Next, an information decoding method according to the present invention comprises a decoding process of frequency component signals and gain control correction information, a waveform signal combining process, and a gain control correcting process of an output waveform signal of the above waveform signal combining process. And the gain control correction amount of the gain control correction processing in the portion where the waveform signal suddenly increases is selected from a plurality of sizes determined based on the contents of the gain control correction information. That is.

Further, the information decoding apparatus according to the present invention comprises a decoding means for decoding the frequency component signal and the gain control correction information, a waveform signal combining processing means for combining the waveform signals, and the above waveform signal combining processing. A gain control correction processing means for performing a gain control correction processing of an output waveform signal from the means, and the gain control correction amount of the gain control correction processing in a portion where the waveform signal suddenly increases is the content of the gain control correction information. It is to be selected from a plurality of sizes determined based on

Further, the information recording medium according to the present invention has frequency component signal information and gain control correction information recorded therein, and the gain control correction information includes information on the gain control correction amount, and a portion where the waveform signal sharply increases. The gain control correction amount in is selected from a plurality of types.

Furthermore, in the information transmission method according to the present invention, the frequency component signal information and the gain control correction information are transmitted, and the gain control correction information includes the information of the gain control correction amount, and the waveform signal rapidly increases. The gain control correction amount in the part is selected from a plurality of types.

In these inventions, the frequency component decomposition process may include a spectrum conversion process for converting a signal on the time axis into a signal on the frequency axis. Also,
The input signal or the output signal may be an acoustic signal.

[0043]

By using a gain control amount that is selectively determined from a plurality of sizes as the gain control amount in the portion where the waveform signal suddenly increases, it is possible to perform gain control according to the degree of amplitude change in the attack portion. , Which enables more efficient encoding, decoding, recording and transmission with higher sound quality.

[0044]

DETAILED DESCRIPTION OF THE PREFERRED EMBODIMENT A preferred embodiment of the present invention will be described below with reference to the drawings.

FIG. 1 is a block circuit diagram of an embodiment of an encoding device to which the information encoding method of the present invention is applied. In FIG. 1, the audio signal input to the encoding device via the input terminal 100 is the band division circuit 1
The band is divided by 01. This band division circuit 101
As the band dividing means in, the dividing means by the filter such as QMF described above may be used, or the means obtained by grouping the spectrum obtained by the spectrum conversion such as MDCT into each band may be used. Alternatively, a means may be used in which spectrum conversion is performed once on a band that has been divided into several bands by the filter, and the spectrum obtained by this is grouped for each band. Further, the width of each band resulting from this band division may be uniform, or may be non-uniform so as to match the critical bandwidth, for example. In addition, in the example of FIG. 1, although it is divided into four bands, of course, if this number is increased,
Alternatively, it may be reduced.

The signal band-divided by the band-dividing circuit 101 is normalized by the normalizing circuits 111, 112, 113, 114 corresponding to each band for each time block, and here, the normalizing coefficient and It is decomposed into a normalized signal. The respective to-be-normalized signals are quantized by the quantization circuits 121, 122, 123, 124 based on the quantization accuracy information output from the quantization accuracy determination circuit 141.
Are quantized by and converted into a normalized / quantized signal. In FIG. 1, the quantization circuits 121, 122, 12 from the quantization accuracy determination circuit 141 are
Among the quantization precision information to 3,124, the quantization precision information sent to the quantization circuit 122 is via the terminal 152,
The quantization precision information sent to the quantization circuit 123 is sent to the corresponding circuit via the terminal 153, and the quantization precision information sent to the quantization circuit 124 is sent to the corresponding circuit via the terminal 154.

The quantizing circuits 121, 122, 123,
Each of the normalized / quantized signals from 124, the normalization coefficients from the normalization circuits 111, 112, 113, 114, and the quantization precision information from the quantization precision determination circuit 141 is a multiplexer. A code string is sequentially formed by 131, and this code string is output from the terminal 103. This code string is then recorded on a recording medium such as a disc, a tape or a semiconductor, or transmitted from a transmission system.

Here, in the example of FIG. 1, the quantization precision determination circuit 141 calculates the quantization precision based on each signal band-divided by the band division circuit 101, but before the band division. It is also possible to calculate from the signal through the terminal 100 of each normalizing circuit 11
It is also possible to calculate based on the normalization coefficient from 1,112,113,114. Further, the calculation in the quantization precision determination circuit 141 can be performed based on an auditory phenomenon such as a masking effect, and the respective quantization precision information is output via the multiplexer 131 as described above. It is later sent to the decoding device. Therefore, the auditory model used in the decoding device can be arbitrarily set.

On the other hand, FIG. 2 shows a block circuit diagram of an embodiment of a decoding device corresponding to the coding device of FIG. 1 to which the information decoding method of the present invention is applied. In FIG. 2, the code information (the code string) input to the terminal 201 of the decoding apparatus according to the present embodiment is sent to the demultiplexer 202, where the quantization accuracy information for each band and the normalization coefficient are used. And the normalized and quantized signals are separated and restored. The quantization accuracy information for each band, the normalization coefficient, and the signal to be normalized / quantized are the signal component configuration circuits 2 corresponding to the respective bands.
11 to 212, 213 and 214, where signal components are formed for each band. The signal components from the respective signal component configuration circuits 211, 212, 213 and 214 are combined by the band combining circuit 221 into an audio signal and output from the terminal 251.

Next, FIG. 3 is a diagram for explaining the gain control operation during the windowing process when the embodiment of the present invention is applied.

Here, in the method described in the above-mentioned conventional example, the gain control amount for amplifying the part immediately before the attack is fixed. For example, in Japanese Patent Laid-Open No. 63-7026, each block is divided into sub blocks,
Amplitude fluctuation between two sub-blocks has a predetermined limit value (20
In the case where the amplitude exceeds about (dB), a method is disclosed in which the amplitude of the signal before the abrupt change in the amplitude is amplified by the encoding means and is corrected by the decoding means.

However, for example, although the signal waveforms SW1 and SW2 in FIG. 3A both include the attack portion, there is a big difference in the way of changing the amplitude. That is, in the signal waveform SW1, there is a waveform signal of a certain level or more immediately before the attack portion, and the generated pre-echo is masked to some extent by the original waveform signal, although not so much after the attack portion. On the other hand, in the signal waveform SW2, the level of the waveform signal immediately before the attack portion is very low, and the generated pre-echo is hardly masked.

Here, when the detection of the attack portion is identified by only one limit value of the level change and the same gain control / gain control correction is attempted, the signal waveform SW
If the limit value and the gain control amount are set optimally for 1, the pre-echo will be heard for the signal waveform SW2. Further, if the limit value and the gain control amount are set optimally for the signal waveform SW2, the signal waveform S
Unnecessary gain control is performed on W1, which causes energy diffusion in the frequency domain, resulting in a decrease in coding efficiency.

Therefore, in the method of the present invention, this problem is solved by changing the gain control amount according to the degree of amplitude change in the attack portion of the signal waveform.

That is, according to the method of the present invention, as shown in FIG.
In (B), the gain control function G1 having a relatively small gain control amount is applied to perform the gain control and the gain control correction, while the gain control amount is relatively small for the signal waveform SW2. A large gain control function G2 is applied to perform gain control and gain control correction processing.

FIG. 3C shows how each quantization noise is generated when the above processing is performed. As shown in (C) of FIG. 3, the quantization noise before the attack portion of the quantization noise of the signal waveform SW1 has a relatively small noise suppression effect by the gain control correction process, and therefore the signal waveform SW2 The quantization noise is larger than the quantization noise before the attack part, but the energy of the quantization noise is small throughout. On the other hand, the energy of the quantization noise throughout the signal waveform SW2 is relatively large, but the quantization noise before the attack portion is kept sufficiently low. Since the pre-echo is a great obstacle to hearing, it is desirable to suppress it by giving priority to reducing the total noise energy.

Next, FIG. 4 shows an example of the flow of processing for detecting an attack part and generating a gain control function when the embodiment of the present invention is actually applied to signal coding. is there. For example, the encoding method of the present invention can be realized by incorporating this processing into the processing corresponding to the attack detection circuit 402 of the encoding apparatus of FIG.

In FIG. 4, for example, a block having a length of 2M is divided into N sub-blocks, and the maximum amplitude value P [I] in the I-th sub-block is divided into K consecutive K-blocks up to the I-th sub-block. Maximum amplitude value Q in sub-block
Compared with [I], if it is equal to or higher than a predetermined ratio, it is considered that an attack portion is detected. Finally, a gain control function having a smooth transition is constructed to prevent energy diffusion when converted into a spectrum.

That is, in the first step S1 in FIG. 4, I out of the sub-blocks obtained by dividing one block into N parts.
The maximum amplitude value Q [I] from K consecutive sub-blocks up to the # 1 sub-block, that is, from I-K + 1 sub-block to the I-th sub-block, is calculated.
The maximum amplitude value P [I] in the No. subblock is calculated. In the next step S3, I = 0 and step S4
In the above, R as the gain control amount is obtained by the ratio of the maximum amplitude Q [I] of the K sub-blocks up to the number I to the maximum amplitude P [I + 1] of the immediately following sub-block. In the next step S5, T is a predetermined threshold value,
When is larger than T, it is determined that the attack portion is detected, and the process proceeds to step S9. If NO, the process proceeds to step S6, I is incremented, and step S
In step 7, it is determined whether or not I has reached the sub-block number N at the end of the block, and step S4 and subsequent steps are repeated until I = N. When YES is determined in the step S7, L = 0 is set in the step S8, that is, there is no attack, and R = 1 is set, and the process proceeds to the step S10. If YES in step S5, that is, if an attack is found, the process proceeds to step S9, L = I is set, and R is an integer value of the value of R obtained in step S4. That is, the length before the attack part in this block is interpreted as L sub-blocks, and the value of R at this time represents the gain control amount. After finishing the process of step S9,
Go to step S10.

In step S10, the gain control function of the sub-block up to the attack position L is set to R, the rest is set to 1, and finally interpolation processing is performed so as to have a smooth transition part, and then the processing is terminated. is doing. That is, in this step S10, the gain control function g (n) is constructed based on the values of L and R, but the function value is smoothly interpolated in the sub-block immediately before the attack part. This is to prevent diffusion of the energy distribution when converted into the frequency domain and enable efficient coding.

As described above, by changing the gain control amount of the attack portion according to the signal level, there is an advantage that the pre-echo can be effectively prevented even when the compression rate is high.

In this example, the gain control is amplified only immediately before the attack portion, but this is due to the use of the effect of forward masking, as described above. However, of course, it is also possible to perform gain control so that amplification is performed in the small amplitude portion at the time of attenuation. For example, when the block length of the spectrum conversion is extremely long and the forward masking effect cannot be expected sufficiently, Amplification may be performed in the small amplitude portion at the time of attenuation. Further, the number of attack parts to be detected does not necessarily have to be one for one block.

When a gain control function that changes abruptly in a stepwise manner is used, the energy is diffused when converted into a spectrum, and the coding efficiency decreases. Therefore, it is desirable that the control function has a shape that changes smoothly to some extent even in the attack portion. However, if that section is not sufficiently short, the pre-echo will be heard, so in consideration of human hearing, the gain control function has a transient section of about 1 msec, and within that section, for example, a sine wave shape is smooth. It is desirable to make changes. In case the attack occurs at the beginning of the next block, the detection range of the attack part is expanded to the sub-block at the beginning of the next block so that the gain control function has a smooth transition part and At times, it is possible to satisfy the condition for allowing waveform elements to interfere with each other between adjacent blocks.

As described above, the method or apparatus of the present invention can be applied to an apparatus for processing an audio waveform converted into a digital signal, and the waveform signal once filed can be processed by a computer. It can also be applied when processing by etc. Further, it is of course possible to record or transmit the code thus obtained on a recording medium. Further, the present invention can be applied to the case where the encoding is always performed at a constant bit rate, and the case where the encoding is performed at a bit rate that temporally changes so that the number of allocated bits differs for each block. Is.

In the above description, the case where the waveform signal digitized in the encoding device is converted into the spectrum signal by using the direct spectrum conversion has been described, but of course, it is once divided by using the band division filter. The method of the present invention can also be applied to the case where spectrum conversion is performed for each band using spectrum conversion.

Next, FIG. 5 shows an example of a recording format when recording information encoded by the method of the present invention on a recording medium or a transmission format when transmitting.

In the example shown in FIG. 5, the code of each block is the attack part detection flag, the spectrum signal code, and the attack part position in addition to those depending on the contents of the attack part detection flag. It is constituted by information and gain control correction function generation information including gain control information. For example, the value of L in FIG. 4 may be recorded as the attack position information, and the value of R in FIG. 4 may be recorded as the gain control amount information. In the actual music signal, the proportion of blocks where the attack part where pre-echo is a problem exists is low, so it is efficient to record the attack position information and gain control amount information only in the block where the attack part actually exists. Is good. However, of course, the gain control correction function generation information may be recorded in all the blocks. In this case, in a block in which no attack portion actually exists, for example, L = 0 and R = 1 may be recorded. Good.

Next, FIG. 6 shows an example of processing in which the decoding means generates the gain control correction function h (n) from the recording information shown in FIG.

For example, the process shown in FIG.
The gain control correction function h (n) generated by being incorporated into the process corresponding to the gain control correction circuit 414 of the first decoding device.
The decoding method according to the present invention can be realized by multiplying by the waveform signal element formed by the inverse spectrum conversion circuit 413. Of course, in the block in which the attack part is not detected, the process of actually multiplying by h (n) may be omitted.

In the example of FIG. 6, step S21
When the attack detection flag is detected in step S1, and the flag is 0, that is, when no attack is detected, the process proceeds to step S22, and the gain control correction function h (n) is set to 1
And finish. If the flag is 1, that is, if an attack is detected, the process proceeds to step S23, where L gain control functions g (n) for L sub blocks from the head of this block are set to R, and the above interpolation process is performed to finally obtain A new gain control function g (n). In the next step S24, the reciprocal 1 / g (n) of the gain control function g (n) is calculated to obtain the gain control correction function h (n).

The method of the present invention can of course be applied to, for example, the method described in the above-mentioned JP-A-3-132228.

Further, not only when the waveform signal is decomposed into frequency components by direct spectrum conversion, but also when the waveform signal once band-divided by the band division filter is decomposed into frequency components by spectrum conversion, of course, The method of the invention can be applied. Further, it can be applied to the case where a waveform signal is decomposed into frequency components by a filter. The frequency component referred to in the present invention includes all those obtained by these processes, but when the pre-echo is applied in relation to the frequency component obtained by the process including the spectrum conversion which is a particularly serious problem, the present invention The method is particularly effective.

Furthermore, the method of the present invention can be applied to an apparatus for processing an audio waveform converted into a digital signal, and the waveform signal once filed is processed by a computer or the like. It can also be applied in some cases. Further, it is of course possible to record or transmit the code thus obtained on a recording medium. Further, the method of the present invention can be applied to the case where the encoding is always performed at a constant bit rate, and the case where the encoding is performed at a bit rate that temporally changes so that the number of allocated bits differs for each block. Is possible.

Although the above description has been made on the case where the quantization noise when the acoustic waveform signal is quantized is made inconspicuous, the method of the present invention makes the generation of the quantization noise of other kinds of signals inconspicuous. The above is also effective, and can be applied to an image signal, for example. However, since the pre-echo in the attack portion of the acoustic signal is a great obstacle to hearing, it is very effective to apply the present invention to the acoustic signal. Further, the method of the present invention can of course be applied to multi-channel acoustic signals.

[0075]

As is apparent from the above description, in the information coding method according to the present invention, in the gain control processing of the input waveform signal for the frequency component decomposition processing, the waveform signal becomes a portion where the waveform signal suddenly increases. Since the gain control amount of the gain control process is selectively determined from a plurality of sizes, gain control can be performed according to the degree of amplitude change of the attack part, resulting in more efficient and better sound quality. High coding is possible.

In the information decoding method according to the present invention, when the gain control correction process of the output waveform signal of the waveform signal synthesis process is performed, the gain control of the gain control correction process in the portion where the waveform signal sharply increases. By using a correction amount selected from a plurality of sizes determined based on the content of the gain control correction information, efficient processing can be performed and a high quality signal can be reproduced.

These effects can be similarly applied to the information coding apparatus and the information decoding apparatus.

By applying this to an acoustic signal or an audio signal, it is possible to prevent the occurrence of pre-echo as well as the efficient processing.

Further, by recording or transmitting the signal encoded by such an encoding method or device in a recording medium, efficient recording or transmission is possible.

[Brief description of drawings]

FIG. 1 is a block circuit diagram showing a schematic configuration of an encoding device to which an embodiment of the present invention is applied.

FIG. 2 is a block circuit diagram showing a schematic configuration of a decoding device to which an embodiment of the present invention is applied.

FIG. 3 is a diagram for explaining the operation of gain control during windowing processing in the embodiment of the present invention.

FIG. 4 is a flowchart schematically showing an example of a processing procedure of gain control function generation in the encoding method according to the embodiment of the present invention.

FIG. 5 is a diagram showing a recording state of a code string obtained by encoding according to the embodiment of the present invention.

FIG. 6 is a flowchart schematically showing an example of part of the processing procedure of the decoding method according to the embodiment of the present invention.

FIG. 7 is a diagram for explaining the operating principle of pre-echo generation in transform coding.

[Fig. 8] Fig. 8 is a diagram for explaining an operation principle of a conventional encoding / decoding technique by changing a conversion window length.

FIG. 9 is a diagram for explaining an operation principle of encoding / decoding using a conventional windowing processing technique.

[Fig. 10] Fig. 10 is a block diagram illustrating a schematic configuration of an encoding device according to a conventional windowing processing technique.

FIG. 11 is a block diagram showing a schematic configuration of a decoding device according to a conventional windowing processing technique.

[Explanation of symbols]

 101 band dividing means 111-114 normalization circuit 121-124 quantization circuit 131 multiplexer 141 quantization precision determination circuit 202 demultiplexer 211-214 signal component configuration circuit 221 band synthesis circuit

Claims (12)

[Claims] 1. A frequency component decomposing process for decomposing an input signal into frequency components, a gain control process of an input waveform signal to the frequency component decomposing process, output information of the frequency component decomposing process, and control information for the gain control. And the gain control amount of the gain control process in a portion where the waveform signal suddenly increases, are selectively determined from a plurality of types of sizes. 2. The information encoding method according to claim 1, wherein the frequency component decomposition process includes a spectrum conversion process for converting a signal on the time axis into a signal on the frequency axis. 3. The information encoding method according to claim 1, wherein the input signal is an acoustic signal. 4. A frequency component decomposition processing means for decomposing an input signal into frequency components, a gain control processing means for performing gain control processing of an input waveform signal to the frequency component decomposition processing means, and the frequency component decomposition processing means. Of the output information and the control information of the gain control described above, and the gain control amount of the gain control process in the portion where the waveform signal suddenly increases can be selected from a plurality of sizes. An information encoding device characterized by being determined according to 1. 5. A waveform signal sharply increases by performing decoding processing of a frequency component signal and gain control correction information, waveform signal synthesis processing, and gain control correction processing of an output waveform signal of the waveform signal synthesis processing. The information decoding method, wherein the gain control correction amount of the gain control correction processing in the part is selected from a plurality of sizes determined based on the content of the gain control correction information. 6. The information encoding method according to claim 6, wherein the frequency component combining process includes an inverse spectrum conversion process for converting a signal on the frequency axis into a signal on the time axis. 7. The information encoding method according to claim 5, wherein the output signal is an acoustic signal. 8. Decoding means for decoding a frequency component signal and gain control correction information, waveform signal synthesis processing means for synthesizing waveform signals, and gain control correction for an output waveform signal from the waveform signal synthesis processing means. A gain control correction processing means for performing processing, and the gain control correction amount of the gain control correction processing in a portion where the waveform signal suddenly increases becomes a plurality of magnitudes determined based on the contents of the gain control correction information. An information decoding device characterized by being selected from among. 9. Frequency component signal information and gain control correction information are recorded, the gain control correction information includes information on a gain control correction amount, and there are a plurality of types of gain control correction amounts in a portion where the waveform signal sharply increases. An information recording medium selected from the above. 10. The information recording medium according to claim 9, wherein the frequency component signal is obtained by a process including spectrum conversion. 11. The information recording medium according to claim 9, wherein the signal is an acoustic signal. 12. Frequency component signal information and gain control correction information are transmitted, the gain control correction information includes information on a gain control correction amount, and there are a plurality of types of gain control correction amounts in a portion where the waveform signal sharply increases. An information transmission method characterized by being selected from among.