JPH03117920A - Digital signal encoding device - Google Patents

Abstract

PURPOSE:To reduce the degradation in S/N on hearing by detecting the output information volume after quantization and correcting the number of assigned bits for quantization based on the error between this detection output and a target value, and the energy of each band to fix the information volume in a prescribed period. CONSTITUTION:The output information volume of a quantizing circuit 24 is detected by a data volume calculating circuit 26, and the error between this detection output and a prescribed target value (the target value of the bit rate for bit rate adjustment) from a terminal 3 is detected by an error detecting circuit 27. Thereafter, the correction value to correct the number of assigned bits of a quantizing circuit 24 based on this error output and the energy of each band is determined by a correction value determining circuit 28, and quantization is performed based on this correction value. Consequently, the information volume in the prescribed period is fixed. Thus, the degradation in S/N on hearing is reduced.

Description

DETAILED DESCRIPTION OF THE INVENTION [Field of Industrial Application] The present invention relates to a digital signal encoding device that encodes an input digital signal.

[Summary of the invention]

The present invention divides an input digital signal into a plurality of frequency bands, selects a wider band width for a higher frequency band, sets an allowable noise level for each band based on the energy of each band, and In a digital signal encoding device that quantizes the components of each band with the number of bits according to the level of the difference between the energy and the set allowable noise level, the amount of output information after quantization is detected, and this detected output and the target value are Based on the error of and the energy of each band,
By correcting the number of allocated bits during quantization and making the amount of information constant over a given period, it is possible to perform bit rate adjustment (bit backing) with a simple configuration and with little signal deterioration. In particular, it is an object of the present invention to provide a digital signal encoding device that can allocate a large number of bits to a frequency band where energy is concentrated, and can improve the perceptual S/N.

[Conventional technology]

In high-efficiency encoding of audio, voice, etc. signals,
2. Description of the Related Art There is an encoding technique using bit allocation that divides an input signal such as audio or voice into a plurality of channels on a time axis or a frequency axis and adaptively allocates the number of bits for each channel. For example, the encoding technology using the above-mentioned bit allocation for audio signals, etc.
Band division coding (sub-band coding: 5BC), which divides and encodes audio signals on the time axis into multiple frequency bands, and converts time-axis signals to signals on the frequency axis (orthogonal transformation) So-called adaptive transform coding (A
TC), or by combining the above SBC and so-called adaptive predictive coding (APC), dividing the time domain signal into bands and converting each band signal to baseband (low band), followed by multi-order linear predictive analysis. There are encoding techniques such as so-called adaptive bit allocation (APC-AB) that performs predictive encoding.

For example, in the band division encoding described above, in order to increase compression efficiency, while keeping the bit rate for each unit time block constant, the number of bits given to each divided band is determined by the time fluctuation of the signal spectrum intensity. It changes dynamically (adaptively) according to the situation. Furthermore, in the adaptive transform encoding described above, the number of allocated bits is dynamically changed on the frequency axis.

[Problem to be solved by the invention]

In high-efficiency coding that adaptively allocates the number of bits as described above, the bit rate per unit block (unit time block or unit frequency block) may not be constant depending on how the number of bits is allocated. There may be an excess or deficiency of the amount. That is, for a given bit rate, there may occur a phenomenon in which there are excess bits in areas where the signal level is low on the time axis or frequency axis, and there are insufficient bits in areas where the signal level is high.

If there is such an excess or deficiency in the number of bits, it is necessary to use a bit rate adjustment method (so-called bit banking) to effectively adjust the number of bits.

This bit rate adjustment means that when there are excess bits in a unit block, the number of bits given to this unit block is reduced, and when there are insufficient bits, the number of bits given is increased, thereby keeping the overall bit rate constant. It is something to be adjusted. It is conceivable that the bit rate adjustment can be performed using functional blocks as shown in FIG. 8, for example. In FIG. 8, an allocated bit number determination function block 90 determines the allocated bit number for quantization with respect to the input digital signal, and a bit rate adjustment function block 91 determines the 'spectrum of a unit block of the input digital signal. Adjust the bit rate (bit backing) according to the strength. Thereafter, the input digital signal is encoded (requantized) in the encoding function block 92 using the number of bits determined by adjusting the bit rate and is output.

However, as for the bit rate adjustment performed by the bit rate adjustment function block 92, there is still no effective method that is simple and does not cause noticeable signal deterioration. Establishment of a method is desired. In particular, when adjusting the bit rate, tonal sources (sine waves, tone burst waves,
With regard to single instrument sounds, etc., noise generated in that band is often misinterpreted as deterioration, and there is a need for a countermeasure to prevent this deterioration.

Therefore, the present invention has been proposed in view of the above-mentioned actual situation, and it is possible to perform pit-lay daily adjustment (bit banking) with a simple configuration in which signal deterioration is not noticeable, and in particular, tones. It is an object of the present invention to provide a digital signal encoding device that can perform signal encoding with little deterioration even when using a certain source.

[Failure to solve the problem]

The digital signal encoding device of the present invention has been proposed to achieve the above-mentioned object. For example, as shown in FIG. A total sum detection circuit 14 and a filter circuit 15 as noise level setting means for selecting a wider band width as the band increases, and setting an allowable noise level for each band based on the energy of each band, and the energy of each band. In a digital signal encoding device having a quantization circuit 24 that quantizes the components of each band with a pip corresponding to the level of the difference between the noise level setting means, the amount of information output from the quantization circuit 24 is detected. The number of bits allocated to the quantization circuit 24 is corrected based on the error between the detection output and the target value and the energy of each band, thereby making the amount of information constant in a predetermined period. be.

[Effect]

According to the present invention, by keeping the overall bit rate constant and at the same time increasing or decreasing the number of allocated bits by a certain amount based on the energy of each band, a large number of bits is allocated to bands where energy is concentrated.

[Example] Hereinafter, an example to which the present invention is applied will be described with reference to the drawings.

The digital signal encoding device of this embodiment converts an input digital signal such as audio or voice by, for example, band division coding (SBC), adaptive transform coding (ATC), adaptive bit allocation (APC-AB), etc. It performs high-efficiency encoding. Therefore, in the device of this embodiment, the input digital signal is divided into a plurality of frequency bands, and the higher the frequency band, the wider the bandwidth is selected. That is, the input digital signal is divided by a so-called critical band that takes into account the human auditory characteristics described later. In addition, as shown in Figure 1, the engineering value (or peak value,
Total sum detection circuit 1 as a noise level setting means for setting the allowable noise level for each band based on the average value)
4, a filter circuit 15, and a nucleation circuit 24 that quantizes the components of each band with the number of bits allocated according to the level of the difference between the energy of each band and the noise level setting means. It is. Here, the amount of information output from the quantization circuit 24 is determined by a data amount calculation circuit 26, which will be described later.
The error between the detection output and a predetermined target value (target value of bit rate for bit rate adjustment) from the terminal 3 is detected by an error detection circuit 27, which will be described later. after that,
A correction value determining circuit 28 (described later) determines a correction value for correcting the number of vias allocated in the quantization circuit 24 based on this error output and the energy of each band, and quantization is performed based on this correction value. ing. By doing this, in the device of this embodiment, the amount of information in a predetermined period is made constant (adjustment of bit rate).

Thereafter, the quantized output from the quantization circuit 24 is outputted from the output terminal 2 of the digital signal encoding device of this embodiment via the panzer memory 25.

For this reason, the signal supplied to the apparatus of FIG.
If the signal has a signal spectrum SS as shown in Figure 2A, it is a tonal source (sine wave, tone burst wave, single instrument sound) in which the signal (spectral intensity, energy) is concentrated in a certain band. etc.), the noise spectrum NS becomes as shown by the dashed line in FIG. 2 with respect to the signal spectrum SS. Therefore, in the device of this embodiment shown in FIG. 1, for example, frequency analysis of the signal is performed to determine the energy ratio for each band,
If the number of bits is left over for a given fixed bit, the bit rate is increased by giving more bits to bands with higher energy in proportion to the signal energy and allocating fewer bits to bands with smaller (fewer) bits. Make adjustments. As a result, as shown in FIG. 2B, the noise spectrum NS is
The noise spectrum is corrected as indicated by the dotted line in FIG. 2B. Conversely, when there is a shortage of bits, bits are taken from a band with less energy in inverse proportion to the signal energy and applied to the band with the insufficient number of bits.

As a result, as shown in FIG. 2C, the noise spectrum NS becomes the noise spectrum N shown by the dotted line in FIG.
It is corrected as Sc.

Here, the digital signal encoding device of this embodiment shown in FIG. 1 performs fast Fourier transform (FFT) on the audio signal 1 audio signal etc. to convert the time axis signal into the frequency axis,
It performs encoding (requantization).

In FIG. 1, an audio signal, for example, is supplied to an input terminal l, and this audio signal on the time axis is transmitted to a fast Fourier transform circuit 11. In this fast Fourier transform circuit 11, the audio signal on the time axis is converted into a signal on the frequency axis at every predetermined time (unit block), and F is made up of a real component value Re and an imaginary component value 1m.
The FT coefficient is obtained. These FFT coefficients are transmitted to the amplitude and phase information generation circuit 12, and the amplitude and phase information generation circuit 1
In step 2, an amplitude value Am and a phase value are obtained from the real component value Re and the imaginary component value Im, and information on this amplitude value Am is output. That is, in general, human hearing is sensitive to amplitude (power) in the frequency domain, but is quite insensitive to phase. Therefore, in this embodiment, only the amplitude value Am is calculated from the output of the amplitude and phase information generation circuit I2. This is taken out and used as an input digital signal in the embodiment of the present invention.

The input digital signal such as the amplitude value Am obtained in this way is transmitted to the band division circuit 13.

This band division circuit 13 divides the input digital signal expressed by the amplitude (I!!Am) into a so-called critical band. This critical band is defined by human hearing characteristics (frequency analysis ability). ), for example, 0 to 16kHz is divided into 24 bands, and the higher the frequency band, the wider the bandwidth is selected.In other words, human hearing has characteristics like a kind of bandpass filter. The bands separated by these filters are called critical bands.The critical bands are shown in FIG.

However, in order to simplify the illustration in FIG. 3, the number of critical bands is 12 (S+ to B
+z).

The amplitude value Am for each band (for example, 24 bands) divided into critical bands by the band division circuit 13 is transmitted to the sum detection circuit 14. In this summation detection circuit 14, the energy of each band (spectral intensity in each band) is determined by the amplitude value A of each band.
It is obtained by taking the summation of m (the peak or average of the amplitude value Am or the summation of energy). The sum detection circuit 1
4, that is, the spectrum of the sum of each band is generally called a park spectrum, and the park spectrum SB of each band is as shown in FIG. 4, for example.

Here, in order to consider the influence of the park spectrum SB on so-called masking, the park spectrum S
B is convolved with a predetermined weighting function (convolution). Therefore, the output of the sum detection circuit 14, that is, the multi-value of the park spectrum SB, is sent to the filter circuit 15. As shown in FIG. 5, the filter circuit 15 includes delay elements (Z-') that sequentially delay input data from the input terminal 100.
01-z~101-+...101!3.10124
(for example, 24 delay elements corresponding to the critical band) and 24 multipliers 102. for example, which multiply the outputs of these delay elements 1011 to 101□ by a filter coefficient (weighting function). ．． 102□・・・Bow 02-13
102□, . . . 102□s, 102□4, and a total adder 104.

At this time, the multipliers 102□3 to 102. I. , for example, the filter coefficients 0, 00 in the multiplier 102.3
00086 to multiplier 102. -, the filter coefficient is 0.
0019, multiplier 102.1 filter coefficient 0.1
5, filter coefficient 1 in multiplier 1028, multiplier 10
2--+ filter coefficient 0.4, and multiplier 1021
Multiply the filter coefficient 0,06 by ゜2, multiplier 102,
, ff is used to multiply the output of each delay element by a filter coefficient of 0.007, thereby performing the convolution process of the park spectrum SB. Through this convolution process, the sum of the parts shown by the dotted line in FIG. 4 is calculated.

The masking mentioned above refers to a phenomenon in which one signal masks another signal so that it cannot be heard due to the characteristics of human hearing.This masking effect includes:
There are masking effects on audio signals on the time axis and masking effects on signals on the frequency axis. That is, due to the masking effect, even if there is noise in the masked portion, this noise will not be heard. Therefore, in an actual audio signal, noise within this masked portion is considered to be acceptable noise.

Thereafter, the output of the filter circuit 15 is sent to a subtracter 16. The subtracter 16 is used to obtain a level α corresponding to an allowable noise level, which will be described later, in the convolved region. Note that the level α corresponding to the permissible noise level (tolerable noise level) is a level that becomes the permissible noise level for each critical band by performing inverse convolution processing, as described later. be. Here, the subtracter 16 is supplied with a tolerance function (a function expressing the masking level) for determining the level α. The level α is controlled by increasing or decreasing this tolerance function. The permissible function is supplied from a function generation circuit 29, which will be described later.

In other words, the level α corresponding to the allowable noise level can be determined by equation (1), where i is the number given in order from the low range of the critical band.

α=S-(n-ai) (1) This (1st
), n and a are constants, a>0, S is the intensity of the convolved park spectrum, and in equation (1), (
n-ai) becomes the tolerance function. In this example, n=38.
With a=1, there was no deterioration in sound quality at this time, and good encoding was achieved.

In this way, the level α is determined and this data is transmitted to the divider 17. The divider 17 is for deconvoluting the level α in the convolved area. Therefore, by performing this inverse convolution process, a masking spectrum can be obtained from the level α. That is, this masking spectrum becomes the allowable noise spectrum. The inverse convolution process described above requires complicated calculations, but in this embodiment, a simplified divider 17 is used to perform the inverse convolution process.

Next, the masking spectrum is transmitted to a subtracter 19 via a synthesis circuit 18. Here, the subtracter 19
The output of the summation detection circuit 14, that is, the park spectrum SB from the summation detection circuit 14 described above is supplied to the delay circuit 21 through a delay circuit 21. Therefore, by performing a subtraction operation between the masking spectrum and the park spectrum SB in the subtracter 19, as shown in FIG. 6, the park spectrum SB is lower than the level indicated by the masking spectrum MS. will be done.

The output of the subtracter 19 is supplied to the quantization circuit 24 via the ROM 20, and the quantization circuit 24 supplies the output via the delay circuit 23 with the number of allocated bits corresponding to the output of the subtracter 19. The current amplitude value Am will be quantized. That is, in other words, the vMR conversion circuit 24 quantizes the components of each band with the number of bits allocated according to the level of the difference between the energy of each of the critical bands and the noise level setting means. I will do it. The delay circuit 21 delays the park spectrum SB from the summation detection circuit 14 by considering the amount of delay in each circuit before the synthesis circuit 18, and the delay circuit 23 delays the park spectrum SB from each circuit before the ROM 20. The amplitude value Am is provided to delay the amplitude value Am in consideration of the amount of delay. Further, the ROM 20 is provided to temporarily store and send out the output of the subtracter 19 at predetermined time intervals during quantization.

Here, when the quantization circuit 24 performs quantization, as described above, the output information amount of the quantization circuit 24 is detected,
Based on the error between the detection output and the target value and the energy of each band, the number of bits allocated during quantization is corrected to keep the amount of information constant in a predetermined period.

In order to do this, in the device of this embodiment,
The data from the panzer memory 25 is sent to the error detection circuit 27 after the data amount is calculated by the data amount calculation circuit 26. The error detection circuit 27 detects an error between the amount of data and a predetermined target value from the terminal 3 (target value of bit rate for adjusting bit rate l),
The error data is transmitted to the correction value determining circuit 28. The correction value determination circuit 2 is also supplied with the output (energy, spectral intensity) of the summation detection circuit 14 via the memory 33 that stores unit block data.

The correction value determining circuit 28 applies a larger correction value to a band with high energy if the error data is positive, and, conversely, gives a small correction value to a band with high energy if the error data is negative. It is something to do. Such a correction value is transmitted to the bit number correction circuit 31. This bit number correction circuit 31 corrects the output of the ROM 20 with the correction value, thereby correcting the number of allocated bits during quantization (adjusting the bit rate), and The amount of information in a time block becomes constant.

For this reason, in bands with high energy, quantization is performed with a large number of bits, and in bands with low energy, quantization is performed with a small number of bits. Here, since the output of the bit number correction circuit 31 is supplied to the quantization circuit 24 via the rounding circuit 32, small changes in the bit number are corrected.

In addition, during the synthesis in the above-mentioned synthesis circuit 18, the so-called minimum audible curve (equal loudness curve) RC, which is the human auditory characteristic as shown in FIG. 7, is supplied from the minimum audible curve generation circuit 22. The data and the masking spectrum MS can be combined. therefore,
By combining this minimum audible curve RC and the masking spectrum MS, the allowable noise level can be reduced up to the shaded area in the figure, and when quantizing, the allocation of the shaded area in the figure It becomes possible to freeze the number of bits. Note that this figure 7 is similar to the above-mentioned figure 3.
It is represented by the critical band shown in the figure, and the signal spectrum SS is also shown at the same time.

However, in this embodiment, it is also possible to adopt a configuration in which the above-described minimum audible curve synthesis process is not performed. That is, in this case, the minimum audible curve generation circuit 222 and the synthesis circuit 18 are unnecessary, and the output from the subtracter 16 is as follows.
After being deconvolved in the divider 17, it is immediately transmitted to the subtracter 19.

That is, in the digital signal encoding device of this embodiment, when encoding an audio signal, the bit rate is adjusted as described above, and the shape of the noise spectrum does not change, so even if the number of bits is frozen. ,
This results in less auditory deterioration. In particular, it becomes possible to perform signal encoding with little deterioration even for a source with tonal characteristics. Furthermore, since the bit rate adjustment algorithm is simple, the hardware can be configured easily.

In addition to so-called adaptive transform coding in which an input digital signal is processed by performing fast Fourier transform as in the embodiment shown in FIG.
) can also be applied to devices that perform in this case,
The signal is filtered using a bandpass filter or the like in the band i1L, and the number of bits allocated to each channel is increased or decreased based on the error between the detected output of the output information amount of the quantization means and the target value, and the energy of each band. . In the case of the band division encoding, the same effects as described above can be obtained.

〔Effect of the invention〕

In the digital signal encoding device of the present invention, the amount of output information after quantization is detected, and the number of bits allocated during quantization is corrected based on the error between the detected output and the target value and the energy of each band. By making the information bond constant over a predetermined period in this manner, it becomes possible to perform bit rate adjustment (bit banking) with a device of simple configuration in which signal deterioration is not noticeable. In particular, it becomes possible to perform signal encoding with little deterioration even for a source with tonal characteristics. Therefore, it becomes possible to reduce the auditory S/N deterioration.

[Brief explanation of drawings]

FIG. 1 is a block circuit diagram showing a schematic configuration of a digital signal encoding device according to an embodiment of the present invention, FIG. 2A is a characteristic diagram showing a noise spectrum before bit rate adjustment, and FIG. 2B is a bit A characteristic diagram showing the noise spectrum after rate adjustment (in the case of excess bits), Figure 2C is a characteristic diagram showing the noise spectrum after bit rate adjustment (in the case of insufficient bits), and Figure 3 shows the critical bunt. Figure 4 is a diagram showing the park spectrum, Figure 5 is a circuit diagram showing the filter circuit, Figure 6 is a diagram showing the masking spectrum, Figure 7 is the minimum audible curve and a diagram combining the masking spectrum, Figure 8 The figure is a functional block diagram for bit rate adjustment. 11... Fast Fourier transform circuit 12...
...... Amplitude phase information generation circuit 3 ...... Band division circuit 4 ...... Sum detection circuit 5 ...... Filter circuit 6 ... ...Subtractor 7...Divider 8...Synthesizing circuit 9...Subtractor O... ROM ■, 23...Delay circuit 2...Minimum audible curve generation circuit 4...
...Style generation circuit 5 ...Buffer memory 6 ...Data amount calculation circuit 7 ...Error detection circuit 8 ... ...Correction value determination circuit 9 ...Function generation circuit 1 ...Bit number correction circuit 2 ...Rounding circuit 3 ... ··memory

Claims (1)

[Claims] A noise generating device that divides an input digital signal into a plurality of frequency bands, selects a wider band width for higher frequency bands, and sets an allowable noise level for each band based on the energy of each band. A digital signal encoding device comprising: a level setting means; and a quantization means for quantizing the components of each band with a number of bits corresponding to the level of the difference between the energy of each band and the noise level setting means, detecting the output information amount of the quantizing means, and correcting the number of bits allocated to the quantizing means based on the error between the detected output and the target value and the energy of each band;
A digital signal encoding device characterized in that the amount of information in a predetermined period is made constant.