JP3041967B2 - Digital signal coding device - Google Patents

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to a digital signal encoding apparatus for encoding an input digital signal.

[0002]

BACKGROUND ART _Audio, in high-efficiency encoding of the signal of voice, etc., audio, adaptive with, the number of bits of each channel is divided into a plurality of channels of the input signal in the time axis or the frequency axis such as voice There is a coding technique based on bit allocation (bit allocation).
For example, the encoding technique based on the above-mentioned bit allocation of an audio signal or the like includes band division encoding (sub-coding) in which an audio signal or the like on a time axis is divided into a plurality of frequency bands and encoded.
Band coding: SBC) or so-called adaptive conversion coding (ATC) in which a signal on the time axis is converted into a signal on the frequency axis (orthogonal conversion), divided into a plurality of frequency bands, and adaptively encoded for each band. Alternatively, the SBC is combined with so-called adaptive prediction coding (APC), a signal on the time axis is divided into bands, and each band signal is converted into a baseband (low band), and then a multi-order linear prediction analysis is performed. There is an encoding technique such as so-called adaptive bit allocation (APC-AB) for predictive encoding.

[0003] For example, in the above-mentioned band division coding, in order to increase the compression efficiency, the number of bits given to each band-divided band is kept constant while the number of bits per unit time block is kept constant. It is dynamically (adaptively) changed according to the time variation of the spectrum intensity. In the adaptive transform coding, the number of allocated bits is dynamically changed on the frequency axis while the bit rate of each unit frequency block is kept constant.

[0004]

However, in the above-described high-efficiency coding in which coding is performed by adaptively allocating the number of bits as described above, the nature of the input signal to be coded and the way of bit allocation are determined. In some cases, the number of bits per unit block (unit time block or unit frequency block) may not be constant. That is, excess or deficiency may occur for a predetermined number of bits per unit block. In other words, the predetermined number of bits for each unit block does not sufficiently satisfy the characteristics required for the decoded signal (insufficient bits), or conversely, the number of bits is May be too much (bit excess) to obtain the characteristics required for the signal of For example, in a case where the number of bits is insufficient in encoding an audio signal, the signal is degraded by encoding because the bit rate is insufficient, and noise is increased when the audio signal is decoded later. Further, if there are extra bits, more processing than necessary is performed, which is not preferable.

[0005] For this reason, conventionally, when the number of bits per unit block is excessive or deficient, so-called bit packing (bit rate adjustment) is performed to keep the number of bits per unit block constant. I have. However, in this bit packing, the number of bits in a unit block is uniformly increased / decreased, for example, to keep the bit rate constant, so that signal degradation becomes noticeable.

Accordingly, the present invention has been proposed in view of the above-described circumstances, and has as its object to provide a digital signal encoding apparatus capable of adjusting a bit rate with little signal degradation. .

[0007]

SUMMARY OF THE INVENTION A digital signal encoding apparatus according to the present invention has been proposed in order to achieve the above-mentioned object, and divides an input digital signal into a plurality of frequency bands and has a high frequency band. Noise level setting means for setting the allowable noise level for each band based on the energy of each band, and a difference between the energy of each band and the output of the noise level setting means. A digital signal encoding device having quantization means for quantizing the components of each band with the number of bits according to the level, detecting an output information amount of the quantization means, and detecting an error between the detected output and a target value, Based on the information of the equal loudness curve, the allowable noise level from the noise level setting means is corrected to make the amount of information constant for a predetermined period. Those were Unishi.

[0008]

According to the present invention, the number of bits is determined according to the difference between the allowable noise level and the energy of each band, and the number of bits is determined based on the error between the output information amount of the quantization means and the target value. Is calculated, and if the number of bits is excessive or deficient, the allowable noise level is corrected to keep the bit rate constant. At the same time, the allowable noise level is corrected in consideration of the equal loudness curve, so that the signal degradation is not noticeable.

[0009]

Embodiments of the present invention will be described below with reference to the drawings. The digital signal encoding apparatus according to the present embodiment converts input digital signals such as audio and voice by, for example, band division encoding (SBC), adaptive conversion encoding (ATC), adaptive bit allocation (APC-AB), and the like. High efficiency coding. Therefore, in the present 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 bandwidth (critical band) in consideration of human auditory characteristics described later. Further, as shown in FIG. 1, the sum detection circuit 14 as a noise level setting means for setting an allowable noise level for each band based on the energy (or peak value, average value) of each band of the critical band. It has a filter circuit 15 and a quantization circuit 24 that quantizes the components of each band with the number of bits assigned according to the level of the difference between the energy of each band and the output of the noise level setting means. is there. Here, a data amount calculation circuit 26 described later detects the output information amount of the quantization circuit 24, and an error between the detected output and a predetermined target value (predetermined bit rate) from the terminal 3 will be described later. It is detected by the comparison circuit 27. Thereafter, the correction value determination circuit 28 determines the correction value of the allowable noise level based on the error information and information of a so-called equal loudness curve described later. The allowable noise level correction circuit 20 corrects the allowable noise level according to the correction value. Therefore, the quantization circuit 24 performs quantization with the number of bits obtained according to the corrected allowable noise level. Thus, in the present embodiment, the number of bits in a predetermined period (the unit block) is made constant.
Thereafter, the quantized output from the quantizing circuit 24 is output from the output terminal 2 of the digital signal encoding device of the present embodiment via the buffer memory 25.

Here, the digital signal encoding apparatus of the present embodiment shown in FIG. 1 performs fast Fourier transform (FFT) on an audio signal, a voice signal, and the like, converts a signal on a time axis into a frequency axis, and encodes the signal. (Requantization). FIG.
, An input terminal 1 is supplied with, for example, an audio signal, and the audio signal on the time axis is transmitted to the fast Fourier transform circuit 11. The fast Fourier transform circuit 11 converts the audio signal on the time axis into a signal on the frequency axis at predetermined time intervals (unit blocks),
An FFT coefficient including the real component value Re and the imaginary component value Im is obtained. These FFT coefficients are transmitted to the amplitude / phase information generation circuit 12, and the amplitude / phase information generation circuit 12 obtains the amplitude value Am and the phase value from the real component value Re and the imaginary component value Im. Will be output. In other words, human hearing is generally sensitive to the amplitude (power) in the frequency domain, but relatively insensitive to the phase. Therefore, in this embodiment, only the amplitude value Am is obtained from the output of the amplitude / phase information generation circuit 12. This is taken out as an input digital signal in the embodiment of the present invention.

The input digital signal such as the amplitude value Am obtained as described above is transmitted to the band dividing circuit 13.
The band dividing circuit 13 divides the input digital signal represented by the amplitude value Am into a so-called critical bandwidth (critical band). The critical band takes into account human auditory characteristics (frequency analysis capability). For example, 0 to 22 kHz is divided into 25 bands, and the higher the frequency band, the wider the bandwidth is selected. That is, human hearing has characteristics like a kind of band-pass filter, and the band divided by each filter is called a critical band.

The amplitude value Am of each band divided into the critical bands by the band dividing circuit 13 is transmitted to the sum detecting circuit 14, respectively. This sum detection circuit 1
In 4, the energy for each band (spectral intensity in each band) is obtained by taking the sum of the amplitude values Am in each band (peak or average of the amplitude values Am or the energy sum). The output of the sum detection circuit 14, that is, the spectrum of the sum of each band is generally called a bark spectrum, and the bark spectrum SB of each band is, for example, as shown in FIG. However, in FIG. 2, the number of the critical bands is represented by 12 bands (B _{1 to} B ₁₂ ) to simplify the illustration.

Here, in order to consider the influence of the bark spectrum SB on so-called masking, a predetermined weighting function is convolved with the bark spectrum SB (convolution). Therefore, the output of the sum detection circuit 14, that is, each value of the bark spectrum SB is sent to the filter circuit 15. The filter circuit 15 includes, for example, a plurality of delay elements for sequentially delaying input data and a plurality of multipliers (for example, 25 corresponding to each band) for multiplying an output from these delay elements by a filter coefficient (weighting function). Multipliers) and a sum adder for summing the outputs of the multipliers. In each of the multipliers of the filter circuit 15, for example, the multiplier M corresponding to an arbitrary band has a filter coefficient of 1, a multiplier M-1 has a filter coefficient of 0.15, and a multiplier M-2 has a filter coefficient of 0.1.
0019 is filtered by a multiplier M-3 with a filter coefficient of 0.00000.
086, a filter coefficient of 0.4 by a multiplier M + 1, a filter coefficient of 0.06 by a multiplier M + 2, and a filter coefficient of 0.007 by a multiplier M + 3, by multiplying the output of each delay element by the bark spectrum SB. Is performed. Here, M is an arbitrary integer of 1 to 25. By this convolution processing, the sum of the parts indicated by the dotted lines in FIG. 2 is obtained. The masking refers to a phenomenon that a certain signal masks another signal and makes it inaudible due to human auditory characteristics.This masking effect includes a masking effect for an audio signal on a time axis. There is a masking effect on signals on the frequency axis. That is, even if there is noise in the masked portion due to the masking effect, this noise will not be heard. For this reason, in an actual audio signal, the noise in the masked portion is regarded as acceptable noise.

Thereafter, the output of the filter circuit 15 is sent to a subtractor 16. The subtracter 16 calculates a level α corresponding to an allowable noise level described later in the convolved area. The level α corresponding to the permissible noise level (permissible noise level) is, as described later, a level at which the permissible noise level of each critical band is obtained by performing inverse convolution processing. is there. Here, an allowance function (a function expressing a masking level) for obtaining the level α is supplied to the subtractor 16. The level α is controlled by increasing or decreasing the allowable function. The permissible function is supplied from a function generation circuit 29 described later. That is, the level α corresponding to the allowable noise level can be obtained by the following equation (1), where i is a number sequentially given from the lower band of the critical band. α = S- (n-ai) (1) In the equation (1), n and a are constants, a> 0, and S is the intensity of the convolution-processed Bark spectrum. , No. (1)
In the equation, (n-ai) is the allowable function. In this embodiment, n = 38,
Since a = 1, there was no deterioration in sound quality at this time, and good encoding was performed.

Thus, the level α is obtained, and this data is transmitted to the divider 17. The divider 17 is for inversely convolving the level α in the convolved region. Therefore, by performing this inverse convolution processing,
A masking spectrum can be obtained from the level α. That is, this masking spectrum becomes an allowable noise spectrum. Note that the above inverse convolution process requires a complicated operation, but in the present embodiment, inverse convolution is performed using a simplified divider 17.

Next, the masking spectrum is transmitted to a subtractor 19 via a synthesis circuit 18. Here, the subtractor 19 outputs the output of the sum detection circuit 14, that is, the bark spectrum S from the sum detection circuit 14 described above.
B is supplied via a delay circuit 21. Therefore, the subtractor 19 performs a subtraction operation between the masking spectrum and the bark spectrum SB, thereby obtaining the signal shown in FIG.
As shown in (1), the bark spectrum SB is masked below the level indicated by the level of the masking spectrum MS.

The output of the subtracter 19 is sent to the ROM 30 via the allowable noise level correction circuit 20. The ROM 30 stores a plurality of pieces of assigned bit number information. The assigned bit number information corresponding to the output of the subtraction circuit 19 (the level of the difference between the energy of each band and the output of the noise level setting means). Is output. The output of the ROM 30 is supplied to the quantization circuit 24. In the quantization circuit 24, the ROM
The quantization of the amplitude value Am supplied through the delay circuit 23 is performed with the number of allocated bits from 30. In other words, in other words, the quantization circuit 24 quantizes the components of each band with the number of bits assigned according to the level of the difference between the energy of each band of the critical band and the output of the noise level setting means. Will be transformed. The delay circuit 21 delays the bark spectrum SB from the sum detection circuit 14 in consideration of the delay amount in each circuit before the synthesis circuit 18, and the delay circuit 23 delays the bark spectrum SB in each circuit before the ROM 30. It is provided to delay the amplitude value Am in consideration of the amount.

Also, at the time of synthesizing by the synthesizing circuit 18, the audible curve generating circuit 22 shown in FIG.
And data indicating a so-called minimum audible curve RC which is a human auditory characteristic as shown in FIG.
And S can be synthesized. At this minimum audible curve, if the absolute noise level is below this minimum audible curve, the noise will not be heard. Further, even if the coding is the same, the minimum audible curve differs depending on, for example, a reproduction volume at the time of reproduction. However, in a realistic digital system, for example, there is not much difference in how to enter music into the 16-bit dynamic range. For example, if quantization noise in the most audible frequency band around 4 kHz is not heard, It is considered that quantization noise below the level of the minimum audible curve is not audible in other frequency bands. Therefore, assuming that the system is used so that noise around 4 kHz of the word length of the system cannot be heard, and an allowable noise level is obtained by synthesizing the minimum audible curve RC and the masking spectrum MS together, In this case, the permissible noise level can be set up to the shaded portion in the figure. In this embodiment, the 4 kHz level of the minimum audible curve is adjusted to the lowest level corresponding to, for example, 20 bits. Also, FIG.
The signal spectrum SS is also shown.

Here, the allowable noise level correction circuit 2
In the case of 0, the allowable noise level from the subtractor 19 is determined based on the detection output of the output information amount (data amount) of the quantization circuit 24 and the error of the target value and the information of the equal loudness curve as described above. Has been corrected. To do this, the data amount from the buffer memory 25 is calculated by the data amount calculation circuit 26,
The signal is sent to the comparison circuit 27. The comparison circuit 27 detects an error between the data amount and a predetermined target value (predetermined bit rate) from the terminal 3, and transmits the error data to the correction value determination circuit 28. Here, as a case where the error data is data indicating a bit number shortage, a case where the data amount is larger than the target value by using a large number of bits per unit block is considered. Can be. Further, as a case where the error data is data indicating the remainder of the number of bits, a case where the number of bits per unit block is small and the amount of data is smaller than the target value can be considered. Therefore, from the correction value determination circuit 28,
In accordance with the error data, data of the correction value for correcting the allowable noise level from the subtractor 19 based on information data of a so-called equal loudness curve is output. By transmitting the correction value as described above to the allowable noise level correction circuit 20, the allowable noise level from the subtractor 19 is corrected. The above-mentioned equal loudness curve relates to human auditory characteristics. For example, the equal loudness curve is obtained by obtaining sound pressures of sounds at each frequency that sounds as loud as a pure sound of 1 kHz and connecting them with a curve. Also called a curve. Further, this equal loudness curve draws substantially the same curve as the minimum audible curve RC shown in FIG. In this equal loudness curve, for example, 1 kHz around 4 kHz
Even if the sound pressure drops by 8 to 10 dB from the point, it sounds the same as 1 kHz, and conversely, it is 1 kHz near 50 kHz.
Unless the sound pressure is about 15 dB higher than the sound pressure, the sound cannot be heard at the same level. For this reason, it can be seen that noise exceeding the level of the minimum audible curve (allowable noise level) preferably has a frequency characteristic given by a curve corresponding to the equal loudness curve. From this, it can be seen that correcting the allowable noise level in consideration of the equal loudness curve is suitable for human auditory characteristics.

As an allowable noise level correction considering the equal loudness curve, for example, when the number of bits is insufficient, the allowable noise level shown in FIG. 4 is changed to a correction level AR1 as shown in FIG. to correct. That is, in FIG. 5, raising the allowable noise level to the correction level AR1 means lowering the number of bits obtained according to the level of the difference between the energy of each band and the output of the noise level setting means. As a result, the shortage of the number of bits per unit block is eliminated. Even in the case of making an adjustment to lower the bit rate in this way, since this adjustment is based on the allowable noise level in consideration of the above equal loudness curve, noise of the decoded speech is minimized. It will be.

If the number of bits is excessive, the allowable noise level shown in FIG. 4 is changed to the correction level AR2 shown in FIG.
To be corrected. That is, in FIG. 6, lowering the allowable noise level to the correction level AR2 is as follows.
The number of bits obtained according to the level of the difference between the energy of each band and the output of the noise level setting means is increased, and as a result, the number of bits remaining per unit block can be used effectively. By increasing the bit rate in this way, the sound quality after decoding can be improved, and the sound quality can be further improved by considering the above equal loudness curve.

By correcting the allowable noise level as described above, the ROM corresponding to the level of the difference between the corrected allowable noise level and the energy of each band is determined.
The number of allocated bits from 30 will also be corrected. In other words, the number of bits allocated at the time of quantization in the quantization circuit 24 is corrected (adjustment of the bit rate), and the information amount in the unit block as a predetermined period is made constant. Become.

In this embodiment, a configuration may be adopted in which the above-described minimum audible curve synthesizing process is not performed. That is, in this case, the minimum audible curve generating circuit 22 and the synthesizing circuit 18 become unnecessary, and the output from the subtractor 16 is inversely convolved by the divider 17 and immediately transmitted to the subtractor 19. Will be done.

As described above, in the digital signal encoding apparatus according to the present embodiment, when encoding an audio signal, the number of bits is adjusted to be too small or too small. Since the bit rate is adjusted based on the correction of the allowable noise level in consideration of the above, even if the number of bits is reduced, for example, the deterioration in audibility can be reduced. The correction of the allowable noise level is
Since it is possible only by changing the level, it can be realized with a simple hardware configuration.

The present invention also employs, for example, band division coding (SBC) in addition to the so-called adaptive transform coding for processing the input digital signal by performing a fast Fourier transform as in the embodiment of FIG. The present invention can be applied to an apparatus for performing the above. In this case, the signal is band-divided by a band-pass filter or the like, and the number of bits to be allocated to each channel is corrected based on the error between the detection output of the output information amount of the quantization means and the target value and the equal loudness curve. It is increased or decreased according to the allowable noise level. In the case of the band division coding, the same effect as described above can be obtained.

[0026]

In the digital signal encoding apparatus according to the present invention, the amount of output information after quantization is detected, and based on the error between the detected output and the target value and information on the equal loudness curve,
By correcting the allowable noise level and stabilizing the amount of information during a predetermined period, it is possible to adjust the bit rate, and to adjust the bit rate. It is possible to perform signal encoding with little S / N deterioration (above).

[Brief description of the drawings]

FIG. 1 is a block circuit diagram showing a schematic configuration of a digital signal encoding device according to one embodiment of the present invention.

FIG. 2 is a diagram showing a bark spectrum.

FIG. 3 is a diagram showing a masking spectrum.

FIG. 4 is a diagram in which a minimum audible curve and a masking spectrum are combined.

FIG. 5 is a characteristic diagram showing a corrected allowable noise level when the number of bits is insufficient.

FIG. 6 is a characteristic diagram showing a corrected allowable noise level when the number of bits is excessive.

[Explanation of symbols]

 11 Fast Fourier transform circuit 12 Amplitude / phase information generating circuit 13 Band dividing circuit 14 Sum detection Circuit 15 ... Filter circuit 16 ... Subtractor 17 ... Divider 18 ... Synthesis circuit 19 ... ... Subtractor 20... Allowable noise level correction circuit 22... Minimum audible curve generation circuit 24... Quantization circuit 25. ······· Buffer memory 26 ······ Data amount calculation circuit 27 ······ Comparison circuit 28 ······ Correction value determination circuit 29 ..... Function generation circuit 30 ..... ROM

Continuation of the front page (56) References JP-A-61-201526 (JP, A) JP-A-63-201700 (JP, A) JP-A-3-263925 (JP, A) JP-A-3-263926 (JP) JP-A-4-104617 (JP, A) JP-A-4-104618 (JP, A) JP-A-4-104606 (JP, A) International publication 90/9064 (WO, A1) International publication 90 / 9022 (WO, A1) (58) Field surveyed (Int. Cl. ⁷ , DB name) H03M 7/30

Claims (1)

(57) [Claims] 1. A noise level that divides an input digital signal into a plurality of frequency bands, selects a wider bandwidth for a higher frequency band, and sets an allowable noise level for each band based on the energy of each band. Setting means, and a digital signal encoding device having 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 output of the noise level setting means, Detecting an output information amount of the quantization means, correcting an allowable noise level from the noise level setting means based on an error between the detected output and a target value and information on an equal loudness curve, and Digital signal encoding apparatus characterized in that: