JP2913696B2 - Digital signal encoding method - Google Patents

Description

Description: TECHNICAL FIELD The present invention relates to a digital signal encoding method for encoding 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 bandwidth for a higher frequency band, sets an allowable noise level for each band based on energy for each band, and In a digital signal encoding method for quantizing 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 the detected output and the target value By correcting the number of allocated bits at the time of quantization in accordance with the error of, and by stabilizing the amount of information in a predetermined period, the bit rate adjustment (bit packing)
The present invention provides a digital signal encoding method capable of performing the following.

[Conventional technology]

In high-efficiency coding of signals such as audio and voice, a bit allocator that divides an input signal such as audio and voice into a plurality of channels along a time axis or a frequency axis and adaptively allocates the number of bits for each channel. There is an encoding technique based on a shion (bit allocation). For example, coding techniques based on the above-mentioned bit allocation of audio signals and the like include band division coding (sub-band coding: SBC) in which an audio signal or the like on the time axis is divided into a plurality of frequency bands and encoded. A so-called adaptive transform coding (AT) that converts a signal on the time axis into a signal on the frequency axis (orthogonal transform), divides the signal into a plurality of frequency bands, and adaptively codes each band.
C) or the above-mentioned SBC and so-called adaptive prediction coding (AP
C), a so-called adaptive bit allocation (APC-AB) in which a signal on the time axis is band-divided, each band signal is converted into a baseband (low band), and then multi-order linear prediction analysis is performed to perform predictive coding. ).

Here, for example, in the above-mentioned band division coding, in order to increase the compression efficiency, while keeping the bit rate per unit time block constant, the number of bits to be given to each band divided band is represented by the time variation of the signal spectrum intensity. Is dynamically (adaptively) changed according to the Also,
In the above adaptive transform coding, the number of allocated bits is dynamically changed on the frequency axis.

[Problems to be solved by the invention]

In the high-efficiency coding that adaptively allocates the number of bits as described above, a unit block (a unit time block or a unit frequency block) depends on how the number of bits is allocated.
The bit rate per hit may not be constant, which may result in excess or deficiency of bits. In other words, for a given bit rate, there may be a phenomenon in which bits are left behind at low signal levels on the time axis or frequency axis, and bits are insufficient at high signal levels.

When there is such an excess or deficiency in the number of bits, it is necessary to use a bit rate adjustment (so-called bit packing) technique for effectively adjusting the number of bits.

This bit rate adjustment is to reduce the number of bits to be given to the unit block when bits remain in the unit block and to increase the number of bits to be given when bits are insufficient, thereby keeping the overall bit rate constant. It is to adjust. It is considered that the bit rate adjustment can be performed using, for example, a functional block as shown in FIG. In FIG. 9, for the input digital signal, the number of allocated bits at the time of quantization is determined by the allocated bit number determining function block 90, and the spectrum intensity of the unit block of the input digital signal is determined by the bit rate adjusting function block 91. Of the bit rate (bit packing) according to. After that, the encoding function block 92 performs encoding (requantization) using the number of bits determined by adjusting the bit rate, and outputs the result.

However, as the bit rate adjustment performed by the bit rate adjustment function block 92, there is still no effective method that is simple and signal deterioration is not conspicuous. The establishment of a method is desired.

Accordingly, the present invention has been proposed in view of the above-described circumstances, and provides a digital signal encoding method capable of performing a bit rate adjustment (bit packing) with a simple configuration in which signal degradation is not noticeable. It is intended to provide.

[Means for Solving the Problems] A digital signal encoding method according to the present invention has been proposed to solve the above-described problems, and divides an input digital signal into a plurality of frequency bands and has a high frequency band. A noise level setting step of selecting a wider bandwidth for each band and setting an allowable noise level for each band based on the energy of each band; and a difference between the energy of each band and the set allowable noise level. And a quantization step of quantizing the components of each band with the number of bits according to the level of the digital signal, wherein the amount of output information obtained by the quantization step is detected, and Depending on the error, the number of bits allocated in the quantization step is corrected so as to increase or decrease evenly in all bands, so that the information amount in a predetermined period is made constant. It is characterized by having made it.

[Operation] According to the present invention, the total bit rate is made constant by increasing or decreasing the number of bits allocated at the time of quantization by a fixed amount according to the error between the detection output of the output information amount of the quantization means and the target value. Is kept.

〔Example〕

Hereinafter, embodiments of the present invention will be described with reference to the drawings.

A digital signal encoding apparatus to which the digital signal encoding method of the present embodiment is applied, an input digital signal such as audio or voice, for example, band division encoding (SBC)
And adaptive transform coding (ATC), adaptive bit allocation (APC-
AB) etc. to perform high-efficiency coding. for that reason,
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, a so-called critical bandwidth (critical band) in consideration of human hearing characteristics described later.
Divides the input digital signal. Also, the first
As shown in the figure, a sum detection circuit 14 and a filter circuit 15 as 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. And a quantization circuit 24 that quantizes the components of each band with the number of bits assigned according to the energy of each band and the level of the difference between the noise level setting means. Here, the output information amount of the quantization circuit 24 is detected by a data amount calculation circuit 26 described later, and the detected output and a predetermined target value (a target value of a bit rate for adjusting a bit rate) from a terminal 3 are output. Is detected by an error detection circuit 27 described later. Thereafter, the error output is interrupted by the divider 28, so that the apparatus of the present embodiment corrects the number of bits allocated to the quantization circuit 24 in accordance with the error output so that the information amount in a predetermined period is kept constant. (Adjustment of bit rate).

By the way, as a result of adjusting the bit rate, for example, if the number of bits is insufficient and the bits are picked up (reduced), the S / N of the audio signal is degraded, and conversely, the number of bits is The S / N will be better if is increased. That is, by performing the bit rate adjustment, the shape of the noise spectrum with respect to the signal spectrum changes, so that good and bad S / N occur. For this reason, in the digital signal encoding device of the present embodiment, the shape of the noise spectrum generated at the time of requantization does not change by keeping the change of the S / N change of the audio signal on the frequency axis constant. Like that.

In other words, as shown in FIG.
On the other hand, the noise spectrum NS is indicated by a dashed line in the figure, but the dynamic bit allocation adjusts the bit allocation on the frequency axis when the bit rate is not constant in the unit block. It is.
For example, by performing a frequency analysis of the signal,
If the number of bits is taken up (decreased) as shown in FIG. 3 by taking up or increasing bits evenly from NS, the noise spectrum shown by the dotted line in FIG.
NS _- is corrected as, in the case of increasing the number of bits are corrected as noise spectrum NS _+.

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 a fast Fourier transform (FFT) on an audio signal, a voice signal, and the like, and converts a time-axis signal into a frequency axis.
The encoding (requantization) is performed.

In FIG. 1, for example, an audio signal is supplied to an input terminal 1, and the audio signal on the time axis is transmitted to a 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, and obtains an FFT coefficient including a real component value Re and an imaginary component value Im. These FFTs
The coefficient is transmitted to the amplitude / phase information generation circuit 12, where the real component value Re and the imaginary component value Im
Then, the amplitude value Am and the phase value are obtained from, and information on the amplitude value Am is output. In other words, general human hearing is sensitive to amplitude (power) in the frequency domain,
Since the phase is very insensitive, in the present embodiment, only the amplitude value Am is extracted from the output of the amplitude / phase information generating circuit 12, and this is used as the 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 dividing circuit 13. The band division circuit 13 divides the input digital signal represented by the amplitude value Am into a so-called critical bandwidth (critical band). The critical band is based on human auditory characteristics (frequency analysis ability), and is, for example, 0.
1616 kHz is divided into 24 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. Here, FIG. 4 shows the critical band. However, in FIG. 4, for simplicity of illustration, the number of the critical bands is represented by 12 bands (B _{1 to} B ₁₂ ).

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, respectively. This sum detection circuit
In 14, the energy for each band (spectral intensity in each band) is obtained by taking the sum of the amplitude values Am of each band (peak or average or the energy sum of the amplitude values Am). 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 as shown in FIG. 5, for example.

Here, in order to consider the influence of the bark spectrum SB on so-called masking, the bark spectrum SB
Is convolved with a function of a predetermined weight (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, as shown in FIG. 6, the delay element (z ^-1) for sequentially delaying input data from the input terminal _{100 101 1, 101 2 ‥‥ 101} m-2 ~101 m + 3 ‥‥ 101 ₂₃ , 101 ₂₄
(The example 24 delay elements corresponding to the critical band) and, for example, 24 multipliers for multiplying the filter coefficients (a function of the weighted) to the output from the delay elements 101 ₁ to 101 ₂₄
102 ₁ , 102 ₂ ‥‥ 102 _{m−3 to} 102 _{m + 3} ‥‥ 102 ₂₃ , 102 ₂₄ and the sum adder 104. At this time, in the multipliers 102 _{m−3 to} 102 _{m + 3} , for example, the multiplier 102 _m−3
Is multiplied by a filter coefficient of 0.0000086, the filter coefficient is 0.0019 by the multiplier 102 _m-2 , and the filter coefficient is 0.15 by the multiplier 102 _m-1.
The multiplier 102 the filter coefficient 1 at _m, a multiplier 102 the filter coefficients 0.4 _{m + 1,} filter coefficient 0.06 at multiplier 102 _{m + 2}
Is further multiplied by a filter coefficient of 0.007 with a multiplier 102 _{m + 3} to the output of each delay element, thereby obtaining the Bark spectrum.
SB convolution processing is performed. By this convolution processing, the sum of the parts indicated by the dotted lines in FIG. 5 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 noise is matched to a masked portion by 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 the subtractor 16. The subtracter 16 calculates a level α corresponding to an allowable noise level described later in the convolved region. The level α corresponding to the allowable noise level (allowable noise level) is, as described later,
By performing the inverse convolution processing, the level is such that the permissible noise level of each critical band is obtained. 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 is
Assuming that the number given in order from the low band of the critical band is i, the critical band can be obtained by Expression (1).

α = S− (n−ai) ‥‥‥‥ (1) In the equation (1), n and a are constants and a> 0, and S is the intensity of the convolution-processed bark spectrum. 1) (n-ai) in the equation is an allowable function. In the present embodiment, n = 38, a = 1, and 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 allowable noise spectrum in the convolved region. Therefore,
By performing the 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 the subtractor 19 via the synthesis circuit 18. Here, the output of the sum detection circuit 14, that is, the bark spectrum SB from the sum detection circuit 14 described above is supplied to the subtractor 19 via the delay circuit 21. Accordingly, by performing the subtraction operation of the masking spectrum and the bark spectrum SB in the subtractor 19, as shown in FIG. 7, the burspectrum SB has a level below the level indicated by each level of the masking spectrum MS. It will be masked.

The output of the subtracter 19 is supplied to the quantization circuit 24 via the ROM 20.
In the quantization circuit 24, the subtracter
Delay circuit with the number of bits allocated based on 19 outputs
The quantization of the amplitude value Am supplied via 23 is performed. In other words, in other words, the quantization circuit 24 quantizes the components of each band with the number of bits allocated according to the energy of each band of the critical band and the level of the difference between the noise level setting means. become. Note that 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
It is provided to delay the amplitude value Am in consideration of the delay amount in each circuit before M20. Further, the ROM 20 is provided for temporarily storing and sending out the output of the subtractor 19 every predetermined time at the time of quantization.

Here, at the time of quantization of the quantization circuit 24, as described above, the output information amount of the quantization circuit 24 is detected, and at the time of quantization according to the error between the detected output and the target value. By correcting the number of allocated bits, the amount of information in a predetermined period is made constant.

To do this, in the present embodiment, the data from the buffer 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 data amount and a predetermined target value (a target value of a bit rate for adjusting a bit rate) from the terminal 3, and the error data is transmitted to a division circuit 28. You. The division circuit 28
Then, the error data is interrupted by the total number of samples in a unit time block (one frame) as a predetermined period,
The division data is transmitted to the adder 31. Therefore, in the adder 31, by adding the division data to the output of the ROM 20, the number of allocated bits at the time of quantization is corrected (adjustment of the bit rate).
The amount of information during the predetermined period is made constant.
Here, since the output of the adder 31 is supplied to the quantization circuit 24 via the rounding circuit 32, a small change in the number of bits is corrected.

At the time of the synthesis by the synthesizing circuit 18, the so-called minimum audible curve (equal loudness curve) RC, which is a human auditory characteristic as shown in FIG. The data and the masking spectrum MS can be synthesized. Therefore, by synthesizing the minimum audible curve RC and the masking spectrum MS together, the allowable noise level can be reduced to the portion indicated by the oblique line in the figure, and the portion indicated by the chain line in the diagram at the time of quantization. Can be reduced. FIG. 8 is represented by the critical band shown in FIG.
SS is also shown.

However, in the present embodiment, a configuration may be adopted in which the above-described minimum audible curve synthesis processing 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 subtracter 16 is deconvolved by the divider 17 and immediately thereafter,
Will be transmitted.

That is, in the digital signal encoding apparatus of the present embodiment, when encoding the audio signal, the bit rate was adjusted as described above, and at this time, the number of bits was reduced because the shape of the noise spectrum did not change. Even in this case, there is little deterioration in hearing. Also, since the algorithm for adjusting the bit rate is easy, the hardware can be easily configured.

In the present invention, in addition to the so-called adaptive transform coding for processing an input digital signal by performing a fast Fourier transform as in the embodiment of FIG.
The present invention can be applied to an apparatus that performs C). In this case, the signal is band-divided by a band-pass filter or the like, and the number of bits allocated to each channel is increased or decreased by a certain amount in accordance with the error between the output output of the quantizer and the target value. In the case of the band division coding, the same effect as described above can be obtained.

[Effects of the Invention] According to the digital signal encoding method of the present invention, the amount of output information after quantization is detected, an error between the detected output and a target value is detected, and in accordance with this detection, The number of bits allocated in the quantization step is corrected so as to increase or decrease evenly in all bands, so that the information amount in a predetermined period is made constant, so that the amount of calculation is small and despite the simple configuration, In addition, it is possible to perform bit rate adjustment (bit packing) in which signal deterioration is not conspicuous. Therefore, for example, even if the number of bits for quantization is reduced, deterioration in audibility can be reduced.

[Brief description of the drawings]

FIG. 1 is a block circuit diagram showing a schematic configuration of a digital signal encoding apparatus to which a digital signal encoding method according to one embodiment of the present invention is applied, and FIG. 2 is a characteristic diagram showing a noise spectrum before bit rate adjustment. 3 is a characteristic diagram showing a noise spectrum after bit rate adjustment, FIG. 4 is a diagram showing a critical band, FIG. 5 is a diagram showing a bark spectrum, FIG. 6 is a circuit diagram showing a filter circuit, and FIG. The figure shows a masking spectrum, FIG. 8 is a view in which a minimum audible curve and a masking spectrum are combined, and FIG. 9 is a functional block diagram for adjusting a bit rate. 11 Fast Fourier transform circuit 12 Amplitude / phase information generation circuit 13 Band division circuit 14 Sum detection circuit 15 Filter circuit 16 Subtractor 17, 28 Divider 18 Synthesis Circuit 19 Subtractor 20 ROM 21, 23 Delay circuit 22 Minimum audible curve generation circuit 24 Quantization circuit 25 Buffer memory 26 Data amount calculation circuit 27 Error detection circuit 29 …… Function generation circuit 31 …… Adder 32 …… Rounding circuit

Continuation of the front page (56) References JP-A-3-117920 (JP, A) JP-A-60-96041 (JP, A) JP-A-62-57621 (JP, A) JP-A-61-110200 (JP, A) , A) Proceedings. IEEE International Conferencing on Acoustics, Speech and Signal Processing (1988) Vol. 5, p. 2524-2527 Proceedings. IEEE International Conferencing on Acoustics, Speech and Signal Processing (1989) Vol. 3, p. 1993-1996 Proceedings of the National Meeting of the Institute of Electronics, Communications and Communication Engineers, 1984-1984 [Volume 1] 1-389-390

Claims (1)

(57) [Claims] 1. A noise level for dividing an input digital signal into a plurality of frequency bands, selecting a wider bandwidth for a higher frequency band, and setting an allowable noise level for each band based on the energy of each band. A digital signal encoding method, comprising: a setting step; and a quantization step of quantizing components of each band with a number of bits corresponding to a level of a difference between the energy of each band and the set allowable noise level. Detecting the amount of output information obtained in the quantization step, detecting an error between the detected output and a target value, and calculating the number of bits allocated in the quantization step in accordance with the detected error. A digital signal encoding method, wherein a correction is performed so as to increase and decrease evenly in a band, and an information amount in a predetermined period is made constant.