JP2005198280A - Digital signal processing system - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide a digital signal processing system for suppressing clip processing to reduce influences of return noise when processing encoded music signals by a processor such as a DSP. <P>SOLUTION: The digital signal processing system wherein a digital signal exceeding a specified scale is converted as a signal within the specified scale by signal processing, comprises steps for: (1) storing a set of a plurality of signal conversion factors (a), (b) in a memory 33; (2) calculating the sum of a gain (g) which is variably set by a parameter input device 20, and a volume level setting value att; (3) reading the set of signal conversion factors (a), (b) out of signal conversion factors stored in the memory 33 on the basis of the sum of the gain (g) and the volume level setting value att; and (4) obtaining an output signal Y by using a DSP 40 to convert the input signal according to a conversion specified by the read signal conversion factors. <P>COPYRIGHT: (C)2005,JPO&NCIPI

Description

  The present invention relates to a digital signal processing system, and particularly, in an audio apparatus, a code (digital signal) exceeding a specified scale (for example, 16 bits (bits)) enters a specified scale by signal processing such as equalizing. The present invention relates to a signal processing method.

  (1) When a music signal encoded (for example, digitized as a 16-bit binary code) is processed by a processor (Processoer) such as a DSP (Digital Signal Processor), the register length is the input signal in the processor. Since it is sufficiently longer than the code length, there is no problem even if the code length in the register exceeds the code length of the input signal (for example, 16 bits) due to processing. For this reason, it must be within a specified scale (for example, 16 bits). For this reason, conventionally, a so-called clip process is performed in which a portion exceeding the specified scale length is forced to become full scale.

  (2) When the encoded signal is boosted by filtering or the like, a signal close to full scale may exceed the full scale by filtering. However, in the processor, as in the case of (1) described above, there is no problem if the code length in the register exceeds the code length of the input signal due to processing, but it is specified at the time of output. The clip processing was done because it had to fit within the scale.

  FIG. 5 is an explanatory diagram of clip processing when the scale length is 8 bits, and FIGS. 5A and 5B are examples of codes in the register before and after the input signal is processed. a) shows an example of the code in the register before the clip process, and (b) shows an example of the code in the register after the clip process. In FIG. 5A, the codes “00101111” to “01111111” in the register are “01011111” to “1111111” in 8 bits and do not exceed the specified scale length (8 bits). As shown in FIG. 5B, the codes are “00101111” to “01111111”, and the codes in the register are not converted before and after the clipping process. However, in FIG. 5A, the codes “10111100” to “10111100” Since “101011000” exceeds the specified scale length (8 bits) and is indicated by 9 bits, the code in the register is converted to full scale “1111111” for 8 bits by clip processing. Therefore, the registers are “01111111” to “01111111” as shown in FIG.

  However, in the case of (1) and (2), in the clip processing method, as shown in FIG. 6 (input / output signal waveform and its spectrum), harmonics having the spectrum of the input signal as a fundamental wave are formed (FIG. 6). (See (b ')). In addition, as the clipped portion increases, the harmonic component becomes larger and becomes a wide range (extends to a high frequency range). Among the components, the frequency component higher than 1/2 of the sampling frequency becomes aliasing noise by D / A conversion, and the sound quality is deteriorated. (In FIG. 6, (a) is the input signal waveform before clip processing, (b) is the input signal waveform after clip processing, and (a ') is the spectrum of the input signal waveform before clip processing.) (B ′) is the spectrum of the input signal waveform after the clip processing). In the case of (2), there is a method of simply attenuating the signal level of the output signal according to the gain of the equalizer as shown in FIG. As a result, there was a problem that the volume was lowered.

  The present invention provides (a) a digital signal processing method that suppresses clip processing when an encoded music signal is processed by a processor such as a DSP and reduces the influence of aliasing noise, and (b) An object of the present invention is to provide a digital signal processing method that can suppress a change in volume feeling as much as possible without performing a clipping process even if an encoded signal exceeds a scale defined by a filter process.

In order to solve the above problems, a digital signal processing system according to the present invention is a digital signal processing system for converting a digital signal exceeding a specified scale as a signal within the specified scale. , (2), (3) and (4).
(1) A plurality of sets of signal conversion coefficients are stored in the memory in advance.
(2) The sum of the variably set signal gain and the set volume level setting value is obtained.
(3) Based on the sum of the signal gain and the volume level setting value obtained in step (2), a set of signal conversions from among a plurality of sets of signal conversion coefficients stored in the memory in step (1) Read the coefficient.
(4) The input signal is converted by a predetermined conversion formula defined by the set of signal conversion coefficients read in step (3), and the output signal Y is obtained.

In the digital signal processing system of the present invention, step (4) is a step of obtaining an output signal Y by the following conversion equation (A) based on the read signal conversion coefficients a and b and the input signal X. Also good.
Y = X · (ab− | X |) (A)

  According to the digital signal processing system of the present invention, without performing clipping processing, a signal conversion coefficient is obtained based on a variably set signal gain and volume level setting value, and the signal conversion coefficient and the input signal are obtained. Based on this, an output signal Y is obtained by a predetermined conversion formula.

  As a result, when the amount of boost in the filter is canceled by the volume level setting value that regulates the output level, the output signal is equal to the input signal, so there is no distortion of the output signal or lowering of the level. Even if the amount is not canceled, the clipping process is not performed, so that the output signal is not significantly distorted, and the volume is not greatly deteriorated as compared with the case where the output signal is simply attenuated according to the gain of the equalizer.

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

  FIG. 1 is a block diagram showing an outline of a digital signal processing system according to the present invention. Signal processing is roughly divided into a volume level set value calculation process 100 for calculating a volume level and reading a code (signal) processing coefficient. A code (signal) conversion process 200 that performs code conversion of the input code (signal) X based on the code processing coefficient obtained to obtain an output code Y.

  FIG. 2 is a block diagram showing the configuration of an embodiment of the apparatus 50 to which the digital signal processing system of the present invention is applied. The apparatus 50 is used for inputting parameters (center frequency f0, Q (sharpness), gain). The parameter input device 20, the microcomputer 30, and the DSP 40 are provided, and the microcomputer 30 shares the sound volume level setting value calculation process 100 and the DSP 40 shares the code conversion process 200.

FIG. 3 is a block diagram for explaining processing by the DSP 40, and FIG. 4 is a diagram showing an example of input / output code conversion characteristics in the digital signal processing system of the present invention.
First, processing by the microcomputer 30 will be described with reference to FIGS. First, a set of code processing coefficients a and b is stored in the memory 33 in advance. The gain g set via the parameter input device 20 and the currently set volume level set value att are added to obtain g + att, and the coefficients a and b stored in the memory 33 are calculated based on the calculation result. Read and transfer to DSP 40. Here, the coefficients a and b correspond to the coefficients in the expression (2) which is the basis of the code conversion in the DSP 40 described later. When (g + att) <0, a = 1.0, b = On the other hand, when 0 and (g + att)> 0, as shown in FIG. 4, a coefficient that becomes a curve having a curve that is compressed by the maximum (g + att) is set. The currently set parameters (center frequency f0, sharpness Q, gain) are rewritten to new values on the memory 33 of the microcomputer 30 each time they are changed.

Next, processing in the DSP 40 will be described with reference to FIG. The conversion formula (A) that is the basis of the code conversion processing performed by the DSP 40 is shown below.
Y = X · (ab− | X |) (A)
Here, X is an input code, Y is an output code, and a and b are parameters.
The coefficients a and b transferred from the microcomputer 30 are stored in the coefficient memory 33 and used for the following processing.

  First, the input code X is transferred to the calculator 45, and the absolute value | X | is obtained. Next, the coefficient b is read from the coefficient memory 47 and transferred to the multiplier 46 to obtain b · | X | and transferred to the computing unit 45. Further, the coefficient a is read from the coefficient memory 47 and transferred to the computing unit 45 to obtain a−b · | X |. Finally, the result of ab · | X | and the input code X are transferred to the multiplier 46, and the obtained value X · (ab · | X |) is output as an output code.

  As a result, when the boost amount g in the filter is canceled by the volume level set value att that defines the output level, the output signal is equal to the input signal, so that the output signal is not distorted or the level is not lowered. On the other hand, even when the boost amount g in the filter is not canceled ((g + att)> 0), since the clipping process is not performed, the output signal is not significantly distorted, and the output signal is simply attenuated according to the gain of the equalizer. Compared with the case of making it, a sound volume does not deteriorate greatly.

It is a block diagram which shows the outline | summary of the digital signal processing system of this invention. It is a block diagram which shows the structure of one Example of the apparatus to which the digital signal processing system of this invention is applied. It is a block diagram for demonstrating the process by DSP. It is a figure which shows the example of the input-output code conversion characteristic in the digital signal processing system of this invention. It is explanatory drawing of a prior art example (clip process). It is a figure which shows the input-output signal waveform by the prior art example (clip process), and its spectrum. It is explanatory drawing of the prior art example (the method of attenuating a signal level simply according to the gain of an equalizer).

Explanation of symbols

20 Parameter input device 30 Microcomputer 33 Memory 40 DSP
45 arithmetic unit 46 multiplier 47 coefficient memory 50 device

Claims (2)

A digital signal processing system for converting a digital signal exceeding a specified scale by signal processing as a signal within the specified scale, wherein the following (1), (2), (3) and (4) A digital signal processing system comprising steps.
(1) A plurality of sets of signal conversion coefficients are stored in the memory in advance.
(2) The sum of the variably set signal gain and the set volume level setting value is obtained.
(3) Based on the sum of the signal gain and the volume level setting value obtained in step (2), a set of signal conversions from among a plurality of sets of signal conversion coefficients stored in the memory in step (1) Read the coefficient.
(4) The input signal is converted by a predetermined conversion formula defined by the set of signal conversion coefficients read in step (3), and the output signal Y is obtained. The digital signal processing system according to claim 1,
A digital signal processing method, wherein step (4) is a step of obtaining an output signal Y by the following conversion equation (A) based on the read signal conversion coefficients a and b and the input signal X.
Y = X · (ab− | X |) (A)

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