JPH10336785A - Audio signal processor, processing method for the same and recording medium for recording program by the same method - Google Patents

Abstract

PROBLEM TO BE SOLVED: To attain simplification or the like of a circuit scale in an audio signal processor that corrects the audio signal by reverse filter characteristics. SOLUTION: This device consists of an analog/digital converter 1, a correction filter 20, digital/analog converter 3, a memory circuit for correction coefficient reservation 5, a control processing unit(CPU) 6 and the like. In a signal processing in the correction filter 20, a circuit scale of an FIR type of filters is reduced and at the same time a high sound quality is gained by dividing an incoming signal into a fixed band and making the FIR filter, which convolutes, correct a phase and an amplitude so that it does not correct a frequency below the fixed band.

Description

DETAILED DESCRIPTION OF THE INVENTION

[0001]

BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to an audio signal processing apparatus used by being connected to a speaker, a signal processing method thereof, and a recorded medium on which the processing method is recorded. And technology for reproducing higher fidelity sound quality.

[0002]

2. Description of the Related Art FIG. 9 is a schematic block diagram showing a state in which a conventional audio signal processing apparatus is connected to a speaker and used, and is shown as a first conventional example. In the figure, an analog audio signal is converted into a digital signal by an analog / digital conversion means (A / D) 1, and the digital signal is inverted in a correction filter 2 in accordance with the response characteristic of a speaker SP connected to a subsequent stage. It is to be corrected.

As shown in FIG. 10, the correction filter 2 includes a plurality of cascaded delay elements d1, d2,.
And the input digital signal directly or by the delay elements d1, d
.., M1200 for multiplying the signal output from the... D1199 by a predetermined coefficient, and an adder a for adding the output signals of the multipliers m1, m2,. Is configured as an FIR filter that performs a convolution operation as a convolver.

In this convolution operation, a predetermined coefficient is read out from a coefficient holding circuit (not shown) through a control circuit (not shown) and supplied to the multipliers m1, m2,. Is performed. As a result, the phase and amplitude characteristics of the incoming signal are corrected, and the signal is converted into an analog signal again by the digital / analog converting means (D / A) 3 and then passed through the amplifying means 4 to the speaker SP. The sound is emitted from

FIG. 11 shows a second conventional example, in which an IIR filter for correcting extreme dips and peaks is cascaded on the subsequent stage of the FIR filter to form a correction filter 2A. In the same manner as in the above-described conventional example, the former stage includes delay elements d1, d2,.
, M1200, and an IIR filter on the subsequent stage is a delay element d1,
d2,... d4, multipliers m1, m2,.
It is composed of

[0006]

By the way, the above-mentioned first method is used.
In both the second prior art and the second conventional example, the FIR filter which is a component of the correction filter is provided with 1200 multipliers (1200 taps) for a sampling frequency of 44.1 kHz in order to secure the resolution on the low frequency side. The convolution operation is performed.

For example, FIG. 12 is a waveform characteristic diagram when an input signal is corrected by the device shown as the first conventional example.
The waveform characteristic has a relatively good resolution up to the low frequency range.

However, the above configuration requires a multiplier of 1200 taps, and if this multiplier is constituted by a DSP (digital signal processor), about five DSPs are required.
However, there is a problem that the circuit scale becomes large and the cost becomes high.

Therefore, it is conceivable to simplify the circuit scale by reducing the number of taps. For example, if the number of taps is reduced to about 200 so as to be accommodated by about one DSP, FIG.
The waveform characteristics shown in FIG. As is clear from the figure, there is a problem that the resolution is deteriorated particularly in a relatively low frequency band of 100 Hz to 700 Hz, as compared with the case where the above-mentioned configuration is made up of 1200 taps (see FIG. 12).

The present invention has been made in view of such a problem, and it is an object of the present invention to prevent degradation of resolution in a low frequency range and to further simplify a circuit scale to reduce costs.

[0011]

The present invention, in order to achieve the above object, comprises the following means (1) to (7). That is,

(1) An audio signal processing apparatus for converting an incoming analog audio signal into a digital signal and correcting the converted digital audio signal in accordance with the response characteristics of a connected speaker in a predetermined frequency band. , Which is supplied with one output signal of the converted digital audio signal and operates as a convolver
First filter means constituted by an R-type filter;
A second filter means provided before or after the first filter means for limiting a predetermined band of the digital audio signal, and the other output signal of the converted digital audio signal is supplied; A delay means for delaying a delay time substantially equal to the delay time of the first filter means, and a signal provided in a preceding or subsequent stage of the delay means for limiting a signal in a band lower than the band of the second filter means. Audio signal processing, comprising: third filter means for adding the output signal of the third filter means or delay means and the output signal of the first or second filter means. apparatus.

(2) An audio signal processing apparatus for converting an incoming analog audio signal into a digital signal and correcting the converted digital audio signal in accordance with the response characteristics of a connected speaker in a predetermined frequency band. A first filter means provided with the converted digital audio signal and configured as an FIR filter operating as a convolver; and a digital audio signal provided before or after the first filter means. A second filter means for limiting a predetermined band of the signal, and a signal delayed by an FIR filter constituting the first filter means are supplied to the second filter means for filtering the delayed signal. Third filter means for limiting a band lower than the band, the third filter means and the first or second Adding means for adding the output signal of the filter means, an audio signal processing apparatus characterized by further comprising.

(3) In the audio signal processing device according to claim 1 or 2, the second and third filter means are I
An audio signal processing device comprising an IR filter.

(4) An audio signal processing method for converting an incoming analog audio signal into a digital signal and correcting the converted digital audio signal in accordance with a response characteristic of a connected speaker in a predetermined frequency band. And converting the converted digital audio signal into F
A first filter operation step for performing an IR-type filter operation, a second filter operation step for limiting a predetermined band of the digital audio signal before or after the first filter operation step, and A delay processing step of delaying the digital audio signal by substantially the same time as the delay time of the signal in the first filter calculation step; and performing the second filter calculation before or after the delay processing step. A third filter operation step for limiting a signal in a band lower than the band obtained, a signal obtained in the third filter operation step or the delay processing step, and a signal obtained in the first or second filter operation step. Processing step of adding an audio signal to the audio signal .

(5) An audio signal processing method for converting an incoming analog audio signal into a digital signal and correcting the converted digital audio signal in accordance with a response characteristic of a connected speaker in a predetermined frequency band. And converting the converted digital audio signal into F
A first filter operation step for performing an IR filter operation, a second filter operation step for limiting a predetermined band of the digital audio signal before or after the first filter operation processing step, and the second filter When the calculation step is a step performed after the first filter calculation step, a signal in a band lower than a band obtained in the second filter calculation step is limited before or after the second filter calculation step. A third filter calculation step to perform the addition, and an addition processing step for adding the signal obtained in the third filter calculation step and the signal obtained in the first or second filter calculation step. How to process audio signals.

(6) The audio signal processing method according to claim 4 or 5, wherein the second and third filter operations are performed by IIR type filter operation processing.

(7) A recorded medium characterized by recording the audio signal processing method according to claim 4, 5 or 6 as a processing program.

[0019]

DESCRIPTION OF THE PREFERRED EMBODIMENTS Embodiments of the present invention will be described with reference to preferred embodiments. FIG. 1 is a basic block diagram when the audio signal processing device of the present invention is applied to a speaker. The left and right channels have the same configuration, and the channels on one side are omitted. In FIG. 1, the audio signal processing apparatus includes an analog-to-digital converter (A / D) 1 for converting an analog audio signal into a digital signal, a correction filter 20 which is a main configuration of the present invention described in detail later, A digital-to-analog converter (D / D) for converting a digital signal to an analog signal.
A) 3. A correction coefficient is determined in advance so that the sound emitted from the speaker becomes an ideal sound in the anechoic chamber, and a memory circuit 5 for holding this data as a correction coefficient is read out. This is an embodiment in which a control circuit (CPU) 6 for supplying a correction coefficient to the filter 20 and an amplifier 4 are schematically configured, and this device is connected to a speaker SP.

The device applied in this manner is
It operates roughly as follows. An analog audio signal supplied from a sound source (not shown) enters the analog / digital converter 1, is converted into a digital signal, and is supplied to the correction filter 20. At the same time, the correction coefficient determined in advance is read from the memory circuit 5 by the control circuit 6 and supplied to the correction filter 20, and based on these coefficients, the incoming digital signal is displayed on the time axis. Convolution operation. As a result, the phase, the amplitude, and the like are corrected, and the corrected signal is returned to the analog signal again in the digital-to-analog converter 3, amplified by the amplifier 4, and then corrected by the speaker SP so as to become an ideal sound. Signal is emitted.

The apparatus according to the present embodiment operates as described above in general. In particular, the apparatus has a feature in the configuration of the correction filter 20, which will be described in detail with reference to FIG. FIG. 2 is a detailed block diagram of the correction filter 20. The correction filter 20 includes an FIR type filter 20-1,
It comprises an IIR filter 20-2 for a high-pass filter, an IIR filter 20-5 for a low-pass filter, a memory holding circuit 20-4 for delay, and an adder 20-3.

The FIR filter 20-1 is
199 cascaded delay elements d1, d2,.
And 200 multipliers m1, m2, 200 for multiplying the input digital signal directly or by multiplying the signal output from each of the delay elements d1, d2,... D199 by a predetermined correction coefficient supplied from the memory circuit 5. .. M200 and an adding means a for adding these output signals.
The IR filter 20-2 includes delay elements d1, d2,.
.., M5 provided to perform multiplication with a coefficient supplied from the memory circuit 5 so as to obtain a high-pass filter characteristic, and an adder a.

On the other hand, delay memory holding means 20-4
The IIR filter 20-5 provided at the subsequent stage is provided to multiply the delay elements d1, d2,... D4 by the coefficient supplied from the memory circuit 5 so as to obtain the low-pass filter characteristic. , M5, and an adder a.

One of the signals input to the correction filter 20 configured as described above is subjected to a convolution operation by the FIR filter 20-1 using the correction coefficient supplied from the memory circuit 5 through the CPU 6. The signal corrected by the convolution operation is output to the IIR filter 20 in the next stage.
In -2, frequencies below a predetermined value, for example, 500 Hz or less are cut. The IIR filter 20-2 operates as a high-pass filter after a predetermined coefficient is supplied from the memory circuit 5 to each multiplier, and this output signal is supplied to the next-stage adder 20-3. ing.

On the other hand, the signal input to the memory holding circuit 20-4 is delayed by a considerable amount to be delayed by each delay element in the FIR filter 20-1, and the output signal is delayed by the IIR filter 20 -5 supplied. This IIR type filter 20-5
In this case, the frequency of 500 Hz or more is cut this time so as to correspond to the IIR filter 20-2 operating as the high-pass filter.

The output and the IIR filter
The output from 20-2 is added in the adder 20-3, and as a result, there is little drop in resolution on the low frequency side as shown in FIG. Equivalent frequency characteristics will be obtained.

That is, in the correction filter 20, in order to simplify the circuit scale, the number of taps of the FIR filter 20-1 is reduced to 200 taps. The band in which the drop is expected is not corrected by the FIR filter 20-1, thereby preventing the band.

Next, as another embodiment of the above-mentioned correction filter, a correction filter 20A having a configuration shown in FIG. 4 can be used. In the configuration shown in the figure, the memory circuit 20-4 is omitted by sharing each delay element of the FIR filter 20-1.

Further, any one of the correction filters 20,
20A also has a configuration in which the IIR filter 20-2 is provided on the subsequent stage side of the FIR filter 20-1, but is not limited thereto.
A configuration may be provided before the filter 20-1. Further, the IIR type filter 20-5 in FIG. 2 may be provided at a stage preceding the memory holding circuit 20-4.

Next, a predetermined program for performing the above-described signal processing is recorded on a recording medium such as a CD-ROM, and the disk drive 40 or the network terminal 41 is used by using the recorded recording medium. Via a personal computer (hereinafter referred to as PC or personal computer)
An embodiment in which signal processing is performed on the side will be described with reference to FIG.

For example, the personal computer 42 used here
Is equipped with MMX (Extended instruction set of P55C by INTEL: an instruction set for specific applications and added mainly for efficiently handling digital signal processing such as image and audio), and includes program data and 1 RAM 42a for temporarily storing PCM data for each sample
-1 and a C having a well-known operation processing unit 42a-2 composed of a multiplier and an accumulator for performing an FIR filter operation, a high-pass / low-pass filter operation, and the like.
A PU (corresponding to a controller) 42a, a RAM 42b for temporarily storing, for example, one frame of PCM data for data processing, and a data converter for converting data input via the disk drive 40 or the network terminal 41 42c, and an audio interface 42d for supplying the processed signal to the speaker SP.

The network terminal 41 has a TCP /
IP (Transmission Control Protocol / Internet P
Data is transmitted or received in packet units using a protocol called "rotocol". This network is known as the Internet. A host is specified by adding a header including an address to the packet.

Next, the operation in the configuration shown in FIG.
This will be described with reference to the flowchart shown in FIG. FIG. 6 shows a case where a program loading command (command) is input from a keyboard (not shown) in a state where the disk on which the signal processing program is recorded is set in the disk drive 40.

When starting in this state, step S1
Indicates that the program is loaded (when a command is input, it is determined that the program is loaded). The program data is read out and read from the RAM 42 in the CPU 42a.
a-1 (step S2), and when the program load is completed, a program load flag is set (step S3) and the process is terminated.

Next, a play command is input from a keyboard (not shown) in a state where a disk on which an audio signal to be corrected is recorded is set in the disk drive 40, and when starting in this state, it is determined in step S1 that the program is not a program. When it is determined (NO in step S1), the process proceeds to step S4, in which, for example, the TOC area is accessed to read a code indicating the type of the disc, and it is determined that the disc has an audio signal recorded (audio source). If YES, the data is read from the data recording area in step 5 to perform signal correction processing, the processed data is set in the audio interface 42d (step S6), and the process returns to step S5 to repeat the above signal processing. . Audio interface 4
2d converts the received data into D at a predetermined sampling cycle.
The analog signal is converted into an analog signal and supplied to the speaker SP. If "NO" in the step S4, the display indicates that the performance cannot be performed (step S7), and ends.

Next, a signal processing program for performing the correction processing in step S5 described above will be described in detail with reference to a flowchart of FIG. 7 for a program corresponding to FIG. First, in step 51, a PCM audio signal is read from the data area and input into the personal computer. The input PCM audio signal is temporarily stored in RAM4.
2b (step S52). Then, the signal already input to the RAM 42b becomes C
It is supplied to the RAM 42a-1 in the PU 42a and is delayed for a predetermined time (step S53). Then, the signal delayed here is supplied to the arithmetic processing unit 42a-2,
The same filter calculation processing as that performed by the low-pass filter is performed (step S54).

On the other hand, the signal for each sample from the RAM 42b is directly supplied to the arithmetic processing section 42a-1 (step S55), and the same filter arithmetic processing as that performed by the FIR filter in FIG. 2 is performed (step S55). Step S5
6). Further, thereafter, a high-pass filter calculation process is performed (step S57).

Then, in step S58, the result of this operation and the result of the low-pass filter operation in step S54 are added (step S58), and finally output as a corrected signal (step S5).
9).

In the above processing steps, step S5
3 and a series of steps S54 and S
The processing order of the series of steps from 55 to S57 may be changed, and in this case, the high-pass filter calculation processing may be performed before the FIR filter calculation processing.

Next, FIG. 8 shows a processing program corresponding to FIG. 4, and in this case, step S5 in FIG.
The processing step 3 is omitted, and is incorporated in a part of the subsequent FIR filter operation processing.
4 is a processing program in which the low-pass filter calculation processing of No. 4 is performed after the FIR filter calculation processing.

Also in this case, the order of the high-pass operation in step S66 and the low-pass filter operation in step S65 in FIG. 8 may be interchanged.
Alternatively, the high-pass calculation processing in step S66 may be performed before step S63.

As described above, according to each embodiment of the present invention, the circuit configuration can be simplified, good frequency characteristics can be corrected, and high-fidelity sound quality can be obtained.

[0043]

According to the present invention, the circuit configuration can be simplified.
In addition, good correction of frequency characteristics is possible, and high-fidelity sound quality can be obtained. In particular, according to the invention of claim 2, the third
Since the signal supplied to the filter means is obtained through the FIR type filter, the circuit scale can be further simplified.

[Brief description of the drawings]

FIG. 1 is a block diagram illustrating a basic configuration of an audio signal processing device according to an embodiment of the present invention.

FIG. 2 shows a first example of a correction filter which is a main component of the present invention.
This is an embodiment of the invention.

FIG. 3 is a waveform chart showing a correction characteristic according to the embodiment.

FIG. 4 is a diagram illustrating a second example of a correction filter that is a main component of the present invention.
This is an embodiment of the invention.

FIG. 5 is a system diagram of a personal computer connected via a disk drive or a network terminal.

FIG. 6 is a control flow in a personal computer.

FIG. 7 is a flowchart of a signal processing executed in the personal computer and corresponding to FIG. 2;

FIG. 8 is one example of a signal processing flow executed in the personal computer and corresponding to FIG. 4;

FIG. 9 is a block diagram showing a basic configuration of a conventional audio signal processing device.

FIG. 10 is a first conventional example of a conventional correction filter.

FIG. 11 is a second conventional example of a conventional correction filter.

FIG. 12 is a characteristic diagram when the correction filter is configured with 1200 taps.

FIG. 13 is a characteristic diagram when the correction filter is configured with 200 taps.

[Explanation of symbols]

 Reference Signs List 1 analog / digital converter (A / D) 3 digital / analog converter (D / A) 4 amplifier 4 5 memory circuit 5 6 control circuit (CPU) 20 correction filter 2 20-1 FIR filter 20-2, 20- 5 IIR filter 20-3 Adder 20-4 Memory holding circuit SP Speaker a Adder d1, d2, ... d199 Delay element m1, m2, ... m200 Multiplier

Claims (7)

[Claims] An audio signal processing apparatus for converting an incoming analog audio signal into a digital signal and correcting the converted digital audio signal according to a response characteristic of a connected speaker in a predetermined frequency band. One output signal of the converted digital audio signal is supplied, and the first filter means is configured by an FIR filter that operates as a convolver. The first filter means is provided on the front or rear side of the first filter means. A second filter means for limiting a predetermined band of the digital audio signal, and the other output signal of the converted digital audio signal is supplied, and the delay time is substantially the same as the delay time of the first filter means. And a delay means provided before or after the delay means.
Third filter means for limiting a signal in a band lower than the band of the filter means, and addition means for adding an output signal of the third filter means or the delay means and the output signal of the first or second filter means. And an audio signal processing device. 2. An audio signal processing apparatus for converting an incoming analog audio signal into a digital signal and correcting the converted digital audio signal according to a response characteristic of a connected speaker in a predetermined frequency band. The converted digital audio signal is supplied,
First filter means constituted by an FIR filter operating as a convolver; and second filter means provided before or after the first filter means for limiting a predetermined band of the digital audio signal. And a third filter means for receiving a signal delayed by an FIR filter constituting the first filter means and limiting a band of the delayed signal lower than a band of the second filter means. An audio signal processing apparatus, comprising: an adder for adding the output signals of the third filter and the first or second filter. 3. An audio signal processing apparatus according to claim 1, wherein said second and third filter means are constituted by IIR filters. 4. An audio signal processing method for converting an incoming analog audio signal into a digital signal and correcting the converted digital audio signal according to response characteristics of a connected speaker in a predetermined frequency band. A first filter operation step of performing an FIR-type filter operation on the converted digital audio signal; and a second step for limiting a predetermined band of the digital audio signal before or after the first filter operation step. (2) a filter operation step, a delay processing step for delaying the converted digital audio signal by substantially the same time as the signal delay time of the first filter operation step, and before or after the delay processing step , A signal in a band lower than the band obtained by the second filter operation. A third filter calculation step for limiting; a signal obtained in the third filter calculation step or the delay processing step; and a signal obtained in the first or second filter calculation step. An audio signal processing method, comprising: an addition processing step. 5. An audio signal processing method for converting an incoming analog audio signal into a digital signal and correcting the converted digital audio signal according to response characteristics of a connected speaker in a predetermined frequency band. A first filter operation step of performing an FIR filter operation on the converted digital audio signal; and a second step of limiting a predetermined band of the digital audio signal before or after the first filter operation processing step. When the second filter operation step is a step performed after the first filter operation step, the second
A third filter operation step of restricting a signal in a band lower than the band obtained in the second filter operation step before or after the filter operation step of (a), a third filter operation step, and the first or second An addition processing step for adding the signal obtained in the filter operation step of (1). 6. The audio signal processing method according to claim 4, wherein the second and third filter operations are performed by IIR.
A method for processing an audio signal, wherein the method is performed by a type filter operation process. 7. A recorded medium characterized by recording the audio signal processing method according to claim 4, 5 or 6 as a processing program.