JP2023119438A - Sound signal processing device - Google Patents

Abstract

To obtain a sound signal processing device capable of reproducing original sound with higher fidelity.SOLUTION: A sound signal processing device 10 includes: a power amplifier 12 for amplifying an input sound signal so as to output a sound signal to be input to a speaker 20; a pre-stage digital filter 14 which is arranged at a front stage of the power amplifier 12 and where an inverse filter with respect to a vibration response waveform on a paper surface of a speaker 20 when a predetermined sound signal is input to the power amplifier 12 is defined as a coefficient; and a post-stage digital filter 16 which is arranged in a post stage of the power amplifier 12 and where an inverse filter with respect to a sound pressure response waveform by the speaker 20 when a predetermined sound signal is input to the pre-stage digital filter 14 while the pre-stage digital filter 14 is arranged is defined as a coefficient.SELECTED DRAWING: Figure 1

Description

The present invention relates to an audio signal processing device.

Conventionally, when constructing an electroacoustic system, emphasis has been placed on frequency characteristics, and efforts have been made to reproduce bass to treble sound evenly and to reduce distortion. However, from the viewpoint of "faithfully reproducing the original sound," the transient characteristics of the speaker (the ability of the speaker to follow the input signal) are important, and it is necessary to consider these transient characteristics when constructing an electroacoustic system. .

FIG. 5 shows a side sectional view showing an example of the structure of a conventional speaker. Also, FIG. 6 shows a waveform diagram showing an example of a sound pressure waveform (here, measured at a point of 1 m in front of the speaker) when an impulse signal is input to the speaker.

Referring to the waveform diagram of FIG. 6, it can be seen that the leading impulse-like waveform is followed by a gently attenuating waveform. This tail-shaped waveform is due to the inertia of the unit shown in FIG. 5 as an example (the property of continuing to vibrate even after the signal is interrupted). As measures for improving such transient characteristics, there are the following two methods.

The first is a passive method, such as reviewing the members of each part of the unit such as cone paper, edges, and dampers, and optimizing the shape, dimensions, and materials of the enclosure. Secondly, as shown in FIG. 7 as an example, it is an active method in which the original sound is set as a target signal and the output signal follows it.

That is, Patent Literature 1 discloses a loudspeaker broadband control device intended to reproduce sound with high fidelity.

This loudspeaker broadband control device includes a band dividing section that divides an input signal into a low frequency component and a high frequency component with a predetermined frequency as a boundary; and a second buffer memory for accumulating the signal obtained by branching the high-frequency component, delaying it for a predetermined time, and outputting it. Further, the speaker broadband control device receives the output from the first buffer memory, the branch signal of the low-frequency component from the band dividing section, and the output from the second low-pass filter section described later, and receives the predetermined A first foreseeing sliding mode control unit that performs sliding mode control by foreseeing the time ahead, an output from the second buffer memory, a high-frequency component branch signal from the band dividing unit, and a high-pass filter unit to be described later. and a second foreseeing sliding mode control unit that receives the output of and performs the sliding mode control by foreseeing the predetermined time ahead.

Further, this speaker broadband control device includes: a first low-pass filter unit connected to the output side of the first foresight sliding mode control unit; and a speaker driven by a combined signal with the signal from the mode control unit. Further, the speaker broadband control device includes a control amount detection unit that detects the vibration output of the speaker, a second low-pass filter unit that captures low-frequency components of the detection signal and outputs a position feedback value, and captures high-frequency components. A detection section is provided comprising a high pass filter section for outputting a velocity feedback value.

JP 2009-206966 A

However, the technique disclosed in Patent Literature 1 temporarily stores an original signal in a buffer memory, and performs preview sliding mode control, which is a type of adaptive control, on the original signal. Therefore, with this technique, a corresponding delay occurs with respect to the original signal when it is stored in the buffer memory. As a result, for example, when loudspeakers are amplified in real time, such as at lectures and concerts, as a result of delays between speech and pronunciation, and accompanying gestures and actions, high fidelity is not always possible. There is a problem that it is not always possible to reproduce sound.

The present disclosure has been made in view of the above circumstances, and an object thereof is to provide an audio signal processing device capable of reproducing original audio with higher fidelity.

According to claim 1, the audio signal processing apparatus according to the present invention comprises an amplifier that amplifies an input audio signal and outputs an audio signal to be input to a speaker; and a pre-stage digital filter whose coefficient is an inverse filter for the vibration response waveform of the cone surface of the speaker when a predetermined audio signal is input.

According to the audio signal processing apparatus according to the present invention as set forth in claim 1, in the preceding stage of the amplifier, an inverse filter coefficient is applied to the vibration response waveform of the cone surface of the speaker when a predetermined audio signal is input to the amplifier. By providing a pre-stage digital filter of , it is possible to reproduce the original sound with higher fidelity.

The audio signal processing apparatus according to the present invention according to claim 2 is the audio signal processing apparatus according to claim 1, wherein the audio signal processing apparatus according to claim 1 is provided in the rear stage of the amplifier, and in a state in which the front digital filter is provided, The digital filter further comprises a post-stage digital filter whose coefficient is an inverse filter for the sound pressure response waveform of the speaker when a predetermined audio signal is input to the digital filter.

According to the audio signal processing apparatus according to the present invention described in claim 2, in a state in which a pre-stage digital filter is provided after the amplifier, a predetermined audio signal is input to the pre-stage digital filter. By providing a post-stage digital filter whose coefficient is the inverse filter for the sound pressure response waveform, it is possible to correct the processing residual of the pre-stage digital filter, and as a result, it is possible to reproduce the original sound with higher fidelity.

According to claim 3, the audio signal processing apparatus according to the present invention is the audio signal processing apparatus according to claim 1 or claim 2, wherein the predetermined audio signal is an impulse signal. .

According to the audio signal processing apparatus according to the present invention described in claim 3, by using the predetermined audio signal as an impulse signal, the audio signal can be processed more effectively than when the audio signal is a signal other than the impulse signal. An inverse filter can be derived easily.

As described above, according to the present invention, original speech can be reproduced with higher fidelity.

1 is a block diagram showing an example of the configuration of an audio signal processing device according to an embodiment; FIG. It is a figure which shows the manufacturing process of the audio signal processing apparatus which concerns on embodiment, and is a block diagram which shows an example of a structure in the case of manufacturing a pre-stage digital filter. It is a figure which shows the manufacturing process of the audio signal processing apparatus which concerns on embodiment, and is a block diagram which shows an example of a structure in the case of manufacturing a post-stage digital filter. FIG. 4A is a diagram showing a manufacturing process of the audio signal processing device according to the embodiment, and is a schematic diagram showing an example of a more detailed manufacturing process when mainly manufacturing a pre-stage digital filter. It is a diagram for explaining a conventional technique, and is a side cross-sectional view showing an example of the structure of a conventional speaker. It is a diagram for explaining a conventional technique, and is a waveform diagram showing an example of a sound pressure waveform when an impulse signal is input to a speaker. FIG. 1 is a diagram for explaining a conventional technique, and is a block diagram showing an example of a configuration of an audio signal processing device using an active improvement method among methods for improving transient characteristics of a speaker.

Embodiments for carrying out the present invention will be described in detail below with reference to the drawings.

First, the configuration of an audio signal processing device 10 according to the present embodiment will be described with reference to FIG. FIG. 1 is a block diagram showing an example of the configuration of an audio signal processing device 10 according to this embodiment.

As shown in FIG. 1, the audio signal processing device 10 according to the present embodiment includes a power amplifier 12, a pre-stage digital filter 14, and a post-stage digital filter 16. As shown in FIG.

The power amplifier 12 according to this embodiment amplifies an input audio signal and outputs an audio signal to be input to the speaker 20 . Further, the pre-stage digital filter 14 according to the present embodiment is provided in the pre-stage of the power amplifier 12, and performs an inverse filter for the vibration response waveform of the cone surface of the speaker 20 when a predetermined audio signal is input to the power amplifier 12. coefficient.

Then, the post-stage digital filter 16 according to the present embodiment is provided in the post-stage of the power amplifier 12, and in a state in which the pre-stage digital filter 14 is provided, the speaker is output when a predetermined audio signal is input to the pre-stage digital filter 14. An inverse filter for the sound pressure response waveform by 20 is used as a coefficient.

Here, the speaker to be applied is not particularly limited, and various speakers capable of measuring vibration during sound reproduction, such as a static speaker, a dynamic speaker, and a piezoelectric element speaker, can be applied.

In addition, in the audio signal processing apparatus 10 according to the present embodiment, an impulse signal is applied as the predetermined audio signal, but the present invention is not limited to this. An audio signal of human voice, an audio signal of music, or the like may be applied as the predetermined audio signal. However, in this form, it is necessary to estimate the impulse response.

Next, a manufacturing process of the audio signal processing device 10 according to the present embodiment will be described with reference to FIGS. 2 to 4. FIG. FIG. 2 is a diagram showing the manufacturing process of the audio signal processing device 10 according to the present embodiment, and is a block diagram showing an example of the configuration when manufacturing the pre-stage digital filter 14. As shown in FIG. FIG. 3 is a diagram showing the manufacturing process of the audio signal processing device 10 according to the present embodiment, and is a block diagram showing an example of the configuration when manufacturing the post-stage digital filter 16. As shown in FIG. Furthermore, FIG. 4 is a diagram showing the manufacturing process of the audio signal processing device 10 according to the present embodiment, and is a schematic diagram showing an example of a more detailed manufacturing process when mainly manufacturing the pre-stage digital filter 14. As shown in FIG. be.

As shown in FIGS. 2 and 4, first, the power amplifier 12 is connected to the target speaker 20, an impulse signal is input to the power amplifier 12, and the cone of the speaker 20 responds to the input of the impulse signal. Measure the vibration response waveform g(t) of the paper (see also FIG. 5). In this embodiment, the laser Doppler vibrometer 40 is used to measure the vibration response waveform g(t), but the present invention is not limited to this. The vibration response waveform g(t) may be measured by applying another measuring device capable of measuring vibration in a non-contact manner.

Next, an inverse filter h(t) for the vibration response waveform g(t), that is, an inverse filter h(t) that satisfies the following equation (1), which is a convolution operation, is obtained. Note that δ(t) in Equation (1) is a delta function.

Next, a digital filter is produced using the obtained inverse filter h(t) as coefficients. As an example, as shown in FIG. 3, the manufactured digital filter is provided as the pre-stage digital filter 14 in the pre-stage of the power amplifier 12, and in this state, the impulse signal is input to the pre-stage digital filter 14, and the sound is generated in the vicinity of the speaker 20. A pressure response waveform p(t) is measured. In this embodiment, an ordinary microphone is used to measure the sound pressure response waveform p(t), but the present invention is not limited to this. A form of measuring the pressure response waveform p(t) may be employed.

In this embodiment, the position for measuring the sound pressure response waveform p(t) is 1 m from the front of the speaker 20 in consideration of the influence of sound reflection from the walls of the room in which the speaker 20 is installed. However, it is not limited to this. Any distance can be applied as the position for measuring the sound pressure response waveform p(t) as long as the distance is such that the reflected sound from the walls of the room where the speaker 20 is installed can be ignored.

Next, an inverse filter hc(t) for the sound pressure response waveform p(t), that is, an inverse filter hc(t) that satisfies the following equation (2), which is a convolution operation, is obtained.

Finally, a digital filter having the obtained inverse filter hc(t) as a coefficient is manufactured, and as an example shown in FIG.

In the audio signal processing apparatus 10 manufactured as described above, as long as the input to the speaker is appropriate (not excessive or the compressor is in a functioning state), these digital filters whose coefficients are once determined are permanent. can be used for

Further, the front-stage digital filter 14 and the rear-stage digital filter 16 may be built in the power amplifier 12, the speaker 20 main body, or other audio equipment (processor or the like).

Since the vibration (vibration velocity) of the surface of the cone paper and the vibration (particle velocity) of the air in contact with it are continuous, the sound pressure waveform is changed by making the vibration waveform of the cone paper follow the original signal by the digital filter 14 in the preceding stage. Dullness can also be corrected. However, if the frequency characteristics of the vibration of the cone paper include peaks and dips, it is possible that a stable and complete inverse filter cannot be obtained. The original signal can be reproduced almost perfectly at the level of the pressure waveform.

A method using only an inverse filter obtained from the sound pressure waveform (most conventional active control belongs to this method) is also conceivable, but the sound pressure waveform is affected by directivity and distance attenuation, etc., and depending on the measurement position Due to the considerable difference, not only is it possible that the obtained inverse filter cannot be sufficiently corrected, but there is also the possibility that some areas will be deteriorated. In addition, complications such as a situation in which the inverse filter cannot be obtained stably due to the influence of the reflected wave (so-called minimum phase condition is not ensured) are assumed.

Further, as described above, the method disclosed in Patent Document 1 temporarily stores an original signal in a buffer memory, and performs preview sliding mode control, which is a kind of adaptive control, on the original signal. There is a corresponding delay with respect to the original signal at the time of storage. Therefore, in this method, for example, in the case of real-time loudspeaking such as lectures and concerts, there is a delay between speech and pronunciation, and accompanying gestures and actions, resulting in discomfort. is also a concern.

As described above, according to this embodiment, an amplifier (in this embodiment, the power amplifier 12) that amplifies an input audio signal and outputs an audio signal to be input to a speaker; and a pre-stage digital filter whose coefficient is an inverse filter for the vibration response waveform of the cone surface of the speaker when a predetermined audio signal is input to the amplifier. Therefore, the original speech can be reproduced with higher fidelity.

Further, according to the present embodiment, in a state in which a pre-stage digital filter is provided after the amplifier and a predetermined audio signal is input to the pre-stage digital filter, the sound pressure response waveform of the speaker is reversed. A post-stage digital filter having the filter as a coefficient is further provided. Therefore, as a result of being able to correct the processing residual of the pre-stage digital filter, the original sound can be reproduced with higher fidelity.

Furthermore, according to this embodiment, an impulse signal is applied as the predetermined audio signal. Therefore, the inverse filter can be derived more easily than when the audio signal is a signal other than the impulse signal.

In the above embodiment, the audio signal processing device 10 includes both the pre-stage digital filter 14 and the post-stage digital filter 16. However, the present invention is not limited to this. For example, a configuration that includes only the pre-stage digital filter 14 may be applied as the audio signal processing device 10 . In this case, although the residual by the pre-stage digital filter 14 described above cannot be corrected, the audio signal processing apparatus 10 can be configured at a lower cost and in a smaller size.

10 Audio signal processing device 12 Power amplifier 14 Pre-stage digital filter 16 Post-stage digital filter 20 Speaker 40 Laser Doppler vibrometer

Claims (3)

an amplifier that amplifies an input audio signal and outputs an audio signal to be input to a speaker;
a pre-stage digital filter provided in the pre-stage of the amplifier and having an inverse filter coefficient for a vibration response waveform of the cone surface of the speaker when a predetermined audio signal is input to the amplifier;
An audio signal processor with A post-stage digital provided after the amplifier and having a coefficient of an inverse filter for a sound pressure response waveform of the speaker when a predetermined audio signal is input to the pre-stage digital filter in a state where the pre-stage digital filter is provided. filter,
The audio signal processing device according to claim 1, further comprising: wherein the predetermined audio signal is an impulse signal;
3. The audio signal processing device according to claim 1 or 2.