JP2005184041A - Apparatus and system for reproducing audio signal - Google Patents

Abstract

<P>PROBLEM TO BE SOLVED: To provide an audio signal processing apparatus which well switches processing with less noise because difference in a propagation delay time never occurs between filters to be switched, and because the phase of a waveform is not deviated. <P>SOLUTION: In the audio signal processing apparatus 1, the output of a second filter 4 is inputted into a first delay device 6 having a delay time of 2.9 ms, and the output of a third filter 5 is inputted into a second delay device 7 having a delay time of 5.8 ms. By doing this, the delay time in the maximum value of impulse response of each filter matches, and a noise to be generated in switching between the filters is reduced. <P>COPYRIGHT: (C)2005,JPO&NCIPI

Description

  The present invention relates to an audio signal processing device and an audio signal reproduction system for processing an audio signal with digital filters having different characteristics and switching and outputting the filter processing output, and more particularly to sound quality and sound image localization obtained by convolution integration in the signal processing device. The present invention relates to an audio signal processing device and an audio signal reproduction system that reduce noise generated when the characteristics of the audio signal are switched.

  By convolving and integrating the impulse response having various characteristics to the input audio signal, an audio output signal having each characteristic can be obtained. A system for enjoying a change in sound and a change in sound quality by turning these impulse responses ON / OFF or switching and listening to each other can be considered.

  However, in such an audio signal reproduction system in which a plurality of convolution integrals having different impulse responses can be switched and heard, unpleasant noise may be generated when the characteristics of each convolution integral are greatly different.

  One factor of noise generated in this way is due to the delay time characteristics of the convolution integrator. In the signal processing apparatus in the conventional example shown in FIG. 10, for example, the filter output switching apparatus 100, the impulse response as shown in FIG. 11 is convoluted with the input speech signal by the convolution integrator configured in the first filter 102, and the second filter. In 103, through processing (signal of only impulse at time 0) is performed. The output from the first filter 102 has a delay time Δt compared to the impulse Ip of the second filter 103 at the maximum signal output portion Max of the impulse response.

  As shown in FIG. 12, the characteristic having the delay time Δt (FIG. 12B) and the through characteristic having the delay time 0 (FIG. 12A) are switched at the time T by the voice switching device 104. In this case, as shown in FIG. 12C, noise is generated due to a waveform shift due to the delay time Δt, and this is heard as a stimulating shock sound.

  In order to avoid this, it is necessary to take measures such as a mute process for eliminating the audio output within the noise generation time in the control system as disclosed in Japanese Patent No. 2626315.

  When such mute processing is performed, it is possible to avoid noise, but there is a problem that processing for that is provided in the control system, and accurate listening comparison by switching is hindered by inserting a silent part. It was.

  In the above description, the case where the convolution integration is the convolution in the time domain has been described. However, when the convolution integration is realized in the frequency domain, in the fast Fourier transform (FFT) operation, wait until samples according to the order of convolution are prepared. There is a need and a time delay for this occurs. For this reason, the noise becomes even larger. Or when the time delay becomes large, the reproduced sound is separated and the same phrase can be heard repeatedly.

Japanese Patent No. 2626315

  As described above, in the switching between the convolution calculation processing result and the through output and other processing results by the conventional filter output switching device, the delay characteristics for each filter are different. There was a problem that repetition occurred. For this reason, it is necessary to add mute processing or the like by the control system. However, when such a mute process is performed, there is a problem that the silent time during which the sound is interrupted becomes long, and accurate sound switching comparison is hindered.

  The present invention has been made in view of the above circumstances, and since there is no propagation delay time difference between the filters for switching the processing output and the phase of the waveform does not shift, an audio signal that can perform a good switching process with less noise. It is an object of the present invention to provide a processing device and an audio signal reproduction system.

  In order to solve the above problems, an audio signal processing device according to the present invention processes an input audio signal with a plurality of filters having different characteristics, and switches each to output an impulse response of each filter. Is provided with the same delay time until the maximum value is output, and the maximum levels of the processing outputs of all the filters are output with substantially the same delay time.

  According to this audio signal processing apparatus, no propagation delay time difference occurs between the filters to be switched, and the phase of the waveform does not shift.

  In order to solve the above problems, an audio signal processing device according to the present invention performs convolution integration processing on an input audio signal with a plurality of filters having different characteristics, and switches each of the filters to output each filter. The delay means for providing the same delay time until the maximum value of the impulse response is output is provided, and the maximum level of the convolution integration processing output of all the filters is output with substantially the same delay time.

  According to this audio signal processing device, the phase of the waveform is not shifted because switching is performed without causing a propagation delay time difference between the filters that perform convolution integration processing on the input audio signal.

  In order to solve the above-described problem, an audio signal processing device according to the present invention performs processing including convolution integration with a constant group delay characteristic on a plurality of filters having different characteristics on an input audio signal, and outputs the audio signal by switching each of them. In the processing apparatus, a delay means is provided to make the delay time until the maximum impulse response value of each filter is output, and the maximum level of the processing output including convolution integral with a constant group delay characteristic of all the filters. Are output with approximately the same delay time.

  According to this audio signal processing apparatus, the phase of the waveform is not shifted because switching is performed without causing a difference in propagation delay time between the filters that perform processing including convolution integration with a constant group delay characteristic on the input audio signal.

  In order to solve the above problems, the audio signal processing device according to the present invention performs processing including convolution integration of the inverse characteristic of the transmission system obtained by calculation or actual measurement with a plurality of filters having different characteristics on the input audio signal, In the audio signal processing apparatus that switches and outputs each, the delay means for making the delay time until the maximum value of the impulse response of each filter is output is provided, and is obtained by calculation or actual measurement of all the filters. The maximum level of the processing output including the convolution integral of the inverse characteristic of the transmission system is output with substantially the same delay time.

According to this audio signal processing apparatus, the processing including the convolution integral of the inverse characteristic of the transmission system obtained by calculation or actual measurement is performed without changing the propagation delay time between the filters that apply to the input audio signal, so that the phase of the waveform Does not slip.
In order to solve the above-described problem, the audio signal processing device according to the present invention performs processing including convolution integration of an impulse response of a transmission system obtained by calculation or actual measurement with a plurality of filters having different characteristics on an input audio signal, In the audio signal processing apparatus that switches and outputs each, the delay means for making the delay time until the maximum value of the impulse response of each filter is output is provided, and is obtained by calculation or actual measurement of all the filters. The maximum level of the processing output including the convolution integral of the impulse response of the transmission system is output with substantially the same delay time.

  According to this audio signal processing device, the processing including the convolution integral of the impulse response of the transmission system obtained by calculation or measurement is performed without switching the propagation delay time between the filters that apply to the input audio signal. Does not slip.

  In order to solve the above problems, an audio signal reproduction system according to the present invention processes an input audio signal with a plurality of filters having different characteristics, and sets a delay time until the maximum value of the impulse response of each filter is output. Provided with the same delay means, the maximum level of the processing output of all the filters is set to substantially the same delay time, and the audio signal processing device that switches and outputs each, and at least two or more drive units divided by frequency band A speaker system.

  The audio signal processing device does not generate a propagation delay time difference between the filters to be switched, and does not cause a shift in the waveform phase.

  According to the audio signal processing apparatus of the present invention, there is provided delay means for processing an input audio signal with a plurality of filters having different characteristics and making the delay time until the maximum value of the impulse response of each filter is output the same. As the maximum processing output level of all filters is set to approximately the same delay time, each is switched and output, so there is no difference in propagation delay time between the filters to be switched, the waveform phase is not shifted, and noise A few good switching processes can be performed.

  According to the audio signal processing device of the present invention, the input audio signal is subjected to convolution integration processing with a plurality of filters having different characteristics, and the delay time until the maximum value of the impulse response of each filter is output is the same. Since the delay means is provided and the maximum level of the convolution integration processing output of all the filters is set to substantially the same delay time and each is switched and output, there is a difference in propagation delay time between the filters that perform the convolution integration processing on the input audio signal. Since the switching is performed without occurrence, the phase of the waveform is not shifted, and therefore, a favorable switching process with less noise can be performed.

  According to the audio signal processing device of the present invention, the input audio signal is subjected to processing including convolution integration with a constant group delay characteristic using a plurality of filters having different characteristics, and the maximum value of the impulse response of each filter is output. Since the delay means for making the delay times of the same is provided, the group delay characteristics of all the filters are set to the same delay time with the maximum level of the processing output including the constant convolution integral being switched, and the group delay characteristics are thus switched. Propagation delay time difference does not occur between the filters that perform processing including constant convolution integration on the input audio signal, and the phase of the waveform does not shift, so that it is possible to perform good switching processing with less noise.

  According to the audio signal processing device of the present invention, the input audio signal is subjected to processing including convolution integration of the inverse characteristic of the transmission system obtained by calculation or actual measurement using a plurality of filters having different characteristics, and the impulse response of each filter The delay means to make the delay time until the maximum value of the output is the same is provided, and the maximum level of the processing output including the convolution integral of the inverse characteristic of the transmission system obtained by calculation or measurement of all the filters is almost Since each is output with the same delay time switched, there is no propagation delay time difference between each filter that performs processing including convolution integration of the inverse characteristics of the transmission system obtained by calculation or actual measurement on the input audio signal, and the waveform Therefore, a favorable switching process with less noise can be performed.

  According to the audio signal processing device of the present invention, the input audio signal is subjected to processing including convolution integration of the impulse response of the transmission system obtained by calculation or measurement using a plurality of filters having different characteristics, and the impulse response of each filter The delay means for making the delay time until the maximum value of the output is the same is provided, and the maximum level of the processing output including the convolution integral of the impulse response of the transmission system obtained by calculation or measurement of all the filters is approximately Since the same delay time is output by switching each, the propagation delay time difference does not occur between the filters that perform processing including convolution integration of the impulse response of the transmission system obtained by calculation or actual measurement on the input audio signal, and the waveform Therefore, a favorable switching process with less noise can be performed.

  The audio signal reproduction system according to the present invention is provided with delay means for processing an input audio signal with a plurality of filters having different characteristics and making the delay time until the maximum value of the impulse response of each filter is output the same. The audio signal processing device for switching and outputting each with the maximum level of the processing output of all filters as substantially the same delay time, and a speaker system having at least two drive units divided by frequency band, In the audio signal processing apparatus, a propagation delay time difference is not generated between the filters to be switched, and the waveform phase is not shifted. Therefore, a favorable switching process with less noise can be performed.

  Several best modes for carrying out the present invention will be described below. The first embodiment is an audio signal processing apparatus 1 as shown in FIG. The audio signal processing device student 1 will be described assuming a digital audio signal as an input signal, but an analog audio signal can be handled in the same manner by first performing A / D conversion processing.

  In FIG. 1, the audio signal processing apparatus 1 adds different characteristics to the digital audio signal input from the input terminal 2 depending on the first filter 3, the second filter 4, and the third filter 5. A signal to which a characteristic is added by one type of filter is switched and selected by the voice switching device 8 according to the operation of the user or the like and supplied to the output terminal 9.

  Further, the first filter 6 is connected to the second filter 4, and the second delay device 7 is connected to the third filter. No delay device is connected to the first filter 3. Therefore, the filter output of the first filter 3 is directly supplied to the selected terminal 8a of the voice switching device 8, and the filter output of the second filter 4 via the first delay device 6 is supplied to the selected terminal 8b. The filter output of the third filter 5 via the second delay device 7 is supplied to the selected terminal 8c.

  As described above, the audio switching device 8 is connected to the selected terminal of the selected terminals 8a to 8c in accordance with the operation of the user or the like, so that the characteristics of the voice switching device 8 are reduced by one type of filter. The added signal is output to the output terminal 9.

For example, if the first filter 3 is an equalizer with 512 taps and a constant group delay, the second filter 4 is an equalizer with a constant tap delay of 256 taps, and the third filter 5 is a through system that does not actually process, the sampling frequency In the case of 44.1 kHz, the delay time until the peak value of the impulse response is transmitted by the first filter 3 is 22.7 μs × 256≈5.8 ms.
It becomes. The delay time of the second filter 4 is
22.7 μs × 128 ≈ 2.9 ms
It becomes. Since the filter 3 is not processed, the delay time is 0 s.

  If the audio signal is output as it is, a propagation time difference due to the group delay characteristic is generated, and a large noise is generated when listening by switching. Therefore, in this audio signal processing apparatus 1, the output of the second filter 4 is input to the first delay device 6 having a delay time of 2.9 ms, and the output of the third filter 5 is supplied to the second delay of 5.8 ms. Input to the device 7. As a result, the delay times at the maximum value of the impulse response of each filter coincide with each other, and it is possible to reduce noise generated when these are switched.

  Next, a second embodiment will be described. In the audio signal processing apparatus 1 according to the first embodiment, a filter constituting an equalizer having a constant group delay characteristic is considered. However, the audio signal reproduction system 10 according to the second embodiment has a reproduction characteristic of a speaker. The delay time is adjusted when switching the filter output of the inverse filter and the filter output of the through system.

  In FIG. 2, the digital audio signal input from the input terminal 11 is supplied to the first signal processing device 12 and the second signal processing device 13. The first signal processing device 12 constitutes an inverse filter of the reproduction characteristic of the speaker, and the second audio signal processing device 13 constitutes a through system filter. The second signal processing device 13 has a built-in or external delay device that generates a delay time corresponding to the delay time by the inverse filter of the first signal processing device 12.

  The filter output of the first signal processing device 12 and the filter output of the second signal processing device 13 are switched by a changeover switch 14. The switching output by the selector switch 14 is converted into an analog audio signal by the D / A converter 15, amplified by the power amplifier 16, and then supplied to the 2-way speaker system 17, for example.

  The two-way speaker system 17 is connected to a low-frequency drive unit 19 connected to the LPF 18 that passes the low frequency band of the analog audio signal amplified by the power amplifier 16 and the HPF 20 that passes the high frequency band of the analog audio signal. The high-frequency drive unit 21. In this 2-way speaker system 17, since the drive surface (acoustic center) 19a of the low-frequency drive unit 19 and the drive surface 21a of the high-frequency drive unit 21 are not aligned, there is a propagation delay time difference Δt between the low frequency and high frequency of the reproduction frequency. Occur. Since the drive surfaces of the drive units are not aligned in this way, if a propagation delay time difference Δt occurs, the phase of the wavefront of the sound wave is shifted depending on the reproduction frequency band, which is not preferable for obtaining a good sound image localization.

  Therefore, in the first signal processing device 12, the total impulse response of the speaker system 17 when both the high frequency drive unit 21 and the low frequency drive unit 19 are driven simultaneously is measured in advance, and the inverse characteristic obtained by the calculation is obtained. Use.

  For example, it is assumed that the speaker system 17 in FIG. 2 has the impulse response in FIG. 3A and the frequency characteristic in FIG. 3B which is an expression in the frequency domain. When the inverse characteristic of the impulse response of FIG. 3A is calculated, the impulse response (reverse impulse response) of FIG. 4A is obtained.

The inverse characteristic of the impulse response is calculated based on the following principle. When the impulse IP shown in FIG. 5A is input to the function A, an impulse response RI is obtained. A transfer function for returning the impulse response RI to the impulse IP as shown in FIG. 5B is an inverse function A- ¹ . When an impulse IP is input to the inverse function A- ¹ as shown in FIG. 5C, an inverse impulse response IRI is obtained. However, in the low sound range, the roll-off is performed based on the reproduction capability of the low frequency drive unit 9, for example, restrictions such as nonlinear distortion characteristics and allowable input level.

  The inverse characteristic (reverse impulse response) IRI of the impulse response is realized by a digital filter in the first signal processing device 12. If this inverse impulse response is input to the speaker system 17 having the function A, the impulse IP is obtained. Thereby, when the characteristics of the speaker are intended at the same measurement point, a flat frequency characteristic as shown in FIG. 6B and an impulse response characteristic close to the impulse shown in FIG. 6A can be obtained.

  The first signal processing device 12 realizes the inverse characteristic of the impulse response using a digital filter 30 as shown in FIG. 7, for example. In the digital filter 30 shown in FIG. 7, the digital audio signal SD is supplied in series to a plurality of delay circuits 32 to 32 through an input terminal 31, and signals obtained from the terminal 31 and the delay circuits 32 to 32 are multiplied by multiplier circuits 33 to 33. 33 and the multiplication output is taken out to the output terminal 35 through the addition circuits 34-34. In this case, the delay circuits 32 to 32 give the digital audio signal SD a delay of one sampling period (one unit period) τ, and the multiplication circuits 33 to 33 use the inverse characteristic of the impulse response as a coefficient. I have it.

  In an ideal two-way speaker system where the drive surface of the low-frequency drive unit and the drive surface of the high-frequency drive unit are aligned, the phase shift described above is ignored when ignoring the time spread of the impulse response of each drive unit itself. Since it should not occur, if an impulse is input, the impulse response should come out as it is, instead of the impulse response having a temporal spread as shown in FIG.

  This is because even in the speaker system 17 shown in FIG. 2, the drive surface 19 a of the low-frequency drive unit 19 and the drive surface 21 a of the high-frequency drive unit 21 are aligned if an impulse is generated and an impulse is output. Is equivalent to

  Therefore, also in the speaker system 17, the deterioration of the transmission characteristics due to the difference in the transmission delay time between the plurality of speaker / drive units is improved, and the same phase property of each unit is ensured. Therefore, if the digital filter 30 shown in FIG. 7 is used in the first signal processing device 12, a speaker reproduction system having good sound image localization and sound quality can be obtained.

  However, when the inverse characteristic as shown in FIG. 4 is obtained by calculation, the peak position of the impulse response has a delay time t as compared with the original impulse response.

  Therefore, when the changeover switch 14 is used to hear the output of the first signal processing device 12 to which such an inverse characteristic is added and the output of the second signal processing device 13 having no delay time, that is, the delay time 0, Noise is generated due to the time difference t. Therefore, in the same way as in the first embodiment, the second signal processing device 13 system to which no reverse characteristic is added has a built-in or external delay device corresponding to the delay time t due to the reverse characteristic, thereby providing an impulse. The delay time difference in the vicinity of the response peak is reduced, and it is possible to reduce noise when the two are switched.

  Next, a third embodiment will be described. This third embodiment is applied to an audio signal processing device that obtains a desired characteristic by convolving, for example, an equalizer, an inverse characteristic of the impulse response, or a room impulse response including reverberation by convolution integration. Thus, the audio signal processing device 40 of FIG. 8 performs the convolution integration on the frequency axis when these characteristics can be switched between each other or switched ON / OFF.

  8, the audio signal processing device 40 includes a first signal processing unit 42 that performs the convolution integration on the audio signal from the input terminal 41 on the frequency axis, and a second signal processing that passes through the input audio signal. And a selector switch 48 that switches between the processing output of the first signal processing unit 42 and the processing output of the second signal processing unit 46 and supplies the processing output to the output terminal 49. The audio signal processing device 40 also includes a delay unit 47 that generates a delay time corresponding to the delay time in the first signal processing unit 42 at the subsequent stage of the second signal processing unit 46.

  The first signal processing unit 42 includes a fast Fourier transform (FFT) unit 43 that converts the input audio signal into data on the frequency axis, a convolution integrator 44 that convolves a filter characteristic G with the output S of the FFT unit 43, And an inverse FFT unit 45 that performs inverse FFT on the output S · G of the integrator 44. The integrator 44 is actually a multiplier that multiplies the output S of the FFT unit 43 and the filter characteristic G in the frequency domain.

  The changeover switch 48 receives the output of the first signal processing unit 42 at the selected terminal a, receives the output of the second signal processing unit 46 delayed by the delay unit 47 at the selected terminal b, and By switching, the selected terminal a or the selected terminal b is selected and either output is supplied to the output terminal 49.

  In the first and second embodiments, the convolution integration is performed on the time axis, and the delay time corresponding to the maximum impulse response level of the output reflects the delay time up to the maximum value of the impulse response almost as it is. Is done. Here, it is assumed that the time delay associated with A / D or D / A is not considered.

However, in the audio signal processing device 40 according to the third embodiment, the convolution integration by the first signal processing unit 42 is performed on the frequency axis using the FFT unit 43. Therefore, in actual processing, impulses are used. In addition to the response delay time, a delay time due to the FFT processing is separately generated. That is, in the FFT operation, it is necessary to wait for samples to be collected according to the order of convolution and the time required for processing. For example, when the time delay for this is 512 taps,
22.7 μs × 512 × 2 ≒ 23.2 ms
This will occur. Therefore, when the output of the first signal processing unit 42 and the output of the second signal processing unit 46 of the through system, for example, are heard by switching the changeover switch 48 according to ON / OFF, the time difference between them is heard. Due to noise and sound repetition. In the audio signal processing device 40 of the third embodiment, in order to avoid this, the time delay due to the FFT processing and the delay time to the peak value of the impulse response itself are given to the second signal processing unit 46. The correction is performed by the connected delay device 47, and the switching with less noise is realized by configuring so that the deviation of the reproduced waveform is reduced even when the switching is heard.

  In the first embodiment, the configuration in which the plurality of filters 3, 4, 5 previously configured in the signal processing device 1 are switched by the audio switching device 8 has been described. Of course, the same applies to the case where a different algorithm is realized by rewriting the filter algorithm with a new algorithm each time switching processing is performed.

  The same algorithm can be used to realize the switching function by switching only the coefficients.

  In the first embodiment, the filter unit and the delay unit have been described separately for the sake of convenience of explanation. However, the characteristics of both are combined, and an interval with an impulse response of 0 is used as a delay means for the first impulse for time t1. By providing it before the response, it can also be realized by using one filter means to which delay information of time t1 is added. A specific example of the original impulse response in this case is shown in FIG. 9A, and a specific example of the impulse response obtained by synthesizing the time delay t1 is shown in FIG. 9B. The combined time delay of t0 + t1 in FIG. 9B is generated by the filter means that combines the above two characteristics.

  This is the same when a through signal, that is, a delay device of t1 is passed through. Instead of the delay device as a delay means, an interval in which the output level of the impulse response is 0 is initially set at time t1, and time t1 This can also be realized by filter means having an impulse response that has an output only at, and thereafter has an output level of 0 again.

  According to this method, for example, a filter algorithm as shown in FIG. 7 can be used in common for all filter characteristics, and the characteristics and on / off switching can be similarly performed only by rewriting coefficients.

  In the first to third embodiments, various impulse responses to be added to the input audio signal in the signal processing apparatus are arbitrary characteristics obtained by measurement or calculation in addition to the inverse characteristics of the impulse response of the speaker system. Transmission characteristics can be mentioned.

  For example, there are characteristics for controlling the sound image localization position when an input audio signal is reproduced by a plurality of speakers, an impulse response characteristic of an arbitrary room, an impulse response characteristic of an electroacoustic transducer, and the like.

  In particular, the impulse response characteristics of electroacoustic transducers are the so-called famous machines, the impulse response characteristics of speakers or headphone systems that have become difficult to obtain, the impulse response characteristics of record hands, the impulse response characteristics of recording and playback devices. The impulse response characteristic of the frequency characteristic adding device, the impulse response characteristic of the voice amplifier, and the like can be mentioned.

1 is a block diagram of an audio signal processing device according to a first embodiment. It is a block diagram of the audio | voice signal reproduction | regeneration system of 2nd Embodiment. It is a characteristic view which shows the impulse response and amplitude frequency characteristic of the speaker system used with the said audio | voice signal reproduction | regeneration system. It is a characteristic view which shows the reverse impulse response and amplitude frequency characteristic of the said speaker system. It is a figure for demonstrating the calculation principle of the reverse characteristic of an impulse response. It is a characteristic view which shows the impulse response close to an impulse, and an amplitude frequency characteristic. It is a circuit diagram which shows the specific example of a digital filter. It is a block diagram of the audio | voice signal processing apparatus of 3rd Embodiment. It is a figure for demonstrating the impulse response which synthesize | combined the characteristic of both a filter and a delay device. It is a block diagram of the conventional filter switching apparatus. It is a characteristic view of the impulse response given to the filter of the filter switching device. It is a figure for demonstrating the noise by the delay time generate | occur | produced by the filter switching apparatus.

Explanation of symbols

  DESCRIPTION OF SYMBOLS 1 Audio | voice signal processing apparatus, 3 1st filter, 4th filter, 5 3rd filter, 6 1st delay device, 7 2nd delay device, 8 Audio switching devices

Claims (8)

In an audio signal processing apparatus that processes input audio signals with a plurality of filters having different characteristics, and switches and outputs them,
The delay means which makes the delay time until the maximum value of the impulse response of each filter is outputted is the same,
An audio signal processing apparatus which outputs the maximum level of the processing output of all filters with substantially the same delay time.   2. The audio signal processing apparatus according to claim 1, wherein the processing by the filter is convolution integration.   2. The audio signal processing apparatus according to claim 1, wherein the processing by the filter includes convolution integration having a constant group delay characteristic.   2. The audio signal processing apparatus according to claim 1, wherein the processing by the filter includes convolution integration of an inverse characteristic of the transmission system obtained by calculation or actual measurement.   2. The audio signal processing apparatus according to claim 1, wherein the processing by the filter includes convolution integration that convolves an impulse response of a transmission system obtained by calculation or measurement.   6. The audio signal processing apparatus according to claim 1, wherein at least one of the plurality of filters performs a process of outputting an input sound as it is.   7. The audio signal processing apparatus according to claim 2, wherein the convolution integration is configured by convolution integration in a frequency domain. The input audio signal is processed by a plurality of filters with different characteristics, and delay means is provided to make the delay time until the maximum impulse response value of each filter is output, and the maximum processing output of all filters is provided. An audio signal processing device that outputs each level with substantially the same delay time, and
An audio signal reproduction system comprising: a speaker system having at least two or more drive units separated by a frequency band.

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