JP2012100117A - Acoustic processing apparatus and method - Google Patents

Abstract

PROBLEM TO BE SOLVED: To provide an acoustic processing apparatus that can suppress grating caused by the reversal of fundamental sound and harmonic wave in a specific listening position, and in which grating sound does not newly occur in a position other than the specific listening position by applying filters.SOLUTION: The acoustic processing apparatus detects fundamental sound frequency f(n) and harmonic wave frequency f(n) from an input regenerative signal. In addition, the acoustic processing apparatus calculates fundamental sound level L(n) and harmonic wave level L(n) of the input regenerative signal and a convolution signal of an input audio characteristic. The acoustic processing apparatus generates a correction filter that satisfies L(n)>L(n) and applies the correction filter to a regenerative signal when the calculated fundamental sound amplitude level L(n) is compared with the harmonic wave amplitude level L(n) and there is L(n) greater than L(n).

Description

  The present invention relates to a sound processing apparatus, and more particularly to a sound processing apparatus and method for suppressing deterioration in sound quality caused by the influence of acoustic characteristics of a listening room, sound pressure frequency characteristics of a speaker, and the like.

  With the advent of large-capacity storage media such as Blu-ray Disks and the expansion of network distribution of high-quality sound sources, high-accuracy and high-dynamic range audio signals can be easily obtained in the home. Along with such high accuracy of audio signals, there is an increasing demand for reproduction equipment for sound reproduction that is closer to the original signal, that is, more realistic and faithful to the original sound. In order to realize a reproduction environment that is faithful to the original sound, the original signal reaches the listener as it is, that is, the sound pressure frequency characteristic (hereinafter referred to as “acoustic path”) that the audio signal follows from the playback device to the listener's ear (hereinafter referred to as “acoustic path”) It is desirable that it is flat.

  However, when reproducing an audio signal at home, it is very difficult to flatten the f characteristic of the acoustic path. There are various factors that change the f-characteristic of the acoustic path, such as the f-characteristic of the speaker, the f-characteristic of the acoustic device on the acoustic path, and the acoustic characteristics of the listening room. This is due to the influence of the acoustic characteristics.

  In the listening room, the listener receives not only the direct sound generated from the speaker but also the reflected sound reflected from the wall, floor, ceiling, etc. Since these sound waves interfere with each other, a particularly large dip or peak occurs. Further, when the listening position changes, the path difference between the direct sound and the reflected sound also changes, so that the position of the peak or dip changes. In other words, the influence of the acoustic characteristics of the listening room has a characteristic that it greatly changes depending on the listening position.

  The audio signal reproduced in this way has various effects on sound quality degradation and consequently hearing. Therefore, in order to realize signal reproduction faithful to the original sound, acoustic processing that corrects the f-specific disturbance in the acoustic path as described above and suppresses adverse effects on hearing becomes very important.

  Conventionally, there has been a method in which the user himself / herself makes the desired f particularly close to the desired f by using an f special adjusting device such as a graphic equalizer that adjusts the f special of the input signal. In addition, there is a method using automatic sound field correction in which the f characteristic of an acoustic path is measured using a measurement signal and the correction is automatically performed so as to make the desired f particularly close (for example, see Patent Document 1).

Japanese Patent Laid-Open No. 62-166698

As shown in FIG. 8A, a human voice or instrument sound is composed of a fundamental tone and harmonics. The fundamental tone is a frequency component corresponding to a so-called pitch (or pitch), and the harmonic is a frequency component N times (N = 2, 3, 4,...) Of the fundamental tone. Usually, the amplitude level of the fundamental tone (hereinafter referred to as “fundamental tone level L _b ”) is larger than the amplitude level of the Nth harmonic (hereinafter referred to as “harmonic level L _N ”), and the fundamental tone level L _b and the harmonic level L _The tone color is determined by the balance of _N.

Here, the balance of the fundamental tone level L _b and harmonic level L _N, from listening experiments as shown in FIG. 8 (B), is relatively large harmonic level L _N than the fundamental tone level L _b in the music signal When this phenomenon occurs (hereinafter referred to as the “reversal phenomenon”), it has been found that the user feels harsh on hearing. This means that when the audio signal is the fundamental tone level L _b and reversal phenomenon collapses balanced harmonic level L _N affected f-characteristic of the acoustic path is generated when it is played, feels the sense of hearing jarring Means that.

  In order to correct such influence, it is necessary to flatten the f characteristic of the acoustic path. As a method of flattening the f characteristic, correction by an inverse filter can be considered. However, a large number of taps are required to realize characteristics that flatten the f-characteristics of the acoustic path with the FIR filter, which is very difficult with the current signal processing performance.

  For this reason, a filter with a short tap has to be used, and peaks and dips remain in the f. That is, in the conventional method, it was not possible to suppress the audible harshness caused by the above-described reversal phenomenon.

  In addition, the f characteristic of the listening room changes depending on the listening position. Therefore, even if an inverse filter that flattens the f characteristic can be realized, there is a possibility that the peak or dip of the f characteristic of the acoustic path at a point other than the listening position (for example, a point shifted by 1 m from the listening position) may be increased. there were. As described above, in the conventional correction by the inverse filter for the f-characteristic of the acoustic path, there is a possibility that the generation of the harsh sound is promoted at a place other than the listening position.

  Therefore, the present invention makes it possible to suppress the generation of an annoying sound due to the inversion of the fundamental tone and the harmonics at a specific listening position. In addition, even at positions other than the specific listening position, the generation of the harsh sound is not promoted by correction.

  An acoustic processing apparatus according to one aspect of the present invention includes a calculation unit that calculates a fundamental frequency and a harmonic frequency of an input acoustic signal, an input unit that inputs acoustic characteristics in at least a part of an acoustic path, and the acoustic signal. A calculation means for performing a convolution calculation with the acoustic characteristics, and a fundamental level that is an amplitude level of the fundamental frequency and a harmonic level that is an amplitude level of the harmonic frequency for the signal obtained by the calculation in the calculation means First level detection means for detecting the fundamental level or the harmonic level so that the fundamental level is higher than the harmonic level when the harmonic level is higher than the fundamental level. And adjusting means for adjusting.

  ADVANTAGE OF THE INVENTION According to this invention, the reversal phenomenon of the fundamental tone level and harmonic level which generate | occur | produce by the influence of the frequency characteristic of an acoustic path can be suppressed, and generation | occurrence | production of the harsh sound on hearing can be suppressed. In addition, correction can be performed at points other than the specific listening position without facilitating the generation of harsh sounds.

(A) And (B) is a block diagram of the acoustic processing apparatus in Embodiment 1, (C) is a block diagram of the detail of the acoustic signal processing part in Embodiment 1. FIG. (A) is an output signal of a convolution FFT processing part, (B) is a figure which shows the example of the characteristic of the band attenuation | damping filter produced | generated by a filter production | generation part. 2 is a flowchart of acoustic processing in the first embodiment. FIG. 3 is a diagram for explaining detection of a fundamental frequency fb and a harmonic frequency fN in the first embodiment. FIG. 10 is a block diagram showing details of an acoustic signal processing unit in the third embodiment. FIG. 9 is a diagram for explaining reverse rotation determination in the third embodiment. The block diagram of the sound processing apparatus in Embodiment 4. FIG. (A) is a figure explaining a fundamental tone and a harmonic, (B) is a figure explaining a reverse phenomenon.

  Hereinafter, the present invention will be described in detail based on preferred embodiments with reference to the accompanying drawings. The configurations shown in the following embodiments are merely examples, and the present invention is not limited to the illustrated configurations.

<Embodiment 1>
FIG. 1A is a block diagram illustrating a configuration of the sound processing apparatus according to the first embodiment. In FIG. 1, the signal analysis processing unit 100 includes a reproduction signal input unit 101, an acoustic characteristic input unit 102, an acoustic signal processing unit 103, and a signal output unit 104.

  The reproduction signal input unit 101 inputs an audio signal as a reproduction signal from an audio reproduction device such as a CD player 1 or a DVD player. If the input acoustic signal is an analog signal, A / D conversion is performed for later digital signal processing.

  The acoustic characteristic input unit 102 receives an impulse response, which is data of acoustic characteristics of a previously measured acoustic path or part thereof, from a device such as a personal computer (PC) 2. As described above, the acoustic path refers to a path followed by an acoustic signal from the acoustic processing device to the listener's ear. Here, the impulse response input to the acoustic characteristic input unit 102 may be in a format that can calculate the impulse response in at least a part of the acoustic path. When input in the form of a frequency spectrum or transfer function, the input signal may be converted into an impulse response and output.

  Note that the acoustic characteristic input unit 102 may have a function that allows the acoustic processing device itself to measure acoustic characteristics instead of inputting acoustic characteristic data from the outside. Such an example will be described with reference to FIG. The signal analysis processing unit 100 shown in the figure has a configuration further including an acoustic characteristic measurement unit 302 and a microphone 313 in addition to the configuration shown in FIG.

  The acoustic characteristic measurement unit 302 includes a measurement signal generation unit 311 and an acoustic characteristic calculation unit 312. The measurement signal generator 311 generates a measurement signal for measuring the indoor impulse response of the listening room. In general, MLS (Maximum Length Sequence) or the like is used as a signal for measuring the indoor impulse response. The measurement signal may be calculated on the spot or may be held in advance in a memory or the like. The impulse response measurement signal generated by the measurement signal generation unit 311 is generated by the speaker 4 and collected by the microphone 313 set at the listening position.

The acoustic characteristic calculation unit 312 calculates an impulse response by taking a cross-correlation from the signal collected by the microphone 313 and the measurement signal. Here, the impulse response calculated in FIG. 1B is the acoustic path through which the measurement signal has passed, that is, the impulse response in which the f characteristics of the equalizer 5 and the speaker 4 installed by the user and the acoustic characteristics of the listening room are integrated. The The calculated impulse response is input to the acoustic characteristic input unit 102.
In this way, the acoustic processing apparatus itself may perform acoustic characteristic measurement.

  The acoustic signal processing unit 103 receives input signals from the reproduction signal input unit 101 and the acoustic characteristic input unit 102, and performs necessary acoustic processing on the input signals from the reproduction signal input unit 101. The signal output unit 104 receives the signal from the acoustic signal processing unit 103, and performs D / A conversion when outputting an analog signal. The output signal is input to an audio device such as an amplifier 3 and is output from the speaker 4.

  Although only one channel path is shown in the figure, when the input signal is a multi-channel signal, the above processing system is provided independently for each channel. For example, if the input signal is a stereo signal, it has a two-channel system.

Next, a signal flow in the acoustic signal processing unit 103 in the above-described embodiment will be described with reference to FIG.
The acoustic signal input to the acoustic signal processing unit 103 from the reproduction signal input unit 101 is input to the frame dividing unit 111. The frame dividing unit 111 divides the input acoustic signal into frames having a predetermined time length. Specifically, a signal is cut out by a window function for later processing, and is output to the FFT processing unit 112, the convolution FFT processing unit 114, and the delay processing unit 120. The FFT processing unit 112 performs FFT processing on the input signal, and outputs the frequency spectrum obtained thereby to the fundamental / harmonic frequency detection unit 113. The fundamental / harmonic frequency detection unit 113 detects the fundamental frequency f _b and the harmonic frequency f _N from the input frequency spectrum and outputs them to the level detection unit 115.

  On the other hand, the convolution FFT processing unit 114 performs a convolution operation between the acoustic signal of one frame output from the frame division unit 111 and the impulse response input from the acoustic characteristic input unit 102, and an FFT process on the result of the convolution operation. The calculated frequency spectrum is output to the level detector 115.

The level detector 115 detects the fundamental level L _b and the harmonic level L _N of the input frequency spectrum from the input frequency spectrum, the fundamental frequency f _b and the harmonic frequency f _N. The detected fundamental tone level L _b and harmonic level L _N are output to the reverse rotation determination unit 116.

Reversal determination unit 116 compares the fundamental tone level L _b input harmonics level L _N, if higher than L _{b N} is present, outputs a control signal for instructing the filter generation process executed to the filter generation unit 117 To do. On the other hand, when all L _N are equal to or less than L _b , a control signal that instructs the filter generation unit 117 to stop the filter generation and application processing is output.

The filter generation unit 117 receives the control signal and performs filter generation processing. When the control signal indicates execution of filter generation processing, the following processing is performed. For example, when the output of the convolution FFT processing unit 114 is as shown in FIG. 2A, the band attenuation that attenuates the harmonic level so that Lb> L _N for all L _N as shown in FIG. 2B. A filter is generated and output to the filter application unit 118. In this embodiment, the harmonic level is attenuated in order to avoid clipping of data due to signal amplification. However, a band amplification filter for amplifying the fundamental level may be generated so that L _b > L _N. Furthermore, both a band amplification filter that amplifies the fundamental sound level and a band attenuation filter that attenuates the harmonic level may be generated so that L _b > L _N. However, care must be taken so that the data is not clipped by the amplification filter.

  The delay processing unit 120 performs delay processing in consideration of the time delay that occurs in the processing up to filter generation, and outputs the result to the filter application unit 118. The filter applying unit 118 applies the input filter to the output of the delay processing unit 120 and outputs the applied filter to the frame combining unit 119.

  When the output of the reverse rotation determination unit 116 is a control signal for instructing filter generation and application stop, the filter generation unit 117 and the filter application unit 118 do not execute the process, and the output of the frame division unit 111 is used as it is. Output to.

  Finally, the frame combination unit 119 combines the outputs of the filter application unit 118 processed for each frame and outputs the combined result to the signal output unit 104.

Next, the flow of processing in the acoustic signal processing unit 103 will be described in detail with reference to FIG.
In S201, the reproduction signal is divided in the frame dividing unit 111. That is, the time waveform signal of the reproduction signal input from the reproduction signal input unit 101 is cut out. Here, the window function that cuts out the signal uses a window function that does not increase or decrease the amount of data when the cut out signals are recombined. The processes in S202 to S211 are performed in units of frames, and the processed signals are combined in S212.

  In S <b> 202, the FFT processing unit 112 performs FFT processing to calculate a frequency spectrum for the output signal from the frame dividing unit 111. Steps S <b> 203 to S <b> 205 are processed by the fundamental / harmonic frequency detection unit 113. The fundamental frequency detection, harmonic frequency calculation, and determination of the presence or absence of undetected peaks in S203 to S205 will be described with reference to FIG.

  First, in S203, maximum peak frequency detection is performed. The maximum value of the amplitude level is detected from the frequency spectrum (FIG. 4A) calculated by the process of S202, and when the value is equal to or higher than the predetermined amplitude level, the frequency of the maximum value is calculated. Here, the predetermined amplitude level is a value set for separating a signal and noise. Therefore, for example, the value may be a half value (about −6 dB) of the maximum amplitude level in FIG. 4A, may be set by the user, or may be determined using other methods. Good.

In the present embodiment, as shown in FIG. 4B, the calculated maximum frequency is set as the fundamental frequency f _b (n). Here, n indicates the number of repetitions of S203 and S204, and is f _b (1) in FIG. _4B because it is the first detection. Details such as the repetition condition will be described later in the description of S205.

In S204, the frequency N times (N = 2, 3, 4,...) With respect to the fundamental frequency f _b (n) is calculated as the _Nth harmonic frequency f _N (n).

f _N (n) = f _b (n) * N (N = 2, 3, 4,...)
Here, the harmonic frequency in the music is not necessarily an integral multiple of the fundamental frequency, and the frequency may be slightly shifted. Therefore, referring to the frequency spectrum of S202 corresponding to the calculated harmonic frequency f _N (n), when there is a spectrum peak within a predetermined range, the frequency corresponding to the peak is set as the harmonic frequency f _N (n). Also good.

In S205, it is determined whether or not there is a spectrum peak exceeding a predetermined amplitude level in addition to the fundamental frequency f _b (n) and the harmonic frequency f _N (n) in the frequency spectrum of S202. In the spectrum (FIG. 4C) obtained by removing the frequency components of the fundamental frequency f _b (1) and the harmonic frequency f _N (1) already detected from the spectrum of S202 shown in FIG. Determine if any signal exceeds the amplitude level. If it is determined that there is a signal exceeding the predetermined amplitude level, the processing of S203 and S204 is performed until there is no signal exceeding the predetermined amplitude level in S205 as shown in FIG.

  The processing from S206 to S207 is performed by the convolution FFT processing unit 114. In S206, the convolution calculation of the impulse response input from the acoustic characteristic input unit 102 and the signal divided in S201 is performed. In S207, FFT processing is performed on the result of the convolution operation in S206, and the frequency spectrum of the signal obtained by convolving the impulse response input from the acoustic characteristic input unit 102 with the signal divided in S201 is calculated. This frequency spectrum shows the result of changing the frequency spectrum of the reproduction signal under the influence of the inputted acoustic characteristics.

The process of S208 is performed by the level detection unit 115. In S208, the fundamental level L _b (n) and the harmonic level L _N in the frequency spectrum calculated in S207 for all fundamental frequencies f _b (n) and all harmonic frequencies f _N (n) detected in S203 to S205. (n) is calculated.

The process of S209 is performed by the reverse rotation determination unit 116. In S209, the fundamental tone level L _b (n) and the harmonic level L _N (n) are compared independently for each n. As a result of comparison for all n, if a reverse phenomenon is found in which the harmonic level L _N (n) is higher than the fundamental tone level L _b (n), a filter generation control signal is sent to the filter generation unit 117. Output. On the other hand, when the reverse phenomenon is not found, a control signal for stopping filter generation and filter application is output to the filter generation unit 117 and the filter application unit 118, and the processing of S210 to S211 is not performed.

The process of S210 is performed by the filter generation unit 117. In S210, the band attenuation filter (notch filter) is set so that the harmonic level L _N (n) is lower than the fundamental level L _b (n) with respect to the harmonic frequency f _N (n) causing the reverse phenomenon detected in S209. ) Is generated. As described above, the band amplification filter may be generated such that the fundamental level L _b (n) is higher than the harmonic level L _N (n) with respect to the fundamental frequency, or both of them may be generated. The filter generated here is preferably designed to have a bandwidth that does not affect the peak adjacent to the peak to be filtered.

  The process of S211 is performed by the filter application unit 118. In S211, the filter generated in S210 is applied to the signal divided in S201. Here, the output of S201 is delayed by the delay processing unit 120 in consideration of the processing time of S202 to S210.

  The process of S212 is performed by the frame combining unit 119. In S212, the outputs of S211 processed for each frame are combined. The combined signal is output to the signal output unit 104.

  According to such a configuration, since the band attenuation filter is applied only to the harmonics of the music, the jarring sound at the listening position can be generated without promoting the generation of the jarring sound on the audibility even at a place other than the listening position. There is an advantage that it can be suppressed.

<Embodiment 2>
Hereinafter, an embodiment when there are a plurality of listening positions will be described.
The configuration of the acoustic signal processing unit is the same as that in FIG. In the present embodiment, for the sake of explanation, the f characteristic (sound pressure frequency characteristic) of the listening point is represented by G _LP (f) (LP = A, B, C,...). Here, A, B, C,... Each indicate a listening position, and f indicates a frequency. When a plurality of listening points f-specific G _LP (f) are input to the acoustic characteristic input unit 102, the convolution FFT processing unit 114 independently calculates G _LP (f) and the output of the frame dividing unit 111 for each listening position LP. FFT processing is performed on the convolution operation and the operation result. In this way, the frequency spectrum is calculated independently for each listening position.

The processing from S208 to S209 is also performed independently for each listening position LP. If a reverse phenomenon is detected in S209, a filter that suppresses the reverse phenomenon is generated. That is, the filter is generated so that the harmonic level L _NLP (n) is relatively smaller than the fundamental level Lb _LP (n) at all listening positions LP. Here, the user gives priority to each listening position LP, and the harmonic level L _NLP (n) is relatively smaller than the fundamental _sound level L _bLP (n) in a place where the priority is high according to the priority. A filter may be generated. Alternatively, the filter may be generated so that the harmonic level L _NLP (n) is relatively smaller than the fundamental level L _bLP (n) at the majority of the listening positions LP.

  Thus, each processing unit operates independently for the listening position LP. In this way, even when there are a plurality of listeners, it is possible to suppress harsh sounds at each listening position.

<Embodiment 3>
FIG. 5 is a block diagram showing the configuration of the acoustic signal processing unit 103 according to the third embodiment of the present invention. A second level detection unit 215 is added to FIG. In the present embodiment, the fundamental / harmonic frequency detection unit 113 may use a method other than the fundamental detection method using maximum peak detection as the fundamental frequency detection method. For example, a scale detected using an existing scale detection (pitch detection) device may be used as a fundamental tone. Further, not only the output of the first level detection unit 115 but also the output of the second level detection unit 215 is input to the reverse rotation determination unit 116. Other configurations are the same as those of the other embodiments.

In the level detection unit 115, the fundamental level L _b (n) detection and the harmonic level L _N (n) detection processing of S208 are performed on the output of the convolution FFT processing unit 114. The level detection unit 215 performs the detection processing of the fundamental level L _sb (n) and the harmonic level L _sN (n) in S208 on the output of the FFT processing unit 112.

The reverse rotation determination unit 116 compares the fundamental level L _b (n) and the harmonic level L _N (n) for n independently of the output of the level detection unit 115. Also, the fundamental level L _sb (n) and the harmonic level L _sN (n) are compared independently for each of the outputs of the level detection unit 215.

An example of the comparison result is shown in FIG. As shown in FIG. _6A, there is one reversal phenomenon in the comparison of L _b (n) and L _N (n) that was not found from the comparison of L _sb (n) and L _sN (n). When it is found as described above, it outputs a filter generation control signal to the filter generation unit 117. On the other hand, when the reverse rotation is already detected in the output of the FFT processing unit 112 as shown in FIG. 6B or when the reverse rotation phenomenon has not occurred, the filter generation unit 117 and the filter application unit 118 are subjected to filter generation and filter application. Output a control signal to stop. Accordingly, the processing from S210 to S211 is not performed.

  The filter generation unit 117 receives the control signal from the reverse rotation determination unit 116 and generates a reverse rotation suppression filter.

As a result of the comparison, the fundamental frequency f _b (n) and the harmonic frequency f _N (n) _satisfying L _sb (n)> L _sN (n) and L _b (n) <L _N (n) Do as follows. That is, the band attenuation filter is generated so that the harmonic level L _N (n) is lower than the fundamental sound level L _b (n) as in the first embodiment. Here, the filter may be generated as follows.

When the result of the comparison is L _sb (n)> L _sN (n) and L _b (n) <L _N (n) and L _N (n)> L _sN (n) as shown in FIG. And do the following: That is, the band attenuation filter (notch filter) is generated so that L _b (n)> L _N (n) with respect to the harmonic frequency f _N (n).

Further, as shown in FIG. 6D, the result of the comparison is L _sb (n)> L _sN (n), L _b (n) <L _N (n), and L _b (n) <L _sb (n). If there is, do the following: That is, a band amplification filter (peak filter) is generated so that L _b (n)> L _N (n) with respect to the fundamental frequency f _b (n). Here, it is desirable that the correction amount, that is, the gain of the correction filter, is set so that the ratio between the fundamental level and the harmonic level after correction is substantially equal to the ratio of the cut-out signal.

According to such a configuration, the band amplification filter is applied to the fundamental tone of the music, and the band attenuation filter is applied to the harmonics. For this reason, there is an advantage that it is possible to suppress the jarring sound at the listening position without promoting the generation of the jarring sound in the sense of hearing even at a place other than the listening position. Furthermore, compared with the first embodiment, the balance between the fundamental level L _b (n) and the harmonic level L _N (n) can be made closer to the balance between the fundamental level L _sb (n) and the harmonic level L _sN (n). Therefore, more ideal correction is possible.

<Embodiment 4>
From the trial listening experiment, it has been found that the high frequency has a high correlation between the f characteristic of the speaker and the audibility, and the low frequency has a high correlation between the f characteristic measured at the listening position and the audibility. Therefore, the f characteristic of the speaker and the f characteristic measured at the listening position may be input independently. In addition, the f-feature of an audio device such as a graphic equalizer that allows the user to adjust the f-feature may be sequentially updated by the user, so it is desirable that the f-feature be input independently. In addition, when a plurality of speakers are installed, an acoustic path is different for each speaker, and thus it is necessary to input a plurality of impulse responses.

  In this way, in the present invention, the acoustic characteristic input to the acoustic characteristic input unit 102 is not limited to one, and it is not always necessary to use all the input acoustic characteristics. Therefore, the following embodiment can be adopted. Good.

FIG. 7 is a block diagram showing a configuration of a sound processing apparatus according to the fourth embodiment of the present invention.
In FIG. 7, the acoustic characteristic input unit 102 in FIG. 1C is replaced with an acoustic characteristic input unit 402. The acoustic characteristic input unit 402 includes a memory unit 412, a selection unit 413, and an impulse response calculation unit 411. Other configurations are the same as those of the first embodiment.

  For example, a PC 2 or the like is connected to the acoustic characteristic input unit 402, and an impulse response in at least a part of the acoustic path is input and stored in the memory unit 412. Here, since the plurality of acoustic characteristics as described above are stored in the memory unit 412, it is possible to specify what acoustic characteristics each acoustic characteristic, such as naming the acoustic characteristics stored in advance. Is desirable.

  The selection unit 413 selects an acoustic characteristic used for correction. Here, the selection unit 413 switches the display device that displays information specifying the acoustic characteristics stored in the memory unit 412 and which acoustic characteristics are selected so that the user can select the acoustic characteristics to be used for correction. A switch or the like may be provided. Further, the selection unit 413 may select an acoustic characteristic to be automatically used according to an input signal from the reproduction signal input unit 101. For example, when multi-channel acoustic characteristics are stored independently for each channel, the acoustic characteristics to be used may be automatically selected depending on whether the reproduction signal input to the reproduction signal input unit 101 is stereo or multi-channel. .

  The impulse response calculation unit 411 calculates and outputs an impulse response used for correction from the acoustic characteristics selected by the selection unit 413. Here, a device for inputting the priority of the acoustic characteristics selected by the impulse response calculation unit 411, the band to be used, and the like is provided so that the user can arbitrarily set those parameters and calculate the impulse response used for correction. You may do it.

  In this way, even when a plurality of acoustic characteristics are input, the user can easily select an impulse response used for correction.

<Other embodiments>
The present invention can also be realized by executing the following processing. That is, software (program) that realizes the functions of the above-described embodiments is supplied to a system or apparatus via a network or various storage media, and a computer (or CPU, MPU, etc.) of the system or apparatus reads the program. It is a process to be executed. In this case, the program and the storage medium storing the program constitute the present invention.

Claims (10)

A calculating means for calculating a fundamental frequency and a harmonic frequency of the input acoustic signal;
Input means for inputting acoustic characteristics in at least part of the acoustic path;
Arithmetic means for performing a convolution operation between the acoustic signal and the acoustic characteristics;
First level detection means for detecting a fundamental level that is an amplitude level of the fundamental frequency and a harmonic level that is an amplitude level of the harmonic frequency for a signal obtained by computation in the computation means;
Adjusting means for adjusting the fundamental level or the harmonic level so that the fundamental level becomes higher than the harmonic level when the harmonic level is higher than the fundamental level;
A sound processing apparatus comprising:   The sound processing apparatus according to claim 1, wherein the adjustment unit is an amplification unit that amplifies the fundamental tone level to a level higher than the harmonic level.   The sound processing apparatus according to claim 1, wherein the adjustment unit is an attenuation unit that attenuates the harmonic level to a level lower than the fundamental level.   The acoustic characteristic includes a sound pressure frequency characteristic for each of a plurality of listening positions, and the computing means, the first level detecting means, and the adjusting means operate independently for each of the plurality of listening positions. The sound processing device according to claim 1, wherein the sound processing device is a sound processing device. The acoustic signal further includes second level detection means for detecting the fundamental level and the harmonic level,
The adjustment means has a harmonic level detected by the second level detection means lower than the fundamental level detected by the second level detection means, and the harmonic level detected by the first level detection means. When the wave level is higher than the fundamental level detected by the first level detection means, the fundamental level detected by the second level detection means and the harmonic level detected by the second level detection means The first level detection unit is configured to attenuate the harmonic level detected by the first level detection unit or to amplify the fundamental level detected by the first level detection unit so as to approach the first level detection unit. The acoustic processing apparatus according to claim 1, wherein the detected harmonic level is adjusted to be lower than the fundamental level detected by the first level detecting means.   The sound processing apparatus according to claim 1, wherein the sound signal is a multi-channel sound signal, and the input unit selects and inputs sound characteristics for each channel.   The sound processing apparatus according to claim 1, wherein the acoustic characteristics include a sound pressure frequency characteristic of a speaker and a sound pressure frequency characteristic of a listening position.   The acoustic processing according to any one of claims 1 to 7, wherein the acoustic characteristic is a sound pressure frequency characteristic of an acoustic path followed by the acoustic signal from the acoustic processing device to a listening ear. apparatus. A calculating step in which the calculating means calculates a fundamental frequency and a harmonic frequency of the input acoustic signal;
An input step in which the input means inputs an acoustic characteristic in at least a part of the acoustic path;
A computing step in which computing means performs a convolution computation between the acoustic signal and the acoustic characteristics;
A level detecting step for detecting a fundamental level, which is an amplitude level of the fundamental frequency, and a harmonic level, which is an amplitude level of the harmonic frequency, for a signal obtained by computation in the computation step; ,
An adjusting step in which, when the harmonic level is higher than the fundamental level, an adjustment means adjusts the fundamental level or the harmonic level or both so that the fundamental level becomes higher than the harmonic level; ,
A sound processing method comprising:   The program for functioning a computer as each means which the sound processing apparatus of any one of Claims 1 thru | or 8 has.

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