JP6805879B2 - Filter generator, filter generator, and program - Google Patents

Description

The present invention relates to a filter generator, a filter generator, and a program.

As a sound image localization technique, there is an out-of-head localization technique in which a sound image is localized on the outside of the listener's head using headphones. In the out-of-head localization technology, the sound image is localized out of the head by canceling the characteristics from the headphones to the ears and giving four characteristics from the stereo speakers to the ears.

In the out-of-head localization reproduction, the measurement signal (impulse sound, etc.) emitted from the speaker of 2 channels (hereinafter referred to as ch) is recorded by a microphone (hereinafter referred to as a microphone) installed in the listener's ear. Then, the processing device creates a filter based on the sound pick-up signal obtained by the impulse response. By convolving the created filter into a 2ch audio signal, out-of-head localization reproduction can be realized.

Patent Document 1 discloses a method of obtaining a set of personalized indoor impulse responses. In Patent Document 1, a microphone is installed near each ear of the listener. Then, the left and right microphones record the impulse sound when the speaker is driven.

Special Table 2008-512015

Regarding the sound quality of the sound field that is reproduced by out-of-head localization processing, the mid-range low range is insufficient, the center localization sound is faint, the vocals are deep, and so on. Was sometimes said.

This hollowing out is caused by the positional relationship between the speaker and the listener. Frequencies in which the difference between the distance from the Lch speaker to the left ear and the distance from the Rch speaker to the left ear are half wavelengths are synthesized in opposite phases. Therefore, at frequencies where the difference in distance is half a wavelength, the sound is heard quietly. In particular, since the center localization signal includes in-phase signals in Lch and Rch, they cancel each other out at the positions of both ears. Such cancellation also occurs due to the influence of indoor reflections.

Normally, when listening to speaker playback, the listener is constantly moving his head even if he thinks he is still, and this phenomenon is hard to notice. However, in the case of the out-of-head localization process, since the spatial transfer function at a certain fixed position is used, the sound synthesized in the opposite phase is presented at the frequency determined by the distance from the speaker.

The present invention has been made in view of the above points, and an object of the present invention is to provide a filter generation device, a filter generation method, and a program capable of generating an appropriate filter.

The filter generation device according to the present invention has a microphone that picks up a measurement signal output from a sound source and acquires the sound pick-up signal, and based on the sound pick-up signal, it corresponds to a transmission characteristic from the sound source to the microphone. The processing unit includes a processing unit that generates a filter, and the processing unit includes an extraction unit that extracts a first signal having a first number of samples from a sample before the boundary sample of the sound collection signal, and the first signal. A signal generation unit that generates a second signal including a direct sound from the sound source with a number of second samples larger than the number of the first samples, and the second signal in the frequency region based on the signal of A conversion unit that converts and generates a spectrum, a correction unit that increases the value of the spectrum in a band below a predetermined frequency to generate a correction spectrum, and an inverse conversion of the correction spectrum into a time region for correction. An inverse conversion unit that generates a signal, and a generation unit that generates a filter using the sound collection signal and the correction signal, and a filter value before the boundary sample is generated by the value of the correction signal. The filter value after the boundary sample and less than the second number of samples is provided with a generation unit that is generated by an added value obtained by adding the correction signal to the sound pick-up signal.

The filter generation method according to the present invention is a filter generation method that generates a filter according to the transmission characteristics by collecting the measurement signal output from the sound source with a microphone, and is a step of acquiring the sound collection signal with the microphone. And the step of extracting the first signal of the first sample number from the sample before the boundary sample of the sound collection signal, and the second including the direct sound from the sound source based on the first signal. In a band of a predetermined frequency or less, a step of generating the signal of the above with a second number of samples larger than the number of the first samples, a step of converting the second signal into a frequency region to generate a spectrum, and a step of generating a spectrum. A step of increasing the value of the spectrum to generate a correction spectrum, a step of inversely converting the correction spectrum into a time region to generate a correction signal, and a filter using the sound collection signal and the correction signal. The filter value before the boundary sample is generated by the value of the correction signal, and the filter value after the boundary sample and less than the second sample number is the sound collection signal. It is provided with a step of generating by an added value obtained by adding the correction signal to the above.

The program according to the present invention is a program for causing a computer to execute a filter generation method for generating a filter according to transmission characteristics by collecting a measurement signal output from a sound source with a microphone. Based on the step of acquiring the sound pick-up signal with the microphone, the step of extracting the first signal of the first sample number from the samples before the boundary sample of the sound pick-up signal, and the first signal. A step of generating a second signal including a direct sound from the sound source with a number of second samples larger than the number of the first samples, and a step of converting the second signal into a frequency region to generate a spectrum. A step, a step of increasing the value of the spectrum in a band below a predetermined frequency to generate a correction spectrum, a step of inversely converting the correction spectrum into a time region to generate a correction signal, and the sound collection. In the step of generating a filter using the signal and the correction signal, the filter value before the boundary sample is generated by the value of the correction signal, and is after the boundary sample and less than the second sample number. The filter value of is provided with a step of generating by an added value obtained by adding the correction signal to the sound collecting signal.

According to the present invention, it is possible to provide a filter generation device, a filter generation method, and a program capable of generating an appropriate filter.

It is a block diagram which shows the out-of-head localization processing apparatus which concerns on this embodiment. It is a figure which shows the structure of the filter generator which generates a filter. It is a control block diagram which shows the structure of the filter generation apparatus. It is a flowchart which shows the filter generation method. It is a waveform diagram which shows the sound collection signal acquired by a microphone. It is an enlarged view of the sound pickup signal for showing the boundary sample d. It is a waveform diagram which shows the direct sound signal generated based on the sample extracted from a sound collection signal. It is a figure which shows the amplitude spectrum of a direct sound signal and the amplitude spectrum after correction. It is a waveform diagram which shows the direct sound signal and the correction signal in an enlarged manner. It is a waveform figure which shows the filter obtained by the processing of this embodiment. It is a figure which shows the frequency characteristic of the corrected filter and the uncorrected filter.

In this embodiment, the filter generator measures the transmission characteristics from the speaker to the microphone. Then, the filter generator generates a filter based on the measured transmission characteristics.

An outline of the sound image localization process using the filter generated by the filter generator according to the present embodiment will be described. Here, the out-of-head localization processing, which is an example of the sound image localization processing apparatus, will be described. The extra-head localization process according to the present embodiment is to perform the extra-head localization process using an individual's spatial acoustic transmission characteristic (also referred to as spatial acoustic transfer function) and external auditory canal transmission characteristic (also referred to as external auditory canal transfer function). The ear canal transmission characteristic is the transmission characteristic from the entrance of the ear canal to the eardrum. In the present embodiment, the extra-head localization process is realized by using the spatial acoustic transmission characteristic from the speaker to the listener's ear and the inverse characteristic of the external auditory canal transmission characteristic when the headphones are worn.

The out-of-head localization processing device according to the present embodiment is an information processing device such as a personal computer, a smartphone, or a tablet PC, and is a processing means such as a processor, a storage means such as a memory or a hard disk, a display means such as a liquid crystal monitor, and the like. It is provided with input means such as a touch panel, buttons, keyboard and mouse, and output means having headphones or earphones.

Embodiment 1.
FIG. 1 shows an out-of-head localization processing device 100 which is an example of the sound field reproducing device according to the present embodiment. FIG. 1 is a block diagram of an out-of-head localization processing device. The out-of-head localization processing device 100 reproduces the sound field for the user U who wears the headphones 43. Therefore, the out-of-head localization processing device 100 performs sound image localization processing on the stereo input signals XL and XR of Lch and Rch. The stereo input signals XL and XR of Lch and Rch are analog audio reproduction signals output from a CD (Compact Disc) player or the like, or digital audio data such as mp3 (MPEG Audio Layer-3). The out-of-head localization processing device 100 is not limited to a physically single device, and some of the processing may be performed by different devices. For example, a part of the processing may be performed by a personal computer or the like, and the remaining processing may be performed by a DSP (Digital Signal Processor) built in the headphones 43 or the like.

The out-of-head localization processing device 100 includes an out-of-head localization processing unit 10, a filter unit 41, a filter unit 42, and headphones 43.

The out-of-head localization processing unit 10 includes convolution calculation units 11 to 12, 21 to 22, and adders 24 and 25. The convolution calculation units 11 to 12 and 21 to 22 perform a convolution process using the spatial acoustic transmission characteristic. Stereo input signals XL and XR from a CD player or the like are input to the out-of-head localization processing unit 10. Spatial acoustic transmission characteristics are set in the out-of-head localization processing unit 10. The out-of-head localization processing unit 10 convolves the spatial acoustic transmission characteristics with respect to the stereo input signals XL and XR of each channel. The spatial acoustic transmission characteristic may be a head-related transfer function HRTF measured by the user U's own head or auricle, or may be a dummy head or a third-party head-related transfer function. These transmission characteristics may be measured on the spot or may be prepared in advance.

The spatial acoustic transmission characteristic has four filters corresponding to the transmission characteristics Hls, Hlo, Hro, and Hrs. A filter corresponding to the four transfer characteristics can be obtained by using a filter generator described later.

Then, the convolution calculation unit 11 convolves the filter corresponding to the transmission characteristic Hls with respect to the stereo input signal XL of the Lch. The convolution calculation unit 11 outputs the convolution calculation data to the adder 24. The convolution calculation unit 21 convolves a filter corresponding to the transmission characteristic H with respect to the stereo input signal XR of Rch. The convolution calculation unit 21 outputs the convolution calculation data to the adder 24. The adder 24 adds two convolution calculation data and outputs the data to the filter unit 41.

The convolution calculation unit 12 convolves a filter corresponding to the transmission characteristic Hlo with respect to the stereo input signal XL of the Lch. The convolution calculation unit 12 outputs the convolution calculation data to the adder 25. The convolution calculation unit 22 convolves a filter corresponding to the transmission characteristic Hrs with respect to the stereo input signal XR of Rch. The convolution calculation unit 22 outputs the convolution calculation data to the adder 25. The adder 25 adds two convolution calculation data and outputs the data to the filter unit 42.

Inverse filters that cancel the headphone characteristics (characteristics between the headphone playback unit and the microphone) are set in the filter units 41 and 42. Then, the inverse filter is convolved in the reproduced signal processed by the out-of-head localization processing unit 10. The filter unit 41 convolves the inverse filter with respect to the Lch signal from the adder 24. Similarly, the filter unit 42 convolves the inverse filter with respect to the Rch signal from the adder 25. The reverse filter cancels the characteristics from the headphone unit to the microphone when the headphone 43 is attached. The microphone may be placed anywhere between the ear canal entrance and the eardrum. The inverse filter may be calculated from the result of measuring the characteristics of the user U himself / herself on the spot, or may prepare in advance an inverse filter calculated from the characteristics of headphones measured using an arbitrary outer ear such as a dummy head. ..

The filter unit 41 outputs the corrected Lch signal to the left unit 43L of the headphones 43. The filter unit 42 outputs the corrected Rch signal to the right unit 43R of the headphones 43. User U is wearing headphones 43. The headphone 43 outputs the Lch signal and the Rch signal toward the user U. As a result, the sound image localized outside the head of the user U can be reproduced.

(Filter generator)
A filter generation device that measures spatial acoustic transmission characteristics (hereinafter referred to as transmission characteristics) and generates a filter will be described with reference to FIG. FIG. 2 is a diagram schematically showing a measurement configuration of the filter generation device 200. The filter generation device 200 may be a device common to the out-of-head localization processing device 100 shown in FIG. Alternatively, a part or all of the filter generation device 200 may be a device different from the out-of-head localization processing device 100.

As shown in FIG. 2, the filter generation device 200 has a stereo speaker 5 and a stereo microphone 2. The stereo speaker 5 is installed in the measurement environment. The measurement environment may be the user U's home room, an audio system sales store, a showroom, or the like. In the measurement environment, sound is reflected by the floor and walls.

In the present embodiment, the processing device of the filter generation device 200 (not shown in FIG. 2) performs arithmetic processing for appropriately generating a filter according to the transmission characteristics. The processing device includes, for example, a music player such as a CD player. The processing device may be a personal computer (PC), a tablet terminal, a smart phone, or the like.

The stereo speaker 5 includes a left speaker 5L and a right speaker 5R. For example, a left speaker 5L and a right speaker 5R are installed in front of the listener 1. The left speaker 5L and the right speaker 5R output an impulse sound or the like for measuring an impulse response. Hereinafter, in the present embodiment, the number of speakers serving as sound sources will be described as 2 (stereo speakers), but the number of sound sources used for measurement is not limited to 2, and may be 1 or more. That is, the present embodiment can be similarly applied to a so-called multi-channel environment such as 1ch monaural or 5.1ch, 7.1ch, etc.

The stereo microphone 2 has a left microphone 2L and a right microphone 2R. The left microphone 2L is installed in the left ear 9L of the listener 1, and the right microphone 2R is installed in the right ear 9R of the listener 1. Specifically, it is preferable to install microphones 2L and 2R at positions of the left ear 9L and the right ear 9R from the entrance of the ear canal to the eardrum. The microphones 2L and 2R pick up the measurement signal output from the stereo speaker 5 and acquire the sound pick-up signal. The microphones 2L and 2R output the sound pick-up signal to the filter generation device 200 described later. The listener 1 may be a person or a dummy head. That is, in the present embodiment, the listener 1 is a concept including not only a person but also a dummy head.

As described above, the impulse response is measured by measuring the impulse sound output by the left and right speakers 5L and 5R with the microphones 2L and 2R. The filter generation device 200 stores the sound collection signal acquired based on the impulse response measurement in a memory or the like. As a result, the transmission characteristic Hls between the left speaker 5L and the left microphone 2L, the transmission characteristic Hlo between the left speaker 5L and the right microphone 2R, the transmission characteristic Hro between the right speaker 5R and the left microphone 2L, and the right speaker The transmission characteristic Hrs between the 5R and the right microphone 2R is measured. That is, the transmission characteristic Hls is acquired by the left microphone 2L collecting the measurement signal output from the left speaker 5L. The transmission characteristic Hlo is acquired by the right microphone 2R collecting the measurement signal output from the left speaker 5L. The transmission characteristic Hro is acquired by the left microphone 2L collecting the measurement signal output from the right speaker 5R. The transmission characteristic Hrs is acquired by the right microphone 2R collecting the measurement signal output from the right speaker 5R.

Then, the filter generation device 200 generates a filter according to the transmission characteristics Hls, Hlo, Hro, and Hrs from the left and right speakers 5L and 5R to the left and right microphones 2L and 2R based on the sound pick-up signal. Specifically, the filter generation device 200 corrects the transmission characteristics Hls, Hlo, Hro, and Hrs. Then, the filter generation device 200 cuts out the corrected transmission characteristics Hls, Hlo, Hro, and Hrs with a predetermined filter length, and performs a predetermined arithmetic process. By doing so, the filter generation device 200 generates the filter as a filter used for the convolution calculation of the out-of-head localization processing device 100. As shown in FIG. 1, the out-of-head localization processing device 100 uses a filter corresponding to the transmission characteristics Hls, Hlo, Hro, and Hrs between the left and right speakers 5L and 5R and the left and right microphones 2L and 2R. Perform external localization processing. That is, the out-of-head localization process is performed by convolving the filter according to the transmission characteristic into the audio reproduction signal.

Further, in the measurement environment, when the measurement signal is output from the speakers 5L and 5R, the sound pick-up signal includes a direct sound and a reflected sound. The direct sound is a sound that directly reaches the microphones 2L and 2R (ears 9L, 9R) from the speakers 5L and 5R. That is, the direct sound is the sound that reaches the microphones 2L and 2R from the speakers 5L and 5R without being reflected by the floor surface or the wall surface or the like. The reflected sound is a sound that is output from the speakers 5L and 5R, is reflected on the floor surface or the wall surface, and reaches the microphones 2L and 2R. Direct sound reaches the ear faster than reflected sound. Therefore, the sound pick-up signals corresponding to the transmission characteristics Hls, Hlo, Hro, and Hrs each include direct sound and reflected sound. Then, the reflected sound appears after the direct sound.

Next, the processing apparatus of the filter generation apparatus 200 and the processing thereof will be described in detail. FIG. 3 is a control block diagram showing a processing device 210 of the filter generation device 200. FIG. 4 is a flowchart showing processing in the processing device 210. The filter generation device 200 performs the same processing on the sound pick-up signals corresponding to the transmission characteristics Hls, Hlo, Hro, and Hrs, respectively. That is, the processing shown in FIG. 4 is performed on each of the four sound pick-up signals corresponding to the transmission characteristics Hls, Hlo, Hro, and Hrs. This makes it possible to generate a filter corresponding to the transfer characteristics Hls, Hlo, Hro, and Hrs.

The processing device 210 includes a measurement signal generation unit 211, a sound collection signal acquisition unit 212, a boundary setting unit 213, an extraction unit 214, a direct sound signal generation unit 215, a conversion unit 216, a correction unit 217, an inverse conversion unit 218, and a generation unit. It is equipped with 219.

The measurement signal generation unit 211 includes a D / A converter, an amplifier, and the like, and generates a measurement signal. The measurement signal generation unit 211 outputs the generated measurement signals to the stereo speakers 5, respectively. The left speaker 5L and the right speaker 5R each output a measurement signal for measuring the transmission characteristics. Impulse response measurement by the left speaker 5L and impulse response measurement by the right speaker 5R are performed respectively. The measurement signal may be an impulse signal, a TSP (Time Streched Pure) signal, or the like. The measurement signal includes a measurement sound such as an impulse sound.

The left microphone 2L and the right microphone 2R of the stereo microphone 2 each collect the measurement signal and output the sound collection signal to the processing device 210. The sound collection signal acquisition unit 212 acquires the sound collection signals from the left microphone 2L and the right microphone 2R (S11). The sound collection signal acquisition unit 212 includes an A / D converter, an amplifier, and the like, and the sound collection signals from the left microphone 2L and the right microphone 2R may be A / D converted and amplified. Further, the sound pick-up signal acquisition unit 212 may synchronously add the signals obtained by a plurality of measurements.

FIG. 5 shows the waveform of the sound pick-up signal. The horizontal axis of FIG. 5 corresponds to the sample number, and the vertical axis is the amplitude of the microphone (for example, the output voltage). The sample number is an integer corresponding to time, and the sample of sample number 0 is the data (sample) sampled at the earliest timing. The sound pick-up signal of FIG. 5 is acquired at a sampling frequency of FS = 48 kHz. The number of samples of the sound pick-up signal in FIG. 5 is 4096 samples. The pick-up signal includes the direct sound of the impulse sound and the reflected sound.

The boundary setting unit 213 sets the boundary sample d of the sound pick-up signal (S12). The boundary sample d is a sample that serves as a boundary between the direct sound and the reflected sound from the speakers 5L and 5R. The boundary sample d is the sample number corresponding to the boundary between the direct sound and the reflected sound, and d is an integer of 0 to 4096. As described above, the direct sound is the sound that reaches the ear of the listener 1 directly from the speakers 5L and 5R, and the reflected sound is reflected from the speakers 5L and 5R on the floor or the wall surface and the ear 2L of the listener 1. It is a sound that reaches 2R. That is, the boundary sample d corresponds to the sample of the boundary between the direct sound and the reflected sound.

FIG. 6 shows the acquired sound pick-up signal and the boundary sample d. FIG. 6 is an enlarged waveform diagram of a part (square frame A) of FIG. For example, in FIG. 6, the boundary sample d = 140.

The boundary sample d can be set by the listener 1. For example, the waveform of the sound pick-up signal is displayed on the display of the personal computer, and the listener 1 specifies the position of the boundary sample d on the display. The boundary sample d may be set by a person other than the listener 1. Alternatively, the processing device 210 may automatically set the boundary sample d. When the boundary sample d is automatically set, the boundary sample d can be calculated from the waveform of the sound collection signal. Specifically, the boundary setting unit 213 obtains the envelope of the sound collection signal by the Hilbert transform. Then, the boundary setting unit 213 sets the envelope sample immediately before the next loudest sound (near the zero cross) as the boundary sample. The pick-up signal before the boundary sample d includes the direct sound that directly reaches the microphone 2 from the sound source. The sound pick-up signal after the boundary sample d includes a reflected sound that is emitted from the sound source and then reflected and reaches the microphone 2.

The extraction unit 214 extracts samples 0 to (d-1) from the sound pick-up signal (S13). Specifically, the extraction unit 214 extracts a sample before the boundary sample of the sound pick-up signal. For example, d samples from 0 to (d-1) samples of the sound pick-up signal are extracted. Here, since the sample number d = 140 of the boundary sample, the extraction unit 214 extracts 140 samples from 0 to 139. The extraction unit 214 may extract a sample from a sample other than the sample number 0. That is, the sample number s of the first sample to be extracted is not limited to 0, and may be an integer larger than 0. The extraction unit 214 may extract samples from sample numbers s to d. The sample number s is an integer of 0 or more and less than d. Hereinafter, the number of samples extracted by the extraction unit 214 will be referred to as the first number of samples. Further, the signal of the first number of samples extracted by the extraction unit 214 is used as the first signal.

The direct sound signal generation unit 215 generates a direct sound signal based on the first signal extracted by the extraction unit 214 (S14). The direct sound signal includes the direct sound and has a sample number larger than d. The number of samples of the direct sound signal is the number of the second sample, and specifically, the number of the second sample is 2048. That is, the number of second samples is half the number of samples of the sound pick-up signal. Here, for the samples from 0 to d, the extracted samples are used as they are. The values are fixed for the samples after the boundary sample d. For example, all the samples d to 2047 are set to 0. Therefore, the number of the second sample is larger than the number of the first sample. FIG. 7 shows the waveform of the direct sound signal. In FIG. 7, the values of the samples after the boundary sample d are constant at 0. The direct sound signal is also referred to as a second signal.

The number of the second sample is 2048, but the number of the second sample is not limited to 2048. When the sampling frequency FS = 48 kHz, the number of the second samples is preferably 256 or more, and the number of the second samples is more preferably 2048 or more in order to obtain sufficient accuracy of the low frequency range. Further, it is preferable to set the number of the second sample so that the direct sound signal has a data length of 5 msec or more, and it is more preferable to set the number of the second sample so that the data length is 20 msec or more.

The conversion unit 216 generates a spectrum from a direct sound signal by FFT (Fast Fourier Transform) (S15). As a result, the amplitude spectrum and the phase spectrum of the direct sound signal are generated. A power spectrum may be generated instead of the amplitude spectrum. When the power spectrum is used, the correction unit 217 corrects the power spectrum in the step described later. The conversion unit 216 may directly convert the sound signal into data in the frequency domain by the discrete Fourier transform or the discrete cosine transform.

Next, the correction unit 217 corrects the amplitude spectrum (S16). Specifically, the correction unit 217 corrects the amplitude spectrum so as to increase the amplitude value in the correction band. The corrected amplitude spectrum is also referred to as a corrected spectrum. In the present embodiment, the phase spectrum is not corrected and only the amplitude spectrum is corrected. That is, the correction unit 217 leaves the phase spectrum as it is without correcting it.

The correction band is a band below a predetermined frequency (correction upper limit frequency). For example, the correction band is a band from the lowest frequency (1 Hz) to 1000 Hz or less. Of course, the correction band is not limited to this band. That is, the correction upper limit frequency can be set to a different value as appropriate.

The correction unit 217 sets the amplitude value of the spectrum in the correction band as the correction level. Here, the correction level is the average level of the amplitude values of 800 Hz to 1500 Hz. That is, the correction unit 217 calculates the average level of the amplitude values of 800 Hz to 1500 Hz as the correction level. Then, the correction unit 217 replaces the amplitude value of the amplitude spectrum in the correction band with the correction level. Therefore, in the correction amplitude spectrum, the amplitude value in the correction band is a constant value.

FIG. 8 shows the amplitude spectrum B before correction and the amplitude spectrum C after correction. In FIG. 8, the horizontal axis is the frequency [Hz] and the vertical axis is the amplitude [dB], which is a logarithmic display. In the corrected amplitude spectrum, the amplitude [dB] of the correction band of 1000 Hz or less is constant. Further, the correction unit 217 leaves the phase spectrum as it is without correcting it.

The band for calculating the correction level is used as the calculation band. The calculation band is a band defined by a second frequency lower than the first frequency from the first frequency. Therefore, the calculation band is a band from the second frequency to the first frequency. In the above example, the second frequency of the calculation band is 1500 Hz, and the first frequency is 800 Hz. Of course, the calculation band is not limited to the band of 800 Hz to 1500 Hz. That is, the first frequency and the second frequency that define the calculation band are not limited to 1500 Hz and 800 Hz, and can be any frequency.

It is preferable that the first frequency that defines the calculation band is higher than the upper limit frequency that defines the correction band. For the first and second frequencies, the values determined by examining the frequency characteristics of the transmission characteristics Hls, Hlo, Hro, and Hrs in advance can be used. Of course, a value other than the average level of amplitude may be used. When determining the first and second frequencies, the frequency characteristics may be displayed and the recommended frequency may be indicated to correct the dip in the mid-low range.

The correction unit 217 calculates the correction level based on the amplitude value of the calculation band. Further, although the correction level in the correction band is set as the average value of the amplitude values in the calculation band, the correction level is not limited to the average value of the amplitude values. For example, the correction level may be a weighted average of the amplitude values. Further, it does not have to be constant over the entire correction band. That is, the correction level may change according to the frequency in the correction band.

As another correction method, the correction unit 217 sets the amplitude level of the frequency lower than the predetermined frequency so that the average amplitude level at the frequency above the predetermined frequency is equal to the average amplitude level at the frequency lower than the predetermined frequency. It may be set to a constant level, or may be moved in parallel in the amplitude value direction while maintaining the general shape of the frequency characteristic. Examples of the predetermined frequency include a correction upper limit frequency.

As yet another correction method, the correction unit 217 may store the frequency characteristic data of the speaker 5L and the speaker 5R in advance, and replace the amplitude level below the predetermined frequency with the frequency characteristic data of the speaker 5L and the speaker 5R. .. Further, the correction unit 217 may store the low frequency characteristic data of the head related transfer function simulated by a hard sphere having the width of the left and right ears of a person (for example, about 18 cm) in advance, and replace it in the same manner. Examples of the predetermined frequency include a correction upper limit frequency.

Next, the inverse transform unit 218 generates a correction signal by IFFT (inverse fast Fourier transform) (S17). That is, the inverse transform unit 218 performs discrete Fourier transform on the corrected amplitude spectrum and the phase spectrum, so that the spectrum data becomes the data in the time domain. The inverse transform unit 218 may generate a correction signal by performing an inverse transform by an inverse discrete cosine transform or the like instead of the inverse discrete Fourier transform. The number of samples of the correction signal is 2048, which is the same as that of the direct sound signal. FIG. 9 shows an enlarged waveform diagram of the direct sound signal D and the correction signal E.

Then, the generation unit 219 generates a filter by using the sound collection signal and the correction signal (S18). Specifically, the generation unit 219 replaces the samples up to the boundary sample d with correction signals. For the samples after the boundary sample d, the correction signal is added to the sound collection signal. That is, the generation unit 219 generates the filter value before the boundary sample d (0 to (d-1)) based on the value of the correction signal. For the filter values after the boundary sample d and less than the second sample (d to 2047), the generation unit 219 generates the filter values by adding the correction signal to the sound pick-up signal. Further, the generation unit 219 generates the filter value of the second sample number or more and less than the sound collection signal sample number based on the sound collection signal value.

For example, the sound pickup signal is M (n), the correction signal is E (n), and the filter is F (n). Here, n is a sample number, which is an integer of 0 to 4095. The filter F (n) is as follows when n is 0 or more and less than d (when 0 ≦ n <d).
F (n) = E (n)
When n is d or more and less than the number of second samples (2048 in this case) (when d ≦ n <number of second samples)
F (n) = M (n) + E (n)
When n is equal to or greater than the number of second samples and less than the number of sound collection signal samples (4096 in this case) (when the number of second samples ≤ n <number of sound collection signal samples)
F (n) = M (n)

If the value of the correction signal E (n) when n is equal to or greater than the second sample is regarded as 0, n is equal to or greater than the number of second samples and less than the number of sound pickup signal samples (4096 in this case). Also in the case of, F (n) = M (n) + E (n). That is, when n is d or more and less than the number of sound collection signal samples (2048 in this case), it can be said that F (n) = M (n) + E (n). FIG. 10 shows a waveform diagram of the filter. The number of filter samples is 4096.

In this way, the generation unit 219 generates the filter by calculating the filter value based on the sound collection signal and the correction signal. Of course, instead of simply adding the sound collection signal and the correction signal, they may be added by multiplying them by a coefficient. FIG. 11 shows the frequency characteristics (amplitude spectrum) of the filter H generated by the above process and the uncorrected filter G. The uncorrected filter G has the frequency characteristics of the sound pick-up signal shown in FIG.

By correcting the transmission characteristics in this way, the sound field in which the center sound image is firmly localized, and the frequency characteristics in which the mid-low range and the high range are well-balanced in terms of hearing are obtained. That is, since the amplitude of the correction band in the mid-low range is increased, an appropriate filter can be generated. It is possible to reproduce a sound field in which so-called hollows do not occur. Further, even when the spatial transfer function of the head of the listener 1 at a fixed position is measured, an appropriate filter can be generated. Therefore, an appropriate filter value can be obtained even for a frequency in which the difference in distance from the sound source to the left and right ears is a half wavelength. Therefore, an appropriate filter can be generated.

Specifically, the extraction unit 214 extracts a sample before the boundary sample d. That is, the extraction unit 214 extracts only the direct sound of the sound collection signal. Therefore, the sample extracted by the extraction unit 214 shows only the direct sound. The direct sound signal generation unit 215 generates a direct sound signal based on the extracted sample. Since the boundary sample d corresponds to the boundary between the direct sound and the reflected sound, the reflected sound can be excluded from the direct sound signal.
Further, the direct sound signal generation unit 215 generates a sound pick-up signal and a direct sound signal having half the number of samples (2048 samples) of the filter. By increasing the number of samples of the direct sound signal, it is possible to accurately correct even in the low frequency range. Further, the number of samples of the direct sound signal is preferably set to the number of samples in which the direct sound signal is 20 msec or more. The sample length of the direct sound signal can be the same as the maximum sound collecting signal (transfer functions Hls, Hlo, Hro, Hrs).

The above processing is performed on four pick-up signals corresponding to the transfer functions Hls, Hlo, Hro, and Hrs. The processing device 210 is not limited to a single physical device. That is, it is also possible to perform a part of the processing of the processing device 210 by another device. For example, a sound collection signal measured by another device is prepared, and the processing device 210 acquires the sound collection signal. Then, the processing device 210 stores the sound pick-up signal in a memory or the like, and performs the above processing.

Part or all of the above processing may be executed by a computer program. The programs described above can be stored and supplied to a computer using various types of non-transitory computer readable media. Non-transient computer-readable media include various types of tangible storage media (tangible storage media). Examples of non-temporary computer-readable media include magnetic recording media (eg, flexible disks, magnetic tapes, hard disk drives), magneto-optical recording media (eg, magneto-optical disks), CD-ROMs (Read Only Memory), CD-Rs, CD-R / W, semiconductor memory (for example, mask ROM, PROM (Programmable ROM), EPROM (Erasable PROM), flash ROM, RAM (Random Access Memory)) is included. The program may also be supplied to the computer by various types of temporary computer readable media (transitory computer readable media). Examples of temporary computer-readable media include electrical, optical, and electromagnetic waves. The temporary computer-readable medium can supply the program to the computer via a wired communication path such as an electric wire and an optical fiber, or a wireless communication path.

Although the invention made by the present inventor has been specifically described above based on the embodiment, the present invention is not limited to the above embodiment and can be variously modified without departing from the gist thereof. Needless to say.

U User 1 Listener 2L Left microphone 2R Right microphone 5L Left speaker 5R Right speaker 9L Left ear 9R Right ear 10 Out-of-head localization processing unit 11 Folding calculation unit 12 Folding calculation unit 21 Folding calculation unit 22 Folding calculation unit 24 Adder 25 Addition Instrument 41 Filter unit 42 Filter unit 43 Headphones 100 Out-of-head localization processing device 200 Filter generation device 210 Processing device 211 Measurement signal generation unit 212 Sound collection signal acquisition unit 213 Boundary setting unit 214 Extraction unit 215 Direct sound signal generation unit 216 Conversion unit 217 Correction unit 218 Inverse conversion unit 219 Generation unit

Claims (5)

A microphone that collects the measurement signal output from the sound source and acquires the sound collection signal,
A processing unit that generates a filter according to the transmission characteristics from the sound source to the microphone based on the sound pick-up signal is provided.
The processing unit
An extraction unit that extracts the first signal of the first number of samples from the sample before the boundary sample of the sound collection signal, and
A signal generation unit that generates a second signal including a direct sound from the sound source with a second sample number larger than the first sample number based on the first signal.
A conversion unit that converts the second signal into a frequency domain to generate a spectrum, and
A correction unit that generates a correction spectrum by increasing the value of the spectrum in a band below a predetermined frequency.
An inverse conversion unit that generates a correction signal by inversely converting the correction spectrum into a time domain.
A generator that generates a filter using the sound pick-up signal and the correction signal, and the filter value before the boundary sample is generated by the value of the correction signal, and is after the boundary sample and the second. A filter generation device including a generation unit for generating a filter value less than the number of samples of the above by adding the correction signal to the sound collection signal. The sound pick-up signal before the boundary sample includes a direct sound directly reaching the microphone from the sound source, and the sound pick-up signal after the boundary sample is emitted from the sound source and then reflected and said. The filter generation device according to claim 1, which includes a reflected sound reaching a microphone. The filter according to claim 1 or 2, wherein the frequency band corrected by the correction unit is defined by a first frequency higher than the predetermined frequency and a second frequency lower than the first frequency. Generator. It is a filter generation method that generates a filter according to the transmission characteristics by collecting the measurement signal output from the sound source with a microphone.
The step of acquiring the sound pick-up signal with the microphone and
A step of extracting the first signal of the first number of samples from the sample before the boundary sample of the sound pick-up signal, and
A step of generating a second signal including a direct sound from the sound source based on the first signal with a second sample number larger than the first sample number.
The step of converting the second signal into the frequency domain to generate a spectrum, and
A step of increasing the value of the spectrum in a band below a predetermined frequency to generate a correction spectrum, and
A step of inversely converting the correction spectrum into a time domain to generate a correction signal,
In the step of generating a filter using the sound pick-up signal and the correction signal, the filter value before the boundary sample is generated by the value of the correction signal, and is after the boundary sample and the second. A filter generation method including a step of generating a filter value less than the number of samples by an added value obtained by adding the correction signal to the sound pick-up signal. A program that causes a computer to execute a filter generation method that generates a filter according to the transmission characteristics by collecting the measurement signal output from the sound source with a microphone.
The filter generation method is
The step of acquiring the sound pick-up signal with the microphone and
A step of extracting the first signal of the first number of samples from the sample before the boundary sample of the sound pick-up signal, and
A step of generating a second signal including a direct sound from the sound source based on the first signal with a second sample number larger than the first sample number.
The step of converting the second signal into the frequency domain to generate a spectrum, and
A step of increasing the value of the spectrum in a band below a predetermined frequency to generate a correction spectrum, and
A step of inversely converting the correction spectrum into a time domain to generate a correction signal,
In the step of generating a filter using the sound pick-up signal and the correction signal, the filter value before the boundary sample is generated by the value of the correction signal, and is after the boundary sample and is the second. A program including a step of generating a filter value less than the number of samples by an added value obtained by adding the correction signal to the sound pick-up signal.