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JP2008197284

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DESCRIPTION JP2008197284
An acoustic characteristic can be accurately corrected even when the number of taps is limited.
In a filter coefficient calculation apparatus 20 according to the present invention, a gain
correction characteristic calculation unit 3 calculates an impulse response corresponding to a
linear phase filter having an inverse characteristic of a gain characteristic of a reproduction
system 17, and in advance, an impulse response is calculated. A temporally continuous impulse
response including peak values, the number of which is the same as the number of filter taps set,
is extracted, and the frequency characteristic of the impulse response is calculated as a gain
correction characteristic. Then, the phase correction characteristic calculation unit 4 normalizes
the gain characteristic of the inverse characteristic from the inverse characteristic of the
frequency characteristic of the reproduction system 17 to calculate the phase correction
characteristic, and the filter coefficient calculation unit 6 corrects the correction characteristic
combination unit 5. The filter coefficient of the reproduction characteristic correction filter is
calculated from the synthetic correction characteristic of the gain correction characteristic
synthesized in step and the phase correction characteristic. [Selected figure] Figure 1
Filter coefficient calculation device, filter coefficient calculation method, control program,
computer readable recording medium, and audio signal processing device
[0001]
The present invention relates to a filter coefficient calculation device, a filter coefficient
calculation method, and a control program for correcting acoustic characteristics in a listening
room or the like of audio output from an audio output device or the like to acoustic
characteristics suitable for a viewing environment using a digital filter. The present invention
relates to a computer readable recording medium and an audio signal processing device.
08-05-2019
1
[0002]
An equalizer is widely used which corrects the response characteristic of the whole reproduction
system including a speaker or the like according to the acoustic characteristic of the listening
room.
The acoustic characteristics of the listening room vary depending on the type of room and the
location of the device that plays the audio. For example, in a Western-style room with wooden
floors, the sound is more echoed, and in bedrooms with large furniture such as beds, the sound is
absorbed but in cases such as tatami rooms without large furniture, the sound is not absorbed
much, and The response is also small. In addition, the acoustic characteristics of the entire
listening room are different between when the speakers are installed parallel to the wall of the
room and when installed at the corners of the room. And an equalizer correct | amends the
output audio | voice to the sound quality suitable for the viewing-and-listening environment in
which acoustic characteristics differ using the filter for sound field control.
[0003]
As a conventional technique for adjusting the sound quality and correcting the response
characteristic of the entire reproduction system, for example, Patent Document 1 discloses an
acoustic characteristic correction apparatus that allows a user to easily set a desired response
characteristic in the reproduction system as a desired characteristic. Is disclosed.
[0004]
The acoustic characteristic correction device described in Patent Document 1 will be described in
more detail below.
FIG. 14 is a diagram showing various characteristics in each process in the case of correcting the
acoustic characteristics in the acoustic characteristic correction device described in Patent
Document 1. As shown in FIG. In the acoustic characteristic correction device described in Patent
Document 1, first, signals for measurement such as a band signal and a TSP signal are
reproduced by a speaker included in a reproduction system to be corrected, and collected by a
microphone, and response characteristics thereof That is, the measurement characteristic (see
FIG. 14A) is calculated. Next, the difference between the desired characteristic (see FIG. 14B) set
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by the user and the measurement characteristic is calculated as a correction characteristic (see
FIG. 14C), and correction is performed as necessary. Further, the determined correction
characteristic is inverse Fourier transformed to obtain a corresponding impulse response (see
FIG. 14D), and the level value at each position on the time axis of the determined impulse
response is equalized by an equalizer (FIR (Finite Impulse Response) )) Set as a filter coefficient.
[0005]
Patent Document 1 describes an embodiment in which linear phase processing inverse Fourier
transform is adopted as a method of obtaining an impulse response from the correction
characteristic by inverse Fourier transform.
[0006]
In the linear phase processing inverse Fourier transform, the correction characteristic is divided
into bands, the power average for each band is calculated, and the power average value is
interpolated into data of 4096 points that can be subjected to Fourier transform by spline
interpolation or the like. Inverse Fourier transform is performed on data in complex format (all
imaginary parts are 0) in which data is set as the real part to calculate an impulse response.
Here, the real part of the above complex form corresponds to an amplitude term, and the
imaginary part corresponds to a phase term. Then, as described above, since all imaginary parts
corresponding to phase terms are set to 0 in complex format data, phase information is not
included in an impulse response obtained by linear phase processing inverse Fourier transform.
[0007]
However, since the filter obtained by linear phase processing inverse Fourier transform, that is,
the linear phase filter does not include phase information, calculation of the filter coefficient is
easy, and although the frequency transfer characteristic is good, the reproduction system It is not
possible to correct the phase shift that occurs.
[0008]
On the other hand, there is a method of correcting the acoustic characteristics of the
reproduction system using an inverse filter including phase information.
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Non-Patent Document 1 describes a method of designing an inverse filter.
[0009]
The outline of the inverse filter will be described below. Assuming that the transfer characteristic
of the regeneration system is C (z), the inverse filter H (z) is represented by H (z) = 1 / C (z). This
equation represents that the output of the reproduction system is made equivalent to the input
by introducing the inverse filter H (z). That is, H (z) is designed such that the impulse response in
the reproduction system is unit impulse (delta function δ (n)). However, since a normal
reproduction system is not a minimum phase shift system but includes a propagation delay, it is
designed by changing the impulse response to δ (n−M). Here, M is called a modeling delay.
[0010]
Further, although H (z) = 1 / C (z) can not be solved as it is depending on the transfer
characteristic of the reproduction system, for example, the approximation of the inverse filter can
be obtained based on the principle of least squares. Then, a generalized expression of the inverse
filter designed based on the principle of the least square described above is H (z) = C <*> (z) / C
<*> (z) C (z). Here, C (z) is represented by a complex number, and C <*> (z) represents a conjugate
complex number of C (z).
[0011]
Various other techniques have been proposed as techniques for correcting response
characteristics using filters for sound field control. For example, Patent Document 2 discloses a
loudness intelligibility improvement device capable of realizing loudness in high definition in an
environment where reverberation is likely to occur. It will be as follows if the loud-speaking
clearness improvement apparatus of patent document 2 is demonstrated in more detail. FIG. 15A
is a diagram showing a flow of processing for improving the intelligibility of the loudspeaker in
the loudspeaker intelligibility improving apparatus described in Patent Document 2. As shown in
FIG. 15, the loudness acuity improvement apparatus described in Patent Document 2 measures
an impulse response in a closed space, and determines whether the reverberation time exceeds a
predetermined time for each 1 / n band. If the reverberation time exceeds a predetermined time,
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the difference energy between the measured impulse response and the calculated impulse
response of the direct sound is calculated and stacked in the memory. FIG. 15 (b) is a diagram
showing differential energy for each 1 / n (octave) frequency band. Furthermore, after
performing determination of reverberation time and stack processing of difference energy for all
1 / n bands, an inverse transfer function is obtained based on the difference energy calculated
for each frequency band, and an equalizer parameter satisfying the transfer function Is set as a
filter. Thereby, according to the loudness intelligibility improvement device described in Patent
Document 2, the volume level in the frequency band having a long reverberation time affecting
the intelligibility can be reduced, so that the original sound quality is not significantly changed. It
is possible to realize high loudness.
[0012]
By the way, in the FIR filter, input data is sequentially delayed by delay elements (buffers), filter
coefficients preset at each delay output are multiplied by a multiplier, each multiplication output
is added by an adder, and output Is represented as a configuration to obtain That is, when
performing signal processing using an FIR filter, product-sum operation processing is performed,
and in order to realize a high-order FIR filter, it is necessary to perform many product-sum
operation processing described above. Then, for signal processing by the FIR filter, a DSP (Digital
Signal Processor) that can execute multiplication and addition in one machine cycle and can
process product-sum operation at high speed is used.
[0013]
The convolution product-sum operation in the FIR filter is expressed by the following equation. y
(n) = h0 x (n) + h1 x (n-1) + h2 x (n-2) + ... + h N x (n-N) where y (n) is the output signal value , X
(ni) (i = 0, 1, ... N) are current and past input signal values, and hi (i = 0, 1, ... N) are filter
coefficients (weights). That is, the output signal value of the FIR filter is represented by a
weighted average of current and past input signal values.
[0014]
Note that the FIR filter includes taps of the number of terms of hi · x (n−i) included in the above
equation (that is, one block composed of the delay element described above, a multiplier, and an
adder) Be The characteristic of the FIR filter is changed by changing the number of taps
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constituting the filter and the value of hi of each tap, and the frequency resolution becomes
higher as the number of taps increases, and the performance of the filter is improved.
[0015]
However, if the number of taps of the FIR filter (ie, the number of filter coefficients) increases,
the number of product-sum operations described above also increases, and processing in the DSP
increases. Therefore, a high performance DSP is required, and the cost required to construct an
FIR filter increases. Therefore, the choice of DSP to be implemented in a product needs to
consider the trade-off between performance and cost. Japanese Patent Application Laid-Open No.
6-327089 (disclosed on November 25, 1994) Japanese Patent Application Laid-Open No. 2003224898 (disclosed on August 8, 2003) http://www.sound.sie.dendai.ac.jp/dsp/Text/PDF/ Chap72.pdf (confirmed on January 25, 2007)
[0016]
As mentioned above, DSPs implemented in products are selected in consideration of the trade-off
between performance and cost. Then, the FIR filter is designed in consideration of the processing
performance of the selected DSP product-sum operation. Therefore, the number of taps of the
FIR filter (ie, the number of filter coefficients) is limited by the specification of the DSP.
[0017]
When the filter coefficient of the FIR filter is determined by the above-described inverse filter,
first, in the reproduction system to be subjected to the sound quality correction, the impulse
response is measured using the TSP method or the like, and the measured impulse response
Calculate the frequency characteristics of Then, based on the calculated frequency characteristic,
the frequency characteristic of the inverse filter is determined, and the inverse Fourier transform
is performed on the frequency characteristic of the inverse filter to obtain an impulse response
corresponding to the inverse filter (hereinafter referred to as an impulse response of the inverse
filter Ask for The impulse response of this inverse filter is set as the filter coefficient of the FIR
filter.
[0018]
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The above-described processing for obtaining the coefficients of the FIR filter is digital signal
processing, and the measurement impulse response is captured as a continuous analog signal
and then sampled and converted into a discrete digital signal. At this time, in order to include
information of high frequency components contained in the original analog signal in the digital
signal, it is necessary to narrow the sampling interval sufficiently, that is, increase the number of
samplings sufficiently. Then, based on the data representing the sampled measurement impulse
response, data representing the impulse response of the above-described inverse filter (that is,
the filter coefficient of the FIR filter) is calculated.
[0019]
At this time, the number of data representing the calculated impulse response of the inverse filter
is equal to the number of data representing the measured impulse response. Then, data
representing the calculated impulse response of the inverse filter is set as a coefficient of the FIR
filter. However, as described above, the number of taps of the FIR filter (ie, the number of filter
coefficients) is limited by the DSP specification. Therefore, it is impossible to use all the data
representing the calculated impulse response of the inverse filter as the coefficients of the FIR
filter. Therefore, the impulse response of the inverse filter is cut out, that is, only a part of data
representing the calculated impulse response of the inverse filter is extracted as the coefficient of
the FIR filter.
[0020]
However, when only a part of data representing the impulse response of the inverse filter is set
as a coefficient of the FIR filter, the data not set as a coefficient is discarded, so the performance
of the FIR filter is degraded. Therefore, when the sound quality is corrected using the FIR filter
obtained in this manner, the error of the impulse response after correction is large, and there is a
problem that a gain difference occurs in the gain frequency characteristic.
[0021]
FIG. 16 is a diagram showing an impulse response of the inverse filter calculated based on the
measurement impulse response (sampling number: 512). The data number of the impulse
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response of the inverse filter shown in FIG. 16 is 512, which is the same as the sampling number
of the measurement impulse response. Here, when the number of taps of the FIR filter is limited
to 256 according to the specification of the DSP, for example, among the impulse responses of
the inverse filter shown in FIG. It will be extracted as a factor. That is, the data in the area
surrounded by the dashed line in FIG. 16 is discarded. In this case, the amplitude of the impulse
response in the region enclosed by the one-dot chain line in FIG. 16 which is truncated is large,
and not negligibly small compared to the magnitude of the amplitude of the entire impulse
response. Therefore, even if the sound quality is corrected by the obtained FIR filter, many errors
are included in the corrected impulse response and its frequency characteristics.
[0022]
The present invention has been made in view of the above problems, and an object thereof is a
filter coefficient calculation device capable of accurately correcting acoustic characteristics even
when the number of taps of a filter is limited. It is an object of the present invention to provide a
filter coefficient calculation method, a control program, a computer readable recording medium,
and an audio signal processing device.
[0023]
A filter coefficient calculation device according to the present invention is a filter coefficient
calculation device for calculating filter coefficients of a reproduction characteristic correction
filter for correcting acoustic characteristics of a reproduction system including a sound field,
which is a gain characteristic of the reproduction system. Linear phase impulse response
calculation means for calculating an impulse response corresponding to a linear phase filter
having the inverse characteristic of the above, and the impulse responses of the same number as
the number of taps of the preset filter among the above-mentioned impulse responses Gain
correction characteristic calculation means for calculating the frequency characteristic of the
temporally continuous impulse response including as a gain correction characteristic, and the
inverse characteristic of the above reproduction system from the inverse characteristic of the
frequency characteristic of the above reproduction system to normalize the inverse characteristic
and phase correction characteristic Phase correction characteristic calculation means for
calculating the phase correction characteristic, and a synthetic correction characteristic obtained
by synthesizing the gain correction characteristic and the phase correction characteristic The
filter coefficients of the filter having, is characterized by comprising a filter coefficient calculating
means for calculating a filter coefficient of the reproduction characteristic correction filter.
[0024]
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According to the above configuration, the filter coefficient calculation device calculates the filter
coefficient of the reproduction characteristic correction filter that corrects the acoustic
characteristic of the reproduction system including the sound field.
For example, when audio is reproduced in a certain room, the transfer characteristic differs
depending on the type and the place of the room, and so on, and the acoustic characteristic such
as time characteristic and frequency characteristic of the reproduced audio is different.
Therefore, by applying a filter to the audio signal that is the source of the audio to be
reproduced, correction is made to acoustic characteristics suitable for the viewing environment,
but the filter coefficient calculation device according to the present invention configures the
filter. Calculate the filter coefficients to be
[0025]
Then, in the filter coefficient calculation device, the linear phase impulse response calculation
means calculates impulse response data corresponding to the linear phase filter as the filter
coefficient of the linear phase filter having the inverse characteristic of the gain characteristic of
the reproduction system. That is, the linear phase impulse response calculation means calculates
the filter coefficient of the filter for correcting the gain characteristic (amplitude frequency
characteristic) of the reproduction system. Here, the gain characteristic of the filter calculated by
the linear phase impulse response calculation means has the inverse characteristic of the gain
characteristic of the reproduction system, and when this filter is applied, the gain characteristic
of the reproduction system is flat. It can be close to the characteristics. Further, the filter
calculated by the linear phase impulse response calculation means is a linear phase filter, which
corrects only the gain characteristic of the reproduction system and does not change the phase
characteristic. Then, the linear phase impulse response calculation means calculates impulse
response data corresponding to the linear phase filter as filter coefficients of the linear phase
filter. At this time, the linear phase impulse response calculation means performs IDFT (Inverse
Discrete Fourier Transform) on the inverse characteristic of the gain characteristic of the
reproduction system, for the impulse response data corresponding to the linear phase filter. It
may be calculated or it may be calculated by IFFT (Inverse Fast Fourier Transform) that performs
IDFT at high speed, and is not particularly limited.
[0026]
Then, the gain correction characteristic calculation means is an impulse response represented by
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temporally continuous impulse response data including a peak value, which is the impulse
response data of the same number as the number of taps of the preset filter among the impulse
response data. The frequency characteristic of is calculated as a gain correction characteristic.
[0027]
Usually, when calculating the acoustic characteristics of the reproduction system, for example,
the impulse response is measured based on the sound actually reproduced in the reproduction
system, but the sampling interval of the measurement impulse response at that time is finer, that
is, sampling The more data, the more accurate the measurement.
Then, for example, the sampling impulse data of the measurement impulse response is subjected
to FFT (Fast Fourier Transform; fast Fourier transformation) to calculate the frequency
characteristic of the reproduction system, and based on the frequency characteristic, the
frequency characteristic of the correction filter is determined. It is IFFTed to calculate impulse
response data corresponding to the filter coefficient. Here, although the impulse response data
corresponding to the calculated filter coefficient is the same number as the sampling data of the
measurement impulse response in the above-mentioned FFT, the number of filter taps, ie, the
number of filter coefficients is It may be limited, and all impulse response data corresponding to
the filter coefficients calculated by IFFT can not be used as the filter coefficients. Therefore, it is
necessary to cut out data to be used as a filter coefficient from the data of impulse response
calculated by IFFT according to the specification of the DSP.
[0028]
Here, conventionally, when calculating the frequency characteristic of the correction filter, the
frequency characteristic of the inverse filter including the gain information and the phase
information has been calculated. In this case, the waveform of the impulse response calculated by
IFFT There is a problem that the error of correction by the finally obtained filter becomes large
because the amplitude (coefficient of the FIR filter) of the impulse response to be truncated is
large when the above extraction is performed because the spread does not converge at both
ends. The
[0029]
On the other hand, the waveform of the impulse response calculated by the gain correction
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characteristic calculation means is concentrated at the center, attenuates in left-right symmetry
about the peak value, and converges at both ends.
Therefore, when the gain correction characteristic calculation means performs the above
clipping, that is, when the impulse response data of the number of taps of the filter set in
advance is taken out of the impulse response data, the amplitude of the impulse response to be
truncated (coefficient of FIR filter The accuracy of the correction by the resulting filter is good
because
[0030]
Then, from the inverse characteristic of the frequency characteristic of the reproduction system,
the phase correction characteristic calculation means normalizes the gain characteristic of the
inverse characteristic to calculate the phase correction characteristic. That is, the phase
correction characteristic calculation means performs the phase correction characteristic by
performing normalization such that the gain in all frequency bands is 1 with respect to the
inverse characteristic of the frequency characteristic of the reproduction system including the
gain information and the phase information. Calculate That is, the phase correction characteristic
is the characteristic of the all-par filter that corrects only the phase characteristic without
changing the gain characteristic.
[0031]
Then, the filter coefficient calculation means calculates the filter coefficient of the filter having
the combined correction characteristic obtained by combining the gain correction characteristic
and the phase correction characteristic as the filter coefficient of the reproduction characteristic
correction filter. That is, the filter coefficient calculation means combines the above gain
correction characteristic and the above phase correction characteristic to calculate a combined
correction characteristic, and for the combined correction characteristic, for example, IDFT
(Inverse Discrete Fourier Transform; discrete Fourier transform) The filter coefficient of the
reproduction characteristic correction filter is calculated by performing an IFFT (Inverse Fast
Fourier Transform) or the like.
[0032]
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In this way, it is possible to calculate a reproduction characteristic correction filter corresponding
to a combined correction characteristic in which a gain correction characteristic corresponding
to a filter that corrects only the gain characteristic and a phase correction characteristic
corresponding to a filter that corrects only the phase characteristic are combined. it can. Then,
according to the reproduction characteristic correction filter, both gain correction and phase
correction can be performed.
[0033]
Therefore, according to the present invention, it is possible to reduce the amplitude (coefficient
of the FIR filter) of the impulse response which is truncated when calculating the gain correction
characteristic for correcting the gain characteristic, and to correct the gain correction
characteristic and the phase characteristic. Since the characteristics are combined, it is possible
to realize a filter that can accurately correct the acoustic characteristics even when the number
of filter taps is limited.
[0034]
A filter coefficient calculation method according to the present invention is a filter coefficient
calculation method for calculating filter coefficients of a reproduction characteristic correction
filter that corrects acoustic characteristics of a reproduction system including a sound field, and
the gain characteristic of the reproduction system A linear phase impulse response calculating
step of calculating an impulse response corresponding to a linear phase filter having an inverse
characteristic of the above, and the impulse responses of the same number as the number of taps
of the filter set in advance among The gain correction characteristic calculation step of
calculating the frequency characteristic of the temporally continuous impulse response including
as a gain correction characteristic, and the inverse characteristic of the above reproduction
system from the inverse characteristic of the frequency characteristic of the reproduction system
normalize the phase characteristic of the inverse characteristic. Obtained by combining the phase
correction characteristic calculating step of calculating The filter coefficients of a filter having the
formed correction characteristic, is characterized in that it contains a filter coefficient calculating
step of calculating a filter coefficient of the reproduction characteristic correction filter.
[0035]
According to the above configuration, the same effects as those of the filter coefficient
calculation device according to the present invention can be obtained.
[0036]
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The filter coefficient calculation device according to the present invention further includes
measurement impulse response calculation means for calculating a measurement impulse
response from sound data obtained by collecting reproduction sound reproduced based on the
measurement signal in the reproduction system. Is preferred.
[0037]
According to the above configuration, the measurement impulse response calculation means
calculates the measurement impulse response from the audio data obtained by collecting the
reproduction sound reproduced based on the measurement signal in the reproduction system.
As a result, the filter coefficient of the reproduction characteristic correction filter can be
calculated based on the impulse response actually measured in the reproduction system.
[0038]
In the filter coefficient calculation device according to the present invention, an exponential
decay window is applied to the exponential decay window such that the reverberation energy of
the measure impulse response becomes smaller than the preset threshold in the measure time
determined in advance. Preferably, the phase correction characteristic calculation unit calculates
an inverse characteristic of the frequency characteristic of the reproduction system from the
exponential attenuation impulse response.
[0039]
According to the above configuration, the attenuation means calculates the exponential decay
impulse response by applying the exponential decay window such that the reverberation energy
of the measurement impulse response becomes smaller than the preset threshold at a
predetermined measurement time. Do.
Then, the phase correction characteristic calculation means calculates an inverse characteristic of
the frequency characteristic of the reproduction system from the exponential decay impulse
response.
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[0040]
As a result, the phase correction characteristic can be calculated based on the impulse response
of the waveform that has sufficiently converged, so that when the reproduction characteristic
correction filter is calculated from the combined correction characteristic, the aliasing
phenomenon occurring due to the influence of cyclic convolution can be reduced. It will be.
Therefore, the reproduction characteristic correction filter can improve the accuracy of the
correction of the acoustic characteristic.
[0041]
The filter coefficient calculation device according to the present invention further comprises
attenuation determination means for determining whether or not the reverberation energy of the
measurement impulse response is smaller than the threshold at the measurement time, and the
attenuation means is the attenuation determination means. Preferably, the exponential decay
window is applied to the measurement impulse response when it is determined that the
reverberation energy of the measurement impulse response is not smaller than the threshold at
the measurement time.
[0042]
According to the above configuration, the attenuation determination means determines whether
the reverberation energy of the measurement impulse response is smaller than the threshold at
the measurement time.
Then, the attenuation means applies the exponential attenuation window when it is determined
that the reverberation energy of the measurement impulse response is not smaller than the
threshold in the measurement time in the attenuation determination means.
This makes it possible to execute processing for applying an exponential decay window as
required.
[0043]
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Preferably, the filter coefficient calculation device according to the present invention further
comprises filter tap number setting changing means for changing the setting of the number of
taps.
[0044]
According to the above configuration, the filter tap number setting changing means can change
the number of taps of the set filter in accordance with the user's designation.
Further, if it is possible to acquire information representing the number of taps of a filter that can
be handled from the DSP, the setting can be changed according to the information of the
acquired number of taps.
[0045]
An audio signal processing device according to the present invention calculates the filter
coefficient calculation device according to any one of claims 1 to 5 and the filter coefficient
calculation means for an audio signal input from the audio signal input device. And a convolution
unit that performs convolution operation processing of the filter coefficients of the reproduction
characteristic correction filter and supplies the convolution unit to an audio output device.
[0046]
According to the above configuration, in the audio signal processing device according to the
present invention, in the filter coefficient calculation device, the filter coefficient calculation
means calculates the filter coefficient of the reproduction characteristic correction filter.
Then, the convolution operation device performs convolution operation processing of the filter
coefficient of the reproduction characteristic correction filter on the audio signal input from the
audio signal input device, and supplies the audio signal to which the synthetic correction
characteristic is provided to the audio output device. Do.
[0047]
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As a result, the audio signal processing device according to the present invention can apply the
synthesis correction characteristic to the audio signal using the reproduction characteristic
correction filter generated in the filter coefficient calculation device.
Therefore, according to the audio signal processing device according to the present invention,
even when the number of taps of the filter is limited, the acoustic characteristics in the
reproduction system can be corrected with high accuracy.
[0048]
The above-mentioned filter punishment calculation device may be realized by a computer. In this
case, a control program for realizing the filter coefficient calculation device in a computer by
operating the computer as the above respective means, and a computer readable recording
medium recording the control program also fall within the scope of the present invention.
[0049]
A filter coefficient calculation device according to the present invention is a filter coefficient
calculation device for calculating filter coefficients of a reproduction characteristic correction
filter for correcting acoustic characteristics of a reproduction system including a sound field,
which is a gain characteristic of the reproduction system. Linear phase impulse response
calculation means for calculating an impulse response corresponding to a linear phase filter
having the inverse characteristic of the above, impulse responses of the same number as the
number of taps of the filter set in advance among the above-mentioned impulse response data
The gain correction characteristic calculating means calculates the frequency characteristic of
the temporally continuous impulse response including the signal as a gain correction
characteristic, and the inverse characteristic of the reproduction system from the inverse
characteristic of the above reproduction system, normalizes the gain characteristic of the inverse
characteristic and corrects the phase Phase correction characteristic calculation means for
calculating characteristics, and a synthetic complement obtained by synthesizing the above gain
correction characteristic and the above phase correction characteristic The filter coefficients of a
filter having a characteristic, and a filter coefficient calculating means for calculating a filter
coefficient of the reproduction characteristic correction filter.
[0050]
Further, a filter coefficient calculation method according to the present invention is a filter
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coefficient calculation method for calculating filter coefficients of a reproduction characteristic
correction filter that corrects acoustic characteristics of a reproduction system including a sound
field, A linear phase impulse response calculation step of calculating impulse response data
corresponding to a linear phase filter having an inverse characteristic of gain characteristics; and
impulse responses of the same number as the number of taps of a preset filter among the
impulse responses, The gain characteristic of the inverse characteristic is normalized from the
inverse characteristic of the frequency characteristic of the above reproduction system, and the
gain correction characteristic calculating step of calculating the frequency characteristic of the
temporally continuous impulse response including the peak value as the gain correction
characteristic A phase correction characteristic calculating step of calculating phase correction
characteristics, combining the gain correction characteristic and the phase correction
characteristic Filter coefficients of a filter having a synthetic correction characteristic obtained Te
a, and a filter coefficient calculating step of calculating a filter coefficient of the reproduction
characteristic correction filter.
[0051]
Therefore, the amplitude (coefficient of the FIR filter) of the impulse response which is discarded
when calculating the gain correction characteristic for correcting the gain characteristic can be
reduced, and the gain correction characteristic and the phase correction characteristic for
correcting the phase characteristic are combined. Since the reproduction characteristic
correction filter is calculated using the combined correction characteristic, it is possible to realize
a filter that can accurately correct the acoustic characteristic even when the number of taps of
the filter is limited.
[0052]
One embodiment of an acoustic characteristic correction device 1 according to the present
invention will be described below with reference to FIGS. 1 to 13.
[0053]
(Acoustic Characteristic Correction Device 1) FIG. 1 is a block diagram showing a configuration of
an acoustic characteristic correction device 1 (audio signal processing device) according to the
present invention.
The acoustic characteristic correction device 1 according to the present invention includes an
acoustic characteristic measurement unit 2 (measurement impulse response calculation unit), a
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gain correction characteristic calculation unit 3 (linear phase impulse response calculation unit,
gain correction characteristic calculation unit), and a phase correction characteristic calculation
unit 4 (phase correction characteristic calculation unit, attenuation unit, attenuation
determination unit), correction characteristic combination unit 5 (filter coefficient calculation
unit), filter coefficient calculation unit 6 (filter coefficient calculation unit), convolution operation
unit 7 (convolution operation unit) And a tap number changing unit 18 (filter tap number setting
changing means).
[0054]
Here, the gain correction characteristic calculation unit 3, the phase correction characteristic
calculation unit 4, the correction characteristic combination unit 5, and the filter coefficient
calculation unit 6 constitute a filter coefficient calculation unit 20 (filter coefficient calculation
device).
[0055]
Further, the acoustic characteristic correction device 1 includes an acoustic characteristic
together with the storage device 8, the microphone 9, the AD converter 10, the source device 11
(audio signal input device), the DA converter 12, the amplifier 13 and the speaker 14 (audio
output device). The correction system 15 is configured.
[0056]
FIG. 2 is a diagram showing a connection state of various devices and the reproduction system
17 which is an object to which the acoustic characteristic is to be corrected in the present
embodiment.
The reproduction system 17 is configured to include the speaker 14 and the listening room 16.
Although illustration of AD converter 10, DA converter 12, and the memory | storage device 8 is
abbreviate | omitted in FIG. 2, it connects with the acoustic characteristic correction apparatus 1,
and comprises the acoustic characteristic correction system 15 like FIG. It is assumed that
Further, in the example shown in FIG. 2, two microphones of the microphone 9a and the
microphone 9b are disposed, but may be one, and there is no particular limitation.
08-05-2019
18
[0057]
The acoustic characteristic correction device 1 corrects the acoustic characteristic of the
reproduction system 17 configured to include the speaker 14 and the listening room 16.
For example, according to the acoustic characteristic correction device 1, it is possible to correct
response characteristics in the time domain such as impulse response, and response
characteristics in the frequency domain obtained by frequency analysis of the impulse response.
The operation of each part constituting the acoustic characteristic correction device 1 will be
described below.
[0058]
The microphone 9 collects voice, converts it into an analog electrical signal, and outputs it to the
AD converter 15. The AD converter 15 converts an analog audio signal representing the audio
input through the microphone into a digital audio signal and outputs the digital audio signal to
the acoustic characteristic measurement unit 2.
[0059]
The acoustic characteristic measurement unit 2 measures the acoustic characteristic of the
reproduction system 17. That is, the acoustic characteristic measurement unit 2 acquires
acoustic characteristic data of the reproduction system 17 based on the audio signal input
through the microphone 9. Then, the acoustic characteristic measurement unit 2 supplies the
acquired acoustic characteristic data to the gain correction characteristic calculation unit 3 and
the phase correction characteristic calculation unit 4. In the present embodiment, the acoustic
characteristic measurement unit 2 measures an impulse response in the measurement of acoustic
characteristics. The impulse response is preferably measured by a TSP (Time Stretched Pulse)
method or a cross spectrum method, but may be measured by a single pulse, and is not
particularly limited. The impulse response measured by the acoustic characteristic measurement
unit 2 is hereinafter referred to as a measurement impulse response.
08-05-2019
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[0060]
The measurement of the measurement impulse response will be described more specifically as
follows. Below, the case where it measures by TSP method is demonstrated to an example. In the
measurement of impulse response by the TSP method, a TSP signal is used. The TSP signal is
stored in the storage unit 8. In addition, the inverse TSP waveform used to convert the response
of the TSP signal into an impulse response is also stored in the storage unit 8. The reverse TSP
waveform is a waveform in which the TSP waveform is reversed in time. Then, when measuring
the impulse response, the acoustic characteristic measurement unit 2 reads the TSP signal from
the storage device 8 and reproduces it through the speaker 14. The sound represented by the
reproduced TSP signal is collected by the microphone 9 and the sound wave form is stored in the
storage unit 8. Then, the waveform of the measurement impulse response can be obtained by
performing a convolution operation of the sound wave form stored in the storage device 8 and
the above-mentioned inverse TSP signal. The convolution operation may be performed in the
convolution unit 7.
[0061]
Although FIG. 2 shows a configuration in which two microphones of the microphone 9a and the
microphone 9b are installed, it is not necessary to measure by two, and one of the microphone
9a and the microphone 9b is used. The measurement impulse response may be measured, and is
not particularly limited.
[0062]
The gain correction characteristic calculation unit 3 creates a gain correction FIR filter based on
the acoustic characteristic data (hereinafter, measured impulse response data) supplied from the
acoustic characteristic measurement unit 2.
The gain correction FIR filter is a filter that corrects only the frequency characteristic of the
amplitude without changing the frequency characteristic of the phase. Here, to create the gain
correction FIR filter more specifically means to calculate the frequency characteristic of the gain
correction FIR filter (hereinafter referred to as gain correction characteristic). Then, the gain
correction characteristic calculation unit 3 outputs data representing the gain correction
characteristic to the correction characteristic combining unit 5. Details of the gain correction
08-05-2019
20
characteristic calculation unit 3 will be described later.
[0063]
The phase correction characteristic calculation unit 4 creates a phase correction FIR filter based
on the acoustic characteristic data (that is, measurement impulse response data) supplied from
the acoustic characteristic measurement unit 2. The phase correction FIR filter is a filter that
corrects only the frequency characteristic of the phase without changing the frequency
characteristic of the amplitude. Here, to create a phase correction FIR filter more specifically
means to calculate the frequency characteristic of the phase correction FIR filter (hereinafter
referred to as a phase correction characteristic). Then, the phase correction characteristic
calculation unit 4 outputs data representing the phase correction characteristic to the correction
characteristic combination unit 5. Details of the phase correction characteristic calculation unit 4
will be described later.
[0064]
The correction characteristic combining unit 5 combines the gain correction characteristic and
the phase correction characteristic to create an FIR filter that corrects the acoustic characteristic
of the reproduction system. Here, creating a filter means, more specifically, calculating a
frequency characteristic of the filter (hereinafter referred to as a combined correction
characteristic). That is, the correction characteristic combining unit 5 combines the above gain
correction characteristic and phase correction characteristic to calculate a combined correction
characteristic, and outputs data representing the combined correction characteristic to the filter
coefficient calculation unit 6.
[0065]
The filter coefficient calculation unit 6 performs inverse Fourier transform (more specifically,
IDFT or IFFT) on the data representing the combined correction characteristic to calculate an
impulse response corresponding to the combined correction characteristic. Each level value on
the time axis of the impulse response corresponding to the combined correction characteristic is
set as a coefficient of the FIR filter that corrects the acoustic characteristic of the reproduction
system. The filter coefficient calculation unit 6 stores data of each level value on the time axis,
which is a coefficient of the FIR filter, in the storage device 8. The filter coefficient calculation
08-05-2019
21
unit 6 can also directly output the data representing the coefficients of the FIR filter to the
convolution operation unit 7.
[0066]
The convolution operation unit 7 applies a synthetic correction characteristic to the audio signal
input from the source device 11, that is, performs convolution operation of the coefficient of the
FIR filter and the audio data, and outputs the audio signal to which the synthetic correction
characteristic is applied. The data is output to the DA converter 12.
[0067]
The DA converter 12 converts the digital audio signal input from the convolution unit 7 into an
analog audio signal and outputs the analog audio signal to the amplifier 13.
The amplifier 13 amplifies an analog audio signal input from the DA converter 12 and outputs
the amplified audio signal to the speaker 14. The speaker 14 converts the amplified analog voice
signal input from the amplifier 13 into voice and outputs the voice.
[0068]
The function of each unit constituting the acoustic characteristic correction device 1 is realized
by the CPU performing processing in accordance with various programs developed in the
memory in cooperation with the operating system. Note that some or all of the functions of the
components that make up the acoustic characteristic correction device 1 may be realized only by
the various programs developed in the CPU and the memory without the intervention of the
operating system. The operating system and various programs are stored in the storage device 8
and read and executed by the CPU. Similarly, various data used in the process executed by the
acoustic characteristic correction device 1 is also stored in the storage device 8 and read by the
CPU as needed.
[0069]
FIG. 3 is a flowchart showing an outline of a flow of processing for correcting acoustic
08-05-2019
22
characteristics performed in the acoustic characteristic correction device 1 according to the
present embodiment. The outline of the flow of the process of correcting the acoustic
characteristic in the acoustic characteristic correction device 1 will be described below with
reference to FIG.
[0070]
First, the acoustic characteristic measurement unit 2 measures an impulse response (that is, the
above-described measurement impulse response) by the TSP method, the cross spectrum
method, or the like (S301).
[0071]
Next, the gain correction characteristic calculation unit 3 creates a gain correction FIR filter
based on the measured impulse response measured in S301 (S302).
More specifically, the gain correction characteristic calculation unit 3 calculates the abovementioned gain correction characteristic.
[0072]
Next, the phase correction characteristic calculation unit 4 creates a phase correction FIR filter
based on the measured impulse response measured in S301 (S303). More specifically, the phase
correction characteristic calculation unit 4 calculates the above-mentioned phase correction
characteristic.
[0073]
Next, the correction characteristic combining unit 5 combines the gain correction characteristic
calculated in S302 and the phase correction characteristic calculated in S303 to calculate a
combined correction characteristic, and the filter coefficient calculation unit 6 calculates the
combined correction characteristic. The filter coefficients of the correction FIR filter are
calculated (S304).
[0074]
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23
Then, the convolution operation unit 7 repeats the convolution operation of the sound signal
input at the source and the filter coefficient calculated in S304 (S305).
Thereby, the sound quality of the sound reproduced based on the sound signal is adjusted. That
is, the acoustic characteristic correction device 1 corrects the acoustic characteristic of the
reproduction system.
[0075]
(Gain Correction Characteristic Calculation Unit 3) The gain correction characteristic calculation
unit 3 performs Fourier transform (more specifically, DFT, acoustic characteristic data (that is,
data representing a measurement impulse response) supplied from the acoustic characteristic
measurement unit 2. FFT) to convert into frequency characteristic data representing the
frequency characteristic Hsp of the reproduction system 17.
[0076]
FIG. 4 is a view showing various characteristics obtained when the gain correction characteristic
is calculated in the gain correction characteristic calculation unit 3. FIG. 4A is a view showing a
sampled measurement impulse response, and FIG. It is a figure which shows the frequency
characteristic of a measurement impulse response, (c) is a figure which shows the impulse
response corresponding to the inverse characteristic of the frequency characteristic of a
measurement impulse response, (d) Gain calculated in the gain correction characteristic
calculation part 3 It is a figure which shows a correction characteristic.
[0077]
Here, in the present embodiment, the number of samplings of the measurement impulse
response in the gain correction characteristic calculation unit 3 is 512.
That is, the measurement impulse response shown in FIG. 4A is represented by 512 sampling
data.
Then, the gain correction characteristic calculation unit 3 Fourier-transforms data representing
08-05-2019
24
the 512 measurement impulse responses to obtain data representing the frequency
characteristic Hsp.
[0078]
Next, the gain correction characteristic calculation unit 3 calculates the frequency characteristic |
Hsp | related to the gain of the frequency characteristic Hsp (corresponding to the gain
characteristic in the claims and hereinafter referred to as the gain frequency characteristic | Hsp
|). The gain frequency characteristic | Hsp | is expressed as an absolute value of the frequency
characteristic Hsp. More specifically, the data representing the frequency characteristic Hsp is
data corresponding to a complex number (hereinafter referred to as complex format data), and
includes real part data and imaginary part data. Accordingly, the gain correction characteristic
calculation unit 3 calculates the absolute value of the complex format data representing the
frequency characteristic Hsp as the gain frequency characteristic | Hsp |. | Hsp | is represented
by equation 1. Here, Hsp <*> is a conjugate complex number of Hsp. FIG. 4 (b) shows gain
frequency characteristics | Hsp |.
[0079]
[0080]
Next, the gain correction characteristic calculation unit 3 averages the gain frequency
characteristic | Hsp | for each predetermined bandwidth (for example, 1/3 octave, 1/6 octave,
etc.) to obtain an average gain frequency characteristic | Hsp Calculate |.
FIG. 4 (b) shows the average gain frequency characteristic | Hsp¯ |. As shown in FIG. 4B, the
average gain frequency characteristic | Hsp | shows a frequency characteristic smoothed
compared to the gain frequency characteristic | Hsp |. By performing averaging such as 1/3
octave or 1/6 octave, it is possible to obtain a gain characteristic frequency characteristic close
to human auditory characteristic. Incidentally, the inverse gain frequency characteristic Hgain to
be described later may be calculated using the gain frequency characteristic | Hsp | without
calculating the average gain frequency characteristic | Hsp 限定 |, and there is no particular
limitation.
08-05-2019
25
[0081]
Furthermore, the gain correction characteristic calculation unit 3 calculates 1 / | Hsp¯ | to
calculate an inverse gain frequency characteristic Hgain (= 1 / | Hsp ||) that shows the inverse
characteristic of the average gain frequency characteristic | Hspsp | Do. That is, assuming that k
is a discrete frequency, Hgain (k) is calculated by Hgain (k) = 1 / | Hsp ((k) |. The data
representing the inverse gain frequency characteristic Hgain is also complex format data, and all
imaginary part data is zero. The inverse gain frequency characteristic Hgain corresponds to the
inverse characteristic of the gain characteristic of the reproduction system in the claims.
[0082]
Then, the gain correction characteristic calculation unit 3 performs inverse Fourier transform on
the inverse gain frequency characteristic Hgain. The real part of the complex form data obtained
by this inverse Fourier transform represents an impulse response corresponding to the inverse
gain frequency characteristic Hgain.
[0083]
Data representing an impulse response corresponding to the inverse gain frequency
characteristic Hgain is a coefficient of an FIR filter that corrects the response characteristic
related to the gain of the reproduction system 17. The data representing the impulse response
corresponding to the inverse gain frequency characteristic Hgain corresponds to the impulse
response data corresponding to the linear phase filter in the claims.
[0084]
Then, the FIR filter corresponding to the inverse gain frequency characteristic Hgain is a filter
that corrects only the frequency characteristic of the amplitude without changing the frequency
characteristic of the phase. Such FIR filters are generally referred to as linear phase FIR filters.
[0085]
08-05-2019
26
By the way, as described above, since the sampling number of the measured impulse response
acquired by the gain correction characteristic calculation unit 3 is 512, data representing an
impulse response corresponding to the inverse gain frequency characteristic Hgain calculated by
the gain correction characteristic calculation unit 3 The number of is also 512. The impulse
response in the range surrounded by the one-dot chain line shown in FIG. 4C is represented by
512 data.
[0086]
Here, the gain correction characteristic calculation unit 3 cuts out an impulse response
corresponding to the inverse gain frequency characteristic Hgain according to the specification
of the DSP that performs the convolution operation. It will be as follows if it extracts more
concretely about extraction of an impulse response.
[0087]
In the present embodiment, the number of taps of the FIR filter is limited to 256 by the
specification of the convolution unit 7 corresponding to the DSP. Therefore, the number of taps
of the FIR filter to be calculated is set to 256, and the data of impulse response that can be finally
used as the filter coefficient of the FIR filter is limited to 256. Therefore, the gain correction
characteristic calculation unit 3 is configured to obtain 256 pieces of data continuous in time
with respect to the peak value (maximum value or minimum value) out of 512 pieces of data
representing an impulse response corresponding to the inverse gain frequency characteristic
Hgain. (Hereinafter, it is referred to as cutout data). That is, 256 pieces of data representing an
impulse response in a range surrounded by a broken line shown in FIG. As shown in FIG. 4C, in
the impulse response corresponding to the inverse gain frequency characteristic Hgain, the
amplitude is concentrated at the center and the waveform converges at both ends.
[0088]
The setting of the number of taps of the filter may be stored in, for example, the storage device 8,
and the gain correction characteristic calculation unit 3 may read out from the storage device 8.
[0089]
08-05-2019
27
By the way, as already explained as a subject, when calculating an impulse response
corresponding to a general inverse filter, it is calculated including not only frequency
characteristics (gain characteristics) related to gain but also information of frequency
characteristics (phase characteristics) related to phase. Be done.
In that case, as shown in FIG. 16, the calculated impulse response has a spread in amplitude
overall, and does not have a waveform converged at both ends. Therefore, for example, when
cutting out 256 pieces of data centering on the peak value out of 512 pieces of data representing
an impulse response, the data in the area surrounded by the dashed line in FIG. 16 is truncated.
In this case, the amplitude (the coefficient of the FIR filter) of the impulse response in the region
surrounded by the one-dot chain line in FIG. 16 which is truncated is not negligible smaller than
the amplitude (the coefficient of the FIR filter) of the entire impulse response. Therefore, even if
the sound quality is corrected by the obtained FIR filter, many errors are included in the
corrected impulse response and its frequency characteristics.
[0090]
On the other hand, the impulse response corresponding to the inverse gain frequency
characteristic Hgain calculated in the gain correction characteristic calculation unit 3 of the
acoustic characteristic correction device 1 according to the present invention is the impulse
response corresponding to the linear phase FIR filter as described above. As shown in FIG. 4C,
the amplitude is concentrated at the center, and the waveform converges at both ends.
[0091]
Therefore, when cutting out 256 data around the peak value, the impulse response data outside
the range enclosed by the broken line in FIG. The coefficient) is negligibly small compared to the
overall amplitude of the impulse response (coefficient of the FIR filter).
That is, in the clipping of the impulse response corresponding to the inverse gain frequency
characteristic Hgain, the truncated amplitude (coefficient of the FIR filter) is smaller than the
impulse response of the general inverse filter. Errors in the acoustic correction that occur are
reduced.
[0092]
08-05-2019
28
However, although an FIR filter created using only information on gain characteristics can
improve the transfer characteristics, a phase shift occurs in the time domain. Therefore, an FIR
filter that corresponds to the inverse gain frequency characteristic Hgain, and combines the
extracted FIR filter with an FIR filter that corrects only the phase characteristic in the frequency
domain.
[0093]
Therefore, the impulse response represented by the 256 pieces of cutout data is subjected to
Fourier transform and converted again into information in the frequency domain. That is, the
gain correction characteristic calculation unit 3 Fourier-transforms the 256 pieces of cutout data
into complex format data representing the frequency characteristic Hgain_256.
[0094]
Then, the gain correction characteristic calculation unit 3 calculates the frequency characteristic
| Hgain_256 | (hereinafter referred to as gain frequency characteristic | Hgain_256 |) related to
the gain of the frequency characteristic Hgain_256 by the same calculation as the gain frequency
characteristic | Hsp |.
[0095]
FIG. 4D shows the gain frequency characteristic | Hgain_256 |.
The gain frequency characteristic | Hgain_256 | shows a gain characteristic reverse to the gain
frequency characteristic shown in FIG. 4 (b). Further, FIG. 4D also shows examples of gain
frequency characteristics when the number of taps is set to 128 and 512.
[0096]
Here, the gain frequency characteristic | Hgain_256 | corresponds to an FIR filter that corrects
the gain characteristic of the reproduction system 17. That is, the gain correction characteristic
08-05-2019
29
calculation unit 3 calculates | Hgain_256 | as the gain correction characteristic.
[0097]
In the present embodiment, the number of taps of the FIR filter (that is, the number of filter
coefficients) to be finally used for the convolution operation with the audio data is preset in the
storage device 8. That is, the gain correction characteristic calculation unit 3 cuts out the impulse
response corresponding to the inverse gain frequency characteristic Hgain based on the number
of taps read from the storage device 8. The number of taps of the FIR filter used for the
convolution operation may be a configuration changeable or specifiable by the user arbitrarily,
and is not particularly limited.
[0098]
(Phase Correction Characteristic Calculation Unit 4) The phase correction characteristic
calculation unit 4 performs Fourier transform on the acoustic characteristic data (that is, data
representing the measurement impulse response) supplied from the acoustic characteristic
measurement unit 2 to obtain the frequency characteristic Hsp_w of the reproduction system.
Convert to frequency characteristic data representing
[0099]
FIG. 5 is a view showing the measurement impulse response sampled in the phase correction
characteristic calculation unit 4.
[0100]
In the present embodiment, the number of taps of the filter is set to 256, and the phase
correction characteristic calculation unit 4 reads the setting of the number of taps from the
storage device 8.
Then, when calculating the phase correction characteristic, 256 data corresponding to the
measurement impulse response are required.
[0101]
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30
Here, in the present embodiment, the sampling number of the measurement impulse response in
the phase correction characteristic calculation unit 4 is set to 64 which is 1⁄4 of the number of
taps (256) of the filter.
Set the value to 0 for the remaining 192 data needed for the Fourier transform. That is, in the
phase correction characteristic calculation unit 4, 256 data including the data obtained by
applying the exponential decay window to the 64 sampling data as data representing the
measurement impulse response and the 192 data whose value is set to 0. Is used.
[0102]
It is not necessary to cut out the data of the measurement impulse response, and the data of the
measurement impulse response may be all used with the number of samplings of the
measurement impulse response being 256, and there is no particular limitation.
[0103]
Further, in the present embodiment, the phase correction characteristic calculation unit 4 applies
an exponential decay window to the measurement impulse response in order to reduce an
aliasing phenomenon that occurs due to the influence of cyclic convolution.
Details of the cyclic convolution will be described later. The measured impulse response shown in
FIG. 5 is represented by data obtained by applying exponential decay windows to 64 sampled
data of the measured impulse response.
[0104]
The exponential decay window for reducing the aliasing phenomenon is expressed, for example,
by the equation w (n) = e <d · n / 64> (n = 0, 1,..., 63). Then, in the present embodiment, this
exponential decay window is applied to the sampled measured impulse response (represented as
hsp (n)) to calculate hsp_w (n), not hsp (n) but hsp_w (n). The phase correction characteristic is
calculated using. hsp_w (n) is calculated by the operation of hsp_w (n) = hsp (n) · w (n) (n = 0,
1,..., 63). However, it is not always necessary to use an exponential decay window, and there is no
08-05-2019
31
particular limitation.
[0105]
Then, the phase correction characteristic calculation unit 4 Fourier-transforms the data
corresponding to these 256 data measurement impulse responses to obtain data representing
the frequency characteristic Hsp_w. The data obtained here is complex format data consisting of
real part data and imaginary part data.
[0106]
Next, the phase correction characteristic calculation unit 4 calculates 1 / Hsp_w to calculate the
frequency characteristic Htemp (= 1 / Hsp_w) corresponding to the inverse filter of the
reproduction system 17. Assuming that the discrete frequency is k, it is calculated by the
calculation of Htemp (k) = Hsp_w <*> (k) / (Hsp_w <*> (k) · Hsp_w (k). Here, Hsp_w <*> (k) is a
complex conjugate of Hsp_w (k). A value is set to both real part data and imaginary part data in
complex format data representing the frequency characteristic Htemp. Here, Htemp corresponds
to the inverse characteristic of the frequency characteristic of the reproduction system in the
claims.
[0107]
Furthermore, the phase correction characteristic calculation unit 4 calculates Htemp / | Htemp |,
normalizes the frequency characteristic Htemp of the inverse filter, and calculates the frequency
characteristic Hap (= Htemp / | Htemp |). Here, the frequency characteristic Hap is represented
by complex format data, and the gain frequency characteristic | Hap | calculated as the absolute
value of the complex format data is 1 for all frequencies, and the gain is constant for all
frequencies. . That is, the frequency characteristic Hap is an all-pass filter, that is, a frequency
characteristic of a filter that corrects only the frequency characteristic of the phase without
changing the frequency of the amplitude.
[0108]
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32
The frequency characteristic Hap corresponds to an FIR filter that corrects the phase
characteristic of the reproduction system 17. That is, the phase correction characteristic
calculation unit 4 calculates the frequency characteristic Hap as the phase correction
characteristic.
[0109]
The details of the cyclic convolution will be described below. As described above, in the present
embodiment, the number of taps of the FIR filter is limited to 256 by the specification of the
convolution unit 7. Therefore, the number of taps (that is, the number of filter coefficients) of the
FIR filter to be finally synthesized is 256, and the data representing the measurement impulse
response which is necessary when performing the Fourier transform for calculating the
frequency characteristic Hsp_w The number is also 256.
[0110]
By the way, when calculating an inverse filter, the measured impulse response is subjected to
Fourier transform to obtain frequency characteristics, and inverse Fourier transform is
performed on the inverse characteristics of the obtained frequency characteristics to calculate an
impulse response corresponding to the inverse filter. Here, the Fourier transform is more
specifically a discrete Fourier transform (DFT) using a fast Fourier transform (FFT). The impulse
response corresponding to the inverse filter determined in this way is one period of a periodic
sequence of numbers which is sequentially shifted by N points in the non-periodic sequence and
repeatedly superimposed. And, if the FFT length is not set long enough, aliasing occurs due to the
influence of cyclic convolution.
[0111]
FIG. 6 is a diagram for explaining the aliasing phenomenon. A portion surrounded by a broken
line in FIG. 6 is one period of a periodic sequence, that is, an impulse response corresponding to
an inverse filter, and shows that a positive time and a negative time coexist.
[0112]
Then, in order not to generate an aliasing phenomenon due to the influence of cyclic convolution,
08-05-2019
33
it is necessary to set the FFT length long enough so that an interval of 0 occurs in the inverse
Fourier transformed response.
[0113]
Therefore, in the present embodiment, the sampling number of the measurement impulse
response is 64 and the 64th sampling point of the measurement impulse response with respect
to 256 of the number of taps of the FIR filter to be obtained (that is, corresponding to the FFT
length). The FFT length is set relatively long by applying an exponential decay window to the
measured impulse response such that the reverberation energy of the impulse response at A
decays less than -60 dB as a predetermined threshold.
The value is set to 0 for the remaining 192 data required for the Fourier transform.
[0114]
That is, as described above, in the present embodiment, this exponential decay window is applied
to the sampled measured impulse response (represented as hsp (n)) to calculate hsp_w (n), not
hsp (n). , Hsp_w (n) to calculate the phase correction characteristic.
[0115]
The exponential decay window is represented, for example, by the equation w (n) = e <d · n / 64>
(n = 0, 1,..., 63).
The reverberation energy of the impulse response can be calculated from the ratio of the energy
of the entire measured impulse response to the energy of the measured impulse response at a
certain sampling point, for example, using the square integral method used when measuring the
reverberation time. . More specifically, reverberation energy can be evaluated by the equation 2
below.
[0116]
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34
[0117]
The influence of the aliasing phenomenon decreases when S calculated by Equation 2 is -60 or
less.
Then, in the present embodiment, with respect to hsp_w (n) used for calculating the phase
correction characteristic by using the equation 2, whether or not the reverberation energy is
sufficiently attenuated at the sampling point of 1⁄4 of the number of taps Evaluate
[0118]
Here, when the attenuation of reverberation energy of hsp_w (n) is evaluated, the d of the
exponential decay window is adjusted so that the influence of the alias is reduced, that is, −60 or
less. Here, when d = 0, it is the same as no exponential decay window. However, if d is too small,
the function approaches the δ function, that is, the phase of Hsp_w approaches 0, and the phase
information decreases. In addition, -60 is a general purpose reference value calculated | required
from examination result, and it is not necessarily limited to this value.
[0119]
As a result, in the impulse response obtained by the inverse Fourier transform of the composite
correction characteristic that is finally synthesized in the correction characteristic synthesis unit
5, it is possible to reduce the aliasing phenomenon that occurs due to the influence of cyclic
convolution.
[0120]
(Synthetic Inverse Filter) In the acoustic characteristic correction device 1, the correction
characteristic synthesis unit 5 synthesizes the gain correction characteristic calculated in the
gain correction characteristic calculation unit 3 and the phase correction characteristic
calculated in the phase correction characteristic calculation unit 4. Then, the combined
correction characteristic H is calculated.
More specifically, the correction characteristic combining unit 5 calculates the combined
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35
correction characteristic H by performing an operation of | Hgain_256 | .Hap. That is, assuming
that the discrete frequency is k, the correction characteristic combining unit 5 calculates H (k) = |
Hgain_256 (k) | · Hap (k) to calculate the combined correction characteristic H (k).
[0121]
Then, the filter coefficient calculation unit 6 performs inverse Fourier transform on the combined
correction characteristic H calculated by the correction characteristic combining unit 5 to
calculate an impulse response corresponding to the combined correction characteristic H. FIG. 7
is a diagram showing an impulse response corresponding to the combined correction
characteristic H. Here, the numbers of complex format data representing | Hgain_256 | and Hap
are both 256, and the complex format data obtained by combining them and the impulse
response calculated by inverse Fourier transform thereof The number of data is also 256.
[0122]
Then, in the acoustic characteristic correction device 1 according to the present invention, an FIR
filter that uses data representing an impulse response corresponding to the synthetic correction
characteristic as a filter coefficient (corresponding to the reproduction characteristic correction
filter in the claims) The acoustic characteristic of the reproduction system 17 is corrected by the
filter). More specifically, the convolution operation unit 7 applies a synthesis correction
characteristic to the speech data by performing a convolution operation of the speech data input
from the source device 11 and the filter coefficient of the synthesis inverse filter. According to
the synthesis inverse filter, both the gain characteristic and the phase characteristic of the
reproduction system 17 can be corrected.
[0123]
Further, as described above, in the convolution unit 7 corresponding to the DSP, the number of
taps of the FIR filter that can be processed is 256. On the other hand, since the number of filter
coefficients of the synthesis inverse filter is also 256, the convolution operation unit 7 can
execute the convolution operation of the synthesis inverse filter.
[0124]
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36
Furthermore, as shown in FIG. 7, the impulse response of the synthetic inverse filter calculated
by the acoustic characteristic correction device 1 according to the present invention is obtained
by cutting 256 samples from the impulse response of the general inverse filter shown in FIG.
Since the waveform is concentrated at the center in comparison, the error after correction due to
the influence of cyclic convolution is small.
[0125]
FIG. 8 is a diagram showing an impulse response in the reproduction system 17. FIG. 8A is a
diagram showing an impulse response when the correction by the synthesis inverse filter is not
performed. FIG. 8B is a case when the correction by the synthesis filter is performed. Is a diagram
showing an impulse response of
FIG. 8B shows an example where the number of taps of the synthesis inverse filter is set to 128
and when it is set to 256. Further, in FIG. 8B, in the case of 256 taps, the impulse response in the
case of performing correction using a synthetic inverse filter calculated by averaging the gain
frequency characteristic | Hsp | The impulse response at the time of performing correction |
amendment by the synthetic | combination inverse filter calculated by octave average is shown.
[0126]
The uncorrected impulse response shown in FIG. 8 (a) shows a waveform having a period
different from that of the unit impulse, whereas the corrected impulse response shown in FIG. 8
(b) has a sharp rise. A waveform close to a unit impulse is shown. That is, correction is performed
such that the impulse response becomes unit impulse by the synthesis inverse filter. In addition,
when the number of taps of the synthesis inverse filter is 128 smaller than 256, that is, the
number of taps is 256 even when the number of data to be truncated when calculating the gain
correction characteristic in the gain correction characteristic calculation unit 3 is larger. It shows
the same impulse response as the case of.
[0127]
FIG. 9 is a diagram showing the frequency characteristics of the gain in the reproduction system
17 when the correction by the synthetic inverse filter is performed, (a) is a diagram showing the
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frequency characteristic of the gain in the entire frequency band, and (b) It is a figure which
shows the frequency characteristic of the gain in a high frequency band. FIGS. 9A and 9B show
an example where the number of taps of the synthesis inverse filter is set to 128 and when it is
set to 256. Also, in the case of 256 taps, the gain frequency characteristic | Hsp | is calculated by
averaging 1/6 octave with the frequency characteristic of the gain when correction is performed
using a synthetic inverse filter calculated by averaging 1/3 octaves. 1 shows the frequency
characteristics of the gain when correction is performed by the synthetic inverse filter.
[0128]
As shown in FIG. 9A, when the correction by the synthetic inverse filter is performed, the
frequency characteristic of the gain is flat in the entire frequency band. In addition, even when
the number of taps of the synthesis inverse filter is 128, which is smaller than 256, the same
correction effect as in the case where the number of taps is 256 (1/3 octave average) can be
obtained. Further, as shown in FIG. 9B, in the high frequency band, even when the number of
taps of the synthesis inverse filter is 128 which is smaller than 256, the number of taps is 256
(1/6 octave average) A correction effect equivalent to
[0129]
(Effects of Phase Correction and Exponential Attenuation Window) In the following, the effects of
phase correction and the effects of using an exponential attenuation window will be described in
more detail. FIG. 10 shows the measurement results of the impulse response by the microphone
9a and the microphone 9b installed in the listening room 16 constituting the reproduction
system 17, and shows the measurement results of the impulse response when the correction by
the FIR filter is not performed. FIG. As shown in FIG. 10, in the case where the correction by the
FIR filter is not performed, the impulse response shows a periodic waveform without being a unit
impulse in any measurement result by the microphone 9a and the microphone 9b.
[0130]
FIG. 11 is a diagram showing the correction effect of the acoustic characteristic in the
reproduction system 17 by the FIR filter calculated based only on the gain correction
characteristic without synthesizing the phase correction characteristic, and (a) synthesizes the
phase correction characteristic It is a figure which shows the impulse response of the FIR filter
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calculated based only on the gain correction characteristic without, and (b) is a measurement
result of the impulse response by the microphone 9a and the microphone 9b. It is a figure which
shows an impulse response. As shown in FIG. 11A, the impulse response of the filter obtained by
inverse Fourier transform of only the gain correction characteristic is such that the waveform is
concentrated at the center, and is attenuated symmetrically about the peak of the center level
value. ing. In this case, even if the clipping is performed according to the limitation of the number
of taps of the FIR filter, the amplitude (coefficient of the FIR filter) of the impulse response to be
truncated is small, and the correction error caused by the clipping is small. However, as shown in
FIG. 11B, in the FIR filter calculated based only on the gain correction characteristic, that is, in
the FIR filter calculated without combining the phase correction characteristic, the impulse
response is a sharp unit impulse of rising. is not.
[0131]
FIG. 12 shows the correction effect of the acoustic characteristic in the reproduction system 17
by the FIR filter calculated based on the composite correction characteristic which combines the
gain correction characteristic and the phase correction characteristic calculated without
adjusting with the exponential attenuation window. It is a figure which shows, (a) is a figure
which shows the measurement impulse response used when synthesize | combining a phase
correction characteristic, (b) shows attenuation of the reverberation energy in each sampling
point about the measurement impulse response shown to (a). It is a figure, and (c) is a figure
showing an impulse response of FIR filter computed based on a synthetic correction
characteristic which compounded a phase amendment characteristic computed without
performing adjustment by an exponential decay window to a gain amendment characteristic, (D)
is a measurement result of the impulse response by microphone 9a and microphone 9b, and is a
figure showing an impulse response before amendment and after amendment.
[0132]
Although the number of samples of the impulse response shown in FIG. 12A is 64, the impulse
response does not converge at the 64th sampling point.
Also, in the example shown in FIG. 12, the exponential decay window is not applied to the
measured impulse response. Therefore, as shown in FIG. 12 (b), the reverberation energy is only
attenuated to -20 db at the 64th sampling point. As a result, as shown in FIG. 12C, the impulse
response of the FIR filter calculated based on the combined correction characteristic in which the
phase correction characteristic calculated without performing the adjustment by the exponential
attenuation window is combined with the gain correction characteristic is Due to the effect of
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circular convolution, the waveform spreads throughout and does not converge at both ends.
When correction is performed using the FIR filter calculated in this manner, as shown in FIG. 12
(d), the rise is sharp compared to the impulse response by the FIR filter not including the phase
correction shown in FIG. 11 (b). Although a waveform close to unit impulse is shown, a pre-echo
is generated before the rising waveform.
[0133]
FIG. 13 shows the correction effect of the acoustic characteristic in the reproduction system 17
by the FIR filter calculated based on the composite correction characteristic which combines the
gain correction characteristic and the phase correction characteristic calculated by performing
adjustment using the exponential attenuation window. It is a figure which shows a measurement
impulse response used when synthesize | combining a phase correction characteristic, and (b) is
a figure which shows attenuation | damping of the reverberation energy in each sampling point
about the measurement impulse response shown to (a). And (c) is a diagram showing an impulse
response of the FIR filter calculated based on a combined correction characteristic in which the
phase correction characteristic calculated by performing adjustment with the exponential
attenuation window is combined with the gain correction characteristic, (d ) Is a measurement
result of the impulse response by the microphone 9a and the microphone 9b, showing the
impulse response before and after the correction.
[0134]
The number of samples of the impulse response shown in FIG. 13A is 64, and an exponential
decay window is applied so that the impulse response converges at the 64th sampling point.
Therefore, as shown in FIG. 13B, at the 64th sampling point, the energy attenuates to -60 db. As
a result, as shown in FIG. 12C, the impulse response of the FIR filter calculated based on the
combined correction characteristic obtained by combining the phase correction characteristic
calculated by performing adjustment using the exponential attenuation window with the gain
correction characteristic is cyclic. The effects of convolution are reduced and the waveform
converges at both ends. When correction is performed using the FIR filter calculated in this
manner, as shown in FIG. 13D, the unit impulse waveform with a sharper rise is obtained as
compared to the waveform of the impulse response shown in FIG. The pre-echo before the rising
waveform is suppressed.
[0135]
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In the phase correction characteristic calculation unit 4, it is not necessary to apply an
exponential decay window to the 64 sampling data of the measurement impulse response. For
example, when the impulse response waveform is sufficiently converged at the 64th sampling
point and the reverberation energy is attenuated to −60 db, the exponential attenuation window
may not be applied, and the configuration is not particularly limited.
[0136]
The phase correction characteristic calculation unit 4 determines whether the reverberation
energy is attenuated to -60 dB at the 64th sampling point based on the measured impulse
response data, and the reverberation energy is not attenuated to -60 dB. It is good also as
composition which applies an exponential decay window only in the case, and limitation is not
carried out.
[0137]
The present invention can also be expressed as follows.
[0138]
(First Configuration) In a sound quality adjustment device comprising a speaker and a
microphone, the device comprises means for acquiring gain characteristics and phase
characteristics, means for synthesizing gain characteristics and phase characteristics in the
frequency domain, and A first configuration comprising means for correcting using gain
characteristics and phase characteristics.
[0139]
Second Configuration The second configuration is characterized in that the device comprises
means for acquiring an impulse response.
[0140]
(Third Configuration) The third configuration is characterized in that the correction means is an
FIR filter whose number of taps is shorter than the impulse response duration time.
[0141]
(Fourth Configuration) A fourth configuration characterized by means for varying the tap length
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of the FIR filter.
[0142]
The present invention is not limited to the embodiments described above, and various
modifications are possible within the scope of the claims.
That is, an embodiment obtained by combining technical means appropriately modified within
the scope of the claims is also included in the technical scope of the present invention.
[0143]
Finally, each block of the acoustic characteristic correction apparatus 1 may be configured by
hardware logic or may be realized by software using a CPU as follows.
[0144]
That is, the acoustic characteristic correction device 1 includes a CPU (central processing unit)
that executes an instruction of a control program that realizes each function, a ROM (read only
memory) that stores the program, and a random access memory (RAM) that expands the
program. And a storage device (recording medium) such as a memory for storing the program
and various data.
And, the object of the present invention is a recording medium in which a program code (an
executable program, an intermediate code program, a source program) of a control program of
the correction value adjusting device 9, which is software that realizes the above-mentioned
function This can also be achieved by supplying the acoustic characteristic correction device 1
and the computer (or CPU or MPU) reading out and executing the program code stored in the
recording medium.
[0145]
Examples of the recording medium include tape systems such as magnetic tapes and cassette
tapes, magnetic disks such as floppy (registered trademark) disks / hard disks, and disks
including optical disks such as CD-ROM / MO / MD / DVD / CD-R. A system, a card system such
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as an IC card (including a memory card) / optical card, or a semiconductor memory system such
as a mask ROM / EPROM / EEPROM / flash ROM can be used.
[0146]
The acoustic characteristic correction device 1 may be configured to be connectable to a
communication network, and the program code may be supplied via the communication network.
The communication network is not particularly limited. For example, the Internet, intranet,
extranet, LAN, ISDN, VAN, CATV communication network, virtual private network, telephone
network, mobile communication network, satellite communication A network etc. are available.
Also, the transmission medium constituting the communication network is not particularly
limited. For example, IEEE1394, USB, power line carrier, cable TV line, telephone line, wired line
such as ADSL line, infrared ray such as IrDA or remote control, Bluetooth ( It is also possible to
use a radio such as a registered trademark), an 802.11 radio, an HDR, a mobile telephone
network, a satellite link, and a terrestrial digital network.
The present invention can also be realized in the form of a computer data signal embedded in a
carrier wave, in which the program code is embodied by electronic transmission.
[0147]
The filter coefficient calculation device according to the present invention can be implemented in
a device that corrects response characteristics of a sound output from an audio output device in
a listening room or the like, and can be suitably used, for example, when a room equalizer is
configured. .
[0148]
It is a block diagram showing composition of an acoustic characteristic amendment device
concerning the present invention.
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It is a figure which shows the connection state of the reproduction system used as the object
which correct | amends an acoustic characteristic in this Embodiment, and various apparatuses.
It is a flowchart which shows the outline | summary of the flow of the process which corrects the
acoustic characteristic performed in the acoustic characteristic correction apparatus which
concerns on this Embodiment.
It is a figure which shows the various characteristics calculated | required in case a gain
correction characteristic is calculated in a gain correction characteristic calculation part, (a) is a
figure which shows the measurement impulse response sampled, (b) is a frequency of a
measurement impulse response It is a figure which shows a characteristic, (c) is a figure which
shows the impulse response corresponding to the reverse characteristic of the frequency
characteristic of a measurement impulse response, (d) It is a figure which shows the gain
correction characteristic calculated in the gain correction characteristic calculation part 3. It is.
It is a figure which shows the measurement impulse response sampled in the phase correction
characteristic calculation part. It is a figure explaining an aliasing phenomenon. It is a figure
which shows the impulse response corresponding to a synthetic | combination correction
characteristic. It is a figure which shows the impulse response in a reproduction | regeneration
system, (a) is a figure which shows the impulse response when not correct | amending by a
synthetic | combination inverse filter, (b) shows the impulse response when correct | amending
by a synthetic | combination filter FIG. It is a figure which shows the frequency characteristic of
the gain in the reproduction | regeneration system at the time of performing correction |
amendment by a synthetic | combination inverse filter, (a) is a figure which shows the frequency
characteristic of the gain in a full frequency band, (b) is a gain in a high frequency band Is a
diagram showing frequency characteristics of It is a figure which is a measurement result of the
impulse response by two microphones installed in the listening room which comprises a
reproduction | regeneration system, Comprising: It is a figure which shows the measurement
result of the impulse response when not correct | amending by a FIR filter. It is a figure which
shows the correction effect of the acoustic characteristic in the reproduction | regeneration
system by the FIR filter calculated based only on the gain correction characteristic without
synthesize | combining a phase correction characteristic, (a) is a gain correction characteristic
without synthesize | combining a phase correction characteristic. FIG. 8B is a diagram showing
measurement results of impulse responses by two microphones and showing impulse responses
before and after correction. FIG. FIG. 17 is a diagram showing the correction effect of acoustic
characteristics in a reproduction system by an FIR filter calculated based on a composite
correction characteristic obtained by combining a gain correction characteristic and a phase
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correction characteristic calculated without performing adjustment using an exponential decay
window, (A) is a figure which shows the measurement impulse response used when synthesize |
combining a phase correction characteristic, (b) is a figure which shows attenuation | damping of
the reverberation energy in each sampling point about the measurement impulse response
shown to (a). c) is a diagram showing an impulse response of the FIR filter calculated based on a
combined correction characteristic in which the phase correction characteristic calculated
without performing the adjustment by the exponential decay window is combined with the gain
correction characteristic, and (d) is 2 It is a measurement result of the impulse response by two
microphones, Comprising: It is a figure which shows the impulse response before correction |
amendment and correction | amendment. FIG. 17 is a diagram showing the correction effect of
the acoustic characteristic in the reproduction system by the FIR filter calculated based on the
composite correction characteristic obtained by combining the gain correction characteristic and
the phase correction characteristic calculated by performing adjustment using the exponential
attenuation window, a) shows the measured impulse response used when combining the phase
correction characteristics, (b) shows the attenuation of reverberant energy at each sampling
point for the measured impulse response shown in (a), (c) Is a diagram showing an impulse
response of an FIR filter calculated based on a composite correction characteristic in which a
phase correction characteristic calculated by performing adjustment using an exponential
attenuation window is combined with a gain correction characteristic, and (d) shows two
microphones It is a measurement result of an impulse response by the above, and is a figure
showing an impulse response before amendment and after amendment.
It is a figure which shows the various characteristics in each process in the case of correct |
amending an acoustic characteristic in the acoustic characteristic correction apparatus of patent
document 1. FIG. It is a figure explaining the loudness intelligibility improvement apparatus of
patent document 2, (a) is a figure which shows the flow of the process which improves the
intelligibility of loudness in the loudness intelligibility improvement apparatus of patent
document 2. (B) is a figure which shows the difference energy for every 1 / n (octave) frequency
band. It is a figure which shows the impulse response of the inverse filter calculated based on the
measurement impulse response (sampling number: 512).
Explanation of sign
[0149]
1 acoustic characteristic correction device (voice signal processing device) 2 acoustic
characteristic measurement unit (measurement impulse response calculation unit) 3 gain
correction characteristic calculation unit (linear phase impulse response calculation unit, gain
correction characteristic calculation unit) 4 phase correction characteristic calculation unit (
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Phase correction characteristic calculation means, attenuation means, attenuation judgment
means) 5 correction characteristic synthesis unit (filter coefficient calculation means) 6 filter
coefficient calculation unit (filter coefficient calculation means) 7 convolution operation unit
(convolution operation device) 8 storage device 9 microphone 10 AD converter 11 source device
(audio signal input device) 12 DA converter 13 amplifier 14 speaker (audio output device) 15
acoustic characteristic correction system 16 listening room 17 reproduction system 18 tap
number changing unit (filter tap number setting changing means) 20 Filter coefficient calculation
unit (filter coefficient calculation device)
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