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JP2007241134

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DESCRIPTION JP2007241134
An object of the present invention is to provide a method capable of determining an appropriate
crossover frequency even if the frequency characteristic of a speaker is uneven. A measurement
signal generation unit sequentially outputs measurement signals to an individual speaker of a
speaker system via an output unit. With respect to each speaker, the signal processing unit 1 first
performs smoothing with a smoothing width in a wide band on a signal corresponding to the
frequency characteristic obtained by the microphone 8 and roughly crosses ignoring dips that
occur partially A range of over frequencies is determined, and then narrow band smoothing is
performed to determine a highly accurate crossover frequency. [Selected figure] Figure 1
Audio signal processing method and reproduction apparatus
[0001]
The present invention performs digital signal processing on an audio signal so as to obtain an
appropriate acoustic characteristic at a listening position, taking into consideration the
characteristics of the speaker, the characteristics of the acoustic space, and the characteristics of
the audio system in a multichannel reproduction system. The present invention relates to an
audio signal processing apparatus capable of determining a crossover frequency.
[0002]
In order to provide high-quality playback sound in multi-channel playback systems, it is
necessary to set the speaker sound pressure level, speaker distance, speaker connection phase,
crossover frequency, frequency characteristics, etc. of each channel to appropriate values. .
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1
In the past, they were set manually, but it was a little difficult for the user to set them properly,
and recently devices that can set them automatically have been used . This device is called an
automatic sound field correction device, and the control unit causes the calculation unit to output
the measurement signal as the reproduction sound from the speaker, and the microphone
installed at the listening position picks up the reproduction sound of the measurement signal.
The calculation unit may calculate and set appropriate values such as the speaker sound pressure
level of each channel, the speaker distance, the speaker connection phase, the crossover
frequency, and the frequency characteristic by analyzing based on the collected sound data. It is
a device that can
[0003]
The automatic sound field correction process will be described with reference to FIGS. FIG. 7
shows a plurality of speakers around the listening point, for example, front left speaker (FL),
front light speaker (FR), center speaker (C), subwoofer speaker (SW), surround left speaker (SL),
surround light A so-called 5.1-channel multi-channel system in which speakers (SR) are arranged,
and automatic sound field correction processing is performed from each speaker so that an
optimum acoustic environment can be obtained at a listening point in such a system. It is a
process of controlling an audio signal to be output.
[0004]
Specifically, the measurement signal is converted into reproduction sound and output from each
speaker, and the reproduction sound is collected by the microphone installed at the listening
point. The measurement signal is filtered by using a filter coefficient that lowers the signal level
toward high frequency to a TSP (Time Stretched Pulse) signal which is an energy-increased signal
by stretching the impulse on the time axis. Use the applied signal. This is to prevent the speaker
(tweeter) from being destroyed when a signal having a high signal level is output at a high
frequency. The reproduced sound output from the speaker is picked up by the microphone, and
the filter coefficient (the filter coefficient of the inverse number of the filter coefficient of the
signal for measurement) for which the signal level decreases toward the low frequency with
respect to the frequency characteristic of the picked up sound Filter processing is performed and
the result is taken as a measurement result. Based on the measurement results, the sound
pressure level of each speaker, the distance between the speakers, the distance between the
listening point and the speaker, the frequency characteristic of the speakers, etc. are calculated.
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Based on the calculation result, the delay time of the audio signal to be output to each speaker,
the frequency band to be output from the speaker, and the like are controlled so that the
optimum acoustic environment can be obtained at the listening point.
[0005]
FIG. 8 is an example of the calculated frequency characteristic of each speaker, and the
horizontal axis represents the frequency and the vertical axis represents the sound pressure level
for the seven speakers described above.
[0006]
Although the method of measuring the reproduction ¦ regeneration limit frequency of a speaker
and setting the cutoff frequency of a high pass filter is described in patent document 1, this
method outputs a sine wave signal of 20 Hz-200 Hz sequentially from a transmitter, The sound
output from the speaker is collected by the microphone, and based on the detection result of the
level for each frequency, the DSP (Digital Signal Processor) plots the value of the level to
determine the reproduction limit frequency of the speaker. is there.
However, in the method described in Patent Document 1, the unevenness of the reproduction
frequency characteristic of the speaker is not taken into consideration. JP 2003-76374 A
[0007]
FIG. 9 is an explanatory diagram of the crossover frequency. The frequency band that can be
output from the speaker differs depending on the size of the speaker itself or the speaker unit.
For example, in the example shown in FIG. 8, the large front speakers (FL, FR) can output audio
signals up to the low frequency range, but the small surround speakers (SL, SR) have higher
output than the front speakers. , The low range that can be output is not low. That is, the
reproduction characteristic in the low range is bad. In such a case, as shown in FIG. 9, an audio
signal of a low frequency band that can not be reproduced by the surround speaker is added to
the audio signal of the front speaker or the subwoofer speaker by automatic sound field
correction processing and output. . As a result, there is no problem that the low-range audio
signal is not output among the audio signals of the surround speakers, and an optimum acoustic
environment can be obtained at the listening point. The crossover frequency is a frequency for
specifying a band to be cut from an audio signal supplied to a surround speaker to be output to
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3
another speaker because the surround speaker can not output.
[0008]
When calculating the crossover frequency, there is a method of obtaining the crossover
frequency by calculating an impulse response from the data collected by the measurement as
described above, and performing frequency analysis using it by using a Fourier transform. . Here,
as described above, the frequency characteristic simply subjected to the Fourier transform is
often extremely uneven, and it is difficult to find an appropriate crossover frequency. Therefore,
a signal corresponding to the Fourier transformed frequency characteristic Smoothing is
performed to smooth the characteristics with severe irregularities. Then, for the signal
corresponding to the smoothed frequency characteristic, first, the level serving as the reference
is determined by averaging the levels in the middle high band or the entire band. Next, from the
reference level, for example, a point 3 dB lower is searched from the high frequency toward the
low frequency, and the frequency 3 dB lower is taken as a crossover frequency.
[0009]
The term "smoothing" as used herein refers to averaging levels for each specific frequency band
(smoothing width). As shown in FIG. 3, smoothed frequency characteristics (solid line) can be
obtained by applying smoothing to the measured frequency characteristics (dotted line) having
irregularities of the speaker. In addition to averaging, other methods such as the least squares
method can also be used as the smoothing method.
[0010]
In the above calculation method, when smoothing the frequency characteristics, if the smoothing
width is performed in a narrow band such as 1/6 octave or 1/3 octave, even partial dips caused
by the inherent characteristics of the speaker, etc. If the dip is as deep as 3 dB or more, the point
is erroneously recognized as a crossover frequency. Also, when smoothing is performed with a
band width (smoothing width) of one octave or more, since those dips are smoothed,
misrecognition at that frequency can be avoided, but on the contrary, it was smoothed too gently.
Therefore, even if the original frequency characteristics are sharply attenuated in the crossover
frequency band, they become gentle attenuation characteristics, and a large error is generated in
the frequency of 3 dB attenuation before and after smoothing. And there is a problem that it is
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not possible to obtain an accurate crossover frequency.
[0011]
According to the present invention, a signal for measurement is supplied to a speaker under
measurement to convert it into a reproduction sound, the reproduction sound is collected by a
microphone to be an electric signal, and the frequency characteristic of the speaker under
measurement is coped with based on the electric signal. In the audio signal processing method
for obtaining a signal to be detected and determining the crossover frequency of the speaker
under measurement, smoothing with a wide band smoothing width and a narrow band
smoothing width for a signal corresponding to the frequency characteristic of the speaker under
measurement The crossover frequency is determined based on the signal corresponding to the
smoothed frequency characteristic by applying different smoothing with at least two steps of
smoothing width such as the smoothing in.
[0012]
Further, the present invention is a reproduction apparatus having a signal input unit, an output
unit, a speaker, a measurement signal generation unit, a signal processing unit, a control unit,
and an operation unit, and an instruction signal to be a processing mode is input from the
operation unit. When it is determined, the measurement signal generation unit generates a
measurement signal under the control of the control unit, and the output unit supplies the
measurement signal to the speaker under the control of the control unit to reproduce sound. The
microphone converts the reproduced sound into an electrical signal, and the signal processor
generates a signal corresponding to the frequency characteristic of the speaker based on the
electrical signal under the control of the controller. The signal is smoothed by applying at least
two different smoothing widths, such as a wide-band smoothing width smoothing and a narrowband smoothing width smoothing, to the signal. Is characterized in that what determines the
cross-over frequency based on a signal corresponding to the grayed frequency characteristics.
[0013]
According to the present invention, it is possible to avoid misrecognition in partial dip and to
determine a highly accurate crossover frequency without a large error, thereby providing an
optimum sound field environment for a listener. .
[0014]
In the present invention, the crossover frequency calculation step is obtained in two stages.
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First, as the first step, the Fourier transformed frequency characteristics are subjected to
smoothing in a wide band such as one octave, and a rough crossover frequency that avoids
misrecognition in partial dips is searched from high frequencies. Obtained at a point of 3 dB
attenuation, a certain band around that frequency is positioned as the optimal crossover
frequency range.
Next, as the second step, the narrow band such as 1/3 octave or 1/6 octave is applied to the
frequency characteristic of the optimal crossover frequency area among the frequency
characteristic similarly subjected to Fourier transform, A more accurate crossover frequency is
searched from the high frequency within the above-mentioned optimum crossover frequency
range, a point of 3 dB attenuation is determined, and the frequency is made the final crossover
frequency.
[0015]
FIG. 1 is a block diagram showing the main part of a playback apparatus according to an
embodiment of the present invention.
In the figure, A is an amplifying device, B is a speaker system, 1 is a signal processing unit, 1a is a
memory, 2 is an output unit, 3 is a measurement signal generating unit, 4 is a sound pickup
signal input unit, 5 is an operation unit, 6 Is a control unit, 7a to 7f are speakers, and 8 is a
microphone. In this embodiment, a multi-channel reproduction system is configured by the
amplification device A and the speaker system B, and external devices such as a DVD
reproduction device, a television, a CD reproduction device, and a VTR, which are not shown, are
connectable. .
[0016]
The signal processing unit 1 performs signal processing such as decoding processing and delay
processing on an audio signal input from an external device or the like. The decoding process is,
for example, a process of converting an audio signal of two channels into an audio signal of six
channels, a process of decompressing a compressed audio signal, and the like. The delay
processing is processing for adding a delay time to each of the decoded audio signals of the
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respective channels in order to give the viewer a sense of presence etc. when the audio signals
are output from the plurality of speakers.
[0017]
An environment in which the signal processing unit 1 listens to an audio signal in a space
surrounded by a plurality of speakers 7 a to 7 f based on a sound collection signal input from the
sound collection signal input unit 4 under the control of the control unit 6 described later.
Automatically set the values of various parameters of acoustic characteristics such as frequency
characteristics, delay time, sound pressure level, crossover frequency etc. of the output signal of
each channel so that the optimum acoustic effect can be obtained at the listener's listening point
in Do. The operation of performing this setting is called automatic sound field correction
processing.
[0018]
The output unit 2 includes a digital-to-analog conversion unit (not shown) and an amplification
unit (not shown), and converts digital audio signals of a plurality of channels input from the
signal processing unit 1 into analog signals after digital-to-analog conversion. The audio signal is
amplified, and a front left channel speaker (hereinafter referred to as FL speaker ) connected
to the output unit 2. 7a, front light channel speaker (hereinafter referred to as "FR speaker". 7b,
center channel speaker (hereinafter referred to as "C speaker". 7c, surround left channel speaker
(hereinafter referred to as "SL speaker". 7d, surround light channel speaker (hereinafter referred
to as "SR speaker". 7e, subwoofer channel speaker (hereinafter referred to as "SW speaker". ) 7f
output respectively.
[0019]
The output unit 2 generates a measurement signal generating unit under the control of the
control unit 6 at the time of automatic sound field correction processing or when the listener
changes the frequency characteristic using the operation unit 5 after the automatic sound field
correction processing. The measurement signals generated by 3 are output to the speakers 7a to
7f of the respective channels.
[0020]
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The measurement signal generation unit 3 generates measurement signals for measuring the
values of various parameters when performing automatic setting of the values of various
parameters of acoustic characteristics in the automatic sound field correction processing under
the control of the control unit 6 Do.
In this embodiment, a signal obtained by filtering a TSP (Time Stretched Pulse) signal is used as a
measurement signal.
[0021]
The sound collection signal input unit 4 receives the sound collection signal input from the
microphone 8 connected to the sound collection signal input unit 4 at the time of automatic
sound field correction processing under the control of the control unit 6. Output to
[0022]
The operation unit 5 is an operation button for switching on / off of automatic sound field
measurement processing, an operation button for outputting a measurement signal, an operation
button for inputting or changing settings of various parameters of acoustic characteristics, etc.
Equipped with
When the listener presses the operation button, the operation unit 5 outputs an instruction signal
corresponding to the pressed operation button to the control unit 6.
[0023]
The control unit 6 comprehensively controls the entire amplification device. The control unit 6
causes the signal processing unit 1 to perform decoding processing, delay processing, and the
like on an audio signal input from an external device, and performs control to output an audio
signal from the output unit 2.
[0024]
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When an instruction signal to turn on automatic sound field correction processing is input from
the operation unit 5, the control unit 6 causes the measurement signal generation unit 3 to
generate a measurement signal, and the measurement signal is transmitted via the signal
processing unit 1. Thus, control is performed to cause the output unit 2 to output each speaker.
Then, the control unit 6 causes the signal processing unit 1 to calculate values of various
parameters of acoustic characteristics based on the collected sound signal input from the
collected sound signal input unit 4, and stores the calculated values in the memory 1a. Control is
performed to set the value as the value of various parameters of the acoustic characteristic.
[0025]
The automatic sound field correction process will be described. As shown in FIG. 1, in a multichannel reproduction system such as a home theater system provided with six speakers 7a to 7f,
acoustic characteristics of a room, frequency characteristics of speakers used, phase
characteristics of each channel, and audio signals from speakers Speaker 7a of each channel at
the listening point depending on the transfer characteristics of the transmission path until the
listener listens to the audio signal, the type and number of speakers used, the installation
position of the speakers, the arrangement of the listening point and each speaker, etc. Because
the phase, roll-off frequency, sound pressure level, distance, and frequency characteristics of the
audio signal arriving from ˜ 7f change, even if the listener listens to the audio signal of the same
sound source, the difference in the above characteristics etc. Sounds different depending on the
timbre, sound field, realism, etc.
[0026]
The audio signal reproduced by the external device and input to the amplification device is
premised on that the environment for listening to the audio signal is an appropriate environment.
For example, the sound pressure level of the output signal of each channel is equal, the distance
between the speakers 7a to 7f, the frequency characteristic of each speaker are appropriate, and
the user listens in an environment where speakers of appropriate types are used as a speaker
configuration. In order to reproduce the audio signal output from the amplification device with
an optimal sound, it is necessary to maintain an appropriate environment.
[0027]
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In order to realize such an environment, the conventional amplification device is equipped with a
sound field correction function, and before the listener listens to the audio signal, the speaker
configuration, the speaker size, the distance between the speakers, the output signal of each
channel Values such as the sound pressure level and the frequency characteristics of the output
signal of each channel are input using the operation unit, and sound field correction is performed
so as to be an appropriate environment. However, the work of these sound field corrections is
not easy to set an appropriate value unless it is a skilled person, and it has been a laborious
operation requiring a lot of time and effort. Therefore, there are conventional amplification
devices provided with an automatic sound field correction function for automatically performing
the above-mentioned sound field correction, and the amplification device of this embodiment also
has an automatic sound field correction function. With this function, sound field correction is a
simple operation for the listener, and the values of various parameters can be set with high
accuracy.
[0028]
The automatic sound field correction processing in the amplification device A of this embodiment
will be specifically described. The automatic sound field correction process includes a
measurement process and a correction process, and the measurement process is performed first.
The measurement process is a process of measuring the current acoustic characteristic at the
listening point. For the processing, the microphones 8 for measurement are arranged at the
listening points with respect to the speakers 7a to 7f arranged as shown in FIG.
[0029]
In the measurement process, based on the control of the control unit 6, the measurement signal
generation unit 3 generates a TSP signal as an original signal of the measurement signal, and a
signal processing unit (for example, a digital signal processor (Digital Signal Processor) : DSP) 1
output. The TSP signal is a signal expanded on the time axis by passing the impulse signal
through a virtual filter having delay times different depending on frequency, and is a signal
having a flat signal level from low frequency to high frequency. The signal processing unit 1
performs filter processing (first filter processing) on the TSP signal generated by the
measurement signal generation unit 3 using a filter coefficient in which the signal level decreases
from low frequency to high frequency. By this filtering process, a signal whose signal level
gradually decreases from low frequency to high frequency is created, and the signal is used as a
measurement signal. This is to prevent the speaker (tweeter) from being destroyed when a signal
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having a high signal level is output at a high frequency. The measurement signal is stored in the
memory 1a. The signal processing unit 1 reads the measurement signal from the memory 1a
based on the control of the control unit 6, is digital-analog converted by the output unit 2, and
outputs the signals in order from the speakers 7a to 7f. The measurement signals output from
the speakers 7 a to 7 f are collected by the microphone 8.
[0030]
The collected signal is input to the collected signal input unit 4. The sound collection signal input
unit 4 includes an analog-digital converter that converts an analog signal into a digital signal,
converts the sound collection signal into a digital signal, and outputs the converted digital sound
collection signal to the signal processing unit 1. The signal processing unit 1 temporarily stores
the digital sound collection signal input from the sound collection signal input unit 4 in the
memory 1 a. The signal processing unit 1 uses a filter coefficient (a filter coefficient that is the
inverse of the filter coefficient of the signal for measurement) in which the signal level decreases
from high frequency to low frequency with respect to the digital sound pickup signal stored in
the memory 1a. Filter processing (second filter processing) is performed. By the second filtering
process, the first filtering process applied to the measurement signal is offset, and the
measurement result of the speaker that outputs the measuring signal is obtained. An impulse
response can be obtained by performing a convolution process on the measurement result using
data obtained by inverting the TSP signal on the time axis. The transfer characteristics until the
signals output from the amplification device are output from the speakers 7a to 7f and the
signals are collected by the microphone 8 correspond to the impulse response obtained here. The
second filtering process is performed, and the signal (measurement result signal) as a result of
the convolution process is stored in the memory 1a. Then, the signal processing unit 1 performs
processing for obtaining the frequency characteristics and the like of each speaker by calculation
based on the measurement result signal stored in the memory 1a.
[0031]
Based on this impulse response, the sound pressure level or peak level of the output signal of
each channel is observed, or FFT (Fast Fourier Transform) analysis is performed, and a speaker
configuration (presence or absence of a speaker, speaker size), and between speakers The phase
relationship, the distance between the speakers, the distance of the speakers to the listening
point, the sound pressure level and the frequency characteristic of the output signal from each
speaker, and the crossover frequency can be obtained.
[0032]
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Then, the process shifts to correction processing.
The correction process sets and corrects various parameters based on the measurement result
obtained by the measurement process.
[0033]
The signal processing unit 1 sets the number of speakers based on the presence or absence of
speakers, corrects the delay time between channels based on the distance between the speakers
or each speaker with respect to the listening point, and Correction of difference in sound
pressure level of output signal among channels based on sound pressure level of output signal,
correction of frequency characteristic of output signal of each channel based on frequency
characteristic of output signal of each speaker, output signal of each speaker The coefficient for
setting the crossover frequency of the output signal of each channel based on the frequency
characteristic of is determined by calculation.
[0034]
The coefficients obtained by calculation are stored in the memory 1a, and when outputting an
audio signal input from an external device, correction of frequency characteristics and phase
correction for the audio signal using the coefficients stored in the memory 1a Signal processing
such as correction of sound pressure level, addition of delay time, and setting of crossover
frequency is performed and output.
[0035]
Here, the determination of the crossover frequency will be described with reference to FIGS.
FIG. 2 is a flowchart.
The signal processing unit 1 performs FFT (Fast Fourier Transform) analysis (step 2) based on
the measured impulse response (step 1). When the signal processing unit 1 performs FFT
analysis on a signal corresponding to the frequency characteristic obtained from the signal
collected from the microphone 8, a signal (dotted line) corresponding to the frequency
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characteristic as shown in FIG. 3 is obtained. This signal is a signal whose frequency is a
parameter and whose level is an output value. In this description, the frequency characteristics of
the surround speaker will be described as an example.
[0036]
Next, this signal is subjected to a smoothing width such as one octave, that is, smoothing with a
wide band smoothing width (step 3) and attenuated 3 dB below the reference level obtained by
averaging the middle and high band sound pressure levels. The high frequency is searched from
the high frequency, and the band of the fixed width of the frequency is made the optimum
crossover frequency range (step 4). In this case, when smoothing is performed on the signal
corresponding to the frequency characteristic obtained by the measurement with a smoothing
width of one octave, a signal (solid line corresponding to the frequency characteristic with
smoothed level irregularities as shown in FIG. 3) ) Is obtained. As shown in FIG. 8, the level of the
reference frequency band (1 kHz to 2 kHz in this embodiment) of the signal corresponding to the
frequency characteristic is used as the reference level, and the frequency characteristic
attenuated by 3 dB or more from the reference level is detected. Do. As shown in FIG. 4, when
there are a plurality of frequency points attenuated by 3 dB or more from the reference level, low
frequency points are selected.
[0037]
Thereafter, the signal (dotted line) corresponding to the frequency characteristic as shown in FIG.
3 obtained by performing the FFT analysis is subjected to smoothing with a narrow band
smoothing width such as a 1/6 octave smoothing width (step 5). In this case, in the frequency of
the selected point, a range of one octave centered on the frequency is defined as an optimal
crossover frequency range, and for the optimal crossover frequency range, for example, Performs
narrowband smoothing such as a 6 octave smoothing width.
[0038]
The frequency which is 3 dB attenuated from the reference level which averaged the sound
pressure level of middle and high bands again is searched from the high frequency in the optimal
crossover frequency zone determined in step 4 this time, and the frequency is finalized Let it be
the crossover frequency (step 6). As a result of smoothing in 1/6 octave, a frequency
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characteristic (solid line) as shown in FIG. 5 is obtained. In the frequency characteristic, a
frequency attenuated by 3 dB from the reference level is detected. As shown in FIG. 6, even if
there are a plurality of frequencies attenuated by 3 dB from the reference level, those
frequencies that do not fall within the optimal crossover frequency area (eg, points indicated as
false recognition in the figure) are erroneously recognized It will not be done. In addition,
when there are a plurality of frequencies attenuated by 3 dB in the optimal crossover frequency
range, the lowest frequency is set as the crossover frequency.
[0039]
The smoothing will be described more specifically. Smoothing is to smooth the level of the
frequency characteristic for each specific frequency band. The measurement result signal (signal
stored in the memory 1a) obtained by the measurement is level data using a frequency as a
parameter, and can be said to be a signal corresponding to the frequency characteristic. For this
signal, FFT processing (frequency analysis) is performed for N points (points), and levels are
obtained for each fs / N [Hz]. The N point is the number of center frequencies of the frequency
band to be averaged. In this embodiment, the smoothing width of one octave is used as the wideband smoothing width, and the level of the frequency band is averaged (smoothing). Specifically,
from low frequency to high frequency, each center frequency (for example, 63, 125, 250, 500,
1k, 2k, 4k, 8k, 16k, ... [Hz]), each The levels in a specific frequency band centered on the center
frequency are averaged to calculate the level in the frequency band (smoothing). For example, in
the case of a center frequency of 1 kHz, the levels in the band of 707 Hz to 1.41 kHz centered on
1 kHz are averaged, and the average value is taken as the level of the center frequency of 1 kHz.
Similarly, the level is determined at other center frequencies. Then, a characteristic obtained by
connecting those levels by a line is taken as a frequency characteristic smoothed by one octave.
[0040]
Thus, after the smoothing in one octave is completed, the optimal crossover frequency range is
detected, and level averaging (smoothing) of the level of the frequency range in the narrow band
within the optimal crossover frequency range is performed. Specifically, when a frequency band
of 50 Hz to 100 Hz (center frequency 75 Hz) is detected as the optimum crossover frequency
area, the smoothing width of the narrow band (1/6 octave) in the frequency band of the optimum
crossover frequency area is detected. Perform smoothing. That is, for each center frequency (53,
59, 67, 75, 84, 94 [Hz]), the levels in a specific frequency band centered on each center
frequency are averaged, and the level in that frequency band is calculate. For example, in the
case of 53 Hz of center frequency, the level of a 50 to 56 Hz band centering on 53 Hz is
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averaged, and let the average value be a level of 53 Hz of center frequencies. Similarly, find the
level at other center frequencies. Then, a characteristic in which those levels are connected by a
line is taken as a frequency characteristic smoothed with a smoothing width of 1/6 octave. For
this frequency characteristic, the crossover frequency is determined based on a level 3 dB
attenuated from the reference level.
[0041]
If no 3 dB attenuation frequency is found in the optimal crossover frequency range, the
frequency closest to 3 dB attenuation is taken as the crossover frequency. The control unit 6 sets
the coefficient of the crossover frequency in the signal processing unit 1 based on the crossover
frequency obtained as a result of the correction processing. The signal processing unit 1 subjects
the audio signal of the channel (surround channel audio signal) output from the surround
speaker to filtering processing that passes the audio signal of a frequency higher than the
crossover frequency. As a result, among the audio signals of the surround channel, the audio
signal of the frequency band higher than the crossover frequency is output from the surround
speaker.
[0042]
In addition, the signal processing unit 1 performs addition processing so as to add an audio
signal of a frequency band lower than the crossover frequency of the surround channel audio
signal to another speaker (for example, a front speaker or a subwoofer speaker). As a result,
among the audio signals of the surround channel, an audio signal of a frequency band lower than
the crossover frequency is output from a speaker other than the surround speaker (for example,
a front speaker or a subwoofer speaker).
[0043]
In this embodiment, the crossover frequency is determined based on a level attenuated by 3 dB
from the reference level, but other values may be selected. Further, in the broad band (one
octave) smoothing and the narrow band (1/6 octave) smoothing, levels attenuated from the
reference level may be set to different levels in each case. Also, although one octave has been
exemplified as the wide band smoothing width, it may be a smoothing width of about 1/3 to 1.5
octaves, and 1/2 to 1/10 of the wide band smoothing width as the narrow band smoothing
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width. The smoothing width may be about an octave.
[0044]
In the present invention, the crossover frequency of the speaker system can be measured, and
the auditory characteristics in the multi-channel reproduction system can be set well.
[0045]
It is a block diagram which shows the principal part of the reproducing ¦ regenerating apparatus
of one Example of this invention.
It is a flowchart which shows operation ¦ movement of the crossover frequency calculation
process of this invention. It is explanatory drawing about the smoothing of crossover frequency
calculation. It is explanatory drawing about the smoothing of crossover frequency calculation. It
is explanatory drawing about the smoothing of crossover frequency calculation. It is explanatory
drawing about the smoothing of crossover frequency calculation. It is explanatory drawing of an
example of arrangement ¦ positioning of the speaker of a multi-channel system. It is explanatory
drawing which shows an example of the calculated frequency characteristic of each speaker. It is
explanatory drawing about a crossover frequency.
Explanation of sign
[0046]
A: amplification device, B: speaker system, 1: signal processing unit, 1a: memory, 2: output unit,
3: measurement signal generation unit, 4: sound pickup signal input unit, 5: operation unit, 6:
control unit 7a: FL speaker 7b: FR speaker 7c: C speaker 7d: SL speaker 7e: SR speaker 7f: SW
speaker 8: microphone
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