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JP2011130212

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DESCRIPTION JP2011130212
PROBLEM TO BE SOLVED: To reduce the influence of reverberation after falling of a signal of a
frequency component generated by a standing wave, which causes a problem in hearing.
SOLUTION: A test signal for measuring a standing wave state emitted in a listening room is
picked up, and a peak position or dip position by the standing wave is specified based on its
frequency characteristic. Next, the burst signal corresponding to the frequency of the peak
position or dip position is emitted and collected. An acoustic signal that calculates an increase
ΔP of the peak that is increased at the falling portion corresponding to the end position of the
burst signal to the peak of the portion of the collected signal corresponding to the steady portion
of the burst signal The frequency of the above peak position or dip position is attenuated with an
amount of attenuation dependent on .DELTA.P. [Selected figure] Figure 5
Sound processing apparatus and method
[0001]
The present invention relates to a sound field correction technique for correcting the influence of
frequency characteristics due to standing waves in a room.
[0002]
When sound is emitted from a sound source such as a speaker in a room such as a home, in
addition to direct sound reaching each place in the room at the shortest distance, reflected sound
from each surface such as a wall, ceiling or floor of the room exists. , These sound waves overlap
each other.
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At this time, for example, in the case of a frequency in which the inter-plane distance is an
integral multiple of a half wavelength of the sound wave between the faces facing in parallel, a
standing wave is generated and a low-pass resonance called booming occurs.
[0003]
In such a case, booming is suppressed by a parametric equalizer, or acoustic characteristics are
measured by a microphone at a listening position in advance, and correction is performed using
the inverse characteristics. In addition to this technology, a technology is also disclosed that
utilizes the directional information of the reflected sound (for example, see Patent Document 1).
[0004]
Unexamined-Japanese-Patent No. 5-83786
[0005]
If frequency characteristics such as a listening room are measured, characteristics as shown in
FIG. 2 are obtained, for example.
A standing wave occurs at a peak where the sound pressure level is increasing and at a dip where
the sound pressure level is decreasing. The standing wave portion is a frequency at which the
sound output from the speaker or the like causes resonance with respect to the size of the room,
and the level fluctuation is not only large with respect to other frequency portions, but also
changes in the time direction Too big.
[0006]
The influence of the standing wave will be described with reference to FIG. In FIG. 3, the signal
33 is a signal of the frequency of the dip portion. The signal 32 is a signal of the frequency of a
flat portion in terms of frequency characteristics, and the signal 31 is a signal when the signal 32
is produced in a burst form. The signal 32 in the flat portion has its sound pressure level sharply
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2
lowered with the fall of the burst signal 31.
[0007]
The signal 33 in the dip portion normally starts to rise at the rising portion in the absence of the
reflected wave. However, since the signal 33 at the dip portion is a frequency at which a standing
wave is generated, when interference with the reflected wave starts, the level is low during burst
signal generation due to the interference between the direct wave and the reflected wave.
Furthermore, since the resonance state is achieved at the standing wave frequency, although the
original burst signal is falling, a signal of a level higher than that during sound generation is
observed. This is because the component of the direct wave is lost at the end of the burst signal,
so only the component of the reflected wave that has been increased by resonance remains, and
the signal at a higher level than the sound generation period is long even though the sound
output is ended It is because time remains. For this reason, the signal component of the dip
portion is at a low level at the time of normal sound generation, and the sound becomes loud at
timing when it should not be sounded, which causes a hearing problem.
[0008]
In addition, there is also a problem that, for the frequency of the standing wave peak portion, a
large reverberation is left forever. In the case of general booming correction, a method is
employed in which a frequency component corresponding to the peak portion of the standing
wave is always attenuated by a fixed amount using a parametric equalizer or the like. However,
when this method is applied to the dip portion, the portion where the original sound which has
already been lowered by the interference is emitted is further attenuated, and there is an adverse
effect such that the sound of that portion can hardly be heard.
[0009]
Therefore, an object of the present invention is to reduce the influence of reverberation after the
fall of the signal of the frequency component generated by the standing wave, which is an
audible problem.
[0010]
According to one aspect of the present invention, there is provided an acoustic processing device
for adjusting an acoustic signal to be output based on acoustic characteristics of a listening room,
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the test signal for measuring a standing wave state in the listening room from a speaker A first
sound collecting means for emitting a sound and collecting the test signal emitted by the
microphone, and a standing wave based on a frequency characteristic of a signal collected by the
first sound collecting means Specification means for specifying a peak position or dip position,
and a burst signal corresponding to the frequency of the peak position or dip position is emitted
from the speaker in the listening room, and the emitted burst signal is collected by the
microphone A second sound collecting means for sounding and a peak of a portion of the signal
collected by the second sound collecting means corresponding to a steady portion of the burst
signal Calculating means for calculating an increase .DELTA.P of the peak increase at the falling
portion corresponding to the end position of the burst signal, and the frequency of the peak
position or dip position of the acoustic signal to be output depending on the .DELTA.P. A sound
processing apparatus is provided, characterized in that it comprises: filter means for attenuating
the amount of attenuation.
[0011]
According to the present invention, it is possible to reduce the influence of the reverberation
after the fall of the signal of the frequency component in which the standing wave is generated,
which causes a problem in hearing.
[0012]
The figure which shows the structure of the sound system in embodiment.
The figure which shows the example of the frequency characteristic in a listening room.
The figure explaining the influence of a standing wave.
BRIEF DESCRIPTION OF THE DRAWINGS The block diagram which shows the structural example
of the sound processing apparatus in embodiment. The timing chart which concerns on
application of an attenuation control signal. The figure explaining generation of an attenuation
control signal. The block diagram showing the example of composition of the filter in an
embodiment. The figure explaining a burst detection waveform. 6 is a flowchart showing
correction coefficient determination processing in the embodiment. The figure which shows
another example of attenuation amount control signal.
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[0013]
Hereinafter, preferred embodiments of the present invention will be described in detail with
reference to the drawings.
[0014]
First Embodiment FIG. 1 is a diagram showing a configuration of an acoustic system in the
present embodiment.
This acoustic system can adjust the outputted acoustic signal based on the acoustic characteristic
of the listening room which is the reproduction sound field space by the configuration and
processing described below. The sound processing device 11 includes a display unit 14, a volume
control 18, a remote control light receiving unit 16, and the like. Audio signals are transmitted
from the sound processing device 11 to the speakers 12L and 12R. The speakers 12L and 12R
are respectively active speakers, and have power amplifiers 17L and 17R, respectively. This
configuration is an example, and a configuration may be employed in which a power amplifier is
provided in the middle instead of the active speaker.
[0015]
A microphone 13 is used to pick up a test signal and the like sent from the sound processing
device 11 to the speakers 12L and 12R. Reference numeral 15 is a remote control device for
controlling the sound processing device 11 and is generally for selecting an audio device (CD,
DVD, etc., not shown) connected to the sound processing device 11 and performing volume
control. is there.
[0016]
FIG. 4 is a diagram showing the configuration of the sound processing apparatus 11. As shown in
FIG. During normal operation, music information from an external audio device connected to the
input switching unit 41 is sent to the output unit 43 via the filter 42. The output unit 43 outputs
music information in an analog manner by a D / A converter (not shown) if it is an apparatus
having LINEOUT. On the other hand, in the case of digital output, the output signal is converted
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into a signal of digital IF such as SPDIF, and music information is output to the speakers 12L and
12R.
[0017]
At the time of operation for determination of the correction coefficient, the input switching unit
41 is connected to the test signal generating unit 44 according to a command from the
calculation control unit 46. The test signal generation unit 44 can output a sweep signal whose
frequency is continuously changed from low frequency to high frequency, white noise, and the
like. Alternatively, a signal using an MLS (maximum length sequence) signal using an M-sequence
signal, which is a type of pseudo random signal, can be output. This signal has a simple
generation method, and at the same time, it is possible to obtain an impulse response at high
speed by using a method such as Hadamard transformation. In measuring characteristics in the
user's listening room etc., it is a short time. It has merits such as being able to calculate with.
[0018]
The microphone 13 can pick up the test signal generated from the speaker 12. The electric signal
output from the microphone 13 is converted into digital data by the A / D converter 48, sent to
the arithmetic control unit 46, and recorded, for example, in the storage unit 50, and analyzed
according to the program by the arithmetic control unit 46 It can be done.
[0019]
In the filter 42, processing as shown in FIG. 5 is performed. Now, for the sake of explanation, it is
assumed that only the signal 51 of the frequency at which a dip is generated due to the standing
wave is inputted. For the input signal 51, the signal observed as a sound wave has a waveform in
which the sound pressure rises after stopping the signal output, as in the signal 54, due to the
resonance characteristic of the room. In order to prevent the sound pressure rise after the signal
output is stopped, the filter 42 is configured to reduce the gain of the falling portion of the input
signal 51 and output the signal.
[0020]
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Specifically, the attenuation amount control signal 53 is generated from the input signal 51 by a
differential operation unit described later. Since the attenuation control signal 53 is synchronized
with the falling characteristics of the input signal, the output signal is delayed by a fixed time ΔT
in order to attenuate only the falling portion of the signal, and the attenuation control signal
Control the amount of attenuation of the frequency notch filter. Thereby, the dashed line portion
of the signal 52 can be attenuated as a solid line portion.
[0021]
By reducing the gain of the falling portion of the output signal in this manner, the conventional
output signal 54 can be made into the signal 55 in which the sound pressure increase after the
output is stopped is suppressed. As described above, by attenuating only the falling portion of
the output signal, a reverberation having a hearing problem can be obtained without lowering
the sound pressure of the rising portion of the signal or the continuous sound portion where the
sound pressure is reduced due to interference. It can act on parts only.
[0022]
FIG. 6 is a diagram showing an outline of generation of an attenuation control signal. Differential
processing is performed on the input signal 61 to extract its fall timing. For that purpose, first, an
envelope signal 62 of the input signal 61 is generated. The differential processing is performed
on the generated envelope signal to obtain a differential signal 63. Of the differential signals, for
example, a pulse signal 64 having a predetermined time width T and amplitude H is generated
for a reverberation time of the listening room at a position synchronized with the negative side
signal related to the fall, and is attenuated The quantity control signal 53 is used.
[0023]
These processes can be realized, for example, by the block configuration of the filter 42 shown in
FIG. The acoustic signal (input signal) to be output is input to a delay circuit 71 to be an output
signal and a band pass filter 73 that discriminates the frequency of the peak position or the dip
position. The signal divided into the specific frequency by the band pass filter 73 passes through
the envelope generation circuit 74 and is input to the differentiation circuit 75. A signal
synchronized with the falling timing of the signal is output from the differentiating circuit 75,
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and an attenuation amount control signal having a pulse width and a gain set by the control
amount setting unit 77 is output to the control signal generating circuit. It is generated by 76.
[0024]
The generated attenuation amount control signal controls the gain of notch filter 72, and the
delay circuit 71 previously set by control amount setting unit 77 controls the gain of the falling
portion of the input signal delayed for a predetermined time. Ru. The delay time needs to be
longer than the pulse width set by the control amount setting unit 77.
[0025]
The pulse width and amplitude of the attenuation control signal may be determined from the
reverberation characteristics of the listening room. For example, consider a case where a signal
as shown in FIG. 8 is obtained as a signal of a frequency corresponding to a dip position for a
burst signal. In this case, following the steady portion where the level is lowered due to
resonance, the signal peak once increases by ΔP and then falls at the falling portion. Therefore,
measure the time until the signal peak increased by ΔP at the falling part falls to a value
equivalent to the peak in the steady part, prepare a table for that time in advance, and determine
the pulse width and height. Good. Alternatively, ΔP may be measured, and the pulse width or
pulse height may be determined such that the value is equal to or less than a preset value, for
example, the same as the portion where the level is low. Thus, the pulse width and amplitude of
the attenuation control signal can be set depending on ΔP.
[0026]
FIG. 9 is a flowchart showing the correction coefficient determination process in the embodiment.
This process is started by instructing the operation mode as the correction coefficient
determination mode via the remote control or the like (S100). Before starting the operation, the
user may place the microphone 13 at a listening point where music is usually viewed, and display
a message prompting the user to connect to the A / D converter 48 on the display unit 14. good.
When the microphone 13 is connected, the input switching unit 41 is instructed to input the
signal from the test signal generating unit 44 (S101).
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[0027]
Next, the correction coefficient is set to an initial value, for example, the pulse width T = 0 and
the height H = 0 (S102). By thus performing the initial setting, the filter 42 does not function at
all, that is, so-called through setting. In such a state, the test signal generation unit 44 generates
a test signal and emits sound from the speaker 12 (S103). The test signal at this time is for
measuring the standing wave state of the listening room, and the test signal at the listening point
is picked up by the microphone 13 using the above-mentioned MLS signal and sweep signal (S
104) (S 104) First sound collection). The recorded data is converted into frequency domain data
using FFT or Hadamard transformation (S105).
[0028]
From the frequency characteristics of the obtained frequency domain data, the peak position and
the dip position by the standing wave are specified (S106). When a dip or the like exceeding the
predetermined level is detected among the identified peak position and dip position, that point is
stored as a correction candidate. In S107, the presence or absence of a correction candidate is
determined from this result. If no correction candidate is found, there is no need to particularly
perform correction or the like, and the process may be ended as it is (S115). If a correction
candidate is found, the burst signal of the frequency to be corrected is output from the test signal
generator 44 in S108.
[0029]
The output burst signal is emitted into the listening room by the output unit 43 and the speaker
12, and is collected by the microphone 13 with the characteristics of the listening room (second
sound collection). The picked up signal is A / D converted by the A / D converter 48, and then
stored in the storage unit 50 via the operation control unit 46 (S109).
[0030]
Next, the reverberation characteristics of the room are analyzed based on the recorded data
(S110). Here, in particular, with respect to the peak of the portion of the signal collected in S109
corresponding to the steady portion of the burst signal, an increase ΔP of the peak increased in
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the falling portion corresponding to the end position of the burst signal is calculated. . In the first
loop, since neither correction coefficient T nor H is set, the characteristic as it is is measured, and
in most cases, the threshold value of ΔP in S111 exceeds the predetermined value. Will be
measured. The threshold value at this time may be set to a value equal to or lower than that of
the portion where the level is reduced due to interference as described above, and the system
may appropriately set the threshold value accordingly. You should decide.
[0031]
If ΔP is not equal to or less than the predetermined value in S111, correction coefficients T and
H are set in S112. Since the values are set to T and H by this, the filter 42 substantially works as
a filter. At this time, the delay time ΔT is also set to the delay circuit 71 in accordance with the
value of T.
[0032]
Next, the process returns to step S108, and while the correction coefficient is set, the burst signal
is generated again. This is recorded (S109), and the reverberation characteristic is analyzed
(S110). Since the data recorded this time includes the effect of the filter 42, the one in which the
reverberation characteristic portion is attenuated is recorded. At this time, if ΔP of the
reverberation characteristic portion is lower than a predetermined value, the correction
coefficient at this time is adopted.
[0033]
If ΔP is a value larger than a predetermined value, the value of the correction coefficient is
increased, the same loop is repeated, and the correction coefficient to be equal to or less than the
predetermined value is determined. If .DELTA.P is determined to be equal to or less than a
predetermined value, values of T, H and .DELTA.T as correction coefficients are stored in S113.
When there are a plurality of frequencies to be corrected, the processing of S108 is repeated to
determine the same correction coefficient, and when the correction coefficients are determined
for all peaks or dips, the processing is ended (S115). .
[0034]
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When the correction coefficient is determined, the input switching unit 41 is changed so that the
normal input passes, and the normal operation is performed, so that the content subjected to the
correction by the correction coefficient determined by the filter 42 It will be possible to
appreciate the At this time, an instruction such as microphone removal may be given to the user
from the display unit 14.
[0035]
Depending on the system, H may be constant and controlled with only the pulse width T. When
the pulse width T is not determined by measurement but is determined by a table or the like, the
attenuation amount to the assumed reverberation is previously defined and stored in the table,
and the value is determined from the reverberation characteristic of the test signal, etc. . In this
case, it is possible to configure a system in which the processing time can be shortened by
determining the coefficients without performing the repetitive loop from S111.
[0036]
Second Embodiment In the above-described first embodiment, the correction is performed in all
cases. However, at the rising of the dip portion, as shown in FIG. 3, the original signal rising
characteristics are obtained before the resonance occurs. If this signal disappears due to the
correction, this frequency signal may not be heard, which may lead to deterioration of the
characteristics.
[0037]
Therefore, the frequency to be corrected may be corrected only after the predetermined time has
elapsed. That is, it is configured to start the operation of the filter after a predetermined time has
elapsed since the acoustic signal to be output is input. The predetermined time for determining
the presence or absence of the correction may be determined from the rising time Tr shown in
FIG. Since Tr is the time until the interference starts, correction is permitted if the signal of the
same frequency continues for a longer time.
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[0038]
Here, the control signal from the differentiating circuit 75 of FIG. 4 is controlled so that the
correction is performed only when the time Td between the positive side portion and the
negative side portion of the differential signal 63 shown in FIG. It may be configured to be sent to
the generation circuit 76.
[0039]
The signal for correction is not limited to the pulse signal as shown in the attenuation control
signal 53 of FIG.
For example, as shown in FIG. 10, it is also possible to adopt a method such as blunting the
falling or rising characteristics of the pulse. As described above, by changing the attenuation by
the filter smoothly, it is possible to end the interference state gently and to reduce the hearing
problems due to the rapid change.
[0040]
In the above, the description has mainly been made regarding the standing wave dip frequency,
but of course the tailing due to resonance occurs in the peak part as well, so the same processing
can be applied. Further, although the description in the drawings relates to one frequency, it is of
course possible to correct a plurality of dips and peaks with the same configuration.
[0041]
Further, although the respective blocks are described as circuits in the description of the
configuration, software processing can also be performed using an LSI for sound processing such
as a digital signal processor (DSP). .
[0042]
Other Embodiments The present invention is also realized by executing the following processing.
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That is, software (program) for realizing the functions of the above-described embodiments is
supplied to a system or apparatus via a network or various storage media, and a computer (or
CPU, MPU or the like) of the system or apparatus reads the program. It is a process to execute.
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