JP2006313953

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DESCRIPTION JP2006313953
PROBLEM TO BE SOLVED: To provide an automatic sound field correction system capable of
accurately measuring and correcting delay without requiring complicated processing for all
speakers having different reproduction bands. SOLUTION: An automatic sound field correction
system according to the present invention comprises a sound field measurement device 1, an
amplifier 2, a speaker 3, a microphone 4 and a microphone amplifier 5. The sound field
measurement apparatus 1 generates different pulse signals according to the band of the speaker
to be measured, and after amplification by the amplifier 2, the speaker 3 reproduces. On the
other hand, when the microphone 4 installed at the listening position records a pulse signal, it is
amplified by the microphone amplifier 5 and then input to the sound field measurement device 1.
The sound field measurement device 1 calculates the delay value to each speaker from the input
data, and applies a correction. By using the present invention, it is possible to correct the delay
accurately without the need for complicated processing such as DFT. [Selected figure] Figure 1
Automatic sound field correction system, automatic sound field correction method and sound
field measurement device
[0001]
The present invention relates to an automatic sound field correction system, an automatic sound
field correction method, and a sound field measurement device that automatically corrects a
deviation of a sound field generated due to a difference in distance from each speaker to a
listening position in an audio system.
[0002]
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Conventionally, in an audio system, it is difficult to output the sound of all the required frequency
bands with a single speaker, so the sound to be reproduced by the band limiting filter is divided
by the frequency band, and the divided sounds are used for the bass The system uses a separate
speaker such as for the middle range and high range.
Also, in order to give a three-dimensional effect to the sound to be reproduced, speakers of two
channels on the left and right are used, or a surround speaker is provided at the back of the
viewer. Here, in order for the listener to listen to the sounds independently generated from the
respective speakers without discomfort, it is necessary that all the sounds from the respective
speakers reach the listening position at the same time.
[0003]
Patent documents 1 to 6 are known as reference documents of prior art related to the present
application. Patent No. 2725838 Patent No. 3148060 Japanese Patent Laid-Open No. 07212896 Japanese Patent Laid-Open No. 11-262081 Japanese Patent Laid-Open No. 11-258034
Japanese Patent Laid-Open No. 2001-224100
[0004]
However, since the distance from each speaker to the listening position is not necessarily
constant, it is necessary to measure the distance from each speaker to the listening position and
to delay the output sound according to the measured distance. Here, as a conventional method of
measuring the distance to the listening position, a method using TSP (Time Stretched Pulse) or
white noise is introduced, but complex processing such as DFT (Discrete Fourier Transform) is
required. Also, in the process of the process, a lot of resources such as memory were required.
[0005]
Another known method is to measure the distance to the listening position using a pulse signal
specialized for the low-range woofer, for example, in a multi-way speaker combining a low-range
woofer and a high-range tweeter. It is done. However, in the case of in-vehicle multi-way
speakers in particular, the positions of the woofer and tweeter are often separated, so in delay
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correction using a pulse signal specialized for woofer, listening from the distance from the
woofer to the listening position and tweeter It is not possible to correct the difference with the
distance to the position.
[0006]
However, if the same pulse signal is output from the woofer and the tweeter in order to
simultaneously measure the listening position of the woofer and the tweeter, the S / N ratio
(Signal to Noise Ratio) is obtained because of the difference in the reproduction band of the
speaker Is degraded. This is because, for example, a pulse signal specialized for the woofer
contains many low frequency components so that a large S / N ratio can be obtained in the
distance measurement of the woofer, but when this pulse signal is passed through a tweeter for
high frequency range, the low frequency component Is largely attenuated by the band limiting
filter. As a result, the output power of the pulse signal decreases, the S / N ratio decreases, and
not only does the distance measurement accuracy deteriorate, but also in an environment where
a noise source such as a car interior is near, the pulse signal Being buried in the noise source, it
becomes impossible to measure the distance.
[0007]
The present invention has been made in consideration of the above circumstances, and an object
thereof is to provide an automatic sound capable of accurately measuring and correcting delay
for all speakers having different reproduction bands without requiring complicated processing. It
is in providing a field correction system.
[0008]
The present invention has been made to solve the above-mentioned problems, and the invention
according to claim 1 comprises a plurality of speakers having different characteristics, and a
sound field forming a sound field at a listening position by the plurality of speakers. An
automatic sound field correction system in a forming apparatus, comprising: pulse signal
generation means for generating different pulse signals according to the characteristics of the
plurality of speakers; and one of the plurality of speakers selected from the plurality of speakers;
Speaker selection means for applying a pulse signal generated according to the characteristics of
the one speaker by the generation means to the one speaker, a microphone installed at the
listening position, an output timing of the pulse signal, and an output signal of the microphone
Delay time measuring means for measuring the sound wave delay time between the one speaker
and the listening position based on An automatic sound field correcting system characterized by
comprising a delay time adjustment means for adjusting the delay time of the signal applied to
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said first speaker in accordance with a constant result.
[0009]
The invention according to claim 2 is the invention according to claim 1, wherein the pulse signal
generation means generates pulse signals for respective bands for each of a plurality of bands.
And pulse signal selection means for selecting any one of the pulse signals for each band output
from the pulse signal generation means for each band according to the characteristics of the one
speaker.
[0010]
Also, in the invention according to claim 3, in the invention according to claim 1 or 2, the delay
time calculation means determines that the output signal of the microphone is the first threshold
from the output start time of the pulse signal. It is characterized in that a delay time is obtained
by calculating a time until it exceeds and subtracting a correction time preset for the pulse signal
from the calculated time.
[0011]
The invention according to a fourth aspect is the invention according to any one of the first to
third aspects, wherein the microphone outputs no sound during a predetermined time before the
pulse signal is output to the speaker. It is characterized by comprising: DC offset calculation
means for calculating a DC offset from the signal; and DC offset subtraction means for
subtracting the calculated DC offset from the output signal of the microphone.
[0012]
In the invention according to claim 5, in the invention according to any one of claims 1 to 4, a
peak of an output signal of the microphone after the pulse signal is applied to the speaker of the
one. Peak detection means for detecting the level; noise detection means for detecting the level of
the output signal of the microphone before the pulse signal is applied to the one speaker;
detection results of the peak detection means and the noise detection means Signal-to-noise ratio
calculation means for calculating the ratio of detection results and whether or not the speaker of
1 is connected based on whether or not the calculation result of the signal-to-noise ratio
calculation means is greater than a second threshold value And a wire connection determination
means for determining.
[0013]
Also, in the invention according to claim 6, in the invention according to claim 5, it is necessary
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to measure the sound wave delay time based on whether the calculation result of the signal to
noise ratio calculation means is larger than a third threshold. A signal-to-noise ratio
determination unit is included to determine whether the signal-to-noise ratio is maintained.
[0014]
The invention according to claim 7 comprises pulse signal generating means for generating pulse
signals having different characteristics, and pulse signal output means for selecting the
characteristics of 1 and outputting the pulse signal generated by the pulse signal generating
means; A response input means for inputting a response to the output pulse signal, a delay time
from an output of the pulse signal to an input at the response input means based on an output
timing of the pulse signal and an input signal of the response input means And a delay time
measuring means for measuring
[0015]
The invention according to claim 8 is an automatic sound field correction method in a sound field
forming apparatus including a plurality of speakers having different characteristics, and forming
a sound field at a listening position by the plurality of speakers, the plurality of speakers A first
process of selecting one of the speakers, a second process of generating a pulse signal according
to the characteristics of the selected one speaker, and adding the pulse signal to the one speaker,
and an output timing of the pulse signal A third process of measuring an acoustic wave delay
time between the one speaker and the listening position based on an output signal of the
microphone installed at the listening position, and the first process according to the
measurement result of the acoustic wave delay time An automatic sound field comprising: a
fourth process of adjusting a delay time of a signal applied to a speaker; and performing the first
to fourth processes on all the plurality of speakers A positive way.
[0016]
According to the present invention, a noise source such as a vehicle interior is located close to
each other to measure the distance to the listening position by outputting pulse signals from the
speakers having different reproduction bands in accordance with the reproduction band of the
speakers. It is possible to make a good measurement of S / N ratio also in various environments.
In addition, it can be easily realized with a small number of resources without requiring
complicated arithmetic processing such as DFT.
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[0017]
Hereinafter, embodiments of the present invention will be described with reference to the
drawings.
FIG. 1 is a block diagram showing the configuration of an automatic sound field correction
system according to an embodiment of the present invention, and FIG. 2 is a block showing a
method for outputting a pulse signal from a sound field measurement apparatus according to an
embodiment of the present invention FIG.
FIG. 3 is a layout diagram showing the layout of speakers in the room.
[0018]
In FIG. 1, a sound field measurement device 1 is a device for measuring a sound field, and is
made into one chip.
The microcomputer 11 of the sound field measurement device 1 is responsible for main control
in the sound field measurement device 1.
A DSP (Digital Signal Processing) unit 12 receives a control signal from the microcomputer 11
and performs digital signal processing.
A D / A converter (Digital-to-Analog Converter) 13 receives a digital signal from the DSP unit 12,
converts it into an analog signal, and outputs the analog signal to the amplifier 2.
An A / D converter (Analog-to-Digital Converter) 14 receives an analog signal from the
microphone amplifier 5, converts it into a digital signal, and outputs the digital signal to the DSP
unit 12.
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[0019]
The amplifier 2 amplifies and outputs an input audio signal.
The speaker 3 reproduces the audio signal input from the amplifier 2 as sound.
A microphone (microphone) 4 is attached to the listening position and collects the sound
reproduced from the speaker 3. The microphone amplifier 5 amplifies an audio signal input from
the microphone 4 and outputs the amplified signal to the A / D converter 14.
[0020]
The audio input unit 6 is for inputting an audio signal from an external audio reproduction
device, and the audio decoding unit 7 is for decoding the audio signal inputted by the audio input
unit 6. The post processor unit 8 adds various acoustic effects to the audio signal decoded by the
audio decoding unit 7. The delay unit (delay time adjustment means) 9 delays the audio signal to
be transmitted to each speaker according to the instruction from the microcomputer 11. In the
embodiment of the present invention, the one-chip sound field measurement apparatus 1, the
amplifier 2, the microphone amplifier 5, the audio input unit 6, the audio decoding unit 7, the
post processor unit 8, and the delay unit 9 It is realized by being incorporated in the device 10).
Although only one amplifier 2, one speaker 3 and one delay unit 9 are shown, in actuality there
are a plurality of channels.
[0021]
FIG. 2 is a view showing a method in which the sound field measurement apparatus 1 of FIG. 1
outputs a pulse signal for sound field measurement, and shows functions performed by the
microcomputer 11 and the DSP unit 12. In FIG. 2, the low frequency pulse signal generation unit
(pulse signal generation unit for each band) 101 of the sound field measurement device 1
generates a pulse signal for output to a speaker that handles a low frequency region such as a
woofer or a subwoofer. The pulse signal generated here is a Raised Cosine type pulse signal and
contains many low frequency components. The high-frequency pulse signal generation unit
(pulse signal generation means for each band) 102 generates a pulse signal for output to a
speaker that handles a high-frequency range such as tweeter, and the pulse signal generated
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here is Raised Cosine Is a type of pulse signal and contains many high frequency components.
[0022]
The pulse selection unit (pulse signal selection unit) 103 selects which of the low-frequency
pulse signal and the high-frequency pulse signal to be output, and the switching signal
generation unit 104 selects the type of pulse signal to be output. Is generated to designate the
pulse select section. The level adjustment unit 105 adjusts the power of the pulse signal to be
output.
[0023]
The speaker selection unit (speaker selection unit) 106 selects which speaker the pulse signal is
to be output to, and the output speaker selection signal generation unit 107 uses the speaker
selection unit 106 to select the type of speaker outputting the pulse signal. It generates an
output speaker selection signal for designating. The CH level adjustment unit 108 adjusts the
power of the pulse signal to be output for each speaker. The amplifier 2 amplifies the power of
the pulse signal output from the CH level adjustment unit 108, and the speaker unit 3 is a
plurality of speakers constituting an audio system.
[0024]
FIG. 3 is a diagram showing an example of arranging eight types of speakers shown in the
speaker unit 3 in the listening room 100. As shown in FIG. In FIG. 3, the installation place of the
microphone 4 is a position where the listener listens to the audio. The R tweeter 301 outputs the
high range in the speaker disposed on the front right side of the listener, and the R woofer 302
outputs the low range of the same speaker. The L tweeter 303 outputs a high range in a speaker
disposed on the front left side of the listener, and the L woofer 304 outputs a low range of the
same speaker. The center 305 is a speaker disposed in the front center of the listener. The R
surround 306 is a speaker disposed on the rear right side of the listener, and the L surround 307
is a speaker disposed on the rear left side of the listener. The subwoofer 308 is a speaker that is
disposed diagonally forward to the right of the listener and outputs a bass range further than the
woofer.
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[0025]
Next, the operation of the above-described embodiment will be described with reference to FIGS.
4 to 8. FIG. 4 is a flow chart showing the procedure performed by the microcomputer 11 and the
DSP unit 12 when performing sound field measurement, and FIG. 5 is a flow chart showing the
procedure in the subroutine shown in step S409 of FIG. Is a flowchart showing the procedure in
the subroutine shown in step S410 of FIG. In FIG. 4, first, the microcomputer 11 selects a speaker
to be measured, and the output speaker selection signal generation unit 107 outputs an output
speaker selection signal to the speaker selection unit 106 (step S401). The speaker selection unit
106 sets a speaker to be output based on the input output speaker selection signal.
[0026]
When the output speaker is determined, the microcomputer 11 selects the type of pulse signal
from the band handled by the speaker, and the switching signal generator 104 outputs a
switching signal to the pulse selector 103 (step S402). The pulse selector 103 selects an output
pulse based on the input switching signal 104. Here, a procedure for outputting a low frequency
pulse signal will be described.
[0027]
When the output destination speaker and the output pulse are determined, the DSP unit 12 starts
measuring the distance to the speaker. In the following, the sampling interval is t seconds, and
time is represented by the number of times of sampling. For example, the delay time of 10
samples represents the time required to perform sampling 10 times, ie, 10 t seconds. In the
distance measurement, as shown in the graph on the upper side of FIG. 7, the signal input from
the microphone 4 through the microphone amplifier 5 is stored in the memory for M samples in
a state where no pulse signal is output (step S403 in FIG. 4). ). After the M samples have elapsed,
a pulse signal is output (step S404), and the N-M samples are stored in the memory (step S405).
At this time, the signal input by the microphone 4 is a graph as shown on the lower side of FIG.
[0028]
The series of measurements (steps S403 to S405) are repeated the number of times set in
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advance by the microcomputer 11 to perform synchronous addition (step S406). The
synchronous addition is to add the data obtained by repeatedly performing the same
measurement while aligning the time axis, and taking the average to suppress the influence of
noise and improve the S / N ratio. When the synchronous addition is completed, the average
value of the measurement results of the first half M samples is determined (DC offset calculating
means), and the average value of the first half M samples determined from the data of the
average values of all N samples is subtracted (DC offset subtracting means, Step S407). This is to
remove the DC offset (DC offset) generated at the time of A / D conversion from the
measurement result.
[0029]
The maximum value is detected from the data of the first half M samples after removing the DC
offset, and the DSP unit 12 stores the larger one of the maximum value and the preset value of
NoiseMin as the noise level (noise detection means, Step S408). Subsequently, a peak level is
detected from the latter half NM samples (peak detection means), and wire connection
determination and S / N ratio determination are performed (step S409). Next, the procedure of
step S409 (subroutine) will be specifically described with reference to FIG.
[0030]
The maximum value is detected from the second half NM samples after removing the DC offset
(step S501), and the DSP unit 12 stores the detected maximum value as a peak level in the
memory and the register (step S502). Next, determination of unconnected and determination of S
/ N ratio are performed from the stored noise level and peak level. In the determination of
unconnected, the ratio of the peak level to the noise level is calculated (signal-to-noise ratio
calculating means), and it is confirmed whether it falls below a preset threshold 1 (connection
determining means, step S503). For example, when the threshold 1 is 1.4, if the peak level is
smaller than 1.4 times the noise level, it is determined as unconnected.
[0031]
If it is determined that the connection is not yet made (step S503: No), the S / N ratio is
subsequently determined. Here, the threshold for delay measurement (threshold 2) and the
threshold for polarity determination (threshold 3) are set in advance, and the ratio of peak level
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to noise level is smaller than the ratio of threshold 2 to threshold 3 or not Make sure. Here,
threshold 2 and threshold 3 are variables. Assuming that threshold 2 is 2.0 and threshold 3 is
0.4, if the peak level is smaller than 5 times the noise level, it is determined that the S / N ratio is
insufficient.
[0032]
If it is determined in step S503 that the value is lower than the threshold 1, the DSP unit 12
outputs an error indicating that the connection is not yet made. If it is determined in step S504
that the S / N ratio is insufficient, an error to that effect is output. The content of the error output
from the DSP unit 12 is displayed on the screen (not shown) of the amplifier device 10 through
the microcomputer 11.
[0033]
Returning to FIG. 4, when the wire connection determination and the S / N ratio determination
are completed, a delay value is finally obtained (delay time measuring means, step S410). The
procedure of step S410 (subroutine) will be specifically described with reference to FIG. In the
present embodiment, NoiseMin is set to a level of −42 dB, and a value obtained by multiplying
the noise level stored in step S409 by the threshold 2 is set as the threshold 4. In the upper
diagram in FIG. 7, the time from when the pulse signal is output after M sampling until the level
(amplitude) of the output pulse signal exceeds threshold 4 for the first time is stored as Δ
sample, and correction is performed in the later procedure Used when. Here, the Δ sample varies
depending on the band of the pulse signal to be used. In this embodiment, in the pulse for each
band, the time (the number of samples) from the output start point of the pulse to the peak is
made to coincide, so for example, for the high frequency band as shown in FIG. The Δ sample
has a large value with the pulse signal of
[0034]
In the procedure for obtaining the delay value, in the lower graph of FIG. 7 representing the level
of the signal input by the microphone 4, first, the DSP unit 12 detects one sample at a time from
the start point (after M samples) where the pulse signal is output It is determined whether the
level exceeds the threshold 4 (step S601 in FIG. 6). If the threshold value 4 is exceeded (step
S602: YES), the Δ sample is subtracted from the elapsed time (D sample) from the output of the
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pulse signal to obtain a delay value (step S603). Thereafter, returning to FIG. 4, the procedure of
sound field measurement for one speaker is completed. On the other hand, when the threshold
value 4 is not exceeded (step S602: No), it is determined whether the determination of the
prescribed number of samples is completed (step S604). Here, if the determination for the
prescribed samples is completed (step S604: Yes), the DSP unit 12 outputs an error indicating
that the delay value could not be measured, assuming that no sample exceeding the threshold 4
is found (step S605), Display on the screen (not shown) of the amplifier device 10 via the
microcomputer 11. If the determination for the prescribed sample is not completed (step S604:
YES), the process returns to step S601 to determine whether the threshold value 4 is exceeded
for the next sample.
[0035]
Essentially, the delay value to be determined is the time from the start point (after M samples) at
which the pulse signal was output to the start point at which the pulse signal was detected by the
microphone 4. However, as shown in the lower diagram of FIG. It is difficult to determine the
start point at which the pulse signal is detected due to the influence of Therefore, a method is
employed in which D samples are measured based on the threshold value 4 and then correction
is performed (Δ sample is subtracted) according to the waveform of the pulse signal. The
number of specified samples in step S604 is an appropriate number equal to or less than the
number of M-N samples.
[0036]
Thus, the measurement for one speaker is completed, and thereafter, the procedure of steps
S401 to S410 is repeatedly performed for each speaker to obtain delay values for all the
speakers. When the delay values for all the speakers are obtained, the microcomputer 11
calculates the amount of delay given to each speaker so as to fit to the speaker having the
maximum delay value. For example, the delay value for R tweeter 301 is 108 samples, the delay
value for R woofer 302 is 264 samples, the delay value for L tweeter 303 is 152 samples, and
the delay value for L woofer 304 is 193 samples. Consider the case of correcting the delay
amount. At this time, the maximum value of the delay value is 264 samples of R woofer 302, the
delay amount of 156 samples for R tweeter 301, the delay amount of 0 samples for R woofer
302, and the delay amount of 112 samples for L tweeter 303 The delay amount of 71 samples is
set to L woofer 304.
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[0037]
The calculated delay amount is output from the microcomputer 11 to the delay device 9 of each
speaker, and the delay device 9 receiving the delay amount gives the audio signal a delay amount
and outputs it to the amplifier 2. In the above example, the audio signal reproduced by R tweeter
301 has a delay of 156 samples, the audio signal reproduced by R woofer 302 has a delay of 0
samples, and the audio signal reproduced by L tweeter 303 has 112 samples The delay 9 gives a
delay of 71 samples to the audio signal reproduced by the L woofer 304. Thus, the arrival time
from each speaker to the microphone 4 is matched by adding a delay when outputting an audio
signal.
[0038]
The sound field automatic correction system according to the embodiment of the present
invention adopts a method using a single pulse signal, and does not require complicated
processing such as DFT. This not only makes it possible to shorten the processing time required
to measure the delay value, but also has the advantage that it can be realized even if the capacity
of the memory required by the sound field measurement device 1 is small.
[0039]
As mentioned above, although the embodiment of the present invention has been described in
detail, the specific configuration is not limited to the present embodiment, and design changes
and the like within the scope of the present invention are also included. For example, although
the threshold value 4 is calculated when the maximum value of noise obtained in step S408
exceeds NoiseMin in the present invention, the threshold value 4 may always be varied according
to the maximum value of noise.
[0040]
The present invention is suitable for use in an automatic sound field correction system which
automatically corrects the bias of the sound field caused by the difference in the distance from
each speaker to the listening position in an audio system.
[0041]
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FIG. 1 is a configuration diagram showing a configuration of an automatic sound field correction
system according to an embodiment of the present invention.
It is the block diagram which showed the method of outputting a pulse signal from the sound
field measurement apparatus of FIG. It is a layout showing the arrangement of the speaker in the
room. It is the flowchart which showed the procedure performed when the microcomputer 11 of
FIG. 1 performs a sound field measurement. It is the flowchart which showed the procedure in
the subroutine shown by FIG.4 S409. It is the flowchart which showed the procedure in the
subroutine shown by FIG.4 S410. It is the graph which showed the pulse signal for low-pass
outputted from speaker 3 of Drawing 1, and the pulse signal which microphone 4 inputs. It is the
graph showing the pulse signal for high frequencies output from the speaker 3 of FIG.
Explanation of sign
[0042]
DESCRIPTION OF SYMBOLS 1 ... sound field measurement apparatus, 2 ... amplifier, 3 ... speaker,
4 ... microphone (microphone), 5 ... microphone amplifier, 9 ... delay device (delay time
adjustment means), 10 ... amplifier apparatus, 11 ... microcomputer, 12 ... DSP unit, 13 ... D / A
converter, 14 ... A / D converter, 101 ... pulse signal generator for low frequency band (pulse
signal generator for each band), 102 ... pulse signal generator for high frequency band (pulse for
each band Signal generation means), 103 ... pulse select part (pulse signal selection means), 104
... switching signal generation part, 105 ... level adjustment part, 106 ... speaker selection part
(speaker selection means), 107 ... output speaker selection signal generation part, 108 ... CH level
adjustment section
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