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JP2010151965

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Notice
This translation is machine-generated. It cannot be guaranteed that it is intelligible, accurate,
complete, reliable or fit for specific purposes. Critical decisions, such as commercially relevant or
financial decisions, should not be based on machine-translation output.
DESCRIPTION JP2010151965
The present invention provides a voice processing apparatus capable of reducing the
intelligibility due to reverberation at the time of reaching a receiver in the case of supplying an
audio signal to a speaker and emitting sound under reverberation. SOLUTION: A band dividing
unit 121 for converting an audio signal into a frequency domain, a gain adjusting unit 122 for
performing gain adjustment for each band on an audio signal converted into a frequency domain
by the band dividing unit 121, and gain adjustment A band combining unit 123 combines the
voice signals of the respective bands subjected to gain adjustment by the unit 122 and converts
them into the time domain, and D // outputs the voice signal converted into the time domain by
the band combining unit 123 to the speaker 1c. And an A converter 13. [Selected figure] Figure 2
Speech processing apparatus and speech processing method
[0001]
The present invention relates to an audio processing apparatus and an audio processing method
for performing audio signal processing on an audio signal emitted by a speaker installed under
reverberation.
[0002]
In disaster prevention broadcasts and in-house broadcasts, audio collected by a microphone is
emitted using a speaker.
08-05-2019
1
However, when a receiver listens to a disaster prevention broadcast, an in-house broadcast, etc.,
since the reverberation is convoluted into the sound directly delivered to the receiver from the
speaker, it may be difficult to hear the content of the talker's speech. This is because the sound
emitted from the speaker passes through various echo paths and reaches the receiver with
various delays.
[0003]
In order to cope with such a problem, measures are taken that the transmitter speaks slowly.
This is to reduce any overlap between direct sound and reverberation of the talker's speech.
[0004]
Further, Patent Document 1 describes a communication device provided with a speech speed
conversion device having a speech speed conversion unit that changes the time axis of the voice
signal of the other party.
[0005]
JP 2006-33789 A
[0006]
As described above, by slowing the talker's speech speed, it is possible to prevent the talker's
utterance contents from being difficult to hear under reverberation, but further improvement is
desired.
[0007]
The present invention has been proposed in view of the above-described circumstances, and in
the case of emitting an audio signal by supplying an audio signal to a speaker under
reverberation, reducing the hearing difficulty due to the reverberation when the receiver is
reached. It is an object of the present invention to provide an audio processing apparatus and an
audio processing method that can
[0008]
As means for solving the above-mentioned problems, the speech processing apparatus according
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2
to the present invention comprises a frequency domain converter for converting an audio signal
into a frequency domain, and an audio signal converted into a frequency domain by the
frequency domain converter. A gain adjustment unit that performs gain adjustment for each
band, a time domain conversion unit that combines audio signals of the respective bands that
have been subjected to gain adjustment by the gain adjustment unit, and converts them into a
time domain; And an output unit for outputting the audio signal converted into the signal to the
speaker.
[0009]
Further, in the audio processing method according to the present invention, gain adjustment is
performed on each band with respect to the audio signal converted into the frequency domain by
the frequency domain conversion step of converting the audio signal into the frequency domain
and the frequency domain conversion step. The adjustment step, the time domain conversion
step of combining the audio signals of the respective bands which have been subjected to the
gain adjustment in the gain adjustment step and converting them into the time domain, and the
audio signal converted into the time domain by the time domain conversion step And an output
step of outputting.
[0010]
The present invention performs gain adjustment for each band with respect to an audio signal
converted to the frequency domain, converts it to the time domain, and outputs it to a speaker,
thereby allowing an adjacent audio signal to be propagated under reverberation. It is possible to
reduce the overlapping of audio signal components in the frequency band to be
Therefore, the present invention can reduce the hearing loss due to the reverberation at the time
of reaching the receiver, in the case of supplying an audio signal to the speaker to emit the sound
under reverberation.
[0011]
Hereinafter, the best mode for carrying out the present invention will be described in detail with
reference to the drawings.
08-05-2019
3
The present invention is not limited to the following embodiments, and it goes without saying
that various modifications can be made without departing from the scope of the present
invention.
[0012]
An audio processing apparatus to which the present invention is applied is an apparatus that
performs audio signal processing on an audio signal emitted by a speaker installed under
reverberation.
The description will be made in the following order.
1.
First embodiment 2.
Second embodiment
[0013]
<1. First Embodiment> A voice processing apparatus to which the present invention is applied
is incorporated and used in a loudspeaker apparatus 1 according to a first embodiment as shown
in FIG.
[0014]
The loud-speaking device 1 according to the first embodiment includes a microphone 1a that
detects a voice of a speaker, a voice processing unit 1b that performs signal processing on a
voice signal detected by the microphone 1a, and a voice processing unit 1b. And a speaker 1c for
emitting the processed audio signal.
[0015]
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The microphone 1a detects the voice of the speaker, converts the detected voice into a voice
signal, and supplies the voice signal to the voice processing unit 1b.
[0016]
The audio processing unit 1 b converts the audio signal into an analog signal by converting the
audio signal supplied from the microphone 1 a into a digital signal, the signal processing unit 12
that performs signal processing on the digital signal, and And a D / A converter 13 for supplying
the speaker 1c.
[0017]
The A / D converter 11 converts the audio signal supplied from the microphone 1 a into a digital
signal, and supplies the audio signal after digital conversion to the signal processing unit 12.
The audio signal supplied to the A / D converter 11 may be not only the microphone 1a but also
a signal supplied from an external audio processing device by line input.
[0018]
Specifically, the signal processing unit 12 performs signal processing to be described later on the
audio signal supplied from the A / D converter 11, and supplies the audio signal subjected to the
signal processing to the D / A converter 13. Do.
[0019]
The D / A converter 13 converts the audio signal supplied from the signal processing unit 12
into an analog signal, and supplies the audio signal after analog conversion to the speaker 1 c.
[0020]
The speaker 1 c is installed under reverberation and emits the audio signal supplied from the
audio processing unit 1 b.
[0021]
In the loudspeaker apparatus 1 having the above configuration, since the speaker 1c is installed
under reverberation, reverberation is convoluted into the voice directly reaching the receiver
08-05-2019
5
from the speaker 1c, and the speech of the transmitter detected by the microphone 1a There are
cases where the sound may be emitted that is difficult to hear.
[0022]
Therefore, in the loudspeaker apparatus 1, the signal processing unit 12 of the audio processing
unit 1b has a configuration as shown in FIG. 2 in order to reduce intelligibility caused by such
reverberation.
[0023]
That is, the signal processing unit 12 includes a band division unit 121 that converts an audio
signal into a frequency domain, a gain adjustment unit 122 that performs gain adjustment for
each frequency band on the audio signal in the frequency domain, and an audio signal whose
gain has been adjusted. And a band combining unit 123 for converting to the time domain.
[0024]
The band division unit 121 converts the voice signal supplied from the A / D converter 11 into a
voice signal in the frequency domain, for example, using a time-frequency conversion method
such as short time Fourier transform for each frame that is a unit time. Do.
Then, the band division unit 121 divides the audio signal converted into the frequency domain
into, for example, bands of 1024 channels, and supplies the audio signal of each band to the gain
adjustment unit 122.
[0025]
Here, in the Fourier transform processing, since it becomes symmetrical in the central channel, it
is sufficient to consider only 512 points that are half of 1024 points. In the following, the lowest
channel is regarded as the first channel, and numbers are assigned in order. The channel of the
region is described as the 512th channel.
[0026]
Further, the band division unit 121 may perform band division by a method of band division
such as a DFT filter bank or a QMF filter bank other than the above-described Fourier transform
processing.
08-05-2019
6
[0027]
The gain adjustment unit 122 performs gain adjustment for each band on the audio signal
supplied from the band division unit 121.
That is, by setting different gains for each channel, the gain adjusting unit 122 can reduce the
overlap of the audio signal components in adjacent frequency bands when the audio signal is
propagated under reverberation.
[0028]
Specifically, in order to more effectively reduce the overlap of audio signal components in
adjacent frequency bands, for example, gain adjustment section 122 adjusts gains for audio
signals in each band as follows in each unit time Change to
[0029]
As a first specific example, the gain adjustment unit 122 performs gain adjustment so as to
alternately reproduce the even channel and the odd channel by changing the gain of each
channel frame by frame.
First, the gain adjustment unit 122 adjusts the amplitude level of the odd channel to a value close
to 0 while reproducing the even channel as it is in a certain frame.
In the next frame, the gain adjustment unit 122 adjusts the odd channel to be reproduced as it is
while setting the amplitude level of the even channel to zero.
In the next frame, the gain adjustment unit 122 adjusts the odd channel to a value close to 0
while reproducing the even channel as it is.
By adjusting the gain in this manner, the odd channels and the even channels are reproduced
08-05-2019
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once every two frames.
As described above, the gain adjustment unit 122 performs the gain adjustment so that the audio
signal in the odd-numbered band and the audio signal in the even-numbered band are alternately
output by the speaker 1 c, and thereby the audio signal in the adjacent frequency band The
overlap of the components can be effectively reduced.
In this way, the gain adjustment unit 122 can realize the improvement in hearing difficulty under
reverberation with simple processing.
[0030]
As a second specific example, the gain adjustment unit 122 may reproduce one channel every
three or more frames by changing the gain of each channel for each frame.
As a specific example, in the case of alternately reproducing one channel out of one set of three
channels, gain adjustment is performed as follows.
First, in a certain frame, the gain adjustment unit 122 adjusts the sound level of the remaining
channels to a value close to 0 while reproducing the channels 1, 4,..., 508, 511.
In the next frame, the gain adjustment unit 122 adjusts the remaining band to a value close to 0
while reproducing the 2, 5,..., 509, 512 channels. In the next band, the gain adjustment unit 122
adjusts the remaining bands to values close to 0 while reproducing the channels 3, 6,..., 510.
Then, the gain adjustment unit 122 repeats these processes.
[0031]
Further, the gain adjustment unit 122 not only selects the channel simply as in the two examples
described above, but also selects an appropriate channel so that reverberations do not overlap
according to the type of audio to be processed actually. It can also be done. In particular, the gain
adjustment unit 122 can multiply the gain at a level that does not affect the sound quality
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8
according to the human auditory characteristics and the masking level. For example, for the
audio signal as shown in FIG. 3A, as shown in FIG. 3B, the gain adjustment unit 122 reduces the
gain of the channel masked by the strong power component based on the masking level. Adjust
to become The gain adjustment unit 122 performs gain adjustment by multiplying the audio
signal of each band converted into the frequency domain by the correction gain set based on the
auditory characteristic, thereby maintaining the sound quality while maintaining the sound
quality. It is possible to reduce the overlap of audio components that are particularly difficult to
hear in terms of auditory characteristics.
[0032]
The audio signal of each band whose gain has been adjusted so as to reduce the overlap of audio
components under reverberation as described above is supplied from the gain adjustment unit
122 to the band synthesis unit 123.
[0033]
The band synthesizing unit 123 synthesizes the voice signals of the respective bands which have
been subjected to the gain adjustment by the gain adjusting unit 122, performs inverse Fourier
transform processing, and converts the signals into the time domain.
Then, the band combining unit 123 supplies the audio signal converted into the time domain to
the D / A converter 13.
[0034]
In this way, the audio signal divided into each band by the band dividing unit 121 can be
recombined with the original time domain audio signal by inverse Fourier transform. Note that
depending on the method, the original signal and the re-synthesized signal may be slightly
different, but they can be configured to have no essential influence.
[0035]
In the loudspeaker apparatus 1 having the signal processing unit 12 configured as described
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9
above, the gain adjustment unit 122 performs gain adjustment for each band with respect to the
audio signal converted into the frequency domain by the band division unit 121, After being
converted to the time domain by the band combining unit 123, the signal is output to the
speaker 1c. By performing such processing, the loudspeaker apparatus 1 can reduce the overlap
of the audio signal components in the adjacent frequency bands when the audio signal is
propagated under reverberation. Therefore, the loudspeaker device 1 can reduce the ambiguity
of the voice due to the reverberation at the time of reaching the receiver when the voice signal is
supplied to the speaker 1 c to emit sound under reverberation.
[0036]
<2. Second Embodiment> A voice processing apparatus to which the present invention is
applied is incorporated and used, for example, in a loudspeaker apparatus 2 according to a
second embodiment as shown in FIG.
[0037]
The loud-speaking device 2 according to the second embodiment includes a microphone 2a that
detects a voice of a speaker, a voice processing unit 2b that performs signal processing on a
voice signal detected by the microphone 2a, and a voice processing unit 2b. And a speaker 2c for
emitting the processed audio signal. Furthermore, in order to measure the reverberation time of
the environment where the speaker 2c emits an audio signal, the loudspeaker 2 includes a
microphone 2d that detects the sound that has been emitted by the speaker 2c and propagated
under the reverberation.
[0038]
The microphone 2a detects the voice of the speaker, converts the detected voice into a voice
signal, and supplies the voice signal to the voice processing unit 2b.
[0039]
The audio processing unit 2 b converts the audio signal into an analog signal by converting the
audio signal supplied from the microphone 2 a into a digital signal, the signal processing unit 22
that performs signal processing on the digital signal, and And a D / A converter 23 for supplying
08-05-2019
10
the speaker 2c.
Furthermore, the audio processing unit 2 b includes an A / D converter 24 that digitizes the
audio signal detected by the microphone 2 d.
[0040]
The A / D converter 21 converts the audio signal supplied from the microphone 2 a into a digital
signal, and supplies the audio signal after digital conversion to the signal processing unit 22. The
audio signal supplied to the A / D converter 21 may be not only the microphone 2a but also a
signal supplied from an external audio processing device by line input.
[0041]
Specifically, the signal processing unit 22 performs signal processing to be described later on the
audio signal supplied from the A / D converter 21 and supplies the audio signal subjected to the
signal processing to the D / A converter 23. Do.
[0042]
The D / A converter 23 converts the audio signal supplied from the signal processing unit 22
into an analog signal, and supplies the audio signal after analog conversion to the speaker 2 c.
[0043]
The speaker 2c is installed under reverberation and emits the audio signal supplied from the
audio processing unit 2b.
[0044]
The microphone 2 d detects the sound emitted from the speaker 2 c and propagated under the
reverberation, and supplies the detected sound signal to the A / D converter 24.
[0045]
In the loudspeaker apparatus 2 configured as described above, since the speaker 2c is installed
under reverberation, reverberation is convoluted into the voice that directly reaches the receiver
from the speaker 2c, and the speech of the transmitter detected by the microphone 2a There are
08-05-2019
11
cases where the sound may be emitted that is difficult to hear.
[0046]
Therefore, the signal processing unit of the voice processing unit 2b in order to appropriately
reduce the intelligibility caused by such reverberation according to the reverberation time of the
environment where the loudspeaker 2c emits the voice signal. Reference numeral 22 has a
configuration as shown in FIG.
[0047]
As in the first embodiment, the signal processing unit 22 includes a band division unit 221 that
converts an audio signal into a frequency domain, and a gain adjustment unit 222 that performs
gain adjustment on the audio signal in the frequency domain for each band. And a band
combining unit 223 configured to convert the gain-adjusted audio signal into a time domain.
Furthermore, as shown in FIG. 5, the signal processing unit 22 according to the second
embodiment performs a speech speed conversion unit 224 that performs speech speed
conversion processing on an audio signal at a stage before the band division unit 221; And a
pitch changing unit 225 that changes the pitch with time based on the fundamental frequency of
[0048]
In the signal processing unit 22 having such a configuration, functions similar to those of the
signal processing unit 12 according to the first embodiment are realized by the band division
unit 221, the gain adjustment unit 222, and the band combination unit 223. The description of
the operation content is omitted.
[0049]
The speech speed conversion unit 224 performs time expansion reproduction using speech
speed conversion to make it easier to hear speech under reverberation, as shown in FIG. 6, the
time supplied from the A / D converter 21. The speech speed conversion process is performed on
the speech signal of the area.
[0050]
08-05-2019
12
Specifically, as shown in FIG. 6, the speech speed conversion unit 224 includes an audio buffer
224a that temporarily holds an audio signal.
The speech speed conversion unit 224 temporarily holds the audio signal supplied from the A /
D converter 21 in the audio buffer 224a in response to a control signal from the control unit 227
described later, and decompresses the speech signal by speech speed conversion processing.
Output to the pitch changing unit 225 connected to the subsequent stage.
As described above, the speech speed conversion unit 224 can make it easy to hear speech under
reverberation by performing time expansion reproduction using speech speed conversion.
[0051]
The speech speed conversion unit 224 constructs a speech speed conversion processing system
using PICOLA (Pointer Interval Controlled OverLap and Add) or the like.
In addition to this, the speech speed conversion unit 224 may perform speech speed conversion
processing by a speech speed conversion method such as Time Domain Harmonic Scaling
(TDHS).
Further, although the speech speed conversion unit 224 is provided at the front stage of the
band division unit 221, the speech speed conversion unit 224 may be provided at the rear stage
of the band synthesis unit 223 or the rear stage of the pitch change unit 225.
[0052]
As shown in FIG. 7, the pitch changing unit 225 is provided with a fundamental frequency
estimating unit 225a, and it is possible to make it easier to hear speech under reverberation by
reproducing while changing the pitch with time. The following process is performed.
[0053]
That is, the fundamental frequency estimation unit 225a estimates the fundamental frequency F0
from the speech signal supplied from the speech speed conversion unit 224 using, for example,
08-05-2019
13
an autocorrelation method or a comb filter, and outputs the estimation result to the pitch change
unit 225. Notice.
[0054]
The pitch changing unit 225 is supplied with the audio signal supplied from the speech speed
converting unit 224 and the fundamental frequency F0 corresponding to this signal.
The pitch changing unit 225 changes the fundamental frequency F0 with time based on the
fundamental frequency F0 to change the pitch of the audio signal.
For example, when the fundamental frequency F0 of the output of the previous frame is the same
as the fundamental frequency F0 of the current frame, the pitch changing unit 225 changes the
fundamental frequency F0 by several Hz up or down.
When the fundamental frequency F0 of the output of the previous frame is different from the
fundamental frequency F0 of the current frame, the pitch changing unit 225 outputs the audio
signal as it is.
In this manner, the pitch changing unit 225 temporally changes the pitch of the supplied audio
signal and supplies it to the band dividing unit 221 connected to the subsequent stage.
[0055]
In addition to the changing method described above, the method of simply changing the
playback speed or changing the playback speed after changing the speech speed and changing
the playback speed may be used to change the fundamental frequency. Other methods such as a
method of operating a band in the frequency domain may be used.
Further, although the pitch changing unit 225 is provided at the front stage of the band dividing
unit 221, it may be provided at the rear stage of the band synthesizing unit 223 or the front
stage of the pitch changing unit 225.
08-05-2019
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[0056]
Speech (vowel) is known to have a harmonic structure. For example, when the fundamental
frequency of the vowel is 100 Hz, a strong component is included in the vicinity of an integral
multiple of the fundamental frequency such as 200 Hz and 300 Hz. The pitch changing unit 225
can reduce the overlap between the direct sound and the reverberation for each frequency band
by changing the above-mentioned strong frequency component, paying attention to such
characteristics of the harmonic structure.
[0057]
The reverberation time measurement unit 226 measures the reverberation time of the
environment in which the speaker 2c is installed, using the audio signal emitted from the speaker
2c and detected by the microphone 2d. That is, the reverberation time measurement unit 226
measures the reverberation time of the environment in which the speaker 2 c is installed using
the audio signal of the microphone 2 d digitally converted by the A / D converter 24, and
transmits the measurement result to the control unit 227. Notice.
[0058]
The control unit 227 switches and performs audio signal processing to be performed on the
voice of the speaker detected by the microphone 2 a according to the reverberation time notified
from the reverberation time measurement unit 226. Performing switching processing in this way
may cause voice distortion by performing voice processing. For example, voice processing is not
performed when the reverberation time is shorter than the threshold, and voice is processed
when the reverberation time is longer than the threshold. It is because it is desirable to process.
[0059]
As a first specific example, according to the flowchart shown in FIG. 8, control unit 227 responds
to the reverberation time notified from reverberation time measurement unit 226 to the voice of
the speaker detected by microphone 2 a. Switch the audio signal processing to be performed.
08-05-2019
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[0060]
In step S1, the control unit 227 determines whether the measured reverberation time is 0.3
seconds or more. If 0.3 seconds or more, the process proceeds to step S3, and if less than 0.3
seconds Go to S2.
[0061]
In step S2, the control unit 227 controls each processing unit so as to supply the audio signal
supplied from the A / D converter 21 directly to the D / A converter 23 without performing any
processing.
[0062]
In step S3, the control unit 227 determines whether the measured reverberation time is 1 second
or more. If 1 second or more, the process proceeds to step S5, and if less than 1 second, the
process proceeds to step S4.
[0063]
In step S4, the control unit 227 alternately outputs the audio signal of the odd-numbered band
and the audio signal of the even-numbered band to the audio signal supplied from the A / D
converter 21 by the speaker 1c. The operation of the gain adjustment unit 222 is controlled so as
to adjust the gain as described above.
The control unit 227 controls the speech speed conversion unit 224 and the pitch change unit
225 to output the supplied audio signal directly to the subsequent stage without performing any
signal processing.
[0064]
In step S5, the control unit 227 determines whether the measured reverberation time is 2
seconds or more, and proceeds to step S7 when 2 seconds or more, and proceeds to step S6
when less than 2 seconds.
[0065]
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In step S6, the control unit 227 changes the gain of each channel with respect to the audio signal
supplied from the A / D converter 21 so as to set one in three channels for every three frames.
The operation of the gain adjustment unit 222 is controlled to reproduce one channel alternately.
The control unit 227 controls the speech speed conversion unit 224 and the pitch change unit
225 to output the supplied audio signal directly to the subsequent stage without performing any
signal processing.
[0066]
In step S7, the control unit 227 controls the operation of the gain adjustment unit 222 so as to
alternately reproduce one channel out of one set of three channels every three frames, and the
speech speed is converted by the speech speed conversion process. The operation of the speech
speed conversion unit 224 is controlled to be slow.
[0067]
As a second specific example, according to the flowchart shown in FIG. 9, control unit 227
responds to the reverberation time notified from reverberation time measurement unit 226 to
the voice of the speaker detected by microphone 2a. Switch the audio signal processing to be
performed.
[0068]
In step S11, the control unit 227 determines whether the measured reverberation time is 0.3
seconds or more. If 0.3 seconds or more, the process proceeds to step S13, and if less than 0.3
seconds Go to S12.
[0069]
In step S12, the control unit 227 controls each processing unit so as to supply the audio signal
supplied from the A / D converter 21 directly to the D / A converter 23 without performing any
processing.
[0070]
In step S13, the control unit 227 determines whether or not the measured reverberation time is
one second or more, and proceeds to step S15 when one second or more, and proceeds to step
S14 when less than one second.
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[0071]
In step S14, with respect to the audio signal supplied from the A / D converter 21, the control
unit 227 alternately outputs the audio signal of the odd-numbered band and the audio signal of
the even-numbered band by the speaker 1c. The operation of the gain adjustment unit 222 is
controlled so as to adjust the gain as described above.
The control unit 227 controls the speech speed conversion unit 224 and the pitch change unit
225 to output the supplied audio signal directly to the subsequent stage without performing any
signal processing.
[0072]
In step S15, the control unit 227 determines whether the measured reverberation time is 2
seconds or more, and proceeds to step S17 when 2 seconds or more, and proceeds to step S16
when less than 2 seconds.
[0073]
In step S16, the control unit 227 changes the gain of each channel with respect to the audio
signal supplied from the A / D converter 21 so as to set one for each of three channels in one set
of three channels. The operation of the gain adjustment unit 222 is controlled to reproduce one
channel alternately.
The control unit 227 controls the speech speed conversion unit 224 and the pitch change unit
225 to output the supplied audio signal directly to the subsequent stage without performing any
signal processing.
[0074]
In step S17, the control unit 227 controls the operation of the gain adjustment unit 222 so as to
alternately reproduce one channel out of one set of three channels every three frames, while
temporally changing the pitch of the audio signal. Control the operation of the speech speed
conversion unit 224.
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[0075]
According to the first and second specific examples, the control unit 227 uses the reverberation
time as an evaluation index of the environment in which the speaker 2c is installed, and controls
the operation of each unit according to the reverberation time to thereby deteriorate the sound
quality. It is possible to reduce the overlap of audio signal components in adjacent frequency
bands while limiting.
That is, the control unit 227 causes deterioration of the sound quality to increase as the number
of frames reproduced one by one increases merely as the reverberation time increases, but
combines with other audio signal processing without merely increasing the number of frames.
This makes it possible to reduce the ambiguity of the voice while limiting the degradation of the
sound quality.
[0076]
As described above, the loud-speaking device 2 according to the second embodiment of the
present invention is an audio ambiguity due to the reverberation at the time of reaching the
receiver when the audio signal is supplied to the speaker 2c to emit sound under reverberation.
Can be reduced.
[0077]
It is a figure showing composition of a loudspeaker system concerning a 1st embodiment.
It is a figure showing the concrete composition of the speech processing part concerning a 1st
embodiment.
It is a figure where it uses for description of the process which concerns on a gain adjustment
part.
It is a figure which shows the structure of the loudspeaker apparatus based on 2nd Embodiment.
08-05-2019
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It is a figure showing the concrete composition of the speech processing part concerning a 2nd
embodiment.
It is a figure for demonstrating the process which concerns on a speech speed conversion part.
It is a figure for demonstrating the process which concerns on a pitch conversion part.
It is a flowchart in which it uses for description of the switching process of the 1st specific
example concerning a control part.
FIG. 10 is a flowchart for describing switching processing of a second specific example according
to a control unit; FIG.
Explanation of sign
[0078]
1, 2 loudspeakers, 1a, 2a, 2d microphones, 1b, 2b audio processing units, 1c, 2c speakers, 11,
21, 24 A / D converters, 12, 22 signal processing units, 13, 23 D / A conversions , 121, 221
band division unit, 122, 222 gain adjustment unit, 123, 223 band synthesis unit, 224 speech
speed conversion unit, 224a voice buffer, 225 pitch change unit, 225a fundamental frequency
estimation unit, 226 reverberation time measurement unit , 227 control unit
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