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DESCRIPTION JPWO2018105077

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DESCRIPTION JPWO2018105077
Abstract: A speech enhancement apparatus extracts a component including a fundamental
frequency (F0) of speech from an input signal and outputs it as a first filter signal, and a first
formant of speech from the input signal (F1). And a second filter (22) for outputting as a second
filter signal, and a component including a second formant (F2) of speech from the input signal,
and outputting as a third filter signal A first mixing section (31) for mixing the first filter signal
and the second filter signal and outputting a first mixed signal, and a first filter signal A second
mixing section (32) for mixing the third filter signal and outputting a second mixed signal, and
delaying the first mixed signal by a first delay amount (D1) for generating a first audio signal
Generating a first delay control unit (41) Has a second delay control unit for generating a second
audio signal a second mixing signal of the second delay amount (D2) by delaying a (42).
Speech enhancement apparatus, speech enhancement method, and speech processing program
[0001]
The present invention relates to a speech enhancement apparatus, speech enhancement method,
and speech processing program for generating a first speech signal for one ear and a second
speech signal for the other ear from an input signal.
[0002]
In recent years, research on an advanced driving support system (ADAS) for assisting driving of a
car has been advanced.
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Important functions of ADAS include, for example, a function of providing clear and easy-to-hear
guidance voices to elderly drivers, and a function of providing comfortable hands-free
communication even under high noise. In the field of television receivers, research is also
underway to improve the ease of listening to broadcast audio flowing from a television when an
elderly person views a television.
[0003]
By the way, in auditory psychology, there is known a phenomenon called auditory masking in
which a sound that would normally be clear can be difficult to hear by being masked (disturbed)
by another sound. As auditory masking, frequency masking where the sound of one frequency
component is masked by a large sound of another frequency component having a nearby
frequency, and frequency masking that is difficult to hear, and the subsequent sound is masked
by a preceding sound. There is time masking that makes it difficult to hear. In particular, elderly
people are susceptible to auditory masking and tend to have a reduced ability to hear vowels and
follow-on sounds.
[0004]
As measures against this, there have been proposed hearing aid methods for persons whose
frequency resolution and time resolution of the hearing have been reduced (see, for example,
Non-Patent Document 1 and Patent Document 1). In these hearing aid methods, in order to
reduce the effects of auditory masking (simultaneous masking), the input signal is divided on the
frequency axis, and the two signals generated by the division are different for the left ear and for
the right ear, respectively. A hearing aid method called binaural hearing aid is used, which allows
one sound to be perceived in the brain of the user (listener) by presenting the characteristic.
[0005]
It has been reported that the binaural separation and hearing aids the user to increase the clarity
of speech. This is because the acoustic signal of the frequency band to be masked (or the acoustic
signal of the time domain) and the acoustic signal of the frequency band to be masked (or the
acoustic signal of the time domain) are presented to different ears respectively. It is considered
that this is because it becomes easy to perceive the sound that has been masked.
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[0006]
D.S. Chaudhari and P.C. Pandey, “Dichotic
Presentation of Speech Signal Using Critical
Filter Bank for Bilateral Sensorineural Hearing
Impairment”, Proc.16th ICA, Seattle Washington
USA, June 1998, vol.1, pp.213−214
[0007]
Patent No. 53511281 (pages 8 to 12, FIG. 7)
[0008]
However, in the above-mentioned conventional hearing aid method, since the pitch frequency
component which is a component of the fundamental frequency of the sound is not presented to
both ears, the hearing aid to which this method is applied is a person with mild hearing loss or a
healthy hearing person. When used, there is a problem that it becomes difficult to hear the voice
due to the loss of the auditory balance between the left ear and the right ear, such as the voice
being biased to one ear side and the voice being double-heard, etc. .
[0009]
Further, the above conventional hearing aid method is applied to an earphone-worn hearing aid
for the hearing impaired, and application to devices other than the earphone-worn hearing aid is
not considered.
That is, the above-mentioned conventional hearing aid method is not considered to be applied to
a loud sound system. For example, in a system in which a loud sound is heard using a twochannel stereo speaker, the sounds emitted by the left and right speakers are left and right ears.
The time to reach each point may be slightly different and the effects of binaural hearing loss
may be reduced.
[0010]
The present invention has been made to solve the problems as described above, and a speech
enhancement apparatus, speech enhancement method, and speech processing program capable
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of generating a speech signal for outputting a clear and easy-to-hear loud speech. Intended to be
provided.
[0011]
The speech enhancement apparatus according to the present invention receives an input signal,
and from the input signal, a first speech signal for a first ear and a second speech signal for a
second ear opposite to the first ear. A first band component of a predetermined frequency band
including a fundamental frequency of speech from the input signal, and the first band component
is used as a first filter signal. A first filter to output and a second band component of a
predetermined frequency band including a first formant of speech are extracted from the input
signal, and the second band component is output as a second filter signal. A third filter that
extracts a third band component of a predetermined frequency band including a second formant
of a voice from the input signal, and outputs the third band component as a third filter signal;
Filter and the A first mixing unit that outputs a first mixed signal by mixing the first filter signal
and the second filter signal, and mixing the first filter signal and the third filter signal A second
mixing unit for outputting a second mixed signal; and a first delay for generating the first audio
signal by delaying the first mixed signal by a predetermined first delay amount. A control unit,
and a second delay control unit that generates the second audio signal by delaying the second
mixed signal by a predetermined second delay amount.
[0012]
The voice enhancement method according to the present invention receives an input signal, and
from the input signal, a first voice signal for a first ear and a second voice signal for a second ear
opposite to the first ear. A first band component of a predetermined frequency band including a
fundamental frequency of speech from the input signal, and the first band component is used as
a first filter signal. Outputting, extracting a second band component of a predetermined
frequency band including a first formant of speech from the input signal, and outputting the
second band component as a second filter signal; Extracting a third band component of a
predetermined frequency band including a second formant of speech from the input signal, and
outputting the third band component as a third filter signal; and the first filter With the signal
Outputting a first mixed signal by mixing with the second filtered signal, and outputting a second
mixed signal by mixing the first filtered signal and the third filtered signal Generating the first
audio signal by delaying the first mixed signal by a predetermined first delay amount; and
secondly generating the second mixed signal as a second predetermined signal. Generating the
second audio signal by delaying the second audio signal.
[0013]
According to the present invention, it is possible to generate an audio signal for outputting a
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clear and easy to hear loud voice.
[0014]
It is a functional block diagram which shows schematic structure of the speech enhancement
apparatus based on Embodiment 1 of this invention.
FIG. 2 (a) is an explanatory view showing the frequency characteristic of the first filter, FIG. 2 (b)
is an explanatory view showing the frequency characteristic of the second filter, and FIG. 2 (c) is
a third filter Explanatory drawing which shows a frequency characteristic, FIG.2 (d) is
explanatory drawing which shows the relationship between fundamental frequency and each
formant, when the frequency characteristic of all the filters is overlap | superposed.
FIG. 3A is an explanatory view showing a frequency characteristic of the first mixed signal, and
FIG. 3B is an explanatory view showing a frequency characteristic of the second mixed signal.
5 is a flowchart illustrating an example of speech enhancement processing (speech enhancement
method) performed by the speech enhancement device according to the first embodiment.
FIG. 2 is a block diagram schematically showing a hardware configuration (in the case of using an
integrated circuit) of the speech enhancement device according to the first embodiment.
FIG. 2 is a block diagram schematically showing a hardware configuration (in the case of using a
program executed by a computer) of the speech enhancement device according to the first
embodiment.
It is a figure which shows schematic structure of the speech enhancement apparatus (when
applied to a car navigation system) which concerns on Embodiment 2 of this invention. It is a
figure which shows schematic structure of the speech enhancement apparatus (when applied to a
television receiver) which concerns on Embodiment 3 of this invention. It is a functional block
diagram which shows schematic structure of the speech enhancement apparatus based on
Embodiment 4 of this invention. It is a functional block diagram which shows schematic structure
of the speech enhancement apparatus based on Embodiment 5 of this invention. FIG. 21 is a
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flowchart showing an example of speech enhancement processing (speech enhancement method)
performed by the speech enhancement device according to the fifth embodiment; FIG.
[0015]
Hereinafter, embodiments of the present invention will be described with reference to the
attached drawings. Note that components given the same reference numerals throughout the
drawings have the same configuration and the same function.
[0016]
<< 1 >> First Embodiment << 1-1 >> Configuration FIG. 1 is a functional block diagram showing a
schematic configuration of the speech enhancement apparatus 100 according to the first
embodiment of the present invention. The speech enhancement apparatus 100 is an apparatus
capable of implementing the speech enhancement method according to the first embodiment and
the speech processing program according to the first embodiment.
[0017]
As shown in FIG. 1, the speech enhancement apparatus 100 mainly includes a signal input unit
11, a first filter 21, a second filter 22, a third filter 23, and a first mixing. A unit 31, a second
mixing unit 32, a first delay control unit 41, and a second delay control unit 42 are provided. In
FIG. 1, 10 is an input terminal, 51 is a first output terminal, and 52 is a second output terminal.
[0018]
The speech enhancement apparatus 100 receives an input signal through the input terminal 10,
and from this input signal, a first speech signal for one (first) ear and a second speech signal for
the other (second) ear , And outputs the first audio signal from the first output terminal 51 and
the second audio signal from the second output terminal 52.
[0019]
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The input signal of the speech enhancement apparatus 100 is, for example, voice, music, noise or
other acoustic signal captured through an acoustic transducer such as a microphone (not shown)
and a sonic vibration sensor (not shown), or a wireless telephone, wired It is a signal obtained by
taking an electrical acoustic signal output from an external device such as a telephone set or a
television receiver through a line cable or the like.
Here, an audio signal collected by a one-channel (monaural) microphone will be described as an
example of an acoustic signal.
[0020]
The operation principle of the speech enhancement apparatus 100 according to the first
embodiment will be described below based on FIG.
[0021]
The signal input unit 11 A / D (analog / digital) converts an acoustic signal included in the input
signal, performs sampling processing at a predetermined sampling frequency (for example, 16
kHz), and performs predetermined frame intervals (for example, 10 ms) , And output to the first
filter 21, the second filter 22, and the third filter 23 as an input signal xn (t) which is a discrete
signal in the time domain.
Here, n represents a frame number assigned to each frame when the input signal is divided into
frames, and t represents a discrete time number (an integer of 0 or more) in sampling.
[0022]
FIG. 2 (a) is an explanatory view showing a frequency characteristic of the first filter 21, FIG. 2
(b) is an explanatory view showing a frequency characteristic of the second filter 22, and FIG. 2
(c) is a third Explanatory drawing which shows the frequency characteristic of the filter 23, FIG.2
(d) is explanatory drawing which shows the relationship between fundamental frequency and
each formant, when the frequency characteristic of all the filters is overlap | superposed.
[0023]
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The first filter 21 receives an input signal xn (t), and from the input signal xn (t), a first of a
predetermined frequency band (pass band) including a fundamental frequency (also referred to
as pitch frequency) F0 of speech. The band component is extracted, and the first band
component is output as a first filter signal y1 n (t).
In other words, the first filter 21 passes the first band component of the frequency band
including the fundamental frequency F0 of speech in the input signal xn (t) and does not pass
frequency components other than the first band component. The first filter signal y1 n (t) is
output. The first filter 21 is formed of, for example, a band pass filter having characteristics as
shown in FIG. In FIG. 2A, fc0 is a cut-off frequency of the lower limit of the pass band of the
band-pass filter constituting the first filter 21, and fc1 is a cut-off frequency of the upper limit of
the pass band. Further, in FIG. 2A, F0 schematically represents the spectral component of the
fundamental frequency. As the band-pass filter, for example, a finite impulse response (FIR) filter,
an infinite impulse response (IIR) filter, or the like can be used.
[0024]
The second filter 22 receives the input signal xn (t), and extracts from the input signal xn (t) a
second band component of a predetermined frequency band (pass band) including the first
formant F1 of speech. , And outputs the second band component as a second filter signal y2n (t).
In other words, the second filter 22 passes the second band component of the frequency band
including the first formant F1 of the voice in the input signal xn (t) and does not pass frequency
components other than the second band component. Thus, the second filter signal y2n (t) is
output. The second filter 22 is formed of, for example, a band pass filter having characteristics as
shown in FIG. 2 (b). In FIG. 2B, fc1 is a cutoff frequency of the lower limit of the pass band of the
band pass filter constituting the second filter 22, and fc2 is a cutoff frequency of the upper limit
of the pass band. Further, in FIG. 2 (b), F1 schematically represents the spectral component of the
first formant. For example, an FIR filter, an IIR filter, or the like can be used as the band pass
filter.
[0025]
The third filter 23 receives the input signal xn (t), and extracts from the input signal xn (t) a third
band component of a predetermined frequency band (pass band) including the second formant
F2 of speech. , And the third band component is output as a third filter signal y3n (t). In other
words, the third filter 23 passes the third band component of the frequency band including the
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second formant F2 of the voice in the input signal xn (t) and does not pass frequency
components other than the third band component. Thus, the third filter signal y3n (t) is output.
The third filter 23 is formed of, for example, a band pass filter having the characteristics as
shown in FIG. In FIG. 2C, fc2 is a cut-off frequency at the lower limit of the pass band of the
band-pass filter constituting the third filter 23. In the example of FIG. 2C, the third filter 23 uses
frequency components higher than the cutoff frequency fc2 as a pass band. However, the third
filter 23 can also be a band pass filter having an upper limit cutoff frequency. Further, in FIG. 2C,
F2 schematically represents the spectral component of the second formant. For example, an FIR
filter, an IIR filter, or the like can be used as the band pass filter.
[0026]
The fundamental frequency of voice F0 is distributed in a band of about 125 Hz to 400 Hz, the
first formant F1 is distributed in a band of about 500 Hz to 1200 Hz, and the second formant F2
is It is known to be distributed in a band of approximately 1500 Hz to 3000 Hz. Therefore, in a
preferable example in the first embodiment, fc0 = 50 Hz, fc1 = 450 Hz, and fc2 = 1350 Hz.
However, these values are not limited to the above examples, and can be adjusted according to
the state of the audio signal included in the input signal. Further, as for the cutoff characteristics
of the first filter 21, the second filter 22 and the third filter 23, as a preferable example in the
first embodiment, the number of filter taps is about 96 in the case of the FIR type filter. It is a
filter, and in the case of the IIR type filter, is a filter having a sixth-order Butterworth
characteristic. However, the first filter 21, the second filter 22, and the third filter 23 are not
limited to these examples, and the first and second output terminals 51 of the speech
enhancement apparatus 100 according to the first embodiment. , 52, and may be appropriately
adjusted in accordance with the hearing characteristics of the user (listener) and an external
device such as a speaker connected thereto.
[0027]
As described above, by using the first filter 21, the second filter 22, and the third filter 23, as
shown in FIG. A band component including the frequency F0, a band component including the
first formant F1, and a band component including the second formant F2 can be separated.
[0028]
FIG. 3 (a) is an explanatory view showing a frequency characteristic of the first mixed signal s1n
(t), and FIG. 3 (b) is an explanatory view showing a frequency characteristic of the second mixed
signal s2n (t).
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[0029]
The first mixing unit 31 mixes the first filtered signal y1n (t) and the second filtered signal y2n
(t) to form a first mixed signal as shown in FIG. 3 (a). Generate s1 n (t).
Specifically, the first mixing unit 31 outputs the first filter signal y1n (t) output from the first
filter 21 and the second filter signal y2n output from the second filter 22. And the first filter
signal y1n (t) and the second filter signal y2n (t) according to the following equation (1) to
output the first mixed signal s1n (t).
s1 n (t) = α · y 1 n (t) + β · y 2 n (t) (1) 0 t t <160
[0030]
In Equation (1), α and β are constants (coefficients) predetermined to perform perceptual
volume correction of the mixed signal. In the first mixed signal s1n (t), since the second formant
component F2 is attenuated, it is desirable to correct the lack of volume in the high region with
the constants α and β. In a preferable example in the first embodiment, α = 1.0 and β = 1.2.
That is, the first mixing unit 31 mixes the first filter signal y1n (t) and the second filter signal y2n
(t) at a predetermined first mixing ratio (ie, α: β). . However, the values of constants α and β
are not limited to the above example, and the externals such as speakers connected to first and
second output terminals 51 and 52 of speech enhancement apparatus 100 according to the first
embodiment. It is possible to adjust appropriately according to the device and the user's hearing
characteristic.
[0031]
The second mixing unit 32 mixes the first filtered signal y1n (t) with the third filtered signal y3n
(t) to form a second mixed signal as shown in FIG. 3 (b). Generate s2n (t). Specifically, the second
mixing unit 32 outputs the first filter signal y1 n (t) output from the first filter 21 and the third
filter signal y 3 n output from the third filter 23. And the first filter signal y1n (t) and the third
filter signal y3n (t) according to the following equation (2) to output a second mixed signal s2n
(t). s2 n (t) = α · y 1 n (t) + β · y 3 n (t) (2) 0 ≦ t <160
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[0032]
In Equation (2), α and β are preset constants for performing perceptual volume correction of
the mixed signal. The constants α and β in equation (2) may be different from those in equation
(1). As with the first mixed signal s1n (t), in the second mixed signal s2n (t), the second formant
component F2 is attenuated, so the volume constant of the high region is corrected by these two
constants. As a preferable example in the first embodiment, α = 1.0 and β = 1.2. That is, the
second mixing unit 32 mixes the first filtered signal y1n (t) and the third filtered signal y3n (t) at
a predetermined second mixing ratio (ie, α: β). . However, the values of constants α and β are
not limited to the above example, and the externals such as speakers connected to first and
second output terminals 51 and 52 of speech enhancement apparatus 100 according to the first
embodiment. It is possible to adjust appropriately according to the device and the user's hearing
characteristic.
[0033]
The first delay control unit 41 generates the first audio signals s 1 to 1 n (t) by delaying the first
mixed signal s 1 n (t) by a predetermined first delay amount. In other words, the first delay
control unit 41 controls the first delay amount that is the delay amount of the first mixed signal
s1n (t) output from the first mixing unit 31, that is, the first delay control unit 41 The time delay
of the mixed signal s1n (t) is controlled. Specifically, the first delay control unit 41 outputs, for
example, the first audio signals s to 1n (t) to which the time delay is added by D1 samples
according to the following equation (3).
[0034]
[0035]
The second delay control unit 42 generates the second audio signals s to 2n (t) by delaying the
second mixed signal s2n (t) by a predetermined second delay amount.
In other words, the second delay control unit 42 controls the second delay amount which is the
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delay amount of the second mixed signal s2n (t) output from the second mixing unit 32, that is,
the second delay control unit 42 The time delay of the mixed signal s2n (t) is controlled.
Specifically, the second delay control unit 42 outputs, for example, the second audio signals s to
2n (t) to which the time delay is added by D2 samples according to the following equation (4).
[0036]
[0037]
In the first embodiment, the first audio signals s to 1n (t) output from the first delay control unit
41 are output to the external device via the first output terminal 51, and the second delay control
is performed. The second audio signals s to 2n (t) output from the unit 42 are output to the
external device via the second output terminal 52.
The external device is, for example, a voice and sound processing device provided in a television
set, a hands-free communication device, and the like. The audio / sound processing device is a
device provided with a signal amplification device such as a power amplifier and an audio output
unit such as a speaker. In addition, when the audio signal subjected to the emphasizing
processing is output and recorded to a recording device such as an IC (integrated circuit)
recorder, the recorded audio signal is output by another audio sound processing device. Is also
possible.
[0038]
Note that the first delay amount D1 (D1 sample) is a time of 0 or more, and the second delay
amount D2 (D2 sample) is a time of 0 or more. It can be a value different from the delay amount
D2. The roles of the first delay control unit 41 and the second delay control unit 42 are as
follows: the first speaker (for example, left speaker) connected to the first output terminal 51; the
first ear (for example, left) of the user And the second speaker (e.g., right speaker) connected to
the second output terminal 52 to the second ear (e.g., the opposite ear of the first ear) of the
user, e.g. Control the first delay amount D1 of the first audio signal s ~ 1n (t) and the second
delay amount D2 of the second audio signal s ~ 2n (t) when the distance to the ear) is different It
is to be. In the first embodiment, the time at which the user hears a sound based on the first
audio signal s ~1 n (t) with the first ear and the second ear based on the second audio signal s ~2
n (t) It is possible to adjust the first delay amount D1 and the second delay amount D2 so as to
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bring the time of listening to the sound closer (preferably, to coincide).
[0039]
<< 1-2 >> Operation Next, an example of the operation (algorithm) of the speech enhancement
device 100 will be described. FIG. 4 is a flowchart showing an example of speech enhancement
processing (speech enhancement method) executed by the speech enhancement apparatus 100
according to the first embodiment.
[0040]
The signal input unit 11 takes in an acoustic signal at a predetermined frame interval (step
ST1A), and sets the first filter 21, the second filter 22, and the third filter as an input signal xn (t)
which is a time domain signal. Execute processing to output to 23. If the sample number t is less
than or equal to the predetermined value T (YES in step ST1B), the process of step ST1A is
repeated until the sample number t becomes the value T. For example, T = 160. However, it is
also possible to set T to a value other than 160.
[0041]
The first filter 21 receives the input signal xn (t), passes only the first band component (low
frequency component) of the frequency band including the fundamental frequency F0 of speech
in the input signal xn (t), The first filter processing for outputting the first filter signal y1 n (t) is
executed (step ST2).
[0042]
The second filter 22 receives the input signal xn (t) and passes only the second band component
(mid-range component) of the frequency band including the first formant F1 of the voice in the
input signal xn (t). The second filter processing for outputting the second filter signal y2n (t) is
executed (step ST3).
[0043]
The third filter 23 receives the input signal xn (t) and passes only the third band component
(high band component) of the frequency band including the second formant F2 of the voice in
the input signal xn (t). The third filter processing for outputting the third filter signal y3n (t) is
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executed (step ST4).
[0044]
The order of the first to third filter processes is not limited to the above order, and may be
random.
For example, the first to third filter processes (steps ST2, ST3 and ST4) may be performed
simultaneously and in parallel, or the second and third filters may be performed before the first
filter process (step ST2). A process (step ST3 or ST4) may be performed.
[0045]
The first mixing unit 31 receives the first filter signal y1n (t) output from the first filter 21 and
the second filter signal y2n (t) output from the second filter 22, and A first mixing process of
mixing the first filter signal y1n (t) with the second filter 22 and outputting a first mixed signal
s1n (t) is executed (step ST5A).
If the sample number t is less than or equal to the value T (YES in step ST5B), the process of step
ST5A is repeated until the sample number t becomes T = 160.
[0046]
The second mixing unit 32 receives the first filter signal y1n (t) output from the first filter 21 and
the third filter signal y3n (t) output from the third filter 23, and A process of mixing the first
filter signal y1n (t) and the third filter signal y3n (t) and outputting a second mixed signal s2n (t)
is executed (step ST6A).
If the sample number t is less than or equal to the value T (YES in step ST6B), the process of step
ST6A is repeated until the sample number t becomes T = 160.
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[0047]
The order of the first and second mixing processes is not limited to the above example, and may
be in any order. For example, the first and second mixing processes (steps ST5A and ST6A) may
be executed simultaneously and in parallel, or the second mixing process (steps ST) before the
execution of the first mixing process (steps ST5A and ST5B) ST6A and ST6B) may be performed.
[0048]
The first delay control unit 41 controls the first delay amount D1 of the first mixed signal s1n (t)
output from the first mixing unit 31, that is, controls the time delay of the signal. Specifically, the
first delay control unit 41 executes a process of outputting the first audio signals s1 to 1n (t)
obtained by adding a time delay of D1 samples to the first mixed signal s1n (t). (Step ST7A). If the
sample number t is less than or equal to the value T (YES in step ST7B), the process of step ST7A
is repeated until the sample number t becomes T = 160.
[0049]
The second delay control unit 42 controls the second delay amount D2 of the second mixed
signal s2n (t) output from the second mixing unit 32, that is, controls the time delay of the signal.
Specifically, the second delay control unit 42 executes a process of outputting the second audio
signals s2n (t) obtained by adding a time delay of D2 samples to the second mixed signal s2n (t).
(Step ST8A). If the sample number t is less than or equal to the value T (YES in step ST8B), the
process of step ST8A is repeated until the sample number t becomes T = 160.
[0050]
The order of the two delay control processes described above may be random. For example, steps
ST7A and ST8A may be performed simultaneously and in parallel, or steps ST8A and ST8B may
be performed before steps ST7A and ST7B.
[0051]
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After the processing of steps ST7A and ST8A, when the speech enhancement processing is
continued (YES in step ST9), the processing returns to step ST1A. On the other hand, when the
speech enhancement process is not continued (NO in step ST9), the speech enhancement process
ends.
[0052]
<< 1-3 >> Hardware Configuration The hardware configuration of the voice emphasizing device
100 is, for example, a computer with a built-in CPU (Central Processing Unit) such as a
workstation, mainframe, personal computer, or microcomputer for use in device integration. Is
possible. Alternatively, the hardware configuration of the speech enhancement apparatus 100
may be realized by an LSI (Large Scale Integrated circuit) such as a digital signal processor (DSP),
an application specific integrated circuit (ASIC), or a field-programmable gate array (FPGA). Good.
[0053]
FIG. 5 is a block diagram schematically showing the hardware configuration (in the case of using
an integrated circuit) of the speech enhancement apparatus 100 according to the first
embodiment. FIG. 5 shows an example of the hardware configuration of the speech enhancement
apparatus 100 configured using an LSI such as a DSP, an ASIC, or an FPGA. In the example of
FIG. 5, the speech enhancement apparatus 100 is composed of an acoustic transducer 101, a
signal input / output unit 112, a signal processing circuit 111, a recording medium 114 for
storing information, and a signal path 115 such as a bus. The signal input / output unit 112 is an
interface circuit that realizes a connection function with the acoustic transducer 101 and the
external device 102. As the acoustic transducer 101, for example, a device such as a microphone
or a sonic vibration sensor that captures and converts acoustic vibration into an electrical signal
can be used.
[0054]
The signal input unit 11, the first filter 21, the second filter 22, the third filter 23, the first mixing
unit 31, the second mixing unit 32, and the first delay control unit 41 shown in FIG. The
respective functions of the second delay control unit 42 can be realized by the signal processing
circuit 111 and the recording medium 114.
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[0055]
The recording medium 114 is used to store various data such as various setting data of the signal
processing circuit 111 and signal data.
As the recording medium 114, for example, volatile memory such as SDRAM (Synchronous
DRAM) or non-volatile memory such as HDD (Hard Disk Drive) or SSD (Solid State Drive) can be
used. The initial state and various setting data can be stored.
[0056]
The first and second audio signals s ~1 n (t) and s ~2 n (t) that have been subjected to
enhancement processing by the voice enhancement device 100 are sent to the external device
102 through the signal input / output unit 112. The external device 102 is, for example, a voice
and sound processing device provided in a television receiver or a hands-free calling device. The
audio / sound processing device is a device provided with a signal amplification device such as a
power amplifier and an audio output unit such as a speaker.
[0057]
FIG. 6 is a block diagram schematically showing the hardware configuration (in the case of using
a program executed by a computer) of the speech enhancement apparatus 100 according to the
first embodiment. FIG. 6 shows an example of the hardware configuration of the speech
enhancement apparatus 100 configured using an arithmetic device such as a computer. In the
example of FIG. 6, the speech enhancement apparatus 100 is configured by a signal input /
output unit 122, a processor 120 incorporating a CPU 121, a memory 123, a recording medium
124, and a signal path 125 such as a bus. The signal input / output unit 122 is an interface
circuit that realizes a connection function with the acoustic transducer 101 and the external
device 102. The memory 123 is a program memory for storing various programs for realizing
the speech enhancement process of the first embodiment, a work memory used when the
processor performs data processing, and a ROM used as a memory for expanding signal data, etc.
It is storage means such as (Read Only Memory) and RAM (Random Access Memory).
11-04-2019
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[0058]
The signal input unit 11, the first filter 21, the second filter 22, the third filter 23, the first mixing
unit 31, the second mixing unit 32, and the first delay control unit 41 shown in FIG. The
respective functions of the second delay control unit 42 can be realized by the processor 120
and the recording medium 124.
[0059]
The recording medium 124 is used to store various data such as various setting data of the
processor 120 and signal data.
As the recording medium 124, for example, volatile memory such as SDRAM, HDD or SSD can be
used. A program including an operating system (OS), various setting data, various data such as
acoustic signal data such as an internal state of a filter can be stored. The data in the memory
123 can also be stored in the recording medium 124.
[0060]
The processor 120 uses the RAM in the memory 123 as a working memory, and operates in
accordance with the computer program (the audio processing program according to the first
embodiment) read from the ROM in the memory 123, as shown in FIG. Signal input unit 11, first
filter 21, second filter 22, third filter 23, first mixing unit 31, second mixing unit 32, first delay
control unit 41, and second Signal processing similar to that of the delay control unit 42 can be
performed.
[0061]
The first and second audio signals s to 1n (t) and s to 2n (t) subjected to the above-described
speech enhancement processing are sent to the external device 102 through the signal input /
output unit 112 or 122.
As the external device, for example, various audio signal processing devices such as a hearing aid
device, an audio storage device, and a hands-free communication device correspond. Also, the
first and second audio signals s ~ 1n (t) and s ~ 2n (t) subjected to the speech enhancement
processing are recorded, and the recorded first and second audio signals s ~ 1n (t) are recorded. ,
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S ~2 n (t) can be output by another audio output device. The speech enhancement apparatus 100
according to the first embodiment can also be realized by executing it as a software program
along with the other apparatus.
[0062]
The speech processing program for executing the speech enhancement apparatus 100 according
to the first embodiment may be stored in a storage device inside a computer that executes a
software program, or may be a CD-ROM (optical information recording medium) or the like. It
may be distributed in a storage medium. It is also possible to acquire a program from another
computer through a wireless and wired network such as a LAN (Local Area Network).
Furthermore, with regard to the acoustic transducer 101 and the external device 102 connected
to the speech enhancement apparatus 100 according to the first embodiment, various data may
be transmitted and received through a wireless and wired network.
[0063]
<1-5> Effects As described above, according to the speech enhancement apparatus 100, the
speech enhancement method, and the speech processing program according to the first
embodiment, the fundamental frequency F0 of speech is presented to both ears while both Since
the ear separation and hearing can be performed, it is possible to generate the first and second
sound signals s to 1 n (t) and s to 2 n (t) that output a clear and easy-to-hear loud voice.
[0064]
Further, according to the voice emphasizing device 100, the voice emphasizing method, and the
voice processing program according to the first embodiment, the first filter signal and the second
filter signal are mixed at an appropriate ratio to produce the first mixed signal. And mixing the
first filter signal and the third filter signal at an appropriate ratio to form a second mixed signal,
and generating a first audio signal s ~1 n (t) based on the first mixed signal; Audio can be output
from the left speaker and the right speaker by the second audio signals s to 2n (t) based on the
two mixed signals.
For this reason, it is possible to eliminate the occurrence of a sense that the voice is biased to one
side or the aural balance of the left and right is disturbed and a sense of discomfort is generated,
and a high quality voice that is clear and easy to hear can be provided.
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[0065]
Further, according to the speech enhancement apparatus 100, the speech enhancement method,
and the speech processing program according to the first embodiment, the first and second
speech signals s to 1n (t) and s to 2n (t) are selected. The second delay amounts D1 and D2 can
be controlled to equalize the arrival times of the sounds output from the plurality of speakers to
the user's ear, so that the voice may be heard biased to one side or the voice may be heard twice.
It is possible to eliminate the occurrence of a sense of incongruity resulting from the collapse of
the aural balance on the left and the right, etc., and to provide a high quality voice that is clear
and easy to hear.
[0066]
Furthermore, there is little discomfort when used by not only normal hearing loss people but also
mild hearing loss people and healthy people, and even when applied to a loudspeaker that uses
speakers etc., the binaural hearing loss effect is reduced It is possible to realize a binaural
hearing aid method that does not do so, and it is possible to provide a high quality speech
enhancement apparatus 100.
[0067]
<< 2 >> Second Embodiment
FIG. 7 is a diagram showing a schematic configuration of a voice emphasizing device 200 (when
applied to a car navigation system) according to Embodiment 2 of the present invention.
In FIG. 7, components that are the same as or correspond to components shown in FIG. 1 are
given the same reference symbols as the reference symbols shown in FIG. 1. The speech
enhancement apparatus 200 is an apparatus capable of implementing the speech enhancement
method according to the second embodiment and the speech processing program according to
the second embodiment. As shown in FIG. 7, the voice emphasizing device 200 according to the
second embodiment includes a car navigation system 600 providing an input signal to the signal
input unit 11 via the input terminal 10, and a left speaker 61. And the right speaker 62, it differs
from the speech enhancement apparatus 100 according to the first embodiment.
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[0068]
The voice emphasizing device 200 according to the second embodiment processes the voice of a
car navigation system having an in-vehicle hands-free calling function and a voice guiding
function. As shown in FIG. 7, the car navigation system 600 has a telephone set 601 and a voice
guiding device 602 for providing a voice message to the driver. The other configuration is the
same as that of the first embodiment.
[0069]
The telephone set 601 is, for example, a device incorporated in the car navigation system 600 or
an external device connected by wire or wirelessly. The voice guide device 602 is, for example, a
device incorporated in the car navigation system 600. The car navigation system 600 outputs the
received voice output from the telephone set 601 or the voice guiding device 602 to the input
terminal 10.
[0070]
Further, the voice guiding device 602 outputs guide voice such as map guidance information to
the input terminal 10. The first audio signals s to 1n (t) output from the first delay control unit
41 are supplied to the L (left) speaker 61 through the first output terminal 51, and the L speaker
61 Output a sound based on the audio signals s ~1n (t). The second audio signals s to 2n (t)
output from the second delay control unit 42 are supplied to the R (right) speaker 62 through
the second output terminal 52, and the R speaker 62 Output a sound based on the audio signals
s ~2n (t) of
[0071]
In FIG. 7, for example, the user (driver) is sitting in the driver's seat of a left-hand drive car, and
the shortest distance between the left ear of the user sitting in the driver's seat and the L speaker
61 is approximately 100 cm. When the shortest distance to the R speaker 62 is about 134 cm,
the difference in distance between the L speaker 61 and the R speaker 62 is about 34 cm. Since
the sound velocity at normal temperature is about 340 m / sec, by delaying the output of the
sound from L speaker 61 by 1 msec, the sound output from L speaker 61 and R speaker 62, that
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is, the received voice or the guide voice of the telephone The time to reach the left ear and the
time to reach the right ear can be matched. Specifically, the first delay amount D1 of the first
audio signal s ~ 1 n (t) provided from the first delay control unit 41 is 1 msec, and the second
delay control unit 42 The second delay amount D2 of the two audio signals s to 2n (t) may be set
to 0 msec (no delay). Note that the values of the first delay amount D1 and the second delay
amount D2 are not limited to the above-described example, and are appropriately determined
according to the usage conditions such as the positions of the L speaker 61 and the R speaker 62
with respect to the user's ear position. It can be changed. Specifically, it can be appropriately
changed according to the use situation such as the distance from the speaker 61 to the left ear
and the distance from the R speaker 62 to the right ear.
[0072]
As described above, according to the speech enhancement apparatus 200, the speech
enhancement method, and the speech processing program according to the second embodiment,
the first and second speech signals s ~1 n (t), s ~2 n (t Of the first and second delay amounts D1
and D2) to adjust the arrival times of the sounds output from the plurality of speakers to the
user's ear, so that the voice may be heard biased to one side or the voice It is possible to
eliminate the occurrence of a sense of incongruity resulting from the disruption of the aural
balance between the left and right, such as double hearing, and to provide a high quality voice
that is clear and easy to hear.
[0073]
In addition, it is possible to realize a binaural hearing aid method that is less incongruent when
used by not only ordinary hearing impaired persons but also mild hearing impaired people and
healthy persons, and the binaural hearing aid effect is not reduced. It is possible to provide a
quality speech enhancement apparatus 200.
The second embodiment is the same as the first embodiment in the points other than the above.
[0074]
<< 3 >> Third Embodiment FIG. 8 is a diagram showing a schematic configuration of a speech
enhancement apparatus 300 (when applied to a television receiver) according to Embodiment 3
of the present invention. In FIG. 8, components that are the same as or correspond to
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components shown in FIG. 1 are given the same reference symbols as the reference symbols
shown in FIG. 1. The speech enhancement apparatus 300 is an apparatus capable of
implementing the speech enhancement method according to the third embodiment and the
speech processing program according to the third embodiment. As shown in FIG. 8, the speech
enhancement apparatus 300 according to the third embodiment includes a television receiver
701 and a pseudo monauralization section 702 that provide an input signal to the signal input
unit 11 via the input terminal 10. And the left (L) channel signal of stereo sound of the television
receiver 701 is supplied to the L speaker 61, and the R (right) channel signal of stereo sound is
transmitted to the R speaker 62. This point is different from the speech enhancement device 100
according to the first embodiment in that it is supplied.
[0075]
The television receiver 701 uses, for example, video content recorded by an external video
recorder that receives broadcast waves or a video recorder built in the television receiver, and is
configured from an L channel signal and an R channel signal. Output stereo signal. The sound of
a television is generally not limited to a stereo signal of two channels, but may be a multi stereo
signal of three or more channels, but here, in order to simplify the description, the case of a
stereo signal of two channels will be described.
[0076]
The pseudo monauralization unit 702 receives the stereo signal output from the television
receiver 701, and adds the antiphase signal of the (L−R) signal to the (L + R) signal, for example,
according to a known method. Extract only the voice of the announcer localized at the center.
Here, the (L + R) signal is a pseudo monaural signal obtained by adding the L channel signal and
the R channel signal, and the (L−R) signal is a signal obtained by subtracting the R channel
signal from the L channel signal. It is a pseudo monaural signal in which the signal to be localized
is attenuated.
[0077]
The voice of the announcer extracted by the pseudo monaural unit 702 is input to the input
terminal 10, the same processing as described in the first embodiment is performed, and the L
channel signal and the R channel signal output from the television receiver 701 are added
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respectively. After that, the sound subjected to the binaural separation and hearing processing is
output from the L speaker 61 and the R speaker 62. With such a configuration, it is possible to
emphasize only the voice of the announcer localized at the center of the stereo signal while
maintaining the conventional stereo sound.
[0078]
In the third embodiment, two channels of stereo signals are illustrated for simplification of the
description, but the method of the third embodiment is also applied to, for example, three or
more channels of multi stereo signals such as 5.1 channel stereo. It is possible and achieves the
same effect as described in the third embodiment.
[0079]
In the third embodiment, the L speaker 61 and the R speaker 62 are described as external
devices of the television receiver 701. However, for example, a speaker built in the television
receiver or an audio device such as headphones may be used.
Also, although the processing before inputting the pseudo monaural unit 702 to the input
terminal 10 is described, the stereo signal output from the television receiver 701 is input to the
input terminal 10, and then the pseudo monaural processing is performed. May be
[0080]
As described above, according to the voice emphasizing device 300, the voice emphasizing
method, and the voice processing program according to the third embodiment, even if the stereo
signal is a binaural signal, the voice of the announcer localized in the center is emphasized. A
separate hearing aid method can be realized.
[0081]
In addition, it is possible to realize a binaural hearing aid method that is less incongruent when
used by not only ordinary hearing impaired persons but also mild hearing impaired people and
healthy persons, and the binaural hearing aid effect is not reduced. It is possible to provide a
quality speech enhancement apparatus 300.
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The third embodiment is the same as the first embodiment in the points other than the above.
[0082]
<< 4 >> Fourth Embodiment In the first to third embodiments, the case where the first audio
signals s to 1n (t) and the second audio signals s to 2n (t) are directly output to the L speaker 61
and the R speaker 62 is described. did. On the other hand, the speech enhancement apparatus
400 according to the fourth embodiment performs crosstalk cancellation processing on the first
audio signals s to 1n (t) and the second audio signals s to 2n (t). It has 70.
[0083]
FIG. 9 is a functional block diagram showing a schematic configuration of the speech
enhancement apparatus 400 according to the fourth embodiment. In FIG. 9, the same or
corresponding components as those shown in FIG. 1 are denoted by the same reference numerals
as those shown in FIG. The speech enhancement apparatus 400 is an apparatus capable of
implementing the speech enhancement method according to the fourth embodiment and the
speech processing program according to the fourth embodiment. As shown in FIG. 9, the speech
enhancement apparatus 400 according to the fourth embodiment differs from the speech
enhancement apparatus 100 according to the first embodiment in that two crosstalk cancelers
(CTC) 70 are provided. . The other configuration is the same as that of the first embodiment.
[0084]
For example, the first audio signal s ~ 1n (t) is an L channel audio (audio to be presented only to
the left ear) signal, and the second audio signal s ~ 2n (t) is an R channel audio (only for the right
ear) Consider the case where it is an audio signal to be presented. The L channel sound is a
sound that is desired to reach only the left ear, but in fact, the crosstalk component of the L
channel sound also reaches the right ear. Also, R channel sound is sound that is desired to reach
only the right ear, but in fact, the crosstalk component of R channel sound also reaches the left
ear. Therefore, the crosstalk canceller 70 subtracts the signal corresponding to the crosstalk
component of L channel audio from the first audio signals s to 1 n (t), and generates the second
signal corresponding to the crosstalk component of R channel audio. The crosstalk component is
canceled by subtracting from the audio signals s ~2n (t). Crosstalk cancellation processing for
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canceling the crosstalk component is a known method such as an adaptive filter.
[0085]
As described above, according to the voice emphasizing device 400, the voice emphasizing
method, and the voice processing program according to the fourth embodiment, the process of
canceling the crosstalk component of the signal output from the first and second output
terminals Because, you can enhance the mutual separation effect of the two sounds that reach
both ears. Therefore, when applied to a loudspeaker system, the binaural hearing aid effect can
be further enhanced, and it is possible to provide a high quality speech enhancement apparatus
400.
[0086]
<< 5 >> Fifth Embodiment Although the case of performing binaural separate hearing aid
processing regardless of the mode of the input signal has been described in the fourth
embodiment, in the fifth embodiment, the input signal is analyzed and both ears of the content
according to the result of this analysis are analyzed. The case of performing separated hearing
aid processing will be described. The speech enhancement apparatus according to the fifth
embodiment performs binaural separation and hearing processing when the input signal is a
vowel.
[0087]
FIG. 10 is a functional block diagram showing a schematic configuration of the speech
enhancement apparatus 500 according to the fifth embodiment. In FIG. 10, components that are
the same as or correspond to components shown in FIG. 9 are assigned the same reference
symbols as the reference symbols shown in FIG. The speech enhancement apparatus 500 is an
apparatus capable of implementing the speech enhancement method according to the fifth
embodiment and the speech processing program according to the fifth embodiment. The speech
enhancement apparatus 500 according to the fifth embodiment is different from the speech
enhancement apparatus 400 according to the fourth embodiment in that a signal analysis unit
80 is provided.
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[0088]
For the input signal xn (t) output from the signal input unit 11, for example, the signal analysis
unit 80 determines whether the input signal is a signal indicating vowel or a vowel by a known
analysis method such as autocorrelation coefficient analysis. An analysis is made as to whether
the signal indicates a non-consonant sound (consonant sound or noise). As a result of analysis of
the input signal, when the input signal is a signal indicating a consonant or noise, the signal
analysis unit 80 stops the outputs of the first mixing unit 31 and the second mixing unit 32 (that
is, filtering is performed The output of the signal performed is stopped), and the input signal xn
(t) is directly input to the first delay control unit 41 and the second delay control unit. The fifth
embodiment is the same as the fourth embodiment regarding the configuration and operation
other than the above.
[0089]
FIG. 11 is a flowchart showing an example of speech enhancement processing (speech
enhancement method) performed by the speech enhancement apparatus 500 according to the
fifth embodiment. In FIG. 11, the same processing steps as in FIG. 4 are assigned the same step
numbers as the step numbers shown in FIG. The voice emphasizing process performed by the
voice emphasizing device 500 according to the fifth embodiment includes the step of
determining whether or not the input signal is a voice signal of a vowel, and the case where the
input signal is not a voice signal of a vowel. Is different from the processing of the first
embodiment in that the processing is advanced to step ST7A. Except for this point, the process in
the fifth embodiment is the same as the process in the first embodiment.
[0090]
As described above, according to the voice emphasizing device 500, the voice emphasizing
method, and the voice processing program according to the fifth embodiment, the binaural
separation and hearing aid processing can be performed according to the mode of the input
signal. It is possible to provide a high quality speech enhancement apparatus 500 without
unnecessarily emphasizing unnecessary consonants and noises.
[0091]
<< 6 >> Modifications In the first to fifth embodiments, the first filter 21, the second filter 22, and
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the third filter 23 execute filtering on the time axis.
However, each of the first filter 21, the second filter 22, and the third filter 23 is an FFT unit (fast
Fourier transform unit), a filter processing unit that performs filter processing on the frequency
axis, and an IFFT unit ( It is also possible to configure with an inverse fast Fourier transform
unit). In this case, each of the filter processing unit of the first filter 21, the filter processing unit
of the second filter 22, and the filter processing unit of the third filter 23 sets the gain of the
spectrum of the passband to 1 and attenuates. This can be realized by setting the gain of the
spectrum of the band to be set to zero.
[0092]
Although the case where the sampling frequency is 16 kHz has been described in the first to fifth
embodiments, the sampling frequency is not limited to this value. For example, it is also possible
to set the sampling frequency to another frequency such as 8 kHz or 48 kHz.
[0093]
In the above-described second and third embodiments, the speech enhancement apparatus has
been described as applied to a car navigation system and a television receiver. However, the
voice emphasizing device according to the first to fifth embodiments is a system or device other
than a car navigation system and a television receiver, and is applicable to a system or device
including a plurality of speakers. The voice emphasizing device according to the first to fifth
embodiments can be applied to, for example, a voice guidance system in an exhibition hall or the
like, a video conference system, a voice guidance system in a train, and the like.
[0094]
In the first to fifth embodiments, various modifications of the constituent elements, addition and
omission of the constituent elements are possible within the scope of the present invention.
[0095]
The speech enhancement apparatus, speech enhancement method, and speech processing
program according to the first to fifth embodiments are applicable to speech communication
systems, speech storage systems, and speech amplification systems.
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[0096]
When applied to a voice communication system, the voice communication system includes, in
addition to the voice emphasizing device according to any one of the first to fifth embodiments, a
signal output from the voice emphasizing device is transmitted and input to the voice
emphasizing device. A communication device for receiving the signal.
[0097]
When applied to a voice storage system, the voice storage system includes a storage device for
storing information in addition to the voice enhancement device according to any of the first to
fifth embodiments, and a first output from the voice enhancement device. And a writing device
for storing the second audio signal s ~ 1n (t) and s ~ 2n (t) in the storage device, and the first and
second audio signals s ~ 1n (t) and s ~ 2n from the storage device And (t) read out and input to
the speech enhancement apparatus.
[0098]
When applied to a voice amplification system, the voice amplification system includes an
amplification circuit for amplifying a signal output from the voice enhancement device, in
addition to the voice enhancement device according to any of the first to fifth embodiments. And
a plurality of speakers for outputting a sound based on the first and second audio signals s ~1 n
(t) and s ~2 n (t).
[0099]
In addition, the voice emphasis device, the voice emphasis method, and the voice processing
program according to the first to fifth embodiments can be applied to a car navigation system, a
mobile phone, an interphone, a television receiver, a handsfree phone system, and a video
conference system. .
When applied to these systems or devices, from the audio signals output from these systems or
devices, a first audio signal s ~1 n (t) for one ear and a second for the other ear Audio signals s to
2n (t) are generated.
The user of the system or apparatus to which the first to fifth embodiments are applied can
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perceive clear speech.
[0100]
DESCRIPTION OF SYMBOLS 10 input terminal, 11 signal input part, 21 1st filter, 22 2nd filter,
23 3rd filter, 31 1st mixing part, 32 2nd mixing part, 41 1st delay control part, 42 Second delay
control unit, 51 first output terminal, 52 second output terminal, 61 L speaker, 62 R speaker,
100, 200, 300, 400, 500 voice emphasis device, 101 acoustic transducer, 111 signal processing
circuit , 112 signal input / output unit, 114 recording medium, 115 signal path, 120 processor,
121 CPU, 122 signal input / output unit, 123 memory, 124 recording medium, 125 signal path,
600 car navigation system, 601 telephone, 602 voice guide device , 701 TV receiver, 702 pseudo
monaural part.
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