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JPH08116585

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DESCRIPTION JPH08116585
[0001]
BACKGROUND OF THE INVENTION 1. Field of the Invention The present invention relates to the
improvement of an apparatus for improving the sound quality of band-limited original signals
such as AM broadcast signals, FM or AM stereo broadcast signals.
[0002]
2. Description of the Related Art Some music sources of original signals have a frequency band
secured up to 20 kHz like CDs, and some are band-limited to 9 kHz or less like AM radio
broadcast signals. . For this reason, in AM radio broadcasting, an audio signal component of 9
kHz or more could not be reproduced from the received signal. As a countermeasure, a method
has been proposed in which a part of the AM radio broadcast reception signal is extracted by a
band pass filter, a signal component of double frequency is generated by a multiplier etc, and this
is added out of the band of the original signal. ing.
[0003]
However, in this case, since only the frequency of the original signal is doubled and added using
a pitch converter, the odd harmonics component to the original signal is created. It is not
possible, and the audio signal component one octave higher than the original signal will be
reproduced rather unnaturally.
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[0004]
An object of the present invention is to solve the above problems by improving the sound quality
by adding a high frequency band component for sound quality improvement to the band-limited
original signal outside the band.
[0005]
In order to achieve the above object, according to a first aspect of the present invention, there is
provided a sound quality improvement apparatus comprising: signal extraction means for
extracting a signal component of a predetermined frequency band from an original signal; Signal
band shifting means for shifting components to a high frequency band, Fourier transform means
for converting the band shifted signal components into frequency data signals, and estimating
the harmonic structure of the spectrum of the original signal based on the frequency data signals
And interpolation processing means for generating an interpolation signal obtained by
interpolating between spectra, inverse Fourier transformation means for converting the
interpolation signal into a time axis waveform signal, and addition means for adding the time axis
waveform signal to the original signal. It is characterized by
[0006]
The second invention is characterized in that in the first invention, the original signal is a
received signal of AM radio broadcast.
[0007]
Furthermore, the sound quality improvement apparatus according to the third aspect of the
present invention is an audio attenuation means for attenuating an audio signal component from
an original signal, a signal extraction means for extracting a signal component of a
predetermined frequency band from a signal having the audio signal component attenuated. And
a signal band shifting means for shifting the extracted signal component to a high frequency
band, and an addition means for adding the band shifted signal component to the original signal.
[0008]
Also, according to a fourth aspect of the present invention, in the third aspect, the original signal
is a stereo signal, and one of the left and right signal components of the stereo signal is inverted
and added to the audio attenuation means. It is characterized in that it is configured as follows.
[0009]
According to a fifth aspect of the present invention, in the third or fourth aspect, the signal band
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shifting means includes a mixer and a local oscillator.
[0010]
In the first and second inventions, a part of the signal component of the band-limited signal such
as an AM radio broadcast signal is extracted as the original signal, and the signal component is
shifted to the high frequency band. For example, the pitch is converted to one octave higher.
The pitch-converted signal component is converted to a frequency component data signal by
performing FFT (Fast Fourier Transform), and after estimating the harmonic structure of the
spectrum of the original signal based on this data signal, to interpolate between the spectra The
interpolation signal is generated, and the interpolation signal is converted to a time axis
waveform signal by inverse FFT and then added to the original signal.
[0011]
In the third, fourth and fifth inventions, after the audio signal component of a band-limited signal
such as an FM stereo broadcast signal is attenuated as an original signal, a part of the signal
components is extracted, The signal component is shifted to a high frequency band, for example,
the pitch is converted to one octave higher and added to the original signal.
[0012]
Embodiments of the present invention will be described below with reference to the drawings.
1 (a) shows an embodiment of the sound quality improvement apparatus according to the
present invention, wherein 1 is an A / D converter, 2 is a band pass filter (BPF), 3 is a pitch
converter, 4 is an FFT processing unit, and 5 is interpolation A processing unit, 6 is an inverse
FFT processing unit, 7 is an adder, and 8 is a D / A converter.
[0013]
The sound quality improvement apparatus 9 described above is incorporated in, for example, an
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AM radio receiver 10 as shown in FIG. 1B, and a reception signal (detection signal) of a bandlimited AM radio broadcast is input. .
The instrument sound is composed of a fundamental frequency and harmonics of integral
multiples thereof.
However, in the AM radio broadcast, the band of the AM broadcast signal is limited to 9 kHz or
less on the broadcast station side, so an audio signal of 9 kHz or more can not be reproduced
from the received signal.
[0014]
Therefore, the detection signal of the received AM broadcast signal is converted into a digital
signal using the A / D converter 1.
Of this digital signal, the signal component of 4 to 8 kHz is extracted by the band pass filter 2
(FIG. 2 (a)), and the frequency of the signal component is shifted to the high frequency band of 8
to 16 kHz using the pitch converter 3 Pitch conversion is performed (FIG. 2 (b)).
Then, the signal component which has been pitch-converted by the FFT processing unit 4 is
converted into a frequency data signal (FIG. 2 (c)), and the power processing is taken by the
interpolation processing unit 5 based on this data signal, the spectrum of the original signal After
estimation of the harmonic structure, the signal of the frequency to be interpolated between the
spectra is generated by calculation.
This interpolation signal rewrites the real part data of the FFT so as to be the average of adjacent
spectra using equation (1) (FIG. 2 (d)).
The interpolation signal obtained by the interpolation processing in this manner is converted into
a time axis waveform signal by the inverse FFT processing unit 6 (FIG. 2 (e)). In addition, the time
axis is adjusted by repeatedly using this signal (FIG. 2 (f)). The time axis waveform signal
subjected to the interpolation processing is added to the AM detection signal as an original signal
by the adder 7, converted again into an analog signal by the D / A converter 8, and added to the
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speaker 12 via the amplifier 11. By this method, sounds of 0 to 16 kHz can be artificially
reproduced.
[0015]
FIG. 3 shows how an actual musical instrument sound is processed. The frequency characteristics
when the signal component extracted by BPF2 (FIG. 3A) is pitch-converted are shown in FIG. 3B,
and the frequency characteristics when the interpolation process is performed after the pitch
conversion are shown in FIG. . Furthermore, FIG. 4 (b) shows the frequency characteristics when
the pitch-converted signal is added to the original signal (FIG. 4 (a)), and FIG. 4 (b) shows the
frequency characteristics when the interpolated signal is added to the original signal. c).
[0016]
As described above, sounds of 9 kHz or more can be artificially reproduced by this sound quality
improvement apparatus. The frequency of the signal component extracted by the band pass filter
2 may be, for example, 3 to 6 kHz, and the signal within 9 kHz may be emphasized. R (k) = √
{(S1 (k) + S2 (k)) / 2} 2-{(I1 (k) + I2 (k)) / 2} 2 (1) where S1 (k) , S2 (k): power spectra I1 (k), I2 (k)
of spectra before and after interpolation: imaginary part RFT (k) of FFT of spectra before and
after interpolation: real part of FFT to interpolate
[0017]
FIG. 5 (a) shows another embodiment of the sound quality improvement apparatus according to
the present invention, wherein 21 is a vocal canceller, 22a and 22b are band pass filters (BPF),
23a and 23b are pitch converters, and 24a and 24b are It is an adder.
[0018]
For example, as shown in FIG. 5 (b), the sound quality improvement apparatus 25 described
above is incorporated in the FM stereo receiver 26, and a band-limited FM stereo broadcast
reception signal (Lch, Rch) is input.
The instrument sound is composed of a fundamental frequency and harmonics of integral
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multiples thereof. However, in the FM radio broadcast, the band of the FM broadcast signal is
limited to 15 kHz or less on the broadcast station side, so the sound of 15 kHz or more can not
be reproduced from the received signal.
[0019]
Therefore, in the above embodiment, the voice component is attenuated or eliminated through
the vocal canceller 1 for the left and right demodulated signals (Lch, Rch) of the received FM
broadcast signal. In a stereo broadcast signal, voice can be removed by adding the Rch and Lch
signals. The 7 to 14 kHz signal components from which the vocal component has been
attenuated or removed are extracted by the band pass filters 2a and 2b, and the respective signal
components are pitched so that the frequency is shifted to 14 to 28 kHz using the pitch
converters 3a and 3b. Convert. The pitch-converted signal is added to the original signal by the
adders 4a and 4b. By this method, an audio signal component of 0 to 28 kHz can be artificially
reproduced. In addition, after the processing procedure of the band pass filters 2a and 2b is
performed, the vocal canceller 1 may be operated. As described above, the sound quality
improvement device can reproduce sounds of 15 kHz or more in a pseudo manner.
[0020]
FIG. 6 shows an example of the configuration of the vocal canceller 21, wherein R1 to R5 are
resistors, AMP1 is an inverting amplifier, AMP2 is an operational amplifier, and one of the left
and right demodulated signals (L, R) is inverted by the inverting amplifier AMP1, By adding to
the other, it is possible to attenuate the vocal signal component localized at substantially the
center of the L and R signals.
[0021]
FIG. 7 shows an example applied to an AM stereo receiver as still another embodiment of the
present invention.
In the figure, 31 represents an RF amplifier, 32 represents a mixer, 33 represents a local
oscillator, 34 represents a band pass filter (BPF), 35 represents an IF amplification, 36 represents
a detector, and the matrix circuit 17 comprises circuits 31 to 37. Is a configuration of a
conventional AM stereo receiver, and L and R signals are given as matrix circuit outputs. Further,
41 is a subtractor, 42 is a BPF, 43 is a mixer, 44 is a local oscillator, 45 is a high pass filter, and
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46 and 47 are adders. Assuming that the high frequency band of the demodulation signal of the
AM stereo receiver is limited by f, the bands of the L signal and the R signal are both as shown in
FIG.
[0022]
The L and R signals are applied to a subtractor 41 as a vocal canceller. The output of the
subtractor 41 may be either an (L-R) signal or an (R-L) signal. The L signal is a signal output from
the left speaker, and the R signal is a signal output from the right speaker. Therefore, by taking
the difference between the L signal and the R signal, the same component contained in both
signals is removed. The same component is a component localized at the center of the left and
right speakers, that is, mainly a vocal (voice) component. Therefore, it is possible to obtain a
signal from which an audio component is removed as the subtractor output signal. The band of
the subtractor output signal is also as shown in FIG.
[0023]
Next, a part of the subtractor output signal is taken out by a band pass filter 42. Assuming that
the band of the band pass filter 42 is fBL to fBH, the band extracted by the band pass filter 42 is
as shown in FIG. The band pass filter output is applied to a mixer (MIX) 43. The mixer 43
multiplies the output signal of the local oscillator (OSC) 44. Since the sum signal and the
difference signal of the input signals are given as the output signal of the mixer 43, assuming
that the OSC output signal has a frequency f0, the MIX output signal appears above and below f0
as shown in FIG. Next, the low-pass signal component of the MIX output signal is attenuated by
the high pass filter 45 to obtain the signal shown in FIG. That is, as the output signal of the highpass filter 45, the audio signal component is removed from the L signal and the R signal, and a
part thereof is extracted to obtain a signal shifted by f0 on the high frequency side. Next, the high
pass filter output signal is added to the adder 46 and the adder 47. The adder 46 adds the high
pass filter output signal and the L signal. The adder 47 adds the high pass filter output and the R
signal. Bands of output signals of the adder 46 and the adder 47 are shown in FIG. Thus, a signal
in which the high frequency signal is added to the L signal at the L 'output terminal is obtained,
and a signal in which the band signal is added to the R signal at the R' output terminal is
obtained.
[0024]
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Although AM stereo radio is taken as the above embodiment, FM stereo radio can be used
similarly. Also, although the upper limit of human audio frequency is less than 20 kHz,
frequencies higher than 20 kHz also affect the mental state of people, so even high frequency
sources such as compact disc with higher frequencies above 20 kHz have higher frequency
bands. Applications such as adding high frequency signals are also conceivable. Thus, it can be
used in any stereo music source.
[0025]
As described above, according to the present invention, a clearer sound than the original sound
can be obtained by adding the interpolation signal constituting the harmonic structure of the
spectrum to the outside of the band of the original signal. It can be played back. In addition, since
the interpolation signal is added after the interpolation processing rather than adding the
harmonic component created simply by the pitch converter as in the prior art, there is no sense
of discomfort in the sound quality. Furthermore, since the pitch conversion signal is added after
the vocal component is attenuated, it is possible to reproduce an instrument sound or the like
more clearly than the original sound and in a natural form.
[0026]
Brief description of the drawings
[0027]
1 is a block diagram showing an embodiment of the present invention.
[0028]
2 is an operation explanatory diagram of the embodiment of FIG.
[0029]
3 is a frequency characteristic diagram showing how an actual instrument sound is processed in
the embodiment of FIG.
[0030]
4 is a frequency characteristic diagram showing how an actual musical instrument is processed
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in the embodiment of FIG.
[0031]
5 is a block diagram showing another embodiment of the present invention.
[0032]
6 is a diagram showing an example of the configuration of the vocal canceller.
[0033]
7 is a block diagram showing another embodiment of the present invention.
[0034]
8 is a diagram showing the L, R signal and the band of the subtractor output signal in the
embodiment of FIG.
[0035]
9 is a diagram showing the band of the BPF output signal in the embodiment of FIG.
[0036]
10 is a diagram showing the band of the MIX output signal in the embodiment of FIG.
[0037]
11 is a diagram showing the band of the HPF output signal in the embodiment of FIG.
[0038]
12 is a diagram showing the band of the output signal of the adder in the embodiment of FIG.
[0039]
Explanation of sign
[0040]
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DESCRIPTION OF SYMBOLS 1 A / D converter 2 BPF 3 Pitch converter 4 FFT process part 5
Interpolation process part 6 Inverse FFT process part 7 Adder 8 D / A converter 9 Sound quality
improvement apparatus 10 AM radio receiver 21 Vocal canceller 22a, 22b BPF 23a, 23b Pitch
converter 24a, 24b Adder
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