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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.
The present invention compensates for the absence of bass components, ie, generates harmonic
components based on bass components, for lack of bass response, due to poor loudspeakers of
bass response. The present invention relates to a bass component compensation method that
compensates for.
BACKGROUND OF THE INVENTION To date, with the increasing popularity of strong, deep bass
sounds, a great deal of effort has been made to increase the power of audio player bass outputs.
Conventionally, this problem is addressed in the design of the speaker and / or the design of the
amplification circuit. The frequency response of the speaker generally rolls off at 12 dB / octave
for frequencies below the resonance point. Thus, the bass response of the loudspeaker can be
improved by appropriately selecting a number of loudspeaker design parameters and materials
to make the resonant frequency as low as possible. Also, by using a linear bass boost circuit, it is
possible to amplify the bass signal and thereby drive the bass power from the amplifier to the
loudspeaker more strongly. At this time, since the roll-off below the resonance point occurs
rapidly, considerable amplification is required.
However, although the method as described above is effective for generating strong bass, it may
not be suitable. For example, due to space constraints, aesthetic reasons, cost, etc., it becomes
difficult to design a low resonant frequency speaker. Also, the inclusion of a linear bass boost
circuit alleviates this problem but is not sufficient and will sacrifice amplifier head room. That is,
there is a problem that clipping occurs in the output signal of the amplifier when overdriven.
The present invention takes into consideration such problems of the conventional bass
reproduction, and obtains a sound that makes the powerful bass feel without being limited by the
space, cost and the like of the speaker and without sacrificing the characteristics of the amplifier.
It is an object of the present invention to provide a bass component compensation method
capable of
The present invention according to claim 1 is characterized in that a low frequency band
component having a low sound pressure in predetermined speaker reproduction is extracted
from an input signal, and the sound by the speaker is extracted based on the extracted low
frequency band component. High frequency pressure harmonics, mixing the generated
harmonics with the input signal, and utilizing the psychoacoustic effects of human auditory on
complex tones to reproduce the bass component due to the reproduction of the speaker It is a
bass region compensation method characterized by compensating.
According to the present invention of claim 4, the bass component having a low sound pressure
in predetermined speaker reproduction is extracted from the input signal, and the level of the
extracted bass component is adjusted to a predetermined level determined in advance. Based on
the bass component, the sound pressure generated by the speaker generates a high frequency
harmonic, and the level of the generated harmonic is readjusted so as to correspond to the
original bass component, Bass component compensation characterized in that the bass
component due to the reproduction of the speaker is compensated by mixing the readjusted
harmonics with the input signal and exploiting the psychoacoustic effects of the human auditory
on complex tones It is a method.
The present invention is applied to any audio player, for example, to generate a harmonic that
can be heard well based on the attenuated bass component, and mixing the harmonic into the
original signal, The lack of bass can be compensated for by making the listener able to sense the
pitch of these lost or weak bass components in the reproduction of the loudspeaker.
Also, the invention can be applied, for example, to any audio playback device, and can be
economically implemented using analog circuitry or digital signal processors.
described below based on the drawings showing the embodiments.
FIG. 1 is a diagram showing the concept of the bass range compensation method of the present
Here, it is assumed that the frequency response of the loudspeaker has a lower resonant
frequency of f0.
As shown in FIG. 1, the loudspeaker response typically rolls off rapidly at 12 dB / octave for
frequencies below f 0.
If one piece of music contains bass components below the 6 dB cut-off frequency, these
components will be significantly attenuated (low sound pressure).
For example, the component of f1 shown in FIG. 1A is hardly reproduced by this speaker.
To solve this problem, a low pass filter is used to filter out all components above frequency f0
and this component is extracted.
Next, the extracted components are processed (details will be described later), and as shown in
FIG. 1 (b), the second, third, and the like 2f1, 3f1, and 4f1, respectively. Generate the fourth
harmonic. These generated harmonics are added to the original music signal. Since these
harmonics have a frequency exceeding f0, they can be reproduced by the speaker without much
attenuation (high sound pressure). If the listener detects this mixed tone, the clear pitch is not an
average of harmonic frequencies. Instead, it is the difference between successive harmonics, ie
the pitch of the tone, which is equal to f1. That is, the low-frequency component of the
fundamental sound is either lost or significantly attenuated (low sound pressure), but it is
perceived by the listener by these psycho-physiological effects of the human auditory system. Ru.
Therefore, with the psycho-psychological bass boost system (hereinafter referred to as PBBS),
with a pitch of f1, it is strong, without accepting that the components of f1 would otherwise be
inaudible or hardly audible. Low-range sound is created.
At this time, not only the power of each harmonic related to the original fundamental sound but
also the number of harmonics becomes an important point of design selection. Too much
harmonics will cause irritating distortion, while too little will defeat this method. The above
configuration should be determined by trial listening and will probably be different for each
speaker to get the best harmony. According to the rule of thumb, the fundamental power and its
generated harmonics are in descending order. In many cases, the fourth and higher harmonics
can be ignored as they have less effect on the quality of the sound. Thus, the amplitude of the
combined output signal should not be greater than the amplitude of the original signal.
Furthermore, there must be no significant degradation of the amplifier head room.
The harmonic components can be generated by applying mathematical operations to the
fundamental components. By squaring the fundamental sound component, a second harmonic
plus a constant term is obtained as shown in (Equation 1). Therefore, by squaring the
fundamental sound and then subtracting the constant term, the pure second harmonic shown in
equation (2) is obtained.
(2) cos 2θ = 2 cos 2 θ−1, where θ = ωtω = the angular frequency of the fundamental sound
signal, the third harmonic is obtained by subtracting the fitting amount of the fundamental sound
from the cube of the fundamental sound by You can get it. Furthermore, the fourth harmonic can
be obtained directly from the second harmonic or the fundamental sound by (Equation 4).
[Equation 3] cos 3 θ = 4 cos 3 θ 3 cos θ
Cos 4 θ = 2 cos 2 2 θ −1 = 8 cos 4 θ−8 cos 2 θ + 1 The present invention generates
harmonics using the mathematical operations described above.
2 and 3 are block diagrams of the PBBS in the first embodiment for implementing this method.
This device includes a DC remover 1, a bass region extractor 2, a normalizer 3, a second
harmonic generator 4, a third harmonic generator 5, and, if necessary, a higher harmonic
generator 6 etc. It consists of The functions of the respective units will be described below. a)
The DC remover 1 input signal S0 often includes a DC component due to a DC offset from the
front stage. In order to remove it, a DC remover 1 is required. Otherwise, the direct current
component will result in undesirable by-products during processing in the subsequent stages.
The DC remover 1 can only be realized by means of a high pass filter with a low 3 dB cut off
frequency so as not to affect the audio signal as 10 Hz. b) Bass Region Extractor 2 The bass
region extractor 2 is used to remove high frequencies from S1 and hold only the bass component
that is the basic sound for generating harmonics. Since the low-frequency component in question
is significantly attenuated by the speaker (low sound pressure), the low-band extractor 2 is
ideally equal to the cut-off frequency of the speaker whose 3 dB cut-off frequency is to be
compensated Low-pass filter. c) Normalizer 3 The normalizer 3 is used to set the amplitude of the
extraction signal S2 to 1 after the extraction of the bass component. The harmonic generation
process is non-linear, ie normalization is required as the amplitude of the generated harmonics is
not proportional to the amplitude of the extracted signal. For example, if the amplitude of the
input signal A cos θ is A = 0.5, then squaring this signal produces a second harmonic whose
amplitude is proportional to A 2 = 0.25. However, if A = 0.05, then A2 = 0.0025, which results in
a much smaller second harmonic compared to the fundamental sound. Normalization offers the
advantage of processing the input signal independent of its original amplitude.
The normalizer 3 comprises an amplitude extractor 8 for deriving the amplitude of the input
signal, and a division function 37 for obtaining a normalized signal by dividing the input signal
by the derived amplitude. In one example of implementation of the amplitude extractor 8, two
input signals S2 are generated using a 90 ° phase difference network consisting of two specially
designed all-pass filters 31, 32. A quadrature signal is generated, namely the Cos output S32 and
the Sin output S31. The Cos output S32 draws out the Sin output S31 with a phase difference of
90 °. Another example of generating two quadrature components utilizes the Hilbert transform.
By squaring these quadrature signals and then summing up, the square of the amplitude S35, ie
equation (5), is obtained.
Next, by applying the square root function 36, the amplitude S36 is derived. In one of the
examples of calculating the square root, approximation by equation (6) is used. The calculated
error is less than 1% if the range is 0.25 <x <1.
If x x = 1.454895 x-1.34491 x 2 + 1. 106 812 x 3-0.536 499 x 4 + 0.1 12 12 16 x 5 + 0.2 0 75
806 x is outside the range of 0.25 to 1, it must be scaled to a value within this range. Applying
this calculation to the scaled values and then multiplying the result by the square root of the
scaling value gives the square root of the original value.
After the amplitude is derived, the normalized signal S3 can be obtained by dividing the input
signal by the derived amplitude A. The division function 37 can be approximated by a long
division method to the precision that the designer considers fit. When implemented using a
digital signal processor, two modifications are recommended.
First, it is desirable to obtain the numerator of the division not from the output S2 of the bass
region extractor 2 but from the output S32 of the 90 ° phase difference network. Since the
denominator of the division (derived A) is affected by the delay and transient of the phase
difference network, more accurate output will be obtained if the numerator is also subjected to
the same processing.
Second, it is desirable to limit the value of the denominator to a predetermined minimum value. If
the intensity of the low frequency signal in question is very low or zero, the numerator and
denominator values will be very small. Dividing the two very small values results in unpredictable
results and may result in undesirable audible effects. This problem can be solved by setting the
denominator to this value whenever the denominator falls below a predetermined minimum
value. The experimental results show that a minimum value of 0.002 is appropriate. d) Second
harmonic generator 4 The second harmonic generator 4 generates the second harmonic by
squaring the normalized input signal S3 and then subtracting the DC term according to (Equation
2) . Alternatively, the DC term may be removed using a DC remover 42 as shown in FIG. By
multiplying the output signal S42 by the signal S36 derived from the amplitude extractor 8 of the
normalizer and a predetermined user-defined constant k2, the level of the generated second
harmonic S4 has a desired ratio with respect to the input signal S0. become. e) Third harmonic
generator 5 and higher harmonics generator 6 The third harmonic generator 5 generates a
normalized input from the cubed signal S52 of the normalized input according to (Equation 3) By
subtracting a specific amount S53, a third harmonic is generated, which level is then adjusted to
the desired ratio. Similarly, the fourth harmonic can be generated based on (Equation 4). Higher
harmonics can also be generated using similar equations.
The final output S7 of the PBBS is obtained by adding these generated harmonics, ie the signals
S4, S5 and S6 (optional), to the original input signal S0. When two or more bass components are
extracted, a square, a third power, and a higher order mathematical manipulation process
provide a mixed output of harmonics + these bass components. The clear pitch of this composite
signal is not well formed. Nevertheless, this method is still effective as it allows the speaker to
produce a reproducible output instead of accepting an almost inaudible bass range.
A poor bass speaker can not play the same sound as that played by a good bass response
speaker, but using PBBS to enhance the feel of the bass sound It will be possible. The bass sound
reproduced by the PBBS has the pitch of the bass component to be compensated.
By utilizing the method described above, PBBS can be easily implemented using a digital signal
processor. This method makes it easy to select harmonics and adjust the signal strength of each
harmonic. The generated harmonics are always proportional to the input signal level within
tolerance regardless of the absolute level of the input signal.
As described above, the PBBS extracts bass components of frequencies lower than the cut-off
frequency of the speaker from the input signal and generates harmonics of the extracted bass
components without causing much attenuation. The speaker makes it possible to reproduce
harmonics of frequencies higher than the cut-off frequency of the speaker.
In order to extract the bass component and generate its harmonics, the method according to the
present embodiment eliminates the DC remover 1 for removing the DC component of the input
signal and the component whose frequency is higher than the cut-off frequency of the speaker A
bass extractor 2; a normalizer 3 for normalizing the extracted bass signal so that it is not related
to the original amplitude; a second harmonic generator 4 for generating harmonics from the
normalized signal; It is comprised from the 3rd harmonics generator 5, the higher harmonics
generator 6 grade ¦ etc.,.
The conventional way to obtain strong bass sounds is to improve with the bass response of the
speaker and / or linear bass boost circuit.
However, in situations where these methods do not fit, PBBS provides an alternative solution to
speakers with poor low frequency response.
In the case of PBBS, harmonics of the bass component are generated and added to the original
signal. Those harmonics whose frequency exceeds the cut-off frequency of the loudspeaker are
reproduced by the loudspeaker without causing significant attenuation. In addition, PBBS takes
advantage of the psychoacoustic effects of human hearing to allow listeners to perceive the pitch
of the missing fundamental. The PBBS concept is simple and can be implemented, for example,
using a digital signal processor. The level of each harmonic is easily adjustable, and the ratio of
each harmonic is constant within tolerance regardless of the level of the input signal.
FIGS. 4 and 5 are block diagrams showing a PBBS for implementing the bass range compensation
method according to the second embodiment of the present invention. That is, the PBBS removes
the DC component from the input signal, the DC remover 1, the bass region extractor 2 for
removing the high frequency component from the input signal and extracting the fundamental
component for generating harmonics, and the fundamental component has a fixed value.
Variable gain adjuster (abbreviated as AGC hereinafter) 103 for adjusting to, second harmonic
generator 104 for producing second harmonic of fundamental sound component, and third
harmonic generation for producing third harmonic of fundamental sound component , A higher
order harmonic generator 106 (optional) that generates higher order harmonics if necessary, a
summer 107 that adds the respective harmonics, and the level of the summed harmonics Inverse
variable gain regulator (hereinafter abbreviated as VIG) 108 for readjustment, the amplifier 109
for adjusting the level of the readjusted harmonic to match the level of the input signal, and the
harmonic for which the level is adjusted And a summer 100 which adds the input signal.
Equations (2) and (3) above are applicable in certain cases where the amplitude of the
fundamental signal is one. Equations (7) and (8) represent general mathematical equations for
the signal of amplitude A.
A 2 cos 2θ = 2 (A cos θ) 2 -A 2
A3cos3θ = 4 (Acosθ) 3-3A3cosθ Here, as apparent from the amplitudes of the fundamental
sound signal (several 7) and (several 8), the amplitudes of harmonics generated using the square
and cubic method are It is not proportional to the amplitude of the fundamental sound of.
Instead, the amplitudes of the second and third harmonics are proportional to the squared and
cubed amplitudes of the fundamental sound, respectively. This condition is undesirable if the
absolute level of the fundamental component varies across the musical material, as the volume of
the harmonic component is not proportional to the volume of the fundamental component. In the
method of the first embodiment described above, as shown in FIG. 2 and FIG. 3, the amplitude of
the fundamental sound is converted to 1 by adopting the normalizer 3, and thereafter, by the
equations (2) and (3) Generate harmonics. The levels of the generated harmonics are then scaled
to the desired ratio to the fundamental signal. All pass filter, multiplication, division, and square
root functions are utilized throughout this normalization process. This is complex and expensive
when using analog circuitry.
In this embodiment, AGC 103 and VIG 108 are used in place of normalizer 3. The AGC 103
adjusts the amplitude of the basic sound to a predetermined constant value before generating the
harmonics according to (Equation 7) and (Equation 8). Since the amplitude A is constant, the
amplitudes of the harmonics are known, so control is easy.
Next, the operation of the harmonic generation method of the second embodiment will be
described with reference to the drawings.
First, the DC remover 1 is used to remove the DC component of the input signal S0 in order to
prevent undesirable side effects during the processing of the subsequent stage.
The DC remover 1 can be configured using a simpler high pass filter whose cut-off frequency is
lower than the audible range.
Next, the bass region extractor 2 removes high frequency components from the signal S1 from
which direct current components have been removed, and holds only low frequency bass
components that are the sources for generating harmonics. The bass range extractor 2 can be
implemented using low-pass filters of matching order and cut-off frequency.
Subsequently, the output S2 of the low frequency range extractor 2 is sent to the AGC 103. The
AGC 103 comprises a variable gain amplifier 131 and a feedback control circuit 132. The
feedback control circuit 132 detects the output S103 of the variable gain amplifier 131 to
generate the output S131 (referred to as the AGC voltage) and controls the gain of the variable
gain amplifier 131 so that the signal S103 spans a wide dynamic range. , A fairly constant
amplitude (indicated by B in FIGS. 4 and 5). With such a constant output, the harmonics
generated in the subsequent stages have known and controllable magnitudes.
Next, the second harmonic generator 104 generates a second harmonic according to (Equation
7). The input S103 is squared by the multiplier 141 and then amplified by the amplifier 142
whose gain is 1 / B. Note that 1 / B is a known value. It becomes clear from the electrical
characteristics of the AGC 103. For convenience, the second harmonic signal S104 is obtained by
removing the DC term of the signal S142 using a DC remover 143 (a simple high pass filter).
Also, the third harmonic generator 105 generates the third harmonic according to (Equation 8).
The multiplier 151 multiplies the signal S103 by the output S142 of the 1 / B amplifier 142 to
obtain the signal S151. Signal S 151 is subjected to level adjustment by amplifier 152 to produce
signal S 152, and signal S 103 is subjected to level adjustment by amplifier 153 to produce
signal S 153. Next, subtracting the signal S153 from the signal S152 produces a third harmonic
signal S154, which is further subjected to level adjustment by the amplifier 155 (so that the gain
is equal to k3). A third harmonic signal S105 adjusted to a desired ratio with respect to the signal
S104 is generated.
Furthermore, the higher order harmonics generator 106 generates the higher order harmonics
component S106 according to the matching mathematical rules. This generator is optional as it
adds to the complexity of implementation and can not add significant differences to the
Next, the summer 107 sums all the harmonic components S104, S105, and S106 to generate a
composite signal S107.
The amplitude of the composite signal S107, which is fairly constant due to the AGC effect, is
adjusted by the VIG 108.
The output S108 swings in proportion to the input signal S0. VIG 108 is controlled by AGC
voltage S 131, and its gain is the inverse of the gain of AGC 103.
The signal S108 is further level-adjusted by the amplifier 109 and becomes the output signal
S109, and the second, third, and higher harmonics have a desired ratio with respect to the input
signal S0. Thereafter, the output signal S109 and the input signal S0 are added by the summing
unit 100 to generate the final output S100.
Match AGC 103 and VIG 108 to make sure that the relationship between their gains is
completely reversed. An example for realizing it is shown in FIG. The AGC 103 includes a variable
gain amplifier 131 and a feedback control circuit 132. The variable gain amplifier 131 is a class
A amplifier with a gain determined by the resistance of the voltage controlled resistor VCR1.
VCR1 is realized by a JFET transistor. The feedback control circuit 132 detects the amplitude of
the signal S103 in both the positive cycle and the negative cycle using an envelope detector
(consisting of D3, D4, R4, C4 and an inverter A3). The amplifier A4 is used to generate an AGC
voltage S131 to control the resistance of VCR1. VIG 108 is comprised of a variable gain amplifier
whose gain is determined by VCR2. VCR2 is controlled by the same AGC voltage S131.
The gains of AGC 103 and VIG 108 are obtained by the following Equations 9 and 10
AGC gain = S103 / S2 = -R2 / (R1‖Rds1)
In the case of VIG gain = S108 / S107 =-(R1dsRds2) / R2 Rds1 = Rds2, that is, when the electrical
characteristics of the two JFET transistors are identical, the former is the reverse of the latter.
Thus, in this design, it is essential to use a well-matched JFET pair.
As described above, instead of the normalizer 3 used in the method of the first embodiment, by
using the AGC 103 and the VIG 108 whose gain is opposite to the gain of the AGC 103, the
method of the first embodiment and Similar sound effects can be realized more easily and at
lower cost, and a PBBS that is extremely advantageous for use in consumer electronics where
cost is a major concern can be realized.
Although the AGC 103 and the VIG 108 are constituted by circuits as shown in FIG. 6 in the
above embodiment, the present invention is not limited to this as long as they have circuit
configurations having similar functions.
As is apparent from the foregoing, the present invention is to obtain a sound that gives a strong
bass without feeling limited by the space and cost of the speaker and without sacrificing the
characteristics of the amplifier. Has the advantage of being able to
Further, according to the present invention, the level of the extracted bass component is adjusted
to a predetermined level, and based on the adjusted bass component, the attenuation in
reproduction by the speaker is small (the sound pressure is high). A simple configuration by
generating harmonics, re-adjusting the levels of the generated harmonics so as to correspond to
the original bass component, and mixing the re-adjusted harmonics with the input signal There is
an advantage that a low cost bass component compensation method can be realized.
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