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JPH0272795

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DESCRIPTION JPH0272795
[0001]
[Industrial Application 1] The present invention relates to a drive device for a speaker provided
with a voice coil. 2. Description of the Related Art In recent years, in audio equipment, digital
audio disc players and digital audio tube recorders that handle digital signals have become
widespread. In these audio devices, sound reproduction is performed using digital signals as
signal light. A digital signal has a wide dynamic range, and has features that can faithfully
represent a waveform from low amplitude (low level) to high amplitude (high level). Along with
this, speakers with higher fidelity are also desired. In general, an electrodynamic speaker
(hereinafter, simply referred to as a speaker) having a voice coil is used to reproduce an acoustic
signal based on a digital signal. This speaker has features such as relatively simple structure,
easy to make good characteristics, and high conversion efficiency. FIG. 7 shows a cross-sectional
view of a conventional speaker. In the figure, a pole piece 2 is fixed to a frame 1. A center ball 3
is inserted through the pole piece 2, and a bottom 4 of the center ball 3 is fixed to the pole piece
2 via an annular magnet 5. A coil bobbin 6 is placed on the end portion of the center ball 3. A
center retainer (corrugation damper) 7 and one end of a diaphragm 8 are fixed to the coil bobbin
6. The other end of the center retainer 7 is fixed to the frame 1. Further, the other end of the
diaphragm 8 is pressure-bonded to the frame 1 by a pressing plate 9. The voice coil 10 is wound
and fixed to the circumferential surface of the coil bobbin 6. In the speaker having the above
configuration, a magnetic air gap 11 is formed between the pole piece 2 and the tip of the center
ball 3. Here, when a signal current flows through the voice coil 10, the coil bobbin 6 vibrates due
to the magnetic force. As a result, the diaphragm 8 vibrates to reproduce sound. Here, assuming
that the length of the voice coil 10 is β [ml, the magnetic flux density of the magnetic gap 11 in
which the voice coil 10 is placed is B [T], and the current flowing through the voice coil 10 is 1
The driving force F [N] generated at the point is expressed by F = B-.beta ... Here, since B and β
are constants, the driving force F generated in the voice coil 10 is proportional to the current I
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flowing in the voice coil 10. By the way, if the mechanical impedance of the vibration system
vibrated by the driving force F generated by the voice coil 10 is constant, then the amplitude of
the vibration system is always proportional to the current flowing through the voice coil 10 .
However, the mechanical impedance of the center retainer 7 nonlinearly changes with the
amount of displacement of the voice coil 10. For this reason, nonlinear distortion occurs to the
sound output at the time of low frequency band or high level signal input where the amplitude of
the diaphragm 8 becomes large. As a result, conventionally, it has been difficult to reproduce
sounds faithful to the input signal. The present invention has been made in view of the above
points, and it is an object of the present invention to provide a speaker drive device capable of
reproducing sound faithful to an input signal regardless of the frequency or level of the input
signal. It is. [Means for Solving the Problems] In a speaker driving device according to the
present invention for driving a speaker provided with a voice coil that vibrates according to an
input signal, a scale plate that vibrates in conjunction with the voice coil, and A difference signal
is generated as a result of the comparison, a detector for detecting the movement amount of the
voice coil facing the scale plate, and a comparator for comparing the detection signal of the
detector with the level of the input signal. In this case, it has a correction control circuit that
performs correction to reduce the difference signal. [Operation] The above apparatus always
detects the amount of movement of the voice coil that vibrates in response to the input signal.
The detection signal obtained by this detection is compared with the level of the input signal. As
a result of the comparison, when a differential signal is generated, the input signal corresponding
to the differential signal is corrected to reduce the differential signal. FIG. 1 is a block diagram of
a speaker driving apparatus according to the present invention. The figure is a speaker drive
device for converting a left channel (L channel) signal of a stereo signal into an acoustic signal. It
goes without saying that the same device is used for the right channel (R channel) signal. In the
figure, a receiver (RD) 20 is connected to an input terminal in to which a digital signal is input
from a digital audio disk or the like. A PLL (phase locked loop) 21 and a serial-to-parallel
converter (spc) 22 are connected to the reception unit 20. The output of the serial to parallel
converter 22 is connected to a digital signal processor (DSP) 23. An output of the digital signal
processor 23 is connected to an address decoder (ADEC) 24 and a first latch circuit (LAI) 25. The
output of the address decoder 24 is connected to the serial-to-parallel converter 22, the second
latch circuit (LA 2) 26 and the detector 27. The output of the second latch circuit 26 is connected
to the digital signal processor 23.
The output of the detector 27 is connected to the second latch circuit 26. The output of the first
latch circuit 25 is connected to a digital to analog converter (DAC) 28. The output of the digitalto-analog converter 28 is connected to the amplifier 30, and the output of the amplifier 30 is
connected to the voice coil 1 o of the speaker 35. Further, the output of the detection unit 351 of
the speaker 35 is connected to the detector 27. A part of the output of the receiver 20 is also
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connected to the digital-to-analog converter 28. The receiving unit 2o is a circuit comprising a
demodulation circuit that demodulates a digital signal of an acoustic component, a pit clock and
the like from an input digital signal. PL L 21 is a circuit that synchronizes the transfer speed (pit
clock) of the digital signal input to the receiving unit 20 with the clock generated in the receiving
unit 20. The serial-to-parallel converter 22 is a circuit that converts serial-type serial sound data
input from the receiving unit 20 into parallel-type parallel sound data. The digital signal
processor 23 is a circuit for outputting various control signals for controlling each part, and
further includes a comparator for performing an operation as described in FIG. 2 and a
correction control circuit later on the input digital signal. It is a circuit composed of a
microprocessor and the like. The at-one decoder 24 is a circuit that receives a control request (an
address indicating a circuit to be controlled) from the digital signal processor 23 and outputs a
control signal to a circuit corresponding to the address. The first latch circuit 25 and the second
latch circuit 26 are circuits that hold input signals at fixed timings. The digital-to-analog
converter 28 is a circuit that receives a digital signal and outputs an analog signal corresponding
to the level of the digital signal. The detector 27 is a circuit that receives an optical signal
(detection signal) output from the detection unit 351 and outputs an electrical signal
corresponding to the level of the optical signal. The amplifier 30 is a circuit that amplifies an
analog signal. The detection unit 351 of the speaker 35 is configured of a not-shown scale plate
or a detector (a light detector) that detects the movement of the voice coil 10. A functional block
diagram of the digital signal processor 23 is shown in FIG. In the figure, the output of the serialto-parallel converter 22 (FIG. 1), that is, parallel acoustic data is input to the input terminal 23A.
On the other hand, the output of the second latch circuit 26 (FIG. 1), that is, parallel detection
data is input to the input terminal 23B.
The parallel sound data is a signal output from the serial-to-parallel converter 22, and the
parallel detection data is a signal output from the second latch circuit 26. The input terminal 23A
is connected to the adder 231 via two delay means Zt and Z2. The input terminal 23 B is
connected to the adder (ADD) 231 via the selection means 232 and the correction means 233.
The output of the adder 231 is connected to the multiplier 234. The output of the multiplier 234
is connected to the correction control circuit 235. The output of the correction control circuit
235 is connected to the output terminal 23C. The correction control circuit 235 is also connected
to the input terminal 23A. The output terminal 23C is connected to the first latch circuit 25.
Here, the comparator 236 is configured by the delay means Z1 and Z2, the adder 231, the
selection means 232, the correction means 233, and the multiplier 234. In the digital signal
processor 23 configured as described above, the selection means 232 is a circuit that selects one
of the plurality of parallel detection data input to the input terminal 23B. The correction means
233 is a circuit that receives the output of the selection means 232 and converts it into a
predetermined acoustic signal based on this output signal. The adder 231 is a circuit that inverts
and adds one of the output of the delay means z2 and the output of the correction means 233,
and outputs the difference signal, and is a circuit that operates substantially as a subtractor. The
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multiplier 234 is a circuit that doubles the difference signal output from the adder 231. FIG. 3
shows a cross-sectional view of a loudspeaker according to the present invention. Compared with
the conventional speaker shown in FIG. 7, the speaker shown in FIG. 3 is newly provided with a
detection unit 351 configured of a scale plate 50 and a light detector 51. The structure of the
other parts is the same as that of the prior art, and the same parts are given the same reference
numerals, and duplicate explanations are omitted. In FIG. 3, a scale plate 50 is attached to the
peripheral surface of the wound coil bobbin 6 of the voice coil 10 so as to extend toward the
bottom 4 in parallel with the axis of the center ball 3. Further, a light detector 51 is attached to
the lower surface of the pole piece 2 so as to face the surface of the scale plate 50. The light
detector 51 emits light toward the surface of the scale plate 50, and detects the reflected light.
The light detector 51 and the scale plate 50 constitute a detection unit 351 shown in FIG. Here,
the detection unit 351 will be described with reference to FIG.
FIG. 4 is a block diagram of a detection unit 351 according to the present invention. In the figure,
the detection unit 351 is composed of a scale plate 50 and a light detector 51 as shown in FIG.
The scale plate 50 is provided with a large number of bits 501. In this bit 501, one horizontal
row indicates position information (signal value information) of n bits of PI to Pn. This position
information indicates the signal value to be input to the power by the horizontal line n of n bits,
and indicates the minimum to maximum of the signal value by a plurality of lines at intervals of
the detected minimum pitch dk. The light detector 51 is provided with a light emitting element
511 formed of a plurality of light emitting diodes, lasers and the like, a diffractive structure 513
formed of a plurality of optical lenses, a grating lens and the like, and a light receiving element
512. The light emitting elements 511 and the light receiving elements 512 are arranged n in
number corresponding to one horizontal row of the bits 501, that is, n bits (one so-called line
sensor). The light emitted from the light emitting element 511 to the bit 501 of the scale plate 5
o is reflected by the bit 501 and detected by the light receiving element 512. By combining the
outputs of the n light receiving elements 512, an n-bit optical signal, that is, a detection signal is
formed. The scale plate 50 is a general term for a device for obtaining position information, and
is not necessarily a plate. For example, an optical card etc. are included. Here, the operation of
the speaker driving apparatus of the present invention shown in FIG. 1 will be described with
reference to FIGS. 5 and 6. FIG. 5 is a time chart showing the operation of the speaker driving
device of the present invention. FIG. 6 is a flow chart showing the operation of the speaker drive
device of the present invention. First, when a digital signal is input to the receiving unit 2o, serial
acoustic data 20a as shown in FIG. 5A is output from the receiving unit 20 to the serial-toparallel converter 22. At the same time, a channel clock (LRCK) 20b as shown in FIG. 5 (b) and a
bit clock 20c as shown in FIG. 5 (d) are output from the receiver 20 to the serial-to-parallel
converter 22. . Based on the channel clock 20b and the bit clock 20c, from the serial / parallel
converter 22, parallel acoustic data (L channel data) 22a in parallel format is directed to the
digital signal processor 23, as shown in FIG. 5 (C). It is output. On the other hand, to the digital
signal processor 23, BI ○ bar pulse 23a falling from high level to low level as shown in FIG. 5 (e)
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is inputted from a not-shown flip flop or the like.
Further, from the digital signal processor 23, as shown in FIG. 5 (f), a DEN bar pulse 23b
consisting of a plurality of pulses repeated at regular intervals, and as shown in FIG. 5 (g), has a
constant period. At the same time, the write enable signal WE bar pulse 23c falling from the high
level to the low level is output. The DEN bar pulse 23 b is a signal output from the digital signal
processor 23 to the second latch circuit 26 via the address decoder 24. Also, the WE bar pulse is
a signal output toward the first latch circuit 25. Now, as shown in FIG. 5 (h), a detection signal
consisting of parallel digital signals, ie, parallel detection data 27a, is input from the detector 27
to the second latch circuit 26. The parallel detection data 27 a is a signal based on the light
signal output from the detection unit 351 of the speaker 35. The parallel detection data 27a is
latched by the second latch circuit 26 in synchronization with the DEN bar pulse shown in FIG. 5
(f) and is input to the digital signal processor 23. In the digital signal processor 23, as shown in
FIG. 5 (i), the first parallel acoustic data (L channel data) 23d is processed as the first parallel
acoustic data (L channel data) 23d until the next WE bar pulse 23c is input. Output to the latch
circuit 25 of FIG. In the first latch circuit 25, the BIO bar pulse 23a output from the digital signal
processor 23 is input until the next BIO bar pulse is input as one parallel acoustic data (L channel
data) 25a (see FIG. 5). The output of the signal as shown in j) is output to the digital-to-analog
converter 28. The parallel acoustic data 25a input to the digital-to-analog converter 28 is
converted into an analog signal, amplified by an amplifier 3.degree., And input to the voice coil
10 of the speaker 35. The pit clock 20c input to the digital-to-analog converter 28 is used as a
reference clock when converting a digital signal into an analog signal. The above operation will
be described with reference to the flowchart shown in FIG. First, when the power is turned on to
the speaker driving device, initialization of each part, that is, system initialization is performed
(step Sl). Next, in the digital signal processor 23, it is determined whether the BrO bar pulse is 0
(low level).
If No, monitoring of the BIO bar pulse is continued. If the result is YES, the digital signal
processor 23 receives parallel acoustic data from the serial to parallel converter 22 (step S3).
Further, the parallel detection data from the second latch circuit 26 is received a predetermined
number of times (three times) at the timing of the DEN bar pulse (step S4). Then, from the three
parallel acoustic data, one having a continuous relationship with the previously input parallel
detection data is selected as the optimum value and the optimum value is corrected to the same
form as the parallel acoustic data (step S5) Step S6). Then, the parallel detection data corrected to
this form is compared with the parallel acoustic data from the serial-to-parallel converter 22 to
detect a difference signal (step S7). For example, the sign of the difference signal is inverted. The
difference signal is multiplied by using such a negative value K and output to the correction
control circuit 235 (step S8). The correction control circuit 235 adds the parallel sound data and
the difference signal (step S9). Output to the first latch circuit 25 (step 5IO). According to the
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above flow, correction of parallel acoustic data input to the voice coil 1 o of the speaker 35 is
performed using a detection signal (parallel detection data) from the detection unit 351 of the
speaker 35. Immediately, if the movement of the voice coil 10 is small compared to the value of
the parallel sound data, the value of the parallel sound data is increased. If the voice coil 10
moves more than the value of the parallel sound data, the opposite is performed. The present
invention is not limited to the above embodiments. The scale plate 50 may be provided with a
large number of slits instead of the bits, and the light receiving element 512 may be configured
by a phototransistor or the like. In this case, light transmitted through the slit is detected to
output a detection signal. Alternatively, a large number of bars may be provided on the surface of
the scale plate 50, and the number of bars may be counted to detect the amount of movement of
the voice coil 10. Further, the arrangement of the position information of the scale plate 50 is not
limited to the embodiment, and it is needless to say that code conversion may be performed to be
recorded or may be recorded to be modulated. The number N of light emitting elements 511 and
light receiving elements 512 may be one. At this time, n bits may be sequentially focused by
rotation of the lens to detect position information. According to the speaker drive device of the
present invention having the above configuration, the position of the voice coil is detected using
the scale plate moving in conjunction with the voice coil of the speaker, and the voice coil is
input to the voice coil based on the position information. Control the level of the acoustic
information to be processed, and eliminate the difference signal between the acoustic
information and the position information, so that the linearity from the low level to the high level
is good regardless of the level of the acoustic information manually input to the voice coil Sound
reproduction can be performed.
[0002]
Brief description of the drawings
[0003]
FIG. 1 is a block diagram of a speaker driving apparatus according to the present invention, FIG.
2 is a functional block diagram of a digital signal processor according to the present invention,
FIG. 3 is a sectional view of a speaker according to the present invention, and FIG. 5 is a time
chart showing the operation of the speaker drive device of the present invention, FIG. 6 is a flow
chart showing the operation of the speaker drive device of the present invention, and FIG. 7 is a
cross-sectional view of the conventional speaker It is.
Reference Signs List 20 reception unit 21 PLL 22 serial-to-parallel converter 23 digital signal
processor 24 address decoder 25 first latch circuit 26. Second latch circuit 27 Detector 28
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Digital-to-analog converter 50 Scale plate 51 Photo-detector. Fig. 3 Fig. 7 Fig. (D) Bit clock (J)
Launch output data
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