CN1021091C - Incremental regulation method according to sound wave form and circuit of electronic piano - Google Patents
Incremental regulation method according to sound wave form and circuit of electronic piano Download PDFInfo
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- CN1021091C CN1021091C CN 89108538 CN89108538A CN1021091C CN 1021091 C CN1021091 C CN 1021091C CN 89108538 CN89108538 CN 89108538 CN 89108538 A CN89108538 A CN 89108538A CN 1021091 C CN1021091 C CN 1021091C
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Abstract
The present invention relates to a method and a circuit for digitally modulating a waveform of a sound source of an electric piano. The present invention adopts a segmental parabola f-[delta](t) to simulate a waveform curve f (t) of the sound source. The variance of f-[delta](t) is controlled by a parameter f-[delta]'(t); in a fixed increment mode, the parameter f-[delta]'(t) is coded into a code stream composed of '0'and '1' to be stored in an ROM, and variance states of the parameter f-[delta]'(t) are divided into three types, i.e. increase, decrease and constant. Through a double-integrating curve fitting decoder designed corresponding to the method, the code stream is converted into a voltage waveform to output. The quality of the waveform curve of the sound source simulated by the present invention achieves the accuracy of a PCM (Pulse Code Modulation) method; compared with the PCM method, the present invention has the advantages of simple circuit and small digital information storage capacity.
Description
The present invention is a kind of pianotron sound wave Ditital modulation method and circuit thereof.
At present, the pianotron sounding circuit mainly adopts two class methods.The one, the method for frequency domain; Mainly be the each harmonic that to form the piano waveform by separately size combinations, as the acoustic source of piano waveform.By the combination the method difference, can be divided into several method again.The advantage of this method is to realize that shortcoming is that the shared deal of harmonic wave is difficult to accurate control with fairly simple circuit, and frequency spectrum changes difficulty, is difficult to adapt to the occasion of piano frequency spectrum dynamic change.The 2nd, time domain approach: generally be that the numerical information of the waveform of piano is deposited in the memory, situation according to button, emit number in the memory by certain frequency, become waveform through D/A, this Ditital modulation method is pulse-code modulation type (PCM), and this method can reach very high precision in theory, can adapt to the situation of waveform dynamic change, but the circuit complexity, price is held high lattice.For reducing price, some improved methods occurred, but be cost all to sacrifice many accuracy of waveform.
The purpose of this invention is to provide and a kind ofly can protect the high precision of acoustic source of piano waveform, it is easy that circuit is realized, low-cost Ditital modulation method and circuit thereof.
The Ditital modulation method that the present invention proposes is a kind of new improvement to time domain approach.The type that belongs to increment modulation method, we are referred to as " double integrator curve-fitting type increment modulation method ".Abbreviate the DM method as.
The delta modulation principle: the increment between very near point of two time intervals on the continuous wave is very little, does not want to be similar to it with a fixing little increment △, and when numeral is 1, expression increases a △, and numeral is 0, and expression reduces a △, so available curve f
△(t) approximate f(t) (f(t) be made as the function of a squiggle), this group 0 and 1 code stream of forming have been represented curve f
△(t), the (see figure 1) also just approximate f(t that represented).If with f
△(t) integration can get f twice
△ 1(t) (see figure 2), f
△ 1(t) smooth, with f(t) fitting degree improve.
The noisiness of delta modulation: noise is defined as original signal f(t) and simulating signal f
△(t) difference (see figure 3).Noise can be divided into overload noise and quantizing noise two parts, and overload noise occurs in the part of signal slope abrupt change, by simulating signal f
△(t) do not catch up with original signal f(t) the caused error of variation (among Fig. 3
Part), quantizing noise is created in the simulating signal maximum slope greater than the part of original signal maximum slope (among Fig. 3
Part), it is caused by a fixed quantisation step by △, when the enough ambassador's simulating signals of △ maximum slope during greater than the original signal maximum slope, overload noise just can be eliminated, and quantizing noise is to change with increment △ value, and △ greatly then quantizing noise is big, and the little then noise of △ is little.
The noise of double integrator delta modulation mode also in like manner, as long as the slope in the above (first order derivative) is become second derivative, △ is that slope delta gets final product by the numerical value increment transformation, so no longer repeat.
The auto-adaptive increment modulation system: therefore, require △ big for eliminating overload noise, little for reducing quantification noise requirements △, if at f(t) part △ that slope is big is also big, at f(t) part △ that slope is little is also little, then can be under the prerequisite that does not produce overload noise lower quantization noise (see figure 4) significantly.This △ is called the auto-adaptive increment modulation system with the delta modulation mode of signal slope variation.The promising too elimination overload noise of double integrator delta modulation mode requires △ big, and is little this contradiction of lower quantization noise requirements △, and the method for solution is to adopt the constant auto-adaptive increment modulation system of △ equally.
How to realize the auto-adaptive increment modulation, concerning double integrator delta modulation mode, at first be to obtain virgin curve f(t) f " (t) at a certain section mean value; control △ (slope delta) then and change; " (t) the big △ of mean value is also big as f thereupon, " (t) the little △ of mean value also reduces f, so just can make simulation curve be in both nonoverloads, makes again under the less situation of quantizing noise.
The present invention is the further improvement to the continuous increment modulation method of self-adaptation, but essence is had any different again.It comprises the steps: to establish acoustic source of piano squiggle function is f(t), get M sampled point: f(t
n), the wave function of remembering adjacent point-to-point transmission is f
n(t), adopt simple curve f
△ n(t) match f
n(t), n=1,2 ..., M, M=2
8Or 2
9, by f
△ n(t) (n=1,2 ..., M) matched curve of Zu Chenging is designated as f
△(t); f
△(t) change is by f
△" variation (t) characterizes, to f
△" (t) weave into the code stream of forming by 0.1, deposit among the read-only memory ROM with incremental form; Above-mentioned code stream converts voltage waveform output to through demoder.Its characteristics are as follows:
Form matched curve f
△(t) sectional curve f
△ n(t) be para-curve, i.e. match f
n(t) curve f
△ n(t) be a kind of para-curve.The basic thought of fitting of parabola curve: establish f
0(t) be general curve, t
0≤ t≤t
1, note f
0" (t) be f
0" mean value (t) is established f
△(t) be a para-curve, satisfy initial value f
0(t
n)
f
△(t
0), f
0" (t)
f
△" (t), t
0≤ t≤t
1, then available f
△(t) the approximate f that replaces
0(t), its approximation ratio is worked as f
0(t) more better.And para-curve f
△(t) with general curve f
0(t) compare, be easy to by processing of circuit.Now use para-curve f
△ n(t) matched curve f
n(t), f
△ n(t) definite method is as follows: establish initial value f
1(t
0)
f
△ 1(t
0), and f
1" (t)
f
△ 1" (t), t
0≤ t≤t
1, final value f is then arranged
1(t
1)=f
△ 1(t
1), because f
1(t
1)=f
2(t
1), f
2(t
1) be f
2(t) at t
1The time initial value, the note f
2(t
1)=f
△ 2(t
1), change f
△ 2" (t) make f
△ 2" (t)
f
2" (t), t
1≤ t≤t
2, final value f is then arranged again
2(t
2)
f
△ 2(t
2) ..., and the like, note initial value f
n(t
N-1)
f
△ n(t
N-1), change f
△ n" (t), make f
△ n" (t)
f
n" (t), t
N-1≤ t≤t
n, final value f is then arranged
n(t
n)
f
△ n(t
n), they initial values as next section.So, then get f until n=M
△ n(t) n=1,2 ... M.
Above-mentioned f
△" change (t) is divided into three kinds of states: increase, minimizing, constant, and realized by following mode in coding: establishing increment △ is a fixed value, continuous two sign indicating numbers are 1, expression f
△" (t) increase a △, have one 1 yard again, f than leading portion
△" (t) increase a △ again.Continuous two sign indicating numbers are 0, expression f
△" (t) reduce by a △ than the last period, have one 0 yard again, f
△" (t) reduce by a △ again; Continuous two sign indicating numbers are 0,1 or 1,0, expression f
△" (t) identical with the last period, if be 0101 ... change, and do not have continuous 0 yard or continuous 1 yard appearance, then f
△" (t) in one section zone, remain unchanged.
The present invention adopts para-curve piecewise fitting acoustic source of piano ripple, and higher degree of accuracy is arranged, and general delta modulation mode has only increases and reduce two states, f among the present invention
△" three kinds of states are adopted in change (t); help reducing quantizing noise; make coding more flexible; in addition; general increment modulation method is conceived to bit by bit go to follow the tracks of virgin curve f(t); its main signal is a coded signal, and it not only will reflect increase or the minimizing of △, also will reflect the size variation of △ itself.Remove match original waveform curve f(t and the present invention is conceived to one section one section), so only need reflect that one section one section fit line parameter (is f
△" (t)) the △ control signal that changes, therefore, transmission only be the △ control signal, behind decoder decode, can produce f
△(t) simulation curve.
According to above-mentioned f
△" (t) three kinds of determined cryptoprinciples of state of Gai Bianing, following mask body provide reflection f
△" (t) definite method of the code stream of the △ control signal of Gai Bianing: investigate virgin curve f(t) at f(t
n) to f(t
N+m) one section, m 〉=1, note f " (t) mean value at this section is f " (t), (1) if f " (t)
f
△" (t
n), and f(t
n)
f
△(t
n), f then
△" (t) remain f
△" (t
n), t
n≤ t≤t
N+m, be encoded to 0101 ... 01(is m sign indicating number altogether), wherein there is not continuous two 0 yard, or two 1 yard; (2) if f
△" (t
n)<f " (t)<f
△" (t
n)+△, then f
△" (t) remain f earlier
△" (t
n) one section constant, increase to f then
△" (t
n)+△, and keep it, f
△" (t
nOne section of)+△ and f
△" (t
n) one section ratio what, by condition f
Δ" (t)
F " (t) determines t
n≤ t≤t
N+m, be encoded to ... 01011010 ... (m sign indicating number altogether), it is 1 continuously that two sign indicating numbers are wherein arranged; (3) if f
△" (t
n)-△<f " (t)<f
△" (t
n), f then
△" (t) remain f earlier
△" (t
n) one section, be reduced to f then
△" (t
n)-△, and keep it, f
△" (t
n) one section and f
△" (t
nThe ratio that)-△ is one section how much, by condition f
Δ" (t)=and f " (t) determine t
n≤ t≤t
N+m, be encoded to ... 010100101 ... (m sign indicating number altogether), it is 0 continuously that two sign indicating numbers are wherein arranged; (4) if virgin curve f(t) increase sharply or reduce t
n≤ t≤t
N+m, then be encoded to ... 1111 ... (m 1 yard), or ... 0000 ..., (m 0 yard).Make f
△Thereby " (t) increase continuously or reduce and make f
△(t) increase sharply or reduce, catch up with f(t) variation and do not produce overload noise.
General piano sound has the following characteristics: A, waveform frequency spectrum and amplitude envelops to determine its tone color; B, each key waveform are inequality, and the bass harmonic wave is abundant, and the high pitch harmonic wave is few, and the waveform of adjacent key is approaching; C, to the end from pronunciation beginning, frequency spectrum changes, and the frequency spectrum of adjacent wave is approaching; D, from pronunciation beginning to the end has foundations, continues, the process of decay on amplitude, the formation amplitude envelops.Condition according to above characteristics and widget capacity, the present invention has taked following setting: (1) is divided into one group to the key of per 3 continuous adjacent, if the waveform of every group of key sound is identical, though the waveform of each key of piano is differentiated, but according to characteristics B, the waveform similarity of adjacent key is so there is no need the waveform of each key is all deposited in the memory.The pianotron of example 88 keys is divided into 29 groups (group is 4 keys at last) to key, and every group deposits a kind of waveform number in and get final product.(2) waveform of each key sound is divided into 32 sections from the beginning time to the end, and the waveform of establishing every period is identical.Though from the pronunciation beginning to the end, waveform is constantly to change, and according to characteristics C as can be known, in the short period scope, different wave shape is very little.
Double integrator curve-fitting type increment modulation method corresponding to above-mentioned pianotron sound wave, the present invention proposes new circuit design, the system chart of sounding circuit as shown in Figure 5, it comprises controlled by single chip microprocessor maincenter 1, keyboard key-position and dynamics testing circuit 2, pedal testing circuit 3, signal superposition circuit 4, acoustics circuit 5, address control circuit 6, read-only memory (ROM) 7, and the channel group 8 between ROM7 and the signal superposition circuit 4, channel group 8 is made up of 6-8 passage.The circuit structure block diagram of each passage as shown in Figure 6, it by reading writing memory (RAM) 9, incorporate into and go here and there out circuit 10, double integrator curve-fitting type delta modulation demoder 11, amplification quantity and select circuit 12, address control circuit 13, clock circuit 14 to form.Two arrows among the figure
The expression data bus, the arrow apostrophe
The presentation address line.
Wherein system controlled by computer maincenter 1 can 8031 single-chip microcomputers be a core, and it is accepted detection information, sends steering order, controlled the work of each passage and other circuit by data bus.Keyboard key-position and dynamics testing circuit 2 are used for detecting the position and the dynamics of institute's button on the keyboard, then these information are sent into the control maincenter.The pedal testing circuit detects the position of pedal, send the control maincenter with detection information then.Read-only memory (ROM) 7 is used to store the waveform number, promptly reflects matched curve f
△(t) parameter (f
△" (t)) bitstream of △ control signal.The effect of address control circuit 6 is the instructions of sending according to control maincenter 1, the address wire of control ROM7.The input data of address control circuit 6 are connected on the control maincenter 1, when by next key, control maincenter 1 forms corresponding start address and termination address according to factors such as the key position of key and dynamics by address control circuit 6, remove to control ROM7, a Wave data in from the start address to the termination address among the ROM7 is sent into corresponding passage.The effect of channel group 8 is to be transformed into the voltage waveform of certain frequency and certain amplitude from the waveform number of ROM7.Each passage is separate in the channel group, can work simultaneously, and the voltage waveform that they produced is sent into acoustics circuit 5 behind signal superposer 4 superpositions.A passage can be for the waveform transformation work of a key, and maximum 10 fingers are pressed key simultaneously when playing the musical instrument.But generally be no more than 8 fingers button simultaneously, so 8 of channels designs are just much of that.Also can be designed to 6 or 7.The effect of reading writing memory in the passage (RAM) 9 is that the waveform number with a certain button accesses from read-only memory 7, deposit among the RAM9 (among Fig. 6, the RAM9 data line connects the data line of ROM7), then according to the given clock frequency output waveform of the given address of address control circuit 13 and clock circuit digital give to incorporate into go here and there out circuit 10.Incorporate that to go here and there out the effect of circuit 10 be 8 bit parallel numbers from RAM9 to be become 1 bit serial number send into demoder 11 into.The function of demoder 11 is to convert the one digit number code stream to voltage waveform.Amplification quantity is selected the amplitude of circuit 12 in order to control waveform, with the effect of generation dynamics and envelope.Amplification quantity is selected the voltage waveform of output line (also being the output line of this passage) output of circuit 12 through envelope modulation, sends into signal superposer 4.The function of clock circuit 14 is to produce the M times of clock frequency to the audio frequency of correspondent button, transmits M 1 figure place sign indicating number with control channel in each sound wave cycle.The function of address control circuit 13 is addresses that control reading writing memory 9 transmits data, and the control reading writing memory is read the number of each waveform one by one in whole phonation, a waveform number repeats to read several times, and then read next waveform number, repeat to read several times again, finish until whole pronunciation.Among Fig. 6, the data bus of circuit block 13,14,11,12 links to each other, is connected on the data bus of control maincenter 1, control maincenter 1 is being controlled respectively: the waveform of address control circuit 13 repeats read-around number, the clock frequency of clock circuit 14, demoder 11 original states and integrating system, and amplify the amplification coefficient of selecting circuit 12.
Demoder 11 circuit structures in the above-mentioned passage as shown in Figure 7, it was by 1: 1 phase inverter K
0, electronic switch 20, integrator 16, integrator 17, add subtract counter 15, decision circuitry 14 form.Wherein:
(1) integrator 16 is by electronic switch 21, electronic switch 22, resistance R
10~R
116(totally 17), capacitor C
10~C
17(totally 8) and amplifier K
1Form, electronic switch 21 is one 16 and selects one switch, selects R
10~R
115In some resistance insert integrator 16, R
17Do not connect, i.e. R
17Resistance value be infinitely great.Electronic switch 22 is eight either-or switch, connects capacitor C respectively
10~C
17, capacitor C
10~C
17Numerical value is that scale-of-two is arranged, i.e. C
10, C
11=2C
10, C
12=4C
10..., they can form 1~256 times of C
10Capacitance.
(2) integrator 17 is by electronic switch 23, resistance R
20~R
28, capacitor C
1With amplifier K
2Form, electronic switch 23 is eight either-or switch, connects R respectively
20~R
27, R
20~R
27Numerical value is that scale-of-two is arranged, i.e. R
20=R
21, 2R
20, R
22=4R
20..., they can form 1-256 doubly to R under the control of switch 23
20Resistance.
(3) decision circuitry 24 is separated by AND circuit 25, not circuit 26, biconditional gate circuit 27, D and is sent out a device 28 and form.
(4) annexation of foregoing circuit is as follows: electronic switch 21 and phase inverter K
0Connected by switch 20, the control end of switch 22 connects data bus by latch 18, and the control end of switch 23 connects data bus by latch 19, and the D termination of d type flip flop 28 is incorporated into and gone here and there out the output terminal of circuit 10, CP termination clock circuit 14.The output terminal Q of add subtract counter 15
0, Q
1, Q
2, Q
3Connect the control end of switch 21, Q
3End also connects the control end of switch 20, the output terminal of CP termination AND circuit 25, and the M termination is incorporated into and is gone here and there out the output terminal of circuit 10, input end D
0, D
1, D
2, D
3Connect data bus, in order to the preset count initial value.
Above-mentioned phase inverter K
0, pass through resistance R
01, R
02, R
03, produce one with input direct voltage signal V
1Numerical value equates, the signal-V of opposite in sign
1The effect of add subtract counter 15 and decision circuitry 24 is that the bit serial sign indicating number from circuit 10 is become control code to △, make continuous two " 0 " sign indicating number or continuous two " 1 " sign indicating number, add subtract counter 15 subtracts 1 or add 1, position by add subtract counter 15 control electronic switches 21, certain resistance is inserted integrator 16, " (t) value that forms corresponding f.The effect of integrator 16 and integrator 17 is the d. c. voltage signal V that make input
1Become parabolic type squiggle f by twice integration
△(t), with the former sound wave curve of match f(t).The resistor network of the capacitance network of integrator 16 and integrator 17 is used to control integral coefficient, so that keep the wave amplitude of output waveform constant under different frequencies.As when waveform frequency is low, long to the duration of charging of integrator integrating capacitor, wave amplitude must rise, and at this moment, by increasing integral coefficient, wave amplitude is remained unchanged.And for example when same waveform as, if frequency decline n doubly, and integral coefficient also increases n doubly, can keep wave amplitude constant.
The element of above-mentioned demoder 11 can adopt following referential data: R
01=R
02=R
03=100K Ω, R
10=R
114=120K Ω, R
11=R
113=180K Ω, R
12=R
112=240K Ω, K
13=R
111=300K Ω, R
14=12
110=360K Ω, R
15=R
19=420K Ω, R
16=R
18=480K Ω, R
17=∞, R
115=60K Ω, R
116=240K Ω, R
20=6.4M Ω, R
21=3.2M Ω, R
22=1.6M Ω, R
23=800K Ω, R
24=400K Ω, R
25=200K Ω, R
26=100K Ω, R
27=50K Ω, R
28=600K Ω, C
10=6400PF, C
11=3200PF, C
12=1600PF, C
13=800PF, C
14=400PF, C
15=200PF, C
16=100PF, C
17=50PF, C
20=500PF.
Whole system operation begins to be described below to the dynamic process that pronunciation finishes from pressing key: after pressing next key, keyboard key-position and dynamics testing circuit 2 detect key position and dynamics, and give control maincenter 1 with information, control maincenter 1 is finished following work according to these information:
1, goes to control the address of read-only memory 7 according to the key position, make the waveform of this key, write the reading writing memory 9 of a passage from read-only memory 7.
2,, produce M doubly to this key sound clock frequency frequently, owing to each waveform is made up of M number sign indicating number, so transmit data frequency M doubly to audio frequency according to key position control clock circuit 14.
3, adjust the integral parameter (see before and state) of demoder 11 according to the key position.
4, select amplification quantity according to dynamics.By measuring the relation of piano dynamics and volume, the form of listing dynamics and volume relation deposits in the computer memory in advance.During work, check in corresponding amplification quantity, go to control amplification quantity again and select circuit 12, to obtain the amplification quantity corresponding with dynamics according to the dynamics table.
5,, in address control circuit 16, set the parameter of waveform multiplicity according to the key position.The waveform multiplicity is to obtain like this: the frequency of establishing institute's button is 200HZ, from pronouncing to finishing is 16 seconds, then common property is given birth to 200 * 16=3200 waveform during this period, in the hypothesis in front, phonation is divided into 32 sections, and every section waveform is identical, can be regarded as to such an extent that the waveform multiplicity is 3200 ÷ 32=100 like this, because the frequency and the tenuto time of each key have nothing in common with each other, so the multiplicity of the waveform of each key is inequality.
6, the system controlled by computer maincenter is finished above work backward channel and is sent the instruction of starting working.During passage work, microcomputer transmits three groups of data to demoder 11, wherein gives latch 18 and latch 19 respectively for two groups, and with the control integral coefficient, another group is sent into add subtract counter 15, removes to control K
1Resistor network, " the initial value (t) of determining f.The one digit number code stream is gone here and there out circuit 10 and is sent into add subtract counter 15, biconditional gate circuit 27, d type flip flop 28 from incorporating into, circuit 27 and circuit 28 form a judgement, in bitstream continuous two 1, or continuous two 0, AND circuit 25 is opened in circuit 27 outputs 1, the CP that makes clock circuit 14 is by not gate 26, send into counter 15 with door 25, counter 15 begins counting, adds 1 or subtract 1, is 1 still to be 0 decision by the M end.M is 1 when being 1 when continuous two of bitstream, and counter 15 makes to add 1 counting; M is Q when being 0 when continuous two of bitstream, and counter 15 makes to subtract 1 counting.When adding 1 counting, control K
1Resistor network change into and reduce by a R
10Value, even f
△" (t) increase a △; When subtracting 1 counting, pass through K
1Resistor network makes f
△" (t) reduce by a △.When in the bitstream when no continuous 1 yard or continuous 0 yard, circuit 27 outputs 0 are blocked and door 25, make counter 15 not count K
1The resistor network resistance constant, i.e. f
△" (t) remain unchanged.The add subtract counter output area is 0~15, and it is a mid point with 8, corresponding to f
△" (t)=0; 9~15 corresponding to f
△" (t)>0,0~7 corresponding to f
△" (t)<0.The Q of the control termination add subtract counter of electronic switch 20
3End, when counter output greater than 8 the time, Q
3Be 1, electronic switch 20 is connected V
1Produce f " (t)>0, on the contrary produce f
△" (t)<0, twice integration by integrator 16 and integrator 17 changes parabolic type matched curve f into
△(t).When M number of input, at K
2Output terminal just export the complete voltage waveform f of one-period
△(t), the frequency of waveform is the 1/M of clock frequency.Address control circuit 13 is exported Wave data next time after making reading writing memory 9 repeat to export the same waveform as several times again, and repeated several times again so continues, and when just simulating piano playing, presses the situation that waveform constantly changes in time behind the key.
In phonation, the system controlled by computer maincenter selects circuit to form the wave-shape amplitude envelope according to envelope data timing controlled amplification quantity.Pedal influences the sound height of all passages, tenuto length, and the system controlled by computer maincenter is done corresponding control according to the detection information of pedal testing circuit to each passage.
When phonation finishes, the system controlled by computer maincenter control channel state of deactivating enters waiting status.
The DM method of the present invention following superiority of having compared with the PCM method:
1, the waveform amount of digital information is few.It is 464KB that this sounding circuit need store waveform numerical information memory cell altogether.Under the situation of roughly the same precision and segmentation precision, it is 12 that the PCM source of sound needs D/A, and sample frequency is 44KHZ, needs the 6MB memory cell, and both differ 13 times.
2, low price.Each passage of PCM source of sound all need be with the D/A converter more than 12, and this is expensive device.12 figure place sign indicating numbers pass on and control all to be needed complicated and expensive circuit also needs high performance low-pass filter, with the filtering quantizing noise.By contrast, the DM source of sound only needs 1 and passes on, and D/A converter is very simple, and when identical precision, both costs differ about 5 times.
3, waveform quality is similar.With the piano waveform is reference waveform, is analog waveform with the demoder output waveform of DM method, and the definition noise is the poor of reference waveform and analog waveform.Then
Signal to noise ratio (S/N ratio) d=20Lg(reference waveform root-mean-square valve/noise root-mean-square valve), draw d>70db when the 4200Hz~500Hz by experiment, when 500Hz~25Hz, d>60db.Such signal to noise ratio (S/N ratio) is gratifying, has generally reached the waveform quality of PCM.Though the noise acoustic ratio is less during low frequency, the resolution to tone color on this band frequency of people's ear is insensitive, has just in time remedied this deficiency.Usually, concerning the sensation of people's ear, millesimal signal to noise ratio (S/N ratio) during low frequency, 1/1000th three poor signal to noise during unlike high frequency.In addition, waveform quality is limited by acoustics circuit, and acoustics circuit will accomplish that degree of distortion is that per mille is quite difficult.Therefore, excessively improve accuracy of waveform, do not have practical significance, it can not improve tone color, can increase cost on the contrary.
Claims (3)
1, a kind of pianotron sound wave double integrator curve-fitting type increment modulation method comprises the steps:
(1) key of per 3 continuous adjacent is divided into one group, the waveform of establishing every group of key sound is identical;
(2) waveform of each key sound is divided into 32 sections from the beginning time to the end, and the waveform of establishing every period is identical;
(3) the squiggle function of establishing key sound is f (t), and (tn, the wave function of remembering adjacent point-to-point transmission are fn (t), adopt simple curve f to get M sampled point: f
△ n(t) match F
n(t), n=1,2 ..., M, M=2
8Or 2
9, note is by f
△ n(t)
(n=1,2 ..., M) matched curve of the acoustic source of piano of Zu Chenging is f
△(t);
(4) change of the matched curve f △ (t) of acoustic source of piano (t) is weaved into the code stream of forming by 0,1 with incremental form by its second derivative f △ " variation (t) characterizes, with f △ ", deposits among the read-only memory ROM;
(5) above-mentioned code stream converts voltage waveform output to through demoder, it is characterized in that:
(1) the matched curve f of composition acoustic source of piano
△(t) sectional curve f
△N (t) is a para-curve, n=1,2 ..., M, it determines that method is as follows: the initial value of certain section function f n (t) is f " (t), establishes fn (t
N-1)=f
△ n(t
N-1), change f
" △ n(t), make f
" △ n(t)=f
" n(t) (f
" n(t) t mean value),
N-1≤ t≤tn then has final value fn (tn)=f
△ n(tn), they initial values as next section ..., so analogize, till n=M.
(2) above-mentioned f
" △(t) variable condition is divided into following three kinds: increase, reduce, constant, realized by following mode in coding: establishing increment △ is fixed value, and continuous two sign indicating numbers are 1, expression f
" △(t) increase a △, continuous two sign indicating numbers are 0, expression f
" △(t) reduce by a △, continuous two sign indicating numbers are 0,1 or 1,0, expression f
" △(t) identical with the last period.
2, according to the said pianotron sound wave of claim 1 double integrator curve-fitting type increment modulation method, it is characterized in that f
△" code stream (t) is determined by following method: the squiggle f(t that investigates former key sound) at f(tn) to f(t
N+m) one section, m 〉=1, note f " (t) mean value at this section is f " is (t).
(1) if f " (t)=f
△" (t
n), and f(t
n)=f
△(t
n), then be encoded to 0101 ... total m the sign indicating number of 01(), f
△" remain the f Δ " (t), t
n≤ t≤t
N+m;
(2) if f
△" (t
n)<f " (t)<f
△" (t
n)+△, then f
△" (t) remain f earlier
△" (t
n) one section constant, increase to f then
△" (t
n)+△, and keep it, make and satisfy f
△" (t)=f " (t), t
n≤ t≤t
N+m, be encoded to ... 01011010 ... (m altogether).
(3) if f
△" (t
n)-△<f " (t)<f
△" (t
n), f then
△" (t) remain f earlier
△" (t
n) one section constant, be reduced to f △ then " (tn)-△, and keep it, make and satisfy f
△" (t)=f ' (t), tn≤t≤t
N+m, be encoded to ... 010100101 ... (m sign indicating number altogether).
(4) if virgin curve f(t) increase sharply or reduce t
n≤ t≤
N+m, then be encoded to ... 1111 ... (m 1 yard) or ... 0000 ... (m 0 yard).
3, a kind of sounding circuit corresponding to pianotron sound wave double integrator curve-fitting type increment modulation method, comprise system controlled by computer maincenter 1, keyboard key-position and dynamics testing circuit 2, pedal testing circuit 3, signal superposition circuit 4, acoustics circuit 5, address control circuit 6, read-only memory (ROM) 7, channel group 8 between ROM7 and the circuit 4, channel group is made up of 6-8 passage, and each passage is by reading writing memory (RAM) 9, incorporates into to go here and there out circuit 10, double integrator curve-fitting type delta modulation demoder 11, amplification quantity is selected circuit 12, address control circuit 13, and clock circuit 14 is formed by connecting.It is characterized in that above-mentioned demoder 11 by 1: 1 phase inverter Ke, electronic switch 20, integrator 16, integrator 17, add subtract counter 15, decision circuitry line 24 is formed, wherein:
(1) integrator 16 is by electronic switch 21, electronic switch 22, resistance R
10-R
116, capacitor C
10-C
17With amplifier K
1Form, electronic switch 21 is one 16 and selects one switch, selects R
10-R
115In some resistance insert integrator 16, R
17Do not insert, electronic switch 22 is eight either-or switch, C
10-C
17Numerical value is that scale-of-two is arranged, and forms 1-256 times of C
10Capacitance;
(2) integrator 17 is by electronic switch 23, resistance R
20-R
28Capacitor C
2With amplifier K
2Form, electronic switch 23 is eight either-or switch; R
20-R
28Numerical value is that scale-of-two is arranged, and forms 1-256 times of R
20Resistance value;
(3) decision circuitry 24 is by AND circuit 25, not circuit 26, and biconditional gate circuit 27, d type flip flop 28 is formed;
(4) electronic switch 21 and phase inverter K
0Connect by electronic switch 20, switch 22 control ends connect data bus by latch 18, and the control end of switch 23 connects data bus by latch 19, and the D termination of d type flip flop 28 is incorporated into and gone here and there out the output terminal of circuit 10, CP termination clock circuit 14, the output terminal Q of add subtract counter 15
0, Q
1, Q
2, Q
3Connect the control end of switch 21, Q
3End also connects the control end of switch 20, the output terminal of CP termination AND circuit 25, and the M termination is incorporated into and is gone here and there out the output terminal of circuit 10, input end D
0, D
1, D
2, D
3Connect data bus.
Priority Applications (1)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
CN 89108538 CN1021091C (en) | 1989-11-09 | 1989-11-09 | Incremental regulation method according to sound wave form and circuit of electronic piano |
Applications Claiming Priority (1)
Application Number | Priority Date | Filing Date | Title |
---|---|---|---|
CN 89108538 CN1021091C (en) | 1989-11-09 | 1989-11-09 | Incremental regulation method according to sound wave form and circuit of electronic piano |
Publications (2)
Publication Number | Publication Date |
---|---|
CN1044183A CN1044183A (en) | 1990-07-25 |
CN1021091C true CN1021091C (en) | 1993-06-02 |
Family
ID=4857616
Family Applications (1)
Application Number | Title | Priority Date | Filing Date |
---|---|---|---|
CN 89108538 Expired - Fee Related CN1021091C (en) | 1989-11-09 | 1989-11-09 | Incremental regulation method according to sound wave form and circuit of electronic piano |
Country Status (1)
Country | Link |
---|---|
CN (1) | CN1021091C (en) |
Cited By (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
WO1997009713A1 (en) * | 1995-09-01 | 1997-03-13 | Shen, Xueliang | A method of processing audio signal for fidelity varying-speed replaying |
-
1989
- 1989-11-09 CN CN 89108538 patent/CN1021091C/en not_active Expired - Fee Related
Cited By (1)
Publication number | Priority date | Publication date | Assignee | Title |
---|---|---|---|---|
WO1997009713A1 (en) * | 1995-09-01 | 1997-03-13 | Shen, Xueliang | A method of processing audio signal for fidelity varying-speed replaying |
Also Published As
Publication number | Publication date |
---|---|
CN1044183A (en) | 1990-07-25 |
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