EP0237798B1 - Elektronisches Musikinstrument - Google Patents

Elektronisches Musikinstrument Download PDF

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Publication number
EP0237798B1
EP0237798B1 EP87102070A EP87102070A EP0237798B1 EP 0237798 B1 EP0237798 B1 EP 0237798B1 EP 87102070 A EP87102070 A EP 87102070A EP 87102070 A EP87102070 A EP 87102070A EP 0237798 B1 EP0237798 B1 EP 0237798B1
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EP
European Patent Office
Prior art keywords
sound data
memory
sound
basic
basic sound
Prior art date
Legal status (The legal status is an assumption and is not a legal conclusion. Google has not performed a legal analysis and makes no representation as to the accuracy of the status listed.)
Expired - Lifetime
Application number
EP87102070A
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German (de)
English (en)
French (fr)
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EP0237798A2 (de
EP0237798A3 (de
Inventor
Rainer Gallitzendörfer
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Individual
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Individual
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Publication of EP0237798A3 publication Critical patent/EP0237798A3/de
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    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
    • G10H7/00—Instruments in which the tones are synthesised from a data store, e.g. computer organs
    • G10H7/02—Instruments in which the tones are synthesised from a data store, e.g. computer organs in which amplitudes at successive sample points of a tone waveform are stored in one or more memories
    • G—PHYSICS
    • G10—MUSICAL INSTRUMENTS; ACOUSTICS
    • G10H—ELECTROPHONIC MUSICAL INSTRUMENTS; INSTRUMENTS IN WHICH THE TONES ARE GENERATED BY ELECTROMECHANICAL MEANS OR ELECTRONIC GENERATORS, OR IN WHICH THE TONES ARE SYNTHESISED FROM A DATA STORE
    • G10H2250/00—Aspects of algorithms or signal processing methods without intrinsic musical character, yet specifically adapted for or used in electrophonic musical processing
    • G10H2250/131—Mathematical functions for musical analysis, processing, synthesis or composition
    • G10H2250/215—Transforms, i.e. mathematical transforms into domains appropriate for musical signal processing, coding or compression
    • G10H2250/235—Fourier transform; Discrete Fourier Transform [DFT]; Fast Fourier Transform [FFT]

Definitions

  • the invention relates to an electronic musical instrument according to the preamble of claim 1.
  • a musical instrument is known from an information document on the synthesizer "PPG Wave 2.2", which was presented at the Frankfurt Music Fair in early 1982.
  • This synthesizer contains a waveform memory, which contains sample values of waveform or spectral profiles of a sound to be generated in a plurality of individually addressable memory locations and has a memory input and readout device, the sample values being organized in the memory in groups of waveforms, each of several Samples of waveforms exist and the memory readout device is dependent on stored sound parameters, such as Envelope curves for the sound, determines the order in which the individual memory groups or the basic sound data stored in them are read out.
  • the "wave tables" are run through depending on an envelope generator.
  • EP-A-0 115 117 shows an electronic musical instrument with sound generation based on waveform memory and with basic waveforms or basic sound data and with an interpolator which can interpolate between different data read from individual memory locations.
  • DE-PS 29 26 548 Another musical instrument is known from DE-PS 29 26 548. There, a waveform generator for sound shaping in an electronic musical instrument is described, which enables dynamic cross-fading from one stored sound to another stored sound. Other electronic musical instruments are described in the prior art recognized in DE-PS 29 26 548, namely DE-AS 22 37 594, DE-OS 28 30 483, DE-OS 28 30 482 and US-PS 38 59 884.
  • US Pat. No. 4,164,020 describes a programmable sound synthesizer which holds a large number of sound data in a memory.
  • the memory is read out by an address generator whose repetition frequency is controlled by an integrator.
  • the rate of integration is itself determined by a "number of tones".
  • several sound parameters such as frequency, waveform, envelope, velocity, decay, etc. can be entered. However, these sound parameters are an unchangeable part of the stored sound data.
  • the sound data is then read out cyclically in the numerical order of the addresses of the sound data entered. In short, only pre-programmed sound sequences with freely programmable, but no longer changeable sound character can be entered, so that one cannot speak of a musical instrument in the narrower sense, but rather of a "sound preserve" based on a record.
  • the object of the invention is to improve the electronic musical instrument of the type mentioned in such a way that it is absolutely freely programmable and thus produces any sounds that can also be changed while the instrument is being played.
  • the basic idea of the invention is therefore to store certain basic sound data and to assign the memory addresses to freely selectable sound parameters.
  • the sound parameters such as B. Velocity, duration of pressing a key, position of a controller, etc. thus determine the memory address to be read out or the sequence of the memory addresses to be read out.
  • the input of the basic sound data can be "synthetic", i. H. via a keypad with the help of a screen, which can also be used to create any kind of art sounds; but it can also be done via a microphone, so that the musical instrument according to the invention bridges the gap between the pure synthesizers and the pure sampling devices.
  • the musical instrument has a central processing unit (CPU) 1, a read-only memory (ROM) 2, a random access memory (RAM) 3, these three modules essentially taking over the sequence control. These components are connected to one another and to further components via lines 25 and 26, one line 25 representing the system data bus and the other line 26 representing the address bus.
  • a screen 5 is connected to these two lines via a screen interface (interface) 4, a keypad (keyboard) 7 via a further interface (interface) 6 and a keyboard 9 via a further interface 8.
  • Both lines 25 and 26 are also connected to inputs of a counter 11, which receives pulses from a clock generator 12.
  • the output of the counter 11 is connected to a waveform memory 14 via an address bus driver 13.
  • the memory 14 is also a random access memory (RAM).
  • an interpolator 15 and a Fourier transformer 16 are connected to both lines 25 and 26. The latter is a commercially available device that performs a Fourier transform on its input signals.
  • the outputs of modules 15 and 16 are also connected to waveform memory 14 via lines 27 and 28, line 27 forming the waveform memory data bus and line 28 the waveform memory address bus.
  • a driver circuit 10 is also connected between lines 27 and 25.
  • a line connection comprising a data latch 17, a digital / analog converter 18, which also contains a low-pass on the output side, an amplifier 19 and a loudspeaker or headphone 20 is connected to line 27. Furthermore, a line connection comprising a microphone 24, a preamplifier 23, an analog / digital converter 22 with an upstream anti-aliasing low-pass filter and a driver 21 is connected to line 27.
  • FIG. 1 The recording or storage of sounds or waveforms is first described with reference to FIG. 1.
  • the user Under control of the keypad 7, which can also contain a so-called joystick, a "mouse” or a light pen, the user generates any waveforms in an input step on the screen, which are here "basic waveforms".
  • An example of such a waveform is shown in FIG. 3.
  • the Y axis is the amplitude of the sound in this first input step, while the X axis is either time or frequency. Either the time profile of a signal or the spectral profile can be specified.
  • this first waveform is stored in the waveform memory 14 if the time profile is input, specifically in a first memory area from the address $ 0000 according to FIG. 4. In a corresponding manner, further ones are then stored Generated basic sound data, which are stored in further memory areas - in FIG. 4 the memory areas $ 1000, $ 2000, $ 3000 ... are stored.
  • the storage is not carried out directly but via the Fourier transformer 16, which first transforms the data entered in the spectral range into the time range.
  • All individual input basic sound data have a constant word length in the waveform memory 14, for example 256 words of 8 bits per waveform.
  • this interpolation or extrapolation results in smooth or dynamic transitions from one input basic waveform to the next input basic waveform. This enables a "dynamic" cross-fading from one sound to another sound.
  • a further “curve” is generated via the keypad 7 and the screen 5, which represents the sound parameter “time” and which determines the sequence in which the individual memory areas are read out.
  • all curve shapes are generally possible.
  • one axis eg Y axis
  • the other axis X-axis
  • a readout cycle then gives a continuously, ie dynamically changing, sound image between at least two input basic sound data.
  • a readout cycle can also extend over more than two basic sound data and the interpolation values in between or the extrapolation values lying outside the range between two adjacent basic sound data. If, on the other hand, you enter a horizontal line, for example, only one sound datum (with 256 words or "samples", for example) is read out, but several times in succession. If you enter a triangle curve, sound data are output in succession with an ascending address number and then with a descending address number. On the other hand, if the curve is very steep, depending on the slope, individual addresses are skipped when reading out. Of course, non-linear functions can also be entered as a curve for reading out.
  • At least one further sound parameter is defined.
  • parameters can be the velocity of the key on the keyboard, the pitch of a tone or another position of a control.
  • the choice of these parameters is based on the knowledge that with many natural musical instruments with changing volume not only the amplitude of the sound produced changes but also the sound character. same for also correspondingly for the pitch, at which not only the frequency of the generated tones changes with many instruments, but also the sound character. This can be explained, for example, by the fact that the body of many instruments has certain natural resonances or that it generates certain non-harmonic vibrations at different pitches and / or volumes.
  • the Y-axis can map the "parameter function” while the X axis maps the parameter itself.
  • the "parameter function” determines the address of the sound data to be read out. In the second exemplary embodiment, it also determines the sound data between which interpolation is to take place and the interpolation step size.
  • Said waveform memory 14, as shown in FIG. 5, is organized as follows.
  • Each of the individual fields shown at the top left in FIG. 5 (here with the addresses 00 to 0F) contains a basic waveform as the basic sound data corresponding to FIG. 3, for example 256 words each.
  • Each column with the fields 00, 01, 02, 03 or 04, 05, 06, 07 or 08, 09, 0A, 0B or 010, 0D, 0E, 0F then contains a "waveform set" corresponding to the 4 explained in connection with FIG.
  • a "waveform set” denotes a large number of related waveforms that are completely read out in a normal read cycle will.
  • parameter 1 then broadly designates the "time" at which the individual fields are read out.
  • the respective column is then set via the adjustment of parameter 2, for example the column with fields 04, 05, 06 and 07.
  • parameter 3 a jump is made to a further "block" which comprises the 16 fields 10 to 1F; 20 to 2F, etc. contains.
  • This parameter 3 can denote, for example, the position of a hand controller.
  • parameter 4 is used to determine the velocity of the individual keys on the keyboard.
  • different instruments e.g. violin, piano, flute etc.
  • other sound effects can be set.
  • the memory organization shown in FIG. 5 realizes a four-dimensional data field.
  • this memory organization can also be used to create an n-dimensional data field, which is particularly useful if you want to introduce even more sound parameters, such as a tremolo, an echo or reverberation, an increase in the amplitudes of certain frequency ranges, etc.
  • the sounds are recorded via the microphone, a piano being recorded.
  • Several (four in the example) pitch ranges are defined for the "Pitch" parameter.
  • a key in the first pitch range is then pressed with a first velocity, the sound waves that arise scanned and digitized and stored under the memory address 00.
  • the same key is then pressed with a different velocity and the digitized sound is stored under the memory address 40.
  • a key with the (four) different velocities is then pressed in the second pitch range, the basic sound data recorded thereby being stored under the addresses 04, 44, 84 and C4.
  • the first line of the matrix of FIG. 5 with the fields 00, 04, 08, 0C, 40 ... 4C, 80 ... 8C, C0 ... CC is then stored.
  • not all sound data in the fields lying side by side with regard to the different parameters also have to be basic sound data. Rather, intermediate values can also be determined here by interpolation. With regard to parameter 2 of FIG. 5, fields 00, 04, 08 and 0C lie next to one another.
  • parameters No. 2, 3 and 4 determine the start address of a row of fields lying next to each other with regard to the remaining parameter 1.
  • parameter # 1 is the "time”. It thus determines the sound character of a sound, which changes dynamically over time, with parameters 2, 3 and 4 initially held in mind.
  • the fields 00, 01, 02, 03 or 08, 09 then become one after the other , 0A, 0B etc. filled with corresponding sound data.
  • the four larger blocks are then assigned the start addresses 00, 40, 80 and CD, with parameter 3 or the controller in its first position.
  • the same process can then be repeated with other controller positions, it being up to the user what function he assigns to parameter 3 or the controller.
  • another instrument can be recorded in the second controller position, parameters 1, 2 and 4 being varied accordingly.
  • artificial sounds can be generated and stored using the keypad 7 and the screen 5.
  • the waveform memory is then read out using the keyboard.
  • This reports (for example, by very quick, cyclical polling of the switching status of the switches assigned to the individual buttons), which button is pressed and the speed at which this was done. This can be measured, for example, in that switching contacts are actuated sequentially when the button is pressed, the time between the successive actuation of the switching contacts being measured and as a measure of the velocity serves.
  • the parameters No. 2 and No. 4 (according to FIG. 5) are thus defined.
  • the other two parameters can be preselected via the keypad 7 or switches, levers, etc. attached to the keyboard 9. With the definition of the parameters, it is then also clearly established which sound data stored in the waveform memory 14 are to be read out.
  • the pitch or frequency is determined by the readout speed, that is by the clock frequency with which the data stored in the memory is read out.
  • Each key is assigned its own readout frequency or clock frequency.
  • the counter 11 can be used as a frequency divider which, depending on the key pressed on the keyboard 9, reduces the (constant) clock frequency generated by the clock generator 12 and with this clock frequency the driver 10 for reading out the sound data from the waveform memory 14 controls.
  • the sound data read out from the waveform memory 14 passes via the data latch 17, which serves as a buffer memory, to the digital / analog converter 18, where they are converted into analog signals and through a low-pass filter built into the digital / analog converter 18 be filtered or smoothed. From there they reach the loudspeaker 20 via the amplifier 19.
  • the "curve” described above, which controls the reading and - more abstractly - determines the "time” parameter, is preferably stored in RAM 3. It is also possible to store them in the waveform memory 14, but then additional memory areas not shown in FIG. 5 must then be provided and either ensured by additional measures It must be ensured that several memory areas can be read out simultaneously or that the memory areas provided for the "time" parameter, which ultimately contain addresses for reading out the waveform curves, can be read out and buffered.
  • the individual stored sound data and sound parameters are arranged hierarchically for input and output.
  • the highest hierarchical level are the sound parameters, for example for velocity, pitch or control position. They assign a parameter to certain addresses of sound data.
  • the X-axis can represent the respective parameter and the Y-axis the associated addresses of sound data.
  • this "curve” for reading out the sound data.
  • this "curve” also defines the memory address under which the individual basic sound data are stored and thus also the size of the gaps between two basic sound data and finally the length of a waveform set or one Sound data set.
  • the X axis corresponds to time and the Y axis corresponds to the address of the individual sound data.
  • Such a curve can have a length of 256 words, for example, which then corresponds to 256 memory addresses.
  • the sound data is then stored in the third (lowest) hierarchy level.
  • the interpolation or extrapolation cannot only be carried out between the basic sound data. Rather, it is also possible to interpolate or extrapolate the sound parameters. In particular with the sound parameter "velocity", the interpolation will be carried out according to an exponential function.
  • FIG. 6 shows a more detailed block diagram of the embodiment of the invention according to Fig. 1.
  • the same reference numerals as in Fig. 1 designate the same parts.
  • the embodiment of FIG. 6 differs from FIG. 1 only by the addition of bus drivers and registers and by the specification of specific, commercially available components for realizing the invention.
  • the interpolator 15 represents a computer subsystem with an independent microcomputer with CPU, ROM and RAM, which is described, for example, in the applicant's US patent 4,348,929.
  • the interpolator 15 is connected to the system bus 25, 26 via bus drivers 31 and 32.
  • the bus driver 31 works unidirectionally and is responsible for the addresses, while the bus driver 32 works bidirectionally and ensures the data transmission.
  • the drivers are controlled by the CPU 1 via D registers 36, 37. Its input “G” switches the driver output into its low-resistance state, the input "DIR” controls the direction of the data transmission; it is connected to the control line "R / W" (Read / Write) of CPU 1. Control is also carried out in conjunction with the D registers.
  • the CPU 1 (Fig. 1) transfers an 8-bit data word to the D register.
  • the corresponding control signals are activated depending on the state (set or reset) of the bits (D0 ... 7 that control the flip-flops Q0 ... 7).
  • An address decoder 38 initiates access to the interpolator 15.
  • the signal on the control line R / W specifies the data direction. Since the data driver may only be activated briefly during a CPU bus cycle, its "G" line is controlled by the address decoder.
  • the data and address lines from the interpolator 15 are also connected to the local RAM bus 27, 28 via second drivers 33 and 34. Here, too, control is again via D registers of type 74 LS 374 (D register 37).
  • the Fourier transformer 16 is also connected to the system bus 25, 26 via bus drivers 41 and 42 and to the RAM 14 via bus drivers 39 and 40.
  • the Fourier transformer likewise constitutes a computer subsystem in the form of an independent microcomputer A circuit board of the type MOS FFT from MEDAV, D-8520 Buckenhof can be used for this purpose, for example.
  • the programmable counter 11 has only address outputs A0 ... 23. It is programmed by CPU 1 via bus 26 (inputs D0 ... 7), its internal registers being addressed via A0 ... 5.
  • the frequency divider output "Prescaler Output” hosts a sound output of the waveforms from the RAM 14 to the D register 17 in synchronism with the pitch.
  • Control input is CLK input in both cases. If the signal at input OC is low, the register outputs are activated.
  • the CPU 1 (FIG. 1) has direct access to the RAM 14 via the drivers 35 and 10.
  • the embodiment of FIG. 2 is similar to that of FIG. 1 in the structure of the block diagram. However, the following differences exist:
  • the waveform memory 14 is here a dual-port RAM which contains both the sound parameters and individual sound data. In this case, only the basic sound data are stored for the sound data, while the interpolation or extrapolation is carried out during sound reproduction and thus virtually in "real time".
  • the RAM 14 of FIG. 2 therefore no longer contains interpolated or extrapolated values.
  • the interpolation or extrapolation is carried out by signal processors 31 and 32 which are connected to the RAM 14 via lines 29 and 30 and which are also connected to the lines 25 and 26.
  • One signal processor 31 processes sound data whose spectral properties change with the pitch.
  • the other signal processor 32 processes all those sound data that do not change with pitch (e.g. blowing noise, resonances, etc.). Both signal processors 31 and 32 each contain a digital / analog converter, which converts the digitally processed signals into analog signals. The analog outputs of the signal processors 31 and 32 are fed to an analog adder 33, which contains a low-pass filter on the output side. From there they reach the loudspeaker 20 via the power amplifier 19.
  • counter 11 in the example of FIG. 2 is a programmable up / down counter and that drivers 10 and 21 are tristate drivers, respectively.
  • the other components of FIG. 2 correspond to those of FIG. 1. In terms of effectiveness, this results the following differences in the embodiment of FIG. 2.
  • the matrix structure of the RAM 14 shown in FIG. 5 is implemented in two ways. One part contains the waveforms that represent the spectral components that change with pitch. The other part contains the curves of the spectral components, which are independent of the pitch.
  • a curve is read from the first part at the speed corresponding to the respective pitch.
  • a curve is read out from the second part with a pitch that is independent of the pitch or at least different than the speed of part 1, in which case both signals are added.
  • the RAM 14 is designed here as a dual-port RAM is therefore that the reading from the two memory parts takes place simultaneously.
  • the main reason for choosing the dual-port RAM is, however, that the CPU 1 with the memories 2 and 3 have access to one port and the signal processors 31 and 32 have access to the other port.
  • Parameter values and status information of the keypad 7 and the keyboard 9 can be input and output via the one port, sound data also running through this port during the sound input (e.g. recording). The sound data then runs to the signal processors 31 and 32 via the other port during playback.
  • the sound waves generated by the instrument are sampled and digitized, with at least two recordings having to be made here, namely one in the low and one in the high instrument sound range.
  • the signal processors then carry out a Fourier transformation for both wave trains with subsequent amount formation. This is followed by a comparison of the two spectra magnitude values. For this purpose, the minimum spectral distance is determined, for example. H. the smallest distance to be observed between two spectral lines according to the resolving power. Then the two absolute values of the spectra are subtracted from each other. The tone-dependent difference is assigned to the first memory section; the rest of the second part of the memory. After a Fourier inverse transformation of the spectral components assumed to be in phase, for example, two sets of waveforms are then available.
  • the readout speed of part 1 is not proportional to part 2 during playback, rather the readout speed for part 2 can even change, the sound image of the instrument can be distorted will.

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  • Engineering & Computer Science (AREA)
  • General Engineering & Computer Science (AREA)
  • Physics & Mathematics (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Electrophonic Musical Instruments (AREA)
  • Organic Low-Molecular-Weight Compounds And Preparation Thereof (AREA)
EP87102070A 1986-02-14 1987-02-13 Elektronisches Musikinstrument Expired - Lifetime EP0237798B1 (de)

Applications Claiming Priority (2)

Application Number Priority Date Filing Date Title
DE19863604686 DE3604686A1 (de) 1986-02-14 1986-02-14 Elektronisches musikinstument
DE3604686 1986-02-14

Publications (3)

Publication Number Publication Date
EP0237798A2 EP0237798A2 (de) 1987-09-23
EP0237798A3 EP0237798A3 (de) 1989-08-30
EP0237798B1 true EP0237798B1 (de) 1996-11-06

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EP87102070A Expired - Lifetime EP0237798B1 (de) 1986-02-14 1987-02-13 Elektronisches Musikinstrument

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US (1) US5298672A (enrdf_load_stackoverflow)
EP (1) EP0237798B1 (enrdf_load_stackoverflow)
AT (1) ATE145083T1 (enrdf_load_stackoverflow)
DE (2) DE3604686A1 (enrdf_load_stackoverflow)

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US5342990A (en) * 1990-01-05 1994-08-30 E-Mu Systems, Inc. Digital sampling instrument employing cache-memory
US5365467A (en) * 1992-12-25 1994-11-15 Yamaha Corporation Signal processor for providing variable acoustic effect
US6072474A (en) * 1995-08-11 2000-06-06 Sharp Kabushiki Kaisha Document processing device
JPH11289169A (ja) * 1998-04-03 1999-10-19 Nec Shizuoka Ltd 電子機器の情報表示窓
JP2001075565A (ja) 1999-09-07 2001-03-23 Roland Corp 電子楽器
JP2001125568A (ja) 1999-10-28 2001-05-11 Roland Corp 電子楽器
AT500124A1 (de) * 2000-05-09 2005-10-15 Tucmandl Herbert Anlage zum komponieren
GB2425526A (en) * 2005-04-27 2006-11-01 Cnh Belgium Nv Bale wrapping apparatus

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EP0237798A2 (de) 1987-09-23
ATE145083T1 (de) 1996-11-15
DE3604686C2 (enrdf_load_stackoverflow) 1988-09-22
DE3751941D1 (de) 1996-12-12
US5298672A (en) 1994-03-29
DE3604686A1 (de) 1987-08-27
EP0237798A3 (de) 1989-08-30

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