EP0448010A2 - Verfahren zur Klangerzeugung mit einem elektronischen Musikinstrument und elektronisches Musikinstrument - Google Patents

Verfahren zur Klangerzeugung mit einem elektronischen Musikinstrument und elektronisches Musikinstrument Download PDF

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Publication number
EP0448010A2
EP0448010A2 EP91104137A EP91104137A EP0448010A2 EP 0448010 A2 EP0448010 A2 EP 0448010A2 EP 91104137 A EP91104137 A EP 91104137A EP 91104137 A EP91104137 A EP 91104137A EP 0448010 A2 EP0448010 A2 EP 0448010A2
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EP
European Patent Office
Prior art keywords
sound
frequency
interpolation filter
musical instrument
features
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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.)
Withdrawn
Application number
EP91104137A
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German (de)
English (en)
French (fr)
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EP0448010A3 (enrdf_load_stackoverflow
Inventor
Reinhard Franz
Christian Dipl.-Phys. Scheidegger
Heino Dipl.-Ing. Wachter
Udo Dipl.-Inform. Spindeler
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Wersi GmbH and Co
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Wersi GmbH and Co
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Publication date
Priority claimed from DE19904008872 external-priority patent/DE4008872C2/de
Priority claimed from DE19904008873 external-priority patent/DE4008873A1/de
Priority claimed from DE19904008875 external-priority patent/DE4008875C1/de
Application filed by Wersi GmbH and Co filed Critical Wersi GmbH and Co
Publication of EP0448010A2 publication Critical patent/EP0448010A2/de
Publication of EP0448010A3 publication Critical patent/EP0448010A3/xx
Withdrawn legal-status Critical Current

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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
    • G10H1/00—Details of electrophonic musical instruments
    • G10H1/02—Means for controlling the tone frequencies, e.g. attack or decay; Means for producing special musical effects, e.g. vibratos or glissandos
    • G10H1/06—Circuits for establishing the harmonic content of tones, or other arrangements for changing the tone colour
    • G10H1/12—Circuits for establishing the harmonic content of tones, or other arrangements for changing the tone colour by filtering complex waveforms
    • G10H1/125—Circuits for establishing the harmonic content of tones, or other arrangements for changing the tone colour by filtering complex waveforms using a digital filter
    • 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
    • G10H7/08—Instruments in which the tones are synthesised from a data store, e.g. computer organs by calculating functions or polynomial approximations to evaluate amplitudes at successive sample points of a tone waveform
    • 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
    • G10H2250/541—Details of musical waveform synthesis, i.e. audio waveshape processing from individual wavetable samples, independently of their origin or of the sound they represent
    • G10H2250/621—Waveform interpolation

Definitions

  • the invention relates to a method for sound generation with an electronic musical instrument, an electronic musical instrument and a device for loading a storage device for an electronic musical instrument.
  • the musical instrument can be designed, for example, as a keyboard musical instrument (keybord) or as a pure sound generation module (expander), which can be controlled via external signals.
  • sounds are reproduced, that is to say, for example when manufacturing a corresponding musical instrument.
  • sound patterns are stored digitally in the form of sampled values in a storage device.
  • the sounds are generated, for example during a lecture by a musician, the samples are read from the memory device, processed, converted to digital-analogue and reproduced as sounds via an audio device.
  • sample Rate Conversion a method which is known under the name “Sample Rate Conversion” (see, for example, Chamberlin "Musical Applications of Microprocessors", pages 470 to 477, Haydn Book Company, Inc., 1980).
  • Sample Rate Conversion it is not possible to produce more than three adjacent semitones from a single "sample”, ie sound pattern which is represented by the sequence of samples without major losses in quality. If you want to recreate a piano sound, ie the sound of a piano or grand piano with a length of 88 keys, about 30 samples or sound patterns are required. The sound of a piano is not only dependent on the pitch, but also on the velocity.
  • the mathematical operation on which the sample rate conversion is based can be implemented by means of a digital interpolation filter, ie the interpolation in the time domain can also be viewed as filtering in the frequency domain.
  • the most suitable filter function for this purpose is a low-pass filter, which allows everything to happen up to half the sampling frequency with which the sound pattern has been sampled, but beyond which it completely suppresses all frequency components. The cutting edge should therefore run practically perpendicular to the blocking frequency.
  • an ideal filter cannot be realized.
  • low-pass filters can be approximated quite well if a filter with a large number of filter poles is used. The higher the number of filter poles, the better the approximation to the ideal filter characteristic.
  • a large number of poles in the digital case has the disadvantage that a predetermined number of arithmetic operations is required per pole, that is to say, for example, an addition and a multiplication per filter pole, so that with many filter poles a correspondingly large number of mathematical operations must be performed by the musical instrument relatively slow despite a lot of effort.
  • This disadvantage is particularly evident when the musical instrument is to be operated polyphonically, ie when a large number of different sound patterns are to be reproduced at the same time.
  • the many mathematical operations need to be performed not only for one sound pattern, but in parallel for a whole series of sound patterns. Even when using extremely fast components, the musical instrument will sooner or later reach a limit beyond which an expansion of the variety of sounds is no longer possible, especially in view of the fact that very large memories have to be used.
  • This object is achieved in a method for sound generation with an electronic musical instrument, in which sound patterns are stored digitally in the form of samples in a preparation section and the samples are read out from the memory device in a sound generation section are digitally interpolated and converted from digital to analog in an interpolation filter, whereby attenuation below the blocking frequency of the interpolation filter is permitted in the digital interpolation and the higher-frequency components of the sound patterns that are attenuated in the interpolation filter are amplified in the preparation section before storage will.
  • a compromise in the filter property is chosen, namely an attenuation is permitted in a range that is actually still the pass band. Frequency components that fall within this range are undesirably heavily damped. The damping becomes stronger the closer the frequencies come to the blocking frequency. For this reason, filters with a relatively steep cut-off characteristic have so far been chosen.
  • the attenuation can be compensated for by a pre-emphasis, which increases or increases the higher-frequency components before the sound patterns are saved. During the interpolation, these parts are damped accordingly, so that despite the poor filter properties, a sound pattern is available at the output of the interpolation filter that practically corresponds to the original.
  • the amplification of the higher-frequency components is preferably carried out before storage with a frequency-dependent gain characteristic which is inversely proportional to the frequency-dependent pass characteristic in the pass band.
  • a frequency-dependent gain characteristic which is inversely proportional to the frequency-dependent pass characteristic in the pass band.
  • the frequency spectrum of the sound patterns is limited to frequencies below a cutoff frequency, the cutoff frequency being less than the cutoff frequency of the interpolation filter.
  • the sound patterns are limited to the audible spectrum, the so-called audio spectrum.
  • the sound patterns are put together from individual sound features, which are stored digitally in the form of sample values, are read out in a controlled manner, are subjected to a sampling rate conversion according to which all sound features are available with a uniform system sampling rate, and are then combined, at least some sound features being components of different types Have frequency, all sound features at the start time of a sound have a predetermined phase relationship to one another and the components of a sound feature at this start time have a predetermined phase relationship to the other components of the same sound feature.
  • This embodiment can also be implemented without the features of the preemphasis.
  • a sound feature is therefore a frequency mix that is stored in the memory in the form of a predetermined number of samples.
  • All sound features preferably have a zero crossing at the start time. This facilitates the establishment of the desired phase relationship.
  • this structure essentially corresponds to the model of a natural musical instrument. With conventional musical instruments, too, the tone begins with an amplitude value of zero.
  • all components of a sound feature have a zero crossing at the start time.
  • the amplitude of a component can develop positively or negatively after the start time. If the amplitude develops negatively in the time range after the start, i.e. the first derivative after the time is negative, this means that the relevant component of the same frequency, which initially assumes an amplitude greater than zero after the start time, is out of phase by 180 °. Such a component can be represented in the frequency spectrum with a negative amplitude. A sound feature with such a component will completely or partially cancel out the component of another sound feature with the same frequency.
  • the "negative" amplitude is of course only used here as a calculation parameter, since the amount of the amplitude is decisive for the ear.
  • Each sound feature is preferably stored with a number of samples which has a predetermined relationship with the highest frequency occurring in the sound feature, the blocking frequency of the interpolation filter being selected depending on the sampling frequency with which the samples were generated from the sound patterns and preferably in on the order of 50% to 60% of the sampling frequency. In this area you have the greatest certainty that no aliasing will occur. It is advantageous that the cutoff frequency is selected in the order of 30% to 50% of the sampling frequency. In this way, practically any blocking or limit frequencies can be realized, so that the characteristics of the interpolation filter also influence the reproduction of the sound pattern is possible.
  • the so-called alias effect can be suppressed by a clever choice of the blocking frequency of the interpolation filter or the cut-off frequency, which causes undesirable disturbances in the reproduction of the sound pattern, especially at high frequencies. It is therefore not necessary to sample all sound features with the same, high sampling frequency, that is to say the system sampling rate, and to store the resulting high number of sampling values. Rather, the number of samples is limited to what is absolutely necessary.
  • the sound features are advantageously subjected to an amplitude control before the assembly.
  • a sound feature that predominantly contains overtones can be amplified in one case to produce a sound that is rich in overtones, but in another case can only be used with a low amplification, so that the overtones are not so clearly audible.
  • different sounds can already be generated by the amplitude control. This also applies to a single sound to show its change over time.
  • the amplitude control is preferably carried out with the aid of digitally stored amplitude envelopes. Changing frequency mixes can then also be realized in one sound. This can lead to interesting beat effects.
  • the amplitude envelopes are composed of envelope characteristics.
  • the technology, the similarities in the individual curves or storing vibrations only once and using them multiple times can also be used for the envelopes.
  • both an amplification and an attenuation of frequency components in the resulting signal can preferably be achieved.
  • This amplification or weakening can be achieved on the one hand by simply superimposing the sound characteristics. If a sound feature has one of the above-mentioned components with negative amplitude, the corresponding frequency, if it also occurs in the other sound feature, will be attenuated. The same effect, but not limited to a single frequency, can be achieved by providing the sound feature with a negative amplitude envelope.
  • the term negative amplitude envelope is also used here for illustrative purposes only introduced. This is to express that sound features that have a negative amplitude envelope are not added to another sound feature when they are joined, but are subtracted from it.
  • the reading out and the sampling rate conversion of a plurality of sound features are carried out by means which are common to a plurality of sound features and which process the sound features in series or in the multiplex method.
  • This can be easily achieved due to the high processing frequencies in relation to the frequencies of the sound features. Strictly speaking, all processing only has to be completed within a period of the system sampling rate of, for example, 44.1 kHz, in order to provide the musical instrument with the instantaneous value of the sound to be reproduced. Assuming a system sampling rate of 44.1 kHz, this is more than 20 ⁇ s.
  • the sampling rate processing of a sample of another sound feature is carried out, which was read out in the previous access to the memory.
  • the calculation of individual values thus takes place in succession, which means that a certain pipelining can be implemented.
  • the individual sound features are preferably assembled with the aid of an accumulator in which instantaneous values of the sound features provided with the uniform system sampling rate are added or subtracted serially, the content being read out at a predetermined point in time. For the "filling" of the battery stands a complete period of the system sampling rate is available. It is therefore not necessary that all sound features of a sound are processed at the same time. Rather, this preferred embodiment opens up the possibility of serial processing of individual sound features in succession, the accumulator making it possible to assemble the individual sound features.
  • each sound is composed of a maximum of sixteen sound features. In many cases, two to eight sound characteristics will suffice. The decision on how many sound features to use is ultimately a question of hearing.
  • the limitation to sixteen sound characteristics means that the amount of data to be stored and managed is limited.
  • a plurality of sets of sound features are put together in parallel for a sound and are superimposed with opposite amplitude control curves.
  • This allows the sound of an instrument to be simulated in a simple manner, which would otherwise be difficult to simulate even with a multitude of sound features.
  • reference is made to a trumpet into which a damper is inserted during the blowing.
  • Another example is a clarinet or saxophone, the players of which greatly change the tension of the lower lip when playing.
  • an interpolation filter is used, the cut-off frequency of which changes automatically as a function of the sampling frequency. This saves storage space in the samples for the sound pattern. If For example, the chamber tone a has only a low harmonic content, for example, if the sound pattern comes from a flute that essentially only has the third harmonic (1760 Hz), a sampling frequency of approximately 3.5 kHz is sufficient to generate the samples. So only about eight samples per wave train. In this example, this results in a memory saving of more than 90%. When reproducing with the system sampling rate of the musical instrument of 44.1 kHz, for example, 100.2 samples must of course then be generated or calculated again. In this context, the advantage of automatically adjusting the cut-off frequency of the digital interpolation filter is particularly evident.
  • the limit frequency is set at a sampling frequency of 3.5 kHz, for example, to a value in the order of 1.4 kHz.
  • the invention is also based on the object of specifying an electronic musical instrument for generating sounds, in which the sounds are reproduced as faithfully as possible with little effort.
  • This object is achieved in an electronic mustikinstrument for generating sounds with a storage device in which sound patterns are stored digitally in the form of samples, a readout device which reads the samples in a controlled manner from the storage device, a digital interpolation filter which carries out a sampling rate conversion and at its output provides the sound characteristics with a system sampling rate that is uniform for the entire musical instrument, and a digital / analog converter, which is connected to the output of the interpolation filter, in which the interpolation filter has a pronounced attenuation below the cutoff frequency, and the samples sound patterns correspond, the higher-frequency components that are attenuated in the interpolation filter are amplified.
  • the musical instrument therefore has the information it needs to generate the sounds stored in it.
  • the readout device reads out the stored samples, which are then processed subsequently. Because the stored values have a different frequency spectrum than the values to be reproduced - namely, according to the invention they are amplified in the upper frequency ranges - the interpolation filter with the pronounced attenuation can be used without problems, without the output of the interpolation filter or the downstream digital / analog Unwanted distortion can be heard.
  • the musical instrument preferably has an accumulator, which is connected to the output of the interpolation filter and adds up values which have been read in serially and outputs the sum, the addition and output taking place within a period of the system sampling rate, and an audio device which produces the output signal of the digital / Makes the analog converter audible.
  • the individual sound features can be overlaid because the interpolation filter ensures a sampling rate conversion.
  • the sound features are still available in digital form at the output of the interpolation filter, but they are all sampled there at the system sampling rate, i.e. within each period of the system sampling rate, an instantaneous value of each sound feature is stable and available for further processing.
  • the individual instantaneous values can therefore be superimposed in each period of the system sampling rate.
  • the output signal then contains the addition or subtraction of the individual frequencies, which leads to the frequency spectrum of the sound to be reproduced.
  • the interpolation filter preferably has 32 or fewer poles. Compared to the commonly used interpolation filters, which have an order of magnitude of 100 or more poles, this saves a considerable amount of computation and processing time and enables the read-out values to be processed quickly, since a smaller number of poles means that fewer computation operations are necessary. In the event that the filter has to be available for several sounds at the same time, for example has to perform the arithmetic operations for a polyphonic sound, a correspondingly greater variety of sounds can be generated simultaneously with the 32 or fewer poles specified. Compared to the otherwise usual hundred or more-pole filters, the variety of sounds can be practically more than tripled. In a particularly preferred embodiment, the interpolation filter has eight poles. A relatively broad transition range is obtained here, i.e. damping sets in relatively early. However, this effect can be easily compensated for by the pre-emphasis provided.
  • An amplitude control device is preferably arranged between the interpolation filter and the accumulator. This amplitude control device ensures that individual sound features have a stronger or weaker influence on the composition of the sounds.
  • readout device interpolation filter, accumulator and optionally amplitude control device are combined in one sound generation module and a plurality of sound generation modules are arranged in parallel.
  • a plurality of sound generation modules arranged in parallel can produce an incomparably larger number of sounds in parallel, without there being any restriction in terms of computing or processing time.
  • a further advantageous effect can be achieved if the individual sound features have a predetermined additional change in the phase shift with respect to one another. This makes it possible for the listener to feel that a large number of instruments are playing at the same time, like an orchestra.
  • the sound generating device preferably has at least one transputer for processing the sound features into sound signals.
  • a transputer such as that sold by the company Inmos, is a programmable circuit component that enables parallel processing of processes at high speed.
  • "Transputer” is an artificial word that emerged from the contraction of "transistor” and "computer”.
  • a transputer thus behaves like a computer on the one hand, but also physically like a semiconductor circuit element.
  • a transputer is a self-contained unit that manages processes internally and only leads the process inputs and outputs to the outside. Transputers allow the parallel processing of processes at high speed.
  • transputer It is irrelevant whether there are actually several units in the transputer that work in parallel, or whether in a special embodiment a unit, ie a single transputer, simulates the parallelism. This is due to the fact that each transputer has local memory that it can access. The transputer can therefore manage itself to a great extent without being dependent on the restrictions of a bus or a controller for an external memory, which may also have to be accessed by other processors. Another reason for the high speed in parallel processing is that the transputer has a very fast interface, a so-called link, through which it can communicate with other transputers, other interfaces or other components. In this case, only the end results of the processes, i.e.
  • a plurality of sound generation modules is preferably provided, the communication between the reading device and the sound generation modules being carried out with the aid of at least one transputer.
  • the advantages of a transputer can be seen here too.
  • the communication between the data input device and the modules has always been limited by the performance of a processor that could either accept many input signals or manage many modules at the same time.
  • transputers the data flow between the data input device and the modules of the processing device can now be controlled much better without being limited to assigning certain signal paths from the data input device to certain modules in the processing device. Rather, all modules can be connected to all elements of the data input device in order to serve the sound generation.
  • the transputer preferably forms at least one digital interpolation filter in each module.
  • the interpolation filter With the help of the interpolation filter, the temporal amplitude sequences of each sound can be simulated from the sampled values, which serve here as support points.
  • the use of the interpolation filter has the advantage that all sounds at the output of the interpolation filter with a uniform sampling rate for the entire musical instrument, the so-called system sampling rate.
  • the further processing of the data still available in digital form can then also be carried out in digital form without the need for conversion to analog form and back again, possibly even several times.
  • the interpolation thus carries out a sample rate conversion.
  • the digital interpolation filter consists of a series connection of several adders and multipliers.
  • transputer has the advantage that each addition and each multiplication can be understood and implemented as a separate process. Since the transputer can carry out several processes in parallel, the individual mathematical operations required for each pole can also be carried out in parallel. Successive values can then be calculated simultaneously in the form of a pipeline.
  • the transputer forms several parallel interpolation filters in each module, which process several sound patterns in parallel. Already with one module, several sounds can be generated in parallel without the need for time slice management, which assigns each interpolation filter only a fraction of the time available in a period of the system sampling rate.
  • the device is provided with a recording device for generating sound patterns in the form of electrical or magnetic signals from sound waves, a scanning device that scans the sound patterns at a sampling frequency at predetermined times and generates samples, and with a storage device that stores the samples in inscribes the memory device, provided, a pre-emphasis device being arranged between the recording device and the scanning device, which amplifies the sound patterns in predetermined frequency ranges.
  • the loading device therefore ensures in advance that the damping carried out by the digital filter is compensated again.
  • the pre-emphasis device amplifies the sound patterns in a frequency range that is below 50% of the sampling frequency.
  • the amplification therefore takes place in a relatively broad frequency band, so that the stored samples, if they are not read out by the electronic musical instrument, could probably not be recognized at all as belonging to the sound patterns.
  • the pre-emphasis device advantageously has a frequency-dependent gain characteristic which is essentially inversely proportional, in the best case even exactly inversely proportional, to the frequency-dependent pass characteristic of the interpolation filter.
  • the greater the attenuation in the interpolation filter as a function of the frequency the greater the amplification or the increase in the same frequencies in the pre-emphasis device.
  • the pre-emphasis device attenuates frequency components above the cut-off frequency.
  • the pre-emphasis device cuts off frequencies above the cut-off frequency or attenuates them considerably.
  • An electronic musical instrument generally has a keyboard 1, operating elements 2 and interfaces 3.
  • the keyboard which comprises one or more manuals with a range of four to eight octaves and / or a pedal, is used by the practicing musician to generate tone sequences. When you press a key, you usually hear a tone. Depressing several keys simultaneously causes the polyphonic generation of several tones.
  • the timbre can be set or changed using the controls 2. Signals from other devices, for example other electronic musical instruments, computers or storage media, can be coupled into the electronic musical instrument via the interfaces 3.
  • Keyboard 1, control element 2 and interfaces 3 are connected to an interface processor 4, which in turn has a memory 5.
  • the interface processor 4 manages the signals received from the keyboard 1, the operating elements 2 and the interfaces 3 and generates a suitable code which is passed on to a main processor 6.
  • the main processor 6 has a memory 7 in which, among other things, the processing instructions for the signals received by the interface processor 4 are stored. After the main processor 6 has processed the signals received by the interface processor 4, it sends 18 addresses and / or commands via a control bus, with the aid of which tone generation modules can generate 8-10 tones.
  • tone generating modules 8-10 can be provided. In principle, their number is only limited by the capacity of the main processor 6. Each tone generation module 8-10 is capable of generating one or more tones simultaneously.
  • the tone is generated digitally, with each tone generation module 8-10 being connected to a control curve memory 11 and to a sample value memory 12, both of which are common to all tone generation modules 8-10 are able to access.
  • the electronic musical instrument emulates other musical instruments, for example a piano, a string instrument or a wind instrument or also a drum set, by means of the tone generation modules 8-10.
  • the information for the sound is stored in the control curve memory 11 and the sample value memory 12.
  • the sound generating modules 8-10 generate digital signals, which they put on an audio bus 17. Also connected to the audio bus 17 are effect processors 13-15 which, if desired, subject the digital output signals of the tone generation modules 8-10 to a digital effect treatment, for example the generation of a reverb, a distortion, a vibrato or other effects.
  • FIG. 2 The structure of a sound generation module is shown schematically in FIG. 2.
  • all sound generation modules 8-10 have the same structure. You can also use the same operating instructions, i.e. Programs in the individual processors can be controlled.
  • the sound generation module 8 has a control processor 19 with memory 20, which receives information from the main processor 6 via the control bus 18.
  • the control processor 19 is connected on the one hand to a control curve processor 21 and on the other hand to a phase processor 22.
  • the control curve processor 21 accesses the control curve memory common to all tone generation modules 8-10, determines the value of a control curve at the required point by interpolation and, if necessary, calculates a final value from a plurality of such values, which it returns to the control processor 19.
  • the control curves are used to determine the frequency sequence, the volume sequence and the relative importance of individual components control the sounds to be generated for a variety of sound components.
  • the individual values can be calculated one after the other, the separation of memory access and processing permitting the simultaneous execution of both functions for successive calls.
  • control curve values can be stored in the control curve memory 11, for example, in the form of successive base values. If the ordinate distance of all base values is the same, only the abscissa values are saved. Otherwise, pairs of values from ordinates and abscissa can be saved.
  • the various tone generation modules 8-10 can access the control curve memory cyclically at specified time intervals in order to rule out access conflicts.
  • individual tone generation modules 8-10 may have different priorities, with the highest priority tone generation module having access to the cam memory 11 before anyone else, if need be.
  • the control processor 19 transfers its output data to a phase processor 22.
  • the phase processor 22 has the task of tracking the phase for each of the sound components for a fixed time interval from the information received from the control processor 19.
  • the phase processor calculates an address from the phase values and thus accesses the sample value memory 12.
  • So-called sound features are stored in the form of sample values in the sample value memory 12.
  • the samples can only include one period of a sound, so-called "waves”, but they can also include sound features of an entire tone, ie the tone from the beginning to the end, so-called “samples”. This can be useful, for example, if a tone changes irregularly from start to decay. This is the case, for example, with a drum cymbal sound.
  • Another example is the blowing behavior of pipe organs, which have a much richer frequency spectrum at the moment of blowing than in the steady state.
  • the data read from the sample value memory 12 with the aid of the addresses generated by the phase processor 22 are forwarded directly to an interpolation filter 23.
  • the calculation of the individual phase values can also take place here one after the other, with the separation of memory access by the phase processor 22 and processing by the interpolation filter 23 making it possible to carry out both functions simultaneously for successive values.
  • the interpolation filter 23 calculates amplitude values from the sample values stored in the sample value memory 12 at a number of support points specified by the phase processor. The support points are determined by a system sampling rate. After passing through the interpolation filter, all tones are available with a uniform system sampling rate. At the times determined by the sampling rate, the instantaneous values of the sounds to be digitally reproduced are therefore always available in the entire musical instrument. They can then simply be added or subtracted without having to pay attention to different phase relationships.
  • An amplitude processor 24, which is also controlled by the control processor 19, is connected to the output of the interpolation filter 23.
  • the amplitude processor 24 has the task of controlling the amplitude of the output signal of the interpolation filter 23. Since the output of the interpolation filter 23 is available in digital form, this means that the individual digital values are subjected to predetermined mathematical operations.
  • the amplitude can be amplified, for example, by multiplication by a factor greater than 1. A weakening takes place by multiplying by a factor smaller than 1. It is also possible to form a "negative" amplitude by multiplying by a negative factor. A negative amplitude is, of course, only to be understood here as a calculation variable, since a difference between a positive and a negative amplitude cannot be made audible.
  • An accumulator is connected to the output of the amplitude processor 24 and can also be controlled by the control processor 19.
  • the accumulator has the task of summing up digital signals which are supplied to it at successive points in time and passing them on to the audio bus 17 at a point in time following the individual summation points in time. Sounds can thus be “put together” in the accumulator 25.
  • An accumulator for each tone generation module is shown. However, it is also possible to provide only one accumulator for the entire musical instrument.
  • the accumulator can also be replaced by an adder, which adds up the digital quantities emerging at its inputs at predetermined times.
  • the sounds are stored in the form of sound features in the sample value memory 12.
  • a sound is composed of a plurality of individual sound features in the accumulator 25. At least one sound feature thereof has components of different frequencies. As a rule, however, most or even all of the sound features will consist of a frequency mix.
  • the individual sound features are stored in successive samples.
  • a special feature is that all sound features are stored with a predetermined phase relationship to one another. In the present exemplary embodiment, all sound features begin with a zero crossing of the amplitude. All frequency components also have a predetermined phase relationship to one another within the sound feature. It is also preferred here that all frequency components have a zero crossing at the start time. With this special regulation it is possible to easily overlay the individual sound characteristics and to create targeted overlay effects.
  • 3a shows a first sound feature with three frequencies f1, f3 and f7, each of which has the significance 80, 30 and 10. For example, this is a frequency spectrum with the fundamental wave f1 and the third and seventh harmonics f3 and f7. Such a sound feature does not necessarily have to evoke a memory of the instrument to be reproduced with the aid of this sound feature.
  • 3b shows a further sound feature which, however, does not contain a fundamental wave, but only the third and seventh harmonics f3 and f7. It is remarkable that the seventh harmonic is shown with a negative amplitude. This means nothing else than that the seventh harmonic f7 has a negative slope at the start time, ie its amplitude immediately less than zero immediately after the start time. It has a phase shift of 180 ° compared to a seventh harmonic, which has a positive slope at the start.
  • FIG. 3c shows the superposition of the two sound features from FIGS. 3a and 3b.
  • the positive frequency component f7 in FIG. 3a is canceled out by the negative sound component f7 in FIG. 3b.
  • the superimposition thus leads to a frequency spectrum which only contains the fundamental wave and the third harmonic, the amplitude of the third harmonic being the sum of the corresponding frequency components from the sound feature from FIG. 3a and the sound feature from FIG. 3b.
  • FIG. 3d shows a further composite frequency spectrum, in which only the sound features from FIGS. 3a and 3b have also been used.
  • the sound feature of FIG. 3b was subtracted from the sound feature of FIG. 3a for the frequency spectrum from FIG. 3d.
  • the third harmonic f3 is therefore only available with the difference.
  • the seventh harmonic f7 is available with the sum of its amplitudes, since the subtracting corresponds to a negative quantity by the addition of its amount.
  • the subtraction can take place, for example, by multiplying the output value of the interpolation filter 23 in the amplitude processor 24 by a factor ( ⁇ 1).
  • 3e shows a third frequency spectrum, which has also been generated from the two sound features according to FIGS. 3a and b.
  • 3a was read unchanged into the accumulator 25, the sound feature according to FIG. 3b was multiplied in the amplitude processor 24 by a factor of 0.5.
  • the frequency components of the third and seventh harmonics are only increased or decreased by an amount attenuated by a factor of 0.5.
  • the use of individual sound features is not necessarily limited to the reproduction of a sound from a single instrument.
  • all tones of an instrument to be reproduced have a certain commonality.
  • the frequency spectrum will not be the same for all tones.
  • the sound feature common to all tones of an instrument can be extracted and, in other sound features, only the differences to the individual tones, which will be different over the range of the instrument, can be additionally stored.
  • it turned out that it is also possible to find out individual sound characteristics that are the same for a group of instruments.
  • by cleverly selecting the sound characteristics practically all sound characteristics can be used for several tones or even sounds. In the present case, only two sound features have been combined for the sake of simplicity.
  • the individual sound features are stored in the sample value memory 12 in the form of sample values. Sound characteristics with only a few overtones are also stored with only a relatively few samples. No information about the highest frequency occurring in the sound feature can be derived from the sound feature itself.
  • the frequency with which the individual sound feature is reproduced results only from the sampling rate conversion in the interpolation filter 23. This should be made clear using an example shown in FIG. 4. Only the sample values marked with an x are stored in the sample value memory 12.
  • the interpolation filter 23 calculates intermediate values at positions predetermined by the phase processor 22, which are identified by a vertical line. These intermediate values are read into the accumulator at the system sampling rate.
  • the accumulator 25 sums up all the values that are applied to the input at discrete times within 22.7 ⁇ s. At the end of these 22.7 ⁇ s, the totalized values are then read out. 4, the sound according to FIG. A is read out within 15 periods of the system sampling rate, in FIG. 4b, however, within 30 periods. The tone according to FIG. 4b thus sounds an octave lower than the tone according to FIG. 4a. In a scale representation, the vertical lines should be much denser. However, the overview would suffer from this. For example, for a representation of a tone with the frequency 440 Hz, approximately 100 periods of the system sampling rate would have to be represented in the figure.
  • a single sound feature can therefore be used for different pitches of a single sound.
  • the frequencies stored in the sound feature only indicate the relationship of the frequencies, for example the relationship of a fundamental frequency to the harmonics. These relative frequencies are only converted into an absolute frequency spectrum by converting the sampling rate in the interpolation filter 23.
  • the information that is stored in a sound feature is limited to the most necessary. The number of stored samples corresponds to twice the highest frequency occurring in the sound feature.
  • Sound patterns that change significantly over time can be stored as sound feature sets that are mixed differently, i.e. time-dependent, with the aid of the amplitude control. It is also possible to crossfade from one sound to another.
  • the amplitude control curves are constructed so that their sum always remains constant. The individual amplitude control curves are therefore opposite.
  • the interpolation filter 23 carries out a "sample rate conversion", as is known, for example, from Chamberlin's "Musical Applications of Microprocessors".
  • the interpolation filter 23 can also be connected to the interface processor 4 via a bus.
  • This processor 4 supplies the interpolation filter 23 with information about the pitch to be generated, for example about the frequency with which the samples are to be reproduced from the memory device 7.
  • the frequency information is transmitted to the interpolation filter 23 in the form of phase information, ie the interpolation filter 23 receives the information via the interface processor 4 which phase distance the individual support points should have from one another.
  • the interpolation filter 23 carries out an interpolation between the individual samples according to its stored filter coefficients.
  • the finished digital signal is fed to a digital / analog converter 26, which converts the digital signal into an analog one.
  • the analog signal is fed to an audio unit 16 which makes the analog but electrical signal audible, that is to say generates sound waves therefrom and couples it to the air.
  • FIG. 6 shows the schematic structure of the interpolation filter 23, eight poles 31, which are basically the same structure, being shown.
  • Each pole receives the phase information via a bus 32, i.e. the information about the position of the support point within the sound pattern at the desired frequency at which the instantaneous value is to be calculated from the sample values.
  • the sampled values read out from the memory device 7 with the aid of a readout device are fed to the digital interpolation filter 23 via the bus 33.
  • the sample values which are supplied via the bus 33 are available to each filter pole, possibly with a time offset.
  • the output of a filter pole is passed on to the input of the next filter pole.
  • Fig. 7 shows schematically the structure of a single filter pole.
  • Filter coefficients are stored in a memory 38, which may be in the form of RAM or ROM, and can be read out under the control of the phase value supplied via bus 32.
  • the filter coefficient memory 38 is connected via bus lines 39, 40 to an interpolator 41, which also receives the phase value information.
  • the filter coefficient memory 38 provides two successive filter coefficients via the bus lines 39, 40, with the aid of which the interpolator 41 can, for example, carry out a linear interpolation.
  • the output of the interpolator 41 is connected to a multiplier 36 which multiplies the output of the interpolator 41 by the samples which are supplied via the bus 33.
  • the output of the multiplier 36 is connected to an adder 37 which adds the output value of the multiplier 36 to the output value of the previous filter pole.
  • the output of the previous filter pole is fed to the second input of the adder 37 via a bus 34.
  • the output of the adder 37 is passed on to the next filter pole via a bus 35. At the last filter pole (filter pole 8), bus 35 corresponds to output 42.
  • the interpolation filter 23 is a filter with a relatively low order, ie it has only 32 or fewer, in the present case even only 8 poles. Filters with such a low order do not cut off sharply at their cut-off frequency f A , but already have a considerable attenuation below it. This state of affairs is shown schematically in FIG. 8. The dependence of the amplitude A on the frequency f is shown in the upper part. It can clearly be seen that the present filter 23 is a low-pass filter which already exerts a maximum attenuation D on the input signal at the end of a pass band 43. The pass band ends here at the cut-off frequency f G , ie the highest frequency occurring in the sound pattern.
  • the purpose of the low-pass filter property of the interpolation filter 23 is to suppress disturbing frequency components which arise from the sampling of the original sound pattern.
  • the cut-off frequency f G must be set such that it lies at least as far below half the sampling frequency f S / 2 of the original sound pattern as the cut-off frequency f A above f S / 2.
  • the problem with the "cheap" design of the filter now arises that, in the case where the blocking region 45 is allowed to begin near the cut-off frequency f G , the frequency components are damped too much, which is actually still completely in the sound pattern to be reproduced should be included.
  • the pass band 44 is shifted further in the direction of a higher frequency, ie if the cut-off frequency f G is left within the transition region or even on its left side, interfering frequencies are also let through, which audibly and disturbingly change the reproduction of the sound pattern.
  • the attenuation in the pass band 43 can be permitted if it is ensured that the corresponding frequencies of the sound patterns have been correspondingly raised or amplified before being stored.
  • 5 shows a suitable arrangement for this.
  • a microphone 46 sound patterns, for example from a conventional musical instrument, are recorded and converted into electrical signals.
  • the output of the microphone 46 is fed to a pre-emphasis device 47, which amplifies selected frequency components of the electrical signal generated by the microphone 46.
  • the structure of the interpolation filter automatically adjusts the cut-off frequency f G to the original sampling rate at which the sound pattern in the sampling device 48 has been sampled.
  • the interpolation filter 23 interpolates according to the filter coefficients between two successive samples, it being irrelevant whether the samples originally followed one another closely in time or were further apart in time.
  • the temporal relationship is only established by the phase information via line 32, with the aid of which the interpolation filter 23 calculates the necessary number of support points that are necessary for further processing at the system sampling rate.
  • the low-pass filter property of the interpolation filter 23 results from the type of interpolation between the two successive ones Sampled values, ie the “relative” cutoff frequency, that is to say the cutoff frequency based on the original sampling rate, and is determined by the filter coefficients stored in the memory 38.
  • the main processor 6 consists of several transputers 50, 51, 52.
  • the individual transputers of this main processor 6 process the incoming input data essentially in parallel, the processing of the individual input data in one of the transputers 50-52 being independent of the processing in the other transputers 50- 52 takes place. So only a very small amount of communication between the individual transputers 50-52 is necessary.
  • the main processor 6 After the main processor 6 has processed the signals received by the interface processor 4, it sends addresses and / or commands via the control bus 18, with the aid of which the tone generation modules 8-10, which are also connected to the control bus 18, can generate tones. A variety of tone generating modules 8-10 can be provided. Up to now, their capacity has been limited by the performance of the managing processor. By using the main processor 6, which can have any number of transputers 50-52, there are practically no more restrictions in the number of tone generation modules 8-10 to be managed.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Multimedia (AREA)
  • Algebra (AREA)
  • General Physics & Mathematics (AREA)
  • Mathematical Analysis (AREA)
  • Mathematical Optimization (AREA)
  • Mathematical Physics (AREA)
  • Pure & Applied Mathematics (AREA)
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  • Electrophonic Musical Instruments (AREA)
EP91104137A 1990-03-20 1991-03-18 Verfahren zur Klangerzeugung mit einem elektronischen Musikinstrument und elektronisches Musikinstrument Withdrawn EP0448010A2 (de)

Applications Claiming Priority (6)

Application Number Priority Date Filing Date Title
DE19904008872 DE4008872C2 (de) 1990-03-20 1990-03-20 Verfahren zum Erzeugen von Klängen und elektronisches Musikinstrument
DE19904008873 DE4008873A1 (de) 1990-03-20 1990-03-20 Elektronisches musikinstrument
DE4008873 1990-03-20
DE4008872 1990-03-20
DE19904008875 DE4008875C1 (enrdf_load_stackoverflow) 1990-03-20 1990-03-20
DE4008875 1990-03-20

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EP0448010A2 true EP0448010A2 (de) 1991-09-25
EP0448010A3 EP0448010A3 (enrdf_load_stackoverflow) 1994-04-06

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JPS5818693A (ja) * 1981-07-28 1983-02-03 ヤマハ株式会社 電子楽器
JPS6044837A (ja) * 1983-08-23 1985-03-11 Victor Co Of Japan Ltd 波形再生装置
JPS6052895A (ja) * 1983-09-02 1985-03-26 ヤマハ株式会社 楽音信号発生装置
JPS6190514A (ja) * 1984-10-11 1986-05-08 Nippon Gakki Seizo Kk 楽音信号処理装置

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