EP0145997B2 - Dispositif de compensation d'erreurs de reproduction de transducteurs électroacoustiques - Google Patents

Dispositif de compensation d'erreurs de reproduction de transducteurs électroacoustiques Download PDF

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Publication number
EP0145997B2
EP0145997B2 EP84114089A EP84114089A EP0145997B2 EP 0145997 B2 EP0145997 B2 EP 0145997B2 EP 84114089 A EP84114089 A EP 84114089A EP 84114089 A EP84114089 A EP 84114089A EP 0145997 B2 EP0145997 B2 EP 0145997B2
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Prior art keywords
signals
circuit
transducer
digital
microcomputer
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German (de)
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EP0145997B1 (fr
EP0145997A2 (fr
EP0145997A3 (en
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Peter Michael Dipl.-Ing. Pfleiderer
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Pfleiderer Peter Michael Dipl-Ing
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Priority claimed from DE19833343027 external-priority patent/DE3343027A1/de
Priority claimed from DE19843418047 external-priority patent/DE3418047C2/de
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    • H—ELECTRICITY
    • H04—ELECTRIC COMMUNICATION TECHNIQUE
    • H04R—LOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R3/00—Circuits for transducers, loudspeakers or microphones
    • H04R3/04—Circuits for transducers, loudspeakers or microphones for correcting frequency response

Definitions

  • the invention relates to a device according to the preamble of claim 1.
  • Electrodynamic pickup systems are mechanical vibration systems, which are characterized by their own values such as spring constant, mass and damping. Speakers, i.e. Wanders that receive electrical signals and emit acoustic signals are e.g. with the help of a voice coil excited and damped vibrations. Conversely, microphones are transducers that convert acoustic signals into electrical signals. With electrodynamic microphones, this is also done with the help of a voice coil attached to a membrane. Electrodynamic pickup systems also pick up mechanical vibrations and generate electrical signals using voice coils. For this reason, there are no fundamental differences between electrodynamic microphones and electrodynamic pickup systems.
  • amplitude frequency response is e.g. the non-linear course with resonance points and the low efficiency at the upper and lower end of the transmission range.
  • a conventional, softly suspended bass loudspeaker of approx. 30 cm ⁇ at 20 Hz built into a closed housing shows only a low sound pressure effect with amplitude values that are too low, but achieves over-loudness with its resonance frequency in the range of approx. 40 - 80 Hz too large amplitude values and decreases again against high frequencies effectiveness in sound transmission due to too small amplitude values.
  • the amplitude ratio is clearly shown by the frequency ratio related to the resonance frequency in FIG. 1 with a different damping factor ⁇ . This representation is known prior art and is not further explained here.
  • the membrane begins to move in the same direction with vibration pulses above and below the resonance frequency, but only reaches low amplitude values in the case of pulses near and below the resonance frequency, especially during the first half of the oscillation period, since the phase shift occurs during the transient process . It is only when the phase shift corresponding to the frequency has taken place that the amplitude values corresponding to the exciting signal are reached, albeit out of phase.
  • Impulse-like vibrations such as striking a guitar string, striking a note on the piano or hitting a drum, show the maximum amplitude when struck the first time and then vibrate in the torn tone frequency.
  • a loudspeaker system or microphone which is operated in the range of its resonance frequency, must first settle slowly with such pulses until it has the phase position corresponding to the frequency, and only then, depending on the quality, usually reaches the maximum amplitude after one or two full oscillation periods.
  • the transducer resonates at least for a period of time predetermined by the phase shift. In the subsequent decay process, the more or less well damped natural or resonant frequency of the transducer can be seen.
  • the human ear only evaluates the volume according to the amplitude of pure sine tones. Mixtures of sounds that always make up music are evaluated using the envelope.
  • FIG. 3 shows the known arrangement of a loudspeaker with a sensor for the membrane movement.
  • the movement of the membrane is scanned capacitively, inductively, piezoelectrically or optically and the electrical actual value signals generated in this way are compared with the setpoint signals.
  • the readjustment is carried out via a differential amplifier.
  • the capacitive motion sensors record not only the entire diaphragm movement but also all partial vibrations of the diaphragm, the inductive sensors move in the strongly changing magnetic field, which is influenced by the excitation winding through which current flows. They therefore only allow a rough detection of errors.
  • the piezo transducers are relatively heavy and, due to their own weight, increase the error to be corrected. They cannot be used for the mid and high range.
  • the optical pickups with their own control electronics are uneconomically expensive.
  • the control loop would start to oscillate if the loop gain was high. To prevent this, the loop gain must be set to small values, e.g. 20, can be reduced, which greatly affects the effectiveness of the feedback.
  • FIG. 5a An equivalent circuit for the actual loudspeaker was used to try to get a better correction signal.
  • the equivalent circuit is used for this purpose in a feedback circuit according to FIG. 5a.
  • the disadvantage of this circuit is that replacement circuits constructed discretely with coils, capacitors and resistors, as well as the electrodynamic converters themselves, already show considerable differences in the assembled end product, even with small component and manufacturing tolerances. Such an equivalent circuit constructed with discrete components is therefore difficult to match to the actual loudspeaker conditions, cannot be tuned and is expensive. 4 may also be arranged inversely in series with the loudspeaker (FIG. 5b), which is known from US Pat. No. 3,988,541.
  • a compensation device of the type mentioned for playback errors of an electroacoustic transducer which is used to straighten the sound pressure curve at the upper and lower end of the transmission range of electrodynamic loudspeakers.
  • the circuit arrangement consists of active components.
  • the individual circuit sections or filters are effective for the separate compensation of errors in different frequency ranges, i.e. Adjustment measures in a circuit section only affect the corresponding frequency range and do not affect the entire transmission range of the converter.
  • the known circuit arrangement cannot fully take into account or compensate for the complex transmission behavior of the converter and its transmission errors. An approximate linearization of the frequency response or sound pressure curve is provided. After the compensation has been carried out, there are still significant phase errors.
  • Circuits to influence the sound pressure curve of loudspeakers in any way are equalizers. These filter circuits are suitable for linearizing the sound pressure curve or for straightening the amplitude frequency response.
  • graphic equalizers the transmission frequency range from 20 to 20,000 Hz is divided into predetermined, frequency-limited sub-ranges, in which corrections are carried out in each case. A setting for other correction areas is not possible.
  • the so-called parametric equalizers of which US Pat. No. 4,052,560 describes a circuit arrangement, also enable the subsequent adjustment of special partial frequency ranges and the filter characteristic, as a result of which adaptation to any correction points can take place.
  • the phase frequency response of the converter is generally not improved, but often the phase error of the equalizer circuits is superimposed on it.
  • the parametric equalizers also allow for very limited improvements in the phase frequency response.
  • a major advantage of the equalizer is the limitation of the setting options of its active filters to relatively small sub-frequency ranges, which makes setting any, even linear, sound pressure curves relatively easy and clear.
  • phase frequency response is influenced by the approximate linearization of the sound pressure curve, but is not properly compensated for.
  • the correction in each case in partial frequency ranges does not take sufficient account of the complex overall relationship of the electrodynamic converter. Instead of compensating the phase errors, these are at most brought to smaller values.
  • Bass equalization in the lower frequency range is described in numerous references.
  • the known circuits often do not record the entire mechanical vibration system of the electrodynamic loudspeaker in its full transmission range.
  • these circuits are often not assigned to a single electrodynamic loudspeaker, but they are upstream of multipath loudspeakers which contain high, medium and bass loudspeakers and a crossover network.
  • crossovers themselves cause phase errors and the phase errors of two sub-frequency loudspeakers produce further errors when they are superimposed, such measures can actually only straighten the sound pressure curve; the compensation of the phase errors is not possible.
  • the invention has for its object to provide a device for compensating playback errors of an electroacoustic, in particular working according to the electrodynamic principle, by which the signals occurring in the electrical section of the transmission path are changed so that the system-related errors are at least largely compensated.
  • the compensation devices should consist of inexpensive active electronic components and setting elements and should be easily and individually adjustable in a wide range to different converter types.
  • the advantages of the compensation circuit are even clearer if one takes into account that the easy adjustability is just as easily possible not only on speakers of the same series type, but even on speakers as different as woofers, mid-range speakers or tweeters.
  • the material cost of the active electronic components and the adjustability of the actuators result in a great cost advantage.
  • the compensation circuit can be used universally, that is to say for all electrodynamic loudspeaker systems and electrodynamic headphones as well as for all electrodynamic microphones and electrodynamic pickups, results in a large field of application with a cost advantage and manufacturing advantage that increase again due to mass or series production.
  • the crossover is constructed according to DE-C-33 04 402 and thus ensures correct settling and all phases in all frequency ranges, the transient response of the bass, mid-range and tweeters results in Sound bursts from sound mixtures, as they are common in music, for example when striking the piano, guitar and drum, there are no phase shifts and no sound changes over the entire reusable loudspeaker box due to overlapping frequency ranges that have experienced different phase shifts.
  • the membranes of the tweeter, midrange and bass loudspeaker remain in the same phase with all suggestions, whether through impulses or through long-lasting tones.
  • Another advantage is that standard loudspeaker specimens can be used in loudspeaker construction. You do not need any special designs, such as with sensors for readjustment or expensive, narrow-tolerance components and special manufacturing processes to maintain certain parameters.
  • Another advantage is that the electrical characteristics of the compensation circuit do not change during operation, as happens with coils and capacitors due to heating during operation. It is also advantageous that non-linearities due to components such as e.g. in the coil due to hysteresis, saturation and eddy current, do not occur in the adjustable compensation circuit with operational amplifiers.
  • the easy and universal tunability is also advantageous if a converter is destroyed and has to be replaced.
  • the compensation circuit provides a high utility value for repairs.
  • a significant advantage of the compensation circuit is also to be emphasized that it can be implemented extremely inexpensively by only a few active components.
  • the small space requirement of the compensation circuit which is easily conceivable in the size of an operational amplifier customary today, compared to the large passive components of a loudspeaker equivalent circuit, e.g. when used in the bass range.
  • FIG. 7 shows a known loudspeaker equivalent circuit diagram with a differentiating stage connected downstream.
  • the values for the example with the bass speaker are determined dynamically on the bass, ie the complex input impedance is measured at different frequencies and the component values for the known equivalent circuit are mathematically calculated from this.
  • the behavior of the equivalent circuit corresponds exactly to that of the loudspeaker itself.
  • R S 6.8 ⁇
  • R1 40 ⁇
  • the voltage U 1 At the input terminals of the speaker or its exact electrical replication by the equivalent circuit, the voltage U 1 is applied, the voltage U 2 can be tapped at the output terminals.
  • the damping function results from the ratio U 1 / U 2, from the phase shift from U i to U2 the phase angle curve.
  • the general mathematical damping function for the example above is:
  • the component values are standardized to simplify the calculation.
  • the freely selectable reference values (index B) are chosen so that the relationships are as simple as possible.
  • This damping function to be compensated for by the compensation circuit as a function of the frequency is shown in FIG. 8a for the example of the bass loudspeaker, but runs in basically the same way for all electrodynamic converters.
  • the phase angle curve to be compensated for by the compensation circuit as a function of the frequency was recorded in FIG. 8b for the example of the bass loudspeaker, but this curve also runs schematically for all electrodynamic converters (see also FIG. 2).
  • Simply reversing equation (3) in order to get the entire loudspeaker behavior in inverse form does not provide a solution, since this function is not stable in terms of circuitry and oscillates.
  • the path to a compensation circuit is shown below which, like the equivalent circuit of the loudspeaker as an analog computer, has similarly complex cross-connections to one another, but only represents the inverse function in a sufficiently good approximation in the loudspeaker transmission range.
  • the inverse function H (p) is applied in a general form as a polynomial in such a way that the numerator from equation (3) with the coefficients that were determined on the loudspeaker comes into the denominator of equation (4) and the new counter in equation (4) is generally applied.
  • the mathematical stability criterion here requires that the degree of the numerator of the polynomial is equal to or greater than the degree of the denominator.
  • the approximation process itself is carried out by suitable selection of the coefficients, which are improved as long as until the desired result is achieved.
  • the coefficient improvement is always gradual and in the overall transmission system.
  • the individual calculation steps can be numerical, with the help of computing computers, done with graphics computers.
  • the change in the coefficient can be assessed directly in terms of the effect on the change in the curve, and the process can thereby be accelerated.
  • the fine adjustment can be carried out with the oscilloscope by correctly setting the phase angle curve.
  • the compensation circuit is connected in series with the electrodynamic loudspeaker system and the entire transmission system comprising the compensation circuit and electrodynamic converter or its exact equivalent circuit is supplied with square-wave signals of different frequencies.
  • the variation of the coefficients corresponds to the adjustment of the adjustable potentiometers of the compensation circuit.
  • the aim of the optimization is to reproduce the square-wave signal form, which is as error-free as possible, from the converter or its equivalent circuit, and thus the transient and decay processes. This can be done optically very well on the oscilloscope compared to the input signal.
  • the error in the range of the sound pressure transmission curve is from 40 - 500 Hz less than 0.1 dB.
  • the error in the phase angle curve is less than ⁇ 10 ° in the range of 80 - 800 Hz.
  • the circuit arrangement according to FIG. 9a has three positive integrators B1, B2 and B3 connected in series according to the degree of derivatives according to equation (5a).
  • the input signal U 1 is introduced into a summer S 1.
  • the returns R0, R1 and R2 are initiated from the circuit, which have the adjustable potentiometers P7, P6 and P5 arranged in their feedback branch.
  • the returned signals are each taken at the outputs of the integrators B1, B2, B3 and inverted with the help of the inverters L0, L1 and L2.
  • Integrators are available as integrated circuit modules (e.g. TL 071 CP or TL 074 from Texas Instruments).
  • circuit arrangements according to FIGS. 9b, 9c, 9d and 9e are modified exemplary embodiments of the circuit arrangement according to FIG. 9a, which can be derived from the circuit arrangement according to FIG. 9a and the mathematical approach.
  • S are summers, B integrators, R feedbacks, A decouplings, P potentiometers that can be set to coefficients and 1 inverter.
  • the three integrators do not follow one after the other, but only two. A third integrator is switched separately.
  • H p p 2nd + c 2nd p + c 1 d 4th p 2nd + d 3rd p + d 2nd ⁇ p + c O d 1 p + d O
  • the modified circuit arrangement according to the invention according to FIG. 9c was realized from the mathematical approach to a fourth-order equation with four integrators arranged one behind the other.
  • H p p 4th + e 3rd p 3rd + e 2nd p 2nd + e 1 p + e O f 4th p 4th + f 3rd p 3rd + f 2nd p 2nd + f 1 p + f O
  • the modified circuit arrangement according to the invention according to FIG. 9d was not implemented with four integrators in series compared to the circuit arrangement according to the invention from FIG. 9c, but rather with two times two integrators arranged one behind the other.
  • H p p 2nd + g 3rd p + g 2nd H 5 p 2nd + h 4th p + h 3rd ⁇ p 2nd + g 1 p + g O H 2nd p 2nd + h 1 p + h O
  • the modified circuit arrangement according to Fig. 9e shows that an embodiment is also possible in which the integrators are not connected in series directly as in Fig. 9a, but each integrator is visible in a circuit diagram closed by feedback and decoupling, and these circuit arrangements are then simply strung together.
  • H p p + i 2nd k 5 p + k 4th ⁇ p + i 1 k 3rd p + k 2nd ⁇ p + i O k 1 p + k O
  • the predistorted signal which is proportional to acceleration or damping, is suitable for being sent directly to the power amplifier for the electrodynamic converter in order to compensate for its own behavior.
  • This type of equivalent circuit constructed with passive components can also be approximated by a compensation circuit according to the invention. Because there is no voice coil influence, you only get a second order approach. The coefficients are determined using the same iteration method.
  • the disadvantages of this circuit arrangement are that current-impressed amplifiers are not common because they are very difficult to dimension correctly and are easily unstable. Damping of the membrane movement by the current of the amplifier is also not possible with current-impressed amplifiers, but it is possible with voltage-impressed amplifiers.
  • a digital computing circuit can also be used. This possibility, which is particularly advantageous when the electrical signals already exist as digital signals when converting electrical into acoustic signals, is described below.
  • FIG. 12 shows the block diagram of a corresponding device which is used to generate a predistorted control signal for the electroacoustic transducer derived from the original input signal.
  • the predistortion must be dependent on the instantaneous course of the input signal and must be such that the inadequacies of the real electrodynamic converter, including the surrounding medium, are compensated for as far as possible.
  • the original input signal U 1 is converted by an analog / digital converter A / D into a sequence of digital signals DS1.
  • the digital signals DS1 which are output with a high repetition frequency (sampling frequency) of 100 kHz, for example, against the highest frequency of the input signal represent the binary coding in each case one of e.g. 128 different amplitude values.
  • Each e.g. 7-bit date thus represents the (instantaneous) amplitude values present at the time of sampling in the time course of the input signal U 1.
  • the sequence of digital signals DS1 is fed to the data inputs of a microcomputer R, which essentially consists of a microprocessor MP, at least one programmable read-only memory PROM and a read / write memory RAM as working memory, and a number of auxiliary devices, which are not dealt with in more detail is known.
  • a microcomputer R which essentially consists of a microprocessor MP, at least one programmable read-only memory PROM and a read / write memory RAM as working memory, and a number of auxiliary devices, which are not dealt with in more detail is known.
  • All the characteristic values relevant to the reproduction quality of the electroacoustic transducer for example an electrodynamic loudspeaker built into a housing with an upstream power amplifier or a microphone, are stored in the read memory PROM. These parameters relate to variables such as slip, mass inertia of the sound-emitting membrane and the upstream air volume, clamping and restoring forces, damping, resonance frequencies, etc., as well as frequency response and internal resistance of the power amplifier, if applicable.
  • the digital signals DS1 entered into the computer which are now referred to as primary digital signals, are converted into secondary digital signals DS2 in accordance with the characteristic converter characteristic values.
  • the computer R requires at least three successive samples of the curve course of the input signal. He can then see both the steepness and the curvature of the curve. The changes in the curve of the input signal U 1 which are of particular interest for the present purpose can be determined by comparison with previous samples.
  • the sequence of the secondary digital signals DS2 is converted by a digital / analog converter D / A connected to the data output of the microcomputer R into an analog control signal U2, which is used to control the electroacoustic converter W.
  • the electroacoustic transducer W is preceded by a power amplifier EV, which only amplifies the analog control signal U2. Since the characteristics of the power amplifier EV, in particular its frequency response and internal resistance, are included in the transmission chain from the original input signal U 1 to the acoustic oscillation, as already mentioned, these variables must also be taken into account together with the characteristic converter characteristic values when calculating the secondary digital signals DS2 .
  • the entire frequency range of the input signal is usually divided into, for example, three sub-frequency ranges.
  • a specially designed loudspeaker is provided for each sub-frequency range.
  • the frequency range is divided by crossovers, which can be designed as LC elements, as filters with operational amplifiers or as digital filters. The latter is particularly useful in connection with digital recording.
  • a delay element DEL is provided in the highest partial frequency range.
  • the electroacoustic transducers and the upstream power amplifiers are designated W1 to W3 and EV1 to EV3.
  • a clock-controlled shift register arrangement can also be provided, which, however, must be preceded by an analog / digital converter and a digital / analog converter.
  • the analog / digital converter can be omitted in connection with digital recording.
  • the shift register arrangement can be replaced by a further microcomputer, the sole task of which is now to delay the signal.
  • the primary digital signals DS11 and DS12 of both partial frequency ranges are the data inputs of a common microcomputer Rg alternately fed and also processed alternately.
  • a prerequisite for this is a sufficiently high processing speed of the microcomputer Rg and, of course, adapted programming.
  • the secondary digital signals output by the microcomputer Rg must be distributed according to their affiliation to the two channels assigned to the low and mid range. This is done with the aid of a multiplexer MUX controlled by the microcomputer Rg. However, the multiplexer MUX can be omitted if the subsequent digital / analog converters D / A1 and D / A2 are equipped for a clock-controlled takeover of the digital input information and the takeover clocks which are synchronous with the data output of the microcomputer R g are mutually out of phase.

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  • Physics & Mathematics (AREA)
  • Engineering & Computer Science (AREA)
  • Acoustics & Sound (AREA)
  • Signal Processing (AREA)
  • Circuit For Audible Band Transducer (AREA)
  • Amplifiers (AREA)

Claims (13)

  1. Dispositif permettant de compenser les erreurs de reproduction d'un transducteur électroacoustique (W) à l'intérieur d'un domaine de fréquences prédéfini, la voie de transmission des signaux à reproduire comprenant une section acoustique et une section électrique, où dans la section électrique est prévu un circuit qui reçoit des signaux d'entrée (U₁) et délivre des signaux de sortie (U₂) modifiés, le circuit prévu dans la section électrique de la voie de transmission étant un circuit de calcul dont les signaux de sortie (U₂) sont déduits de ses signaux d'entrée (U₁), suivant une fonction complexe d'atténuation (H(p)) en considérant l'amplitude et la phase,
    le transducteur (W) ayant une fonction complexe d'atténuation (H₁(p)) qui est valable pour l'ensemble du domaine de fréquences prédéfini du transducteur (W) et qui décrit les propriétés complexes du transducteur en ce qui concerne sa réponse en fréquence en fonction de l'amplitude et de la phase,
    la fonction d'atténuation (H₁ (p)) du transducteur (W) étant définie par les paramètres du transducteur qui décrivent ses propriétés inhérentes, telles que le glissement, l'inertie d'une membrane rendue mobile et du volume d'air, les forces de contrainte et de rappel, l'atténuation, les fréquences de résonnance, etc,
    la fonction complexe d'atténuation (H(p)) du circuit de calcul étant inverse de la fonction complexe d'atténuation (H₁(p)) du transducteur dans la mesure où les pôles de la fonction de transfert (1/H₁(p)) du transducteur (W) qui dégradent sa réponse, sont reproduits par approximation par des points d'annulation dans la fonction de transfert (1/H(p)) du circuit de calcul et de ce fait créent de nouveaux pôles par la fonction de transfert du circuit de calcul,
    caractérisé en ce que le circuit de calcul présente des éléments de réglage (p) pour faire tendre, par approximation, la fonction complexe de transfert (1/H(p)) du circuit de calcul vers la fonction complexe d'atténuation (H₁(p)) du transducteur et
    en ce que les coefficients du circuit de calcul, pouvant être choisis librement, sont choisis de telle manière que, lors d'une alimentation du système de transmission combiné constitué d'un circuit de calcul et d'un transducteur avec des signaux rectangulaires, de fréquences différentes, une reproduction des signaux rectangulaires, la plus exempte possible d'erreur en comparaison avec le signal d'entrée, est obtenue au moyen d'une observation optique des signaux de sortie sur un oscilloscope.
  2. Dispositif suivant la revendication 1, caractérisé en ce que le transducteur électroacoustique (W) sert à la transformation de signaux électriques en signaux acoustiques et en ce que le circuit de calcul est disposé en amont du transducteur (W) suivant le sens de la transmission.
  3. Dispositif suivant la revendication 1, caractérisé en ce que le transducteur électroacoustique (W) sert à la transformation de signaux acoustiques en signaux électriques et en ce que le circuit de calcul est disposé en aval du transducteur (W) suivant le sens de la transmission.
  4. Dispositif suivant la revendication 2, caractérisée en ce qu'en tant que circuit de calcul, il est prévu un microcalculateur (R), à fonctionnement numérique, auquel sont envovoyés des signaux d'entrée (U₁) se présentant sous la forme de successions de signaux numériques primaires (DS1) et qui délivre des successions de signaux numériques secondaires (DS2), en ce qu'il est associé au microcalculateur (R) une mémoire morte programmable (PROM) dans laquelle sont rangées les valeurs caractéristiques correspondant au transducteur (W) et un programme permettant de convertir les signaux numériques primaires (D1) en signaux numériques secondaires (DS2) et en ce qu'il est en outre prévu un convertisseur numérique/analogique (D/R) permettant de convertir les successions de signaux numériques secondaires (DS2) en signaux analogiques de sortie (U₂).
  5. Dispositif suivant la revendication 4, caractérisé en ce qu'il est prévu un convertisseur analogique/numérique (A/D) permettant de convertir des signaux d'entrée (U₁), se présentant sous la forme de signaux analogiques, en successions de signaux numériques primaires (DS1).
  6. Dispositif suivant la revendication 5, caractérisé en ce qu'il est prévu des aiguillages de fréquences (FW1 à FW3) permettant de diviser le domaine de fréquences des signaux d'entrée (U₁) en plusieurs domaines partiels de fréquences déterminés à l'avance, en ce que, pour chaque domaine partiel de fréquences, il est prévu un amplificateur final (EV1 à EV3) et un transducteur électroacoustique (W1 à W3) et en ce qu'au moins dans le domaine partiel de fréquences le plus bas, il est disposé une unité de correction (K1, K2) constituée d'un microcalculateur (R), d'un convertisseur numérique/analogique (D/A) et éventuellement d'un convertisseur analogique/numerique (A/D), tandis que, dans les autres domaines partiels de fréquences, il est prévu des dispositifs (DEL) pour le retard des signaux.
  7. Dispositif suivant la revendication 6, caractérisé en ce que les signaux numériques primaires (DS11, DS12) du domaine de fréquences le plus bas et au moins du domaine de fréquences plus élevé suivant sont envoyés, en étant décalés dans le temps, aux entrées de données d'un microcalculateur (Rg) commun et en ce qu'aux sorties de données du microcalculateur (Rg), il est raccordé un multiplexeur (MUX), commandé par le microcalculateur (Rg ), qui commute alternativement sur les entrées des convertisseurs numérique/analogique (D/A1, D/A2) correspondants les signaux numériques secondaires (DS21, DS22) associés au domaine de fréquences le plus bas et au moins au domaine de fréquences plus élevé suivant.
  8. Dispositif suivant la revendication 6, caractérisé en ce que les signaux numériques primaires (DS11, DS12) du domaine de fréquences le plus bas et au moins du domaine de fréquences plus élevé suivant sont envoyés, en étant décalés dans le temps, aux entrées de données d'un microcalculateur (Rg) commun, en ce que les entrées des convertisseurs numérique/analogique (D/A1, D/A2) correspondant au domaine de fréquences le plus bas et au moins au domaine de fréquences plus élevé suivant sont connectées en parallèle et sont reliées aux sorties de données du microcalculateur (Rg) et en ce que le transfert des signaux numériques secondaires (DS21, DS22) dans les convertisseurs numérique/analogique (D/A1, D/A2) peut être commandé, d'une manière décalée dans le temps, par des signaux délivrés par le microcalculateur (Rg).
  9. Dispositif suivant la revendication 1, caractérise en ce que le circuit de calcul est réalise sous la forme d'un circuit analogique qui est constitué de plusieurs intégrateurs (B), de plusieurs éléments de réglage (P) et d'au moins deux circuits additionneurs (S), en ce que les signaux d'entrée (U₁) sont appliqués sur une entrée du premier circuit additionneur (S) et les autres entrées sont reliées, par l'intermédiaire d'inverseurs (I) et d'éléments de réglage (P), à des sorties d'au moins l'un des intégrateurs (B) connectés en aval du premier circuit additionneur (S), en ce que la sortie du premier circuit additionneur (S) et les sorties des intégrateurs (B) sont reliées, par l'intermédiaire d'autres éléments de réglage (P), aux entrées du second circuit additionneur (S), sur la sortie duquel le signal de sortie (U₂) peut être prélevé, et en ce que le nombre des intégrateurs (B) contenus dans le circuit de calcul est égal au degré de la fonction H(p) à l'aide de laquelle le comportement propre, complexe, du transducteur (W) fait l'objet d'une approximation, sous forme inverse, en ce qui concerne la réponse en amplitude à la fréquence et la réponse en phase à la fréquence.
  10. Dispositif suivant la revendication 9, caractérisé en ce que le nombre des intégrateurs (B) connectés directement en série est chaque fois égal au degré des facteurs de la fonction mathématique (H(p) éventuellement décomposée en facteurs, en ce qu'à chaque groupe d'intégrateurs (B) connectés directement en série, il est associé un premier et un second circuits additionneurs (S), ainsi que des éléments de réglage (P) correspondants à des inverseurs (I), et en ce que la sortie du second circuit additionneur (S) d'un groupe précédent est reliée à une entrée du premier circuit additionneur (S) d'un groupe suivant.
  11. Dispositif suivant l'une des revendications 9 à 10, caractérisé en ce que la structure du circuit de calcul correspond à une fonction mathématique du troisième degré.
  12. Dispositif suivant l'une des revendications 1, 2 et 9 à 11, caractérisé par sa combinaison avec un circuit connu de régulation de la membrane.
  13. Dispositif suivant l'une des revendications 1, 2 et 4 à 12, caractérisé en ce qu'un courant est appliqué à la bobine mobile d'un haut-parleur.
EP84114089A 1983-11-28 1984-11-22 Dispositif de compensation d'erreurs de reproduction de transducteurs électroacoustiques Expired - Lifetime EP0145997B2 (fr)

Applications Claiming Priority (4)

Application Number Priority Date Filing Date Title
DE3343027 1983-11-28
DE19833343027 DE3343027A1 (de) 1983-11-28 1983-11-28 Verfahren und schaltungsanordnung zur verbesserung der wiedergabequalitaet von elektroakustischen wandlern
DE19843418047 DE3418047C2 (de) 1984-05-15 1984-05-15 Einrichtung zur Kompensation von Wiedergabefehlern eines elektroakustischen Wandlers
DE3418047 1984-05-15

Publications (4)

Publication Number Publication Date
EP0145997A2 EP0145997A2 (fr) 1985-06-26
EP0145997A3 EP0145997A3 (en) 1987-09-30
EP0145997B1 EP0145997B1 (fr) 1991-11-06
EP0145997B2 true EP0145997B2 (fr) 1996-01-10

Family

ID=25815960

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Application Number Title Priority Date Filing Date
EP84114089A Expired - Lifetime EP0145997B2 (fr) 1983-11-28 1984-11-22 Dispositif de compensation d'erreurs de reproduction de transducteurs électroacoustiques

Country Status (4)

Country Link
US (1) US4675835A (fr)
EP (1) EP0145997B2 (fr)
JP (1) JPH07114519B2 (fr)
DE (1) DE3485242D1 (fr)

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US4914750A (en) * 1987-07-13 1990-04-03 Avm Hess, Inc. Sound transducer
US4953112A (en) * 1988-05-10 1990-08-28 Minnesota Mining And Manufacturing Company Method and apparatus for determining acoustic parameters of an auditory prosthesis using software model
US5247467A (en) * 1989-08-16 1993-09-21 Hewlett-Packard Company Multiple variable compensation for transducers
DE4111884A1 (de) * 1991-04-09 1992-10-15 Klippel Wolfgang Schaltungsanordnung zur korrektur des linearen und nichtlinearen uebertragungsverhaltens elektroakustischer wandler
US5493620A (en) * 1993-12-20 1996-02-20 Pulfrey; Robert E. High fidelity sound reproducing system
DE19917584A1 (de) * 1999-04-19 2000-10-26 Siemens Ag Flächenlautsprecher und Verfahren zu dessen Betrieb
DE10045201C2 (de) * 2000-09-13 2002-08-14 Siemens Ag Akustische Wiedergabeeinrichtung
US6771781B2 (en) * 2001-05-08 2004-08-03 Daniel A. Chattin Variable damping circuit for a loudspeaker
US7825986B2 (en) * 2004-12-30 2010-11-02 Mondo Systems, Inc. Integrated multimedia signal processing system using centralized processing of signals and other peripheral device
US7653447B2 (en) * 2004-12-30 2010-01-26 Mondo Systems, Inc. Integrated audio video signal processing system using centralized processing of signals
US8015590B2 (en) 2004-12-30 2011-09-06 Mondo Systems, Inc. Integrated multimedia signal processing system using centralized processing of signals
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US10423229B2 (en) 2017-08-17 2019-09-24 Google Llc Adjusting movement of a display screen to compensate for changes in speed of movement across the display screen

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DE10236307B4 (de) * 2002-03-16 2005-04-28 Joerg Seiffert Schaltung zur Korrektur der akustischen Gruppenlaufzeit und des frequenzabhängigen Phasenverhaltens für Schallwandler

Also Published As

Publication number Publication date
EP0145997B1 (fr) 1991-11-06
US4675835A (en) 1987-06-23
DE3485242D1 (de) 1991-12-12
JPS60134699A (ja) 1985-07-17
EP0145997A2 (fr) 1985-06-26
JPH07114519B2 (ja) 1995-12-06
EP0145997A3 (en) 1987-09-30

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