EP3606098B1 - Method and circuit arrangement for operating a condenser microphone - Google Patents

Method and circuit arrangement for operating a condenser microphone Download PDF

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Publication number
EP3606098B1
EP3606098B1 EP19189032.6A EP19189032A EP3606098B1 EP 3606098 B1 EP3606098 B1 EP 3606098B1 EP 19189032 A EP19189032 A EP 19189032A EP 3606098 B1 EP3606098 B1 EP 3606098B1
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digital
dsb
audio signal
signal
carrier signal
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German (de)
French (fr)
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EP3606098A1 (en
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Lars Urbansky
Udo Zölzer
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Helmut Schmidt Universitaet
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Helmut Schmidt Universitaet
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    • HELECTRICITY
    • H04ELECTRIC COMMUNICATION TECHNIQUE
    • H04RLOUDSPEAKERS, MICROPHONES, GRAMOPHONE PICK-UPS OR LIKE ACOUSTIC ELECTROMECHANICAL TRANSDUCERS; DEAF-AID SETS; PUBLIC ADDRESS SYSTEMS
    • H04R19/00Electrostatic transducers
    • H04R19/04Microphones

Definitions

  • the invention relates to a method and a circuit arrangement for operating a condenser microphone using a high-frequency method with amplitude modulation.
  • a condenser microphone In a condenser microphone, sound waves are known to be recorded in the form of changes in the capacitance of a condenser capsule, which has a membrane moved by the sound waves and at least one rigid counter-electrode.
  • standard capacitor converters which have only one counter electrode
  • balanced capacitor converters, or push-pull converters in which the diaphragm is sandwiched between two counter electrodes, resulting in two capacitors.
  • a membrane deflection in one direction caused by impinging sound waves leads to a reduction in the air gap to one counter-electrode and a simultaneous increase in the air gap to the other counter-electrode. This results in a correspondingly opposing increase or decrease in the capacitances of the two capacitors (push-pull converter).
  • the mass of the diaphragm is very low, it can follow the sound waves that strike it very quickly and precisely, so that condenser microphones have superior physical properties, especially compared to dynamic microphones, and can therefore generate AF output signals with high sound quality, impulse fidelity, sensitivity and low noise.
  • condenser microphones require an external power supply for the condenser capsule in order to be able to convert the changes in capacitance into an output signal.
  • this method has various disadvantages. On the one hand, leakage currents can occur, especially in environments with high humidity, which lead to a poor signal-to-noise ratio.
  • the low capacitance of the capacitor capsule requires an impedance converter with a very high input resistance for further processing of the output voltage.
  • one of the known high-frequency methods is preferably used, in which the changing capacitance of the capacitor capsule is used to modulate the frequency, the phase or the amplitude of a high-frequency electrical carrier signal (HF signal).
  • HF signal high-frequency electrical carrier signal
  • the condenser capsule In frequency modulation, the condenser capsule is part of a resonant circuit, so changes in the capacitance of the condenser capsule caused by the sound waves result in corresponding changes in the resonant frequency.
  • this method requires a very frequency-stable oscillator so that the audio signal obtained after demodulation is not disturbed.
  • the capacitor capsule is part of an LC parallel resonant circuit, so that the changes in the capacitance of the capacitor capsule lead to corresponding changes in the resonant frequency of the LC circuit and thus to phase shifts in the externally generated and coupled RF signal.
  • This method also requires a very frequency-stable HF signal and a high Q factor of the LC circuit to achieve a large phase shift or microphone sensitivity.
  • Another disadvantage is that the phase response has a relatively small linear range, so that in the case of a high sound pressure level, the change in capacitance and the resulting phase shifts are no longer proportional to each other.
  • the capacitor capsule together with two windings of the same type forms an HF transformer bridge circuit which is tuned to the frequency of an HF carrier signal which is generated by an HF oscillator which is coupled via another winding. Due to the movement of the membrane and the resulting change in capacitance, a proportional double sideband amplitude modulated (DSB-AM) audio output voltage with a suppressed HF carrier signal can be generated at the bridge branch of the circuit.
  • DSB-AM proportional double sideband amplitude modulated
  • This method is particularly suitable for the symmetrical capacitor converters or push-pull converters mentioned at the outset, in which the two capacitors are connected as a capacitive voltage divider in one of the bridge arms.
  • an integrated circuit which, to generate a microphone output signal, has a preamplifier for an input signal generated with a microphone membrane or a microphone element, in relation to which the membrane is movable, a charge pump and a low-pass filter, the low-pass filter is provided for filtering the pump voltage and for generating a bias voltage for the membrane or the microphone element.
  • the invention is based on the object of creating a method and a circuit arrangement for operating a condenser microphone with which the above-mentioned superior physical properties of such a microphone can be better utilized and converted into AF output signals with high sound fidelity.
  • a digital microphone is to be created with the smallest possible analog outlay, in which the dynamic range and the signal-to-noise ratio of the AF output signals is at least equivalent to or even better than in known, analog-operated microphones.
  • An essential principle of the invention is to use digital signal processing in combination with a suitable analog circuit for detecting the changes in capacitance generated by impinging sound waves to obtain an AF output signal that is characterized in particular by a high dynamic range and a high signal-to-noise ratio.
  • the frequency of the analog HF carrier signal can be very precisely matched to the resonance of the analog circuit part, and a spectrally very pure RF carrier signal can also be generated.
  • the digital HF carrier signal is preferably subjected to noise shaping before the digital/analog conversion, which not only reduces the resolution or word width, but also shifts the quantization noise into frequency ranges that Capturing the relevant LF bandwidth of the microphone are not required. As a result, a high signal-to-noise ratio is maintained in the relevant frequency band.
  • the amplitude of the digital HF carrier signal can preferably be adjusted in order to change or adapt the sensitivity of the microphone, in particular as a function of a sound pressure level that occurs.
  • the sensitivity can be adjusted digitally and multi-channel analog-to-digital converters are not required to achieve high dynamic range or high gain.
  • This amplitude is preferably changed automatically by means of a sound pressure level detector/predictor as a function of a detected or predicted sound pressure level, so that the sensitivity of the microphone is adjusted accordingly and overloading of the analog circuit part is thus avoided.
  • an additional signal-to-noise ratio and dynamic range can also be obtained by the fact that the recorded audio signal is digitized, preferably oversampled, and after demodulation is clocked down to a lower sampling frequency (for example to a standard sampling frequency of 48 kHz).
  • a condenser microphone is preferably used in the analog circuit part of the circuit arrangement according to the invention and this is preferably operated according to the high-frequency method mentioned at the outset with amplitude modulation, the basic function and implementation of this mode of operation should first be described using the figure 1 be explained.
  • figure 1 shows an exemplary circuit arrangement with a condenser microphone with a symmetrical converter (push-pull converter), which has a first and a second capacitor C 1 (t), C 2 (t), whose capacitances are changed in opposite directions as explained above by incoming sound waves s(t).
  • a condenser microphone with a symmetrical converter push-pull converter
  • the circuit arrangement works according to the high-frequency method with amplitude modulation mentioned at the outset, which is preferably used in connection with the invention, and comprises a first transformer Tr1, a second transformer Tr2, an HF generator HF, a first and a second diode D1, D2 and a charging capacitor Ca.
  • a first winding W1 is located on the primary side of the first transformer Tr1, via which the HF voltage (HF carrier signal) generated by the HF generator HF is injected.
  • the first transformer Tr1 On a first secondary side, the first transformer Tr1 has a second and an identical third winding W2, W3. The two windings W2, W3 together with the two capacitors C 1 (t), C 2 (t) form a bridge circuit. A fourth and an identical fifth winding W4, W5 is located on a second secondary side of the first transformer Tr1, the outer ends of which are connected to one another via two diodes D1, D2 connected in series and which together form a synchronous rectifier.
  • the second transformer Tr2 has one side connected between the center tap between the second and third windings W2, W3 and ground, while the other side is connected between the center tap between the fourth and fifth windings W4, W5 and one of the output terminals.
  • the other output connection is tapped between the two diodes D1, D2.
  • the charging capacitor Ca is connected in parallel with the output terminals, and the output voltage x AF (t) is applied to this capacitor in the form of the demodulated audio signal (baseband signal).
  • the HF bridge circuit formed by the first and the second capacitor C 1 (t), C 2 (t) and the second and the third winding W2, W3 is tuned to the frequency of the HF carrier signal generated by the HF generator HF.
  • the bridge is balanced and no HF voltage is produced at its output.
  • an HF voltage is produced that is proportional to the membrane deflection proportional output voltage x AF (t) is present.
  • figure 2 shows a schematic block diagram of an exemplary first embodiment of a circuit arrangement according to the invention, which essentially consists of three components, namely a digital circuit part DC, an analog circuit part AC and a mixed analog/digital circuit part MC, the latter being connected between the digital and the analog circuit part is and connects both together.
  • the digital circuit part DC includes the digital components of the circuit arrangement. These are essentially an HF generator CG for generating a digital oversampled HF carrier signal x c,h (k) with a first high sampling rate f S,1 and with a first high resolution, which corresponds to a first large word length of w 0 bits , an optional unit NS for noise shaping of the HF carrier signal x c,h (k) and for generating an HF carrier signal x c (k) with the same first sampling rate f S,1 and with a second resolution that is lower than the first resolution, the corresponds to a correspondingly smaller second word width of w 1 bit.
  • an HF generator CG for generating a digital oversampled HF carrier signal x c,h (k) with a first high sampling rate f S,1 and with a first high resolution, which corresponds to a first large word length of w 0 bits
  • an optional unit NS for noise shaping of the HF carrier signal x
  • the digital circuit part DC also includes a demodulator Dm for demodulating an applied digital DSB-AM audio output signal x DSB (m), which has a second sampling rate f S,2 and a third resolution or third word length of w 2 bits, and for Generation of a demodulated digital (baseband) audio signal x dem (m) with unchanged (second) sampling rate f S,2 and a fourth resolution or fourth word width of w 3 bits.
  • a demodulator Dm for demodulating an applied digital DSB-AM audio output signal x DSB (m), which has a second sampling rate f S,2 and a third resolution or third word length of w 2 bits, and for Generation of a demodulated digital (baseband) audio signal x dem (m) with unchanged (second) sampling rate f S,2 and a fourth resolution or fourth word width of w 3 bits.
  • the digital circuit part includes an optional decimator Dc for decimation or reduction of the second sampling rate f S,2 of the demodulated audio signal x dem (m) and for generating a digital audio output signal x AF (n) with a standard audio sampling frequency of for example 48 kHz (third sampling rate fs) with a desired fifth resolution or a fifth word width of w 4 bits.
  • Dc decimation or reduction of the second sampling rate f S,2 of the demodulated audio signal x dem (m) and for generating a digital audio output signal x AF (n) with a standard audio sampling frequency of for example 48 kHz (third sampling rate fs) with a desired fifth resolution or a fifth word width of w 4 bits.
  • This digital circuit part D-C is preferably implemented in the form of an FPGA (field programmable gate array) and represents a digital signal processor (DSP). It is preferably implemented using a corresponding program which is executed on a data processing system.
  • FPGA field programmable gate array
  • DSP digital signal processor
  • the analog/digital circuit part MC comprises a digital/analog converter D/A for generating an analog HF carrier signal x c (t) from the supplied digital HF carrier signal x c (k), and an analog/digital converter A/ D for generating a digital DSB-AM audio signal x DSB (m) with said third resolution or third word length of w 2 bits from a supplied analog DSB-AM audio signal x DSB (t), which is generated with the microphone unit Cm.
  • D/A digital/analog converter
  • A/ D for generating an analog HF carrier signal x c (t) from the supplied digital HF carrier signal x c (k)
  • an analog/digital converter A/ D for generating a digital DSB-AM audio signal x DSB (m) with said third resolution or third word length of w 2 bits from a supplied analog DSB-AM audio signal x DSB (t), which is generated with the microphone unit Cm.
  • the analog circuit part AC contains the microphone unit Cm with a condenser microphone and is used to convert the changes in the capacitance of the microphone generated by the incident sound waves s(t) (LF audio signal) by means of the analog HF carrier signal x c (t) supplied into the convert analog DSB-AM audio signal x DSB (t) with suppressed HF carrier signal x c (t).
  • the HF generator CG generates an oversampled (oversampled) digital HF carrier signal x c,h (k) with said first high sampling rate f S,1 and the first high resolution, which corresponds to the first large word width of w 0 bits .
  • the level of the carrier frequency f r of the HF carrier signal x c,h (k) is suitably selected in particular as a function of the electrical properties of the analog circuit part AC (in particular its resonant frequency, see below).
  • oversampling means that the signal is generated with a sampling rate that is higher than that required according to the sampling theorem for detecting the RF carrier frequency f r (also called oversampling).
  • the (first) sampling rate f S,1 of the HF carrier signal x c,h (k) with the frequency f r is consequently greater than the sampling rate of 2 * f r required according to the sampling theorem.
  • the first high resolution of the HF carrier signal x c,h (k) mentioned (and the associated first large word length of w o bits) is selected so large that the signal/noise ratio in the relevant frequency range is sufficiently high and even then still large is enough if there is a gain adjustment of the RF carrier signal. This principle is explained below with reference to figure 5 still to be explained.
  • Such an HF generator CG can, for example, be a synthesizer that works according to the known DDS (direct digital synthesis) method, or can be implemented in the form of a look-up table (LUT).
  • a DDS synthesizer has the advantage that the carrier frequency can easily be adapted to the electrical properties such as the resonant frequency of the analog circuit part AC, while using a look-up table requires less digital circuitry and thus lower hardware costs.
  • other HF generators are conceivable that meet the stated requirements.
  • the relatively large first word length of w 0 bits of the RF carrier signal x c,h (k) is preferably correspondingly reduced to the second word length of w 1 bits.
  • the digital HF carrier signal x c,h (k) is first preferably fed to the unit NS for noise shaping (noise shaping), with which the quantization noise in an is, as is known, concentrated in areas outside the relevant frequency band and is thus shifted into frequency ranges that are not required to cover the relevant AF bandwidth (ie usually around 40 kHz symmetrically around the carrier frequency).
  • noise shaping noise shaping
  • the quantization noise in an is, as is known, concentrated in areas outside the relevant frequency band and is thus shifted into frequency ranges that are not required to cover the relevant AF bandwidth (ie usually around 40 kHz symmetrically around the carrier frequency).
  • figure 3 shows the HF generator CG and a possible equivalent circuit diagram of the unit NS for noise shaping.
  • the HF generator CG can alternatively be dimensioned in such a way that it generates a digital HF carrier signal x c (k) with a reduced second word width of w 1 bit, adapted to the resolution of the digital/analog converter, so that the unit NS for noise shaping can then be dispensed with.
  • the digital HF carrier signal x c (k) is fed to the digital/analog converter D/A, with which an analog HF carrier signal x c (t) is generated therefrom in a manner known per se.
  • This analog HF carrier signal x c (t) is fed to the microphone unit Cm and is used to convert the changes in the capacitance of the microphone caused by the incident sound waves s(t) (LF audio signal) into a DSB-AM audio signal x DSB (t) implement, preferably using a high-frequency method with amplitude modulation and suppressed carrier frequency.
  • LF audio signal LF audio signal
  • DSB-AM audio signal x DSB (t) preferably using a high-frequency method with amplitude modulation and suppressed carrier frequency.
  • the analog DSB-AM audio signal x DSB (t) generated by the microphone unit Cm is then fed to the analog/digital converter A/D. In order to generate a high-quality digital audio signal, this has the highest possible (third) resolution mentioned with the third word width of w 2 bits. Furthermore, the DSB-AM audio signal x DSB (t), since it is a bandpass signal, is (over)sampled with the said second sampling rate f S,2 , which in turn is higher than it is after the Sampling theorem for the bandwidth of 40 kHz occurring in the bandpass signal is required.
  • This is preferably implemented with an analog/digital converter which works according to the known method of successive approximation (SAR—successive approximation register).
  • Such SAR analog / digital converters have the advantage that they have a relatively low converter latency, which is based on the below for the figure 5 described gain control is of particular importance.
  • a very high signal-to-noise ratio can be achieved by using such high-performance analog/digital converters.
  • the digital DSB-AM audio signal x DSB (m) present at the output of the analog/digital converter A/D with the mentioned high (third) resolution with the third word width of w 2 bits and the mentioned second sampling rate f S,2 is then supplied to the demodulator Dm, with which it is synchronously demodulated, so that a digital baseband signal x dem (m) with the second sampling rate f S,2 and the fourth resolution or fourth word length of w 3 bits results.
  • the demodulator Dm is preferably a coherent demodulator whose HF demodulator signal x d,h (m) is generated digitally with the second sampling rate f S,2 and a desired sixth resolution or sixth word width of w 5 bits and is used for demodulation. Since the frequency of the HF carrier signal x c,h (k) is known, the HF demodulator signal can be generated in a correspondingly frequency-synchronous manner.
  • figure 4 shows an example of a block diagram of such a demodulator, which has a demodulator/generator unit DmG and preferably a unit PhE for phase estimation, at the input of which the demodulated digital baseband signal x dem (m) is present and whose output is connected to the demodulator/generator unit DmG is to shift or adapt the phase of the HF demodulator signal x d,h (m) generated by the demodulator/generator unit DmG, with which the DSB-AM audio signal x DSB (m) is demodulated, to achieve coherent demodulation .
  • This is necessary because the phase of the HF carrier signal x DSB (t) generated by the microphone unit Cm and modulated with the audio signal is usually not known.
  • the unit PhE for phase estimation can alternatively be dispensed with.
  • the frequency of the HF demodulator signal x d,h (m) generated by the demodulator/generator unit DmG is equal to the frequency of that generated by the HF generator CG HF carrier signal x c,h (k), while the sampling rate f S,2 corresponds to that of the analog/digital converter A/D.
  • the demodulated digital baseband signal x dem (m) now has a bandwidth of about 20 kHz and is highly oversampled with the (second) sampling rate f S,2 , it is preferably filtered with the following decimator Dc and downsampled to to generate a digital AF audio signal x AF (n) with a customary or standard sampling frequency of, for example, 48 kHz (third sampling rate fs).
  • an additional signal-to-noise ratio and dynamic range are achieved. For example, assuming oversampling by a factor of 100 (corresponding to an analog-to-digital converter A/D sampling frequency of 4.8 MHz) and white quantization noise is present, this results in an additional signal-to-noise ratio and dynamic range of 20 dB, which corresponds to an additional resolution of about 3 bits.
  • the decimation process can be performed in several stages.
  • the decimator Dc can be implemented, for example, by using FIR (finite impulse response) filter topologies such as CIC (cascaded integrator comb) filters or polyphase filters.
  • FIR finite impulse response
  • IIR infinite impulse response filters
  • a main criterion in the design of these filters is to preserve a linear phase in the audio frequency band.
  • figure 5 shows a schematic block diagram of an exemplary second embodiment of a circuit arrangement according to the invention, which can also be derived from the three above said components, namely a digital circuit part DC, an analog circuit part AC and a mixed analog/digital circuit part MC.
  • this embodiment additionally includes an optional first correction unit Corr1 connected between the output of the analog/digital converter A/D and the input of the demodulator Dm, a sound pressure level detector/predictor SPL-DP, which is also connected to the output of the analog/ Digital converter A / D and further connected to the first correction unit Corr1, and a unit GC for gain control, which is controlled by the sound pressure level detector / predictor SPL-DP and by means of a first multiplier M1 generated by the HF generator CG HF carrier signal x c,h (k) with a first amplification factor g 1 (k) and by means of a second multiplier M2 the output signal of the first correction unit Corr1 with a second amplification factor g 2 (m).
  • a first correction unit Corr1 connected between the output of the analog/digital converter A/D and the input of the demodulator Dm
  • a sound pressure level detector/predictor SPL-DP which is also connected to the output of the analog/ Digital converter A
  • an optional second correction unit Corr2 is provided, which is connected to the output of the optional decimator Dc.
  • the unit NS for noise shaping is again optional here.
  • the analog output level of microphones can usually be adjusted, for example by attenuating the recorded AF signal.
  • a particular advantage of the embodiment according to the invention is that such adjustability or damping of the LF signal is not required.
  • the sensitivity of the microphone is directly proportional to the amplitude of the analog HF carrier signal x c (t) and the HF carrier signal is generated in the digital domain according to the invention, the sensitivity can be digitally increased and decreased without additional analog circuits and thus the dynamic range of the microphone can be expanded or optimized accordingly.
  • the ability to digitally manipulate the RF carrier signal is one of the key benefits of generating the RF carrier signal in the digital domain. Since the RF carrier signal is generated in the frequency range of interest with a high signal-to-noise ratio as explained above, its intensity can be reduced before it is thereafter subjected to noise-shaping in the unit NS, thereby maintaining an optimum signal-to-noise ratio.
  • the adjustment of the sensitivity preferably before the noise shaping namely by the mentioned application of the digital HF carrier signal x c,h (k) with the first amplification factor g 1 (k) ⁇ 1 (whereby the HF generator CG can be driven directly with the first amplification factor g 1 (k) accordingly).
  • the maximum sound pressure level that can be handled would be 114 dBSPL.
  • the amplitude of the RF carrier signal can be adjusted in such a way that a gain factor g 1 ( k) of about 1/32.
  • the above-mentioned, first high resolution or the first large word length of w 0 bits of the HF carrier signal x c,h (k) can be selected, for example, so that it provides the lost 5 bits before the noise shaping, ie the RF carrier signal x c,h (k) before noise shaping is resolved 5 bits higher than the RF carrier signal x c (k) after noise shaping.
  • so-called “gain ranging” is used, in which the level range of the microphone is automatically adapted to a recorded sound pressure level and the dynamic range of the entire system is thus automatically changed or correspondingly increased and decreased.
  • this problem is solved in the circuit arrangement according to FIG figure 5 This is solved by automatically setting or adjusting the sensitivity of the microphone when high sound pressure levels occur or are predicted.
  • the sound pressure level detector/predictor SPL-DP is provided for this purpose, which has the unit GC for amplification control, which uses the first amplification factor g 1 (k) mentioned for applying the HF generator CG or the HF carrier signal x c,h ( k) generates, controls.
  • the sound pressure level detector/predictor SPL-DP is used to record or predict particularly high sound pressure levels. Since, as already mentioned above, a SAR analog/digital converter A/D is preferably used and the digital DSB-AM bandpass signal x DSB (m) is heavily oversampled, the system can be switched to high respond to sound pressure levels.
  • the unit GC generates not only the first amplification factor g 1 (k) for applying the HF carrier signal x c,h (k), but also the second amplification factor gz(m), with which the digital DSB -AM audio signal x DSB (m) is amplified or multiplied at the input of the demodulator Dm.
  • the unit GC generates the second amplification factor g 2 (m) in such a way that the amplitude adjustment or multiplication of the HF carrier signal x c,h (k) with the first amplification factor g 1 (k) is compensated again and thus a corrected digital DSB-AM audio signal x ⁇ DSB (m) is generated, which is fed to the demodulator Dm.
  • the second amplification factor g 2 (m) is preferably not only determined by inverting the first amplification factor g 1 (k), but the internal delays and filter properties of the circuit and the change in the sampling rate in included in the signal path between the first multiplier M1 or the unit NS for noise shaping and the second multiplier M2.
  • the analog/digital converter A/D can thus also be referred to as a floating-point A/D converter.
  • the optional first correction unit Corr1 is preferably used to correct any remaining incorrect sample values due to different filter characteristics between the digital/analog conversion and the Compensate for analog/digital conversion, in particular when the factor gz(m) is determined only by inverting the first amplification factor g 1 (k) or does not correspond precisely enough to the delayed, filtered and inverted value of g 1 (k).
  • the circuit can be precisely measured so that the required correction can be calculated and applied. Another possibility is to work with predicted samples at the points that are critical in this regard.
  • the first amplification factor g 1 (k), which is applied to the digital HF carrier signal x c,h (k), can be either a constant factor or a factor that falls monotonously over time. In this case, it must only be ensured that no overmodulation (clipping) occurs in the analog circuit part AC due to the high sound pressure level.
  • the optional second correction unit Corr2 can also be provided, with which the temporary AF audio signal x ⁇ AF (m) output by the decimator Dc is further corrected and the digital AF audio signal xAF (n) is generated.
  • FIGs 6 to 8 show the preferred use of a condenser microphone with a push-pull converter.
  • a condenser microphone with standard transducer C 1 (t) and reference condenser C 2 shows figure 9 a condenser microphone with standard transducer C 1 (t) and reference condenser C 2 .
  • This standard microphone can be used instead of the push-pull microphone in the Figures 6 to 8 be used by connecting them to the contact points K1, K2, K3, each of which has the same name. The following explanations therefore apply to both types of microphones.
  • the condenser microphone is preferably operated using a high-frequency method with amplitude modulation, in particular double sideband amplitude modulation (DSB-AM).
  • DSB-AM double sideband amplitude modulation
  • the selection of the specific circuit with which the microphone is operated depends on the properties of the microphone, the intended application, the required recording quality and other criteria.
  • an (analog) DSB-AM audio signal x DSB (t) is thus preferably generated with the analog circuit part AC.
  • the analog circuit part AC includes a transformer Tr with a primary side L p and a secondary side L s , which has a center tap or is formed by two identical inductances L s1 , L s2 connected in series.
  • Parallel to the secondary side L s is a capacitive voltage divider, namely the series connection of the two capacitors C 1 (t) and C 2 (t) of the push-pull microphone or the series connection of the capacitor C 1 (t) of the standard microphone and connected to the reference capacitor C 2 , creating an RF bridge circuit in this way.
  • the RF carrier signal x c (t) generated as explained above is applied to the primary side L p of the transformer Tr.
  • the DSB-AM audio signal x DSB (t) (microphone signal with suppressed RF carrier signal x c (t)) present at the output connection of the bridge is processed according to the Figures 6 to 8 on the membrane between the two capacitors C 1 (t), C 2 (t) of the (push-pull) microphone or on the Contact point K3 decoupled.
  • the decoupled DSB-AM audio signal x DSB (t) is fed directly to the analog/digital converter A/D of the mixed analog/digital circuit part MC after impedance conversion and amplification.
  • the DSB-AM audio signal x DSB (t) is preferably coupled out via a resonant circuit tuned to the output connection of the bridge.
  • the DSB-AM audio signal x DSB (t) is coupled out as a current signal i x (t) via a resonant circuit with a series connection of an inductor L and a resistor R connected in series to the output terminal of the bridge.
  • the generated current i x (t) is according to in figure 7 shown embodiment preferably recorded or measured by means of a transimpedance amplifier, which is realized by an inverting circuit and feedback amplifier V (operational amplifier) by connecting a capacitor C f in parallel with a resistor R f , and at the output of which the DSB-AM audio signal x DSB (t) (voltage signal) is present.
  • V operation amplifier
  • the inductance L determines the resonant frequency f r of the circuit (which corresponds to the lowest source impedance), while the frequency bandwidth (which is 40 kHz for the DSB-AM audio signal) and thus the Q factor of the circuit is only determined by the resistance R becomes.
  • An advantage of this design is that due to the low source impedance, a DSB-AM audio signal x DSB (t) with improved noise characteristics can be achieved.
  • the DSB-AM audio signal x DSB (t) is coupled out as a voltage signal u x (t) via a resonant circuit with a series circuit made up of an inductor L and a resistor R connected in parallel to the output connection of the bridge.
  • the generated voltage u x (t) is calculated according to in figure 8
  • the embodiment shown is preferably recorded or measured by means of an amplifier, which is realized by an amplifier V (operational amplifier) with non-inverting wiring and feedback by a voltage divider R 1/ R 2 , and at the output of which the DSB-AM audio signal x DSB (t ) is present.
  • V operational amplifier

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Description

Die Erfindung betrifft ein Verfahren und eine Schaltungsanordnung zum Betreiben eines Kondensatormikrofons nach einem Hochfrequenzverfahren mit Amplitudenmodulation.The invention relates to a method and a circuit arrangement for operating a condenser microphone using a high-frequency method with amplitude modulation.

Bei einem Kondensatormikrofon werden Schallwellen bekanntlich in Form von Änderungen der Kapazität einer Kondensatorkapsel erfasst, die eine durch die Schallwellen bewegte Membran und mindestens eine starre Gegenelektrode aufweist. Unterschieden wird dabei u. a. zwischen Standard-Kondensatorwandlern, die nur eine Gegenelektrode aufweisen, und symmetrischen Kondensatorwandlern oder Gegentaktwandlern, bei denen die Membran zwischen zwei Gegenelektroden angeordnet ist, sodass zwei Kondensatoren entstehen. Eine durch auftreffende Schallwellen verursachte Membranauslenkung in einer Richtung führt dabei zu einer Verkleinerung des Luftspaltes zu der einen Gegenelektrode und einer gleichzeitigen Vergrößerung des Luftspaltes zu der anderen Gegenelektrode. Dies hat eine entsprechend gegenläufige Vergrößerung bzw. Verkleinerung der Kapazitäten der beiden Kondensatoren (Gegentaktwandler) zur Folge.In a condenser microphone, sound waves are known to be recorded in the form of changes in the capacitance of a condenser capsule, which has a membrane moved by the sound waves and at least one rigid counter-electrode. Among other things, a distinction is made between between standard capacitor converters, which have only one counter electrode, and balanced capacitor converters, or push-pull converters, in which the diaphragm is sandwiched between two counter electrodes, resulting in two capacitors. A membrane deflection in one direction caused by impinging sound waves leads to a reduction in the air gap to one counter-electrode and a simultaneous increase in the air gap to the other counter-electrode. This results in a correspondingly opposing increase or decrease in the capacitances of the two capacitors (push-pull converter).

Da die Masse der Membran sehr gering ist, kann sie den auftreffenden Schallwellen sehr genau und schnell folgen, sodass Kondensatormikrofone insbesondere gegenüber dynamischen Mikrofonen überlegene physikalische Eigenschaften aufweisen und damit NF-Ausgangssignale mit hoher Klangqualität, Impulstreue, Empfindlichkeit und Rauscharmut erzeugt werden können.Since the mass of the diaphragm is very low, it can follow the sound waves that strike it very quickly and precisely, so that condenser microphones have superior physical properties, especially compared to dynamic microphones, and can therefore generate AF output signals with high sound quality, impulse fidelity, sensitivity and low noise.

Allerdings benötigen Kondensatormikrofone eine externe Spannungsversorgung für die Kondensatorkapsel, um die Kapazitätsänderungen in ein Ausgangssignal umsetzen zu können. In diesem Zusammenhang werden zwei grundsätzlich unterschiedliche Verfahren unterschieden:
Bei dem Niederfrequenz-Verfahren wird die Kondensatorkapsel über einen hochohmigen Widerstand mit einer Vorspannung aufgeladen. Die sich durch die auftreffenden Schallwellen ändernde Kapazität der Kondensatorkapsel führt bei im Wesentlichen konstanter Kondensatorladung zu einer entsprechenden proportionalen Änderung der Ausgangsspannung an der Kondensatorkapsel. Dieses Verfahren hat jedoch verschiedene Nachteile. Zum einen können insbesondere in Umgebungen mit hoher Luftfeuchtigkeit Leckströme auftreten, die zu einem schlechten Signal/Rauschverhältnis führen, zum anderen ist aufgrund der geringen Kapazität der Kondensatorkapsel ein Impedanzwandler mit sehr hohem Eingangswiderstand für die weitere Verarbeitung der Ausgangsspannung erforderlich.
However, condenser microphones require an external power supply for the condenser capsule in order to be able to convert the changes in capacitance into an output signal. In this context, a distinction is made between two fundamentally different procedures:
In the low-frequency process, the capacitor capsule is charged with a bias voltage via a high-impedance resistor. Which are caused by the impinging sound waves Changing capacitance of the capacitor capsule leads to a corresponding proportional change in the output voltage at the capacitor capsule when the capacitor charge is essentially constant. However, this method has various disadvantages. On the one hand, leakage currents can occur, especially in environments with high humidity, which lead to a poor signal-to-noise ratio. On the other hand, the low capacitance of the capacitor capsule requires an impedance converter with a very high input resistance for further processing of the output voltage.

Zur Vermeidung dieser Nachteile wird bevorzugt eines der bekannten HochfrequenzVerfahren angewendet, bei dem die sich ändernde Kapazität der Kondensatorkapsel zur Modulation der Frequenz, der Phase oder der Amplitude eines hochfrequenten elektrischen Trägersignals (HF-Signal) benutzt wird.To avoid these disadvantages, one of the known high-frequency methods is preferably used, in which the changing capacitance of the capacitor capsule is used to modulate the frequency, the phase or the amplitude of a high-frequency electrical carrier signal (HF signal).

Bei der Frequenzmodulation ist die Kondensatorkapsel Teil eines Resonanzkreises, sodass die durch die Schallwellen verursachten Änderungen der Kapazität der Kondensatorkapsel zu entsprechenden Änderungen der Resonanzfrequenz führen. Dieses Verfahren erfordert jedoch einen sehr frequenzstabilen Oszillator, damit das nach der Demodulation gewonnene Audiosignal nicht gestört ist.In frequency modulation, the condenser capsule is part of a resonant circuit, so changes in the capacitance of the condenser capsule caused by the sound waves result in corresponding changes in the resonant frequency. However, this method requires a very frequency-stable oscillator so that the audio signal obtained after demodulation is not disturbed.

Bei der Phasenmodulation ist die Kondensatorkapsel Teil eines LC Parallel-Resonanzkreises, sodass die Änderungen der Kapazität der Kondensatorkapsel zu entsprechenden Änderungen der Resonanzfrequenz des LC-Kreises und damit zu Phasenverschiebungen des extern erzeugten und eingekoppelten HF-Signals führen. Auch für dieses Verfahren ist ein sehr frequenzstabiles HF-Signal sowie zur Erzielung einer großen Phasenverschiebung bzw. Mikrofonempfindlichkeit ein hoher Q-Faktor des LC-Kreises erforderlich. Ein weiterer Nachteil besteht darin, dass die Phasen-Antwort nur einen relativ geringen linearen Bereich aufweist, sodass im Falle eines hohen Schalldruckpegels die Änderung der Kapazität und die sich ergebenden Phasenverschiebungen nicht mehr proportional zueinander sind.With phase modulation, the capacitor capsule is part of an LC parallel resonant circuit, so that the changes in the capacitance of the capacitor capsule lead to corresponding changes in the resonant frequency of the LC circuit and thus to phase shifts in the externally generated and coupled RF signal. This method also requires a very frequency-stable HF signal and a high Q factor of the LC circuit to achieve a large phase shift or microphone sensitivity. Another disadvantage is that the phase response has a relatively small linear range, so that in the case of a high sound pressure level, the change in capacitance and the resulting phase shifts are no longer proportional to each other.

Bei der Amplitudenmodulation bildet die Kondensatorkapsel schließlich zusammen mit zwei gleichartigen Wicklungen eine HF-Transformator-Brückenschaltung, die auf die Frequenz eines HF-Trägersignals abgestimmt ist, das von einem über eine weitere Wicklung angekoppelten HF-Oszillator erzeugt wird. Durch die Bewegung der Membran und die dadurch bewirkte Veränderung der Kapazität kann am Brückenzweig der Schaltung eine dazu proportionale Doppelseitenband-amplitudenmodulierte (DSB-AM) Audio-Ausgangsspannung mit unterdrücktem HF-Trägersignal erzeugt werden.Finally, in amplitude modulation, the capacitor capsule together with two windings of the same type forms an HF transformer bridge circuit which is tuned to the frequency of an HF carrier signal which is generated by an HF oscillator which is coupled via another winding. Due to the movement of the membrane and the resulting change in capacitance, a proportional double sideband amplitude modulated (DSB-AM) audio output voltage with a suppressed HF carrier signal can be generated at the bridge branch of the circuit.

Dieses Verfahren ist insbesondere für die eingangs genannten symmetrischen Kondensatorwandler oder Gegentaktwandler geeignet, bei denen die beiden Kondensatoren als kapazitiver Spannungsteiler in einen der Brückenarme geschaltet werden.This method is particularly suitable for the symmetrical capacitor converters or push-pull converters mentioned at the outset, in which the two capacitors are connected as a capacitive voltage divider in one of the bridge arms.

Aus der DE 4300379A1 und der DE 102010000686A1 sind Beispiele für solche HF-Transformator-Brückenschaltungen mit symmetrischen Kondensatorwandlern bekannt.From the DE 4300379A1 and the DE 102010000686A1 examples of such HF transformer bridge circuits with symmetrical capacitor converters are known.

Bei Standard-Kondensatorwandlern muss hingegen für den Spannungsteiler ein entsprechender Referenz-Kondensator in den Brückenarm geschaltet werden.In the case of standard capacitor converters, on the other hand, a corresponding reference capacitor must be switched into the bridge arm for the voltage divider.

Aus der US 6,697,493 B1 ist ein Verfahren und eine Schaltungsanordnung bekannt, mit dem/der ein auf einen kapazitiven Schallwandler auftreffendes akustisches Signal in ein digitales Signal umgewandelt wird. Dazu wird der Schallwandler mit einem Gegen-Signal in der Weise beaufschlagt, dass dieser im Wesentlichen in seinem Ruhezustand verbleibt, wenn auf diesen ein akustisches Signal auftrifft. Jede Auslenkung aus dem Ruhezustand in der einen oder anderen Richtung wird in ein digitales Signal "1" bzw. "0" umgesetzt, aus dem das Gegen-Signal abgeleitet wird und das Informationen über das akustische Signal enthält.From the US 6,697,493 B1 a method and a circuit arrangement are known with which an acoustic signal impinging on a capacitive sound transducer is converted into a digital signal. For this purpose, the sound transducer is subjected to a counter-signal in such a way that it essentially remains in its idle state when an acoustic signal strikes it. Each deflection from the resting state in one direction or the other is converted into a digital signal "1" or "0" from which the counter-signal is derived and which contains information about the acoustic signal.

In der US 2008/0219474 A1 wird eine integrierte Schaltung beschrieben, die zur Erzeugung eines Mikrofon-Ausgangssignals einen Vorverstärker für ein mit einer Mikrofonmembran oder einem Mikrofonelement, demgegenüber die Membran beweglich ist, erzeugtes Eingangssignal, eine Ladungspumpe und einen Tiefpassfilter aufweist, wobei der Tiefpassfilter zur Filterung der Pumpspannung und zur Erzeugung einer Vorspannung für die Membran oder das Mikrofonelement vorgesehen ist.In the U.S. 2008/0219474 A1 an integrated circuit is described which, to generate a microphone output signal, has a preamplifier for an input signal generated with a microphone membrane or a microphone element, in relation to which the membrane is movable, a charge pump and a low-pass filter, the low-pass filter is provided for filtering the pump voltage and for generating a bias voltage for the membrane or the microphone element.

In " IEEE Sensors Journal", Vol. 11, Nr. 2, Februar 2011, Seiten 296 bis 304 wird ein digitales CMOS-Silizium-Kondensatormikrofon beschrieben, bei dem der akustische Sensor zusammen mit den elektrischen Schaltkreisen auf einem Chip integriert ist. Alle Komponenten werden mit der gleichen Versorgungsspannung betrieben, wobei eine Ladungspumpe nicht erforderlich ist.In " IEEE Sensors Journal", Vol. 11, No. 2, February 2011, pages 296 to 304 describes a digital CMOS silicon condenser microphone in which the acoustic sensor is integrated together with the electrical circuits on one chip. All components are operated with the same supply voltage, whereby a charge pump is not required.

Der Erfindung liegt die Aufgabe zugrunde, ein Verfahren und eine Schaltungsanordnung zum Betreiben eines Kondensatormikrofons zu schaffen, mit dem/der die oben genannten überlegenen physikalischen Eigenschaften eines solchen Mikrofons noch besser ausgenutzt und in NF-Ausgangssignale mit hoher Klangtreue umgesetzt werden können. Dabei soll insbesondere ein digitales Mikrofon mit kleinstmöglichem analogen Aufwand geschaffen werden, bei dem der Dynamikbereich und das Signal/Rauschverhältnis der NF-Ausgangssignale mindestens gleichwertig oder sogar besser ist als bei bekannten, analog betriebenen Mikrofonen.The invention is based on the object of creating a method and a circuit arrangement for operating a condenser microphone with which the above-mentioned superior physical properties of such a microphone can be better utilized and converted into AF output signals with high sound fidelity. In particular, a digital microphone is to be created with the smallest possible analog outlay, in which the dynamic range and the signal-to-noise ratio of the AF output signals is at least equivalent to or even better than in known, analog-operated microphones.

Gelöst wird diese Aufgabe mit einem Verfahren gemäß Anspruch 1 und einer Schaltungsanordnung gemäß Anspruch 8.This object is achieved with a method according to claim 1 and a circuit arrangement according to claim 8.

Ein wesentliches Erfindungsprinzip besteht demnach darin, durch digitale Signalverarbeitung in Kombination mit einer geeigneten analogen Schaltung zur Erfassung der durch auftreffende Schallwellen erzeugten Kapazitätsänderungen ein NF-Ausgangssignal zu gewinnen, das sich insbesondere durch einen hohen Dynamikbereich und ein hohes Signal/Rauschverhältnis auszeichnet.An essential principle of the invention is to use digital signal processing in combination with a suitable analog circuit for detecting the changes in capacitance generated by impinging sound waves to obtain an AF output signal that is characterized in particular by a high dynamic range and a high signal-to-noise ratio.

Vorteile der erfindungsgemäßen Lösungen bestehen darin, dass durch die reduzierte Anzahl an analogen Komponenten und insbesondere durch den Einsatz eines digitalen synchronen Demodulators keine spürbaren Nichtlinearitäten eingebracht werden. Ferner wird auch das elektrische 1/f-Rauschen anhand der analogen Bandpassverarbeitung weitgehend reduziert bzw. vermieden.Advantages of the solutions according to the invention are that no noticeable non-linearities are introduced due to the reduced number of analog components and in particular due to the use of a digital synchronous demodulator. Furthermore, that will also Electrical 1/f noise is largely reduced or avoided using analog bandpass processing.

Durch die digitale Erzeugung eines (digitalen) HF-Trägersignals insbesondere in Kombination mit einer möglichst hohen Abtastrate und/oder mit einer möglichst hohen Auflösung bzw. großen Wortbreite kann die Frequenz des analogen HF-Trägersignals sehr exakt auf die Resonanz des analogen Schaltungsteils abgestimmt werden, und es kann außerdem ein spektral sehr reines HF-Trägersignal erzeugt werden.Through the digital generation of a (digital) HF carrier signal, in particular in combination with the highest possible sampling rate and/or with the highest possible resolution or large word width, the frequency of the analog HF carrier signal can be very precisely matched to the resonance of the analog circuit part, and a spectrally very pure RF carrier signal can also be generated.

Die abhängigen Ansprüche haben vorteilhafte Weiterbildungen der Erfindung zum Inhalt.The dependent claims relate to advantageous developments of the invention.

Da Digital/Analog-Wandler üblicherweise eine begrenzte Auflösung aufweisen, wird das digitale HF-Trägersignal vor der Digital/Analogwandlung vorzugsweise einer Rauschformung unterworfen, mit der nicht nur die Auflösung bzw. Wortbreite reduziert, sondern auch das Quantisierungsrauschen in Frequenzbereiche verschoben wird, die zur Erfassung der relevanten NF-Bandbreite des Mikrofons nicht erforderlich sind. Dadurch bleibt in dem relevanten Frequenzband ein hohes Signal/Rauschverhältnis erhalten.Since digital/analog converters usually have a limited resolution, the digital HF carrier signal is preferably subjected to noise shaping before the digital/analog conversion, which not only reduces the resolution or word width, but also shifts the quantization noise into frequency ranges that Capturing the relevant LF bandwidth of the microphone are not required. As a result, a high signal-to-noise ratio is maintained in the relevant frequency band.

Vorzugsweise ist die Amplitude des digitalen HF-Trägersignals einstellbar, um eine Veränderung bzw. Anpassung der Empfindlichkeit des Mikrofons insbesondere in Abhängigkeit von einem auftretenden Schalldruckpegel vorzunehmen. Somit kann die Empfindlichkeit digital eingestellt werden, und es sind keine Mehrkanal-Analog/Digital-Wandler erforderlich, um einen hohen Dynamikbereich bzw. eine hohe Verstärkung zu erzielen.The amplitude of the digital HF carrier signal can preferably be adjusted in order to change or adapt the sensitivity of the microphone, in particular as a function of a sound pressure level that occurs. Thus, the sensitivity can be adjusted digitally and multi-channel analog-to-digital converters are not required to achieve high dynamic range or high gain.

Bevorzugt wird diese Amplitude mittels eines Schalldruckpegel-Detektors/Prädiktors in Abhängigkeit von einem erfassten oder vorhergesagten Schalldruckpegel automatisch so verändert, dass eine entsprechende Anpassung der Empfindlichkeit des Mikrofons vorgenommen und damit eine Übersteuerung des analogen Schaltungsteils vermieden wird. Schließlich kann auch dadurch, dass das erfasste Audiosignal bevorzugt überabgetastet digitalisiert und nach der Demodulation auf eine geringere Abtastfrequenz heruntergetaktet wird (zum Beispiel auf eine Standard-Abtastfrequenz von 48 kHz), ein zusätzlicher Signal/Rauschabstand und Dynamikbereich gewonnen werden.This amplitude is preferably changed automatically by means of a sound pressure level detector/predictor as a function of a detected or predicted sound pressure level, so that the sensitivity of the microphone is adjusted accordingly and overloading of the analog circuit part is thus avoided. Finally, an additional signal-to-noise ratio and dynamic range can also be obtained by the fact that the recorded audio signal is digitized, preferably oversampled, and after demodulation is clocked down to a lower sampling frequency (for example to a standard sampling frequency of 48 kHz).

Weitere Einzelheiten, Merkmale und Vorteile der Erfindung ergeben sich aus der folgenden Beschreibung von beispielhaften und bevorzugten Ausführungsformen anhand der Zeichnung. Es zeigt:

Fig. 1
eine bekannte HF-AM-Schaltung mit einem Kondensatormikrofon mit Gegentaktwandler;
Fig. 2
ein Blockschaltbild einer ersten Ausführungsform einer erfindungsgemäßen Schaltungsanordnung;
Fig. 3
eine erste Komponente des digitalen Schaltungsteils;
Fig. 4
eine zweite Komponente des digitalen Schaltungsteils;
Fig. 5
ein Blockschaltbild einer zweiten Ausführungsform einer erfindungsgemäßen Schaltungsanordnung;
Fig. 6
eine erste Ausführungsform eines analogen Schaltungsteils mit Gegentakt-Kondensatormikrofon;
Fig. 7
eine zweite Ausführungsform eines analogen Schaltungsteils mit Gegentakt-Kondensatormikrofon;
Fig. 8
eine dritte Ausführungsform eines analogen Schaltungsteils mit Gegentakt-Kondensatormikrofon; und
Fig. 9
ein alternativ in den Schaltungsteilen gemäß den Figuren 6 bis 8 einsetzbares Kondensatormikrofon mit Standardwandler.
Further details, features and advantages of the invention result from the following description of exemplary and preferred embodiments with reference to the drawing. It shows:
1
a known RF AM circuit with a push-pull condenser microphone;
2
a block diagram of a first embodiment of a circuit arrangement according to the invention;
3
a first component of the digital circuit portion;
4
a second component of the digital circuit portion;
figure 5
a block diagram of a second embodiment of a circuit arrangement according to the invention;
6
a first embodiment of an analog circuit part with a push-pull condenser microphone;
7
a second embodiment of an analog circuit part with a push-pull condenser microphone;
8
a third embodiment of an analog circuit part with a push-pull condenser microphone; and
9
an alternative in the circuit parts according to the Figures 6 to 8 Deployable condenser microphone with standard converter.

Da in dem analogen Schaltungsteil der erfindungsgemäßen Schaltungsanordnung bevorzugt ein Kondensatormikrofon eingesetzt und dieses bevorzugt nach dem eingangs genannten Hochfrequenzverfahren mit Amplitudenmodulation betrieben wird, soll zunächst die grundsätzliche Funktion und Realisierung dieser Betriebsart anhand der Figur 1 erläutert werden.Since a condenser microphone is preferably used in the analog circuit part of the circuit arrangement according to the invention and this is preferably operated according to the high-frequency method mentioned at the outset with amplitude modulation, the basic function and implementation of this mode of operation should first be described using the figure 1 be explained.

Figur 1 zeigt eine beispielhafte Schaltungsanordnung mit einem Kondensatormikrofon mit symmetrischem Wandler (Gegentaktwandler), der einen ersten und einen zweiten Kondensator C1(t), C2(t) aufweist, deren Kapazitäten wie oben erläutert durch eintreffende Schallwellen s(t) gegenläufig verändert werden. figure 1 shows an exemplary circuit arrangement with a condenser microphone with a symmetrical converter (push-pull converter), which has a first and a second capacitor C 1 (t), C 2 (t), whose capacitances are changed in opposite directions as explained above by incoming sound waves s(t).

Die Schaltungsanordnung arbeitet nach dem eingangs genannten, im Zusammenhang mit der Erfindung bevorzugt angewendeten Hochfrequenzverfahren mit Amplitudenmodulation und umfasst einen ersten Übertrager Tr1, einen zweiten Übertrager Tr2, einen HF-Generator HF, eine erste und eine zweite Diode D1, D2 sowie einen Ladekondensator Ca.The circuit arrangement works according to the high-frequency method with amplitude modulation mentioned at the outset, which is preferably used in connection with the invention, and comprises a first transformer Tr1, a second transformer Tr2, an HF generator HF, a first and a second diode D1, D2 and a charging capacitor Ca.

Auf der Primärseite des ersten Übertragers Tr1 befindet sich eine erste Wicklung W1, über die die von dem HF-Generator HF erzeugte HF-Spannung (HF-Trägersignal) eingekoppelt wird.A first winding W1 is located on the primary side of the first transformer Tr1, via which the HF voltage (HF carrier signal) generated by the HF generator HF is injected.

Auf einer ersten Sekundärseite weist der erste Übertrager Tr1 eine zweite und eine gleiche dritte Wicklung W2, W3 auf. Die beiden Wicklungen W2, W3 bilden zusammen mit den beiden Kondensatoren C1(t), C2(t) eine Brückenschaltung. Auf einer zweiten Sekundärseite des ersten Übertragers Tr1 befindet sich eine vierte und eine gleiche fünfte Wicklung W4, W5, deren äußere Enden über zwei in Reihe geschaltete Dioden D1, D2 miteinander verbunden sind und die zusammen einen Synchrongleichrichter bilden.On a first secondary side, the first transformer Tr1 has a second and an identical third winding W2, W3. The two windings W2, W3 together with the two capacitors C 1 (t), C 2 (t) form a bridge circuit. A fourth and an identical fifth winding W4, W5 is located on a second secondary side of the first transformer Tr1, the outer ends of which are connected to one another via two diodes D1, D2 connected in series and which together form a synchronous rectifier.

Der zweite Übertrager Tr2 liegt schließlich mit einer Seite zwischen dem Mittenabgriff zwischen der zweiten und der dritten Wicklung W2, W3 und Masse, während die andere Seite zwischen den Mittenabgriff zwischen der vierten und der fünften Wicklung W4, W5 und einen der Ausgangsanschlüsse geschaltet ist. Der andere Ausgangsanschluss wird zwischen den beiden Dioden D1, D2 abgegriffen. Parallel zu den Ausgangsanschlüssen ist der Ladekondensator Ca geschaltet, an dem die Ausgangsspannung xAF(t) in Form des demodulierten Audiosignals (Basisband-Signal) anliegt.Finally, the second transformer Tr2 has one side connected between the center tap between the second and third windings W2, W3 and ground, while the other side is connected between the center tap between the fourth and fifth windings W4, W5 and one of the output terminals. The other output connection is tapped between the two diodes D1, D2. The charging capacitor Ca is connected in parallel with the output terminals, and the output voltage x AF (t) is applied to this capacitor in the form of the demodulated audio signal (baseband signal).

Die durch den ersten und den zweiten Kondensator C1(t), C2(t) sowie die zweite und die dritte Wicklung W2, W3 gebildete HF-Brückenschaltung ist auf die Frequenz des von dem HF-Generator HF erzeugten HF-Trägersignals abgestimmt. Solange sich die Membran der Kondensatorkapsel in der mittleren Ruhelage befindet (d.h. keine Schallwellen auftreffen), ist die Brücke abgeglichen, und an deren Ausgang entsteht keine HF-Spannung. Wenn die Membran durch eintreffende Schallwellen s(t) ausgelenkt wird, entsteht eine zu der Membranauslenkung proportionale HF-Spannung, die zur Demodulation über den Synchrongleichrichter D1, D2 auf den Ladekondensator Ca geführt wird und an dem dann die zu der Membranbewegung in Betrag und Phase proportionale Ausgangsspannung xAF(t) anliegt.The HF bridge circuit formed by the first and the second capacitor C 1 (t), C 2 (t) and the second and the third winding W2, W3 is tuned to the frequency of the HF carrier signal generated by the HF generator HF. As long as the membrane of the condenser capsule is in the middle rest position (ie no sound waves impinge), the bridge is balanced and no HF voltage is produced at its output. When the membrane is deflected by incoming sound waves s(t), an HF voltage is produced that is proportional to the membrane deflection proportional output voltage x AF (t) is present.

Figur 2 zeigt ein schematisches Blockschaltbild einer beispielhaften ersten Ausführungsform einer erfindungsgemäßen Schaltungsanordnung, die sich im Wesentlichen aus drei Komponenten, nämlich einem digitalen Schaltungsteil D-C, einem analogen Schaltungsteil A-C und einem gemischt analog/digitalen Schaltungsteil M-C zusammensetzt, wobei letzterer zwischen den digitalen und den analogen Schaltungsteil geschaltet ist und beide miteinander verbindet. figure 2 shows a schematic block diagram of an exemplary first embodiment of a circuit arrangement according to the invention, which essentially consists of three components, namely a digital circuit part DC, an analog circuit part AC and a mixed analog/digital circuit part MC, the latter being connected between the digital and the analog circuit part is and connects both together.

Der digitale Schaltungsteil D-C umfasst die digitalen Komponenten der Schaltungsanordnung. Dieses sind im Wesentlichen ein HF-Generator CG zur Erzeugung eines digitalen überabgetasteten HF-Trägersignals xc,h(k) mit einer ersten hohen Abtastrate fS,1 sowie mit einer ersten hohen Auflösung, die einer ersten großen Wortbreite von w0 Bit entspricht, eine optionale Einheit NS zur Rauschformung des HF-Trägersignals xc,h(k) und zur Erzeugung eines HF-Trägersignals xc(k) mit gleicher erster Abtastrate fS,1 und mit einer gegenüber der ersten Auflösung geringeren zweiten Auflösung, die einer entsprechend geringeren zweiten Wortbreite von w1 Bit entspricht.The digital circuit part DC includes the digital components of the circuit arrangement. These are essentially an HF generator CG for generating a digital oversampled HF carrier signal x c,h (k) with a first high sampling rate f S,1 and with a first high resolution, which corresponds to a first large word length of w 0 bits , an optional unit NS for noise shaping of the HF carrier signal x c,h (k) and for generating an HF carrier signal x c (k) with the same first sampling rate f S,1 and with a second resolution that is lower than the first resolution, the corresponds to a correspondingly smaller second word width of w 1 bit.

Der digitale Schaltungsteil D-C umfasst ferner einen Demodulator Dm zur Demodulation eines anliegenden digitalen DSB-AM Audio-Ausgangssignals xDSB(m), das eine zweite Abtastrate fS,2 und eine dritte Auflösung bzw. dritte Wortbreite von w2 Bit aufweist, und zur Erzeugung eines demodulierten digitalen (Basisband-) Audiosignals xdem(m) mit unveränderter (zweiter) Abtastrate fS,2 und einer vierten Auflösung bzw. vierten Wortbreite von w3 Bit. Der digitale Schaltungsteil umfasst schließlich einen optionalen Dezimator Dc zur Dezimation bzw. Verminderung der zweiten Abtastrate fS,2 des demodulierten Audiosignals xdem(m) und zur Erzeugung eines digitalen Audio-Ausgangssignals xAF(n) mit einer Standard-Audio-Abtastfrequenz von zum Beispiel 48 kHz (dritte Abtastrate fs) mit einer gewünschten fünften Auflösung bzw. einer fünften Wortbreite von w4 Bit.The digital circuit part DC also includes a demodulator Dm for demodulating an applied digital DSB-AM audio output signal x DSB (m), which has a second sampling rate f S,2 and a third resolution or third word length of w 2 bits, and for Generation of a demodulated digital (baseband) audio signal x dem (m) with unchanged (second) sampling rate f S,2 and a fourth resolution or fourth word width of w 3 bits. Finally, the digital circuit part includes an optional decimator Dc for decimation or reduction of the second sampling rate f S,2 of the demodulated audio signal x dem (m) and for generating a digital audio output signal x AF (n) with a standard audio sampling frequency of for example 48 kHz (third sampling rate fs) with a desired fifth resolution or a fifth word width of w 4 bits.

Dieser digitale Schaltungsteil D-C wird vorzugsweise in Form eines FPGA (fieldprogrammable gate array) realisiert und stellt einen digitalen Signalprozessor (DSP) dar. Die Realisierung erfolgt vorzugsweise durch ein entsprechendes Programm, das auf einem System zur Datenverarbeitung ausgeführt wird.This digital circuit part D-C is preferably implemented in the form of an FPGA (field programmable gate array) and represents a digital signal processor (DSP). It is preferably implemented using a corresponding program which is executed on a data processing system.

Der analog/digitale Schaltungsteil M-C umfasst einen Digital/Analog-Wandler D/A zur Erzeugung eines analogen HF-Trägersignals xc(t) aus dem zugeführten digitalen HF-Trägersignal xc(k), sowie einen Analog/Digital-Wandler A/D zur Erzeugung eines digitalen DSB-AM Audiosignals xDSB(m) mit der genannten dritten Auflösung bzw. dritten Wortbreite von w2 Bit aus einem zugeführten analogen DSB-AM Audiosignal xDSB(t), das mit der Mikrofoneinheit Cm erzeugt wird.The analog/digital circuit part MC comprises a digital/analog converter D/A for generating an analog HF carrier signal x c (t) from the supplied digital HF carrier signal x c (k), and an analog/digital converter A/ D for generating a digital DSB-AM audio signal x DSB (m) with said third resolution or third word length of w 2 bits from a supplied analog DSB-AM audio signal x DSB (t), which is generated with the microphone unit Cm.

Der analoge Schaltungsteil A-C beinhaltet schließlich die Mikrofoneinheit Cm mit einem Kondensatormikrofon und dient dazu, die durch die auftreffenden Schallwellen s(t) (NF-Audiosignal) erzeugten Änderungen der Kapazität des Mikrofons mittels des zugeführten analogen HF-Trägersignals xc(t) in das analoge DSB-AM Audiosignal xDSB(t) mit unterdrücktem HF-Trägersignal xc(t) umzusetzen.Finally, the analog circuit part AC contains the microphone unit Cm with a condenser microphone and is used to convert the changes in the capacitance of the microphone generated by the incident sound waves s(t) (LF audio signal) by means of the analog HF carrier signal x c (t) supplied into the convert analog DSB-AM audio signal x DSB (t) with suppressed HF carrier signal x c (t).

Diese Schaltungsanordnung arbeitet im Detail wie folgt:
Der HF-Generator CG erzeugt ein über-abgetastetes (oversampled) digitales HF-Trägersignal xc,h(k) mit der genannten ersten hohen Abtastrate fS,1 und der ersten hohen Auflösung, die der ersten großen Wortbreite von w0 Bit entspricht.
This circuit arrangement works in detail as follows:
The HF generator CG generates an oversampled (oversampled) digital HF carrier signal x c,h (k) with said first high sampling rate f S,1 and the first high resolution, which corresponds to the first large word width of w 0 bits .

Die Höhe der Trägerfrequenz fr des HF-Trägersignal xc,h(k) wird insbesondere in Abhängigkeit von den elektrischen Eigenschaften des analogen Schaltungsteils A-C (insbesondere dessen Resonanzfrequenz, siehe unten) geeignet gewählt.The level of the carrier frequency f r of the HF carrier signal x c,h (k) is suitably selected in particular as a function of the electrical properties of the analog circuit part AC (in particular its resonant frequency, see below).

Unter dem Begriff "Über-Abtastung" ist zu verstehen, dass das Signal mit einer Abtastrate erzeugt wird, die höher ist, als sie gemäß dem Abtasttheorem zur Erfassung der HF-Trägerfrequenz fr erforderlich ist (auch Oversampling genannt). Die (erste) Abtastrate fS,1 des HF-Trägersignals xc,h(k) mit der Frequenz fr ist demzufolge größer als die nach dem Abtasttheorem benötigte Abtastrate von 2fr.The term "oversampling" means that the signal is generated with a sampling rate that is higher than that required according to the sampling theorem for detecting the RF carrier frequency f r (also called oversampling). The (first) sampling rate f S,1 of the HF carrier signal x c,h (k) with the frequency f r is consequently greater than the sampling rate of 2 * f r required according to the sampling theorem.

Die genannte erste hohe Auflösung des HF-Trägersignals xc,h(k) (und die damit verbundene erste große Wortbreite von wo Bit) wird so groß gewählt, dass das Signal/Rauschverhältnis im relevanten Frequenzbereich hinreichend hoch ist und auch dann noch groß genug ist, wenn eine Verstärkungsanpassung des HF-Trägersignals erfolgt. Dieses Prinzip wird nachfolgend anhand der Figur 5 noch erläutert werden.The first high resolution of the HF carrier signal x c,h (k) mentioned (and the associated first large word length of w o bits) is selected so large that the signal/noise ratio in the relevant frequency range is sufficiently high and even then still large is enough if there is a gain adjustment of the RF carrier signal. This principle is explained below with reference to figure 5 still to be explained.

Ein solcher HF-Generator CG kann zum Beispiel ein Synthesizer sein, der nach dem bekannten DDS (direct digital synthesis) Verfahren arbeitet, oder in Form einer Nachschlagetabelle (LUT - look-up table) realisiert werden. Ein DDS-Synthesizer hat den Vorteil, dass die Trägerfrequenz in einfacher Weise an die elektrischen Eigenschaften wie zum Beispiel die Resonanzfrequenz des analogen Schaltungsteils A-C angepasst werden kann, während bei Einsatz einer Nachschlagetabelle der digitale Schaltungsaufwand und damit die Kosten für die Hardware geringer sind. Darüber hinaus sind weitere HF-Generatoren denkbar, die die genannten Anforderungen erfüllen.Such an HF generator CG can, for example, be a synthesizer that works according to the known DDS (direct digital synthesis) method, or can be implemented in the form of a look-up table (LUT). A DDS synthesizer has the advantage that the carrier frequency can easily be adapted to the electrical properties such as the resonant frequency of the analog circuit part AC, while using a look-up table requires less digital circuitry and thus lower hardware costs. In addition, other HF generators are conceivable that meet the stated requirements.

Da Digital/Analog-Wandler mit hoher Abtastrate üblicherweise eine relativ begrenzte Auflösung aufweisen, wird die relativ große erste Wortbreite von w0 Bit des HF-Trägersignals xc,h(k) vorzugsweise entsprechend auf die zweite Wortbreite von w1 Bit reduziert.Since digital/analog converters with a high sampling rate usually have a relatively limited resolution, the relatively large first word length of w 0 bits of the RF carrier signal x c,h (k) is preferably correspondingly reduced to the second word length of w 1 bits.

Um dies zu erreichen, ohne das Signal/Rauschverhältnis in dem relevanten Frequenzband wesentlich zu vermindern, wird das digitale HF-Trägersignal xc,h(k) zunächst vorzugsweise der Einheit NS zur Rauschformung (noise shaping) zugeführt, mit der das Quantisierungsrauschen in an sich bekannter Weise in Bereichen außerhalb des relevanten Frequenzbandes konzentriert und damit in Frequenzbereiche verschoben wird, die zur Erfassung der relevanten NF-Bandbreite (d.h. üblicherweise etwa 40 kHz symmetrisch um die Trägerfrequenz) nicht erforderlich sind. Dadurch bleibt in dem relevanten Frequenzband ein hohes Signal/Rauschverhältnis erhalten, und es wird ein entsprechend dem Auflösevermögen des Digital/Analog-Wandlers D/A re-quantifiziertes HF-Trägersignal xc(k) mit der reduzierten zweiten Wortbreite von w1 Bit erzeugt.In order to achieve this without significantly reducing the signal-to-noise ratio in the relevant frequency band, the digital HF carrier signal x c,h (k) is first preferably fed to the unit NS for noise shaping (noise shaping), with which the quantization noise in an is, as is known, concentrated in areas outside the relevant frequency band and is thus shifted into frequency ranges that are not required to cover the relevant AF bandwidth (ie usually around 40 kHz symmetrically around the carrier frequency). As a result, a high signal-to-noise ratio is maintained in the relevant frequency band, and an RF carrier signal x c (k) re-quantified according to the resolution of the digital/analog converter D/A is generated with the reduced second word length of w 1 bit .

Figur 3 zeigt den HF-Generator CG und ein mögliches Ersatzschaltbild der Einheit NS zur Rauschformung. Die damit realisierte z-Übertragungsfunktion lautet: X c z = X c , h z + E z * 1 H z

Figure imgb0001
figure 3 shows the HF generator CG and a possible equivalent circuit diagram of the unit NS for noise shaping. The thus realized z-transfer function is: X c e.g = X c , H e.g + E e.g * 1 H e.g
Figure imgb0001

Dies bedeutet, dass der Quantisierungsfehler e(k) durch H(z) spektral gewichtet beaufschlagt wird und somit der Re-Quantisierungsfehler für das stark über-abgetastete HF-Trägersignal in dem relevanten Frequenzband minimiert werden kann.This means that the quantization error e(k) is spectrally weighted by H(z) and the re-quantization error for the heavily oversampled HF carrier signal can thus be minimized in the relevant frequency band.

Wenn man hingegen (je nach Anwendungsfall) ein etwas geringeres Signal/Rauschverhältnis in dem relevanten Frequenzband in Kauf nehmen kann, so kann alternativ der HF-Generator CG auch so dimensioniert werden, dass er ein digitales HF-Trägersignal xc(k) mit einer reduzierten, an das Auflösevermögen des Digital/Analog-Wandlers angepassten zweiten Wortbreite von w1 Bit erzeugt, sodass man dann auf die Einheit NS zur Rauschformung verzichten kann.On the other hand, if (depending on the application) you can accept a slightly lower signal-to-noise ratio in the relevant frequency band, the HF generator CG can alternatively be dimensioned in such a way that it generates a digital HF carrier signal x c (k) with a reduced second word width of w 1 bit, adapted to the resolution of the digital/analog converter, so that the unit NS for noise shaping can then be dispensed with.

In beiden Fällen wird das digitale HF-Trägersignal xc(k) dem Digital/Analog-Wandler D/A zugeführt, mit dem daraus in an sich bekannter Weise ein analoges HF-Trägersignal xc(t) erzeugt wird.In both cases, the digital HF carrier signal x c (k) is fed to the digital/analog converter D/A, with which an analog HF carrier signal x c (t) is generated therefrom in a manner known per se.

Dieses analoge HF-Trägersignal xc(t) wird der Mikrofoneinheit Cm zugeführt und dient dazu, die durch die auftreffenden Schallwellen s(t) (NF-Audiosignal) erzeugten Änderungen der Kapazität des Mikrofons in ein DSB-AM Audiosignal xDSB(t) umzusetzen, und zwar vorzugsweise nach einem Hochfrequenzverfahren mit Amplitudenmodulation und unterdrückter Trägerfrequenz. Hierzu wird Bezug genommen auf die weiter unten noch folgenden Erläuterungen im Zusammenhang mit den Figuren 6 bis 9.This analog HF carrier signal x c (t) is fed to the microphone unit Cm and is used to convert the changes in the capacitance of the microphone caused by the incident sound waves s(t) (LF audio signal) into a DSB-AM audio signal x DSB (t) implement, preferably using a high-frequency method with amplitude modulation and suppressed carrier frequency. In this regard, reference is made to the explanations below in connection with the Figures 6 to 9 .

Das von der Mikrofoneinheit Cm erzeugte analoge DSB-AM Audiosignal xDSB(t) wird dann dem Analog/Digital-Wandler A/D zugeführt. Um ein qualitativ hochwertiges digitales Audiosignal zu erzeugen, hat dieser die genannte möglichst hohe (dritte) Auflösung mit der dritten Wortbreite von w2 Bit. Ferner wird das DSB-AM Audiosignal xDSB(t), da es sich um ein Bandpass-Signal handelt, mit der genannten zweiten Abtastrate fS,2 (über-) abgetastet (oversampling), die wiederum höher ist, als es nach dem Abtasttheorem für die in dem Bandpass-signal vorkommende Bandbreite von 40 kHz erforderlich ist.The analog DSB-AM audio signal x DSB (t) generated by the microphone unit Cm is then fed to the analog/digital converter A/D. In order to generate a high-quality digital audio signal, this has the highest possible (third) resolution mentioned with the third word width of w 2 bits. Furthermore, the DSB-AM audio signal x DSB (t), since it is a bandpass signal, is (over)sampled with the said second sampling rate f S,2 , which in turn is higher than it is after the Sampling theorem for the bandwidth of 40 kHz occurring in the bandpass signal is required.

Realisiert wird dies vorzugsweise mit einem Analog/Digital-Wandler, der nach dem bekannten Verfahren der sukzessiven Approximation (SAR - successive approximation register) arbeitet.This is preferably implemented with an analog/digital converter which works according to the known method of successive approximation (SAR—successive approximation register).

Solche SAR-Analog/Digital-Wandler haben den Vorteil, dass sie eine relativ geringe Wandler-Latenz aufweisen, die für die nachfolgend noch anhand der Figur 5 beschriebene Verstärkungsregelung von besonderer Bedeutung ist. Außerdem kann durch den Einsatz solcher Hochleistungs-Analog/Digital-Wandler ein sehr hohes Signal/Rauschverhältnis erzielt werden.Such SAR analog / digital converters have the advantage that they have a relatively low converter latency, which is based on the below for the figure 5 described gain control is of particular importance. In addition, a very high signal-to-noise ratio can be achieved by using such high-performance analog/digital converters.

Bekannt und geeignet sind Ausführungen solcher Wandler mit zum Beispiel einer Auflösung von bis zu 18 Bit bei einer Abtastrate von bis zu 15 MHz oder mehr.Designs of such converters with, for example, a resolution of up to 18 bits at a sampling rate of up to 15 MHz or more are known and suitable.

Das am Ausgang des Analog/Digital-Wandlers A/D anliegende digitale DSB-AM Audiosignal xDSB(m) mit der genannten hohen (dritten) Auflösung mit der dritten Wortbreite von w2 Bit und der genannten zweiten Abtastrate fS,2 wird dann dem Demodulator Dm zugeführt, mit dem es synchron demoduliert wird, sodass sich ein digitales Basisbandsignal xdem(m) mit der zweiten Abtastrate fS,2 und der vierten Auflösung bzw. vierten Wortbreite von w3 Bit ergibt.The digital DSB-AM audio signal x DSB (m) present at the output of the analog/digital converter A/D with the mentioned high (third) resolution with the third word width of w 2 bits and the mentioned second sampling rate f S,2 is then supplied to the demodulator Dm, with which it is synchronously demodulated, so that a digital baseband signal x dem (m) with the second sampling rate f S,2 and the fourth resolution or fourth word length of w 3 bits results.

Der Demodulator Dm ist vorzugsweise ein kohärenter Demodulator, dessen HF-Demodulatorsignal xd,h(m) mit der zweiten Abtastrate fS,2 und einer gewünschten sechsten Auflösung bzw. sechsten Wortbreite von w5 Bit digital erzeugt und zur Demodulation verwendet wird. Da die Frequenz des HF-Trägersignals xc,h(k) bekannt ist, kann das HF-Demodulatorsignal entsprechend frequenzsynchron generiert werden.The demodulator Dm is preferably a coherent demodulator whose HF demodulator signal x d,h (m) is generated digitally with the second sampling rate f S,2 and a desired sixth resolution or sixth word width of w 5 bits and is used for demodulation. Since the frequency of the HF carrier signal x c,h (k) is known, the HF demodulator signal can be generated in a correspondingly frequency-synchronous manner.

Figur 4 zeigt beispielhaft ein Blockschaltbild eines solchen Demodulators, der eine Demodulator/Generator-Einheit DmG und vorzugsweise eine Einheit PhE zur Phasenabschätzung aufweist, an deren Eingang das demodulierte digitale Basisbandsignal xdem(m) anliegt und deren Ausgang mit der Demodulator/Generator-Einheit DmG verbunden ist, um die Phase des von der Demodulator/Generatoreinheit DmG erzeugten HF-Demodulatorsignals xd,h(m), mit dem das DSB-AM Audiosignal xDSB(m) demoduliert wird, zur Erzielung einer kohärenten Demodulation entsprechend zu verschieben bzw. anzupassen. Dies ist erforderlich, weil die Phase des von der Mikrofoneinheit Cm erzeugten, mit dem Audiosignal modulierten HF-Trägersignals xDSB(t) üblicherweise nicht bekannt ist. figure 4 shows an example of a block diagram of such a demodulator, which has a demodulator/generator unit DmG and preferably a unit PhE for phase estimation, at the input of which the demodulated digital baseband signal x dem (m) is present and whose output is connected to the demodulator/generator unit DmG is to shift or adapt the phase of the HF demodulator signal x d,h (m) generated by the demodulator/generator unit DmG, with which the DSB-AM audio signal x DSB (m) is demodulated, to achieve coherent demodulation . This is necessary because the phase of the HF carrier signal x DSB (t) generated by the microphone unit Cm and modulated with the audio signal is usually not known.

Sofern jedoch die exakte Phase des HF-Trägersignals xDSB(m) am Eingang der Demodulator/Generator-Einheit DmG zum Beispiel durch vorherige Phasenmessung des Gesamtsystems ermittelt wird, kann alternativ auf die Einheit PhE zur Phasenabschätzung verzichtet werden.However, if the exact phase of the HF carrier signal x DSB (m) at the input of the demodulator/generator unit DmG is determined, for example by prior phase measurement of the overall system, the unit PhE for phase estimation can alternatively be dispensed with.

Die Frequenz des von der Demodulator/Generator-Einheit DmG erzeugten HF-Demodulatorsignals xd,h(m) ist dabei gleich der Frequenz des von dem HF-Generator CG erzeugten HF-Trägersignals xc,h(k), während die Abtastrate fS,2 derjenigen des Analog/DigitalWandlers A/D entspricht.The frequency of the HF demodulator signal x d,h (m) generated by the demodulator/generator unit DmG is equal to the frequency of that generated by the HF generator CG HF carrier signal x c,h (k), while the sampling rate f S,2 corresponds to that of the analog/digital converter A/D.

Durch diese digitale Demodulation des HF-Trägersignals (Bandpass-Signal) xDSB(m) kann eine nahezu ideale Linearität erzielt werden. Die Genauigkeit dieser Linearität ist nur abhängig von der internen Bit-Auflösung, wobei alle nachteiligen analogen Einflüsse vermieden werden.This digital demodulation of the HF carrier signal (bandpass signal) x DSB (m) makes it possible to achieve almost ideal linearity. The accuracy of this linearity depends only on the internal bit resolution, avoiding all adverse analog influences.

Da das demodulierte digitale Basisbandsignal xdem(m) nun eine Bandbreite von etwa 20 kHz aufweist und in hohem Maße mit der (zweiten) Abtastrate fS,2 über-abgetastet ist, wird es vorzugsweise mit dem nachfolgenden Dezimator Dc gefiltert und heruntergetaktet, um ein digitales NF-Audiosignal xAF(n) mit einer üblichen bzw. Standard-Abtastfrequenz von zum Beispiel 48 kHz (dritte Abtastrate fs) zu erzeugen.Since the demodulated digital baseband signal x dem (m) now has a bandwidth of about 20 kHz and is highly oversampled with the (second) sampling rate f S,2 , it is preferably filtered with the following decimator Dc and downsampled to to generate a digital AF audio signal x AF (n) with a customary or standard sampling frequency of, for example, 48 kHz (third sampling rate fs).

Dabei wird gleichzeitig ein zusätzlicher Rauschabstand und Dynamikbereich erzielt. Wenn man zum Beispiel von einer Überabtastung mit einem Faktor von 100 ausgeht (der einer Abtastfrequenz des Analog/Digitalwandlers A/D von 4,8 MHz entspricht) und weißes Quantisierungsrauschen vorliegt, so ergibt sich ein zusätzlicher Signal/Rauschabstand und Dynamikbereich von 20 dB, was einer zusätzlichen Auflösung von etwa 3 Bit entspricht.At the same time, an additional signal-to-noise ratio and dynamic range are achieved. For example, assuming oversampling by a factor of 100 (corresponding to an analog-to-digital converter A/D sampling frequency of 4.8 MHz) and white quantization noise is present, this results in an additional signal-to-noise ratio and dynamic range of 20 dB, which corresponds to an additional resolution of about 3 bits.

Der Dezimationsvorgang kann in mehreren Stufen durchgeführt werden. Der Dezimator Dc kann zum Beispiel durch Einsatz von FIR (finite impulse response) Filter-Topologien wie zum Beispiel CIC (cascaded integrator comb) Filtern oder Polyphasen-Filtern realisiert werden. Zur Minimierung der Dezimations-Latenz können insbesondere IIR (infinite impulse response) Filter eingesetzt werden. In diesem Fall besteht ein Hauptkriterium bei dem Entwurf dieser Filter darin, eine lineare Phase in dem Audiofrequenzband zu bewahren.The decimation process can be performed in several stages. The decimator Dc can be implemented, for example, by using FIR (finite impulse response) filter topologies such as CIC (cascaded integrator comb) filters or polyphase filters. In particular, IIR (infinite impulse response) filters can be used to minimize the decimation latency. In this case, a main criterion in the design of these filters is to preserve a linear phase in the audio frequency band.

Figur 5 zeigt ein schematisches Blockschaltbild einer beispielhaften zweiten Ausführungsform einer erfindungsgemäßen Schaltungsanordnung, die sich ebenfalls aus den drei oben genannten Komponenten, nämlich einem digitalen Schaltungsteil D-C, einem analogen Schaltungsteil A-C und einem gemischt analog/digitalen Schaltungsteil M-C zusammensetzt. figure 5 shows a schematic block diagram of an exemplary second embodiment of a circuit arrangement according to the invention, which can also be derived from the three above said components, namely a digital circuit part DC, an analog circuit part AC and a mixed analog/digital circuit part MC.

Gleiche bzw. einander entsprechende Komponenten und Signale sind in diesem Blockschaltbild mit gleichen Bezeichnungen versehen wie in dem Blockschaltbild der Figur 2.Identical or mutually corresponding components and signals are given the same designations in this block diagram as in the block diagram of FIG figure 2 .

Im Vergleich zu dem digitalen Schaltungsteil D-C der in Figur 2 gezeigten Schaltungsanordnung umfasst diese Ausführungsform zusätzlich eine optionale, zwischen den Ausgang des Analog/Digital-Wandlers A/D und den Eingang des Demodulators Dm geschaltete erste Korrektureinheit Corr1, einen Schalldruckpegel-Detektor/Prädiktor SPL-DP, der ebenfalls mit dem Ausgang des Analog/Digital-Wandlers A/D und ferner mit der ersten Korrektureinheit Corr1 verbunden ist, sowie eine Einheit GC zur Verstärkungsregelung, die durch den Schalldruckpegel-Detektor/Prädiktor SPL-DP angesteuert wird und mittels eines ersten Multiplizierers M1 das von dem HF-Generator CG erzeugte HF-Trägersignal xc,h(k) mit einem ersten Verstärkungsfaktor g1(k) und mittels eines zweiten Multiplizierers M2 das Ausgangssignal der ersten Korrektureinheit Corr1 mit einem zweiten Verstärkungsfaktor g2(m) beaufschlagt.Compared to the digital circuit part DC of the in figure 2 circuit arrangement shown, this embodiment additionally includes an optional first correction unit Corr1 connected between the output of the analog/digital converter A/D and the input of the demodulator Dm, a sound pressure level detector/predictor SPL-DP, which is also connected to the output of the analog/ Digital converter A / D and further connected to the first correction unit Corr1, and a unit GC for gain control, which is controlled by the sound pressure level detector / predictor SPL-DP and by means of a first multiplier M1 generated by the HF generator CG HF carrier signal x c,h (k) with a first amplification factor g 1 (k) and by means of a second multiplier M2 the output signal of the first correction unit Corr1 with a second amplification factor g 2 (m).

Weiterhin ist eine optionale zweite Korrektureinheit Corr2 vorgesehen, die mit dem Ausgang des optionalen Dezimators Dc verbunden ist. Die Einheit NS zu Rauschformung ist auch hier wiederum optional.Furthermore, an optional second correction unit Corr2 is provided, which is connected to the output of the optional decimator Dc. The unit NS for noise shaping is again optional here.

Mit dieser Ausführungsform kann insbesondere der Tatsache Rechnung getragen werden, dass hohe Schalldruckpegel zu einer Übersteuerung des analogen Schaltungsteils A-C sowie des Analog/Digital-Wandlers A/D führen können. Deshalb ist üblicherweise der analoge Ausgangspegel von Mikrofonen einstellbar, indem z.B. das aufgenommene NF-Signal gedämpft wird.With this embodiment, account can be taken in particular of the fact that high sound pressure levels can lead to overdriving of the analog circuit part AC and of the analog/digital converter A/D. Therefore, the analog output level of microphones can usually be adjusted, for example by attenuating the recorded AF signal.

Ein besonderer Vorteil der erfindungsgemäßen Ausführungsform besteht darin, dass eine solche Einstellbarkeit bzw. Dämpfung des NF-Signals nicht erforderlich ist.A particular advantage of the embodiment according to the invention is that such adjustability or damping of the LF signal is not required.

Da nämlich die Empfindlichkeit des Mikrofons direkt proportional zu der Amplitude des analogen HF-Trägersignals xc(t) ist und erfindungsgemäß das HF-Trägersignal in der digitalen Ebene erzeugt wird, kann die Empfindlichkeit ohne zusätzliche analoge Schaltungen digital erhöht und vermindert und somit der Dynamikbereich des Mikrofons entsprechend erweitert bzw. optimiert werden.Since the sensitivity of the microphone is directly proportional to the amplitude of the analog HF carrier signal x c (t) and the HF carrier signal is generated in the digital domain according to the invention, the sensitivity can be digitally increased and decreased without additional analog circuits and thus the dynamic range of the microphone can be expanded or optimized accordingly.

Die Möglichkeit, das HF-Trägersignal auf digitale Weise zu verändern, ist einer der wesentlichen Vorteile der Erzeugung des HF-Trägersignals in der digitalen Ebene. Da das HF-Trägersignal in dem interessierenden Frequenzbereich mit einem wie oben erläutert hohen Signal/Rauschverhältnis erzeugt wird, kann seine Intensität vermindert werden, bevor es danach der Rauschformung in der Einheit NS unterworfen wird, um dadurch ein optimales Signal/Rauschverhältnis beizubehalten.The ability to digitally manipulate the RF carrier signal is one of the key benefits of generating the RF carrier signal in the digital domain. Since the RF carrier signal is generated in the frequency range of interest with a high signal-to-noise ratio as explained above, its intensity can be reduced before it is thereafter subjected to noise-shaping in the unit NS, thereby maintaining an optimum signal-to-noise ratio.

Somit erfolgt also, wie es auch in den Figuren 3 und 5 dargestellt ist, die Einstellung der Empfindlichkeit vorzugsweise vor der Rauschformung, und zwar durch die genannte Beaufschlagung des digitalen HF-Trägersignals xc,h(k) mit dem ersten Verstärkungsfaktor g1(k) ≤ 1 (wobei natürlich auch der HF-Generators CG direkt mit dem ersten Verstärkungsfaktor g1(k) entsprechend angesteuert werden kann).Thus, as is also the case in the Figures 3 and 5 is shown, the adjustment of the sensitivity preferably before the noise shaping, namely by the mentioned application of the digital HF carrier signal x c,h (k) with the first amplification factor g 1 (k) ≤ 1 (whereby the HF generator CG can be driven directly with the first amplification factor g 1 (k) accordingly).

Unter der Annahme eines Spannungspegels von -20 dBFS bei einem 94 dBSPL Eingangspegel würde der maximal verarbeitbare Schalldruckpegel 114 dBSPL betragen. Um diesen Bereich um zum Beispiel 30 dB auf einen Pegel von 144 dBSPL zu erweitern (sofern das betreffende Mikrofon einen solchen Schalldruckpegel erfassen kann), kann die Amplitude des HF-Trägersignals in der Weise eingestellt werden, dass sich im Ruhezustand ein Verstärkungsfaktor g1(k) von etwa 1/32 ergibt.Assuming a voltage level of -20 dBFS at a 94 dBSPL input level, the maximum sound pressure level that can be handled would be 114 dBSPL. To extend this range by, for example, 30 dB to a level of 144 dBSPL (provided the microphone in question can detect such a sound pressure level), the amplitude of the RF carrier signal can be adjusted in such a way that a gain factor g 1 ( k) of about 1/32.

Durch Anwendung eines solchen Faktors g1(k) vermindert sich zwar die Ausgangsauflösung um etwa 5 Bit. Dieser Verlust kann jedoch durch die oben erläuterte Rauschformung wieder weitgehend kompensiert werden, sodass das HF-Trägersignal weiterhin ein hinreichend hohes Signal/Rauschverhältnis aufweist. Da g k dB = 20 log 10 32 30 dB

Figure imgb0002
ist, wird die Empfindlichkeit des Mikrofons vermindert, was zu dem erforderlichen zusätzlichen Dynamikbereich für den Analog/Digital-Wandler A/D führt.The application of such a factor g 1 (k) reduces the output resolution by about 5 bits. However, this loss can again be largely compensated for by the noise shaping explained above, so that the HF carrier signal continues to have a sufficiently high signal-to-noise ratio. There G k dB = 20 log 10 32 30 dB
Figure imgb0002
the sensitivity of the microphone is reduced, resulting in the required additional dynamic range for the analog-to-digital converter A/D.

In diesem Fall kann die oben erwähnte, erste hohe Auflösung bzw. die erste große Wortbreite von w0 Bit des HF-Trägersignals xc,h(k) beispielsweise so gewählt werden, dass diese die verlorenen 5 Bit bereits vor der Rauschformung bereitstellt, d.h. das HF-Trägersignal xc,h(k) vor der Rauschformung ist um 5 Bit höher aufgelöst, als das HF-Trägersignal xc(k) nach der Rauschformung.In this case, the above-mentioned, first high resolution or the first large word length of w 0 bits of the HF carrier signal x c,h (k) can be selected, for example, so that it provides the lost 5 bits before the noise shaping, ie the RF carrier signal x c,h (k) before noise shaping is resolved 5 bits higher than the RF carrier signal x c (k) after noise shaping.

Darüber hinaus werden bevorzugt jedoch weitere Maßnahmen getroffen, um den im Vergleich zu dem Analog/Digital-Wandler A/D sehr hohen Dynamikbereich von modernen bzw. zukünftigen Mikrofonen von zwischen 140 und 150 dB oder mehr weitgehend verlustfrei verarbeiten zu können.In addition, however, further measures are preferably taken in order to be able to process the very high dynamic range of modern or future microphones of between 140 and 150 dB or more in comparison to the analog/digital converter A/D largely without losses.

Dabei wird erfindungsgemäß von dem so genannten "Gain ranging" Gebrauch gemacht, bei dem der Pegelbereich des Mikrofons automatisch an einen aufgenommenen Schalldruckpegel angepasst und damit der Dynamikbereich des gesamten Systems automatisch verändert bzw. entsprechend erhöht und vermindert wird.According to the invention, so-called "gain ranging" is used, in which the level range of the microphone is automatically adapted to a recorded sound pressure level and the dynamic range of the entire system is thus automatically changed or correspondingly increased and decreased.

Bekannt ist es, zu diesem Zweck Analog/Digital-Wandler mit mehreren Kanälen einzusetzen, die jeweils unterschiedliche Verstärkungsfaktoren aufweisen, sodass mehrere Analog/Digital-gewandelte Signale entstehen, die dann auf der digitalen Ebene entsprechend kombiniert und nachverarbeitet werden müssen, um ein Audiosignal mit einem erhöhten Dynamikbereich, d.h. einer höheren Bitauflösung, zu erzeugen. Dabei wird jedoch das Signal/Rauschverhältnis nicht verbessert, was als nachteilig angesehen wird.It is known to use analog/digital converters with multiple channels for this purpose, each of which has different amplification factors, resulting in multiple analog/digital converted signals, which then have to be combined and post-processed accordingly on the digital level in order to create an audio signal with to generate an increased dynamic range, ie a higher bit resolution. However, this does not improve the signal-to-noise ratio, which is viewed as a disadvantage.

Erfindungsgemäß wird dieses Problem bei der Schaltungsanordnung gemäß Figur 5 durch eine automatische Empfindlichkeitseinstellung bzw. -anpassung des Mikrofons beim Auftreten oder der Vorhersage von hohen Schalldruckpegeln gelöst.According to the invention, this problem is solved in the circuit arrangement according to FIG figure 5 This is solved by automatically setting or adjusting the sensitivity of the microphone when high sound pressure levels occur or are predicted.

Zu diesem Zweck ist der Schalldruckpegel-Detektor/Prädiktor SPL-DP vorgesehen, der die Einheit GC zur Verstärkungsregelung, die den genannten ersten Verstärkungsfaktor g1(k) zur Beaufschlagung des HF-Generators CG bzw. des HF-Trägersignals xc,h(k) erzeugt, ansteuert.The sound pressure level detector/predictor SPL-DP is provided for this purpose, which has the unit GC for amplification control, which uses the first amplification factor g 1 (k) mentioned for applying the HF generator CG or the HF carrier signal x c,h ( k) generates, controls.

Der Schalldruckpegel-Detektor/Prädiktor SPL-DP dient zur Erfassung oder Vorhersage von insbesondere hohen Schalldruckpegeln. Da, wie oben bereits erwähnt wurde, vorzugsweise ein SAR-Analog/Digital-Wandler A/D eingesetzt wird, und das digitale DSB-AM Bandpass-Signal xDSB(m) stark über-abgetastet ist, kann das System weitgehend verzögerungsfrei auf hohe Schalldruckpegel reagieren.The sound pressure level detector/predictor SPL-DP is used to record or predict particularly high sound pressure levels. Since, as already mentioned above, a SAR analog/digital converter A/D is preferably used and the digital DSB-AM bandpass signal x DSB (m) is heavily oversampled, the system can be switched to high respond to sound pressure levels.

Die Einheit GC erzeugt dabei nicht nur den ersten Verstärkungsfaktor g1(k) für die Beaufschlagung des HF-Trägersignals xc,h(k), sondern auch den zweiten Verstärkungsfaktor gz(m), mit dem mittels des zweiten Multiplizierer M2 das digitale DSB-AM Audiosignal xDSB(m) am Eingang des Demodulators Dm verstärkt bzw. multipliziert wird. Die Einheit GC erzeugt den zweiten Verstärkungsfaktor g2(m) in der Weise, dass damit die Amplitudenanpassung bzw. Multiplikation des HF-Trägersignals xc,h(k) mit dem ersten Verstärkungsfaktor g1(k) wieder kompensiert und somit ein korrigiertes digitales DSB-AM Audiosignal x~ DSB(m) erzeugt wird, das dem Demodulator Dm zugeführt wird.The unit GC generates not only the first amplification factor g 1 (k) for applying the HF carrier signal x c,h (k), but also the second amplification factor gz(m), with which the digital DSB -AM audio signal x DSB (m) is amplified or multiplied at the input of the demodulator Dm. The unit GC generates the second amplification factor g 2 (m) in such a way that the amplitude adjustment or multiplication of the HF carrier signal x c,h (k) with the first amplification factor g 1 (k) is compensated again and thus a corrected digital DSB-AM audio signal x ~ DSB (m) is generated, which is fed to the demodulator Dm.

Um eine möglichst exakte Kompensation zu erzielen, wird der zweite Verstärkungsfaktor g2(m) vorzugsweise nicht nur durch Invertierung des ersten Verstärkungsfaktors g1(k) ermittelt, sondern es werden zusätzlich auch die internen Verzögerungen und Filtereigenschaften der Schaltung sowie die Änderung der Abtastrate in dem Signalpfad zwischen dem ersten Multiplizierer M1 oder der Einheit NS zur Rauschformung und dem zweiten Multiplizierer M2 mit eingerechnet.In order to achieve compensation that is as exact as possible, the second amplification factor g 2 (m) is preferably not only determined by inverting the first amplification factor g 1 (k), but the internal delays and filter properties of the circuit and the change in the sampling rate in included in the signal path between the first multiplier M1 or the unit NS for noise shaping and the second multiplier M2.

Durch diese automatische Einstellung der Empfindlichkeit in beiden Zweigen werden im Ergebnis die in dem Digital/Analog-Wandler D/A verlorenen Bits zu dem Analog/DigitalWandler A/D überführt, was zu einem erhöhten Dynamikbereich führt. Das Signal/Rauschverhältnis nach der Analog/Digital-Wandlung bleibt, ebenso wie bei dem oben beschriebenen bekannten Ansatz, weitgehend unverändert. Der Analog/Digital-Wandler A/D kann somit auch als Gleitkomma- (floating-point) A/D-Wandler bezeichnet werden.As a result of this automatic setting of the sensitivity in both branches, the bits lost in the digital/analog converter D/A are transferred to the analog/digital converter A/D, which leads to an increased dynamic range. The signal/noise ratio after the analog/digital conversion remains largely unchanged, just as in the known approach described above. The analog/digital converter A/D can thus also be referred to as a floating-point A/D converter.

Die optionale erste Korrektureinheit Corr1 dient dabei vorzugsweise dazu, vor der Beaufschlagung des digitalen DSB-AM Audiosignals xDSB(m) mit dem genannten zweiten Verstärkungsfaktor gz(m) eventuell verbliebene fehlerhafte Abtastwerte auf Grund unterschiedlicher Filtercharakteristiken zwischen der Digital/Analog-Wandlung und der Analog/DigitalWandlung zu kompensieren, und zwar insbesondere dann, wenn der Faktor gz(m) nur durch Invertierung des ersten Verstärkungsfaktors g1(k) ermittelt wird oder nicht genau genug dem verzögerten, gefilterten und invertierten Wert von g1(k) entspricht. Dafür kann der Schaltkreis beispielsweise exakt ausgemessen werden, sodass die benötigte Korrektur errechnet und angewendet werden kann. Eine weitere Möglichkeit besteht darin, an den insoweit kritischen Stellen mit prädizierten Abtastwerten zu arbeiten. The optional first correction unit Corr1 is preferably used to correct any remaining incorrect sample values due to different filter characteristics between the digital/analog conversion and the Compensate for analog/digital conversion, in particular when the factor gz(m) is determined only by inverting the first amplification factor g 1 (k) or does not correspond precisely enough to the delayed, filtered and inverted value of g 1 (k). For example, the circuit can be precisely measured so that the required correction can be calculated and applied. Another possibility is to work with predicted samples at the points that are critical in this regard.

Der erste Verstärkungsfaktor g1(k), mit dem das digitale HF-Trägersignal xc,h(k) beaufschlagt wird, kann dabei sowohl ein konstanter Faktor, als auch ein zeitlich monoton abfallender Faktor sein. Es muss dabei nur sichergestellt werden, dass in dem analogen Schaltungsteil A-C durch die hohen Schalldruckpegel keine Übersteuerungen (clipping) auftreten.The first amplification factor g 1 (k), which is applied to the digital HF carrier signal x c,h (k), can be either a constant factor or a factor that falls monotonously over time. In this case, it must only be ensured that no overmodulation (clipping) occurs in the analog circuit part AC due to the high sound pressure level.

Sofern erforderlich, kann schließlich noch die optionale zweite Korrektureinheit Corr2 vorgesehen sein, mit der das von dem Dezimator Dc ausgegebene temporäre NF-Audiosignal x~ AF(m) weiterhin korrigiert und das digitale NF-Audiosignal xAF(n) erzeugt wird.Finally, if necessary, the optional second correction unit Corr2 can also be provided, with which the temporary AF audio signal x˜AF (m) output by the decimator Dc is further corrected and the digital AF audio signal xAF (n) is generated.

Zur Erläuterung des analogen Schaltungsteils A-C wird schließlich noch auf die Figuren 6 bis 8 Bezug genommen. Diese Figuren zeigen jeweils den bevorzugten Einsatz eines Kondensatormikrofons mit Gegentaktwandler. Als Alternative zu dem Einsatz eines solchen Gegentakt-Mikrofons zeigt Figur 9 ein Kondensatormikrofon mit Standardwandler C1(t) und Referenzkondensator C2. Dieses Standard-Mikrofon kann anstelle des Gegentakt-Mikrofons in den Figuren 6 bis 8 durch Verbindung mit den jeweils gleich lautenden Kontaktpunkten K1, K2, K3 verwendet werden. Die nachfolgenden Erläuterungen gelten somit für beide Arten von Mikrofonen.Finally, to explain the analog circuit part AC, reference is made to Figures 6 to 8 referenced. These figures each show the preferred use of a condenser microphone with a push-pull converter. As an alternative to using such a push-pull microphone shows figure 9 a condenser microphone with standard transducer C 1 (t) and reference condenser C 2 . This standard microphone can be used instead of the push-pull microphone in the Figures 6 to 8 be used by connecting them to the contact points K1, K2, K3, each of which has the same name. The following explanations therefore apply to both types of microphones.

Wie bereits erwähnt wurde, wird das Kondensatormikrofon bevorzugt mit einem Hochfrequenzverfahren mit Amplitudenmodulation, insbesondere der Doppelseitenband-Amplitudenmodulation (DSB-AM), betrieben.As already mentioned, the condenser microphone is preferably operated using a high-frequency method with amplitude modulation, in particular double sideband amplitude modulation (DSB-AM).

Die Auswahl der konkreten Schaltung, mit der das Mikrofon betrieben wird, erfolgt in Abhängigkeit von den Eigenschaften des Mikrofons, dem vorgesehenen Anwendungsfall, der geforderten Aufnahmequalität und anderen Kriterien.The selection of the specific circuit with which the microphone is operated depends on the properties of the microphone, the intended application, the required recording quality and other criteria.

Mit dem analogen Schaltungsteil A-C wird also bevorzugt ein (analoges) DSB-AM Audiosignal xDSB(t) erzeugt. Hierzu umfasst der analoge Schaltungsteil A-C einen Übertrager Tr mit einer Primärseite Lp und einer Sekundärseite Ls, die eine Mittenanzapfung aufweist bzw. durch zwei gleiche, in Reihe geschaltete Induktivitäten Ls1, Ls2, gebildet ist.An (analog) DSB-AM audio signal x DSB (t) is thus preferably generated with the analog circuit part AC. For this purpose, the analog circuit part AC includes a transformer Tr with a primary side L p and a secondary side L s , which has a center tap or is formed by two identical inductances L s1 , L s2 connected in series.

Parallel zu der Sekundärseite Ls ist ein kapazitiver Spannungsteiler, nämlich die Reihenschaltung aus den beiden Kondensatoren C1(t) und C2(t) des Gegentakt-Mikrofons bzw. die Reihenschaltung aus dem Kondensator C1(t) des Standard-Mikrofons und dem Referenzkondensator C2 geschaltet, sodass auf diese Weise eine HF-Brückenschaltung entsteht.Parallel to the secondary side L s is a capacitive voltage divider, namely the series connection of the two capacitors C 1 (t) and C 2 (t) of the push-pull microphone or the series connection of the capacitor C 1 (t) of the standard microphone and connected to the reference capacitor C 2 , creating an RF bridge circuit in this way.

Das wie oben erläutert erzeugte HF-Trägersignal xc(t) wird an die Primärseite Lp des Übertragers Tr angelegt.The RF carrier signal x c (t) generated as explained above is applied to the primary side L p of the transformer Tr.

Das am Ausgangsanschluss der Brücke anliegende DSB-AM Audiosignal xDSB(t) (Mikrofonsignal mit unterdrücktem HF-Trägersignal xc(t)) wird gemäß den Figuren 6 bis 8 an der Membran zwischen den beiden Kondensatoren C1(t), C2(t) des (Gegentakt-) Mikrofons bzw. an dem Kontaktpunkt K3 ausgekoppelt. Es ist jedoch ebenso möglich, die Auskopplung an der Mittenanzapfung der Sekundärseite Ls des Übertragers Tr vorzunehmen, während die Membran bzw. der Kontaktpunkt K3 selbst an eine Referenzspannung gekoppelt ist.The DSB-AM audio signal x DSB (t) (microphone signal with suppressed RF carrier signal x c (t)) present at the output connection of the bridge is processed according to the Figures 6 to 8 on the membrane between the two capacitors C 1 (t), C 2 (t) of the (push-pull) microphone or on the Contact point K3 decoupled. However, it is also possible to perform the decoupling at the center tap of the secondary side L s of the transformer Tr, while the membrane or the contact point K3 itself is coupled to a reference voltage.

Bei der einfachsten Ausführung gemäß Figur 6 wird das ausgekoppelte DSB-AM Audiosignal xDSB(t) nach Impedanzwandlung und Verstärkung direkt dem Analog/Digitalwandler A/D des gemischt analog/digitalen Schaltungsteils M-C zugeführt.In the simplest version according to figure 6 the decoupled DSB-AM audio signal x DSB (t) is fed directly to the analog/digital converter A/D of the mixed analog/digital circuit part MC after impedance conversion and amplification.

Vorzugsweise wird das DSB-AM Audiosignal xDSB(t) jedoch über einen auf den Ausgangsanschluss der Brücke abgestimmten Resonanzkreis ausgekoppelt.However, the DSB-AM audio signal x DSB (t) is preferably coupled out via a resonant circuit tuned to the output connection of the bridge.

Bei der Ausführung gemäß Figur 7 wird das DSB-AM Audiosignal xDSB(t) über einen Resonanzkreis mit einer in Reihe zu dem Ausgangsanschluss der Brücke geschalteten Reihenschaltung aus einer Induktivität L und einem Widerstand R als Stromsignal ix(t) ausgekoppelt.When executing according to figure 7 the DSB-AM audio signal x DSB (t) is coupled out as a current signal i x (t) via a resonant circuit with a series connection of an inductor L and a resistor R connected in series to the output terminal of the bridge.

Der erzeugte Strom ix(t) wird gemäß der in Figur 7 gezeigten Ausführung vorzugsweise mittels eines Transimpedanz-Verstärkers erfasst bzw. gemessen, der durch einen invertierend beschalteten und durch die Parallelschaltung einer Kapazität Cf mit einem Widerstand Rf rückgekoppelten Verstärker V (Operationsverstärker) realisiert ist, und an dessen Ausgang das DSB-AM Audiosignal xDSB(t) (Spannungssignal) anliegt.The generated current i x (t) is according to in figure 7 shown embodiment preferably recorded or measured by means of a transimpedance amplifier, which is realized by an inverting circuit and feedback amplifier V (operational amplifier) by connecting a capacitor C f in parallel with a resistor R f , and at the output of which the DSB-AM audio signal x DSB (t) (voltage signal) is present.

Dabei bestimmt die Induktivität L die Resonanzfrequenz fr des Kreises (die mit der niedrigsten Quellenimpedanz korrespondiert), während die Frequenzbandbreite (die bei dem DSB-AM Audiosignal bei 40 kHz liegt) und damit der Q-Faktor des Kreises nur durch der Widerstand R bestimmt wird.The inductance L determines the resonant frequency f r of the circuit (which corresponds to the lowest source impedance), while the frequency bandwidth (which is 40 kHz for the DSB-AM audio signal) and thus the Q factor of the circuit is only determined by the resistance R becomes.

Ein Vorteil dieser Ausführung besteht darin, dass auf Grund der niedrigen Quellenimpedanz ein DSB-AM Audiosignal xDSB(t) mit verbesserten Rauscheigenschaften erzielt werden kann.An advantage of this design is that due to the low source impedance, a DSB-AM audio signal x DSB (t) with improved noise characteristics can be achieved.

Bei der Ausführung gemäß Figur 8 wird das DSB-AM Audiosignal xDSB(t) über einen Resonanzkreis mit einer parallel zu dem Ausgangsanschluss der Brücke geschalteten Reihenschaltung aus einer Induktivität L und einem Widerstand R als Spannungssignal ux(t) ausgekoppelt.When executing according to figure 8 the DSB-AM audio signal x DSB (t) is coupled out as a voltage signal u x (t) via a resonant circuit with a series circuit made up of an inductor L and a resistor R connected in parallel to the output connection of the bridge.

Die erzeugte Spannung ux(t) wird gemäß der in Figur 8 gezeigten Ausführung vorzugsweise mittels eines Verstärkers erfasst bzw. gemessen, der durch einen nicht-invertierend beschalteten und durch einen Spannungsteiler R1/R2 rückgekoppelten Verstärker V (Operationsverstärker) realisiert ist, und an dessen Ausgang wiederum das DSB-AM Audiosignal xDSB(t) anliegt.The generated voltage u x (t) is calculated according to in figure 8 The embodiment shown is preferably recorded or measured by means of an amplifier, which is realized by an amplifier V (operational amplifier) with non-inverting wiring and feedback by a voltage divider R 1/ R 2 , and at the output of which the DSB-AM audio signal x DSB (t ) is present.

Ein Vorteil dieser Ausführung besteht darin, dass das Spannungssignal ux(t) eine (abhängig vom Q-Faktor) höhere Empfindlichkeit aufweist, dafür jedoch auch die Quellenimpedanz höher ist.An advantage of this design is that the voltage signal u x (t) has a higher sensitivity (depending on the Q factor), but the source impedance is also higher.

Claims (12)

  1. Method for operating a condenser microphone according to a high-frequency method with amplitude modulation, characterized by the following steps:
    - generating a digital HF carrier signal (xc,h(k); xc(k)),
    - generating an analogue HF carrier signal (xc(t)) by digital-to-analogue conversion of the digital HF carrier signal (xc,h(k); xc(k)),
    - generating an analogue DSB-AM audio signal (xDSB(t)) by double-sideband amplitude modulation (DSB-AM) of the analogue HF carrier signal (xc(t)) by means of the microphone,
    - generating a digital DSB-AM audio signal (xDSB(m)) by analogue-to-digital conversion of the analogue DSB-AM audio signal (xDSB(t)), and
    - demodulating the DSB-AM digital audio signal (xDSB(m)) and generating a baseband digital audio signal (xdem(m)).
  2. The method according to claim 1,
    wherein the digital HF carrier signal (xc,h(k); xc(k)) is generated with an oversampling at a first sampling rate (fS,1).
  3. The method according to claim 1,
    wherein the digital HF carrier signal (xc,h(k)) is generated at a first high resolution that is greater than a maximum resolution of the digital-to-analogue conversion, and wherein, prior to the digital-to-analogue conversion, a digital HF carrier signal xc(k) is generated which has a reduced second resolution adapted to the maximum resolution of the digital-to-analogue conversion, and wherein a reduction of the signal/noise ratio in the DSB-AM frequency band of the audio signal caused by the reduction of the first resolution is at least largely compensated by noise shaping.
  4. The method according to claim 1,
    wherein the amplitude of the digital HF carrier signal (xc,h(k); xc(k)) is digitally adjustable in order to modify the sensitivity of the microphone.
  5. The method according to claim 1,
    wherein a sound pressure level in the digital DSB-AM audio signal (xDSB(m)) is captured or predicted and, as a function of this sound pressure level, a first gain factor (g1(k)) is generated which is applied to the amplitude of the digital HF carrier signal (xc,h(k); xc(k)) in order to adapt the sensitivity of the microphone to the captured or predicted sound pressure level.
  6. The method according to claim 5,
    wherein a second gain factor (g2(k)) is generated by inverting the first gain factor (g1(k)) in such a manner that, by applying it to the digital DSB-AM audio signal (xDSB(m)), the modification of the amplitude of the digital HF carrier signal (xc,h(k); xc(k)) by the first gain factor (g1(k)) is at least largely compensated.
  7. The method according to claim 1,
    wherein the digital DSB-AM audio signal (xDSB(m)) is generated by an analogue-to-digital conversion of the analogue DSB-AM audio signal (xDSB(t)) with an oversampling at a second sampling rate (fS,2).
  8. A circuit arrangement for operating a condenser microphone according to a method according to one of claims 1 to 7, comprising:
    - a digital HF generator (CG) for generating a digital HF carrier signal (xc,h(k); xc(k)),
    - a digital-to-analogue converter (D/A) for generating an analogue HF carrier signal (xc(t)) from the digital HF carrier signal (xc,h(k); xc(k)),
    - a microphone unit (Cm) with a condenser microphone for generating an analogue DSB-AM audio signal (xDSB(t)) by double-sideband amplitude modulation of the analogue HF carrier signal (xc(t)) by means of the microphone,
    - an analogue-to-digital converter (A/D) for generating a digital DSB-AM audio signal (xDSB(m)) from the analogue DSB-AM audio signal (xDSB(t)), and
    - a digital demodulator (Dm) for demodulating the digital DSB-AM audio signal (xDSB(m)) and for generating a digital baseband audio signal (xdem(m)).
  9. The circuit arrangement according to claim 8,
    comprising a noise shaping unit (NS) for generating a digital HF carrier signal (xc(k)) with a reduced resolution adapted to the digital-to-analogue converter (A/D), as well as for noise shaping the HF carrier signal in order to at least largely compensate a signal-to-noise ratio in the DSB-AM frequency band of the audio signal that has been adversely affected by the reduction of the resolution.
  10. The circuit arrangement according to claim 8, comprising:
    - a sound pressure level detector/predictor (SPL-DP) for capturing or predicting a sound pressure level in the digital DSB-AM audio signal (xDSB(m)), and
    - a gain control unit (GC) which is controlled by the sound pressure level detector/predictor (SPL-DP) and which generates a first gain factor (g1(k)) with which the amplitude of the digital HF carrier signal (xc,h(k); xc(k)) can be modified by means of a first multiplier (M1) in order to adapt the sensitivity of the microphone to the captured or predicted sound pressure level.
  11. The circuit arrangement according to claim 10,
    wherein the gain control unit (GC) generates a second gain factor (g2(m)) by inverting the first gain factor (g1(k)) in such a manner that, by applying it to the digital DSB-AM audio signal (xDSB(m)) by means of a second multiplier (M2), the modification of the amplitude of the digital HF carrier signal (xc,h(k); xc(k)) by the first gain factor (g1(k)) is at least largely compensated.
  12. The circuit arrangement according to claim 8,
    wherein the microphone unit (Cm) comprises a HF transformer bridge circuit (Ls1, Ls2) with a capacitive voltage divider formed by the microphone in order to generate the analogue DSB-AM audio signal (xDSB(t)), wherein the DSB-AM audio signal (xDSB(t)) is decoupled via a resonant circuit tuned to an output terminal of the bridge.
EP19189032.6A 2018-08-02 2019-07-30 Method and circuit arrangement for operating a condenser microphone Active EP3606098B1 (en)

Applications Claiming Priority (1)

Application Number Priority Date Filing Date Title
DE102018118795.5A DE102018118795B3 (en) 2018-08-02 2018-08-02 Method and circuit arrangement for operating a condenser microphone

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EP3606098A1 EP3606098A1 (en) 2020-02-05
EP3606098B1 true EP3606098B1 (en) 2023-03-29

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DE (1) DE102018118795B3 (en)

Family Cites Families (5)

* Cited by examiner, † Cited by third party
Publication number Priority date Publication date Assignee Title
DE4300379C2 (en) 1993-01-09 1996-02-08 Sennheiser Electronic Circuit for high-frequency condenser microphones
DE19612068A1 (en) * 1996-03-27 1997-10-02 Neumann Gmbh Georg Method and arrangement for converting an acoustic signal into an electrical signal
WO2005055406A1 (en) * 2003-12-01 2005-06-16 Audioasics A/S Microphine with voltage pump
JP4823134B2 (en) * 2007-04-25 2011-11-24 株式会社オーディオテクニカ Condenser microphone
DE102010000686B4 (en) 2010-01-05 2018-05-09 Sennheiser Electronic Gmbh & Co. Kg condenser microphone

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EP3606098A1 (en) 2020-02-05

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