Method, electrical audio signal circuit, and microphone arrangement for calibrating a microphone cutoff frequency

Abstract

Method in an electrical audio signaling circuit comprising a feedback loop (280) with a digital filter (285) coupled to a current-to-digital-to-analog converter (240, 320), the method comprising: providing (560, 625) an output signal from the current-to-digital-to-analog converter (240, 320) to analog elements of the electrical audio signaling circuit, wherein the output signal from the current-to-digital-to-analog converter (240, 320) is based on a reference signal input to the current-to-digital-to-analog converter (240, 320) while no output from the digital filter (285) is input to the current-to-digital-to-analog converter (240, 320); comparing (565, 630) an output signal of the electrical audio signaling circuit with a reference; and calibrating (570, 635) the electrical audio signal circuit, such that the output signal of the electrical audio signal circuit corresponds to the reference,wherein the calibration of the electrical audio signal circuit enables more precise control of a cutoff frequency of a microphone signal, while the output of the digital filter (285) is input into the current-to-digital-to-analog converter (240, 320).

Description

CROSS REFERENCE TO RELATED APPLICATIONSThe present application claims the benefit and priority of U.S. Provisional Patent Application No. 62 / 459,031, filed February 14, 2017.BACKGROUNDThe following description is intended to make the invention comprehensible to the reader. The information or references cited herein are not to be understood as prior art.Microphones are widely used in a wide variety of applications, such as smartphones, mobile telephones, tablets, headsets, hearing aids, sensors, automobiles, etc. Noise reduction in these microphones is an important function for obtaining excellent sound quality. The existing microphones have certain limitations due to their configuration and operation.DE 11 2017 003 403 T5 and DE 11 2017 003 709 T5 disclose a microphone arrangement with a transducer element and a processing circuit. US 2015 / 0 078 569 A1 relates to audio signal circuits for processing microphone signals. US 2012 / 0 038 500 A1 discloses a sigma-delta modulator having a quantizer followed by a digital integrator for generating an integrated digital signal from a quantized signal. The output of the digital integrator is coupled to a digital-to-analog converter in the sigma-delta modulator feedback loop.A disadvantage of this is that the sensitivity of the microphone and the cut-off frequency of the microphone cannot be controlled precisely. The invention is based on the object of at least partially eliminating this shortcoming.BRIEF DESCRIPTION OF THE DRAWINGSFIG. 1 is a schematic view of a microphone assembly. FIG. 2 is a simplified schematic view of an electrical circuit configuration of the microphone arrangement of FIG. 1. FIG. 3 is another simplified schematic view of the electrical circuit structure of the microphone array of FIG. 1. FIG. 4 is a flow chart showing steps for calibrating a cut-off frequency of the microphone assembly of FIG. 1. FIG. 5 is another flow chart showing steps for calibrating an amplifier of the electrical circuitry of FIGS. 2 and 3. FIG. 6 is another flow chart showing steps for calibrating a current-to-digital-to-analog converter of the electrical circuitry of FIGS. 2 and 3. FIG. 7A is a Cartesian graph showing the effect on beam formation using two microphone arrays with varying cutoff frequencies. FIG. 7B is a polar diagram showing a comparison of one composite beam formed using two microphone arrays with tuned cutoff frequencies with another composite beam formed using two microphone arrays with non-tuned cutoff frequencies.DETAILED DESCRIPTIONThe invention relates to a method in an electrical audio signal circuit according to claim 1, an electrical audio signal circuit according to claim 9 and a microphone arrangement according to claim 14, The invention relates generally to a system and a method for more precisely setting a cut-off frequency of a microphone signal. The cut-off frequency is generally controlled by a digital filter in a feedback loop, however precise control of the cut-off frequency requires knowledge of the loop gain. The invention therefore further relates to a system and method for setting the loop gain of a microphone signal processing or electrical circuitry for better control of the cut-off frequency of the microphone. The processing circuit includes an amplifier for amplifying a microphone signal, an analog-to-digital converter for converting the amplified microphone signal to a digital signal, and a decimator for down-sampling a frequency of the digital signal. The output of the decimator is then provided to a host processor of a device (e.g., a smartphone) for further processing and use. The processing circuit also includes a feedback loop and, in particular, an internal feedback loop. The feedback loop returns the output of the decimator to a combining block via a current to digital to analog converter (IDAW). The combining block receives an acoustic signal from a transducer of the microphone and combines the acoustic signal with the output of the feedback loop to obtain the microphone signal which is then input to the amplifier.The loop gain of the processing circuit may be adjusted by controlling the amplifier or the IDAW. The amplifier may be controlled by adjusting the gain of the amplifier. The IDAW may be controlled by analog adjustment of a source current of the current to digital to analog converter. The IDAW may also be controlled using digital adjustment of a gain before the IDAW (e.g., at an input of the IDAW). By adjusting the loop gain of the processing circuit using the amplifier or the IDAW, the sensitivity of the microphone and the cut-off frequency of the microphone can be precisely controlled.Precise control of the cut-off frequency of the microphone signal improves sound quality, enables filtering of low frequencies that can overload the amplifier, and enables the manufacture of microphones with closely tuned / identical cut-off frequencies. Microphones with better tuned cutoff frequencies allow for better beamforming performance using multiple microphones.FIG. 1 shows a microphone assembly 100 that includes a microelectromechanical system (MEMS) acoustic sensor 105 and a MEMS die, respectively, and a processing circuit 110 that converts acoustic signals (e.g., changes in barometric pressure) to electrical signals. The MEMS sensor 105 may be implemented as a capacitive sensor or as a piezoelectric sensor. In FIG. 1, the MEMS die is a capacitive sensor having a back plate 135 and a membrane 140. Alternatively, another acoustic sensor may be used. The microphone assembly 100 includes a housing 115 defining an enclosed volume 145. The housing 115 includes a base 120 and a cover 125 attached thereto that encloses and protects the MEMS sensor 105 and the processing circuitry 110 disposed therein. An opening 150 in the housing 115 allows the MEMS sensor 105 to sense changes in air pressure outside the housing. The base 120 may be embodied as a layered material such as FR4 with embedded conductors for forming a circuit board. The cover 125 may be embodied as a metal can or layered FR4 material that may also include conductors embedded therein. The cover 125 or the cover can also be formed from other materials, such as plastic or ceramic, and can contain an electromagnetic shield. In some embodiments, the housing 115 includes external contacts on one surface that form an external device interface for integration with a host device in a reflow or wave soldering operation. In one embodiment, the interface includes power, ground, clock, data, and select contacts. The individual contacts of the interface generally depend on the protocol with which data is communicated between the microphone array 100 and the host device. Such protocols are, for example, PDM, SoundWir, I2S and I2C, but further protocols can also be developed in the future.The processing circuit 110 (also referred to as electrical circuit, audio signal processing circuit, or electrical audio signal circuit) is configured to receive the acoustic signal from the MEMS sensor 105. The MEMS sensor 105 may be operatively connected to the processing circuitry 110 using one or more bond wires 130. In other embodiments, other connection mechanisms such as vias, traces, electrical connectors, etc. may be used to electronically connect the MEMS sensor 105 to the processing circuit 110. The processing circuit 110 receives the acoustic signal from the MEMS sensor 105 for further processing. And after processing the acoustic signal, the processing circuitry passes the processed acoustic signal to another computing or host device (e.g., a smartphone) for use therein.Only those components of the microphone assembly 100 that are required for understanding the invention will be explained herein. Also, some other components, such as motors, charge pumps, current sources, filters, resistors, etc., that are desirable or necessary for the functions described herein are discussed in the following description.In addition, various variations will be explained in the following description. For example, the processing circuitry 110 and the MEMS sensor 105 are shown as separate components, but in some embodiments the processing circuitry and the MEMS sensor may be integrated into a single component. In some embodiments, the MEMS sensor 105 and the processing circuit 110 may both be formed from a semiconductor die using, for example, complementary mixed signal metal oxide semiconductor devices. In other embodiments, other techniques may be used to form the MEMS sensor 105 and the processing circuitry 110. In some embodiments, processing circuit 110 may be configured as an application specific integrated circuit (ASIC).As shown in the simplified electrical circuit diagram of FIG. 2, a processing circuit 200 includes an amplifier 205 having a first input node 210 and a second input node 215. The first input node 210 receives a microphone signal from a combining block 220 that combines an analog acoustic signal 225 of a transducer 230 (also referred to as acoustic transducer) with an output 235 of a current digital-to-analog converter (IDAW) 240. The second input node 215 of the amplifier 205 is connected to a virtual ground. Processing circuit 200 is shown for simplicity as a single end system (i.e., having a single output from transducer 230 and a single output from IDAW 240). In other embodiments, the processing circuit 200 may be configured as a differential system.In a differential system, rather than connecting the second input node 215 to the virtual earth, the amplifier 205 may be used to support a differential signal from the transducer 230. In a differential system, transducer 230 is configured to generate two outputs. The second output from transducer 230 may be an inverted output as compared to acoustic signal 225. Accordingly, the IDAW 240 is configured to generate two outputs in a differential system, where the second output may be an inverted signal compared to the output 235 of the IDAW. In a differential system, the second output from transducer 230 may be combined with the second output from IDAW 240 in a combination block similar to combination block 220, and the output of the combination block may be provided to second input node 215. Thus, in a differential system, the second input node 215 may be connected to a negative differential output (not shown) of the transducer 230 via a negative node (not shown) that also includes a negative output (not shown) from the IDAW 240.In some embodiments, the combining block 220 is a summing block that sums the acoustic signal 225 from the transducer 230 with the output 235 from the IDAW 240. In some embodiments, instead of using the combining block 220, the output 235 from the IDAW 240 may be directly connected to the acoustic signal 225 from the transducer 230. By a "direct connection" is meant herein a connection via an electrically conductive path without intervening active devices such as transistors, but possibly via passive components such as resistors, capacitors, conductive traces, wires, etc. In other embodiments, other mechanisms may be used for combining acoustic signal 225 from transducer 230 with output 235 from IDAW 240. The combining block 220 has been described as a summing block, and in other embodiments, other mechanisms for combining the acoustic signal 225 from the transducer 230 with the output 235 from the IDAW 240 may be used. In some embodiments, acoustic signal 225 and output 235 may be subtracted from each other rather than added together. Other functions may also be used to combine acoustic signal 225 with output 235.By combining acoustic signal 225 from transducer 230 with output 235 from IDAW 240, the low frequencies that are otherwise input to the amplifier may be filtered. Such filtering prevents the amplifier 205 from being overloaded with low frequency components (e.g., noise components) of the acoustic signal 225. By reducing the overload on amplifier 205, the amplifier can receive and process the full dynamic range of the microphone signal without unacceptable distortion. In some embodiments, these low frequencies in acoustic signal 225 from transducer 230 are suppressed using output 235 from IDAW 240. The output 235 supplies a low frequency anti-phase component that cancels or suppresses the low frequency components of the acoustic signal 225, such that the microphone signal input to the first input node 210 is substantially free of the low frequency components of the acoustic signal 225.The amplifier 205 receives the microphone signal at the first input node 210 without (or substantially without) the low frequency components and amplifies the microphone signal to an amplified microphone signal 245 which is then input to an analog-to-digital converter (ADC) 250. The amplifier 205 may be configured with a specified gain. The "gain" of amplifier 205 is understood to mean the gain capability of the amplifier, which in some embodiments is expressed as a ratio between the output of the amplifier (e.g., amplified microphone signal 245) to the input of the amplifier (e.g., the microphone signal at first input node 210). In some embodiments, the gain of amplifier 205 is adjusted to obtain a more precise cut-off frequency of microphone assembly 100.Various variations in the amplifier 205 are possible. In some embodiments, amplifier 205 is a differential amplifier that generates amplified microphone signal 245 in a differential or balanced format having a positive signal component 255 and a negative signal component 260. In other embodiments, the amplifier 205 may be a standard amplifier that generates the amplified microphone signal 245 in a single-ended format. In some embodiments, amplifier 205 may be an AC amplifier or a DC amplifier. Generally speaking, amplifier 205 may be any amplifier suitable for performing the functions described herein. Only a single instance of the amplifier 205 is shown here, but in some embodiments, multiple instances of the amplifier may be connected in series or other topologies may be used. Accordingly, in some embodiments, amplifier 205 may use multiple amplifier stages, filters, or other components that may be considered necessary or desirable for obtaining amplified microphone signal 245 to perform the functions described herein.The amplified microphone signal 245 from the amplifier 205 is input to the ADC 250. The ADC 250 is configured to receive, sample, and quantize the amplified microphone signal 245 and generate a corresponding digital microphone signal 265, which is then input to a decimator 270. The ADC 250 receives an analog signal (e.g., the amplified microphone signal 245) and converts the analog signal to a digital signal (e.g., the digital microphone signal 265).The ADC 250 may also be configured in various ways. In some embodiments, the ADC 250 is configured to output the digital microphone signal in a multi-bit format. In other embodiments, the ADC 250 is configured to generate the digital microphone signal 265 in a single bit format. In some embodiments, the ADC 250 is based on a sigma-delta converter (ΣΔ), while in other embodiments the ADC is based on another type of converter such as a flash ADC, a data encoded ADC, a Wilkinson ADC, a pipelined ADC, etc. The ADC 250 may also be configured to generate the digital microphone signal 265 at a specific sampling frequency or sampling rate. In some embodiments, the sampling frequency of the digital microphone signal may be between 2 MHz and 20 MHz and about 3.072 MHz. In other embodiments, the sampling frequency may vary.The ADC 250 may input the digital microphone signal 265 to the decimator 270. Decimator 270 samples down the digital microphone signal 265 to reduce the data size (also referred to as data rate) of the digital microphone signal. In some embodiments, decimator 270 may downsample digital microphone signal 265 by reducing the sampling frequency of the digital microphone signal. For example, decimator 270 may downsample digital microphone signal 265 from a 3.072 MHz frequency to about 384 kHz. In other embodiments, decimator 270 may sample to other frequencies depending on the sampling frequency of digital microphone signal 265 and the desired down-sampled frequency. After down sampling, decimator 270 outputs a down sampled microphone signal 275.Although not shown, in some embodiments, the down-sampled microphone signal 275 may be sent as an input to other components (e.g., an interpolator, a digital-to-digital converter, etc.) for further processing before being input to a host processor of a received device (e.g., smart phones) for use.The down-sampled microphone signal 275 may also be provided to a feedback loop 280. The feedback loop 280 returns the down-sampled microphone signal 275 to the first input node 210 of the amplifier 205 via at least one digital loop filter 285, the IDAW 240 and the combining block 220.The down-sampled microphone signal 275 may be input to the digital loop filter 285 of the feedback loop 280. The digital loop filter 285 filters the downsampled microphone signal 275 according to an adjustable or fixed transfer function to generate a digital feedback signal 290, which is then input to the IDAW 240. The IDAW 240 may be applied to the source current or sink current from a capacitive element (not shown) connected to the output 235 of the IDAW. The IDAW 240 may convert the digital feedback signal 290 to an analog signal. The IDAW 240 may also be configured in other ways. In some embodiments, the IDAW 240 may consist of a plurality of individually controllable current generators, which may be selectively configured for a source or sink current. Other configurations of the IDAW 240 are also possible within the scope of the invention.The output 235 of the IDAW 240 is fed back to the combining block 220. Thus, feedback loop 280 feeds an analog feedback signal (e.g., output 235) back to combining block 220 where it is combined with acoustic signal 225 from transducer 230 and input back to first input node 210 of amplifier 205.Only the digital loop filter 285 and the IDAW 240 are shown in the feedback loop 280, and additional or different components may also be provided in some embodiments. In some embodiments, the feedback loop 280 may include pulse width and pulse amplitude modulators, filters or other digital-to-analog converters, amplifiers, etc. Accordingly, only amplifier 205, ADC 250, and decimator 270 are shown in the processing circuit external to feedback loop 280, but other components such as filters, modulators, etc., which are considered desirable or required for performing the functions described herein, may also be used. Further, only individual instances of the ADC 250, decimator 270, digital loop filter 285, and IDAW 240 are shown, however, in some embodiments, multiple instances of one or more of these components may also be used in the processing circuit 200. In some embodiments, one or more of the components described above may be integrated into a single component.By using the feedback loop 280 and combining the output 235 from the feedback loop with the acoustic signal 225 from the transducer 230 in the combining block 220 prior to inputting the microphone signal to the first input node 210 of the amplifier 205, the processing circuit 200 effectively prevents low frequency overload at the amplifier 205 and / or the ADC 250. The low frequency components in the acoustic signal 225 from the transducer 230 may be unwanted noise components caused, for example, by exposure of the microphone assembly 100 to intense sub-sound or low frequency sound such as generated by wind, large machinery, etc. By suppressing these low frequency components in acoustic signal 225, processing circuit 200 eliminates susceptibility of amplifier 205 to overload caused by a low frequency component and distortion caused by saturation and nonlinearity of active gain elements such as transistors of the amplifier.The undesired low frequency components in the acoustic signal 225 from the transducer 230 may be expressed by a cut-off frequency. In some embodiments, the cut-off frequency may be set such that all frequency components below the cut-off frequency are considered undesired low frequency components that may be suppressed using the output 235, wherein the sound quality of the resulting microphone signal is not degraded, but may even be improved. In this way, the noise floor of the microphone assembly 100 may be adjusted by varying the cutoff frequency response of the transducer 230. The cut-off frequency of the microphone may also vary depending on the application of the microphone assembly 100.Conventionally, the cut-off frequency of a microphone (e.g., the microphone array 100) is adjusted using an analog filter. When an analog filter is used, the cut-off frequency is usually set during manufacture and cannot be changed after setting. Thus, the cut-off frequency cannot be easily programmed when an analog filter is used. In other conventional approaches, a digital filter may be used in the microphone (e.g., in the microphone array 100). The digital filter is also usually programmed during manufacture to set a specific cut-off frequency and may or may not be programmable thereafter. However, as mentioned above, knowledge of the loop gain is required for precise control of the cut-off frequency.In FIG. 3, a processing circuit 300 includes a transducer 305 whose acoustic signal 310 is combined with an output 315 of an IDAW 320 in a combining block 325 before being input as a microphone signal to a first input node 330 of an amplifier 335. The amplifier 335 amplifies the microphone signal to generate an amplified microphone signal 340, which is then input to an ADC 345. The ADC 345 converts the amplified microphone signal 340, which is analog, to a digital microphone signal 350.The digital microphone signal 350 from the ADC 345 is input to a decimator 355 for down sampling. The downsampled microphone signal 360 from decimator 355 is then passed for further processing and use (e.g., to a host processor of a receiving device). In contrast to the processing circuit 200 of FIG. 2 including the feedback loop 280, the feedback loop in the processing circuit 300 of FIG. 3 is broken or, in other words, open. In particular, the downsampled microphone signal 360 from the decimator 355 is not fed back to the combining block 325 via a digital loop filter (e.g., the digital loop filter 285) and the IDAW 320. Instead, in the processing circuit 300, a wave generator 365 that generates a reference waveform or signal having a known amplitude and frequency is connected to an input 370 of the IDAW 320.The reference signal from the wave generator 365 may be used for setting the loop gain by calibrating the IDAW 320, and more specifically by calibrating a source current or gain of the IDAW. By calibrating the source current or gain of the IDAW 320, the cut-off frequency of the microphone array 100 can be set more precisely. Wave generator 365 may be any type of wave generator that can generate sinusoidal or quadratic waveforms.The processing circuit 300 may be configured with various variations. Similar to the processing circuit 200, the configuration of the various components of the processing circuit 300 may vary in various embodiments. Other or additional components that may be necessary or desirable for performing the functions described herein may also be used in the processing circuit 300.Referring now to FIG. 4, a flowchart is shown illustrating steps of a process 400 for adjusting the cut-off frequency of a microphone signal (e.g., the microphone array 100). The cut-off frequency of the microphone signal may be adjusted by adjusting a loop gain of the processing circuit (e.g., the processing circuit 200). The loop gain of the processing circuit may be adjusted at least in part by calibrating the IDAW (e.g., the IDAW 240) or at least in part by calibrating the amplifier (e.g., the amplifier 205). Furthermore, by calibrating the amplifier, the sensitivity of the microphone can be adjusted.After starting the process in step 405, the feedback loop of the processing circuit is broken in step 410. In Figure 2, the feedback loop 280 is broken by separating the downsampled microphone signal 275 from the decimator 270 and removing the digital loop filter 285 from the feedback loop. In FIG. 4, the amplifier is calibrated in step 415. In some embodiments, the amplifier may be calibrated before the feedback loop is broken. The sequence of steps 410 and 415 can thus be changed or reversed. The amplifier is calibrated by adjusting a gain of the amplifier. In some embodiments, the cutoff frequency is also dependent on the magnitude of the input capacitance in the transducer and parasitic capacitances of the MEMS sensor. The effect of the input capacitance and parasitic capacitances on the cut-off frequency can also be reduced by adjusting the gain of the amplifier. The calibration of the amplifier is explained in FIG. 5.In step 420, the wave generator is inserted into the processing circuit. In Figure 2, the wave generator is inserted by removing the digital loop filter 285 from the feedback loop 280 and connecting the wave generator to the input of the IDAW as shown in Figure 3. In FIG. 4, the IDAW is calibrated in step 425. The IDAW is calibrated by adjusting a current of the IDAW, either by adjusting the value of a current source (analog adjustment) or by adjusting a gain of the input to the IDAW (digital adjustment). The calibration of the IDAW is explained in Figure 6. After calibrating the IDAW, the feedback loop is restored in step 430. In FIG. 2, the feedback loop 280 is recovered by disconnecting the wave generator and connecting the digital loop filter 285 back into the feedback loop to receive the down-sampled microphone signal 275 from the decimator 270 and output the digital feedback signal 290 to the IDAW 240. This restores the configuration of the processing circuit 200 after calibrating the IDAW. The process of FIG. 4 then ends in step 435.Process 400 shows calibration of the amplifier prior to calibration of the IDAW, wherein in some embodiments the IDAW may also be calibrated prior to calibration of the amplifier. Preferably, by calibrating the amplifier prior to calibrating the IDAW, the cut-off frequency to be set by calibrating the IDAW may be determined accurately (or substantially accurately) to also account for the various capacitance tolerances (e.g., parasitic capacitance) that are accounted for during calibration of the amplifier. Further, in some embodiments, the calibration of the amplifier is performed using an external tone (e.g., a signal) and the calibration of the IDAW for adjusting the cut-off frequency is performed using an internal test tone (e.g., a signal). Thus, the amplifier may be calibrated during any step of process 400 as long as the amplifier may be calibrated using an external tone and the IDAW may be calibrated using an internal test tone. For example, in some embodiments, the amplifier may be calibrated between step 420 and step 425 or before step 410 as discussed above. In some embodiments, the calibrations of the amplifier and the IDAW may occur simultaneously (or substantially simultaneously) using two separate frequency signals.Additionally, in some embodiments, the calibrations of the amplifier and the IDAW and controlling the cut-off frequency of the microphone signal may be performed either internally on a chip or externally on a host processor.Adjusting the cut-off frequency of the microphone signal includes two types of calibrations: a first calibration of the amplifier and a second calibration of the IDAW. In other words, adjusting the cut-off frequency includes calibrating the processing circuit by calibrating the amplifier and calibrating the IDAW.Referring now to FIG. 5, a flowchart is shown illustrating steps of a process 500 for calibrating the amplifier 335. The purpose of calibrating amplifier 335 is to adjust the sensitivity of the microphone (e.g., microphone assembly 100) such that the level above the cutoff frequency is equal. The amplifier 335 is calibrated by adjusting the gain of the amplifier. In some embodiments, the gain of amplifier 335 is adjusted to also adjust (e.g., reduce) the effect of the parasitic capacitance and other tolerance capacitances of microphone array 100 to obtain an acoustic output (e.g., down-sampled microphone signal 275) with a higher quality. When the amplifier 335 generates a differential output (e.g., the amplified microphone signal 340), the negative and positive components (e.g., the positive signal component 255 and the negative signal component 260 of the amplified microphone signal 245) of the differential output may have a parasitic capacitance associated therewith. In these cases, the gain of the amplifier 335 for the negative and positive components must be adjusted to account for the parasitic capacitance of these components.After the process is started in step 505, an acoustic signal is generated in step 510. The acoustic signal is the acoustic signal 310 from the transducer 305. In some embodiments, the acoustic signal may be generated using a charge pump. In other embodiments, other mechanisms for generating the acoustic signal may be used.The acoustic signal 310 is combined with the output 315 of the IDAW 320 in the combining block 325 to obtain a microphone signal. In embodiments where feedback loop 280 is broken before calibration of amplifier 335, output 315 from IDAW 320 may be equal to or near zero (e.g., because wave generator 365 does not generate an input signal for the IDAW, which in turn does not generate output 315), such that the output (e.g., the microphone signal) of combining block 325 substantially corresponds to acoustic signal 310 input via first input node 330 of amplifier 335. In embodiments where feedback loop 280 is broken after calibrating amplifier 335, acoustic signal 310 may be combined with output 315 from IDAW 320 before being input as the microphone signal via first input node 330 of the amplifier. In some embodiments, a 1 kHz signal having a sound pressure level of four decibels (94 dBSPL @ 1 kHz) may be used as the acoustic signal 310. In other embodiments, an acoustic signal having a different intensity may be used.The acoustic signal is amplified by the amplifier 335 in step 515 to obtain an amplified microphone signal. The amplified microphone signal may be a differential signal having a positive signal component and a negative signal component. The amplified microphone signal is converted to a digital microphone signal by the ADC 345 and downsampled by the decimator 355. Prior to the beginning of the calibration process of amplifier 335, in some embodiments, it is determined whether or not processing circuit 300 is operating as desired.The output (e.g., down-sampled microphone signal 360) of decimator 355 may be measured to confirm that processing circuit 300 is operating as desired. For example, in some embodiments, when a response (e.g., an acoustic response) from the output (e.g., down-sampled microphone signal 360) of decimator 355 is measured in response to the microphone signal, processing circuit 300 operates correctly. If no response is measured from the output (e.g., down-sampled microphone signal 360) of decimator 355, then there may be a malfunction of processing circuit 300 and it must be repaired before amplifier 335 can be calibrated.When a response is measured from the output (e.g., down-sampled microphone signal 360) of decimator 355, then the process to calibrate amplifier 335 begins in step 520. In some embodiments, the output of the ADC 345 may be used to determine whether or not the processing circuit 300 is operating as desired, and also to take the various measurements described below for calibrating the amplifier. The amplifier 335 may be configured to generate a differential signal having a positive signal component and a negative signal component. The gain of the amplifier 335 for the negative signal component and the positive signal component is adjusted to calibrate the amplifier.Therefore, in step 520, the negative signal component of the amplified microphone signal 340 is gated out. In some embodiments, the negative signal component may be masked by obstructing that component with a virtual earth. In other embodiments, other mechanisms for gating out the negative signal component may be used such that the value of the negative signal component is substantially equal to zero. By gating out the negative signal component of the amplified microphone signal 340, only the positive signal component of the amplified microphone signal is input to the ADC 345. The ADC 345 converts the positive signal component of the amplified microphone signal 340 to a digital signal which is then downsampled by the decimator 355 to generate the downsampled microphone signal 360.The downsampled microphone signal 360 of decimator 355 is measured in step 525. In some embodiments, the output of the ADC 345 is measured in step 525. In some embodiments, the down-sampled microphone signal 360 or the output of the ADC 345 may be measured by measuring the level (e.g., an effective value) around a test sound, either with filtering or using a fast Fourier transform value. In other embodiments, other mechanisms may be used for measuring the downsampled microphone signal 360 or the output of the ADC 345. By measuring the downsampled microphone signal 360 of decimator 355 (or the output of ADC 345), the sensitivity of microphone assembly 100 may be calculated.The sensitivity of the microphone assembly 100 is dependent on the acoustic signal 310, the down-sampled microphone signal 360 of the decimator 355 (or the output signal of the ADC 345), and the gain of the amplifier 335. By measuring the downsampled microphone signal 360 of decimator 355 (or the output of ADC 345) for acoustic signal 310 in step 510 and knowing the gain of amplifier 335, the sensitivity of microphone assembly 100 may be calculated. If the calculated sensitivity is not within a desired range, the gain of the amplifier may be adjusted until the desired sensitivity of the microphone assembly 100 is obtained.The calibration of the amplifier 335 thus results in an adjustment of the sensitivity of the microphone arrangement 100. By adjusting the sensitivity of the microphone assembly 100, the noise levels to which the microphone assembly 100 is sensitive can be adjusted. The sensitivity of the microphone arrangement 100 can be adjusted such that the microphone arrangement can ignore frequency components below the limit frequency of the microphone arrangement. Depending on the application in which the microphone assembly 100 is used, high or low sensitivity may be desirable. Generally, a microphone array with a higher sensitivity generates a higher output voltage for the down-sampled microphone signal 275, thus requiring a lower gain at the amplifier 335. In contrast, a microphone arrangement with a lower sensitivity generates a lower output voltage at the down-sampled microphone signal 275, so that a higher gain is required at the amplifier 335. Accordingly, the sensitivity of the microphone assembly 100 may be adjusted to obtain the desired cut-off frequency.Thus, in step 530, the calculated sensitivity of the microphone assembly 100 is compared to a predetermined value or reference value corresponding to the desired sensitivity of the microphone assembly. If the calculated sensitivity does not match (i.e., does not match or fall within a range of) the desired sensitivity, the gain of amplifier 335 is adjusted until the desired sensitivity value of microphone assembly 100 is obtained. In some embodiments, the gain of amplifier 335 may be changed in increments of one quarter of a decibels (0.25 dB). In other embodiments, the gain of amplifier 335 may be changed in other increments. Each time a change in the gain of amplifier 335 is repeated, downsampled microphone signal 360 is measured and the sensitivity of microphone array 100 is calculated until the desired sensitivity of the microphone array is obtained.In some embodiments, instead of incrementally changing the gain of the amplifier 335 until the desired sensitivity is obtained, a look-up table may be used to determine the gain of the amplifier in accordance with a desired level of sensitivity of the microphone assembly 100. When using a look-up table, instead of incrementally increasing or decreasing the gain of the amplifier 335 and calculating the sensitivity value of the microphone assembly 100, the look-up table may be used to determine the correct gain of the amplifier in a single step for the desired sensitivity value. In some embodiments, if the desired sensitivity value is not included in the lookup table, the lookup table may still be used to find a gain that is closer to the desired sensitivity, and then the gain of the amplifier may be incrementally changed until the desired sensitivity is obtained, thereby minimizing the number of repetitions. The desired sensitivity of the microphone arrangement 100 can thus be obtained more quickly using a look-up table.Once the gain of amplifier 335 has been adjusted for the positive signal component of amplified microphone signal 340, the negative signal component that was gated out in step 520 is gated in again in step 535. In some embodiments, the negative signal component may be re-gated by cancelling the connection to the virtual earth or reversing the mechanism used to gate the component.In step 540, the positive signal component of the amplified microphone signal 340 is gated out. Again, the positive signal component can be gated out by connecting this component to a virtual ground or using some other mechanism. In this case, the negative signal component of the amplified microphone signal 340 is inserted again and the positive signal component is inserted. The process for adjusting the gain of amplifier 335 is repeated for the negative signal component in step 545 and step 550 until the calculated sensitivity of microphone assembly 100 corresponds to the predetermined value or reference value, which corresponds to the desired sensitivity of the microphone assembly.In some embodiments, the process for adjusting the gain of the negative signal component amplifier 335 may be started by setting the gain of the amplifier to a gain value determined in the positive signal component step 530. In other embodiments, a gain value other than a starting point may be used. Again, in some embodiments, a look-up table may be used. Further, in some embodiments, the gain value of amplifier 335 may be the same for the negative and positive signal components. In other embodiments, the gain value for the negative signal component may be different from the gain value of the positive signal component. The gain of amplifier 335 is continuously adjusted until the desired microphone sensitivity is obtained.After the desired sensitivity of the microphone arrangement 100 is obtained at the negative signal component of the output of the amplifier 335, the positive signal component which was masked in the step 540 is again masked in the step 555. If the negative and positive signal components are both re-gated, the down-sampled microphone signal 360 (or the output of the ADC 345) is again measured in step 560. If the downsampled microphone signal 360 (or the output of the ADC 345) corresponds to the predetermined value (corresponding to the desired sensitivity of the microphone array 100) in step 565 in step 560, the calibration of the amplifier 335 is completed in step 570 and the process 500 is terminated in step 575. If, in step 565, the sensitivity of the microphone assembly 100 is not as desired, the process 500 returns to step 520 and the gain of the amplifier 335 is again set for the positive and negative signal components of the amplified microphone signal 340, respectively, until the desired sensitivity is obtained.Various modifications of the process 500 are possible within the scope of the invention. For example, in process 500, the negative signal component of amplified microphone signal 340 is first gated out and the gain of positive signal component amplifier 335 is adjusted, while in other embodiments, the positive signal component may be first gated out to adjust the gain of the negative signal component amplifier first. In addition, in embodiments where amplifier 335 generates a single-ended amplified microphone signal, the gain of the amplifier may also be adjusted without fading / re-fading signal components.Process 500 uses a look-up table with gain values of amplifier 335 corresponding to different sensitivity values, and in some embodiments, other or additional look-up tables may also be used. For example, in some embodiments, a look-up table may be used with values of the down-sampled microphone signal 360 (or values of the output of the ADC 345) corresponding to the sensitivity of the microphone assembly 100, such that, instead of calculating the sensitivity of the microphone after measuring the down-sampled microphone signal (or the output of the ADC 345), the look-up table may be used to determine the sensitivity corresponding to the measured down-sampled microphone signal (or the output of the ADC). Additionally, in some embodiments, the desired sensitivity for the negative signal component (e.g., the predetermined value in step 530) may be different from the desired sensitivity (e.g., the predetermined value in step 550) of the positive signal component, which in turn may be different from the desired overall sensitivity (e.g., the predetermined value in step 565). In some embodiments, the predetermined values may be set in step 530 and step 550 such that the predetermined value is obtained in step 565.In some embodiments, it may be advantageous to wait for some (e.g., a few fractions of a second) between different steps of the process 500. For example, in some embodiments, it may be advantageous to wait for approximately twenty milliseconds (20 ms) between gating out the negative or positive signal components and measuring the down-sampled microphone signal 360 at the output of decimator 355. By waiting between two steps, trace signals from the previous repetition may be resolved, thereby preventing erroneous measurements of the down-sampled microphone signal 360. Other variations for performing the process 500 may be provided in other embodiments.In some embodiments, the control of the calibration process of the amplifier 335 may be external, wherein the control and measurements are performed externally for calibrating the amplifier or internally at startup or upon a request made by a host processor for a calibration of the microphone assembly 100. An advantage of the process 500 is that another MEMS element (e.g., the microphone array 100) may be calibrated with possibly different gains in the positive and negative sides using the process described above.Referring now to FIG. 6, a flowchart is shown illustrating steps of a process 600 that may be used for calibrating the IDAW 320. After starting in step 605, acoustic signal 310 is deactivated in step 610. In some embodiments, the acoustic signal 310 may be deactivated by deactivating the charge pump of the microphone array 100. In other embodiments, other mechanisms for disabling acoustic signal 310 may be used. Preferably, the process 600 is performed in a quiet environment to avoid interference from scattering sound or noise signals that may interfere with the calibration of the IDAW320. In some embodiments, the calibration of the IDAW 320 may be performed in a sound enclosure (if calibrated in a sound enclosure, the charge pump may not need to be deactivated). In other embodiments, other mechanisms for blocking unwanted noise or sound may be used prior to calibrating the IDAW 320.Prior to calibrating the IDAW 320, the feedback loop 280 is broken and the wave generator 365 is inserted into the broken feedback loop. Specifically, wave generator 365 is connected to input 370 of IDAW 320. In step 615, wave generator 365 generates a sinusoidal signal for input to IDAW 320 via input 370. In some embodiments, wave generator 365 may be used to generate a 1 kHz sine wave without attenuation. In other embodiments, signals having different intensities may be generated by wave generator 365. It has been described above that the wave generator 365 generates a sinusoidal signal, wherein in other embodiments the wave generator may also generate a quadratic signal that may be used for calibrating the IDAW 320.The signal generated by wave generator 365 is input to IDAW 320 via input 370. In step 620, the output (e.g., down-sampled microphone signal 360) of decimator 355 or the output of ADC 345 is measured. Again, the output of decimator 355 or the output of ADC 345 may be measured by measuring the root mean square around the test tone, either by filtering or using a fast Fourier transform measurement. From the measured value of the downsampled microphone signal 360 (or the value of the output of the ADC 345), it is determined whether a source current of the IDAW 320 or a gain of an input to the IDAW needs to be adjusted to obtain the desired cut-off frequency of the microphone assembly 100. Generally, either the source current or the gain of the input to the IDAW 320 is used to calibrate the IDAW.In step 625 and step 630, the value of the downsampled microphone signal 360 (or the value of the output of the ADC 345) is compared to a predetermined value or reference value. If the value of the downsampled microphone signal 360 (or the value of the output of the ADC 345) is outside a specific percentage of the predetermined value, then the source current or gain of the input to the IDAW 320 is readjusted. In some embodiments, a percentage between 30 and 46 percent may be used for comparing the down-sampled microphone signal 360 (or the value of the output of the ADC 345) with the predetermined value. In other embodiments, a percentage between 1 and 5 percent may be used. In other embodiments, different percentage ranges may be used for comparing the down-sampled microphone signal 360 (or the value of the output of the ADC 345) with the predetermined value.The source current of the IDAW 320 or the gain of the input to the IDAW is incrementally adjusted until the value of the downsampled microphone signal 360 (or the value of the output of the ADC 345) corresponds to (i.e., matches or falls within a specific range of) the predetermined value. Thus, the IDAW 320 may be adjusted by adjusting a current of the IDAW by either adjusting a value of the source current of the IDAW (analog adjustment) or adjusting the gain of the input to the IDAW (digital adjustment).The predetermined value is a value of the down-sampled microphone signal 360 (or the value of the output of the ADC 345) that corresponds to the desired cut-off frequency of the microphone array 100. By adjusting the source current of the IDAW 320 or the gain of the input to the IDAW until the down-sampled microphone signal 360 (or the value of the output of the ADC 345) corresponds to the predetermined value, the desired cut-off frequency can be obtained. In some embodiments, the predetermined value may be determined from a look-up table, where the look-up table may include a corresponding predetermined value for the down-sampled microphone signal 360 for each desired cut-off frequency. In other embodiments, other mechanisms for determining the predetermined value may be used.In some embodiments, a specific range of the predetermined value may be defined such that when the value of the down-sampled microphone signal 360 (or the value of the output of the ADC 345) falls within this range, the IDAW 320 is considered calibrated. In some embodiments, a range of 30 to 46 percent or a range of 1 to 5 percent of the predetermined value may be used, while in other embodiments, other value ranges may be used. In other embodiments, no ranges may be used, so in this case, the value of the downsampled microphone signal 360 (or the value of the output of the ADC 345) is to correspond very close to the predetermined value. The acceptable range of the downsampled microphone signal 360 (or the value of the output of the ADC 345) may vary depending on the desired cutoff frequency and the predetermined value to which the downsampled microphone signal is compared, in various embodiments.In step 630, the value of the downsampled microphone signal 360 (or the value of the output of the ADC 345) is compared to the predetermined value. If the value of the downsampled microphone signal 360 (or the value of the output of the ADC 345) is within the acceptable range of the predetermined value, then the process 600 proceeds to step 635, where the calibration of the IDAW 320 is considered complete. If, in step 630, the value of the downsampled microphone signal 360 (or the value of the output of the ADC 345) is not within the specified range of the predetermined value, the process 600 returns to step 625, in which the source current of the IDAS 320 or the gain of the input to the IDAW is adjusted and the value of the downsampled microphone signal 360 (or the value of the output of the ADC 345) is remeasured. The process of adjusting the source current or gain of the IDAW 320 and measuring the value of the downsampled microphone signal 360 (or the value of the output of the ADC 345) continues until the value of the downsampled microphone signal (or the value of the output of the ADC) corresponds to the predetermined value.Similar to the calibration of amplifier 335, in some embodiments, a look-up table may be used for calibrating IDAW 320. The look-up table may be used to determine the value of the source current or gain of the IDAS 320. In some embodiments, the look-up table may include cut-off frequencies and corresponding source current or gain values for the IDAW 320. Knowing the cut-off frequency, the corresponding source current or gain value of the IDAW 320 or the closest source current or gain value of the IDAW corresponding to the desired cut-off frequency can be determined from the look-up table, thereby accelerating calibration of the IDAW.After calibrating the IDAW 320 in step 635, the charge pump or other mechanism supplying the acoustic signal 310 is activated (if it was deactivated) in step 640. The process 600 is ended in step 645.By calibrating the amplifier 335 and the IDAW 320, the loop gain of the processing circuit 200 is adjusted to set a desired cut-off frequency of the microphone array 100. By setting the cut-off frequency of the microphone array 100, frequencies below the cut-off frequency may be filtered out by the processing circuit 200 (e.g., at the combining block 220) to filter out at least a portion of the noise in the acoustic signal 225 from the transducer 230 and improve the quality of the down-sampled microphone signal 275. In some embodiments, the calibration of the amplifier 335 and the IDAW 320 may be performed at startup (e.g., when the microphone assembly 100 is mounted or calibrated in the factory), during manufacture of the microphone, or also periodically after mounting, when the microphone is not used by the application in which the microphone is integrated.FIG. 7A is a graph 700 showing the effect on beam formation using two microphones with varying cutoff frequencies. The graph 700 shows a signal 705 of a first microphone and a signal 710 of a second microphone, wherein each microphone has a varying cut-off frequency. The first microphone signal 705 has a cutoff frequency of about 30 Hz and the second microphone signal 710 has a cutoff frequency of about 35 Hz. Although there is only a 5 Hz difference between the cut-off frequencies of the signal 705 and the signal 710, the small difference between the cut-off frequencies may introduce a phase difference 715 between the signals when the signals are combined together for beamforming.A difference in the phases of the signal 705 and the signal 710 may result in sound distortion and noise in the resulting beam (e.g., the composite beam when the beams are combined by the microphones), which may not be acceptable in certain applications such as hearing aids. The phase difference 715 between the signal 705 and the signal 710 may be eliminated or at least minimized by tuning the cutoff frequencies of these signals.In FIG. 7B, a polar diagram 720 is shown showing how certain frequencies are reproduced as they enter a microphone from various angles. In the polar diagram 720, a first composite beam 725 and a second composite beam 730 are shown. The first composite beam 725 is formed by at least two microphone signals having different cut-off frequencies, while the second composite beam 730 is formed by at least two microphone signals having the same (or substantially the same) cut-off frequencies. The curve of the first composite beam 725 appears in the polar diagram 720 as an omnidirectional response that picks up sound including noise from all directions. The curve of the second composite beam 730 appears in the polar diagram 720 as a kidney-shaped response, meaning that the composite beam picks up sound from a front direction and some sound from a rear direction, while filtering out sound from other directions. The first composite beam 725 and the second composite sound 730 thus generate different responses to a same acoustic signal of approximately 200 Hz.In some applications such as hearing aids, better sound quality with minimal noise distortions is preferred. In these cases, a kidney-shaped reaction from a composite beam can be advantageous, which transmits a certain sound and excludes another sound. The kidney-shaped response can be obtained by tuning the cut-off frequencies of the beams constituting the composite beam in combination. By tuning the cut-off frequencies of signals from two or more microphones for generating a composite beam, excellent sound quality can be obtained.The invention has been explained above with reference to various embodiments. The invention is not, however, limited to the embodiments described herein, which may be modified and varied in various ways without, therefore, departing from the scope of the invention as defined by the following claims.

Claims

A method in an electrical audio signal circuit including a feedback loop (280) having a digital filter (285) coupled to a current-to-digital-to-analog converter (240, 320), the method comprising: providing (560, 625) an output signal from the current-to-digital-to-analog converter (240, 320) to analog elements of the electrical audio signal circuit, wherein the output signal from the current-to-digital-to-analog converter (240, 320) is based on a reference signal input to the current-to-digital-to-analog converter (240, 320) while no output of the digital filter (285) is input to the current-to-digital-to-analog converter (240, 320); comparing (565, 630) an output signal of the electrical audio signal circuit to a reference, and calibrating (570, 635) the electrical audio signal circuit so that the output signal of the electrical audio signal circuit corresponds to the reference, wherein the calibration of the electrical audio signal circuit enables more precise control of a cut-off frequency of a microphone signal while the output of the digital filter (285) is input to the current-to-digital-to-analog converter (240, 320).The method of claim 1, further comprising. amplifying (625) the output signal from the current to digital to analog converter (240, 320), converting the amplified signal to a digital signal, wherein the output signal of the electrical audio signal circuit is the digital signal, and calibrating (635) the electrical audio signal circuit until the digital signal corresponds to the reference.The method of claim 1, further comprising: amplifying (625) the output signal from the current to digital to analog converter (240, 320); converting the amplified signal to a digital signal; down-sampling the digital signal to a down-sampled signal, wherein the output signal of the electrical audio signal circuit is the down-sampled signal; and calibrating the electrical audio signal circuit until the down-sampled signal corresponds to the reference.The method of claim 1, wherein calibrating the electrical audio signal circuit comprises calibrating (635) the current-to-digital-to-analog converter (240, 320).The method of claim 4, further comprising: combining an analog acoustic signal with the output from the current to digital to analog converter (240, 320) while the output of the digital filter (285) is input to the current to digital to analog filter (240, 320); and gating the analog acoustic signal prior to calibrating the current to digital to analog converter (240, 320).The method of claim 1, wherein providing the reference signal input to the current-to-analog-to-digital converter comprises providing a reference signal having a known frequency and amplitude.The method of claim 1, further comprising: amplifying the output signal from the current to digital to analog converter (240, 320) with an amplifier (205, 335); and calibrating the electrical audio signal circuit by calibrating (570) the amplifier (205, 335).The method of claim 7, wherein calibrating the amplifier (205, 335) comprises: adjusting the gain of the amplifier (205, 335) using a first output component of the amplifier (205, 335) until a desired sensitivity is obtained, and adjusting the gain of the amplifier (205, 335) using a second output component of the amplifier (205, 335) until the desired sensitivity is obtained.An electrical audio signal circuit (200, 300) configured to perform the method of claim 1, comprising: a combining block (220, 325) configured to combine an analog acoustic signal with an output from a current to digital to analog converter (240, 320) to obtain a microphone signal, an amplifier (205, 335) configured to amplify the microphone signal to an amplified microphone signal, and an analog to digital converter (250, 345) configured to convert the amplified microphone signal to a digital microphone signal, a feedback loop (280) including a digital filter (285) coupled to an output of the analog to digital converter and to an input of the current to digital to analog converter (240, 320), wherein the analog-to-digital converter and the amplifier (205, 335) are configured to be controlled so as to adjust the loop gain of the electrical audio signal circuit during calibration until a desired cut-off frequency for the digital microphone signal is reached.The electrical audio signal circuit of claim 9, further comprising a reference signal generator (365) configured to input a reference signal to the current to digital to analog converter (240, 320) while the digital filter (285) is separate from the current to digital to analog converter (240, 320), wherein the loop gain is calibrated while the reference signal is applied to the current to digital to analog converter (240, 320) and the analog acoustic signal is gated out.The electrical audio signal circuit of claim 9, further comprising a decimator (270, 355) configured to convert the digital microphone signal to a down-sampled microphone signal, wherein the decimator (270, 355) is coupled to the analog-to-digital converter and the digital filter (285).The electrical audio signal circuit of claim 9, wherein the current to digital to analog converter (240, 320) contributes to the loop gain and the current to digital to analog converter (240, 320) is calibrated to provide the desired cut-off frequency for the digital microphone signal.The electrical audio signal circuit of claim 9, wherein the amplifier (205, 335) contributes to the loop gain and the amplifier (205, 335) is calibrated to provide the desired sensitivity.A microphone arrangement (100) configured to perform the method of claim 1, wherein the microphone arrangement (100) comprises: an acoustic transducer (230, 305); an electrical audio signal circuit (200, 300) connected to the acoustic transducer (230, 305) and configured to receive an acoustic signal from the acoustic transducer (230, 305), wherein the electrical audio signal circuit comprises: a combining block (220, 325) configured to combine the acoustic signal with an output from a current digital-to-analog converter (240, 320) to obtain a microphone signal; an amplifier (205, 335) configured to amplify the microphone signal to an amplified microphone signal; an analog-to-digital converter (250, 345) configured to convert the amplified microphone signal to a digital microphone signal, and a digital filter (285) coupled to the analog-to-digital converter and the current-to-digital-to-analog converter (240, 320), wherein the amplifier (205, 335) is configured to be controlled to adjust a gain until a desired sensitivity is reached during calibration, and wherein the current-to-digital-to-analog converter (240, 320) is configured to be controlled to adjust a gain until a desired cut-off frequency of the microphone signal is reached during calibration.The microphone assembly of claim 14, further comprising a microphone housing (115) configured to enclose and hold the acoustic transducer (230, 305) and the electrical audio signal circuit.The microphone assembly of claim 14, further comprising a base (120) configured to mount the acoustic transducer (230, 305) and the electrical audio signal circuit.The microphone assembly of claim 14, wherein the acoustic transducer comprises a microelectromechanical (MEMS) sensor (105).The microphone assembly of claim 14, wherein the electrical audio signal circuit further comprises: a decimator (270, 355) configured to downsample the digital microphone signal to a downsampled microphone signal, wherein the downsampled microphone signal is configured to be input to a receiving device.

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