System and method for calibrating microphone cut-off frequency

A system and method in an audio signal electrical circuit including a feedback loop with a digital filter coupled to a current digital to analog converter (IDAC) includes providing an output signal from the IDAC to analog elements of the audio signal electrical circuit, the output signal from the IDAC based upon a reference signal input to the IDAC when an output of the digital filter is not input to the IDAC. The system and method also include comparing an output signal of the audio signal electrical circuit to a reference, and calibrating the audio signal electrical circuit to correspond the output signal of the audio signal electrical circuit to the reference. Calibration of the audio signal electrical circuit enables more precise control of a cut-off frequency of a microphone signal when the output of the digital filter is input to the IDAC.

BACKGROUND

Microphones are now widely used in a variety of applications, such as, in smartphones, mobile phones, tablets, headsets, hearing aids, sensors, automobiles, etc. Noise reduction in these microphones is a crucial feature for obtaining superior sound quality. Present day microphones have limitations due to their configuration and the way they operate.

DETAILED DESCRIPTION

The present disclosure relates generally to a system and method for more precisely setting a cut-off frequency of a microphone signal. The cut-off frequency is controlled generally by a digital filter in a feedback loop, but precise control of the cut-off frequency requires knowledge of the loop gain. Thus the disclosure also pertains to a system and method for setting or adjusting the loop gain of a microphone signal processing or electrical circuit to better control the microphone's cut-off frequency. The processing circuit includes an amplifier for amplifying a microphone signal, an analog to digital converter for converting the amplified microphone signal into 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, particularly, an internal feedback loop. The feedback loop returns the output of the decimator back to a combination block via an electrical current digital to analog converter (IDAC). The combination block receives an acoustic signal from a transducer of the microphone and combines that acoustic signal with the output of the feedback loop to obtain the microphone signal, which is then input into the amplifier.

The loop gain of the processing circuit may be adjusted by controlling the amplifier or the IDAC. The amplifier may be controlled by adjusting a gain of the amplifier. The IDAC may be controlled by an analog adjustment of a source current of the current digital to analog converter. The IDAC may also be controlled using a digital adjustment of a gain before (e.g., at an input) the IDAC. By adjusting the loop gain of the processing circuit by using the amplifier or the IDAC, the sensitivity of the microphone and the cut-off frequency of the microphone may be precisely controlled.

Precise control of the cut-off frequency of the microphone signal improves sound quality, enables filtering of low-frequencies that may overload the amplifier, and permits the manufacture of microphones with closely matched/identical cut-off frequencies. Microphones with better matched cut-off frequencies enable better multi-microphone beamforming performance.

FIG. 1is a microphone assembly100having a microelectomechanical systems (MEMS) acoustic sensor105or die and a processing circuit110that converts acoustic signals (e.g., changes in air pressure) into electrical signals. The MEMS sensor105may be implemented as a capacitive or condenser sensor, or as a piezoelectric sensor. InFIG. 1, the MEMS die is a capacitive sensor with a backplate135and a diaphragm140. Alternatively, some other acoustic sensor may be used. The microphone assembly100includes a housing115defining an enclosed volume145. The housing115includes a base120and a cover125fastened thereto, that encloses and protects the MEMS sensor105and the processing circuit110disposed therein. A port150in the housing115permits the MEMS sensor105to sense changes in air pressure outside the housing. The base120may be embodied as a layered material like FR4 with embedded conductors forming a PCB. The cover125may be embodied as a metal can, or a layered FR4 material, which may also include embedded conductors. The cover125or lid may also be formed from other materials like plastics and ceramics and may include electromagnetic shielding. In some embodiments, the housing115includes external contacts on a surface thereof forming 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 particular contacts constituting the interface however depend generally on the protocol with which data is communicated between the microphone assembly100and the host device. Such protocols include but are not limited to PDM, SoundWire, I2S, and I2C among other known and future protocols.

The processing circuit110(also referred to herein as an electrical circuit, an audio signal processing circuit, or audio signal electrical circuit) is configured to receive the acoustic signal from the MEMS sensor105. The MEMS sensor105may be operationally connected to the processing circuit110using one or more bond wires130. In other embodiments, other connecting mechanisms such as, vias, traces, electrical connectors, etc. may be used to electronically connect the MEMS sensor105to the processing circuit110. The processing circuit110receives the acoustic signal from the MEMS sensor105for further processing and after processing the acoustic signal, the processing circuit provides the processed acoustic signal to another computing or host device (e.g., a smartphone) for use.

Only those components of the microphone assembly100that are necessary for a proper understanding of the present disclosure are disclosed herein. Several other components, such as, motors, charge pumps, power sources, filters, resistors, etc., that are desirable or necessary for performing the functions described herein, are contemplated and considered within the scope of the present disclosure.

Additionally, several variations are contemplated and considered within the scope of the present disclosure. For example, although the processing circuit110and the MEMS sensor105are shown as separate components, in some embodiments, the processing circuit and the MEMS sensor may be integrated together into a single component. In some embodiments, either or both the MEMS sensor105and the processing circuit110may be constructed from a semiconductor die using, for example, mixed-signal complementary metal-oxide semiconductor devices. In other embodiments, other techniques may be used to construct the MEMS sensor105and the processing circuit110. In some embodiments, the processing circuit110may be configured as an application specific integrated circuit (ASIC).

InFIG. 2, a simplified electrical schematic of a processing circuit200includes an amplifier205having a first input node210and a second input node215. The first input node210receives a microphone signal from a combination block220, which combines an analog acoustic signal225of a transducer230(also referred to herein as an acoustic transducer) and an output235of an electrical current digital to analog converter (IDAC)240. The second input node215of the amplifier205is connected to virtual ground. The processing circuit200is shown as a single ended system (e.g., a single output from the transducer230and a single output from the IDAC240) for simplicity. In other embodiments, the processing circuit200may be configured as a differential system.

In a differential system, instead of connecting the second input node215to virtual ground, the amplifier205may be used to support a differential signal from the transducer230. In a differential system, the transducer230is configured to generate two outputs. The second output from the transducer230may be an inverted output compared to the acoustic signal225. Likewise, the IDAC240is configured to generate two outputs in a differential system, and the second output may be an inverted signal compared to the output235of the IDAC. In a differential system, the second output from the transducer230may be combined with the second output from the IDAC240in a combination block similar to the combination block220, and the output of the combination block may be provided to the second input node215. Thus, in a differential system, the second input node215may be connected to a negative differential output (not shown) of the transducer230through a negative node (not shown), which also includes a negative output (not shown) from the IDAC240.

In some embodiments, the combination block220is a summation block that sums up the acoustic signal225from the transducer230with the output235from the IDAC240. In some embodiments, instead of using the combination block220, the output235from the IDAC240may be directly connected to the acoustic signal225from the transducer230. As used herein, “directly connected” means through an electrically conductive path without any intervening active devices like transistors, but possibly through passive components like resistors, capacitors, electrical traces, wires, etc. In other embodiments, other mechanisms of combining the acoustic signal225from the transducer230with the output235from the IDAC240may be used. Although the combination block220has been described as a summation block, in other embodiments, other mechanisms of combining the acoustic signal225from the transducer230with the output235from the IDAC240may be used. In some embodiments, the acoustic signal225and the output235may be subtracted from one another instead of being added. Other functions to combine the acoustic signal225with the output235may be used as well.

By combining the acoustic signal225from the transducer230with the output235from the IDAC240, the low frequencies that may otherwise be input into the amplifier may be filtered. Such filtering prevents the amplifier205from being overloaded with low frequency components (e.g., noise components) of the acoustic signal225. By reducing overload at the amplifier205, the amplifier is able to receive and process the full dynamic range of the microphone signal without unacceptable distortion. In some embodiments, these low frequencies in the acoustic signal225from the transducer230are suppressed by using the output235from the IDAC240. The output235supplies an anti-phase low frequency component that cancels or suppresses the low frequency components of the acoustic signal225, such that the microphone signal that is input into the first input node210is substantially devoid of the low frequency components of the acoustic signal225.

The amplifier205receives the microphone signal at the first input node210without (or substantially without) the low frequency components and amplifies the microphone signal into an amplified microphone signal245, which is then input into an analog-to-digital converter (ADC)250. The amplifier205may be configured with a specified gain. The “gain” of the amplifier205means the amplifying ability of the amplifier that, in some embodiments, is expressed as a ratio of the output of the amplifier (e.g., the amplified microphone signal245) to the input of the amplifier (e.g., the microphone signal at the first input node210). In some embodiments, the gain of the amplifier205is adjusted to attain a more precise cut-off frequency of the microphone assembly100.

Variations are contemplated in the amplifier205. In some embodiments, the amplifier205is a fully-differential amplifier that generates the amplified microphone signal245in a differential or balanced format having a positive signal component255and a negative signal component260. In other embodiments, the amplifier205may be a standard amplifier that generates the amplified microphone signal245in a single-ended format. In some embodiments, the amplifier205may be an alternating current amplifier or a direct current amplifier. Generally speaking, the amplifier205may be any amplifier that is suitable for performing the functions described herein. Also, although only a single instance of the amplifier205is shown, in some embodiments, multiple instances of the amplifier connected in series or other topologies may be used. Likewise, in some embodiments, the amplifier205may use multiple gain stages, filters, or other components that may be deemed necessary or desirable in obtaining the amplified microphone signal245to perform the functions described herein.

The amplified microphone signal245from the amplifier205is input into the ADC250. The ADC250is configured to receive, sample, and quantize the amplified microphone signal245and generate a corresponding digital microphone signal265, which is then input into a decimator270. Thus, the ADC250receives an analog signal (e.g., the amplified microphone signal245) and converts that analog signal into a digital signal (e.g., the digital microphone signal265).

The ADC250may also be configured in a variety of ways. In some embodiments, the ADC250is adapted to output the digital microphone signal in a multibit format. In other embodiments, the ADC250is configured to generate the digital microphone signal265in a single bit format. In some embodiments, the ADC250is based on a sigma-delta converter (IA), while in other embodiments, the ADC is based on any other type of a converter, such as a flash ADC, a data-encoded ADC, a Wilkinson ADC, a pipeline ADC, etc. The ADC250may be also configured to generate the digital microphone signal265at a specific sampling frequency or sampling rate. In some embodiments, the sampling frequency of the digital microphone signal265may lie between 2 MHz and 20 MHz, such as 3.072 MHz. In other embodiments, the sampling frequency may vary.

The ADC250may input the digital microphone signal265into the decimator270. The decimator270down-samples the digital microphone signal265to reduce the size of data (also referred to as data rate) of the digital microphone signal. In some embodiments, the decimator270may down-sample the digital microphone signal265by reducing the sampling frequency of the digital microphone signal. For example, the decimator270may down-sample the digital microphone signal265from a 3.072 MHz frequency to about 384 KHz. In other embodiments, the decimator270may down-sample to other frequencies depending upon the sampling frequency of the digital microphone signal265and the down-sampled frequency that is desired. After down-sampling, the decimator270outputs a down-sampled microphone signal275.

Although not shown, in some embodiments, the down-sampled microphone signal275may be transmitted as input to other components (e.g., an interpolator, a digital-to-digital converter, etc.) for further processing before being input into a host processor of a receiving device (e.g., smartphone) for use. The down-sampled microphone signal275may also be provided to a feedback loop280. The feedback loop280provides the down-sampled microphone signal275back to the first input node210of the amplifier205via at least a digital loop filter285, the IDAC240, and the combination block220.

The down-sampled microphone signal275may be input into the digital loop filter285of the feedback loop280. The digital loop filter285filters the down-sampled microphone signal275in accordance with an adjustable or fixed transfer function to generate a digital feedback signal290, which is then input into the IDAC240. The IDAC240may be used to either source current to or sink current from a capacitive element (not shown) connected to the output235of the IDAC. The IDAC240may convert the digital feedback signal290, which is digital in nature, to a signal that is analog in nature. The IDAC240may also be configured in a variety of ways. In some embodiments, the IDAC240may be composed of a plurality of individually controllable current generators that may be selectively configured to source or sink current. Other configurations of the IDAC240are also contemplated and considered within the scope of the present disclosure.

The output235of the IDAC240is fed back into the combination block220. Thus, the feedback loop280supplies an analog feedback signal (e.g., the output235) back to the combination block220, where it is combined with the acoustic signal225from the transducer230and input back into the first input node210of the amplifier205.

Although only the digital loop filter285and the IDAC240have been shown in the feedback loop280, in some embodiments, additional or different components may be provided. In some embodiments, the feedback loop280may include pulse-width and pulse-amplitude modulators, filters, other digital-to-analog converters, amplifiers, etc. Likewise, although only the amplifier205, the ADC250, and the decimator270have been shown in the processing circuit outside of the feedback loop280, other components such as, filters, modulators, etc. that may be deemed desirable or necessary to perform the functions described herein may be used. Additionally, although only single instances of the ADC250, the decimator270, the digital loop filter285, and the IDAC240are shown, in some embodiments, multiple instances of one or more of those components may be used in the processing circuit200. In some embodiments, one or more of the components described above may be integrated together as a single component.

Again, by using the feedback loop280and combining the output235from the feedback loop with the acoustic signal225from the transducer230in the combination block220before inputting the microphone signal into the first input node210of the amplifier205, the processing circuit200effectively prevents low-frequency overload at the amplifier205and/or the ADC250. The low frequency components in the acoustic signal225from the transducer230may be undesirable noise components that may be caused by, for example, exposure of the microphone assembly100to intense subsonic or low frequency sounds generated by wind noise, large machinery, etc. By suppressing these low frequency components in the acoustic signal225, the processing circuit200eliminates the vulnerability of the amplifier205to low frequency component induced overload and distortion caused by saturation and non-linearity of active amplification elements like transistors of the amplifier.

The undesirable low frequency components in the acoustic signal225from the transducer230may be expressed in terms of a cut-off frequency (also referred to as a roll-off frequency). In some embodiments, the cut-off frequency may be set such that all frequency components below the cut-off frequency are deemed as the undesirable low frequency components that may be suppressed by using the output235without compromising, and possibly improving, the sound quality of the resulting microphone signal. Thus, the noise floor of the microphone assembly100may be adjusted by varying the cut-off frequency response of the transducer230. Depending upon the application in which the microphone assembly100is used, the cut-off frequency of the microphone may vary.

Conventionally, the cut-off frequency of a microphone (e.g., the microphone assembly100) has been adjusted using an analog filter. When using an analog filter, the cut-off frequency is typically set during manufacture and once set, cannot be changed. Thus, the cut-off frequency is not easily programmable when an analog filter is used. In other conventional approaches, a digital filter in the microphone (e.g., the microphone assembly100) may be used. The digital filter is also typically programmed during manufacture to set a specific cut-off frequency and, may or may not, be programmable. As noted above, however, precise control of the cut-off frequency requires knowledge of the loop gain.

InFIG. 3, a processing circuit300includes a transducer305whose acoustic signal310is combined with an output315of an IDAC320in a combination block325before being input as a microphone signal into a first input node330of an amplifier335. The amplifier335amplifies the microphone signal to generate an amplified microphone signal340, which is then input into an ADC345. The ADC345converts the amplified microphone signal340, which is analog in nature, to a digital microphone signal350.

The digital microphone signal350from the ADC345is input into a decimator355for down-sampling. Down-sampled microphone signal360from the decimator355is then forwarded for additional processing and use (e.g., to a host processor of a receiving device). In contrast to the processing circuit200ofFIG. 2, which includes the feedback loop280, the feedback loop in the processing circuit300ofFIG. 3is broken or, in other words, open. Specifically, the down-sampled microphone signal360from the decimator355is not fed back into the combination block325via a digital loop filter (e.g., the digital loop filter285) and the IDAC320. Rather, in the processing circuit300, a wave generator365that generates a reference waveform or signal having a known amplitude and frequency, is connected to an input370of the IDAC320.

The reference signal from the wave generator365may be used to set the loop gain by calibrating the IDAC320and, particularly, by calibrating a source current or gain of the IDAC. By calibrating the source current or gain of the IDAC320, the cut-off frequency of the microphone assembly100may be set more precisely. The wave generator365may be any type of wave generator that is capable of generating sinusoidal or square wave forms.

The processing circuit300may be configured to include several variations. Similar to the processing circuit200, the configuration of the various components of the processing circuit300may vary from one embodiment to another. Other or additional components that may be needed or desired to perform the functions described herein may also be used in the processing circuit300.

InFIG. 4, a flowchart outlining operations of a process400for adjusting the cut-off frequency of a microphone signal (e.g., from the microphone assembly100) is shown. The cut-off frequency of the microphone signal may be adjusted by adjusting a loop gain of the processing circuit (e.g., the processing circuit200). The loop gain of the processing circuit may be adjusted, at least in part, by calibrating the IDAC (e.g., the IDAC240) or at least in part by calibrating the amplifier (e.g., the amplifier205). Further, by calibrating the amplifier the sensitivity of the microphone may be adjusted.

After starting at operation405, the feedback loop of the processing circuit is broken at operation410. InFIG. 2, the feedback loop280is broken by disconnecting the down-sampled microphone signal275from the decimator270and removing the digital loop filter285from the feedback loop. InFIG. 4, the amplifier is calibrated at operation415. In some embodiments, the amplifier may be calibrated before breaking the feedback loop. Thus, the order of the operation410and operation415may be switched or reversed. The amplifier is calibrated by adjusting a gain of the amplifier. In some embodiments, the cut-off frequency is also dependent upon the size of the input capacitator in the transducer and any parasitic capacitances of the MEMS sensor. The impact of the input and parasitic capacitances on the cut-off frequency may also be attenuated by adjusting the gain of the amplifier. The calibration of the amplifier is discussed inFIG. 5.

At operation420, the wave generator is inserted into the processing circuit. InFIG. 2, the wave generator is inserted by removing the digital loop filter285from the feedback loop280and connecting the wave generator at the input of the IDAC, as shown inFIG. 3. InFIG. 4, at operation425, the IDAC is calibrated. The IDAC is calibrated by adjusting a current of the IDAC, either by adjusting the value of a current source (analog adjustment) or by adjusting a gain of the input to the IDAC (digital adjustment). The calibration of the IDAC is discussed inFIG. 6. After calibrating the IDAC, the feedback loop is restored at operation430. InFIG. 2, the feedback loop280is restored by disconnecting the wave generator and connecting the digital loop filter285back into the feedback loop to receive the down-sampled microphone signal275from the decimator270and to output the digital feedback signal290into the IDAC240. Thus, the configuration of the processing circuit200is restored after calibrating the IDAC. The process400ofFIG. 4then ends at operation435.

Although the process400shows the calibration of the amplifier before the calibration of the IDAC, in some embodiments, the IDAC may be calibrated before the amplifier is calibrated. Advantageously, by calibrating the amplifier before calibrating the IDAC, the cut-off frequency that is desired to be set by calibrating the IDAC may be accurately (or substantially accurately) determined to also account for the various capacitance tolerances (e.g., the parasitic capacitance) that are accounted for during calibration of the amplifier. Further, in some embodiments, the calibration of the amplifier occurs using an external tone (e.g., signal), and the calibration of the IDAC to adjust the cut-off frequency occurs using an internal test tone (e.g., signal). Thus, the amplifier may be calibrated along any operation of the process400so long as the amplifier may be calibrated using an external tone, and the IDAC may be calibrated using an internal test tone. For example, in some embodiments, the amplifier may be calibrated between the operation420and the operation425, or as discussed above, before the operation410. In some embodiments, the calibrations of the amplifier and the IDAC may occur at the same (or substantially same) time using two separate frequency signals.

Additionally, in some embodiments, the calibrations of the amplifier and the IDAC, and controlling of the cut-off frequency of the microphone signal may be made either internally on chip or externally on a host processor.

Thus, adjusting the cut-off frequency of the microphone signal involves two types of calibrations: a first calibration of the amplifier and a second calibration of the IDAC. In other words, adjusting the cut-off frequency involves calibrating the processing circuit by calibrating the amplifier and calibrating the IDAC.

InFIG. 5, a flowchart outlining operations of a process500for calibrating the amplifier335is shown. The purpose of calibrating the amplifier335is to adjust a sensitivity of the microphone (e.g., the microphone assembly100) such that the level above the cut-off frequency is the same. The amplifier335is calibrated by adjusting the gain of the amplifier. In some embodiments, the gain of the amplifier335is adjusted to also adjust (e.g., reduce) the impact of the parasitic capacitance and other tolerance capacitances of the microphone assembly100to achieve a higher quality acoustic output (e.g., the down-sampled microphone signal275). When the amplifier335generates a differential output (e.g., the amplified microphone signal340), each of the negative and positive components (e.g., the positive signal component255and the negative signal component260of the amplified microphone signal245) of the differential output may have a parasitic capacitance associated therewith. In those cases, the gain of the amplifier335for each of the negative and positive components may need to be adjusted to account for the parasitic capacitance of those components.

Thus, after starting at operation505, an acoustic signal is generated at operation510. The acoustic signal is the acoustic signal310from the transducer305. In some embodiments, the acoustic signal may be generated using a charge pump. In other embodiments, other mechanisms to generate the acoustic signal may be used.

The acoustic signal310is combined with the output315of the IDAC320in the combination block325to obtain a microphone signal. In those embodiments in which the feedback loop280is broken before the calibration of the amplifier335, the output315from the IDAC320may be zero or close to zero (e.g., because the wave generator365is not generating an input signal for the IDAC, which in turn is not generating the output315), such that the output (e.g., the microphone signal) of the combination block325is substantially equivalent to the acoustic signal310, which is input along the first input node330of the amplifier335. In those embodiments in which the feedback loop280is broken after the amplifier335is calibrated, the acoustic signal310may be combined with the output315from the IDAC320before being input as the microphone signal along the first input node330of the amplifier. In some embodiments, a ninety four decibel sound pressure level one kilohertz (94 dBSPL @ 1 kHz) signal may be used as the acoustic signal310. In other embodiments, an acoustic signal of a different intensity may be used.

The acoustic signal is amplified by the amplifier335at operation515to 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 into a digital microphone signal by the ADC345, and down-sampled by the decimator355. Before starting the calibration process of the amplifier335, in some embodiments, it is determined whether the processing circuit300is operating as intended or not.

The output (e.g., the down-sampled microphone signal360) of the decimator355may be measured to confirm that the processing circuit300is operating as intended. For example, in some embodiments, if there is a response (e.g., an audible response) measured from the output (e.g., the down-sampled microphone signal360) of the decimator355in response to the microphone signal, the processing circuit300is operating properly. If no response is measured from the output (e.g., the down-sampled microphone signal360) of the decimator355, then the processing circuit300may be malfunctioning and may need to be repaired before the amplifier335may be calibrated.

If a response is measured from the output (e.g., the down-sampled microphone signal360) of the decimator355, then at operation520, the process of calibrating the amplifier335starts. In some embodiments, the output of the ADC345may be used for determining whether the processing circuit300is operating as intended or not, as well as for making the various measurements described below for calibrating the amplifier. The amplifier335may be configured to generate a differential signal having a positive signal component and a negative signal component. The gain of the amplifier335for each of the negative signal component and the positive signal component is adjusted to calibrate the amplifier.

Thus, at the operation520, the negative signal component of the amplified microphone signal340is muted. In some embodiments, the negative signal component may be muted by connecting that component to a virtual ground. In other embodiments, other mechanisms to mute the negative signal component may be used such that the value of the negative signal component is essentially zero. By muting the negative signal component of the amplified microphone signal340, only the positive signal component of the amplified microphone signal is input into the ADC345. The ADC345converts the positive signal component of the amplified microphone signal340into a digital signal, which is then down-sampled by the decimator355to produce the down-sampled microphone signal360.

The down-sampled microphone signal360of the decimator355is measured at operation525. In some embodiments, the output of the ADC345is measured at the operation525. In some embodiments, the down-sampled microphone signal360or the output of the ADC345may be measured by measuring the level (e.g., a root mean square value) around a test tone either with filtering or using a fast Fourier transform value. In other embodiments, other mechanisms of measuring the down-sampled microphone signal360or the output of the ADC345may be used. By measuring the down-sampled microphone signal360of the decimator355(or the output of the ADC345), a sensitivity of the microphone assembly100may be calculated.

The sensitivity of the microphone assembly100is a function of the acoustic signal310, the down-sampled microphone signal360of the decimator355(or the output signal of the ADC345), and the gain of the amplifier335. By measuring the down-sampled microphone signal360of the decimator355(or the output of the ADC345) for the acoustic signal310of the operation510, and knowing the gain of the amplifier335, the sensitivity of the microphone assembly100may be calculated. If the calculated sensitivity is not within a desired range, the gain of the amplifier335may be adjusted until the desired sensitivity of the microphone assembly100is achieved.

Thus, calibration of the amplifier335results in an adjustment of the sensitivity of the microphone assembly100. By adjusting the sensitivity of the microphone assembly100, the noise levels to which the microphone assembly100may be sensitive may be adjusted. The sensitivity of the microphone assembly100may be adjusted such that the microphone assembly may ignore frequency components below the cut-off frequency of the microphone assembly. Depending upon the application in which the microphone assembly100is being used, a high or a low sensitivity may be desired. Generally speaking, a microphone assembly with a higher sensitivity produces a higher output voltage at the down-sampled microphone signal275and, therefore, requires a smaller gain at the amplifier335. On the other hand, a microphone assembly with a lower sensitivity produces a lower output voltage at the down-sampled microphone signal275and, therefore, requires, a bigger gain at the amplifier335. Accordingly, the sensitivity of the microphone assembly100may be adjusted to achieve the cut-off frequency that is desired.

Thus, at operation530, the calculated sensitivity of the microphone assembly100is compared with a pre-determined or reference value, which corresponds to the desired sensitivity of the microphone assembly. If the calculated sensitivity does not correspond to (e.g., match or fall within a range of) the desired sensitivity, the gain of the amplifier335is adjusted until the desired sensitivity value of the microphone assembly100is achieved. In some embodiments, the gain of the amplifier335may be changed in increments of one quarter of a decibel (0.25 dB). In other embodiments, the gain of the amplifier335may be changed in other increments. In each iteration of changing the gain of the amplifier335, the down-sampled microphone signal360is measured and the sensitivity of the microphone assembly100calculated, until the desired sensitivity of the microphone assembly is achieved.

In some embodiments, instead of incrementally changing the gain of the amplifier335until the desired sensitivity is achieved, a look-up table may be used to determine the gain of the amplifier corresponding to a desired level of the sensitivity of the microphone assembly100. By using a look-up table, instead of having to incrementally increase or decrease the gain of the amplifier335, and calculate the sensitivity value of the microphone assembly100, 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 in the look-up table, the look-up 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 achieved, thereby minimizing the number of iterations. Thus, the desired sensitivity of the microphone assembly100may be achieved faster by using a look-up table.

Once the gain of the amplifier335is adjusted for the positive signal component of the amplified microphone signal340, the negative signal component that was muted at the operation520is unmuted at operation535. In some embodiments, the negative signal component may be unmuted by removing the virtual ground connection or reversing the mechanism that was used to mute the component.

At operation540, the positive signal component of the amplified microphone signal340is muted. Again, the positive signal component may be muted by connecting that component to virtual ground or by using another mechanism. At this point, the negative signal component of the amplified microphone signal340is unmuted and the positive signal component is muted. The process of adjusting the gain of the amplifier335is now repeated for the negative signal component at operation545and operation550until the calculated sensitivity of the microphone assembly100corresponds to the pre-determined or reference value, which reflects the desired sensitivity of the microphone assembly.

In some embodiments, the process of adjusting the gain of the amplifier335for the negative signal component may be started by setting the gain of the amplifier to a gain value that was determined at the operation530for the positive signal component. In other embodiments, a different gain value may be used as a starting point. Again, a look-up table may be used in some embodiments. Also, in some embodiments, the gain value of the amplifier335may be the same for both 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 the amplifier335is continuously adjusted until the desired microphone sensitivity is achieved.

After achieving the desired sensitivity of the microphone assembly100on the negative signal component of the output of the amplifier335, at operation555, the positive signal component that was muted at the operation540is unmuted. With both the negative and positive signal components unmuted, the down-sampled microphone signal360(or the output of the ADC345) is again measured at operation560. If the down-sampled microphone signal360(or the output of the ADC345) at the operation560corresponds with the pre-determined value (that reflects the desired sensitivity of the microphone assembly100) at operation565, the calibration of the amplifier335is complete at operation570and the process500ends at operation575. If, at the operation565, the sensitivity of the microphone assembly100is not as desired, the process500returns to the operation520and the gain of the amplifier335is again set for each of the positive and negative signal components of the amplified microphone signal340, until the desired sensitivity is attained.

Various modifications to the process500are contemplated and considered within the scope of the present disclosure. For example, although the process500describes muting the negative signal component of the amplified microphone signal340and adjusting the gain of the amplifier335for the positive signal component first, in other embodiments, the positive signal component may be muted to adjust the gain of the amplifier for the negative signal component first. Additionally, in those embodiments in which the amplifier335generates a single-ended amplified microphone signal, the gain of the amplifier may be adjusted without having to mute/unmute signal components.

Although the process500has been described as using a look-up table with values of gain of the amplifier335corresponding to various sensitivity values, in some embodiments, other or additional look-up tables may be used. For example, in some embodiments, a look-up table with values of the down-sampled microphone signal360(or values of the output of the ADC345) corresponding to the sensitivity of the microphone assembly100may be used, such that instead of calculating the sensitivity of the microphone each time after measuring the down-sampled microphone signal (or the output of the ADC345), 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 pre-determined value at the operation530) may be different from the desired sensitivity (e.g., the pre-determined value at the operation550) of the positive signal component, which in turn may be different from the overall sensitivity (e.g., the pre-determined value at the operation565) that is desired. In some embodiments, the pre-determined values at the operation530and the operation550may be set such that the pre-determined value at the operation565is achieved.

In some embodiments, it may be desirable to wait for some time (e.g. a few fractions of a second) in between various operations of the process500. For example, in some embodiments, it may be desirable to wait for about twenty milliseconds (20 msec) between muting the negative or the positive signal components and measuring the down-sampled microphone signal360at the output of the decimator355. By waiting between two operations, any trace signals from the previous iteration may be dissipated, preventing erroneous measurements of the down-sampled microphone signal360. Other variations to perform the operations of the process500may be made in other embodiments.

In some embodiments, the control of the calibration process of the amplifier335may be external where the control and measurements are made externally to calibrate the amplifier, or may be internal on startup or when a host processor requests calibration of the microphone assembly100. A benefit of the process500is that a differential MEMS element (e.g., the microphone assembly100) with possible different gains in each of the positive and negative sides may be calibrated using the process discussed above.

InFIG. 6, a flowchart outlining operations of a process600may be used to calibrate the IDAC320. After starting at operation605, the acoustic signal310is disabled at operation610. In some embodiments, the acoustic signal310may be disabled by disabling the charge pump of the microphone assembly100. In other embodiments, other mechanisms of disabling the acoustic signal310may be used. It is desirable to perform the process600in a quiet environment to avoid interferences from any stray sound or noise signals that may impact the calibration of the IDAC320. In some embodiments, the calibration of the IDAC320may be performed in a sound box (the charge pump may not need to be disabled when calibrating within a sound box). In other embodiments, other mechanisms to block undesirable noise and sound may be used before calibrating the IDAC320.

Before calibrating the IDAC320, the feedback loop280is broken and the wave generator365is inserted into the broken feedback loop. Specifically, the wave generator365is connected to the input370of the IDAC320. At operation615, the wave generator365generates a sinusoidal signal to be input into the IDAC320via the input370. In some embodiments, the wave generator365may be used to generate a one kilohertz (1 kHz) sinusoidal wave with no attenuation. In other embodiments, signals of other intensities may be generated by the wave generator365. Furthermore, although the wave generator365has been described as generating a sinusoidal signal, in other embodiments, the wave generator may generate a square signal that may be used to calibrate the IDAC320.

The signal generated by the wave generator365is input into the IDAC320via the input370. At operation620, the output (e.g., the down-sampled microphone signal360) of the decimator355or the output of the ADC345is measured. Again, the output of the decimator355or the output of the ADC345may be measured by measuring the root mean square (RMS) value around the test tone either by filtering or using the fast Fourier transform measurement. From the measured value of the down-sampled microphone signal360(or the value of the output of the ADC345), it is determined whether a source current of the IDAC320or a gain of an input to the IDAC needs to be adjusted to attain the desired cut-off frequency of the microphone assembly100. Generally, either the source current or the gain of the input to the IDAC320is used to calibrate the IDAC.

At operation625and operation630, the value of the down-sampled microphone signal360(or the value of the output of the ADC345) is compared with a pre-determined or reference value. If the value of the down-sampled microphone signal360(or the value of the output of the ADC345) is outside a specific percentage of the pre-determined value, then the source current or gain of the input to the IDAC320is adjusted again. In some embodiments, a percentage of between, and including, thirty and forty six percent (30-46%) may be used for comparing the down-sampled microphone signal360(or the value of the output of the ADC345) with the pre-determined value. In other embodiments, a percentage of between, and including, one and five percent (1-5%) may be used. In yet other embodiments, different percentage ranges may be used for comparing the down-sampled microphone signal360(or the value of the output of the ADC345) with the pre-determined value.

The source current of the IDAC320or the gain of the input to the IDAC is incrementally adjusted until the value of the down-sampled microphone signal360(or the value of the output of the ADC345) corresponds with (e.g., matches with or falls within the specified range of) the pre-determined value. Thus, the IDAC320may be calibrated by adjusting a current of the IDAC by adjusting either a value of the source current of the IDAC (analog adjustment) or by adjusting the gain of the input to the IDAC (digital adjustment).

The pre-determined value is a value of the down-sampled microphone signal360(or the value of the output of the ADC345) that corresponds to the desired cut-off frequency of the microphone assembly100. By adjusting the source current of the IDAC320or the gain of the input to the IDAC until the down-sampled microphone signal360(or the value of the output of the ADC345) corresponds to the pre-determined value, the desired cut-off frequency may be achieved. In some embodiments, the pre-determined value may be determined from a look-up table, such that for each desired cut-off frequency, the look-up table may include a corresponding pre-determined value for the down-sampled microphone signal360. In other embodiments, other mechanisms to determine the pre-determined value may be used.

In some embodiments, a specific range of the pre-determined value may be defined, such that if the value of the down-sampled microphone signal360(or the value of the output of the ADC345) falls within that range, the IDAC320is deemed calibrated. In some embodiments, a range of thirty to forty-six percent (30-46%) or a range of one to five percent (1-5%) of the pre-determined value may be used, while in other embodiments, other range values may be used. In other embodiments, no ranges may be used and it may be desirable for the value of the down-sampled microphone signal360(or the value of the output of the ADC345) to correspond very closely to the pre-determined value. Thus, the range of acceptability of the down-sampled microphone signal360(or the value of the output of the ADC345) may vary from one embodiment to another, depending upon the desired cut-off frequency and the pre-determined value that the down-sampled microphone signal is compared against.

Thus, at the operation630, the value of the down-sampled microphone signal360(or the value of the output of the ADC345) is compared with the pre-determined value. If the value of the down-sampled microphone signal360(or the value of the output of the ADC345) is within the acceptable range of the pre-determined value, then the process600proceeds to operation635, where the calibration of the IDAC320is deemed complete. If at the operation630, the value of the down-sampled microphone signal360(or the value of the output of the ADC345) is not within the specified range of the pre-determined value, the process600returns to the operation625, where the source current of the IDAC320or the gain of the input to the IDAC is adjusted and the value of the down-sampled microphone signal360(or the value of the output of the ADC345) is measured again. The process of adjusting the source current or the gain of the IDAC320and measuring the value of the down-sampled microphone signal360(or the value of the output of the ADC345) is continued until the value of the down-sampled microphone signal (or the value of the output of the ADC) corresponds to the pre-determined value.

Similar to the calibration of the amplifier335, in some embodiments, a look-up table may be used to calibrate the IDAC320. The look-up table may be used to determine the value of the source current or the gain of the IDAC320. In some embodiments, the look-up table may include cut-off frequencies and corresponding source current or gain values for the IDAC320. By knowing the cut-off frequency, the corresponding source current or gain value of the IDAC320, or the closest source current or gain value of the IDAC corresponding to the desired cut-off frequency may be determined from the look-up table, making the calibration of the IDAC go faster.

After calibrating the IDAC320at the operation635, the charge pump or other mechanism that supplies the acoustic signal310is enabled (if it was disabled), at operation640. The process600ends at operation645.

By calibrating the amplifier335and the IDAC320, the loop gain of the processing circuit200is adjusted to set a desired cut-off frequency of the microphone assembly100. By setting the cut-off frequency of the microphone assembly100, frequencies below the cut-off frequency may be filtered out by the processing circuit200(e.g., at the combination block220), thereby filtering out at least some of the noise in the acoustic signal225from the transducer230and improving the quality of the down-sampled microphone signal275. In some embodiments, the calibration of the amplifier335and the IDAC320may be performed at start up (e.g., when the microphone assembly100is assembled or factory calibrated), during manufacture of the microphone, or even periodically after assembling when the microphone is not being used by the application in which the microphone is incorporated.

FIG. 7Ais a graph700illustrating impact on beamforming using two microphones with varying cut-off frequencies. The graph700shows a signal705of a first microphone and a signal710of a second microphone, with each microphone having a varying cut-off frequency. The signal705of the first microphone has a cut-off frequency of about thirty hertz (30 Hz) and the signal710of the second microphone has a cut-off frequency of about thirty-five hertz (35 Hz). Even though there is only a five hertz (5 Hz) difference between the cut-off frequencies of the signal705and the signal710, the small difference in the cut-off frequencies may introduce a phase difference715between the signals when the signals are combined together for beamforming.

Difference in the phases of the signal705and the signal710may lead to sound distortion and noise in the resulting beam (e.g., the composite beam when the beams from the microphones are combined) that may not be acceptable in certain applications, such as in hearing aids. The phase difference715between the signal705and the signal710may be removed or at least minimized by matching the cut-off frequencies of those signals.

InFIG. 7B, a polar plot720shows how certain frequencies are reproduced when they enter a microphone from different angles. The polar plot720plots a first composite beam725against a second composite beam730. The first composite beam725is formed from at least two microphone signals having different cut-off frequencies, while the second composite beam730is formed from at least two microphone signals having same (or substantially same) cut-off frequencies. The plot of the first composite beam725appears on the polar plot720as an omni-directional response that picks up sound, including noise, from all directions. The plot of the second composite beam730appears on the polar plot720as a cardioid response, meaning that the composite beam picks up sounds from a front direction and some sound from a rear direction, while filtering out sounds from other directions. Thus, for a same acoustic signal of about two hundred hertz (200 Hz), the first composite beam725and the second composite beam730generate different responses.

In some applications, such as hearing aids, a superior sound quality with minimal noise distortions may be preferred. In those cases, a cardioid response from a composite beam may be preferred to allow certain sounds while excluding certain sounds. The cardioid response may be obtained by matching the cut-off frequencies of the beams that combine together to make up the composite beam. By matching the cut-off frequencies of signals from two or more microphones to generate a composite beam, a superior sound quality may be achieved.