Abstract: Methods and systems to produce audio output signals from audio input signals. In one embodiment, a first portion of the audio input signals can be pre-processed, with the output used to modulate ultrasonic carrier signals, thereby producing modulated ultrasonic signals. The modulated ultrasonic signals can be transformed into a first portion of the audio output signals, which is directional. Based on a second portion of the audio input signals, a standard audio speaker can output a second portion of the audio output signals. Another embodiment further produces distortion compensated signals based on the pre-processed signals. The distortion compensated signals can be subtracted from the second portion of the audio input signals to generate inputs for the standard audio speaker to output the second portion of the audio output signals. In yet another embodiment, noise can be added during pre-processing of the first portion of the audio input signals.

Abstract: The present invention provides methods and systems for narrow bandwidth digital processing of an input audio signal. Particularly, the present invention includes a high pass filter configured to filter the input audio signal. A first compressor then modulates the filtered signal in order to create a partially processed signal. In some embodiments, a clipping module further limits the gain of the partially processed signal. A splitter is configured to split the partially processed signal into a first signal and a second signal. A low pass filter is configured to filter the first signal. A pass through module is configured to adjust the gain of the second signal. A mixer then combines the filtered first signal and the gain-adjusted second signal in order to output a combined signal. In some embodiments, a tone control module further processes the combined signal, and a second compressor further modulates the processed signal.

Abstract: An audio signal measurement method for a speaker and an electronic apparatus having the speaker are provided. The electronic apparatus further has a processing circuit and a power amplifier. The processing circuit is coupled to the speaker and configured to execute a time domain to frequency domain transform according to a voltage value of an audio signal and a current value of current feedback from the speaker so as to obtain a frequency response curve. The power amplifier is coupled to the speaker and configured to drive the speaker according the voltage value of the audio signal. The processing circuit is capable of determining whether the frequency response curve is located within a predetermined area such that the processing circuit generates a signal when the frequency response curve is located out of the predetermined area. Thereby, the electronic apparatus may measure its transducer distortion and acoustic box leakage.

Abstract: A sound correcting apparatus includes an air-conduction microphone, a bone-conduction microphone, a processor, and a memory. The air-conduction microphone picks up an air conduction sound using aerial vibrations. The bone-conduction microphone picks up a bone conduction sound using bone vibrations of a user. The processor calculates a ratio of a voice of the user for the air conduction sound to a noise. The memory stores a correction coefficient for making a frequency spectrum of the bone conduction sound identical with a frequency spectrum of the air conduction sound which corresponds to the ratio that is equal to or greater than a first threshold. The processor corrects the bone conduction sound using the correction coefficient, and generates an output signal from the corrected bone conduction sound when the ratio is less than a second threshold.

Abstract: A collar/headset microphone cable control device is provided with a sound receiving part and a radio frequency emitting part. The sound receiving part includes a microphone sound receiving end for picking up the sound signal inputs, and a microphone cable control end electrically connected to the microphone sound receiving end for controlling the sound signal transmission state. The radio frequency emitting part includes a substrate, a microphone signal receiving element, a built-in sound receiving end, an external sound signal receiving element, a signal emitting element, buttons, indicator lights, a display element and a bus element that all are provided on the substrate and are electrically interconnected. The microphone signal receiving element picks up sound signals from the microphone cable control end. The built-in sound receiving end receives additional sound signals. A user can easily control the sound volume level and muteness on the microphone sound source cable.

Abstract: A system for processing signals to enhance patient audibility of a plurality of signals in an MRI environment is provided. The system includes an acoustic measuring device for measuring sound power levels generated by the MRI and a principal frequency component identifier for identifying principal frequencies measured by the acoustic measuring device. The system also includes an audio equalizer for controlling the amplitude and frequency of each of the plurality of signals in accordance with the principal frequencies. Further provided by the system is an attenuator for attenuating an overall sound level of the signals being processed and a dynamic range compression processor.

Abstract: A method for convolving an input signal with an impulse response function, the impulse response function being partitioned into a plurality of time segments of equal size, the method including transforming a segment of an input signal into the frequency domain to generate a frequency spectrum of the segment of the input signal; multiplying the frequency spectrum of the segment of the input signal with a frequency spectrum of each of the segments of the impulse response function; scaling the results from the multiplication of frequency spectra; accumulating the scaled results; and performing an inverse transform on the accumulated signals to generate a desired convolved signal in the time domain. The scaling includes performing a bitwise shift operation on the multiplication results, and performing the bitwise shift operation includes adding a bit to the multiplication results before the bitwise shift operation.

Abstract: An acoustic apparatus includes a high frequency driver that has a first front volume and a low frequency driver that has a second front volume. The first front volume and the second front volume communicate with each other to form a common front volume. At least one acoustic resistance is placed between the first front volume and the second front volume. The acoustic resistance acts as a low pass filter.

Abstract: A communication system is provided. A receiver receives a plurality of audio signals, wherein a frequency of each of the audio signals is selected from a frequency group formed by at least three frequencies. A signal detector coupled to the receiver is configured to obtain the frequency of each of the audio signals. A processor coupled to the signal detector is configured to convert the frequency of each of the audio signals into a digital signal having a first logic level or a second logic level. Two adjacent audio signals of the audio signals have different frequencies, and at least one frequency of the frequency group is used to dynamically represent the first logic level or the second logic level.

Abstract: A canalphone system may include a canalphone housing, and a first high frequency driver carried within the canalphone housing. The system may also include a second high frequency driver carried within the canalphone housing where the second high frequency driver is tuned with the first high frequency driver to deliver lower distortion than a standard canalphone high frequency driver and/or lower distortion than two standard canalphone high frequency drivers that are not tuned with each other.

Abstract: A sound processing apparatus includes a sound input unit configured to input a sound signal, a signal processor having a band-limiting filter for limiting a passing band of the sound signal, and being configured to apply a predetermined signal processing on a sound signal output from the band-limiting filter, and a controller configured to control a band width of the passing band of the band-limiting filter. The controller changes the band width of the passing band according to a level of the sound signal, and makes a speed at the time of expanding the band width faster than a speed at the time of narrowing the band width.

Abstract: Provided are a method for estimating a spectrum density of diffused noises. Also provided is a processor for implementing the method. The processor includes at least two sound receiving units and a spectrum density estimating unit for estimating spectrum density.

Abstract: A system is provided that produces haptic effects. The system receives an audio signal that includes a low-frequency effects audio signal. The system further extracts the low-frequency effects audio signal from the audio signal. The system further converts the low-frequency effects audio signal into a haptic signal by shifting frequencies of the low-frequency effects audio signal to frequencies within a target frequency range of a haptic output device. The system further sends the haptic signal to the haptic output device, where the haptic signal causes the haptic output device to output one or more haptic effects.

Abstract: In a method of reproducing audio signals using an electronic device, the electronic device decodes an audio file to generate original digital audio signals, and attenuates the original digital audio signals having frequencies that are lower than a preset cutoff frequency. The electronic device further amplifies each of attenuated digital audio signals and the original digital audio signals having frequencies that are higher than the preset cutoff frequency with a preset gain. Once the electronic device converts the amplified digital audio signals into analog audio signals, and outputs the analog audio signals to an audio amplifier, the audio amplifier amplifies the analog audio signals to drive a speaker to output the analog audio signals.

Abstract: Psychoacoustic Bass Enhancement (PBE) is integrated with one or more other audio processing techniques, such as active noise cancellation (ANC), and/or receive voice enhancement (RVE), leveraging each technique to achieve improved audio output. This approach can be advantageous for improving the performance of headset speakers, which often lack adequate low-frequency response to effectively support ANC.

Abstract: The application describes a method and an apparatus to prevent clipping of an audio signal when protection against signal clipping by received audio metadata is not guaranteed. The method may be used to prevent clipping for the case of downmixing a multichannel signal to a stereo audio signal. According to the method, it is determined whether first gain values (4) based on received audio metadata are sufficient for protection against clipping of the audio signal. The audio metadata is embedded in a first audio stream (1). In case a first gain value (4) is not sufficient for protection, the respective first gain value (4) is replaced with a gain value sufficient for protection against clipping of the audio signal. Preferably, in case no metadata related to dynamic range control is present in the first audio stream (1), the method may add gain values sufficient for protection against signal clipping.

Abstract: An electrical outlet strip is controlled by sound or an audio signal. The electrical outlet strip can include a plurality of outlets having a selector switch for either each outlet or one selector switch that controls all the outlets. The selector switch can control whether the outlets are powered directly be the input power, or if the outlets are controlled by the audio signal or sound. When the outlets are controlled by the audio signal or sound, the audio signal or sound can be filtered into a plurality of bands, one for each outlet, for example. There audio signal or sound can be input to the outlet strip by various mechanisms, including WiFi, Bluetooth®, FM wireless receiver, via an audio jack, via a microphone, or the like.

Abstract: Techniques are described herein that are capable of spectrally shaping audio signal(s) for audio mixing. Spectrally shaping an audio signal means modifying a frequency spectrum of the audio signal. A frequency spectrum of an audio signal is a representation of the audio signal in the frequency domain. For instance, a frequency spectrum may be represented using multiple frequency bands. The frequency spectrum may be modified by modifying characteristic(s) (e.g., magnitude, phase, etc.) one or more of the frequency bands.

Abstract: The present invention relates to coding of audio signals, and in particular to high frequency reconstruction methods including a frequency domain harmonic transposer. A system and method for generating a high frequency component of a signal from a low frequency component of the signal is described.

Abstract: Microphones are provided at an air inlet of the engine and a vehicle-cabin-side wall surface of an engine room, and engine sounds are picked up. The engine sound is processed by a signal processing section, and the processed engine sound is output from a speaker provided in a vehicle cabin. The signal processing section is provided with a filter which simulates a sound insulation characteristic of the vehicle cabin and a transformation section for processing the engine sound according to driving condition. A spectrum transformation characteristic of the transformation section is determined according to values detected by a vehicle speed sensor, an engine speed sensor, and an accelerator depression sensor, and a spectrum of the engine sound is transformed by means of specification of the spectrum transformation characteristic, thereby enhancing an engine sound.

Abstract: The invention relates to a method for generating sound for a rotating machine, including a step (E1) of determining the frequencies and amplitudes of n partials and/or harmonics (i) pertaining to the sound of a rotating machine, characterized in that the method includes a step (E2) of determining values (a1) and a step (E7-E8) of calculating a synthetic sound for the rotating machine, said synthetic sound being composed from the n partials and/or harmonics (i), while the frequency thereof is entirely or partially shifted by the values (ai).

Abstract: Mixer and one or more external devices, related to audio signal processing, are interconnected via a communication network, such as a LAN. For a particular one of the external devices for which communication with the mixer via the communication network is to be established, a network address unique to the particular external device and a communication protocol to be used for data communication with the particular external device are designated by entry operation by a user. Logical connection is established such that data communication is performed, in accordance with the designated communication protocol, between the mixer and the particular external device corresponding to the designated network address. Thus, data communication is performed, via the communication network, between the mixer and the particular external device for which the logical connection has been established.

Abstract: Integrated processing of multimedia signals can eliminate unnecessary signal processors and converters without losing the functionality of typical home entertainment system components. The integrated multimedia system includes a main player that captures and processes signals digitally. The main player may adjust the audio signal to provide audio output of equal loudness across all frequencies by accounting for sensitivity of the human ear for sounds of varying frequencies. The main player can also account for perceived differences in loudness based on the angle of a listener to a speaker by detecting the position of a user and making an adjustment accordingly. The invention further provides a speaker that has embedded performance characteristics or an identifier that allows the system to provide an optimal speaker driving current for a particular system or determine how that speaker would be best implemented in the integrated system.

Abstract: Various methods and apparatus for processing audio signals are disclosed herein. The assembly may be attached, adhered, or otherwise embedded into or upon a removable oral appliance to form a hearing aid assembly. Such an oral appliance may be a custom-made device which can enhance and/or optimize received audio signals for vibrational conduction to the user. Received audio signals may be processed to cancel acoustic echo such that undesired sounds received by one or more intra-buccal and/or extra-buccal microphones are eliminated or mitigated. Additionally, a multiband actuation system may be used where two or more transducers each deliver sounds within certain frequencies. Also, the assembly may also utilize the sensation of directionality via the conducted vibrations to emulate directional perception of audio signals received by the user. Another feature may include the ability to vibrationally conduct ancillary audio signals to the user along with primary audio signals.

Abstract: According to one embodiment, a sound signal processor includes: a connector; an input module; and a generator. The connector is connectable with an earphone. The input module receives and processes a plurality of sound signals corresponding to sound of a plurality of times output from the earphone, respectively. The generator generates, by using first data indicating a frequency characteristic of a first sound signal among the received and processed sound signals and second data indicating a frequency characteristic of a second sound signal among the received and processed sound signals, correction data correcting a frequency characteristic of the earphone to be a target frequency characteristic set as a target. The first data is used for a first frequency band lower than or equal to a reference. The second data is used for a second frequency band higher than the reference.

Abstract: An acoustic chamber includes a loudspeaker, which includes at least two membranes that each reproduce a different frequency band, and a filter that makes it possible to generate a plurality of activation signals from an audio signal source. The activation signals are each applied to an actuator of one of the membranes. The acoustic chamber has an operating range having a variable and controlled directivity, each frequency of which belongs to at least two frequency bands reproduced by the membranes. The acoustic chamber obtains a directivity control signal, and the filter makes it possible to dose, for each frequency of the operating range and depending on the directivity control signal, the contribution of each one of the at least two membranes reproducing the frequency.

Abstract: An active noise control (ANC) system may be implemented to both sides of a fan, such that both directions of the noise emitting are treated to reduce the overall noise. The impact on airflow is minimal, and the technique is very effective in a broad range of low frequencies. Passive sound-absorbing materials may be included for attenuation of high frequencies. The resulting quiet fan produces a low level of noise compared to any other device based on fan, which produces the same capacity of airflow. The quiet fan may be incorporated in any mechanical system which requires airflow induction such as: computers, air conditioners, machines, and more.

Abstract: The initial values of parameter estimates are set, including reverberation parameter estimates, which includes a regression coefficient used in a linear convolutional operation for calculating an estimated value of reverberation included in an observed signal, source parameter estimates, which includes estimated values of a linear prediction coefficient and a prediction residual power that identify the power spectrum of a source signal, and noise parameter estimates, which include noise power spectrum estimates. Then, the maximum likelihood estimation is used to alternately repeat processing for updating at least one of the reverberation parameter estimates and the noise parameter estimates and processing for updating the source parameter estimates until a predetermined termination condition is satisfied.

Abstract: A feedback-controlled microphone includes a microphone body and a membrane operatively connected to the body. The membrane is configured to be initially deflected by acoustic pressure such that the initial deflection is characterized by a frequency response. The microphone also includes a sensor configured to detect the frequency response of the initial deflection and generate an output voltage indicative thereof. The microphone additionally includes a compensator in electric communication with the sensor and configured to establish a regulated voltage in response to the output voltage. Furthermore, the microphone includes an actuator in electric communication with the compensator, wherein the actuator is configured to secondarily deflect the membrane in opposition to the initial deflection such that the frequency response is adjusted. An acoustic beam forming microphone array including a plurality of the above feedback-controlled microphones is also disclosed.

Abstract: Sound processing processes and sound processing units are disclosed. The processes and units have particular application to auditory prostheses. In an embodiment, after input sound signals are processed into channels, an algorithm is applied to selectively increase the modulation depth of the envelope signals. In another embodiment, a sound processing unit includes a filter bank configured to process a sound signal to generate envelope signals in each of a plurality of spaced frequency channels. The speech processing unit also includes a broadband envelope detector configured to measure the envelope of at least one broadband signal, and a channel modulation module configured to use the at least one broadband envelope signal to modulate the channel envelope signals to generate modulated envelope signals.

Abstract: The invention relates to a method for adapting sound pressure levels in at least one listening location, the sound pressure being generated by a first and a second loudspeaker, each loudspeaker having a supply channel arranged upstream thereto, where at least the supply channel of the second loudspeaker modifies the phase of an audio signal transmitted therethrough according to a phase function. The method includes supplying an audio signal to the supply channels and thus generating an acoustic sound signal; measuring the acoustic sound signal at each listening location and providing corresponding electrical signals representing the measured acoustic sound signal; estimating updated transfer characteristics for each pair of loudspeaker and listening location; calculating an optimum offset phase function based on a mathematical model using the estimated transfer characteristics; updating the phase function by superposing the optimal offset phase function thereto.

Abstract: The disclosure includes a system and method for sonically customizing an audio reproduction device. The system includes a processor and a memory storing instructions that when executed cause the system to: determine an application environment associated with an audio reproduction device associated with a user; determine one or more sound profiles based on the application environment; provide the one or more sound profiles to the user; receive a selection of a first sound profile from the one or more sound profiles; and generate tuning data based on the first sound profile, the tuning data configured to sonically customize the audio reproduction device.

Abstract: A processor according to embodiments comprises an on-board sample rate converter for converting a source audio signal that is sampled at a first sampling rate to an output audio signal that is sampled at a second sampling rate. The sample rate converter utilizes a master clock signal in converting the audio signal. The sample rate converter selects the master clock signal from available reference clock signals, such as an on-chip system clock or a bus interface clock, and scales the frequency of the selected clock signal to generate the master clock signal with the frequency of the second sampling rate.

Abstract: A device and method for manipulating an audio signal includes a windower for generating a plurality of consecutive blocks of audio samples, the plurality of consecutive blocks including at least one padded block of audio samples, the padded block having padded values and audio signal values, a first converter for converting the padded block into a spectral representation having spectral values, a phase modifier for modifying phases of the spectral values to obtain a modified spectral representation and a second converter for converting the modified spectral representation into a modified time domain audio signal.

Abstract: An audio signal processing method and circuitry that processes an input audio signal by filtering the input audio signal with a high pass filter to produce a filtered audio signal, which is input to a compressor. A first intermediate audio signal is produced based on the compressor output signal. The filtered audio signal is also input to a harmonics generator that produces harmonics of the filtered audio signal. A second intermediate audio signal is produced based on such harmonics. A third intermediate signal is produced based upon the input audio signal. An output audio signal is produced by combining the first intermediate audio signal, the second intermediate audio signal and the third intermediate audio signal.

Abstract: The technology in one embodiment includes an audio-processing unit, for processing an input audio signal using a pre-established audio feature. The unit includes a processor, an audio-device input in operative communication with the processor, and a system-user interface being in operative communication with the processor. The unit also includes a computer-readable medium being in operative communication with the processor and comprising a tone-manipulation component and computer-executable instructions that, when executed by the processor, cause the processor to perform various operations.

Abstract: The present invention discloses a driving circuit for electronic horn and a driving method for the electronic horn. The driving circuit includes an oscillating circuit, which generates a signal having oscillating frequency. Based on the signal, said driving circuit generates a driving signal to drive the electronic horn to produce sound. It is characterized in that said oscillating circuit includes a variable capacitor, and the oscillating frequency is changed by adjusting capacitance of the variable capacitor so as for the frequency of the driving signal to be consistent with working frequency of the electronic horn. The present invention overcomes the problem of mismatch between the frequency of circuit's driving signal and the horn's sounding diaphragm and horn's tone inflexion due to resistance change caused by vehicle vibration.

Abstract: According to one embodiment, a monitoring apparatus includes an acquisition unit, an analysis unit, a calculation unit, a storage, a determination unit. The acquisition unit acquires an environmental sound. The analysis unit performs frequency analysis to extract characteristic frequency components. The calculation unit calculates first values of metrics. The storage stores contour data generated by second values of the metrics. The determination unit determines whether or not there is a first measurement point in which a first value and a second value match, if there is no first measurement point or the change is not less than the threshold value, determines that the machines is abnormal.

Abstract: An absolute-value circuit outputs the absolute value of a signal SIN? that corresponds to an input signal SIN. A clipping circuit clips the signal SIN? that corresponds to the input signal, to a positive limit value and to a negative limit value. A first multiplier multiplies the signal SIN? that corresponds to the input signal, by a predetermined coefficient. A first adder subtracts the output signal of the first multiplier from the output signal of the clipping circuit. A second adder sums a signal that corresponds to the output signal of the first adder and a signal that corresponds to the output signal of the absolute-value circuit. A third adder sums the input signal SIN and a signal that corresponds to the output signal of the second adder.

Abstract: The present invention relates to audio coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR). A system and a method for generating a high frequency component of a signal from a low frequency component of the signal is described. The system comprises an analysis filter bank providing a plurality of analysis subband signals of the low frequency component of the signal. It also comprises a non-linear processing unit to generate a synthesis subband signal with a synthesis frequency by modifying the phase of a first and a second of the plurality of analysis subband signals and by combining the phase-modified analysis subband signals. Finally, it comprises a synthesis filter bank for generating the high frequency component of the signal from the synthesis subband signal.

Abstract: Systems, methods, and apparatus to filter audio are disclosed. An example device includes first and second audio speakers having first audio characteristics, a third audio speaker having second audio characteristics, wherein the third speaker is positioned between the first and second audio speakers, a first audio filter to process an audio input signal to have a first frequency response including a first cutoff frequency, the first audio filter to output a first audio output signal to the first audio speaker, and a second audio filter to process the audio input signal to have a second frequency response to compensate for interference between the first and second frequency responses caused by a position of the first audio speaker relative to the second audio speaker.

Abstract: Devices and methods for digital sound leveling are disclosed. The subject devices and methods can be used to reduce the risk of noise induced hearing loss. According to an embodiment, an audio file can be parsed into frames that are filtered using an A-weighted digital filter and scaled to a desired power level. Accordingly, the dynamic range of an audio output at a particular volume level can be controlled.

Abstract: The present invention provides a signal filtering apparatus, which comprises a control circuit and a filter for receiving a transmitted input signal and generating an output signal. The filter comprises multiple filter taps for processing the transmitted input signal corresponding to different timings with different filter coefficients, respectively, to generate the output signal. The control circuit is configured to shrink at least part of the low-frequency-response filter taps having filter coefficients less than a predetermined value.

Abstract: An audio processing device comprises a multitude of electric input signals, each electric input signal being provided in a digitized form, and a control unit receiving said digitized electric input signals and providing a resulting enhanced signal. The control unit is configured to determine the resulting enhanced signal from said digitized electric input signals, or signals derived therefrom, according to a predefined scheme.

Abstract: An acoustic apparatus which outputs test signals from a plurality of speakers configuring a multichannel reproduction system and adjusts acoustic characteristics by picking up response signals from the speakers using a microphone, includes a test signal storage unit storing test signals output from the speakers; a response signal storage unit storing response signals from the speakers picked up by the microphone; a parameter calculation unit calculating acoustic adjustment parameters including at least a polarity reversal parameter and a phase filter parameter based on the response signals from the speakers stored in the response signal storage unit; and an acoustic adjustment parameter storage unit storing the acoustic adjustment parameters calculated by the parameter calculation unit. The parameter calculation unit calculates the polarity reversal parameter using a low-pass component of the response signals.

Abstract: The present invention relates to an audio equipment (1) and a signal processing method thereof, wherein the mismatches that can occur at the packet boundaries in an audio signal, which is processed in packets by utilizing the missing fundamental phenomenon and which is formed again by the concatenating of these packets, are eliminated.

Abstract: An audio processing system processes an audio signal that may come from one or more microphones. The audio processing system may use information from one or more non-acoustic sensors to improve a variety of system characteristics, including responsiveness and quality. Those audio processing systems that use spatial information, for example to separate multiple audio sources, are undesirably susceptible to changes in the relative position of any audio sources, the audio processing system itself, or any combination thereof. Using the non-acoustic sensor information may decrease this susceptibility advantageously in an audio processing system.

Abstract: Process for synthesis of a synthesized audio signal, in which at least one audio signal of contact is produced for each excitation contact of a sequence of contacts carried out on a vibrating body (2). A partial attenuation signal of remanence is generated from at least one vibration signal representative of the vibration of the vibrating body (2) generated by at least one excitation contact, named partial contact, the partial attenuation signal of remanence being representative, of a partial attenuation of at least one remanent audio signal of contact, resulting from an excitation contact that is previous to said partial contact. The synthesized audio signal after said partial contact is produced by mixing the audio signal of contact of said partial contact and each remanent audio signal of contact affected by the partial attenuation signal of remanence. The invention extends to a device (3) for synthesizing said synthesized audio signal.

Abstract: An electronic signal processor for processing signals includes a complex first filter, one or more gain stages and a second filter. The first filter is characterized by a frequency response curve that includes multiple corner frequencies, with some corner frequencies being user selectable. The first filter also has at least two user-preset gain levels which may be alternately selected by a switch. Lower frequency signals are processed by the first filter with at least 12 db/octave slope, and preferably with 18 db/octave slope to minimize intermodulation distortion products by subsequent amplification in the gain stages. A second filter provides further filtering and amplitude control. The signal processor is particularly suited for processing audio frequency signals.

Abstract: This document discusses, among other things, apparatus and methods including an analog-to-digital controller (ADC) configured to receive an enable signal and to provide an ADC output signal to control logic, wherein the control logic is configured to provide a control voltage to a control input of a switch. In an example, the control voltage includes the ADC output signal when the ADC output signal is below a first threshold or above a second threshold. In certain examples, the control logic is configured to transition the control voltage from the first threshold to the second threshold when the ADC output signal is between the first and second thresholds.