Abstract: Disclosed is an encoding device, wherein the energy information of a given layer is efficiently encoded using a scalable encoding method in which the band to be encoded is selected in each layer, and the quality of decoded signals can be enhanced. An encoding device (101) is equipped with: a second layer encoding unit (205) which generates a second layer encoded information included in which is the first band information of said band; a second layer decoding unit (206) which generates a first decoding signal by using the second layer encoded information; an adding unit (207) which generates a second input signal by using the first decoding signal; and a third layer encoding unit (208) which generates a third layer encoded information included in which is a second band information obtained by selecting a second band to be quantized in the second input signal, and a corrected gain (energy information).

Abstract: A telephone user's speech volume is monitored in relation to a minimum volume sufficient for a remote conversant, with whom the user communicates, to audibly perceive the user's speech. Upon detecting that the user's volume exceeds the level sufficient to communication, it is determined whether the user moderates the volume, without being prompted, to the sufficient level. Upon determining that the user moderates the speaking voice without being prompted, positive reinforcement is provided to the user. Results are recorded over multiple phone calls. A frequency of the user moderating the speaking volume without being prompted is tracked from the recorded results. Upon the frequency of the user moderating the volume without being prompted rising over the tracked phone calls, the quality and/or quantity of the positive reinforcement may be improved. The user is thus trained or treated, e.g., audiologically or therapeutically, to self-modulate the volume.

Abstract: A method for accurately estimating and improving the speech intelligibility from a loudspeaker (LS) is disclosed. A microphone is placed in a desired position and using an adaptive filter, an estimate of the clean speech signal at the microphone is generated. By using the adaptive-filter estimate of the clean speech signal and measuring the background noise in the enclosure an accurate Speech Intelligibility Index (SII) or Articulation Index (AI) measurement at the microphone position is obtained. On the basis of the estimated speech intelligibility measurement, a decision can be made if the LS signal needs to be modified to improve the intelligibility.

Abstract: A method for accurately estimating and improving the speech intelligibility from a loudspeaker (LS) is disclosed. A microphone is placed in a desired position and using an adaptive filter, an estimate of the clean speech signal at the microphone is generated. By using the adaptive-filter estimate of the clean speech signal and measuring the background noise in the enclosure an accurate Speech Intelligibility Index (SII) or Articulation Index (AI) measurement at the microphone position is obtained. On the basis of the estimated speech intelligibility measurement, a decision can be made if the LS signal needs to be modified to improve the intelligibility.

Abstract: Disclosed is an encoding device which can accurately specify a band having a large error among all the bands by using a small calculation amount. A first position identifier uses a first layer error conversion coefficient indicating an error of a decoding signal for an input signal so as to search for a band having a large error in a relatively wide bandwidth in all the bands of the input signal and generates first position information indicating the identified band. A second position identifier searches for a target frequency band having a large error in a relatively narrow bandwidth in the band identified by the first position identifier and generates second position information indicating the identified target frequency band. An encoder encodes a first layer decoding error conversion coefficient contained in the target frequency band.

Abstract: Disclosed is an encoding device which can accurately specify a band having a large error among all the bands by using a small calculation amount. A first position identifier uses a first layer error conversion coefficient indicating an error of a decoding signal for an input signal so as to search for a band having a large error in a relatively wide bandwidth in all the bands of the input signal and generates first position information indicating the identified band. A second position identifier searches for a target frequency band having a large error in a relatively narrow bandwidth in the band identified by the first position identifier and generates second position information indicating the identified target frequency band. An encoder encodes a first layer decoding error conversion coefficient contained in the target frequency band.

Abstract: The present proposes new methods and an apparatus for enhancement of source coding systems utilizing high frequency reconstruction (HFR). It addresses the problem of insufficient noise contents in a reconstructed highband, by Adaptive Noise-floor Addition. It also introduces new methods for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The present invention is applicable to both speech coding and natural audio coding systems.

Abstract: A vector joint encoding/decoding method and a vector joint encoder/decoder are provided, more than two vectors are jointly encoded, and an encoding index of at least one vector is split and then combined between different vectors, so that encoding idle spaces of different vectors can be recombined, thereby facilitating saving of encoding bits, and because an encoding index of a vector is split and then shorter split indexes are recombined, thereby facilitating reduction of requirements for the bit width of operating parts in encoding/decoding calculation.

Abstract: In a multiband compressor 100, a level calculation unit 121 calculates a signal level inputted for each of bands, a gain calculation unit 122 calculates a gain value from the calculated signal level, and a gain limitation unit 130 limits a gain value by comparison with a gain value of the other band in a compressor for each band. With this configuration, provided is a multiband compressor capable of achieving a balance between the quality of sound and the effect of enhancing the sound level at a high level.

Abstract: A method for coding and decoding an audio signal or speech signal and an apparatus adopting the method are provided.

Abstract: A method implemented by processing and other audio components of an electronic device provides a smart audio output volume control, which correlates a volume level of an audio output to that of an audio input that triggered generation of the audio output. According to one aspect, the method includes: receiving an audio input that triggers generation of an audio output response from the user device; determining an input volume level corresponding to the received audio input; and outputting the audio output response at an output volume level correlated to the input volume level. The media output volume control level of the device is changed from a preset normal level, including from a mute setting, to the determined output level for outputting the audio output. Following, the media output volume control level is automatically reset to a pre-set volume level for normal media output.

Abstract: By monitoring the wind noise in a location in which a cellular telephone is operating and by applying noise reduction and/or cancellation protocols at the appropriate time via analog and/or digital signal processing, it is possible to significantly reduce wind noise entering into a communication system.

Abstract: The present invention relates to a signal processing apparatus and method, a program, and a data recording medium configured such that the playback level of an audio signal can be easily and effectively enhanced without requiring prior analysis. An analyzer 21 generates mapping control information in the form of the root mean square of samples in a given segment of a supplied audio signal. A mapping processor 22 takes a nonlinear function determined by the mapping control information taken as a mapping function, and conducts amplitude conversion on a supplied audio signal using the mapping function. In this way, by conducting amplitude conversion of an audio signal using a nonlinear function that changes according to the characteristics in respective segments of an audio signal, the playback level of an audio signal can be easily and effectively enhanced without requiring prior analysis. The present invention may be applied to portable playback apparatus.

Abstract: Systems, methods, apparatus, and machine-readable media for voice activity detection in a single-channel or multichannel audio signal are disclosed.

Abstract: A method for enhancing speech includes extracting a center channel of an audio signal, flattening the spectrum of the center channel, and mixing the flattened speech channel with the audio signal, thereby enhancing any speech in the audio signal. Also disclosed are a method for extracting a center channel of sound from an audio signal with multiple channels, a method for flattening the spectrum of an audio signal, and a method for detecting speech in an audio signal. Also disclosed is a speech enhancer that includes a center-channel extract, a spectral flattener, a speech-confidence generator, and a mixer for mixing the flattened speech channel with original audio signal proportionate to the confidence of having detected speech, thereby enhancing any speech in the audio signal.

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: Methods of, apparatuses for, and computer readable media having instructions thereon that when executed cause carrying out methods of determining and modifying the perceived loudness of a frequency domain audio signal where the frequency resolution, and corresponding temporal coverage of the frequency domain information is not constant. The frequency (and thus temporal) resolution of the perceived loudness processing is maintained constant at the longest block size. One method includes a block combiner and a loudness modification interpolator.

Abstract: A method and device are provided for the objective evaluation of voice quality of a speech signal. The device includes: a module for extracting a background noise signal, referred to as a noise signal, from the speech signal; a module for calculating the audio parameters of the noise signal; a module for classifying the background noise contained in the noise signal on the basis of the calculated audio parameters, according to a predefined set of background noise classes; and a module for evaluating the voice quality of the speech signal on the basis of at least the resulting classification relative to the background noise in the speech signal.

Abstract: Described herein are systems, methods, and apparatus for determining audio context between an audio source and an audio sink and selecting signal profiles based at least in part on that audio context. The signal profiles may include noise cancellation which is configured to facilitate operation within the audio context. Audio context may include user-to-user and user-to-device communications.

Abstract: A voice enhancement device including an earpiece configured to be positioned in an ear canal of a user. A microcontroller is operatively coupled to the earpiece. The microcontroller is configured to selectively provide at least multitalker babble. An accelerometer is located within the earpiece and operatively coupled to the microcontroller. The accelerometer is configured to detect speech by the user and communicate with the microcontroller to provide the multitalker babble to the earpiece during the detected speech by the user. A method of making the voice enhancement device, and a method for increasing vocal loudness in a patient using the voice enhancement device are also disclosed.

Abstract: In response to a first envelope within a kth frequency band of a first channel, a speech level within the kth frequency band of the first channel is estimated. In response to a second envelope within the kth frequency band of a second channel, a noise level within the kth frequency band of the second channel is estimated. A noise suppression gain for a time frame n is computed in response to the estimated speech level for a preceding time frame, the estimated noise level for the preceding time frame, the estimated speech level for the time frame n, and the estimated noise level for the time frame n. An output channel is generated in response to multiplying the noise suppression gain for the time frame n and the first channel.

Abstract: A voice separation means 82 separates an input voice of a volume adjusted by an input volume adjustment means 81, into a voice recognition voice and a monitoring voice. A monitoring volume adjustment means 83 adjusts a volume of the monitoring voice. An output volume adjustment means 84 adjusts a volume of an output voice and causes an output device to output the output voice of the adjusted volume, the output voice being a voice obtained by synthesizing a synthetic voice and the monitoring voice of the volume adjusted by the monitoring volume adjustment means 83, the synthetic voice being a voice synthesized from information generated as a result of voice recognition of the voice recognition voice. A control means 85 instructs the monitoring volume adjustment means 83 to adjust the volume of the monitoring voice so that an amplification factor of the volume of the output voice with respect to the volume of the input voice does not exceed 1.

Abstract: The present invention relates to a method of evaluating intelligibility of a degraded speech signal received from an audio transmission system conveying a reference speech signal. The method comprises sampling said reference and degraded signals into reference and degraded signal frames, and forming frame pairs by associating reference and degraded signal frames with each other. For each frame pair a difference function representing disturbance is provided, which is then compensated for specific disturbance types for providing a disturbance density function. Based on the density function of a plurality of frame pairs, an overall quality parameter is determined. The method provides for weighing disturbances in silent periods dependent on the loudness of the reference signal.

Abstract: Adaptive Gain Control (AGC) is performed directly in a coded domain. A Coded Domain Adaptive Gain Control (CD-AGC) system modifies at least one parameter of a first encoded signal, resulting in corresponding modified parameter(s). The CD-VQE system replaces the parameter(s) of the first encoded signal with the modified parameter(s), resulting in a second encoded signal. In a decoded state, the second encoded signal approximates a target signal that is a function of two signals, including the first encoded signal and a third encoded signal, in at least a partially decoded states. Thus, the first encoded signal does not have to go through intermediate decode/re-encode processes, which can degrade overall speech quality. Computational resources required for a complete re-encoding are not needed. Overall delay of the system is minimized. The CD-AGC system can be used in any network in which signals are communicated in a coded domain, such as a Third Generation (3G) wireless network.

Abstract: A telecommunication device is disclosed, comprising: a microphone array comprising a plurality of microphones, wherein each microphone receives an analogue acoustic signal; a position sensing device for determining how the telecommunication device is positioned in three-dimensions with respect to a user's mouth; at least one analogue/digital converter for converting each analogue acoustic signal into a digital signal; a digital signal processor for performing signal processing on the received digital signals comprising a controller, a plurality of delay circuits for delaying each received signal based on an input from the controller and a plurality of preamplifiers for adjusting the gain of each received signal based on a gain input from the controller, wherein the controller selects the appropriate delay and gain values applied to each received signal to remove noise from the received signals based on the determined position of the telecommunication device.

Abstract: An audio signal processing device is designed to enhance the low-pitch register of an audio signal by generating harmonics causing a missing fundamental effect with a light load of processing but without damaging an audio waveform. The audio signal processing device includes a filtering part (e.g. a band-pass filter configured of a high-pass filter and a low-pass filter) that extracts a low-pitch signal from an audio signal input thereto; a dynamic range compression part that compresses a dynamic range of the low-pitch signal by use of a time-variant gain relative to a peak of the low-pitch signal, which is detected via a peak hold operation using a predetermined time constant, thus producing a compressed signal; and an adder that adds the compressed signal to the audio signal so as to produce a processed audio signal including harmonics.

Abstract: A method for decoding an audio signal having a bandwidth that extends beyond a bandwidth of a CELP excitation signal in an audio decoder including a CELP-based decoder element. The method includes obtaining a second excitation signal having an audio bandwidth extending beyond the audio bandwidth of the CELP excitation signal, obtaining a set of signals by filtering the second excitation signal with a set of bandpass filters, scaling the set of signals using a set of energy-based parameters, and obtaining a composite output signal by combining the scaled set of signals with a signal based on the audio signal decoded by the CELP-based decoder element.

Abstract: A method of compensating for noise in a receiver having a first receiver unit and a second receiver unit, the method includes receiving a first transmission at the first receiver unit, the first transmission having a first signal component and a first noise component; receiving a second transmission at the second receive unit, the second transmission having a second signal component and a second noise component; determining whether the first noise component and the second noise component are incoherent and; only if it is determined that the first and second noise components are incoherent, processing the first and second transmissions in a first processing path, wherein the first processing path is configured to compensate for incoherent noise.

Abstract: An original loudness level of an audio signal is maintained for a mobile device while maintaining sound quality as good as possible and protecting the loudspeaker used in the mobile device. The loudness of an audio (e.g., speech) signal may be maximized while controlling the excursion of the diaphragm of the loudspeaker (in a mobile device) to stay within the allowed range. In an implementation, the peak excursion is predicted (e.g., estimated) using the input signal and an excursion transfer function. The signal may then be modified to limit the excursion and to maximize loudness.

Abstract: In a speech enhancement apparatus, a generator part generates a value representing likelihood of a consonant from an input audio signal, and a calculator part generates a consonant/vowel discriminating signal for discriminating a consonant portion and a vowel portion based on the generated value, detects a first signal level of the vowel portion and a second signal level of the consonant portion based on the audio signal and the consonant/vowel discriminating signal, and outputs a level-related signal. A determining part determines a gain coefficient that exceeds one when the second signal level is smaller than the first signal level based on the level-related signal so that the gain coefficient increases as the second signal level becomes smaller than the first signal level. A multiplier part multiplies the audio signal by the gain coefficient to output an audio signal having an emphasized consonant portion.

Abstract: Systems and methods for adaptive power control are provided. In exemplary embodiments, a primary signal is received. A noise power level of the primary signal is then estimated. The noise power level may then be compared to at least one power threshold. Subsequently, a large power consuming system is controlled based on the comparison of the noise power level to the power threshold.

Abstract: An encoder whereby the bit efficiency of encoding can be improved, thereby improving the qualities of signals as decoded. In the encoder: a time-frequency converting unit (101) converts signals, which are to be encoded, to frequency domain signals; an adaptive spectrum formation encoding unit (102) determines an effective range in the frequency band of the frequency domain signals; and a pulse vector encoding unit (103) pulse vector encodes only the signal components within the effective range.

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: An adaptive equalization system that adjusts the spectral shape of a speech signal based on an intelligibility measurement of the speech signal may improve the intelligibility of the output speech signal. Such an adaptive equalization system may include a speech intelligibility measurement module, a spectral shape adjustment module, and an adaptive equalization module. The speech intelligibility measurement module is configured to calculate a speech intelligibility measurement of a speech signal. The spectral shape adjustment module is configured to generate a weighted long-term speech curve based on a first predetermined long-term average speech curve, a second predetermined long-term average speech curve, and the speech intelligibility measurement. The adaptive equalization module is configured to adapt equalization coefficients for the speech signal based on the weighted long-term speech curve.

Abstract: Systems and methods are described to automatically balance acoustic channel sensitivity. A long-term power level of a main acoustic signal is calculated to obtain an averaged main acoustic signal. Segments of the main acoustic signal are excluded from the averaged main acoustic signal using a desired voice activity detection signal. A long-term power level of a reference acoustic signal is calculated to obtain an averaged reference acoustic signal. Segments of the reference acoustic signal are excluded from the averaged reference acoustic signal using a desired voice activity detection signal. An amplitude correction signal is created using the averaged main acoustic signal and the averaged reference acoustic signal.

Abstract: There is disclosed an encoder apparatus whereby, when a band expanding technique for encoding, based on the spectral data of a lower frequency portion, the spectral data of a higher frequency portion is applied to a lower layer in a hierarchical encoding/decoding system, an efficient encoding can be performed in an upper layer as well, thereby improving the decoded-signal quality. In an encoder apparatus (101), a second layer decoder unit (207) calculates a spectrum (differential spectrum), which is to be encoded in a third layer encoder unit (210) that is an upper layer of the second layer decoder unit (207), by applying such an ideal gain (first gain parameter a1) that minimizes the energy of the differential spectrum.

Abstract: A method of speech enhancement in a room (10) includes the steps of capturing audio signals from a speaker's voice by a microphone (12), estimating an ambient noise level in the room from the captured audio signals, processing the captured audio signals by an audio signal processing unit (20), estimating a reverberation level, determining the gain to be applied to the captured audio signals by the audio signal processing unit according to a comparison between the estimated ambient noise level and the estimated reverberation level, and generating sound according to the processed audio signals by a loudspeaker arrangement (24) located in the room, wherein the reverberation level is the level of reverberant components of the sound generated by the loudspeaker arrangement.

Abstract: Provided are a method and apparatus for encoding and decoding a high frequency signal by using a low frequency signal. The high frequency signal can be encoded by extracting a coefficient by linear predicting a high frequency signal, and encoding the coefficient, generating a signal by using the extracted coefficient and a low frequency signal, and encoding the high frequency signal by calculating a ratio between the high frequency signal and an energy value of the generated signal. Also, the high frequency signal can be decoded by decoding a coefficient, which is extracted by linear predicting a high frequency signal, and a low frequency signal, and generating a signal by using the decoded coefficient and the decoded low frequency signal, and adjusting the generated signal by decoding a ratio between the generated signal and an energy value of the high frequency signal.

Abstract: In one configuration, erasure of a significant frame of a sustained voiced segment is detected. An adaptive codebook gain value for the erased frame is calculated based on the preceding frame. If the calculated value is less than (alternatively, not greater than) a threshold value, a higher adaptive codebook gain value is used for the erased frame. The higher value may be derived from the calculated value or selected from among one or more predefined values.

Abstract: The invention relates to a method for determining a quality indicator representing a perceived quality of an output signal of an audio system with respect to a reference signal. The reference signal and the output signal are processed and compared. The processing includes dividing the reference signal and the output signal into mutually corresponding time frames, and includes scaling the intensity of the reference signal towards a fixed intensity level, and then performing measurements on time frames within the scaled reference signal for determining reference signal time frame characteristics. Further on, the loudness of the output signal is scaled towards a fixed loudness level in the perceptual loudness domain. Finally, the loudness of the reference signal is scaled from a loudness level corresponding to the output signal related intensity level towards a loudness level related to the loudness level of the scaled output signal in the perceptual loudness domain.

Abstract: An apparatus for decoding data segments representing a time-domain data stream, a data segment being encoded in the time domain or in the frequency domain, a data segment being encoded in the frequency domain having successive blocks of data representing successive and overlapping blocks of time-domain data samples. The apparatus includes a time-domain decoder for decoding a data segment being encoded in the time domain and a processor for processing the data segment being encoded in the frequency domain and output data of the time-domain decoder to obtain overlapping time-domain data blocks. The apparatus further includes an overlap/add-combiner for combining the overlapping time-domain data blocks to obtain a decoded data segment of the time-domain data stream.

Abstract: A method includes receiving a first value of a mixing factor. The first value corresponds to a first portion of an audio signal received at an audio encoder. The method includes receiving a second value of the mixing factor. The second value corresponds to a second portion of the audio signal. The method also includes generating a third value of the mixing factor at least partially based on the first value and the second value and mixing an excitation signal with modulated noise based on the third value. Another method includes determining a first set of spectral frequency values corresponding to an audio signal and determining a second set of spectral frequency values that approximates the first set of spectral frequency values. A gain value corresponding to at least a portion of the audio signal is adjusted based on a difference between the first set and the second set.

Abstract: A particular method includes determining, based on an inter-line spectral pair (LSP) spacing corresponding to an audio signal, that the audio signal includes a component corresponding to an artifact-generating condition. The method also includes, in response to determining that the audio signal includes the component, adjusting a gain parameter corresponding to the audio signal. For example, the gain parameter may be adjusted via gain attenuation and/or gain smoothing.

Abstract: A particular method includes determining, based on spectral information corresponding to an audio signal that includes a low-band portion and a high-band portion, that the audio signal includes a component corresponding to an artifact-generating condition. The method also includes filtering the high-band portion of the audio signal and generating an encoded signal. Generating the encoded signal includes determining gain information based on a ratio of a first energy corresponding to filtered high-band output to a second energy corresponding to the low-band portion to reduce an audible effect of the artifact-generating condition.

Abstract: An echo suppression system and method, and a computer-readable storage medium that is configured with instructions that when executed carry out echo suppression. Each of the system and the method includes the elements of a linear echo suppressor having a reference signal path, with a nonlinearity introduced in the reference signal path to introduce energy in spectral bands. Unlike an echo canceller, the echo suppression system and method are relatively robust to errors in the introduced nonlinearity.

Abstract: According to an embodiment, a method of reducing noise in a signal received at a processing stage of an acoustic system includes, at the processing stage identifying at least one frequency which causes a system gain of the acoustic system to be above an average system gain of the acoustic system; providing a noise attenuation factor for reducing noise in the signal for the at least one frequency, the noise attenuation factor for the at least one frequency based on the system gain for that frequency; and applying the noise attenuation factor to a component of the signal at that frequency.

Abstract: Provided are, among other things, systems, methods and techniques for detecting whether a transient exists within an audio signal. According to one representative embodiment, a segment of a digital audio signal is divided into blocks, and a norm value is calculated for each of a number of the blocks, resulting in a set of norm values for such blocks, each such norm value representing a measure of signal strength within a corresponding block. A maximum norm value is then identified across such blocks, and a test criterion is applied to the norm values. If the test criterion is not satisfied, a first signal indicating that the segment does not include any transient is output, and if the test criterion is satisfied, a second signal indicating that the segment includes a transient is output. According to this embodiment, the test criterion involves a comparison of the maximum norm value to a different second maximum norm value, subject to a specified constraint, within the segment.

Abstract: A decoding apparatus decodes a first encoded data that is encoded from a low-frequency component of an audio signal, and a second encoded data that is used when creating a high-frequency component of an audio signal from a low-frequency component and encoded in accordance with a certain bandwidth, into the audio signal. In the decoding apparatus, a high-frequency component detecting unit divides the high-frequency component into bands with a certain interval range correspondingly to the certain bandwidth, and detects magnitude of the high-frequency components corresponding to each of the bands. A high-frequency component compensating unit compensates the high-frequency components based on the magnitude of the high-frequency components corresponding to each of the bands detected by the high-frequency component detecting unit.

Abstract: A method is disclosed for controlling a voice-activated device by interpreting a spoken command as a series of voiced and non-voiced intervals. A responsive action is then performed according to the number of voiced intervals in the command. The method is well-suited to applications having a small number of specific voice-activated response functions. Applications using the inventive method offer numerous advantages over traditional speech recognition systems including speaker universality, language independence, no training or calibration needed, implementation with simple microcontrollers, and extremely low cost. For time-critical applications such as pulsers and measurement devices, where fast reaction is crucial to catch a transient event, the method provides near-instantaneous command response, yet versatile voice control.

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.