Abstract: The present disclosure provides an audio circuit capable of inhibiting a current when mute is deactivated. An output terminal of a class D amplifier circuit is connected to an electroacoustic conversion element through a low-pass filter. An output node of a bridge circuit is connected to the output terminal. An integrator integrates and outputs, in a non-mute period in which a mute control signal is negated, a difference between an input signal and a feedback signal corresponding to an output signal generated at the output terminal, and outputs a predetermined bias voltage in a mute period in which the mute control signal is asserted. A PWM comparator compares the output of the integrator with a periodic voltage. A driver switches, in the non-mute period, the bridge circuit according to an output of the PWM comparator, and fixes an output of the bridge circuit in the mute period.

Abstract: A vibration generating apparatus comprises a vibration apparatus and a vibration driving circuit including a driving signal generator configured to supply a driving signal to the vibration apparatus, wherein the driving signal generator is configured to adjust a frequency-based gain compensation value based on at least one of a circuit internal temperature value of the vibration driving circuit and a temperature prediction value of the vibration apparatus corresponding to a current value of an nth driving signal, compensate for a frequency-based gain value based on the adjusted frequency-based gain compensation value, compensate for an (n+1)th driving signal based on the compensated frequency-based gain value, and supply the compensated (n+1)th driving signal to the vibration apparatus.

Abstract: The technology described in this document can be embodied in a method that includes receiving an input signal captured by one or more sensors associated with an active noise reduction (ANR) headphone, and determining one or more characteristics of a first portion of the input signal. Based on the one or more characteristics of the first portion of the input signal, a gain of a variable gain amplifier (VGA) disposed in an ANR signal flow path can be adjusted, and accordingly, a set of coefficients for a tunable digital filter disposed in the ANR signal flow path can be selected. The method further includes processing a second portion of the input signal in the ANR signal flow path using the adjusted gain and selected set of coefficients to generate a second output signal for the electroacoustic transducer of the ANR headphone.

Abstract: An audio system includes a digital audio interface, a first filter stage, a second filter stage, a controller, and a first path selector. The controller is communicatively coupled to the digital audio interface and configured to determine whether an audio signal received at the digital audio interface is below a first signal threshold. The controller selects one of a first signal path that passes through the first and second filter stages and a second signal path that bypasses the first filter stage based upon the determination. The first path selector is communicatively coupled to the controller and at least one of the first filter stage and the second filter stage and is configured to generate the selected one of the first and second signal paths.

Abstract: A digitally controlled microphone with audio compressor includes a capsule; a compressor circuitry having a compressor engine, a digitally controlled root mean square (RMS) detector and a digitally controlled analog processor; a bypass switch to the compressor circuitry; an analog to digital converter (ADC); a digital processor that sends digital audio to a host and receives a second digital audio signal; a digital to analog converter (DAC) and a summing circuit that combine the audio with the audio from the host and send it to a digitally-controlled headphone amplifier and a headphone jack; a microphone body; a serial host interface connector; and hardware controls on the microphone body.

Abstract: An audio system for use in a confined space comprises N acoustic transducers, where N is an integer and N?2. Each acoustic transducer is configured to receive an audio signal for a respective channel of the audio system and to generate an acoustic signal therefrom, and is operably coupled to a respective filter having a phase response. The phase response of at least one of the filters is different from the phase response of at least one other of the filters at a frequency f having a corresponding wavelength ?=d/(n+½), where n is an integer and n?0 and d is a characteristic separation of two or more of the N acoustic transducers when in the confined space. Compensation is thereby achieved for a reduction in acoustic power that would otherwise occur over a range of frequencies of width ?f around frequency f.

Abstract: A method and apparatus are disclosed for providing a video signature representative of a content of a video signal. A method and apparatus are further disclosed for providing an audio signature representative of a content of an audio signal. A method and apparatus for detecting lip sync are further disclosed and take advantage of the method and apparatus disclosed for providing a video signature and an audio signature.

Abstract: The implementations disclosed herein provide for a storage device including a preamplifier that dynamically adjusts at least one of a rise time and fall time of an analog write current pulse based on a length of a corresponding write transition and/or characteristics of a media location where the write transition is to be recorded.

Abstract: Disclosed herein are methods and apparatuses designed to reduce the power usage of a hearing prosthesis. To reduce the power usage of the prosthesis, an external portion may transmit a signal having a plurality of transmission parameters to an internal component. The external portion receives an indication of an implant power parameter associated with the plurality of transmission parameters. After receiving the indication, the external portion adjusts at least one transmission parameter. After adjusting at least one parameter, the external portion transmits a signal to the internal component based on the adjusted parameters and responsively receives an indication of an implant power parameter associated with the adjusted parameters. Based on the received indication of the implant power parameter for a plurality of transmissions, the external portion determines a preferred set of transmission parameters. Finally, the external portion is operated based on the preferred set of transmission parameters.

Abstract: Provided are methods and systems for noise suppression within multiple time-frequency points of spectral representations. A multi-feature cluster tracker is used to track signal and noise sources and to predict signal versus noise dominance at each time-frequency point. Multiple features, such as binaural and monaural features, may be used for these purposes. A Gaussian mixture model (GMM) is developed and, in some embodiments, dynamically updated for distinguishing signal from noise and performing mask-based noise reduction. Each frequency band may use a different GMM or share a GMM with other frequency bands. A GMM may be combined from two models, with one trained to model time-frequency points in which the target dominates and another trained to model time-frequency points in which the noise dominates. Dynamic updates of a GMM may be performed using an expectation-maximization algorithm in an unsupervised fashion.

Abstract: A signal processing apparatus includes an absolute value unit configured to convert an audio signal into absolute values, a representative value calculation unit configured to calculate representative values of consecutive sample values included in blocks of the audio signal which has been converted into the absolute values using at least maximum sample values among values of the samples included in the blocks for individual blocks, an average value calculation unit configured to determine a section which includes a predetermined number of consecutive blocks as a frame and calculate a maximum value of the representative values of the blocks included in the frame and an average value of the representative values of the blocks included in the frame, and a detector configured to detect click noise in the frame on the basis of a ratio of the maximum value to the average value.

Abstract: The application relates to a compressor circuit for a listening device, the compressor circuit comprising a pair of input terminals and a pair of output terminals, the compressor circuit being adapted for receiving an electric input signal representing an audio signal with an input voltage swing Vipp at the input terminals and for providing a possibly compressed version of the input signal as an output signal with an output voltage swing Vopp at the output terminals, the compressor circuit comprising. The application further relates to a listening device and to the use of the compressor circuit. The object of the present application is to provide a compressor circuit with a proper functioning also at relatively low input levels.

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: In an embodiment of the invention, the voice coil of an electro dynamic transducer is protected against thermal overload by estimating the temperature of a magnet in the electro dynamic transducer. When a power limit based on the temperature of the magnet and on a predetermined voice coil temperature limit is reached by an audio signal, the power applied to the voice coil is reduced.

Abstract: A signal processing apparatus includes: one or more detection means for detecting movement of a diaphragm of a speaker in correspondence with feedback methods that are different feedback methods; analog-to-digital conversion means for converting one or more detection signals acquired by the detection means into a digital form; feedback signal generating means for generating feedback signals corresponding to the feedback methods using the digital detection signals; synthesis means for combining an audio signal to be output as a driving signal of the speaker with the feedback signals; correction equalizer means for setting an equalizing characteristic to allow a sound reproduced by the speaker to have a target frequency characteristic by changing the digital audio signal; feedback operation setting means for setting feedback methods in which a feedback operation up to combining the audio signal with the feedback signal is performed and the feedback operation is not performed equalizing characteristic changing a

Abstract: A frequency band of an audio input signal is analyzed to determine if a transient is present. When transients are detected, modifications are made to the intensity levels corresponding to the frequency band for a brief time period.

Abstract: It is described a method for controlling an adaptation of a behavior of an audio device (100) to a current acoustic environmental condition. The method comprises (a) monitoring an audio output signal (x(t), x?(t)) being provided to an acoustic output device (110) of the audio device (100) for outputting an acoustic output signal, (b) measuring an audio input signal (z(t)) being provided by an acoustic input device (120) of the audio device (100), wherein the audio input signal (z(t)) is indicative for a feedback portion of the acoustic output signal and for the current acoustic environmental condition, (c) determining a relation between the audio output signal (x?(t)) and the audio input signal (z(t)) and (d) adapting the behavior of the audio device (100) based on the determined relation.

Abstract: Systems and methods of limiting peak audio power in mobile devices may include a high pass filter and a burst module to detect a burst load condition in a mobile device. The burst module can also apply the high pass filter to an audio signal of the mobile device in response to the burst load condition to obtain a filtered signal, and transmit the filtered audio signal to a speaker of the mobile device.

Abstract: Feedback whistle in hearing devices is intended to be able to be suppressed without loss of output of the useful signal. To this end, it is provided to establish or predetermine a frequency range which is susceptible to feedback. From an input signal which has a spectral component in the frequency range susceptible to feedback, a predeterminable component is substituted with a synthetic signal. Mixing-in a synthetic signal is also possibly used to widen the spectrum of an input signal, which is limited.

Abstract: Apparatus and methods for reducing misinterpretation of gesture-based input to portable electronic devices are described.

Abstract: A processing system generates an output signal which includes desired signal components, and reduces or eliminates tonal noise. The output signal may be provided to any subsequent signal processing system, including voice recognition systems, pitch detectors, and other processing systems. The subsequent processing systems are less likely to mistake tonal input signal noise for desired signal content, to needlessly consume computational resources to analyze noise, and to take spurious actions induced by the tonal noise.

Abstract: A signal processing apparatus is configured to change volume level or frequency characteristics of an input signal with a limited bandwidth in a first frequency range. The apparatus includes: an information extracting unit configured to extract second frequency characteristic information from a collection signal with a limited bandwidth in a second frequency range different from the first frequency range; a frequency characteristic information extending unit configured to estimate first frequency characteristic information from the second frequency characteristic information extracted by the information extracting unit, the first frequency characteristic information including the first frequency range; and a signal correcting unit configured to change volume level or frequency characteristics of the input signal according to the first frequency characteristic information obtained by the frequency characteristic information extending unit.

Abstract: Disclosed herein is a digital filter circuit for producing a noise reduction signal for reducing noise based on a noise signal outputted from a microphone which collects the noise, including: an analog/digital conversion section; a first digital filter section; an arithmetic operation processing section; a second digital filter section; and a digital/analog conversion section. The first digital filter section and/or the second digital filter section are configured such that a predetermined attenuation amount is obtained within a predetermined range in the proximity of a sampling frequency around the sampling frequency.

Abstract: The present invention discloses a circuit with three-stage of power-on sequence used for suppressing the pop noise in audio system. It mainly comprises a first resistor (R1); a capacitor (Cout); a first switch (SW1); a second switch (SW2); a soft start device; a first feedback amplifier; and a second feedback amplifier. By using the three-stage of power-on sequence, the present invention can effectively suppress the pop noise when the audio driver is power on.

Abstract: A sound processing apparatus includes a sound determination portion operable to determine whether an input sound includes a first sound emitted from a particular source based on location information of the source, a sound separation portion operable to separate the input sound into the first sound and a second sound emitted from a source different from the particular source if the sound determination portion determines that the input sound includes the first sound, and a sound mixing portion operable to mix the first sound and the second sound separated by the sound separation portion at a prescribed volume ratio.

Abstract: A signal processing apparatus is configured to change volume level or frequency characteristics of an input signal with a limited bandwidth in a first frequency range. The apparatus includes: an information extracting unit configured to extract second frequency characteristic information from a collection signal with a limited bandwidth in a second frequency range different from the first frequency range; a frequency characteristic information extending unit configured to estimate first frequency characteristic information from the second frequency characteristic information extracted by the information extracting unit, the first frequency characteristic information including the first frequency range; and a signal correcting unit configured to change volume level or frequency characteristics of the input signal according to the first frequency characteristic information obtained by the frequency characteristic information extending unit.

Abstract: An acoustic signal level limiter provides a telephone handset/headset user with protection against loud audible signals generated within a communications system. The acoustic signal limiter comprises an acoustic signal level attenuation circuit and at least one acoustic signal level relay circuit. Once activated, the acoustic signal level attenuation circuit creates an attenuation network that attenuates an electrical acoustic signal transmitted through the communications system. Additional acoustic signal level relay circuits are activated to further attenuate the electrical acoustical signal to prevent the acoustic signal level attenuation circuit from operating in a deep saturation mode and provide further hearing safety for the telephone headset/handset user. A fuse in series and “Zener Zap” shunting transistor diodes may provide additional failsafe protection.

Abstract: A system for limiting the excursion of an audio speaker. The system may include first lowpass filter circuitry configured to receive a system input signal and to generate a first signal. The system may further include clipping circuitry configured between the first lowpass filter circuitry and the first combiner circuitry. The system may include first highpass filter circuitry configured to receive the system input signal and to generate a second signal. The system may include second lowpass filter circuitry configured to receive a combined output signal from the first combiner circuitry. The system may include allpass filter circuitry configured to receive the system input signal and to generate a fourth signal. The system may include second highpass filter circuitry configured to receive the fourth signal from the allpass filter circuitry. The second combiner circuitry may be configured to generate an excursion limiting output signal.

Abstract: This specification describes technologies relating to editing audio data. In general, one aspect of the subject matter described in this specification can be embodied in methods that include receiving an audio signal including digital audio data; receiving an input identifying particular audio data of the audio signal corresponding to a noise pulse; and replacing the audio data corresponding to the detected noise pulse using interpolation of adjacent audio data to generate an edited audio signal. Other embodiments of this aspect include corresponding systems, apparatus, and computer program products.

Abstract: A circuit of reducing a pop-up noise in a digital amplifier includes a switch unit and a switch signal generator. The switch unit is coupled in parallel to an output load between an output node of the digital amplifier and a reference node. The switch unit controls a current flowing through the output load by forming a conduction path between the output node and the reference node in response to a switch signal. The switch signal generator generates the switch signal in response to a switch control signal indicating a power-on or a power-off. The pop-up noise is reduced by the conduction path that is formed when the digital amplifier is powered on or off.

Abstract: The invention provides a microphone circuit. In one embodiment, the microphone circuit comprises a transducer, a biasing resistor, a pre-amplifier, and a switch circuit. The transducer is coupled between a ground and a first node for converting a sound into a voltage signal output to the first node. The biasing resistor is coupled between the ground and the first node. The pre-amplifier is biased with a biasing voltage and coupled between the first node and a second node, and amplifies the voltage signal to obtain an output signal at the second node. The switch circuit is coupled between the first node and the ground, couples the first node to the ground when the microphone circuit is reset, and decouples the first node from the ground after a voltage status of the microphone circuit is stable, thus clamping a voltage of the first node to the ground to prevent generation of a popping noise when the microphone circuit is reset.

Abstract: An audio testing system is configured for receiving an audio signal from the an audio emitting device. The system samples the audio signal and obtains sampling points from the audio signal for determining if the audio signal has been distorted. A related method is also disclosed.

Abstract: An audio control apparatus. The audio control apparatus comprises a processing module, an amplification module, and a control module. The processing module receives and processes an audio signal according to a control signal, and generates a processed signal. The amplification module coupled to the processing module, amplifies the processed signal to generate an output signal. The control module coupled to the processing module and the amplification module, receives the output signal to generate the control signal.

Abstract: A method for clipping and post-clipping processing an audio signal, includes clipping an audio signal to provide a clipped audio signal; filtering, by a first filter, the audio signal to provide a filtered unclipped audio signal; and filtering, by a second filter, the clipped audio signal to provide a filtered clipped audio signal. The method further includes differentially combining the filtered clipped audio signal and the clipped audio signal to provide a differentially combined audio signal; and combining the filtered unclipped audio signal and the differentially combined audio signal to provide an output signal.

Abstract: The hearing device to be worn by a user of the hearing device comprises an input transducer for transducing incoming acoustical sound into audio signals; a signal processor for processing audio signals; an output transducer for transducing audio signals into outgoing acoustical sound. Said signal processor is designed such that there exists an input SPL range of at least a portion of said incoming acoustical sound, in which an increase in input SPL of said at least one portion of said incoming acoustical sound results in a decrease in output SPL of at least a portion of said outgoing acoustical sound, wherein SPL stands for sound pressure level. Through this, at high input SPL the hearing device's power consumption can be reduced by making use of direct sound, which propagate as sound waves from outside the user's ear canal to the user's ear drum.

Abstract: An amplifier power-down apparatus for suppressing transient signals in a circuit, for example an audio circuit, comprising a reference voltage generator circuit for generating a reference voltage. The reference voltage generator circuit comprises a capacitor for maintaining the reference voltage at a desired level. The amplifier power-down apparatus comprises a discharge control circuit for controlling the operation of the reference voltage generator circuit during power-down. The discharge control circuit comprises an amplifier with a positive feedback path for controlling a current flow from the capacitor during power-down.

Abstract: An amplifier start-up apparatus for reducing transient signals in a circuit, for example an audio circuit, comprising a reference voltage generator circuit for generating a reference voltage. The reference voltage generator circuit comprises a capacitor for maintaining the reference voltage at a desired level. The amplifier start-up apparatus comprises a control circuit for controlling the operation of the reference voltage generator circuit during power-up. The control circuit comprises an amplifier with a positive feedback path for controlling a current flow to the capacitor.

Abstract: Protection method and system for use with speaker systems. The method and system being suitable for use in any number of environments to protect speakers from overload and other undesirable operating conditions. The method and system optionally being suitable for use in maintaining system sound integrity while protecting the speakers from overload and/or other undesirable operating conditions.

Abstract: A pop noise suppression apparatus for eliminating popping noise generated upon initiation or shutdown of an audio output circuit comprises a switch component and a control circuit. The switch component allows the audio output circuit to provide audio through the output of the audio output circuit. The control circuit provides a mute signal for a first period of time in to response initiation or shutdown of the audio circuit. The control circuit comprises a capacitor to be charged upon initiation of the audio output circuit or to be discharged upon shutdown of the audio output circuit. A length of the first period of time during which the mute signal is provided depends on a second period of time to charge or discharge the capacitor.

Abstract: A method and an apparatus for designing a sound compensation filter of a portable terminal are provided. The method includes synchronizing a signal input through a microphone of the system and a test signal, estimating a loss interval of the synchronized signal, compensating for a frame signal delayed by a signal loss in a time axis when the signal loss of the estimated loss interval is greater than a threshold and restoring the loss interval of the signal.

Abstract: The invention makes use of impulse responses of the performance venue to process a recording or other signal so as to emulate that recording having being recorded in the performance venue. In particular, by measuring or calculating the impulse responses of a performance venue such as an auditorium between an instrument location within the venue and one or more soundfield sampling locations, it then becomes possible to process a “dry” signal, being a signal which has little or no reverberation or other artifacts introduced by the location in which it is captured (such as, for example, a close microphone studio recording) with the impulse response or responses so as to then make the signal seem as if it was produced at the instrument location in the performance venue, and captured at the soundfield sampling location.

Abstract: A voice enhancement logic improves the perceptual quality of a processed voice. The voice enhancement system includes a noise detector and a noise attenuator. The noise detector detects a wind buffet and a continuous noise by modeling the wind buffet. The noise attenuator dampens the wind buffet to improve the intelligibility of an unvoiced, a fully voiced, or a mixed voice segment.

Abstract: The present invention discloses a multi-level output signal converter, which is connected to an audio amplifier. The audio amplifier comprises a comparing/measuring device, an encoder and an output unit. The multi-level output signal converter comprises a timing processing unit and a multi-level converter. The timing processing unit is connected to the comparing/measuring device and the encoder. The timing processing unit includes a plurality of flip-flops and a timing summing element. The flip-flop receives a first signal from the comparing/measuring device and outputs the first signal to the timing summing element. The encoder converts the first signal into a second signal. The multi-level converter is connected to the encoder and the output unit. The encoder transmits the second signal to the multi-level converter, and the multi-level converter thus outputs a third signal to the output unit.

Abstract: A speaker array and microphone arrays positioned on both sides of the speaker array are provided. A plurality of focal points each serving as a position of a talker are set in front of the microphone arrays respectively symmetrically with respect to a centerline of the speaker array, and a bundle of sound collecting beams is output toward the focal points. Difference values between sound collecting beams directed toward the focal points that are symmetrical with respect to the centerline are calculated to cancel sound components that detour from the speaker array to microphones. Then, it is estimated based on totals of squares of peak values of the difference values for a particular time period that the position of the talker is close to which one of the focal points, and the position of the talker is decided by comparing the totals of the squares of the peak values of the sound collecting beams directed to the focal points that are symmetrical mutually.

Abstract: A noise reduction system includes multiple transducers that generate time domain signals. A transforming device transforms the time domain signals into frequency domain signals. A signal mixing device mixes the frequency domain signals according to a mixing ratio. Frequency domain signals are rotated in phase to generate phase rotated signals. A post-processing device attenuates portions of the output based on coherence levels of the signals.

Abstract: A system and method to detect and remediate unacceptable levels of speech intelligibility evaluates received test audio transmitted across and received in a space or region of interest. Intelligibility is improved by altering the rate, pitch, amplitude and frequency bands energy during presentation of the speech signal.

Abstract: A system and method to detect and measure remediated speech intelligibility by evaluating received test audio transmitted across and received in a space or region of interest. Remediation of the test audio may include altering the rate, pitch, amplitude and frequency bands energy during presentation of the speech signal.

Abstract: An audio output apparatus includes: an audio codec outputting an analog audio signal corresponding to a digital audio signal from a system controller; a switch unit having a first end coupled to the audio codec through a capacitor, and a grounded second end; and a switch controller triggered by a trigger signal to output a control signal to a control end of the switch unit such that the switch unit couples the capacitor to ground in response to the control signal. The trigger signal is generated by one of the system controller, the audio codec, and a power circuit supplying electric power to the system controller, the audio codec and the switch controller upon occurrence of a condition associated with pop noise, and is outputted to the switch controller before the pop noise is generated, such that the pop noise is conducted to ground via the switch unit.

Abstract: The present disclosure generally relates to a system for limiting the excursion of an audio speaker. The system may include first lowpass filter circuitry configured to receive a system input signal and to generate a first signal. The system may further include clipping circuitry configured between the first lowpass filter circuitry and the first combiner circuitry. The system may include first highpass filter circuitry configured to receive the system input signal and to generate a second signal. The system may include second lowpass filter circuitry configured to receive a combined output signal from the first combiner circuitry. The system may include allpass filter circuitry configured to receive the system input signal and to generate a fourth signal. The system may include second highpass filter circuitry configured to receive the fourth signal from the allpass filter circuitry. The second combiner circuitry may be configured to generate an excursion limiting output signal.

Abstract: This disclosure describes circuit configurations that may be used for active noise cancellation in the digital domain. In particular, this disclosure proposes the use a down sample unit and an up sample unit, rather than memory-based delay circuits, to achieve one or more desired delays in digital adaptive noise cancellation circuits or other circuits that use delay for signal processing. The delay achieved by the down sample unit and the up sample unit may be tunable so as to allow flexibility in producing the necessary delay for different active noise cancellation circuit configurations. Many different adaptive noise cancellation circuit configurations are discussed, and the techniques may also be useful for other types of circuits, such as low-latency equalization circuits.