Abstract: The present disclosure relates to circuitry comprising audio amplifier circuitry for receiving an audio signal to be amplified; and first and second output nodes for outputting first and second differential output signals. The circuitry further comprises common mode buffer circuitry configured to receive a common mode voltage and to selectively output the common mode voltage to the first and second output nodes.

Abstract: Example embodiments provide a process that includes one or more of receiving an audio signal from a feedback path of a feedback compressor circuit, determining whether an auxiliary attenuation value applied to the feedback compressor circuit has changed since a last audio signal was received, responsive to determining the auxiliary value has changed, determining a current operational state value of the LPF needs to be modified based on the changed auxiliary attenuation value, modifying the operational state value of the LPF, and applying the audio signal to the modified LPF.

Abstract: Encoding/decoding an audio signal having one or more audio components, wherein each audio component is associated with a spatial location. A first audio signal presentation (z) of the audio components, a first set of transform parameters (w(f)), and signal level data (?2) are encoded and transmitted to the decoder. The decoder uses the first set of transform parameters (w(f)) to form a reconstructed simulation input signal intended for an acoustic environment simulation, and applies a signal level modification (?) to the reconstructed simulation input signal. The signal level modification is based on the signal level data (?2) and data (p2) related to the acoustic environment simulation. The attenuated reconstructed simulation input signal is then processed in an acoustic environment simulator. With this process, the decoder does not need to determine the signal level of the simulation input signal, thereby reducing processing load.

Abstract: Example embodiments provide a process that includes one or more of receiving an audio signal at a feedback compressor circuit, receiving an auxiliary attenuation signal from an auxiliary attenuation source, determining a threshold power level based on a value of the auxiliary attenuation signal, determining an output power level of the audio signal exceeds the threshold power level, combining the audio signal with the auxiliary attenuation signal from the auxiliary attenuation source and a compressed attenuation signal from the feedback compressor circuit to create a combination signal, and generating an audio output signal of the feedback compressor circuit based on the combination signal.

Abstract: A transducer system has a transducer configured to receive an incident signal and produce an output signal as a function of the incident signal. As known in the art, the output signal has a ?3 dB point. The transducer system also has circuitry operatively coupled with the transducer. The circuitry includes an amplifier with an output configured to produce an amplified signal as a function of the output signal. In addition, the circuitry further has a positive feedback loop and a negative feedback loop that both are electrically coupled between the transducer and the amplifier. The positive feedback loop is configured to move the ?3 dB point in a first frequency direction. Conversely, the negative feedback loop is configured to move the ?3 dB point in a second frequency direction. Preferably, the first and second frequency directions are different.

Abstract: A public address system includes audio inputs from a moderator/presenter and from one or more participants. In an embodiment, the moderator speaks first and then selects participants to speak, utilizing audio captured by participant devices. A central signal processor is configured to receive the audio inputs and to utilize a configured acoustic model to provide for acoustic echo cancellation (AEC) and feedback control (FBC) during various phases of a presentation or conference. Audio signals from the presenter and/or participants, that have been processed to remove echo, are utilized as reference signals during various phases of the audio presentation that utilize the acoustic model for either AEC or FBC. The system utilizes the knowledge that the best learning occurs during the far talking state to learn the echo path in the canceler mode (AEC) vs. the feedback mode (FBC), which usually can only train in a double-talking mode.

Abstract: Transfer functions can describe responses of microphones or ears to sounds at different locations on a sphere. The transfer functions can be compressed by determining, based on transfer functions, a) one or more basis transfer functions, and b) spherical harmonics coefficients that describe variations of the transfer functions with respect to spherical coordinates. Other aspects are described and claimed.

Abstract: A disclosed method for wellsite operations includes obtaining a spectral decomposition, of a seismic data associated with a geological formation. The spectral decomposition includes a first spectral representation generated using a first operator and a second spectral representation generated using a second operator. The method also includes determining a first characteristic of the first operator and a second characteristic of the second operator. The method further includes determining at least one acceptable operator based on the first characteristic and the second characteristic. The method also includes generating a geological model feature using the at least one acceptable operator.

Abstract: Encoding/decoding an audio signal having one or more audio components, wherein each audio component is associated with a spatial location. A first audio signal presentation (z) of the audio components, a first set of transform parameters (w(f)), and signal level data (?2) are encoded and transmitted to the decoder. The decoder uses the first set of transform parameters (w(f)) to form a reconstructed simulation input signal intended for an acoustic environment simulation, and applies a signal level modification (?) to the reconstructed simulation input signal. The signal level modification is based on the signal level data (?2) and data (p2) related to the acoustic environment simulation. The attenuated reconstructed simulation input signal is then processed in an acoustic environment simulator. With this process, the decoder does not need to determine the signal level of the simulation input signal, thereby reducing processing load.

Abstract: A device can receive an audio signal and determine a measure of correlation between the audio signal and a microphone signal. The audio signal can be attenuated based on the measure of correlation. The audio signal can be used to drive one or more speakers of the device. Other aspects are described and claimed.

Abstract: Multi-channel audio signal processing includes receiving a left stereo audio signal from a first channel and a right stereo audio signal from a second channel; up-mixing the left and the right stereo audio signals to generate an up-mixed audio signal for a third channel; de-correlating the up-mixed audio signal from the left and the right stereo audio signals to generate a de-correlated up-mixed audio signal; providing the left and right stereo signals and the de-correlated up-mixed audio signal to first, second, and third loudspeakers respectively to generate first, second, and third sound signals, respectively; picking up the first, second and third sound signals with a microphone to generate a microphone output signal; and adaptively filtering the microphone output signal with an acoustic echo canceller based on the left, the right, and the de-correlated up-mixed audio signal to generate an echo compensated microphone signal.

Abstract: The present disclosure relates to tracking of acoustic feedback events of a hearing assistance device, such as a hearing aid. Information about the acoustic feedback events is stored for analysis. Such information is useful for programming acoustic feedback cancellers and other parameters of a hearing assistance device.

Abstract: A signal processing device comprises: input transducer(s) configured to convert input(s) to an input signal; output transducer(s) configured to convert an output signal to output(s); a signal processing circuit configured to at least subtract a feedback estimation signal from the input signal to produce a feedback compensated signal; and an adaptive feedback estimator.

Abstract: Technology described in this document can be embodied in an earpiece of an active noise reduction (ANR) device. The earpiece includes a plurality of microphones, wherein each of the plurality of microphones is usable for capturing ambient audio to generate input signals for both an ANR mode of operation and a hear-through mode of operation of the ANR device. The earpiece further includes a controller configured to: process a first subset of microphones from the plurality of microphones to generate input signals for the ANR mode of operation, process a second subset of microphones from the plurality of microphones to generate input signals for the hear-through mode of operation, detect that a particular microphone of the second subset is acoustically coupled to an acoustic transducer of the ANR device in the hear-through mode of operation, and in response to the detection, process the input signals from the second subset of microphones without using input signals from the particular microphone.

Abstract: An acquisition system includes a processor, one or more sensors operatively coupled to the processor where the one or more sensors acquire at the ear, on the ear or within an ear canal, one or more of acceleration, blood oxygen saturation, blood pressure or heart-rate, and the one or more sensors configured to monitor a biological state or a physical motion or both for an event. The event can be a detection of a discrepancy when compared with a set of reference data by the one or more sensors or the biological state or the event can be one of a detection of an abrupt movement of a headset operatively coupled to the processor, a change in location of an earpiece operatively coupled the processor, a touching of the headset, a recognizing of a voice command, a starting or ending of a phone call, or a scheduled time.

Abstract: A processing module for a noise control circuit may include a howling detector configured to receive an input signal and to determine a linearity metric based on the input signal, the linearity metric comprising a measure of the linearity of a logarithmic representation of the energy of the input signal. The processing module may also include a gain adjuster configured to adjust the gain of a noise control unit.

Abstract: For many hearing assistive devices, the user's speech is received at a larger amplitude signal than the speech of someone speaking to the user. Since the user's speech is also picked up by the microphone and feed through the speaker causing an acoustic feedback effect, the user may have to constantly adjust the volume of the hearing assistive device to achieve a more comfortable volume based on where the speech is coming from. Therefore, mitigating the acoustic feedback effect of assistive hearing devices can generate a more efficient and comfortable hearing device.

Abstract: Encoding/decoding an audio signal having one or more audio components, wherein each audio component is associated with a spatial location. A first audio signal presentation (z) of the audio components, a first set of transform parameters (w(f)), and signal level data (?2) are encoded and transmitted to the decoder. The decoder uses the first set of transform parameters (w(f)) to form a reconstructed simulation input signal intended for an acoustic environment simulation, and applies a signal level modification (?) to the reconstructed simulation input signal. The signal level modification is based on the signal level data (?2) and data (p2) related to the acoustic environment simulation. The attenuated reconstructed simulation input signal is then processed in an acoustic environment simulator. With this process, the decoder does not need to determine the signal level of the simulation input signal, thereby reducing processing load.

Abstract: A system configured to improve noise cancellation by reducing attenuation of local speech in proximity to a device. When the local speech is present in both a target signal and a reference signal, performing noise cancellation to remove the reference signal inadvertently attenuates the local speech. To prevent this, the system may perform first noise cancellation to identify frequency bands associated with the local speech and may generate a modified reference signal based on the frequency bands. For example, the system may generate the modified reference signal by applying attenuation to first frequencies associated with the local speech and/or gain to second frequencies that are not associated with the local speech. The system may generate final output audio data by performing noise cancellation using the modified reference signal.

Abstract: Example embodiments disclosed herein relate to a estimation of reverberant energy components from audio sources. A method of estimating a reverberant energy component from an active audio source (100) is disclosed. The method comprises determining a correspondence between the active audio source and a plurality of sample sources by comparing one or more spatial features of the active audio source with one or more spatial features of the plurality of sample sources, each of the sample sources being associated with an adaptive filtering model (101); obtaining an adaptive filtering model for the active audio source based on the determined correspondence (102); and estimating the reverberant energy component from the active audio source over time based on the adaptive filtering model (103). Corresponding system (800) and computer program product (900) are also disclosed.

Abstract: Provided is a mobile terminal and method for controlling a sound (audio) to be output, the mobile terminal including at least one application including sound (audio) data and requesting an output sound, a sound (audio) control unit to determine an output sound when a plurality of requests for the output sound is input by a plurality of applications being executed, and a sound (audio) output unit to output the determined sound by forming a virtual output path for each of the sound data to be output, and blocking other virtual output paths aside from a virtual output path mapped to the sound determined to be output by the sound control unit.

Abstract: A howling suppression device includes a first speaker and a second speaker provided in a room, and a sound signal generator configured to supply the first speaker with a first sound signal generated based on an output signal of a microphone, and configured to supply the second speaker with a second sound signal generated so that a part or all of a sound emitted from the first speaker is canceled out at the microphone by a sound emitted from the second speaker based on the output signal of the microphone.

Abstract: A method of operating a hearing system for conducting telephone calls as well as a hearing system capable of performing the proposed method. The hearing system includes a communication device (4) with a communication device microphone (3?) for picking up a first sound signal and providing a primary sidetone signal. The hearing system further includes a hearing device with a hearing device microphone for picking up a second sound signal and providing a secondary sidetone signal. A first amplification is applied to the primary sidetone signal and a second amplification is applied to the secondary sidetone signal. Adjusting of the first amplification is dependent on adjusting of the second amplification, or alternatively, adjusting of the second amplification is dependent on adjusting of the first amplification. The primary and secondary sidetone signals are then combined and subsequently output by a hearing device loudspeaker.

Abstract: A system includes a feedback filter configured to receive an audio signal from a microphone and to process the audio signal based on a feedback path between a loudspeaker and the microphone. The system also includes an adaptive filter coupled to the feedback filter. The system further includes a signal combiner coupled to the microphone and to the adaptive filter.

Abstract: A method which includes receiving a first modulated signal from a first user device, the first modulated signal being generated based upon a time-frequency transformation of first user data. The method further includes receiving a second modulated signal from a second user device, the second modulated signal being generated based upon a time-frequency transformation of second user data. The first modulated signal and the second modulated signal are then processed.

Abstract: A system and method for measuring the performance of a plurality of transducers integrated in one or more loudspeakers is described. The method simultaneously drives each transducer to emit sounds corresponding to distinct orthogonal test signals. A listening device senses sounds produced by the orthogonal test signals and analyzes the sensed audio signal to determine the performance of each transducer. By using orthogonal test signals, the multiple transducers may be measured and/or characterized simultaneously and with limited affect from extraneous noises.

Abstract: A method of generating audio output includes displaying a graphical user interface (GUI) at a user device. The GUI represents an area having multiple regions and multiple audio capture devices are located in the area. The method also includes receiving audio data from the multiple audio capture devices. The method further includes receiving an input indicating a selected region of the multiple regions. The method also includes generating, at the user device, audio output based on audio data from a subset of the multiple audio capture devices. Each audio capture device in the subset is located in the selected region.

Abstract: Active acoustic filter systems and methods are disclosed. An active acoustic filter system includes a memory storing one or more stored sound profiles and a respective set of processing parameters associated with each of the one or more stored sound profiles. A processor coupled to the memory generates processed sound by processing ambient sound in accordance with a set of processing parameters retrieved from the memory based on results from comparing an ambient sound profile to at least one of the one or more stored sound profiles.

Abstract: A method includes obtaining a plurality of audio channels using a plurality of microphone outputs having at least one audio control channel and at least one audio output channel. When a keyword is detected on the audio control channel using voice recognition, adaptive filtering is performed to attenuate the keyword from the audio output channel. An apparatus operative to perform the method includes a plurality of microphones that provide a plurality of audio channels with at least one audio output channel and at least one audio control channel. Voice command recognition logic is operatively coupled to the plurality of microphones to receive the at least one audio control channel. The voice command recognition logic detect keywords on the audio control channel and filter logic with at least one adaptive filter performs adaptive filtering to attenuate the keyword from the at least one audio output channel.

Abstract: Disclosed is a method and system for multi-receiving terminal echo cancellation Echo filters performs echo filtering on receiving terminal signals in M channels to obtain filtered receiving terminal signals in M channels. The filtered receiving terminal signals in M channels are subtracted from a transmitting terminal signal to obtain a system output signal in which receiving terminal echoes have been cancelled. The receiving terminal signals in M channels are buffered. A decorrelation matrix according to each of the receiving terminal signals in M channels is calculated. The decorrelation matrix decomposes the buffered receiving terminal signals in M channels into decorrelated receiving terminal signals in M channels. Update amounts of the echo filters are calculated according to the decorrelation matrix, the decorrelated receiving terminal signals in M channels and the system output signal that is fed back.

Abstract: Example methods and apparatus to verify presentation of media content are disclosed. A disclosed example apparatus for use with a set-top box (STB) and a media presentation device includes an audio input interface to receive a first audio signal associated with a program selected by a user via the STB, a microphone to receive a free-field radiating second audio signal output by at least one of the media presentation device or an audio system associated with the media presentation device, a comparator to compare the second audio signal to the first audio signal to form an output, and an output interface to provide a value indicative of whether the program selected by the user via the STB is presented at the media presentation device.

Abstract: A method, device and computer program product for processing audio signals, the method including determining beamformer filter coefficients to be applied to the audio signals; applying the beamformer filter coefficients to the audio signals; outputting the filtered audio signals from an audio output comprising a plurality of speakers coupled with the device; and receiving at a microphone coupled with the device, the filtered audio signals output from the audio output. The filter coefficients are determined such that the filtered audio signals are suppressed when received at the microphone.

Abstract: A variable-bandwidth adaptive notch filter which cancels howling from an input signal with a bandwidth varying according to a howling frequency to generate an output signal.

Abstract: An acoustic feedback suppression apparatus includes an amplitude measurement section for measuring the amplitude of a sound signal input from a microphone, an amplitude attenuation section for attenuating the amplitude of a sound signal output to a speaker, and an acoustic feedback determination section for determining whether or not acoustic feedback is occurring based on the sound signal input from the microphone. If it is determined that acoustic feedback is occurring, a sound-deadening control section enables sound-deadening control so that the amplitude of the sound signal output to the speaker is attenuated to a sound-deadening level. A sound-deadening release section releases the sound-deadening control if the amplitude of the sound signal input from the microphone is more than specified threshold amplitude and if it is determined that acoustic feedback is not occurring while the sound-deadening control is conducted.

Abstract: A method and audio signal processor for detecting feedback in an electrical amplification system divides a frequency spectrum into a plurality of frequency bands, classifies a tone present in an incoming audio signal into one of the plurality of frequency bands based on a frequency of the tone, selects a threshold value based on the frequency band into which the tone is classified, and disqualifies the tone from potentially being feedback if an amplitude of the tone is greater than an amplitude of a harmonic of the tone by less than the threshold value. The tone is deemed to be potentially feedback if the amplitude of the tone is greater than an amplitude of a plurality of harmonics of the tone by at least the threshold value.

Abstract: Devices and methods are disclosed that allow for selective acoustic near-field nulls for microphone arrays. One embodiment may take the form of an electronic device including a speaker and a microphone array. The microphone array may include a first microphone positioned a first distance from the speaker and a second microphone positioned a second distance from the speaker. The first and second microphones are configured to receive an acoustic signal. The microphone array further includes a complex vector filter coupled to the second microphone. The complex vector filter is applied to an output signal of the second microphone to generate an acoustic sensitivity pattern for the array that provides an acoustic null at the location of the speaker.

Abstract: Provided is an electronic device including: a housing; a vibration body disposed in the housing; and a microphone disposed in the housing. By vibrating the vibration body, the electronic device transmits air conduction sound and vibration sound that is generated by the vibration. During the generation of the sounds, howling reduction processing is performed, so that usability of the electronic device is improved.

Abstract: Disclosed herein is an apparatus of suppressing howling, including: a microphone sensing a voice of a user to generate a voice signal; a memory storing a command for suppressing howling of the voice signal; and a processor suppressing the howling of the voice signal depending on the command, wherein the command includes a command allowing the following steps to be performed: calculating a first ratio, which is a ratio between a power of the current frame signal for the howling candidate frequency and the short-section average, and a second ratio, which is a ratio between the long-section average and the short-section average; determining the howling candidate frequency to be a howling frequency of the current frame signal; and suppressing the howling of the voice signal depending on the howling frequency.

Abstract: A howling canceller applied to an acoustic system having a speaker and a microphone comprises: a filter insertion unit for inserting a notch filter at a frequency of an audio signal picked up by the microphone; a setting unit for setting the insertion time of the notch filter on the basis of the frequency at which the notch filter is inserted; and a releasing unit for, when the insertion time set by the setting unit has elapsed, releasing the notch filter, the insertion time of which has elapsed. The setting unit sets the insertion time of the notch filter to be shorter as the frequency at which the notch filter is inserted increases.

Abstract: A listening device for processing an input sound to an output sound, includes an input transducer for converting an input sound to an electric input signal, an output transducer for converting a processed electric output signal to an output sound, a forward path being defined between the input transducer and the output transducer and including a signal processing unit for processing an input signal in a number of frequency bands and an SBS unit for performing spectral band substitution from one frequency band to another and providing an SBS-processed output signal, and an LG-estimator unit for estimating loop gain in each frequency band thereby identifying plus-bands having an estimated loop gain according to a plus-criterion and minus-bands having an estimated loop gain according to a minus-criterion.

Abstract: A microphone module and method for suppressing feedback in a microphone. The module has a casing with a hollow bore therethrough and a microphone mounted in one end of the bore. The other end of the bore is completely covered by a film mounted onto the top of the casing. The film has at least one slit therethrough in the film portion that covers the other end of the bore. The method includes introducing a sound wave to a film having at least one slit therethrough that separates the film into at least two parts; generating a sound wave from each film part; and conveying the generated sound waves in a sound tube to a microphone as a rejoined sound wave.

Abstract: Method and apparatus for entrainment containment in digital filters using output phase modulation. Phase change is gradually introduced into the acoustic feedback canceller loop to avoid entrainment of the feedback canceller filter. Various embodiments employing different output phase modulation approaches are set forth and time and frequency domain examples are provided. Additional method and apparatus can be found in the specification and as provided by the attached claims and their equivalents.

Abstract: A method is disclosed for acoustic feedback attenuation at a telecommunications terminal. A speakerphone equipped with a loudspeaker and two microphones is featured. Signals from the two microphones are subjected to a calibration stage and then to a runtime stage. The purpose of the calibration stage is to match the microphones to each other by advantageously using both magnitude and phase equalization across the frequency spectrum of the microphones. During the runtime stage, the microphones monitor the ambient sounds received from sound sources, such as the speakerphone's users and the loudspeaker itself, during a conference call. The speakerphone applies the generated set of filter coefficients to the optimized microphone's signals. By combining the signal from the reference microphone with the filtered signal from the optimized microphone, the speakerphone is able to attenuate the sounds from the loudspeaker that would otherwise be transmitted back to other conference call participants.

Abstract: A pseudo noise superimposing unit superimposes a pseudo noise (M-sequence) to an audio signal picked up by a microphone and outputs the superimposed signal to an amplifying system. An calculating unit calculates a correlation value between the audio signal picked up by the microphone and the pseudo noise. The calculating unit estimates a gain of a closed loop based on the correlation value. A gain control unit suppresses a gain of the audio signal based on the estimated gain of the closed loop.

Abstract: Devices and methods are disclosed that allow for selective acoustic near-field nulls for microphone arrays. One embodiment may take the form of an electronic device including a speaker and a microphone array. The microphone array may include a first microphone positioned a first distance from the speaker and a second microphone positioned a second distance from the speaker. The first and second microphones are configured to receive an acoustic signal. The microphone array further includes a complex vector filter coupled to the second microphone. The complex vector filter is applied to an output signal of the second microphone to generate an acoustic sensitivity pattern for the array that provides an acoustic null at the location of the speaker.

Abstract: Systems, methods, and other embodiments associated with selective notch filtering for howling suppression are described. According to one embodiment, an apparatus includes a howling detector that detects howling by performing a time domain analysis of speech signals to identify a speech signal that may be exhibiting howling and performing frequency domain analysis to confirm that the speech signal is exhibiting howling. The apparatus also includes a suppression selector configured to select a suppression technique, a signal processor configured to process the speech signal according to a selected suppression technique. The apparatus outputs, without suppression-related processing, speech signals that are not identified based on the time domain analysis.

Abstract: Methods and systems for masking audio noise are disclosed. One apparatus includes a silence detector configured to detect a period of substantial silence in an audio signal; a masking noise source operably coupled to the silence detector, the masking noise source configured to generate a noise signal in response to the silence detector detecting the period of substantial silence; and at least one combining device operably coupled to the masking noise source, the at least one combining device configured to contribute to combining the audio signal and the noise signal. A method includes detecting a period of substantial silence in an audio signal; and combining masking noise with the audio signal during the period of substantial silence.

Abstract: Acoustic feedback in an audio device is reduced or eliminated via a feedback compensator. The feedback compensator disposes of at least one adaptive filter whose adaptation is governed by a step-size parameter p. A step-size parameter p of the adaptive filter is set at a high level to eliminate acoustic feedback and the parameter p is decreased when a critical excitation signal is detected. This way to proceed allows for efficient elimination of feedback whistling while required environment signals are retained.

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: Provided is a sound emission and collection device capable of estimating the azimuth of a sound source (such as a main utterer) precisely without any processing load. The sound emission and collection device (1) is connected with another sound emission and collection device via a network or the like. The sound emission and collection device (1) receives a sound signal from another sound emission and collection device, as a sound emission signal (FE), and emits the same from a speaker (SP). The sound emission and collection device (1) collects the sound at microphones (MIC1 to MIC3), and produces sound collection beam signals (NE1 to NE3) of different azimuths. The sound emission and collection device down-samples the individual sound collection beam signals (NE1 to NE3), and filters out the echoes of the down-sampled sound collection beam signals (DNE1 to DNE3).