Abstract: A computing device is configured to (i) receive an indication of an operational limit of a given playback device, where the given playback device is included in a group of at least two playback devices that are configured for synchronous playback of audio content, (ii) receive a first audio signal for playback by the group of at least two playback devices, (iii) based on the operational limit of the given playback device, apply a filter to the first audio signal to generate a second audio signal, and (iv) transmit the second audio signal to at least the given playback device for synchronous playback by the group of at least two playback devices.

Abstract: A method performed by a wearable audio output device worn by a user is provided for controlling external noise attenuated by wearable audio output device. A speech is detected from a user wearing the wearable audio output device, wherein the audio output device has active noise reduction turned on. It is determined, based on the detecting, that the user desires to speak to a subject in the vicinity of the user. In response to the determining, a level of noise reduction is reduced to enable the user to hear sounds external to the audio output device. It is determined that the user desires to speak to the subject by detecting at least one condition of a plurality of conditions.

Abstract: An air-pulse generating device includes a film structure including a flap pair. The film structure is actuated to perform a common mode movement, so as to form an amplitude-modulated ultrasonic air pressure variation with an ultrasonic carrier frequency. The film structure is actuated to perform a differential mode movement, so as to form an opening at a rate synchronous with the ultrasonic carrier frequency. The air-pulse generating device produces a plurality of air pulses according to the amplitude-modulated ultrasonic air pressure variation.

Abstract: One aspect of the present disclosure relates to a noise masking method through variable masking sound level conversion, capable of adjusting an interval for changing a level of a masking sound for masking noise to an optimal interval so that the masking sound is not perceived as noise, by determining a fluctuation trend of a level of audible noise and adjusting the interval for changing the level of the masking sound for masking the noise into an optimal environment through a control unit.

Abstract: An communication system supports communication paths within an environment by receiving speech signals of a speaker and playing it back for one or more listeners. Signal processing tasks are split into a microphone related part and into a loudspeaker related part. A sound processing system suitable for use in an environment having multiple acoustic zones includes a plurality of microphone communication instances coupled and a plurality of loudspeaker instances.

Abstract: Techniques are described for detecting and correcting mismatched microphone sensitivities in a microphone array without knowledge of the acoustic excitation source(s). Mismatch is detectable based on time and/or frequency domain analysis of each microphone's long term exposure to a real-world sound field that includes an acoustic source and a non-acoustic source, and is corrected by adjusting the amount of amplification applied to at least one microphone signal. In the time domain, sensitivity matching can be performed by using an average of all microphone signals as a reference signal. In some embodiments, the reference signal is the root mean square of the average. Alternatively, a single microphone can be selected as a reference. In some embodiments, sensitivity mismatch is detected and corrected at specific frequencies based on comparing frequency components of amplified microphone signals. Sensitivity matching can be repeated to ensure that the microphones remain sensitivity-matched over time.

Abstract: Embodiments included herein generally relate to managing low frequencies of an output signal. For example, a method may include: measuring a sound pressure level (SPL) of a speaker; dynamically selecting a cutoff frequency for operating a first filter configured to provide a first bass component of a bass element of an output signal and a second filter configured to provide a second bass component of the bass element; providing the cutoff frequency to the first filter to configure the first filter to generate the first bass component based on the cutoff frequency; providing the cutoff frequency to the second filter to configure the second filter to generate the second bass component based on the cutoff frequency; and generating an output signal based on at least one of the first and second bass components.

Abstract: A method for audio signal processing of microphone signals of a headphone. An audio signal from a first microphone of a headphone is filtered by an ANC system to produce a first filtered signal. Dynamic range control is performed upon the first filtered signal to produce a first dynamic range adjusted signal that drives a speaker of the headphone. Other aspects are also described and claimed.

Abstract: A method and system of estimating human perception of audibility of audio alerts in the presence of background noise. In the audibility estimation system, a microphone generates an input signal corresponding to an audio alert. A processor receives the input signal and generates an audibility metric representing human perception of audibility of the audio alert based on a comparison between a background noise estimate and an audio alert estimate, and causes an action to be taken based on the audibility metric.

Abstract: Various methods and systems are related to pulse-based automatic gain control. In one example, pulse-based automatic gain control (AGC) includes a variable gain amplifier (VGA) configured to amplify an analog input signal to generate an analog output signal based upon an amplification control signal; an integrate-and-fire (IF) sampler configured to generate a pulse train corresponding to the analog output signal; and a gain adjustment configured to generate the amplification control signal based upon a comparison of time between pulses of the pulse train to a reference time. In another example, a method includes determining time between pulses of a pulse train corresponding to an analog output signal from a VGA; generating an amplification control signal based upon a comparison of the time between pulses of the pulse train to a reference time; and adjusting amplification of the VGA in response to the amplification control signal.

Abstract: A display apparatus includes a display panel configured to display a first image in a first direction and configured to display a second image in a second, different direction, a sound output unit including sound output modules to output a sound, and a sound focusing unit. The sound focusing unit is configured to receive first and second audio signals synchronized with video data signals of the first image and the second image, apply a direction focusing weight to the first and second audio signals, and to provide the sound output modules with a sound signal having the direction focusing weight applied thereto. Furthermore, the sound output unit is configured to receive the sound signal corresponding to the sound output modules, to output a first sound corresponding to the first image in the first direction, and to output a second sound corresponding to the second image in the second direction.

Abstract: There is provided an audio processing device including a memory, and a processor coupled to the memory and the processor configured to detect a first acoustic feature amount and a second acoustic feature amount of an input audio, calculate a coefficient for the second acoustic feature amount based on a time change amount by calculating the time change amount of the first acoustic feature amount, and calculate a statistical amount for the second acoustic feature amount based on the coefficient.

Abstract: Systems and methods to provide distortion sensing, prevention, and/or distortion-aware bass enhancement in audio systems can be implemented in a variety of applications. Sensing circuitry can generate statistics based on an input signal received for which an acoustic output is generated. In various embodiments, the statistics can be used such that a multi-notch filter can be used to provide input to a speaker to generate the acoustic output. In various embodiments, the statistics from the sensing circuitry can be provided to a bass parameter controller coupled to bass enhancement circuitry to operatively provide parameters to the bass enhancement circuitry. The bass enhancement circuitry can provide a bass enhanced signal for generation of the acoustic output, based on the parameters. Various combinations of a multi-notch filter and bass enhancement circuitry using statistics from sensing circuitry can be implemented to provide an enhanced acoustic output.

Abstract: A solid-state semiconductor guitar amplifier system/method mimicking the audio performance characteristics of conventional vacuum tube guitar amplifiers is disclosed. The disclosed system/method incorporates solid-state semiconductor circuitry implementing an input audio preamplifier (IAP) having asymmetric gain control (AGC) that feeds wave shape transformer (WST) circuitry implementing a piecewise/diode breakpoint (PDB) transform that emulates a conventional vacuum tube voltage-current transfer (VIT) characteristic. A breakpoint threshold controller (BTC) determines the offset associated with application of the PDB operation to the audio signal.

Abstract: Systems and methods to provide distortion sensing, prevention, and/or distortion-aware bass enhancement in audio systems can be implemented in a variety of applications. Sensing circuitry can generate statistics based on an input signal received for which an acoustic output is generated. In various embodiments, the statistics can be used such that a multi-notch filter can be used to provide input to a speaker to generate the acoustic output. In various embodiments, the statistics from the sensing circuitry can be provided to a bass parameter controller coupled to bass enhancement circuitry to operatively provide parameters to the bass enhancement circuitry. The bass enhancement circuitry can provide a bass enhanced signal for generation of the acoustic output, based on the parameters. Various combinations of a multi-notch filter and bass enhancement circuitry using statistics from sensing circuitry can be implemented to provide an enhanced acoustic output.

Abstract: Systems and methods to provide distortion sensing, prevention, and/or distortion-aware bass enhancement in audio systems can be implemented in a variety of applications. Sensing circuitry can generate statistics based on an input signal received for which an acoustic output is generated. In various embodiments, the statistics can be used such that a multi-notch filter can be used to provide input to a speaker to generate the acoustic output. In various embodiments, the statistics from the sensing circuitry can be provided to a bass parameter controller coupled to bass enhancement circuitry to operatively provide parameters to the bass enhancement circuitry. The bass enhancement circuitry can provide a bass enhanced signal for generation of the acoustic output, based on the parameters. Various combinations of a multi-notch filter and bass enhancement circuitry using statistics from sensing circuitry can be implemented to provide an enhanced acoustic output.

Abstract: An apparatus comprising at least one processor and at least one memory including computer program code. The at least one memory and the computer program code is configured to, with the at least one processor, cause the apparatus at least to determine a loudness estimate of a first audio signal, generate a parameter dependent on the loudness estimate; and control the first audio signal dependent on the parameter.

Abstract: A method of processing an audio signal is provided. The method includes acquiring location information and performance information of a speaker configured to output an audio signal, selecting a frequency band based on the location information, determining a section to be strengthened from the selected frequency band with respect to the audio signal based on the performance information, and applying a gain value to the determined section.

Abstract: A sound signal processing apparatus includes: first amplifier which amplifies a sound signal, depending on an input level of the sound signal; and second amplifier which amplifies with a gain the sound signal amplified by first amplifier. Second amplifier automatically decreases the gain when a level of the sound signal is greater than an upper limit threshold, and second amplifier automatically increases the gain when the level of the sound signal is lower than the upper limit threshold and when the gain is lower than a predetermined value. When second amplifier increases the gain, second amplifier changes the gain at a first speed when a user has performed an operation to decrease the input level, and second amplifier changes the gain at a second speed lower than the first speed when the user has not performed the operation to decrease the input level.

Abstract: An audio system receives an input signal and, if the input signal has a sparse representation in the frequency domain, comprising components at at least one frequency of interest, the input signal is filtered in at least one band pass filter, such that only components at the or each frequency of interest are passed to an output. The operation is able in some case to analyze the input signal to determine whether the input signal has a sparse representation in the frequency domain.

Abstract: A speaker system featuring automatic volume adjustment to avoid excess distortion includes a first converter for converting an input signal to an input frequency domain signal; and a second converter for converting a microphone signal to an environmental frequency domain signal. A distortion analyzer receives output of the first and second converter and determines at least one distortion level and provides at least one gain adjustment signal used to control a corrector to adjust gain of a speaker driver. The system performs a method of reducing distortion by automatically adjusting volume in each of a plurality of frequency bands including converting an input signal to a frequency domain input signal; converting a microphone signal to an environmental frequency domain signal; determining a distortion level; generating a gain adjustment signal; and adjusting gain of a signal provided to a speaker driver.

Abstract: Signal processing methods for predicting the intelligibility of speech, e.g., in the form of an index that correlate highly with the fraction of words that an average listener (amongst a group of listeners with similar hearing profiles) would be able to understand from some speech material are proposed. Specifically, solutions to the problem of predicting the intelligibility of speech signals, which are distorted, e.g., by noise or reverberation, and which might have been passed through some signal processing device, e.g., a hearing aid are described. In summary, the disclosure present solutions to the following problems: 1. Monaural, non-intrusive intelligibility prediction of noisy/processed speech signals 2. Binaural, non-intrusive intelligibility prediction of noisy/processed speech signals 3. Monaural and binaural intelligibility enhancement of noisy speech signals.

Abstract: A digital-to-analog converter (DAC) circuit includes a pair of output stages, each including a DAC configured to convert a digital audio signal into an analog audio signal. A low-pass filter circuit includes an operational amplifier in signal communication with the DAC. The operation amplifier generates a filtered analog signal based on the analog audio signal. An amplifier network generates an amplified audio signal based on the filtered analog signal. The operational amplifier includes a feedback circuit path including a first node connected to the output of the amplifier network and a second node connected to the input of the operational amplifier. The amplifier network is electrically nested in the feedback circuit path.

Abstract: A volume controller may maintain a volume mapping including a plurality of zones of volume level, each zone defined according to a range of included volume levels and specifying a volume ramp up control rate and a volume ramp down control rate for adjustment of volume levels within the range, determine, according to the volume mapping based on a current volume level and a direction of a requested volume change, a step value for adjusting the current volume level, and adjust the current volume level according to the step value.

Abstract: A first fixed filter extracts high-frequency and low-frequency sound signals outside the audible band, and a level integrator integrates levels of the sound signals to obtain a first integrated value. With a second fixed filter, the level integrator integrates levels of sound signals in a low frequency band within the audible band to obtain a second integrated value. A control unit determines whether the high-frequency sound signals are dominant by comparing the first integrated value with the second integrated value. When the high-frequency sound signals are dominant, the control unit attenuates high-frequency speaker drive signals using a variable filter.

Abstract: A down-mixing method of 5.1 audio input channels for two channel replay by DRC processing of the LFE and the LS and RS channel signal before mixing, and by filtering LS and RS channels with BRTFs measured from placing a loud speaker at a corner of a room and measuring head at the diagonal corner of the room.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: A device includes circuitry configured to determine noise shaping parameters for one or more transmit signals on one or more transmit channels based on transmission protocol and spectral mask criteria. Frequency positions are identified for the one or more transmit channels based on the noise shaping parameters, and the noise shaping parameters are applied to the one or more transmit signals.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: A method of allocating a fixed resource includes displaying an interface control in a user interface display. The interface control includes at least three continuous value control elements associated with a group including at least three categories. Each continuous value control element is associated with a value associated with a category of the group. The operator of the values associated with the at least three continuous value control elements is equal to a fixed quantity. In response to a first user input, a first continuous value control element is locked to fix the value associated with the first continuous value control element. In response to a second user input, a second continuous value control element is adjusted to change the value associated with the second continuous value control element.

Abstract: A method of encoding a time-domain audio signal is presented. A device transforms the time-domain signal into a frequency-domain signal including a sequence of sample blocks, wherein each block includes a coefficient for each of multiple frequencies. The coefficients of each block are grouped into frequency bands. For each frequency band of each block, a scale factor is estimated for the band, and the energy of the band for the block is compared with the energy of the band of an adjacent sample block, wherein the blocks may be adjacent to each other in either or both of an interchannel and a temporal sense. If the ratio of the band energy for the first block to the band energy for the adjacent block is less than some value, the scale factor of the band for the first block is increased. The coefficients of the band for each block are quantized based on the resulting scale factor. The encoded audio signal is generated based on the quantized coefficients and the scale factors.

Abstract: A method (100) for matching characteristics of two or more transducer systems (202, 208). The method involving: receiving input signals from a set of said transducer systems; determining if the input signals contain a pre-defined portion of a common signal which is the same at all of said transducer systems; and balancing the characteristics of the transducer systems when it is determined that the input signals contain the pre-determined portion of the common signal.

Abstract: Equalizer controller and controlling method are disclosed. In one embodiment, an equalizer controller includes an audio classifier for identifying the audio type of an audio signal in real time; and an adjusting unit for adjusting an equalizer in a continuous manner based on the confidence value of the audio type as identified.

Abstract: The technology disclosed can provide capabilities such as using vibrational sensors and/or other types of sensors coupled to a motion-capture system to monitor contact with a surface that a user can touch. A virtual device can be projected onto at least a portion of the surface. Location information of a user contact with the surface is determined based at least in part upon vibrations produced by the contact. Control information is communicated to a system based in part on a combination of the location on the surface portion of the virtual device and the detected location information of the user contact. The virtual device experience can be augmented in some implementations by the addition of haptic, audio and/or visual projectors.

Abstract: An audio processor has an active noise cancellation (ANC) processor to compute an S-filter, which estimates a transfer function from an earpiece speaker, that is to be worn by a user, to an error microphone. The ANC processor adaptively produces an anti noise signal for reducing ambient noise that can be heard by the user. An adaptive equalization (EQ) processor has an EQ filter controller that adjusts an EQ filter, while audio user content is being filtered in accordance with the EQ filter before being fed to the speaker and then heard by the user. Other embodiments are also described and claimed.

Abstract: A system or method for controlling speaker acoustic output in a multi-speaker audio system having multiple listening zones include substantially simultaneously adjusting gain and at least one additional parameter, such as filtering and/or other signal processing parameters of at least a first speaker relative to at least a second speaker in response to a change in balance and/or fade settings of the audio system across a range of balance and fade settings except for a maximum or minimum setting, and muting at least one speaker in response to the maximum or minimum balance or fade setting.

Abstract: A method for reducing parasitic vibrations of a loudspeaker environment while maintaining the perception of the low frequencies of an original electric sound signal for broadcast after processing by the loudspeaker having a cut-off frequency. A vibration frequency band is identified that causes the loudspeaker to vibrate. A low frequency band of the original sound signal having a frequency close to the cut-off frequency of the loudspeaker is identified as an upper limit. At least one harmonic signal from the isolated low frequency band of the original sound signal is generated. The original sound signal and the harmonic signal are combined to obtain a recombined signal. The vibration frequency band is removed from the recombined signal to obtain a signal for broadcast by the loudspeaker.

Abstract: Audio systems for a vehicle and methods of operating an audio device in a vehicle cabin are provided. A method of operating an audio device directs an acoustic signal to at least one vehicle cabin receiver (VCR) of the audio device; measures sound pressure levels (SPLVCM) for acoustic energy received by at least one vehicle cabin microphone (VCM) of the audio device during a time period ?t; calculates a sound pressure level (SPL) dose (SPL_Dose?t) during the time period ?t using the sound pressure levels SPLVCM of the at least one VCM; combines the SPL dose (SPL_Dose?t) with an SPL dose of a previous time period to form a total SPL dose (SPL_Dosetotal); and performs an action to modify operation of the audio device when the total SPL dose is greater than a predetermined threshold.

Abstract: In certain embodiments, systems for receiving one or more echoes are provided. The system comprises a first attenuator, a first amplifier, and a second attenuator. The first attenuator is configured to receive the one or more echo signals, and generate a corresponding set of first attenuated echo signals, respectively, based on a number of signal strengths of the one or more echo signals. The first amplifier is configured to receive and amplify the set of first attenuated echo signals to thereby generate a set of first amplified echo signals corresponding to the one or more first attenuated echo signals, respectively. The second attenuator is configured to receive the set of first amplified echo signals and generate a set of second attenuated echo signals corresponding to the set of first amplified echo signals, respectively, based on a number of signal strengths of the set of first amplified echo signals, respectively.

Abstract: A selector selects an analog audio signal input to one input port from among analog audio signals input to multiple input ports according to an instruction from the user. An analog gain control circuit amplifies the analog audio signal received from the selector, with a corresponding one of the gains set for the respective input ports. An analog gain control circuit is configured to gradually change its gain when the gain is switched. An A/D converter converts an output signal of the analog gain control circuit into a digital audio signal. A first audio signal processing circuit is monolithically integrated on a signal semiconductor substrate.

Abstract: A volume amplitude limiting device includes a volume adjustment unit, an amplitude limiting unit, and a control unit. The volume adjustment unit adjusts the amplitude of an input sound signal to a larger degree as the adjustment value increases and outputs the resulting sound signal. The amplitude limiting unit limits the amplitude of the sound signal output from the volume adjustment unit to an output limitation value and outputs the resulting signal to a speaker. The control unit sets an adjustment value for the volume adjustment unit and also sets an output limitation value for the amplitude limiting unit according to an adjustment value.

Abstract: Sound pressure level limiter with anti-startle feature for audio systems are disclosed. The anti-startle feature may be implemented with a delta incident detector for detecting delta acoustic incidents that exceed a predetermined acoustic startle boundary, a delta limiter for determining an anti-startle gain, and an amplifier to apply the anti-startle gain to the input signal. The delta incident detector may detect delta incidents based on an estimated true SPL delivered by a transducer to a predetermined datum point. The estimated true SPL may be determined by based on a measured receiving frequency response of the transducer. An SPL limiter may also determine an SPL gain in response to detecting an SPL acoustic incident that exceeds a predetermined SPL threshold, and an amplifier may apply the SPL gain to the input signal to reduce it below the threshold.