Abstract: A signal processing apparatus includes a control section, a signal processing section connected with a plurality of signal processing elements and performs signal processing for enhancing or attenuating an input signal in a specific frequency band, and a crossfade signal section including a crossfade signal processing element capable of replacing at least one of the signal processing elements, wherein the control section controls any one of the signal processing elements among the plurality of signal processing elements, and the crossfade signal processing element, to crossfade to the crossfade signal processing element having the signal processing element as a new characteristic, to perform processing for replacing any one of the signal processing elements by the crossfade signal processing element, and to perform the processing on remaining signal processing elements of the plurality of signal processing elements in the signal processing section.

Abstract: An input audio signal is separated into input audio signal components (X(k,b)). A set of two or more band branches provides output audio signal components (Y(k,b)). The set of band branches comprises one or more compressor branches Each compressor branch compresses a respective input audio signal component (X(k,b)) into a respective output audio signal component (Y(k,b)). A summed audio signal (y(t)) is generated by summing the output audio signal components (Y(k,b)). A residual audio signal (v(t)) is a difference between the input audio signal and the summed audio signal(y(t)) A virtual bass signal (w(t)) comprises one or more harmonics of the residual audio signal (v(t)). An output audio signal is generated by summing the summed audio signal (y(t)) and the virtual bass signal (w(t)).

Abstract: A sound source separation apparatus includes: a separation-matrix processor that transforms a plurality of observation signals corresponding to sounds being propagated from a plurality of sound sources into a frequency-domain signal group the separation-matrix processor updating a separation matrix based on the frequency-domain signal group and transforming the updated separation matrix into time-series filter coefficients to output; a filter-coefficient transformer that partially removes non-causal components from the filter coefficients to transform the filter coefficients, and a separator that supplies the filter coefficients to a filter group, the separator generating a plurality of separation signals separated from the plurality of observation signals corresponding to the separation matrix.

Abstract: Example techniques may involve calibration with multiple recording devices. An implementation may include detecting, via a microphone, one or more calibration sounds as emitted by one or more playback devices of one or more zones during a calibration sequence. The implementation may further include determining a first response, the first response representing a response of a given environment to the one or more calibration sounds as detected by the first recording device and receiving data indicating a second response, the second response representing a response of the given environment to the one or more calibration sounds as detected by a second recording device. The implementation may also include determining a calibration for the one or more playback devices based on the first response and the second response and sending, to the one or more zones, an instruction that applies the calibration to playback by the one or more playback devices.

Abstract: A remote speaker microphone with touch volume control is provided. The remote speaker microphone includes a housing; a microphone; a speaker; and an interface configured to connect the speaker and the microphone. The remote speaker microphone further includes a touch sensor configured to, in response to receiving touch input, control a volume level of the speaker from a first volume level to a lower volume level, the first volume level being one of a plurality of discrete volume levels.

Abstract: Methods and systems are provided for adjusting a crossover frequency between a plurality of audio speakers rendering audio content. In one example, a first subset of a plurality of audio speakers may be rendering a first sub-range of a range of audio frequencies of an audio content, and a second subset of speakers of the plurality of audio speakers may be rendering a second sub-range of the range of audio frequencies. In this example, the first sub-range and the second sub-range may be substantially separated at the crossover frequency. In one case, a characteristic of the audio content may be determined, and the crossover frequency may be adjusted based on the determined characteristic to help improve the audio content rendering quality by the respective subsets of audio speakers in the plurality of audio speakers.

Abstract: Systems and methods for calibrating a playback device include (i) outputting first audio content; (ii) capturing audio data representing reflections of the first audio content within a room in which the playback device is located; (iii) based on the captured audio data, determining an acoustic response of the room; (iv) connecting to a database comprising a plurality of sets of stored audio calibration settings, each set associated with a respective stored acoustic room response of a plurality of stored acoustic room responses; (v) querying the database for a stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located; and (vi) applying to the playback device a particular set of stored audio calibration settings associated with the stored acoustic room response that corresponds to the determined acoustic response of the room in which the playback device is located.

Abstract: The reliability and resilience of a speaker in a portable device may be improved by enabling the device to determine a status of an enclosure of a transducer. The impedance profiles of transducers vary depending on the enclosure status, whether the enclosure is leaky, ported, or sealed. Determination of parameters related to an impedance of a transducer may aid in a determination of the status of the enclosure. Once an enclosure status is determined, that information may be used to determine an excursion model for the transducer. Adapting the excursion model or other audio processing parameters can improve protection and performance of the transducer.

Abstract: A microphone assembly includes an acoustic transducer element configured to convert sound into a microphone signal in accordance with a transducer frequency response including a first highpass cut-off frequency. The microphone assembly additionally includes a processing circuit including a signal amplification path configured to receive, sample and digitize the microphone signal to provide a digital microphone signal. A frequency response of the signal amplification path includes a second highpass cut-off frequency which is higher than the first highpass cut-off frequency of the acoustic transducer element.

Abstract: A wearable includes an article of clothing, an intelligent control and a plurality of sets of vibration motors operatively connected to the intelligent control and positioned on the article of clothing to provide tactile feedback wherein each of the sets of vibration motors is associated with a different frequency range of audio which may be communicated to the wearable via Bluetooth or otherwise.

Abstract: A voltage regulator system, comprising: a switched capacitor (SC) regulator that operates at a switching frequency and receives an input voltage; and a controller configured to control an operation of the SC regulator by adjusting the switching frequency of the SC regulator based on efficiency. In some embodiments, the switching frequency is swept to determine a best efficiency. In some embodiments, the switching frequency is swept at each of a plurality of values for the input voltage. In some embodiments, the system includes further one or more switches in series with the SC regulator. In some embodiments, the SC regulator includes an output terminal that is coupled to a battery.

Abstract: A group of devices connected on a network operate in conjunction with one another to present content. Individual devices may be wirelessly connected to presentation devices such as speakers. Skew is a delay between when content is sent to the presentation device and then output. Drift is a difference in timing of output between two or more presentation devices not synchronized to a common clock. Non-zero values of skew and drift may result in poor content presentation by the group. An actual rate at which data is transferred to a presentation device is measured and compared to an expected rate. This difference is analyzed to determine a drift value. The content to be delivered to the presentation device is resampled. During presentation the combination of the drift and the resampled content reduces skew. The drift value may also be used to lookup a skew value used to delay initial presentation.

Abstract: Method for determining distance to target using a multitone nonlinear radar system comprising providing a transmitter that transmits a signal comprising at least two predetermined frequency components; receiving transmitted signal upon reflection from target; determining the phase relationships of the frequency components when signal strikes target; determining distance the signal has travelled to target based upon the phase relationship of the frequency signal components at the time of reflection from target; computing the distance to target.

Abstract: A computerized method for filtering a digital audio file to generate an output audio file that induces optimal health and cognitive ability in a listener of a playback of the output audio file is described herein. The method includes the steps of identifying a plurality of target frequencies that span within an octave, identifying a plurality of mid-point frequencies that are situated at mid-points between any two adjacent target frequencies, applying a peaking filter to the digital audio file centered around the plurality of mid-point frequencies to produce highest frequency attenuation at the plurality of mid-point frequencies, and generating the output audio file.

Abstract: A system and method automates the tuning of one or more multiband compressors or multiband limiters to minimize loudspeaker distortion. The system and method render one or more test loudspeaker signals that vary in frequency and in amplitude out of a loudspeaker and records the responses of the loudspeaker at a microphone. The system and method measure a distortion of the loudspeaker, relative to a frequency and an amplitude, with respect to the one or more test loudspeaker signals and a microphone signal and calculate tunable parameters of the multiband compressor that includes cutoff frequencies in response to the measured distortion.

Abstract: A method, and a corresponding apparatus, for processing an input signal comprise filtering the input signal to separate a passband frequency component of the input signal from a stopband frequency component of the input signal, and adjusting relative signal power values of the passband frequency component and the stopband frequency component of the input signal based at least in part on signal values of a number of samples associated with the input signal. In the case of audio signals, for example, such processing is used for spectral expansion of the input signal by enhancing the power of the stopband, or low and high frequencies, component with respect to the power of the passband component of the input signal. As a result, a better audio quality is achieved.

Abstract: A method for characterizing a loudspeaker comprises: filtering an input test signal by applying the inverse of a linear model describing an excursion of the loudspeaker in response to a given input signal, the linear model containing only linear terms; applying the filtered input test signal to the loudspeaker; measuring the excursion of the loudspeaker in response to the filtered input test signal; and, based on the measured excursion, determining one or more non-linear parameters for a non-linear model describing the excursion of the loudspeaker in response to a given input signal.

Abstract: Embodiments of the invention provide methods and apparatus for processing audio input signals, for example, so as to linearize the output of a given loudspeaker. An audio processor is provided, for modifying an audio signal to be provided to a loudspeaker, the audio processor comprising: a first filter stage, for applying to an audio signal a linear model describing an excursion of the loudspeaker in response to a given input signal, the linear model containing only linear terms, and for generating one or more excursion signals; a plurality of second filters for receiving the one or more excursion signals, each of the second filters configured to apply to a respective one of a plurality of frequency bands in the one or more excursion signals the inverse of a model describing an excursion of the loudspeaker in response to a given input signal; and a combiner for combining the outputs of each of the plurality of second filters.

Abstract: Among other things, a sound processing device system is disclosed to assist a hearing-impaired human listener recognize speech sounds or phonemes. The device system may be configured at least to generate an output audio signal at least by transposing and causing a negative rank ordering of frequency of at least a portion of the input audio signal. Compression also may be performed on the at least the portion of the input audio signal as part of generating the output audio signal. The negative rank ordering may be performed on a high-frequency portion of the input audio signal that becomes a low-frequency portion of the output audio signal by the transposing. The low-frequency portion of the output audio signal may represent an inverted ordering of frequencies or frequency segments present in the high-frequency portion of the input audio signal.

Abstract: The present invention provides methods and systems for digital processing of an input audio signal. Specifically, the present invention includes a high pass filter configured to filter the input audio signal to create a high pass signal. A first filter module then filters the high pass signal to create a first filtered signal. A first compressor modulates the first filtered signal to create a modulated signal. A second filter module then filters the modulated signal to create a second filtered signal. The second filtered signal is processed by a first processing module. A band splitter splits the processed signal into low band, mid band, and high band signals. The low band and high band signals are modulated by respective compressors. A second processing module further processes the modulated low band, mid band, and modulated high band signals to create an output signal.

Abstract: An embodiment of the invention provides a method for monitoring for and detecting a leak in a line where a mobile detector having an audio sensor and an odor sensor transiting along the line. A sample audio reading is obtained by the audio sensor; and, a sample odor reading is obtained by the odor sensor. A processor compares the sample audio reading to a baseline audio reading to determine whether the line is leaking. The processor compares the sample odor reading to a baseline odor reading to determine whether the line is leaking. The location of the mobile detector is identified by a GPS device on the mobile detector when the processor determines that the line is leaking. The location of the mobile detector, the sample audio reading, and the sample odor reading is sent to a central monitoring center by a communications device on the mobile detector.

Abstract: Example embodiments disclosed herein relate to modelling a frequency response characteristic of an electro-acoustic transducer. A method includes obtaining at least one measurement of the frequency response characteristic for at least one electro-acoustic transducer of the category. A model of a frequency response characteristic specific to a category of electro-acoustic transducers is generated at least in part based on perceptual importance of a frequency band, an averaged, normalized or microphone compensated measurement such that the distortion of the model is optimized. A further method for estimating a frequency response characteristic of an electro-acoustic transducer is based on the generated model and the sensitivity of the electro-acoustic transducer or headphone. Corresponding system and computer program product are also disclosed.

Abstract: A presentation moderator assembly for facilitating nonverbal and nonphysical communication between a moderator and a presenter includes a base unit that is positioned in view of a presenter. The base unit has a first light emitter that is selectively illuminated to communicate to the presenter that the presenter may begin speaking. The base unit has a second light emitter that is selectively illuminated to communicate to the presenter that the presenter should begin finishing speaking. The base unit has a third light emitter that is selectively illuminated to communicate to the presenter to stop speaking. A remote unit is carried by a moderator and the remote unit is in electrical communication with the base unit. The remote unit selectively turns on each of the first, second and third light emitters.

Abstract: An active sound management system comprises a transducer that senses an actual sound or vibration from a sound generating source and generates a transducer signal in response to sensing the actual sound or vibration; a harmonic extractor device that extracts a plurality of harmonics from the transducer signal; and a harmonic modifier device that adjusts a feature of the extracted harmonic to be within a predetermined threshold with respect to a target harmonic corresponding to a desired sound.

Abstract: This invention describes methods of Audio Signal Personalized Enhancement for optimal listening to music. A combination of hearing test, audio signal compensation, and signal processing algorithms are implemented jointly in an audio Device. An audio signal is processed and tuned within the entire range of audible frequencies to actively compensate for both the characteristics of the listening device, and the specific hearing characteristics of the listener, resulting in a much improved music listening experience. A randomized self-test algorithm is used to analyze user's hearing ability in each ear within a preselected dynamic volume range at multiple frequencies spanning the audible spectrum.

Abstract: Systems and methods for handling silence in audio streams are disclosed. In one aspect, a transmitter detects a halt in an audio stream. After detection of the halt in the audio stream, the transmitter embeds a silence signal into the audio stream and transmits the silence signal to associated receivers. The associated receivers may respond to the embedded silence signal by “playing” silence or by using the silence signal to activate a silence protocol. In either event, the associated receivers do not receive the original audio halt and do not produce an unwanted audio artifact.

Abstract: The present invention relates to a device for controlling a loudspeaker (14) in an enclosure, comprising: an input for an audio signal (Saudio_ref) to be reproduced; an output for supplying an excitation signal from the loudspeaker. It comprises a control unit comprising: means (24, 25) for calculating a desired dynamic value (Aref) of the loudspeaker diaphragm based on the audio signal (Saudio_ref) to be reproduced and the structure of the enclosure; means (26) for calculating a plurality of desired dynamic values (Aref, dAref/dt, Vref, Xref) of the loudspeaker diaphragm at each moment based on only the desired dynamic value (Aref); a mechanical model (36) of the loudspeaker; and means (70, 80, 90) for calculating the excitation signal of the loudspeaker at each moment, without feedback loop, from the mechanical model (36) of the loudspeaker and desired dynamic values (Aref, dAref/dt, Vref, Xref).

Abstract: An estimate of a fractional chirp rate of a signal may be computed by using multiple frequency representations of the signal. A first frequency representation may be computed using a first fractional chirp rate and a first score may be computed using the first frequency representation that indicates a match between the first fractional chirp rate and a fractional chirp rate of the signal. A second frequency representation may be computed using a second fractional chirp rate and a second score may be computed using the second frequency representation that indicates a match between the second fractional chirp rate and the fractional chirp rate of the signal. The fractional chirp rate of the signal may be estimated using the first score and the second score, for example, by selecting a fractional chirp rate corresponding to a highest score.

Abstract: Systems and methods for handling silence in audio streams are disclosed. In one aspect, a transmitter detects a halt in an audio stream. After detection of the halt in the audio stream, the transmitter embeds a silence signal into the audio stream and transmits the silence signal to associated receivers. The associated receivers may respond to the embedded silence signal by “playing” silence or by using the silence signal to activate a silence protocol. In either event, the associated receivers do not receive the original audio halt and do not produce an unwanted audio artifact.

Abstract: A multi-band limiter, a sound recording apparatus, and a program are provided having a multi-band limit function with a flat frequency characteristic while removing a dead zone of a multi-band limiter. A cross-over filter unit divides an input sound signal into a low-band component, a middle-band component, and a high-band component so that the adjacent bands cross at ?6 dB.

Abstract: Embodiments of systems and methods are described for reducing undesired leakage energy produced by a non-front-facing speaker in a multi-speaker system. For example, the multi-speaker system can include an array of forward-facing speakers, one or more upward-facing speakers, and/or one or more side-facing speakers. Filters coupled to any two of the speakers in the multi-speaker system can generate audio signals output by the coupled speakers to reduce, attenuate, or cancel a portion of an audio signal output by one or more non-front-facing speakers that acoustically propagates along a direct path from the respective non-front-facing speaker to a listening position in a listening area in front of the multi-speaker system.

Abstract: The present invention provides a vibration audio system for transmitting an audio signal outputted from a sound source to a listener in the form of vibration while reducing output level of the signal and power consumption. The system includes an envelope detection unit (204) for detecting an envelope signal of the audio signal outputted from a sound source, a vibration transmission member for allowing the listener to perceive vibration of a low-frequency sound outputted from a low-frequency output speaker that outputs audio signals, and a frequency conversion unit (205) for generating an audio signal frequency-converted on the basis of resonant frequencies by multiplying the envelope signal by sine waves having the same frequencies as resonance frequencies obtained from an impulse response of the low-frequency output speaker disposed in the vibration transmission member. The audio signal frequency-converted by the frequency conversion unit (205) is outputted from the low-frequency output speaker.

Abstract: The invention relates to amplifiers of audio power which operate with loudspeaker systems. The main technical result is significant extension of the reproducible frequency range without reduction of the loudspeaker system fundamental resonance frequency. This result can be obtained by means of adding only one low power resistor and one low capacitance capacitor to a typical power amplifier. In the described embodiment this is achieved by means of a rise of the amplifier FR at low frequencies using positive feedback and the quasi-resonance effect.

Abstract: An unknown content audio signal that is input to an audio power amplifier and speaker that have a known gain and speaker sensitivity, is buffered. A portion of the buffered signal is provided to an automatic content recognition (ACR) system, and then a content identifier, of the unknown content signal, as provided by the ACR system is received. Previously determined metadata is also received, for a reference audio content signal that has been assigned to the content identifier. The metadata includes a previously determined reference measure of an audio signal characteristic for the reference content. An input measure of the audio signal characteristic is computed, for the unknown content, and compared with the reference measure. On that basis, and also based on the known gain and speaker sensitivity, at least one of a scalar gain and spectral shaping filtering that is being applied to the unknown content is adjusted. Other embodiments are also described and claimed.

Abstract: Techniques are disclosed for systems and methods to provide wind sensor motion compensation for wind sensors mounted to moving platforms. A wind sensor motion compensation system may include a wind sensor, a wind sensor accelerometer, one or more additional sensors, actuators, controllers, user interfaces, and/or other modules mounted to or in proximity to a vehicle. The wind sensor motion compensation system may be implemented with one or more logic devices adapted to receive sensor signals and determine a sensor-motion compensated wind velocity. The logic devices may be adapted to receive a wind sensor acceleration and a relative wind velocity from a wind sensor, determine a wind sensor velocity from the wind sensor acceleration, and determine a sensor-motion compensated relative wind velocity from the wind sensor velocity and the relative wind velocity.

Abstract: Provided are, among other things, systems, methods and techniques for processing an audio signal to add virtual bass. In one representative embodiment, an apparatus includes: (a) an input line that inputs an original audio signal in the time domain; (b) a bass extraction filter that extracts a bass portion of the original audio signal, which also is in the time domain; (c) an estimator that estimates a fundamental frequency of a bass sound within the bass portion; (d) a frequency translator that shifts the bass portion by a positive frequency increment that is an integer multiple of the fundamental frequency estimated by the estimator, thereby providing a virtual bass signal; (e) an adder having (i) inputs coupled to the original audio signal and to the virtual bass signal and (ii) an output; and (f) an audio output device coupled to the output of the adder.

Abstract: Provided are, among other things, systems, methods and techniques for processing an audio signal to add virtual bass. In one representative embodiment, an apparatus includes: (a) an input line that inputs an original audio signal; (b) a transform module that transforms the original audio signal into a set of frequency components; (c) a bass extraction filter that extracts a bass portion of such frequency components; (d) an estimator that estimates a fundamental frequency of a bass sound within such bass portion; (e) a frequency translator that shifts the bass portion by a frequency that is an integer multiple of the fundamental frequency estimated by the estimator, thereby providing a virtual bass signal; (f) an adder having (i) inputs coupled to the original audio signal and to the virtual bass signal and (ii) an output; and (g) an audio output device coupled to the output of the adder.

Abstract: Disclosed are a method and electronic device for voice recognition. The voice recognition method includes recognizing, in a first processor using low power mode, a voice signal inputted through a microphone, entering an active state and performing voice recording in a second processor if the recognized voice signal is a previously set keyword, and performing voice recognition in the second processor if the end of a voice input is determined during the voice recording.

Abstract: A voice signal processing apparatus and a voice signal processing method are provided. A first sampling point of an mth original frequency-lowered signal frame phase-matched to the sampling point corresponding to a phase reference sampling point number is determined according to the phase reference sampling point number of an (m?1)th original frequency-lowered signal frame corresponding to a middle sampling point of an (m?1)th renovating frequency-lowered signal frame. The q consecutive sampling points starting from the first sampling point are used as the sampling points of an mth renovating frequency-lowered signal frame.

Abstract: Example techniques may involve calibration with multiple recording devices. An implementation may include detecting, via a microphone, one or more calibration sounds as emitted by one or more playback devices of one or more zones during a calibration sequence. The implementation may further include determining a first response, the first response representing a response of a given environment to the one or more calibration sounds as detected by the first recording device and receiving data indicating a second response, the second response representing a response of the given environment to the one or more calibration sounds as detected by a second recording device. The implementation may also include determining a calibration for the one or more playback devices based on the first response and the second response and sending, to the one or more zones, an instruction that applies the calibration to playback by the one or more playback devices.

Abstract: Separation of speech and background from an audio mixture by using a speech example, generated from a source associated with a speech component in the audio mixture, to guide the separation process.

Abstract: An apparatus for processing a medical image includes: a receiver configured to receive a signal having a plurality of frequency bands; an image reconstructor configured to segment the signal into a first signal of a first frequency band and a second signal of a second frequency band based on a signal strength, and configured to generate a first reconstructed image of the first frequency band and a second reconstructed image of the second frequency band; and an image synthesizer configured to synthesize the first reconstructed image and the second reconstructed image.

Abstract: Technologies pertaining to calibration of filters of an audio system are described herein. A mobile computing device is configured to compute values for respective filters, such as equalizer filters, and transmit the values to a receiver device in the audio system. The receiver device causes audio to be emitted from a speaker based upon the values for the filters.

Abstract: A voice signal processing apparatus and a voice signal processing method are provided. A last sampling point of an mth original frequency-lowered signal frame is determined according to a phase reference sampling point number of the mth original frequency-lowered signal frame. Here, the phase reference sampling point number corresponds to a middle sampling point of an mth renovating frequency-lowered signal frame, and the last sampling point is phase-matched with a sampling point corresponding to the phase reference sampling point number in the mth original frequency-lowered signal frame. P consecutive sampling points starting from the last sampling point are applied as sampling points of an (m+1)th renovating frequency-lowered signal frame.

Abstract: One or more techniques and/or systems are provided for communicating between two or more drivers respectively controlling and/or managing different channels through which data is transferred between a wireless communication device and a computer system and/or between a controller of the computer system and an application of the computer system. Typically, at least one of the channels is configured to transmit time sensitive data (e.g., such as audio data) while another channel is configured to transmit time insensitive data (e.g., such as call control data). A device driver interface may be configured to provide a medium through which the two or more drivers can communicate. The techniques and/or systems find particular application with respect to Bluetooth headsets used in combination with a computer system comprising a system on chip architecture, but other applications are also contemplated.

Abstract: Devices related to digital MEMS loudspeakers are discussed. Such devices may include an air pressure source, MEMS valves coupled to the air pressure source, and an audio modulator coupled to the MEMS valve to receive an audio signal and to control the MEMS valves via a modulation signal to provide an acoustic output.

Abstract: A computing system automates the configuration and management of a live sound system that includes a processor and memory for building in a GUI of a display a representation of the live sound system for a venue. The system loads a venue template that includes loudspeaker arrays and related properties including a setup configuration of the loudspeaker arrays and tuning data for constituent loudspeakers that are operable to provide an audio coverage pattern for the venue. The system overlays on top of the representation of the loudspeaker arrays a wiring circuit representation indicating interconnections of the loudspeakers that define bandpass inputs for each array. The system generates a plurality of amplifiers in the representation to drive the arrays, and associates amplifier channels of the amplifiers with the bandpass inputs. The amplifier channels include representations of output channels of DSPs coupled with respective amplifier channels.

Abstract: A speech signal processing system is described for use with automatic speech recognition and hands free speech communication. A signal pre-processor module transforms an input microphone signal into corresponding speech component signals. A noise suppression module applies noise reduction to the speech component signals to generate noise reduced speech component signals. A speech reconstruction module produces corresponding synthesized speech component signals for distorted speech component signals. A signal combination block adaptively combines the noise reduced speech component signals and the synthesized speech component signals based on signal to noise conditions to generate enhanced speech component signals for automatic speech recognition and hands free speech communication.

Abstract: A mobile multi-function device that includes a speaker, two or more microphones, and a beamformer processor is described. The beamformer processor uses the microphones to perform beamforming operations. One of the microphones shares a receiver acoustic opening with the speaker while the other microphone uses a separate acoustic opening. The receiver acoustic opening may be an earpiece opening that is held to the ear of a user while conducting a phone call with the device and provides acoustic input and output paths for the microphone and the speaker, respectively.

Abstract: An apparatus for driving loudspeakers of a sound system is provided. The sound system has at least two loudspeakers of a basic system and at least three loudspeakers of a focus system, each of the loudspeakers having a position in an environment. The apparatus has a basic channel provider for providing basic system audio channels to drive the loudspeakers of the basic system, and a focused source renderer for providing focus system audio channels to drive the loudspeakers of the focus system. The focused source renderer is configured to calculate a plurality of delay values for the loudspeakers of the focus system based on the positions of the loudspeakers of the focus system and based on a position of a focus point, and to generate at least three focus group audio channels for at least some of the loudspeakers of the focus system based on the plurality of delay values and based on a focus audio base signal to provide the focus system audio channels.