Abstract: A signal processing apparatus has a first memory in which plural pieces of FIR coefficient data used for implementing an FIR filter algorithm are stored, a second memory which stores plural pieces of input data to be subjected to the FIR filter algorithm, and a processor implements the FIR filter algorithm using the plural pieces of FIR coefficient data stored in the first memory and the plural pieces of input data stored in the second memory as many times as the number corresponding to a designated filter order, in which filter algorithm each piece of coefficient data and each piece of input data are multiplied together and resultant products are summed up. The signal processing apparatus is provided, which can implement plural sorts of FIR filter algorithms of filter order which can be changed flexibly.

Abstract: A method for estimating the acoustic reverberations in an environment comprising the following steps: a measurement step in which one acoustic signal emitted in the environment is captured; a step for determination of acoustic energy decay rate distribution during which an acoustic energy decay rate distribution is determined from the acoustic signal captured in step (a); an estimation step during which a reverberation time and a reverberation level of sound in the environment are estimated by regression from the characteristic function of the acoustic energy decay rate distribution determined in step (b).

Abstract: Holographic sound is recorded and reproduced by way of a single monaural recording per left and right ear recorded. This is accomplished by determining the phase shift of frequencies recorded after dividing the sound into discrete frequencies in a recording device having resonators, each resonating at a different frequency, placed in a circular arrangement and divided into discrete channels by non-resonant material. The resonators are placed in a pseudo-randomized arrangement within the recording device and the circle of resonators is in front of a microphone which records the sound monaurally. Playback is then by way of arranging speakers or transducers into micro perforated sheets which amplify the sound, the arrangement of speakers/transducers around a central point. The sound is then played back directionally based on the position where the sound originally was recorded from and the position of the particular transducer around the central point.

Abstract: Systems and methods are provided for conducting conference calls using doppler-based, i.e., reverberation-based techniques. The embodiments comprise a call device performing operations to join a call session hosted on a session server; receive sensor data comprising an audio signal from a first microphone and location information associated with the first microphone; determine a reverberation parameter associated with the location information; generate a first processed audio signal based on the audio signal and the reverberation parameter; and transmit the first processed audio signal to the session server. The session server may perform operations to receive a respective processed audio signal; determine a sound quality parameter of the respective processed audio signal; generate a balanced audio signal based on the sound quality parameter and the received processed audio signal; and transmit the balanced audio signal to a remote call device belonging to a second party.

Abstract: In embodiments of the present invention improved capabilities are described for a computer-based method for reproducing a performance sound quality in a rehearsal space, the method comprising accessing a computer stored multi-dimensional sound signature for an audience location in a performance space, receiving sound data from a sound input device in the rehearsal space, modifying the sound data to match a sound characteristic of the multi-dimensional sound signature, and transmitting the modified sound data through a sound output device in the rehearsal space.

Abstract: Systems and methods are provided for conducting conference calls using doppler-based, i.e., reverberation-based techniques. The embodiments comprise a call device performing operations to join a call session hosted on a session server; receive sensor data comprising an audio signal from a first microphone and location information associated with the first microphone; determine a reverberation parameter associated with the location information; generate a first processed audio signal based on the audio signal and the reverberation parameter; and transmit the first processed audio signal to the session server. The session server may perform operations to receive a respective processed audio signal; determine a sound quality parameter of the respective processed audio signal; generate a balanced audio signal based on the sound quality parameter and the received processed audio signal; and transmit the balanced audio signal to a remote call device belonging to a second party.

Abstract: Provided are a sound effect processing method and device. The sound effect processing method includes: invoking a pre-loaded plug-in unit manager to acquire a sound effect processing parameter supported by each preloaded sound effect plug-in unit; acquiring a sound effect configuration file pre-configured by the plug-in unit manager; displaying sound effect index identifiers corresponding to various sound effect modes; according to a selected sound effect index identifier, determining a selected sound effect mode, and according to an adjustment interface, acquiring adjusted parameter control data; sending the adjusted parameter control data to a selected sound effect plug-in unit; and invoking the plug-in unit manager to send data to be processed to the selected sound effect plug-in unit, and according to the adjusted parameter control data, conducting sound effect processing on the data to be processed by the selected sound effect plug-in unit.

Abstract: A system, article, and method of acoustic dereverberation factoring the actual non-ideal acoustic environment.

Abstract: A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter. The encoded audio bitstream is then decoded to obtain a decoded frequency domain representation of the M audio channels, and at least a portion of the frequency domain representation is decorrelated with an all-pass filter having a fractional delay. The all-pass filter is attenuated at locations of a transient. A matrixed version of the decorrelated signals are summed with a matrixed version of the decoded frequency domain representation to obtain N audio signals that collectively having N audio channels where M is less than N.

Abstract: An effect providing apparatus is provided which ostensibly increases the number of effects that can be simultaneously provided. When switching to flanger processing is performed while filter processing in accordance with a periodic signal (LFO signal) is being performed in an effect processing section that can provide only one effect, a CPU advances the periodic signal (LFO signal) from a phase at the time of the switching, and performs the filter processing in accordance with the advancing periodic signal when the flanger processing is ended. As a result of this configuration, the number of effects that can be simultaneously provided is ostensibly increased.

Abstract: A method and apparatus may be used to perform personalized audio virtualization. The apparatus may include a speaker, a headphone (over-the-ear, on-ear, or in-ear), a microphone, a computer, a mobile device, a home theater receiver, a television, a Blu-ray (BD) player, a compact disc (CD) player, a digital media player, or the like. The apparatus may be configured to receive an audio signal, scale the audio signal, and perform a convolution and reverberation on the scaled audio signal to produce a convolved audio signal. The apparatus may be configured to filter the convolved audio signal and process the filtered audio signal for output.

Abstract: A delay line is constructed by combining a phase generator and a fabric. The phase generator splits a digital input signal in multiple incrementally delayed versions, which are input to the fabric. The fabric has an array of node filters. Inputs of filters in the first array column are inputs of the fabric. A node filter has a delay element and a cross-coupling element, whose output signals are added or subtracted to form a filter output signal. A node filter in a row is concatenated to the previous filter in the row through its delay element. Inputs of cross-coupling elements are connected to other array rows. Outputs of node filters form the outputs of the fabric. Delay times of delay elements and cross-coupling elements are nominally equal. Drive strengths of cross-coupling elements may be lower than drive strengths of delay elements.

Abstract: An information processing device includes: a reflecting surface determining section configured to determine a reflecting surface as an object for reflecting a sound; a reflecting surface information obtaining section configured to obtain reflecting surface information indicating a reflection characteristic of the determined reflecting surface; and an output control portion configured to output a directional sound according to the obtained reflecting surface information to the determined reflecting surface.

Abstract: Methods and devices for processing and voice operated control are provided. The method can include performing a non-difference comparison between a first received sound and a second received sound, determining if speech exists based on the comparison, and transmitting or providing a decision that the speech is present to at least one among the device, a cell phone, a media player, or a portable computing device. Other embodiments are disclosed.

Abstract: A reverberator for reverberating an audio signal includes a feedback delay loop processor for delaying at least two different frequency subband signals representing the audio signal by different loop delays to obtain reverberated frequency subband signals.

Abstract: An effect giving device that can solve a problem that sound is localized inside a head when using a headphone and a problem that timbre changes depending on kind of the headphone is provided. A DSP 10 performs a speaker adjusting equalizer process that processes an entire bandwidth with seven bands. When output selecting means select a headphone 320, the DSP 10 performs a headphone adjusting equalizer process of the entire bandwidth with three bands of the equalizer process, and processes a process of other bands by replacing it to a lateralization sense alleviating process for the headphone 320. Since the headphone 320 and a speaker 318 are exclusively used, using an identical program storage area efficiently ensures an alleviation of a lateralization sense when using the headphone 320 and an adjustment of the timbre.

Abstract: A sound effect producing apparatus includes an input portion, a memory portion, a pseudo sound reflection producing portion, and an effect provision portion. The input portion performs inputting of an audio signal. The memory portion stores sound effect information that includes production information for producing a pseudo sound reflection corresponding to a sound reflection generated in a predetermined acoustic space and sound source position information showing a sound source position of the pseudo sound reflection. The pseudo sound reflection producing portion produces a pseudo sound reflection based on the production information. The effect provision portion performs a process of localizing the pseudo sound reflection using a predetermined direction as a reference, based on the audio signal and the sound source position information.

Abstract: A mixing signal processing technique modifies digital audio recordings to simulate the linear and nonlinear effects of propagation and mixing of sounds in air. When multiple sounds or complex sounds comprised of multiple frequencies in the audible spectrum propagate in such a nonlinear medium, they transfer energy into sound at new frequencies given by the sums and differences of the original signal frequencies. The mixing signal processing technique may improve the ability of a system to reproduce the effects of a live performance using a digital audio recording.

Abstract: Example techniques may involve multiple calibrations for one or more playback devices. An example implementation may involve detecting, via a microphone, calibration sounds as emitted by one or more playback devices during a calibration sequence, perhaps by recording first samples while the microphone is in motion through a given environment and recording second samples while the microphone is stationary at one or more particular locations. The implementation may also include determining a first calibration for the one or more playback devices based on at least the first samples of the calibrations sounds and determining a second calibration for the one or more playback devices based on at least the second samples of the calibrations sounds. The implementation may further include applying at least one of (a) the first calibration or (b) the second calibration to playback by the one or more playback devices.

Abstract: An audio renderer comprises a receiver (801) receiving input data comprising early part data indicative of an early part of a head related binaural transfer function; reverberation data indicative of a reverberation part of the transfer function; and a synchronization indication indicative of a time offset between the early part and the reverberation part. An early part circuit (803) generates an audio component by applying a binaural processing to an audio signal where the processing depends on the early part data. A reverberator (807) generates a second audio component by applying a reverberation processing to the audio signal where the reverberation processing depends on the reverberation data. A combiner (809) generates a signal of a binaural stereo signal by combining the two audio components. The relative timing of the audio components is adjusted based on the synchronization indication by a synchronizer (805) which specifically may be a delay.

Abstract: Provided is a signal processing apparatus including a feature detection unit configured to detect, from an input signal, a detection signal including at least one of audience-generated-sound likelihood and music likelihood, a reverberation adding unit configured to add long or short reverberations to the input signal based on a detected tone being moderate or ordinary tone respectively, and a vicinity-sound generation unit configured to generate vicinity sound based on the detection signal.

Abstract: An apparatus for generating a first multitude of output signals based on at least one audio source signal having a delay network and a feedback processor. The delay network includes a second multitude of delay paths, each delay path having a delay line and an attenuation filter. Each delay line is configured for delaying delay line input signals and for combining the at least one audio source signal and a reverberated audio signal to obtain a combined signal, wherein the attenuation filter of a delay path is configured for filtering the combined signal from the delay line of the delay path to obtain an output signal. The first multitude of output signals includes the output signal. The feedback processor is configured for reverberating the first multitude of output signals to obtain a third multitude of the reverberated audio signals including the reverberated audio signal.

Abstract: An audio object including audio content and object metadata is received. The object metadata indicates an object spatial position of the audio object to be rendered by audio speakers in a playback environment. Based on the object spatial position and source spatial positions of the audio speakers, initial gain values for the audio speakers are determined. The initial gain values can be used to select a set of audio speakers from among the audio speakers. Based on the object spatial position and a set of source spatial positions at which the set of audio speakers are respectively located in the playback environment, a set of non-negative optimized gain values for the set of audio speakers is determined. The audio object at the object spatial position is rendered with the set of optimized gain values for the set of audio speakers.

Abstract: Methods, systems, and computer readable media for utilizing parallel adaptive rectangular decomposition (ARD) to perform acoustic simulations are disclosed herein. According to one method, the method includes assigning, to each of a plurality of processors in a central processing unit (CPU) cluster, ARD processing responsibilities associated with one or more of a plurality of partitions of an acoustic space and determining, by each processor, pressure field data corresponding to the one or more assigned partitions. The method further includes transferring, by each processor, the pressure field data to at least one remote processor that is assigned to a partition that shares an interface with at least one partition assigned to the transferring processor and receiving, by each processor from the at least one remote processor, forcing term values that have been derived by the at least one remote processor using the pressure field data.

Abstract: Provided are methods and systems for generating Direct-to-Reverberant Ratio (DRR) estimates. The methods and systems use a null-steered beamformer to produce accurate DRR estimates across a variety of room sizes, reverberation times, and source-receiver distances. The DRR estimation algorithm uses spatial selectivity to separate direct and reverberant energy and account for noise separately. The formulation considers the response of the beamformer to reverberant sound and the effect of noise. The DRR estimation algorithm is more robust to background noise than existing approaches, and is applicable where a signal is recorded with two or more microphones, such as with mobile communications devices, laptop computers, and the like.

Abstract: A circuit for driving a fan motor includes: a control input interface circuit configured to generate a first digital value indicating an input duty ratio; a duty calculation unit configured to generate a duty command value linearly increasing with a slope with respect to the first digital value; a digital pulse width modulator configured to generate a control pulse having an output duty ratio corresponding to the duty command value; an output circuit configured to drive a fan motor based on the control pulse; a lock protection circuit configured to switch between an enable state and a disable state and to stop supply of power to the fan motor when lock of the fan motor is detected in the enable state; and a torque-off determination unit configured to switch the lock protection circuit to the disable state in a torque-off state.

Abstract: Mobile devices may capture audio signals indicative of test audio received by an audio capture device of the mobile device; and send the captured audio and the zone designation to a sound processor to determine equalization settings for speakers of the zone of the venue. An audio filtering device may receive the captured audio signals from the mobile devices; compare each of the captured audio signals with the test signal to determine an associated reliability of each of the captured audio signals; combine the captured audio signals into zone audio data; and transmit the zone audio data and associated reliability to a sound processor configured to determine equalization settings for the zone based on the captured audio signals and the test signal.

Abstract: Methods and systems are provided for gaming headset with enhanced off-screen awareness. One or more components may be identified or detected in input audio signals, which may meet particular positioning criteria, relating to a user of the system when the user is receiving output audio signals corresponding to the input audio signals. The input audio signals may then be processed to adjust characteristics corresponding to the one or more components and/or one or more other components in the output audio signals, as perceived by the user when receiving the output audio signals. The positioning criteria may include field-of-view of the user, with the one or more components corresponding to sounds in (or having source(s) within) the field-of-view of the user. The processing may be configured to enhance the one or more other components and/or to subtract or suppress the one or more components.

Abstract: A headphone with adjustable speaker drivers and a microphone can be used to determine and adjust sound pressure levels. The speaker drivers can be adjusted manually or wirelessly via a mobile device with a wireless connection to the headphone. Processing of audio and microphone data via the headphone can also be used to help determine and adjust the sound pressure levels.

Abstract: A system and method of matching reverberation in teleconferencing environments. When the two ends of a conversation are in environments with differing reverberations, the method filters the reverberation so that when both signals are output at the near end (e.g., the audio signal from the far end and the sidetone from the near end), the reverberations match. In this manner, the user does not perceive an annoying difference in reverberations, and the user experience is improved.

Abstract: Existing post-filtering methods for microphone array speech enhancement have two common deficiencies. First, they assume that noise is either white or diffuse and cannot deal with point interferers. Second, they estimate the post-filter coefficients using only two microphones at a time, performing averaging over all the microphones pairs, yielding a suboptimal solution. The provided method describes a post-filtering solution that implements signal models which handle white noise, diffuse noise, and point interferers. The method also implements a globally optimized least-squares approach of microphones in a microphone array, providing a more optimal solution than existing conventional methods. Experimental results demonstrate the described method outperforming conventional methods in various acoustic scenarios.

Abstract: A vehicle approach alert device includes: a first output sound voltage waveform signal generator having a first output sound generator that generates a first signal corresponding to output sound data read out from a memory device, and a first sound volume variable device that controls a voltage level of the first signal; a second output sound voltage waveform signal generator having a second output sound generator that generates a second signal corresponding to the output sound data read out from the memory device, and a second sound volume variable device that controls a voltage level of the second signal; a mixer circuit that mixes the first signal and the second signal; and an amplifier that outputs a vehicle approach alert sound with flowing current, which corresponds to a voltage of a mixed signal, through a sound output body.

Abstract: An apparatus for determining a measure for a perceived level of reverberation in a mix signal consisting of a direct signal component and a reverberation signal component, has a loudness model processor having a perceptual filter stage for filtering the dry signal component the reverberation signal component or the mix signal, wherein the perceptual filter stage is configured for modeling an auditory perception mechanism of an entity to obtain a filtered direct signal, a filtered reverberation signal or a filtered mix signal. The apparatus furthermore has a loudness estimator for estimating a first loudness measure using the filtered direct signal and for estimating a second loudness measure using the filtered reverberation signal or the filtered mix signal, where the filtered mix signal is derived from a superposition of the direct signal component and the reverberation signal component.

Abstract: Systems and processes for compensating for changes in a theatre sound system positioned in a theatre are described. A subsequent response of a loudspeaker to a test signal is captured and compared to a previously obtained signature response of the loudspeaker to the test signal. An audio signal can be processed based on the comparison to compensate for changes to loudspeaker performance, or otherwise.

Abstract: A method performed by an audio decoder for reconstructing N audio channels from an audio signal containing M audio channels is disclosed. The method includes receiving a bitstream containing an encoded audio signal having M audio channels and a set of spatial parameters, the set of spatial parameters including an inter-channel intensity difference parameter and an inter-channel coherence parameter. The encoded audio bitstream is then decoded to obtain a decoded frequency domain representation of the M audio channels, and at least a portion of the frequency domain representation is decorrelated with an all-pass filter having a fractional delay. The all-pass filter is attenuated at locations of a transient. A matrixed version of the decorrelated signals are summed with a matrixed version of the decoded frequency domain representation to obtain N audio signals that collectively having N audio channels where M is less than N.

Abstract: Provided is a signal processing device including an input unit inputting a sound collection signal of a sound of a user collected by user-surrounding microphones with inward directivity in a reproduction environment, an acoustic-signal processing unit obtaining a reproduction signal of a sound field in a measurement environment by performing a process on the sound collection signal based on a transfer function measured based on a result of sequential emission of predetermined sounds from speakers at positions set on a first closed surface with outward directivity in the measurement environment and collection of the sounds with microphones at positions on a second closed surface with outward directivity in the measurement environment and indicates how the sounds are transferred to the positions on the second closed surface, and a sound-emission control unit causing user-surrounding speakers to emit sound based on the reproduction signal with inward directivity in the reproduction environment.

Abstract: A mixing signal processing technique modifies digital audio recordings to simulate the linear and nonlinear effects of propagation and mixing of sounds in air. When multiple sounds or complex sounds comprised of multiple frequencies in the audible spectrum propagate in such a nonlinear medium, they transfer energy into sound at new frequencies given by the sums and differences of the original signal frequencies. The mixing signal processing technique may improve the ability of a system to reproduce the effects of a live performance using a digital audio recording.

Abstract: A reverberation suppression apparatus includes: an instantaneous value calculation unit that calculates an instantaneous value/instantaneous values in an envelope of values correlating with the absolute value or the square of an input signal; a reverberation estimation unit that calculates an exponential moving average of the instantaneous value(s) as an estimated reverberation component; a gain derivation unit that derives a gain corresponding to the input signal according to the estimated reverberation component and the instantaneous value(s) when the each instantaneous value is larger than the estimated reverberation component, and derives a lower limit of the gain as the gain corresponding to the input signal when the each instantaneous value is smaller than the estimated reverberation component; a smoothing unit that performs a smoothing process on the gain; and a gain processing unit that applies the gain to amplitude adjustment of the input signal thereafter.

Abstract: Apparatus and methods are disclosed for acoustic optimization. An example playback device includes a first transducer to at least one of output sound waves and receive sound waves, a second transducer to at least one of output sound waves and receive sound waves, and an acoustic grille positioned in relation to the first transducer, where the acoustic grille is to reflect sound waves received at a first angle of incidence.

Abstract: A fabric for delaying digital signals in continuous time has an array of node filters. Inputs of filters in the first column are inputs of the fabric. A node filter has a delay element and a cross-coupling element, whose output signals are added or subtracted to form an output signal of the filter. A node filter in a row is concatenated to the previous filter in the row through its delay element. Inputs of cross-coupling elements are connected to other rows of the array. Outputs of node filters form the outputs of the fabric. Delay times of delay elements and cross-coupling elements are nominally equal. Drive strengths of cross-coupling elements may be lower than drive strengths of delay elements. A delay line is constructed by combining a phase generator and a fabric, where the phase generator splits a digital input signal in multiple incrementally delayed versions for the fabric inputs.

Abstract: In an augmented reality environment, a speaker array is centrally located within an area to generate sound for the environment. The speaker array has a spherical or hemispherical body and speakers mounted about the body to emit sound in multiple directions. A controller is provided to select sets of speakers to form beams of sound in determined directions. The shaped beams are output to deliver a full audio experience in the environment from the fixed location speaker array.

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

Abstract: A system may be for the correction of phase and amplitude errors. The system may receive a first input signal and a second input signal and supply a first output signal and a second output signal. The system may include two adders that supply the first and second output signals, respectively. The two adders may be configured for computing a sum of the first and second input signals, and multiplying the weighted sum by a third coefficient. Moreover, the first coefficient or the second coefficient of the first adder may be variable to enable correction of the phase errors, and the third coefficient of the second adder may be variable to enable correction of the amplitude errors.

Abstract: A head-related transfer function (HRTF) convolution method arranged, when an audio signal is reproduced acoustically by an electro-acoustic conversion unit disposed in a nearby position of both ears of a listener, to convolute an HRTF into the audio signal, which allows the listener to listen to the audio signal such that a sound image is localized in a perceived virtual sound image localization position, the method including the steps of: measuring, when a sound source is disposed in the virtual sound image localization position, and a sound-collecting unit is disposed in the position of the electro-acoustic conversion unit, a direct-wave direction HRTF regarding the direction of a direct wave, and reflected-wave direction HRTFs regarding the directions of selected one or more reflected waves, from the sound source to the sound-collecting unit, separately beforehand; and convoluting the obtained direct-wave direction HRTF, and the reflected-wave direction HRTFs into the audio signal.

Abstract: A method and apparatus may be used to perform personalized audio virtualization. The apparatus may include a speaker, a headphone (over-the-ear, on-ear, or in-ear), a microphone, a computer, a mobile device, a home theater receiver, a television, a Blu-ray (BD) player, a compact disc (CD) player, a digital media player, or the like. The apparatus may be configured to receive an audio signal, scale the audio signal, and perform a convolution and reverberation on the scaled audio signal to produce a convolved audio signal. The apparatus may be configured to filter the convolved audio signal and process the filtered audio signal for output.

Abstract: A system and method of improving the efficiency in the power consumption of an audio system. In essence, the technique is to adjust the power delivered from the power supply to the analog section, such as the power amplifier, in response to the volume level indicated by the volume control module and/or in response to the detected characteristic of the input audio signal. Thus, in this manner, the analog section is operated in a manner that is related to the level of the signal it is processing. Additionally, the system and method also relate to a technique of adjusting the dynamic ranges of the digital signal and the analog signal to improve the overall dynamic range of the system without needing to consume additional power.

Abstract: A system and method of matching reverberation in teleconferencing environments. When the two ends of a conversation are in environments with differing reverberations, the method filters the reverberation so that when both signals are output at the near end (e.g., the audio signal from the far end and the sidetone from the near end), the reverberations match. In this manner, the user does not perceive an annoying difference in reverberations, and the user experience is improved.

Abstract: An audio processing device includes a reverb property estimating unit that estimates a reverb property at each frequency on the basis of a first audio signal and a second audio signal representing sounds of the first audio signal output by an audio output unit and collected by an audio input unit, a gain calculating unit that determines an attenuating ratio for a component of the first audio signal at each frequency such that the larger the reverb property at the frequency is, the larger the attenuating ratio for the component at the frequency becomes, and a correcting unit that attenuates the first audio signal at the each frequency in accordance with the attenuating ratio determined for each frequency.

Abstract: Methods and apparatuses for representing a sound field in a physical space are provided and described as embodied in a system that includes a sound transducer array along with a touch surface-enabled display table. The array may include a group of transducers (multiple speakers and/or microphones). The array may be configured to perform spatial processing of signals for the group of transducers so that sound rendering (in configurations where the array includes multiple speakers), or sound pick-up (in configurations where the array includes multiple microphones), may have spatial patterns (or sound projection patterns) that are focused in certain directions while reducing disturbances from other directions.

Abstract: A method for continuously estimating reverberation decay comprising receiving a sequence of audio data samples. Determining whether a plateau is present in the sequence of audio data samples. Generating one or more reverberation parameters from the sequence of audio data samples if it is determined that the plateau is present.