Abstract: A system is provided to protect a loudspeaker (144) by controlling a level of an applied audio signal. A control signal is generated by applying an input audio signal (115) to the collective operations of a gain control system (100). The gain control system (100) uses the input audio signal (115) in conjunction with at least one parameter to derive an estimated stress associated with the loudspeaker (144). The estimated stress is compared with a protection threshold stress (127). If the protection threshold stress is exceeded, a gain applied by a gain component (134) is selectively adjusted to modify the input audio signal (115). The resulting gain-controlled audio signal (116) is employed to drive the loudspeaker (144).

Abstract: Provided are systems and methods for dynamic detection of changes in signal bandwidth associated with audio samples received by a communication device during handovers, where the communication device is moved through geographical areas served by different wireless network technologies. Based on the detection, operation parameters of an audio processor of the communication device may be adjusted or switched to achieve optimal performance of audio enhancing procedures (e.g., noise suppression) performed by the audio processor and to preserve minimal power consumption.

Abstract: A method of configuring an array of speaker elements is disclosed. The method computes a sound that pressure level at various points in the venue and is evaluated by various objective functions. The configuration of a candidate array is changed, for example by the orientation or position of the speakers and the sound field is recalculated. The process is then iterated until an acceptable configuration is found. The real physical array of speakers is then configured in that manner. The method also provides a 3D plot of the sound pressure level displayed against freguency and position in the venue.

Abstract: There is provided a method for determining a road surface state which can accurately estimate the road surface state on which a vehicle is running at one of finely classified road surface states based on a sound detection signal from a ground contact surface sound detection unit for detecting a sound in the vicinity of the ground contact surface of the tire of the running vehicle. The method calculates a wave profile of frequency dispersion by executing a 1/N octave analysis of a detected sound signal from a sound detection unit 20 for estimating a road surface state. The state of the road surface on which the vehicle is running is determined whether the calculated wave profile of frequency dispersion satisfies predetermined determination conditions of the respective road surface states. A microphone housed in a container is used to detect said sound, and the determining step is adapted to apply the determination conditions to the measured wave profile including a resonance caused by the container.

Abstract: An open air, wide area noise cancellation system and method provides improved identification and characterization of noise sources then generating noise cancelling sound waves.

Abstract: A device to measure or to estimate the acoustical output of the loudspeaker of a earphone, earplug, headphone or ear muff, and to measure the ambient environmental sounds where appropriate. The loudspeaker is connected to an electronic audio device. The acoustical output of the loudspeaker may be measured by means of a microphone placed in close proximity to the loudspeaker; or estimated by means of measuring the output of the audio amplifier (in the electronic audio device) and using the output/input transfer function that relates the acoustical output of the loudspeaker to the output of the audio amplifier; or estimated by means of the output the digital signal processor (in the electronic audio device) and using the output/input transfer function that relates the acoustical output of the loudspeaker to the output of the digital signal processor.

Abstract: There is provided a sound processing device including an input signal processing unit configured to calculate a first acoustic feature quantity indicating a likelihood being of a sinusoidal wave of a signal in each time frequency domain and a second acoustic feature quantity different from the first acoustic feature quantity based on an input signal of content to be identified, a reference signal processing unit configured to calculate the first acoustic feature quantity and the second acoustic feature quantity based on a reference signal of content prepared in advance, and a matching processing unit configured to calculate a similarity between the input signal and the reference signal based on the first and second acoustic feature quantities of the input signal and the first and second acoustic feature quantities of the reference signal.

Abstract: Sound data identification techniques are described. In one or more implementations, common sound data and uncommon sound data are identified from a plurality of sound data from a plurality of recordings of an audio source using a collaborative technique. The identification may include recognition of spectral and temporal aspects of the plurality of the sound data from the plurality of the recordings and sharing of the recognized spectral and temporal aspects to identify the common sound data as common to the plurality of recordings and the uncommon sound data as not common to the plurality of recordings.

Abstract: A system is provided and includes a plurality of acoustic devices disposed in locations arrayed throughout a defined space, each one of the plurality of acoustic devices being receptive of acoustical attributes such as sound or noise levels generated in the defined space and configured to issue signals reflective of the generated acoustical attributes and an acoustic data unit disposed in signal communication with each of the plurality of acoustic devices. The acoustic data unit is receptive of the signals issued from the plurality of acoustic devices and configured to convert the signals into digital acoustic data and to output the digital acoustic data in a serialized format compatible with a network protocol.

Abstract: The disclosure relates to a method for evaluating structure-borne sound during a collision of a vehicle. The method includes filtering a structure-borne sound signal which represents a structure-borne sound propagating in the vehicle, having a plurality of filter rules. Each of the filter rules is configured to generate a filtered structure-borne sound signal from the structure-borne sound signal. The method further includes combining the filtered structure-borne sound signals into an evaluation signal.

Abstract: An audio processing apparatus is provided. A microphone array includes microphone units. Amplifier modules each receives and amplifies an input signal from one microphone unit to generate amplified signals. A compensation module receives adjusted gains corresponding to the amplifier modules, obtains a gain difference between the adjusted gains, and adjusts one amplified signal according to the gain difference to obtain a compensated signal.

Abstract: An embodiment of an apparatus for computing control information for a suppression filter for filtering a second audio signal to suppress an echo based on a first audio signal includes a computer having a value determiner for determining at least one energy-related value for a band-pass signal of at least two temporally successive data blocks of at least one signal of a group of signals. The computer further includes a mean value determiner for determining at least one mean value of the at least one determined energy-related value for the band-pass signal. The computer further includes a modifier for modifying the at least one energy-related value for the band-pass signal on the basis of the determined mean value for the band-pass signal. The computer further includes a control information computer for computing the control information for the suppression filter on the basis of the at least one modified energy-related value.

Abstract: A mobile device moves by calculating a distance between a sound source and the mobile device using a sound source direction estimation technique. The mobile device moves by a reference distance in a direction perpendicular to a direction in which the mobile device faces the sound source when call sound of the sound source is generated, outputs voice to instruct to the sound source to generate recall sound, checks a directional angle of the mobile device when recall sound is generated by the sound source, calculates the distance between the sound source and the mobile device according to the reference distance and the directional angle of the mobile device, and moves to the vicinity of the sound source.

Abstract: The present invention relates generally to a method and system for defining a reference sound position and producing an indicia proximate thereto in relation to one or more sound characteristics. The present invention, in one or more embodiments, provides for displaying a holographic image at a reference sound location which is determined in relation to the identification of one or more target sounds being associated with one or more identified sound characteristics. In other embodiments, the present invention provides for an indicia to appear to be interactive with a reference sound location and may be used in a variety of environments including but not limited to rock theatrics, homeland security and residential security.

Abstract: Systems and methods for classification of audio environments are disclosed. In one embodiment, a method of classifying an audio environment comprises sampling the audio environment to obtain sampled audio data in accordance with a first time interval, computing features of the sampled audio data, inferring an audio cluster identifier from the features of the sampled audio data in accordance with a second time interval, and updating an audio environment model using the features of the sampled audio data in accordance with a third time interval.

Abstract: A computer-implemented method of providing matching information to prospective first and second users is performed at a server, the method including: receiving first audio data and first user information; receiving second audio data, and second user information; storing a first timestamp of the first audio data and first user information and a second timestamp of the second audio data and second user information; performing analysis processing on the first audio data to obtain first feature information, and performing analysis processing on the second audio data to obtain second feature information; judging whether the first timestamp and the second timestamp and the first feature information and the second feature information match; and if the first and second times of receipt and the first and second feature information match, sending the first user information to the second user and sending, by the server, the second user information to the first user.

Abstract: A system includes a processor and a wireless transceiver in communication with the computer processor and configured to communicate with a wireless device. In this embodiment, the processor is configured to receive a wireless audio input request through the transceiver, generated by a remote process. The processor is further configured to package and send vehicle audio input to the requesting process responsive to the request.

Abstract: Audible signals are created and emitted that provide a human user with improved sound localization cues to quickly and efficiently find the emitting device. Different emitted audible signals are sequentially emitted in response to receiving an activation signal. The different audible signals have been observed to efficiently help a human locate emitting devices that are 1) inside a sealed enclosure, such as a box; 2) in close proximity to the user; and 3) that is a large distance from the user.

Abstract: Public address systems or other systems for emitting audio signals, like music, speech or announcements, in different locations like supermarkets, schools, universities, auditoriums are widely known. These systems usually comprise an audio source, for example a microphone or a recorder, and a plurality of loudspeakers, which are locally distributed in the locations, for emitting the audio signal from the audio source.

Abstract: A testing system tests an audio connection between an audio source and a loudspeaker. The system includes a loudspeaker that converts a reference signal into a sound. An adaptive filter processes the reference signal to minimize an error signal. A decision circuit analyzes the error signal and the received signal to determine signal correlation. When the signals are not correlated, a defect is detected.

Abstract: A system is provided and includes a plurality of acoustic devices disposed in locations arrayed throughout a defined space, each one of the plurality of acoustic devices being receptive of acoustical attributes such as sound or noise levels generated in the defined space and configured to issue signals reflective of the generated acoustical attributes and an acoustic data unit disposed in signal communication with each of the plurality of acoustic devices. The acoustic data unit is receptive of the signals issued from the plurality of acoustic devices and configured to convert the signals into digital acoustic data and to output the digital acoustic data in a serialized format compatible with a network protocol.

Abstract: There is provided a method of operating a device, the device comprising a plurality of audio sensors and being configured such that when a first audio sensor of the plurality of audio sensors is in contact with a user of the device, a second audio sensor of the plurality of audio sensors is in contact with the air, the method comprising obtaining respective audio signals representing the speech of a user from the plurality of audio sensors; and analyzing the respective audio signals to determine which, if any of the plurality of audio sensors is in contact with the user of the device.

Abstract: A method and apparatus for providing simultaneous enhancement of transmission loss and absorption coefficient using activated cavities is presented. A layer of material is provided, and a backing plate having a plurality of cavities on the top surface of said backing plate, is disposed adjacent a top surface of said layer of material. A screen is disposed along the top surface of said cavities on said backing plate and at least one cavity includes an actuator disposed within the cavity and a control system receiving a signal from the microphone and receiving a signal from the accelerometer and providing a drive signal to the actuator to provide an acoustic output to provide simultaneous insertion loss and absorption which serves to minimize a linear combination of the signal from the microphone and the signal from the accelerometer.

Abstract: An apparatus for extracting a signal from convolutive mixtures includes a receiving unit which includes two or more receivers and receives a signal; a transfer function calculation unit which calculates transfer functions for demixing; and a demixing unit which demixes the received signal using the calculated transfer functions. The transfer function is determined such that a signal is extracted from a source closest to the receivers, and is calculated on the basis of a transfer function for a path to each receiver being approximated to a delta function as closer to the source.

Abstract: The present invention relates to means and methods of automated difference recognition between speech and music signals in voice communication systems, devices, telephones, and methods, and more specifically, to systems, devices, and methods that automate control when either speech or music is detected over communication links. The present invention provides a novel system and method for monitoring the audio signal, analyze selected audio signal components, compare the results of analysis with a pre-determined threshold value, and classify the audio signal either as speech or music.

Abstract: Methods and apparatus for audio signal clip detection are disclosed. The clip detectors may receive audio signals, from which peak reference signals, indicative of the highest voltage of the received audio signals, may be derived. The received audio signals may also be differentiated and phase-lagged to produce differentiated audio signals which may, in turn, be rectified to produce rectified differentiator signals. The rectified differentiator signals and the peak reference signals may be compared to thereby produce clip detect signals indicative of whether the received audio signals are clipped. The clip detect signals may then be used to indicate whether the received audio signal are clipped.

Abstract: An automatic gain control derives the gain from the distance between the sound source and the microphone. The distance-based automatic gain control normalizes signal level changes caused by a speaker not maintaining a constant distance with respect to the microphone. Also, a proximity-effect compensation that derives the adaptive filter from the distance between the sound source and the microphone. The proximity-effect compensation corrects frequency response changes due to undesired proximity-effect variations. Determination of the distance between a sound source and a microphone permits accurate compensation for both frequency response changes and distance-related signal level changes.

Abstract: A system is provided for configuring an audio system for a given space. The system may statistically analyze potential configurations of the audio system to configure the audio system. The potential configurations may include positions of the loudspeakers, numbers of loudspeakers, types of loudspeakers, listening positions, correction factors, or any combination thereof. The statistical analysis may indicate at least one metric of the potential configuration including indicating consistency of predicted transfer functions, flatness of the predicted transfer functions, differences in overall sound pressure level from seat to seat for the predicted transfer functions, efficiency of the predicted transfer functions, or the output of predicted transfer functions.

Abstract: An apparatus comprising: a receiver configured to receive at least one audio signal from a recording apparatus; the receiver further configured to receive at least one orientation indicator from the recording apparatus, each orientation indicator associated with one at least audio signal; a recording direction determiner configured to determine a recording orientation of the recording apparatus dependent on the at least one audio signal; a relative distance determiner configured to determining a relative distance of the recording apparatus from a sound source dependent on the at least one audio signal; and a relative position determiner configured to determining a relative position of the recording apparatus dependent on the orientation indicator and relative distance.

Abstract: A sound signal processing apparatus includes a sliding operation detecting section to which a sound signal collected by a microphone is inputted, and which determines start and end of a sliding operation by a determination process using a sliding sound signal component in the inputted sound signal, the sliding sound signal component being produced by the sliding operation on the microphone itself or its vicinity, and a control section that performs a predetermined control process that is set with respect to the sliding operation, during a period from the start to the end of the sliding operation determined by the sliding operation detecting section.

Abstract: Systems and methods are provided to create an acoustic map of a space containing multiple acoustic sources. Source localization and separation takes place by sampling an ultra large microphone array containing over 1020 microphones. The space is divided into a plurality of masks, wherein each masks represents a pass region and a complementary rejection region. Each mask is associated with a subset of microphones and beamforming filters that maximize a gain for signals coming from the pass region of the mask and minimizes the gain for signals from the complementary region according to an optimization criterion. The optimization criterion may be a minimization of a performance function for the beamforming filters. The performance function is preferably a convex function. A processor provides a scan applying the plurality of masks to locate a target source. Processor based systems to perform the optimization are also provided.

Abstract: An information processor has a sound acquisition unit configured to acquire a sound in a surrounding environment, a volume judging unit configured to judge whether a volume level of the sound acquired by the sound acquisition unit is higher than a reference level previously set, and a first reproduction position information recorder configured to record reproduction position information showing a current reproduction position of digital content, when the volume judging unit judges that the volume level of the acquired sound is higher than the reference level.

Abstract: Sounds sensed by a microphone of a device include sounds from a cooling fan of the device that varies based on the speed of the cooling fan, and other sounds used by a program of the device such as voice inputs. The sound level of sounds used by the program is determined, and the speed of the fan is adjusted so that a desired cooling level is attained while keeping the fan speed low enough that the noise from the fan does not interfere with the sounds used by the program.

Abstract: An apparatus, method and a computer program are provided. The apparatus is configured to process an electrical output signal from a loudspeaker to detect a user input signal.

Abstract: The use of sound pattern matching methods and techniques to analyse driver behavior and accidents and provide alerts to drivers, passengers, insurers or accident management companies. While a driver is operating a vehicle on the road, sound signatures are captured from a smartphone, in car microphone, retrofitted telematics unit or other device with audio capture capabilities. The sound signatures captured are recorded, analysed and matched against known and/or previously recorded sound patterns to help determine circumstances relating to a vehicle accident event or incident. Examples of recorded and matched sound signatures include the “rumbling” sound when driving over cats eyes or edging strips on roads, the skidding sound when braking harshly or an airbag explosion.

Abstract: A system and method may be configured to analyze audio information derived from an audio signal. The system and method may track sound pitch across the audio signal. The tracking of pitch across the audio signal may take into account change in pitch by determining at individual time sample windows in the signal duration an estimated pitch and a representation of harmonic envelope at the estimated pitch. The estimated pitch and the representation of harmonic envelope may then be implemented to determine an estimated pitch for another time sample window in the signal duration with an enhanced accuracy and/or precision.

Abstract: A volume correction device includes: a variable gain means for controlling a gain, given to an input audio signal, according to a gain control signal; a consecutive relevant sounds interval detection means for detecting a consecutive relevant sounds interval, during which a group of temporally adjoining consecutive relevant sounds is present, in the input audio signal; a mean level detection means for detecting the mean level of the input audio signal attained during the consecutive relevant sounds interval, and whose time constant for mean level detection is set to a smaller value during the leading period of the consecutive relevant sounds interval than during the remaining period; a gain control signal production means for producing the gain control signal, so that the mean level will be equal to a reference level, and feeding the gain control signal to the variable gain means.

Abstract: An approach is provided for efficiently capturing, processing, presenting, and/or associating audio objects with content items and geo-locations. A processing platform may determine a viewpoint of a viewer of at least one content item associated with a geo-location. Further, the processing platform and/or a content provider may determine at least one audio object associated with the at least one content item, the geo-location, or a combination thereof. Furthermore, the processing platform may process the at least one audio object for rendering one or more elements of the at least one audio object based, at least in part, on the viewpoint.

Abstract: In reverberant environments, reflected waves including an echoic sound and a muffled sound affect and disable recognition of sound arrival directions. As a result, the subjective clearness of the sounds deteriorates. In order to enhance the clearness of a reproduced sound in a reverberant environment, a pre-processing filter unit corrects an input sound signal portion having a frequency band relating to human auditory recognition on a sound wave arrival direction, and speakers reproduce the sound signal. The correction involves attenuating an input sound signal in the frequency band portion, based on the relationship between the frequencies of the input sound signal and the magnitude of influence to the recognition of the sound wave arrival direction. This attenuation is achieved by filtering using filter coefficients that are set by a first filter characteristic setting unit using hearing characteristic parameters that are set by a hearing characteristic setting unit.

Abstract: An audio signal processing system including a time-frequency conversion unit which converts an audio signal in time domain into frequency domain in frame units so as to calculate a frequency spectrum of the audio signal, a spectral change calculation unit which calculates an amount of change between a frequency spectrum of a first frame and a frequency spectrum of a second frame before the first frame based on the frequency spectrum of the first frame and the frequency spectrum of the second frame, and a judgment unit which judges the type of the noise which is included in the audio signal of the first frame in accordance with the amount of spectral change.

Abstract: Embodiments of the present invention provide a power supply unit capable of achieving noise reduction while maintaining efficiency under the light load state. A FET driver controls switching elements in either one of a Pulse Width Modulation (PWM) mode, an intermittent mode having a lower operating frequency than that in the PWM mode, and a noise-free mode having a higher operating frequency than an audible frequency range. The FET driver operates first in the intermittent mode under the light load state. A microphone collects noise generated from the surroundings of a power supply unit. When the level of an audio signal collected by the microphone exceeds a predetermined level, the FET driver transitions from the intermittent mode to the noise-free mode. In accordance with such an embodiment, the FET driver operates in the noise-free mode only when noise is actually generated.

Abstract: There is provided a audio pacing device including a sensing unit, such as heart rate monitoring unit, to obtain a parameter, such as heart-beat rate (pulse), of a user in physical exercise, a memory to store a plurality of audio signals having predetermined tempo indications (e.g.

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: An acoustic monitoring system includes a portable acoustic detection device, a sound analysis device and a confidence level device. The portable acoustic detection device is capable of receiving sound at one or more locations near a sound-producing device. The sound analysis device receives the sound from the portable sound detection device, determines a diagnosis based on a comparison between the sound and pre-recorded sound data, and outputs the diagnosis to the portable acoustic detection device. The sound analysis device also determines a corrective action for inhibiting the sound, which is also output to the portable sound detection device. The confidence level device determines a confidence level of the diagnosis indicating a likelihood that the diagnosis is successfully diagnosed.

Abstract: An uncomfortable sound pressure estimation system includes: a biological signal measurement section for measuring an electroencephalogram signal of a user; a sound stimulation output section for presenting a sound stimulation group to the user, the sound stimulation group including a first sound, a second sound, and a third sound which are pure tones of the same frequency and which consecutively decrease in sound pressure within a predetermined range; an extraction section for, from the electroencephalogram signal in a predetermined zone defined based on a point of presenting at least one of the second sound and the third sound as a starting point, extracting an N1-P2-amplitude related or wavelet-coefficient related characteristic amount of event-related potential of the electroencephalogram signal; and a determination section for determining an uncomfortable sound pressure at the frequency of the sound stimulation group based on the characteristic amount extracted by the extraction section.

Abstract: A method for checking availability of audio interface between a display apparatus and an AV apparatus in an AV system, and an AV apparatus using the same are provided. The checking includes transmitting a command to check an audio interface to a second AV apparatus where an audio from a first AV apparatus is output, and receiving a result of checking the audio interface in response to the command from the second AV apparatus. Therefore, a user can recognize whether or not an audio interface has a problem in an AV system.

Abstract: An audio input signal is processed to produce a processed audio output signal. An audio input signal is received as an original signal. The audio input signal is dynamically filtered to produce a first stage signal consisting of a selected frequency band of the input signal. Gain applied to the first stage signal is dynamically controlled in response to a control signal to produce a second stage signal. The control signal is derived from the first stage signal. Processing the original signal in combination with the second stage signal to produce a processed audio output signal. Processing the original signal in combination with the second stage signal and the first stage signal to produce a processed audio output signal.

Abstract: A smart phone application is configured to connect to the internal microphone to continuously monitor ambient audio picked up by the microphone by keeping the microphone always on in order to detect the presence of music. When a song is detected by audio analysis an acoustic fingerprint is created and compared against a song database of acoustic fingerprints. When the artist and title of a song is obtained by the smart phone application, the application stores that information in a listing.

Abstract: A method for determining a semantic concept associated with an audio signal captured using an audio sensor. A data processor is used to automatically analyze the audio signal using a plurality of semantic concept detectors to determine corresponding preliminary semantic concept detection values, each semantic concept detector being adapted to detect a particular semantic concept. The preliminary semantic concept detection values are analyzed using a joint likelihood model based on predetermined pair-wise likelihoods that particular pairs of semantic concepts co-occur to determine updated semantic concept detection values. One or more semantic concepts are determined based on the updated semantic concept detection values. The semantic concept detectors and the joint likelihood model are trained together with a joint training process using training audio signals, at least some of which are known to be associated with a plurality of semantic concepts.

Abstract: An audio signal processing apparatus includes a tapping detection unit to which an audio signal picked up by a microphone is input and that detects a tapping input operation on the basis of an energy increase/decrease determination process for making a determination as to whether an increase and a decrease of an energy level of low-frequency components of the audio signal occur within a first time period; and a control unit that performs a certain control process that is set for a tapping input operation in response to a tapping input operation being detected by the tapping detection unit.