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

Abstract: A connector mating assurance system includes a microphone configured to be located in a vicinity of a mating zone for electrical connectors. The microphone is configured to detect an audible sound when the electrical connectors are mated. An output unit is connected to the microphone and receives audio signals from the microphone. The output unit processes the audio signals for mating assurance. The output unit may provide feedback to an assembler based on the audio signals. The output unit may determine if the electrical connectors are properly mated based on the audio signals. The microphone may be held by the assembler proximate to the assembler's hand when assembling the electrical connectors.

Abstract: A game headset receives a plurality of audio channels during play of a particular game, monitors one or more of the plurality of audio channels and detects an occurrence of one or more particular sounds in the plurality of audio channels during the monitoring of the one or more of the plurality of audio channels. In response to the detecting, the game headset triggers playback of one or more of a plurality of voice commands that corresponds to the one or more particular sounds. The voice commands may be predefined and associated with the one or more particular sounds in a data structure. The voice commands may instruct the listener of the game headset to perform an action in the particular game. The characteristics of the one or more sounds may include direction, intensity, and/or frequency of the particular one or more sounds.

Abstract: A sound field control apparatus includes at least two main microphones; for each main microphone, a set of at least two sub microphones arranged such that the at least two sub microphones are placed in different axis directions about each of the main microphones; a filtering unit; and a filter coefficient calculating unit configured to calculate a filter coefficient for the filtering unit. A filter coefficient used to control sound pressure levels and air particle velocities of an output audio signal is calculated on the basis of a sound pressure level detected by each main microphone and the difference between the sound pressure level detected by the main microphone and that detected by each of the corresponding sub microphones.

Abstract: A signal processing apparatus includes an absolute value unit configured to convert an audio signal into absolute values, a representative value calculation unit configured to calculate representative values of consecutive sample values included in blocks of the audio signal which has been converted into the absolute values using at least maximum sample values among values of the samples included in the blocks for individual blocks, an average value calculation unit configured to determine a section which includes a predetermined number of consecutive blocks as a frame and calculate a maximum value of the representative values of the blocks included in the frame and an average value of the representative values of the blocks included in the frame, and a detector configured to detect click noise in the frame on the basis of a ratio of the maximum value to the average value.

Abstract: Systems and methods for a bone-conduction transducer array configured to provide spatial audio are described, in which the bone-conduction transducer array may be coupled to a head-mountable device so as to provide sound, for example, to a wearer of the head-mountable device. Audio information and a vibration transducer from an array of vibration transducers coupled to the head-mountable computing device may be caused to vibrate based at least in part on the audio signal so as to transmit a sound. Information indicating a movement of the wearable computing device toward a given direction may be received. One or more parameters associated with causing the at least one vibration transducer to emulate the sound from the given direction may then be determined, wherein the one or more parameters are representative of a correlation between the audio information and the information indicating the movement.

Abstract: Systems and methods are disclosed that may be implemented to locally and remotely record, control, monitor and analyze sounds generated in an information handling system environment, e.g., such as in a server environment. Among other things, remote monitoring and observation of sound in a data center environment may be implemented to assist administrators with data center management and forensics analysis related to physical chassis events (e.g., such as chassis access and/or chassis intrusion), as well as providing live streaming, and local technician recorded sound bite capabilities. A sound policy may also be implemented involving server sound optimizations that works in conjunction with system thermal and power algorithms.

Abstract: Techniques for improved text-to-speech processing are disclosed. The improved text-to-speech processing can convert text from an electronic document into an audio output that includes speech associated with the text as well as audio contextual cues. One aspect provides audio contextual cues to the listener when outputting speech (spoken text) pertaining to a document. The audio contextual cues can be based on an analysis of a document prior to a text-to-speech conversion. Another aspect can produce an audio summary for a file. The audio summary for a document can thereafter be presented to a user so that the user can hear a summary of the document without having to process the document to produce its spoken text via text-to-speech conversion.

Abstract: A hearing protector comprises two protective muffs (1) with passive noise damping, a microphone (3) disposed in at least the one protective muff (1) and connected to an analog and a digital signal processing device (5 and 7, respectively), the signal processing devices (5, 7) being connected to loudspeakers (4) disposed interiorly in the protective muffs (1) for extinguishing interior noise in the protective muffs (1). The analog signal processing device is rehearsed for processing non-repeatable noise, while the digital signal processing device is rehearsed for processing repeatable noise. According to the invention, a further microphone (8) is disposed outside the passive noise damping and is connected to the digital signal processing device for tracing and locking onto repeatable noise.

Abstract: A sound processing device includes a storage unit configured to store first operation data corresponding to a motion of a mechanical apparatus and a first sound feature value corresponding to the motion in correlation with each other, a noise estimating unit configured to estimate a third sound feature value corresponding to a noise component based on a second sound feature value corresponding to an acquired sound signal, a sound feature value processing unit configured to calculate a target sound feature value from which the noise component is removed based on the second sound feature value and the third sound feature value, and an updating unit that updates the first sound feature value stored in the storage unit based on detected second operation data and the third sound feature value estimated by the noise estimating unit.

Abstract: Techniques are described herein that use sensors (e.g., microphones) for noise reduction in a mobile communication device. For example, one technique enables a first sensor that is initially configured to be a speech sensor to be used as a noise reference sensor. This technique also enables a second sensor that is initially configured to be a noise reference sensor to be used as a speech sensor. Another technique enables a primary sensor and/or a secondary sensor in a handset of a mobile communication device to be used as a speech sensor while a sensor in a headset of the mobile communication device is used as a noise reference sensor, or vice versa. In yet another technique, a secondary sensor in a mobile communication device is configured to be a directional sensor.

Abstract: A watermark generator for providing a watermark signal in dependence on binary message data includes an information processor configured to provide, in dependence on a single message bit of the binary message data, a 2-dimensional spread information representing the message bit in the form of a set of time-frequency-domain values. The watermark generator also includes a watermark signal provider configured to provide the watermark signal on the basis of the 2-dimensional spread information. A watermark detector, methods and computer programs are also described.

Abstract: An auditory display apparatus is provided that places sounds such that sounds whose fundamental frequencies are close to each other are not adjacent to each other. A sound transmission/reception section receives sound data. A sound analysis section analyzes the sound data, and calculates a fundamental frequency of the sound data. A sound placement section compares the fundamental frequency of the sound data with a fundamental frequency of adjacent sound data, and places the sound data such that a difference in fundamental frequency is maximized. A sound management section manages a placement position of the sound data. A sound mixing section mixes the sound data with the adjacent sound data. A sound output section outputs the sound data obtained by the mixture to a sound output device.

Abstract: Implementations and techniques for audio fingerprint differences for end-to-end quality of experience measurement are generally disclosed.

Abstract: A method, apparatus and computer program for: receiving a spatial audio input; determining a direction of interest from the spatial audio input; and generating a spatial audio output dependent on the spatial audio input and the direction of interest.

Abstract: Systems and methods for logging acoustic incidents. An exemplary system includes a transducer configured to receive an electrical signal from an electrical signal source and produce a corresponding acoustic signal, a recording system operable to record the electrical signal from the electrical signal source, and an incident estimation and control system coupled to the recording system to estimate an acoustic incident and to instruct the recording system to record the electrical signal that is producing the acoustic incident. The system can replay the recorded electrical signal through the same transducer that was in use at the time of the acoustic incident, to exactly reproduce the sound that occurred.

Abstract: Peak reduction and power limitations are used to prevent distortion and protect components. In a cellular telephone, peak reduction can be based on battery power level to prevent electrical distortion from saturation. In addition peak reduction can be used to prevent mechanical distortion such as rub and buzz. Dynamic range compression can be used for peak reduction. In another application dynamic range compression can be used to control the power output to protect a speaker from damage. One example of a dynamic range compressor/peak limiter comprises a look-ahead buffer and an analysis engine. For example, the look-ahead buffer holds a window of samples of a signal. The analysis engine selects a gain envelope function on the basis of the samples, for example, by selecting the Pth sample in the buffer whenever that sample exceeds a given threshold.

Abstract: A method and device are presented for identifying machines, such as vehicles, based on acoustic machine signatures. An acoustic sensor generates machine-acoustic data about the machine. A machine-signature identifier receives the machine-acoustic data, determines a machine signature from the machine-acoustic data, and identifies the machine based on the machine signature. A machine-signature database, configured to store multiple machine signatures and/or machine-signature templates, may receive and process queries about machine signatures for machine identification. The machine-signature identifier may generate and send an instruction based on the identified machine. The instruction may instruct application of electronic countermeasures or may permit a vehicle to enter a secured area. An acoustic modulator may generate the machine-acoustic data to act as a machine signature. The acoustic modulator may generate the machine signature using an ultrasonic carrier.

Abstract: A headset having game, chat and microphone audio signals is provided with a programmable signal processor for individually modifying the audio signals and a memory configured to store a plurality of user-selectable signal-processing parameter settings that determine the manner in which the audio signals will be altered by the signal processor. The parameter settings collectively form a preset, and one or more user-operable controls can select and activate a preset from the plurality of presets stored in memory. The parameters stored in the selected preset can be loaded into the signal processor such that the sound characteristics of the audio paths are modified in accordance with the parameter settings in the selected preset.

Abstract: A method comprising receiving an indication of an input indicative of the user not facing the apparatus, determining an ambient sound non-limitation directive associated with the audio uplink based, at least in part, on the input, receiving other audio information, determining ambient sound non-limited audio information based, at least in part, on the other audio information, and causing transmission of the ambient sound non-limited audio information by way of the audio uplink is disclosed.

Abstract: Sound is outputted, without giving the sense of incongruity to the user, even if reproduction of animation is stopped by the user. An animation acquiring section 11 acquires animation data D1 representing an animation generated in advance based on a setting operation by the user, and sound data D2 representing a sound to be reproduced in association with the animation. A sound output control section 12 calculates stop time sound information representing a feature of the sound at a point of time at which reproduction of the animation is stopped, using sound attribute information D4, in response to input of a stop command by the user, and determines a predetermined output method of the sound that matches the animation, based on the calculated stop time sound information, to reproduce the sound by the determined output method.

Abstract: Example methods disclosed herein to monitor media presented by a media device include adjusting, with a circuit, an inbound audio signal input to the media device based on (i) a level of an outbound audio signal generated by the media device and (ii) a level of the inbound audio signal to produce an adjusted inbound audio signal. Such disclosed example methods also include combining, with the circuit, the adjusted inbound audio signal with a sampled version of the outbound audio signal to determine a return signal. Such disclosed example methods further include processing the return signal to determine first identification information to identify first presented media having an audio portion corresponding to the outbound audio signal generated by the media device.

Abstract: A low power sound recognition sensor is configured to receive an analog signal that may contain a signature sound. Sparse sound parameter information is extracted from the analog signal. The extracted sparse sound parameter information is processed using a sound signature database stored in the sound recognition sensor to identify sounds or speech contained in the analog signal, wherein the sound signature database comprises a plurality of sound signatures each representing an entire word or multiword phrase.

Abstract: A method for adjusting the performance of an electroacoustic transducer includes receiving, by gain adjustment circuit, a displacement signal corresponding to a relative motion between a magnetic structure of the electroacoustic transducer and a voice coil of the electroacoustic transducer. The method includes detecting, by the gain adjustment circuit, a displacement signal value of the displacement signal as one of meeting or exceeding a displacement signal threshold. The method includes modifying, by the gain adjustment circuit, a loop gain of an active noise reduction loop associated with the electroacoustic transducer when the displacement signal value of the displacement signal one of meets or exceeds the displacement signal threshold.

Abstract: Systems and methods of alert and notification transmission are contemplated in which a transceiver provides an alert/notification to a user(s) wearing a headset that provides hearing protection device with an integrated transceiver (e.g., radio) and on-board siren-voice alert function board. Typically, this device would be used in a loud noise area (>+70 dB) that would otherwise prevent the user from hearing an alert and notification, such as a siren tone and/or Public Address announcement. While the alert and notification is silent, the device is contemplated to provide sufficient hearing protection (>+15 dB)from the ambient surroundings. It is also contemplated that it could play local music through an audio input port/other receiving method. When an alert and notification command is communicated to a headset device's address (or globally) through the transceiver, the device(s) would then proceed to play, within the headset, the desired alert and notification message.

Abstract: An arrangement and method for assessing the audibility and annoyance of at least one distortion component dn(t) with n=1, . . . , N in the output signal p(t) of a device under test, by generating a virtual auralization output signal pA(t) at the output of an auralization system. The output signal pA(t) contains the distortion component dn(t) at an adjustable magnitude according to a scaling factor Sn provided from a control input, and is supplied to a perceptive model and to a reproduction system used by a listener. The auralization system receives the distortion component dn(t) from a separator which receives a test signal xT(t) from the output of a microphone and a reference signal xR(t) from a reference system.

Abstract: A watermark signal provider provides a watermark signal suitable for being hidden in an audio signal when the watermark signal is added to the audio signal, such that the watermark signal represents watermark data. The watermark signal provider includes a psychoacoustical processor for determining a masking threshold of the audio signal; and a modulator for generating the watermark signal from a superposition of sample-shaping functions spaced apart from each other at a sample time interval of a time-discrete representation of the watermark data, each sample-shaping function being amplitude-weighted with a respective sample of the time-discrete representation, multiplied by a respective amplitude weight depending on the masking threshold, the modulator being configured such that the sample time interval is shorter than a time extension of the sample-shaping functions; and the respective amplitude weight also depends on samples of the time-discrete representation neighboring the respective sample in time.

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: Improvements in a sound activated music accessory device are disclosed where the accessory device further enhances the listening experience by providing an interaction device. The device can be an item that is seen in the movie or music video and can be included with the CD/DVD or can be sold as a separate item. The activation and or deactivation are synchronized with the embedded sound that is placed into the soundtrack. The sound, tone, frequency or complex blended sound or sounds are subtle or non-obvious to a person listening to the music but are detected by the accessory device to begin a sequence or stop a sequence to correspond the a particular part of the soundtrack. The activation can mimic what is occurring in a movie, video or soundtrack. The accessory device to be a bracelet, necklace, hat, headband, apparel, plush or rigid toy.

Abstract: The present invention relates to a signal processing apparatus and method, a program, and a data recording medium configured such that the playback level of an audio signal can be easily and effectively enhanced without requiring prior analysis. An analyzer 21 generates mapping control information in the form of the root mean square of samples in a given segment of a supplied audio signal. A mapping processor 22 takes a nonlinear function determined by the mapping control information taken as a mapping function, and conducts amplitude conversion on a supplied audio signal using the mapping function. In this way, by conducting amplitude conversion of an audio signal using a nonlinear function that changes according to the characteristics in respective segments of an audio signal, the playback level of an audio signal can be easily and effectively enhanced without requiring prior analysis. The present invention may be applied to portable playback apparatus.

Abstract: A personal audio device, such as a wireless telephone, includes noise canceling circuit that adaptively generates an anti-noise signal from a reference microphone signal and injects the anti-noise signal into the speaker or other transducer output to cause cancellation of ambient audio sounds. An error microphone may also be provided proximate the speaker to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit uses the reference and/or error microphone, optionally along with a microphone provided for capturing near-end speech, to determine whether one of the reference or error microphones is obstructed by comparing their received signal content and takes action to avoid generation of erroneous anti-noise.

Abstract: A microphone array apparatus includes: an acquisition unit configured to acquire samples from a sound signal inputted from each of a plurality of microphones, at predetermined time intervals; an operation unit configured to calculate a value based on volumes of the sound signal possessed by a plurality of the samples for each of the sound signals inputted from the plurality of microphones; a correlation coefficient calculator configured to calculate a coefficient of correlation between the sound signals, on the basis of the values calculated for the respective sound signals; and a gain calculator configured to calculate reduction gain for the sound signals inputted from the plurality of microphones, on the basis of the coefficient of correlation.

Abstract: A system for controlling an electronic device, including an accessory coupled to the device having a control interface for receiving at least one input and a tone generator, the accessory configured to, in response to an input, generate a corresponding control tone using the tone generator, and send the control tone and a calibration sequence to the device. The device is operable to receive the calibration sequence and control tone. The device includes a zero-crossing detector operable to determine at least one of the start or ending of each of the calibration sequence and the control tone and having a circuit operable to determine the periods of the calibration sequence and the control tone based on the zero-crossing detector output.

Abstract: There is provided a sound detecting apparatus including: a feature value extracting unit which extracts a feature value per every predetermined time from an input time signal; a feature value maintaining unit which maintains a feature value sequence of a predetermined number of detection target sound items; and a comparison unit which respectively compares a feature value sequence extracted by the feature value extracting unit with a feature value sequence of the maintained predetermined number of detecting target sound items and obtains detection results of the predetermined number of detection target sound items every time the feature value extracting unit newly extracts a feature value, wherein the feature value extracting unit includes a time frequency transform unit and a likelihood distribution detecting unit, smooths the obtained likelihood distribution in frequency and time directions, and extracts a feature value per the predetermined time.

Abstract: Systems and methods are provided herein relating to audio matching. A compact digest can be generated based on sets of triples, where triples are groupings of three interest points that meet threshold criteria. The compact digest can be used in identifying a potential audio match. A full digest can then be used in verifying the potential match. By using a compact digest to perform audio matching, the audio matching system can be scaled to encompass millions or billions of reference audio samples while still using the full digest to maintain accuracy.

Abstract: Audio systems for a vehicle and methods for increasing auditory situation awareness in a vehicle are provided. An audio system includes at least one ambient microphone disposed on the vehicle, a processor and at least one loudspeaker. The at least one ambient microphone is configured to capture ambient sound external to the vehicle and to produce an ambient sound signal. The processor is configured to receive the ambient sound signal and an audio content signal, and to mix the ambient sound signal with the audio content signal to generate a mixed output signal. The at least one loudspeaker is configured to reproduce the mixed output signal in the vehicle cabin.

Abstract: Methods and systems are provided for receiving by a first electronic device audio content, determining audio quality of the received audio content, which includes measuring and/or estimating quality related parameters associated with the received audio content, and communicating the quality related parameters to a second electronic device, from which the audio content was transmitted. The quality related parameters may be used to control audio related functions in the second electronic device. The audio related functions may comprise noise reduction, noise suppression, cancellation, distortion reduction, equalization, compression, enhancement and/or audio rate conversion. Determining the audio quality of the received audio content may be based on one or more noise related thresholds. The quality related parameters may be communicated to the second electronic device over a dedicated control channel setup between the devices.

Abstract: A method of enhancing audio sound. The method includes sensing an acoustic signal using a microphone in an electronic device. The acoustic signal is emitted in response to a primary sound signal and transmitted as a sound wave through a space. The method further includes receiving, using an antenna in the electronic device, a wireless signal encoded with the primary sound signal. An impulse response for the space is estimated based on the sensed acoustic signal and the primary sound signal encoded within the received wireless signal. A delay between the sensed acoustic signal and the primary sound signal encoded within the received wireless signal is calculated based on the estimated impulse response. The primary sound signal encoded within the received wireless signal is delayed using the calculated delay and reproduced to enhance the acoustic signal heard by a user of the electronic device.

Abstract: An exemplary uncomfortable sound pressure evaluation system consecutively presents sound stimulation groups to a user. Each sound stimulation group includes a sound stimulation, and the sound stimulation groups differ in frequency from one another. The system includes: an extraction section configured to extract, for each sound stimulation group, information concerning event-related potential relating to the sound stimulation; an estimation section configured to estimate an uncomfortable sound pressure (UCL) of the user from the information concerning event-related potential; and a correction section configured to determine whether the estimated UCL is higher than a predefined maximum UCL or not, and if a proportion of those determination results which indicate the UCL to be higher than the maximum UCL is smaller than predetermined, correcting each UCL determined as higher than the maximum UCL to a sound pressure equal to or less than the maximum UCL.

Abstract: A method and a device for determining an amplitude noise and/or phase noise contained within a signal performs a quadrature mixing of the signal containing the amplitude noise and/or the phase noise into the baseband or intermediate-frequency band and an amplitude demodulation downstream of the quadrature mixing in order to determine the amplitude noise contained within the signal and a phase demodulation downstream of the quadrature mixing in order to determine the phase noise contained within the signal.

Abstract: A system and method for detection of sound of interest amongst plurality of other dynamically varying sounds is disclosed. In one embodiment, a spectrum detector identifies dominant spectrum energy frequency by detecting the dominant spectrum energy band present in spectrum of sound energy. A modified mel filter bank is designed by revising spectral positioning of the first mel filter bank and the second mel filter bank according to the identified dominant frequency. A feature extractor extracts the features from first mel filter bank, second mel filter bank and the modified mel filter bank which are further classified in order to detect the sound of interest.

Abstract: In one arrangement, a person's position in a movie auditorium is discerned by detection of audio signals from two or more speakers, and the person is then assigned a role or task in a game based on this discerned location. In another arrangement, a person's current position in a home listening environment is similarly discerned; this position information is used to control the balance of complementary audio channels of a home audio system, so that the audio “sweet spot” is located at the person's current position. A great number of other features and arrangements are also detailed.

Abstract: Methods and systems are provided for enhancing speech intelligibility in electronic devices. During outputting of acoustic signal via an electronic device, measurement of forces applied by user of the electronic device against the device (or enclosure thereof) may be obtained. The force measurements may be used to assess and/or estimate the listening intelligibility experienced by the user. Further, the force measurements may be used to control or adjust a listening intelligibility stage applied during generation and/or processing of the acoustic signals that are outputted via the electronic device. In some instances, an audio input, corresponding to ambient noise affecting intelligibility, may be obtained, and may be used to control or assist in controlling the listening intelligibility stage.

Abstract: Disclosed herein are systems, methods, and non-transitory computer-readable storage media for automatically analyzing, modifying, and mixing a plurality of audio signals. The modification of the audio signals takes place to avoid spectral collisions which occur when more than one signal simultaneously occupies one or more of the same frequency bands. The modifications mask out some signals to allow others to exist unaffected. Also disclosed herein is a method for displaying the identified spectral collisions superimposed on graphical waveform representations of the analyzed signals.

Abstract: Device for localizing acoustic sources and/or measuring their intensities, intended in particular to be used in confined spaces, the device including at least one acoustic antenna having a first series of microphones delimiting a first volume, and a second series of microphones delimiting a volume containing at least substantially the first series of microphones.

Abstract: Methods and apparatuses for comparing a level of a signal carrying ambient acoustic noise with a threshold level; and based on results of the comparison, ignoring time intervals identified as noise burst in estimating ambient noise levels.

Abstract: Unlike sound based pressure waves that go everywhere, air turbulence caused by wind is usually a fairly local event. Therefore, in a system that utilizes two or more spatially separated microphones to pick up sound signals (e.g., speech), wind noise picked up by one of the microphones often will not be picked up (or at least not to the same extent) by the other microphone(s). Embodiments of methods and apparatuses that utilize this fact and others to effectively detect and suppress wind noise using multiple microphones that are spatially separated are described.

Abstract: There is provided an information processing apparatus including an information amount gain calculation unit that, on the assumption that a factor that could cause a state transition in a state transition model occurs and the state transition takes place as a result of an occurrence of the factor, determines an information amount gain, which is an information amount obtained by causing the factor to occur regarding a causal relationship between the factor and the state transition and observing a result thereof, an evaluation value calculation unit that determines an evaluation value to evaluate effectiveness of causing each of the factors to occur based on the information amount gain to acquire the causal relationship between the factor and the state transition, and a decision unit that decides the factor to be occurred based on the evaluation value.

Abstract: The selective filtering earplugs allow a user to filter most external and environmental sounds, only allowing a set of pre-approved sounds to be heard by the user. The earplugs include a sound acquisition module and at least one filtering earplug. The sound acquisition module includes a microphone for receiving environmental sounds and a controller for converting the received environmental sounds to a digital signal. First memory is provided for storing the digital signal and second memory is provided for storing a database of pre-approved sound signals. The controller compares the digital signal with the database of pre-authorized sound signals. If the digital signal matches a pre-authorized sound signal of the database, then a transmitter transmits the digital signal. The filtering earplug includes a receiver for receiving the digital signal and a converter for converting the digital signal into an audio signal.

Abstract: In one aspect, the invention divides an audio signal into auditory events, each of which tends to be perceived as separate and distinct, by calculating the spectral content of successive time blocks of the audio signal, calculating the difference in spectral content between successive time blocks of the audio signal, and identifying an auditory event boundary as the boundary between successive time blocks when the difference in the spectral content between such successive time blocks exceeds a threshold. In another aspect, the invention generates a reduced-information representation of an audio signal by dividing an audio signal into auditory events, each of which tends to be perceived as separate and distinct, and formatting and storing information relating to the auditory events. Optionally, the invention may also assign a characteristic to one or more of the auditory events. Auditory events may be determined according to the first aspect of the invention or by another method.