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.

Abstract: It is inter alia disclosed to check whether an audio signal captured from an environment of the apparatus comprises arriving sound from an audio source of interest, and to provide a direction identifier being indicative on the direction of the arriving sound from the audio source of interest via a user interface when said check yields a positive result.

Abstract: A system and method may be provided to segment and/or classify an audio signal from transformed audio information. Transformed audio information representing a sound may be obtained. The transformed audio information may specify magnitude of a coefficient related to energy amplitude as a function of frequency for the audio signal and time. Features associated with the audio signal may be obtained from the transformed audio information. Individual ones of the features may be associated with a feature score relative to a predetermined speaker model. An aggregate score may be obtained based on the feature scores according to a weighting scheme. The weighting scheme may be associated with a noise and/or SNR estimation. The aggregate score may be used for segmentation to identify portions of the audio signal containing speech of one or more different speakers. For classification, the aggregate score may be used to determine a likely speaker model to identify a source of the sound in the audio signal.

Abstract: A system and method are provided for processing sound signals. The processing may include identifying individual harmonic sounds represented in sound signals, determining sound parameters of harmonic sounds, classifying harmonic sounds according to source, and/or other processing. The processing may include transforming the sound signals (or portions thereof) into a space which expresses a transform coefficient as a function of frequency and chirp rate. This may facilitate leveraging of the fact that the individual harmonics of a single harmonic sound may have a common pitch velocity (which is related to the chirp rate) across all of its harmonics in order to distinguish an the harmonic sound from other sounds (harmonic and/or non-harmonic) and/or noise.

Abstract: SPL monitoring systems are provided. A SPL monitoring system includes an audio transducer configured to receive sound pressure, a logic circuit which calculates a safe time duration over which a user can receive current sound pressure values and an indicator element which produces a notification when an indicator level occurs. An SPL monitoring information system includes a database which stores data such as a list of earpiece devices and associated instrument response functions. The logic circuit compares a request with the data in the database and retrieves a subset of data and sends it to an output control unit. The output control unit sends the subset of data to a sending unit.

Abstract: Methods of operating an audio device are provided. A method includes calculating estimated sound pressure levels (SPLs) for drive signals directed to an ear canal receiver (ECR) during a time increment ?t; calculating an estimated SPL_Dose during the time increment ?t using the estimated sound pressure levels; and calculating a total SPL_Dose at a time t of the audio device using the estimated SPL_Dose.

Abstract: The bandwidth of a codec is compared with a determined bandwidth of the output signal produced by the codec. If the bandwidths match, then an audio profile of the apparatus is set to correspond to the established bandwidths and a corresponding icon is rendered on a display of the apparatus. If the signal bandwidth is narrower, than the codec bandwidth, then the audio profile of the apparatus is set to a narrower audio profile and a corresponding icon is rendered on the display of the apparatus. The bandwidth of the signal processed by the codec may be determined in several different ways and the narrower audio profile includes reduced bandwidth, reduced gain, or reduced dynamic range.

Abstract: A method determines the classification of an audio signal in dependence on a comparison of two difference sums of audio features over time periods of different length. Thus, an adequately exact yet quickly reacting adaptation of the classification in changing hearing situations is ensured. The method is advantageously used in a hearing aid. The audio signal is processed in different ways on the basis of the classification.

Abstract: In a sound source detection apparatus that detects a sound source of a detection subject on the basis of collected sounds, sounds are collected by at least one sound collector, an autocorrelation between sounds collected in time series by the sound collector is calculated, and a determination as to whether or not the sound source of the detection subject exists is made on the basis of the autocorrelation. More particularly, sounds are preferably collected respectively by two or more sound collectors such that the existence of the sound source of the detection subject is determined by determining whether or not the autocorrelations of the sounds collected by the two or more sound collectors satisfy a predetermined condition. By using the autocorrelation to detect the sound source of the detection subject in this manner, high robustness against an S/N ratio is exhibited, leading to an improved detection performance when detecting the sound source of the detection subject.

Abstract: An audio signal measurement method for a speaker and an electronic apparatus having the speaker are provided. The electronic apparatus further has a processing circuit and a power amplifier. The processing circuit is coupled to the speaker and configured to execute a time domain to frequency domain transform according to a voltage value of an audio signal and a current value of current feedback from the speaker so as to obtain a frequency response curve. The power amplifier is coupled to the speaker and configured to drive the speaker according the voltage value of the audio signal. The processing circuit is capable of determining whether the frequency response curve is located within a predetermined area such that the processing circuit generates a signal when the frequency response curve is located out of the predetermined area. Thereby, the electronic apparatus may measure its transducer distortion and acoustic box leakage.

Abstract: Certain aspects relate to providing an at least one audio source to at least one user. Certain aspects relate to selectively modifying an at least one first sound source to be provided to the at least one user, wherein the at least one first sound source is combined with an at least one second sound source, and wherein the selectively modifying is performed relative to the at least one audio source based at least in part on at least some specific information of the at least one first sound source. Other aspects relate to selectively modifying the at least one first sound source to be provided to the at least one user relative to the at least one second sound source based at least in part on at least some specific information of the at least one first sound source.

Abstract: An alert system for generating a signal indicating at least one aspect of sound within a workspace environment, the system comprising at least a first sensor positioned one of within and proximate a workspace, the at least a first sensor sensing at least one aspect of sound generated within the workspace and generating a signal indicating the at least one aspect, a communication device located within the workspace, the communication device configured to generate a signal perceivable within the workspace and a processor receiving signals from the at least a first sensor and linked to the communication device, the processor programmed to perform the steps of identifying when the at least one aspect exceeds a threshold value and driving the communication device to indicate that the at least one aspect exceeds the threshold value.

Abstract: Embodiments of the present invention disclose a method, computer program product, and system for real-time determination of attentiveness of an audience in a room to a speaker. A computer determines amounts of wireless activity of a computing device within the room of the audience over time during the presentation. Based in part on the amounts of wireless activity of the computing device, the computer determines and initiates display of measures of attentiveness of the audience over the time during the presentation, to display changes in the measures of attentiveness of the audience to the speaker during the presentation. In another embodiment, the computer determines amounts of attentiveness of the audience within the room over time during the presentation utilizing sensors located throughout the room.

Abstract: A sound signal processing device that is capable of suitably extracting main sound from mixed sound in which unnecessary sound (for example, leakage sound and reverberant sound) is mixed with the main sound. More specifically, a mixed sound signal in the time domain including first sound and second sound, and a target sound signal in the time domain including sound corresponding to at least the second sound, which have temporal relation in their entirety or in part, are each divided into a plurality of frequency bands. A level ratio between the two signals is calculated at each frequency. Based on the level ratio, a signal of the first sound that is included in the mixed sound signal is extracted.

Abstract: A loudspeaker control apparatus of the present invention includes: an operation reception section that receives a user operation; a mode selection section that selects between two operation modes, based on the user operation, either a normal mode in which operations of a plurality of loudspeakers and lights are controlled independently or an inspection mode in which each of the plurality of loudspeakers is inspected to see whether it is operating normally; and a control section that controls the operations of the plurality of loudspeakers and lights based on the operation mode, and the control section, in the inspection mode, sequentially causes the plurality of loudspeakers to output an inspection tone and sequentially causes lights which correspond to the loudspeakers from which the inspection tone is being outputted to be turned on or to blink.

Abstract: A process for scanning and/or synchronizing audio/video events is described. According to the process, a signal is acquired and divided into a plurality of segments corresponding to different moments of the signal. A spectrogram is generated and peaks are located in the spectrogram. Transition peaks are located among said peaks, and the bands of such transition peaks are combined in one or more transitions to which hashes correspond. The hashes are associated with the time at which the transitions occur in the signal. Means for scanning and/or synchronizing audio/video events are also disclosed.

Abstract: The suppression of off-axis audio in an audio environment is provided. Off-axis audio may be considered audio that does not originate from a region of interest. The off-axis audio is suppressed by comparing a phase difference between signals from two microphones to a target slope of the phase difference between signals originating from the region of interest. The target slope can be adapted to allow the region of interest to move with the location of a human speaker such as a driver.

Abstract: An instrument panel for a vehicle that includes a housing, a microphone and an actuator. The microphone is configured to detect a characteristic sound in the housing. The actuator is coupled to the housing, and the actuator is configured to emit the characteristic sound into the housing when actuated.

Abstract: A disclosed method selects a plurality of fewer than all of the channels of a multichannel signal, based on information relating to the direction of arrival of at least one frequency component of the multichannel signal.

Abstract: Systems and methods are provided herein relating to interactive gaming within a media sharing service. Game data, such as sets of notes extracted from the audio track of user generated videos or from audio samples, can be generated based on videos containing musical content or from audio content. A device can use the game data to facilitate an interactive game during playback of the user generated videos or audio samples. Players can press buttons, for example, corresponding to notes as the video with musical content is played within the game interface. Players can be scored for accuracy, and can play with other players in a multiplayer environment. In this sense, user generated video content or audio content can be transformed and used within a gaming interface to increase interaction and engagement between users in a media sharing service.

Abstract: The present document relates to audio source coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), as well as to digital effect processors, e.g. exciters, where generation of harmonic distortion add brightness to the processed signal, and to time stretchers where a signal duration is prolonged with maintained spectral content. A system and method configured to generate a time stretched and/or frequency transposed signal from an input signal is described. The system comprises an analysis filterbank (101) configured to provide an analysis subband signal from the input signal; wherein the analysis subband signal comprises a plurality of complex valued analysis samples, each having a phase and a magnitude. Furthermore, the system comprises a subband processing unit (102) configured to determine a synthesis subband signal from the analysis subband signal using a subband transposition factor Q and a subband stretch factor 5?.

Abstract: An earpiece (100) and a method (640) for acoustic management of multiple microphones is provided. The method can include capturing an ambient acoustic signal from an Ambient Sound Microphone (ASM) to produce an electronic ambient signal, capturing in an ear canal an internal sound from an Ear Canal Microphone (ECM) to produce an electronic internal signal, measuring a background noise signal, and mixing the electronic ambient signal with the electronic internal signal in a ratio dependent on the background noise signal to produce a mixed signal. The mixing can adjust an internal gain of the electronic internal signal and an external gain of the electronic ambient signal based on the background noise characteristics. The mixing can account for an acoustic attenuation level and an audio content level of the earpiece.

Abstract: Provided are a method for estimating a spectrum density of diffused noises. Also provided is a processor for implementing the method. The processor includes at least two sound receiving units and a spectrum density estimating unit for estimating spectrum density.

Abstract: A method includes receiving, at a processor, a first data frame at a first time from a first microphone. The method also includes receiving a second data frame at the first time from a second microphone. The method further includes calculating a power ratio of the first microphone and the second microphone based on the first data frame and the second data frame in response to determining that the first data frame and the second data frame are noise data frames.

Abstract: Psychoacoustic Bass Enhancement (PBE) is integrated with one or more other audio processing techniques, such as active noise cancellation (ANC), and/or receive voice enhancement (RVE), leveraging each technique to achieve improved audio output. This approach can be advantageous for improving the performance of headset speakers, which often lack adequate low-frequency response to effectively support ANC.

Abstract: A difference signal calculating unit of a noise detecting device calculates a difference between the amplitudes of a residual signal at each sample timing and a residual signal at the preceding sample timing. A difference signal comparing unit determines whether or not an impulsive noise is present on the basis of the difference signal at the current sample timing, and the difference signal at each sample timing within a predetermined duration from the current sample timing.

Abstract: Quality of industrial products is evaluated by evaluating non-stationary operation sound, which is a kind of operation sound, from an aspect of tone, using closely simulated evaluation levels of evaluation of non-stationary sound by used of a human sense of hearing. An operation sound of a conforming product sample is converted into sound waveform data by a sound collecting unit, and the sound waveform data is input into a computer via an A-D converter, and then converted into psychoacoustic parameters. Data of pseudo conforming products is additionally obtained from the psychoacoustic parameters of a plurality of conforming product samples by making use of deviation in the data of the conforming product samples. Threshold data is obtained using thresholds and masking data for evaluation calculated from psychoacoustic parameters of data of the conforming product samples and the pseudo conforming products by a statistical technique.

Abstract: A disclosed example method to determine a location of an audience member involves generating a correlation analysis result based on correlating first audio samples from a stationary audio detector with second audio samples from a portable audio detector carried by the audience member and determining via a neural network the location of the audience member based on the correlation analysis result.

Abstract: A device, system and method for enabling the measuring of audiences to media consumed through the use of digital media terminals. An example disclosed herein includes a loop-back device that allows the production/extraction of content identification information from media items rendered by a digital media terminal that can be seamlessly and homogenously implemented in a wide variety of media distribution platforms and devices, including Internet and mobile networks.

Abstract: Method and devices for providing surround audio signals are provided. Surround audio signals are received and are binaurally filtered by at least one filter unit. In some embodiments, the input surround audio signals are also processed by at least one equalizing unit. In those embodiments, the binaurally filtered signals and the equalized signals are combined to form output signals.

Abstract: An audio processing apparatus includes an audio signal acquisition unit which acquires an audio signal of a musical piece, a feature value extraction unit which extracts a predetermined type of feature value from the audio signal acquired by the audio signal acquisition unit in time series, a change point detection unit which detects a change point in which the amount of change of the feature value extracted in time series by the feature value extraction unit is changed to be greater than a predetermined threshold value, a hook analysis unit which analyzes a hook place of the audio signal based on the feature value extracted by the feature value extraction unit in block units with the change point detected by the change point detection unit as a boundary, and a hook information output unit which outputs the hook place analyzed by the hook analysis unit as hook information.

Abstract: Embodiments herein provide analysis and transmission of environmental sounds (e.g., outdoor sounds generated from an environment surrounding an enclosure) to an indoor occupant. Specifically, a sound analyzer is configured to receive a sensor output containing the environmental sounds, and analyze, in real-time, the environmental sounds to determine attributes of each of the environmental sounds. Each attribute is then compared to a collection of known sounds, and classified based on the comparison. The environmental sounds are then output and transmitted to the indoor occupant as at least one of: an audio transmission containing one or more of the environmental sounds, and a display of information about the one or more environmental sounds. The environmental sounds can be broadcast to the indoor occupant in real-time, and/or may be recorded for later analysis and use.

Abstract: A sound signal processing apparatus includes an observed signal analysis unit that receives as an observed signal a sound signal for channels obtained by a sound signal input unit formed of microphones and estimates a sound direction and a sound segment of a target sound which is sound to be extracted and a sound source extraction unit that receives the sound direction and sound segment of the target sound estimated by the observed signal analysis unit and extracts the sound signal for the target sound. The observed signal analysis unit includes a short time Fourier transform unit that generates an observed signal in time-frequency domain by applying short time Fourier transform to the sound signal for the channels received and a direction/segment estimation unit that receives the observed signal generated by the short time Fourier transform unit and detects the sound direction and sound segment of the target sound.

Abstract: A method and apparatus to evaluate a quality of an audio signal, in which the number of effective channels is determined for each of a reference signal of a current frame and a test signal indicative of the reference signal that has passed through an audio codec, and an audio quality evaluation score of the current frame is calculated by evaluating an audio quality of the current frame based on the determined number of effective channels for each of the reference signal and the test signal by means of a predetermined evaluator.

Abstract: A portable device for use with audio or visual content (e.g., an MP3 player-like device, or a camera-enabled device) is equipped with one or more features/capabilities, including: a user interface through which a consumer can instruct that content be delivered to a different consumer's device, the ability to derive identifiers from content and take action(s) based thereon, the ability to count playbacks of content; and/or the ability to interact with remote web services. A variety of other features and arrangements are also detailed.

Abstract: The present invention discloses a vision-aided hearing assisting device, which includes a display device, a microphone and a processing unit. The processing unit includes a receiving module, a message generating module and a display driving module. The processing unit is electrically connected to the display device and the microphone. The receiving module receives a surrounding sound signal, which is generated by the microphone. The message generating module analyzes the surrounding sound signal according to a present-scenario mode to generate a related message related with the surrounding sound signal. The display driving module drives the display device to display the related message.

Abstract: The present invention relates to a method and system for analysing audio (eg. music) tracks. A predictive model of the neuro-physiological functioning and response to sounds by one or more of the human lower cortical, limbic and subcortical regions in the brain is described. Sounds are analysed so that appropriate sounds can be selected and played to a listener in order to stimulate and/or manipulate neuro-physiological arousal in that listener. The method and system are particularly applicable to applications harnessing a biofeedback resource.

Abstract: A method of compensating for noise in a receiver having a first receiver unit and a second receiver unit, the method includes receiving a first transmission at the first receiver unit, the first transmission having a first signal component and a first noise component; receiving a second transmission at the second receive unit, the second transmission having a second signal component and a second noise component; determining whether the first noise component and the second noise component are incoherent and; only if it is determined that the first and second noise components are incoherent, processing the first and second transmissions in a first processing path, wherein the first processing path is configured to compensate for incoherent noise.

Abstract: A system and method allow for extracting target signals from the overall measurement data without knowing how and where these data are collected, the locations and characteristics of target sources as well as those of random background noise sources. In essence, it uses an innovative and advanced signal processing technique to reveal certain critical information from a mixture of data that is hard to obtain otherwise. In particular, it allows for denoising the measured data that have been contaminated by various interfering and background noise, making it possible to extract certain target information that may be otherwise difficult to observe. The only assumption made in this method is that the target signal is incoherent with respect to all interfering signals and background noise. The more information about a target signal is available, the more complete the extracted signal becomes.

Abstract: An original loudness level of an audio signal is maintained for a mobile device while maintaining sound quality as good as possible and protecting the loudspeaker used in the mobile device. The loudness of an audio (e.g., speech) signal may be maximized while controlling the excursion of the diaphragm of the loudspeaker (in a mobile device) to stay within the allowed range. In an implementation, the peak excursion is predicted (e.g., estimated) using the input signal and an excursion transfer function. The signal may then be modified to limit the excursion and to maximize loudness.

Abstract: An apparatus may include an acoustic transducer, a housing, a microphone, and an acoustical conduit. The acoustic transducer may include a diaphragm having a front and a back, the diaphragm configured to mechanically vibrate in response to an audio signal, thereby producing sound from the front of the diaphragm. The housing may be configured to mechanically support the acoustic transducer such that the front faces an exterior of the housing and the back faces an interior of the housing. The microphone may be disposed in the interior of the housing and may be configured to sense combined sound produced by the acoustic transducer and ambient sound proximate to the acoustic transducer. The acoustical conduit may be coupled to and extend from the microphone and pass adjacent the acoustic transducer such that the microphone senses sound proximate to the front of the diaphragm.

Abstract: A portable terminal, a hearing aid, and a method of indicating the position of a sound source in the portable terminal are provided. The method includes receiving sound signals of a plurality of sound sources, determining respective distances between the plurality of sound sources and the portable terminal, and respective directions of the plurality of sound sources from the portable terminal, based on the received sound signals of the plurality of sound sources, determining positions of the plurality of sound sources on the touch screen based on the determined distances and directions of the plurality of sound sources, and displaying the determined positions of the plurality of sound sources on the touch screen.

Abstract: A personal noise monitoring system and method for operating the same. Sound pressure data from inside the ears of a user of hearing protection device (HPD) are acquired from microphones in the HPD and reported to a master control unit. Simultaneously, the master control unit receives sound pressure data from the ambient environment in which the user is located. The data are reported to a central server for processing. The central server may use the data to determine in near real time or real time whether the user of the HPD is at risk for exposure to excess sound levels and to take remedial action while the user is still in the ambient environment.

Abstract: Automatic measurements are made of audio presence and level in an audio signal by direct processing of an MPEG data stream representing the audio signal, without reconstructing the audio signal. Sub-band data is extracted from the data stream, and the extracted sub-band data is dequantized and denormalized. An audio level for the dequantized and denormalized sub-band data is measured without reconstructing the audio signal. Channel characteristics are used in measuring the audio level of the sub-band data, wherein the channel characteristics are used to weight the measured levels. The measured levels are compared against at least one threshold to determine whether an alarm should be triggered.

Abstract: A system and method may be provided to segment and/or classify an audio signal from transformed audio information. Transformed audio information representing a sound may be obtained. The transformed audio information may specify magnitude of a coefficient related to energy amplitude as a function of frequency for the audio signal and time. Features associated with the audio signal may be obtained from the transformed audio information. Individual ones of the features may be associated with a feature score relative to a predetermined speaker model. An aggregate score may be obtained based on the feature scores according to a weighting scheme. The weighting scheme may be associated with a noise and/or SNR estimation. The aggregate score may be used for segmentation to identify portions of the audio signal containing speech of one or more different speakers. For classification, the aggregate score may be used to determine a likely speaker model to identify a source of the sound in the audio signal.

Abstract: The present invention discloses a microphone detection circuit for detecting whether an external device includes a microphone. The microphone detection circuit comprises: an audio circuit for receiving an analog input signal which is a signal from the external device or a preset signal. The audio circuit comprises: an analog-to-digital conversion unit to generate a digital audio signal according to the analog input signal; a decision unit, coupled to the analog-to-digital conversion unit, to determine whether the digital audio signal satisfies a predetermined threshold to thereby generate an analysis result, wherein if the digital audio signal satisfies the predetermined threshold, the analysis result indicates that the external device includes a microphone; and a control unit, coupled to the decision unit, to control an operation of the audio circuit according to the analysis result.

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.

Abstract: A signal correction apparatus receives an input audio signal (serving as a first sound reception means). The signal correction apparatus computes, at every frequency, first power that indicates magnitude of sound represented by the input audio signal (serving as a first power computation means). The signal correction apparatus estimates a correction function that is a continuous function defining a relation between each frequency and a correction coefficient used to approximate the first power computed at that frequency to the reference power predetermined for that frequency (serving as a correction function estimation means). The signal correction apparatus multiplies the computed first power by the correction coefficient acquired in accordance with the relation defined by the estimated correction function so as to correct the first power at every frequency (serving as a power correcting means).

Abstract: A home theater system, an audio output adjusting device of a home theater system and methods thereof allow a prescribed or three-dimension audio configuration state for at least one prescribed position even when the prescribed location and/or a configuration of the speakers in a home theater system are modified. The device can include a checker for analyzing at least one of volume and delay time of a test audio signal outputted from speakers and providing the analyzed information to a controller. The controller can control output level and timing of audio signals subsequently output from the speakers based on the analyzed information.