Abstract: Systems and methods are described that perform audio source localization in a manner that provides increased robustness and responsiveness in the presence of acoustic echo. The systems and methods calculate a difference between a signal level associated with one or more of the audio signals generated by a microphone array and an estimated level of acoustic echo associated with one or more of the audio signals. This information is then used to determine whether and/or how to perform audio source localization. For example, a controller may use the difference to determine whether or not to freeze an audio source localization module that operates on the audio signals. As another example, the audio source localization module may incorporate the difference (or the estimated level of acoustic echo used to calculate the difference) into the logic that is used to determine the location of a desired audio source.

Abstract: Systems and methods may provide for using an audio front end of a mobile device to sampled audio from an audio signal during a first portion of a periodic detection window, and reducing a power consumption of one or more components of the audio front end during a second portion of the periodic detection window. Additionally, a determination may be made as to whether voice activity is present in the audio signal based at least in part on the sampled audio. In one example, the length of the first portion and the length of the second portion are defined by a duty cycle of the periodic detection window.

Abstract: There is provided a notification control device including a determination unit configured to determine whether a sound collection unit configured to collect voice in a microphone is blocked, and a notification controller configured to cause, when it is determined that the sound collection unit of the microphone is blocked, a determination result to be reported to a user. When a voice signal indicating the voice collected by the microphone is recorded, the notification controller causes the determination result to be reported to the user in one or more notification ways not including an auditory notification way. When the voice signal is not recorded, the notification controller causes the determination result to be reported to the user in one or more notification ways from among one or more notification ways including the auditory notification way.

Abstract: A system for removing noise from an audio signal is described. For example, noise caused by content playing in the background during a voice command or phone call may be removed from the audio signal representing the voice command or phone call. By removing noise, the signal to noise ratio of the audio signal may be improved.

Abstract: Methods, apparatus, and articles of manufacture for reducing spillover in a media monitoring system are disclosed. An example method includes identifying media associated with media monitoring data. The media monitoring data is received from a first meter associated with a first media presentation device. The example method includes identifying an expected sound pressure level associated with the first meter. The example method includes comparing the expected sound pressure level to an actual sound pressure level collected from the media by the first meter to determine if spillover occurred. The example method includes crediting the media as a media exposure if spillover did not occur.

Abstract: A communication component modifies production of an audio waveform at determined modification segments to thereby mitigate the effects of a delay in processing and/or receiving a subsequent audio waveform. The audio waveform and/or data associated with the audio waveform are analyzed to identify the modification segments based on characteristics of the audio waveform and/or data associated therewith. The modification segments show where the production of the audio waveform may be modified without substantially affecting the clarity of the sound or audio. In one embodiment, the invention modifies the sound production at the identified modification segments to extend production time and thereby mitigate the effects of delay in receiving and/or processing a subsequent audio waveform for production.

Abstract: A computationally implemented system and method that is designed to, but is not limited to: electronically determining positioning status of one or more portions of one or more humans relative to one or more locations of demodulation of one or more acoustic ultrasonic signals into one or more acoustic audio signals said one or more acoustic ultrasonic signals originating from a portable electronic device; and electronically alerting said one or more humans from said portable electronic device regarding said positioning status of one or more portions of one or more humans relative to one or more locations of demodulation of one or more acoustic ultrasonic signals into one or more acoustic audio signals when said positioning status includes a first characterization. In addition to the foregoing, other method aspects are described in the claims, drawings, and text forming a part of the present disclosure.

Abstract: A sound quality correcting apparatus includes: an input module; a feature quantity calculator; a score calculator; a modulation spectrum power calculator; a score corrector; and a signal corrector. The input module receives an input audio signal. The feature quantity calculator calculates feature quantities of the input audio signal for each of a plurality of first intervals having a certain time length. The score calculator calculates a score value for each the first interval based on the feature quantities. The modulation spectrum power calculator calculates a power value, at a certain modulation frequency, of a modulation spectrum of the input audio signal. The score corrector corrects score values in the plurality of first intervals that belong to a second interval if a power value calculated in the second interval is larger than or equal to a certain value. The signal corrector corrects the audio signal based on the score values.

Abstract: The present invention relates to a medical feedback system (100) based on sound analysis in a medical environment. With a sound scene analyzer (SSA, 10) being capable of analyzing and classifying an audio signal so as to obtain a list of one or more sound sources in the medical environment surrounding the patient, and a sound-level analyzer (SLA, 20) being capable of providing an indicator for perceived levels of corresponding sound from the list of sound sources, advanced sound analysis is possible. Finally, a sound classifier (SC, 30) is arranged for classifying said list of one or more sound sources in a medical environment with respect to a degree of avoidability, and generating a corresponding feedback signal (FEED, 50) for appropriate action by medical personal, patients and/or visitors.

Abstract: Devices and methods for scoring viewer's interactions with content broadcast on a presentation device by processing at least one audio signal received by a microphone proximate the viewer and the presentation device, to generate at least one audio signature, which is compared to at least two different reference audio signatures.

Abstract: Devices and methods that match audio signatures to programming content stored in a remote database.

Abstract: According to one embodiment, a monitoring apparatus includes an acquisition unit, an analysis unit, a calculation unit, a storage, a determination unit. The acquisition unit acquires an environmental sound. The analysis unit performs frequency analysis to extract characteristic frequency components. The calculation unit calculates first values of metrics. The storage stores contour data generated by second values of the metrics. The determination unit determines whether or not there is a first measurement point in which a first value and a second value match, if there is no first measurement point or the change is not less than the threshold value, determines that the machines is abnormal.

Abstract: A system that incorporates teachings of the present disclosure may include, for example, a mobile device having a controller to detect a location of a user by detecting at least one of a network associated with the user and a location of the mobile device, wherein the location of the user includes a media processor, transmit a request for an audio command intended for the media processor to a server on a communications network after detecting the location of the user, wherein the audio command is transmitted by the server to the media processor, detect an audio signal emitted from the media processor, wherein the audio signal is caused by the audio command, and determine an identity of the media processor by associating the detected audio signal with the media processor. Other embodiments are disclosed.

Abstract: A method for monitoring sound in a process system is provided that includes sensing sound at a specified location, capturing the sensed sound as sound data, and forwarding the sound data from the specified location to a process control system. The sound data is analyzed at the process control system and a determination is made regarding whether or not to issue a warning based on the analysis of the sound data. For a particular embodiment, the analysis of the sound data includes comparing the sound data to at least one stored sound, and the determination regarding whether or not to issue a warning based on the analysis of the sound data is made based on a level of similarity between the sound data and the stored sound.

Abstract: Systems and methods for audio difference detection are described. One embodiment includes a memory component that stores logic that, when executed by a processor, causes a vehicle amplifier to receive a test input signal for a generic test sequence to determine a standard input signal for producing a desired output response signal, store the standard input signal with standard amplifier settings associated with the standard input signal, and receive an audio signal from a head unit of the vehicle, where the head unit is part of the vehicle audio system for the vehicle in which the vehicle amplifier is utilized. In some embodiments, the logic causes the vehicle amplifier to determine whether there is a difference between the audio signal and the standard input signal and in response to determining there is a difference, and adjust vehicle amplifier settings of the vehicle amplifier, such that vehicle speakers output the desired output response signal.

Abstract: A sound source detecting system that detects a predetermined sound source on the basis of sounds collected by sound collectors includes: a noise extracting unit that extracts a noise from signals of the collected sounds; and a noise suppressing unit that suppresses a signal component of the noise extracted by the noise extracting unit from the signals of the collected sounds. The sound source detecting unit detects a location of the predetermined sound source, such as an approaching vehicle, using information about the sounds having the signals of which the noise is suppressed.

Abstract: An audio device and a volume adjusting method are provided. The audio device includes a speed sensor, a first FIFO buffer unit, and a second FIFO buffer unit. The audio device generates audio signals by playing multimedia files and stores audio signals to the first FIFO buffer unit. The audio device collects ambient sound signals and stores collected sound signals to the second FIFO buffer unit. The audio device further analyzes stored audio signals and sound signals to determine a waveform of environmental noise signals, and determines a SPL of the environmental noise signals according to the determined waveform. The audio device then compares the determined SPL with a preset SPL and compares the sensed speed with a preset speed if the determined SPL is greater than the preset SPL, and adjusts the volume of audio signals according to a comparison result between the sensed speed and the preset speed.

Abstract: A display apparatus includes a display part to display an image thereon, a speaker to output sound, an audio input part to receive an audio signal, an audio signal processor to process the audio signal input from the audio input part through the speaker and to output the audio signal, a control signal output part to output a predetermined control signal, a detector to detect whether the audio signal is input through the audio input part, and a controller to control the audio signal processor to control the audio signal according to the control signal from the control signal output part if the audio signal is input by the detector, and to control display conditions of the image displayed on the display part according to the control signal from the control signal output part if the control signal is not input by the detector.

Abstract: A switch controller is provided that uses one or more capacitors to generate a slow turn on/slow turn off switch control signals to suppress audible switching noise in an audio switch. In some embodiments, an analog inverter and a capacitor are used to generate the switch control signals, while in other embodiments two capacitors are used to generate the switch control signals. To conserve power between switching states, routing logic is provided that ties the switch control signals to respective voltage rails and disables selected portions of the switch controller.

Abstract: The perception of musical sound with a music component and a speech component is intended to be improved. To this end, a method is proposed for controlling a binaural hearing system with a left hearing-device for a left ear and a right hearing-device for a right ear, which method contains the below described steps. First of all, the hearing system determines a hearing situation with the music component and the speech component. Thereupon, one of the two hearing devices is switched into a music mode and, at the same time, the other one of the two hearing devices is switched into a speech mode. The hearing-aid wearer himself/herself can then decide which component of the sound he/she would rather listen to.

Abstract: Methods and systems for detecting the presence and frequency of clipping in an audio signal are provided. A clipping detection algorithm detects the presence of hard and soft clipping using histograms with intervals of samples, rather than attempting to identify the clipping value. Therefore, it is not essential to the algorithm that there be a large number of bins. Furthermore, the bins may be non-uniformly distributed since the number of samples belonging to lower amplitudes is of little importance. The detection algorithm is also configured to determine the severity and/or perceptual effect of any clipping found to be present in the signal by calculating the ratio of clipped samples to non-clipped samples. Temporal information on the occurrence of clipping in the signal is also used to evaluate perceptual effect.

Abstract: An RMS detector uses the concept of the k-NN (classifying using nearest neighbors)-algorithm in order to obtain RMS values. A rms detector using first-order regressor with a variable smoothing factor is modified to penalize samples from center of data in order to obtain RMS values. Samples which vary greatly from the background noise levels, such as speech, scratch, wind and other noise spikes, are dampened in the RMS calculation. When background noise changes, the system will track the changes in background noise and include the changes in the calculation of the corrected RMS value. A minimum tracker runs more often (e.g. two or three times) than the rate as in prior art detectors and methods, tracks the minimum rms value, which is to compute a normalized distance value, which in turn is used to normalize the smoothing factor.

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: Methods and system for a sealing test comprising the steps of: sealing a predefined volume of gas; sampling a predefined frequency range, whereby there is no need to inject a predefined sound wave while sampling the predefined frequency range; and determining whether the predefined volume of gas is sealed or unsealed according to the sampled frequency range.

Abstract: A content reproduction device including: a microphone that collects noise in the surroundings of a casing; a feature amount extractor that extracts a plurality of feature amounts; a distance calculator that calculates an intervector distance between the extracted feature amount vector and a feature amount vector with the same dimensions which is set in advance as a feature amount of a waveform of a music signal; a determinator that determines whether or not music is included in the sounds collected by the microphone; a processor that processes the signal of the sounds collected by the microphone to change the volume or frequency characteristics of the sounds collected by the microphone; and an adder that adds and outputs the signal of the sounds collected by the microphone and the signal of sounds of reproduced content.

Abstract: In an example signal clustering apparatus, a feature of a signal is divided into segments. A first feature vector of each segment is calculated, the first feature vector having has a plurality of elements corresponding to each reference model. A value of an element attenuates when a feature of the segment shifts from a center of a distribution of the reference model corresponding to the element. A similarity between two reference models is calculated. A second feature vector of each segment is calculated, the second feature vector having a plurality of elements corresponding to each reference model. A value of an element is a weighted sum and segments of second feature vectors of which the plurality of elements are similar values are clustered to one class.

Abstract: An audio processing system that includes an audio filter having one or more elements capable of having state, such as a capacitor, an inductor or a delay. A saturation detector is configured to detect saturation of the audio filter and to generate an output when saturation of the filter is detected, such as a switch control signal. A switch is connected to the audio filter and the saturation detector, wherein the state of one or more of the elements of the audio filter is changed when the saturation detector provides the output to the switch, such as when the switch shorts the element and causes the energy stored in the element to be dissipated.

Abstract: An earpiece includes an Ambient Sound Microphone (ASM) configured to measure a background noise signal, an Ear Canal Receiver (ECR) configured to deliver audio to an ear canal, and an optional Ear Canal Microphone (ECM) configured to convert an internal sound within an ear canal of a user to an internal sound signal, and a processor operatively coupled to the ASM, the ECR and optionally the ECM. Note, internal sounds can also include residual background noise related to ambient sounds in the environment. The processor is configured to maintain a natural audio level based on a combined measurement of the background noise signal and the internal sound signal. Other embodiments are disclosed.

Abstract: A method and device for performing audio recognition, including: collecting a first audio document to be recognized; initiating calculation of first characteristic information of the first audio document, including: conducting time-frequency analysis for the first audio document to generate a first preset number of phase channels; and extracting at least one peak value characteristic point from each phase channel of the first preset number of phrase channels, where the at least one peak value characteristic point of each phase channel constitutes the peak value characteristic point sequence of said each phase channel; and obtaining a recognition result for the first audio document, wherein the recognition result is identified based on the first characteristic information, and wherein the first characteristic information is calculated based on the respective peak value characteristic point sequences of the preset number of phase channels.

Abstract: An acoustic apparatus which outputs test signals from a plurality of speakers configuring a multichannel reproduction system and adjusts acoustic characteristics by picking up response signals from the speakers using a microphone, includes a test signal storage unit storing test signals output from the speakers; a response signal storage unit storing response signals from the speakers picked up by the microphone; a parameter calculation unit calculating acoustic adjustment parameters including at least a polarity reversal parameter and a phase filter parameter based on the response signals from the speakers stored in the response signal storage unit; and an acoustic adjustment parameter storage unit storing the acoustic adjustment parameters calculated by the parameter calculation unit. The parameter calculation unit calculates the polarity reversal parameter using a low-pass component of the response signals.

Abstract: Embodiments of the present invention provide systems, methods, and computer-readable media for presenting a mute indicator to a caller when the caller is attempting to provide input to a call that has been muted. In particular, a determination is made that a mute function is engaged during a call. An area surrounding a calling system, such as a phone, is then monitored by a monitoring system to determine whether a caller is attempting to provide input to the call. A determination is then made that a caller is attempting to provide input to the call. The determination may be based on a caller providing a threshold volume of voice input within a threshold area of proximity near the calling system. A mute indicator is then presented to the caller to notify the caller that the call has a mute function engaged.

Abstract: Methods and apparatus to collect media identifying data are disclosed. An example method includes collecting audio data through a sound hole. The sound hole has a chamfered opening. The sound hole has an effective length-to-width ratio which is equal to or less than about 1.25 and analyzing the audio data to obtain media identifying data.

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 measure the output of the transducer in order to control the adaptation of the anti-noise signal and to estimate an electro-acoustical path from the noise canceling circuit through the transducer. A processing circuit that performs the adaptive noise canceling (ANC) function also either adjusts the frequency response of the anti-noise signal with respect to the reference microphone signal, and/or by adjusting the response of the adaptive filter independent of the adaptation provided by the reference microphone signal.

Abstract: A method includes determining by a sensor of a mobile device that a first ambient sound level represents a quiet context for the mobile device, providing a audio signal to a speaker of the mobile device in response to determining that the first ambient sound level represents a quiet context, determining, by the sensor, that a second ambient sound level represents a first noisy context for the mobile device, wherein the second ambient sound level is louder than the first ambient sound level, modifying the audio signal into a beam forming audio signal in response to determining that the second ambient sound level represents the first noisy context, and providing the beam forming audio signal to the speaker.

Abstract: Provided is an audio processing device comprising: a feature data generation unit which generates, for each unit section of an audio signal, section feature data expressing features of the audio signal in the unit section; a feature variation calculation unit which calculates, for each unit section, a feature variation value quantifying temporal variation of the features in the unit section, by setting the unit section as a target section and using section feature data of unit sections close to the target section; and a section judgment unit which judges, for each unit section, whether the unit section is a feature unit section including a variation point of the features, based on comparison of a threshold value and the feature variation value. Through the above, the audio processing device can detect feature unit sections from an audio signal of an AV content or the like.

Abstract: Embodiments for measuring content coherence and embodiments for measuring content similarity are described. Content coherence between a first audio section and a second audio section is measured. For each audio segment in the first audio section, a predetermined number of audio segments in the second audio section are determined. Content similarity between the audio segment in the first audio section and the determined audio segments is higher than that between the audio segment and all the other audio segments in the second audio section. An average of the content similarity between the audio segment in the first audio section and the determined audio segments is calculated. The content coherence is calculated as an average, the maximum or the minimum of the averages calculated for the audio segments in the first audio section. The content similarity may be calculated based on Dirichlet distribution.

Abstract: A device (100) for processing audio data, wherein the device (100) comprises a first audio reproduction unit (102) adapted for reproducing a first part of the audio data and adapted to be attached to a left ear (106) of a user (110), a second audio reproduction unit (104) adapted for reproducing a second part of the audio data and adapted to be attached to a right ear (108) of the user (110), a detection unit (112) adapted for detecting a left/right inversion of the first audio reproduction unit (102) and the second audio reproduction unit (104), and a control unit (114) adapted for controlling the first audio reproduction unit (102) for reproducing the second part of the audio data and for controlling the second audio reproduction unit (104) for reproducing the first part of the audio data upon detecting the left/right inversion.

Abstract: Control devices are provided. A control device includes an input, an output, and a processor. The processor is configured to receive an input signal through the input. The input signal is generated from a user interface on a media device. The processor sends a control signal through the output to a sound reproduction device. The output signal controls audio generated by the sound reproduction device. The input, output, and processor are not included in the media device and are not included in the sound reproduction device.

Abstract: Noise control for a vehicle is implemented by identifying sources of noise that are indicative of vehicle noise producing components. The sources of noise include a thermal management device of a hybrid-electric component of the vehicle and non-hybrid-electric components of the vehicle. The noise control is further implemented by determining a decibel value of the noise produced by each of the vehicle noise producing components, selecting the decibel value of one of the vehicle noise producing components having a highest decibel value as a mask limit, and determining an operational limit for the thermal management device. The operational limit includes a lowest threshold operating value which, when implemented, maintains a threshold level operating efficiency of the hybrid-electric component.

Abstract: A system and method for mapping acoustic activities is disclosed. One method comprises receiving acoustic measurements recorded for a predetermined interval at a plurality of telephonic computing devices that are located at a determined location. A map of the acoustic measurements is created based on a location of each of the plurality of telephonic computing devices at which the acoustic amplitude measurement is recorded. An amplitude and/or frequency of the acoustic measurements can be displayed on the map relative to the location of the telephonic computing device at the selected location to form an acoustic map.

Abstract: A matching system receives probe audio samples for comparison to references of a data store. Comparisons are generated to determine a sufficient match for a portion or a first amount of the probe sample. Ranking scores are assigned to the resulting match references. The match references are retained, unless meeting a score threshold. Comparisons are continually generated with second amounts of the probe sample and the retained references are updated with further matching references assigned ranking scores. The retained results are merged and determined to satisfy a score threshold for release as outputted results for matching references.

Abstract: A sound pickup device is provided that includes a first housing, a second housing, a first microphone, and a second microphone. The second housing is coupled to the first housing and is configured to change positions with respect to the first housing. The first microphone is mounted on the first housing and is configured to output a first audio signal based on sound picked up by the first microphone. The second microphone is mounted on the second housing and is configured to output a second audio signal based on sound picked up by the second microphone.

Abstract: According to one embodiment, an acoustic characteristic correction coefficient calculation apparatus includes a frequency converter, a smoother, a frequency inverter, a cutter, and a calculator. The frequency converter is configured to convert a first impulse response corresponding to an input acoustic signal to a frequency domain. The smoother is configured to smooth amplitude and phase corresponding to the frequency domain converted by the frequency converter. The frequency inverter is configured to convert a frequency characteristic smoothed by the smoother to a time domain. The cutter is configured to cut out a second impulse response configured by the time domain obtained by converting the frequency characteristic by use of the frequency inverter by a preset tap number. The calculator is configured to calculate a correction coefficient used to correct an acoustic characteristic based on a result cut out by the cutter.

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: Methods and systems for beam forming an audio signal based on a location of an object relative to the listening device, the location being determined from positional data deduced from an optical image including the object. In an embodiment, an object's position is tracked based on video images of the object and the audio signal received from a microphone array located at a fixed position is filtered based on the tracked object position. Beam forming techniques may be applied to emphasize portions of an audio signal associated with sources near the object.

Abstract: A method for determining an inverse filter for altering the frequency response of a loudspeaker so that with the inverse filter applied in the loudspeaker's signal path the inverse-filtered loudspeaker output has a target frequency response, and optionally also applying the inverse filter in the signal path, and a system configured (e.g., a general or special purpose processor programmed and configured) to determine an inverse filter. In some embodiments, the inverse filter corrects the magnitude of the loudspeaker's output. In other embodiments, the inverse filter corrects both the magnitude and phase of the loudspeaker's output. In some embodiments, the inverse filter is determined in the frequency domain by applying eigenfilter theory or minimizing a mean square error expression by solving a linear equation system.

Abstract: The present invention discloses methods, apparatus and systems for individualizing music, audio and speech adaptively, intelligently and interactively according to a listener's personal hearing ability, unique hearing preference, characteristic feedback, and real-time surrounding environment.

Abstract: A system of remotely monitoring the sound level at the location of a monitored person. The system includes a first device positioned with a monitored person, a monitoring server that is a hub for communication within the system, and a second device for a monitoring person that is remote from the first device and the monitored person. The first device is configured to detect the occurrence of sound exceeding a sound threshold. The first device then places a call to at least one of the monitoring server and the second device indicating that the sound threshold has been exceeded in the area of the monitored person.

Abstract: Systems, methods, and computer-readable storage media for facilitating recognition of real-time content are provided. In embodiments, a new audio fingerprint associated with live audio being presented is received. In accordance with the received audio fingerprint, at least one previously received fingerprint associated with the live audio from a real-time index is removed. Thereafter, the real-time index is updated to include the new audio fingerprint associated with the live audio being presented. Such a real-time index having the new audio fingerprint can be used to recognize the live audio being presented and, thereafter, an indication of the recognized live audio can be provided to the user device.

Abstract: An audio apparatus is provided. The audio apparatus includes at most one electroacoustic transducer; and an audio controller, coupled to the electroacoustic transducer, for actively controlling the electroacoustic transducer to function as a loudspeaker or a microphone, wherein the loudspeaker converts output electrical signals to output sounds, and the microphone converts input sounds to input electrical signals.