Abstract: Embodiments described herein provide methods and apparatus for predicting the excursion of a transducer. The method comprises determining whether an input signal to the transducer is a periodic signal; and responsive to determining that the input signal is a periodic signal, calculating the predicted excursion based on a direct current (“DC”) offset associated with the transducer.

Abstract: A cable apparatus, and systems and methods, are provided for mixing ambient sound signals generated by the cable apparatus with a monitor mix signal received by the cable apparatus for transmission to first and second in-ear monitoring devices. The cable apparatus may be connected between an audio signal transceiver and the first and second in-ear monitoring devices of an audio monitoring system. The cable apparatus comprises an electrical communications channel that includes first and second connectors for coupling with the first and second in-ear monitoring devices and a third connector for coupling with the audio signal transceiver. The cable apparatus is configured to generate first and second ambient sound signals associated with first and second microphones coupled along the electrical communications channel of the cable apparatus proximate to the first and second connectors. The cable apparatus actively mixes the first and second ambient sound signals with the monitor mix signal.

Abstract: A lighting controller includes: an information acquiring unit configured to acquire music piece information including at least information on a beat position in music piece data; a lighting controller configured to control a lighting fixture with reference to a lighting-effect change point, a minimum unit of the lighting-effect change point being defined by the beat position; and an operation interlock control unit configured to apply, in response to a predetermined operation applied on a music piece reproduction apparatus configured to reproduce the music piece data as a music piece or a music piece reproduction controller configured to control the music piece reproduction apparatus, a predetermined lighting effect corresponding to the operation or a process executed by the operation to the control of the lighting fixture.

Abstract: A method or apparatus for delivering audio programming such as music to listeners may include identifying, capturing and applying a listener's audiometric profile to transform audio content so that the listener hears the content similarly to how the content was originally heard by a creative producer of the content. An audio testing tool may be implemented as software application to identify and capture the listener's audiometric profile. A signal processor may operate an algorithm used for processing source audio content, obtaining an identity and an audiometric reference profile of the creative producer from metadata associated with the content. The signal processor may then provide audio output based on a difference between the listener's and creative producer's audiometric profiles.

Abstract: A device for acoustic echo cancellation includes a modulator, a speaker, a microphone, a demodulator, and an adaptive filter. The modulator duplicates a far-end signal to a frequency range that is higher than the far-end signal to be a first frequency-shifted signal and generates a modulated signal according to the far-end signal and the first frequency-shifted signal. The speaker generates a sound signal according to the modulated signal. The microphone generates a microphone signal according to a near-end signal and an echo signal. The echo signal is a convolution of the sound signal with a room impulse response. The demodulator extracts a demodulated signal and an echo-reference signal from the microphone signal. The adaptive filter generates a recovered signal to recover the near-end signal according to the demodulated signal and the echo-reference signal.

Abstract: A method and an apparatus control sound signal output. The method of controlling a sound signal output in a terminal includes: storing receiver identifications and corresponding sound signal control parameters associated with each other by the terminal; providing a receiver selection interface capable of selecting one from a list of the receiver identifications and the receiver identifications included in the list by the terminal; receiving one receiver identification through the receiver selection interface; converting a digital sound signal into an analog sound signal by applying a sound signal control parameter corresponding to the received receiver identification; and outputting the analog sound signal.

Abstract: Provided is an electronic device that includes a first processor for receiving an audio signal, performing first voice recognition on the audio signal, and transferring a driving signal to a second processor based on a result of the first voice recognition. The second processor performs second voice recognition based on a voice signal by the first voice recognition or the audio signal, in response to the driving signal.

Abstract: A monitoring system can include an earpiece, a database with stored earpiece characteristics data, and a microphone to receive a plurality of signals where each signal of the plurality of signals can represents a respective sound pressure level of sound pressure values over a time duration. The system can also include a processor and a memory coupled processor, the memory having computer instructions which when executed by the processor causes the processor to perform the operations. The operations can include determining exposure time duration when a signal of the plurality of signals exceeds a sound pressure level threshold value, retrieving a subset of data of the stored earpiece characteristics data, and modifying an acoustic output of the earpiece in accordance with the subset of data of the stored earpiece characteristics data.

Abstract: A failure detection apparatus includes: an output portion that outputs from an audio output device a confirmation signal having a frequency in a band other than the audible band; an input portion to which an input signal is inputted from an audio input device; and a failure detection portion that detects whether the audio output device and the audio input device are operating normally on the basis of the confirmation signal and the input signal.

Abstract: The application discloses an eyeball movement capturing method, which includes: acquiring a real-time image shot by a photographic device and extracting a real-time facial image; inputting the real-time facial image into a pretrained eyeball mean shape and recognizing n1 orbit feature points and n2 eyeball feature points; and calculating a movement direction and movement distance of an eyeball in the real-time facial image according to x and y coordinates of the (n1+n2) eyeball feature points in the real-time facial image. The application also discloses an electronic device and a computer-readable storage medium. According to the application, movement information of the eyeball in the real-time facial image is calculated according to the coordinates of the eye feature points to implement real-time capturing of an eyeball movement.

Abstract: Methods, systems, and media for providing sound level information for a particular location are provided. In some embodiments, the method comprises: receiving, from a user device, a request for sound level indications associated with a geographic region; in response to receiving the request, identifying a plurality of locations associated with the geographic region; identifying, for each location in the plurality of locations, at least one sound level reading corresponding to the location; generating, for each location in the plurality of locations, sound level indications based on the at least one sound level reading corresponding to the location; and transmitting at least one of the sound level indications and an identifier of at least one location in the plurality of locations corresponding to the at least one of the sound level indications to the user device.

Abstract: Various embodiments provide systems and methods which disclose a device which may be used to determine an audio event based on receiving an audio signal. Based on the determined audio event, audio activity may be tracked. The tracked audio activity may be summarized based on a summary mode. Feedback of the summarized tracked audio activity may be based on a feedback mode.

Abstract: An apparatus including a recording system including at least one microphone; a controller including a processor and a non-transitory memory, where the controller is configured to automatically adjust at least one recording parameter for a sound signal from the at least one microphone based, at least partially, upon a distance between the apparatus and a sound source; and a distance determinator configured to determine the distance between the apparatus and the sound source and provide the determined distance to the controller so as to adjust the at least one recording parameter, where the distance determinator includes at least one sensor and does not include the at least one microphone, where the apparatus is a portable electronic device.

Abstract: A device can be worn by a user and can include a microphone to analyze music and other sound in the surrounding environment. In one embodiment, the device can translate audio into a haptic and/or light of the sound's or music's bassline, in real-time. In one embodiment, no music is recorded by the device. The haptic or vibrational representation of the music can be generated by a motor. In some examples, the light representation of the music can be generated via a red/green/blue light emitting diode.

Abstract: Methods and systems for passwordless authenticating of a user are provided. A first request to access a first content is received from a first user of a first device. A first challenge is sent to the first device in response to the first request. The first challenge is resolved using a first private key associated with the first user for the first content. A first response, including a resolved challenge, is received from the first device. It is determined whether the first response is an acceptable response to the first challenge. Access to the first content is granted to the first user in response determining that the first response is an acceptable response.

Abstract: Exemplary aspects of the present disclosure assist in phase alignment for systems having multiple audio sources. For example, in a system having plural microphones, phase alignment may also be assisted by sampling the microphones at the appropriate time relative to when the samples are placed on the audio bus. Further, phase shifts between audio samples are reduced or eliminated by keeping a sample delay constant for samples from the same microphone. Such manipulation of the audio samples reduces phase shifts which reduces the likelihood of an audio artifact capable of being detected by the human ear and thus improves consumer experience.

Abstract: Disclosed is an apparatus and method for extracting a sound source from a multi-channel audio signal. A sound source extracting method includes transforming a multi-channel audio signal into two-dimensional (2D) data, extracting a plurality of feature maps by inputting the 2D data into a convolutional neural network (CNN) including at least one layer, and extracting a sound source from the multi-channel audio signal using the feature maps.

Abstract: A method of measuring state of vigilance of a pilot by a measuring device to measure time between which a sound wave is emitted by a loudspeaker in the piloting station of an aircraft and is received by the microphone of a headset of a pilot and to calculate coordinates of a sphere centered on the loudspeaker and on the perimeter of which the microphone is situated. Calculation of points of intersection of spheres centered on different loudspeakers provides possible microphone positions in the space of the piloting station. These positions are those of the head of the pilot when the headset is used. Via the knowledge of at least one possible position of the head of the pilot in the piloting station, the measuring device can measure the state of vigilance of the pilot and to alert in case of a confirmed lack of vigilance of the pilot.

Abstract: A method for determining the presence of unwanted signal components in an acoustic signal, comprises: receiving a first microphone signal derived from an acoustic signal, applying the first microphone signal to a first signal processing path having a first transfer function to provide a first output, receiving a second microphone signal derived from the acoustic signal, applying the second microphone signal to a second signal processing path having a second transfer function to provide a second output, wherein the second transfer function has a different degree of linearity from the first transfer function, and determining the presence of unwanted signal components in the acoustic signal based on a comparison of the first output and the second output.

Abstract: In some embodiments, a method for processing an audio signal in an audio processing apparatus is disclosed. The method includes receiving an audio signal and a parameter, the parameter indicating a location of an auditory event boundary. An audio portion between consecutive auditory event boundaries constitutes an auditory event. The method further includes applying a modification to the audio signal based in part on an occurrence of the auditory event. The parameter may be generated by monitoring a characteristic of the audio signal and identifying a change in the characteristic.

Abstract: An acoustic spatial adjustment system may be provided that includes: a wireless communication module which receives a voice signal from a mobile terminal connected thereto by wireless communication; an adjustment controller which filters the voice signal in order to generate a sound source signal for focusing an output of the voice signal transmitted from the wireless communication module on a bright zone (BZ); an amplifier which divides the sound source signal transmitted from the adjustment controller into multi-channels and amplifies; and a plurality of speakers which output the sound source signal transmitted from the amplifier. As a result, the sound can be focused on a specific zone in the vehicle by using existing mounted speakers in the vehicle or by installing only additional speakers for performance improvement. Also, the individually distinguished sound (audio signal of a stereo system, CD, DVD, etc., a telephone voice signal connected by Bluetooth, etc.

Abstract: A method includes wirelessly receiving audio content via a first transceiver included in a housing of a first device and producing a first audible signal from the audio content via a speaker included in the housing of the first device. The audio content is wirelessly transmitted via the first transceiver to a second transceiver integrated within a first remote speaker. A second audible signal is produced from the audio content via the first remote speaker. The method also includes wirelessly relaying the audio content via the second transceiver to a third transceiver integrated within a second remote speaker and producing a third audible signal from the audio content via the second remote speaker. The second remote speaker is located outside of the transmission range of the first transceiver.

Abstract: When transferring waveform data from a waveform region in a high-capacity flash memory to a waveform buffer in a RAM, first, whether a non-ring buffer region that can receive the waveform data can be allocated in the RAM is determined, and if so, the waveform data is transferred to the allocated non-ring buffer region. If such a non-ring buffer region cannot be allocated, then whether a ring buffer region that can receive the waveform data can be allocated in the RAM is determined, and if so, the waveform data is transferred to the ring-buffer region in a ring buffer operational manner.

Abstract: Systems, methods, and devices for monitoring industrial equipment using audio are described herein. One system includes two computing devices. The first computing device can receive, from an audio sensor, audio sensed during operation of industrial equipment, extract a plurality of features from the audio, determine whether any portion of the audio is anomalous, and send, upon determining a portion of the audio is anomalous, the anomalous portion of the audio to the second, remotely located, computing device. The second computing device can provide the anomalous portion of the audio to a user to determine whether the anomalous portion of the audio corresponds to a fault occurring in the equipment, and receive, from the user upon determining the anomalous portion of the audio corresponds to a fault occurring in the equipment, input indicating the anomalous portion of the audio corresponds to the fault to learn fault patterns in the equipment.

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: A device can be worn by a user and can include a microphone to analyze music and other sound in the surrounding environment. In one embodiment, the device can translate audio into a haptic and/or light of the sound's or music's bassline, in real-time. In one embodiment, no music is recorded by the device. The haptic or vibrational representation of the music can be generated by a motor. In some examples, the light representation of the music can be generated via a red/green/blue light emitting diode.

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

Abstract: Provided is a user authentication method including reproducing sound data of which a sound source in a first position of a space around a user is virtually localized using a Head-Related Transfer Function (HRTF) of the user toward the user, acquiring a second position of the space around the user, the second position being estimated by the user who has listened to the reproduced sound data as a position of the sound source; and authenticating the user according to a coincidence between the first position and the second position.

Abstract: A method for compensating for acoustic influence of a listening room on an acoustic output from an audio system including at least a left and a right loudspeaker, the method comprising determining a left frequency response and a right frequency response, designing a left compensation filter FL, and a right compensation filter FR, and during playback applying the left and right filters to left and right channel inputs. According to the invention, a target response in the listening position is simulated, and the left and right compensation filters are designed to filter transfer functions based on the simulated target function multiplied by an inverse of the left/right frequency responses. By relying on a simulated target instead of relying on an empirical approach, the general impact of a room can be more accurately captured by the target functions.

Abstract: A method for improved virtual localization of sound comprises making a sound at an origin point, recording the sound with two or more recording devices at, two or more different distances from the origin point, generate a head-related transfer function (HRTF) for each of signals received from the two or more recording devices at the two or more different distances from the origin point, convolving a waveform with a localized HRTF generated using at least one of the HRTFs, and drive a speaker with the convolved waveform.

Abstract: Described herein are methods, systems, and devices which in some embodiments may be used to detect noise produced by a vehicle and identify the vehicle that is the source of the noise.

Abstract: A hearing protection device configured to measure the noise exposure of a user and to provide a time remaining estimate until a maximum noise exposure level is achieved. Generally, the hearing protection device may comprise a sound sealing section, a microphone configured to generate a detected sound signal indicative of the measured sound level at the user's ear canal, a memory/storage configured to record the detected sound signal as sound exposure history, a timer configured to measure the amount of time the hearing protection device is exposed to sound, and a processor configured to receive a timer signal from the timer and the detected sound signal to calculate a time remaining estimate until reaching the maximum noise exposure level. Additionally, the maximum noise exposure level may be based on industry regulations or personalized hearing thresholds.

Abstract: The technology relates to processing one or more audio streams. The processing may include segmenting the one or more audio streams into structural components comprising a tonal stream and a transient stream. The tonal stream and the transient stream may be processed and combined to provide an output sound signal. The one or more audio streams may include a first audio stream from a first sound source and a second audio stream from a second sound source. The tonal stream may comprise sound from the first audio stream and the transient stream may comprise sound from the second audio stream. One or more sound effects may be added to the tonal stream or transient stream prior to combining.

Abstract: The disclosed technologies calibrate audio output based on a computer-based hearing evaluation. In an embodiment, the disclosed technologies perform operations that include playing an electronic file that contains segments of digital music that have different playback attributes, wherein playback attribute include at least one of volume data, frequency data, left/right data; during the playing of the electronic file, recording timestamp data and user account data that are associated with a user input; mapping the timestamp data to a set of playback attributes of a segment of the digital music; determining calibration data based on the timestamp data and the set of playback attributes; storing the calibration data in a user profile that is associated with the user account data; when different digital music is played, automatically calibrating the playing of the different digital music based on the user profile that is associated with the user account data.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding of a vector of parameters in an audio coding system. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system. According to the disclosure, a modulo differential approach for coding and encoding a vector of a non-periodic quantity may improve the coding efficiency and provide encoders and decoders with less memory requirements. Moreover, an efficient method for encoding and decoding a sparse matrix is provided.

Abstract: The present invention relates in one aspect to a microphone array signal processing system comprising a digital buffer coupled to a signal input and configured to store first and second digital audio signals. A beamformer analyzer is configured to, in response to a first voice trigger, determine noise statistics based on the first signal segment of the first digital audio signal and a first signal segment of the second digital audio signal. A coefficients calculator is configured to calculate a first set of fixed beamformer coefficients of a beamforming algorithm. The beamforming algorithm is configured for applying the first set of fixed beamformer coefficients to the first signal segments of the first and second digital audio signals retrieved from the digital buffer to produce a noise reduced digital audio signal.

Abstract: To resolve a problem that a listener feels that sense of localization and sense of connecting with the other channels are lost in case that the listener listens to an audio that is output from a ceiling reflection type speaker in an audio processing device that outputs an analog audio signal to speakers including the ceiling reflection type speaker that makes the audio reflect at a ceiling. A DSP 5 performs low-pass filter processing that extracts low frequency component from a digital audio signal, high-pass filter processing that extracts high frequency component from the digital audio signal, delay processing that delays the low frequency component of the digital audio signal that is extracted by the low-pass filter processing, and composition processing that composes the low frequency component of the digital audio signal that is delayed by the delay processing and the high frequency component of the digital audio signal that is extracted by the high-pass filter processing as audio signal processing.

Abstract: A method and a system for determining an interference value. The method receives a sound signal and an interferer signal and forms a pair of a portion of the sound signal and the interferer signal. The portions have a predetermined time duration but the method is capable of determining the interference value swifter, so that the interference value may be output in real time.

Abstract: Systems and methods for monitoring and controlling sound levels across multiple regions are provided. Some systems and methods may include a sound level sensor in a first region determining a sound level within the first region for sound originating from a source in a second region and a sound level display device in the second region displaying a visual indicator of the sound level in the first region. Additionally or alternatively, some systems and methods may include controlling a volume level of the source.

Abstract: A diagnosis method includes giving an instruction to present at least two sound collection positions of an abnormal sound of an electric apparatus to an outside; comparing a current position of the diagnosis device and one sound collection position of the at least two sound collection positions; obtaining sound data indicative of an abnormal sound of the electric apparatus collected by a microphone, based on a comparison result and storing the sound data in a sound data storage portion so as be correlated with log data and the one sound collection position; and diagnosing an abnormal sound of the electric apparatus by using at least two items of sound data and at least two sound collection positions, based on a determination result whether sound collection is completed and a determination result whether diagnosis of an abnormal sound is possible.

Abstract: Techniques and mechanisms for exchanging information between a mobile device and another device which is to serve as an accessory to the mobile device. In an embodiment, the accessory device and the mobile device are coupled to one another via an audio connector of the mobile device. The mobile device receives via the audio connector encoded signals which represent, in a first frequency range, information generated by a sensor of the accessory device. The first frequency range is within an audible frequency range and outside of a telephony frequency range of the mobile device. In another embodiment, the encoded signals are provided to telephony logic of the host device during a telecommunication exchange between the host device and a remote device. Due to its representation in the first frequency range, some sensor information is prevented from being represented as an audio signal in the telecommunication exchange.

Abstract: Systems and methods are presented for efficient cross-fading (or other multiple clip processing) of compressed domain information streams on a user or client device, such as a telephone, tablet, computer or MP3 player, or any consumer device with audio playback. Exemplary implementation systems may provide cross-fade between AAC/Enhanced AAC Plus (EAACPlus) information streams or between MP3 information streams or even between information streams of unmatched formats (e.g. AAC to MP3 or MP3 to AAC). Furthermore, these systems are distinguished by the fact that cross-fade is directly applied to the compressed bitstreams so that a single decode operation may be performed on the resulting bitstream. Moreover, using the described methods, similar cross fade in the compressed domain between information streams utilizing other formats of compression, such as, for example, MP2, AC-3, PAC, etc. can also be advantageously implemented.

Abstract: A hearing device includes: at least one microphone for converting sound received by the at least one microphone into an audio signal; a sound impulse detector configured for detecting a presence of an impulse in the audio signal; and a signal processor configured for processing the audio signal into a processed audio signal in response to the presence of the impulse in the audio signal as detected by the sound impulse detector; and a receiver coupled to the signal processor for converting the processed audio signal into an output sound signal for emission towards an eardrum of a user; wherein the sound impulse detector is configured for operation in a frequency domain for detecting presence of the impulse in the audio signal.

Abstract: A fuzzy entropy based noisy signal processing method and an iterative singular spectrum analysis (SSA) soft threshold de-noising method are disclosed. The method employs FuzzyEn, which is used to quantify the system complexity in chaos theory, to characterize a noise floor, which provides a more effective path for processing of noisy signal; its fuzzy entropy spectrum based iterative singular spectrum analysis soft threshold (SSA-IST) de-noising method outperforms the conventional truncated singular spectrum, wavelet transform and empirical mode decomposition de-noising approaches in de-noising performance.

Abstract: Methods and systems for a transportation vehicle are provided. For example, one method includes storing a key on a device of a transportation vehicle and a headset; generating a challenge by the device using the key when the headset is connected to the device; presenting the challenge to the headset by the device; creating a response to the challenge by the headset; presenting the response to the device; validating the response by the device; and enabling a data line for the headset by the device when the response is valid.

Abstract: A diagnostic device includes an acquisition unit, first and second measurement units, a generation unit, and a guidance unit. The acquisition unit accepts a produced sound as input and acquires sound information. The first measurement unit measures a distance from a standard position. The second measurement unit measures a change in orientation from a standard state. The generation unit generates appropriate position information, the appropriate position information being the distance from the standard position to the diagnostic device measured by the first measurement unit, and the change in orientation from the standard state measured by the second measurement unit. The guidance unit uses the appropriate position information to guide the diagnostic device to an appropriate position when acquiring the sound information.

Abstract: A method or apparatus for delivering audio programming such as music to listeners may include identifying, capturing and applying a listener's audiometric profile to transform audio content so that the listener hears the content similarly to how the content was originally heard by a creative producer of the content. An audio testing tool may be implemented as software application to identify and capture the listener's audiometric profile. A signal processor may operate an algorithm used for processing source audio content, obtaining an identity and an audiometric reference profile of the creative producer from metadata associated with the content. The signal processor may then provide audio output based on a difference between the listener's and creative producer's audiometric profiles.

Abstract: The invention relates to a background noise estimator and a method therein, for supporting sound activity detection in an audio signal segment. The method comprises reducing a current background noise estimate when the audio signal segment is determined to comprise music and the current background noise estimate exceeds a minimum value. This is to be performed when an energy level of an audio signal segment is more than a threshold higher than a long term minimum energy level, lt_min, which is determined over a plurality of preceding audio signal segments, or, when the energy level of the audio signal segment is less than a threshold higher than lt_min, but no pause is detected in the audio signal segment.

Abstract: Pressure sensing earbuds and systems are disclosed. The earbuds can include one or more pressure sensors to determine the size and shape of a user's ear. The pressure signals can be relayed back to a processor, which may use them to dynamically optimize the volume levels delivered for frequencies over the audible range for a particular user.

Abstract: A system and method of detecting and analyzing a threat in a confined environment is disclosed. An audio board detects and analyzes audio signals which are then transmitted and analyzed to determine the location of a gunshot in a confined location and the type of firearm being shot.