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 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: 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 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: 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 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.

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: The present disclosure relates to a method, apparatus, and system for speaker verification. The method includes: acquiring an audio recording; extracting speech signals from the audio recording; extracting features of the extracted speech signals; and determining whether the extracted speech signals represent speech by a predetermined speaker based on the extracted features and a speaker model trained with reference voice data of the predetermined speaker.

Abstract: Systems, apparatus, and methods for collecting, interpreting, and utilizing noise exposure data may include sensors to obtain an analog signal representative of impulse noise sound pressure and an analog signal representative of continuous noise sound pressure. At least one ADC may generate digital signals by sampling the analog signals at rates equal to or greater than twice the reciprocal of a minimum impulse noise rise time. Accelerometers may obtain data in close proximity to and remote from the sensors. At least one processor may include a first combining node to combine the digital signals to represent both the continuous noise and the impulse noise, a shock-artifact detection filter to identify a time frame including a shock artifact based on the accelerometry data, a frequency filter to generate a background-removed audio signal, an adaptive filter to estimate the shock artifact, and a second combining node to produce a shock-artifact-free audio signal.

Abstract: Systems, methods, and devices for determining a frequency of a received audio signal. A device may include a microphone, a speaker, and a processor configured to perform operations including receiving, using the microphone, a first audio signal having a predetermined frequency. The operations may also include, for each test frequency of a plurality of test frequencies and for each test phase of a plurality of test phases, generating a second audio signal having the test frequency and the test phase, outputting, using the speaker, the second audio signal, receiving, using the microphone, a combined audio signal being a combination of the first audio signal and the second audio signal, determining an amplitude of the combined audio signal, and determining that the predetermined frequency is within a threshold range of the test frequency when the amplitude of the combined audio signal is below a threshold.

Abstract: The technology described in this document can be embodied in a computer-implemented method that includes receiving identification information associated with a geographic location. The identification information includes one or more features that affect an acoustic environment of the geographic location at a particular time. The method also includes determining one or more parameters representing at least a subset of the one or more features, and estimating at least one acoustic parameter that represents the acoustic environment of the geographic location at the particular time. The at least one parameter can be estimated using a mapping function that generates the estimate of the at least one acoustic parameter as a weighted combination of the one or more parameters. The method further includes presenting, using a user-interface displayed on a computing device, information representing the at least one acoustic parameter estimated for the geographic location for the particular time.

Abstract: In some embodiments, the disclosed subject matter involves a system and method for dynamic object identification and environmental changes for use with autonomous vehicles. For efficient detection of changes for autonomous, or partially autonomous vehicles, embodiments may use a technique based on background removal and image subtraction which use motion detection rather than full object identification for all objects in an image. Road side units proximate to a road segment or virtual road side units in the cloud, other vehicles or mobile device (e.g., drones) are used to retrieve and store background images for a road segment, to be used by the autonomous vehicle. Other embodiments are described and claimed.

Abstract: Real-time pitch detection of voiced musical notes involves converting sound waves, produced by a voiced rendition of one or more musical notes, to a time domain electronic audio signal. The electronic audio signal is processed to determine a true pitch of the time domain electronic audio signal. True pitch information is displayed in real-time, concurrent with the voiced rendition of each musical note. A pitch indicator conveys to a user information concerning the true pitch which has been determined. The true pitch is determined by segmenting the electronic audio signal into a plurality of audio signal samples and applying a constant-Q transform. Additional processing steps are applied to reduce pitch detection errors.

Abstract: A system and method for performing microphone gating operations, is disclosed. The system and method include a transmitter configured to emit a transmit signal towards an object and a receiver configured to receive a reflected signal from the object, the reflected signal corresponding to the transmit signal. The system and method also include a controller configured to instruct the transmitter to emit the transmit signal and receive the reflected signal from the receiver. The controller is further configured to detect motion of the object based upon the reflected signal and turn a microphone on or off based upon the motion of the object.

Abstract: A protected extended playback mode protects the integrity of audio and side information of a spatial audio signal and sound object and position information of audio objects in an immersive audio capture and rendering environment. Integrity verification data for audio-related data determined. An integrity verification value is computable dependent on the transmitted audio-related data. The integrity verification value can be compared with the integrity verification data for verifying the audio-related data transmitted in the audio stream for generating a playback signal having a mode dependent on the verification of the audio-related data A transmitting device transmits that integrity verification data and the audio-related data in an audio stream for reception by a receiving device. The audio stream, including the audio-related data and integrity verification data are received by the receiving device.

Abstract: An active awareness control system and method for a helmet with a rigid shell that spatially divides a shell interior from a shell ambiance includes receiving at least one playback-sound signal, generating in the shell interior from the at least one playback-sound signal playback sound, and receiving and processing at least one ambient-sound signal representative of sound occurring in the shell ambience to detect the at least one desired-sound signal. The generation of the at least one playback sound is put on hold when the at least one desired-sound pattern is detected.

Abstract: A signal processor for determining a plurality of drive signals for driving a plurality of loudspeakers to cancel a reverberation effect in a listening area, wherein the signal processor is configured to determine from one or more measured audio signals a plurality of measured physical coefficients in a basis of physical sound functions, such that a sum of the physical sound functions, weighted with the plurality of measured physical coefficients approximates the one or more measured audio signals, wherein at least half of the plurality of measured physical coefficients are zero, determine a residual error between the plurality of measured physical coefficients and a plurality of desired physical coefficients, estimate a transfer function describing a transformation from the plurality of desired physical coefficients to the plurality of measured physical coefficients, based on the determined residual error, and update the plurality of drive signals based on the estimated transfer function.

Abstract: A method for separating an audio signal into a harmonic signal component and a transient signal component is disclosed. The method includes the steps of: transferring the audio signal into a frequency space in order to obtain a transferred audio signal in dependence on frequency and time and applying a non-linear smoothing filter to the transferred audio signal over frequency to obtain a filtered transient signal in which the harmonic signal component is suppressed relative to the transient signal component. The method further includes applying the non-linear smoothing filter to the transferred audio signal over time to obtain a filtered harmonic signal in which the transient signal component is suppressed relative to the harmonic signal component and determining the harmonic signal component and the transient signal component based on the filtered harmonic signal and the filtered transient signal.

Abstract: A system and method for quieting unwanted sound. As a non-limiting example, various aspects of this disclosure provide a system and method, for example implemented in a premises-based or home audio system, for quieting unwanted sound at a particular location.

Abstract: A sound field coding system and method that provides flexible capture, distribution, and reproduction of immersive audio recordings encoded in a generic digital audio format compatible with standard two-channel or multi-channel reproduction systems. This end-to-end system and method mitigates any impractical need for standard multi-channel microphone array configurations in consumer mobile devices such as smart phones or cameras. The system and method capture and spatially encode two-channel or multi-channel immersive audio signals that are compatible with legacy playback systems from flexible multi-channel microphone array configurations.

Abstract: Techniques are described for performing adaptive noise suppression to improve handling of both speech signals and music signals at least up to super wideband (SWB) bandwidths. The techniques include identifying a context or environment in which audio data is captured, and adaptively changing a level of noise suppression applied to the audio data prior to bandwidth compressing (e.g., encoding) based on the context. For a valid speech context, an audio pre-processor may set a first level of noise suppression that is relatively aggressive in order to suppress noise (including music) in the speech signals. For a valid music context, the audio pre-processor may set a second level of noise suppression that is less aggressive in order to leave the music signals undistorted. In this way, a vocoder at a transmitter side wireless communication device may properly encode both speech and music signals with minimal distortions.

Abstract: Methods and apparatus relating to a location aware power management scheme for an always-on-always-listen voice recognition system are described. In an embodiment, logic performs ambient noise trigger level analysis for a location and causes storage of an ambient noise trigger level threshold value for the location based on the ambient noise trigger level analysis. Furthermore, logic determines whether to cause modification to a state of an audio processor in response to detection of the audio processor at the location and comparison of a detected sound level at the location and the stored ambient noise trigger level threshold value. Other embodiments are also disclosed and claimed.

Abstract: A binaural hearing device adapted to assist a recipient to segregate sounds from local and remote sources. Segregation can be achieved with two environmental microphones that are configured to mix right/left ambient sounds and divert them to the ear on one side of the recipient.

Abstract: An example method, and example devices and computer-readable media related to the example method, are disclosed herein. The method includes, as a computing device is moving within an environment of a playback device, capturing, via a microphone of the computing device, one or more calibration sounds played by the playback device. The method further includes identifying one or more sections of data such that each of the one or more sections of the data correspond to a respective calibration sound of the one or more calibration sounds and using the one or more sections of the data to determine a frequency response of the playback device. The method further includes determining one or more parameters of an audio processing algorithm based on the frequency response of the playback device and a target frequency response, and sending, to the playback device, the one or more parameters.

Abstract: The signal processing functions of a hearing aid as described above necessarily cause some delay between the time the audio signal is received by the microphone or wireless receiver and the time that the audio is actually produced by the output transducer. In some situations, signal processing incorporating longer delays may be better able to improve signal-to-noise ratio (SNR) or other functional parameters for a hearing aid wearer, but a balance should be struck between these positive effects of delay and other negative effects. The techniques described herein address the problem of balancing the positive and negative effects of delay.

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: Provided is a measurement device that evaluates an acoustic device 100 including a vibrating element and configured to allow sound to be heard by vibration transmission. The measurement device includes: an ear model unit 50 including an ear model 51 that is molded after a human ear and an artificial cartilage unit 54 that is joined to the ear model 51; and a vibration detector 56 disposed in the ear model unit 50.

Abstract: An apparatus, method, system, and program product are disclosed for data replication. One method includes determining a predicted match ratio. The predicted match ratio may predict a percentage of a current version of a file that matches a prior version of the file. The method may include performing a delta transfer if the predicted match ratio is greater than a predetermined threshold. The delta transfer may transfer a portion of the current version of the file that has changed relative to the prior version of the file. The method may include performing a transfer of the current version of the file if the predicted match ratio is less than the predetermined threshold.

Abstract: The described examples provide for a sound recognition and notification process and apparatus that provides a notification to a hearing impaired person via a portable device in response to the recognition of a selected sound. A processor of the sound notification apparatus is configured to compare the detected sound data to a model for recognition. If the sound data matches the model, features are extracted from the detected sound data and used by the processor to generate reference data, e.g. an updated sound model. The reference is retained in a memory of the apparatus to facilitate subsequent sound recognition without having to rely upon an external device for analysis. Recognition of a sound, based on a substantial match to stored reference data, may trigger transmission of a notification message indicating an occurrence of the corresponding sound.

Abstract: Method and apparatus are disclosed for accelerometer-based external sound monitoring. A vehicle includes accelerometers, speakers, and an infotainment head unit. The accelerometers are affixed to windows of the vehicle. The speakers are located inside the vehicle in the vicinity of the accelerometers. Each of the speakers is uniquely associated with one of the accelerometers. The infotainment head unit, when the vehicle is not in motion and when a signal is received from one of the accelerometers, plays the signal only on the associated one of the speakers.

Abstract: The present invention discloses a method of adjusting ambient sounds for an earphone. The earphone collects ambient sounds using at least one microphone. The method includes: determining an adjustment type and an adjustment amount for the collected ambient sounds; and adjusting, according to the determined adjustment type, the collected ambient sounds by the determined adjustment amount. The present invention also provides an earphone.

Abstract: A sound pressure level (SPL) monitoring information system includes a database configured to store data including at least one of a list of earpiece devices and associated instrument response functions, an audiogram compensation information, or an earpiece frequency response function. The system further includes an SPL monitoring system comprising a microphone. The SPL monitoring system can be configured to receive a plurality of signals, each signal representing a respective SPL of sound pressure values over a time duration for a particular frequency band, the SPL monitoring system configured to determine at least one of: (a) an exposure time duration when at least one of the signals exceeds a SPL threshold value for the particular frequency band or (b) a recovery time duration when at least one of the signals is less than the SPL threshold value for the particular frequency band. Other embodiments are disclosed.

Abstract: A computer system may include at least one processor and a non-transitory, tangible, computer-readable storage medium having instructions stored thereon that, in response to execution by the at least one processor, cause the processor to perform operations including: (i) receiving, from a client device associated with a user within a user environment, sound data collected from the user environment; (ii) analyzing the sound data collected from the user environment; (iii) identifying, based upon the analyzing, a noise characteristic associated with the sound data and a duration of the sound data; (iv) comparing the noise characteristic and the duration to a predetermined noise threshold; (v) determining whether the sound data represents unsafe noise conditions based upon the comparison; and/or (vi) notifying the client device when the sound data is determined to represent unsafe noise conditions. The client device may generate an audible, haptic, and/or visible alert in response.

Abstract: A method for determining equalization filter parameters for a headphone, the method includes determining a composite response curve based on an average of amplitude response values measured from a plurality of measurement locations, the plurality of measurement locations, in cumulative, substantially spanning at least the headphone transducer, and determining the equalization filter parameters based on the determined composite response curve.

Abstract: A system includes a first source interface, a second source interface, a content mixer, a first communication interface (CI), and a second CI. The first source interface and the second source interface are configured to receive a first audio signal (FS) and a second audio signal (SS), respectively. The content mixer is configured to generate a portion of a terrestrial broadcast stream (TBS portion) that includes the FS and the SS. A first segment of the FS overlaps a second segment of the SS. The content mixer is configured to generate a portion of an internet radio stream (IR portion) corresponding to the TBS portion. The IR portion includes the first segment and omits the SS. The first CI is configured to send the TBS portion to a terrestrial radio station. The second CI is configured to send the IR portion to a content replacer. Other aspects are also disclosed.

Abstract: A security system includes a transducer within a monitored space responsive to audio signals therein. An analog-to-digital converter couples the transducer to the input of a digital signal processor. Time-staggered digital samples are stored, and are each multiplied by the current sample in a corresponding series of multipliers, and accumulated in a like series of accumulators. Accumulated sums, generated by the series of accumulators during known, undisturbed periods, are stored. During active monitoring, the processor compares currently generated accumulated sums to the previously stored accumulated sums generated during undisturbed periods. The processor creates an alarm signal when current values for the accumulated sums do not correspond to accumulated sums previously stored during undisturbed periods.

Abstract: A measurement device and measurement method for evaluating an acoustic device that allows a sound to be transmitted via vibration transmission by being held by a human ear includes an ear model unit having an ear model modeled after a human ear and an artificial external ear canal unit continuous with the ear model. The ear model unit is formed from rubber, a same material as an auricle model conforming to IEC 60318-7 or IEC 60268-7, or a material having a Shore hardness of from 30 to 60. The ear model is provided with an auricle or a hole which holds the acoustic device. A microphone is configured to measure an air-conducted component generated within an artificial ear canal of the artificial ear canal unit by vibration transmitted to the ear model unit from the acoustic device held by the auricle or the hole.

Abstract: A method for context-dependent emergency reporting includes receiving at a processor of a cellular phone a signal indicative of detecting a contextual cue. The exemplary method also includes receiving at the processor of the cellular phone a signal indicative of detecting a confirmatory cue. The exemplary method further includes the processor of the cellular phone identifying an emergency in response to detecting both the confirmatory cue and the contextual cue; and, in response to identifying the emergency, the processor of the cellular phone alerting a pre-registered recipient of the emergency via a wireless communication module of the cellular phone.

Abstract: According to one embodiment, a media system communicates with an aggregate device that includes multiple media output devices. When providing media data for presentation, the system adjusts for device clock drift by resampling the media data provided to a media output device based at least in part on a device clock rate difference between a device clock of one of the media output devices and a device clock of another of the media output devices.

Abstract: A noise detector, a method of detecting noise and an analysis/alert engine. In one embodiment, the noise detector includes: (1) a vibration sensor configured to derive a raw signal from noise proximate the noise detector and (2) a noise score generator having a processor and memory, coupled to the vibration sensor and configured to generate a noise score from the raw signal, the noise score being insufficient to reproduce a content of the raw signal.

Abstract: An audio mixer system and method is provided. The audio mixer system includes an audio interface, an audio signal processor, and a control circuit. The control circuit is configured to recognize a source type of an audio signal at the audio interface based on analyzing the audio signal. The control circuit is configured to control a setting of a signal path of the audio signal from an input of the audio interface through the audio signal processor to an output of the audio interface based on the recognized source type.

Abstract: The present invention relates to a method of evaluating quality of a degraded speech signal received from an audio transmission system conveying a reference speech signal. The method comprises sampling said signals into reference and degraded signal frames, and forming frame pairs by associating reference and degraded signal frames with each other. For each frame pair a difference function representing disturbance is provided, which is then compensated for specific disturbance types for providing a disturbance density function. Based on the density function of a plurality of frame pairs, an overall quality parameter is determined. The method provides for compensating the overall quality parameter for the effect that the impact of noise in frequency bands where there is only marginal speech activity when compared to natural speech is not correctly modelled in the current measurement standards.

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 technology for estimating a delay between a far-end audio signal and a near-end audio signal for acoustic echo cancellation is disclosed. A copy of the far-end signal is stored in a speaker buffer and organized in chunks, and a copy of the near-end signal is stored in a microphone buffer and organized in chunks. Cross correlation is performed on each pair of speaker chunks and microphone chunks based on ?-PHAse Transform (“PHAT”) generalized cross correlation (“GCC”). A peak correlation value can be obtained for each pair of the chunks. Offset values corresponding to the peak correlation values are collected and clustered. A best cluster is selected and the offset value represented by the selected cluster is identified as the estimated delay. Acoustic echo cancellation can be performed on the near-end signal based on the estimated delay.

Abstract: Embodiments described herein provide for smart configuration of audio settings for a playback device. According to an embodiment, while a playback device is a part of a first zone group that includes the playback device and at least one first playback device, the playback device applies a first audio setting. The embodiment also includes the playback device joining a second zone group that includes the playback device and at least one second playback device. The embodiment further includes the playback device applying a second audio setting based on an audio content profile corresponding to the second zone group.

Abstract: According to one embodiment, an electronic device includes a receiver and a hardware processor. The receiver is configured to receive an audio signal. The hardware processor is configured to enable a first function comprising separating the audio signal into a voice signal and a background sound signal and emphasizing or suppressing either the voice signal or the background sound signal and enable a second function comprising giving an acoustic effect to the audio signal. The hardware processor is further configured to receive an user operation to turn on either the first function or the second function and restrict the second function, if the first function is turned on.

Abstract: A system for intelligent acoustic monitoring. The system includes a microphone to capture environmental acoustic data and a processor coupled to the microphone. The processor is configured to receive and perform acoustic analysis on the captured acoustic data to generate an acoustic signature, based on a result of the acoustic analysis, identify an event indicated by the acoustic signature, and perform a remedial action based on the identified event.