Abstract: The program of the present invention is provided for an information terminal equipped with a touch panel that is used for displaying a musical score thereon, the information terminal performing music in accordance with performance sequence data corresponding to the musical score. The program displays the musical score on the touch panel, determines a start bar for a performance sequence as the start point for performing from any bar region of the musical score by sliding a touch position in the right direction on the particular bar region, and determines an end bar for the performance sequence as an end point for ending the performance in any bar region of the musical score by sliding the touch position in the left direction on the particular bar region. Thus, the user can intuitively designate a range of performance.

Abstract: A musical instrument switching system includes a) a control circuit controllable for combining and routing the analog audio, the control circuit comprising a plurality of instrument pickups for transmitting analog audio, and a controller connector for allowing connection of the control circuit to a computing device, and b) a software program, installable in the computing device, for modifying the characteristics of the musical instrument.

Abstract: There are disclosed automatic mixers and methods for creating a surround audio mix. A set of rules may be stored in a rule base. A rule engine may select a subset of the set of rules based, at least in part, on metadata associated with a plurality of stems. A mixing matrix may mix the plurality of stems in accordance with the selected subset of rules to provide three or more output channels.

Abstract: A method and apparatus for attenuating sound. Air, through which acoustic waves are traveling, is received within a core comprised of a plurality of cells. The sound created by the acoustic waves is attenuated using a set of channels through a number of cell interfaces between cells of the plurality of cells by allowing the air to flow between the cells of the plurality of cells through the set of channels.

Abstract: An active vibration noise control apparatus is provided. When a vehicle speed changes thereby to change frequency characteristics (peak-amplitude frequency) of vibration noise, the active vibration noise control apparatus refers to a vehicle speed versus frequency correspondence table representing a correspondence relation between a vehicle speed of a vehicle and a frequency of a reference signal, and changes the frequency of the reference signal that is used by an adaptive notch filter.

Abstract: A soundproof panel has a sandwich structure, which includes two outer walls, a core, and modifying elements that are maintained in a fractal distribution by the core. The panel has an improved trade-off between sound attenuation efficiency and the weight of the panel.

Abstract: A smart pillow unit and processes for providing active noise cancellation and biofeedback.

Abstract: An audio processor (200) for a hearing device has a noise estimator (100), an adaptive noise canceller (110), and a dynamic range compressor (120) to provide a gain according to a signal strength of an input signal. A signal strength processor (130) determines a signal strength dependent noise cancellation parameter, and the adaptive noise canceller provides noise cancellation of the input signal according to an estimated noise from the noise estimator, and according to the signal strength dependent noise cancellation parameter, so as to provide less noise cancellation when the input signal has higher signal strength than when the input signal has a lower signal strength. This helps retain the desirable environment or ambient type noise, present when the signal strength is higher, while providing more cancellation of undesirable circuit type noise when there is low input signal strength.

Abstract: A method and system for a noise cancellation comprises an amplifier in communication with the three or more speakers disposed in an area. A system controller produces a driver signal for each of the speakers in response to a signal from at least one microphone detecting sound in the area and communicates the driver signals to the amplifier. The amplifier drives each speaker with the driver signal produced for that speaker. In response to the driver signals, the speakers emit sound that combined produces a substantially uniform sound pressure field for a particular zone within the area. The substantially uniform sound pressure field produced by the speakers has a magnitude and phase adapted to attenuate a noise field in the area corresponding to the sound detected by the at least one microphone.

Abstract: Disclosed an acoustically transparent material including an acoustic wave steering material, and methods for fabrication and use thereof. The materials are specially designed structures of homogenous isotropic metals. These structures are constructed to propagate waves according to Pentamode elastic theory. The metamaterial structures are two-dimensional, intended to propagate acoustic waves in the plane in a manner which closely emulates the propagation of waves in water. The acoustically transparent materials described herein have particular utility as acoustic wave steering materials and acoustic cloaks.

Abstract: An acoustic lens or diffractive acoustic device, including but not limited to, a sub-wavelength thickness lens or diffuser, comprising an array of Helmholtz resonators (HRs) that provide perfect or near-perfect sound transmission through a rigid barrier. HRs are arranged in a line or an array confined within a waveguide and oriented so that one neck protrudes onto each side of the barrier. Extraordinary acoustic transmission (EAT) occurs when radiation (such as EM or acoustic radiation) incident on the barrier perforated with sub-wavelength holes is transmitted at a rate higher than expected based on the areal coverage fraction of the holes. Transmission is independent of the direction of sound on the barrier and the relative placement of the necks.

Abstract: A sound synthesizing apparatus includes a waveform storing section which stores a plurality of unit waveforms extracted from different positions, on a time axis, of a sound waveform indicating a voiced sound, and a waveform generating section which generates, for each of a first processing period and a second processing period, a synthesized waveform by arranging the plurality of unit waveforms on the time axis, wherein the second processing period is an immediately succeeding processing period after the first processing period.

Abstract: Systems, methods, and computer-readable storage media for providing for intelligent switching of languages and/or pronunciations in a text-to-speech system. As the system receives text, the text is analyzed to identify portions which should have speech constructed using a pronunciation distinct from the remaining portions of the text. The text-to-speech system uses multiple pronunciation dictionaries to generate and produce speech corresponding to the text, where the identified portions of the text are in a different language or have a different accent from the remainder of the text. Having generated speech corresponding to the text in multiple languages, accents, or dialects, the system combines the portions, then communicates the speech to the text recipient.

Abstract: A home appliance and an operation method thereof are disclosed. The operation method of the home appliance includes entering a voice recognition mode, receiving a voice data through a microphone, recognizing the received voice date, and, in a case in which the recognized voice data contains information related to another home appliance, transmitting the recognized voice data to the corresponding home appliance. Consequently, sharing of voice data between home appliances is achieved.

Abstract: Systems and methods are described for adding entries to a custom lexicon used by a speech recognition engine of a speech interface in response to user interaction with the speech interface. In one embodiment, a speech signal is obtained when the user speaks a name of a particular item to be selected from among a finite set of items. If a phonetic description of the speech signal is not recognized by the speech recognition engine, then the user is presented with a means for selecting the particular item from among the finite set of items by providing input in a manner that does not include speaking the name of the item. After the user has selected the particular item via the means for selecting, the phonetic description of the speech signal is stored in association with a text description of the particular item in the custom lexicon.

Abstract: An apparatus and a method are provided for building a spoken language understanding model. Labeled data may be obtained for a target application. A new classification model may be formed for use with the target application by using the labeled data for adaptation of an existing classification model. In some implementations, the existing classification model may be used to determine the most informative examples to label.

Abstract: A system, method, and computer-readable medium for performing ambiguous utterances identification operations by extrapolating statements of the utterance using machine learning based linguistic analysis. More specifically, in certain embodiments, the ambiguous utterances identification operations are performed by generating an ambiguous utterance repository that is indexed by and contains individuals, regions, tweets, blogs, and latest trends. This ambiguous utterance repository is then linked to a lexicon engine that stores linguistic semantics for particular demographics. The ambiguous utterances identification operations also can capture the latest trends in ambiguous utterances occurring happening in certain demographics.

Abstract: A speech recognition method and system enables user-configurable speech commands. For a given speech command, the speech recognition engine provides a mechanism for the end-user to select speech command terms to use in substitution for the given speech command. The speech recognition engine, or module thereof, forms a replacement command for the given speech command from the user-selected speech command terms. The speech recognition engine thereafter is synonymously responsive to user utterance of the replacement command as though the user issued/voiced the associated counterpart given speech command.

Abstract: Techniques for tailoring beamforming techniques to environments such that processing resources may be devoted to a portion of an audio signal corresponding to a lobe of a beampattern that is most likely to contain user speech. The techniques take into account both acoustic characteristics of an environment and heuristics regarding lobes that have previously been found to include user speech.

Abstract: Methods, systems, and apparatus, including computer programs encoded on computer storage media, for handing off a user conversation between computer-implemented agents. One of the methods includes receiving, by a computer-implemented agent specific to a user device, a digital representation of speech encoding an utterance, determining, by the computer-implemented agent, that the utterance specifies a requirement to establish a communication with another computer-implemented agent, and establishing, by the computer-implemented agent, a communication between the other computer-implemented agent and the user device.

Abstract: A speech processing engine of an HSC device transcribes user utterance input and displays resulting text on a microdisplay. The speech processing engine further provides to the user text editing functions and auto suggestions. The user interface utilizes head gesture control and natural language for text editing. Standard language dictionaries, user's dictionary and user's address book contacts data are utilized.

Abstract: Systems and vehicles that provide speech recognition system status notifications are described herein. Systems described herein include one or more processors, one or more memory modules, and machine readable instructions. In some embodiments described herein, when executed by the one or more processors, the machine readable instructions cause the speech recognition system to transform the acoustic vibrations received by a microphone into a speech input signal, determine to remotely process the speech input signal with a remote computing system, and provide a remote processing notification. In some embodiments described herein, when executed by the one or more processors, the machine readable instructions cause the speech recognition system to output an audible sequence with a speaker and provide a barge-in availability notification on a display.

Abstract: An electronic device is provided. The electronic device includes a processor configured to perform automatic speech recognition (ASR) on a speech input by using a speech recognition model that is stored in a memory and a communication module configured to provide the speech input to a server and receive a speech instruction, which corresponds to the speech input, from the server. The electronic device may perform different operations according to a confidence score of a result of the ASR. Besides, it may be permissible to prepare other various embodiments speculated through the specification.

Abstract: The application relates to HFR (High Frequency Reconstruction/Regeneration) of audio signals. In particular, the application relates to a method and system for performing HFR of audio signals having large variations in energy level across the low frequency range which is used to reconstruct the high frequencies of the audio signal. A system configured to generate a plurality of high frequency subband signals covering a high frequency interval from a plurality of low frequency subband signals is described.

Abstract: A method and apparatus for enhancing modulation of certain speech sounds, such as trill sounds, are provided for radios which utilize digital vocoders. A digitized speech stream is sampled and the sampling is adjusted to determine, detect and enhance trill nulls in the digitized voice stream by one or more of: frame shifting the digitized speech input stream prior to vocoding, time expanding a digitized speech steam prior to vocoding, time compressing a digitized speech output stream after vocoding, and/or modulation enhancement and filtering of the a digitized speech output stream after vocoding.

Abstract: Deep scattering spectral features are extracted from an acoustic input signal to generate a deep scattering spectral feature representation of the acoustic input signal. The deep scattering spectral feature representation is input to a speech recognition engine. The acoustic input signal is decoded based on at least a portion of the deep scattering spectral feature representation input to a speech recognition engine.

Abstract: The invention relates to a method and an apparatus for processing an audio signal, wherein the method comprises the steps of: filtering an audio signal into at least two frequency band signals, generating for each frequency band signal a plurality of sub-band signals, wherein for at least one frequency band signal the plurality of sub-band signals are generated using a time to frequency domain transform and for the at least one other frequency band the plurality of sub-band signals for the other frequency band are generated using a sub-band filterbank, and the apparatus comprises at least one processor and at least one memory including computer program code, the at least one memory and the computer program code being configured to, with the at least one processor, cause the apparatus to perform the method.

Abstract: A method performed in an audio decoder for decoding M encoded audio channels representing N audio channels is disclosed. The method includes receiving a bitstream containing the M encoded audio channels and a set of spatial parameters, decoding the M encoded audio channels, and extracting the set of spatial parameters from the bitstream. The method also includes analyzing the M audio channels to detect a location of a transient, decorrelating the M audio channels, and deriving N audio channels from the M audio channels and the set of spatial parameters. A first decorrelation technique is applied to a first subset of each audio channel and a second decorrelation technique is applied to a second subset of each audio channel. The first decorrelation technique represents a first mode of operation of a decorrelator, and the second decorrelation technique represents a second mode of operation of the decorrelator.

Abstract: An apparatus for generating a frequency enhancement signal has: a calculator for calculating a value describing an energy distribution with respect to frequency in a core signal; and a signal generator for generating an enhancement signal having an enhancement frequency range not included in the core signal, from the core signal, wherein the signal generator is configured for shaping the enhancement signal or the core signal so that a spectral envelope of the enhancement signal or of the core signal depends on the value describing the energy distribution with respect to frequency in the core signal.

Abstract: In a speech coding scheme based on a speech production model, such as a CELP-based scheme, an object of the present invention is to provide a decoding method that can reproduce natural sound even if the input signal is a noise-superimposed speech. The decoding method includes a speech decoding step of obtaining a decoded speech signal from an input code, a noise generating step of generating a noise signal that is a random signal, and a noise adding step of outputting a noise-added signal, the noise-added signal being obtained by summing the decoded speech signal and a signal obtained by performing, on the noise signal, a signal processing that is based on at least one of a power corresponding to a decoded speech signal for a previous frame and a spectrum envelope corresponding to the decoded speech signal for the current frame.

Abstract: An apparatus for processing an encoded signal, the encoded signal having an encoded audio signal having information on a pitch delay or a pitch gain, and a bass post-filter control parameter, has: an audio signal decoder for decoding the encoded audio signal using the information on the pitch delay or the pitch gain to obtain a decoded audio signal; a controllable bass post-filter for filtering the decoded audio signal to obtain a processed signal, wherein the controllable bass post-filter has the variable bass post-filter characteristic controllable by the bass post-filter control parameter; and a controller for setting the variable bass post-filter characteristic in accordance with the bass post-filter control parameter included in the encoded signal.

Abstract: An electronic device is provided. The electronic device includes a reception unit configured to receive an audio signal, a bandwidth change unit configured to gradually change a bandwidth of the received audio signal from a first bandwidth to a second bandwidth during a preset time, when the received audio signal is changed, and an audio output unit configured to output the received audio signal.

Abstract: To improve the intelligibility of speech for users with high-frequency hearing loss, the present systems and methods provide an improved frequency lowering system with enhancement of spectral features responsive to place-of-articulation of the input speech. High frequency components of speech, such as fricatives, may be classified based on one or more features that distinguish place of articulation, including spectral slope, peak location, relative amplitudes in various frequency bands, or a combination of these or other such features. Responsive to the classification of the input speech, a signal or signals may be added to the input speech in a frequency band audible to the hearing-impaired listener, said signal or signals having predetermined distinct spectral features corresponding to the classification, and allowing a listener to easily distinguish various consonants in the input.

Abstract: Described are noise suppression techniques applicable to various systems including automatic speech processing systems in digital audio pre-processing. The noise suppression techniques utilize a machine-learning framework trained on cues pertaining to reference clean and noisy speech signals, and a corresponding synthetic noisy speech signal combining the clean and noisy speech signals. The machine-learning technique is further used to process audio signals in real time by extracting and analyzing cues pertaining to noisy speech to dynamically generate an appropriate gain mask, which may eliminate the noise components from the input audio signal. The audio signal pre-processed in such a manner may be applied to an automatic speech processing engine for corresponding interpretation or processing.

Abstract: Embodiments of the invention include a device for reducing noise. The device may include a storage configured to store noise data; a processor configured to: classify a segment of noise utilizing noise data which was accumulated prior to initiation of a communication session; estimate the segment of noise, utilizing information received from the noise classification; and select a noise profile which accounts for a user's current context based on a context defined by the data which was accumulated prior to initiation of the communication session.

Abstract: Speech recognition error may be reduced or eliminated when background noise is detected at a customer's location. For example, when background noise is detected at the customer's location, the customer may be prompted to use dual-tone multi-frequency (DTMF).

Abstract: Methods and systems are provided for separating a target speech from a plurality of other speeches having different directions of arrival. One of the methods includes obtaining speech signals from speech input devices disposed apart in predetermined distances from one another, calculating a direction of arrival of target speeches and directions of arrival of other speeches other than the target speeches for each of at least one pair of speech input devices, calculating an aliasing metric, wherein the aliasing metric indicates which frequency band of speeches is susceptible to spatial aliasing, enhancing speech signals arrived from the direction of arrival of the target speech signals, based on the speech signals and the direction of arrival of the target speeches, to generate the enhanced speech signals, reading a probability model, and inputting the enhanced speech signals and the aliasing metric to the probability model to output target speeches.

Abstract: Apparatus, including a sensor which is configured to be fixed to a neck of an operator of equipment in a location suitable for sensing a voiceless alaryngeal speech vibration generated by the operator during operation of the equipment, The apparatus further includes a processor which is configured to receive and process a signal output by the sensor so as to measure the voiceless alaryngeal speech vibration and so as to generate a control signal for the equipment responsively to the measured voiceless alaryngeal speech vibration.

Abstract: A headset computer device provides user voice indication of location of the device. The user may implicitly or explicitly present by voice input his and hence the HSC device location. A voice driven location module is coupled to the voice recognition engine, a map database and GPS of the HSC device. Based on user voiced indications of 3D space location, the voice driven location module determines device location and resets 3D space location accordingly.

Abstract: An apparatus includes a function module, a strength module, and a filter module. The function module compares an input signal, which has a component, to a first delayed version of the input signal and a second delayed version of the input signal to produce a multi-dimensional model. The strength module calculates a strength of each extremum from a plurality of extrema of the multi-dimensional model based on a value of at least one opposite extremum of the multi-dimensional model. The strength module then identifies a first extremum from the plurality of extrema, which is associated with a pitch of the component of the input signal, that has the strength greater than the strength of the remaining extrema. The filter module extracts the pitch of the component from the input signal based on the strength of the first extremum.

Abstract: Implementations disclosed herein provide a method comprising comparing high-latency data sectors of a storage band, the high-latency data sectors having latency above a predetermined threshold, with target sectors for storing new data to determine one or more of the high-latency data sectors that may be skipped during retrieval of at-rest data from the storage band.

Abstract: A method of manufacturing a plasmon generator includes the steps of: forming an initial film made of a metal polycrystal and including a pre-plasmon-generator portion that later becomes the plasmon generator; heating the initial film with heating light so that a plurality of crystal grains constituting the metal polycrystal grow at least in the pre-plasmon-generator portion; stopping the heating of the initial film; and forming the plasmon generator by processing the initial film after the step of stopping the heating. The step of forming the plasmon generator includes the step of providing the pre-plasmon-generator portion with a front end face that generates near-field light.

Abstract: An apparatus comprises a recording transducer configured for microwave-assisted magnetic recording (MAMR) and a magnetic recording medium. The recording transducer comprises a write pole configured to generate a write magnetic field and a write-assist arrangement proximate the write pole. The write-assist arrangement is configured to generate a radio frequency assist magnetic field at a plurality of different frequencies. The magnetic recording medium comprises a plurality of tracks, wherein adjacent tracks of the medium have different ferromagnetic resonant frequencies.

Abstract: Implementations disclosed herein provide an apparatus comprising a write pole, and a two-layer front shield formed on the write pole, the front shield comprising, a first dielectric material formed on the first layer of the front shield, an active shield control (ASC) device formed between the two layers of the front shield on the first dielectric material configured to synchronize the response to a magnetomotive force (MMF) of a write pole and the front shield, and a second dielectric material formed on the ASC device, wherein the second layer of the front shield is formed only on top of the second dielectric material and the first layer of the front shield.

Abstract: A magnetic recording-reproducing head includes, on a support member, a first head section having a function of executing a recording process and having a function of executing a reproducing process, and one or more second head sections each having a function of executing an erasing process and having no function of executing a reproducing process.

Abstract: A thermal assisted magnetic recording head executing magnetic recording while locally heating a magnetic recording medium includes a plasmon generator generating surface plasmon and generating near-field light from the surface plasmon at an end surface situated on an air bearing surface facing the magnetic recording medium, a main pole being in contact with the plasmon generator and exposed on the air bearing surface, a metal protective layer situated on an opposite side to the plasmon generator when viewed from the main pole and positioned to overlap with a part of the main pole when viewed from one side in a down track direction, and an overcoat protective layer covering the metal protective layer. The overcoat protective layer is formed on a flat surface at least at a position where it overlaps with the main pole when viewed from one side in the down track direction, and the metal protective layer configures a part of the flat surface.

Abstract: In order to prevent leaking of lubricant, the bearing device includes a rolling bearing having an outer ring and an inner ring, a shaft fixed to the inner ring, and a sleeve fixed to the outer ring. A flange part is formed on at least one end part of the shaft. The flange part includes a disk part extending from the shaft toward an outside in a radial direction and a cylindrical part extending from the disk part along an axial direction. The cylindrical part is disposed at an outside of the outer ring, and a labyrinth gap which bends at two places is formed by gaps.

Abstract: A method for calibrating a tape drive is disclosed herein. The method includes writing a calibration track onto a tape cartridge using a first tape drive. The method includes reading the calibration track using a plurality of tape drives to measure a plurality of write element offset values, in which each write element offset value corresponds to each of the plurality of tape drives. The method includes determining a calibration factor based on an average write element offset value, in which the calibration factor is used to align a write element of a drive head of the first tape drive.

Abstract: In one embodiment, a computer-implemented method for simulating a defect signal in a time based servo system includes determining a start position for a spike, determining an end position for the spike, determining a servo channel, determining an amplitude of the spike, initiating an acquire lock process, creating the spike in the determined servo channel from the determined start position to the determined end position, and storing servo data including at least a portion thereof generated during the spike. The spike has the determined amplitude. In another embodiment, a computer program product for generating defect signals in a storage system includes a computer readable storage medium having program instructions embodied therewith, wherein the computer readable storage medium is not a transitory signal per se. The program instructions are executable by a controller to perform the foregoing method.

Abstract: A write head is positioned over a disk surface of a hard disk drive during a self-servo write process. A virtual window mode is employed for positioning the write head, in which the read channel of the hard disk drive continuously searches for servo spirals, but spiral detection is disabled except during specific time intervals, or “virtual windows.” Each virtual window is associated with one specific servo spiral on the disk surface, and has a predetermined duration. The duration and timing of these virtual windows may be selected to ensure that spiral detection is enabled while a read head of the hard disk drive passes over the associated servo spiral, even when servo spiral placement is not ideal.