Abstract: A method and apparatus for controlling a display of a portable electronic device are provided. The method comprises generating a screen comprising a list on the display. The list comprises one or more row elements, with each row element comprising one or more column sub-elements. At least a portion of one of the column sub-elements of at least one of the row elements is truncated. Upon selection of a row element, the screen is enabled for scrolling of that row element. Enabling scrolling comprises loading the truncated portion of the selected row element into a pre-buffer. The method further comprises receiving a scrolling input, and updating the screen on the display to scroll the row element, bringing the truncated portion into view. The selected row is scrolled while maintaining the series of column sub-elements in the row without reference to other rows.

Abstract: A system, for determining the velocity of a hammer of a keyboard instrument, with a light transceiver and processing circuitry. The light transceiver being configured to transmit a light signal to a hammer to measure a hammer velocity, receive a reflected light signal from the hammer indicative of the velocity of the hammer and send an electrical signal to the processing circuitry, where the electrical signal is based on the reflected light signal from the hammer. The processing circuitry being configured to receive and process the electrical signal so that a time interval between the electrical signal passing through a first trigger point and a second trigger point and the velocity of the hammer can be determined.

Abstract: An adjustable guitar saddle is provided. The adjustable guitar saddle includes saddle piece. Each saddle piece includes a top portion and a bottom portion. At least the bottom portion may be sized to fit within a saddle bridge channel of a guitar. The saddle piece may include at least one bottom threaded aperture running through the bottom portion towards the top portion. At threaded screw may mate with the threaded aperture. The threaded screw rotates and adjusts the height of the adjustable guitar saddle relative to the saddle bridge. The bottom portion of the saddle piece may be placed within the saddle bridge channel and a guitar string may rest on the top portion.

Abstract: A mapping estimation apparatus includes a mapping adjuster. The mapping adjuster estimates mappings which correlate a plurality of subset data items with respective parts of universal set data including union of the plurality of subset data items based on the plurality of subset data items and the universal set data. The mapping adjuster estimates a mode of selecting a plurality of codomain data items from the universal set data and modes of mappings applied to the plurality of subset data items so as to have a maximum probability that data items obtained by selecting a plurality of codomain data items of which a subset of union is the universal set data from the universal set data and applying the mappings to the plurality of subset data items as domains will be respectively the plurality of codomain data items.

Abstract: In one embodiment, a system for analyzing melodies is disclosed that may receive data for a musical composition, the data including a sequence of notes that make up a melody contained in the musical composition, and store, in a database, a pitch and a duration for at least one note in the sequence of notes in association with the melody. In addition, the system may determine at least one pitch differential and at least one duration ratio for at least two consecutive notes in the sequence of notes, and store, in the database, the at least one pitch differential and the at least one duration ratio in association with the melody. The system may also generate at least one metric associated with the melody based on the at least one pitch differential and/or the at least one duration ratio.

Abstract: A passive feedback mechanism by which sound produced by a musical instrument is enhanced by utilizing the reverberations from the sound to dynamically and passively alter the magnetic field in the vicinity of an electronic pickup is disclosed. This is accomplished by utilizing a component or material adhered to the instrument with the property to alter the magnetic field as the material vibrates in response to sound reverberation, one embodiment being a ferromagnetic surface coating. An electromagnetic pickup, sensitive to these variations in the magnetic field, is installed on or within the instrument. The changes in the magnetic field, in response to the music, are captured by the pickup and transformed into an electric signal with thus produce a unique and enhanced sound.

Abstract: A music generating system comprising sensors configured to acquire data sets related to physiological and environmental condition of a user and a computing device comprising a processor and a memory. The computing device is configured to read, by a data acquisition module, the data sets received from the sensors, to perform, by a data analysis module, a value range analysis of the data sets received to normalize the data sets, to map, with the help of the data analysis module, each of the normalized data sets to families of sounds stored in a data store, to create, by the data analysis module, individual music pieces on the basis of a time series analysis and a frequency analysis of the normalized data sets mapped to the families of sounds and to merge, by a music generation module, the individual music pieces to create a final piece of music.

Abstract: The invention relates to a vibration plate for a striking mechanism of musical watches or music boxes. According to the invention, the vibration plate includes at least two parts, joined to each other, manufactured in at least two materials, a first part forming a first set of strips made of a first material so as to produce sounds in a first frequency range, and at least a second part forming a second set of strips made of a second material so as to produce sounds in a second frequency range.

Abstract: A method for noise cancellation. The method includes providing a sound signal output device having a listening device output, a microphone input, and signal processing circuitry. The signal processing circuitry is adapted for causing the sound signal output device to perform in a noise-cancellation mode, including making available to the sound signal output device, at the microphone input, a noise-containing signal which is representative of ambient acoustical noise, deriving a noise-cancelling signal from the noise-containing signal, and making available to the personal listening device, at the listening device output, the noise-cancelling signal.

Abstract: The subject matter herein field is for multiple and complimentary systems in the structure of a cellular device case so as to provide sound jamming and/or sound insulation, and/or powered noise cancellation as means of restriction the entry of useful sound into the microphone(s) of cellular smartphone devices, as a means of citizen counter-action of the “constant surveillance” characteristics inherent in the “always on,” microphone systems as described under Background of the Subject matter, above.

Abstract: The present invention relates to an ultrasound probe (60) comprising: an ultrasound transducer (12) having an emission surface (24) for generating ultrasound waves, and an acoustical lens (12) with a first part (64) having an inner surface (66) facing the emission surface (24), wherein the inner surface (64) comprises a plurality of protrusions (74) and/or recesses (76) for scattering reflections of ultrasound waves.

Abstract: Disclosed herein are systems, methods, and non-transitory computer-readable storage media for reducing latency in web-browsing TTS systems without the use of a plug-in or Flash® module. A system configured according to the disclosed methods allows the browser to send prosodically meaningful sections of text to a web server. A TTS server then converts intonational phrases of the text into audio and responds to the browser with the audio file. The system saves the audio file in a cache, with the file indexed by a unique identifier. As the system continues converting text into speech, when identical text appears the system uses the cached audio corresponding to the identical text without the need for re-synthesis via the TTS server.

Abstract: A systems, article, and method of automatic speech recognition using on-the-fly word lattice generation with word histories.

Abstract: Pairs of feature vectors are obtained that represent speech. Some pairs represent two samples of speech from the same speakers, and other pairs represent two samples of speech from different speakers. A neural network feeds each feature vector in a sample pair into a separate bottleneck layer, with a weight matrix on the input of both vectors tied to one another. The neural network is trained using the feature vectors and an objective function that induces the network to classify whether the speech samples come from the same speaker. The weights from the tied weight matrix are extracted for use in generating derived features for a speech processing system that can benefit from features that are thus transformed to better reflect speaker identity.

Abstract: Methods and systems for providing help prompts to a user of an automated dialog system are presented. In some embodiments, a computing device may receive a help request from the user of an automated dialog system. The help request may comprise a user request for information about one or more capabilities of the automated dialog system. The computing device may identify information expected to be input by the user to request that the automated dialog system perform its one or more capabilities. A natural language help prompt may be generated to provide guidance to the user to provide the identified information expected to be input.

Abstract: A system and method for processing an audio signal captured from a microphone may reproduce a known audio signal with an audio transducer into an acoustic space. The known audio signal may include content from one or more audio sources. A microphone audio signal may be captured from the acoustic space where the microphone audio signal comprises the known audio signal and one or more unknown audio signals. Processing control information may be accessed. The known audio signal may be reduced in the microphone audio signal responsive to the processing control information where the processing control information indicates one or more characteristics of a downstream audio processor that processes the microphone audio signal.

Abstract: An information-sharing system has a server connected to a network, the server having a processor and a coupled data repository, with software executing on the processor from a non-transitory medium providing system intelligence, a plurality of computerized communication devices coupled to the network, each having a microphone, a speaker, and a display screen, each executing coded instructions providing local intelligence at least presenting interactive interfaces to a user. The user is enabled by the coded instructions executing at the computerized communication device to record an audio or audio-video input which the system, by use of natural language processing, determines to be either a voice command or a Shout, as a situational report local to the user, and if the enunciation is determined to be a Shout is transmitted to the Server, where it is processed and transmitted to other users according to determinations made by the server intelligence.

Abstract: A voice recognition method and device, for improving efficiency and accuracy of voice recognition. The method comprises: receiving a trigger message of an operation class to be operated for operating on a mobile terminal, wherein the operation class is a class divided according to the service function of the mobile terminal (S101); receiving voice keyword information and determining a voice keyword from the voice keyword information (S102); and retrieving a keyword library under an operation class entry to be operated in accordance with the voice key word, and returning a search result (S103).

Abstract: A system and method for providing a voice assistant including receiving, at a first device, a first audio input from a user requesting a first action; performing automatic speech recognition on the first audio input; obtaining a context of user; performing natural language understanding based on the speech recognition of the first audio input; and taking the first action based on the context of the user and the natural language understanding.

Abstract: A speech processing apparatus, method and non-transitory computer-readable storage medium are disclosed. A speech processing apparatus may include a memory storing instructions, and at least one processor configured to process the instructions to calculate an acoustic diversity degree value representing a degree of variation in types of sounds included in a speech signal representing a speech, on a basis of the speech signal, and compensate for a recognition feature value calculated to recognize specific attribute information from the speech signal, using the acoustic diversity degree value.

Abstract: Provided are systems and methods for operating a dialog system in a cross-platform environment. The method comprises receiving, by a server comprising at least one processor and a memory storing processor-executable codes, a first request from a first client device to initiate operation of the dialog system. The first client device is identified based at least on the first request. Based on the identification, a first set of predetermined settings associated with a user and the first client device is applied to the dialog system. The operation of the dialog system according to the first set of predetermined settings is initiated and the dialog system is connected to the first client device.

Abstract: An illustrative method includes receiving, at a processor of a computing device, an audio voice signal of a first call participant during a first call, where the first call is a communication across a communication network. The method further includes determining an identity of the first call participant and determining a speech to text profile associated with the identity of the first call participant, where the speech to text profile includes at least one rule for transcribing a word in the audio voice signal into text. The method further includes generating a text output, where the text output is a transcribed version of a plurality of words identified in the audio voice signal of the first call participant. At least one of the plurality of words identified is identified using the at least one rule.

Abstract: Provided is a lightweight computational device that is configured to be in communication with a cloud both directly and via a smart computational device. The lightweight computational device receives a voice command from a user, wherein the lightweight computational device does not have adequate processing power to convert the voice command to a text command. The voice command is transmitted from the lightweight computational device to a smart computational device, wherein the smart computational device uses voice recognition to convert the voice command to a text command in the smart computational device, and transmits the text command for being processed by that cloud that provides at least one of voice recognition service and other services. The lightweight computational device receives a data response for the user from the cloud, via the smart computational device, based on the other services provided by the cloud.

Abstract: Provided is a lightweight computational device that is configured to be in communication with a cloud both directly and via a smart computational device. The lightweight computational device receives a voice command from a user, wherein the lightweight computational device does not have adequate processing power to convert the voice command to a text command. The voice command is transmitted from the lightweight computational device to a smart computational device, wherein the smart computational device uses voice recognition to convert the voice command to a text command in the smart computational device, and transmits the text command for being processed by that cloud that provides at least one of voice recognition service and other services. The lightweight computational device receives a data response for the user from the cloud, via the smart computational device, based on the other services provided by the cloud.

Abstract: Systems and methods for voice searching media content based on metadata or subtitles are provided. Metadata associated with media content can be pre-processed at a media server. Upon receiving a vocal command representative of a search for an aspect of the media content, the media server performs a search for one or more portions of the media content relevant to the aspect of the media content being searched for. The media performs the search by matching the aspect of the media content being searched for with the pre-processed metadata.

Abstract: To detect events in an audio stream, frames of an audio signal (e.g., frames generated by a codec for a voice call or music stream) are received. Based on information in the frames, an index is used to look up an entry in a table associated with the codec. Each entry in the table indicates a likelihood that a frame matches a sound model element. The likelihood is used in the search for a sound bite, word, and/or phrase in the audio signal. The process of dynamic programming is used to find the combined likelihood for a match of the word, phrase, and/or sound bite to a region of the audio stream. Upon detection of the word, phrase, and/or sound bite in the audio stream, an event is generated, such as, notifying a person or logging the event in a database.

Abstract: A post-encoding bitrate reduction system and method for generating one more scaled compressed bitstreams from a single encoded plenary file. The plenary file contains multiple audio object files that were encoded separately using a scalable encoding process having fine-grained scalability. Activity in the data frames of the encoded audio object files at a time period are compared with each other to obtain a data frame activity comparison. Bits from an available bitpool are assigned to all of the data frames based on the data frame activity comparison and corresponding hierarchical metadata. The plenary file is scaled down by truncating bits in the data frames to conform to the bit allocation. In some embodiments frame activity is compared to a silence threshold and the data frame contains silence if the frame activity is less than or equal to the threshold and minimal bits are used to represent the silent frame.

Abstract: In general, techniques are directed to performing normalization with respect to ambient higher order ambisonic audio data. A device configured to decode higher order ambisonic audio data may perform the techniques. The device may include a memory and one or more processors. The memory may be configured to store an audio channel that provides a normalized ambient higher order ambisonic coefficient representative of at least a portion of an ambient component of a soundfield. The one or more processors may be configured to perform inverse normalization with respect to the audio channel.

Abstract: An apparatus for and a method of processing a multi-channel audio signal using space information. The apparatus includes: a main coding unit down mixing a multi-channel audio signal by applying space information to surround components included in the multi-channel audio signal, generating side information using the multi-channel audio signal or a stereo signal of a down-mixed result, coding the stereo signal and the side information, and transmitting the coded result as a coding signal; and a main decoding unit receiving the coding signal, decoding the stereo signal and the side information using the received coding signal, up mixing the decoded stereo signal using the decoded side information, and restoring the multi-channel audio signal.

Abstract: A digital watermark embedding device includes a synthesized voice generating unit that outputs a synthesized voice according to an input text and outputs phoneme-based alignment regarding phonemes included in the synthesized voice; an estimating unit that estimates whether or not a potentially risky expression is included in the input text, and outputs a potentially risky segment in which the potentially risky expression is estimated to be included; an embedding control unit that associates the potentially risky segment with the phoneme-based alignment, and decides and outputs an embedding time for embedding a watermark in the synthesized voice; and an embedding unit that embeds a digital watermark in the synthesized voice at a time specified as the embedding time for the synthesized voice.

Abstract: A noise filling method is provided that includes detecting a frequency band including a part encoded to 0 from a spectrum obtained by decoding a bitstream; generating a noise component for the detected frequency band; and adjusting energy of the frequency band in which the noise component is generated and filled by using energy of the noise component and energy of the frequency band including the part encoded to 0.

Abstract: The present proposes new methods and an apparatus for enhancement of source coding systems utilising high frequency reconstruction (HFR). It addresses the problem of insufficient noise contents in a reconstructed highband, by Adaptive Noise-floor Addition. It also introduces new methods for enhanced performance by means of limiting unwanted noise, interpolation and smoothing of envelope adjustment amplification factors. The present invention is applicable to both speech coding and natural audio coding systems.

Abstract: According to one embodiment, an electronic device includes circuitry configured to perform a process for suppressing a noise of a sound signal by a first suppression amount when a first reproduction speed of the sound signal is set to a first value by user, wherein the circuitry is configured to perform a process for suppressing a noise of a sound signal by a second suppression amount larger than the first suppression amount when a second reproduction speed of the sound signal is set to a second value lower than the first value by a user, and the circuitry is configured to reproduce a noise-suppressed sound signal in accordance with the first reproduction speed or the second reproduction speed set by a user.

Abstract: This specification describes, among other things, a computer-implemented method. The method can include receiving a stream of audio data at a computing device. The stream of audio data can be segmented into a plurality of audio segments. Respective intensity levels are determined for each of the plurality of audio segments. For each of the plurality of audio segments and based on the respective intensity levels, a determination can be made as to whether the audio segment includes a speech signal. Selective gain control can be performed on the stream of audio data by automatically adjusting a gain of particular ones of the plurality of audio segments that are determined to include a speech signal.

Abstract: The application relates to a hearing device comprising a) an input unit for delivering a time varying electric input signal representing an audio signal comprising at least two sound sources, b) a cyclic analysis buffer unit of length A adapted for storing the last A audio samples, c) a cyclic synthesis buffer unit of length, where L is smaller than A, adapted for storing the last L audio samples, which are intended to be separated in individual sound sources, d) a database having stored recorded sound examples from said at least two sound sources, each entry in the database being termed an atom, the atoms originating from audio samples from first and second buffers corresponding in size to said synthesis and analysis buffer units, where for each atom, the audio samples from the first buffer overlaps with the audio samples from the second buffer, and where atoms originating from the first buffer constitute a reconstruction dictionary, and where atoms originating from the second buffer constitute an analysis d

Abstract: An electronic device for reproducing data and a method thereof are provided. The electronic device includes a memory; and a processor. The processor is configured to determine a spatial characteristic of a space where content is reproduced, store the spatial characteristic in the memory, and change a sound of the content based on the spatial characteristic when the content is reproduced.

Abstract: Methods, systems, and apparatus, including computer programs encoded on computer storage media, for identifying the language of a spoken utterance. One of the methods includes receiving input features of an utterance; and processing the input features using an acoustic model that comprises one or more convolutional neural network (CNN) layers, one or more long short-term memory network (LSTM) layers, and one or more fully connected neural network layers to generate a transcription for the utterance.

Abstract: A method for assessing suicide risk for a human subject including receiving recorded voice data of the subject; and classifying the subject as suicidal or non-suicidal based upon a computerized analysis of one or more nonverbal characteristics of the speech data, especially features associated with a breathy phonation type. The analysis of the nonverbal characteristics of the voice data can include an analysis of acoustic characteristics of speech, and/or an analysis of prosodic and voice quality-related features of the voice data. Related apparatus, systems, techniques and articles are also described.

Abstract: An estimate of a pitch of a signal may be computed by using correlations of frequency portions of a frequency representation of the signal. An initial pitch estimate may be obtained and frequency portions of the frequency representation may be identified using multiples of the initial pitch estimate. Correlations of the frequency portions may be computed, and a score for the initial pitch estimate may be determined using the correlations. A second pitch estimate may be determined using the first score, and the process may be repeated.

Abstract: A method for making an interferometric taper waveguide (I-TWG) with high critical dimension uniformity and small line edge roughness for a heat assisted magnetic recording (HAMR) head, wherein the method includes creating an I-TWG film stack with two hard mask layers on top of an I-TWG core layer sandwiched between two cladding layers, defining a photoresist pattern over the I-TWG film stack using deep ultraviolet lithography, transferring the pattern to the first hard mask layer using reactive ion etching (RIE), forming a temporary I-TWG pattern on the second hard mask layer using RIE, transferring the temporary pattern to the I-TWG core using RIE, refilling the cladding layer, and planarizing using chemical mechanical planarization (CMP).

Abstract: A magnetoresistive element according to an embodiment includes: a multilayer element including a first magnetic layer, a magnetization direction of the first magnetic layer being pinned, a nonmagnetic layer disposed on the first magnetic layer, a second magnetic layer disposed in a first region on the nonmagnetic layer, a magnetization direction of the second magnetic layer being pinned and antiparallel to the magnetization direction of the first magnetic layer, and a third magnetic layer disposed in a second region that is different from the first region on the nonmagnetic layer near one of two opposite end faces of the nonmagnetic layer, a magnetization direction of the third magnetic layer being changeable by an external magnetic field, a lower face of the nonmagnetic layer being in contact with an upper face of the first magnetic layer.

Abstract: A magnetic head according to an embodiment includes: a non-local spin valve element with three terminals including a nonmagnetic base layer, a first terminal including a first magnetic layer, a second terminal including a second magnetic layer, and a third terminal including a third magnetic layer; and a slider including a first external lead terminal connecting to the first terminal, a second external lead terminal connecting to the second terminal, and a third external lead terminal connecting to the third terminal, in an operation, a first current being caused to flow from the second external lead terminal to the third terminal via the second terminal and the nonmagnetic base layer, and a second current that is lower than the first current being caused to flow to the first terminal.

Abstract: A stack includes a substrate and a magnetic recording layer. Disposed between the substrate and magnetic recording layer is an MgO—Ti(ON) layer.

Abstract: According to one embodiment, a magnetic recording medium includes a substrate, and a magnetic recording layer structure positioned above the substrate, the magnetic recording layer structure including: a first magnetic recording layer having a first plurality of magnetic grains surrounded by a first segregant; a second magnetic recording layer positioned above the first magnetic recording layer, the second magnetic recording layer having a second plurality of magnetic grains surrounded by a second segregant; and a third magnetic recording layer positioned above the second magnetic recording layer, the third magnetic recording layer having a third plurality of magnetic grains surrounded by a third segregant, where at least the first segregant is primarily a combination of carbon and a second component, and where the second segregant is primarily carbon.

Abstract: A magnetic recording medium includes a flexible substrate, an amorphous seed layer, an under layer containing Ru, and a recording layer having a granular structure. The seed layer is provided between the substrate and the under layer. The under layer has a thickness in a range of 5 to 50 nm.

Abstract: Implementations disclosed herein provide a method of reducing the topography at the alignment and overlay marks area during the writer pole photolithography process in order to reduce the wafer scale variation and reduce the writer pole photolithography process rework rate. In one implementation, an intermediate stage of a wafer for writer pole formation is generated by removing a part of at least one metallic writer pole layer on top of an intermediate stage writer pole wafer to form a recovery trench, depositing an optically transparent material on top of the wafer, wherein the thickness of the optically transparent material is higher than a target recovery trench topography, forming a photoresist pattern on top of the optically transparent material over the recovery trench, etching the optically transparent material, and removing the photoresist pattern and at least part of the remaining optically transparent material.

Abstract: Apparatus for recording data and method for making the same. In accordance with some embodiments, a magnetic layer is supported by a substrate and comprises a magnetic magnetic material, a non-magnetic material, and an energy assisted segregation material. The segregation material enhances segregation of the non-magnetic material into grain boundaries within the layer at an elevated, moderate energy level.

Abstract: Systems and methods for a multi-speed programmable batch scrubber are illustrated. The disclosed system includes a brush configured to scrub a disk at a plurality of speeds during cleaning of the disk, a combing system configured to support the disk during cleaning, a cleaning solution dispenser, and a first deionized water dispenser configured to dispense water on the brush during cleaning. The disk is cleaned in a plurality of predetermined cleaning steps, each step including predetermined parameter values such as the brush RPM, the step duration, the dispensing state of the cleaning solution dispenser, and the dispensing state of the deionized water dispenser.

Abstract: Systems and methods for encoding and playing back video at adjustable playback speeds by interpolating frames to achieve smooth playback in accordance with embodiments of the invention are described. One embodiment includes a source encoder that includes a processor, memory including an encoder application, where the encoder application directs the processor to: select a subset of frames from a first video sequence; generate motion vectors describing frames from the first video sequence that are not part of the selected subset of frames, where each motion vector describes movement between a frame in the subset of frames and a frame not included in the subset of frames; store the motion vectors; decimate frames not included in the subset of frames from the first video sequence to generate a second video sequence having a nominal frame rate less than the frame rate of the first video sequence; and encode the second video sequence at the nominal frame rate.

Abstract: Embodiments described herein include systems and methods for automatically creating compilation videos from an original video based on metadata associated with the original video. For example, a method for creating a compilation video may include determining a relevance score for video frames in an original video; selecting a plurality of relevant video frames from the original video based on the relevance score; selecting a plurality of video clips from the original video based on the relevance scores of the video frames; and creating a compilation video from the plurality of video clips. Each of the plurality of video clips, for example, may include at least one relevant video frame from the plurality of relevant video frames.