Abstract: A display controller includes a local contrast enhancement circuit that includes a plurality of scale circuits. Each scale circuit includes a block separation circuit configured to divide input display data into sub blocks, a calculation circuit configured to calculate a feature value that characterizes each of the sub blocks, and a storage device configured to store the feature value. The number and size of the sub blocks differ among the scale circuits.

Abstract: According to an aspect, an electronic device includes a first face, a second face, a display unit arranged on the first surface, a notification unit arranged on the second surface, an attitude detecting unit, and a control unit. The attitude detecting unit detects whether attitude of the electronic device is a first attitude, in which the first surface faces upward in the vertical direction, or a second attitude, in which the first surface faces downward in the vertical direction. When the attitude detecting unit detects the second attitude, the control unit prevents displaying on the display unit and enables the notification unit to give a notification.

Abstract: In accordance with embodiments of the present disclosure, an information handling system may include a processor, a user interface, and a flexible integrated circuit. The user interface may include a user interface comprising a display, a touch sensor overlaid on the display and configured to detect tactile touches on the user interface, a cover overlaid on the touch sensor, and an antenna array comprising a plurality of antennas. The flexible integrated circuit may be configured to couple the antenna array to the processor, the flexible integrated circuit comprising a radio-frequency module for controlling radio-frequency communications to and from the antenna array and a baseband controller for controlling communication of information between the processor and the antenna.

Abstract: Disclosed herein is a computing device configured to send display data to a display through a near-field communication (NFC) interface. The computing device includes a chassis, a primary display, and a near-field communication interface to transmit display data to a secondary display.

Abstract: A data processing apparatus has a compression circuit and an output interface. The compression circuit has a pre-processor and a compressor. The pre-processor receives a first input display data in a first color domain, and performs a color format conversion upon the first input display data to generate a second input display data in a second color domain, wherein the second color domain is different from the first color domain. The compressor performs compression in the second color domain, wherein the compressor is arranged to compress the second input display data into a compressed display data in the second color domain. The output interface packs an output display data derived from the compressed display data into an output bitstream, and outputs the output bitstream via a display interface.

Abstract: A device for generating a display image on a composite glass pane is described. The device has: a composite glass pane with a first pane and a second pane, which are connected to each other via an intermediate layer that contains at least one luminescent pigment; a laser projector, which contains at least one laser and an optical system for deflecting a radiation of the laser, the radiation being directed within a region of the composite glass pane; a photodetector, which detects luminescent radiation of the luminescent pigment; and a control unit, which electronically connects the photodetector and the laser projector.

Abstract: In a method of avoiding interference in an integrated capacitive sensor device and display device, a first transmitter signal having a first frequency is transmitted with a combination electrode of the integrated capacitive sensor device and display device. The combination electrode is configured for both capacitive sensing and display updating. A shift is made from transmitting the first transmitter signal with the combination electrode to transmitting a second transmitter signal with the combination electrode. The shift is made by changing a duration of at least one non-display update time period of the display device. The second transmitter signal has a second frequency, and the shifting occurs based at least in part upon an amount of interference.

Abstract: Methods, apparatuses, and computer readable storage media for increasing the weight of a character are described. In one aspect, the method includes receiving a vector character description and an emboldening parameter of an initial character. Vertices of an emboldened representation of the initial character are calculated by moving each vertex of the initial character based on a combination of the emboldening parameter and a stroke width for each vertex associated with the vertex, such that each stroke of the emboldened representation of the initial character is thickened based on the width of the stroke. The emboldened representation of the initial character are formed using the calculated vertices to increase the weight of the initial character.

Abstract: An input system comprises a human machine interface device comprising a scroll device adapted to feed scroll pulses to a controller residing in the human machine interface device, for scrolling through contents on a display of a host computer. The controller is adapted and programmed to communicate with the host computer and adapted and programmed to sense, in a sequence of scroll pulses, elapsed time spans between each of a plurality of the scroll pulses resulting from a user operating a scroll device being a part of a human-machine interface device which is adapted to communicate with the host computer, and a direction of the operation of the scroll device by the user.

Abstract: A stringed instrument bridge assembly for a stringed instrument having a plurality of strings of different gauges, the bridge assembly including a stringed instrument bridge; and a plurality of string-specific saddles arranged within the stringed instrument bridge. Each of the plurality of string-specific saddles includes a respective notch structured and arranged for accommodating therein a respective string of the plurality of strings. Each of the respective notches is configured in a string-specific manner to accommodate therein a string at least one of: configured to produce a particular note, having specific string gauge, and within a range of string gauges configured to produce a particular note.

Abstract: A guitar pick is of a larger size than traditional picks and shaped as a roughly equilateral triangle. The pick has holes therethrough, with 3 of the holes equidistant from each vertex of the pick. A fourth hole may extend through the pick's center.

Abstract: A compact assembly for retaining a bass drum in position on a playing surface while it is played by a drummer seated on a drum stool. The compact assembly includes a mat component having a length and a width. The length of the mat component is sufficient to allow both a bass drum and a drum stool to be operatively positioned on opposite ends thereof while the mat component is disposed in an open operative configuration. A retention component comprises a retention member secured to a portion of one end of the mat component, and a cushion member is affixed to at least one side of the retention member. The cushion member engages a portion of the bass drum when the bass drum is disposed in an operative orientation on the mat component, thereby retaining the bass drum in position while a drummer is playing.

Abstract: A vibrato assembly for a keyboard percussion instrument comprises a motor and a vibrato fan or disc assembly. A motor (510) is responsive to a controller (1000), a run/stop switch (1035), and a potentiometer (1020). Pulleys (100A, 100B?, 515) and a timing belt (520) couple rotation of the motor to an assembly of resonator or vibrato fans (720) via one or more magnets (120) or magnetically susceptible members to isolate vibrations. A sensor is activated by one of the magnets to sense the rotational position of a pulley when the fans are oriented vertically. When a run/stop switch 1035 is in a “stop” condition, sensor 525 signals controller 1000 to stop the motor, thereby leaving the fans vertical so that resonant tubes (725) are in an open condition when the vibrato function is stopped.

Abstract: An improved apparatus for displaying sheet music is an electronic tablet having the ability to upload and download sheet music to the device as well as having the ability to provide a music composer with the tools to put the music easily and conveniently in printed form. Sound recognition allows the tablet to recognize audio sequences of music and locate the place in the electronic music sheets on the tablet to keep the proper location displayed for the user-performer, including automatically turning electronic pages during the playing of a piece of music.

Abstract: A foot-operated pedal for programming audio and other effects, including an HMI or Human-Machine Interface that digitally communicates with a CPU, providing the parameters of effects set by the user, where a PC Interface is responsible for the mediation of the communication between the CPU and an external computer used to program the settings and adjustments of the device, with the audio interface having the function of promoting the compatibility between the electrical signal from the musical instrument and the CPU. A graphical user interface to facilitate the creation of an audio effect using icons, a given configuration of icons and interconnections being modeled and employed to generate a given audio effect.

Abstract: A motor vehicle includes a loudspeaker, a voice recognition module, and a microphone producing a microphone signal based upon words uttered by a human passenger within a passenger compartment of the motor vehicle. An electronic processor is communicatively coupled to the microphone, the loudspeaker, and the voice recognition module. The electronic processor receives the microphone signal and communicates with the voice recognition module to thereby ascertain the words uttered by the human passenger. The electronic processor retrieves Karaoke music corresponding to the ascertained words uttered by the human passenger, and plays the Karaoke music on the loudspeaker.

Abstract: A stringed instrument having a flex action tremolo. The assembly includes a neck of the stringed instrument that is capable of pivoting to cause the stings of the instrument to slack or tighten depending on the movement of the neck. A metal bar extends across a portion of the hollow portion of the body and is connected to the neck. There is at least one spring-loaded connection between the metal bar and the inside surface of the hollow portion of the body. This allows the metal bar and neck to pivot at one or more angles relative to the longitudinal axis of the neck, while maintaining the tuning of the guitar when the neck is returned to a neutral position.

Abstract: A device for reading waveform data of a musical tone from a primary storage device and transferring the read waveform data to a secondary storage device for tone reproduction includes a processor configured to perform: retrieving, for each waveform of a plurality of waveforms that represent a musical tone stored in the primary storage device, segment group information from the primary storage device; retrieving the plurality of waveforms that represent the musical tone from the primary storage device, the waveform group retrieval process retrieving a waveform or waveforms, among the plurality of waveforms, that have the same segment group information as a group; and writing, as a single group, the waveform or waveforms, among the plurality of waveforms, that have the same segment group information onto one of prescribed storage segments that are storage regions of prescribed sizes in the secondary storage device.

Abstract: Sound insulation constructions, multilayer constructions for acoustically insulating a source of sound from a receiver that include one or more sound insulation constructions, structures comprising one or more sound insulation constructions and/or multilayer constructions, and a method for acoustically insulating a source of sound from a receiver. In some embodiments, the sound insulation construction includes a first layer and a second layer. The first layer can include a bonded fiber nonwoven web exhibiting a work of compression of at least about 0.7 kJ/m3 and an airflow resistance of no greater than 10,000 Rayls/m. The second layer can exhibit an airflow resistance of greater than 10,000 Rayls/m. In some embodiments, the method includes coupling the first layer to a surface of a vehicle to attenuate sound in at least a portion of the vehicle.

Abstract: One embodiment provides a method, including: detecting, using one or more audio capture devices, nuisance audio; receiving, from one or more device sensors, contextual information; determining a mitigating audio signal based on the nuisance audio and contextual information; thereafter, emitting, from one or more audio source devices, mitigating audio into an interior space. Other aspects are described and claimed.

Abstract: The technology described in this document can be embodied in a computer-implemented method that includes receiving a portion of a feedback signal of an active noise control (ANC) system, and processing the portion of the feedback signal using an adaptive line enhancer (ALE) filter to detect a tonal signature. The method also includes determining, by one or more processing devices, that the tonal signature represents an unstable condition, and responsive to determining that the tonal signature represents an unstable condition, generating one or more control signals for adjusting one or more parameters of the ANC system, to mitigate the unstable condition.

Abstract: A computing device with a microphone system is disclosed. The computing device includes a microphone system with an environment microphone and a noise microphone. The environment microphone picks up an environment microphone signal which includes (1) a desired signal component based on desired sound and (2) a noise component based on noise from a noise source. The noise microphone picks up a noise microphone signal based on the noise, and is configured such that contributions to the noise microphone signal from the desired sound, if present, are attenuated relative to the environment microphone. A controller receives and processes time samples from the noise microphone signal to yield a noise estimation of the noise component. The estimation is subtracted from the environment microphone signal to yield and end-user output.

Abstract: An apparatus in a surrounding liquid and methods of using same. The apparatus includes an acoustic Fresnel zone plate comprising a compliant frame, a substrate connected to the frame, and an acoustic diffraction grating connected to the substrate. The acoustic diffraction grating includes a plurality of concentric rings or a plurality of concentric islands. The substrate is acoustic-impedance-mismatched with the acoustic diffraction grating. The acoustic Fresnel zone plate includes at least one acoustic focal spot. The substrate is acoustic-impedance-mismatched with the surrounding liquid and the acoustic diffraction grating is acoustic-impedance-matched with the surrounding liquid, or the substrate is acoustic-impedance-matched with the surrounding liquid and the acoustic diffraction grating is acoustic-impedance-mismatched with the surrounding liquid.

Abstract: An interactive system may be implemented in part by an audio device located within a user environment, which may accept speech commands from a user and may also interact with the user by means of generated speech. In order to improve performance of the interactive system, a user may use a separate device, such as a personal computer or mobile device, to access a graphical user interface that lists details of historical speech interactions. The graphical user interface may be configured to allow the user to provide feedback and/or corrections regarding the details of specific interactions.

Abstract: The disclosure describe a system and method for detecting one or more segments of desired speech utterances from an audio stream using timings of events from other modes that are correlated to the timings of the desired segments of speech. The redundant information from other modes results in a highly accurate and robust utterance detection.

Abstract: The subject matter of the disclosure is embodied in a method that includes receiving input speech data from a speaker in a first language, and estimating, based on a universal speech model, a speaker transform representing speaker characteristics associated with the input speech data. The method also includes accessing a speaker-independent speech model for generating speech data in a second language that is different from the first language. The method further includes modifying the speaker-independent speech model using the speaker transform to obtain a speaker-specific speech model, and generating speech data in the second language using the speaker-specific speech model.

Abstract: An electronic device with one or more processors and memory includes a procedure for training a digital assistant. In some embodiments, the device detects an impasse in a dialog between the digital assistant and a user including a speech input. During a learning session, the device utilizes a subsequent clarification input from the user to adjust intent inference or task execution associated with the speech input to produce a satisfactory response. In some embodiments, the device identifies a pattern of success or failure associated with an aspect previously used to complete a task and generates a hypothesis regarding a parameter used in speech recognition, intent inference or task execution as a cause for the pattern. Then, the device tests the hypothesis by altering the parameter for a subsequent completion of the task and adopts or rejects the hypothesis based on feedback information collected from the subsequent completion.

Abstract: A method for adapting a phonetic dictionary for peculiarities of a speech of an at least one speaker, comprising generating search pronunciations for a search term, retrieving audio sections from an audio database for each search pronunciation, audibly presenting to a person the audio sections of the speech of the at least one speaker, and updating the phonetic dictionary based on acceptability of the audio sections determined from judgments by the person regarding intelligibility of the audio sections in audibly pronouncing the provided at least one word, wherein the method is performed on an at least one computerized apparatus configured to perform the method.

Abstract: One or more processors receive recording data of a meeting between a professional and a client. One or more processors analyze the recording data to make one or more determinations. One or more processors identify one or more characteristics of the professional based on the one or more determinations. One or more processors match the one or more characteristics of the professional to one or more preferences of an individual seeking one or more professionals. One or more processors build a profile of the professional based on the one or more characteristics and store the profile in a database. One or more processors search the database for one or more profiles that provide a match of the one or more preferences of the individual seeking one or more professionals and display the one or more profiles.

Abstract: The technology described in this document can be embodied in a computer-implemented method that includes receiving, at a processing system, a first signal including an output of a speaker device and an additional audio signal. The method also includes determining, by the processing system, based at least in part on a model trained to identify the output of the speaker device, that the additional audio signal corresponds to an utterance of a user. The method further includes initiating a reduction in an audio output level of the speaker device based on determining that the additional audio signal corresponds to the utterance of the user.

Abstract: Techniques for identifying a location of a voice-controlled device within an environment. After identifying a location of the device, the device may receive a voice command from a user within the environment and may determine a response to the command based in part on the location, may determine how to output a response based in part on the location or may determine how to interact with the user based in part on the location.

Abstract: A method for reducing response time in a speech interface including constructing a partially completed word sequence from a partially received utterance from a speaker received by an audio sensor, modeling a remainder portion using a processor based on a rich predictive model to predict the remainder portion, and responding to the partially completed word sequence and the predicted remainder portion using a natural language vocalization generator with a vocalization, wherein the vocalization is prepared before a complete utterance is received from the speaker and conveyed to the speaker by an audio transducer.

Abstract: Methods, systems, and apparatus for receiving, by a voice action system, data specifying a new voice action for an application different from the voice action system. A voice action intent for the application is generated based at least on the data, wherein the voice action intent comprises data that, when received by the application, requests that the application perform one or more operations specified for the new voice action. The voice action intent is associated with trigger terms specified for the new voice action. The voice action system is configured to receive an indication of a user utterance obtained by a device having the application installed, and determines that a transcription of the user utterance corresponds to the trigger terms associated with the voice action intent. In response to the determination, the voice action system provides the voice action intent to the device.

Abstract: The invention relates to a customer interaction management system that comprises a memory that stores customer profile data and customer interaction data; a voice response input that receives a voice pattern from a customer; and a computer processor, coupled to the memory and the voice response input, programmed to: retrieve customer voice data from a current customer interaction via a voice channel; retrieve data from one or more other interactions via one or more other channels; compare customer voice data to a customer baseline, where the customer baseline is developed from one or more prior customer interactions; generate, using a speech analytics engine, a customer score that indicates customer sentiment based on the customer voice data and data from one or more other interactions; during the current customer interaction, update the customer score based on customer progress data; and develop one or more actions, in response to the customer score.

Abstract: Features are disclosed for generating intent-specific results in an automatic speech recognition system. The results can be generated by utilizing a decoding graph containing tags that identify portions of the graph corresponding to a given intent. The tags can also identify high-information content slots and low-information carrier phrases for a given intent. The automatic speech recognition system may utilize these tags to provide a semantic representation based on a plurality of different tokens for the content slot portions and low information for the carrier portions. A user can be presented with a user interface containing top intent results with corresponding intent-specific top content slot values.

Abstract: A method includes detecting an initiation command to begin outputting text data received from a voice recognition system to a text buffer of an aircraft. The method includes recognizing user speech as the text data and outputting the text data to the text buffer for presentation to a user. The method further includes detecting an end command to end outputting the text data received from the voice recognition system to the text buffer. The method also includes receiving a confirmation input from the user, the confirmation input indicating that the user has confirmed accuracy of the text data of the text buffer. The method additionally includes receiving a destination selection from the user, the destination selection indicating a destination selected by the user for the text data of the text buffer. The method further includes outputting the text data of the text buffer to the destination selected by the user.

Abstract: A Transcription Engine is configured to broadcast over the Internet and stream text associated with the broadcast to registered and authenticated end users who may be hearing impaired or may have difficulty understanding the language used in the broadcast. The Transcription Engine comprises an authentication server, a database server and a Transcription server. End users are authenticated automatically by the Transcription Engine prior to being given access to a live or recorded broadcast.

Abstract: Systems and methods of validating transcriptions of natural language content using crowdsourced validation jobs are provided herein. In various implementations, a transcription pair comprising natural language content and text corresponding to a transcription of the natural language content may be gathered. A group of validation devices may be selected for reviewing the transcription pair. A crowdsourced validation job may be created for the group of validation devices. The crowdsourced validation job may be provided to the group of validation devices. One or more votes representing whether or not the text accurately represents the natural language content may be received from the group of validation devices. Based on the one or more votes received, the transcription pair may be stored in a validated transcription library, which may be used to process end-user voice data.

Abstract: An incremental speech recognition system. The incremental speech recognition system incrementally decodes a spoken utterance using an additional utterance decoder only when the additional utterance decoder is likely to add significant benefit to the combined result. The available utterance decoders are ordered in a series based on accuracy, performance, diversity, and other factors. A recognition management engine coordinates decoding of the spoken utterance by the series of utterance decoders, combines the decoded utterances, and determines whether additional processing is likely to significantly improve the recognition result. If so, the recognition management engine engages the next utterance decoder and the cycle continues. If the accuracy cannot be significantly improved, the result is accepted and decoding stops.

Abstract: A computer speech output control method, system, and non-transitory computer readable medium, include a computer speech output control system, including a computer speech output unit configured to output a computer speech, a human speech monitoring circuit configured to determine whether a human conversation is occurring, an interruption priority setting circuit configured to set a priority setting for when the human conversation can be interrupted by the computer speech, and an interruption determining circuit configured to determine whether to cause the computer speech output unit to output the computer speech based on the priority setting and a status of the human conversation.

Abstract: In general, techniques are described for transitioning an ambient higher order ambisonic coefficient. A device comprising a memory and a processor may be configured to perform the techniques. The processor may obtain, from a frame of a bitstream of encoded audio data, a bit indicative of a reduced vector. The reduced vector may represent, at least in part, a spatial component of a sound field. The processor may also obtain, from the frame, a bit indicative of a transition of an ambient higher-order ambisonic coefficient. The ambient higher-order ambisonic coefficient may represent, at least in part, an ambient component of the sound field. The reduced vector may include a vector element associated with the ambient higher-order ambisonic coefficient in transition. The memory may be configured to store the frame of the bitstream.

Abstract: When compressing an HOA data frame representation, a gain control (15, 151) is applied for each channel signal before it is perceptually encoded (16). The gain values are transferred in a differential manner as side information. However, for starting decoding of such streamed compressed HOA data frame representation absolute gain values are required, which should be coded with a minimum number of bits. For determining such lowest integer number (?e) of bits the HOA data frame representation (C(k)) is rendered in spatial domain to virtual loudspeaker signals lying on a unit sphere, followed by normalization of the HOA data frame representation (C(k)). Then the lowest integer number of bits is set to (AA).

Abstract: A challenge of audio watermarking systems in which an acoustic path is involved is the robustness against microphone pickup in case of surrounding noise. The strength of phase-based watermarking is increased by determining a masking threshold for a current frequency bin in a frequency/phase representation changing the phase based on that masking threshold and an allowed phase change value, calculating an allowed magnitude change value for the current frequency bin and calculating from an audio quality level value a magnitude change scaling factor for the magnitude change value, and increasing its magnitude accordingly.

Abstract: Several embodiments include a remote tracker for a videography drone. The remote tracker can include a spatial information sensor and a microphone configured to capture audio data surrounding the remote tracker. The remote tracker can also include a logic control component configured to decorate the audio data with location-based metadata or temporal metadata. A network interface of the remote tracker can communicate with the videography drone, including streaming the audio data captured by the microphone to the videography drone.

Abstract: The present technology relates to a device, a method, and a program for expanding a frequency band, which are capable of obtaining high-quality sound with a small processing amount. A low band extraction band-pass filter processing unit passes a predetermined band of a low band of an input signal and generates a low band sub band signal. A band-pass filter calculation circuit calculates band-pass filter coefficients of band-pass filters having sub bands of high bands as a pass band based on an estimate value of high band sub band power, and an addition unit obtains one filter coefficient by adding the band-pass filter coefficients. A poly-phase configuration level adjustment filter performs up-sampling and level adjustment by performing filtering on a flattened signal obtained from a low band sub band signal using the filter coefficient obtained by the addition unit, and generates a high band signal. An addition unit obtains an output signal by adding the high band signal to the low band signal.

Abstract: A method comprising: receiving an utterance, an original pitch contour of the utterance, and a target pitch contour for the utterance, wherein the utterance comprises a plurality of consecutive frames, and wherein at least one of said frames is a voiced frame; calculating an original intensity contour of said utterance; generating a pitch modified utterance based on the target pitch contour; calculating an intensity modification factor for each of said frames, based on said original pitch contour and on said target pitch contour, to produce a sequence of intensity modification factors corresponding to said plurality of consecutive frames; calculating a final intensity contour for said utterance by applying said intensity modification factors to said original intensity contour; and generating a coherently modified speech signal by time dependent scaling of the intensity of said pitch modified utterance according to said final intensity contour.

Abstract: A method comprising: receiving an utterance, an original pitch contour of the utterance, and a target pitch contour for the utterance, wherein the utterance comprises a plurality of consecutive frames, and wherein at least one of said frames is a voiced frame; calculating an original intensity contour of said utterance; generating a pitch modified utterance based on the target pitch contour; calculating an intensity modification factor for each of said frames, based on said original pitch contour and on said target pitch contour, to produce a sequence of intensity modification factors corresponding to said plurality of consecutive frames; calculating a final intensity contour for said utterance by applying said intensity modification factors to said original intensity contour; and generating a coherently modified speech signal by time dependent scaling of the intensity of said pitch modified utterance according to said final intensity contour.

Abstract: A voice signal processing method and apparatus, which are used to process a voice signal collected by a microphone of a terminal in order to meet requirements of the terminal in different application modes for the voice signal generated after the processing. The method includes collecting at least two voice signals, determining a current application mode of a terminal, determining, according to the current application mode from the voice signals, voice signals corresponding to the current application mode, and performing, in a preset voice signal processing manner that matches the current application mode, beamforming processing on the corresponding voice signals.

Abstract: A system for and method of characterizing a target application acoustic domain analyzes one or more speech data samples from the target application acoustic domain to determine one or more target acoustic characteristics, including a CODEC type and bit-rate associated with the speech data samples. The determined target acoustic characteristics may also include other aspects of the target speech data samples such as sampling frequency, active bandwidth, noise level, reverberation level, clipping level, and speaking rate. The determined target acoustic characteristics are stored in a memory as a target acoustic data profile. The data profile may be used to select and/or modify one or more out of domain speech samples based on the one or more target acoustic characteristics.

Abstract: There are provided systems and methods for generating a visually consistent alternative audio for redubbing visual speech using a processor configured to sample a dynamic viseme sequence corresponding to a given utterance by a speaker in a video, identify a plurality of phonemes corresponding to the dynamic viseme sequence, construct a graph of the plurality of phonemes that synchronize with a sequence of lip movements of a mouth of the speaker in the dynamic viseme sequence, use the graph to generate an alternative phrase that substantially matches the sequence of lip movements of the mouth of the speaker in the video.