Abstract: A display auxiliary device includes a light receiving sensor that receives display light from a display target, and a CPU that determines whether or not a switching display synchronous signal is included in the received display light, shields both a left-eye liquid crystal shutter and a right-eye liquid crystal shutter during a light reception period of time of the switching display synchronous signal based on the switching display synchronous signal, and performs a shutter switching operation of the left-eye liquid crystal shutter for the left-eye image and the right-eye liquid crystal shutter for the right-eye image to be switched and displayed, when the switching display synchronous signal is determined to be included.

Abstract: A source driver apparatus and an operating method thereof are provided. The source driver apparatus can drive a plurality of source lines of a display panel, wherein the display panel further comprising a gate driver apparatus. The source driver apparatus includes driving channels and a delay control circuit. The driving channels output source driving signals. The delay control circuit controls the driving channels to change delay times of the source driving signals within the same period, such that the delay times of the source driving signals respectively correspond to distances from the source lines to the gate driver apparatus.

Abstract: Provided is a liquid crystal display device that includes: pixels, a pixel control unit, and a common voltage generation unit. The pixel includes: a display element; a first switching unit configured to sample each frame data; a first holding unit configured to form an SRAM, and to hold sub-frame data sampled; a second switching unit configured to output the sub-frame data; and a second holding unit configured to form a DRAM, and configured of which stored content is rewritten by the sub-frame data. The pixel control unit is configured to repeat writing the sub-frame data into the first holding units, and to operate to rewrite the stored content in the second holding units. The common voltage generation unit is configured to change a voltage value of a common voltage from a first voltage value to a second voltage value determined based on at least the one frame period.

Abstract: A driving circuit comprising a signal edge cutting circuit is described. The signal edge cutting circuit comprises a first switch unit, a second switch unit and a third switch unit wherein the third switch unit decreases a voltage amplitude of the scanning signal by an edge-cutting resistor for implementing the signal edge cutting procedure of the scanning signal. The present invention further provides an LCD apparatus and employs the third switch unit for eliminating the image sticking phenomenon of the display image advantageously.

Abstract: A GOA (Gate On Array) circuit applied to a liquid crystal display device is disclosed. The liquid crystal display device has a plurality of scan lines. The GOA circuit has a plurality of cascaded shift register units. An (N)th level shift register unit controls charge to an (N)th level scanning line. The (N)th level shift register unit includes a forward-rearward scan circuit, a pull-up circuit, an bootstrap capacitor circuit, a gate signal point electricity leakage preventing circuit, and a pull-down sustain circuit. The bootstrap capacitor circuit, the gate signal point electricity leakage preventing circuit, and the pull-down sustain circuit are connected together with each other to form a gate signal point, so as to raise the stability of the gate signal point and to decrease the use of switches.

Abstract: A liquid crystal panel driving apparatus that produces a normal display without damaging the liquid crystal panel even when each of a plurality of timing controllers receives an abnormal display data signal. In an exemplary apparatus, each of the timing controllers transmits an abnormality detection signal, which is generated by detecting an abnormality in the display data signal, to other timing controllers via an abnormality detection line, and each of the timing controllers outputs image data associated with the abnormality detection signal stored in an image data memory.

Abstract: A display device includes: a first display panel including: a display area including a first edge, a second edge and a third edge, where the first and second edges are disposed opposite to each other, and the third edge is connected to the first and second edges, a peripheral area around the display area, and a plurality of pixels disposed in the display area; a first common voltage transmitting line extending along the third edge, where the first common voltage transmitting line transmits a first common voltage to the display area through a plurality of input points sequentially disposed along the third edge; and a second common voltage transmitting line extending along the third edge, where the second common voltage transmitting line transmits a second common voltage to the display area through a supplementary input point, which is adjacent to the second edge or the third edge.

Abstract: A display device is capable of preventing damage to a driver integrated circuit (IC) when a malfunction occurs therein, the display device including a display panel including a gate line, a data line, a first dummy line, and a second dummy line; a first data driver integrated circuit connected to one side of the data line; a second data driver integrated circuit connected to another side of the data line; a first power supply configured to apply a part of first enable signals to the first data driver integrated circuit, and to apply a part of second enable signals to the second data driver integrated circuit through the second dummy line; and a second power supply configured to apply the rest of the second enable signals to the second data driver integrated circuit, and to apply the rest of the first enable signals to the first data driver integrated circuit through the first dummy line.

Abstract: According to an embodiment of the present invention, an image processing apparatus includes a processor, and a computer-readable medium in which a computer program product is stored. The computer program product causes the processor to perform operations to acquire an image, set an area in the image, change the area set in the image, acquire information on an image near at least one of the area before the change and the area after the change, correct an image in at least one of the area before the change and the area after the change based on the acquired information, and control a display unit to sequentially display the images in the area before the change and the area after the change based on the result of the correction.

Abstract: A shift register unit and a driving method, a gate drive circuit and a display device, the shift register unit comprises a first transistor (T1), an input module (10), a first control module (20), a second control module (30) and an output module (40). The shift register unit and driving method, gate drive circuit and display device are capable of solving the problem that the decrease of drive capacity of the shift register unit is caused by loss of the threshold voltage.

Abstract: The present invention has a technical objective of providing a distinctive display format for machine-readable dot patterns which does not harm the aesthetics of images displayed on a display device. The dot pattern is made machine readable by providing the periphery of dots with a contrast control region and/or an anti-aliasing region which are distinguishable from the surrounding image. Thus enabled is the embedding of the two-dimensional codes which cannot be recognized by audiences yet machine readable on video or images on the screen of a display device. An audience can easily acquire information defined by the dot patterns using a digital camera, a video camera, a portable telephone camera, a web camera, or the like to image the screen of the display device or to capture the screen as data.

Abstract: Embodiment relate to a display panel that modulates 3D image data based on pixel data of adjacent lines. The display panel includes data lines, gate lines, and a plurality of pixels; a data modulation unit. The modulation unit modulates kth pixel data of a jth line in 3D image data based on the kth pixel data of the jth line and kth pixel data of a line adjacent to the jth line. The data driving circuit converts the modulated 3D image data into analog data voltages and outputs the analog data voltages to the data lines. A gate driving circuit sequentially outputs gate pulses to the gate lines.

Abstract: The technology provides a dimming module for a near-eye display (NED) device that controls an amount of ambient light that passes through the transmissive near-eye display to a user. The dimming module includes at least one electrochromic cell that enables variable density dimming so that the NED device may be used in an augmented reality (AR) and/or virtual reality (VR) application. The electrochromic cell may be a monochrome electrochromic cell having stacked layers of, a monochrome electrochromic compound layer and insulator sandwiched between a pair of transparent substrates and conductors. A current may be applied to the conductor layers to control the amount of dimming in response to a dimming value. A NED device having a dimming module may be included in a visor, or other type of head-mounted display (HMD). The dimming module may be flat and supported by a flat waveguide mount in the user's field of view.

Abstract: The invention provides a driving method and a driving system for a display panel. The driving method comprises: partitioning the display panel into N display sub-areas, each of which is used for displaying the 1/N sub-picture of the whole frame of picture to be displayed and correspondingly controlled by one timing controller, wherein N?2 and N is an integer, and the data signal of the whole frame of picture to be displayed is allocated to each of the timing controllers. According to the driving method and the driving system, by allocating the data signal of the whole frame of picture to be displayed to each of timing controllers, the backlight brightness provided by the different backlight sub-areas is the same, and then a joint or boundary phenomenon occurring among the different backlight sub-areas is avoided.

Abstract: A display unit has a flexible display section, a detection section for detecting a deflection amount and a deflection direction of the display section, a determination section for determining a visible portion and a non-visible portion of the display section based on the deflection amount and the deflection direction, and a control section for controlling display contents of the display section. The control section either prevents an image display on the non-visible portion or displays one of a fixed image or a pre-set moving image on the non-visible portion. The display unit also includes a displacement sensor located in the same region as the display section.

Abstract: Image information displayed on an electronic device can be adjusted based at least in part upon a relative position of a viewer with respect to a device. In some embodiments, image stabilization can be provided such that an image remains substantially consistent from the point of view of the viewer, not the display element of the device. The image can be stretched, rotated, compressed, or otherwise manipulated based at least in part upon the relative viewing position. Similarly, the viewer can move relative to the device to obtain different views, but views that are consistent with the viewer looking at an object, for example, through a piece of glass. The device can overlay information on the image that will adjust with the adjusted image. Three-dimensional modeling and display can be used to offset parallax and focus point effects.

Abstract: A data processing device according to embodiments comprises a data converting unit, a selecting unit, a managing unit, a updating unit, and a controller. The data converting unit is configured to convert update-data for updating at least a part of an electronic paper into processed update-data to be displayed. The selecting unit is configured to select an update-control-information identifier to be used for updating the electronic paper with the processed update-data. The managing unit is configured to store the processed update-data and a selected update-control-information identifier on a first memory. The updating unit is configured to instruct a drawing step of the electronic paper using the processed update-data and the update-control-information identifier stored on the first memory.

Abstract: A sound resonator device for internal connection to a guitar or other musical instrument that has strings tensioned over a bridge piece. One embodiment of resonator includes a plurality of wound coils made of a harmonically predetermined (consistent or varying) thickness or gauge of metal wire secured to the instrument interior. A series of dampering devices are added for muting the effect of such sound resonation when desired. The dampering devices are activated using either a slider button (for felt dampers) or a rotating knob (for fiber threads).

Abstract: An apparatus comprises a flute, a base, and a thumb rest. The base is attached to the flute and defines a channel. The thumb rest is inserted into the channel and defines an elongate body. The thumb rest is positioned such that a flutist may rest a right thumb against the thumb rest while using conventional fingering to play the flute.

Abstract: One or more drumsticks may be optimally balanced using a grip point of the drumstick by calculating a radius of gyration. The radius of gyration is a controlling parameter wherein the radius of gyration correlates with the grip point, which is itself based on a particular gripping style and hand size. Given the radius of gyration, the center of gravity of the one or more drumsticks may be re-located at a point along the drumstick closely aligned to the radius of gyration. The re-location may be performed by adding a weight to the butt end of the drumstick. Balanced drumstick pairs provide optimum performance, control and handling characteristics for a drummer.

Abstract: A musical instrument pedal with a drumstick angle adjustment structure comprises a drumstick, a rotation shaft, a stand frame, a pedal base, a pedal, and a chain. The drumstick is fixed to the rotation shaft. The stand frame includes a vertical rod pivotally coupled to the rotation shaft. The pedal base includes a fixed member connected with the stand frame to secure the stand frame, a mobile member slidable with respect to the fixed member, and a screw fastening the fixed member to the mobile member. Two ends of the chain are respectively connected with the pedal and the rotation shaft. Thereby, the relative position of the mobile member and the fixed member can be varied to drag the chain to rotate the rotation shaft, change the drumstick angle, vary the force strength of beating the drum skin, and generate different drum sounds for different melodies.

Abstract: The concept of the invention is to take the said top snare assembly of the said double snare system, and to configure the snare system inside of a drum head. By using the said invention, modifications to the said snare drum will not be required and will not compromise the integrity of the original design of the said drum manufacturers' marching snare drum.

Abstract: Systems, including methods and apparatus, for generating audio effects based on accompaniment audio produced by live or pre-recorded accompaniment instruments, in combination with melody audio produced by a singer. Audible broadcast of the accompaniment audio may be delayed by a predetermined time, such as the time required to determine chord information contained in the accompaniment signal. As a result, audio effects that require the chord information may be substantially synchronized with the audible broadcast of the accompaniment audio. The present teachings may be especially suitable for use in karaoke systems, to correct and add sound effects to a singer's voice that sings along with a pre-recorded accompaniment track.

Abstract: Techniques are described that relate to various aspects of converting the mechanical energy of instrument strings to digital representations for use in a variety of applications.

Abstract: Embodiments disclosed herein describe systems and methods for asymmetrical bobbin configurations. Embodiments of asymmetrical bobbin configurations are configured to allow manufacturers and users to modify, adjust, change, etc. the tone of pickups. The asymmetrical bobbin configurations are configured to boost frequencies on strings on an instrument, wherein based on the configurations of the bobbins different frequencies may be modified.

Abstract: A system for modifying an audio signal comprises a computer including a first digital signal processor (DSP). The first DSP includes a first module and a second module. The first module is coupled to the interface to provide a musical instrument with a set of testing input signals and obtain a set of testing output signals from the musical instrument via the interface. The second module is configured to perform functions including: analyzing the set of testing output signals to obtain a set of parameters, constructing an acoustic transducer to model characteristics of the musical instrument based on the set of parameters, and receiving the audio signal to modify the audio signal using the acoustic transducer, wherein the first DSP module obtains the set of parameters at least by identifying occurrences when an output level state determined at least based on the set of testing output signals significantly changes.

Abstract: A microphone or instrument sound pickup for various musical instruments and singers is coupled to an amplifier, the output of which is coupled to a speaker system. A microphone is positioned within the output range of the speaker system and is coupled to one input of a comparator circuit. The remaining input of the comparator circuit is coupled to the microphones or pickups. An additive device receives the output signal of the comparator and the originating signal from the microphones or pickups to form a correcting audio signal which is applied to the amplifier. In an alternate embodiment, an adaptive preamplifier is interposed between the originating microphones and audio pickups and the amplifier. The adaptive preamplifier receives the output of the comparator and imposes alteration upon the signal processed through the preamplifier for application to the amplifier.

Abstract: There is provided a mounting system for a fish finding device to a watercraft which has a support for supporting a fish finding device on a watercraft. The support has a top, a bottom and a peripheral side wall which form a receptacle housing for a battery. Mounting means are positioned at the top of the support to receive a fish finder. Securing means are provided on the bottom of the support for mounting the support to a surface on a watercraft. Mounting means are provided for detachably securing a transducer to a watercraft.

Abstract: The present invention aims to provide a composition for vibration damping materials which provides excellent vibration damping property. The present invention relates to an emulsion composition for vibration damping materials including: a vibration damping modifier including a compound that has 7 or more carbon atoms, a boiling point of 190° C. or higher, and at least two ether groups or at least two ester groups in the molecule; and a polymer emulsion.

Abstract: A vehicle audio system includes: a tune knob configured to select a frequency of a desired radio broadcast; a microcomputer configured to detect a broadcast frequency input through the tune knob and to determine an optimal sampling frequency of a full digital amplifier based on the detected broadcast frequency; a tuner configured to receive only a broadcast signal corresponding to the frequency selected through the tune knob; and the full digital amplifier configured to generate a digital signal by sampling the broadcast signal received by the tuner at the optimal sampling frequency determined by the microcomputer and to amplify and output the generated digital signal.

Abstract: A control signal filter 2 receives a sound source signal determined by a control frequency specified in conformity with the vibration/noise source that produces vibration/noise, and outputs a control signal. A filter coefficient update unit 4 updates coefficients of the control signal filter 2 in response to a sound source signal and an error signal. A signal-to-interference ratio measuring unit 5 outputs a signal-to-interference ratio determined from the vibration/noise and the interference contained in the error signal in response to the control frequency and error signal. An update controller 6 adjusts an update step of the filter coefficient update unit 4 in accordance with the signal-to-interference ratio.

Abstract: Techniques for improved text-to-speech processing are disclosed. The improved text-to-speech processing can convert text from an electronic document into an audio output that includes speech associated with the text as well as audio contextual cues. One aspect provides audio contextual cues to the listener when outputting speech (spoken text) pertaining to a document. The audio contextual cues can be based on an analysis of a document prior to a text-to-speech conversion. Another aspect can produce an audio summary for a file. The audio summary for a document can thereafter be presented to a user so that the user can hear a summary of the document without having to process the document to produce its spoken text via text-to-speech conversion.

Abstract: A method and a device that preprocess a speech signal are disclosed, which include extracting at least one frame corresponding to a speech recognition range from frames included in a speech signal, generating a supplementary frame to supplement speech recognition with respect to the speech recognition range based on the at least one extracted frame, and outputting a preprocessed speech signal including the supplementary frame along with the frames of the speech signal.

Abstract: A method for speech retrieval includes acquiring a keyword designated by a character string, and a phoneme string or a syllable string, detecting one or more coinciding segments by comparing a character string that is a recognition result of word speech recognition with words as recognition units performed for speech data to be retrieved and the character string of the keyword, calculating an evaluation value of each of the one or more segments by using the phoneme string or the syllable string of the keyword to evaluate a phoneme string or a syllable string that is recognized in each of the detected one or more segments and that is a recognition result of phoneme speech recognition with phonemes or syllables as recognition units performed for the speech data, and outputting a segment in which the calculated evaluation value exceeds a predetermined threshold.

Abstract: A method for speech retrieval includes acquiring a keyword designated by a character string, and a phoneme string or a syllable string, detecting one or more coinciding segments by comparing a character string that is a recognition result of word speech recognition with words as recognition units performed for speech data to be retrieved and the character string of the keyword, calculating an evaluation value of each of the one or more segments by using the phoneme string or the syllable string of the keyword to evaluate a phoneme string or a syllable string that is recognized in each of the detected one or more segments and that is a recognition result of phoneme speech recognition with phonemes or syllables as recognition units performed for the speech data, and outputting a segment in which the calculated evaluation value exceeds a predetermined threshold.

Abstract: A system and method are provided for receiving speech and/or non-speech communications of natural language questions and/or commands and executing the questions and/or commands. The invention provides a conversational human-machine interface that includes a conversational speech analyzer, a general cognitive model, an environmental model, and a personalized cognitive model to determine context, domain knowledge, and invoke prior information to interpret a spoken utterance or a received non-spoken message. The system and method creates, stores, and uses extensive personal profile information for each user, thereby improving the reliability of determining the context of the speech or non-speech communication and presenting the expected results for a particular question or command.

Abstract: Techniques for generating language models. The techniques include: obtaining language data comprising training data and associated values for one or more metadata attributes, the language data comprising a plurality of instances of language data, an instance of language data comprising an instance of training data and one or more metadata attribute values associated with the instance of training data; identifying, by processing the language data using at least one processor, a set of one or more of the metadata attributes to use for clustering the instances of training data into a plurality of clusters; clustering the training data instances based on their respective values for the identified set of metadata attributes into the plurality of clusters; and generating a language model for each of the plurality of clusters.

Abstract: Systems and methods are described for personifying communications. According to at least one embodiment, the computer-implemented method for personifying a natural-language communication includes observing a linguistic pattern of a user. The method may also include analyzing the linguistic pattern of the user and adapting the natural-language communication based at least in part on the analyzed linguistic pattern of the user. In some embodiments, observing the linguistic pattern of the user may include receiving data indicative of the linguistic pattern of the user. The data may be one of verbal data or written data. Written data may include at least one of a text message, email, social media post, or computer-readable note. Verbal data may include at least one of a recorded telephone conversation, voice command, or voice message.

Abstract: An apparatus and method for recognizing a speech, and an apparatus and method for generating a noise-speech recognition model are provided. The speech recognition apparatus includes a location determiner configured to determine a location of the apparatus, a noise model generator configured to generate a noise model corresponding to the location by collecting noise data related to the location, and a noise model transmitter configured to transmit the noise model to a server.

Abstract: A system and method are provided for improving speech recognition accuracy. Contextual information about user speech may be received, and then speech recognition analysis can be performed on the user speech using the contextual information. This allows the system and method to improve accuracy when performing tasks like searching and navigating using speech recognition.

Abstract: A voice recognition terminal is provided to be able to communicate with a server capable of voice recognition for recognizing voice, and includes a voice input acceptance portion accepting voice input from a user, a voice recognition portion carrying out voice recognition of the voice input accepted, a response processing execution portion performing processing for responding to the user based on a result of voice recognition of the voice input accepted, and a communication portion transmitting the voice input accepted by the voice input acceptance portion to the server and receiving a result of voice recognition in the server. The response processing execution portion performs the processing for responding to the user based on the result of voice recognition determined as more suitable, of the result of voice recognition by the voice recognition portion and the result of voice recognition received from the server.

Abstract: Efficient empirical determination, computation, and use of an acoustic confusability measure comprises: (1) an empirically derived acoustic confusability measure, comprising a means for determining the acoustic confusability between any two textual phrases in a given language, where the measure of acoustic confusability is empirically derived from examples of the application of a specific speech recognition technology, where the procedure does not require access to the internal computational models of the speech recognition technology, and does not depend upon any particular internal structure or modeling technique, and where the procedure is based upon iterative improvement from an initial estimate; (2) techniques for efficient computation of empirically derived acoustic confusability measure, comprising means for efficient application of an acoustic confusability score, allowing practical application to very large-scale problems; and (3) a method for using acoustic confusability measures to make principled

Abstract: Embodiments are disclosed for an in-vehicle computing system comprising a communication interface communicatively coupled to a query response server, a processor, and a storage device. The storage device configured to collect and store vehicle and user interaction data and configured to store instructions executable by the processor to receive a voice query from a user, translate the voice query into a text query, send the text query to a query response server, receive a result from the query response server, the result including one or more placeholders representing data to be retrieved from a storage device local to the head unit, replace the one or more placeholders with information from the storage device to form an updated result; and present the updated result to the user.

Abstract: The present disclosure discloses an information processing method and an electronic device. The electronic device comprises a first processing unit and a second processing unit. The second processing unit is capable of executing at least one application program. The information processing method comprises: collecting first sound information; when the first sound information comprises first information which matches with audio data preset in the first processing unit, generating a first instruction, and transmitting the first instruction to the second processing unit; when it is determined that there is a first application which meets a predetermined condition in the at least one application program, generating, by the second processing unit, a second instruction based on the first instruction and the first application; and executing the second instruction in the first application.

Abstract: A method of operating a speech processing system is provided. The method includes translating a portion of a speech record into a plurality of possible words associated with a plurality of contexts, and determining a plurality of correctness values based on a plurality of probabilities that each of the plurality of possible words is correct for each of the plurality of contexts. The method also includes determining which of the plurality of possible words is a correct translation of the portion of the speech record based on the plurality of correctness values.

Abstract: A method is described for improving the accuracy of a transcription generated by an automatic speech recognition (ASR) engine. A personal vocabulary is maintained that includes replacement words. The replacement words in the personal vocabulary are obtained from personal data associated with a user. A transcription is received of an audio recording. The transcription is generated by an ASR engine using an ASR vocabulary and includes a transcribed word that represents a spoken word in the audio recording. Data is received that is associated with the transcribed word. A replacement word from the personal vocabulary is identified, which is used to re-score the transcription and replace the transcribed word.

Abstract: Embodiments of the present invention relate to speaker identification using spatial information. A method of speaker identification for audio content being of a format based on multiple channels is disclosed. The method comprises extracting, from a first audio clip in the format, a plurality of spatial acoustic features across the multiple channels and location information, the first audio clip containing voices from a speaker, and constructing a first model for the speaker based on the spatial acoustic features and the location information, the first model indicating a characteristic of the voices from the speaker. The method further comprises identifying whether the audio content contains voices from the speaker based on the first model. Corresponding system and computer program product are also disclosed.

Abstract: Method for text-dependent Speaker Recognition using a speaker adapted Universal Background Model, wherein the speaker adapted Universal Background Model is a speaker adapted Hidden Markov Model comprising channel correction.

Abstract: A method and device for decoding a signal. The method for decoding a signal includes: obtaining spectral coefficients of sub-bands from a received bitstream by means of decoding; classifying sub-bands in which the spectral coefficients are located into a sub-band with saturated bit allocation and a sub-band with unsaturated bit allocation; performing noise filling on a spectral coefficient that has not been obtained by means of decoding and is in the sub-band with unsaturated bit allocation, so as to restore the spectral coefficient that has not been obtained by means of decoding; and obtaining a frequency domain signal according to the spectral coefficients obtained by means of decoding and the restored spectral coefficient. Therefore, a sub-band with unsaturated bit allocation in a frequency domain signal may be obtained by classification, thereby improving signal decoding quality.

Abstract: The invention provides a highly efficient technique for encoding a multi-channel audio signal. The invention relies on the basic principle of encoding a first signal representation of one or more of the multiple channels in a first encoder and encoding a second signal representation of one or more of the multiple channels in a second, multi-stage, encoder. This procedure is significantly enhanced by providing a controller for adaptively allocating a number of encoding bits among the different encoding stages of the second, multi-stage, encoder in dependence on multi-channel audio signal characteristics.