Abstract: A bistable display system includes a plurality of pixels arranged in pixel rows and pixel columns. Each pixel has a bistable material between first and second transparent and conductive substrates. A bistable display method includes driving the plurality of pixels having at least one target pixel and at least one non-target pixel and applying a voltage difference across at least one target column and at least one target row to switch the at least one target pixel between transparent and opaque states.

Abstract: An information processing method and an electronic device, the method comprising: acquiring a first value of a first parameter by the first sensor; acquiring a second value of the first parameter by the second sensor; judging whether a difference value between the first value and the second value is larger than or equal to a first threshold or not; determining a third value based on a first algorithm, the first value and the second value, and reporting the third value as a value of the first parameter to the processing unit, when the difference value is larger than or equal to the first threshold; determining a fourth value based on a second algorithm, the first value and the second value, and reporting the fourth value as the value of the first parameter to the processing unit, when the difference value is smaller than the first threshold.

Abstract: In a digital signage device, a control unit extracts a face region and a torso region from an image of the object to be displayed, compares the extracted face region and torso region to a face region and torso region extracted from a screen image on an image display unit in use, and then calculates, on the basis of these comparison results, a first adjustment factor for adjusting the size of the face region and a second adjustment factor for adjusting the size of the torso region in the image of the object to be displayed, the respective sizes of the face region and torso region in the image of the object to be displayed being separately adjusted on the basis of the calculated first adjustment factor and second adjustment factor in order to display an adjusted image on the image display unit.

Abstract: Embodiments of the present invention provide client-side scene movement using imagery generated by a game server. Embodiments of the present invention predictively render additional imagery surrounding the present field of view. The predictive scene imagery may be on all sides of the current field of view. Embodiments of the present invention determine the amount of predictive scene imagery generated according to a likelihood of use. In addition to client-adjusted rotation, embodiments of the present invention may predictively translate the field of view. Translation is moving the point of view forward, backward or side-to-side. Predictive translation imagery may be communicated to the game server for use in local translation functions.

Abstract: An image processing circuit and a method thereof. The image processing circuit includes a light sensor, an image control unit, a Gamma curve LUT, a backlight control unit and an image processing unit. The light sensor senses an intensity of ambient light and outputs an intensity signal. The image control unit receives an image signal and determines an image luminance value corresponding to a first image data of the image signal according to the intensity signal. The Gamma curve LUT outputs a luminance value signal corresponding to the image luminance value. The backlight control unit outputs a backlight luminance signal corresponding to the image luminance value. The image processing unit outputs a second image data corresponding to the first image data according to the luminance value signal and the backlight luminance signal.

Abstract: Techniques are presented for adjusting a brightness of a front light that illuminates a display of an electronic device. Typically, page updates on certain types of displays, such as electronic paper displays, result in a relatively large but rapid change in brightness. Therefore, by adjusting the brightness of the front light during the update (e.g., when some, a majority, or all of the pixels are in the white state) and then adjusting the brightness back to another state upon or near completion of the update, the overall brightness perceived by the user remains more uniform. Increasing the uniformity of this perceived brightness in turn decreases the jarring effect of an update and increases the experience of the user.

Abstract: A programmable level shifter for providing upshifted control signals to an active matrix display based on logic-level control signals received from a timing controller. The programmable level shifter includes a programmable state machine, level-shifting output drivers, and a programming interface. The programmable state machine is configured to receive at least one control signal from a timing controller. The state machine generates, based on said at least one control signal, a plurality of outputs for driving gate drivers of the active matrix display. The level-shifting output drivers convert the plurality of outputs generated by the programmable state machine to a higher-magnitude voltage level. The programming interface facilitates the programming of aspects of the programmable state machine.

Abstract: A display panel includes a timing controller embedded data driver and a first data driver. The timing controller embedded data driver includes an image processing part and an internal data driving part. The image processing part generates a first data signal corresponding to a first display area and a second data signal corresponding to a second display area based on input image data. The internal data driving part generates a second data voltage based on the second data signal to output the second data voltage to the second display area. The first data driver is disposed at a first side of the timing controller embedded data driver. The first data driver receives the first data signal from the timing controller embedded data driver and generates a first data voltage based on the first data signal to output the first data voltage to the first display area.

Abstract: A method for adjusting characters of a page includes determines a selected page on a display device of an electronic device. Sizes of characters on the selected page are acquired. An adjustment ratio of the characters on the selected page is computed according to a predetermined calculation method. A size of each of the characters on the selected page is adjusted according to the acquired sizes of the characters on the selected page and the computed adjustment ratio of the characters on the selected page. Each of the characters on the selected page on the displaying device is displayed with the adjusted size of each of the characters on the selected page.

Abstract: A method for rendering video frames by a computing device having a software stack with an application layer and a kernel layer comprises various steps. First, a system reference time is initialized. A triggering of an interrupt signal in the kernel layer is waited for. Next, it is determined whether to update the system reference time as a function of a render function from the application layer. A next video frame in the kernel layer is rendered by the computing device as a function of the determined system reference time and the next video frame. The steps after the initializing step and starting at the waiting step are recursively performed.

Abstract: An electric stringed instrument for enhancing the musical or artistic performance of a user is provided. The instrument includes a body with a cutout such that a center portion of the body is open, a neck affixed to an outer edge of the body, and a plurality of strings having first ends operably connected to the neck and second ends operably connected to the body, the plurality of strings being oriented such that the strings extend over the cutout of the body. The user may maneuver the instrument and strum the plurality of strings on any portion of the strings located within the cutout of the body, thereby enhancing the sound of the instrument and the performance of the user.

Abstract: A string clamping and tuning mechanism for stringed instruments is disclosed, where the string tension actuates one or more levers to generate two points of clamping force along the string. In a first embodiment, a single lever is used to generate two points of clamping force along the string. In a second embodiment, the clamping force is less concentrated at a first point of clamping force than a second point of clamping force to optimize the holding power of the mechanism. In a third embodiment, a second lever with a screw adjusted stop causes an increase or decrease in the tension in the string.

Abstract: A guitar support stand includes a mounting plate that is attachable to a stable support such as a musical piece of equipment having a substantial weight. A guitar support member is releasably attached to the mounting plate and includes at its upper end a support bracket for supporting the neck portion of a guitar.

Abstract: An instrument cushion and support device includes a main body having a curved top surface, a curved bottom surface, a proximal end and a distal end. The main body includes a rigid central core and soft padding located between the curved top and bottom surfaces. A slip guard extends upward from the curved top surface of the main body at a location adjacent to the distal end, and a neck support extends upward from the top surface of the main body at a location adjacent to the proximal end.

Abstract: Specialized functional components are formed from a single continuous one-piece pliable block. A raised main body is between laterally spaced mounting wings. Each mounting wing terminates laterally in an insertion wedge end. The exterior surface of the main body is formed with a plurality of stick slots. Each stick slot is in a curved configuration with an upwardly facing opening. The exterior surface of each mounting wing is formed with a tapered section adjacent to each insertion wedge end. The exterior surface of each mounting wing is formed with positioning lock ridges between the tapered sections and the main body.

Abstract: An electronic musical instrument is provided with a voice sensor for detecting a voice uttered by a user, when the user blows into the musical instrument with a voice, a breath sensor for detecting at least one of a blow pressure and a blow volume in a body of the musical instrument, when the user blows into the musical instrument with a voice, and a musical tone controlling unit for controlling generation of a musical tone based on at least one of outputs of the voice sensor and the breath sensor.

Abstract: Disclosed are a method and apparatus for producing sounds in a virtual world, as well as a sound card. The method comprises the steps of determining a spring mass model of an object in the virtual world based on a 3D model of the object; analyzing force components produced on the 3D model by a collision in the virtual world; and generating sounds produced by the collision according to the spring mass model and the force components. By considering sound material of the object in the virtual world, the method, apparatus, and sound card may produce sounds in the virtual world more vividly and in real time.

Abstract: The ultrasonic-transducer mounting structure includes: a housing including: a body part which is formed into a hollow cylindrical shape and is provided at its front surface with a transmission surface allowing an ultrasonic wave to pass therethrough and is designed to accommodate an ultrasonic transducer therein; and plural connection pieces provided to a side surface of the body part; and a holder fixed to a rear surface of a bumper and designed to hold the housing such that the transmission surface of the housing is exposed via an opening of the bumper. The holder includes plural reception pieces to which the plural connection pieces are detachably coupled respectively. The plural reception pieces are fixed to the rear surface of the bumper to surround the opening.

Abstract: Systems and methods of limiting peak audio power in mobile devices may include a high pass filter and a burst module to detect a burst load condition in a mobile device. The burst module can also apply the high pass filter to an audio signal of the mobile device in response to the burst load condition to obtain a filtered signal, and transmit the filtered audio signal to a speaker of the mobile device.

Abstract: This sound insulating structure (1) is provided with: a panel (2); a sound insulating material (3) laminated to the panel (2); a reinforcing material (4) provided between the panel (2) and the sound insulating material (3) and bonded to the panel (2); and an air layer (S) formed between the panel (2) and the reinforcing material (4). The reinforcing material (4) has a plurality of through-holes (5).

Abstract: A voice converting apparatus and a voice converting method are provided. The method of converting a voice using a voice converting apparatus including receiving a voice from a counterpart, analyzing the voice and determining whether the voice abnormal, converting the voice into a normal voice by adjusting a harmonic signal of the voice in response to determining that the voice is abnormal, and transmitting the normal voice.

Abstract: A method of and system for speech synthesis. First and second text inputs are received in a text-to-speech system, and processed into respective first and second speech outputs corresponding to stored speech respectively from first and second speakers using a processor of the system. The second speech output of the second speaker is adapted to sound like the first speech output of the first speaker.

Abstract: Disclosed herein are systems, computer-implemented methods, and computer-readable storage media for unit selection synthesis. The method causes a computing device to add a supplemental phoneset to a speech synthesizer front end having an existing phoneset, modify a unit preselection process based on the supplemental phoneset, preselect units from the supplemental phoneset and the existing phoneset based on the modified unit preselection process, and generate speech based on the preselected units. The supplemental phoneset can be a variation of the existing phoneset, can include a word boundary feature, can include a cluster feature where initial consonant clusters and some word boundaries are marked with diacritics, can include a function word feature which marks units as originating from a function word or a content word, and/or can include a pre-vocalic or post-vocalic feature. The speech synthesizer front end can incorporates the supplemental phoneset as an extra feature.

Abstract: A method for adapting a language model for a context of a domain, comprising obtaining textual contents from a large source by a request directed to the context of the domain, discarding at least a part of the textual contents that contain textual terms determined as irrelevant to the context of the domain, thereby retaining, as retained data, at least a part of the textual contents that contain textual terms determined as relevant to the context of the domain, and adapting the language model by incorporating therein at least a part of the textual terms of the retained data, wherein the method is performed on an at least one computerized apparatus configured to perform the method and equipped for communication with the large source, and an apparatus for performing the same.

Abstract: A system and method are provided for adding user characterization information to a user profile by analyzing user's speech. User properties such as age, gender, accent, and English proficiency may be inferred by extracting and deriving features from user speech, without the user having to configure such information manually. A feature extraction module that receives audio signals as input extracts acoustic, phonetic, textual, linguistic, and semantic features. The module may be a system component independent of any particular vertical application or may be embedded in an application that accepts voice input and performs natural language understanding. A profile generation module receives the features extracted by the feature extraction module and uses classifiers to determine user property values based on the extracted and derived features and store these values in a user profile. The resulting profile variables may be globally available to other applications.

Abstract: A wearable computing device receives user preferences, wherein the user preferences include information detailing one or more types of information to display based on one or more criteria being satisfied. The wearable computing device determines information to display on the wearable computing device based on the user preferences. The wearable computing device displays the determined information.

Abstract: Methods and systems are provided for adapting a speech system. In one example a method includes: logging speech data from the speech system; detecting a user characteristic from the speech data; and selectively updating a language model based on the user characteristic.

Abstract: In some embodiments, a recognition result produced by a speech processing system based on an analysis of a speech input is evaluated for indications of potential errors. In some embodiments, sets of words/phrases that may be acoustically similar or otherwise confusable, the misrecognition of which can be significant in the domain, may be used together with a language model to evaluate a recognition result to determine whether the recognition result includes such an indication. In some embodiments, a word/phrase of a set that appears in the result is iteratively replaced with each of the other words/phrases of the set. The result of the replacement may be evaluated using a language model to determine a likelihood of the newly-created string of words appearing in a language and/or domain. The likelihood may then be evaluated to determine whether the result of the replacement is sufficiently likely for an alert to be triggered.

Abstract: The present invention relates to a speech recognition method and system based on user personalized information. The method comprises the following steps: receiving a speech signal; decoding the speech signal according to a basic static decoding network to obtain a decoding path on each active node in the basic static decoding network, wherein the basic static decoding network is a decoding network associated with a basic name language model; if a decoding path enters a name node in the basic static decoding network, network extending is carried out on the name node according to an affiliated static decoding network of a user, wherein the affiliated static decoding network is a decoding network associated with a name language model of a particular user; and returning a recognition result after the decoding is completed. The recognition accuracy rate of user personalized information in continuous speech recognition may be raised by using the present invention.

Abstract: Methods, systems and devices are provided for controlling a speech recognition process on a computing device. A computing device may receive audio signals from a microphone and muscle movement signals from a muscle movement detector, such as an electromyography sensor. The computing device may determine whether the audio signals satisfy an audio characteristic criterion indicative of speech and whether the head muscle activity signals satisfy a muscle movement criterion indicative of the user speaking. The computing device may perform voice recognition processing on the audio signals through a voice recognition algorithm in response to determining that both the audio signals and the head muscle activity signals satisfy their respective criterion indicative of the user speaking, and not perform voice recognition processing of audio signals while either audio signals or head muscle activity signals do not satisfy their respective criterion indicative of the user speaking.

Abstract: A voice dialogue method performed by a voice dialog system includes: a voice signal generation unit; a voice dialog agent unit; a voice output unit; and a voice input control unit, the method including: a step of, by the voice signal generation unit, receiving a voice input and generating a voice signal based on the received voice input; a step of, by the voice dialog agent unit, performing voice recognition processing on the voice signal and performing processing based on a result of the voice recognition processing to generate a response signal; a step of, by the voice output unit, outputting a voice based on the response signal; and a step of, when the voice output unit outputs the voice, by the voice input control unit, keeping the voice signal generation unit, for predetermined period after output of the voice, a receivable state in which a voice input is receivable.

Abstract: Embodiments disclosed herein provide a wireless controller which shows a voice, motion, or an image complying with or not complying with a user's command and controls an external device in accordance with the user's command. According to an embodiment, a wireless controller includes a main body provided in a shape of a flowerpot, and includes a voice recognition unit, a control unit generating a signal for controlling an object to be controlled, which is designated by a voice recognized in the voice recognition unit, in accordance with the voice, and a communication unit outputting the control signal generated in the control unit to the object to be controlled; and an indicator provided at the main body in a shape of at least one of a stem, a leaf, a flower, and a tree, and showing a motion corresponding to the voice recognized in the voice recognition unit.

Abstract: Examples disclose a low power integrated circuit to receive and digitize an audio stream. Further, the examples provide the low power integrated circuit to compare the digitized audio stream to a keyword and store the digitized audio stream in a memory. Additionally, the examples also disclose upon recognition of the keyword in the digitized audio stream, the low power integrated circuit transmits a signal to a processor to increase power and analyze the digitized audio stream.

Abstract: A vehicle based system and method for receiving voice inputs and determining whether to perform a voice recognition analysis using in-vehicle resources or resources external to the vehicle.

Abstract: Devices, methods, and computer program products for facilitating enhanced social interactions using a mobile device are disclosed. A method for facilitating an enhanced social interaction using a mobile device includes receiving an audio input at the mobile device, determining a salient portion of the audio input, receiving relevant information associated with the salient portion, and presenting the relevant information via the mobile device.

Abstract: The present invention relates to a method and apparatus for speaker-calibrated speaker detection. One embodiment of a method for generating a speaker model for use in detecting a speaker of interest includes identifying one or more speech features that best distinguish the speaker of interest from a plurality of impostor speakers and then incorporating the speech features in the speaker model.

Abstract: An audio receiving system includes logic configured to reduce the accumulation of delays caused by the late arrival of audio packets. This logic is configured to accelerate or decelerate presentation of a resulting audio stream in response to the detection of late packets. The acceleration is discontinued once the effects of the late packets have been compensated for. The audio receiving system is typically applied to applications in which lag is undesirable. These can include web conferencing, telepresence, and online video games.

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: An audio coding terminal and method is provided. The terminal includes a coding mode setting unit to set an operation mode, from plural operation modes, for input audio coding by a codec, configured to code the input audio based on the set operation mode such that when the set operation mode is a high frame erasure rate (FER) mode the codec codes a current frame of the input audio according to a select frame erasure concealment (FEC) mode of one or more FEC modes. Upon the setting of the operation mode to be the High FER mode, the one FEC mode is selected, from the one or more FEC modes predetermined for the High FER mode, to control the codec by incorporating of redundancy within a coding of the input audio or as separate redundancy information separate from the coded input audio according to the selected one FEC mode.

Abstract: The present invention relates to a method and device for encoding or decoding an object audio signal or rendering the object audio signal in a three-dimensional space. The method for processing an audio signal, according to one aspect of the present invention, comprises the steps of: generating a first object signal group and a second object signal group obtained by classifying a plurality of object signals according to a determined method; generating a first down-mix signal for the first object signal group; generating a second down-mix signal for the second object signal group; generating first object extraction information in correspondence with the first down-mix signal with respect to object signals included in the first object signal group; and generating second object extraction information in correspondence with the second down-mix signal with respect to object signals included in the second object signal group.

Abstract: A spread spectrum data hiding for audio signals is described. A set of pseudo-random noise sequences is added to an audio signal according to a data to be embedded. A masking curve is used to shape the added noise. A transient detection step can be used to control whether a shaped noise sequence is to be added or not. Embedded information is detected by first performing a whitening step and then performing a phase-only correlation with a same set of pseudo-random noise sequences. A detection method that is based on correlation of multiplexed noise sequences with a noise sequence embedded in the audio is also described.

Abstract: Some embodiments relate to techniques for encoding an audio signal represented by a plurality of frames including a first frame. The techniques include using at least one computer hardware processor to perform: obtaining an initial discrete spectral representation of the first frame; obtaining a primary discrete spectral representation of the initial discrete spectral representation at least in part by estimating a phase envelope of the initial discrete spectral representation and evaluating the estimated phase envelope at a discrete set of frequencies; calculating a residual discrete spectral representation of the initial discrete spectral representation based on the initial discrete spectral representation and the primary discrete spectral representation; and encoding the residual discrete spectral representation using a plurality of codewords.

Abstract: A method includes separating, at a device, an input audio signal into at least a low-band signal and a high-band signal. The low-band signal corresponds to a low-band frequency range and the high-band signal corresponds to a high-band frequency range. The method also includes selecting a non-linear processing function of a plurality of non-linear processing functions. The method further includes generating a first extended signal based on the low-band signal and the non-linear processing function. The method also includes generating at least one adjustment parameter based on the first extended signal, the high-band signal, or both.

Abstract: Provided is a method and apparatus for encoding/decoding an audio signal. Sections which are not used to output noise components near important spectral components and sub-bands which are not used to output noise components, are determined to be encoded or decoded, so that the efficiency of encoding and decoding an audio signal increases, and sound quality can be improved using less bits.

Abstract: An audio signal transmission device for encoding an audio signal includes an audio encoding unit that encodes an audio signal and a side information encoding unit that calculates and encodes side information from a look-ahead signal. An audio signal receiving device for decoding an audio code and outputting an audio signal includes: an audio code buffer that detects packet loss based on a received state of an audio packet, an audio parameter decoding unit that decodes an audio code when an audio packet is correctly received, a side information decoding unit that decodes a side information code when an audio packet is correctly received, a side information accumulation unit that accumulates side information obtained by decoding a side information code, an audio parameter missing processing unit that outputs an audio parameter upon detection of audio packet loss, and an audio synthesis unit that synthesizes decoded audio from the audio parameter.

Abstract: A system for processing audio data comprising a linear demixing system configured to receive a plurality of sub-band audio channels and to generate an audio output and a noise output. A spatial likelihood system coupled to the linear demixing system, the spatial likelihood system configured to receive the audio output and the noise output and to generate a spatial likelihood function. A sequential Gaussian mixture model system coupled to the spatial likelihood system, the sequential Gaussian mixture model system configured to generate a plurality of model parameters. A Bayesian probability estimator system configured to receive the plurality of model parameters and a speech/noise presence probability and to generate a noise power spectral density and spectral gains. A spectral filtering system configured to receive the spectral gains and to apply the spectral gains to noisy input mixtures.

Abstract: Methods and systems are provided for enhancing speech intelligibility in electronic devices. During outputting of acoustic signal via an electronic device, measurement of forces applied by user of the electronic device against the device (or enclosure thereof) may be obtained. The force measurements may be used to assess and/or estimate the listening intelligibility experienced by the user. Further, the force measurements may be used to control or adjust a listening intelligibility stage applied during generation and/or processing of the acoustic signals that are outputted via the electronic device. In some instances, an audio input, corresponding to ambient noise affecting intelligibility, may be obtained, and may be used to control or assist in controlling the listening intelligibility stage.

Abstract: The present invention relates to a system for processing of an audio signal relating to a diver in a deep diving environment. The system comprises an input device structured to receive the signal, a pitch changing module configured to change the pitch of the received signal, a plurality of processing modules collectively configured to process the pitch changed signal and produce a processed signal. The present invention is further directed to a method for processing of an audio signal relating to a diver in a deep diving environment.

Abstract: An audio communication system includes a generation unit that superimposes an addition sound having a volume level determined on the basis of a voice acquired by a voice acquisition unit on an input voice acquired by the voice acquisition unit of a transmission terminal and generates a synthesis sound and a transmission unit that transmits a signal of the synthesis sound generated by the generation unit to a reception terminal.

Abstract: A wireless communication device comprises a wireless communication interface, a processing system, and a user interface. The wireless communication interface is configured to wirelessly receive a mixed media signal comprising a plurality of audio streams originating from different source devices. The processing system is configured to isolate the audio streams and generate graphical representations of each one of the audio streams. The user interface is configured to display the graphical representations of each one of the audio streams and receive audio stream modification instructions from a user. The processing system is configured to process the audio stream modification instructions to generate at least one modified audio stream based on the audio stream modification instructions. The user interface is configured to display a graphical representation of the modified audio stream and audibly output a modified mixed media signal comprising the modified audio stream.