Abstract: In order to provide an optical apparatus capable of adjusting a light amount in each of a plurality of optical systems with a simple structure, the optical apparatus according to the present invention includes a plurality of optical systems each of which includes a stop, a storing unit configured to store information indicating a relationship between target aperture diameters of the stop and target light amounts in the respective optical systems, and a controller configured to adjust a light amount by changing an aperture diameter of the stop in the respective optical systems based on the information.

Abstract: A display device includes: a display panel with pixels arranged in a first direction and a second direction; and a light source. Each pixel includes: a first sub-pixel including a color filter for a first color; and a second sub-pixel arranged adjacent to the first sub-pixel in the first direction and including a color filter for a complementary color between a second color and a third color. The light source includes: a first light emitter configured to emit light in the first color; a second light emitter configured to emit light in the second color; and a third light emitter configured to emit light in the third color. One frame period includes: a first light emission period of causing the first and second light emitters to emit light simultaneously; and a second light emission period of causing the first and third light emitters to emit light simultaneously.

Abstract: A wearable computing device includes an outer covering, a housing, an electronic display configured to display an image, and a display drive integrated circuit (DDIC) comprising a processor and a memory device. The memory device(s) includes a blanking time programmed therein based on one or more parameters of the image and/or the DDIC, which generally refers to the time period in which the DDIC receives pixel data of the image from an external controller. The memory device stores instructions that when executed by the processor cause the processor to perform operations, including receiving an indication to rotate the image by a certain angle, upon receipt of the indication, starting to receive a transmission of the pixel data of the image from the external controller, and completing the transmission of the pixel data of the image during the blanking time and before the DDIC displays the image on the electronic display so as to avoid a tearing effect of the image on the electronic display.

Abstract: A source driver includes an output buffer. The output buffer includes: an amplifying unit; a first current control unit that includes a first constant current source and a second constant current source; and a second current control unit that includes a third constant current source and a fourth constant current source. The first constant current source is disposed on a first supply line. The second constant current source is disposed on a second supply line. The third constant current source is connected in parallel to the first supply line, supplies the first power supply voltage to the amplifying unit, and allows turning on and off the supply. The fourth constant current source is connected in parallel to the second supply line, supplies the second power supply voltage to the amplifying unit, and allows turning on and off the supply.

Abstract: A voltage generation unit includes first to k-th amplifiers that individually receiving first to k-th reference voltages having mutually different voltage values, individually amplify these reference voltages with gain 1, and output the reference voltages. The generation unit generates plural gradation voltages by dividing voltages between respective voltages output from the first to k-th amplifiers. A decoder unit selects one gradation voltage corresponding to the luminance level represented by the pixel data piece among the gradation voltages and generates a signal having the one gradation voltage as the drive signal for driving a display device. Each amplifier includes a response-speed increase circuit that includes at least one transistor in which a source and a back gate are connected to an output terminal of the amplifier, a predetermined electric potential is applied to a drain, and the reference voltage received by the amplifier is received at a gate.

Abstract: One embodiment provides a method comprising collecting ambient light sensing (ALS) data indicative of one or more ambient lighting conditions of a user within proximity of a display device, and collecting user preferences and characteristics (UPC) of the user. The UPC include a user-preferred display correlated color temperature (CCT). The user-preferred display CCT represents a CCT the user prefers for the display device. The method further comprises transmitting to a remote server the ALS data and the UPC, and receiving from the remote server one or more models constructed based in part on some of the ALS data and some of the UPC. The method further comprises automatically adjusting one or more color temperature settings of the display device based on the one or more models.

Abstract: A controller includes a frame memory configured to store an image frame, a frame memory controller coupled to the frame memory and configured to obtain image data from the image frame. The image data is associated with a color component of the image frame. The controller also includes a dither noise mask generator configured to provide dither noise masks according to dither noise levels for the image data, and a bit plane generator coupled to the frame memory controller and the dither noise mask generator and configured to generate bit planes based on the dither noise masks for the image data.

Abstract: One embodiment provides a method comprising determining a first representation of a source gamut of an input content in a first (2D) device-independent color space, determining a second representation of a target gamut of a display device in a second 2D device-independent color space, and determining a color transition protection zone (TPZ) based on the source gamut and the target gamut. The method further comprises utilizing a color gamut mapping (CGM) module to perform, based on the TPZ, linear color gamut compression from the first 2D device-independent color space to the second 2D device-independent color space if the target gamut is narrower than the source gamut. The method further comprises utilizing the same CGM module to perform, based on the TPZ, linear color gamut extension from the first 2D device-independent color space to the second 2D device-independent color space if the target gamut is wider than the source gamut.

Abstract: A mura compensation method of a display panel and a display panel are provided. The display panel includes a display area. The display area includes a regular display sub-area and a function display sub-area. The mura compensation method of the display panel includes: performing an initial compensation for each pixel unit in the display area; acquiring an actual grayscale value and an actual brightness value of each pixel unit in the function display sub-area; and determining a secondary compensation value of each pixel unit in the function display sub-area and performing a secondary compensation.

Abstract: A method is disclosed for receiving a synchronization signal from a display circuit configured to display a series of frames, each frame comprising a plurality of tiles of pixels, determining, based on the received synchronization signal, that the display circuit has consumed data corresponding to one or more tiles of a frame, identifying a predetermined number of tiles that are subsequent to the one or more tiles consumed by the display circuit based on the synchronization signal, determining that one or more tiles of the identified tiles require an update, selectively rendering the determined tiles, and transmitting the rendered tiles to the display circuit.

Abstract: Accessories to improve the tonal qualities of a guitar or other stringed instruments include a bridge sound plate, a bridge support, a stress tube, and one or more bridge pin bolts. A bridge pin bolt may include a head, a shank, a thread, and a nut, the head of the bridge pin bolt having a void therethrough extending from a front of the head to a back of the head to receive a guitar string therethrough, the void extending upwardly from the front of the head to the back of the head. One or more of the accessories may be formed of brass or another material with good tonal properties. A sound plate may be adjustable or tunable. The accessories may be used independently, or may be used together as a system of accessories.

Abstract: A musical instrument system includes a sound element medium including a plurality of sound-element information areas each having sound element information on a sound element, a reader configured to read the sound element information, a generator configured to generate musical sound information based on the read sound element information, and an audio unit configured to produce a sound based on the generated musical sound information, each of the sound-element information areas having a display corresponding to the sound element information, and the sound element information relating to both or one of pitch and tone.

Abstract: Systems and methods for dewatering slurries having relatively high solids content such as earthen slurries are disclosed. In some embodiments, one or more transducer devices configured for acoustic spectroscopy and/or electroacoustic spectroscopy are used to determine a parameter related to the particle size distribution (e.g., specific surfaced area) and/or zeta potential of the slurry.

Abstract: In a soundproof structure body including a tubular tube body having an opening portion, and a resonance type soundproof structure, in which a phase difference ?, at an upstream of the resonance type soundproof structure, between a reflected wave in the resonance type soundproof structure and a reflected wave of a transmitted wave transmitted through the resonance type soundproof structure and reflected by the opening portion satisfies Inequation |???|??/3 with respect to a resonance frequency of the resonance type soundproof structure. This soundproof structure body can effectively offset a reflected wave from a resonance type soundproof structure body by opening end reflection by appropriately specifying positions of the resonance type soundproof structure and the opening end portion of a duct, a tube line, or the like to improve an absorbance of a single resonance type soundproof structure.

Abstract: Noise cancellation systems and methods are provided that receive a reference signal representative of a noise to be reduced and provide a noise cancellation signal based upon the reference signal. Some examples include a feedback signal representative of a residual noise in the environment. At least one of the reference signal, the noise cancellation signal, or the feedback signal is filtered to remove components based upon a rotational rate of a rotating equipment associated with the environment, such as an engine of a vehicle. Accordingly, the noise cancellation system may not interfere with sounds related to the engine.

Abstract: A device for superscattering a target acoustic wave may include a body having an outer surface, at least one resonator being defined within the body and extending to an opening defined within the outer surface and configured to cause the superscattering of the target acoustic wave impinging upon the body, and a motor connected to the body and configured to selectively rotate the body.

Abstract: An information processing apparatus including a controller that controls, based on a detected state of the apparatus, outputting of sound that is performed by the apparatus, wherein the controller sequentially changes, according to an amount of change in the state, a mode of outputting a synthetic sound that can be output by the apparatus in a normal state is provided. Furthermore, an information processing method including, by a processor, based on a detected state of an apparatus, controlling outputting of sound that is performed by the apparatus, wherein the controlling includes, according to an amount of change in the state, sequentially changing a mode of outputting a synthetic sound that can be output by the apparatus in a normal state is provided.

Abstract: A notification appliance is disclosed and includes a sound engine that generates sound according to an acoustic pattern and the power control that regulates input voltage to the sound engine that is matched to the acoustic pattern. A method of powering a sounder is also disclosed and includes providing a constant input current and regulating an input voltage to a phase of a sound engine corresponding with an acoustic signal generated by a sound engine.

Abstract: In some implementations, a system may receive an audio stream associated with a call between a user and an agent. The system may process, by the device and using a speech alteration model, speech from a first channel of the audio stream to alter the speech from having a first speech characteristic to having a second speech characteristic, wherein the speech alteration model is trained based on reference audio data associated with the first speech characteristic and the second speech characteristic and based on reference speech data associated with the first speech characteristic and the second speech characteristic. The system may extract the speech from the first channel that has the first speech characteristic. The system may provide, within a second channel of the audio stream, altered speech that corresponds to the speech and that has the first speech characteristic.

Abstract: A method for estimating an embedding capacity includes receiving, at a deterministic reference encoder, a reference audio signal, and determining a reference embedding corresponding to the reference audio signal, the reference embedding having a corresponding embedding dimensionality. The method also includes measuring a first reconstruction loss as a function of the corresponding embedding dimensionality of the reference embedding and obtaining a variational embedding from a variational posterior. The variational embedding has a corresponding embedding dimensionality and a specified capacity. The method also includes measuring a second reconstruction loss as a function of the corresponding embedding dimensionality of the variational embedding and estimating a capacity of the reference embedding by comparing the first measured reconstruction loss for the reference embedding relative to the second measured reconstruction loss for the variational embedding having the specified capacity.

Abstract: A method for mining feature information, an apparatus for mining feature information and an electronic device are disclosed. The method includes: determining a usage scenario of a target device; obtaining raw audio data including real scenario data, speech synthesis data, recorded audio data and other media data; generating target audio data of the usage scenario by simulating the usage scenario based on the raw audio data; and obtaining feature information of the usage scenario by performing feature extraction on the target audio data.

Abstract: Provided is an information processing apparatus including a first dictionary that has registered a plurality of utterance expressions, a first voice recognition unit that executes a first voice recognition process using the first dictionary, and a dictionary update unit that updates utterance expressions to be registered to the first dictionary based on usage performance of utterance of a user, in which the dictionary update unit deletes an utterance expression having a low usage performance of utterance from the first dictionary or transfers the utterance expression to a second dictionary that has registered a plurality of utterance expressions not registered in the first dictionary.

Abstract: An electronic device is provided. The electronic device includes a processor and a memory operatively connected to the processor. The memory may store instructions that, when executed, cause the processor to receive a voice input of a user, to extract a feature from the voice input of the user, to select an acoustic model through comparison with the extracted feature, and to learn the feature of the voice input by performing fine-tuning on the selected acoustic model.

Abstract: The present disclosure is directed towards methods, systems and apparatuses for providing a human-machine interface and assistant that are particularly useful for financial trading and in researching trading opportunities in marketplaces that are otherwise difficult to research while taking into account a broad context of the query including past queries and specific preferences or restrictions of the user issuing the query. The present disclosure provides tools for interpreting voice queries and iteratively improving, through direct and inferred corrections, the translation and execution of queries for information and trading opportunities.

Abstract: A wearable device collects a fingerprint pattern input by a user and speech input by the user. The wearable device sends the fingerprint pattern to an electronic device, to enable the electronic device to perform authentication on the fingerprint pattern input by a user. The wearable device sends the speech to the electronic device, and, upon a determination that the authentication succeeds, the electronic device is enabled to execute a function corresponding to the speech.

Abstract: Methods, systems, and apparatuses for predicting an end of a command in a voice recognition input are described herein. The system may receive data comprising a voice input. The system may receive a signal comprising a voice input. The system may detect, in the voice input, data that is associated with a first portion of a command. The system may predict, based on the first portion and while the voice input is being received, a second portion of the command. The prediction may be generated by a machine learning algorithm that is trained based at least in part on historical data comprising user input data. The system may cause execution of the command, based on the first portion and the predicted second portion, prior to an end of the voice input.

Abstract: A method including transcribing, into digital tokens, utterances from a conversation between an agent and a person. The method also includes embedding the digital tokens into an utterances tensor including sequences of the digital tokens. The method also includes obtaining a metadata tensor by encoding metadata related to the utterances into the metadata tensor. The method also includes executing a machine learning model which takes, as input, the utterances tensor and the metadata tensor, and which outputs a predicted source article predicted to be related to the utterances. The method also includes generating an interactive link to the predicted source article.

Abstract: The present disclosure discloses a speech recognition method and apparatus, and relates to the field of speech and deep learning technologies. A specific implementation scheme involves: acquiring candidate recognition results with first N recognition scores outputted by a speech recognition model for to-be-recognized speech, N being a positive integer greater than 1; scoring the N candidate recognition results based on pronunciation similarities between candidate recognition results and pre-collected popular entities, to obtain similarity scores of the candidate recognition results; and integrating the recognition scores and the similarity scores of the candidate recognition results to determine a recognition result corresponding to the to-be-recognized speech from the N candidate recognition results. The present disclosure can improve recognition accuracy.

Abstract: Methods and systems are disclosed herein for improvements relating to compressed automatic speech recognition (ASR) systems. The ASR system may comprise a compressed acoustic engine and an adaptive decoder. The adaptive decoder may be dynamically compiled based on characteristics of the compressed acoustic engine and a current state of the application device. In some embodiments, a dynamic command list is used to manage context-specific commands. Two or more commands recognized by the adaptive decoder may be confusable due to compression of the ASR system. Alternate commands may be determined that are semantically equivalent but phonetically different than the confusable commands to reduce classification error of the adaptive decoder. An alternate command may replace one or more of the confusable commands in the adaptive decoder. In some embodiments, a user interface is displayed to a user of the ASR system to select the alternate command for replacement in the decoder.

Abstract: A computer-implemented method includes generating an empirically derived acoustic confusability measure by processing example utterances and iterating from an initial estimate of the acoustic confusability measure to improve the measure. The method can further include using the acoustic confusability measure to selectively limit phrases to make recognizable by a speech recognition application.

Abstract: Disclosed is an electronic device. An electronic device according to an embodiment comprises: a display; a microphone; a wireless communication circuit; a processor operatively connected to the display, the microphone, and the communication circuit; and a memory operatively connected to the processor, wherein the memory may store a plurality of user-customized tasks mapped to one of a selected word, phrase, or sentence, and may store instructions that, when executed, cause the processor to receive a first voice input according to user's utterance through the microphone, obtain context information related to the electronic device when one of the word, phrase, or sentence is included in the first voice input, select one task from among the plurality of tasks on the basis of at least in part on the context information, and perform the selected one task by using at least a part of the electronic device. Other embodiments understood through the present specification are also possible.

Abstract: Systems and methods for generating responses to user input such as dialogues, and images are discussed. The system may generate, by a response generation module of at least one server, an optimal generated response to the user communication by applying an generative adversarial network. In some embodiments, the generative adversarial network may include a hierarchical recurrent encoder decoder generative adversarial network including a generator and a discriminator component.

Abstract: Technologies are disclosed for interacting with a virtual assistant to coordinate, recommend and perform actions. According to some examples, a user may use their voice to interact with a virtual assistant to receive recommendations relating to determining when to perform one or more actions. For example, a user may interact with a virtual assistant to request a recommendation as to when they should leave for the office, leave the office for the day, perform a task, and the like. The recommendation system accesses selected data sources (e.g., calendars, task lists, traffic, transportation schedules, maps, . . . ) to obtain data used in generating the recommendation. In addition, to providing a recommended time, the virtual assistant may also recommend actions to perform. The virtual assistant may also provide notifications to one or more other users that includes information relating to the user leaving.

Abstract: The disclosure deals with a system and method for improved general task-oriented virtual assistants (VAs). The presently disclosed framework incorporates discovery of knowledge from online sources to accomplish tasks (open world), user-specific knowledge for personalization, and domain-specific knowledge for context adaptation to recommend and assist the users over procedural tasks such as cooking and Do-it-Yourself (DIY) tasks. The approach also focuses on content curation for fault-tolerant execution to ensure the end goal is reached despite common failures.

Abstract: Methods, apparatus, systems, and computer-readable media are provided for using shortcut command phrases to operate an automated assistant. A user of the automated assistant can request that a shortcut command phrase be established for causing the automated assistant to perform a variety of different actions. In this way, the user does not necessarily have to provide an individual command for each action to be performed but, rather, can use a shortcut command phrase to cause the automated assistant to perform the actions. The shortcut command phrases can be used to control peripheral devices, IoT devices, applications, websites, and/or any other apparatuses or processes capable of being controlled through an automated assistant.

Abstract: Systems and processes for providing a virtual assistant service are provided. In accordance with one or more examples, a method includes receiving, from an accessory device communicatively coupled to the first electronic device, a representation of a speech input representing a user request. The method further includes detecting a second electronic device and transmitting, from the first electronic device, a representation of the user request and data associated with the detected second electronic device to a third electronic device. The method further includes receiving, from the third electronic device, a determination of whether a task is to be performed by the second electronic device in accordance with the user request; and in accordance with a determination that a task is to be performed by the second electronic device, requesting the second electronic device to performed the task in accordance with the user request.

Abstract: Methods and systems for rendering lists of instructions and performing actions associated with those lists are described herein. In some embodiments, an individual may request that a voice activated electronic device associated with their user account assist in performing a task using a list of instructions. The list of instructions may include metadata that indicates actions capable of being performed by additional Internet of Things (“IoT”) devices. When the instructions are rendered, an instructions speechlet may recognize the metadata and may cause one or more of the IoT devices to perform a particular action. Furthermore, the metadata may also correspond to content capable of being rendered by the voice activated electronic device to assist the individual in performing a particular step of the instructions.

Abstract: The present disclosure discloses a method and device of generating acoustic features, speech model training, and speech recognition. By acquiring the acoustic information vector of the current speech frame and the information weight of the current speech frame, and according to the accumulated information weight corresponding to the previous speech frame, the retention rate corresponding to the current speech frame, and the information weight of the current speech frame, the accumulated information weight corresponding to the current speech frame can be obtained. The retention rate is the difference between 1 and a leakage rate.

Abstract: A detection device provides a voice signal of a person and includes a light source, a first planar carrier medium, a second planar carrier medium, a sensor device, and an evaluation device. The first planar carrier medium includes an input coupling region and an output coupling region. The second planar carrier medium includes a light input coupling region and a light output coupling region. Light from the light source is emitted into the second planar carrier medium and output in the direction of a front neck region of the person, and the light reflected on the neck region is input into the first planar carrier medium and output out of the first planar carrier medium. The output light is detected by the sensor device which provides sensor data to the evaluation device which converts the sensor data into vibrational data and provides the voice signal.

Abstract: Presented are a combined learning method and device using a transformed loss function and feature enhancement based on a deep neural network for speaker recognition that is robust in a noisy environment. A combined learning method using a transformed loss function and feature enhancement based on a deep neural network, according to one embodiment, can comprise the steps of: learning a feature enhancement model based on a deep neural network; learning a speaker feature vector extraction model based on the deep neural network; connecting an output layer of the feature enhancement model with an input layer of the speaker feature vector extraction model; and considering the connected feature enhancement model and speaker feature vector extraction model as one mode and performing combined learning for additional learning.

Abstract: This relates to systems and processes for using a virtual assistant to control electronic devices. In one example process, a user can speak an input in natural language form to a user device to control one or more electronic devices. The user device can transmit the user speech to a server to be converted into a textual representation. The server can identify the one or more electronic devices and appropriate commands to be performed by the one or more electronic devices based on the textual representation. The identified one or more devices and commands to be performed can be transmitted back to the user device, which can forward the commands to the appropriate one or more electronic devices for execution. In response to receiving the commands, the one or more electronic devices can perform the commands and transmit their current states to the user device.

Abstract: What is described is a method for loss concealment of spatial audio parameters, the spatial audio parameters having at least a direction of arrival information; the method having the following steps: receiving a first set of spatial audio parameters having at least a first direction of arrival information; receiving a second set of spatial audio parameters, having at least a second direction of arrival information; and replacing the second direction of arrival information of a second set by a replacement direction of arrival information derived from the first direction of arrival information, if at least the second direction of arrival information or a portion of the second direction of arrival information is lost or damaged.

Abstract: An apparatus comprising means configured to: receive a primary track comprising at least one audio signal; receive at least one secondary track, each of the at least one secondary track comprising at least one audio signal, wherein the at least one secondary track is based on the primary track; and decode and render the primary track and the at least one secondary track using spatial audio decoding.

Abstract: The present disclosure discloses an inter-channel phase difference parameter encoding method, where a current frame is obtained; a signal type and a previous IPD parameter encoding scheme of a previous frame are obtained; a current IPD parameter encoding scheme is obtained at least based on the signal type of the previous frame and the previous IPD parameter encoding scheme; and an IPD parameter of the current frame is processed based on the current IPD parameter encoding scheme.

Abstract: A method for applying a watermark signal to a speech signal to prevent unauthorized use of speech signals, the method may include receiving an original speech signal; determining a corresponding spectrogram of the original speech signal; selecting a phase sequence of fixed frame length and uniform distribution; and generating an encoded watermark signal based on the corresponding spectrogram and phase sequence.

Abstract: An apparatus for determining a predetermined characteristic related to an artificial bandwidth limitation processing of an audio signal has a slope evaluator configured for evaluating a slope of a spectrum of the audio signal to obtain a slope evaluation result. The apparatus has a frequency evaluator configured for evaluating a cut-off frequency of the spectrum of the audio signal to obtain a frequency evaluation result, and has a processor for providing an information indicating that the audio signal has the predetermined characteristic dependent on an evaluation of the slope evaluation result and an evaluation of the frequency evaluation result.

Abstract: A codec allowing for switching between different coding modes is improved by, responsive to a switching instance, performing temporal smoothing and/or blending at a respective transition.

Abstract: A method includes receiving text input data for conversion into synthesized speech and determining, using a hotword-aware model trained to detect a presence of a hotword assigned to a user device, whether a pronunciation of the text input data includes the hotword. The hotword is configured to initiate a wake-up process on the user device for processing the hotword and/or one or more other terms following the hotword in the audio input data. When the pronunciation of the text input data includes the hotword, the method also includes generating an audio output signal from the text input data and providing the audio output signal to an audio output device to output the audio output signal. The audio output signal when captured by an audio capture device of the user device, configured to prevent initiation of the wake-up process on the user device.

Abstract: Disclosed is an operation device that executes a noise removal process of removing noise from a collected audio signal collected by a microphone. In an acting state where noise generation possibly occurs, the operation device transmits noise generation information indicating that the operation device is in the acting state where noise generation possibly occurs, and changes the details of the noise removal process, according to the noise generation information.

Abstract: In an audio signal processing method and system for echo suppression, selection of a target audio processing mode is controlled based on strength of a speaker signal. In the method and system, a control signal is generated based on the strength of the speaker signal, and the target audio processing mode is controlled based on the control signal to perform signal processing on a microphone signal so as to obtain better voice quality. When the speaker signal does not exceed a threshold, the system selects a first mode, and performs signal processing on a first audio signal and a second audio signal to obtain a first target audio; or when the speaker signal exceeds a threshold, the system selects a second mode, and performs signal processing on a second audio signal to obtain a second target audio, and the mode can be switched based on the speaker signal.