Abstract: An electronic device may capture a voice command from a user. The electronic device may store contextual information about the state of the electronic device when the voice command is received. The electronic device may transmit the voice command and the contextual information to computing equipment such as a desktop computer or a remote server. The computing equipment may perform a speech recognition operation on the voice command and may process the contextual information. The computing equipment may respond to the voice command. The computing equipment may also transmit information to the electronic device that allows the electronic device to respond to the voice command.

Abstract: Systems and techniques for automated voice assistant personality selector are described herein. A task may be identified that is to be completed by a user of a voice-enabled virtual assistant. A response may be output in connection with the task using a default personality for the voice-enabled virtual assistant selected based on the task. A task completion checkpoint may be determined for the task. It may be identified that the task completion checkpoint has not been reached. A personality profile of the user may be compared to personality data of a set of voice-enabled virtual assistant profiles corresponding to respective members of a set of available voice personalities for the voice-enabled virtual assistant. An escalation personality may be selected for the voice-enabled virtual assistant based the comparison and the task. Commands for facilitating user completion of the task may be transmitted via the voice-enabled virtual assistant using the selected escalation personality.

Abstract: An electronic device is provided. The electronic device includes a microphone configured to receive a voice, communication circuitry configured to communicate with an external electronic device, a memory, and a processor operatively connected to the microphone, the communication circuitry, and the memory. The memory stores instructions that, when executed by the processor, cause the electronic device to perform voice recognition for voice data corresponding to the voice, and determine whether to perform voiceprint recognition for the voice data based on whether a first expression is included in a result of the voice recognition.

Abstract: Disclosed is an augmented reality system to generate and cause display of an augmented reality interface at a client device. Various embodiments may detect speech, identify a source of the speech, transcribe the speech to a text string, generate a speech bubble based on properties of the speech and that includes a presentation of the text string, and cause display of the speech bubble at a location in the augmented reality interface based on the source of the speech.

Abstract: A system and method of updating automatic speech recognition parameters on a mobile device are disclosed. The method comprises storing user account-specific adaptation data associated with ASR on a computing device associated with a wireless network, generating new ASR adaptation parameters based on transmitted information from the mobile device when a communication channel between the computing device and the mobile device becomes available and transmitting the new ASR adaptation data to the mobile device when a communication channel between the computing device and the mobile device becomes available. The new ASR adaptation data on the mobile device more accurately recognizes user utterances.

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

Abstract: A user interface device includes a sound input unit which acquires an inputted sound, a sound recognition unit which acquires a result of predetermined recognition processing on the acquired sound, a speech generation unit which generates a speech corresponding to the result of the recognition processing, and a speech output unit which outputs the speech generated by the speech generation unit.

Abstract: An encoder for encoding an audio signal has a predictor, a factorizer, a transformer and a quantize and encode stage. The predictor is configured to analyze the audio signal to obtain prediction coefficients describing a spectral analog of the audio signal or a fundamental frequency of the audio signal and subject the audio signal to an analysis filter function dependent on the prediction coefficients to output a residual signal of the audio signal. The factorizer is configured to apply a matrix factorization onto an audiocorrelation or covariance matrix of synthesis filter function defined by the prediction coefficients to obtain factorized matrices. The transformer is configured to transform the residual signal based on the factorized matrices to obtain a transformed residual signal. The quantize and decode stage is configured to quantize the transformed residual signal to obtain a quantized transformed residual signal or an encoded quantized transformed residual signal.

Abstract: An apparatus for generating a bandwidth extended signal from a bandwidth limited audio signal, the bandwidth limited audio signal The patch generator is configured to perform a harmonic patching algorithm to obtain the patched signal. The signal manipulator is configured for manipulating a signal before patching or the patched signal. The timely preceding bandwidth limited time block timely precedes the current bandwidth limited time block in the plurality of consecutive bandwidth limited time blocks of the bandwidth limited audio signal. The combiner is configured for combining the bandwidth limited audio signal having the core frequency band and the manipulated patched signal having the upper frequency band to obtain the bandwidth extended signal.

Abstract: A method comprising: receiving lattice vector quantised parameter data, the parameter data representing at least one audio signal; determining within the data at least one bit error; and controlling the decoding of the data to generate an audio signal based on the determining of the bit error.

Abstract: Embodiments of the present disclosure provide a method and a device for determining a comment, a server and a storage medium. The method includes: obtaining text content of an event; determining an event tag of the event according to the text content; comparing the event tag with comment tags in a comment graph, to determine candidate comments, the comment graph being generated according to other events and corresponding comments, and the comment graph including comments and corresponding comment tags; and screening the candidate comments, to determine a candidate comment satisfying a preset condition as the comment of the event.

Abstract: The present disclosure relates to speech recognition systems and methods using an adaptive incremental learning approach. More specifically, the present disclosure relates to adaptive incremental learning in a self-taught vocal user interface.

Abstract: A method for phoneme identification. The method includes receiving an audio signal from a speaker, performing initial processing comprising filtering the audio signal to remove audio features, the initial processing resulting in a modified audio signal, transmitting the modified audio signal to a phoneme identification method and a phoneme replacement method to further process the modified audio signal, and transmitting the modified audio signal to a speaker. Also, a system for identifying and processing audio signals. The system includes at least one speaker, at least one microphone, and at least one processor, wherein the processor processes audio signals received using a method for phoneme replacement.

Abstract: An encoder for encoding frequency transform coefficients of a harmonic audio signal include the following elements: A peak locator configured to locate spectral peaks having magnitudes exceeding a predetermined frequency dependent threshold. A peak region encoder configured to encode peak regions including and surrounding the located peaks. A low-frequency set encoder configured to encode at least one low-frequency set of coefficients outside the peak regions and below a crossover frequency that depends on the number of bits used to encode the peak regions. A noise-floor gain encoder configured to encode a noise-floor gain of at least one high-frequency set of not yet encoded coefficients outside the peak regions.

Abstract: A method, computer system, and computer program product for determining a server conversational state in an interactive dialog between a server and an administrator is provided. The embodiment may include receiving a query statement from a user. The embodiment may also include updating a context map to reflect a context of the received query statement. The embodiment may further include transmitting a request to one or more cognitive engines capable of resolving the received query statement. The embodiment may also include processing the transmitted request in each of the one or more cognitive engines. The embodiment may further include updating the context map with a resolution operation performed by the one or more cognitive engines. The embodiment may also include displaying a response to the user.

Abstract: An application automatically composed using natural language processing. A natural language input comprising one or more application requirements is received via an interface. The natural language input is parsed to extract one or more chunks, each chunk representing one of the application requirements, and at least one of the chunks representing at least one of one or more main functionalities described by the application requirements. A coarse architecture logically arranging the main functionalities to satisfy the application requirements is inferred according to the chunks. Existing assets corresponding to the chunks are identified, each asset associated with at least one of the main functionalities. The identified assets are assembled according to the coarse architecture. The assembled assets are deployed as an application.

Abstract: In one aspect, a computerized method useful for implementing and managing an Automated assistant translator between an Automated assistant service and a plurality of third-party messaging platforms includes the step of deploying an Automated assistant in a plurality of third-party messaging platforms. The method includes the step of, with the Automated assistant, receiving a conversational input from a human user via a first third-party messaging platform of the plurality of third-party messaging platforms. The method includes the step of parsing the conversation input. The method includes the step of determining a request for additional information in the conversation input. The method includes the step of retrieving a set of information relating to the request for additional information in the conversation input. The method includes the step of generating a natural-language output that includes the information.

Abstract: A method for processing audio data includes obtaining a noise frame of an audio signal, and decomposing the current noise frame into a noise low-band signal and a noise high-band signal; and encoding and transmitting the noise low-band signal by using a first discontinuous transmission mechanism, and encoding and transmitting the noise high-band signal by using a second discontinuous transmission mechanism. According to the present disclosure, different processing manners are used for the high-band signal and the low-band signal, calculation loads and encoded bits may be saved under a premise of not lowering subjective quality of a codec, and bits that are saved may help to achieve an objective of reducing a transmission bandwidth or improving overall encoding quality.

Abstract: In a transmit method, a set of data eigenvectors that are based on a Prometheus Orthonormal Set (PONS) code construction and orthogonal to each other are stored, wherein the data eigenvectors are mapped to unique multi-bit words. A pilot sequence representing a pilot eigenvector that is based on the PONS code construction and orthogonal to each of the data eigenvectors is generated. Input data are grouped into multi-bit words and data eigenvectors among the data eigenvectors are selected based on the multi-bit words. A spread data sequence including the selected data eigenvectors and that is synchronized to the pilot sequence is generated. An acoustic signal including the synchronized pilot sequence and the spread data sequence is generated. The acoustic signal is transmitted.

Abstract: A computer-implemented method for language generation of a flow diagram, which receives a flow diagram. A plurality of geometric shapes within the flow diagram is identified. A plurality of text elements within the flow diagram is identified. The plurality of text elements and corresponding geometric shapes are associated. The association between the plurality of geometric shapes are identified. A diagram matrix based on the associations between the plurality of geometric shapes is generated. A linear language representation of the diagram matrix is generated.