Abstract: A spectrum coding method includes quantizing spectral data of a current band based on a first quantization scheme, generating a lower bit of the current band using the spectral data and the quantized spectral data, quantizing a sequence of lower bits including the lower bit of the current band based on a second quantization scheme, and generating a bitstream based on a upper bit excluding N bits, where N is 1 or greater, from the quantized spectral data and the quantized sequence of lower bits.

Abstract: This disclosure describes, in part, techniques and systems for generating and outputting immersive, multi-device content items in user environment, such as connected homes, offices, and the like. For example, the techniques and systems may output different portions of content on different devices within a user environment based on information such as respective capabilities of the devices, a current location of the user within the environment, a time of day, which user(s) are present in the environment, and/or the like.

Abstract: A speech-processing system may provide access to multiple virtual assistants. Speech-processing systems may perform actions for or on behalf of users with the aid of skills; e.g., a shopping skill, navigation skill, communications skill, etc. Some skills may be associated with more than one assistant. The speech-processing system may determine which assistant to invoke upon receiving a command from a user device. The identity of the virtual assistant is propagated to the skill and the device, as well as other components of the speech-processing system. In some cases, however, a multi-assistant skill may determine that an assistant other than the one initially selected by the speech-processing system is to handle the command. The skill may send the identity of the new assistant back to the speech-processing system. The speech-processing system may restart the command dissemination process to provide each component of the system with the updated assistant identity.

Abstract: A computer-implemented method for facilitating navigation of an oil-gas domain application using a virtual assistant integrated within the oil-gas domain application includes generating a trained model for responding to utterances received from a user via a virtual assistant integrated within an oil-gas domain application. The trained model links the utterances to respective actions and responses; receiving a user utterance via the virtual assistant integrated within the oil-gas domain application. The method further includes determining a response to the user utterance using the trained model, wherein the response is associated with performing an action within the oil-gas domain application; and providing the response to the virtual assistant to cause the virtual assistant to execute the action within the oil-gas domain application.

Abstract: Systems and methods are provided to determine consensus values for duplicate fields in a document or form.

Abstract: A computer server system comprises a communications module; a processor coupled with the communications module; and a memory coupled to the processor and storing processor-executable instructions which, when executed by the processor, configure the processor to receive, via the communications module and from a server associated with a first device, a request to perform an operation; determine that the first device cannot perform the operation; send, via the communications module and to the server associated with the first device, a signal causing the first device to output a message indicating that the first device cannot perform the operation and requesting authentication from a second device; receive, via the communications module and from the second device, a signal including authentication information; and send, via the communications module and to the second device, a signal including a selectable option to perform the operation.

Abstract: Speech fluency evaluation and feedback tools are described. A computing device such as a smartphone may be used to collect speech (and/or other data). The collected data may be analyzed to detect various speech events (e.g., stuttering) and feedback may be generated and provided based on the detected speech events. The collected data may be used to generate a fluency score or other performance metric associated with speech. Collected data may be provided to a practitioner such as a speech therapist or physician for improved analysis and/or treatment.

Abstract: A method for exiting a voice skill, an apparatus, a device, and a storage medium are provided by embodiments of the present disclosure, wherein a user voice instruction is received; a target exit intention corresponding to the user voice instruction is identified according to the user voice instruction and a grammar rule of a preset exit intention; and a corresponding operation is executed on a current voice skill of a device according to the target exit intention. The embodiments of the present disclosure refine and expand the user's exit intention. After the target exit intention to which the user voice instruction belongs is identified, the corresponding operation is executed according to the target exit intention so as to meet the users' different exit requirements for the voice skills, enhance the fluency and convenience of user interaction with the device and improve the user's exit experience when using the voice skills.

Abstract: A method for encoding a high frequency signal includes determining a signal type of a high frequency signal of a current frame, smoothing and scaling time envelopes of the high frequency signal of the current frame and obtaining time envelopes of the high frequency signal of the current frame that require to be encoded when the high frequency signal of the current frame is a non-transient signal and a high frequency signal of the previous frame is a transient signal, and quantizing and encoding the time envelopes of the high frequency signal of the current frame that require to be encoded, and frequency information and signal type information of the high frequency signal of the current frame.

Abstract: Aspects relate to apparatuses and methods for selectively inserting text into a video resume. An exemplary apparatus includes a processor and a memory communicatively connected to the processor, the memory containing instructions configuring the processor to receive a video resume from a user, divide the video resume is into temporal sections, acquire a plurality of textual inputs from a user, wherein the plurality of textual inputs pertains to the same user of received video resume, classify the plurality of textual inputs to corresponding temporal sections of the received video resume and display, as a function of the classification, the received video resume with a corresponding plurality of textual inputs.

Abstract: A robot according to the present disclosure comprises: a microphone; a camera disposed to face a predetermined direction; and a processor configured to: inactivate driving of the camera and activate driving of the microphone, if a driving mode of the robot is set to a user monitoring mode; acquire a sound signal through the microphone; activate the driving of the camera based on an event estimated from the acquired sound signal; confirm the event from the image acquired through the camera; and control at least one constituent included in the robot to perform an operation based on the confirmed event.

Abstract: A voice inputting device inputs a voice operation of a user, and transmits voice data based on the voice operation to a first cloud server. The first cloud server receives the voice data from the voice inputting device, analyzes the received voice data, and determines an operational skill level of the user and the details of the voice operation. A second cloud server generates a control command for an air conditioner based on the operational skill level and the details of the voice operation determined by the first cloud server, and transmits the generated control command to the air conditioner.

Abstract: Systems and methods are provided herein for avoiding inadvertently trigging a voice assistant with audio played through a speaker. An audio signal is captured by sampling a microphone of the voice assistant at a sampling frequency that is higher than an expected finite sampling frequency of previously recorded audio played through the speaker to generate a voice data sample. A quality metric of the generated voice data sample is calculated by determining whether the generated voice data sample comprises artifacts resulting from previous compression or approximation by the expected finite sampling frequency. Based on the calculated quality metric, it is determined whether the captured audio signal is previously recorded audio played through the speaker. Responsive to the determination that the captured audio signal is previously recorded audio played through the speaker, the voice assistant refrains from being activated.

Abstract: A voice command system according to a first disclosure comprises a gateway apparatus having an interface configured to receive a voice command, and a controller configured to perform a registration process of registering a speaker permitted to receive the voice command. The controller is configured to perform an authentication process of rejecting a reception of the voice command when a speaker of the voice command is not registered, and permitting a reception of the voice command when a speaker of the voice command is registered. The controller is configured to perform the authentication process for each voice command.

Abstract: An apparatus for selecting one of a first encoding algorithm having a first characteristic and a second encoding algorithm having a second characteristic for encoding a portion of an audio signal to obtain an encoded version of the portion of the audio signal has a first estimator for estimating a first quality measure for the portion of the audio signal, which is associated with the first encoding algorithm, without actually encoding and decoding the portion of the audio signal using the first encoding algorithm. A second estimator is provided for estimating a second quality measure for the portion of the audio signal, which is associated with the second encoding algorithm, without actually encoding and decoding the portion of the audio signal using the second encoding algorithm. The apparatus has a controller for selecting the first or second encoding algorithms based on a comparison between the first and second quality measures.

Abstract: The present disclosure is generally directed to the generation of voice-activated data flows in interconnected network. The voice-activated data flows can include input audio signals that include a request and are detected at a client device. The client device can transmit the input audio signal to a data processing system, where the input audio signal can be parsed and passed to the data processing system of a service provider to fulfill the request in the input audio signal. The present solution is configured to conserve network resources by reducing the number of network transmissions needed to fulfill a request.

Abstract: The disclosure herein relates to methods and systems for enabling human-robot interaction (HRI) to resolve task ambiguity. Conventional techniques that initiates continuous dialogue with the human to ask a suitable question based on the observed scene until resolving the ambiguity are limited. The present disclosure use the concept of Talk-to-Resolve (TTR) which initiates a continuous dialogue with the user based on visual uncertainty analysis and by asking a suitable question that convey the veracity of the problem to the user and seek guidance until all the ambiguities are resolved. The suitable question is formulated based on the scene understanding and the argument spans present in the natural language instruction. The present disclosure asks questions in a natural way that not only ensures that the user can understand the type of confusion, the robot is facing; but also ensures minimal and relevant questioning to resolve the ambiguities.

Abstract: Provided is a system, server, and method for speech recognition capable of collectively setting a plurality of setting items for device control through an utterance of a single sentence provided in the form of natural language. The system includes: a home appliance configured to receive a speech command that is generated through an utterance of a single sentence for control of the home appliance; and a server configured to receive the speech command in the single sentence from the home appliance and interpret the speech command of the single sentence through multiple intent determination.

Abstract: Disclosed herein is a system for facilitating accomplishing tasks based on a natural language conversation. Accordingly, the system may include a direct graph unit. Further, the direct graph unit may include a directed graph. Further, the directed graph models a non-linearity of the natural language conversation. Further, the directed graph may include a set of nodes connected by at least one edge. Further, the system may include a context-encoded language understanding unit may include a learning unit and an inferring unit. Further, the learning unit may be configured for receiving a plurality of inputs. Further, the learning unit may be configured for generating a model based on the plurality of inputs. Further, the inferring unit may be configured for receiving a plurality of inputs. Further, the inferring unit may be configured for generating an output based on the plurality of inputs and the model.

Abstract: Disclosed is a system for speech recognition text enhancement fusing multi-modal semantic invariance, the system includes an acoustic feature extraction module, an acoustic down-sampling module, an acoustic feature extraction module, an acoustic down-sampling module, an encoder and a decoder fusing multi-modal semantic invariance; the acoustic feature extraction module is configured for frame-dividing processing of speech data, dividing the speech data into short-term audio frames with a fixed length, extracting thank acoustic features from the short-term audio frames, and inputting the acoustic features into the acoustic down-sampling module for down-sampling to obtain an acoustic representation; inputting the speech data into an existing speech recognition module to obtain input text data, and inputting the input text data into the encoder to obtain an input text encoded representation; inputting the acoustic representation and the input text encoded representation into the decoder to fuse.