28 results about "Speech interface" patented technology

A method for voice-based interaction between a user (e.g., a shopper) and online stores includes capturing an initial voice input from a user and converting the voice input to text. The text is analyzed and initial search terms are created such as product name, product description, product category, product brand name, product manufacturer and retailer. One or more searches are conducted using the initial search terms to identify one or more retailers meeting the search terms. One or more identified online retailers are accessed and searches are executed at the one or more identified retailers, generating search results. The identified online retailers have a user interface allowing purchases of the identified product or item. The search results are returned to the user.

The invention relates to the technical field of intelligent voice data processing, and discloses an intelligent voice interface data calling method. The method comprises the following steps: receiving a voice instruction stream input by a user, and converting the voice instruction stream into a structured semantic unit set through a multi-level semantic analysis engine; and dynamically layering the instruction intention according to the semantic association degree in the structured semantic unit set to generate an intention sequence with priority marks. And based on the priority mark of the intention sequence, extracting a candidate data block set matched with the intention of each level from a distributed data storage cluster. And performing cross-modal feature alignment on the candidate data block set, and calculating the semantic coverage degree of each data block and the corresponding intention hierarchy. And screening out target data blocks according to a semantic coverage threshold, and recombining the data blocks according to a priority sequence of the intention sequence. And loading the recombined target data block set to a real-time processing channel of the voice interface to generate an interactive voice response data stream.

The utility model relates to interface EMC protection technical field, concretely is a low cost voice protection circuit, a low cost voice protection circuit, including speech interface RJ11, voice protection circuit, speech matching circuit and speech SLIC chip that electrically connect gradually, voice protection circuit includes the primary protection circuit and secondary protection circuit that set gradually along signal transmission direction, the input end of primary protection circuit connects speech interface RJ11's tip end and ring, and the input end of secondary protection circuit is connected to the output end of primary protection circuit. The present application sets up the hierarchical protection framework containing primary protection circuit and secondary protection circuit between speech interface and SLIC chip, and combines the hierarchical configuration of matching resistance group, and forms the low cost, high reliability speech signal protection system.

A method and system for use of context information to disambiguate an automation-configuration command. In an example method, a computing system receives a voice command uttered by a user into a voice-interface device, the voice command describing an Internet-of-Things (IoT) automation. Further, in response to receiving the voice command, the computing system determines, based on context information not specified by the voice command, which of multiple IoT devices should be a subject of an IoT rule that implements the described IoT automation, and provisions the IoT rule with the determined IoT device as the subject of the IoT rule. In example implementations, the context information could be based on network signaling between devices, ambient audio in the user's environment, and / or one or more other factors.

A dialysis machine includes a user interface for providing visual information and / or spoken information to a user. For example, in some implementations, the user interface may be configured to provide visual information related to an action, such as showing the action being partially or fully completed, and a speaker can provide spoken instructions to assist the user in machine set-up, calibration and / or operation. Such instructions can be particularly useful in a home dialysis setting. In some implementations, the speaker can provide spoken alarms that are related to alarm conditions. The spoken alarms may include patient and / or dialysis machine identifying information. The verbosity of the spoken instructions and / or the spoken alarms may be adjustable, and both may be accompanied by visual information displayed by the dialysis machine (e.g. visual alarms, images and / or video).

This invention discloses a speech conversion method, apparatus, system, and medium based on dependency inversion. The method includes: receiving a user's speech conversion request; invoking a preset abstract speech interface according to the speech conversion request; obtaining the currently injected speech SDK in the abstract speech interface; processing the speech conversion request according to the function of the currently injected speech SDK; and returning the corresponding speech conversion result. By pre-setting an abstract speech interface on which both the application's main program and the underlying speech SDK modules depend, the required speech SDK is flexibly injected into the abstract speech interface based on dependency injection to fulfill the user's speech conversion request. This makes the application's main program no longer dependent on the underlying speech SDK, reducing the dependencies between components and eliminating the impact of changes in the underlying modules on the application's operation. This improves the flexibility of speech conversion functionality while also enhancing the stability of mobile application operation.

The present invention proposes a method and system that converts a procedure document into a smart procedure system capable of voice interfacing. The system of the present invention may include a conversion module that converts the BPMN process model of a procedure expressed in XML, automatically generated via NLP-based technology, into a smart procedure system capable of voice interfacing. The conversion module may be configured as a client-server architecture composed of a mobile application client on a smart device and a server system providing backend services.

This invention relates to the field of voice interface control technology, specifically to an AI outbound calling system and its control method and medium, comprising modules for physical link construction, interaction state recognition, flow path control, and intervention and collaborative response. By smoothly connecting analog signal gaps through impedance matching and gain compensation, a synchronous and continuous uplink and downlink voice signal sequence is generated; audio energy jumps are identified to pinpoint call state transition periods; frequency distribution characteristics of AI and human mode switching are analyzed to extract abnormal frequency segments of routing switching; and response synchronization flags are located by combining human intervention trend positioning to ultimately determine the core interactive control result. This invention improves the time consistency and compatibility of audio transmission across hardware terminals, and achieves efficient collaborative response and closed-loop control of interaction states between AI and human modes by accurately identifying flow paths and intervention response locations.

This disclosure describes, in part, context-driven device arbitration techniques to select a speech interface device from multiple speech interface devices to provide a response to a command included in a speech utterance of a user. In some examples, the context-driven arbitration techniques may include executing multiple pipeline instances to analyze audio signals and device metadata received from each of the multiple speech interface devices which detected the speech utterance. A remote speech processing service may execute the multiple pipeline instances and analyze the audio signals and / or metadata, at various stages of the pipeline instances, to determine which speech interface device is to respond to the speech utterance.

This invention belongs to the field of communication noise detection and processing technology, and relates to a method and system for testing low-noise power supplies using waveforms. The method connects the power supply under test (DUT) and a voice gateway, and powers them on in real time. The voice gateway is connected to an external noise-shielded environment via its voice interface. Within this environment, the voice gateway is modulated to a periodic state of comfortable noise or a dial tone, and the sound is played. The playing sound is acquired and processed, and based on the processed sound and the sound signal waveform, the noise level of the DUT is detected in real time. This transforms uncontrollable human-induced noise into an electrical signal measurable by an oscilloscope, accurately determining whether the power supply meets requirements. In other words, by utilizing a clean environment, the unmeasurable power frequency noise on the telephone line is converted into a differential-mode signal measurable by an oscilloscope through a physical sound isolation process, providing testers with an objective experimental result.

Systems and methods for integration of speech processing functionality with organization systems are disclosed. For example, a voice interface application may be created to enable a voice interface functionality for devices associated with an organization. Space identifiers of spaces of the organization may be created and associated with the voice interface application. Devices associated with the space identifiers may be enabled for utilizing the voice interface application and may be set up utilizing wireless network identifiers associated with the spaces and / or the organization.

In a method for human speech processing in an automatic speech recognition (ASR) system, human speech is received at a speech interface of the ASR system, wherein the ASR system comprises embedded componentry for onboard processing of the human speech and cloud-based componentry for remote processing of the human speech. A keyword is identified at the speech interface within a first portion of the human speech. Responsive to identifying the keyword, a second portion of the human speech is analyzed to identify at least one command, the second portion following the first portion. The at least one command is identified within the second portion of the human speech. The at least one command is selectively processed within at least one of the embedded componentry and the cloud-based componentry.

The present invention relates to a soft, flexible, and communicative device for patients with laryngectomy and its method thereof The present invention discloses the fabrication of soft, lightweight, skin-compatible, and reusable device for silent speech interfaces (SSIs). It further comprises the unique feature to recognize speech-related information using surface electromyography (sEMG). sEMG-based speech recognition operates on signals recorded from a set of sEMG sensors that are strategically located on the face (with the help of desired form factor) and measure muscle activity associated with the phonation, resonation, and articulation of speech. sEMG sensors are printed onto strategical locations on soft, stretchable, and biocompatible membrane shaped into a form-factor of a face mask. The silent speech interfaces (SSIs) can also be deployed in acoustically challenging environment or where privacy / confidentially is a desirable such as in defense or military application and it is cost effective with reliable speech recognition accuracy of 96.2%(>95 %).

The utility model provides a testing device and system. The testing device comprises a testing host, a control panel, a telephone set and an off-hook and on-hook assembly, a first communication interface of the test host is in wired connection with a communication interface of the control panel, an input / output interface of the control panel is in wired connection with a control end of the off-hook and on-hook assembly, and an operation end of the off-hook and on-hook assembly is arranged opposite to an insertion spring of the telephone so as to press or release the insertion spring of the telephone under the driving of the off-hook and on-hook assembly; the telephone connection interface of the telephone is connected with the voice interface of the to-be-tested call gateway module through a telephone line, and the second communication interface of the test host is in wired connection with the communication interface of the to-be-tested call gateway module. Therefore, a tester does not need to carry out operations such as connection and hanging up, the test efficiency and the test precision are improved, and the test labor cost is reduced.

To provide a technique related to a voice interface (I / F) control for operating a thermal appliance by voice.SOLUTION: A control device that controls a voice I / F includes an input I / F that receives an input signal from the outside, a computing unit that determines, when the input signal is a voice signal, whether or not the voice signal represents a prepared operation word, and an output I / F that outputs the control signal to the outside. The computing unit receives a voice input signal from the input I / F, and outputs, through the output I / F, a control signal requesting confirmation of execution of the voice command corresponding to a predetermined operation word when the computing unit determines that the signal represents a predetermined operation word. After transmission of the control signal, a second input signal is received via the input I / F, and when it is determined that the second input signal is an acknowledgment signal confirming the execution of the voice command, the computing unit outputs the control signal to instruct the execution of the voice command through the output I / F.SELECTED DRAWING: Figure 9

Techniques for processing a voice initiated request by a web server are presented. The techniques may include receiving, by a web server, request data representing a voice command to a user device, the request data including an identification of a requested webpage; determining, by the web server, that a response to the request data will continue a voice interaction; and providing, by the web server and to the user device, data for a voice enabled webpage associated with the requested webpage, where the data for the voice enabled webpage is configured to invoke a voice interface for the user device.

Each vehicle in the fleet has an automotive assistant and an external voice interface. A person authorized to interact with an automotive assistant of any vehicle from the fleet is detected as being external to the vehicle from the fleet. After it has been determined that the person is indeed authorized, an automotive assistant in the vehicle communicates with the person. It accomplishes this by receiving an utterance of the person or by sending an utterance to the person.

Memory degradation detection and assessment includes capturing human utterances using a voice interface and generating a corpus of human utterances for a user, selecting human utterances from the corpus of human utterances based on a meaning of the human utterances determined by a computer processor through natural language processing. Contextual information for one or more of the corpus of human utterances is determined based on data generated in response to signals sensed by one or more sensing devices operatively coupled with the computer processor. Patterns in the corpus of human utterances are identified based on pattern recognition performed by the computer processor using one or more machine learning models. Changes in memory functioning of the user are identified based on the pattern recognition. The changes identified based on the contextual information are classified as to whether the changes are likely due to a memory impairment of the user.

Digitally assisted mobile assistance system with at least one rolling or flying mobile carrier platform, at least one computing unit, at least one voice interface, at least one audio unit, at least one camera unit and at least one artificial intelligence for interaction with at least one user.

An automotive assistant in an infotainment system of a vehicle includes a hierarchy that receives a top-level query via a speech interface and that provides a top-level response to the query. The hierarchy includes a top-level agent and a first and second domain with a related domain specific query. Both domains are queried using prompts comprising natural language. The top level response is formulated based on first and second domain specific activities.

A method and system for use of context information to disambiguate an automation-configuration command. In an example method, a computing system receives a voice command uttered by a user into a voice-interface device, the voice command describing an Internet-of-Things (IoT) automation. Further, in response to receiving the voice command, the computing system determines, based on context information not specified by the voice command, which of multiple IoT devices should be a subject of an IoT rule that implements the described IoT automation, and provisions the IoT rule with the determined IoT device as the subject of the IoT rule. In example implementations, the context information could be based on network signaling between devices, ambient audio in the user's environment, and / or one or more other factors.

An apparatus for interacting with an occupant in a vehicle includes an infotainment system that has been integrated into the vehicle, an automotive assistant that is configured to execute in the infotainment system, a speech interface that is configured to receive, from the occupant, an original utterance that is to be processed by the automotive assistant, and a classifier that determines that the original utterance has either anaphora or ellipsis. A model that provides a response to the occupant based on a prompt that includes the original utterance and that has been augmented by a function specification selected based at least in part on a rewritten utterance that has been derived from the original utterance.

The invention discloses a dedusting alarm system and method for a steel plant, and the system comprises an industrial control picture module, an alarm triggering module, an alarm information binding module, an alarm traversing module, a voice synthesis module and an alarm broadcasting module, and is used for monitoring the operation parameters of a primary dedusting fan and a secondary dedusting fan; when the parameter exceeds a preset threshold value, generating an alarm record and displaying the alarm record in an alarm column; unconfirmed alarm records in the alarm bar are traversed, and corresponding alarm information content is obtained; and the alarm information content is converted into voice through a voice interface for broadcasting. The voice alarm content directly comprises specific alarm point information, equipment numbers and abnormal types, all related personnel in an area can alert potential risks in advance through voice broadcast of pre-alarm information, extension from post-event alarm to pre-event early warning is achieved, passive response is changed into active defense, and the safety of the system is improved. And especially in a production area with gas risk, the overall safety level is greatly improved.

In various embodiments, a method for deriving at least one parameter (P) for setting a user interface (82) of a vehicle (F) and / or at least one vehicle system (83) is described, comprising the steps of: converting a message (M) received from a user (1) into a query (Q) using a speech interface coupled to the user interface (82); transforming the query (Q) into a vector embedding using a first trained model (3); performing a similarity search in at least one data source (4) to find at least one relevant piece of information (I) therein that satisfies a similarity criterion with respect to the query (Q);Generating a response (R) using a second trained model (7) based on the message (M), which contains at least one relevant piece of information (I) and a current environmental context, which includes at least one piece of vehicle information (F) and / or at least one piece of user information (B); deriving at least one parameter (P) from the generated response (R) to configure the user interface (B) and / or the at least one vehicle system (F).

A configuration system receives, from an endpoint node of a communications network, information about a desired bot configuration. The endpoint node and the configuration system are in the communications network. The configuration system sends a request comprising a system prompt and the received information to a generative model. The configuration system receives a response to the request, the response comprising a plurality of further system prompts for implementing the desired bot configuration. For each of the plurality of further system prompts, the configuration system triggers instantiation of a bot at a node of the communications network. The instantiated bot comprises the further system prompt. The configuration system sends configuration to a voice interface, to configure the voice interface such that a telephony call associated with the endpoint node has at least one of the instantiated bots as a participant on the telephony call.

A replay detector executing in a vehicle prevents a voice interface from taking action in accordance with audio input received from a microphone when the audio input belongs to a first class of audio input, and allows the voice interface to take action in accordance with the audio input when the audio input belongs to a second class of audio input. The audio input belonging to the first class is formed by acoustic signals that have been produced by technology in an environment of the microphone. The audio input belonging to the second class is formed by acoustic signals that have been produced by at least one person in the environment.