Abstract: A method of enhancing dialog intelligibility in an audio signal, comprising determining a speech confidence score that the audio content includes speech content, determining a music confidence score that the audio content includes music correlated content, in response to the speech confidence score, and applying a user selected gain of selected frequency bands of the audio signal to obtain a dialogue enhanced audio signal. The user selected gain is smoothed by an adaptive smoothing algorithm, an impact of past frames in said smoothing algorithm being determined by a smoothing factor, the smoothing factor being calculated in response to the music confidence score, and having a relatively higher value for content having a relatively higher music confidence score and a relatively lower value for speech content having a relatively lower music confidence score, so as to increase the impact of past frames on the dialogue enhancement of music correlated content.

Abstract: A speech interface device is configured with “hybrid” capabilities, which allows the speech interface device to perform actions in response to user speech, even when the speech interface device is unable to communicate with a remote system over a wide area network (e.g., the Internet). A hybrid request selector of the speech interface device sends audio data representing user speech to both a remote speech processing system and a local speech processing component executing on the speech interface device, and then waits for a response from either or both components. The local speech processing component may start execution based on the audio data and subsequently suspend the execution until further instruction from the hybrid request selector. The hybrid request selector can then determine which response to use, and, depending on which response is chosen, may instruct the local speech processing component to either continue or terminate the suspended execution.

Abstract: A process may include detecting, via a controller, one or more microphones and one or more speakers in an area, measuring audio performance levels of the one or more microphones and the one or more speakers to identify one or more of a noise floor and a reverberation level, identifying an initial room performance rating based on the audio performance levels, applying optimized speaker tuning levels to the one or more speakers and the one or more microphones, measuring, via the one or more microphones, optimized audio performance levels of the one or more speakers based on the applied optimized speaker tuning levels, and generating a report to identify an optimized room performance rating based on the applied optimized speaker tuning.

Abstract: Embodiments provide a method for processing an audio signal, including: performing a cascaded lapped critically sampled transform on two partially overlapping blocks of samples of the audio signal, to obtain sets of subband samples; identifying one or more sets of subband samples that in combination represent the same region of the time-frequency plane; performing time-frequency transforms on the identified one or more sets of subband samples, to obtain one or more time-frequency transformed subband samples, each of which represents the same region in the time-frequency plane; performing a weighted combination of two corresponding sets of subband samples or time-frequency transformed versions thereof, to obtain aliasing reduced subband representations of the audio signal.

Abstract: Methods, apparatus, systems, and computer-readable media are provided for providing context specific schema files that allow an automated assistant to broker human-to-computer dialogs between a user and an application that is separate from the automated assistant. The context specific schema file can provide the automated assistant with sufficient data to be responsive to user queries without necessarily communicating with a remote device, such as a server. Multiple different context specific schema files can be made available to the automated assistant according to a context in which a user is interacting with the automated assistant. In this way, latency otherwise exhibited by the automated assistant can be mitigated by providing the automated assistant with the information needed to respond to a user without continually retrieving the information over a network.

Abstract: A computer mediated communication system includes a plurality of communication devices, a wireless transceiver, a computer system configured to wirelessly communicate with the communication devices via the wireless transceiver and mediate communications among the communication devices. The computer system a memory coupled with a processor.

Abstract: Disclosed is an electronic device including a communication interface, a memory, a microphone, a speaker, a display, a main processor, and a sub-processor activating the main processor by recognizing a wake-up word included in a voice input. The at least one memory stores instructions that, when executed, cause the main processor to receive a first voice input to register the wake-up word, when the first voice input does not include a specified word, to receive a second voice input including a word identical to the first voice input, through the microphone, to generate a wake-up word recognition model for recognizing the wake-up word, and to store the generated wake-up word recognition model in the at least one memory, and when the first voice input includes the specified word, to output information for requesting a third voice input, through the speaker or the display.

Abstract: Implementations set forth herein relate to pre-emptively initializing an automated assistant in a vehicle according to certain indications, in order to reduce latency while also seeking to preserve computational resources. In some implementations, data for effectuating one or more features of an automated assistant can be loaded into memory of a computing device based on vehicle interaction data. For example, the vehicle interaction data can characterize instances in which the user, from within their vehicle, invoked the automated assistant within a threshold period of time of an application completing an operation. Based on the vehicle interaction data, subsequent instances of the operation being completed while the user is in the vehicle can cause data to be loaded into memory in order to pre-emptively prepare the automated assistant to be utilized by the user.

Abstract: A voice wake-up method is provided. The method includes: collecting a first voice signal in an environment in which the first electronic device is located. If audio is being played in the environment when the first voice signal is collected, obtaining, in a wired or wireless communication manner, an audio signal corresponding to the audio, determining a first false wake-up result based on the first voice signal and the audio signal; receiving a second false wake-up result sent by the second electronic device; determining a third false wake-up result based on the first false wake-up result and the second false wake-up result; wherein the third false wake-up result is used to indicate whether a wake-up operation needs to be performed on a to-be-woken-up device in a local area network; sending the third false wake-up result to another electronic device other than the first electronic device in the local area network.

Abstract: In one embodiment, a computer-implemented method for navigating a content item is disclosed. The method includes presenting, via a user interface of a media player, the content item and time-synchronized text pertaining to the content item, receiving a voice command to play a portion of the content item performed by a performer, based on the voice command, using the media player (i) to initiate playback of the content item such that the content item is played at a timestamp associated with the portion of the content item performed by the performer and (ii) to present the time-synchronized text associated with the portion.

Abstract: Real-time modification of audio of humans allows for the audio to be modified so that an expression of a subject human may be changed. Customer service agents may have more successful interactions with customers if they provide vocalization attribute in their speech that are appropriate, such as to provide a particular emotional state. By determining an appropriate vocalization attribute, and any deviation from a customer service agent's current vocalization attribute, a modification to the audio of the customer service agent's speech may be determined and applied. As a result, agents may not have a vocalization attribute that is best suited to successfully resolve a purpose of the interaction, altered to have the customer be presented with the customer service agent's speech having the best-suited vocalization attribute.

Abstract: Audio communication apparatus comprises a set of two or more audio communication nodes; each audio communication node comprising: an audio encoder controlled by encoding parameters to generate encoded audio data to represent a vocal input generated by a user of that audio communication node, the encoded data being agnostic to which user who generated the vocal input; and an audio decoder controlled by decoding parameters to generate a decoded audio signal as a reproduction of a vocal signal generated by a user of another of the audio communication nodes, the decoding parameters being specific to the user of that other of the audio communication nodes.

Abstract: The disclosed technology relates to methods, accent conversion systems, and non-transitory computer readable media for real-time accent conversion. In some examples, a set of phonetic embedding vectors is obtained for phonetic content representing a source accent and obtained from input audio data. A trained machine learning model is applied to the set of phonetic embedding vectors to generate a set of transformed phonetic embedding vectors corresponding to phonetic characteristics of speech data in a target accent. An alignment is determined by maximizing a cosine distance between the set of phonetic embedding vectors and the set of transformed phonetic embedding vectors. The speech data is then aligned to the phonetic content based on the determined alignment to generate output audio data representing the target accent.

Abstract: In one embodiment, a method includes receiving, from a client system of a user, a user input comprising a plurality of n-grams, parsing the user input to identify one or more overall intents, hidden intents, and slots associated with the one or more n-grams, wherein at least one of the hidden intents is non-resolvable for being associated with partial slot information corresponding to an n-gram that has not been resolved to a particular entity identifier, wherein the partial slot information is associated with two more entity identifiers of two or more entities, respectively, sending, to the client system, instructions for prompting the user to select one of the entities to be associated with the non-resolvable hidden intent, resolving the non-resolvable hidden intent based on the entity identifier of the entity selected by the user, and generating a response to the user input based on the resolved hidden intent.

Abstract: Methods, apparatus, systems and articles of manufacture are disclosed for signal identification using a low power watermark. Example apparatus for media identification based on watermarks includes a first processor to determine, in response to receiving a signal, if a first watermark is present in the signal using a first processing technique. The example first processor is further to provoke, in response to the first watermark being present in the signal, a second processing technique on a signal processor. The signal processor is to extract a second watermark in the signal using the second processing technique.

Abstract: A method (600) for decoding an encoded audio signal (102) is described. The encoded audio signal (102) comprises a sequence of frames. Furthermore, the encoded audio signal (102) is indicative of a plurality of different dynamic range control (DRC) profiles for a corresponding plurality of different rendering modes. Different subsets of DRC profiles from the plurality of DRC profiles are comprised within different frames of the sequence of frames, such that two or more frames of the sequence of frames jointly comprise the plurality of DRC profiles.

Abstract: The disclosure provides a method for controlling an intelligent device and an intelligent device. The method comprises: receiving a voice input; determining a service instruction based on the received voice input; determining a target serviced object for which the service instruction is intended; determining a target execution element of the service instruction based on the target serviced object; and controlling the intelligent device to perform an action corresponding to the service instruction based on the target execution element. In addition, the process of determining the target serviced object for which the service instruction is intended may be performed based on an artificial intelligence model.

Abstract: A morphing interface system updates, that is, morphs a display on a client device as a user provides portions of input and additionally provides suggested selections for a user based on the received user input. The system receives a first portion of user input and generates intent suggestions for the user based on the user input. The intent suggestions, which represent predicted likely intents of the user, are provided to the user for selection. The user may select an intent suggestion or may provide additional user input. Based on the user response, the system determines whether an intent is selected or if additional information is needed. When an intent is selected, the interface morphs into an interface to provide predicted entity suggestions for the user to select entity values as inputs to execution of the intent.

Abstract: Systems and methods are provided for determining importance and urgency of a task based on acoustic features of audio input associated with the task. The determining includes classifying the task into one or more classes associated with importance, urgency, and priority of the task. The classification may use a trained machine learning model of acoustic features and embedding for a neural network. The task classifier uses feature acoustics of either or both the foreground and background audio. The feature acoustics include a pitch, a tone, and a volume over a time duration of the audio input. A combination of the acoustic features determines a class associated with the task. The machine learning model includes a regression model of acoustic features over time and a model with embedding for a neural network.

Abstract: A method for optimizing speech recognition includes receiving a first acoustic segment characterizing a hotword detected by a hotword detector in streaming audio captured by a user device, extracting one or more hotword attributes from the first acoustic segment, and adjusting, based on the one or more hotword attributes extracted from the first acoustic segment, one or more speech recognition parameters of an automated speech recognition (ASR) model. After adjusting the speech recognition parameters of the ASR model, the method also includes processing, using the ASR model, a second acoustic segment to generate a speech recognition result. The second acoustic segment characterizes a spoken query/command that follows the first acoustic segment in the streaming audio captured by the user device.

Abstract: A system and method for concurrent multi-path processing of audio signals for automatic speech recognition is presented. Audio information defining a set of audio signals may be obtained (502). The audio signals may convey mixed audio content produced by multiple audio sources. A set of source-specific audio signals may be determined by demixing the mixed audio content produced by the multiple audio sources. Determining the set of source-specific audio signals may comprises providing the set of audio signals to both a first signal processing path and a second signal processing path (504). The first signal processing path may determine a value of a demixing parameter for demixing the mixed audio content (506). The second signal processing path may apply the value of the demixing parameter to the individual audio signals of the set of audio signals (508) to generate the individual source-specific audio signals (510).

Abstract: An electronic apparatus is provided. The electronic apparatus includes an interface configured to receive a first audio signal from a first microphone set and receive a second audio signal from a second microphone set provided at a position different from that of the first microphone set; a processor configured to: obtain a plurality of first sound-source components based on the first audio signal and a plurality of second sound-source components based on the second audio signal; identify a first sound-source component, from among the plurality of first sound-source components, and a second sound-source component, from among the plurality of second sound-source components, that correspond to each other; identify a user command based on the first sound-source component and the second sound-source component; and control an operation corresponding to the user command.

Abstract: A method, system, and computer readable medium for decomposing an audio signal into different isolated sources. The techniques and mechanisms convert an audio signal into K input spectrogram fragments. The fragments are sent into a deep neural network to isolate for different sources. The isolated fragments are then combined to form full isolated source audio signals.

Abstract: A guide sentence generation device includes an acquisition unit that acquires, from a storage unit, staircase information about a staircase existing on a path on which a user moves; and a generation unit that generates a guide sentence for walking on the staircase and a guide sentence for walking after going up or down the staircase based on the staircase information and the path.

Abstract: A method for a soft acceptance of a hotword receives audio data characterizing a soft hotword event detected by a hotword detector in streaming audio captured by a user device. The method also processes the audio data to determine that the audio data corresponds to a query specifying an action to perform on the user device. Without triggering performance of the action on the user device or the other device, the method provides a notification for output from the user device where the notification prompts a user associated with the user device to provide an affirmative input indication in order to trigger performance of the action on the user device or the other device and, when the user fails to provide the affirmative input indication, instructs the user device or the other device to not perform the action specified by the query.

Abstract: In one embodiment, a method includes receiving, from a client system of a user, a user input comprising a plurality of n-grams, parsing the user input to identify one or more overall intents, hidden intents, and slots associated with the one or more n-grams, wherein at least one of the hidden intents is non-resolvable for being associated with partial slot information corresponding to an n-gram that has not been resolved to a particular entity identifier, wherein the partial slot information is associated with two more entity identifiers of two or more entities, respectively, sending, to the client system, instructions for prompting the user to select one of the entities to be associated with the non-resolvable hidden intent, resolving the non-resolvable hidden intent based on the entity identifier of the entity selected by the user, and generating a response to the user input based on the resolved hidden intent.

Abstract: The present disclosure discloses a voice conversion method. The method includes: obtaining a to-be-converted voice, and extracting acoustic features of the to-be-converted voice; obtaining a source vector corresponding to the to-be-converted voice from a source vector pool, and selecting a target vector corresponding to the target voice from the target vector pool; obtaining acoustic features of the target voice output by the voice conversion model by using the acoustic features of the to-be-converted voice, the source vector corresponding to the to-be-converted voice, and the target vector corresponding to the target voice as an input of the voice conversion model; and obtaining the target voice by converting the acoustic features of the target voice using a vocoder. In addition, a voice conversion apparatus and a storage medium are also provided.

Abstract: A system configured to improve a voice quality during a communication session by performing bandwidth extension on a narrowband speech signal to generate a wideband speech signal with higher audio quality. For example, a system can extend a speech bandwidth from a narrowband signal having a first bandwidth (e.g., 4 kHz) to a wideband signal having a second bandwidth (e.g., 8 kHz or higher). To perform bandwidth extension, the system may include cascaded neural networks, such as two or more sub-pixel convolutional neural networks (CNNs) connected in series. In some examples, a first sub-pixel CNN may extend the speech bandwidth from 4 kHz to 6 kHz and a second sub-pixel CNN may extend the speech bandwidth from 6 kHz to 8 kHz. Alternatively, the system may use three or more cascaded neural networks and/or may extend the speech bandwidth above 8 kHz without departing from the disclosure.

Abstract: Systems and methods for voice authentication are disclosed. In an embodiment, a computer system may determine that a user is eligible for establishing a voice authentication capability for a user account during a real-time audio communication between a user device corresponding to the user and a communication system associated with an electronic service provider. The computer system may enhance a recording quality of a portion of the real-time audio communication and record a voice sample for the portion of the real-time audio communication at the enhanced recording quality. The computer system may generate a voiceprint based on the voice sample and enable the voice authentication capability such that the user can be authenticated by voice in future audio communications with the communication system in a minimally intrusive fashion where normal conversation can be used to capture voice samples which can be compared to the voiceprint to authenticate the user.

Abstract: The present disclosure provides method and apparatus for responding in a voice conversation by an electronic conversational agent. A voice input may be received in an audio upstream. In response to the voice input, a primary response and at least one supplementary response may be generated. A primary voice output may be generated based on the primary response. At least one supplementary voice output may be generated based on the at least one supplementary response. The primary voice output and the at least one supplementary voice output may be provided in an audio downstream, wherein the at least one supplementary voice output is provided during a time period adjacent to the primary voice output in the audio downstream.

Abstract: Methods of and systems for forming an image of a subterranean region of interest are disclosed. The method includes obtaining an observed seismic dataset and a seismic velocity model for the subterranean region of interest and generating a simulated seismic dataset based on the seismic velocity model and the source and receiver geometry of the observed seismic dataset. The method also includes forming a plurality of time-windowed trace pairs from the simulated and the observed seismic datasets, and forming an objective function based on a penalty function and a cross-correlation between the members of each pair. The method further includes determining a seismic velocity increment based on the extremum of the objective function and forming an updated seismic velocity model by combining the seismic velocity increment and the seismic velocity model, and forming the image of the subterranean region of interest based on the updated seismic velocity model.

Abstract: A method of generating content in association with an information search and retrieval system. It begins by receiving a query from a user. The query is semantically-searched to identify a context. A conversation history between the user and the system is identified. An enriched query is then generated by associating to the query both the context and at least a portion of the conversation history. The enriched query is then evaluated/processed by a generative-AI. In response, information associated with the enriched query is received from the generative-AI. A response to the query is then generated using the information, e.g., by passing the information back to the user, by modifying (e.g., editing or supplementing) the information to generate modified information and passing the modified information back to the user, or by dismissing the information. If sensitive information is identified in the utterance, it is masked prior to generating the enriched query.

Abstract: A method for determining a current viewing direction of a user of a pair of data glasses having a virtual retina scan display. The method includes at least the method steps: projecting at least substantially parallel infrared laser beams onto an eye of a user of the data glasses, acquiring two-dimensional images from the infrared laser beams reflected back by the eye of the user, and determining pupil contours in the acquired two-dimensional images. The instantaneous viewing direction of the user of the data glasses is ascertained from a comparison of an instantaneous elliptical shape of the pupil contour with an elliptical shape of a reference pupil contour.

Abstract: An audio processing method applied to a first terminal is described, and includes: in response to receiving of audio data input by a user at the first terminal, and determination that a voice change function is turned on, determining change parameters; and based on the change parameters, performing change processing on the audio data.

Abstract: A method includes, while displaying a first set of text content via a display device, determining an engagement value that characterizes a level of user engagement with respect to the first set of text content. The method includes, in accordance with a determination that the engagement value satisfies a threshold, replacing the first set of text content with a second set of text content via the display device. The first set of text content is different from the second set of text content. The method includes in accordance with a determination that the engagement value does not satisfy the threshold, maintaining display of the first set of text content via the display device.

Abstract: Implementations set forth herein relate to pre-emptively initializing an automated assistant in a vehicle according to certain indications, in order to reduce latency while also seeking to preserve computational resources. In some implementations, data for effectuating one or more features of an automated assistant can be loaded into memory of a computing device based on vehicle interaction data. For example, the vehicle interaction data can characterize instances in which the user, from within their vehicle, invoked the automated assistant within a threshold period of time of an application completing an operation. Based on the vehicle interaction data, subsequent instances of the operation being completed while the user is in the vehicle can cause data to be loaded into memory in order to pre-emptively prepare the automated assistant to be utilized by the user.

Abstract: Embodiments of the invention are directed to a system, method, or computer program product for multimodal and distributed database system structured for dynamic latency reduction. In this regard, the invention comprises a unified data layer structured to map a plurality of data storage mechanisms to a common abstraction and a query engine structured for heterogenous domain based data extraction without requiring input of schema-based queries. In some embodiments, the invention comprises determining (i) one or more data components and (ii) one or more associated data domains associated with the first domain-based query by parsing the user input based on derived metadata from data dictionaries associated with a unified data layer system component. Moreover, the invention is configured to extract stored data from each of a plurality of databases based on the associated one or more data domains.

Abstract: Methods, apparatus, systems, and computer-readable media are provided for providing context specific schema files that allow an automated assistant to broker human-to-computer dialogs between a user and an application that is separate from the automated assistant. The context specific schema file can provide the automated assistant with sufficient data to be responsive to user queries without necessarily communicating with a remote device, such as a server. Multiple different context specific schema files can be made available to the automated assistant according to a context in which a user is interacting with the automated assistant. In this way, latency otherwise exhibited by the automated assistant can be mitigated by providing the automated assistant with the information needed to respond to a user without continually retrieving the information over a network.

Abstract: A Natural Language Command system, which receives Natural Language Commands either over a voice system or over a text system. The commands are associated with the session, and thus can be modified.

Abstract: An image processing method determines a geometric transform of a suspect image by efficiently evaluating a large number of geometric transform candidates in environments with limited processing resources. Processing resources are conserved by using complementary methods for determining a geometric transform of an embedded signal. One method excels at higher geometric distortion, and specifically, distortion caused by greater tilt angle of a camera. Another method excels at lower geometric distortion, for weaker signals. Together, the methods provide a more reliable detector of an embedded data signal in image across a larger range of distortion while making efficient use of limited processing resources in mobile devices.

Abstract: A virtual assistant system comprising an interface configured to receive user input and provide a response to the user and a processor configured to run machine executable code. A memory storing non-transitory machine executable code configured to process the user input to generate two or more primary interpretations and one or more secondary interpretations based on one or more of the two or more primary interpretations. The code is also configured to process the primary interpretations and alternative interpretations to generate results which lead to two or more terminal states and then score the two or more terminal states to rank the two or more terminal states such that a top ranked terminal state is the top result, which is presented to the user. A transceiver may communicate over a network to a second device configured to assist the virtual assistant system in generating the top result for the user.

Abstract: Techniques that facilitate a dialogue system based on contextual information are provided. In one example, a system includes a contextual information component and a dialogue routing component. The contextual information component determines contextual information associated with a user identity based on a statement related to communication information received by a computing device associated with the user identity. The dialogue routing component generates a path traversal for a dialogue system based on the contextual information to facilitate generation of a response to the statement by the dialogue system.

Abstract: A method, system, and computer readable medium for decomposing an audio signal into different isolated sources. The techniques and mechanisms convert an audio signal into K input spectrogram fragments. The fragments are sent into a deep neural network to isolate for different sources. The isolated fragments are then combined to form full isolated source audio signals.

Abstract: An analysis filter bank corresponding to multiple sub-bands, which performs frequency-division filtering on an input signal to generate multiple sub-band signals, the analysis filter bank comprising: a sub-band response pre-compensator which performs a linear filtering on the input signal to generate a response pre-compensated signal, multiple sub-filters with different central frequencies, which perform complex-type first-order infinite impulse response filtering respectively on the response pre-compensated signal to generate multiple sub-filter signals, and multiple binomially-combining and rotating devices based on a set of binomial weights, each of which performs a weighted summation on at least two of the sub-filter signals with the set of binomial weights, and rotates a weighted-summation result with a rotating phase according to a corresponding sub-band central frequency to generate one of the sub-band signals, wherein the at least two of the sub-filter signals are generated by at least two of the sub-

Abstract: An apparatus includes a receiver and a decoder. The receiver is configured to receive a bitstream that includes a first frame and a second frame. The first frame includes a first portion of a mid channel and a first quantized stereo parameter. The second frame includes a second portion of the mid channel and a second quantized stereo parameter. The decoder is configured to generate a first portion of a channel based on the first portion of the mid channel and the first quantized stereo parameter. The decoder is configured to, in response to the second frame being unavailable for decoding operations, estimate the second quantized stereo parameter based on stereo parameters of one or more preceding frames and generate a second portion of the channel based on the estimated second quantized stereo parameter. The second portion of the channel corresponds to a decoded version of the second frame.

Abstract: A method performed by a head-mounted device can include, based on a front-facing camera included in the head-mounted device capturing an image of a wearable device, configuring the head-mounted device to receive input via the wearable device, determining that a gesture received by the wearable device includes a request to launch an application, and, in response to determining that the gesture includes the request to launch the application, launching the application.

Abstract: A system of reducing transmissions of packetized data in a voice activated data packet based computer network environment is provided. A natural language processor component can parse an input audio signal to identify a request and a trigger keyword. Based on the input audio signal, a direct action application programming interface can generate a first action data structure, and a content selector component can select a content item. An interface management component can identify candidate interfaces and determine if prior instances of the packetized data was transmitted to the candidate interfaces. The interface management component can prevent the transmission of the packetized data if determined to be redundant, such as having previously received the data, and instead transmit it to a separate client device of a different device type.

Abstract: Generally discussed herein are devices, systems, and methods for custom wake word selection assistance. A method can include receiving, at a device, data indicating a custom wake word provided by a user, determining one or more characteristics of the custom wake word, determining that use of the custom wake word will cause more than a threshold rate of false detections based on the characteristics, rejecting the custom wake word as the wake word for accessing a personal assistant in response to determining that use of the custom wake word will cause more than a threshold rate of false detections, and setting the custom wake word as the wake word in response to determining that use of the custom wake word will not cause more than the threshold rate of false detections.

Abstract: An intelligence-driven virtual assistant for automated documentation of new ideas is provided. During a brainstorming session, one or more user participants may discuss and identify one or more ideas. Such ideas may be tracked, catalogued, analyzed, developed, and further expanded upon through use of an intelligence-driven virtual assistant. Such virtual assistant may capture user input data embodying one or more new ideas and intelligently process the same in accordance with creativity tool workflows. Such workflows may further guide development and expansion upon a given idea, while continuing to document, analyze, and identify further aspects to develop and expand.

Abstract: Embodiments of the invention provide a method, system and computer program product for online ordering using conversational interfaces. In an embodiment of the invention, the method includes storing customer information corresponding to a customer and responsive to receiving a message with text or speech and an image from the customer, identifying an intent type from the text or speech using Natural Language Understanding, identifying a product or service from the image using image classification techniques and transmitting a product detail message to the customer with the product or service and corresponding pricing using Natural Language Generation. The method further includes responsive to receiving an affirmative message from the customer in response to the product detail message identified as affirmative using Natural Language Understanding, automatically completing a purchase of the product or service with the customer information and transmitting a receipt message to the customer with an order receipt.