Abstract: Methods, computer program products, and systems are presented. The method computer program products, and systems can include, for instance: examining social network data of a user, wherein the social network data specifies social network connections of the user; obtaining from the user voice data defining a vocal utterance request of the user; converting the voice data defining the vocal utterance request of the user into a text based message; subjecting the text based message to natural language processing; determining a response message to the vocal utterance request of the user, wherein the determining the response data to the vocal utterance request of the user is performed in dependence on the examining of the social network data of the user; and presenting the response message to the user.

Abstract: The present document relates to audio source coding systems which make use of a harmonic transposition method for high frequency reconstruction (HFR), as well as to digital effect processors, e.g. exciters, where generation of harmonic distortion add brightness to the processed signal, and to time stretchers where a signal duration is prolonged with maintained spectral content. A system and method configured to generate a time stretched and/or frequency transposed signal from an input signal is described. The system comprises an analysis filterbank configured to provide an analysis subband signal from the input signal; wherein the analysis subband signal comprises a plurality of complex valued analysis samples, each having a phase and a magnitude. Furthermore, the system comprises a subband processing unit configured to determine a synthesis subband signal from the analysis subband signal using a subband transposition factor Q and a subband stretch factor S.

Abstract: In general, techniques are described for quantizing spatial components based on bit allocations determined for psychoacoustic audio coding. A device comprising a memory and one or more processors may perform the techniques. The memory may store a bitstream including an encoded foreground audio signal and a corresponding quantized spatial component. The one or more processors may perform psychoacoustic audio decoding with respect to the encoded foreground audio signal to obtain a foreground audio signal, and determine, when performing the psychoacoustic audio decoding, a first bit allocation for the encoded foreground audio signal. The one or more processors may also determine, based on the first bit allocation, a second bit allocation, and dequantize, based on the second bit allocation, the quantized spatial component to obtain a spatial component. The one or more processors may reconstruct, based on the foreground audio signal and the spatial component, scene-based audio data.

Abstract: An apparatus for generating a decoded two-channel signal, comprising: a parametric decoder for providing parametric data for a second set of second spectral portions and a two-channel identification identifying for a second spectral portion of the second set of second spectral portions either a first two-channel representation for the second spectral portion of the second set of second spectral portions or a second two-channel representation for the second spectral portion of the second set of second spectral portions, the second two-channel representation being different from the first two-channel representation; and a frequency regenerator for regenerating the second spectral portion of the second set of second spectral portions depending on a first spectral portion of a first set of first spectral portions, the parametric data for the second spectral portion of the second set of second spectral portions and the two-channel identification for the second spectral portion of the second set of second spectral

Abstract: A method, computer system, and a computer program product for generating a conversational bot for an application programming interface (API) is provided. The present invention may include parsing an API schema. The present invention may include generating sentences for the conversational bot from the parsed API schema. The present invention may include constructing the conversational bot by training a deep learning model. The present invention may include receiving a natural language expression from a user. The present invention may include determining whether the natural language expression is enough to activate the bot.

Abstract: An apparatus for encoding directional audio coding parameters including diffuseness parameters and direction parameters includes: a parameter calculator for calculating the diffuseness parameters with a first time or frequency resolution and for calculating the direction parameters with a second time or frequency resolution; and a quantizer and encoder processor for generating a quantized and encoded representation of the diffuseness parameters and the direction parameters.

Abstract: A speech signal processing method and apparatus. The method includes preprocessing a speech signal that is in a first frequency band and that is collected by an ear canal speech collector, to obtain a first speech signal; preprocessing a speech signal that is in a second frequency band and that is collected by at least one external speech collector, to obtain an external speech signal, where frequency ranges of the first frequency band and the second frequency band are different; performing correlation processing on the first speech signal and the external speech signal to obtain a second speech signal; and outputting a target speech signal, where the target speech signal includes the first speech signal and the second speech signal.

Abstract: An apparatus for encoding directional audio coding parameters comprising diffuseness parameters and direction parameters having a parameter calculator (100) for calculating the diffuseness parameters with a first time or frequency resolution and for calculating the direction parameters with a second time or frequency resolution; and a quantizer and encoder processor (200) for generating a quantized and encoded representation of the diffuseness parameters and the direction parameters.

Abstract: Systems and methods for item recommendation are described. One or more embodiments of the systems and methods include generating a hidden vector representation for each word of a source document; removing at least one word from the source document based on the hidden vector representation using a summarization network to obtain a summary document; filtering a plurality of candidate documents based on the source document to obtain a plurality of filtered candidate documents; comparing the summary document to each of the filtered candidate documents to obtain a ranking score for each of the filtered candidate documents; and identifying a relevant candidate document from the filtered candidate documents based on the ranking score.

Abstract: Systems are provided for multilingual speech data processing. A language identification module is configured to analyze spoken utterances in an audio stream and to detect at least one language corresponding to the spoken language utterances. The language identification module detects that a first language corresponds to the first portion of the audio stream. A first transcription of the first portion of the audio stream in the first language is generated and stored in a cache. A second transcription of a second portion of the audio stream in the first language is also generated and stored. When the second portion of the audio stream corresponds to a second language, a third transcription is generated in the second language using a second speech recognition engine configured to transcribe spoken language utterances in the second language. Then, the second transcription is replaced with the third transcription in the cache and any displayed instances.

Abstract: A method for using a neural network model for natural language processing (NLP) includes receiving training data associated with a source domain and a target domain; and generating one or more query batches. Each query batch includes one or more source tasks associated with the source domain and one or more target tasks associated with the target domain. For each query batch, class representations are generated for each class in the source domain and the target domain. A query batch loss for the query batch is generated based on the corresponding class representations. An optimization is performed on the neural network model by adjusting its network parameters based on the query batch loss. The optimized neural network model is used to perform one or more new NLP tasks.

Abstract: A plurality of audio datasets associated with captured audio are provided to a plurality of automatic speech recognition engines, wherein each of the automatic speech recognition engines is configured to recognize speech of a first language. Word error rate estimates that comprise at least one word error rate estimate for each of the plurality of audio datasets are determined from outputs of the plurality of automatic speech recognition engines. From the word error rate estimates, audio in the plurality of audio datasets is determined to include speech in a second language.

Abstract: An encoder operable to filter audio signals into a plurality of frequency band components, generate quantized digital components for each band, identify a potential for pre-echo events within the generated quantized digital components, generate an approximate signal by decoding the quantized digital components using inverse pulse code modulation, generate an error signal by comparing the approximate signal with the sampled audio signal, and process the error signal and quantized digital components. The encoder operable to process the error signal by processing delayed audio signals and Q band values, determining the potential for pre-echo events from the Q band values, and determining scale factors and MDCT block sizes for the potential for pre-echo events.

Abstract: A system determines an event location of an event within an indoor environment based on an event sound generated by the event. The system employs time-reversal techniques based on a received event sound to identify the event location as being in the vicinity of one of a plurality of locator devices at locator locations in the environment. The system includes a base array located within the environment that receives an indication that an event has been detected. Upon receiving the event sound, the system generates a time-reversed event sound for each transceiver and transmits via each transceiver the time-reversed event sound for that transceiver. When a locator device receives a time-reversed event sound, the locator device determines whether the event is in the vicinity of that locator location of the locator device and, if so, outputs an indication that the event occurred at that locator location.

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: An information processing method applied to a computation circuit is disclosed. The computation circuit includes a communication circuit and an operation circuit. The method includes controlling, by the computation circuit, the communication circuit to obtain a voice to be identified input by a user; controlling, by the computation circuit, the operation circuit to obtain and call an operation instruction to perform voice identification processing on the voice to be identified to obtain target text information corresponding to the voice to be identified. The operation instruction is a preset instruction for voice identification.

Abstract: Provided are a sound output control apparatus, a sound output control system, a sound output control method, and a program that can appropriately thin out the output of pieces of sound data. A sound data reception section receives a plurality of pieces of sound data transmitted from transmission apparatuses that are different from each other. A selection section selects a portion of the plurality of pieces of sound data on the basis of at least one of a result of a voice activity detection process performed on each of the pieces of sound data or moving averages of volumes of sounds represented by the pieces of sound data. A sound data transmission section outputs the selected portion of the pieces of sound data.

Abstract: Techniques disclosed herein are directed towards streaming keyphrase detection which can be customized to detect one or more particular keyphrases, without requiring retraining of any model(s) for those particular keyphrase(s). Many implementations include processing audio data using a speaker separation model to generate separated audio data which isolates an utterance spoken by a human speaker from one or more additional sounds not spoken by the human speaker, and processing the separated audio data using a text independent speaker identification model to determine whether a verified and/or registered user spoke a spoken utterance captured in the audio data. Various implementations include processing the audio data and/or the separated audio data using an automatic speech recognition model to generate a text representation of the utterance.

Abstract: Controlling a concealment method for a lost audio frame associated with a received audio signal is provided. At least one bin vector of a spectral representation for at least one tone is obtained, wherein the at least one bin vector includes three consecutive bin values for the at least one tone. Whether each of the three consecutive bin values has a complex value or a real value is determined. Responsive to the determination, the three consecutive bin values are processed to estimate a frequency of the at least one tone based on whether each bin value has a complex value or a real value.

Abstract: Systems and methods for enabling voice-based interactions with electronic devices can include a data processing system maintaining a plurality of device action data sets and a respective identifier for each device action data set. The data processing system can receive, from an electronic device, an audio signal representing a voice query and an identifier. The data processing system can identify, using the identifier, a device action data set. The data processing system can identify a device action from device action data set based on content of the audio signal. The data processing system can then identify, from the device action dataset, a command associated with the device action and send the command to the for execution device for execution.