Abstract: Systems and methods for processing emergency communications are provided. A system may receive an emergency communication initiated by an emergency communicator. The system may detect a data field action in response to an emergency receiver entering a data input based on the emergency communication. The system may capture a timestamp of when the data field action occurred. The system may generate a communication snippet based on the action timestamp and a snippet length. The communication snippet may be configured to provide context from the emergency communication to the data input. The system may transmit the communication snippet and the data input to an emergency responder.

Abstract: Systems and methods for processing emergency communications are provided. A system may receive an emergency communication initiated by an emergency communicator. The system may detect a data field action in response to an emergency receiver entering a data input based on the emergency communication. The system may capture a timestamp of when the data field action occurred. The system may generate a communication snippet based on the action timestamp and a snippet length. The communication snippet may be configured to provide context from the emergency communication to the data input. The system may transmit the communication snippet and the data input to an emergency responder.

Abstract: A communication system with a noise cancellation (NC) assembly providing adaptive or dynamic noise cancellation. The NC assembly includes a localizer module determining, during a communication session (active speaking or during idle times), a location of the active talker. The NC assembly includes a beam generator forming a beam in the determined direction of the active talker to enhance the active talker speech. Once the NC assembly has determined the position of the active talker, the NC assembly assigns a microphone of the microphone array or generated beam in that active direction to be the “active signal” source. The NC assembly assigns a second microphone or beam to be the noise source for NC purposes, and this source may be selected to be in acoustic shadow of the first microphone used as the active signal source or may be the farthest away in its position from the active talker's position.

Abstract: A method, system and computer-readable storage medium for performing a cognitive information processing operation.

Abstract: In one aspect, a computer-implemented method includes receiving signals corresponding to wavelengths of light detected by an optical sensor placed in proximity to a patient's body, and for each received signal: separating the signal into an AC signal and a DC signal; separating the AC signal into component signals; analyzing the component signals through a fractional phase transformation to identify a desired component signal and harmonic signals associated with the desired component signal; smoothing the desired component signal, the harmonic signals, and the DC signal; and combining the smoothed desired component signal, the smoothed harmonic signals, and the smoothed DC signal to generate a modulation signal. A modulation ratio signal is generated based on the modulation signals derived from the signals, and a peripheral oxygen saturation (SpO2) of the patient's body is determined based on the modulation ratio signal.

Abstract: A device configured to process data comprised in data messages passing on message buses of a rolling stock comprises: a universal input interface receiving data messages complying with the three following physical layers: RS232; RS485; CAN. From the message buses, the data messages comprise data; a processing engine receiving a remote requested configuration comprising one or more processing rules; a standardizing unit decoding the data messages into standardized data streams in function of the remote requested configuration; and wherein the processing engine further applies one or more of the one or more processing rules of the standardized data streams in function of the remote requested configuration.

Abstract: Computerized systems are provided for determining an identity of one or more users that use a same audio source, such as a microphone. The identity of one or more users that use a same audio source can based on generating a list of participant candidates who are likely to participate in an associated event, such as a meeting. For instance, embodiments can generate one or more network graphs of a meeting invitee any only voice input samples of the meeting invitee's N closest connections are compared to an utterance to determine the identity of the user associated with the utterance. One or more indicators that identify the users who are using the same audio source, as well as additional information or metadata associated with the identified user can be caused to be presented.

Abstract: An apparatus for spatial audio signal encoding, the apparatus comprising at least one processor and at least one memory including a computer program code, the at least one memory and the computer program code configured to, with the at least one processor, cause the apparatus at least to: determine, for two or more audio signals, at least one spatial audio parameter for providing spatial audio reproduction, the at least one spatial audio parameter comprising a direction parameter with an elevation and an azimuth component; define a spherical grid generated by covering a sphere with smaller spheres, wherein the centres of the smaller spheres define points of the spherical grid; and convert the elevation and azimuth component of the direction parameter to an index value based on the defined spherical grid.

Abstract: [Object] To achieve displaying of a text in a more flexible and highly readable manner in accordance with a situation. [Solution] According to the present disclosure, an information processing device is provided. The information processing device includes a calculator that calculates, on the basis of context data to be entered, a recognition difficulty score used for display control of a target text. An information processing method is further provided. The information processing method includes allowing a processor to calculate, on the basis of context data to be entered, a recognition difficulty score used for display control of a target text.

Abstract: There is provided methods and apparatuses for decoding and encoding of audio signals. In particular, a method for decoding includes receiving a waveform-coded signal having a spectral content corresponding to a subset of the frequency range above a cross-over frequency. The waveform-coded signal is interleaved with a parametric high frequency reconstruction of the audio signal above the cross-over frequency. In this way an improved reconstruction of the high frequency bands of the audio signal is achieved.

Abstract: An interactive music audition method, apparatus and terminal are provided. The method includes: generating audition inquiry information according to audition requirement information, wherein the audition inquiry information includes a plurality of audition music options associated with the audition requirement information; generating a plurality of audition inquiry voices corresponding to the respective audition music options based on the audition inquiry information, and playing the generated audition inquiry voices; acquiring music selection information for the generated audition inquiry voices; and playing audition music according to the music selection information. Not only the interaction experience between a user and a smart device is improved, but also the accuracy of mining a user's interest is increased.

Abstract: A touch screen controller disclosed herein includes a circuit configured to generate a digital touch voltage comprises of samples, at a base sampling rate. The touch screen controller also includes a digital processing unit configured to analyze a first subset of samples of the digital touch voltage samples to determine noise content thereof, the first subset of samples corresponding to samples at a first investigated sampling rate that is a first function of the base sampling rate. The digital processing unit is also configured to analyze a second subset of samples of the digital touch voltage to determine noise content thereof, with the second subset of samples corresponding to samples at a second investigated sampling rate that is a second function of the base sampling rate, and determine a preferred sampling rate from among the first and second investigated sampling rates as a function of determined noise content thereof.

Abstract: An apparatus for encoding an audio signal includes: a converter for converting the audio signal into a spectral representation; a scale parameter calculator for calculating a first set of scale parameters from the spectral representation: a downsampler for downsampling the first set of scale parameters to obtain a second set of scale parameters, a second number of scale parameters in the second set of scale parameters being lower than a first number of scale parameters in the first set of scale parameters; a scale parameter encoder for generating an encoded representation of the second set of scale parameters; a spectral processor for processing the spectral representation using a third set of scale parameters, the third set of scale parameters having a third number of scale parameters being greater than the second number of scale parameters, the spectral processor being configured to use the first set of scale parameters or to derive the third set of scale parameters from the second set of scale parameters o

Abstract: A sound discriminating method comprises sensing a sound signal; changing the sensed sound signal into an electrical signal; and determining whether the electrical signal is a predetermined sound by analyzing the electrical signal.

Abstract: A sound collection direction is decided based upon an area of an object in a captured image obtained by image capturing of a periphery and a sound collection target position input as a position of a sound collection target. A noise direction is decided based upon an arrangement of the object in the captured image. A sound collected from the periphery is separated into a sound in the sound collection direction and a sound in the noise direction, and noise canceling on the sound in the sound collection direction is performed using the sound in the noise direction.

Abstract: A method of training a non-verbal sound class detection machine learning system, the non-verbal sound class detection machine learning system comprising a machine learning model configured to: receive data for each frame of a sequence of frames of audio data obtained from an audio signal; for each frame of the sequence of frames: process the data for multiple frames; and output data for at least one sound class score representative of a degree of affiliation of the frame with at least one sound class of a plurality of sound classes, wherein the plurality of sound classes comprises: one or more target sound classes; and a non-target sound class representative of an absence of each of the one or more target sound classes; wherein the method comprises: training the machine learning model using a loss function.

Abstract: An encoding method, a decoding method, an encoding apparatus, a decoding apparatus, a transmitter, a receiver, and a communications system, where the encoding method includes dividing a to-be-encoded time-domain signal into a low band signal and a high band signal, performing encoding on the low band signal to obtain a low frequency encoding parameter, performing encoding on the high band signal to obtain a high frequency encoding parameter, obtaining a synthesized high band signal; performing short-time post-filtering processing on the synthesized high band signal to obtain a short-time filtering signal, and calculating a high frequency gain based on the high band signal and the short-time filtering signal.

Abstract: A personalized news service provides personalized news programs for its users by generating personalized combinations of audible versions of news stories derived from text-based based versions of the news stories. The audible versions may be generated from the text-based version by a text-to-speech system, or may by recording a person reading aloud the text-based version. To acquire recordings, the personalized news service can make a determination that a particular news story has a threshold extent of popularity. The news service can then transmit a request to a remote recording station for a recording of a verbal reading of the particular news story. The news service can then receive the requested recording from the remote recording station.

Abstract: A device includes a receiver and a decoder. The receiver is configured to receive one or more upmix parameters, one or more inter-channel bandwidth extension parameters, one or more inter-channel prediction gain parameters, and an encoded audio signal. The encoded audio signal includes an encoded mid signal. The decoder is configured to generate a synthesized mid signal based on the encoded mid signal. The decoder is also configured to generate a synthesized side signal based on the synthesized mid signal and the one or more inter-channel prediction gain parameters. The decoder is further configured to generate one or more output signals based on the synthesized mid signal, the synthesized side signal, the one or more upmix parameters, and the one or more inter-channel bandwidth extension parameters.

Abstract: A smart nasometer according to an embodiment of the present invention includes: a hardware unit worn on a head of a user for measuring nasal and oral sounds and providing feedback for the user; and a computational unit for receiving and processing speech signals of the nasal and oral sounds measured by the hardware unit, wherein the hardware unit includes: a microphone unit for separately measuring the nasal and oral sounds in a non-touched state of the user's philtrum, wherein the computational unit includes: a nasalance adjustment unit for adjusting a nasalance of the nasal and oral sounds measured by the microphone unit.

Abstract: A method of regenerating wideband speech from narrowband speech, the method comprising: receiving samples of a narrowband speech signal having a first range of frequencies; identifying, based on a characteristic of the narrowband speech signal, frequencies in the first range of frequencies to translate into a target band of a regenerated speech signal; modulating the identified frequencies in the first range of frequencies of the received samples of the narrowband speech signal with a modulation signal, the modulation signal having a modulating frequency adapted to upshift the identified frequencies in the first range of frequencies into the target band; filtering the modulated samples, using a target band filter, to form the regenerated speech signal in the target band; and combining the narrowband speech signal with the regenerated speech signal to produce a new wideband speech signal.

Abstract: A speech signal processing method of a user terminal includes: receiving a speech signal, detecting a personalized information section including personal information in the speech signal, performing data processing on the personalized information section of the speech signal by using a personalized model generated based on the personal information, and receiving, from a server, a result of the data processing performed by the server on a general information section of the speech signal that is different than the personalized information section of the speech signal.

Abstract: This invention discloses a speech analysis/synthesis method and a simplified form of such a method. Based on a harmonic model, the present method decomposes the parameters of the harmonic model into glottal source characteristics and vocal tract characteristics in its analysis stage and recombines the glottal source and vocal tract characteristics into harmonic model parameters in its synthesis stage.

Abstract: A pitch extraction device includes a processor configured to perform a process including: dividing a first bit stream in encoded data into a plurality of sections each having a prescribed section length, the encoded data being obtained by performing entropy encoding on a residual signal calculated by performing linear prediction analysis on a sound signal; allocating a first value or a second value to each of the plurality of sections in the first bit stream in accordance with a bit value in each of the plurality of sections; generating a second bit stream obtained by re-encoding the first bit stream according to the first value and the second value that have been allocated to each of the plurality of sections in the first bit stream; and calculating a fundamental frequency of the sound signal in accordance with an autocorrelation of the second bit stream.

Abstract: A speech/audio coding apparatus includes a receiver that receives a time-domain speech input signal. The apparatus also includes a processor that transforms a time-domain speech input signal into a frequency-domain spectrum, and divides a frequency region of the spectrum in an extended band into a plurality of bands. The processor sets a limited band for each divided band in the current frame, a width of the limited band in the current frame being narrower than the divided band and the limited band including a first frequency. The processor further encodes the spectrum in the limited band within each divided band in the current frame, wherein the width of the limited band is predetermined and is set to 31.

Abstract: The present disclosure discloses a method including: performing a silence detection on a speech to be decoded; cutting the speech to be decoded off to obtain a target speech if detecting that the speech to be detected is a silent speech; resetting tail features of the target speech with preset tail features of silent frames; and performing a CTC decoding process on the target speech reset. In embodiments, when a large number of blank frames are carried in the speech to be decoded, the speech to be decoded is cut off, and the tail features of the target speech is placed with the tail features of the silent frames such that there may be one CTC peak when the CTC decoding process is performed on the tail features of the target speech. Therefore, a last word of text content may be displayed rapidly on a screen.

Abstract: An image data transfer system includes a receiver and a transmitter configured to sequentially receive compressed image data and sequentially transmit transmission data corresponding to the compressed image data to the receiver. The transmitter is configured to, in transmitting a specific transmission data, perform data comparison of bits of a compressed image body data of a specific compressed image data with bits of a previous transmission data transmitted over signal lines allocated to the compressed image body data, incorporate the compressed image body data of the specific compressed image data or the bit-inverted data corresponding thereto into the specific transmission data, in response to the result of the data comparison, and incorporate the compression code of the specific compressed image data into the specific transmission data independently of the result of the data comparison.

Abstract: A speech/audio encoding device for selectively allocating bits for higher precision encoding. The speech/audio encoding device receives a time-domain speech/audio input signal, transforms the speech/audio input signal into a frequency domain, and quantizes an energy envelope corresponding to an energy level for a frequency spectrum of the speech/audio input signal. The speech/audio encoding device further groups quantized energy envelopes into a plurality of groups, determines a perceptual significant group including one or more significant bands and a local-peak frequency, and allocates bits to a plurality of subbands corresponding to the grouped quantized energy envelopes, in which each of the subbands is obtained by splitting the frequency spectrum of the speech/audio input signal. The speech/audio encoding device encodes the frequency spectrum using the bits allocated to the subbands.

Abstract: A device for noise estimation comprises a first microphone capturing a nominal speech signal, and a second microphone capturing a nominal noise signal. A generalized sidelobe canceller of the device applies spatial noise reduction, and comprises a blocking matrix filter to adaptively process the nominal speech signal to produce a speech cancellation signal, a node for subtracting the speech cancellation signal from the nominal noise signal to produce a noise reference signal, a noise cancellation filter to adaptively filter the noise reference signal to produce a noise cancellation signal; and a node for subtracting the noise cancellation signal from the nominal speech signal to produce a speech reference signal.

Abstract: The present invention provides a diagram building system adapted for processing a signal with a time period. The diagram building system comprises a inputting device for receiving the signal; a computing device, dividing the signal into a plurality of window scales according to one of time interval scales; decomposing the window scales via HHT algorithm to generate a plurality of quantized windows according to different components; then, calculating the value of quantized windows with the same single-frequency component through a quantifying function to generate a plurality of specific frequency values; an outputting device, sequentially arranging the specific frequency values according to the time interval scales and the single-frequency components to form a visual diagram.

Abstract: Example embodiments disclosed herein relate to filter coefficient updating in time domain filtering. A method of processing an audio signal is disclosed. The method includes obtaining a predetermined number of target gains for a first portion of the audio signal by analyzing the first portion of the audio signal. Each of the target gains is corresponding to a linear subband of the audio signal. The method also includes determining a filter coefficients for time domain filtering the first portion of the audio signal so as to approximate a frequency response given by the target gains. The filter coefficients are determined by iteratively selecting at least one target gain from the target gains and updating the filter coefficient based on the selected at least one target gain. Corresponding system and computer program product for processing an audio signal are also disclosed.

Abstract: Methods and systems are provided for customizing an action. In some implementations, voice input is received from a user and a context is determined from the voice input. Potential contextual data is identified based on the context and the voice input. A level of confidence is determined for an association of the potential contextual data and the context. An action is performed based on the voice input, the potential contextual data, and the level of confidence. The potential contextual data is used to customize the action.

Abstract: A speech/audio decoding apparatus is provided that includes a receiver that receives encoded data including a limited-band mode flag, and a memory that stores information on a position of a maximum amplitude spectrum frequency of a previous frame in a divided band. The speech/audio decoding apparatus also includes a processor that identifies whether a decoding band is encoded using a limited-band mode based on the decoded limited-band mode flag. Additionally, the processor decodes the spectrum in a limited band within each of the divided bands in a current frame using the stored information. Furthermore, the limited-band mode is set at an encoder side, when a difference between a first frequency with a first maximum amplitude in a spectrum of the divided band in a preceding frame and a second frequency with a second maximum amplitude in a spectrum of the divided band in the current frame is below a threshold.

Abstract: Provided are a method and apparatus for encoding an audio signal and a method and apparatus for decoding an audio signal, in which errors generated during encoding and decoding of the audio signal are reduced to enhance the audio quality of a reconstructed audio signal. The method of encoding the audio signal includes detecting a pitch of the audio signal, determining a filter coefficient based on the detected pitch, performing second filtering on the audio signal, based on the determined filter coefficient; and encoding an audio signal resulting from the second filtering.

Abstract: An apparatus for generating a decoded two-channel signal includes: an audio processor for decoding an encoded two-channel signal to obtain a first set of first spectral portions; a parametric decoder for providing parametric data for a second set of second spectral portions and a two-channel identification identifying either a first or a second different two-channel representation for the second spectral portions; and a frequency regenerator for regenerating a second spectral portion depending on a first spectral portion of the first set of first spectral portions, the parametric data for the second portion and the two-channel identification for the second portion.

Abstract: A method implemented in a fronthaul communication unit, comprising applying, via a processor of the fronthaul communication unit, a plurality of first frequency-domain windowing (FDW) functions on a plurality of first communication channel signals to produce a plurality of first windowed signals, aggregating, via the processor, the plurality of first windowed signals to produce a first aggregated signal, and transmitting, via a frontend of the fronthaul communication unit, the first aggregated signal to a corresponding fronthaul communication unit over a fronthaul communication link to facilitate fronthaul communication.

Abstract: A frame error concealment (FEC) method is provided. The method includes: selecting an FEC mode based on states of a current frame and a previous frame of the current frame in a time domain signal generated after time-frequency inverse transform processing; and performing corresponding time domain error concealment processing on the current frame based on the selected FEC mode, wherein the current frame is an error frame or the current frame is a normal frame when the previous frame is an error frame.

Abstract: A meeting summarization method, system, and computer program product, include recording meeting audio of a meeting, capturing notes including a time stamp from each of a plurality of users associated with the meeting, synchronizing the recorded meeting audio of the meeting and each of the notes of each of the plurality of users based on a correlation between the time stamp, and analyzing the synchronized meeting audio and notes to determine highlights of the meeting based on a co-occurrence of notes between the plurality of users.

Abstract: Methods and systems are provided for customizing an action. In some implementations, voice input is received from a user and a context is determined from the voice input. Potential contextual data is identified based on the context and the voice input. A level of confidence is determined for an association of the potential contextual data and the context. An action is performed based on the voice input, the potential contextual data, and the level of confidence. The potential contextual data is used to customize the action.

Abstract: Embodiments of the present application proposes a frequency envelope vector quantization method and apparatus, where the method includes: dividing N frequency envelopes in one frame into N1 vectors; quantizing a first vector in the N1 vectors by using a first codebook, to obtain a code word corresponding to the quantized first vector, where the first codebook is divided into 2B1 portions; determining, according to the code word corresponding to the quantized first vector; determining a second codebook according to the codebook of the ith portion; and quantizing a second vector in the N1 vectors based on the second codebook. In the embodiments of the present application, vector quantization can be performed on frequency envelope vectors by using a codebook with a smaller quantity of bits. Therefore, complexity of vector quantization can be reduced, and an effect of vector quantization can also be ensured.

Abstract: Present disclosure discloses a method and an apparatus for encoding a stereo phase parameter, which relate to the field of information technologies and can improve an effect of stereo audio phase information. The method includes: first, acquiring a global stereo phase parameter of a current frame; then, determining a value of the global stereo phase parameter of the current frame, and adjusting the value of the global stereo phase parameter of the current frame according to a determining result of the value of the global stereo phase parameter of the current frame; and finally, encoding an adjusted value of the global stereo phase parameter of the current frame. The embodiments of the present disclosure are applicable to recovering stereo phase information.

Abstract: The present invention relates to an apparatus and method for transmitting/receiving streaming data using multiple paths, in which the streaming data is smoothly reproduced without being interrupted, and more particularly, to an apparatus and method for transmitting/receiving streaming data using multiple paths, in which exchange of the streaming data is performed in real-time using the multiple paths regardless of obstacles. The method for transmitting streaming data using multiple paths includes managing and maintaining a path list including sequence information about a transmission path capable of transmitting data, framing the streaming data, and transmitting the framed streaming data via the transmission path according to the sequence information.

Abstract: An apparatus and a method to encode and decode a speech signal using an encoding mode are provided. An encoding apparatus may select an encoding mode of a frame included in an input speech signal, and encode a frame having an unvoiced mode for an unvoiced speech as the selected encoding mode.

Abstract: A system and methods is provided for providing SIP based voice transcription services. A computer implemented method includes: transcribing a Session Initiation Protocol (SIP) based conversation between one or more users from voice to text transcription; identifying each of the one or more users that are speaking using a device SIP_ID of the one or more users; marking the identity of the one or more users that are speaking in the text transcription; and providing the text transcription of the speaking user to non-speaking users.

Abstract: Methods and apparatus for dynamically suppressing low frequency non-speech audio events, such as road bumps, without suppressing speech formants. In exemplary embodiments of the invention, maximum powers in first and second windows are computed and used to determine whether dampening should be applied, and if so, to what extent.

Abstract: A method for recognizing a voice context for a voice control function in a vehicle. The method encompasses reading in a gaze direction datum regarding a current gaze direction of an occupant of the vehicle; allocating the gaze direction datum to a viewing zone in an interior of the vehicle in order to obtain a viewing zone datum regarding a viewing zone currently being viewed by the occupant; and determining, by utilization of the viewing zone datum, a voice context datum regarding a predetermined voice context allocated to the viewing zone currently being viewed.

Abstract: A device may receive a speech signal. The device may determine acoustic feature parameters for the speech signal. The acoustic feature parameters may include phase data. The device may determine circular space representations for the phase data based on an alignment of the phase data with given axes of the circular space representations. The device may map the phase data to linguistic features based on the circular space representations. The linguistic features may be associated with linguistic content that includes phonemic content or text content. The device may provide a synthetic audio pronunciation of the linguistic content based on the mapping.

Abstract: A matrix is generated that stores sinusoidal components evaluated for a given sample rate corresponding to the matrix. The matrix is then used to convert an audio signal to chroma vectors representing of a set of “chromae” (frequencies of interest). The conversion of an audio signal portion into its chromae enables more meaningful analysis of the audio signal than would be possible using the signal data alone. The chroma vectors of the audio signal can be used to perform analyzes such as comparisons with the chroma vectors obtained from other audio signals in order to identify audio matches.

Abstract: Disclosed herein are systems, methods, and tangible computer readable-media for detecting synthetic speaker verification. The method comprises receiving a plurality of speech samples of the same word or phrase for verification, comparing each of the plurality of speech samples to each other, denying verification if the plurality of speech samples demonstrate little variance over time or are the same, and verifying the plurality of speech samples if the plurality of speech samples demonstrates sufficient variance over time. One embodiment further adds that each of the plurality of speech samples is collected at different times or in different contexts. In other embodiments, variance is based on a pre-determined threshold or the threshold for variance is adjusted based on a need for authentication certainty. In another embodiment, if the initial comparison is inconclusive, additional speech samples are received.

Abstract: A linear prediction coefficient of a signal represented in a frequency domain is obtained by performing linear prediction analysis in a frequency direction by using a covariance method or an autocorrelation method. After the filter strength of the obtained linear prediction coefficient is adjusted, filtering may be performed in the frequency direction on the signal by using the adjusted coefficient, whereby the temporal envelope of the signal is shaped. This reduces the occurrence of pre-echo and post-echo and improves the subjective quality of the decoded signal, without significantly increasing the bit rate in a bandwidth extension technique in the frequency domain represented by SBR.