Abstract: A virtual assistant system for communicating with customers uses human intelligence to correct any errors in the system AI, while collecting data for machine learning and future improvements for more automation. The system may use a modular design, with separate components for carrying out different system functions and sub-functions, and with frameworks for selecting the component best able to respond to a given customer conversation.

Abstract: A method for aligning a translation of original caption data with an audio portion of a video is provided. The method includes identifying, by a processing device, original caption data for a video that includes a plurality of caption character strings. The processing device identifies speech recognition data that includes a plurality of generated character strings and associated timing information for each generated character string. The processing device maps the plurality of caption character strings to the plurality of generated character strings using assigned values indicative of semantic similarities between character strings. The processing device assigns timing information to the individual caption character strings based on timing information of mapped individual generated character strings. The processing device aligns a translation of the original caption data with the audio portion of the video using assigned timing information of the individual caption character strings.

Abstract: A machine learning method for training a learning model includes: transforming a first audio type of audio data into a first image type of image data, wherein a first audio component and a second audio component are mixed in the first audio type of audio data, and the first image type of image data corresponds to the first audio type of audio data; transforming a second audio type of audio data into a second image type of image data, wherein the second audio type of audio data includes the first audio component without mixture of the second audio component, and the second image type of image data corresponds to the second audio type of audio data; and performing machine learning on the learning model with training data including sets of the first image type of image data and the second image type of image data.

Abstract: A data processing device applies a first convolutional neural network layer to pieces of data included in a mini-batch to obtain a first feature map of each of the pieces of data, independently calculates a first statistic for each of the pieces of data based on the first feature maps, calculates a normalization parameter for each of the pieces of data based on the first statistic of each of the pieces of data and a cumulative statistic, normalizes the first feature map of each of the pieces of data by using a normalization parameter of each of the pieces of data to obtain a normalized feature map of each of the pieces of data, and applies a second convolutional neural network layer to the normalized feature map of each of the pieces of data to obtain a second feature map of each of the pieces of data.

Abstract: Embodiments of the disclosure provide systems and methods for speech detection. The method may include receiving a multichannel audio input that includes a set of audio signals from a set of audio channels in an audio detection array. The method may further include processing the multichannel audio input using a neural network classifier to generate a series of classification results in a series of time windows for the multichannel audio input. The neural network classifier includes a causal temporal convolutional network (TCN) configured to determine a classification result for each time window based on portions of the multichannel audio input in the corresponding time window and one or more time windows before the corresponding time window. The method may additionally include determining whether the multichannel audio input includes one or more speech segments in the series of time windows based on the series of classification results.

Abstract: An automation network provides packet-based communication between the host and a client, wherein the client determines output values from the host in the event of errors in the communication between the host and the client, where the determination of output data can be performed in a separate local processing module in accordance with a less complex method than on the host, such that it becomes possible to perform complex open-loop and closed-loop control tasks on the host even in the case of mobile clients or other clients that are difficult to wire.

Abstract: An apparatus for processing an input audio signal relies on a cascade of filterbanks, the cascade having a synthesis filterbank for synthesizing an audio intermediate signal from the input audio signal, the input audio signal being represented by a plurality of first subband signals generated by an analysis filterbank, wherein a number of filterbank channels of the synthesis filterbank is smaller than a number of channels of the analysis filterbank. The apparatus furthermore has a further analysis filterbank for generating a plurality of second subband signals from the audio intermediate signal, wherein the further analysis filterbank has a number of channels being different from the number of channels of the synthesis filterbank, so that a sampling rate of a subband signal of the plurality of second subband signals is different from a sampling rate of a first subband signal of the plurality of first subband signals.

Abstract: An audio packet error concealment system includes an encoding unit for encoding an audio signal consisting of a plurality of frames, and an auxiliary information encoding unit for estimating and encoding auxiliary information about a temporal change of power of the audio signal. The auxiliary information is used in packet loss concealment in decoding of the audio signal. The auxiliary information about the temporal change of power may contain a parameter that functionally approximates a plurality of powers of subframes shorter than one frame, or may contain information about a vector obtained by vector quantization of a plurality of powers of subframes shorter than one frame.

Abstract: A voice correction method implemented by a computer, the method includes: obtaining first voice information which is voice information recorded when noise is generated and on which noise suppression processing is performed and second voice information indicating voice information recorded in an environment in which no noise is generated, and generating emphasized information by emphasizing a component of a band corresponding to a band having a low signal noise ratio (SNR) of the first voice information, among bands of the second voice information; performing machine learning on a model based on the first voice information and the emphasized information; and generating corrected voice information by correcting third voice information on which noise suppression processing is performed, based on the machine-learned model.

Abstract: A voice synthesis method includes: sequentially acquiring voice units comprising at least one of diphone or a triphone in accordance with synthesis information for synthesizing voices; generating statistical spectral envelopes using a statistical model built by machine learning in accordance with the synthesis information for synthesizing the voices; and concatenating the sequentially acquired voice units and modifying a frequency spectral envelope of each voice unit in accordance with the generated statistical spectral envelope, thereby synthesizing a voice signal based on the concatenated voice units having the modified frequency spectra.

Abstract: A method for writing to electrically erasable and programmable non-volatile memory and a corresponding integrated circuit are disclosed. In an embodiment a method includes operatively connecting a filter circuit belonging to a communication interface to an oscillator circuit, wherein the communication interface is physically connected to a bus, generating, by the oscillator circuit, an oscillation signal and regulating the oscillation signal by the filter circuit so as to generate a clock signal for timing a write cycle.

Abstract: Embodiments are disclosed for transliterating text entries across different script systems. A method according to some embodiments includes steps of: receiving an input string in a first script system input using a keyboard; segmenting, using a probabilistic model, the input string into phonemes that correspond to characters or sets of characters in a second script system; converting the phonemes in the first script system into the characters or sets of characters in the second script system, the characters or sets of characters forming a word or a word prefix in the second script system; and outputting the word or the word prefix in the second script system.

Abstract: An apparatus and method are disclosed for processing an audio signal. The apparatus includes an input interface, a digital filterbank having an analysis part and a synthesis part, a first phase shifter, a spectral envelope adjuster, a second phase shifter, and an output interface. The first phase shifter and the second phase shifter reduce a complexity of the digital filterbank, which includes both analysis and synthesis filters that are complex-exponential modulated versions of a prototype filter.

Abstract: A subtitle management system is provided that analyzes and adjusts subtitles for video content to improve the experience of viewers. Subtitles may be optimized or otherwise adjusted to display in particular regions of the video content, to display in synchronization with audio presentation of the spoken dialogue represented by the subtitles, to display in particular colors, and the like. Subtitles that are permanently integrated into the video content may be identified and addressed. These and other adjustments may be applied to address any of a variety of subtitle issues and shortcomings with conventional methods of generating subtitles.

Abstract: Disclosed is a voice recognition apparatus connected via a network and sharing a voice recognition function. The voice recognition apparatus includes: a microphone configured to receive a voice signal from a user's speech; a communicator configured to communicate with at least one external voice recognition apparatus; a voice recognizer configured to determine a wake-up word involved in the voice signal; and a controller configured to transmit the voice signal to the external voice recognition apparatus corresponding to the determined wake-up word. Thus, it is possible to overcome a limited voice recognition distance caused by a physical characteristic of a microphone and expand a spatial range where voice recognition is possible, thereby providing various voice recognition services to a user in more places.

Abstract: Systems, devices and methods are described herein for segmentation of content, and more specifically for segmentation of content in a content management system. In one aspect, a method may include receiving content associated with speech, text, or closed captioning data. The speech, the text, or the closed captioning data may be analyzed to derive at least one of a topic, subject, or event for at least a portion of the content. The content may be divided into two or more content segments based on the analyzing. At least one of the topic, the subject, or the event may be associated with at least one of the two or more content segments based on the analyzing. At least one of the two or more content segments may then be published such that each of the two or more content segments is individually accessible.

Abstract: A method and system implements a repeater in a link of a communication medium. The method and system enables a counter to count alternations of a clock signal received from a host or device over the link, compares a value of the counter to a reference count, adjusts a frequency selection based on the comparison of the value of the counter to the reference count, and locks the frequency selection in response to the counter matching the reference count.

Abstract: A non-transitory computer-readable recording medium records a program for causing a computer to execute an utterance impression determination process. The utterance impression determination process includes specifying a current fundamental frequency from a voice signal which is received, calculating a relaxation value by changing the current fundamental frequency in chronological order so that the change in the current fundamental frequency becomes moderate, and evaluating the voice signal based on a degree of a magnitude of a difference between at least one feature amount associated with the current fundamental frequency and the relaxation value corresponding to the feature amount.

Abstract: Methods and apparatus provide acoustic detection for automated devices such as respiratory treatment apparatus. In some embodiments of the technology, acoustic analysis of noise or sound pulses, such as a cepstrum analysis, based on signals of a sound sensor (104) permits detection of obstruction (O) such as within a patient interface, mask or respiratory conduit (108) or within patient respiratory system. Some embodiments further permit detection of accessories such as an identification thereof or a condition of use thereof, such as a leak. Still further embodiments of the technology permit the detection of a patient or user who is intended to use the automated device.

Abstract: An apparatus for processing an input audio signal relies on a cascade of filterbanks, the cascade having a synthesis filterbank for synthesizing an audio intermediate signal from the input audio signal, the input audio signal being represented by a plurality of first subband signals generated by an analysis filterbank, wherein a number of filterbank channels of the synthesis filterbank is smaller than a number of channels of the analysis filterbank. The apparatus furthermore has a further analysis filterbank for generating a plurality of second subband signals from the audio intermediate signal, wherein the further analysis filterbank has a number of channels being different from the number of channels of the synthesis filterbank, so that a sampling rate of a subband signal of the plurality of second subband signals is different from a sampling rate of a first subband signal of the plurality of first subband signals.

Abstract: During text-to-speech processing, a speech model creates output audio data, including speech, that corresponds to input text data that includes a representation of the speech. A spectrogram estimator estimates a frequency spectrogram of the speech; the corresponding frequency-spectrogram data is used to condition the speech model. A plurality of acoustic features corresponding to different segments of the input text data, such as phonemes, syllable-level features, and/or word-level features, may be separately encoded into context vectors; the spectrogram estimator uses these separate context vectors to create the frequency spectrogram.

Abstract: A computer system constructs a decision-predictive recipient profile using sensatory data tied to an online profile of a recipient. After obtaining base sensatory data tied to the online profile of the recipient, the system may filter the base sensatory data by searching the base sensatory data for one or more machine-cognizable characteristics. The filtered sensatory data may be provided to an execution group, which may review displays of the sensatory data. Responsive to the displays of the sensatory data, the execution group may generate descriptors of the content of the filtered sensatory data and send the descriptors to the system. The system may process the descriptors to generate or augment the decision-predictive recipient profile.

Abstract: A camera device and camera system for video-based workplace safety is provided. The camera device includes at least one imaging sensor configured to capture one or more video sequences in a workplace environment having a plurality of machines therein. The video camera further includes a processor. The processor is configured to generate a plurality of embedding vectors based on a plurality of observations. The observations include (i) a subject, (ii) an action taken by the subject, and (iii) an object on which the subject is taking the action on. The subject and object are constant. The processor is further configured to generate predictions of one or more future events based on one or more comparisons of at least some of the plurality of embedding vectors. The processor is configured to generate a signal for initiating an action to the at least one of the plurality of machines to mitigate harm.

Abstract: A method and apparatus of acoustic processing for a mobile device having a haptic actuator is described. A vibration drive signal for driving a haptic actuator is received. A vibration noise output from a haptic actuator is detected. At least one vibration noise metric from the detected vibration noise output and the vibration drive signal is generated. The vibration noise output level is adapted in dependence of the at least one vibration noise metric.

Abstract: Embodiments are disclosed for transliterating text entries across different script systems. A method according to some embodiments includes steps of: receiving an input string in a first script system input using a keyboard; segmenting, using a probabilistic model, the input string into phonemes that correspond to characters or sets of characters in a second script system; converting the phonemes in the first script system into the characters or sets of characters in the second script system, the characters or sets of characters forming a word or a word prefix in the second script system; and outputting the word or the word prefix in the second script system.

Abstract: An audio processing system has a buffer, a first digital signal processing module that uses a first lookahead, a second digital signal processing module that uses a second, greater lookahead, and a cross-fader. The cross-fader fades between the output of the first digital signal processing module to the output of the second digital signal processing module, based on lookahead depth of data of the audio signal in the buffer. Other aspects are also described and claimed.

Abstract: The present invention is a system and method for increasing the playback speed of audio waves. The system analyzes an audio wave to detect a first silent section that has a length greater than a minimum short pause length required to distinguish between words. The system then calculates a new playback speed of the first silent section so that the total playback time for the first silent section is less than or equal to the minimum short pause length and controls an audio playback device to play the audio wave in a manner so that the first silent section is played back at the new playback speed. In another embodiment, the system analyzes spoken words, phonemes by phonemes, and increases the spoken word playback speed by dynamically reducing the length of each phoneme and inter-syllable silent pauses. Thus, the system functions equally well for all languages and accents.

Abstract: One embodiment is a method of presenting an audio or audio-visual work which includes: (a) detecting media work content properties in an audio portion of the audio or audio-visual work using a media work content properties detection apparatus; (b) associating a presentation rate of the audio of the audio portion of the audio or audio-visual work with the detected media work content properties; and (c) presenting the portion of the audio or audio-visual work using the media work content properties detection apparatus so that the audio is presented at the presentation rate; wherein the media work content properties comprise one or more indicia of words of interest; and wherein the audio or audio-visual work includes conversations.

Abstract: According to some embodiments, an analog processing portion may receive an audio signal from a microphone. The analog processing portion may then convert the audio signal into sub-band signals and estimate an energy statistic value, such as a Signal-to-Noise Ratio (“SNR”) value, for each sub-band signal. A classification element may classify the estimated energy statistic values with analog processing such that a wakeup signal is generated when voice activity is detected. The wakeup signal may be associated with, for example, a battery-powered, always-listening audio application.

Abstract: A waveform data structure includes a plurality of types of frames having different data sizes. Each of the plurality of types of frames includes an auxiliary information area and a data area. The auxiliary information area includes an area for storing common effective-bit length data for a section of waveform samples, and an area for storing an identifier for identifying one of the plurality of types of frames. The data area is an area for storing extracted waveform samples which are extracted from the waveform samples based on the common effective-bit length. The number of the extracted waveform samples is determined based on the common effective-bit length.

Abstract: In accordance with an example embodiment of the present invention, disclosed is a method and an apparatus thereof for selecting a packet loss concealment procedure for a lost audio frame of a received audio signal. A method for selecting a packet loss concealment procedure comprises detecting an audio type of a received audio frame and determining a packet loss concealment procedure based on the audio type. In the method, detecting an audio type comprises determining a stability of a spectral envelope of signals of received audio frames.

Abstract: An apparatus and method are disclosed for processing an audio signal. The apparatus includes an input interface, a digital filterbank having an analysis part and a synthesis part, a first phase shifter, a spectral envelope adjuster, a second phase shifter, and an output interface. The first phase shifter and the second phase shifter reduce a complexity of the digital filterbank, which includes both analysis and synthesis filters that are complex-exponential modulated versions of a prototype filter.

Abstract: A gain adjustment apparatus for use in decoding of audio that has been encoded with separate gain and shape representations includes an accuracy meter configured to estimate an accuracy measure of the shape representation, and to determine a gain correction based on the estimated accuracy measure. An envelope adjuster further included in the apparatus is configured to adjust the gain representation based on the determined gain correction.

Abstract: An interpreting assistant system which provides to a user captions of auditory communications in the user's vicinity. The interpreting assistant system includes a smart microphone transmitter that defines an input device which converts auditory communications into audio signals and transmit the signals a translation device, with a smart phone defining the translation device which generates a text transcript from the audio signals and send the transcript file to a display device, with the display device being defined by a wearable display interface which displays the transcript for a user to see. When in use, the interpreting assistant system provides for the display of a real time transcription and display of auditory communications such as spoken words for a user that may have hearing difficulties.

Abstract: The computational resources that are needed to apply a transform-based filterbank to a limited-bandwidth audio signals are reduced by performing an integrated process of combining real-valued input data into complex-valued data and applying a short transform to the complex-valued data, applying a bank of very short transforms to the output of the integrated process, and deriving a sequence of real-valued output data from the outputs of the bank of very short transforms.

Abstract: Embodiments are disclosed for transliterating text entries across different script systems. A method according to some embodiments includes steps of: receiving an input string in a first script system input using a keyboard; segmenting, using a probabilistic model, the input string into phonemes that correspond to characters or sets of characters in a second script system; converting the phonemes in the first script system into the characters or sets of characters in the second script system, the characters or sets of characters forming a word or a word prefix in the second script system; and outputting the word or the word prefix in the second script system.

Abstract: A system for exploiting visual information for enhancing audio signals via source separation and beamforming is disclosed. The system may obtain visual content associated with an environment of a user, and may extract, from the visual content, metadata associated with the environment. The system may determine a location of the user based on the extracted metadata. Additionally, the system may load, based on the location, an audio profile corresponding to the location of the user. The system may also load a user profile of the user that includes audio data associated with the user. Furthermore, the system may cancel, based on the audio profile and user profile, noise from the environment of the user. Moreover, the system may include adjusting, based on the audio profile and user profile, an audio signal generated by the user so as to enhance the audio signal during a communications session of the user.

Abstract: This disclosure describes systems, methods, and apparatus for precluding transmission of messages that breach one or more compliance or legal framework. In particular, messages, whether digital, written, or verbal, can be stopped from leaving a device on which they are created, thereby preventing non-compliant messages from reaching intermediary servers that could constitute a compliance violation even if the message never reached a recipient.

Abstract: A method includes receiving, at a content server from a media device, a request for media content at a first playback rate. The media content is available to the content server at a second playback rate that is different from the first playback rate. The method includes generating modified media content by modifying a first portion of the media content to have a second format corresponding to a third media playback rate. The first portion having a first media characteristic. The third playback rate is different than the first playback rate and is different than the second playback rate. The third playback rate is selected such that the modified media content has a third format corresponding to the first playback rate. The method further includes sending the modified media content from the content server to a media device.

Abstract: Captured vocals may be automatically transformed using advanced digital signal processing techniques that provide captivating applications, and even purpose-built devices, in which mere novice user-musicians may generate, audibly render and share musical performances. In some cases, the automated transformations allow spoken vocals to be segmented, arranged, temporally aligned with a target rhythm, meter or accompanying backing tracks and pitch corrected in accord with a score or note sequence. Speech-to-song music applications are one such example. In some cases, spoken vocals may be transformed in accord with musical genres such as rap using automated segmentation and temporal alignment techniques, often without pitch correction. Such applications, which may employ different signal processing and different automated transformations, may nonetheless be understood as speech-to-rap variations on the theme.

Abstract: Hearing device configuration and hearing treatment using categorical perception; systems and methods for categorical perception based configuration of hearing devices and hearing treatment.

Abstract: The disclosure relates to an audio classifier comprising: a first processor having hard-wired logic configured to receive an audio signal and detect audio activity from the audio signal; and a second processor having reconfigurable logic configured to classify the audio signal as a type of audio signal in response to the first processor detecting audio activity.

Abstract: An audio packet error concealment system includes an encoding unit for encoding an audio signal consisting of a plurality of frames, and an auxiliary information encoding unit for estimating and encoding auxiliary information about a temporal change of power of the audio signal. The auxiliary information is used in packet loss concealment in decoding of the audio signal. The auxiliary information about the temporal change of power may contain a parameter that functionally approximates a plurality of powers of subframes shorter than one frame, or may contain information about a vector obtained by vector quantization of a plurality of powers of subframes shorter than one frame.

Abstract: In accordance with an example embodiment of the present invention, disclosed is a method and an apparatus thereof for selecting a packet loss concealment procedure for a lost audio frame of a received audio signal. A method for selecting a packet loss concealment procedure comprises detecting an audio type of a received audio frame and determining a packet loss concealment procedure based on the audio type. In the method, detecting an audio type comprises determining a stability of a spectral envelope of signals of received audio frames.

Abstract: A processor is disclosed. The processor includes a first-receiver-node for receiving a first-receiver-signal, a second-receiver-node for receiving a second-receiver-signal, a first-output-node for coupling to a digital-baseband-processor, a second-output-node for coupling to the digital-baseband-processor and a first-active-data-pipe extending between the first-receiver-node and the first-output-node. The first-active-data-pipe includes a first-analog-to-digital-converter comprising a first-ADC-input coupled to the first-receiver-node and a first-ADC-output coupled to the first-output-node. The first-analog-to-digital-converter is configured to provide a first-digital-signal to the first-output-node. The processor comprises a first-reference-node and a configurable-data-pipe extending between the second-receiver-node and the second-output-node.

Abstract: An apparatus for processing an input audio signal relies on a cascade of filterbanks, the cascade having a synthesis filterbank for synthesizing an audio intermediate signal from the input audio signal, the input audio signal being represented by a plurality of first subband signals generated by an analysis filterbank, wherein a number of filterbank channels of the synthesis filterbank is smaller than a number of channels of the analysis filterbank. The apparatus furthermore has a further analysis filterbank for generating a plurality of second subband signals from the audio intermediate signal, wherein the further analysis filterbank has a number of channels being different from the number of channels of the synthesis filterbank, so that a sampling rate of a subband signal of the plurality of second subband signals is different from a sampling rate of a first subband signal of the plurality of first subband signals.

Abstract: A method of detecting a particular abnormal sound in an environment with background noise is provided. The method includes acquiring a sound from a microphone, separating abnormal sounds from the input sound based on non-negative matrix factorization (NMF), extracting Mel-frequency cepstral coefficient (MFCC) parameters according to the separated abnormal sounds, calculating hidden Markov model (HMM) likelihoods according to the separated abnormal sounds, and comparing the likelihoods of the separated abnormal sounds with a reference value to determine whether or not an abnormal sound has occurred. According to the method, based on NMF, a sound to be detected is compared with ambient noise in a one-to-one basis and classified so that the sound may be stably detected even in an actual environment with multiple noises.

Abstract: A system, article, and method of random access compression of transducer data for automatic speech recognition decoding.

Abstract: A mobile terminal with a built-in voice message searching function includes: a voice recording module configured to record a voice searching signal from a user and to send the voice searching signal to the pre-processing module for pre-processing, a pre-processing module configured to pre-process the voice searching signal, and to send the pre-processed signal to the matching module for signal matching, a matching module configured to extract a characteristic parameter of the pre-processed signal, to calculate a similarity of the extracted characteristic parameter with a characteristic parameter of a stored voice message, and to send the voice message with a similarity higher than or equal to a threshold to the result outputting module, and a result outputting module configured to display the voice message with the similarity higher than or equal to the threshold on a screen of the mobile terminal.

Abstract: Embodiments of the invention are directed to systems and methods for voice filtering. In some embodiments, an original voice segment from a user may be received. The received original voice segment may be modified using a first predetermined algorithm. The modified voice segment may be sent to an authentication server. At the authentication server, the modified voice segment may be reconstructed into the original voice segment using a second predetermined algorithm. The user may be authenticated for a transaction based at least in part on the reconstructed original voice segment.