Abstract: An embodiment of the invention may include a method, computer program product and system for communicating information to a user via a graphical user interface of a computer. An embodiment may include displaying a visual cue corresponding to a suggestion for text substitution within a text representation of a media file, wherein the suggestion for text substitution is generated in response to identification of any one or combination of a repeated word, a repeated phrase, and a filler, and wherein the suggestion for text substitution is based on an aggregation of one or more of characteristics of the text representation of the media file.

Abstract: Systems and methods for podcast playback in a system including a playback device and a mobile device as a system controller are disclosed. In one embodiment, a playback system comprising a first playback device and a mobile device, the mobile device comprising computer-readable medium having stored thereon instructions executable to perform a method comprising capturing user input selecting an alarm function, capturing user input selecting a time for playing an alarm on the first playback device, capturing user input selecting a podcast channel, updating the graphical user interface to reflect the selected podcast channel, capturing user input specifying what order to play podcast episodes from the selected podcast channel, and starting playback of a first podcast episode on the first playback device according to the specified order to play podcast episodes by the previous user input and the selected time for playing an alarm.

Abstract: The speech correction system includes a storage device, an audio receiver and a processing device. The processing device includes a speech recognition engine and a determination module. The storage device is configured to store a database. The audio receiver is configured to receive an audio signal. The speech recognition engine is configured to identify a key speech pattern in the audio signal and generate a candidate vocabulary list and a transcode corresponding to the key speech pattern; wherein the candidate vocabulary list includes a candidate vocabulary corresponding to the key speech pattern and a vocabulary score corresponding to the candidate vocabulary. The determination module is configured to determine whether the vocabulary score is greater than a score threshold. If the vocabulary score is greater than the score threshold, the determination module stores the candidate vocabulary corresponding to the vocabulary score in the database.

Abstract: A gaming headset receives a plurality of audio channels comprising game audio channels and a chat audio channel during play of a particular game. The gaming headset monitors the received audio channels for predefined words that are associated with particular sounds in a data structure, and in response to detecting predefined words, filters out at least a portion of the detected predefined words from the received plurality of audio channels. The monitoring compares sounds on the received audio channels with the particular sounds in the data structure and also performs signal analysis on the audio channels during game play to detect the occurrence of the predefined words. The filtering mutes one or more of the plurality of audio channels so that the detected occurrence of the one of the predefined words is not output via speakers of the gaming headset.

Abstract: Detecting a replay attack on a voice biometrics system comprises receiving a speech signal; forming an autocorrelation of at least a part of the speech signal; and identifying that the received speech signal may result from a replay attack based on said autocorrelation. Identifying that the received speech signal may result from a replay attack may be achieved by: comparing the autocorrelation with a reference value; and identifying that the received speech signal may result from a replay attack based on a result of the comparison of the autocorrelation with the reference value, or by: supplying the autocorrelation to a neural network trained to distinguish autocorrelations formed from speech signals resulting from replay attacks from autocorrelations formed from speech signals not resulting from replay attacks.

Abstract: Embodiments of the present disclosure disclose a method and apparatus for generating a text-to-speech model. A specific implementation of the method includes: obtaining a training sample set, a training sample including sample text information, sample audio data corresponding to the sample text information, and a fundamental frequency of the sample audio data; obtaining an initial deep neural network; and using the sample text information of the training sample in the training sample set as an input, and using the sample audio data corresponding to the input sample text information and the fundamental frequency of the sample audio data as an output, to train the initial deep neural network using a machine learning method, and defining the trained initial deep neural network as the text-to-speech model.

Abstract: This disclosure relates to configuration tools for interactive agents, sometimes referred to as bots, chatbots, virtual robots, or talkbots. Such interactive agents utilize slots for organizing and storing data received as inputs and displayed as outputs. These slots can be configured such that a slot is temporary and does not persist beyond its source dialog. Slots can also be configured such that a slot is pre-populated with information contain in incoming passed parameters.

Abstract: A data processing device includes: a digital signal processor; at least one processor; and at least one memory device configured to store a plurality of instructions, which when executed by the at least one processor, cause the at least one processor to operate to: output a first determination result relating to a scene of content through use of sound data; select processing for the sound data by a first selection method based on the first determination result; determine an attribute of the content from among a plurality of attribute candidates; and select the processing by a second selection method, which is different from the first selection method, based on a determination result of the attribute, wherein the digital signal processor is configured to execute the processing selected by the at least one processor on the sound data.

Abstract: In one embodiment, a computer program product includes a computer readable storage medium having program instructions embodied therewith. The program instructions are executable by a processing circuit to cause the processing circuit to receive collected data from one or more data collection devices, the collected data being aggregated over at least one month and comprising audio data including first voice input of a user of the one or more data collection devices. The program instructions also cause the processing circuit to store the audio data to a computer readable storage medium; determine an identity of the user based on comparing second voice input to the first voice input; and analyze the audio data for indications of hearing loss in the user over the period of time. The analysis includes determining a user's emotion during production of the audio input.

Abstract: A system and apparatus for recording and archiving diverse communications over radio transmissions. The system and apparatus enables unattended airports within a geofenced area to generate a useful archive of all radio communications made by Automatic Dependent Surveillance-Broadcast (ADS-B) equipped aircraft and ground personnel. A combination of hardware and software components are provided to record and store radio transmissions in computer files. Once stored, the computer files may then be replayed for training and investigation purposes. Likewise, users may generate custom reports based upon the data embodied in the computer files.

Abstract: Traditional audio feedback elimination systems may attempt to reduce the effect of the audio feedback by simply scaling down the audio volume of the signal frequencies that are prone to howling. Other traditional feedback elimination systems may also employ adaptive notch filtering to detect and “notch” the so-called “singing” or “howling” frequencies as they occur in real-time. Such devices may typically have several knobs and buttons needing tuning, for example: the number of adaptive parametric equalizers (PEQs) versus fixed PEQs; attack and decay timers; and/or PEQ bandwidth. Rather than removing the singing frequencies with PEQs, the devices described herein attempt to holistically model the feedback audio and then remove the entire feedback signal. Two advantages of the devices described herein are: 1.) the system can operate at a much larger loop-gain (and hence with a much higher loudspeaker volume); and 2) setup is greatly simplified (i.e., no tuning knobs or buttons).

Abstract: There is provided an audio encoding apparatus including a memory, and a processor coupled to the memory and the processor configured to determine whether a tone is included in a boundary between a low-frequency that is a frequency bandwidth below a predetermined frequency of an input signal and a high-frequency that is a frequency bandwidth above the predetermined frequency of the input signal, suppress a tone in one of the low-frequency and the high-frequency, encode the input signal having the low-frequency to generate a low-frequency code, encode the input signal having the high-frequency to generate a high-frequency code, and generate an encoded stream by multiplexing the low-frequency code and the high-frequency code.

Abstract: A user interface is presented for a hearing-impaired user to make selections that influence how that user participates in a video conference and how other participants in that video conference interact with the hearing-impaired user. For example, through appropriate user interface selections, the hearing-impaired user may specify that he/she is to always preview (for editing purposes, etc.) any sign language translations made by a video conferencing system before those translations are released to the other participants, whether as text and/or speech. Through other user interface selections, the user may configure linguistic and/or playback characteristics associated with the sign-language-to-speech translations, with speech signals being produced so as to include certain effects when played out at the endpoints of the other participants and/or emulate a desirable video conference persona from the standpoint of the hearing-impaired user.

Abstract: A method for generating a signature is disclosed. As part of the method, a first number of bits are identified in respective rows of an audio signature matrix that are determined to be the strongest bits in the row, bits of the audio signature matrix are replaced with respective cells having values depending on whether the respective bits are included in the first number of bits, a set of uniformly distributed numbers are generated within a range of numbered locations corresponding to cells of the audio signature matrix; numerical distances are determined, from respective numbers in the set of uniformly distributed numbers, to the numbered locations of the matrix, associated with nearest occurrences of a first value. A set of integers is generated based on the distances.

Abstract: A car audio output control device and a method therefor are disclosed. When an acoustic source is generated by the peripheral device, the control device controls a car audio output such that when a device acoustic source and an identification code are received while a source acoustic source self-generated by a car audio is outputted, the control device selects one output mode matched with the received identification code from among the plurality of output modes and controls the source acoustic source and the device acoustic source according to the selected output mode so as to change the car audio output.

Abstract: Methods and apparatus for audio identification during a performance are disclosed herein. An example method includes transforming a segment of audio into a log-frequency spectrogram based on a transform using a logarithmic frequency resolution. The log-frequency spectrogram is transformed into a binary image, each pixel of the binary image corresponding to a time frame and frequency channel pair. A matrix product of the binary image and a plurality of reference fingerprints is computed. Based on a number of frequency ranges represented in the binary image, the matrix product to form a similarity matrix is computed. Selecting an alignment of a line in the similarity matrix that intersects one or more bins in the similarity matrix with the largest calculated Hamming similarities. A reference fingerprint is selected based on the alignment.

Abstract: Embodiments for extraction and summarization of decision discussions of a communication by a processor. The decision elements may be grouped together according to similar characteristics. The decision elements may be linked, and sentiments of the discussion participants towards each of the decision elements may be analyzed. A summary of the plurality of the decision elements may be provided via an interactive graphical user interface (GUI) on one or more Internet of Things (IoT) devices. The summary of the decision elements may be linked to domain knowledge. The summary may be enhanced using a domain knowledge.

Abstract: Apparatus, systems, articles of manufacture, and methods are disclosed for multiple scrambled layers for audio watermarking. An example system includes a scrambler executing instructions to: divide a watermark into a plurality of watermark symbols; map the watermark symbols to a plurality of frequency bins in a plurality of frequency clumps according to a first distribution scheme to create a first watermark layer having a first combination of the frequency bins and a second watermark layer having a second combination of the frequency bins, the first combination of the frequency bins and the second combination of the frequency bins partially overlap. The scrambler also is to determine a sequence for shifting watermark symbols among the frequency bins, and generate a second distribution scheme to map the watermark symbols in accordance with the sequence. The example system also includes a transceiver to communicate the second distribution scheme to a device.

Abstract: A software input string is received and tokenized into a sequence of tokens. The sequence of tokens is applied to a trained sequence-dependent lock/unlock classifier so that each of the tokens is classified as a token that should be locked, or remain unlocked, for subsequent translation. The software input string is converted to a converted string, in which the locked tokens are identified and the converted string is submitted for machine translation. A machine translation result is received and converted so that the locked tokens are replaced in the machine translation result, to obtain a translated software string.

Abstract: In one embodiment, an apparatus comprising a first audio looping device is provided. The first audio looping device is electrically coupled to a computing device. The first audio looping device is programmed to receive a first audio signal from a musical instrument and to store the first audio signal. The first audio looping device is further programmed to playback the stored first audio signal as a first audio loop a number of times and to transmit the first audio loop to a second audio looping device via the first computing device. The first audio looping device is further programmed to receive a second audio loop from the second audio looping device via the computing device.