Abstract: Systems and methods in accordance with various embodiments of the present disclosure provide improved techniques to process and manage media content and associated intellectual property rights associated with the media content. Intellectual property rights associated with media content can include copyright, trademarks, licenses to composition, synchronization, performance, recordings, etc. In particular, various embodiments provide media licensing management and monetization based on media licensing using a centralized registry of media content and associated asset rights.

Abstract: A marking method on image combined with sound signal, a terminal apparatus, and a server are provided. In the method, a first image is displayed. A selection command is detected. A target sound signal is embedded into a speech signal so as to generate a combined sound signal. The combined sound signal is transmitted. The selection command corresponds to a target region in the first image, and the selection command is generated selecting the target region through an input operation. The target sound signal corresponds to the target region of the selection command, and the speech signal is obtained by receiving sound. Accordingly, all attendants in the video conference are able to make makings on a shared screen.

Abstract: A method, including an action of receiving first data based on data based on ambient sound captured with a first microphone, and further including an action of receiving second data based on data based on the ambient sound captured with a second microphone, wherein the first microphone is a part of a hearing prosthesis, the second microphone is part of an indoor sound capture system or indoor sound capture sub-system, and the method further comprises comparing the first data to the second data.

Abstract: The event detection system and method employs at least one biometric capture device or mechanism disposed within a controlled-environment area configured to capture biometrics of residents of the controlled-environment facility within the controlled-environment area. The system also identifies the conditions that define an occurrence of an event. The system monitors the conditions within the facility to detect the conditions associated with an event. The computing system detects such events as delivery of meals, medicines, commissary items, and other goods and services. Such delivery of meals, medicines, and commissary items may be identified by proximity of a resident with personnel or identifying the location of the resident.

Abstract: In a content management system for customizing and controlling data distribution, data items can be customized and dispersed to disparate reception terminals according to various control mechanisms where they can convey relevant information to local or remote stakeholders. Data items can be integrated into existing data distribution structures in real time according to applicable control mechanisms and distribution protocols. Data resources can be efficiently used in distributing data under predefined and configurable conditions. Control mechanisms and distribution protocols can be determined based on an interested user's recognized status within the system. Design of the distribution system allows for informed delivery of essential information in a controlled and automated fashion such that users can interact with the system in receiving or distributing current and relevant information across the distribution network for others to access.

Abstract: Methods, devices, and systems are provided for synchronizing a source device with a sink device. In some examples, the source device plays first audio using, e.g., an electro-acoustic transducer. The source device transmits a stream of packets to the sink device to be used by the sink device for playing second audio, where the playing of the second audio is to be synchronized within a predefined tolerance with the playing of the first audio. In response to determining there is a delay in average packet arrival times of the stream of packets at the sink device, the source device adjusts the playing of the first audio to maintain synchronization with the playing of the second audio within the predefined tolerance.

Abstract: An active noise reduction system includes a canceling sound output device configured to output a canceling sound, a plurality of noise microphones configured to generate a plurality of noise signals based on a noise, and a controller configured to control the canceling sound output device based on the plurality of noise signals, wherein the controller is configured to acquire the plurality of noise signals output from the plurality of noise microphones, select a reference signal and an error signal from among the plurality of noise signals, the reference signal corresponding to the noise, the error signal corresponding to an error between the noise and the canceling sound, generate a correction reference signal by removing a component of the canceling sound from the reference signal, and generate a control signal based on the correction reference signal, the control signal being a signal for controlling the canceling sound output device.

Abstract: A method operates a hearing instrument having an acousto-electric first input transducer and an electro-acoustic output transducer. A first input signal is generated by the first input transducer from an ambient sound. The first input signal and/or a first intermediate signal derived from the first input signal is resolved into a multiplicity of frequency bands. An output signal is generated from the first input signal, or from the first intermediate signal, by frequency-selective signal processing. A relevant subset of frequency bands is determined from the aforementioned multiplicity such that, in each frequency band of the relevant subset, an output sound generated from the output signal by the output transducer makes a contribution that lies above a predefined or desired threshold, further, with the aid of signal components of the first input signal, or of the first intermediate signal.

Abstract: A first device includes a memory configured to store instructions and one or more processors configured to receive audio signals from multiple microphones. The one or more processors are configured to process the audio signals to generate direction-of-arrival information corresponding to one or more sources of sound represented in one or more of the audio signals. The one or more processors are also configured to and send, to a second device, data based on the direction-of-arrival information and a class or embedding associated with the direction-of-arrival information.

Abstract: The present disclosure provides methods, devices and computer program products for encoding and decoding of a vector of parameters in an audio coding system. The disclosure further relates to a method and apparatus for reconstructing an audio object in an audio decoding system. According to the disclosure, a modulo differential approach for coding and encoding a vector of a non-periodic quantity may improve the coding efficiency and provide encoders and decoders with less memory requirements. Moreover, an efficient method for encoding and decoding a sparse matrix is provided.

Abstract: Methods, apparatus and articles of manufacture to identify sources of network streaming services are disclosed. An example method includes receiving a first audio signal that represents a decompressed second audio signal, identifying, from the first audio signal, a parameter of an audio compression configuration used to form the decompressed second audio signal, and identifying a source of the decompressed second audio signal based on the identified audio compression configuration.

Abstract: This disclosure describes an apparatus and method of an embodiment of an invention that is improves Direction of Arrival (DOA) determinations. This embodiment of the apparatus includes a plurality of microphones coupled together as a microphone array used for beamforming, the plurality of microphones are positioned at predetermined locations and produce audio signals to be used to form a directional pickup pattern; a processor, memory, storage, and a power supply operably coupled to the microphone array, the processor configured to execute the following steps: processing an algorithm for a Direction of Arrival (DOA) determination; supplemental processing that improves the DOA processing.

Abstract: Provided are a method and an apparatus for controlling power consumption of a speaker, a storage medium, and an electronic device. The method includes: acquiring environment state information through an environment information acquisition device; analyzing the environment state information through an environment state analysis module to generate an environment state result; receiving a mode selection result input by a user; generating a power consumption control parameter through a power consumption parameter calculation module according to the environment state result and the mode selection result; and controlling power consumption of a speaker through a power consumption suppression module according to the power consumption control parameter. In this way, the power consumption of the speaker can be dynamically controlled, thereby further saving the power consumption.

Abstract: A detection unit (30) detects an input start timing of a command by a gesture on an operation target with a temporal change. A command processing unit (31) performs processing of a command recognized from the gesture based on the state of the operation target at the input start timing detected by the detection unit (30).

Abstract: An audio analyzer configured to obtain spectral domain representations of two or more input audio signals. Additionally the audio analyzer is configured to obtain directional information associated with spectral bands of the spectral domain representations and to obtain loudness information associated with different directions as an analysis result. Contributions to the loudness information are determined in dependence on the directional information.

Abstract: A system and method implementing an acoustic mapping suite that equips intelligent vehicles with the ability to identify and respond to primary sounds of interest in their environment. The invention performs sound separation, localization and classification techniques, while also generating a sound-based map of the vicinity as well as making inferences and qualitative assessments about sound sources with information over time.

Abstract: An exemplary system includes a memory storing instructions and one or more processors communicatively coupled to the memory and configured to execute the instructions to perform a process comprising: accessing fitting data representative of fitting parameters set by a fitting application to fit a hearing device to a user; determining user behavior data indicative of a hearing intention of the user in an auditory scene where the user is located; determining auditory scene data representative of information about the auditory scene; and implementing, based on the fitting data, the user behavior data, and the auditory scene data, a sound processing delay scheme for use by the hearing device.

Abstract: Examples of the disclosure describe systems and methods for estimating acoustic properties of an environment. In an example method, a first audio signal is received via a microphone of a wearable head device. An envelope of the first audio signal is determined, and a first reverberation time is estimated based on the envelope of the first audio signal. A difference between the first reverberation time and a second reverberation time is determined. A change in the environment is determined based on the difference between the first reverberation time and the second reverberation time. A second audio signal is presented via a speaker of a wearable head device, wherein the second audio signal is based on the second reverberation time.

Abstract: Methods, systems, and media for identifying a plurality of sets of coordinates for a plurality of devices are provided. In some embodiments, the method comprises: identifying each device in a plurality of devices associated with a user account; instructing the plurality of devices to perform an audio sequence; receiving a plurality of transit times from the plurality of devices; determining a plurality of distances based on the plurality of transit times; determining a plurality of sets of coordinates based on the plurality of distances; associating to each of the plurality of devices a corresponding unique one of the plurality of sets of coordinates; and causing at least one of the plurality of devices to play spatial audio determined from the plurality of sets of coordinates.

Abstract: The present disclosure relates to an artificial intelligence-based similar sound source search system and method for searching for a matching sound source by identifying timbre, pitch, and chord information using a convolutional neural network (CNN) and comprises a sound source input unit, a sound source analysis unit, a sound source information extraction unit, and a matching sound source search unit.

Abstract: A method, computer program product, and computing system for generating a plurality of acoustic relative transfer functions associated with a plurality of audio acquisition devices of an audio recording system deployed in an acoustic environment. At least a pair of the plurality of acoustic relative transfer functions from time frames may be compared. A change in the acoustic environment may be detected based upon, at least in part, the comparison of the plurality of acoustic relative transfer functions from at least the pair of time frames.

Abstract: An in-vehicle violence detection system includes a speech and non-speech audio recognition module capturing occupant threat words and non-speech audio events. In-vehicle accelerometers generate data analyzed in a shaking movement recognition module. A heart rate and breathing rate detection module measures physiological changes in occupant heart rates and breathing rhythms. An in-vehicle semantic scene recognition module analyzes occupant non-verbal interactions. Occupant threat indicators including an audible threat indicator are generated by the speech and non-speech audio event recognition module. Visual threat indicators are generated by the in-vehicle semantic scene recognition module. Physiological threat indicators are generated by the heart rate and breathing rate detection module. Vibration-based threat indicators are generated by the shaking movement recognition module.

Abstract: Provided is a filter coefficient optimization technology that makes it possible to design a stable beamformer having a good quality by considering the relationship of a filter coefficient between adjacent frequency bins. A filter coefficient optimization apparatus includes an optimization unit that calculates an optimum value of a filter coefficient w={w1, . . .

Abstract: A noise cancellation processing method, device and apparatus are provided. The noise cancellation processing method includes: collecting first voice data in a surrounding environment by using a noise-cancelling earphone in response to detecting that the noise-cancelling earphone is in a wearing state and a noise cancellation mode is enabled; extracting to-be-recognized voiceprint feature information according to the first voice data; identifying similarities between registered voiceprint feature information stored in a registered voiceprint database and the to-be-recognized voiceprint feature information entry by entry; and in response to at least one of the similarities being greater than a first preset threshold, performing a preset action in the noise-cancelling earphone.

Abstract: A monitoring and inspection system for a work area includes a non-visual sensory detection sensor, such as a microphone or vibration detection sensor, and a processor. The sensor is configured to sense sounds or vibrations generated in the work area during the performance of an action that are then received by the processor. The processor analyzes the received acoustic and/or vibrational signals and compares the received signals to an expected signal to verify that the identified acoustic and/or vibration signature of the detected signal is an acoustic and/or vibration signature associated with the operational step that was performed to confirm that the operational step has been performed, and that it has been performed properly.

Abstract: Provided are a sound model generation device and the like that make it possible to more easily generate a sound model capable of distinguishing sound events using a plurality of features. A concatenating unit concatenates a plurality of features of sound signals that are learning data and generates a concatenated feature. A learning unit learns the generated concatenated feature for generating a sound model for distinguishing a sound event from a sound signal.

Abstract: Introduced here are systems and methods to increase accuracy of speech recognition by utilizing multiple sensors placed at multiple speech transmission regions such as a user's lips, throat, ear canal, etc. The various speech transmission regions are better at transmitting certain phonemes, and a sensor placed close to a particular speech transition region can more accurately detect the phoneme transmitted through the particular speech transmission region. For example, a microphone placed close to the lips can better detect labial phonemes such as m, n, p, and b, than a microphone placed close to the throat. Further, disclosed here are ways to reduce energy consumption while performing speech recognition.

Abstract: In one aspect, a network microphone device includes a plurality of microphones and is configured to detect sound via the one or more microphones. The network microphone device may capture sound data based on the detected sound in a first buffer, and capture metadata associated with the detected sound in a second buffer. The network microphone device may classify one or more noises in the detected sound and cause the network microphone device to perform an action based on the classification of the respective one or more noises.

Abstract: A venting device includes a first flap, which is configured to be actuated to swing upward during a rising time, and a second flap, which is disposed opposite to the first flap and configured to be actuated to swing downward during a falling time, a first actuating portion disposed on the first flap, and a second actuating portion disposed on the second flap. The venting device configured to form a vent is disposed within a wearable sound device or to be disposed within the wearable sound device. The vent is formed via applying a first voltage to the first actuating portion and applying a second voltage on the second actuating portion, such that the venting device gradually forms the vent.

Abstract: Apparatus, methods and computer-readable medium are provided for processing wind noise. Audio input is processed by receiving an audio input. A wind noise level representative of a wind noise at the microphone array is measured using the audio input and a determination is made, based on the wind noise level, whether to perform either (i) a wind noise suppression process on the audio input on-device, or (ii) the wind noise suppression process on the audio input on-device and an audio reconstruction process in-cloud.

Abstract: A system, method, and computer-readable medium are disclosed for improved threat sensitivity in a gaming application. As audio streams are received during a gaming scenario, a threat profile is selected that best matches the audio streams. The threat profile is a machine language cluster of noise used to adjust speaker equalization profile to emphasize threat noises. Digital signal processing (DSP) coefficients of the speaker equalization profile are adjusted as the gaming application continues.

Abstract: A computing system can implement a computer-implemented method for providing selective control of an acoustic environment of a user. The computer-implemented method can include determining a selected audio source of a plurality of audio sources in an environment of a user based at least in part on a selection user input at a peripheral device, where the peripheral device is in communication with the computing system. The computer-implemented method can include receiving a configuration user input at the peripheral device, where the configuration user input is indicative of an intent to configure an audio characteristic associated with the selected audio source. The computer-implemented method can include, in response to receiving the configuration user input, configuring the audio characteristic associated with the selected audio source based at least in part on the configuration user input.

Abstract: A portable birdfinding device comprising a plurality of PDM microphone sensors (e.g. a microphone array) coupled to processing electronics that electronically filter the acoustic signals to detect sounds from birds, and to interpret the acoustic signals to identify the angle-of-arrival (e.g. azimuth and elevation) and distance from which the bird sound is coming. Additional electronic processing may provide an identification of the bird species. In some embodiments, the microphone array may have sensors spaced so that sounds from distant birds have a different time of flight to reach the different microphone sensors, while having the spacing between microphones be less than half of the wavelength for sounds made by a predetermined bird species. The microphone array may be positioned onto an object wearable by the birdwatcher, such as a hat or a jacket. In some embodiments, the electronic processing may be carried out using a smartphone.

Abstract: A microphone unit includes: an audio data acquisition unit that acquires speech as audio data; an audio data registration unit that registers verification audio data obtained by extracting a feature point from the audio data; an evaluation audio data acquisition unit that acquires speech that is input to a first microphone as evaluation audio data; a verification unit that verifies whether or not a speaker who uttered speech that is based on the evaluation audio data is a speaker who uttered speech that is based on the verification audio data, based on the verification audio data and a feature point extracted from the evaluation audio data; and a verification result output unit that outputs a result of verification performed by the verification unit.

Abstract: Techniques for selecting an audio profile for an audio output device include generating a plurality of vector representations, wherein each vector representation of the plurality of vector representations is based on a candidate audio profile of a plurality of candidate audio profiles; clustering the plurality of vector representations into a plurality of clusters; selecting a first candidate audio profile that is representative of the plurality of candidate audio profiles included in a first cluster of the plurality of clusters; presenting, to a user, a plurality of audio test patterns, wherein each audio test pattern is rendered based on the first candidate audio profile; receiving, from the user, at least one response based on the plurality of audio test patterns; and determining an audio profile for an audio output device based on the at least one response of the user.

Abstract: A method for presenting a multimedia stream including a first audio stream and a second audio stream, comprising: receiving the first audio stream, wherein the first audio stream comprises a set of first audio slices sequentially located therein, wherein each first audio slice comprises a timestamp and a grade value; receiving the second audio stream, wherein the second audio stream comprises a set of second audio slices sequentially located in the second stream, and aligned in time with one of the first audio slice; presenting the first audio stream according to the timestamp of the first set of first audio slices; receiving a set of control commands including a first threshold value; determining whether the first threshold value is lower than the grade value of the first audio slice; and presenting the second audio slice aligned with the first audio slice.

Abstract: An abnormal sound specifying device may include an arithmetic device configured to access a learned model of artificial intelligence and an output device. The arithmetic device may perform: specifying frequency-time data of sound recorded at a vehicle; causing the learned model to specify a type of abnormal sound included in the sound based on the frequency-time data and causing the learnt model to specify a basis range from the frequency-time data, the basis range indicating a frequency range and a time range that are used to specify the type of the abnormal sound; designating a designated range indicating frequency and time ranges; and determining whether to cause the output device to output the type of the abnormal sound in a determination process, the determination process including, as a determination element, at least a determination on whether the basis range and the designated range overlap each other.

Abstract: A device for authenticating a user is described that comprises a sensor configured to measure the movement of a user in response to the interaction of the user with a displayed image and controller circuitry configured to authenticate the user in response to a positive comparison between the movement of the user and a stored movement associated with the user.

Abstract: A statistical audibility prediction (SAP) method for predicting the audibility of a signal over time at a listening location, the signal from a signal source in the presence of a concurrent masking sound or masker from a masker source. The method includes receiving, via a processor over a plurality of auditory channels, a specific loudness of the signal and masker at the listening location. The method includes calculating for each auditory channel a standard deviation of a distribution of the specific loudness of the signal and masker, and calculating, via the processor, corresponding channel-specific detectability indices (d?t,i) for each auditory channel as a function of their standard deviations. The corresponding channel-specific detectability indices are then summed to produce a total detectability index (d?t), which may be output as an electronic signal that indicates the predicted audibility vs. time, e.g., to a downstream process and/or system or offline.

Abstract: A voice-controlled device includes a microphone to receive a set of sound waves that includes speech uttered by a user and other sound, and to output a first audio signal that includes a contribution from the speech uttered by the user and a contribution from the other sound. The device also includes a receiver to receive an electromagnetic signal and to output a second audio signal obtained from the electromagnetic signal. An audio pre-processor of the device processes the first audio signal using the second audio signal to reduce the contribution from the other sound in a processed audio signal. The voice-controlled device then provides the processed audio signal to a speech recognition module to determine a voice command issued by the user.

Abstract: Systems and methods for contextual awareness in dynamic device groups are disclosed. For example, a dynamic device group may be generated while output of content is occurring. When a user provides user input to alter the output of the content, contextual data indicating the devices in the dynamic device group when the user input is received may be generated and utilized by an application to determine which devices are to receive a command to perform an action responsive to the user input.

Abstract: Techniques for implementing an augmented audio conditioning (AAC) system are described herein. In some examples, the AAC system can store conditioning data comprising crowd noise experiences associated with context-relevant environments and/or actions associated with an activity. The AAC system can detect an action of a user who is training in a conditioning environment and determine that the action is associated with the activity. In some examples, the AAC system can also determine an association between the action of the user and audio data representing a crowd noise experience of a context-relevant environment during an event. Furthermore, the AAC system can, in response to detecting the action of the user, output the audio-conditioning data into the conditioning environment to simulate the crowd noise experience of the context-relevant environment during the event.

Abstract: A computer system for facilitating voice communication within a mixed-reality environment identifies a first location of a first user within a mixed-reality environment. The first user is associated with a first electronic communication device, which is capable of facilitating voice communication between one or more users within the mixed-reality environment. The computer system identifies a second location of a second user within the mixed-reality environment. The second user is associated with a second electronic communication device, which is capable of facilitating voice communication between one or more users within the mixed-reality environment. The computer system determines a vocal distance attribute between the first user and the second user and determines, based upon the vocal distance attribute, a communication state for the first electronic communication device and the second electronic communication device.

Abstract: An acoustic system includes an in-vehicle device including a vehicle-side communicator; and an information processing device configured to be worn at an ear of a user and to communicate with the vehicle-side communicator. The information processing device includes: a sound pickup configured to pick up external sound; a sound emitter configured to emit sound to the ear of the user; a communicator configured to communicate with the in-vehicle device; and a computer. The computer is programmed to: determine, based on a result of communication with the in-vehicle device by the communicator, whether or not a predetermined condition is satisfied in a state where the information processing device is in a vehicle; and when determining that the predetermined condition is satisfied, cause the sound emitter to emit the external sound picked up by the sound pickup.

Abstract: A system configured to improve sound source localization (SSL) processing by reducing a number of direction vectors and grouping the direction vectors into direction cells is provided. The system performs clustering to generate a smaller set of direction vectors included in a delay-direction codebook, reducing a size of the codebook to the number of unique delay vectors. In addition, the system groups the direction vectors into direction cells having a regular structure (e.g., predetermined uniformity and/or symmetry), which simplifies SSL processing and results in a substantial reduction in computational cost. The system may also select between multiple codebooks and/or dynamically adjust the codebook to compensate for changes to the microphone array. For example, a device with a microphone array fixed to a display that can tilt may adjust the codebook based on a tilt angle of the display to improve accuracy.

Abstract: A system includes a memory and a processor. The memory is configured to store a machine learning (ML) model. The processor is configured to (i) obtain a set of training audio signals that are labeled with respective levels of distortion, (ii) convert the training audio signals into respective images, (iii) train the ML model to estimate the levels of the distortion based on the images, (iv) receive an input audio signal, (v) convert the input audio signal into an image, and (vi) estimate a level of the distortion in the input audio signal, by applying the trained ML model to the image.

Abstract: An electronic device comprising circuitry configured to perform (402; 702; 1204) source separation (201) based on a received audio input to obtain a separated source, to perform onset detection (202) on the separated source to obtain an onset detection signal and to mix (405; 706; 1207) the audio signal with the separated source based on the onset detection signal to obtain an enhanced separated source.

Abstract: A device that includes an adaptive acoustic detection circuit and an acoustic sensor device such as a microphone is described. The device includes in addition to the sensor a circuit configured to detect when an input stimulus to the sensor satisfies an adaptive threshold, and further configured to produce a signal upon detection that causes adjustment of performance of the device, wherein the adaptive threshold is a threshold value that varies over time in accordance with detected changes to sound of an environment in which the device is located.

Abstract: Disclosed embodiments may relate to systems and methods for monitoring and/or mapping noise data from a plurality of noise monitoring devices. In some embodiments, the plurality of noise monitoring devices may include hearing protection devices configured to detect noise, and typically may communicate such noise data (which may also include location) so that the noise data can be pooled. The pooled noise data from the plurality of noise monitoring devices may then be used to the benefit of one or more of such noise monitoring devices.

Abstract: Methods and apparatus for pre-scaling media content to facilitate audience measurement are disclosed. An example method includes encoding a media content sample in accordance with a first encoding configuration and playing the encoded sample. The example method also includes attempting to detect the codes in the sample, and computing a ratio between the codes encoded in the sample and the codes detected when the sample is played.