Abstract: An electronic device according to various embodiments of the present invention may comprise: a gyro sensor for sensing rotation of an electronic device; a memory for storing a plurality of head related transfer functions corresponding to a plurality of directions; and a processor configured to generate a stereo signal by applying, to an audio signal, a first head related transfer function corresponding to a first direction, generate a stereo signal by applying, to the audio signal, a second head related transfer function corresponding to a third direction symmetric to a second direction with reference to the first direction when the electronic device rotates from the first direction to the second direction, and generate a stereo signal by applying the first head related transfer function to the audio signal when a pre-designated time elapses after the electronic device rotates from the first direction to the second direction.

Abstract: Provided are a detection method for a sound source and a detection device therefor, which are capable of accurately detecting the position of a sound source through use of measurement results of sound intensities. In the detection method, Iall=?(Ix2+Iy2+Iz2), which is a total value of the sound intensities Ix, Iy, and Iz in three axial directions (x-, y-, and z-axis directions) orthogonal to each other at a sound receiving point (P) in a sound field, is calculated, and then the position of the sound source is estimated after extracting the sound source by performing processing with averaged values within a predetermined peak width instead of the magnitude of a peak in a sound intensity waveform of the total value, or the position of the sound source is estimated after extracting the sound source with attention being given to a spatial travel speed of a sound intensity or a sound particle velocity of the total value.

Abstract: A mechanism is provided to implement a summarization mechanism for summarizing an identified natural language document using hyperbolic embeddings. Responsive to receiving a query from a user for a summarization of the identified natural language document, the summarization mechanism produces a hyperbolic embedding model of embeddings of the query. The summarization mechanism compares the embeddings of the query to each of a set of embeddings associated with a set of sentences of the identified natural language document. Responsive to identifying a subset of embeddings associated with the set of sentences of the identified natural language document having a semantic specificity to a subset of embeddings associated with the query, the summarization mechanism adds the sentence to a summary of the identified natural language document. The summarization mechanism then outputs the summary to the user.

Abstract: There is disclosed active acoustic filter systems and methods. A processor is disposed within a housing configured to interface with a user's ear. A memory stores data defining one or more locations and a respective set of location-based processing parameters associated with each of the one or more locations. A personal computing device external to the housing is coupled to the processor via a first wireless communications link. The personal computing device determines a current location of the active acoustic filter system. The processor generates digitized processed sound by processing digitized ambient sound in accordance with a set of location-based processing parameters retrieved from the memory, the retrieved set of location-based processing parameters associated with the current location of the active acoustic filter system as determined by the personal computing device.

Abstract: The sound pick-up apparatus has a first area sound pick-up unit configured to acquire, on a basis of an input signal from a microphone array unit capable of forming microphone arrays with three or more different directionalities, area sound pick-up outputs based on two or more patterns of combinations of the microphone arrays, and a second area sound pick-up unit configured to output, as an area sound pick-up result, a result obtained by integrating the area sound pick-up outputs of the respective patterns which are acquired by the first area sound pick-up unit.

Abstract: An acoustic environment identification system is disclosed that can use neural networks to accurately identify environments. The acoustic environment identification system can use one or more convolutional neural networks to generate audio feature data. A recursive neural network can process the audio feature data to generate characterization data. The characterization data can be modified using a weighting system that weights signature data items. Classification neural networks can be used to generate a classification of an environment.

Abstract: A control apparatus includes a determination unit configured to perform detection processing for detecting a volume of a sound obtained by sound obtaining unit corresponding to imaging unit, determine whether the volume is not less than a threshold, and execute volume determination processing for outputting a determination result, a change control unit configured to perform first change control for changing an imaging direction in the imaging unit, and a control unit configured to control to change control associated with the volume determination processing when the first change control is executed by the change control unit and the second change control is executed by the change control unit, so as not to set a state in which the first change control is performed, upon execution of the second change control.

Abstract: Capturing, encoding and transcoding an acoustic field, such as a three-dimensional acoustic field, comprising a device made up of two microphones, directional analysis and encoding means of said acoustic field, and optionally means for transcoding said acoustic field.

Abstract: An audio device includes a signal processor that processes an audio signal based on a predetermined parameter, a level adjuster that adjusts a level of the audio signal, a speaker that receives the audio signal and outputs sound, an interface that receives an instruction to switch the parameter, and a controller that decreases gain of the level adjuster when receiving the instruction to switch the parameter, and then performs gain control to increase the gain of the level adjuster so that a change amount of the gain of the level adjuster for a predetermined period of time is less than or equal to a predetermined value.

Abstract: A method for processing an audio signal in accordance with a room impulse response is described. The audio signal is processed with an early part of the room impulse response separate from a late reverberation of the room impulse response, wherein the processing of the late reverberation has generating a scaled reverberated signal, the scaling being dependent on the audio signal. The processed early part of the audio signal and the scaled reverberated signal are combined.

Abstract: A method and device for sound source positioning using a microphone array. The method comprises: determining a horizontal axis that a microphone array rotates around as a reference axis; calculating, according to a sound emitted by a sound source that is collected by the microphone array, to obtain a first sound source estimated value indicating a sound source position in a three-dimensional space; acquiring an inclination angle between a plane in which the microphone array is located when it is rotating and a horizontal plane in which the reference axis is located; and according to the first sound source estimated value and the inclination angle, calculating out a second sound source estimated value on a horizontal plane corresponding to the first sound source estimated value, and using the second sound source estimated value as the determined sound source position.

Abstract: An audio data processing device according to an aspect of the present disclosure includes: a sound field effect data generator configured to add sound field effect data to audio data by arithmetic operation processing using a parameter, at least one processor, and at least one memory device that stores a plurality of instructions, which when executed by the at least one processor, causes the at least one processor to operate to: analyze a scene for the audio data, recognize switching of the scene based on an analysis result of the scene, gradually decrease both an input gain and an output gain of the sound field effect data generator, and gradually increase both the input gain and the output gain after changing the parameter.

Abstract: Personal audio systems and methods are disclosed. A personal audio system includes a voice activity detector to determine whether or not an ambient audio stream contains voice activity, a pitch estimator to determine a frequency of a fundamental component of an annoyance noise contained in the ambient audio stream, and a filter bank to attenuate the fundamental component and at least one harmonic component of the annoyance noise to generate a personal audio stream. The filter bank implements a first filter function when the ambient audio stream does not contain voice activity, or a second filter function when the ambient audio stream contains voice activity.

Abstract: The present embodiments relates to a display panel configured to display images, a plurality of sound generating actuators in a plurality of sound output areas, the sound generating actuators configured to vibrate the display panel to generate sound, and a partition that is between each of the plurality of sound output areas and surrounds at least one of the plurality of sound output areas.

Abstract: An external extended device, an audio processing method and an audio playback method are provided. The external extended device is configured to receive power supplied and a signal transmitted by a television device and the external extended device includes: an integrated physical interface connecting the external extended device and the television device; and a sound-mixing processing chip electrically connected with the integrated physical interface and configured to: acquire an accompaniment audio signal transmitted by the television device via the integrated physical interface; perform sound-mixing processing on a user voice signal gathered by a microphone device and the accompaniment audio signal; and transmit a sound-mixed signal to a power amplifier circuit of the television device via the integrated physical interface.

Abstract: A sound system for use with a non-acoustically transparent video screen (1) is disclosed, comprising at least one loudspeaker enclosure (10) comprising a loudspeaker (15) arranged so that the main direction of propagation of the sound (33) which it emits is directed towards said screen, the sound being reflected by the surface of the screen towards an audience (2). The enclosure (10) further comprises a second sound emitter (16/20) situated at the rear of said loudspeaker (15), whereby the loudspeaker enclosure has a directivity of the cardioid type.

Abstract: A multi-channel audio decoder for providing at least two output audio signals on the basis of an encoded representation is configured to perform a weighted combination of a downmix signal, a decorrelated signal and a residual signal, to obtain one of the output audio signals. The multi-channel audio decoder is configured to determine a weight describing a contribution of the decorrelated signal in the weighted combination in dependence on the residual signal. A multi-channel audio encoder for providing an encoded representation of a multi-channel audio signal is configured to obtain a downmix signal on the basis of the multi-channel audio signal, to provide parameters describing dependencies between the channels of the multi-channel audio signal, and to provide a residual signal. The multi-channel audio encoder is configured to vary an amount of residual signal included into the encoded representation in dependence on the multi-channel audio signal.

Abstract: An overlay network platform facilitates a multi-party conference. End users participate in the conference using client-based web browser software. According to this disclosure, an enhanced “audio” experience for a user participating in the conference is provided by rendering different participants' audio in the conference at different positions in a three-dimensional (3D) space. In operation, and given a set of audio streams that comprise the conference and that are being received, a three-dimensional (3D) position for each audio stream in a 3D soundscape is computed. In one embodiment, the 3D position for each audio stream is computed by identifying a target area region of space, and then determining the 3D position for each stream in the target area region of space. The relative 3D positions may be determined based on contextual information related to the content of an audio stream, as well as a computed “relevance” of one stream versus another.

Abstract: A time heuristic audio control system, comprises a receiver for receiving time-based data from a personal computing device and a memory storing one or more sets processing parameters comprising instructions for processing the ambient sound based upon the time-based data. The system further includes a processor coupled to the memory and the receiver configured to adjust the ambient sound as directed by a selected set of processing parameters retrieved from the memory to create adjusted audio, the selected set of processing parameters retrieved based upon the time-based data and at least one speaker for outputting the adjusted audio.

Abstract: A system, method and computer-readable storage device provides an improved speech processing approach in which hyper parameters used for speech recognition are modified dynamically or in batch mode rather than fixed statically. The method includes estimating, via a model trained on audio data and/or metadata, a set of parameters useful for performing automatic speech recognition, receiving speech at an automatic speech recognition system, applying, by the automatic speech recognition system, the set of parameters to processing the speech to yield text and outputting the text from the automatic speech recognition system.