Abstract: An apparatus comprising: a perception sorter configured to perceptually order at least two object orientated audio signal channels; and a selective channel processor configured to process at least one of the at least two object orientated audio signal channels based on the order of the at least two object orientated audio signal channels.

Abstract: A method comprises providing at least one processing unit comprising a decomposing section and a playback section; receiving, at the decomposing section, audio data generated via an array of microphones, the audio data representing an acoustic scene; decomposing the audio data into a plurality of signals representing components of the acoustic scene arriving from a plurality of directions, using the decomposing section; and rendering the audio components for a listener based on the plurality of directions of the audio components, using the playback section.

Abstract: Provided is an audio signal processing device including a signal processing section that changes, at a time of generating and outputting 2-channel audio signals to be subjected to sound reproduction by two electroacoustic transducing means located at positions in the vicinities of both ears of a listener, from audio signals of a plurality of and more than two channels, virtual sound image localization positions on a circle around the listener, across the virtual sound image localization positions, the virtual sound image localization position that is supposed for each of the plurality of channels of audio signals provided on the circle.

Abstract: The disclosure includes a system and method for sonically customizing an audio reproduction device. The system includes a processor and a memory storing instructions that when executed cause the system to determine an application environment associated with an audio reproduction device associated with a user; determine one or more sound profiles based on the application environment; provide the one or more sound profiles to the user; receive a selection of a first sound profile from the one or more sound profiles; and generate tuning data based on the first sound profile, the tuning data configured to sonically customize the audio reproduction device.

Abstract: A method improves computer performance to provide binaural sound in a telephone call. The method includes providing the telephone call with a voice of a calling party and a coordinate location that describes where a voice of the calling party should localize as binaural sound to the called party. The voice of the calling party is convolved to localize at the coordinate location.

Abstract: An audio system comprises at least one processor configured to receive multi-channel audio data and comprising a plurality of output channels. Control instructions are established based on preset positions of a plurality of speakers connectable to the output channels and further based on virtual positions of the channels of the multi-channel audio data. The channels of the multi-channel audio data are routed to the output channels based on the control instructions.

Abstract: Systems and methods facilitating removal of content from audio files are described. A method includes identifying a sound recording in a first audio file, identifying a reference file having at least a defined level of similarity to the sound recording, and processing the first audio file to remove the sound recording and generate a second audio file. In some embodiments, winner-take-all coding and Hough transforms are employed for determining alignment and rate adjustment of the reference file in the first audio file. After alignment, the reference file is filtered in the frequency domain to increase similarity between the reference file and the sound recording. The frequency domain representation (FR) of the filtered version is subtracted from the FR first audio and the result converted to a time representation of the second audio file. In some embodiments, spectral subtraction is also performed to generate a further improved second audio file.

Abstract: The present disclosure relates to a positioning method and apparatus in a three-dimensional space of reverberation. The method includes the following steps: constructing a three-dimensional space frame surrounding a point sound source and a sound receiver, and acquiring barycentric coordinates of the three-dimensional space frame; establishing an X-Y coordinate system by using the barycentric coordinates as the origin; determining a vector from the sound receiver to the barycentric coordinates and a facing angle between the sound receiver and the barycentric coordinates according to the established X-Y coordinate system; and obtaining reverberation intensity through calculation according to the vector from the sound receiver to the barycentric coordinates and the facing angle between the sound receiver and the barycentric coordinates, and using the reverberation intensity as a control parameter.

Abstract: A device comprising one or more processors is configured to determine a plurality of segments for each of a plurality of binaural room impulse response filters, wherein each of the plurality of binaural room impulse response filters comprises a residual room response segment and at least one direction-dependent segment for which a filter response depends on a location within a sound field; transform each of at least one direction-dependent segment of the plurality of binaural room impulse response filters to a domain corresponding to a domain of a plurality of hierarchical elements to generate a plurality of transformed binaural room impulse response filters, wherein the plurality of hierarchical elements describe a sound field; and perform a fast convolution of the plurality of transformed binaural room impulse response filters and the plurality of hierarchical elements to render the sound field.

Abstract: A device for calculating loudspeaker signals using a plurality of audio sources, an audio source including an audio signal, includes a forward transform stage for transforming each audio signal to a spectral domain to obtain a plurality of temporally consecutive short-term spectra, a memory for storing a plurality of temporally consecutive short-term spectra for each audio signal, a memory access controller for accessing a specific short-term spectrum for a combination consisting of a loudspeaker and an audio signal based on a delay value, a filter stage for filtering the specific short-term spectrum by using a filter, so that a filtered short-term spectrum is obtained for each audio signal and loudspeaker combination, a summing stage for summing up the filtered short-term spectra for a loudspeaker to obtain summed-up short-term spectra, and a backtransform stage for backtransforming summed-up short-term spectra for the loudspeakers to a time domain to obtain the loudspeaker signals.

Abstract: In some embodiments, portable speakers can be small and lightweight and can communicate with one or more audio device over wired or wireless connections. In some embodiments, portable speakers achieve reduced complexity as compared to typical high fidelity systems (e.g., by including a reduced number of speaker drivers and amplifiers), while still maintaining high fidelity stereo audio playback, thereby achieving both portability and high quality audio capability. For instance, certain implementations of the speaker include two primary speakers disposed on opposing faces of the speaker enclosure (e.g., full or mid-range drivers) and two tweeters, also disposed on opposing faces. Primary speakers can be disposed on respective ends of the housing and each can output a different stereo channel. Each tweeter can be positioned on different face of the housing. The speaker system according to some embodiments generates a mono high frequency signal to drive the tweeters.

Abstract: An electronic device includes processing circuitry configured to produce an audio sound from a virtual sound source position. The circuitry also acquires from a user-input a sound position information of a perceived sound source position, and then controls an enable or disable of a function of the electronic device based on a relationship between the virtual sound source position and the perceived sound source position.

Abstract: Provided is an audio processing device including a narration canceling section configured to generate a narration canceling signal by removing a narration component from an input signal, and a reverberation adding section configured to add a reverberation effect to the narration canceling signal.

Abstract: The invention provides a method and device of extracting a sound source acoustic image body in 3D space. The method includes: determining a spatial position of a sound source acoustic image and determining a speaker beside the spatial position where the sound source acoustic image is located according to the determined spatial position (?, ?, ?) of the sound source acoustic image; calculating a correlation of signals of all sound tracks of the selected speaker in the horizontal direction and the vertical direction, and obtaining and storing a parameter set {ICH, ICv, Min{ICH, ICv}} of a acoustic image body, wherein the Min{ICH, ICv} is a smaller value between ICH and ICV. The expression parameters of the acoustic image body obtained in the present invention are used for providing technical support for accurately restoring the size of the sound source acoustic image in a 3D audio live system, which solves the technical problem that the restored acoustic image in a 3D audio is excessively narrow at present.

Abstract: A system calibrates one or more head-related transfer functions (HRTFs) for a user. An indicator is presented on a head-mounted display, where the indicator prompts the user to turn the user's head to view the indicator and effectively change the user's head orientation. The head orientation corresponds to positions of both ears, thus a position of the indicator prompting to change the user's head orientation is associated with corresponding positions of both ears. Responsive to the indicator being viewed by the user, a sound source at a fixed position transmits a test sound and the test sound is received at microphones coupled to the user's ears. By analyzing the test sound received at the microphones, a unique HRTF associated with a relative position between the sound source and each ear can be obtained.

Abstract: Example disclosed media source detection methods include determining first trending coefficients for a first audio signal obtained from a first media source in communication with a media device, wherein the first trending coefficients correspond to a set of frequency bands. Example disclosed media source detection methods also include determining second trending coefficients for a second audio signal obtained from monitoring media presented by the media device, wherein the second trending coefficients correspond to the set of frequency bands.

Abstract: An embodiment in accordance with the present invention provides a system and method for binaural rendering of complex acoustic scenes. The system includes a computing device configured to process a sound recording of the acoustic scene to produce a binaurally rendered scene for the listener. The system also includes a position sensor configured to collect motion and position data for a head of the user and also configured to transmit said motion and position data to the computing device. A sound delivery device is configured to receive the binaurally rendered acoustic scene from the computing device and to transmit the acoustic scene to the ears of the listener. In the system, the computing device is further configured to utilize the motion and position data from the inertial motion sensor to process the sound recording of the acoustic scene with respect to the motion and position of the user's head.

Abstract: Method and devices for providing surround audio signals are provided. Surround audio signals are received and are binaurally filtered by at least one filter unit. In some embodiments, the input surround audio signals are also processed by at least one equalizing unit. In those embodiments, the binaurally filtered signals and the equalized signals are combined to form output signals.

Abstract: The invention relates to a method for processing a multichannel sound in a multichannel sound system, wherein the input signals L and R are decoded, preferably as stereo signals. The aim of the invention is to develop the method such that a further improvement of the spatial reproduction of the input signals L and R is achieved on the basis of an extraction of direction components. According to the invention, this is achieved in that the signals R and L are decoded at least into two signals of the form nL-mR, in which n, m=1, 2, 3, 4.

Abstract: A reproducing device includes a reproducing unit configured to reproduce a contents sound, an output unit configured to output the contents sound which has been reproduced using the reproducing unit to a headphone, a talk detection unit configured to detect that a headphone fitter who fits the headphone talks with a person and an image normal position control unit configured to move the normal position of the image of the contents sound which has been reproduced using the reproducing unit to an arbitrary position when the talk detecting unit has detected that the headphone fitter has started talking with the person.

Abstract: Signals in an automobile audio system having at least two near-field speakers located close to an intended position of a listener's head are adjusted such that in a first mode, audio signals are distributed to the near-field speakers according to a first filter that causes the listener to perceive a wide soundstage, and in a second mode, the audio signals are distributed to the near-field speakers according to a second filter that causes the listener to perceive a narrow soundstage. A user input of a variable value is received and, in response, distribution of the audio signals is transitioned from the first mode to the second mode, the extent of the transition being variable based on the value of the user input.

Abstract: An audio-signal processing device includes a decoding unit that decodes a compressed audio stream to obtain audio signals for a predetermined number of channels; a signal processing unit that generates 2-channel audio signals including left-channel audio signals and right-channel audio signals, on the basis of the predetermined-number-of-channels audio signals; and a coefficient setting unit that sets filter coefficients corresponding to the impulse responses for the digital filters, on the basis of format information of the compressed audio stream. The signal processing unit uses digital filters to convolve impulse responses for paths from sound-source positions of the channels to the left and right ears of a listener with the corresponding predetermined-number-of-channels audio signals and adds corresponding results of the convolutions for the channels to generate the left-channel audio signals and the right-channel audio signals.

Abstract: The present invention relates to an audio signal processing apparatus and an audio signal processing method which perform binaural rendering.

Abstract: An information processing apparatus includes: position detection means for detecting a position of a client unit held by a user on the basis of a signal outputted from the client unit; conversion means for variably setting a parameter value used to convert at least one of a sound signal and a video signal on the basis of the position of the client unit detected by the position detection means and converting the signal using the parameter value; and output means for outputting the signal after conversion by the conversion means.

Abstract: A method captures audio impulse responses of a person wearing electronic microphones in a left ear and a right ear. A smartphone generates a sound while it is away from but proximate to the person. The microphones capture the audio impulse responses that are used to convolve sound so it localizes away from but proximate to the person.

Abstract: A method and apparatus are disclosed to determine individualized head-related transfer function (HRTF) parameters for a user. The technique can include determining HRTF data of a user by using transform data of the user, where the transform data is indicative of a difference, as perceived by the user, between a sound of a direct utterance by the user and a sound of an indirect utterance by the user. The technique may further involve producing an audio effect tailored for the user by processing audio data based on the HRTF data of the user.

Abstract: The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing an audio signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing binaural rendering for reproducing multi-channel or multi-object audio signals in stereo with a low calculation amount.

Abstract: The present invention relates to a method and an apparatus for processing a signal, which are used for effectively reproducing an audio signal, and more particularly, to a method and an apparatus for processing a signal, which are used for implementing binaural rendering for reproducing multi-channel or multi-object audio signals in stereo with a low calculation amount.

Abstract: The present technique relates to a sound processing apparatus, a sound processing method, and a program which are configured to obtain higher quality sound. In order to obtain an objective characteristic, the sound processing apparatus previously holds a characteristic of the sound processing apparatus, generates an inverse characteristic filter of the characteristic of the sound processing apparatus, and generates a correction filter from the obtained filter and a filter having the objective characteristic. The sound processing apparatus uses the generated correction filter to subject a sound signal to be reproduced to filter processing, and reproduces sound based on the obtained sound signal. Therefore, a characteristic to be originally obtained can be achieved. The present technique can be applied to the sound processing apparatus.

Abstract: A signal processing method and system are provided for delivering spatialized sound using highly optimized inverse filters to deliver narrow localized beams of sound from the included speaker array. The inventive method can be used to provide private listening areas in a public space and provide spatialization of source material for individual users to create a virtual 3D audio effect. In a binaural mode, a speaker array provides two targeted beams aimed towards the primary user's ears—one discrete beam for the left ear and one discrete beam for the right ear.

Abstract: A method for responding in an augmented reality (AR) application of a mobile device to an external sound is disclosed. The mobile device detects a target. A virtual object is initiated in the AR application. Further, the external sound is received, by at least one sound sensor of the mobile device, from a sound source. Geometric information between the sound source and the target is determined, and at least one response for the virtual object to perform in the AR application is generated based on the geometric information.

Abstract: A conversation assistance system with a bi-lateral array of microphones arranged externally of a space that does not include any array microphones, where the space has a left side, a right side, a front and a back, the array comprising a left side sub-array of multiple microphones and a right side sub-array of multiple microphones, where each microphone has a microphone output signal, and a processor that creates from the microphone output signals a left-ear audio signal and a right-ear audio signal. The left-ear audio signal is created based on the microphone output signals from one or more of the microphones of the left-side sub-array and one or more of the microphones of the right-side sub-array and the right-ear audio signal is created based on the microphone output signals from one or more of the microphones of the left-side sub-array and one or more of the microphones of the right-side sub-array.

Abstract: The system and method of the present invention rely on combining the Speakers+Room binaural Impulse Response(s) (SRbIR) with a special kind of crosstalk cancellation (XTC) filter—one that does not degrade or significantly alter the SRbIR's spectral and temporal characteristics that are required for effective head externalization. This unique combination leads to a 3D audio filter for headphones that allows the emulation of the sound of crosstalk-cancelled speakers through headphones, and allows for fixing the perceived soundstage in space using head tracking and thus solves the major problems for externalized and robust 3D audio rendering through headphones. Furthermore, by taking advantage of a well-documented psychoacoustic fact, this system and method can produce universal 3D audio filters that work for all listeners i.e. independent of the listener's head related transfer function (HRTF).

Abstract: An audio signal reproduction device generates, from an obtained audio signal, first reproduction signals for the first speaker group, sounds from which are localized at first virtual sound positions, and second reproduction signals for the second speaker group, sounds from which are localized at second virtual sound positions substantially the same as the first virtual sound positions. The audio signal reproduction device generates the first reproduction signals and the second reproduction signals so that at least phases or sound pressure values of a first sound and a second sound are different at a listening position, the first sound being indicated by the first reproduction signals, and localized at a first position among the first virtual sound positions, the second sound being indicated by the second reproduction signals, localized at a substantially same position as the first position, and substantially the same as the first sound.

Abstract: Provided is a surround component generation device capable of generating surround components for multichannel reproduction based on two-channel audio signals with a small amount of arithmetic operation. The surround component generation device includes: multipliers for changing amplitudes of audio input signals; subtractors for subtracting outputs of the multipliers from the input signals; coefficient updaters for updating coefficients of the multipliers; subtractors for generating surround components based on the input signals and the output signals of the multipliers; output switches for switching and outputting the output signals of the subtractors; and output switching controllers for controlling the output switches. Each of the coefficient updaters updates the coefficient of the multiplier based on the output of the subtractor, and each of the output switching controllers switches the output switch based on the coefficient updated by the coefficient updater.

Abstract: A printed article (2) and electronic device (3) are acoustically coupled. The printed article is able to receive user input by touching an area or areas (4) of the printed article with their finger and output an acoustic signal (5) which is detected, decoded and processed by the electronic device. Thus, the printed article can be used to provide remote user input and/or data, such as a universal resource identifier (URI), to the electronic device.

Abstract: A conferencing system includes a near-eye display device that displays video received from a remote communication device of a communication partner. An audio stream is transmitted to the remote communication device. The audio stream includes real-world sounds produced by one or more real-world audio sources captured by a spatially-diverse microphone array and virtual sounds produced by one or more virtual audio sources. A relative volume of background sounds in the audio stream is selectively reduced based, at least in part, on real-world positioning of corresponding audio sources, including real-world and/or virtualized audio sources.

Abstract: The present application provides method and apparatus for a binaural hearing assistance system using a monaural audio signal input. The system, in various examples, provides adjustable delay/phase adjustment and sound level adjustment. Different embodiments are provided for receiving the monaural signal and distributing it to a plurality of hearing assistance devices. Different relaying modes are provided. Special functions are supported, such as telecoil functions. The system also has examples that account for a head-related transfer function in providing advanced sound processing for the wearer. Other examples are provided that are described in the detailed description.

Abstract: A method of providing an audio signal comprising spatial information relating to a location of at least one virtual source (202) in a sound field with respect to a first user position comprises obtaining a first audio signal comprising a plurality of signal components, each of the signal components corresponding to a respective one of a plurality of virtual loudspeakers (200a-e) located in the sound field; obtaining an indication of user movement; determining a plurality of panned signal components by applying, in accordance with the indication of user movement, a panning function of a respective order to each of the signal components; and outputting a second audio signal comprising the panned signal components.

Abstract: A sound field control device according to the present disclosure includes a display target object position information acquisition unit for acquiring position information of a viewer from information obtained by imaging, and a virtual sound source position control unit for controlling a virtual sound source position on the basis of the position information. Thus, it becomes possible to optimally adjust virtual sound source reproduction in consideration of size or orientation of a head. Accordingly, it becomes possible to provide a sound field without unnatural feeling to viewers.

Abstract: Examples disclosed herein relate to an electronic gaming device including one or more processors, a memory, and one or more display devices. The electronic gaming device further includes a left surround audio device, a right surround audio device, a dialog enhancing center channel speaker, a low frequency effects device, a left speaker, and a right speaker. The combination of the left surround audio device; the right surround audio device; the dialog enhancing center channel speaker; the low frequency effects device; the left speaker; and/or the right speaker produce one or more sound effects in a vertical direction.

Abstract: A surround signal generating device includes: a surround signal generating unit which subtracts a sound signal obtained by multiplying a first gain by an inputted sound signal of a right channel from an inputted sound signal of a left channel so as to generate a surround signal of the left channel, and which subtracts a sound signal obtained by multiplying a second gain by the inputted sound signal of the left channel from the inputted sound signal of the right channel so as to generate a surround signal of the right channel; and a gain setting unit which sets the first gain and the second gain so that a correlation between the surround signal of the left channel and the surround signal of the right channel which are generated by the surround signal generating unit realizes a correlation between surround signals of an actual content having a surround sound.

Abstract: An apparatus for generating a merged audio data stream is provided. The apparatus includes a demultiplexer for obtaining a plurality of single-layer audio data streams, wherein each input audio data stream includes one or more layers, wherein the demultiplexer is adapted to demultiplex each one of one or more input audio data streams having one or more layers into two or more demultiplexed audio data streams having exactly one layer. Furthermore, the apparatus includes a merging module for generating the merged audio data stream based on the plurality of single-layer audio data streams. Each layer of the input data audio streams, of the demultiplexed audio data streams, of the single-layer data streams and of the merged audio data stream includes a pressure value of a pressure signal, a position value and a diffuseness value as audio data.

Abstract: Systems, methods, and apparatus for a unified approach to encoding different types of audio inputs are described.

Abstract: System and methods are provided for audio management. Initial head-related transfer function (HRTF) parameters indicating an initial virtual configuration of a plurality of audio sources are determined. A first user operation is detected through a user interface. Target HRTF parameters are generated in response to the first user operation. A target virtual configuration of the plurality of audio sources is determined based at least in part on the target HRTF parameters.

Abstract: Disclosed herein, among other things, are systems and methods for improved telecommunication for hearing instruments. One aspect of the present subject matter includes a hearing assistance method. The method includes receiving a first signal from a first hearing assistance device, receiving a second signal from a second hearing assistance device, and processing the first signal and the second signal to produce an output signal for use in telecommunication. In various embodiments, processing the first signal and the second signal includes comparing the first signal and the second signal, and selecting one or more of the first hearing assistance device and the second hearing assistance device for use in telecommunication based on the comparison. According to various embodiments, processing the first signal and the second signal includes combining the first signal and the second signal algorithmically to produce the output signal.

Abstract: A method for separating audio sources and an audio system using the same are provided. The method introduces the concept of a residual signal to separate a mixed audio signal into audio sources, and separates an audio signal corresponding to at least two of the audio sources as a residual signal and processes the audio signal separately. Therefore, audio separation performance can be improved. In addition, the method re-separates a separated residual signal and adds the separated residual signals to corresponding audio sources. Therefore, audio sources can be separated more safely.

Abstract: A method for encoding multi-channel HOA audio signals for noise reduction comprises steps of decorrelating the channels using an inverse adaptive DSHT, the inverse adaptive DSHT comprising a rotation operation and an inverse DSHT, with the rotation operation rotating the spatial sampling grid of the iDSHT, perceptually encoding each of the decorrelated channels, encoding rotation information, the rotation information comprising parameters defining said rotation operation, and transmitting or storing the perceptually encoded audio channels and the encoded rotation information.

Abstract: Systems and techniques are provided for detecting the location of a user, stopping media output from a first media output device, and selecting a second media output device for output of media, based on the detection. Media output from a first media output device may be stopped based on the detection or based on media output from the second media output device. A media control device used to select or control the media output may remain proximate to the first media output device as the second media output device is selected. For example, a first output device may output a media component selected by a user located proximate to the first output device. The user may exit the location and enter a second location proximate to a second output device. The media component may be output from the second output device based on detecting the user at the second location.

Abstract: In one disclosed aspect, modifications are applied to an audio signal at least partly in response to auditory events. In another aspect, an input channel is divided into auditory events by detecting changes in a measurable characteristic of the input signal.