Abstract: A multimodal system and process processed one or more sensor signals and extracts features from the one or more sensor signals through a spatiotemporal correlation between consecutive frames of an image or video sequence. The multimodal system and process determines the movement and direction of the features through an image subtraction, or a coherence measure and synthesizes a musical instrument signal in response to the detected movement and direction or the triggers. The imaginary musical instrument signal is added to an infotainment signal within a vehicle.

Abstract: Technology is described for providing audio for digital content. The digital content may be provided to a plurality of devices. Each device may be associated with a profile. At least one group of profiles may be identified from the plurality of devices that share an affiliation. The group of profiles may be represented as a group of environment objects in the digital content. A location may be identified within the digital content that corresponds to the group of environment objects. Audio may be received for the devices. The audio may be received while the digital content is being transmitted to the plurality of devices. The audio may be transmitted to the plurality of devices for directional audio playback. The audio may be directed to correspond with the virtual location of the at least one group of environment objects in the digital content.

Abstract: A system and method for applying an individual volume control setting to individual sound zones as set by a user. When the individual volume control setting differs from a master volume control setting, the individual volume control setting is applied to the low frequency component of the audio signal. Headrest and optional height speakers for each individual sound zone's independent volume setting are applied to the mid and high frequency components of the audio signal.

Abstract: An apparatus for adapting input audio information, encoding one or more audio objects, to obtain adapted audio information is provided. The input audio information includes two or more input audio downmix channels and further includes input parametric side information. The adapted audio information includes one or more adapted audio downmix channels and further includes adapted parametric side information. The apparatus includes a downmix signal modifier for adapting, depending on adaptation information, the two or more input audio downmix channels to obtain the one or more adapted audio downmix channels. Moreover, the apparatus includes a parametric side information adapter for adapting, depending on the adaptation information, the input parametric side information to obtain the adapted parametric side information.

Abstract: In general, techniques are described for identifying a codebook to be used when compressing spatial components of a sound field. A device comprising one or more processors may be configured to perform the techniques. The one or more processors may be configured to identify a Huffman codebook to use when compressing a spatial component of a plurality of spatial components based on an order of the spatial component relative to remaining ones of the plurality of spatial components, the spatial component generated by performing a vector based synthesis with respect to a plurality of spherical harmonic coefficients.

Abstract: A method provides binaural sound to a person through electronic earphones. The binaural sound localizes to a sound localization point (SLP) in empty space that is away from but proximate to the person. When an event occurs, the binaural sound switches or changes to stereo sound, to mono sound, or to altered binaural sound.

Abstract: In accordance with an example aspect of the present invention, an apparatus is provided, the apparatus comprising at least one processing core, at least one memory including computer program code, the at least one memory and the computer program code being configured to, with the at least one processing core, cause the apparatus at least to derive, from first digital information describing a system impulse response caused by a loudspeaker to a first location in a first room, second digital information describing a cancelling impulse response, determine, for samples comprised in the second digital information, at least one of a delay value describing how much the samples are to be shifted in time and a gain factor describing how much the samples' magnitude is to be adjusted, so that an energy associated with a sum of the system impulse response and the cancelling impulse response is reduced, and employ the determined at least one of the delay value and the gain factor to generate a cancelling signal, and to ad

Abstract: Systems and methods are described for storing and reusing previously generated/calculated acoustic environment data. By reusing acoustic environment data, the systems and methods described herein may avoid the increased overhead in generating/calculating acoustic environment data for a location when this data has already been generated and is likely accurate. In particular, the time and complexity involved in determining reverberation/echo levels, noise levels, and noise types may be avoided when this information is available in storage. This previously stored acoustic environment data may not be limited to data generated/calculated by the same audio device. Instead, in some embodiments an audio device may access a centralized repository to leverage acoustic environment data generated/calculated by other audio devices.

Abstract: A binaural audio calibration method, system, and computer program product for using behavioral data and sensor data to calibrate binaural audio to a specific user and creating a personalized binaural audio which can lead to greater immersion and allow user attention to be more effectively controlled.

Abstract: A system for receiving multiple conversations or messages and for playing the multiple conversations or messages with a mobile device and wireless earpieces. The system may determine various presentation parameters based on various characteristics of the received messages and may play the messages such that audio appears at distinguishing locations around the user. The system may change how messages and/or conversations are played in response to recognize a change in the focus of the user based on detected user inputs, such as body movement gestures.

Abstract: A method of elevation rendering an audio signal includes receiving multichannel signals including a height input channel signal of a predetermined elevation angle, obtaining first elevation rendering parameters for a height input channel signal of a standard elevation angle, obtaining a delayed height input channel signal by applying a predetermined delay to a height input channel signal, updating the first elevation rendering parameters based on the predetermined elevation angle, obtaining second elevation rendering parameters based on the label of the height input channel signal and labels of two output channel signals, and elevation rendering the multichannel signals and the delayed height input channel signal to output a plurality of output channel signals of an elevated sound image, based on the updated first elevation rendering parameters and the second elevation rendering parameters.

Abstract: A speaker system has a speaker array including a plurality of first speakers and a first filter array. A peak of the directivity of reproduced sounds output from the speaker array. And the peak being in a direction in which the plurality of first speakers are arranged, is shifted toward a wall surface due to first filter processing performed by the first filter array.

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 expedites processing and playing of binaural sound during an electronic communication between a first user and a second user. An electronic device of the first user convolves sound into binaural sound for the second user before the binaural sound transmits to the electronic device of the second user. In this way, the binaural sound is already convolved and ready to play upon receipt at the electronic device of the second user.

Abstract: Embodiments of the present invention disclose a control method of a scene sound effect and related products. The method comprises: when turned on, an electrical device starts a service with a monitoring function; the electronic device monitors a newly created view in the electronic device by means of the service with the monitoring function to determine whether a view is newly created in the electronic device or not, the characteristic of the newly created view having a mapping relationship with an application in the electronic device; if yes, the electronic device determines the application that has the mapping relationship with the characteristic of the newly created view in the electronic device according to the mapping relationship; the electronic device obtains a scene sound effect corresponding to the application, and sets the current sound effect of the electronic device as the scene sound effect. The scene sound effect is not set manually during the process.

Abstract: System, method, and apparatus including receiving an identification of content played a playback system and an identification of location of the playback system. A playlist of content is associated with the location. A request is received to retrieve the playlist of content and the requested playlist of content is sent to a playback system.

Abstract: One embodiment provides a device comprising a speaker driver including a diaphragm. The device further comprises a controller configured to receive a source signal for reproduction via the speaker driver, determine an estimated displacement of the diaphragm resulting from the reproduction of the source signal, and generate a control voltage based on the estimated displacement and threshold information relating to safe displacement of the diaphragm. An actual displacement of the diaphragm during the reproduction of the source signal is controlled based on the control voltage.

Abstract: Example techniques may involve performing aspects of a spectral calibration using an applied spatial calibration. An example implementation may include receiving data representing spatial filters that correspond to respective playback configurations. The implementation may also involve causing the audio drivers to output calibration audio that is divided into a repeating set of frames, the set of frames including a respective frame for each playback configuration. Causing the audio drivers to output the calibration audio may involve causing an audio stage to apply, during each frame, the spatial filter corresponding to the respective playback configuration. The implementation may also include receiving data representing spectral filters that correspond to respective playback configurations, the spectral filters based on the calibration audio output by the audio drivers.

Abstract: A method comprising determining a spatial audio signal; determining an apparatus motion parameter; and stabilizing the spatial audio signal dependent on the apparatus motion parameters.

Abstract: A method provides binaural sound to a person through electronic earphones. The binaural sound localizes to a sound localization point (SLP) in empty space that is away from but proximate to the person. When an event occurs, the binaural sound switches or changes to stereo sound, to mono sound, or to altered binaural sound.

Abstract: A method generates a sound localization point (SLP) where binaural sound externally localizes to a person during a telephone call. The method determines a location of a handheld portable electronic device (HPED) with respect to the person and saves this location as the SLP. During the telephone call, a voice of another person is processed so the voice externally localizes to the person as the binaural sound at the SLP.

Abstract: A method for processing an audio signal in accordance with a room impulse response is described. The audio signal is separately processed with an early part and a late reverberation of the room impulse response, and the processed early part of the audio signal and the reverberated signal are combined. A transition from the early part to the late reverberation in the room impulse response is reached when a correlation measure reaches a threshold, the threshold being set dependent on the correlation measure for a selected one of the early reflections in the early part of the room impulse response.

Abstract: Methods and apparatus assist listeners in distinguishing between electronically generated binaural sound and physical environment sound while the listener wears a wearable electronic device that provides the binaural sound to the listener. The wearable electronic device generates a visual alert or audio alert when the electronically generated binaural sound occurs.

Abstract: A bass management system and method for mitigating bass management errors by using explicit information available in the object audio rendering process and deriving the correct subwoofer contribution for each audio object. Embodiments of the bass management system and method are used to maintain the correct balance of the bass reproduced by the subwoofer relative to the sound coming out of the other speakers. The system and method are useful for a variety of different speaker configurations, including speaker configurations having different speaker sub-zones. Power-normalized gain coefficients for each speaker are combined and the power of the combined gain coefficients is computed and used to obtain a power-preserving subwoofer contribution coefficient. This subwoofer contribution coefficient is applied to the bass portion of the audio signal and audio objects to determine the contribution of a particular subwoofer.

Abstract: A composite speaker module includes a vibration damper sheet, a first sound guide assembly, a first speaker driver, a second sound guide assembly, and a second speaker driver. The vibration damper sheet includes a first surface and a second surface opposite to the first surface. The first sound guide assembly includes a first sound guide and a first bracket. The first sound guide disposed on the first surface. The first bracket is connected to the first sound guide and extends away from the vibration damper sheet. The first speaker driver is disposed on the first bracket. The second sound guide assembly includes a second sound guide and a second bracket. The second sound guide disposed on the second surface. The second bracket is connected to the second sound guide and extends along a direction away from the vibration damper sheet. The second speaker driver is disposed on the second bracket.

Abstract: A system including base stations determines head-related transfer functions (HRTFs) for a user. Each base station is located at a distinct location within a local area and includes a speaker configured to emit a test sound in accordance with calibration instructions. A depth camera assembly determines depth information describing a position of a head-mounted display (HMD) in the local area relative to the locations of the base stations. A microphone is placed in an ear canal of a user wearing the HMD, and generates a respective audio sample from the test sound emitted by the speaker of each base station. A controller determines the relative position of the HMD using the depth information, generates the calibration instructions based on the relative position of the HMD, and determines the HRTFs based on the audio samples.

Abstract: A method for processing an audio signal is performed at a computing device. The method includes the following steps: receiving a digital stereo audio input signal; extracting localization cues from the digital stereo audio input signal; generating a left-side component and a right-side component from the digital stereo audio input signal, at least partially, in accordance with the localization cues; performing crosstalk cancellation to the left-side component and the right-side component, respectively, to obtain a crosstalk-cancelled left-side component and a crosstalk-cancelled right-side component; and generating a digital stereo audio output signal including the crosstalk-cancelled left-side component and the crosstalk-cancelled right-side component.

Abstract: The disclosure relates to an audio signal processing apparatus for processing a plurality of audio signals defining a plurality of audio signal spectra, the audio signals to be transmitted to a listener in such a way that the listener perceives the audio signals to originate from virtual positions of a plurality of audio signal sources. The audio signal processing apparatus comprises a selector configured to select a spatial arrangement of the virtual positions of the audio signal sources relative to the listener from a plurality of possible spatial arrangements, and a filter configured to filter the plurality of audio signals on the basis of the selected spatial arrangement.

Abstract: An in-vehicle noise control system is provided. The system includes speakers that output navigation guidance sound and audio content reproduced at a head unit of a vehicle. An audio module receives the navigation guidance sound and an audio content signal to generate an opposite waveform with respect to the navigation guidance sound and output noise offset waveforms in which tuning functions are calculated based on positions of the speakers to the opposite waveform. An amplifier is connected between the audio module and the speakers. The audio module outputs the navigation guidance sound and the audio content signal to one of the speakers located at a driver's seat and outputs the noise offset waveforms and the audio content signal to the remaining speakers.

Abstract: Embodiments are described for a system of rendering object-based audio content through a system that includes individually addressable drivers, including at least one driver that is configured to project sound waves toward one or more surfaces within a listening environment for reflection to a listening area within the listening environment; a renderer configured to receive and process audio streams and one or more metadata sets associated with each of the audio streams and specifying a playback location of a respective audio stream; and a playback system coupled to the renderer and configured to render the audio streams to a plurality of audio feeds corresponding to the array of audio drivers in accordance with the one or more metadata sets.

Abstract: A multi-object audio encoding and decoding apparatus supporting a post downmix signal may be provided. The multi-object audio encoding apparatus may include: an object information extraction and downmix generation unit to generate object information and a downmix signal from input object signals; a parameter determination unit to determine a downmix information parameter using the extracted downmix signal and the post downmix signal; and a bitstream generation unit to combine the object information and the downmix information parameter, and to generate an object bitstream.

Abstract: An out-of-head localization processing apparatus according to the embodiments includes a filter selection unit configured to select a preset filter, an out-of-head localization processing unit configured to perform out-of-head localization processing using the preset filter selected, headphones configured to output, to a user, a signal of a test sound source, an input unit configured to accept a user input, a sensor unit, a three-dimensional coordinate calculation unit configured to calculate three-dimensional coordinates of a localized position of a sound image based on a detection signal from the sensor unit, and an evaluation unit configured to evaluate, based on the three-dimensional coordinates of each of the preset filters, a filter optimal for the user from the plurality of preset filters.

Abstract: A measurement device according to this embodiment includes a peak detection unit that detects positive and negative peaks included in a sound pickup signal, a sign determination unit that determines a positive or negative sign based on amplitudes at the negative and positive peaks detected by the peak detection unit, a peak group separation unit that separates a peak sequence composed of the peaks with the determined sign into a plurality of peak groups, a maximum amplitude detection unit that detects a maximum amplitude in each of the plurality of peak groups, a signal cutout unit that cuts out the sound pickup signal at cutout timing based on the maximum amplitude and generates a plurality of cutout signals, and a signal adding unit that adds the plurality of cutout signals.

Abstract: Embodiments are provided for equalizing audio data for output by a speaker of an electronic device based on a local position or orientation of the electronic device. According to certain aspects, the electronic device can determine (858, 868) its local position based on various sensor data, and identify (870, 872) an appropriate equalization setting. In some cases, the electronic device can modify (876, 880) the equalization setting based on acoustic and/or optical data. The electronic device can apply (882) the modified or unmodified equalization setting to an audio signal and cause the speaker to output (886) the audio signal with the applied equalization setting.

Abstract: Method and system for providing virtual surround sound on headphones using input audio. Embodiments herein relate to sound processing and more particularly to providing surround sound on headphones. Embodiments herein disclose a method and system for simulating surround sound on a headphone, by emulating multiple speakers in 3D space by processing audio using Head Related Transfer Function (HRTF) filters and other audio processing filters, wherein the input to the headphone is stereo input.

Abstract: One embodiment provides a method comprising determining an actual elevation of a sound source. The actual elevation is indicative of a first location at which the sound source is physically located relative to a first listening reference point. The method further comprises determining a desired elevation for a portion of an audio signal. The desired elevation is indicative of a second location at which the portion of the audio signal is perceived to be physically located relative to the first listening reference point. The desired elevation is different from the actual elevation. The method further comprises, based on the actual elevation, the desired elevation and the first listening reference point, modifying the audio signal, such that the portion of the audio signal is perceived to be physically located at the desired elevation during reproduction of the audio signal via the sound source.

Abstract: Audio signals are received. The audio signals include left and right surround channels. The audio signals are played back using far-field loudspeakers distributed around a space having a plurality of listener positions. The left and right surround channels are played back by a pair of far-field loudspeakers arranged at opposite sides of the space having the plurality of listener positions. An audio component coinciding with or approximating audio content common to the left and right surround channels is obtained. The audio component is played back using at least a pair of near-field transducers arranged at one of the listener positions. Associated systems, methods and computer program products are provided. Systems, methods and computer program products providing a bitstream comprising the audio signals and the audio component are also provided, as well as a computer-readable medium with data representing such audio content.

Abstract: Apparatus comprising a processor configured to: determine a first orientation value of a head (101) of a user (100) of the apparatus relative to a further body part (111) of the user (100) using at least one orientation sensor (105); and control a 3D audio reproduction function of the apparatus based on the first orientation value.

Abstract: To provide an information processing device, an information processing method, and a program capable of sharing a space while maintaining the degree of freedom of a visual line. An information processing device according to the present disclosure includes: a control unit configured to perform control in a manner that a display image generated based on image information which is generated through imaging of an imaging device mounted on a moving object moving in a space, imaging-device posture information which is information regarding a posture of the imaging device, and user view information which is obtained from a user manipulation device manipulated by a user and specifies a region that the user desires to view is displayed in a display region viewed by the user.

Abstract: Disclosed is a control method for an electronic device. An electronic device according to an embodiment comprises: at least one speaker; and a processor. The processor may be configured to: obtain sound source data; obtain first sound source data, corresponding to a first designated frequency band, from the sound source data by using a filter; generate second sound source data by applying sound effect to at least a portion of sound source data corresponding to a second designated frequency band among the sound source data; generate synthesized sound source data corresponding to the sound source data by synthesizing the first sound source data and the second sound source data; and output the synthesized sound source data through the at least one speaker. Other embodiments may also be possible.

Abstract: Methods and systems for designing binaural room impulse responses (BRIRs) for use in headphone virtualizers, and methods and systems for generating a binaural signal in response to a set of channels of a multi-channel audio signal, including by applying a BRIR to each channel of the set, thereby generating filtered signals, and combining the filtered signals to generate the binaural signal, where each BRIR has been designed in accordance with an embodiment of the design method. Other aspects are audio processing units configured to perform any embodiment of the inventive method. In accordance with some embodiments, BRIR design is formulated as a numerical optimization problem based on a simulation model (which generates candidate BRIRs) and at least one objective function (which evaluates each candidate BRIR), and includes identification of a best one of the candidate BRIRs as indicated by performance metrics determined for the candidate BRIRs by each objective function.

Abstract: A signal processing system includes a filter unit, a conversion unit, a decomposition unit, and an estimation unit. The filter unit applies, to a plurality of time series input signals, N filters estimated by independent component analysis of the input signals to output N output signals. The conversion unit converts the output signals into nonnegative signals each taking on a nonnegative value. The decomposition unit decomposes the nonnegative signals into a spatial basis that includes nonnegative three-dimensional elements, that is, K first elements, N second elements, and I third elements, a spectral basis matrix of I rows and L columns that includes L nonnegative spectral basis vectors expressed by I-dimensional column vectors, and a nonnegative L-dimensional activity vector. The estimation unit estimates sound source signals representing signals of the signal sources based on the output signals using the spatial basis, the spectral basis matrix, and the activity vector.

Abstract: The disclosure relates to an apparatus for generating a sound field on the basis of an input audio signal. The apparatus comprises a plurality of transducers, wherein each transducer is configured to be driven by a transducer driving signal ql of the respective transducer; a plurality of filters configured to generate for each transducer the transducer driving signal ql of the respective transducer; and a control unit configured to provide or receive a first transducer driving signal vector q0 of dimension L such that the gradient of J(q;?) with respect to q is zero in (q0;?0), the control unit is further configured to provide a second transducer driving signal vector {tilde over (q)} of dimension L such that the gradient of the cost function J(q;?) with respect to q is [approximately] zero in ({tilde over (q)}; {tilde over (?)}), the control unit is configured to provide the second transducer driving signal vector {tilde over (q)}.

Abstract: Methods for dialog enhancing audio content, comprising providing a first audio signal presentation of the audio components, providing a second audio signal presentation, receiving a set of dialog estimation parameters configured to enable estimation of dialog components from the first audio signal presentation, applying said set of dialog estimation parameters to said first audio signal presentation, to form a dialog presentation of the dialog components; and combining the dialog presentation with said second audio signal presentation to form a dialog enhanced audio signal presentation for reproduction on the second audio reproduction system, wherein at least one of said first and second audio signal presentation is a binaural audio signal presentation.

Abstract: In one embodiment, a gain redistribution application restructures gains associated with multiple microphones included in a head-worn audio system to minimize near field noise. In response to a sound generated by a sound source, the microphones generate input signals. The gain redistribution application performs mixing operations on the input signals to generate an output signal that mitigates near field noise associated with the same side of the head as the sound source. Subsequently, the gain redistribution application transmits the output signal to a speaker that targets the same side of the head as the sound source. Advantageously, by reducing the gain associated with an input signal received via a microphone located on the same side of the head as the sound source, the gain redistribution application reduces near field noise transmitted to the user during operation in a more comprehensive fashion relative to conventional designs.

Abstract: A method and device for determining the location-dependent pulse responses in signal transmission from a transmitter in a transmission volume to a receiver in a reception volume, wherein either the transmitter or the receiver is a device fixed in a predetermined location and the other is a movable device includes: continuously emitting a band-limited signal by the transmitter; continuously capturing the signal and recording the signal with time indexing by the receiver; moving the movable device during the emission and capturing of the signal along a trajectory within the transmission or reception volume while continuously capturing the location coordinates of the movable device and recording the location coordinates with time indexing; and numerically solving a linear system of equations, the unknowns of which system of equations represent the pulse responses at discrete sampling points in the transmission or reception volume associated with the movable device.

Abstract: A transfer characteristics measurement unit measures first transfer characteristics from left and right sound sources to left and right microphones, respectively. An environmental measurement unit picks up environmental measurement signals output from the left and right sound sources with use of the left and right microphones, sets an amplitude level of transfer characteristics measurement signals and a tap length of the transfer characteristics, picks up sounds with use of the left and right microphones in a state where no sound is output from the left and right sound sources, and measures second transfer characteristics. A correction unit corrects a low frequency range of the first transfer characteristics based on the second transfer characteristics.

Abstract: A method provides binaural sound to a person through electronic earphones. The binaural sound localizes to a sound localization point (SLP) in empty space that is away from but proximate to the person. When an event occurs, the binaural sound switches or changes to stereo sound, to mono sound, or to altered binaural sound.

Abstract: An audio system is described that receives a piece of sound program content for playback from a content distribution system. The piece of sound program content may include a multi-channel dialog signal and a combined multi-channel music and effects signal. The audio system may determine a first set of directivity patterns for the multi-channel dialog signal and a second set of directivity patterns for the combined multi-channel music and effects signal. The first set of directivity patterns associated with channels of the dialog signal may have higher directivity indexes than the second set of directivity patterns associated with corresponding channels of the music and effects signal. By associating dialog components with a higher directivity than music and effects components, the system increases the intelligibility of dialog for a piece of sound program content while allowing music and effects to retain conventional directivity having a typical ratio of direct-to-reverberant sound energy.

Abstract: A phase control signal generation device generating a phase control signal for each of frequency bands for an audio signal converted into a frequency domain, the phase control signal generation device comprising: a setting change means that is able to change setting of a propagation delay time for each of predetermined frequency bands; a difference obtaining means that obtains a difference between propagation delay times before and after setting change; an updating means that updates a phase control amount of the frequency band for which the propagation delay time is changed, based on the obtained difference; and a phase control signal generating means that generates a phase control signal of each frequency band by performing a smoothing process for the phase control amount in a frequency domain using the updated phase control amount.