Abstract: A method for creating a head-related impulse response (HRIR) for use in rendering audio for playback through headphones comprises receiving location parameters for a sound including azimuth, elevation, and range relative to a head of a listener, applying a spherical head model to the azimuth, elevation, and range input parameters to generate binaural HRIR values, computing a pinna model using the azimuth and elevation parameters to apply to the binaural HRIR values to pinna modeled HRIR values, computing a torso model using the azimuth and elevation parameters to apply to the pinna modeled HRIR values to generate pinna and torso modeled HRIR values, and computing a near-field model using the azimuth and range parameters to apply to the pinna and torso modeled HRIR values to generate pinna, torso and near-field modeled HRIR values.

Abstract: A semiconductor device includes a data processing unit that processes input data and outputs processed data, a logic inversion unit that receives the processed data, inverts the processed data based on a determination result signal to be transmitted to a data bus, and an inversion determination unit that compares the input data which has not been processed by the data processing unit with the output data of the logic inversion unit corresponding to a preceding input data, and generates the determination result signal based on a comparison result.

Abstract: A measuring unit obtains a head-related impulse response of a user based on a sound signal which is collected by a microphone worn on an ear of the user in a state where a predetermined sound as a measurement signal is outputted from a speaker. A feature amount extraction unit extracts a feature amount of a frequency characteristic corresponding to the head-related impulse response. A characteristic selection unit selects a head-related transfer function from a database, where head-related transfer functions of many people are respectively made in association with feature amounts of head-related transfer functions, based on the extracted feature amount.

Abstract: A first level converter performs level conversion for a first digital signal into a second digital signal. A second level converter performs level conversion for a third digital signal into a fourth digital signal. An arithmetic unit generates a fifth digital signal obtained by subtracting the fourth digital signal from the second digital signal, and a sixth digital signal obtained by adding the first digital signal and the fourth digital signal to each other. A first DA converter performs DA conversion for the fifth digital signal into a first analog signal. A second DA converter performs DA conversion for the sixth digital signal into a second analog signal. An adder adds the first analog signal and the second analog signal to each other to generate a third analog signal.

Abstract: Embodiments of the invention are directed to systems and methods for speech transaction processing. A location of a device associated with a user may be determined. A reference to a voice model associated with the user stored within a database may be retrieved, based at least in part on the location. A voice segment may be received from the user. Using the reference, the voice segment may be compared to the voice model stored within the database. A determination may be made whether the user is authenticated for the transaction based on the comparing step.

Abstract: A system for listening to binaural audio through a plurality of speakers having at least two pair of speakers, incorporating applying at least two Crosstalk Cancellation Filter to a corresponding at least two binaural signals to create a corresponding at least two pair of speaker signals, and inputting the at least two pairs of speakers signals to a corresponding at least two pairs of speakers of a plurality of speakers. The invention also relates to a system and method for listening to binaural audio through a plurality of speakers by dividing the speakers into groups, generating a Crosstalk Cancellation Filter for each group, and distributing the binaural audio among the speaker groups. The invention also relates to a system for placing a binaural audio signal onto a plurality of pairs of cross talk cancelled loudspeakers.

Abstract: An apparatus, method and system of enhancing a sound effect of VR glasses is disclosed. The method may include detecting angular velocity data of a turning of the VR glasses through a gyroscope sensor and detecting turning direction data of the VR glasses through a compass sensor when a user wears the VR glasses to watch video; acquiring, by a mobile terminal, the angular velocity data and the turning direction data, judging a change of the turning direction of the VR glasses using the turning direction data, controlling a change of a volume level according to the change of the turning direction, and accordingly adjusting an audio output of the mobile terminal, including volumes of left and right sound channels.

Abstract: Embodiments relate to, for a scene comprising a representation of at least one object and at least one sound source: obtaining a decomposition of the at least one object, the decomposition comprising at least one geometric component; modelling at least one interaction of the at least one object and the at least one sound source using the at least one geometric component; and, in dependence on the modelling of the at least one interaction, processing an audio input associated with the at least one sound source to obtain an audio output.

Abstract: An audio apparatus and a method of converting an audio signal are provided. The method includes: receiving a first audio signal including a plurality of channels; comparing audio signals of the plurality of channels to estimate a source position of the first audio signal; localizing a source of the first audio signal toward a three-dimensional (3D) position having an elevation component based on the estimated source position; converting the first audio signal into a second audio signal including the plurality of channels and at least one channel having, based on the localized source, a different elevation from the plurality of channels; and outputting the second audio signal.

Abstract: An audio system and method of extracting indoor reflection characteristics, the method including generating a detection signal based on an input signal and a modulation code, outputting a sound signal generated based on the detection signal through a plurality of speakers, measuring sound signals output through the plurality of speakers, or sound signals reflected by a wall of a space in which the plurality of speakers is installed, and extracting reflection characteristics of the space based on the measured sound signals and the modulation code, is disclosed.

Abstract: A method generates a sound localization point (SLP) where binaural sound externally localizes to a person during a telephone call. The method determines coordinates of 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 convolved so the voice externally localizes to the person as the binaural sound at the SLP.

Abstract: A smartphone having two microphones is used for determining the direction of a loudspeaker in a surround system setup. This is performed using smartphone rotation in azimuth and polar angle direction while capturing in its microphones a test signal from a current one of the loudspeakers. From the microphone signals a corresponding TDOA value is calculated, and the smartphone is rotated until that TDOA value is nearly zero, resulting in a loudspeaker direction information.

Abstract: An information processing method and an electronic device are provided. The method includes: detecting a first physiological parameter of a user; obtaining a change in the first physiological parameter of the user based on the first physiological parameter of the user; controlling, based on the change in the first physiological parameter of the user, an image outputting unit to output the adjusted image information; determining an audio adjustment parameter for an audio outputting unit based on the change in the first physiological parameter; and adjusting output power of the audio outputting unit based on the audio adjustment parameter, and controlling the audio outputting unit to output audio information at the adjusted output power, so that the outputted audio information coincides with the outputted adjusted image information.

Abstract: An audio processing device includes a plurality of loudspeakers outputting audio for each band, a correction filter correcting an audio signal including a plurality of bands in accordance with characteristics of the plurality of loudspeakers, and a plurality of band division filters dividing the audio signal corrected by the correction filter into bands of the loudspeakers so that a phase difference of phase characteristics is approximately 0 degree or approximately 180 degrees. The correction filter is an inverse filter set with an impulse response based on the audio outputted for each band from the plurality of loudspeakers via the plurality of band division filters.

Abstract: A first sound is output by a first transducer in an earcup located at a pinna. Respective second sound based on the output of the first audio sound is detecting by each of one or more microphones in the earcup located at the pinna. Based on the respective detected second sound, a non-linear transfer function is determined which characterizes how sound is transformed via the pinna. A signal indicative of one or more audio cues for spatializing third sound is generated based on the determined non-linear transfer function.

Abstract: A display device and a control method thereof are disclosed. The display device comprises: a body with a display on the front side; at least one speaker provided on at least one side of the body; at least one sensor configured to measure the distance from the body to an adjacent face; and a controller configured to change the quality of sound output through the at least one speaker based on the distance measured by the at least one sensor. Accordingly, sound can be altered and output based on the distance between the display device and the adjacent face.

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: A virtual sound source position determination unit (122) determines a position of the virtual sound source from which a sound represented by a sound signal is output based on a detected motion of a user (31). A sound signal processing unit (123) performs stereophonic sound processing on the sound signal based on the determined position of the virtual sound source. The sound signal processing unit (123) reproduces the sound signal subjected to the stereophonic sound processing. Thus, the sound signal is reproduced such that the sound suitable for the motion of the user is output.

Abstract: A method and an electronic device are provided for simultaneously outputting different audios with directionality. The method includes outputting a first audio, and when a request to output a second audio is detected, changing at least one attribute value of the first audio and the second audio and outputting the first audio and the second audio.

Abstract: A signal processing device includes phase rotation units which rotate a phase of a signal A and generate two signals having a phase difference of ?, and a control unit which performs transition of ? over time. The control unit controls phases so that ? is approximately 0 degrees at a time point T0 and ? is approximately 180 degrees at a time point T1.

Abstract: A Head-Related Transfer Function. The Head-Related Transfer Function includes listener-specific and general components. The listener-specific component includes listener-specific, vertical variations in the Head-Related Transfer Function. The general component includes non-listener-specific, lateral variations in the Head-Related Transfer Function.

Abstract: The present invention relates to a method and an apparatus for processing an audio signal, and more particularly, to a method and an apparatus for processing an audio signal, which synthesize an object signal and a channel signal and effectively perform binaural rendering of the synthesized signal.

Abstract: Techniques for optimizing a player based on the addition of a second player are disclosed. In an embodiment, when a first player no longer needs to play certain audio frequencies due to the addition of a second player, the gain of the first player is automatically increased as part of the setup process. In another embodiment, when a first player needs to play certain audio frequencies, for example due to the removal of a second player, the gain of the first player is automatically decreased. Many other embodiments are disclosed.

Abstract: Provided are methods and systems for updating a sound field in response to user movement. The methods and systems are less computationally expensive than existing approaches for updating a sound field, and are also suitable for use with arbitrary loudspeaker configurations. The methods and systems provide a dynamic binaural sound field rendering realized with the use of “virtual loudspeakers.” Rather than loudspeaker signals being fed into the physical loudspeakers, the signals are instead filtered with left and right HRIRs (Head Related Impulse Response) corresponding to the spatial locations of these loudspeakers. The sums of the left and right ear signals are then fed into the audio output device of the user.

Abstract: The disclosure relates to an audio signal processing apparatus for modifying a stereo image of a stereo signal. The apparatus includes a panning index modifier configured to apply a mapping function to at least all panning indexes of stereo signal time-frequency segments that are within a frequency bandwidth, a first panning gain determiner configured to determine modified panning gains for time-frequency signal segments of the first and second audio signal based on the modified panning indexes, and a re-panner configured to re-pan the stereo signal according to ratios between the modified panning gains and panning gains of the first and second audio signal that correspond to the modified panning gains in time and frequency.

Abstract: A non-transitory computer readable storage medium with instructions executable by a processor identify a center component, a side component and an ambient component within right and left channels of a digital audio input signal. A spatial ratio is determined from the center component and side component. The digital audio input signal is adjusted based upon the spatial ratio to form a pre-processed signal. Recursive crosstalk cancellation processing is performed on the pre-processed signal to form a crosstalk cancelled. The center component of the crosstalk cancelled signal is realigned to create the final digital audio output.

Abstract: Various aspects of this disclosure describe setting an audio compressor threshold using averaged audio measurements. Examples include calculating one or more average values of amplitude values of an audio file, and setting a threshold used in the audio compressor based on the calculated thresholds. Samples of the audio file with amplitude values above the threshold are attenuated, while samples of the audio file with amplitude values below the threshold are not attenuated. The threshold can be set equal to a calculated average value, or from a function of one or more calculated average values. Different audio channels comprising the audio file can be processed to set a respective compressor threshold for each audio channel.

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 non-causally shifting a channel includes estimating comparison values at an encoder. Each comparison value is indicative of an amount of temporal mismatch between a previously captured reference channel and a corresponding previously captured target channel. The method also includes smoothing the comparison values to generate smoothed comparison values based on historical comparison value data and a smoothing parameter. The method further includes estimating a tentative shift value based on the smoothed comparison values. The method also includes non-causally shifting a target channel by a non-causal shift value to generate an adjusted target channel that is temporally aligned with a reference channel. The non-causal shift value is based on the tentative shift value. The method further includes generating, based on reference channel and the adjusted target channel, at least one of a mid-band channel or a side-band channel.

Abstract: A method including determining location of at least one second device relative to a first device, where at least two of the devices are configured to play audio sounds based upon audio signals; and mixing at least two of the audio signals based, at least partially, upon the determined location(s).

Abstract: Accurate modeling of acoustic reverberation can be essential to generating and providing a realistic virtual reality or augmented reality experience for a participant. In an example, a reverberation signal for playback using headphones can be provided. The reverberation signal can correspond to a virtual sound source signal originating at a specified location in a local listener environment. Providing the reverberation signal can include, among other things, using information about a reference impulse response from a reference environment and using characteristic information about reverberation decay in a local environment of the participant. Providing the reverberation signal can further include using information about a relationship between a volume of the reference environment and a volume of the local environment of the participant.

Abstract: An audio file playing method and an apparatus are disclosed and are used to: when an audio file is played, expand a quantity of audio channel signals in the audio file and improve a playing effect of the audio file. The method is as follows: after the audio file is obtained, determining, whether the audio file includes an audio channel signal that can be played by the mobile device; if the audio file includes the audio channel signal that can be played by the mobile device, directly playing the audio channel signal. Therefore, when multiple mobile devices are used to play a same audio file, the mobile devices can avoid performing a same operation, thereby increasing a quantity of audio channels of the audio file, expanding a sound field of the audio file, and improving a playing effect of the audio file.

Abstract: When a channel signal, such as a 22.2 channel signal, is rendered into a 5.1 channel signal, a three-dimensional (3D) audio may be reproduced by using a two-dimensional (2D) output channel, however, when an elevation angle of an input channel is different from a standard elevation angle, if elevation rendering parameters according to the standard elevation angle are used, distortion may occur in a sound image.

Abstract: Pattern identification using convolution is described. In one or more implementations, a representation of a pattern is obtained that is described using data points that include frequency coordinates, time coordinates, and energy values. An identification is made as to whether sound data described using irregularly positioned data points includes the pattern, the identifying including use of a convolution of the frequency or time coordinates to determine correspondence with the representation of the pattern.

Abstract: Audio encoding methods/terminals, audio decoding methods/terminals, and audio codec systems are provided. A plurality of audio signals that are continuous is obtained. it is determined whether each audio signal of the plurality of audio signals includes a designated signal type, according to an audio parameter of each audio signal. A marked audio encoding stream is obtained by performing a marking to each audio signal as having or not having the designated signal type. The marking is used, at a decoding terminal, to perform an enhancement-process to one or more audio signals having the designated signal type. The enhancement-process is not performed to audio signals that do not have the designated signal type.

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: The present technology relates to an information processing device and method for allowing a sound image to be localized with higher precision, and a program. When a target sound image is outside a mesh, the target sound image is moved in a vertical direction while a position in a horizontal direction of the target sound image remains fixed, so that the target sound image is present on a boundary of the mesh. Specifically, a mesh detection unit detects a mesh including a position in the horizontal direction of the target sound image. A candidate position calculation unit calculates a position that is a movement target of the target sound image, based on loudspeaker positions that are at opposite ends of an arc of the detected mesh that is a destination, and the position in the horizontal direction of the target sound image. As a result, the target sound image can be moved onto a boundary of the mesh. The present technology is applicable to a sound processing device.

Abstract: Techniques of performing binaural rendering involve generating separate locations of virtual sources on the sphere for each ear of a listener. Along these lines, consider a set of actual audio sources that are not equidistant from a central point. To provide a listener with ambisonic audio, a sphere is defined with the listener at its center. When a source is not on the surface of the sphere, respective rays from the source to each of the listener's ears may not intersect the sphere at the same point. Rather, to provide a more accurate representation of the actual source, virtual loudspeakers are placed at each of the sphere intersections, a first virtual loudspeaker propagating audio to the left ear, a second virtual loudspeaker propagating audio to the right ear.

Abstract: A magnetic sensor mounted on a headband of the personal audio delivery device may output a sensor signal indicative of an interaction between a magnetic field of a transducer of a personal audio delivery device and the magnetic sensor. A head size of a head on which the personal audio delivery device is worn is calculated based on the sensor signal from the magnetic sensor. Based on the head size, a non-linear transfer function is identified which characterizes how sound is transformed via the head with the calculated head size. An output signal is generated indicative of one or more audio cues to facilitate spatialization of sound associated with the output signal based on the identified non-linear transfer function. The sound associated with the output signal is output by the transducer of the personal audio delivery device.

Abstract: A hearing aid system comprising a pair of hearing devices, e.g. hearing aids, worn at the ears of a user receives a target signal generated by a target signal source and transmitted through an acoustic channel to microphones of the hearing aid system. Due to (potential) additive environmental noise, a noisy acoustic signal is received at the microphones of the hearing system. An essentially noise-free version of the target signal is simultaneously transmitted to the hearing devices of the hearing system via a wireless connection.

Abstract: To prevent the measurement of the position of a speaker from not being performed in a case where the position of the speaker is changed, and prevent the measurement of the position of the speaker from being performed more than necessary. An AV amplifier (1) according to an aspect of the present invention includes: a control unit (105) that detects a change in the installation position of at least one speaker (2) that supplies an audio signal; and a measurement unit (106) that measures the installation position of at least the speaker (2) of which the installation position is changed among the plurality of speakers (2), automatically or according to a user operation, after the change is detected by the control unit (195).

Abstract: Embodiments of the present invention provide a method and an apparatus for simulating a sound in a virtual scenario, and a terminal. The method includes: determining whether a virtual object is in a falling state; detecting, when the virtual object is in the falling state, whether a virtual object collides with another virtual object in a virtual scenario; and invoking sound data when the virtual object collides with the another virtual object, and simulating a sound according to the invoked sound data.

Abstract: A method and apparatus for decompressing a Higher Order Ambisonics (HOA) signal representation is disclosed. The apparatus includes an input interface that receives an encoded directional signal and an encoded ambient signal and an audio decoder that perceptually decodes the encoded directional signal and encoded ambient signal to produce a decoded directional signal and a decoded ambient signal, respectively. The apparatus further includes an extractor for obtaining side information related to the directional signal and an inverse transformer for converting the decoded ambient signal from a spatial domain to an HOA domain representation of the ambient signal. The apparatus also includes a synthesizer for recomposing a Higher Order Ambisonics (HOA) signal from the HOA domain representation of the ambient signal and the decoded directional signal. The side information includes a direction of the directional signal selected from a set of uniformly spaced directions.

Abstract: The disclosure relates to binaural hearing instruments and more particularly to reduction of processing time required in a binaural hearing aid system. According to the disclosure, there is provided a method comprising mono-directional transmission of data blocks comprising audio and/or information frames from one hearing instrument to the other hearing instrument or vice versa in a binaural hearing aid. According to the disclosure, the direction of transmission is determined by a quantity characterizing the presence of usable information content in the sound signal picked up by the hearing instruments of the binaural hearing aid. It is proposed to use one or more of local SNR, local voice activity detection indication, local level, local speech intelligibility estimate to determine the direction of transmission, although other quantities may be used.

Abstract: Audio processor having an input interface, a detector interface, a mixer and an output interface. The input interface receiving at least two input audio channels, each input audio channel being associated with a predetermined reproduction position of at least two loudspeakers on at least one loudspeaker axis. The detector interface receiving a position signal indicating an information on a position of the at least two loudspeakers with respect to an ear axis of a listener, wherein the ear axis and the at least one loudspeaker axis have an angle to each other, being greater than 0° and lower than 180°. The mixer mixing the at least two input audio channels to obtain the at least two output channels depending on the position signal.

Abstract: An example method is performed by a media playback system comprising a plurality of audio drivers having a first radiation pattern. The method includes receiving data representing audio content, where each datum of the data indicates (i) a frequency and (ii) an amplitude corresponding to the frequency. The method further includes, for each audio driver of the plurality of audio drivers, determining a transfer function; processing each datum of the data based on (i) the frequency indicated by the given datum and (ii) the determined transfer function; and providing, to the given audio driver, a respective signal representing the data processed for the given audio driver, thereby causing the plurality of audio drivers to play back the audio content according to a second radiation pattern that is different from the first radiation pattern. An example media playback system and an example non-transitory computer-readable medium related to the example method is also disclosed herein.

Abstract: A speech processing device includes a sound source localization unit configured to determine a sound source position from acquired speech, a reverberation suppression unit configured to suppress a reverberation component of the speech to generate dereverberated speech, a feature quantity calculation unit configured to calculate a feature quantity of the dereverberated speech, a feature quantity adjustment unit configured to multiply the feature quantity by an adjustment factor corresponding to the sound source position to calculate an adjusted feature quantity, and a speech recognition unit configured to perform speech recognition using the adjusted feature quantity.

Abstract: In at least one embodiment, an apparatus for enhancing speech prompts in a vehicle is provided. The apparatus includes an audio processor that is electrically coupled to a plurality of loudspeakers in a vehicle. The audio processor being programmed to provide entertainment data to the plurality of loudspeakers for playback in the vehicle and to receive a speech prompt indicative of only a spoken audio output to a driver in the vehicle. The audio processor is further programmed to mute the entertainment data that is played back on a first loudspeaker of the plurality of loudspeakers in response to the speech prompt and to provide the speech prompt to the first loudspeaker for playback to the driver in response to the speech prompt.

Abstract: Provided is a method and apparatus for synthesizing a separated sound source, the method including generating spatial information associated with a sound source included in a frame of a stereo audio signal, and synthesizing a separated frequency-domain sound source from the frame of the stereo audio signal based on the spatial information, wherein the spatial information includes a frequency-azimuth plane representing an energy distribution corresponding to a frequency and an azimuth of the frame of the stereo audio signal.

Abstract: Monaurally-recorded mono or stereo recordings may be processed and converted into binaurally-recorded audio recordings. An analog process of performing this involves output of at least subsets of the monaurally-recorded recording, such as isolated instrument/vocal tracks, to be played to a dummy with two microphones. A digital process of performing this includes simulating audio input from simulated locations corresponding to audio sources. A neural network process of performing this includes training a neural network using speakers and microphones and then automating conversion from monaural audio to binaural audio based on the training of the neural network. The neural network can also be trained with output speakers to eliminate or reduce dead zones and/or speaker crosstalk.