Abstract: Described herein are systems and method for providing an immersive audio experience in a computer-generated virtual environment. An audio-mixer is placed at a location in a domain of the virtual environment and receives a separate audio feed from each of the audio source nodes directly connected to the audio-mixer. The audio-mixer mixes the received audio feeds in dependence on corresponding location, position and intrinsic loudness information to generate a separate spatialized mixed audio feed for each of the directly connected audio listener nodes. The audio-mixer sends, to each of the audio listener nodes directly connected to the audio-mixer, the separate spatialized mixed audio feed generated for the audio listener. An audio-mixer also receives and sends non-spatialized mixed audio feeds to/from other audio-mixers. The location of the audio-mixer is preferably moved to account for audio source nodes moving and/or changes to which audio source nodes are directly connected to the audio-mixer.

Abstract: Soundfield signals such as e.g. Ambisonics carry a representation of a desired sound field. The Ambisonics format is based on spherical harmonic decomposition of the soundfield, and Higher Order Ambisonics (HOA) uses spherical harmonics of at least 2nd order. However, commonly used loudspeaker setups are irregular and lead to problems in decoder design. A method for improved decoding an audio soundfield representation for audio playback comprises calculating (110) a function (W) using a geometrical method based on the positions of a plurality of loudspeakers and a plurality of source directions, calculating (120) a mode matrix (?) from the loudspeaker positions, calculating (130) a pseudo-inverse mode matrix (?+) and decoding (140) the audio soundfield representation. The decoding is based on a decode matrix (D) that is obtained from the function (W) and the pseudo-inverse mode matrix (?+).

Abstract: The present invention is intended to be used in a variety of situations where the user desires to have a high-quality portable audio device. On the market today, there cannot be found a rugged, portable audio device aimed at average users and containing expected elements such as (1) high-quality audio components, (2) an internal power source, (3) an integrated fool-proof charging and power management system and (4) the ability to protect the internal components while still retaining the intended purpose of the original container. The present invention contains the elements above and utilizes them in a way that is different from all previous inventions. All other inventions appear to be expensive, hefty in size and weight, require separate purchase of power sources and charging systems and eliminate the intended purpose of the original container.

Abstract: An apparatus comprising: a housing containing a speaker for delivering sound waves; and a sensing element for detecting air movement in the housing and further providing a control signal based on the detected air movement. The control signal is sent to a host apparatus. A user interface operation dependent on the control signal is preformed, such as filtering, muting or amplifying an input audio signal to the speaker or switching off the host apparatus. The housing comprises compressible material such that when the compressible material is displaced air movement in the housing is detected by the sensing element.

Abstract: An infrasonic stethoscope for monitoring physiological processes of a patient includes a microphone capable of detecting acoustic signals in the audible frequency bandwidth and in the infrasonic bandwidth (0.03 to 1000 Hertz), a body coupler attached to the body at a first opening in the microphone, a flexible tube attached to the body at a second opening in the microphone, and an earpiece attached to the flexible tube. The body coupler is capable of engagement with a patient to transmit sounds from the person, to the microphone and then to the earpiece.

Abstract: Disclosed is a computer system comprising an audio processing module, an echo cancellation module and a control module. The audio processing module is configured to process an audio signal and output the processed audio signal to e.g. a speaker (or similar) of the system. The echo cancellation module is configured to cancel echo from an audio signal received via e.g. a microphone (or similar) of the system. The control module is configured to detect a condition indicative of nonlinear amplitude processing by the audio processing module and control said echo cancellation by the echo cancellation module based on the detected condition.

Abstract: The present invention regards a hearing system configured to be worn by a user comprising an environment sound input unit, an output transducer, and electric circuitry. The environment sound input unit is configured to receive sound from the environment of the environment sound input unit and to generate sound signals representing sound of the environment. The output transducer is configured to stimulate hearing of a user. The electric circuitry comprises a spatial filterbank. The spatial filterbank is configured to use the sound signals to generate spatial sound signals dividing a total space of the environment sound in subspaces. Each spatial sound signal represents sound coming from a subspace. The subspaces may (in particular modes of operation) be either fixed, or dynamically determined, or a mixture thereof.

Abstract: The invention relates to a method and the associated device 1 for separating one or more particular digital audio source signals (si) contained in a mixed multichannel digital audio signal (smix) obtained by mixing a plurality of digital audio source signals (s1, . . . , sp). According to the invention: the modulus of the amplitude or the normalized power of the particular source signal(s) (si) is determined from representative values of said particular source signal(s) contained in the mixed signal; and then linearly constrained minimum variance spatial filtering is performed on the mixed signal in order to obtain each particular source signal (s?i), said filtering being based on the distribution of said particular source signal between at least two channels of the mixed signal, and the modulus of the amplitude or the normalized power of said particular source signal is used as a linear constraint of the filter.

Abstract: An exemplary uncomfortable sound pressure estimation system includes an HTL section for receiving information of hearing threshold levels respectively for a plurality of frequencies; a UCL measurement determination section for determining at least two measurement points at which to measure uncomfortable sound pressures, the at least two measurement points defining a first and a second frequency concerning a first and a second hearing threshold level; a UCL measurement section for measuring uncomfortable sound pressures with respect to the at least two measurement points; a calculation section for calculating a criterion concerning auditory characteristics by using the respective hearing threshold levels and uncomfortable sound pressures for the at least two points; and a UCL estimation section for estimating an uncomfortable sound pressure at a frequency corresponding to a given acquired hearing threshold level, according to the predetermined criterion.

Abstract: An audio data processing device includes a mixing unit that mixes first audio data and second audio data to generate mixed audio data, a first output unit that outputs, to a first speaker, audio data not having undergone the mixing, a second output unit that outputs, to a second speaker, the mixed audio data generated by the mixing unit, a processor which is provided between a point of branching to the mixing unit and the first output unit or between the mixing unit and the second output unit and adds an acoustic effect, and a delay unit which is provided between the branching point and the first output unit or between the mixing unit and the second output unit and delays the audio data so that a relationship of timing between input, to the first speaker, of the audio data output from the first output unit and input, to the second speaker, of the audio data output from the second output unit coincides with a relationship of timing of input between the first audio data and the second audio data.

Abstract: Systems and methods for removing or reducing distortion in an ultrasonic audio system can include receiving a first audio signal, wherein the first audio signal represents audio content to be reproduced using the ultrasonic audio system; calculating a first error function for the ultrasonic audio system, the first error function comprising an estimate of distortion introduced by reproduction of the audio content by the ultrasonic audio system; transforming the first audio signal into a first pre-conditioned audio signal by combining the first error function with the first audio signal; and modulating the transformed audio signal onto an ultrasonic carrier.

Abstract: A system and method of matching reverberation in teleconferencing environments. When the two ends of a conversation are in environments with differing reverberations, the method filters the reverberation so that when both signals are output at the near end (e.g., the audio signal from the far end and the sidetone from the near end), the reverberations match. In this manner, the user does not perceive an annoying difference in reverberations, and the user experience is improved.

Abstract: The method of playing back a multichannel audio signal via a playback device comprises a plurality of loudspeakers arranged at fixed locations of the device and define a spatial window for sound playback relative to a reference spatial position. The method comprises for at least one sound object extracted from the signal, estimating a diffuse or localized nature of the object and estimating its position relative to the window. The audio signal is played back via the loudspeakers of the device during which playback treatment is applied to each sound object for playing back via at least one loudspeaker of the device, which treatment depends on the diffuse or localized nature of the object and on its position relative to the window, and includes creating at least one virtual source outside the window from loudspeakers of the device when the object is estimated as being diffuse or positioned outside the window.

Abstract: A display device includes a housing, a protective glass, a display panel, a vibration glass layer, and a fastening frame. The display panel is positioned on the housing. The vibration glass layer includes a transparent portion on the display panel and a sound generation portion extending from sides of the transparent portion to the housing. The fastening frame is positioned on the vibration glass layer between the vibration glass layer and the protective glass. The fastening frame includes a base portion and rib portions. The base portion surrounds the transparent portion. The rib portions extend from the base portion to the housing. The rib portions divide the sound generation portion into sound generation regions. The sound generation regions have different areas.

Abstract: Embodiments described herein may involve dynamically adjusting the equalization of a playback device based on the environment in which the playback device is operating. An example embodiment may involve detecting, by at least one microphone, an audio signal. The example embodiment may also involve selecting a playback device environment from among two or more playback device environments based on the detected audio signal. The example embodiment may further involve identifying an audio setting corresponding to the selected playback device environment; and applying the audio setting when playing media content.

Abstract: An audio processing device includes a reverb property estimating unit that estimates a reverb property at each frequency on the basis of a first audio signal and a second audio signal representing sounds of the first audio signal output by an audio output unit and collected by an audio input unit, a gain calculating unit that determines an attenuating ratio for a component of the first audio signal at each frequency such that the larger the reverb property at the frequency is, the larger the attenuating ratio for the component at the frequency becomes, and a correcting unit that attenuates the first audio signal at the each frequency in accordance with the attenuating ratio determined for each frequency.

Abstract: The electronic stethoscope device has a Medical Acoustic Collector (MAC) used to auscultate a patient. The MAC microphone communicates with a bio-signal filter/amplifier that communicates with a transmitter. A plurality of separable auditory implements includes earbuds and an earpiece. Each of the earbuds and the earpiece are removably docked within the MAC. The auditory implements communicate with the MAC via a Wireless Personal Area Network (WPAN) which includes but is not limited to a Bluetooth® an Infrared Data Acquisition, and a Body Area Network. The device also selectively communicates with existing wireless media capable devices.

Abstract: An approach to processing of acoustic signals acquired at a user's device include one or both of acquisition of parallel signals from a set of closely spaced microphones, and use of a multi-tier computing approach in which some processing is performed at the user's device and further processing is performed at one or more server computers in communication with the user's device. The acquired signals are processed using time versus frequency estimates of both energy content as well as direction of arrival. In some examples, a non-negative matrix or tensor factorization approach is used to identify multiple sources each associated with a corresponding direction of arrival of a signal from that source. In some examples, data characterizing direction of arrival information is passed from the user's device to a server computer where direction-based processing is performed.

Abstract: Disclosed are an apparatus and method of processing an audio signal to optimize audio for a room environment. One example method of operation may include recording the audio signal generated within a particular room environment and processing the audio signal to create an original frequency response based on the audio signal. The method may also include identifying a target sub-region of the frequency response which has a predetermined area percentage of a total area under a curve generated by the frequency response, determining whether the target sub-region is a narrow energy region, creating a filter to adjust the frequency response, and applying the filter to the audio signal.

Abstract: An audio signal processing method includes: obtaining an L signal including a sound localized closer to the left as a major component and an R signal including a sound localized closer to the right as a major component; extracting a first signal which is a component of a sound included in the L signal and localized closer to the right and a second signal which is a component of a sound included in the R signal and localized closer to the left; generating a first output signal by subtracting the first signal from the L signal and adding the second signal to the L signal and a second output signal by subtracting the second signal from the R signal and adding the first signal to the R signal; and outputting the first output signal and the second output signal.