Abstract: Metadata comprising a set of gain values for creating a dominance effect is automatically generated. Automatically generating the metadata includes receiving multiple audio streams and a dominance criterion for at least one of the audio streams. A set of gains is computed for one or more audio streams based on the dominance criterion for the at least one audio stream and metadata is generated with the set of gains.

Abstract: A sound system and method for capturing and reproducing sounds produced by a plurality of sound sources are disclosed. Sounds produced by the plurality of sound sources may be received and converted separately to separate audio signals without mixing the audio signals. The audio signals may be separately stored on a recording medium without mixing the audio signals. Such audio signals may be read from the recording medium and separately amplified using an amplification network of amplifiers. Loudspeakers in a loudspeaker network may be used to separately reproduce the sounds from the amplified audio signals. Components within a signal path may be dynamically controlled by a controller. The amplifiers and/or loudspeakers for each signal path may be customized based on the characteristics and complexities of the original sound to be reproduced on each signal path.

Abstract: A piezoelectric vibration module includes a piezoelectric element, a wiring member connected to the piezoelectric element and drawn out to the outside, and an elastic plate bonded to one surface of the piezoelectric element, wherein the elastic plate is made of a silicone rubber.

Abstract: A method, device and system enhance an audio signal in a receiving device. The method comprises acoustically propagating a target signal from an acoustic source along an acoustic propagation path, providing a propagated acoustic signal at the receiving device; converting the received propagated acoustic signal to a propagated electric signal, the received propagated acoustic signal comprising the target signal, noise and possible other sounds from the environment as modified by the propagation path from the acoustic source to the receiving device; wirelessly transmitting a signal comprising the target audio signal to the receiving device; receiving the wirelessly transmitted signal in the receiving device; retrieving a streamed target audio signal from the wirelessly received signal comprising the target audio signal; and estimating the target signal from the propagated electric signal and the streamed target audio signal using an adaptive system.

Abstract: A signal processor includes: a first adaptive filter that takes a first signal as input and generates a first pseudo signal; a first subtractor that subtracts the first pseudo signal from a second signal to supply a first differential signal as output; a second adaptive filter that takes the first signal as input to generate a second pseudo signal; a second subtractor that subtracts the second pseudo signal from the second signal to supply a second differential signal as output; a first step size control circuit that generates a first step size used in updating the first adaptive filter in accordance with the relation between the second pseudo signal and the second differential signal; and a second step size control circuit that generates a second step size used in updating the second adaptive filter in accordance with the relation between the first signal and the second signal.

Abstract: The object of the present invention is that the surplus vibration involved therewith simultaneously generating a sound wave by the vibration of the vibrating cone of the loudspeaker unit is generated in the loudspeaker system, which has impaired the sound quality of the loudspeaker system. The purpose of the present invention is to provide a technique for improving the sound quality by reducing the this surplus vibration which are the loudspeaker frame vibration and the cabinet vibration like a front baffle vibration. The present invention is characterized in being configured with a viscoelastic material layer upon rear portions such as the frame surface of the loudspeaker unit and inside and outside surfaces of each surface configuring the loudspeaker cabinet, and additionally suppresses vibration by addition of a heavy member or pressurization caused by constraint of a binding member.

Abstract: Earbud headphones configured for use with a portable media player that include a main body portion with an extended curvature configuration. In one example implementation, the earbud headphones include a speaker housing and an extension that form a gap between the speaker housing and the extension. A user can secure a portion of the user's outer ear in the gap (i.e., between the extension and speaker housing), such that the earbud is securely and comfortably held in place within the user's ear. Additionally, the earbud headphones can include interchangeable components, such that a user can customize the size, fit, and style of the earbud headphones.

Abstract: An embodiment of the present invention provides a method for generating a downmixed signal, including: performing a time-frequency transform on a received left sound channel signal and a received right sound channel signal to obtain a frequency domain signal, and dividing the frequency domain signal into several frequency bands; calculating a sound channel energy ratio and a sound channel phase difference of each frequency band; calculating a phase difference between the downmixed signal and a first sound channel signal in each frequency band according to the sound channel energy ratio and the sound channel phase difference; and calculating a frequency domain downmixed signal according to the left sound channel signal, the right sound channel signal, and the phase difference between the downmixed signal and the first sound channel signal in each frequency band. This method effectively improves quality of stereo encoding and decoding.

Abstract: A method of processing an audio signal, the method including receiving a downmix signal including at least one object signal, and object information extracted when the downmix signal is generated, the at least one object signal including a background object and at least one independent object; receiving mix information including mode selection information, the mix information for controlling the at least object signal; receiving enhanced object information corresponding to a residual signal generated when the at least one object signal is downmixed to the downmix signal; generating downmix processing information by using the object information and the mix information; generating an output signal by applying the downmix processed information to the downmix signal; and extracting the background object and the at least one independent object from the downmix signal, by using the enhanced object information.

Abstract: Media content typically includes closed captioning information such as subtitles in domestic and foreign languages. Techniques and mechanisms provide that closed captioning information may be toggled on/off using menu options and preferences as well as automatically managed by intelligently monitoring the environment surrounding a device. Device sensors such as microphones and vibration monitors determine the noise level of an environment as well as the spectral characteristics of the noise to determine whether the noise profile would interfere with the video playback experience. A particular environmental noise profile could automatically trigger the display of closed captioning information or present an easy access, otherwise unavailable toggle to display closed captioning information associated with a video stream.

Abstract: The present document relates to audio decoding and/or audio transcoding. In particular, the present document relates to a scheme for efficiently decoding a number M of audio channels from a bitstream comprising a higher number N of audio channels. In this context a method and system for merging a first and a second source set of spectral band replication (SBR) parameters to a target set of SBR parameters is described. The first and second source set comprise a first and second frequency band partitioning, respectively, which are different from one another. The first source set comprises a first set of energy related values associated with frequency bands of the first frequency band partitioning. The second source set comprises a second set of energy related values associated with frequency bands of the second frequency band partitioning. The target set comprises a target energy related value associated with an elementary frequency band.

Abstract: Systems and methods for using biometrics to select music preference are provided. A system for using biometrics to select music preferences for a user in a vehicle, comprises a music selection module electrically coupled to at least one biometric sensor in the vehicle, wherein the at least one biometric sensor senses a characteristic of the user and outputs data for the sensed characteristic to the music selection module, and wherein the music selection module selects a music selection for the user based on the sensed characteristic data, and a controller module electrically coupled to the music selection module to control playing of the music selection, wherein the controller module receives an output including the music selection from the music selection module.

Abstract: Provided is an electronic device including a housing and an acoustic device configured to generate vibration sound that is transmitted through vibration of a part of a human body. The acoustic device includes a panel configured to be joined to the housing; a piezoelectric element configured to be joined to the panel for vibrating the panel; and a vibration adjusting unit configured to be joined to the panel and vibrated together with the panel by the piezoelectric element. With the above configuration, a frequency characteristic is improved.

Abstract: There is provided a variable directional microphone including dynamic microphone units that is small in size and has good directional frequency response. In a variable directional microphone 1A in which a unidirectional first dynamic microphone unit (front-side unit) 1F and a second dynamic microphone unit (rear-side unit) 1R, which has substantially the same configuration as that of the front-side unit 1F and is provided with an output adjusting means of sound signal, are provided as a pair; the front-side unit 1F and the rear-side unit 1R are arranged coaxially so that the directivity axes thereof are directed to the directions 180° opposite to each other; and the output signals of the front-side unit 1F and the rear-side unit 1R are generated via a signal synthesis circuit, one rear air chamber 1b is used in common by the front-side unit 1F and the rear-side unit 1R.

Abstract: In one embodiment, the invention is a microphone system with an internal test circuit. The system includes a microphone having a housing with an acoustic port, a first transducer, a second transducer, a controller, and a current source. The system also includes an acoustic assembly with a cover and an acoustic pressure source positioned in the cover. When the acoustic assembly is positioned over the acoustic port, an acoustic chamber is formed, and a signal can be applied to the acoustic pressure source such that a first set of measurements can be taken. The acoustic assembly can be removed and replaced with an acoustic cover to take a second set of measurements. Based on the first and second measurements, sensitivities of the first and second transducers can be determined. In another embodiment, the invention provides a method for calibrating the sensitivity of a microphone.

Abstract: A sound reproduction system includes one or more arrays of drive units, coupled with sound processing allowing improved steerability, directional control, width control, and/or beam overlay. A speaker column may comprise two arrays facing one another, the drive units being perpendicular to the speaker unit front, with the acoustic output of each line array being compressed, turned and output from an elongate slot. The drive units may be staggered with respect to those in the opposing array. The arrays may be arc-shaped else straight. Selective delays to simulate an arced pattern. Differential delays applied to the drive units in a sub-array allow beam steering. Additional drive units, such as high frequency drivers, may be added along the length of the elongate slot or elsewhere to increase the dynamic frequency range of the speaker system. A collocated sound processor and amplifier output stage may be integrated with the speaker unit.

Abstract: Systems and methods for using biometrics to select music preference are provided. A system for using biometrics to select music preferences for a user in a vehicle, comprises a music selection module electrically coupled to at least one biometric sensor in the vehicle, wherein the at least one biometric sensor senses a characteristic of the user and outputs data for the sensed characteristic to the music selection module, and wherein the music selection module selects a music selection for the user based on the sensed characteristic data, and a controller module electrically coupled to the music selection module to control playing of the music selection, wherein the controller module receives an output including the music selection from the music selection module.

Abstract: Whereas a high masking effect can be secured in a space to which a masking sound is emitted, the degree of a discomfort a person existing in the space suffers can be reduced. In superimposition processing, a CPU 21 extracts sound signals in different intervals of a sound signal X12-n of a human voice, superimposes the extracted sound signals on each other on the time axis, and outputs a resulting superimposed sound signal X13-n. In shift and addition processing, the CPU 12 interchanges a sound signal, before a reference position, of a sound signal X16-n and a sound signal, after the reference position, of the sound signal X16-n (shift processing) and outputs a sound signal X17-n obtained by adding together a shift-processed sound signal X16?-n and the original, non-shift-processed sound signal X16-n.

Abstract: A processing circuit may implement an adaptive filter having a response that generates an anti-noise signal from a reference microphone signal, one or more filters for modeling an electro-acoustic path of the anti-noise signal from a location of an error microphone to an eardrum of a listener and having a response that generates a filtered reference microphone signal from the reference microphone signal, one or more filters for modeling an acoustic path of ambient audio sounds from the location of the error microphone to the eardrum and having a response that generates a synthesized playback corrected error signal based on the error microphone signal, wherein the synthesized playback corrected error signal is indicative of ambient audio sounds present at the eardrum, and a coefficient control block that shapes the response of the adaptive filter in conformity with the filtered reference microphone signal and the synthesized playback corrected error signal by adapting the response of the adaptive filter to min

Abstract: A method for continuously monitoring noise exposure level of a person via measuring sound level within an ear canal of the person using a sound measurement means linked to a sound level recording device external to the ear canal, wherein the sound measurement means may be shielded from the environment by a hearing protective device, and wherein the cable does not interfere with the ability of the hearing protective device to reduce noise exposure level in the ear canal. Also provided is a system comprising a sound measurement means mounted within an ear canal of the person, an external sound level recording means and a linking cable.