摘要:
The present disclosure generally relates to an audio interface arrangement. In one embodiment, a handheld device comprises the audio interface arrangement (not shown). The audio interface arrangement comprises at least two audio connecting means, wherein each of the at least two audio connecting means is adapted to connect a respective audio accessory to the audio interface arrangement. The audio interface arrangement further comprises an accessory determining means, which is coupled to said at least two audio connecting means. The accessory determining means is adapted to determine which type of audio accessory is connected to the respective audio connecting means. Hereby it is made possible to provide a handheld device that offers a user with a possibility to connect several different audio accessories to the handheld device, which accessories may be connected to the handheld device at the same time.
摘要:
A sound masking system according to the invention is disclosed in which one or more sound masking loudspeaker assemblies are coupled to one or more electronic sound masking signal generators. The loudspeaker assemblies in the system of the invention have a low directivity index and preferably emit an acoustic sound masking signal that has a sound masking spectrum specifically designed to provide superior sound masking in an open plan office. Each of the plurality of loudspeaker assemblies is oriented to provide the acoustic sound masking signal in a direct path into the predetermined area in which masking sound is needed. In addition, the sound masking system of the invention can include a remote control function by which a user can select from a plurality of stored sets of information for providing from a recipient loudspeaker assembly an acoustic sound masking signal having a-selected sound masking spectrum.
摘要:
An audio signal processing apparatus includes: an obtaining unit which obtains a stereo signal including an R signal and an L signal; a control unit which generates a processed R signal and a processed L signal by performing (i) a first process of convolving pairs of right- and left-ear head related transfer functions into the R signal so that a sound image of the R signal is localized at each of two or more different positions at a right side of a listener; and (ii) a second process of convolving pairs of right- and left-ear head related transfer functions into the L signal so that a sound image of the L signal is localized at each of two or more different positions at a left side of the listener; and an output unit which outputs the processed R signal and the processed L signal.
摘要:
In a system and method for maintaining the spatial stability of a sound field a balance gain may be calculated for two or more microphone signals. The balance gain may be associated with a spatial image in the sound field. Signal values may be calculated for each of the microphone. The signal values may be signal estimates or signal gains calculated to improve a characteristic of the microphone signals. The differences between the signal values associated with each microphone signal may be limited although some difference between signal values may be allowable. One or more microphone signals are adjusted responsive to the two or more balance gains and the signal gains to maintain the spatial stability of the sound field. The adjustments of one or more microphone signals may include mixing of two or more microphone. The signal gains are applied to the two or more microphone signals.
摘要:
An augmented-reality audio system generates information regarding the acoustic environment by sampling audio signals. Using a Gaussian mixture model or other technique, the system identifies the location of one or more audio sources, with each source contributing an audio component to the sampled audio signals. The system determines a reverberation time for the acoustic environment using the audio components. In determining the reverberation time, the system may discard audio components from sources that are determined to be in motion, such as components with an angular velocity above a threshold or components having a Doppler shift above a threshold. The system may also discard audio components from sources having an inter-channel coherence above a threshold. In at least one embodiment, the system renders sounds using the reverberation time at virtual locations that are separated from the locations of the audio sources.
摘要:
The present invention provides a tablet woofer, including: a vibration diaphragm, at least four mutually independent driving units, and a housing for accommodating and securing the vibration diaphragm and the driving unit. The driving unit includes independent voice coil units and magnetic circuit units, the voice coil units being attached and fixed to the vibration diaphragm. The woofer further includes a FPCB communicating with an internal circuit and an external circuit, and the FPCB is disposed at two sides of the housing, the part of the FPCB disposed on the side of the housing close to the voice coil units being electrically connected to the voice coil units, the part of the FPCB disposed on the side of the housing far away from the voice coil units being electrically connected to the external circuit. The woofer can solve the problems of thick and heavy voice coils and magnet performance surplus.
摘要:
In accordance with embodiments of the present disclosure, a method for processing audio information in an audio device may include reproducing audio information by generating an audio output signal for communication to at least one transducer of the audio device, receiving at least one input signal indicative of ambient sound external to the audio device, detecting from the at least one input signal a near-field sound in the ambient sound, and modifying a characteristic of the audio information reproduced to the at least one transducer in response to detection of the near-field sound.
摘要:
Processor-implemented methods and systems for visually-assisted mixing of audio using a spectral analyzer are disclosed. The system calculates and displays a spectral view for each track in an arrangement in a multi-track view. A user can then request modification of the spectral view. In response to this request for modification, the system automatically adjusts associated mixing parameters for the modified track so that the spectral output of the track substantially matches the user-requested modified spectral view. This allows a user to visually mix an arrangement by imputing a desired spectral result and having the program make the changes necessary to achieve it.
摘要:
A method for identifying the position of loudspeaker boxes in a loudspeaker box arrangement comprises the operation of a first loudspeaker box in the loudspeaker box arrangement as an acoustic test signal generator; the reception of the acoustic test signal from the first loudspeaker box at other loudspeaker boxes in the loudspeaker box arrangement; and the ascertainment of a positional relationship between the signal-generating first loudspeaker box and at least one of the other loudspeaker boxes in the loudspeaker box arrangement on the basis of the acoustic test signal received in the other loudspeaker boxes.
摘要:
A directional coding conversion method and system includes receiving input audio signals that comprise directional audio coded signals into which directional audio information is encoded according to a first loudspeaker setup and extracting the directional audio coded signals from the received input audio signals. The method and system further includes decoding, according to the first loudspeaker setup, the extracted directional audio coded signals to provide at least one absolute audio signal and corresponding absolute directional information and processing the at least one absolute audio signal and the absolute directional information to provide first output audio signals coded according to a second loudspeaker setup.