摘要:
The present invention includes a filtering unit configured to perform filtering on input audio signals in accordance with coefficients set for a plurality of audio outputs, respectively, and a plurality of speakers configured to output audio in response to audio signals filtered for the plurality of audio outputs. The filtering unit performs the filtering on the input audio signals in accordance with the coefficients adjusted so that the sound pressures and the sound pressure gradients at a plurality of control points set on a boundary plane of an enclosed space surrounding the plurality of speakers reach values corresponding to a desired sound field. This allows producing a desired sound field on the boundary plane of the enclosed space surrounding the sound sources by making use of the properties of the Kirchhoff-Helmholtz integral equation that requires that the sound sources are present in an enclosed space.
摘要:
Provided are a de-reverberation control method and apparatus for a device equipped with a microphone. The method includes: reverberation parameters which indicate, at respective moments, reverberation levels of a room environment where the device is located are acquired from an audio signal played by the device,; and a de-reverberation mode adopted by the device is dynamically adjusted according to the reverberation levels indicated by the reverberation parameters at different moments and preset correspondences between reverberation levels and de-reverberation modes. By adopting a dynamic de-reverberation mode, the method and the apparatus disclosed herein significantly improve the rate of the recognition of a device for the voice of the user.
摘要:
Provided are a de-reverberation control method and apparatus for a device equipped with a microphone. The method includes: reverberation parameters which indicate, at respective moments, reverberation levels of a room environment where the device is located are acquired from an audio signal played by the device,; and a de-reverberation mode adopted by the device is dynamically adjusted according to the reverberation levels indicated by the reverberation parameters at different moments and preset correspondences between reverberation levels and de-reverberation modes. By adopting a dynamic de-reverberation mode, the method and the apparatus disclosed herein significantly improve the rate of the recognition of a device for the voice of the user.
摘要:
An apparatus for mapping a first input channel and a second input channel of an input channel configuration to at least one output channel of an output channel configuration, wherein each input channel and each output channel has a direction in which an associated loudspeaker is located relative to a central listener position, wherein the apparatus is configured to map the first input channel to a first output channel of the output channel configuration. The apparatus is further configured to at least one of a) map the second input channel to the first output channel, comprising processing the second input channel by applying at least one of an equalization filter and a decorrelation filter to the second input channel, and b) despite of the fact that an angle deviation between a direction of the second input channel and a direction of the first output channel is less than an angle deviation between a direction of the second input channel and the second output channel and/or is less than an angle deviation between the direction of the second input channel and the direction of the third output channel, map the second input channel to the second and third output channels by panning between the second and third output channels.
摘要:
The invention relates to a method in which, primarily in listening rooms, instead of the spatial sound of the respective environment, the spatial sound of a third room can be made substantially enveloping in the perception of the listener. The third room spatial acoustics perceived instead of the listening room spatial acoustics act in the same way in terms of hearing physiology as the spatial acoustics of natural rooms, namely enveloping in terms of the spatial sound (in distinction to direct sound). This type of spatial enveloping effect can also be incorporated emotionally by the listener into the auditory event, and has no sweet spot problem (there is a large preferred listening area instead of a preferred listening spot) and lets deep frequencies take effect in a way through which subwoofer speakers can generally be omitted.
摘要:
Disclosed is a method for determining filter coefficients of an audio precompensation controller for the compensation of an associated sound system, including N≧2 loudspeakers, including estimating, for each one of at least a pair of the loudspeakers, a model transfer function at each of M control points distributed in Z≧2 spatially separated listening zones in a listening environment of the sound system. The method also includes determining, for each of the M control points, a zone-dependent target transfer function at least based on the zone affiliation of the control point; and determining the filter coefficients of the audio precompensation controller at least based on the model transfer functions and the target transfer functions of the M control points. Consequently, an audio precompensation controller for an associated sound system can be obtained that enables improved and/or customized sound reproduction in two or more listening zones simultaneously.
摘要:
A system and method for processing and enhancing utility of a sound mask noise signal, including generating, by a signal processor, the sound mask noise signal by modulating a noise signal with embedded additional information; outputting, by a plurality of audio speakers, sound signals comprising the sound mask noise signal with the embedded additional information; and receiving, by one or more microphones, the outputted sound signals comprising the sound mask noise signal, wherein an impulse response between each audio speaker and each microphone is measured in real time based on the embedded additional information.
摘要:
Various methods and systems for estimating a room impulse response between an audio source and an array of microphones are described. In one example, a method includes receiving audio signals at a microphone of an array of microphones. The audio signals correspond to each of the microphones in the array of microphones. The method also includes determining a room impulse response in a subspace that is compatible with a geometry of the array of microphones based on the received audio signals.