69 results about "Wave field synthesis" patented technology

A device for determining a reproduction position of a source of sound for audio-visual reproduction of a film scene from a plurality of individual pictures with regard to a reproduction surface having a predetermined width and a projection source having a projection reference point comprises means for providing a recording position of the source of sound, a camera position during recording, and an aperture angle of the camera during recording. In addition, provision is made of means for transforming the recording position of the source of sound to a camera coordinate system, the origin of which is defined, in relation to a camera aperture, to obtain a recording position of the source of sound in the camera coordinate system. Means for calculating the reproduction position of the source of sound in relation to the projection reference point determines whether the aperture angle of the camera equals a predetermined aperture angle, and whether or not a source of sound is located within the visual range of the camera. If the current aperture angle of the camera differs from the predetermined standard aperture angle, the reproduction position of the source of sound is spaced toward a viewer, or away from the viewer, by a distance which depends on the ratio of the standard aperture angle to the current aperture angle. Hereby, automatable sound-source positioning is achieved so as to provide not only a visually realistic, but also an acoustically realistic situation in a reproduction room using wave-field synthesis methods.

The following coding scenario is addressed: A number of audio source signals need to be transmitted or stored for the purpose of mixing wave field synthesis, multi-channel surround, or stereo signals after decoding the source signals. The proposed technique offers significant coding gain when jointly coding the source signals, compared to separately coding them, even when no redundancy is present between the source signals. This is possible by considering statistical properties of the source signals, the properties of mixing techniques, and spatial hearing. The sum of the source signals is transmitted plus the statistical properties of the source signals which mostly determine the perceptually important spatial cues of the final mixed audio channels. Source signals are recovered at the receiver such that their statistical properties approximate the corresponding properties of the original source signals. Subjective evaluations indicate that high audio quality is achieved by the proposed scheme.

The apparatus for determining an impulse response in an environment in which a speaker and a microphone are placed works using an audio signal. Means for spectrally coloring a test signal, which preferably is a pseudonoise signal, works using a psychoacoustic masking threshold of the audio signal to obtain a colored test signal, which is embedded in the audio signal to obtain a measuring signal, which can be fed to the speaker. Means for determining the impulse response preferably performs a cross-correlation of a reaction signal received via the microphone from the environment and the test signal or the colored test signal. With this, an impulse response of an environment may also be determined during the presentation of an audio piece to provide an optimal description of environment for a wave-field synthesis.

An audio reproduction system is divided into a central wave-field synthesis module and a plurality of loudspeaker modules disposed in a distributed way, wherein synthesis signals for the individual loudspeakers as well as corresponding channel information associated to the synthesis signals are calculated in the central wave-field synthesis module. The synthesis signals for a loudspeaker as well as associated channel information will then be transmitted to respective loudspeaker modules via a transmission path, wherein every loudspeaker module obtains the synthesis signals and associated channel information intended for the loudspeaker associated to the loudspeaker module. A distributed audio rendering and digital/analog converting takes place in the loudspeaker module to generate the actually analog loudspeaker signals in a distributed way in spatial proximity to every loudspeaker. The division into a central wave-field synthesis module and the plurality of distributed loudspeaker modules allows that audio reproduction systems that are scalable with regard to the price can be generated in order to offer systems of different size scalable in price particularly for cinema reproduction rooms varying strongly in size.

Three-dimensional (3D) audio content creation and rendering systems and methodologies are presented here. A disclosed method of processing 3D audio assigns audio source objects to 3D video objects, links audio tracks to assigned audio source objects, and performs wave field synthesis on the linked audio tracks to generate 3D audio data representing a 3D spatial sound field. A disclosed method of processing 3D audio during playback of 3D video content obtains 3D audio data and 3D video data for a frame of 3D video content, applies device-specific parameters to the 3D audio data to obtain transformed 3D audio data scaled to a presentation device, and processes the transformed 3D audio data to render audio information for an array of speakers associated with the presentation device.

A perceptually motivated spatial decomposition for two-channel stereo audio signals, capturing the information about the virtual sound stage, is proposed. The spatial decomposition allows to re-synthesize audio signals for playback over other sound systems than two-channel stereo. With the use of more front loudspeakers, the width of the virtual sound stage can be increased beyond +/−30° and the sweet spot region is extended. Optionally, lateral independent sound components can be played back separately over loudspeakers on the two sides of a listener to increase listener envelopment. It is also explained how the spatial decomposition can be used with surround sound and wavefield synthesis based audio system. According to the main embodiment of the invention applying to multiple audio signals, it is proposed to generate multiple output audio signals (y₁ . . . y_M) from multiple input audio signals (x₁, . . . , x_L), in which the number of output is equal or higher than the number of input signals, this method comprising the steps of: —by means of linear combinations of the input subbands X₁(i), . . . , X_L(i), computing one or more independent sound subbands representing signal components which are independent between the input subbands, —by means of linear combinations of the input subbands X₁(i), . . . , X_L(i), computing one or more localized direct sound subbands representing signal components which are contained in more than one of the input subbands and direction factors representing the ratios with which these signal components are contained in two or more input subbands, —generating the output subband signals, Y₁(i) . . . Y_M(i), where each output subband signal is a linear combination of the independent sound subbands and the localized direct sound subbands—converting the output subband signals, Y₁(i) . . . Y_M(i), to time domain audio signals, y₁ . . . y_M.

For a level correction in a wave field synthesis system having a wave field synthesis module and an array of loudspeakers for providing sound to a presentation region, a correction value which is based on a set amplitude state in a presentation region is determined, the set amplitude state depending on a position of the virtual source or a type of the virtual source, and the actual amplitude state in the presentation region depending on the component signals for the loudspeakers due to the virtual source. The correction value determined is fed to a manipulator manipulating the audio signal associated to the virtual source before feeding to the wave field synthesis module, or the component signals for the individual loudspeakers due to the virtual source are manipulated to reduce a deviation between a set amplitude state and an actual amplitude state at one point or several points in the presentation region. Thus, level artifacts due to the finite number of loudspeakers in a wave field synthesis system are at least reduced such that a more pleasant sound experience for a listener is obtained.

Provided is an apparatus and method for reproducing a wave field using a loudspeaker array. A loudspeaker array may be configured in front of and behind a listener, and a wave field synthesis rendering and a three-dimensional sound image localization rendering may be performed based on a position of a sound source.

For reducing Doppler artifacts in the wave-field synthesis due to delay changes from one time to a second time, first, the delay for the first time and the delay for the second time are determined. Then, a value of an audio signal delayed by the first delay for the current time and the value for the audio signal delayed by the second delay for the current time are determined. Then, the first value is weighted by a first weighting factor and a second value is averaged with a second weighting factor, whereupon the two weighted values are added up to obtain a discrete value for the current time of the component in a loudspeaker signal for a loudspeaker based on a virtual source. Thus, by knowing a delay present at a later time, panning is obtained from a delay to a subsequent delay, which reduces undesired Doppler artifacts.

A sound system obtains a desired sound field from an array of sound sources arranged on a panel. The desired sound field allows a listener to perceive the sound as if the sound were coming from a live source and from a specified location. Setup of the sound system includes arranging a microphone array adjacent the array of sound sources to obtain a generated sound field. Arbitrary finite impulse response filters are then composed for each sound source within the array of sound sources. Iteration is applied to optimize filter coefficients such that the generated sound field resembles the desired sound field so that multi-channel equalization and wave field synthesis occur. After the filters are setup, the microphones may be removed.

The invention discloses a replay method of sound fields in a three-dimensional local space. In the method, based on a continuous sound source concept, to-be-replayed sound fields and replay sound fields from continuously-distributed replay sound sources are expanded to form spherical harmonic functions; coefficients of the spherical harmonic functions are matched to acquire driving functions of the continuously-distributed replay sound sources; the continuously-distributed replay sound sources and the driving functions thereof are discretized to obtain driving functions of an actually-discretized replay sound source array; and the driving functions are fed back to the replay sound sources for replaying the sound fields and suitable for any type of sound sources. Compared with a spherical harmonic function expansion method, the replay method is used to avoid the problems of matrix inversion and errors possibly caused by the matrix inversion and expand replay regions; and compared with a wave field synthesis method, the replay method can be used to improve the replay accuracy in micro-regions in which replay centers are contained.

A device for determining an impulse response in an environment provided with a speaker (10) and a microphone (12) operates using an audio signal. The means (20) for spectrally coloring a test signal, preferably a pseudo-noise signal, operates using a psychoacoustic masking threshold of the audio signal to obtain a colored test signal, and embedded in the audio signal to obtain a test signal that can be fed to the loudspeaker (10). And, preferably, the means (30, 32) for calculating the impulse response can perform a cross-correlation of a response signal received from the environment via the microphone and the test signal, or the colored test signal, Thereby, the impulse response of an environment can also be determined during the presentation of an audio signal to provide an ideal description of the environment for a wave-scene synthesis.

An apparatus for controlling a wave field synthesis renderer with audio objects includes a provider for providing a scene description, wherein the scene description defines a temporal sequence of audio objects in an audio scene and further includes information on the source position of a virtual source as well as on a start or an end of the virtual source. Furthermore, the audio object includes at least a reference to an audio file associated with the virtual source. The audio objects are processed by a processor, in order to generate a single output data stream for each renderer module, wherein both information on the position of the virtual source and the audio file itself are included in mutual association in this output data stream. With this, high portability on the one hand and high quality due to secure data consistency on the other hand are achieved.