260 results about "Virtual source" patented technology

Multi-gate NOR flash thin-film transistor (TFT) string arrays are organized as three dimensional stacks of active strips. Each active strip includes a shared source sublayer and a shared drain sublayer that is connected to substrate circuits. Data storage in the active strip is provided by charge-storage elements between the active strip and a multiplicity of control gates provided by adjacent local word-lines. The parasitic capacitance of each active strip is used to eliminate hard-wire ground connection to the shared source making it a semi-floating, or virtual source. Pre-charge voltages temporarily supplied from the substrate through a single port per active strip provide the appropriate voltages on the source and drain required during read, program, program-inhibit and erase operations. TFTs on multiple active strips can be pre-charged separately and then read, programmed or erased together in a massively parallel operation.

An in vivo imaging device having an illumination system that creates a virtual source within a tissue region of a subject in a non-invasive manner. The illumination system transforms a maximum amount of illumination energy from a light source into a high contrast illumination pattern. The illumination pattern is projected onto the object plane in a manner that maximizes the depth to which clear images of sub-surface features can be obtained. The high intensity portion of the illumination pattern is directed onto the object plane outside the field of view of an image capturing device that detects the image. In this configuration, scattered light from within the tissue region interacts with the object being imaged. This illumination technique provides for a high contrast image of sub-surface phenomena such as vein structure, blood flow within veins, gland structure, etc.

A network routing system is provided that may establish a communication session between a local device and a remote destination. To establish a communication session, the local device generates a message directed to the remote destination. A traffic processor intercepts the message and redirects the message to a predefined network server in a secure fashion. The network server receives the message, and extracts the address for the remote destination, as well as message data. The network server has a pool of available virtual address that are registered with the network server. The network server associates a virtual source address with the real address of the local device or traffic processor, and opens a communication session. The virtual address is set as the source address for the message, and the message is sent to the remote destination. Since the source address is virtual, the real address for the local device or traffic processor are not sent to the remote location. If the remote location sends a return message, the return message will be directed to the virtual address. The network server receives the return message, and using the virtual address information, redirects the message to the real address for the traffic processor in a secure fashion. The traffic processor sets the address on the message to show it was sent from the remote destination, and forwards the message to the local device.

A method for sound field reproduction into a listening area of spatially encoded first audio input signals according to sound field description data using an ensemble of physical loudspeakers. The method includes computing reproduction subspace description data from loudspeaker positioning data describing the subspace in which virtual sources can be reproduced with the physically available setup. Then, second and third audio input signals with associated sound field description data, in which second audio input signals include spatial components of the first audio input signals located within the reproducible subspace and third audio input signals include spatial components of the first audio input signals located outside of the reproducible subspace. A spatial analysis is performed on second audio input signals to extract fourth audio input signals corresponding to localizable sources within the reproducible subspace with associated source positioning data. Components of second audio input signals after spatial analysis are merged with third audio input signals into fifth audio input signals with associated sound field description data for reproduction within the reproducible subspace. Loudspeaker alimentation signals are computed from fourth and fifth audio input signals.

Aspects of the invention provide methods, computer-readable media, and apparatuses for re-panning multiple audio signals by applying spatial cue coding. Sound sources in each of the signals may be re-panned before the signals are mixed to a combined signal. Processing may be applied in a conference bridge that receives two omni-directionally recorded audio signals. The conference bridge subsequently re-pans one of the signals to the listeners left side and the signal to the right side. The source image mapping and panning may further be adaptively based on the content and use case. Mapping may be done by manipulating the directional parameters prior to directional decoding or before directional mixing. Directional information that is associated with an audio input signal is remapped order to compress input source positions into virtual source positions. The virtual sources may be placed with respect to actual loudspeakers using binaural cue panning.

A sound reproducing apparatus and a sound reproducing method. The sound reproducing apparatus includes an actual listening environment feature function database where an actual listening space feature function is stored for correcting the virtual source in response to a feature of an actual listening space provided at the time of listening; and an actual listening space feature correcting unit of reading out the actual listening space feature function stored in the actual listening environment feature function database, and correcting the virtual source based on the reading result. Accordingly, causes of each distortion may be removed to provide sounds having the best quality.

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.

A nitride read only memory cell comprising a silicon-germanium layer with a pair of source/drain regions. A strained silicon layer is formed overlying the silicon-germanium layer such that the pair of source/drain regions is linked by a channel that is generated in the strained silicon layer during operation of the cell. A nitride layer is formed overlying the substrate. The nitride layer has at least one charge storage region. The nitride layer may be a planar layer, a planar split gate nitride layer, or a vertical split nitride layer. A control gate is formed overlying the nitride layer. Ballistic direct injection is used to program the memory cell. A first charge storage region of the nitride layer establishes a virtual source/drain region in the channel. The virtual source/drain region has a lower threshold voltage than the remaining portion of the channel.

A method of recording sound for reproduction by a plurality of loudspeakers, or for processing sound for reproduction by a plurality of loudspeakers, is described in which some of the reproduced sound appears to a listener to emmanate from a virtual source which is spaced from the loudspeakers. A filter means (H) is used either in creating the recording, or in processing the recorded signals for supply to loudspeakers, the filter means (H) being created in a filter design step in which: a) a technique is employed to minimise error between the signals (w) reproduced at the intended position of a listener on playing the recording through the loudspeakers, and desired signals (d) at the intended position, wherein: b) said desired signals (d) to be produced at the listener are defined by signals (or an estimate of the signals) that would be produced at the ears of (or in the region of) the listener in said intended position by a source at the desired position of the virtual source. A least squares technique may be employed to minimise the time averaged error between signal reproduced at the intended position of a listener and the desired signal, or it may be applied to the frequency domain.

Method of imaging seismic data recorded using a set of seismic receivers positioned to receive seismic responses from a subsurface formation upon activating a seismic source. The seismic responses are direction-sensitive responses. A virtual signal received by receiver k from the virtual source at the position of receiver m is determined by processing involving cross correlating at least part of the responses of a receiver m with at least part of the responses of a seismic receiver k. Part of direction-sensitive responses or part of the virtual signal that contains a component of the wave field at receiver m traveling in a first direction and part of the direction-sensitive responses or part of the virtual signal that contains a component of the wave field at the receiver k traveling in a second direction may be removed.

The invention aims at providing a sound field rebuilding method. The sound field rebuilding method comprises the following steps that firstly, a sound pressure value of a measuring face in a measuring sound field is measured; secondly, different sound sources are endowed with a certain weight through the centroid method according to the actual positions of the sound sources and the sound intensity, the positions and number of multi-ball virtual source intensities and the position coordinates of each virtual ball are calculated; thirdly, zero filling expanding is carried out on the measured sound pressure value of the measuring face to obtain the sound pressure value of an expanding face of the measuring face; fourthly, a transfer matrix between the sound pressure value of the expanding face of the measuring face and the virtual ball sound pressure is calculated; fifthly, a transfer matrix between the sound pressure value of a rebuilding face and the sound pressure value of the multi-ball virtual equivalent source intensities is built; sixthly, the transfer matrix obtained in the fourth step and the transfer matrix obtained in the fifth step are combined, and the sound pressure value of the rebuilding face is obtained. According to the sound field rebuilding method, multi-ball region two-dimensional fast Fourier conversion algorithm is introduced, a virtual source intensities configuration and the measuring face do not need to be conformal, and the calculation efficiency and calculation precision are very high.

The invention provides a method and device for simulating detection efficiency of a calibration detector by adopting a virtual source. The virtual source is used for replacing gas for efficiency calibration, and the efficiency split principle, an collimation conical point source and a virtual point source in a Monte Carlo program and a line source are ingeniously applied to aerial survey Monte Carlo calibration simulation work. The method and device can replace a body source to do the daily calibration work of labs and field instruments, that is, energy calibration and efficiency calibration are achieved through one-time experiment calibration, and therefore calibration work links are reduced, and source purchasing cost is lowered.

A seismic velocity profile in a region of interest in a subsurface formation is determined using at least the following steps. (a) Activating a seismic source at a location n, thereby exciting a wave in the subsurface formation.

(b) Recording a wave signal trace u_nm(t) against time t, at a seismic receiver m.

(c) Recording a wave signal trace u_nk(t) against time t at a seismic receiver k.

(d) Cross correlating the wave signal traces u_nm(t) and u_nk(t) to obtain u^conv_nmnk(t). (e) Repeating these steps, for different locations n;

(f) Summing u^conv_nmnk(t) over all locations n, to obtain a signal trace u^vs_mk(t) which corresponds to the signal received by the seismic receiver k from the virtual source at the position of seismic receiver m;

(g) Deriving the seismic velocity based on the time of first arrival of the wave in u^vs_mk(t) and the predetermined distance between the seismic receiver m and the seismic receiver k.

Provided is a method and apparatus for encoding/decoding a multi-channel audio signal. The apparatus for encoding a multi-channel audio signal includes a frame converter for converting the multi-channel audio signal into a framed audio signal; means for downmixing the framed audio signal; means for encoding the downmixed audio signal; a source location information estimator for estimating source location information from the framed multi-channel audio signal; means for quantizing the estimated source location information; and means for multiplexing the encoded audio signal and the quantized source location information, to generate an encoded multi-channel audio signal.

Disclosed herein is a method of reconstructing a sound field. The method comprises receiving measured values indicative of a first acoustic quantity measured at a set of measurement locations; defining a set of virtual source locations; and computing a second acoustic quantity for at least one target location from one or more wave functions each representative of a respective sound field originating from a respective one of the defined set of virtual source locations; wherein the one or more wave functions are weighted by respective one or more weighting factors, and wherein computing comprises determining the one or more weighting factors from a least-norm fit of the one or more wave functions to the received measured values.

Provided is a method of separating an audio object that includes extracting virtual source location information and an audio signal from a bitstream, separating an object included in the audio signal based on a virtual source location, mapping objects of a previous frame and objects of a current frame located at the virtual source location, and extracting the mapped objects between continuous frames.

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.