29018 results about "Time domain" patented technology

A method for the detection of analytes using resonant mass sensors or sensor arrays comprises frequency encoding each sensor element, acquiring a time-domain resonance signal from the sensor or sensor array as it is exposed to analyte, detecting change in the frequency or resonant properties of each sensor element using a Fourier transform or other spectral analysis method, and classifying, identifying, and/or quantifying analyte using an appropriate data analysis procedure. Frequency encoded sensors or sensor arrays comprise sensor elements with frequency domain resonance signals that can be uniquely identified under a defined range of operating conditions. Frequency encoding can be realized either by fabricating individual sensor elements with unique resonant frequencies or by tuning or modifying identical resonant devices to unique frequencies by adding or removing mass from individual sensor elements. The array of sensor elements comprises multiple resonant structures that may have identical or unique sensing layers. The sensing layers influence the sensor elements' response to analyte. Time-domain signal is acquired, typically in a single data acquisition channel, and typically using either (1) a pulsed excitation followed by acquisition of the free oscillatory decay of the entire array or (2) a rapid scan acquisition of signal from the entire array in a direct or heterodyne configuration. Spectrum analysis of the time domain data is typically accomplished with Fourier transform analysis. The methods and sensor arrays of the invention enable rapid and sensitive analyte detection, classification and/or identification of complex mixtures and unknown compounds, and quantification of known analytes, using sensor element design and signal detection hardware that are robust, simple and low cost.

A number of systems and methods, alone, or in combination, achieve various levels of protection against unauthorized modification and distribution of digital content. This encompasses at least unauthorized study, modification, monitoring, reconstruction, and any other means for subversion from the originally intended purpose and license model of the digital content. The invention combines a number of techniques that in whole, or in part, serve to protect such content from unauthorized modification, reconstructive engineering, or monitoring by third parties. This is accomplished by means of methods which protect against subversion by specific tools operating on specific platforms as well as general tools operating on general platforms. Specific time domain attacks are identified, code modification can be identified and reversed, and virtual and emulated systems are identified. In addition, identification of in-circuit emulator tools (both software and hardware), debuggers, and security threats to running programs can be achieved.

An audio encoder applies an adaptive block-encoding process to segments of audio information to generate frames of encoded information that are aligned with a reference signal conveying the alignment of a sequence of video information frames. The audio information is analyzed to determine various characteristics of the audio signal such as the occurrence and location of a transient, and a control signal is generated that causes the adaptive block-encoding process to encode segments of varying length. A complementary decoder applies an adaptive block-decoding process to recover the segments of audio information from the frames of encoded information. In embodiments that apply time-domain aliasing cancellation (TDAC) transforms, window functions and transforms are applied according to one of a plurality of segment patterns that define window functions and transform parameters for each segment in a sequence of segments. The segments in each frame of a sequence of overlapping frames may be recovered without aliasing artifacts independently from the recovery of segments in other frames. Window functions are adapted to provide preferred frequency-domain responses and time-domain gain profiles.

A control channel configuration method and apparatus is provided for supporting Inter-Cell Interference Coordination (ICIC) in an OFDM-based communication system. The control channel configuration method includes determining a Resource Block (RB) to be used for configuring control channels; configuring the control channels by mapping the control channels in a data channel region within the RB; and transmitting the configured control channels, wherein configuring the control channels includes mapping the control channels in units of Resource Element Groups (REGs) formed by binding one or more Resource Elements (REs) in a time domain-preferred allocation manner within the same RB.

The present invention is directed toward a pulse oximetry system for the determination of a physiological parameter capable of removing motion artifacts from physiological signals comprises a hardware subsystem and a software subsystem. The software subsystem is used in conjunction with the hardware subsystem to perform a method for removing a plurality of motion artifacts from the photo-plethysmographic data and for obtaining a measure of at least one physiological parameter from the data. The method comprises acquiring the raw photo-plethysmographic data, transforming the data into the frequency domain, analyzing the transformed data to locate a series of candidate cardiac spectral peaks (primary plus harmonics), reconstructing a photo-plethysmographic signal in the time domain with only the candidate cardiac spectral peaks (primary plus harmonics), computing the second order derivative of the reconstructed photo-plethysmographic signal, analyzing the candidate second order derivative photo-plethysmographic signal to determine the absence or presence of cardiac physiologic signal characteristics, and finally selecting the best physiologic candidate from the series of potential cardiac spectral peaks (primary plus harmonics) based upon a second derivative scoring system. This scoring system is preferentially based upon second derivative processing analysis, but can be equally applied using the first, third, fourth or other similar derivative processing analysis.

Representations of spatial audio scenes using higher-order Ambisonics HOA technology typically require a large number of coefficients per time instant. This data rate is too high for most practical applications that require real-time transmission of audio signals. According to the invention, the compression is carried out in spatial domain instead of HOA domain. The (N+1)² input HOA coefficients are transformed into (N+1)² equivalent signals in spatial domain, and the resulting (N+1)² time-domain signals are input to a bank of parallel perceptual codecs. At decoder side, the individual spatial-domain signals are decoded, and the spatial-domain coefficients are transformed back into HOA domain in order to recover the original HOA representation.

A noise suppression device receives data representative of a noise-corrupted signal which contains a speech signal and a noise signal, divides the received data into data frames, and then passes the data frames through a pre-filter to remove a dc-component and the minimum phase aspect of the noise-corrupted signal. The noise suppression device appends adjacent data frames to eliminate boundary discontinuities, and applies fast Fourier transform to the appended data frames. A voice activity detector of the noise suppression device determines if the noise-corrupted signal contains the speech signal based on components in the time domain and the frequency domain. A smoothed Wiener filter of the noise suppression device filters the data frames in the frequency domain using different sizes of a window based on the existence of the speech signal. Filter coefficients used for Wiener filter are smoothed before filtering. The noise suppression device modifies magnitude of the time domain data based on the voicing information outputted from the voice activity detector.

Techniques are provided to suppress noise and interference using an array microphone and a combination of time-domain and frequency-domain signal processing. In one design, a noise suppression system includes an array microphone, at least one voice activity detector (VAD), a reference generator, a beam-former, and a multi-channel noise suppressor. The array microphone includes multiple microphones—at least one omni-directional microphone and at least one uni-directional microphone. Each microphone provides a respective received signal. The VAD provides at least one voice detection signal used to control the operation of the reference generator, beam-former, and noise suppressor. The reference generator provides a reference signal based on a first set of received signals and having desired voice signal suppressed. The beam-former provides a beam-formed signal based on a second set of received signals and having noise and interference suppressed. The noise suppressor further suppresses noise and interference in the beam-formed signal.

Techniques for reducing peak-to-average power in multicarrier transmitters employ peak cancellation with subcarriers that are impaired by existing channel conditions. The use of Carrier Interferometry (CI) coding further improves the effectiveness of peak reduction. CI coding can also be impressed onto pulse sequences in the time domain, which enhances spectral selection and facilitates peak-power control.

A method is disclosed for compressing a frame of data representing parameter values, a time at which each parameter value was recorded, and an orientation of a sensor at the time each parameter value was recorded. Generally the method includes performing a two-dimensional transform on the data in the orientation domain and in a domain related to the recording time. In one embodiment, the method includes calculating a logarithm of each parameter value. In one embodiment, the 2-D transform includes generating a Fourier transform of the logarithm of the parameter values in the azimuthal domain, generating a discrete cosine transform of the transform coefficients in the time domain. This embodiment includes quantizing the coefficients of the Fourier transform and the discrete cosine transform. One embodiment of the method is adapted to transmit resistivity measurements made by an LWD instrument in pressure modulation telemetry so that while-drilling images of a wellbore can be generated. The one embodiment includes encoding the quantized coefficients, error encoding the encoded coefficients, and applying the error encoded coefficients to the pressure modulation telemetry.

Disclosed are systems and methods which provide interference mitigation by making alternative resources available within areas served by wireless communication links. Embodiments provide multiple channel availability in establishing wireless communication links to facilitate interference mitigation. Time domain techniques, spatial processing techniques, and/or frequency domain techniques may be implemented for spectrum management. Embodiments provide wireless base station configurations in which all or a plurality of base station sectors use a same frequency channel and/or in which each sector or a plurality of sectors use all frequency channels. Multi-channel strategies may be implemented such as to provide dynamic selection of a “best” frequency channel, to provide transmission of identical data on multiple channels for combining/selection at the receiver, and/or to provide for dividing the data for transmission on multiple channels.

The invention relates to the field of communication, provides an uplink DMRS (Demodulation Reference Signal) transmitting method, device and system and solves the problem that the time-frequency resource overhead of the uplink demodulation reference signal is excessive. The method comprises the steps that a base station configures a resource, a pattern or a parameter set required for transmitting the uplink DMRS for a user terminal; and the base station transmits or indicates the resource, the pattern or the parameter set required for transmitting the uplink DMRS to the user terminal via a bit domain or a high-level signaling of downlink control information. The resource, the pattern or the parameter set comprises a time domain position, a frequency domain position, a subframe configuration or a cycle, and a hopping mode. The technical scheme is applicable to a long term evolution system, and an uplink DMRS transmitting mechanism with the low time-frequency resource overhead is realized.

A virtual image is registered among a perceived real world background. Tracking light is scanned into the real world environment, which includes at least one detector pair. A first time and a second time at which the tracking light impinges on the first detector is detected, in which the first time and second time occurs within adjacent scan lines. A time at which a horizontal scan line edge (e.g., beginning of scan line or end of scan line) is encountered is derived as occurring one half way between the first time and the second time. The horizontal location of the first detector then is determined within a specific scan line inferring the scan line edge time. The vertical location of the detector is determined within a scan frame by measuring time duration using the beginning of the frame. By determining a location independently from the temporal resolution of the augmented imaging system, the temporal location of the detector is identified to a sub-pixel/sub-line precision. The augmented image is registered either to a 3D real world spatial coordinate system or to a time domain coordinate system based upon tracked position and orientation of the user.

PCT No. PCT/DK95/00089 Sec. 371 Date Dec. 27, 1996 Sec. 102(e) Date Dec. 27, 1996 PCT Filed Feb. 27, 1995 PCT Pub. No. WO95/23493 PCT Pub. Date Aug. 31, 1995A method and apparatus for simulating the transmission of sound from sound sources to the ear canals of a listener encompasses novel head-related transfer functions (HTFs), novel methods of measuring and processing HTFs, and novel methods of changing or maintaining the directions of the sound sources as perceived by the listener. The measurement methods enable the measurement and construction of HTFs for which the time domain descriptions are surprisingly short, and for which the differences between listeners are surprisingly small. The novel HTFs can be exploited in any application concerning the simulation of sound transmission, measurement, simulation, or reproduction. The invention is particularly advantageous in the field of binaural synthesis, specifically, the creation, by means of two sound sources, of the perception in the listener of listening to sound generated by a multichannel sound system. It is also particularly useful in the designing of electronic filters used, for example, in virtual reality systems, and in the designing of an "artificial head" having HTFs that approximate the HTFs of the invention as closely as possible in order to make the best possible representation of humans by the artificial head, thereby making artificial head recordings of optimal quality.

Various embodiments relate to the systems and methods for creating events, creating points of interest, associating points of interest with geographic locations, associating events with points of interest or with geographic locations, discovering events in a geographic area given a time frame, tagging and rating events and points of interest and sharing events among users. A particular embodiment includes a computer-implemented method including associating geographical locations or points of interest with events having geographical and temporal attributes in a geo-temporal data store, receiving a geo-temporal search query specified by a user, and producing search results including items from the geo-temporal data store having a pre-determined degree of relevance to the geo-temporal search query in both a geographical and temporal domain.

The invention discloses a motion vector forecasting method in a resolution quantizing structure, wherein, forecasting motion vectors of macro blocks in enhancement layers are acquired by utilization of relativity of motion vectors of a time domain, a space domain and an interlayer domain. The realization process is that: on the time domain, candidate motion vectors of the time domain are acquired from motion vectors of macro blocks with the same position with the prior frame; on the space domain, candidate motion vectors of the space domain are acquired from motion vectors of adjacent blocks; on the interlayer domain, candidate motion vectors of the interlayer domain are acquired from corresponding motion vectors with low space domain resolution hierarchy; candidate motion vectors with minimum motion estimation cost are selected to be forecasting motion vectors by selection among the candidate motion vectors of the time domain, the space domain and the interlayer domain. The invention has the advantages of capability of acquiring more accurate forecasting motion vectors, capability of compressing code rate of motion vectors, and obtaining of performance gain.

A reduction in a peak-to-mean envelope power ratio of a multicarrier signal, represented as a time domain signal, is achieved by applying (76) an offset, indicative of a difference (74) between a mean signal level and a midpoint level of the time domain signal, to the time domain signal. Alternatively, constellation values of positive and negative frequency components are modified by differing functions to produce a modified data set. Preferably, the negative frequency components are set to a predetermined value (namely zero (124)) to provide an alternate coding scheme Once the multicarrier signal has been converted into a time domain representation, real and imaginary parts (126-128) of the modified data set that consequently only contain positive frequency components and zeros are compared with one another to identify (130) which of the real and imaginary parts has a lower peak-to-average signal ratio (134) for the time domain representation. Then, based upon which of the peak-to-average signal ratios is lowest (136), either the real and imaginary part of the time domain signal is selected for subsequent transmission (138).

A depolarization waveform classifier based on the Modified lifting line wavelet Transform is described. Overcomes problems in existing rate-based event classifiers. A task for pacemaker/defibrillators is the accurate identification of rhythm categories so correct electrotherapy can be administered. Because some rhythms cause rapid dangerous drop in cardiac output, it's desirable to categorize depolarization waveforms on a beat-to-beat basis to accomplish rhythm classification as rapidly as possible. Although rate based methods of event categorization have served well in implanted devices, these methods suffer in sensitivity and specificity when atrial/ventricular rates are similar. Human experts differentiate rhythms by morphological features of strip chart electrocardiograms. The wavelet transform approximates human expert analysis function because it correlates distinct morphological features at multiple scales. The accuracy of implanted rhythm determination can then be improved by using human-appreciable time domain features enhanced by time scale decomposition of depolarization waveforms.

The invention relates to a method and system for wideband digital pre-distortion linearization, which is used to overcome the influence of memory effect in radio frequency power amplifier, to expand digital pre-distortion linearization bandwidth, and to improve digital pre-distortion linearization performance. The method and system can get an in-band pre-distortion signal and an out-of-band pre-distortion signal according to the characteristic parameter of the amplifier; the in-band pre-distortion signal is up-converted and the up-converted signal is added to the out-of-band pre-distortion signal, which is not up-converted, then the combined signal is inputted to the power amplifier as an input signal; a part of the output signal from the power amplifier, serving as a feedback signal, can be compared with the original input signal, and the characteristic parameter of the amplifier for generating the in-band pre-distortion signal and the out-of-band pre-distortion signal is adaptively regulated according to the comparison result, so that the waveform of time domain or the frequency domain of the feedback signal can be close to that of the original input signal as much as possible.

Detecting a linear impairment in a cable under test by using a random signal transmitted down the cable. The impairment causes a reflected signal to be combined with the random signal. The combined signal extends over a plurality of sub-bands. A method and apparatus perform the steps of: (a) receiving the combined signal from a test point upstream from the impairment; (b) tuning to each sub-band and receiving a part of the combined signal within each sub-band; (c) determining an autocorrelation function of each part of the combined signal of each sub-band, to produce a plurality of autocorrelation functions; (d) combining the autocorrelation functions to form a combined function; (e) detecting the reflected signal from the combined function; and (f) determining, from the combined function, a time delay associated with the reflected signal and the distance from the test point to the impairment.