714 results about "Gain factor" patented technology

The present invention comprises one or more microprocessors programmed to execute methods for improving the performance of an analyte monitoring device including prediction of glucose levels in a subject by utilizing a predicted slower-time constant (1 / k2). In another aspect of the invention, pre-exponential terms (1 / c2) can be used to provide a correction for signal decay (e.g., a Gain Factor). In other aspects, the present invention relates to one or more microprocessors comprising programming to control execution of (i) methods for conditional screening of data points to reduce skipped measurements, (ii) methods for qualifying interpolated / extrapolated analyte measurement values, (iii) various integration methods to obtain maximum integrals of analyte-related signals, as well as analyte monitoring devices comprising such microprocessors. Further, the present invention relates to algorithms for improved optimization of parameters for use in prediction models that require optimization of adjustable parameters.

The envelope of a decorrelated signal derived from an original signal can be shaped without introducing additional distortion, when a spectral flattener is used to spectrally flatten the spectrum of the decorrelated signal and the original signal prior to using the flattened spectra for deriving a gain factor describing the energy distribution between the flattened spectra, and when the so derived gain factor is used by an envelope shaper to timely shape the envelope of the decorrelated signal.

A method and an apparatus for data transmission in a mobile telecommunication system supporting an enhanced uplink service are provided. A Transport Format Combination (TFC) selector determines TF information for data to be transmitted through a first data channel not supporting Hybrid Automatic Repeat reQuest (HARQ) and a second data channel supporting HARQ, and determines gain factors for the first and second data channel, and first and second control channel carrying control information for the first and second data channel. The gain factors are input to a physical channel transmission controller, and the physical channel transmission controller reduces the gain factor for the second channel if total transmit power required for transmission of the channels exceeds the predetermined maximum allowed power. A gain scaler adjusts transmit powers of the channels using the scaled gain factor and gain factors for the first data channel, the first control channel and the second control channel.

Techniques like Huffman coding can be used to represent digital audio signal components more efficiently using non-uniform length symbols than can be represented by other coding techniques using uniform length symbols Unfortunately, the coding efficiency that can be achieved by Huffman coding depends on the probability density function of the information to be coded and the Huffman coding process itself requires considerable processing and memory resources. A coding process that uses gain-adaptive quantization according to the present invention can realize the advantage of using non-uniform length symbols while overcoming the shortcomings of Huffman coding. In gain-adaptive quantization, the magnitudes of signal components to be encoded are compared to one or more thresholds and placed into classes according to the results of the comparison. The magnitudes of the components placed into one of the classes are modified according to a gain factor that is related to the threshold used to classify the components. Preferably, the gain factor may be expressed as a function of only the threshold value. Gain-adaptive quantization may be used to encode frequency subband signals in split-band audio coding systems. Additional features including cascaded gain-adaptive quantization, intra-frame coding, split-interval and non-overloading quantizers are disclosed.

A method and apparatus is provided for determining an effective thermal mass of a patient. The effective thermal mass is employed to determine a gain factor used in a feedback control system controlling patient temperature. The method begins by inducing hypothermia or hyperthermia in at least a selected portion of the patient with a device having a heat transfer surface. Next, power is transferred between the device and the patient. A change in temperature over time, which arises in the selected portion of the patient, is measured while performing the step of inducing hypothermia or hyperthermia. Finally, an effective thermal mass is calculated based on the measured power and the measured temperature change over time.

Method and apparatus for non-invasively determining the time onset (Tonset) and end (Tend) of patient inspiratory efforts. A composite pressure signal is generated comprising the sum of an airway pressure signal, a gas flow pressure signal obtained by applying a gain factor (Kf) to a signal representing gas flow rate and a gas volume pressure signal obtained by applying a gain factor (Kv) to a signal representing volume of gas flow. Kf and Kv values are adjusted to result in a desired linear trajectory of composite pressure signal baseline in the latter part of the exhalation phase. The current composite pressure signal is compared with (i) selected earlier composite pressure signal values and / or (ii) value expected at current time based on extrapolation of composite pressure signal trajectory at specified earlier times and / or (iii) the current rate of change in the composite pressure signal with a selected earlier rates of change. The differences obtained by the comparison are compared with selected threshold values. Tonset is identified when at least one of the differences exceeds the threshold values.

A method and apparatus provides high-accuracy measurements of an electrical parameter across a broad range of parameter input values. In one embodiment, an intelligent electronic device (IED), e.g., a digital electrical power and energy meter, with a plurality of independently-adjustable gain factors measures a parameter, and calculates and stores calibration factors associated with known values of the measured parameter. The IED or meter applies the stored calibration factors when measuring unknown values of the measured parameter, to improve the accuracy of the measurement.

A technique is provided for use by a Multi-Radio Management Resource (MRRM) component of a multi-radio wireless communication system for controlling the handover of a mobile terminal between different radio access technologies (RAT). In one example, all suitable RATs having coverage areas currently covering the location of a mobile terminal are identified. The performance gain that might be achieved via a handover to one of the other RATs is then determined by the MRRM based on various performance gain factors. Handover costs that will be incurred as a result of the handover are also explicitly calculated. Then, a cost-adjusted gain is determined by the MRRM based on the performance gain and the handover costs. A handover is only triggered by the MRRM if the cost-adjusted gain exceeds a minimum threshold. The speed and trajectory of the mobile terminal may also be considered.

A multimode optical fibre communications system, and in particular to a system in which non-linearities in the propagation of the signal through a multimode optical communications channel degrade the signal presented to the receiver. The system includes an optical transmitter unit for connection to a multimode optical fibre transmission link. The transmitter unit has a data input for receiving an input data signal, a data signal processing circuit and a source of optical radiation. The data signal processing circuit is arranged to receive the input data signal from the data input and to provide a processed data signal to the source of optical radiation and the source of optical radiation is arranged to generate from this an optical signal for transmission by a multimode optical fibre. The data processing circuit is arranged to provide from the input data signal a non-inverted data signal and an inverted data signal, receive a control signal for controlling the generation of the processed data signal, apply a controllable delay in accordance with the control signal to at least one of the non-inverted and inverted data signals, and combine the non-inverted and inverted signals after the application of the controllable delay(s) and gain factor(s) to generate the processed data signal.

A multi-channel decoder for generating a binaural signal from a downmix signal using upmix rule information on an energy-error introducing upmix rule for calculating a gain factor based on the upmix rule information and characteristics of head related transfer function based filters corresponding to upmix channels. The one or more gain factors are used by a filter processor for filtering the downmix signal so that an energy corrected binaural signal having a left binaural channel and a right binaural channel is obtained.

A computer implemented method for fusing images taken by a plurality of cameras is disclosed, comprising the steps of: receiving a plurality of images of the same scene taken by the plurality of cameras; generating Laplacian pyramid images for each source image of the plurality of images; applying contrast normalization to the Laplacian pyramids images; performing pixel-level fusion on the Laplacian pyramid images based on a local salience measure that reduces aliasing artifacts to produce one salience-selected Laplacian pyramid image for each pyramid level; and combining the salience-selected Laplacian pyramid images into a fused image. Applying contrast normalization further comprises, for each Laplacian image at a given level: obtaining an energy image from the Laplacian image; determining a gain factor that is based on at least the energy image and a target contrast; and multiplying the Laplacian image by a gain factor to produce a normalized Laplacian image.

This invention provides a method and a system for coding / decoding frequency domain parameter stereos and multiple tracks, which converts signals of right and left tracks to a residual signal of a sum-difference tack and a residual signal of an estimated difference track, carries out time frequency conversion to residual signals of the estimated difference track, the original difference track and the original sum track to pick up the HF component and low frequency component of the residual signal of the original difference track, which is coded in quantization to utilize the HF component to get estimated residual RF component of the right and left tracks and gets the original residual RF component of the right and left tracks to be controlled in gain to get a gain coefficient of the difference of the two tracks and coded in quantization and finally multiplexes the coded data and edge information to get a frequency domain parameter stereo bit stream.

Adaptive noise control for reducing power of an acoustic noise signal radiated from a noise source to a listening position comprises providing an electrical reference signal correlated with the acoustic noise signal; filtering the electrical reference signal with an adaptive filter to provide an electrical output signal; multiplying the electrical output signal of the adaptive filter by a gain factor to provide a first electrical compensation signal; filtering and multiplying the electrical output signal of the adaptive filter by the inverse of the gain factor to provide a second electrical compensation signal, the second gain factor being equal to 1 subtracted by the first gain factor; radiating the first electrical compensation signal to the listening position with an acoustic transducer; sensing a residual electrical error signal at the listening position; adding the second electrical compensation signal to the electrical error signal to provide a compensated error signal; and adapting filter coefficients of the adaptive filter as a function of the compensated error signal and the reference signal.

Methods, devices and systems for generating ultrashort optical linear chirped pulses with very high power by amplifying the pulses so that their temporal duration is longer than the storage time of the amplifying medium. The additional gain factor is related to the ratio of the storage time to the stretched pulse. A preferred embodiment connects a mode locked laser source that generates optical pulses whose duration is stretched with a chirped fiber Bragg grating. Embodiments include methods, devices and systems causing an extreme chirped pulse amplifier (XCPA) effect in an oscillator.

A method and an apparatus for data transmission in a mobile telecommunication system supporting an enhanced uplink service are provided. A Transport Format Combination (TFC) selector determines TF information for data to be transmitted through a first data channel not supporting Hybrid Automatic Repeat reQuest (HARQ) and a second data channel supporting HARQ, and determines gain factors for the first and second data channel, and first and second control channel carrying control information for the first and second data channel. The gain factors are input to a physical channel transmission controller, and the physical channel transmission controller reduces the gain factor for the second channel if total transmit power required for transmission of the channels exceeds the predetermined maximum allowed power. A gain scaler adjusts transmit powers of the channels using the scaled gain factor and gain factors for the first data channel, the first control channel and the second control channel.

An amplifier having an active and passive gain stage connect to a load for driving a load according to a system analog input. A first embodiment of the amplifier in accordance with the present invention includes a logic network connected between a comparator network and a switching system, wherein the comparator network connects to the passive gain stage. Specifically, the active gain stage may include an active filter connected to receive an analog or digital input and provide a difference between the analog or digital input and the feedback signal relative to the gain factor of a gain unit connected to the active filter. The passive gain stage includes a passive filter. The logic network generates at least one switching signal which controls the switching system that includes at least one switching device to selectively provide power to the load. An output signal from the switching system provides output for the amplifier and is fed back to the active gain stage. In another embodiment, the output is a two-level signal and the passive and active filters are second order low pass filters, where the gain factor is about 25 or more. In yet another embodiment, the gain factor is approximately 250. Moreover, the amplifier may include a digital delta-sigma modulator connected to supply a two level input.

A method of encoding a digital signal, particularly an audio signal, which predicts favorable scalefactors for different frequency subbands of the signal. Distortion thresholds which are associated with each of the frequency subbands of the signal are used, along with transform coefficients, to calculate total scaling values, one for each of the frequency subbands, such that the product of a transform coefficient for a given subband with its respective total scaling value is less than a corresponding one of the distortion thresholds. In an audio encoding application, the distortion thresholds are based on psychoacoustic masking. The invention may use a novel approximation for calculating the total scaling values, which obtains a first term based on a corresponding distortion threshold, and obtains a second term based on a sum of the transform coefficients. Both of these terms may be obtained using lookup tables. The total scaling values can be normalized to yield scalefactors by identifying one of the total scaling values as a minimum nonzero value, and using that minimum nonzero value to carry out normalization. Encoding of the signal further includes the steps of setting a global gain factor to this minimum nonzero value, and quantizing the transform coefficients using the global gain factor and the scalefactors.

A programmable state machine is incorporated into the core of a network processor (NP) to monitor the utilization of different processing elements in the NP and to control the power state of each element as a function of past and predicted utilization. The state machine can be used to control a centralized power management control unit or to control a distributed power management unit where each processing element includes its own state machine. The function of the power management state machine can be implemented in any combination of software and / or hardwired logic, depending on the system design requirements. The monitoring and control are implemented through the use of a power management state change algorithm. The determination of the power state of a processing element accommodates interdependencies between the elements. It also makes adjustments in gain factors in response to actual performance and utilization of the network processor.

A speech coding system is provided with time-domain noise attenuation. The speech coding system has an encoder operatively connected to a decoder via a communication medium. A preprocessor processes a digitized speech signal from an analog-to-digital converter. Speech coding systems are used to encode and decode a bitstream. Gains from the speech coding are adjusted by a gain factor Gf that provides time-domain background noise attenuation.

A method and an apparatus for data transmission in a mobile telecommunication system supporting an enhanced uplink service are provided. A Transport Format Combination (TFC) selector determines TF information for data to be transmitted through a first data channel not supporting Hybrid Automatic Repeat reQuest (HARQ) and a second data channel supporting HARQ, and determines gain factors for the first and second data channel, and first and second control channel carrying control information for the first and second data channel. The gain factors are input to a physical channel transmission controller, and the physical channel transmission controller scales-down the gain factor for the second channel if total transmit power required for transmission of the channels exceeds the predetermined maximum allowed power. A gain scaler adjusts transmit powers of the channels using the scaled gain factor and gain factors for the first data channel, the first control channel and the second control channel.

A wireless communicator, including a modem operable to transmit and receive voice communication phone calls, a power amplifier coupled with the modem operable to dynamically apply a variable gain factor to voice communications transmitted by the modem, to produce an appropriate power output, and a controller coupled with the modem including a radiation monitor operable to monitor the power output of the power amplifier, the controller being operable to restrict operation of the modem, based on data provided by the radiation monitor, when the cumulative power output produced by the power amplifier exceeds a pre-designated daily cumulative power limit.

A system and method for controlling an internal combustion engine for low emissions include an inner feedback control loop to control the engine fuel / air ratio with feedback provided by a first exhaust gas sensor and an outer feedback control loop that modifies the fuel / air ratio reference provided to the inner feedback control loop based on feedback signals provided by the first exhaust gas sensor and a second exhaust gas sensor. The fuel / air ratio reference signal controller adapts to the oxygen storage capacity of the catalyst by modeling the catalyst as an integrator with an unknown gain and estimating the catalyst gain based on the first and second exhaust gas sensor signals. Using the estimated catalyst gain, an adaptive controller gain factor is determined and subsequently used to determine the fuel / air ratio reference signal provided to the fuel / air ratio controller of the inner feedback control loop.

A system and method for calibrating a vehicle heading sensor, such as a yaw-rate sensor, when GPS signals are not available using a bias update model that employs a bias gain factor. In order for the bias update model to be accurate, the vehicle should be traveling relatively straight. One embodiment of the present invention uses three thresholds to determine if the vehicle is traveling straight. These thresholds include a yaw-rate threshold, a steering wheel angle threshold and a wheel speed threshold. If all three of the thresholds indicate that the vehicle is traveling straight, then the update bias model can be used to calibrate the yaw-rate sensor.

A fourth order delta sigma analog-to-digital converter is presented, comprising a passive delta sigma modulator including a passive filter, a quantizer, and a digital-to-analog converter in a first feedback loop, and an active filter having a large gain factor in a second feedback loop around the passive delta-sigma modulator.

The invention discloses a device-to-device (D2D) cluster data sharing method under a cellular environment. The method comprises the following steps that: step1. before D2D cluster retransmission begins, each mobile equipment in the D2D cluster firstly acquires a packet error matrix (PEM) which belongs to the mobile equipment itself; step 2. a retransmission gain factor of the each mobile equipment in the D2D cluster is calculated; step 3. the mobile equipment with the highest retransmission gain factor is taken as a transmitter in the D2D cluster retransmission; step 4. the transmitter retransmits a data packet needed to be retransmitted in the cluster and updates the PEM of the each mobile equipment; step 5. if the updated PEM still contains a ''0'' element, the step 1 to the step 4 are repeated till the PEMs of all the mobile equipment in the D2D cluster are ''1'' matrixes. In the method of the invention, through effectively reducing retransmission times in the D2D cluster, related signaling costs of D2D communication under the cellular environment can be substantially saved and simultaneously system time delay can be shortened. The method is beneficial to realization of a real time multimedia service.

The invention discloses a speech enhancement method using a speech existence probability, after the processing of the invention, the speech quality is higher, and the noise reduction amount is larger.The method is realized by the following technical scheme that: on the basis of a MIS measure speech enhancement method, the speech existence probability is used, an inputted noisy speech is sampled,framed and windowed, and then a noisy speech spectrum is obtained through fast Fourier transform (FFT); the noise estimation of the obtained speech spectrum is carried out, based on a non-stationary noise minimum search algorithm of statistical information, the smoothing between front and back frame noise estimation values is carried out by using inter-frame correlation, and a noise power spectrumis estimated; at the same time, speech prior signal-to-noise ratio estimated values obtained in several front and back frames are smoothened; then the speech existence probability and a MIS measure gain factor are combined, the obtained noisy speech spectrum is multiplied by a new gain factor, and a spectrum of the enhanced speech is obtained; and the inverse fast Fourier transform (IFFT) is performed to obtain an enhanced time domain speech signal.