236 results about "Gain function" patented technology

A storage pixel sensor disposed on a semiconductor substrate comprises a photosensor. At least one nonlinear capacitive element is coupled to the photosensor. At least one nonlinear capacitive element is arranged to have a compressive photocharge-to-voltage gain function. An amplifier has an input coupled to the nonlinear capacitor and an output. Other, non-capacitive elements may be employed to produce a compressive photo-charge-to-voltage gain having at least one breakpoint.

(A2) An extended depth of field (DOF) imaging system (10) is disclosed that has a corresponding extended depth of focus (DOF′) by virtue of its optical system (20) having a select amount of spherical aberration. The imaging system has an image processing unit (54) adapted to process the raw images and perform contrast enhancement to form processed images. The image processing includes restoring the defocused modulation transfer functions (MTFs) using a gain function (G) and the amount of defocus. The imaging system can include an illumination system (60) that illuminates the object being imaged to establish a distance (DH) between the optical system and the object, where distance DH is used in the restoring of the MTF. An iris-recognition (I-R) system based on the enhanced DOF imaging system is also disclosed.; Optical system embodiments for use in the DOF imaging system that can provide select amounts of spherical aberration—and thus select increases in DOF—without increasing the adverse impact of other aberrations on image formation are also disclosed.

A dynamic hearing protection method and device with which a hearing protection device specific differential head related transfer function corresponding to the difference between an open ear resonance measurement of the user's ear canal or an ear canal approximating the user's ear canal without the hearing protection device being worn in the ear canal and an open ear resonance measurement with the hearing protection device being worn in the ear canal as a function of the angle of sound incidence is determined and when using the hearing protection device, a frequency-dependent gain function to the captured audio signals selected according to the presently determined angle of sound incidence with each of the gain functions is determined by inverting the previously determined HPD specific differential HRTF for the respective sound incidence angle, to compensate for the effect of the presence of the hearing protection device on the open ear resonance.

A frequency-warped processing system using either sample-by-sample or block processing is provided. Such a system can be used, for example, in a hearing aid to increase the dynamic-range contrast in the speech spectrum, thus improving ease of listening and possibly speech intelligibility. The processing system is comprised of a cascade of all-pass filters that provide the frequency warping. The power spectrum is computed from the warped sequence and then compression gains are computed from the warped power spectrum for the auditory analysis bands. Spectral enhancement gains are also computed in the warped sequence allowing a net compression-plus-enhancement gain function to be produced. The speech segment is convolved with the enhancement filter in the warped time-domain to give the processed output signal. Processing artifacts are reduced since the frequency-warped system has no temporal aliasing.

A pulsed radar level gauge, comprising a transmitter for generating and transmitting a signal in the form of a pulse train, means for receiving pulses reflected in said tank, and to form an analogue tank signal based on these reflections, an analogue to digital converter for sampling and digitizing said tank signal and forming a digital time domain reflectometry (TDR) signal, and means for varying the output power of said transmitter during the transmission of said pulse train, in order to adjust an amplitude of said analogue tank signal to match the dynamics of said A / D-converter. A gain function applied on the receiver side is thus replaced by varying the output power on the transmitter side. This allows the signal-to-noise ratio to be improved, as the noise will not be amplified together with the signal.

An electronic device for suppressing noise in an audio signal is described. The electronic device includes a processor and instructions stored in memory. The electronic device receives an input audio signal and computes an overall noise estimate based on a stationary noise estimate, a non-stationary noise estimate and an excess noise estimate. The electronic device also computes an adaptive factor based on an input Signal-to-Noise Ratio (SNR) and one or more SNR limits. A set of gains is also computed using a spectral expansion gain function. The spectral expansion gain function is based on the overall noise estimate and the adaptive factor. The electronic device also applies the set of gains to the input audio signal to produce a noise-suppressed audio signal and provides the noise-suppressed audio signal.

The invention relates to a method of estimating noise power spectral density PSD in an input sound signal comprising a noise signal part and a target signal part. The invention further relates to a system to its use. The object of the present invention is to provide a scheme for estimating the noise PSD in an acoustic signal consisting of a target signal contaminated by acoustic noise. The problem is solved by a method comprising the steps of d) providing a digitized electrical input signal to a control path and performing; d1) storing a number of time frames of the input signal each comprising a predefined number N2 of digital time samples xn (n=1, 2, . . . , N2), corresponding to a frame length in time of L2=N2 / fs; d2) performing a time to frequency transformation of the stored time frames on a frame by frame basis to provide corresponding spectra Y of frequency samples; d3) deriving a periodogram comprising the energy content |Y|2 for each frequency sample in a spectrum, the energy content being the energy of the sum of the noise and target signal; d4) applying a gain function G to each frequency sample of a spectrum, thereby estimating the noise energy level |Ŵ|2 in each frequency sample, |Ŵ|2=G·|Y|2; d5) dividing the spectra into a number Nsb2 of sub-bands, each sub-band comprising a predetermined number nsb2 of frequency samples, and assuming that the noise PSD level is constant across a sub-band; d6) providing a first estimate |{circumflex over (N)}|2 of the noise PSD level in a sub-band based on the non-zero noise energy levels of the frequency samples in the sub-band; d7) providing a second, improved estimate |Ñ|2 of the noise PSD level in a sub-band by applying a bias compensation factor B to the first estimate, |Ñ|2=B·|{circumflex over (N)}|2. The invention may e.g. be used in listening devices, e.g. hearing aids, mobile telephones, headsets, active earplugs, etc.

The invention provides a real time voice denoising method and device; the method comprises the following steps: generating a frequency domain zone noise voice signal according to a voice input received by a voice receiver; calculating a logarithm spectrum posterior signal to noise ratio according to the frequency domain zone noise voice signal, wherein the logarithm spectrum posterior signal to noise ratio refers to a ratio between logarithm of a power spectrum of a present frame frequency domain zone noise voice signal and a logarithm of a previous frame noise power estimation value; obtaining a noise power spectrum estimation value according to the logarithm spectrum posterior signal to noise ratio and based on a weight noise estimation algorithm; generating a Wiener filtering gain function according to the noise power spectrum estimation value, and filtering the frequency domain zone noise voice signal according to the gain function, thus generating a frequency domain denoising voice signal; generating a time domain denoising voice signal according to the frequency domain denoising voice signal, and the time domain denoising voice signal is further processed by the voice receiver. Correspondingly, the invention also provides the real time voice denoising device.

A method of removing reverberation from audio signals is disclosed. The method comprises spectro-temporally analyzing the first audio signal and the second audio signal to derive an energy function of time for a plurality of frequency bands. The method further comprises determining a delay stability between the energy function of time for the first audio signal and the second audio signal in each band, determining a gain function in each band based on the delay stability, adjusting the energy of the first audio signal and the second audio signal using the gain function within each band, and resynthesizing audio signals from the energy in each band of the first audio signal and the second audio signal.

The invention relates to an echo cancellation method and system for a speech recognition front end. The echo cancellation method mainly comprises the steps of time-frequency domain signal transformation, echo signal and residual signal estimation, sub-band signal power spectrum and cross correlation coefficient calculation, nonlinear processing based on a gain function of cross correlation coefficients, frequency-time domain transformation and the like to finally output signals without echoes. The single-channel echo cancellation method is adopted for processing frequency-domain echoes under avehicle-mounted condition or in other application scenes, and has the advantages of small calculation amount and high convergence speed.

The invention discloses a temperature compensating circuit of a radio frequency power amplifier with stable gain function. The temperature compensating circuit comprises a control circuit, wherein thecontrol circuit generates control voltage V<tf> which can be changed along with the change of the chip temperature of the radio frequency power amplifier so as to regulate the feedback quantity of afeedback loop of the radio frequency power amplifier, the feedback loop is connected between the input end and the output end of the radio frequency power amplifier or the control circuit generates control voltage V<bias> which can be changed along with the change of the temperature of the radio frequency power amplifier to regulate the bias current of a bias circuit of the radio frequency power amplifier, and the bias circuit is connected with the input end of the radio frequency power amplifier. On the premise that the indexes of efficiency, and the like of the radio frequency power amplifier are not influenced, the temperature compensating circuit can reduce the grain change along with temperature change, change the feedback quantity of the feedback loop, improve the linearity index ofthe amplifier simultaneously and achieve simple structure and low cost.

A sound processing apparatus includes a target sound emphasizing unit configured to acquire a sound frequency component by emphasizing target sound in input sound in which the target sound and noise are included, a target sound suppressing unit configured to acquire a noise frequency component by suppressing the target sound in the input sound, a gain computing unit configured to compute a gain value to be multiplied by the sound frequency component using a gain function that provides a gain value and has a slope that are less than predetermined values when an energy ratio of the sound frequency component to the noise frequency component is less than or equal to a predetermined value, and a gain multiplier unit configured to multiply the sound frequency component by the gain value computed by the gain computing unit.

The invention discloses a self-adaptive de-noising and characteristic enhancing method of an SAR (Synthetic Aperture Radar) image, which mainly aims at overcoming the defects that the traditional method has poor de-noising property and can not self-adaptively select parameters in the experiment. The method comprises the following steps: firstly, carrying out logaritmim and then ME-curvelet transformation on an original SAR image; secondly, self-adaptively selecting and optimizing parameters in improved gain functions by adopting an improved PSO (Particle Swarm Optimization) algorithm according to the provided evaluation criterion; and finally, carrying out the nonlinear transformation on ME-curvelet coefficients, ME-curvelet inverse transformation and exponential transformation by adopting the improved gain functions to obtain the final SAR image subjected to the de-noising and characteristic enhancement. By using the method, the noise can be removed while the characteristics are enhanced, the complexity for processing can be reduced and the better de-noising and characteristic enhancing effects of the SAR image can be achieved.

A method for individually fitting a hearing instrument to a user, comprising at least one microphone for generating an input audio signal from ambient sound, an audio signal processing unit for processing the input audio signal into a processed output audio signal, and a transducer for stimulation of the human auditory system according to the processed output audio signal as input to said transducer is described, the method comprising: providing the user with the hearing instrument and starting operation of the hearing instrument; pre-defining a desired target loudness function, wherein loudness perception of a stimulus by the user when using the hearing instrument is defined as function of frequency and input sound pressure level at the microphone; measuring for a given measurement parameter set of perceived loudness levels and frequencies or frequency bands the respective transducer input audio signal level to be applied to the transducer input in order to achieve the respective perceived loudness level at the respective frequency or frequency band, said measurement parameter set comprising at least a low loudness level, an intermediate loudness level and a high loudness level, and said intermediate loudness level being measured for a larger number of frequencies or frequency bands and with a finer frequency resolution than said low and high loudness levels; calculating an individual gain function to be implemented in the audio signal processing unit in order to achieve the pre-defined target loudness function by taking into account the measured transducer input audio signal levels; and operating the hearing instrument with the individual gain function.

The invention discloses a two-channel beam forming speech enhancement method based on noise mixed coherence. Adaptive beam forming can effectively inhibit directional noise signals under the reverberation-free condition, but the inhibiting effect of the adaptive beam forming is greatly reduced in the presence of reverberation. Aiming at the problem, the invention provides the two-channel beam forming method based on noise mixed coherence. Considering that coherence and scattering noises exist simultaneously in a sound field, in the method provided by the invention, the hypothesis of replacing the traditional scattering sound field with a mixed noise field is provided, specifically, the method comprises the following steps: firstly, estimating the noise coherence in the mixed noise field, estimating a power spectrum of the noises by utilizing the noise coherence, and calculating a gain function of a frequency domain filtering by using the estimated result for the power spectrum of the noises; and carrying out frequency domain filtering processing on the noises and reverberant signals, and carrying out further processing on the residual noises by adopting a minimum variance distortionless response beam forming device. The experiment proves that the quality of the speech enhanced by adopting the method provided by the invention is obviously improved compared with that enhanced by adopting the traditional method.

A programmable gain amplifier having three separately programmable amplifiers. A programmable transconductance amplifier is followed by a programmable transimpedance amplifier, then a programmable switched capacitor amplifier. In one embodiment, this programmable gain amplifier is implemented in an analog front-end (AFE) circuit. One AFE embodiment provides a coarse pre-gain offset a black reference level sampler, and a fine post-gain offset in the programmable switched capacitor amplifier. In one embodiment, an ADC reference is sampled, and is subtracted directly from the video signal in the switched capacitor amplifier so that the zero level of the video signal is made to correspond to the zero level of the ADC. In another embodiment, a piece-wise linear approximation of an exponential gain function is implemented by programming the transconductance and transimpedance amplifiers to provide exponential jumps in gain, while the switched capacitor amplifier is programmed to interpolate as needed between the exponential gain levels.

A processor for enhancing an input luminance signal including: a means for calculating a chroma edge value associated with the input luminance signal; a means for measuring a luminance gradient associated with the input luminance signal; a peaking filter for processing the input luminance signal; a gain adjustment circuit having a gain function that is adjustable relative to the calculated chroma edge value, the gain adjustment circuit being operable to adjust the magnitude of the output of the peaking filter wherein: in a first range of luminance gradients, the output is an attenuated version of the input; in a second range of luminance gradients the output is directly proportional to the input; in a third range of luminance gradients, the output is inversely proportional to the input; wherein said enhanced luminance signal is the sum of the input and output of the peaking filter.

The present invention discloses a hypersonic aerocraft neural network composite learning non-backstepping control method. The technical problem is solved that a current hypersonic aerocraft control method is bad in practicality. The technical scheme comprises: performing transformation of an attitude subsystem strict feedback form, obtaining an output feedback form, employing a high-gain observer to perform estimation of newly defined variables, and providing basis for subsequent design of a controller; allowing the controller to consider the lump nondeterminacy of the system, and only requiring one neural network to perform approximation, wherein the controller is simple in design and is convenient for engineering realization; aiming at control of unknown cases of a gain function, designing the controller based on the parameter linearization expression mode; and introducing system modeling errors, and constructing a neural network composite updating rule and a parameter adaptive composite updating rule to realize fast tracking of a hypersonic aerocraft. The effective estimation of unknown states is realized based on the high-gain observer, the repeat design of the virtual controlled quantity is not needed so as to simple the design of the controller. the realization is easy, and the practicality is good.

The invention relates to a rotary compressor with enthalpy-gaining function. An upper bearing or a lower bearing of the rotary compressor is provided with one or more gas orifices; the gas orifices are matched with a pipeline so as to allow an air cylinder of the compressor to communicates with the external part of the compressor through the gas orifices; the gas orifices are provided with a one-way valve for allowing external fluid to enter the air cylinder through the pipeline and the gas orifices and preventing the fluid in the air cylinder from discharging from the gas orifices and the pipeline. The rotary compressor of the invention improves the structure of conventional rotary compressor by enabling a gaseous refrigerant to be injected into the compressor air cylinder through the pipeline, thus increasing the suction temperature of the rotary compressor, improving the compression efficiency and saving energy.

The invention relates to a method for reducing a noise by using the hearing threshold of impaired hearing. The method comprises the following steps of: dividing an input speech signal into N subband signals, compensating the hearing, respectively reducing the noise of the subband signals, and comprehensively processing various paths of input subband signals into a path of output signals by a weighted stacking filter bank. The invention provides a generalized spectral subtraction method according to noise levels and hearing threshold adjustment parameters, wherein a posteriori signal-to-noise ratio is replaced by an a priori signal-to-noise ratio to adjust a gain function, so that musical noise generated by noise reduction can be effectively reduced. The noise reduction gain function is related to the signal-to-noise ratio, and is adjusted according to the noise levels and the hearing threshold of a patient; when the hearing threshold is high, the suppression to the noise is low, and speech distortion is reduced simultaneously; and when the hearing threshold is low, the suppression to the noise is increased so as to improve listening comfort, so that the method has pertinence to different hearing loss conditions and different noise conditions, and makes the suppression effect and the speech distortion relatively balanced.

A method of improving the production of a mature gas or oil field, the field comprising a plurality of existing wells, the method comprising the steps of providing a field simulator capable of predicting a production of the field in function of a given scenario, a scenario being a set of data comprising production parameters of the existing wells and, the case may be, location and production parameters of one or more new wells, determining drainage areas of the existing wells using the field simulator, determining locations of candidate new wells such that drainage areas of the candidate new wells, determined using the field simulator, do not overlap with the drainage areas of the existing wells, optimizing the value of a gain function which depends on the field production by determining a set of wells out of a plurality of sets of wells, which optimize the value of said gain function, each set of wells of said plurality of sets of wells comprising the existing wells and new wells selected among the candidate new wells.

The invention discloses an audio-reverberation inhibiting device and an inhibiting method thereof. The device includes a reverberation-time blind estimation module, a later-period reverberation power spectrum estimation module, a spectral-subtraction module and a complex-cepstrum-domain filtering module. A reverberation voice estimates a reverberation time through the reverberation-time blind estimation module. The later-period reverberation power spectrum estimation module establishes a reverberation statistical model through the estimated reverberation time and carries out analysis processing on the reverberation voice so that a later-period reverberation power spectrum is obtained. The spectral-subtraction module includes a gain function structure and a spectral-subtraction implementation module and a spectral-subtraction gain function is constructed firstly through use of a reverberation-voice power spectrum and a later-period reverberation power spectrum. Then the spectral-subtraction gain function and the reverberation voice are input into the spectral-subtraction implementation module so that an earlier-period voice is obtained. Finally, the earlier-period voice is input into the complex-cepstrum-domain filtering module so that a reverberation-removed voice is obtained. The audio-reverberation inhibiting device and the inhibiting method thereof is low in calculation complexity, convenient to handle in a real-time manner and capable of inhibiting audio reverberation obviously and improving voice quality efficiently.

The invention discloses a high-performance large-scale MIMO downlink transmission channel estimation method, which mainly comprises the following steps: a base station generates a large-scale beam setto cover the whole cell through beam shaping, and all antennas of the base station synchronously send pilot sequences to a target user; the base station generates a large-scale beam set to cover thewhole cell. After acquiring the pilot information, the user constructs the compressed sensing channel estimation problem according to the designed pilot matrix and dictionary. According to the block sparsity of the channel gain function after over-complete dictionary sampling, the compressed sensing recovery algorithm based on block sparsity is used to estimate the channel vector, and the estimation result is the required channel vector. The invention improves the sparsity of the channel vector under the overcomplete dictionary by using the multi-band modulation DPSS matrix, improves the performance of the channel recovery, reduces the operation complexity of the algorithm, and reduces the pilot overhead through the use of the non-orthogonal pilot matrix.

The invention relates to an on-site programmable device FPGA logic unit model and a general bin packing algorithm thereof. The module is analyzed and modeled according to realizable subscriber circuit logic function type in logic unit, and can be divided into three stages according to mutual driving case, the first stage is a searching table, the second stage is a special device, and the third stage is a sequential device, and the model can widely describe present commercial FPGA chip logic unit structure. Based on the module, the invention provides a general logic unit bin packing algorithm from bottom to top, which respectively establish user-defined logic devices for a device using the special device as core, a device using the sequential device as core and a device using the searching table as core in subscriber circuit, and carries out bin packing for the self- established devices by employing gain function calculation. The algorithm is a universal algorithm for treating bin packing problem of various logic units, has wide representation, versatility, practicality and excellent time spending performance, and is suitable for large-scale subscriber circuit.

A loudspeaker drive circuit uses a dynamic range compressor to implement a non-linear gain function between the input signal to the dynamic range compressor and an output signal from the dynamic range compressor. The output is used to drive a loudspeaker, and the operating parameters of the dynamic range compressor are varied in dependence on a criterion, such as the estimated voice coil temperature, the power consumption or the acoustical distortion.

The present invention discloses a hypersonic aerocraft neural network composite learning control method based on robustness design. The objective of the invention is to solve the technical problem that a current hypersonic aerocraft control method is bad in practicality. The technical scheme comprises: performing transformation of an attitude subsystem strict feedback form, obtaining an output feedback form, employing a high-gain observer to perform estimation of newly defined variables, and providing basis for subsequent design of a controller; allowing the controller to consider the lump nondeterminacy of the system, and only requiring one neural network to perform approximation, wherein the controller is simple in design and is convenient for engineering realization; and considering control of unknown cases of a gain function, introducing the upper and lower bound information, and designing a robustness item to ensure stability of the system. Because the strict feedback form is transformed to an output feedback form to effectively avoid approximation of virtual control amount required for future through adoption of the neural network; aiming at the system nondeterminacy, the robustness item is designed to ensure the stability of the system; and modeling errors are constructed to design a neural network composite learning updating rule so as to improve the neural network learning speed.

The invention relates to a howling suppression method based on feedback signal spectrum estimation. The method comprises the following steps: firstly, carrying out framing, windowing and Fourier transform on a signal collected by a microphone and a loudspeaker output signal respectively so as to obtain a corresponding frequency domain signal; then, according to the microphone frequency domain signal and the loudspeaker output frequency domain signal, calculating a cross-power spectrum between the microphone and the loudspeaker output signal and a power spectrum of the loudspeaker output signal; dividing the cross-power spectrum by the power spectrum so as to obtain estimation of a feedback path transfer function amplitude spectrum; using the obtained estimation of the feedback path transfer function spectrum amplitude to calculate a current frame feedback signal amplitude spectrum; and then using the microphone signal amplitude spectrum and the obtained feedback signal amplitude spectrum to calculate a gain function; finally, according to the obtained gain function, carrying out correction on the microphone frequency domain signal so as to realize howling suppression. By using the method of the invention, convergence and tracking speeds are fast, robustness is good and algorithm complexity is low, which is good for real-time realization.

The invention discloses a vertical switching method based on Bayesian decision in a heterogeneous wireless network. The present invention proposes the vertical switching method based on the Bayesian decision for the problem that a mobile terminal is frequently switched between different networks in a heterogeneous wireless network environment. A prior probability of terminal switching is calculated by using received signal strength (Received Signal Strength, RSS) of the terminal; a posterior probability of the terminal switching is calculated in combination with the Bayesian formula; a networkparameter is determined by using an analytic hierarchy process and a entropy method, comprising: weights of an available bandwidth, delay, a bit error rate and delay jitter, and income functions of different actions are constructed in different states of the mobile terminal based on a simple weighting method; and the posterior probability and the income function of the switching are comprehensively considered to construct a posterior gain function of the switching, and switching decision is performed based on the posterior gain function. By adoption of the vertical switching method, the number of switching is effectively reduced, the "ping-pong effect" in the switching process is effectively inhibited, and the resource utilization rate of the system is improved.

An inverted pendulum self-adaptive iterative learning inversion control method is characterized in that: aiming at an inverted pendulum system containing unknown input saturation, a self-adaptive iterative learning inversion controller is designed through utilization of a neural network and an inversion control method in combination with self-adaptive iterative learning control; the construction of an integral lyapunov function solves the control problem caused by derivation of a unknown gain function; based on the median theorem, a hyperbolic tangent function is adopted to approximate an input saturation term; then, a radial basis function neural network is adopted to approximate and compensate uncertain unknown items of a system, and two combined self-adaptive laws are adopted to updatethe weight of the neural network and the bound of estimation errors. Under the condition that the system has input saturation, the invention provides the control method which can compensate the unknown uncertainty of the system, solve the control problem caused by derivation of the unknown gain function and realize two-norm convergence of a tracking error of the system to be near zero within limited iteration times.

A multiple channel steered spatialized signal is generated from a signal input modified according to respective spatialization gain functions to generate a plurality of audio channels. An echo cancellation signal is applied to a return path using a combined spatialization and echo path estimate. The estimate is derived from the gain functions applied to the respective channels. When the gain functions applied in the respective channels are changed, for instance to represent a different apparent position of the sound source, a new estimate of the echo paths is generated, based on a previous estimate of the echo path and on the new gain functions.