Showing papers on "Sampling (signal processing) published in 2001"

VD-AUTO-SMASH imaging.

Abstract: Recently a self-calibrating SMASH technique, AUTO-SMASH, was described. This technique is based on PPA with RF coil arrays using auto-calibration signals. In AUTO-SMASH, important coil sensitivity information required for successful SMASH reconstruction is obtained during the actual scan using the correlation between undersampled SMASH signal data and additionally sampled calibration signals with appropriate offsets in k-space. However, AUTO-SMASH is susceptible to noise in the acquired data and to imperfect spatial harmonic generation in the underlying coil array. In this work, a new modified type of internal sensitivity calibration, VD-AUTO-SMASH, is proposed. This method uses a VD k-space sampling approach and shows the ability to improve the image quality without significantly increasing the total scan time. This new k-space adapted calibration approach is based on a k-space-dependent density function. In this scheme, fully sampled low-spatial frequency data are acquired up to a given cutoff-spatial frequency. Above this frequency, only sparse SMASH-type sampling is performed. On top of the VD approach, advanced fitting routines, which allow an improved extraction of coil-weighting factors in the presence of noise, are proposed. It is shown in simulations and in vivo cardiac images that the VD approach significantly increases the potential and flexibility of rapid imaging with AUTO-SMASH.

Optimal sub-Nyquist nonuniform sampling and reconstruction for multiband signals

Abstract: We study the problem of optimal sub-Nyquist sampling for perfect reconstruction of multiband signals. The signals are assumed to have a known spectral support /spl Fscr/ that does not tile under translation. Such signals admit perfect reconstruction from periodic nonuniform sampling at rates approaching Landau's (1967) lower bound equal to the measure of /spl Fscr/. For signals with sparse /spl Fscr/, this rate can be much smaller than the Nyquist rate. Unfortunately the reduced sampling rates afforded by this scheme can be accompanied by increased error sensitivity. In a previous study, we derived bounds on the error due to mismodeling and sample additive noise. Adopting these bounds as performance measures, we consider the problems of optimizing the reconstruction sections of the system, choosing the optimal base sampling rate, and designing the nonuniform sampling pattern. We find that optimizing these parameters can improve system performance significantly. Furthermore, uniform sampling is optimal for signals with /spl Fscr/ that tiles under translation. For signals with nontiling /spl Fscr/, which are not amenable to efficient uniform sampling, the results reveal increased error sensitivities with sub-Nyquist sampling. However, these can be controlled by optimal design, demonstrating the potential for practical multifold reductions in sampling rate.

A low-power low-noise ultrawide-dynamic-range CMOS imager with pixel-parallel A/D conversion

Abstract: A CMOS image sensor with pixel-parallel analog-to-digital (A/D) conversion fabricated with different array sizes and photodiode types in a three-metal 0.5-/spl mu/m process is presented. Nominal power dissipation is 40 nW per pixel at V/sub DD/=3.3 V. A/D conversion results from sampling a free-running photocurrent-controlled oscillator to give a first-order /spl Sigma/-/spl Delta/ sequence. The sensor displays dynamic range capability of greater than 150000:1 and exhibits fixed pattern noise correctable to within 0.1% of signal.

Display device and driving method thereof

Abstract: In an active matrix display device utilizing electro-optical elements such as organic EL elements capable of obtaining a high image quality with a low power, each pixel circuit is provided with: a sampling circuit for sampling a signal voltage on a signal wiring line synchronously with a scan pulse; a reference voltage; and a comparator circuit. In the pixel circuit, the sampled signal voltage is compared with the reference voltage and the display time of each EL device is controlled by the period until the relation between the signal voltage and reference voltage is inverted, to thereby control the light emission time in one frame period.

AC coupling technique for Josephson waveform synthesis

Abstract: We demonstrate a new bias technique that uses low-pass and high-pass filters to separate the current paths of the sampling and signal frequencies in a Josephson waveform synthesizer. This technique enables the output voltage of the array to be directly grounded by removing the low-frequency common mode signal that previously prevented direct measurement of the array voltage with low-impedance instruments. We directly measure the harmonic spectra of 1 kHz and 50 kHz synthesized sine waves. We also use a thermal transfer standard to compare the rms voltages of synthesized sine waves at frequencies from 1 kHz to 50 kHz. Finally, we describe a new circuit that should enable a significant increase in output voltage by allowing several distributed arrays to be biased in parallel at high frequency, while combining their low frequency output voltages in series.

Method for whitening colored noise in a communication system

Abstract: A filter settings generation operation includes sampling colored noise present at the input of a receiver to produce a sampled signal. The sampled signal is spectrally characterized across a frequency band of interest to produce a spectral characterization of the sampled signal. This spectral characterization may not include a signal of interest. The spectral characterization is then modified to produce a modified spectral characterization. Filter settings are then generated based upon the modified spectral characterization. Finally, the input present at the receiver is filtered using the filter settings when the signal of interest is present to whiten colored noise that is present with the signal of interest. In modifying the spectral characterization, pluralities of spectral components of the spectral characterization are independently modified to produce the modified spectral characterization. Modifications to the spectral characterization may be performed in the frequency domain and/or the time domain. Particular modifications include amplifying spectral components, weighting spectral components based upon prior spectral components, and averaging spectral components with prior spectral components.

Multiple channel adaptive data recovery system

Abstract: A microprocessor controlled data recovery unit with an adjustable sampling and signal comparison level. The data recovery unit includes a data channel and a monitor channel. The monitor channel samples an incoming data stream in a varying manner. The results of the sampling in the monitor channel are used to adjust the sampling and comparing of the signal in the data channel. The data recovery unit includes a PLL based clock recovery unit in one embodiment, and in another embodiment the clock signal is derived by the microprocessor.

Mixed echo train acquisition displacement encoding with stimulated echoes : An optimized DENSE method for in vivo functional imaging of the human heart

Abstract: Cardiac functional imaging based on displacement encoding with stimulated echoes (DENSE) has been proposed as a high-resolution method for evaluating cardiac contractility (1–5). This was first combined with a segmented echo-planar imaging readout (fast-DENSE), and its application has been demonstrated with normal volunteers (6). However, two issues arose with respect to utilizing this methodology for routine patient cardiac functional evaluation. First, the signal-to-noise ratio (SNR) of the images acquired was insufficient for encoding the entire systolic interval. Specifically, only the last 100–150 ms of systole were captured for functional evaluation. Second, the breath-holds required for such exams were on the order of 26 heartbeats. This certainly posed a problem for evaluating function in patients with myocardial infarction. The solution to both issues is presented in this work. It involves a new way of sampling the stimulated echo (STE) signal and improved resolution by suppressing the FID signal. This method, mixed echo train acquisition DENSE (meta-DENSE), is applied successfully in both normal volunteers and patients to quantitatively evaluate cardiac function at high resolution in situ. In our previous work with fast-DENSE, the duration of the encoding interval and data acquisition speed were both constrained by the SNR, as already mentioned (Fig. 1). Since DENSE is based on STEs, there is an inherent 50% signal loss. In addition, myocardial strain alters the shape of the voxels during the encoding period. As a result, intravoxel dephasing can further lower the SNR, especially where strain is high along the longest dimension of the voxel, i.e., along the slice axis (6,7). Intravoxel dephasing signal loss is accentuated if strong TE gradient pulses are applied along the slice axis (Fig. 1). Such pulses are routinely used in stimulated echo acquisition mode (STEAM) experiments to eliminate unwanted signal contributions that can cause artifacts. The previous implementation (fast-DENSE) used a low RF flip angle segmented-EPI readout. The position-encoded magnetization was recalled in small portions onto the XY-plane and then discarded. Despite its ability to acquire data fast, the EPI-based readout is inefficient for acquiring DENSE data. Unlike other EPI applications, with DENSE the available magnetization that contains useful phase-encoded displacement information is limited. A fast readout scheme that recycles the available magnetization is more suitable for sampling the STE in DENSE experiments. One approach to this goal has been to utilize a fast-spin-echo (FSE) readout (8,9). That method yielded 3D static displacement strain images. However, there were significant artifacts with the normal FSE readout. The source of these artifacts can be traced to the mixing of the original STEAM signal with additional STEs that are created by the FSE pulse train (10). FIG. 1 Fast-DENSE image acquired with a segmented EPI readout. Note the signal loss due to intravoxel dephasing in the myocardium (arrow). This magnitude image was acquired with an encoding interval of 100 ms covering the end portion of systole. Alternative FSE readouts applied to diffusion weighting can be utilized to sample the STEAM signal. In one such approach, diffusion weighting was accomplished via the Stejskal Tanner sequence (11) and read out was performed in a single-shot via phase-insensitive rapid acquisition with relaxation enhancement (RARE) preparation (12). This preparation, via a dephasing gradient, divides the signal evenly between a Meiboom Gill (MG) (13) and an orthogonal component. The latter is then nutated onto the z-axis, where it remains during image acquisition, allowing the MG component to be imaged free of artifacts. It is important to note that if a signal has already lost its non-MG component, such a phase-insensitive RARE preparation is no longer required. One such signal is that generated by the STEAM preparation in DENSE, as will be shown. Therefore, the STEAM signal can be sampled by a train of 180° pulses with no artifacts. Another approach to sampling the STEAM signal is split acquisition of fast spin-echo signals for diffusion imaging (14), which has also utilized an FSE readout. However, SPLICE reduces artifacts by reconstructing two separate magnitude images from two echo trains (14) created by a readout gradient imbalance rather than exploiting the inherent MG properties of the STEAM signal. Indeed, the two echo trains of SPLICE can be forced to coincide with the STE and the stimulated anti-echo (STAE) (15) of STEAM and result in artifact-free images, as will be described later. In this work, the underlying cause of artifacts generated in STEAM displacement imaging with a conventional FSE readout is explained. The use of the STAE along with the conventional notion of the STE will facilitate the description process. It will be shown that the DENSE preparation is inherently compatible with a specialized readout train of 180° refocusing pulses, and as such artifact-free phase images can be obtained. The STEAM signal, which is comprised of two components, is RF-refocused so that not only is phase-contrast consistency maintained but also any additional STEs, created by imperfect 180°s during the readout, combine coherently with its two components. Additionally, an inversion-recovery nulling method will be used to eliminate unencoded water and fat signal contributions, which can cause artifacts when low encoding strengths are used in conjunction with high resolution. Finally, data acquired with this new method from normal volunteers and a patient in 14 heartbeats will be shown.

PLL circuit, data detection circuit and disk apparatus

Abstract: A PLL circuit is disclosed which extracts phase difference information of a high S/N ratio from a readout signal uses the phase difference information for PLL control. An A/D converter samples the input signal to produce a digital signal. A pattern string detector identifies a type of an input pattern string formed from a plurality of successive sample values successively outputted from the A/D converter and outputs pattern string identification information which indicates an identification result. A phase difference generator outputs phase difference information which indicates a phase error of the output of the A/D converter based on the pattern string identification information and the output of the A/D converter. A loop filter, a D/A converter and a voltage controlled oscillator generate a clock signal from the phase difference information to control the sampling timing of the A/D converter.

Integrated time domain reflectometry (TDR) tester

Abstract: An integrated TDR for locating transmission line faults. An integrated circuit comprises a transmitter, a path coupled to the transmitter, and a TDR receiver integrated with the transmitter for analyzing a reflected signal from the path. The TDR receiver compares the reflected signal with a variable reference signal to generate a logic state at a sampling instant determined by a timebase generated by a sampling circuit. The reflected signal equals the variable reference signal when the logic state transitions. The reference signal and the corresponding timebase value are recorded at the logic state transition. A waveform is generated from the recorded reference signal and its corresponding timebase value. A reference point for the waveform is determined. The location of a fault on the transmission line can be determined from the timebase value difference between the reference point and the fault.

Polyphase channelization system

Abstract: A polyphase channelizer converts an intermediate frequency wideband signal into a complex signal that is sampled by parallel analog-to-digital converters having a bank of samplers respectively clocked by staggered clocking signals for respective converters for feeding I and Q quadrature samples to a polyphase filter bank of finite impulse response filters driving a fast Fourier transform processor for providing channelized digital signal outputs. The parallel analog-to-digital converters can operate at lower speeds but are parallel connected for effectively operating at required higher speeds. The channelizer channelizes the wideband signal using a polyphase clock for enabling high speed sampling and converting through low speed parallel analog to digital converters.

High-speed serial data transceiver and related methods

Abstract: A high-speed serial data transceiver includes multiple receivers and transmitters for receiving and transmitting multiple analog, serial data signals at multi-gigabit-per-second data rates. Each receiver includes a timing recovery system for tracking a phase and a frequency of the serial data signal associated with the receiver. The timing recovery system includes a phase interpolator responsive to phase control signals and a set of reference signals having different predetermined phases. The phase interpolator derives a sampling signal, having an interpolated phase, to sample the serial data signal. The timing recovery system in each receiver independently phase-aligns and frequency synchronizes the sampling signal to the serial data signal associated with the receiver. A receiver can include multiple paths for sampling a received, serial data signal in accordance with multiple time-staggered sampling signals, each having an interpolated phase.

Broadband design of a probe analysis system

Abstract: A radio frequency (RF) probe analysis system has a broadband design. The analysis system includes a sampling unit for generating digital power signals based on a plurality of analog signals. The analog signals characterize power delivered from an RF power delivery system to a plasma chamber. The analysis system further includes a digital processing unit for generating a digital spectrum signal based on the digital power signals. The sampling unit simultaneously samples a first plurality of frequencies from the analog signals such the that digital spectrum signal defines signal levels for the first plurality of frequencies. The sampling unit may also simultaneously sample a second plurality of frequencies from the analog signals such that the digital spectrum signal further defines signal levels for the second plurality of frequencies. The broadband architecture of the sampling unit enables closed loop control of power delivered to the chamber to tolerances unachievable through conventional approaches.

Communication receiver with signal processing for beam forming and antenna diversity

Abstract: The present invention provides a technique to reduce the number of RF components by using digital signal processing functions in wireless receivers having two or more antennas, which results in the reduction of system cost. The reduction occurs by using a multiplexer or switch to sample the signal received by each of the antennas. The sampling rate of the multiplexer or switch is greater than the Nyquist required sampling rate (Fs) of the received signal bandwidth. The sampled signal is a multiplexed single analog (RF) signal, which only requires one chain of receiver components. A signal processor then is able to demultiplex the received signal at a lower frequency and can perform several functions including antenna diversity and beam forming utilizing digital signals at IF or baseband frequencies.

Ground-penetrating imaging and detecting radar

Abstract: A ground-penetrating radar comprises a single resonant microstrip patch antenna (RMPA) that is driven by a three-port directional coupler. A reflected-wave output port is buffered by a wideband isolation amplifier and a reflected-wave sample is analyzed to extract measured values of the real and imaginary parts of the load impedance-the driving point impedance of RMPA. Each such port will vary in a predictable way according to how deeply an object is buried in the soil. Calibration tables can be empirically derived. Reflections also occur at the interfaces of homogeneous layers of material in the soil. The reflected-wave signals are prevented from adversely affecting transmitted-signal sampling by putting another wideband isolation amplifier in front of the input port of the directional coupler. A suppressed-carrier version of the transmitted signal is mixed with the reflected-wave sample, and the carrier is removed. Several stages of filtering result in a DC output that corresponds to the values of the real and imaginary parts of the load impedance. The suppressed-carrier version of the transmitted signal is phase shifted 0° or 90° to select which part is to be measured at any one instant.

Wide dynamic range operation for CMOS sensor with freeze-frame shutter

Abstract: Wide dynamic range operation is used to write a signal in a freeze-frame pixel into the memory twice, first after short integration and then after long integration. The wide dynamic range operation allows the intra-scene dynamic range of images to be extended by combining the image taken with a short exposure time with the image taken with a long exposure time. A freeze-frame pixel is based on voltage sharing between the photodetector PD and the analog memory. Thus, with wide dynamic range operation, the resulting voltage in the memory may be a linear superposition of the two signals representing a bright and a dark image after two operations of sampling.

Estimation of timing offsets in parallel A/D converters

Abstract: A parallel analog-to-digital converter device has parallel converter cells signal (u) appearing at an input line. The cells are clocked by a common clock signal delayed by delay elements for the respective cell. Timing errors of the sampling times are determined in a calculation unit receiving the output signals of the cells. The output signal of the total converter is compensated by a compensation device for the determined timing errors to provide corrected sampled values on the output line of the converter. In the calculation unit sums of squared differences of the sampled values for successive cells are calculated which by simple operations provide estimates of the timing errors. In the calculation unit also a variance of noise superposed on the sampled values can be estimated and then used to correct the calculated sums for the variance. The input signal is basically unknown and the determination of the timing errors works without using any calibration signal. However, the energy spectrum of the signal should be concentrated to lower frequencies. By compensating for the timing errors the accuracy of the A/D converter is significantly improved.

System and method for estimating a frequency of slider airbearing resonance

Abstract: A method and apparatus for estimating the value of a slider airbearing resonance frequency involves obtaining a readback signal from a data storage medium over a plurality of complete airbearing periods and estimating the value of an airbearing resonance frequency using the readback signal. In one embodiment, a discrete signal segment comprising a plurality of frequency transform components is produced using the readback signal information, and the value of the airbearing resonance frequency is estimated using spectral leakage in the discrete signal segment. A ratio of the magnitudes of a first DFT component to a second DFT component is computed at each of a plurality of sampling rates. Each of these sampling rates is defined by a number of samples per average airbearing cycle multiplied by a frequency falling within a range of expected airbearing frequencies associated with a given implementation. The second DFT component is related to the slider airbearing resonance frequency, and the first DFT component is a DFT component adjacent to or non-adjacent to the second DFT component. The airbearing resonance frequency value is estimated using a minimum of the ratios, which may also constitute DFT component power ratios. A number of different frequency transform techniques may be employed, including Discrete Fourier Transform, Fast Fourier Transform, and Short-Time DFT techniques. One of several frequency transform approaches may be implemented depending on whether the detected airbearing signal is stationary or non-stationary. The airbearing resonance frequency methodology may be implemented in-situ a data storage system.

Driver circuit of a display device

Abstract: In a driver circuit of a display device handling a digital image signal, there is provided a driver circuit with a structure in which the timing of holding the image signal in a latch circuit is not influenced by a delay of a sampling pulse. A pre-charge TFT ( 102 ) is turned ON in a return line period and an input terminal of a holding portion ( 101 ) is set as Hi (VDD). When there is input to all the three signals, the sampling pulse, and a multiplex signal and the digital image signal which are input from the outside, TFTs ( 104 to 106 ) all turn ON, and the potential of the input terminal of the holding portion ( 101 ) becomes a Lo potential. Thus, holding of the digital image signal is performed. A sampling pulse width is wider than a pulse width of the two signals input from the outside, and the output periods of the two signals input from the outside are completely included in an output period of the sampling pulse. Thus, even if a slight delay is generated, there is no influence on the holding timing, and the holding timing may be easily determined.

Blind estimation of timing errors in interleaved AD converters

Abstract: Parallel A/D converter structures is one way to increase the sampling rate. Instead of increasing the sample rate in one A/D converter, several A/D converters with lower sampling rate can be used instead. A problem in these structures is that the time between samples is usually not equal because there are errors in the delays between the A/D converters. We present a method to estimate the timing offset errors. The estimation algorithm works without any special calibration signal, instead the normal input signal is used. The only assumption that we need on the input signal is that most of the energy is concentrated to a low pass band, below about 1/3 of the Nyquist frequency. Simulations of the time-interleaved A/D converter show that the method estimates the errors with high accuracy.

Liquid crystal display device, driving method for liquid crystal display devices, and inspection method for liquid crystal display devices

Abstract: Using technology which uses a single shift register and simultaneously generates multiple pulses, this invention is a liquid crystal display device which rapidly drives data lines It is possible to increase the frequency of the shift register output signal without changing the frequency of the shift register operation clock If the shift register output signals, by means of analog switches, are used to determine the video signal sampling timing, high speed data line driving can be realized Additionally, if the output signals of the shift register mentioned above are used to determine the video signal latch timing in a digital driver, high speed latching of the video signal can be realized Consequently, even if the driving circuits of the liquid crystal display matrix are composed of TFTs, high speed operation of the driving circuits is possible without increasing power consumption The shift register can also be used to inspect the electrical characteristics of the data lines and analog switches

Broadcast transmission system with sampling and correction arrangement for correcting distortion caused by amplifying and signal conditioning components

Abstract: A transmission system (14) broadcasts a signal, having (14), a power amplifier (20) that causes non-linear distortion. A pre-amp component, such as a band-pass filter (32), causes linear distortion. A high power filter (38) is located downstream of the power amplifier (20) and causes linear distortion. A linear equalizer (42) compensates for the distortion caused by the high power filter (38). A non-linear corrector (44) compensates for the distortion caused by the power amplifier (20), and is located downstream of the linear equalizer (42). A linear equalizer (46) compensates for the distortion caused by the pre-amp component (e.g., 32). The compensating components (42-46) are located upstream of the distorting, pre-amp component (e.g., 32).

Design of oversampled uniform DFT filter banks with delay specification using quadratic optimization

Abstract: Subband adaptive filters have been proposed to avoid the drawbacks of slow convergence and high computational complexity associated with time domain adaptive filters. Subband processing introduces transmission delays caused by the filter bank and signal degradations due to aliasing effects. One efficient way to reduce the aliasing effects is to allow a higher sample rate than critically needed in the subbands and thus reduce subband signal degradation. We suggest a design method, for uniform DFT filter banks with any oversampling factor, where the total filter bank group delay may be specified, and where the aliasing and magnitude/phase distortions are minimized.

Dither signal insertion inversely proportional to signal level in delta-sigma modulators

Abstract: A method is disclosed to operate a sigma-delta modulator of a type that includes a quantizer. The method has steps of (a) sampling an amplitude of an input signal to the sigma-delta modulator; and (b) controlling the switching of a capacitance bank in accordance with the sampled amplitude of the input signal for generating a dither signal at an input of the quantizer. The dither signal is generated to have a pseudorandom amplitude that is inversely proportional to the sampled amplitude of the input signal. The step of controlling and generating operates a linear feedback shift register to switch individual ones of a plurality of capacitances of the bank of capacitances in and out of a capacitance network. In one embodiment the step of operating the at least one linear feedback shift register turns a linear feedback shift register clock signal on and off as a function of the amplitude of the input signal. In one embodiment the step of sampling operates at least one window detector, and the dither signal is turned off and on depending on a relationship between the amplitude of the input signal and voltage thresholds of the window detector. In another embodiment the step of sampling operates a rectifier that rectifies the input signal to provide a rectified output signal, and the step of controlling and generating pseudorandomly applies the rectified output signal to the bank of capacitances for controlling an amount of current that is transferred between the input of the quantizer and the bank of capacitances.

Harmonic tunnelling: tracking non-stationary noises during speech.

Abstract: This paper presents a novel noise robust front-end algorithm and evaluates its performance on the Aurora 2 database. Most algorithms aimed at improving the performance of recognisers in background noise make an estimate of the noise spectrum that is then used to obtain an improved estimate of the spectrum of the underlying speech. In the case of stationary noises it is sufficient to take an average noise spectrum from the period before the speech utterance and/or to use a speech/non-speech detector to update this estimate using the noise sampled from any gaps in an utterance. For nonstationary noises where the noise spectrum changes faster than the duration of a typical utterance (e.g. within 0.5s) then there can be substantial differences between the estimated and actual noise spectrum for a particular frame, leading to poor performance. The algorithm presented here provides an improved estimate of the noise that can be tracked throughout the duration of the speech utterance, by making use of the harmonic structure of the voiced speech spectrum. This running estimate of the noise is obtained by sampling the noise spectrum in the gaps (or “tunnels”) between the harmonic spectral peaks. Compared to the ETSI standard MFCC front-end [1], the proposed algorithm delivers an average improvement in performance of 43.93% on the Aurora 2 database [2]. 1. Front-end Algorithm Overview A summary of the algorithm is shown in figure 1 and the details of the processing performed by each block described in the section 2. Figure 1: Block diagram of front-end algorithm The spectrum of the signal is first obtained by taking an FFT. The peaks in this spectrum are then determined from spectral derivatives. Each of these candidate peaks are analysed to categorise them as a peak coming from either a voiced speech harmonic or noise. The noise spectrum at a peak categorised as speech is estimated by interpolation from the adjacent noise spectra in the surrounding “tunnels”. These frame based noise measurements contribute to the running average of the noise spectrum in the Mel domain. Whilst this noise estimate could be used in many alternative algorithms, in this implementation it is used for an SNR dependent spectral subtraction. The remaining processing blocks are spectral normalisation performed in the Mel domain, normalising with the long term average of the spectrum and also by the frame energy. Finally a cube root compression is performed followed by cosine transform to produce 12 cepstral coefficients and a log energy measure. 2. Front-end Algorithm Details

Multi-wavelength locking method and apparatus for wavelength division multiplexing (WDM) optical communication system

Abstract: Disclosed is a multi-wavelength locking method for a wavelength division multiplexing (WDM) optical communication network, and in particular, a multi-wavelength locking method and apparatus for a WDM optical communication system that can lock wavelengths of optical signals by producing pilot tones by applying a sine-wave current to a plurality of transmission lasers having different wavelengths, passing the optical signal through a Fabry-Perot etalon filter, and then Fourier-transforming the filtered optical signal. The multi-wavelength locking method includes frequency-modulating an optical signal by applying a small and specified sine-wave current to a bias current for driving WDM lasers, detecting pilot tones produced after filtering the optical signal through a filtering section and converting the detected signal into a digital signal by performing a sampling of the detected signal, detecting a magnitude and phase of the pilot tones by performing a fast Fourier transform, providing Fourier-transformed data as a first derivative signal of the filtering section, and locking an operation wavelength of WDM channels by controlling the temperature or current if resonance frequencies of the filtering section coincide with respective standard frequency.

System and method for analyzing baby cries

Abstract: This invention provides a system for analyzing baby cries capable of diagnosing a cause of cry of a baby based on a cry from the baby. A microphone ( 1 ) picks up a cry from a baby as an audio signal. At a certain sampling frequency, an A/D converter ( 2 ) samples the audio signal received by the microphone ( 1 ) to A/D convert it. An audio analyzer ( 3 ) analyzes the audio signal samples by the A/D converter ( 2 ) and computes a characteristic quantity based on a frequency spectrum. A cause-of-cry assumption unit ( 4 ) assumes a cause of cry based on the characteristic quantity of the audio signal derived at the audio analyzer ( 3 ). Finally, an assumed result display ( 5 ) displays the assumed result from the cause-of-cry assumption unit ( 4 ).

Ultrasonic diagnostic apparatus and method for processing ultrasonic signal

Abstract: An ultrasonic diagnostic apparatus, which can acquire images with different frequencies, comprises a probe made up of a plurality of elements which transmit ultrasonic waves and receive reflected ultrasonic waves, a digital conversion part which digitizes a plurality of received signals, a first mixing part which multiplies the output of the digital conversion part and a first digital reference signal, a first filter part which extracts a signal having a predetermined center frequency out of the output of the first mixing part, a digital delay part which delays the output of the first filter part, an adding part which adds a plurality of outputs of the digital delay part, a second mixing part which multiplies the output of the adding part and a second digital reference signal, a detection part which detects the output of the second mixing part, a conversion part which converts the output of the detection part into an image signal, and a display device which displays the output signal of the conversion part. This configuration dispenses with changes in frequency characteristics of the filter part.

Flexible TDMA system architecture

Abstract: A wireless TDMA communication platform for processing a communication signal is disclosed herein. The platform includes a sampler for sampling a TDMA signal, received from a transmission channel; a derotator (304) for correcting for frequency offset in the sampled TDMA signal; a matched filter (306) for correcting for the response of the transmission channel in the received TDMA signal; an equalizer (307) to which is applied an output signal from the matched filter; a deinterleaver (309) to deinterleave the received TDMA signal; and a channel decoder (310) for decoding the received TDMA signal after it is deinterleaved.

Method and apparatus for sample rate pre-and post-processing to achieve maximal coding gain for transform-based audio encoding and decoding

Abstract: The invention relates to a method and apparatus for achieving maximal coding gain for audio transmission. More particularly, at a chosen sample rate and frequency range value, an audio input signal is downsampled to the sample rate, encoded and transmitted at a given bit rate. At the receiving end, the downsampled signal is decoded and upsampled to the original or other suitable sample rate. The upsampled signal is then audibly output. Since resampling ratios using “small” numbers prove to be more computationally efficient, this method and apparatus supports resampling ratios which imply both standard and non-standard sampling ratios in the codec.