Showing papers on "Sampling (signal processing) published in 1995"

Effects of undersampling on the proper interpretation of modulation transfer function, noise power spectra, and noise equivalent quanta of digital imaging systems

Abstract: The proper understanding of modulation transfer function (MTF), noise power spectra (NPS), and noise equivalent quanta (NEQ) in digital systems is significantly hampered when the systems are undersampled. Undersampling leads to three significant complications: (1) MTF and NPS do not behave as transfer amplitude and variance, respectively, of a single sinusoid, (2) the response of a digital system to a delta function is not spatially invariant and therefore does not fulfill certain technical requirements of classical analysis, and (3) NEQ loses its common meaning as maximum available SNR2 (signal-to-noise) at a particular frequency. These three complications cause the comparisons of MTF and NEQ between undersampled digital systems to depend on the frequency content of the images being evaluated. A tutorial of MTF, NPS, and NEQ concepts for digital systems is presented, along with a complete theoretical treatment of the above-mentioned complications from undersampling.

Time-of-flight radio location system

Abstract: A bi-static radar configuration measures the direct time-of-flight of a transmitted RF pulse and is capable of measuring this time-of-flight with a jitter on the order of about one pico-second, or about 0.01 inch of free space distance for an electromagnetic pulse over a range of about one to ten feet. A transmitter (53) transmits a sequence of electromagnetic pulses in response to a transmit timing signal, and a receiver (58) samples the sequence of electromagnetic pulses with controlled timing response to the samples. A timing circuit supplies the transmit timing signal to the transmitter and supplies the receive timing signal to the receiver. The receive timing signal causes the receiver to sample the sequence of electromagnetic pulses such that the time between transmission of pulses in the sequence and sampling by the receiver sweeps over a range of delays.

Progressive scanning conversion apparatus

Abstract: A progressive scanning conversion apparatus for converting an interlaced scan video signal into a progressive scan video signal by performing interpolation based on original pixels which are obtained in a matrix by sampling the interlaced scan video signal in a current field, original pixels which are obtained in a matrix by sampling the interlaced scan video signal in a previous field which is immediately prior to the current field, and original pixels which are obtained in a matrix by sampling the interlaced scan video signal in a next field which is immediately following the current field The apparatus includes a motion vector detector for detecting a motion vector between two of the current field, the previous field, and the next field; a motion estimation circuit for estimating a pixel value in the current field, from one of a pixel in the previous field and a pixel in the next field using the motion vector detected by the vector detection means; and an interpolated pixel generator for generating an interpolated pixel used for conversion by multiplying the pixel value obtained by the motion estimation means and the corresponding pixel value in the current field by a weighting factor and summating the multiplication results

128 x 128 CMOS photodiode-type active pixel sensor with on-chip timing, control, and signal chain electronics

Abstract: A 128 X 128 element CMOS active pixel image sensor (APS) with on-chip timing, control, and signal chain electronics has been designed, fabricated and tested The chip is implemented in 12 micrometers n-well process with a 192 micrometers pixel pitch The sensor uses a photodiode-type CMOS APS pixel with in-pixel source follower, row selection and reset transistors The sensor operates from a +5 V supply and requires only a clock signal to produce video output The chip performs correlated double sampling (CDS) to suppress pixel fixed pattern noise, and double delta sampling (DDS) to suppress column fixed pattern noise The on-chip control circuitry allows asynchronous control of an inter frame delay to adjust pixel integration On-chip control is also provided to select the readout of any size window of interest

Sampled amplitude read channel employing interpolated timing recovery

Abstract: A sampled amplitude read channel for reading information stored on a magnetic medium by detecting digital data from a sequence of discrete time interpolated sample values, the interpolated sample values generated by interpolating a sequence of discrete time channel sample values generated by sampling pulses in an analog read signal from a magnetic read head positioned over the magnetic medium. A write VFO generates a write clock for writing digital data to the magnetic medium at a predetermined baud rate for a selected zone, and upon read back, the write VFO generates a sampling clock at a frequency slightly higher than the write frequency. A sampling device samples the analog read signal at the sampling clock rate to generate a sequence of discrete time channel samples that are not synchronized to the baud rate. The channel samples are equalized by a discrete time equalizing filter according to a predetermined partial response (PR4, EPR4, EEPR4, etc.). An interpolating timing recovery circuit, responsive to the equalized channel samples, computes an interpolation interval τ and, in response thereto, generates interpolated sample values substantially synchronized to the baud rate. The timing recovery circuit also generates a data clock for clocking a discrete time sequence detector for detecting the digital data from the interpolated sample values.

Aliasing reduction in staring infrared imagers utilizing subpixel techniques

Abstract: We introduce and analyze techniques for the reduction of aliased signal energy in a staring infrared imaging system. A standard staring system uses a fixed two-dimensional detector array that corresponds to a fixed spatial sampling frequency determined by the detector pitch or spacing. Aliasing will occur when sampling a scene containing spatial frequencies exceeding half the sampling frequency. This aliasing can significantly degrade the image quality. The aliasing reduction schemes presented here, referred to as microscanning, exploit subpixel shifts between time frames of an image sequence. These multiple images are used to reconstruct a single frame with reduced aliasing. If the shifts are controlled, using a mirror or beam steerer for example, one can obtain a uniformly sampled microscanned image. The reconstruction in this case can be accomplished by a straightforward interlacing of the time frames. If the shifts are uncontrolled, the effective sampling may be nonuniform and reconstruction becomes more complex. A sampling model is developed and the aliased signal energy is analyzed for the microscanning techniques. Finally, a number of experimental results are presented that illustrate the perlormance of the microscanning methods.

Direct downconverter circuit for demodulator in digital data transmission system

Abstract: A digital downconverter circuit for a digital data transmission system is provided, comprising (i) a bandpass filter circuit for filtering a received analog intermediate frequency (IF) carrier signal onto which baseband information has been modulated, and outputting a bandpass analog IF signal; (ii) an analog-to-digital converter for converting the bandpass analog IF signal into a bandpass digital IF signal; (iii) a phase shifter device for outputting a complex pair of phase shifted baseband signals operating at a local baseband frequency; (iv) a mixer device for mixing the bandpass digital IF signal separately with each of the complex pair of phase shifted baseband signals and outputting a complex combined baseband/bandpass signal comprising inphase and quadrature components; (v) filtering circuitry for recovering the baseband information onto the phase shifted baseband signals; and (vi) sampling circuitry for sampling the information-bearing recovered baseband signals and outputting a sampled complex baseband output signal. The phase shifter device, the mixer device, the filtering circuitry, and the sampling circuitry are all implemented on a gallium arsenide (GaAs) application specific integrated circuit (ASIC). The filtering circuitry includes a single half-band pre-filter and a multi-pole low pass filter. The analog-to-digital converter operates at the sampling clock frequency, which is about 512 megahertz (MHz), and the IF carrier signal operates at about 52-176 MHz. An automatic gain control circuit controls the amplitude of the bandpass analog IF signal prior to its being converted into the bandpass digital IF signal.

Split frequency band signal digitizer and method

Abstract: A split frequency band signal digitizer (10) includes a first mixer (18) for translating a split-band signal (200) about a multiple of the sampling frequency The split-band signal so translated is then communicated to an analog-to-digital converter (26) The digitized signal is then analyzed to recover an original portion of the split-band signal and a second portion of the signal aliased into a single Nyquist band

Digital signal processing apparatus

Abstract: A specific pattern in an inputted digital signal is detected (13) and the input digital signal is sampled and held (14) in accordance with such a detection output. A clock (15-17) whose phase is synchronized with the input digital signal is generated on the basis of the sampling output. With such a construction as mentioned above, the phase of the clock can be optimally controlled by a simple construction. The clock can be precisely extracted from the input digital signal, so that the operation of a circuit is also stable.

Adaptive spread spectrum receiver

Abstract: A sampling signal SPS of a received signal which is subjected to spectrum spreading is inputted to matching filters 121 to 124 and is compressed back. A spreading code of a desired wave is used as a compressing code in the filter 121, and a plurality of diffusion codes orthogonal to the spreading code of the desired wave and orthogonal to one another are used for compression in the filters 122 to 124. The outputs x1 to x4 of the filters 121 to 124 are multiplied by weighting coefficients w1 to w4, respectively, and the productions are linearly combined and outputted as a composite signal DCS. The signals x1 to x4 multiplied by the weighting coefficients w1 to w4 and the composite signal DCS are inputted to a coefficient control part 37, and the weighting coefficients w1 to w4 are determined by an algorithm which minimizes the mean power of the composite signal DCS under the restriction condition of these weighting coefficients w1 to w4.

Synchronous detection of concurrent servo bursts for fine head position in disk drive

Abstract: Synchronous detection of fine position servo information within a partial response maximum likelihood (PRML) data channel Servo information (17) is recorded as a pair or series of fractional-track-width sinewave concurrent burst patterns producing an on-track phase generating a position error signal varying linearly about track centerline and at least one off-track phase generating a position error signal related to track boundary Head (26) generates on-track phase and off-track phase analog signals that are gain normalized (40) and synchronously quantized (46) Mulfiplying by a normalization factor from a correlation signal generator (64) during servo sampling intervals provides normalized samples, integrating (66) on-track phase normalized samples provides on-track position error signal (82), and integrating (66) off-track phase normalized samples provides an off-track position error signal (88) A discrete matched filter is also used

Method and apparatus for decoding bar code data from a video signal and application thereof

Abstract: A new technology in bar code scanning and digitizing is provided where a bar code within a composite video signal may be detected and decoded. In a preferred embodiment, the composite video signal is converted into a form suitable for providing to a bar code decoder, in order that any bar code within the video signal may be decoded by the decoder. The output of the bar code decoder may thereafter be provided to, for example, a computer for the further processing. For purposes of the present invention, the composite video signal which is decoded may be generated by a video camera, a video tape recorder, a television signal, or any other source which generates such a video signal. A sampling of the multitude of possible applications of the present invention is also provided, whereby bar codes detected by, for example, video cameras in different environments may be decoded and processed by a computer.

Symbol and frame synchronization in both a TDMA system and a CDMA

Abstract: A digital radio communication system includes a receiver for receiving a signal stream that includes data frames, each frame including a data signal sequence and a synchronizing signal sequence. The communication system synchronizes the receiver by employing the signal stream. The receiver comprises: a sampling circuit for sampling symbol levels in the synchronizing signal sequence; cross correlation circuitry for comparing values derived from the sampled symbol levels with an expected set of values and producing an error value output; and correction circuitry that is responsive to the error output and produces a sample control output to the sample circuitry to alter the times of sampling of the symbol levels so as to reduce the error output and achieve time synchronization with the received synchronizing signal sequence. A further embodiment of the invention is described in relation to a CDMA receiving system.

A generalization of nonuniform bandpass sampling

Abstract: Nth-order nonuniform sampling is described for generalized bandpass signal frequency position, bandwidth, sampling rate, frequency-shift and phase-shift. A bandpass extension to the Nyquist criterion is derived, showing that restrictions on bandpass frequency position for odd orders of nonuniform sampling tend to zero as N tends to infinity. Bandpass interpolants based on the sinc function are derived for the generalized Nth-order sampled bandpass signals. It is shown that, for minimum (Nyquist) rate sampling, these interpolants are comprised of N bandpass filters, each with independent phase. The number of bandpass filters comprising the interpolant is found to decrease as the sample rate increases. The advantage of describing Nth-order sampling as the Nth replication and uniform sampling of a signal is demonstrated. Finally, digital implementation of the Nth-order bandpass sampling interpolants is discussed. It is established that it is not practicable to attempt to perform nonuniform bandpass sampling at the theoretical minimum rate, where the interpolation is to be performed digitally. >

Variable sample rate adc

Abstract: A method and apparatus for analog-to-digital conversion using sigma-delta modulation of the temporal spacing between digital samples are provided. The method and apparatus include sigma-delta modulation of the time-base such that errors produced by non-uniform sampling are frequency-shaped to a high frequency region where they are reduced by conventional digital filtering techniques. In one embodiment, a sigma-delta ADC receives an analog input signal and converts the analog input signal to digital samples at an oversampling rate. A decimator, coupled to the sigma-delta ADC, receives the digital samples and decimates the digital samples to produce the digital samples at a preselected output sample rate, less than the oversampling rate. An ADC sample rate control circuit, coupled to the ADC, receives a frequency select signal representing the preselected output sample rate, and produces a noise-shaped clock signal for controlling operation of the ADC at the oversampling rate. The control circuit includes a sigma-delta modulator for sigma-delta modulating the frequency select signal. A randomizer/suppressor circuit, under control of the output of the sigma-delta modulator, receives an input clock signal and adjusts the frequency of the clock signal to produce a noise-shaped clock signal for controlling the oversampling rate of the ADC.

Active matrix display device and its driving method

Abstract: To restrict a potential oscillation in a video line caused by a high speed sampling rate, the active matrix display device is comprised of gate lines X in row, signal lines Y in column and liquid crystal pixels LC of matrix arranged at each of the crossing points of both lines. The V driver 1 scans in sequence each of the gate lines X and selects the liquid crystal pixels LC in one line for every one horizontal period. The H driver 4 performs a sampling of the video signal VSIG for each of the signal lines Y and performs a writing of the video signal VSIG in the liquid crystal pixels LC in one selected line within one horizontal period. The precharging means 5 supplies the predetermined precharging signal VPS to each of the signal lines Y just before writing the video signal VSIG for the liquid crystal pixels LC in one line. With such an arrangement as above, it is possible to reduce the charging or discharging amount in each of the signal lines Y when the video signal VSIG is sampled and further to restrict the potential oscillation in the video line 2.

Data collection and processing for digital AC power system monitor/analyzer

Abstract: A digital monitor/analyzer (1) for an ac electrical power system (3) samples the current and voltage waveforms at a slow rate for performing metering functions, and at a high rate for analyzing harmonic content. Sampling is performed in sampling frames (35) comprising a plurality of repetitions (37₁-37₄) of sampling for a selected number of cycles followed by a delay (δ) of a fraction of a cycle. This effects equivalent sampling at a higher rate for slow speed sampling over a frame (35). High speed sampling is performed over only one repetition (37₃) of the selected number of cycles in a frame (35) so that it can be carried out sychronously as required for the harmonic analysis. The high speed sampling rate is an integer multiple of the slow speed rate so that slow speed data can be extracted from the high speed samples for continuous metering. Fourier analysis of the harmonic content of the waveforms is performed (61) on the odd interrupts while all of the other tasks are apportioned out (59) over the even interrupts of a sampling frame. Both the odd and even interrupts initiate sampling.

Radio communication apparatus

Abstract: A radio communication apparatus comprises a first/second modulator for modulating a first/second signal according to a first/second frequency, a first demodulator for detecting the received first modulation signal and regenerating the carrier, a clock regenerator for regenerating a clock from the detection output from the first demodulator, an amplitude sampler for sampling the amplitude of the received first modulation signal according to the regenerated clock, a second demodulator for controlling the frequency of the received second modulation signal and executing demodulation for the signal, and a gain controller for executing gain control for the output from the second demodulator according to the output from the amplitude sampler.

Method of measuring speed of a mobile unit and a receiver for use in a wireless communication system

Abstract: A method of measuring speed of a mobile unit for use in a wireless communication system 20. The method includes the steps of receiving a radio frequency (RF) signal from the mobile unit 202, measuring signal quality of the RF signal to produce a received quality signal 204, sampling the received quality signal during a first time period to produce a first group of samples 206, sampling the received quality signal during a second time period to produce a second group of samples 208, calculating a variation in signal quality of the RF signal in response to the first and second group of samples 210, and determining a speed measurement in response to the variation in signal quality 212.

Rate-distortion performance in coding bandlimited sources by sampling and dithered quantization

Abstract: The rate-distortion characteristics of a scheme for encoding continuous-time band limited stationary sources, with a prescribed band, is considered. In this coding procedure the input is sampled at Nyquist's rate or faster, the samples undergo dithered uniform or lattice quantization, using subtractive dither, and the quantizer output is entropy-coded, The rate-distortion performance, and the tradeoff between the sampling rate and the quantization accuracy is investigated, utilizing the observation that the coding scheme is equivalent to an additive noise channel. It is shown that the mean-square error of the scheme is fixed as long as the product of the sampling period and the quantizer second moment is kept constant, while for a fixed distortion the coding rate generally increases when the sampling rate exceeds the Nyquist rate. Finally, as the lattice quantizer dimension becomes large, the equivalent additive noise channel of the scheme tends to be white Gaussian, and both the rate and the distortion performance become invariant to the sampling rate. >

Sampling phase synchronizing apparatus and bidirectional maximum likelihood sequence estimation scheme therefore

Abstract: A sampling phase synchronizing apparatus includes a received signal memory for storing a received signal partially including training codes, a channel response calculator for calculating the time response of a channel (to be referred to as a channel response hereinafter) unique to the reception time using a partial sequence of the training codes included in the received signal stored in the received signal memory, a power ratio calculator for calculating the ratio between the power of the channel response calculated by the channel response calculator, and the power of a portion of the channel response, a power ratio memory for storing the power ratio calculated by the power ratio calculator, and a sampling phase deciding section for determining a sampling phase using the power ratio stored in the power ratio memory. The apparatus samples the received signal with reference to the sampling phase determined by the sampling phase deciding section.

Use of the Ar2+ signal as a diagnostic tool in solid sampling electrothermal vaporization inductively coupled plasma mass spectrometry

Abstract: The utility of the Ar2+ signal (at mass-to-charge ratio m/z= 80) as a diagnostic tool in solid sampling electrothermal vaporization inductively coupled plasma mass spectrometry (ETV–ICP-MS) is reported. Simultaneous monitoring of the argon dimer signal with the signal(s) of the analyte element(s) indicated that non-spectral interferences, caused by matrix components co-volatilizing with the analyte element(s), can strongly affect the analyte signal profiles in solid sampling ETV–ICP-MS of samples of biological or environmental origin. This observation led to a more profound understanding of why, for a given matrix, the signal profiles strongly differ from one element to another, and why, for a given element, the signal profile is seen to be strongly dependent on the matrix. These matrix effects were also observed to cause a curvature in the sample mass response curves (analyte signal intensity as a function of sample mass). It is shown that, at least in some instances, the use of the Ar2+ signal as an internal standard allows (i) this non-linearity to be corrected for and (ii) accurate analysis results to be obtained. Finally, it is demonstrated that simultaneous registration of the argon dimer and the analyte signal(s) is useful during optimization of ashing and vaporization temperatures.

Pitch lag estimation using frequency-domain lowpass filtering of the linear predictive coding (LPC) residual

Abstract: A pitch estimation device and method utilizing a multi-resolution approach to estimate a pitch lag value of input speech. The system includes determining the LPC residual of the speech and sampling the LPC residual. A discrete Fourier transform is applied and the result is squared. A lowpass filtering step is carried out and a DFT on the squared amplitude is then performed to transform the LPC residual samples into another domain. An initial pitch lag can then be found with lower resolution. After getting the low-resolution pitch lag estimate, a refinement algorithm is applied to get a higher-resolution pitch lag. The refinement algorithm is based on minimizing the prediction error in the time domain. The refined pitch lag then can be used directly in the speech coding.

Method and apparatus for determining and selectively displaying valid measurement information

Abstract: An electronic test instrument adapted for displaying only meaningful information notwithstanding the intermittent arrival of valid input signals due to probing operations is provided. Two independent measurement processes measure the input signal simultaneously. The first measurement process operates in a similar fashion to a digital storage oscilloscope (DSO) by successively sampling the input signal to produce waveform information which are selectively sent to an LCD display device which graphically displays the waveforrn. The second measurement process continually performs a stability assessment of the input signal by collecting a series of stability measurements of a selected input signal parameter, creating a moving average of the series, and comparing each new stability measurement to the moving average relative to stability criteria. The stability decision controls the flow of waveform information to the display, thereby ensuring that only meaningful information is displayed based on waveform scans conducted when the input signal is stable.

Fluorescence decay measurement by calculation of inner product

Abstract: An apparatus and its accompanying method measures fluorescence decay, fluorescence amplitude and fluorescence polarization. The instrument includes a light emitting diode or laser diode light source, an amplified photodiode detector, and electronics and software for calculating the phase of the fluorescence from the computed inner product of the scattering and fluorescence signals. The electronics for measuring fluorescence decay includes two signal synthesizers, two downconverters, a simultaneously sampling analog-to-digital converter and means for calculating the inner product of the downconverted waveforms. A signal source, preferably a direct digital synthesizer, is used in combination with a digital-to-analog converter to provide a controllable output signal. That signal drives a modulatable light source, which is a light emitting diode, a laser diode or a combination of the two. Excitation light from the light source excites the sample fluorophore which is immobilized in a solid material. Fluorescence is detected using an amplified photodiode. The received signal is downconverted using a pair of mixers. The downconverted signals are digitized and stored. Preferably, the signals are digitized using a two-channel synchronous analog-to-digital converter controlled by a digital signal processor. Data from each channel are multiplied and summed to provide the inner product.

Gabor's expansion and the Zak transform for continuous-time and discrete-time signals : critical sampling and rational oversampling

Abstract: Gabor's expansion of a signal into a discrete set of shifted and modulated versions of an elementary signal is introduced and its relation to sampling of the sliding-window spectrum is shown. It is shown how Gabor's expansion coefficients can be found as samples of the sliding-window spectrum, where - at least in the case of critical sampling - the window function is related to the elementary signal in such a way that the set of shifted and modulated elementary signals is bi-orthonormal to the corresponding set of window functions. The Zak transform is introduced and its intimate relationship to Gabor's signal expansion is demonstrated. It is shown how the Zak transform can be helpful in determining the window function that corresponds to a given elementary signal and how it can be used to find Gabor's expansion coefficients. The continuous-time as well as the discrete-time case are considered, and, by sampling the continuous frequency variable that still occurs in the discrete-time case, the discrete Zak transform and the discrete Gabor transform are introduced. It is shown how the discrete transforms enable us to determine Gabor's expansion coefficients via a fast computer algorithm, analogous to the well-known fast Fourier transform algorithm. Not only Gabor's critical sampling is considered, but also the case of oversampling by a rational factor. An arrangement is described which is able to generate Gabor's expansion coefficients of a rastered, one-dimensional signal by coherent-optical means.

Two-loop modulator for sigma-delta analog to digital converter

Abstract: A two-loop modulator has been designed for a superconductive sigma-delta analog to digital converter. In contrast to semiconductor modulators, which use high-gain amplifiers in the signal feed forward path, the superconductive modulator used digital gain in the signal feedback path. The use of superconductive electronics to precisely feed back a single flux quantum into the second integrator loop and multiple flux quanta into the first integrator loop is a key to this circuit. In simulations of a 40 GHz sampling rate, the modulator obtained a 98 dB signal to noise ratio on the dc-60 MHz band. The modulator tolerated thermal noise well, obtaining 98 dB SNR on the dc-4 MHz band, while sampling at a rate of 4 GHz. The modulator tolerated clock timing jitter better than Nyquist-rate A/D converters, obtaining equivalent performance with 3 times as much rms jitter. Compared to single-loop sigma-delta and oversampled lobe-counting A/D converters, the two-loop modulator can achieve equivalent performance at a significantly reduced sampling and digital filter rate. >

Digital to analog conversion using non-uniform sample rates

Abstract: A method and apparatus for digital to analog conversion using sigma-delta modulation of the temporal spacing between digital samples. The method and apparatus of the present invention provides for sigma-delta modulation of the time base such that errors produced by non-uniform sampling are frequency-shaped to a region (i.e., shifted to higher frequencies) where they can be removed by conventional filtering techniques. In one embodiment, the digital data is interpolated by a fixed ratio and then decimated under control of a sigma-delta modulated frequency selection signal that represents, on average, the data rate of the incoming digital data stream. The frequency signal selection number is modulated using an n-th order m-bit sigma-delta modulator. Data thus emerges from the interpolation/decimation process at the clock rate of the n-th order m-bit sigma-delta modulator. The method and apparatus converts the data rate of the incoming digital data stream to the data rate of the n-th order m-bit sigma-delta modulator.

Image pickup device

Abstract: PURPOSE:To pick up an image by selecting the high resolution mode or the usual mode with an image pickup element having number of picture elements available of image pickup at high resolution. CONSTITUTION:The operation of image pickup is attained in the two image pickup mode by using a switch 9 to select either of synchronizing signals with two different frequencies from synchronizing signal generators 7, 8 in the image pickup sensor 10. The generator 8 generates a synchronizing signal for a video signal to access a sensor for video CCD or the like similarly and is used to read 768X494 picture elements in matching with the video signal. An image signal by the sensor 10 is given to a sample-and-hold circuit and an AGC circuit 13, in which sampling and holding and AGC(automatic gain control) are implemented and the resulting signal is converted into digital data by an A/D converter 14, a chrominance processing section 31 converts the data into luminance and two color difference data and stored in a memory 35 under the control of a memory controller 36. An image signal of the standard video system is outputted through the selection of the image pickup mode, and a high resolution image signal is outputted through the mode selection and stored in the memory.