Showing papers on "Digital signal published in 1992"

Advanced Digital Signal Processing

Abstract: Outline:

Digital communication system using superposed transmission of high speed and low speed digital signals

Abstract: A digital communication system using superposed transmission of high and low speed digital signals capable of transmitting superposed high and low speed digital signals through an identical frequency band efficiently by increasing a simultaneously transmittable number of channels in the low speed digital signals, while achieving the practically reasonable bit error rate performances for both the high speed digital signals and the low speed digital signals. In this system, the low speed digital signals are transmitted in a form of spread spectrum signals and the high speed digital signals are cancelled out from the superposed transmission signals in obtaining the output low speed digital signals by using the phase inverted replica of the high speed digital signals to be combined with the superposed transmission signals, so that it becomes possible to increase a number of channels for the low speed digital signals, while achieving the practically reasonable bit error rate performances for both the high speed digital signals and the low speed digital signals.

Signal processing in the context of chaotic signals

Abstract: Signals generated by chaotic systems represent a potentially rich class of signals both for detecting and characterizing physical phenomena and in synthesizing new classes of signals for communications, remote sensing, and a variety of other signal processing applications. Since classical techniques for signal analysis do not exploit the particular structure of chaotic signals there is both a significant challenge and an opportunity in exploring new classes of algorithms matched to chaotic signals. The authors outline a variety of signal processing issues associated with the analysis and synthesis of chaotic signals. In addition two examples are described in detail, illustrating some possible ways in which the characteristics of chaotic signals and systems can be exploited. One example is a binary signaling scheme using chaotic signals. The second example is the use of synchronized chaotic systems for signal masking and recovery. >

Digital filtering: background and tutorial for psychophysiologists.

Abstract: Digital filtering offers more to psychophysiologists than is commonly appreciated. An introduction is offered here to foster the explicit design and use of digital filters. Because of considerable confusion in the literature about terminology important to both analog and digital filtering, basic concepts are reviewed and clarified. Because some time series concepts are fundamental to digital filtering, these are also presented. Examples of filters commonly used in psychophysiology are given, and procedures are presented for the design and use of one type of digital filter. Properties of some types of digital filters are described, and the relative advantages of simple analog and digital filters are discussed.

Automatic detection of seizures using electroencephalographic signals

Abstract: A patient monitoring system and a method of operating same, including automatically detecting a seizure in the patient, by detecting an electrical discharge in the patient's brain; converting the detected electrical discharge into a digital signal; inputting the digital signal into a microprocessor; detecting a seizure by dividing the digital signals into time segments, preprocessing each time segment including standardizing the signal, reducing the signal in each time segment to a feature, the feature providing information about whether a seizure is occurring using the feature from each time segment; and indicating that a seizure is occurring.

System for simultaneous analog and digital communications over an analog channel

Abstract: At a transmitting end, frequencies used to a construct a digital signal are substantially removed from an analog signal by a notch filter circuit to produce an interim signal which is then combined with the digital signal as by combining to produce a composite analog signal that is transmitted to a receiving end. At the receiving end the frequencies used to construct the digital signal are substantially removed from the composite analog signal by a notch filter circuit. In this way the digital signal can be transmitted simultaneously with the analog signal without errors that could be introduced by the analog signal, and with only a slight change to the frequency spectrum of the analog signal. This can be used in telephone communications for sending caller ID digital data over the same line carrying a voice signal. Caller ID data is sent FSK encoded and only those frequency used for the mark and the space are attenuated in the received voice signal before it reaches the telephone's speech circuit. For PSK encoded data method, a single notch filter removes the carrier frequency from the voice signal at the receiving end. In this way, the user is not subjected to the audible frequencies used to transmit the caller ID data, but the overall fidelity of the voice signal is only insignificantly reduced.

Adaptive power control for a spread spectrum communications system and method

Abstract: An apparatus for adaptive-power control of a spread-spectrum transmitter of a mobile station operating in a cellular-communications network using spread-spectrum modulation. A base station transmits a first spread-spectrum signal. A mobile station has an automatic-gain-control circuit for generating an AGC-output signal, from a received signal. The received signal includes the first spread-spectrum signal and an interfering signal. The mobile station also has a correlator for despreading the AGC-output signal, a power-measurement circuit responsive to processing the received signal with the despread AGC-output signal for generating a received-power level, a comparator coupled to the power-measurement circuit for generating a comparison signal by comparing the received-power level to a threshold level, a transmitter for transmitting a second spread-spectrum signal, an antenna, and a variable-gain device responsive to the comparison signal for adjusting a transmitter-power level of the second spread-spectrum signal.

Digital data converter

Abstract: A decoder for recovering an analog signal from its digital representation produced by an encoder employing adaptive-delta-modulation first decodes audio bit-stream (17) in the digital domain with digital delta demodulation (15), to a pulse-code-modulation (PCM) format digital signal. Post-processing (35), complementary to encoder processing is performed digitally on the PCM signal. The output from post-processor (35) is up-sampled, and using delta modulation, a noise shaped single-bit highly oversampled output is produced by converter (14). Applying the single-bit output to lowpass filter (11) re-constructs the decoded analog signal, which is audio output (12). By fully exploiting digital signal processing techniques, the decoder can be manufactured at low cost and exceed the performance of existing analog implementations.

Device for the coherent demodulation of time-frequency interlaced digital data, with estimation of the frequency response of the transmission channel and threshold, and corresponding transmitter

Abstract: A coherent demodulation device, for the demodulation of a digital signal of the type constituted by digital elements distributed in the time-frequency space and transmitted in the form of symbols constituted by a multiplex of N orthogonal carrier frequencies modulated by a set of the digital elements and broadcast simultaneously, the digital signal comprising reference elements, having a value and a position, in the time-frequency space, that are known to the demodulation device, comprising means for the estimation, by Fourier transform, of the frequency response of the transmission channel at any instant, carrying out the transformation of the received samples, corresponding to reference elements, from the frequency domain to the temporal domain, the multiplication in the temporal domain of the transformed samples by a rectangular temporal window (f n) and the reverse transformation, after the multiplication, of the obtained samples from the temporal domain into the frequency domain, the estimation means comprising means for the thresholding of the samples in the temporal domain, providing for the systematic elimination of the samples below a certain threshold.

Integrated opto-chemical sensors

Abstract: Integrated opto-chemical sensors have promising prospects, for example by having the potential to be realized as very sensitive small monolithic smart multisensor systems with a digital signal output. Here the main accent will be laid on the optical principles underlying chemo-optical waveguiding sensors, focusing on linear evanescent field sensors. Sensing principles and systems based on interferometry, surface plasmon resonance and luminescence quenching will be treated in more detail. Materials and technologies applied to integrated optic sensors are mentioned.

Bit allocation device for transformed digital audio broadcasting signals with adaptive quantization based on psychoauditive criterion

Abstract: A device for compressing a digital audio signal, includes a device for allocating bits available for the transmission or storage of the signal, controlling means for the adaptive quantization of the signal, in order to enable a major reduction in the bit rate while at the same time preserving the quality of the starting signal to the maximum extent. The device includes means for allocating a specific number of bits for the expression of the coefficients of each frequency band of a transformed digital audio signal, as a function of a piece of auxiliary information corresponding to a description of the spectrum of the signal, said device being informed by means for the prior elimination of spectral components of said transformed signal as a function of a psychoauditive criterion.

System for measuring position by using global positioning system and receiver for global positioning system

Abstract: A GPS position measuring system includes at least one mobile station and a base station associated with the mobile station. The mobile station is provided with an antenna for receiving one or more GPS signals from the corresponding GPS satellites, an amplifying and frequency converting circuit for amplifying the received signal and converting the amplified GPS signal into a signal on an IF band, an analog-to-digital converter for converting the signal on the IF band into a digital signal, and a writing unit for writing the digital signal into a memory card in order to transfer the digital signal to the base station.

Method and apparatus for providing time diversity

Abstract: The invention provides a method and apparatus for transmitting digital signal information to a receiver using a plurality of antennas. The invention involves applying a channel code to a digital signal producing one or more symbols. A plurality of symbol copies is made and each copy is weighted by a distinct time varying function. Each antenna transmits a signal based on one of the weighted symbol copies. Any channel code may be used with the invention, such as a convolutional channel code or block channel code. Weighting provided to symbol copies may involve application of an amplitude gain, phase shift, or both. The present invention may be used in combination with either or both conventional interleavers and constellation mappers.

High level software synthesis for signal processing systems

Abstract: For the design of complex digital signal processing systems, block diagram oriented simulation has become a widely accepted standard Current research is concerned with the coupling of heterogenous simulation engines and the transition from simulation to the implementation of digital signal processing systems Due to the difficulty in mastering complex design spaces high level hardware and software synthesis is becoming increasingly important The authors concentrate on the block diagram oriented software synthesis of digital signal processing systems for programmable processors, such as digital signal processors (DSP) They present the synthesis environment DESCARTES illustrating novel optimization strategies Furthermore they discuss goal directed software synthesis, by which code is interactively or automatically generated, which can be adapted to the application specific needs imposed by constraints on memory space, sampling rate or latency >

Digital video signal resolution converting apparatus using an average of blocks of a training signal

Abstract: A digital video signal converting apparatus for converting a first digital video signal having a first resolution to a second digital video signal having a second resolution higher than the first resolution, comprises; block segmentation circuit for converting the first digital video signal into a block format, memory having a mapping table stored therein and having address terminals to which the first digital video signal in a block format is supplied and output terminals from which the second digital video signal in block format is output, and block separation circuit for converting the second digital video signal in a block format into a digital video signal in a raster scan order, wherein the mapping table in the memory is generated by training utilizing a plurality of images the training step being performed by generating first and second digital video signal corresponding to each of the plurality of images, converting each of the first and second digital video signals into a block format, and selecting the first digital video signal in a block format is an address signal for the mapping table and inputting the second digital video signal in a block format to a memory area corresponding to the address, and generating data of the mapping table from the signal stored in the memory area.

Synchronous digital signal to asynchronous digital signal desynchronizer

Abstract: Improved jitter performance is realized in a desynchronizer for obtaining an asynchronous digital signal, e.g., a DS3 signal, from a received synchronous digital signal, e.g., a SONET STS-1 signal. The improved jitter performance results from the use of a unique dynamic bit leaking arrangement in conjunction with a digital phase locked loop and desynchronizing elastic store. An optimum bit leak interval is obtained by controllably leaking a greater number of shorter interval STS-1 bits than the number of received pointer adjustment bits.Additionally, the affect of random pointer adjustments and the superposition of randomly received pointer adjustments on a periodic sequence of received pointeradjustments is minimized by employing a "static" queue of pointer adjustment bits to be leaked. The queue is dynamically maintained at its "static" count so that there are always bits in the queue to be leaked at the desired optimum bit leak interval even in the presence of randomly received pointer adjustments.

Delta sigma analog-to-digital converter with increased dynamic range

Abstract: A delta sigma analog-to-digital (A/D) converter includes a digitally-controlled multiplying digital-to-analog converter (MDAC) in a feedback configuration. The MDAC is driven by a digital signal obtained from the output (or an intermediate output) of the A/D converter. An incremental feedback quantum to the first stage integrator is a function of the input values that immediately precede it. In the most general implementation, a table look-up permits an arbitrary relation between the input values and feedback quantum size. In another implementation, the A/D converter output (or intermediate output) signal drive the MDAC and the compression curve of the A/D converter bears a square-root relationship to the input analog signal; a linear relationship is restored by squaring the output signal. In a third implementation, the MDAC is driven by a digital signal obtained from the output (or an intermediate output) of the A/D converter together with an added small positive constant number. In this implementation, the compression curve starts out approximately linear and approaches a square-root relationship at the high end of the scale, while a linear relationship is restored by providing the feedback loop, in the digital domain, with the same value of signal as employed in the analog domain.

Coding apparatus for digital signals

Abstract: In a coding apparatus for a digital signal adapted for implementing, every variable length block, floating processing to an input digital signal by using a block floating processing circuit thereafter to orthogonally transform signal components which have undergone such processing by using orthogonal transform circuits (e.g., DCT circuits), the block floating processing circuit is constructed so as to determine the length of a variable length block and a floating coefficient of the block floating processing on the basis of the same index, e.g., a maximum absolute value in that block. Thus, a quantity subject to processing can be reduced. In addition, there may be employed such a configuration to divide, every critical bands, spectrum signals on the frequency base from DCT (Discrete Cosine Transform) circuits to determine, every respective critical bands, allowed noises in which the masking is taken into consideration to compare these allowed noises and a minimum audible curve from a minimum audible curve generator at a comparator. When the minimum audible curve is grater than an allowed noise at that time, this minimum audible curve is considered as an allowed noise to divide the critical band into smaller bands to carry out bit allocation every respective smaller bands, and to rase or set a flag. Thus, an accurate allowed noise level can be provided without increasing auxiliary information.

Method and apparatus for signal processing

Abstract: A digital demodulator for a NICAM 728 system signal has sampling and digitizing means converting the signal into a succession of digital samples. The digital samples pass through an anti-aliasing digital comb filter to a selector (decimator) that selects one sample in three and feeds it to a second digital filter. Digital matched filters for the in-phase (I) and quadrature (Q) components of the signal effect the demodulation. Carrier and symbols I tracking is carried out digitally. A pulse-width modulated automatic gain control signal is produced for controlling the amplitude of the signal applied to the sampling and digitizing means.

Multiple use digital transmitter/transceiver with time multiplexing

Abstract: A multiple use radio frequency (RF) transmitter comprises means (2, 8, 22) for generating a plurality of different digital information signals and means (12) for generating respective digital carrier signals for said information signals, said carrier signals having higher frequencies than their respective information signals. Further included are means (10) for modulating said digital carrier signals with their respective digital information signals and means (24) for accumulating said modulated signals to a composite digital signal. Digital-to-analog converter (DAC) means (26) are common to each of said modulated signals for converting said composite digital signal to a composite signal in analog format. Further provided are means (30) for altering the frequency of said composite signal to a desired radio frequency (RF) band and means (32) for excluding frequencies outside said desired RF band from said composite signal.

Radio phone composable of separate modules

Abstract: The present invention relates to a radio phone composed of separate modules, comprising a basic module (1) and at least one additional module (2 or 3) detachably plugged thereto. The basic module (1) comprises at least the components and functions which are common to phones operating both in analogue and digital mode. The additional module (2 or 3) includes the main part of the electrical circuits required in transmitting and receiving an analogue or a digital signal. The phone operates as a single-mode phone when one of the additional modules (2 or 3) is plugged into the basic module (1), and as a dual-mode phone when both the first additional module (2) and the second additional module (3) have been plugged into the basic module (1).

Apparatus with transient-dependent bit allocation for compressing a digital signal

Abstract: A digital input signal is divided into signal components in plural frequency ranges by band division filters. The signal components thus obtained are divided in time into signal components in blocks in the respective frequency bands, and are also sent to a transient detector, which determines a transient state for each blocks. Changing an allowed noise level calculating circuit in response to the determined transient states, or any similar method, alters the quantizing bit numbers allocated to each block by an adaptive bit allocation circuit. The allocated bit numbers are altered depending upon the transient state of each block to improve the signal-to-noise ratio in blocks including a transient. Quantizing noise pre echo occurring before the transient, a phenomenon that is offensive to the ear, is prevented.

Method and apparatus for deskewing digital data

Abstract: In a computer system, parallel streams of digital data are transmitted from a source to a destination in bursts or packets. At the beginning of each burst all the parallel data signals contain a start bit. Each data signal is received by a deskewing buffer which transmits the data signal through a delay line with multiple taps. At the beginning of each clock cycle the signal value Data(i) at each tap (i) in the delay line is latched. Each resulting latched signal value LData(i) is compared with the latched signal value LData(i+1) for the next tap down the delay line to generate a set of comparison signals C(i). When the start bit of a new burst is received, one of the comparison signals will have a distinct value from all the others, thereby indicating the delay line tap at which the phase of the received data signal is approximately synchronized with the receiver's clock signal. The data stored in the deskewing buffer's latches represents the phase of the received digital signal and is retained until the end of the burst transmission. A multiplexer which outputs a selected one of data signals from the tapped delay line in accordance with the values of the comparison signals. The selected data signal is sampled and latched at each clock cycle, thereby generating a deskewed data signal that is synchronized with both the receiver's clock signal and also with the other parallel data streams.

An adaptive controller for the active absorption of sound

Abstract: An adaptive system is presented that allows a secondary acoustic source to become an active absorber of sound at the end of a closed duct. The system can also be generalized in order to achieve other termination impedances. The system consists of a loudspeaker, two microphones, and signal processing hardware including a digital signal microprocessor. The signals from the microphones are processed to obtain an error signal that represents the difference between the actual and the desired acoustic impedance of the termination. An absorbing termination requires, for example, that the microphone pair acts as a unidirectional probe picking up the sound reflected from the active termination only. This signal is used as the error signal that the digital controller is required to minimize. A simple analysis shows that this can be done adaptively using the ‘‘filtered‐X’’ LMS algorithm. A simple experimental setup is used to obtain an absorbing termination which is shown to work with periodic, random, and transient input signals.

Digital adaptive transversal filter for spread spectrum receivers

Abstract: A digital adaptive transversal filter includes an interface circuit for regulating and digitizing analog input signals, which comprise of multiple spread-spectrum signals, additive thermal noise and additive multiple narrowband interferers, to provide multi-bit digital input signals. A digital finite impulse response filter having a set of variable digital weight coefficients responds to the multi-bit digital input signals to generate digital output signals which contain a reduced amount of narrowband interference. A digital weight generator responds to the digital input and digital output signals for updating the digital weight coefficients, and a reset generator periodically resets the digital weight coefficients to zero initial values. In a preferred embodiment an 8-tap, 8-bit digital adaptive transversal filter operating at 22 MHz was implemented with discrete components and a processing gain of 30 dB was demonstrated.

System of very-long-distance digital transmission by optical fiber with compensation for distortions at reception

Abstract: A system for very-long-distance transmission of a digital signal between a transmitter station and a receiver station, wherein the transmitter and receiver stations are connected by a monomode optical fiber with negative chromatic dispersion at the operating wavelength of the system, having a length of at least one thousand kilometers. The receiver station comprises device to compensate for the distortions due to the non-linear effects and to the chromatic dispersion introduced by the transmission line, the compensation device carrying out a positive chromatic dispersion of the received signal, the amplitude of the positive chromatic dispersion being a function notably of the amplitude of the negative chromatic dispersion induced by the optical fiber as well as of the mean on-line optical power of the signal transmitted on the optical fiber.

Digital data recording/playback system utilizing EEPROM and ROM memories as a storage medium

Abstract: A digital data storage system which does not require the use of moving, mechanical components, and which utilizes semiconductor memory elements. In one embodiment, the digital data storage system includes a ROM, a system control microcomputer, a digital signal processor (DSP), and a D/A converter. In operation, the DSP is responsive to control signals generated by the system control microcomputer for reading out digital data, e.g., digital audio data, stored in the ROM, and decoding the read-out digital data. The D/A converter functions to convert the decoded read-out digital data into an analog output signal, e.g., an analog audio signal, and to supply the analog output signal to an output terminal. The digital data storage system of this embodiment is a playback-only system. In another embodiment, the digital data storage system includes all of the elements of the above-described embodiment, and further includes an A/D converter and an EEPROM, to facilitate the recording of digital data. In operation, during a record mode of operation, the A/D converter functions to convert an input analog signal, e.g., an analog audio signal, into an input digital data signal, and the DSP functions, in response to the control signals, to write the input digital data signal into the EEPROM. The digital data storage device of this embodiment functions as a record/playback system.

Composite direct digital synthesizer

Abstract: A low-power digital frequency synthesizer combining direct digital frequency synthesis techniques with serrodyne frequency translation principles to produce a wideband frequency response with high spectral purity. A conventional direct digital synthesizer is used to generate a high-resolution analog carrier signal from a low-speed digital clock signal. The carrier signal is phase modulated by a low-resolution signal generated from a high-speed digital clock signal. The modularity signal is a higher frequency signal than the carrier signal. The phase modulation is accomplished by exact decoded attenuators. The spectral purity of the resulting high-resolution output signal is unobtainable by conventional direct digital synthesizers, while providing significant power savings.

Anti-piracy video tape system has anti-piracy components added to the luminance signal and the color difference signals

Abstract: An anti-piracy signal generating circuit for generating anti-piracy signal components on a video signal reproduced by a first video unit, the signal components being interleaved to a frequency fH in the band of the video signal in accordance with the video signal components and a signal superimposing circuit for superimposing the anti-piracy signal components intermittenly on the luminance signal, first color difference signal, and second color difference signal.