Abstract: A digital cordless telephone system using lossless pulse code modulation (PCM) for encoding an audio signal. By using an efficiently implemented and uncompressed encoding scheme, the system substantially reduces implementation costs and improves the quality of transmission for white signals such as modem signals. A transceiver implementation includes a PCM coder, an RF transmitter, an RF receiver, and a PCM decoder. Also presented is a method for communicating an audio signal from a transmitter unit to a remote receiver unit. The method includes steps of sampling an audio signal into a PCM data stream, modulating the PCM data stream onto a carrier, transmitting the carrier, receiving the carrier, demodulating the PCM data stream from the carrier, and generating a reconstructed audio signal from the PCM data stream.

Abstract: Nonlinear distortion compensating data is calculated by approximation equations at a compensating coefficient calculation section using an amplitude value of transmission quadrature base band signals obtained by a power calculating section, wherein by carrying out nonlinear distortion compensation at a nonlinear distortion compensating section using the data, it is possible to compensate nonlinear distortion generated in an amplifier of the transmission system without use of any memory table, and a nonlinear distortion compensating section can be obtained, which has a great effect of decrease distortion.

Abstract: An improved technique for processing digital audio signals is provided wherein adaptation of predictor coefficients in an ADPCM environment is caused to converge in a rapid and computationally efficient manner. The technique employs a whitening filter to generate a filtered reconstructed signal which is utilized to update, or adapt, the prediction coefficients of a pole-based predictor.

Abstract: In a line driver that utilizes a digital-to-analog converter (DAC) to generate a current that is used to form an output voltage VOD, variations in the output voltage VOD are minimized by a calibration circuit that senses the output voltage VOD, compares the output voltage VOD to a reference voltage, and then increments or decrements the bias current fed into the DAC to match the output voltage VOD to the reference voltage.

Abstract: A system and method for source coding a signal to localize transmission errors to a set of samples is disclosed. The signal comprises a plurality of signal elements (SEs) with each SE having a plurality of components. The signal is divided into a plurality of data sets with each data set having a set of SEs. Each SE component of a data set is grouped into a plurality of divisions with each SE component having a plurality of bits. The plurality of bits of the SE components are distributed from the plurality of divisions across a generated bitstream. In one embodiment, this is used in the transmission of video signals over a potentially lossy communications channel.

Abstract: A method and apparatus for controlling and shaping the spectrum of transmitted signal samples having predetermined frequency characteristics and allowable transmit signal levels. A running measure of unwanted components is measured and, for each block of the transmitted samples, an objective function based on the previously calculated running measure is computed. Depending upon the measurements, redundancy is determined and selected for each block of the transmitted samples. Accordingly, at least one redundant sample may be added or, the sign of the sample modified, at structured or randomized locations within the block to optimize the objective function.

Abstract: A system and method for shifting the relative phases of a PCM data frame and a network RBS frame by one or more symbols is disclosed. The method includes determining whether or not RBS is present. If RBS is present, the slots affected by RBS are identified. Next it is determined what data slots, if any, are affected by encoding. The PCM data frame is shifted so the RBS constraint and the encoding constraint do not coincide at any particular data slot. For one particular PCM upstream modulation scheme, the digital modem adjusts the relative phases of the data-mode frame and network RBS frame such that a minimum number of trellis-modified symbols fall on network RBS affected slots. This information is transmitted to an analog modem. The amount of shift can be conveyed to an analog modem in the same data sequence used to send upstream constellation sets and mapping parameters during an initial training sequence.

Abstract: A method of selecting a PCM codeword set. A communication system includes an encoder and a decoder. The encoder has a digital connection to the digital portion of the telephone network. The digital portion of the telephone network is connected by an analog loop to the decoder. The method includes transmitting sequence of PCM codewords from the encoder and receiving the sequence as a sequence of analog voltages. A noise level it determined and a minimum separation value is selected in response to the noise level. A set of PCM codewords is selected such that no two PCM codewords within the set are separated by less than minimum separation value.

Abstract: A method for obtaining a reliable estimate of the transmitter clipping error compliant with T1.413 ADSL standard is disclosed. An architecture is disclosed that uses the clipping error estimate at the receiver to reconstruct a frequency-domain compensation signal. The method for computing the compensation signal is disclosed along with an asymmetric digital subscriber line modem supporting T1.413 standard transmission/reception functions over a discrete multi-tone communications system capable of estimating clipping errors and computing clipping compensation signals.

Abstract: A circuit for at least substantially eliminating at least one external interference signal from a radio communications signal received by a code division multiple access (CDMA) mobile phone including a receiving terminal and a low noise amplifier comprises: a plurality of damping devices operatively coupled between the receiving terminal and the low noise amplifier; a plurality of switches respectively operatively coupled between the plurality of damping devices and the receiving terminal; an interference detector operatively coupled to the receiving terminal for detecting the at least one external interference signal from the received radio communications signal; and a switch driver operatively coupled between the plurality of switches and the interference detector, the switch driver generating a switching signal in response to the interference detector to select at least one of the switches in accordance with the at least one detected interference signal in order to vary a gain associated with the received

Abstract: A recording and/or reproducing apparatus for recording and/or reproducing digital audio signal on a recording medium such as a magnetic tape, which includes a recording and/or reproducing unit, an analog-to-digital convertor, a decimation filter, and noise shaper. The recording and/or reproducing unit records and/or reproduces signal on a recording medium at a speed double relative moving speed of a head and the recording medium at least. The analog-to-digital convertor converts input analog audio signal into digital signal with a sampling frequency which is above the audio frequency band, and sufficiently higher than the maximum frequency capable of recording by the recording and/or reproducing unit. The decimation filter reduces the sampling frequency of the digital signal output from the analog-to-digital convertor, and converts it into digital signal of m-bit (m>n).

Abstract: A calibration circuit adjusts a differential output voltage from a line driver circuit when the differential output voltage falls outside a specified tolerance range. The calibration circuit includes a sample and hold circuit which samples the differential output voltage and holds a representative signal. A comparator compares the held signal with a reference voltage signal. When the held signal is greater than the reference voltage signal the comparator outputs a LOW signal and when the held signal is less than the reference voltage signal the comparator outputs a HIGH signal. The comparator output signal is stored in a memory circuit of a control logic. The control logic instructs an up/down counter to increment when the comparator output is LOW and to decrement when the comparator output is HIGH. A calibration current source sinks a unit of calibration current when the comparator output is LOW and sources a unit of calibration current when the comparator output is HIGH.

Abstract: Signal points from a PCM-derived constellation are selected for transmission via a modulation technique which employs different levels of redundancy coding--including the possibility of no redundancy coding--for respective different sub-constellations of the overall PCM-derived constellation. The coding that is employed for at least one of the sub-constellations is carried out independently from any coding that is employed for any of the other sub-constellations. In preferred embodiments, the sub-constellations are non-overlapping portions of the overall PCM-derived constellation, the redundancy codes are trellis codes, and the trellis codes employed in conjunction with sub-constellations having increasingly smaller minimum distance between signal points provide respectively increasing amounts of decibel gain in that minimum distance in order to compensate for that increasingly smaller minimum distance.

Abstract: A codec supports PCM modem communications (e.g., V.90, X2, or K56flex protocols) over a universal digital loop carrier (UDLC) having an analog interface to the central office (CO). The codec includes an analog-to-digital converter (ADC) that is synchronized to the CO clock, and an equalizer that has been trained to provide a transfer function to compensate for the distortion introduced by the CO line interface reconstruction filter and connecting circuits, to minimize resampling error. The features allow the codec to output a PCM stream that accurately reproduces the PCM data at the central office, at data rates sufficient to support PCM modem communications, as well as voice and conventional modem communications.

Abstract: A method of allocating bits to discrete frequencies in a discrete multitone modulator comprises the steps of allocating bits to frequencies on a per frequency basis such that the bits are successively allocated until a maximum power level for that frequency is exceeded. Then, a total power level for bits allocated to a plurality of transmit frequencies is calculated to see if a maximum permissible total power level is exceeded. If the maximum permissible total power level is not exceeded, then the process is complete. If the maximum total power level is exceeded, successive bits are removed until the total power level is no longer exceeded. In one bit removal method, bits are removed in order of the amount of power the bit would consume of the transmit power spectrum. The algorithm may be implemented in the combination of transmitter elements including tone ordering circuitry, gain scaling circuitry and the inverse discrete Fourier transform modulator of the discrete multitone data transmitter.

Abstract: An input data signal (102) in a digital signal processing system gets adaptively quantized by performing a LMS error algorithm on a signal of transformed coefficients (113), determining paths of minimum error (202), and forming a three-dimensional matrix (206) representing these paths. Centroids are computed (208) based on the paths of minimum error (206), and these computed centroids are used as part of a next iteration to represent new quantization states and levels.

Abstract: Enhanced performance is realized in a wireless transmission arrangement that is subject to fading by employing a perspective that combines array signal processing with channel coding. Specifically, in accordance with our invention, antennas at the transmitter are partitioned into small groups, and individual space-time codes are used to transmit information from each group of antennas. At the receiver, an individual space-time code is decoded by a linear array processing technique that suppresses signals transmitted by other groups of antennas by treating them as interference.

Abstract: A digital telephone (60) comprises send equipment (40) and receive equipment (50), coupled via a signal splitter/combiner (61) onto a digital telephone line (62) for communication with a remote party (90). The send equipment (40) comprises an A-law map (12) and a mu-law map (43). Receive equipment (50) comprises an inverse A-law map (14) and an inverse mu-law map (52). Maps (12,43) are selectable by the user (80). Similarly, inverse maps (14,52) are also selectable by the user (80). Receive equipment (50) may further comprise an un-quick-fix module (55) for removing unwanted insertions of quick fixes (25) inserted into the telephone line (62) by international telephone companies. Receive equipment (50) may further comprise a digital signal processor (57), computer (56), a set of pre-stored templates (58) corresponding to possible incoming telephone characteristics, and a caller parameter decipher module (51).

Abstract: A maximum a posteriori (MAP) filter is applied as a loop filter during encode/decode processing to generate filtered data used to encode/decode other data. In one embodiment, the MAP filter is applied iteratively to a decoded frame to generate the resulting filtered frame. When applied as a loop filter, the filtered frame is used to generate reference data used to encode or decode one or more other video frames.

Abstract: A system and method for reducing quantizing noise is presented. In some Internet access and other dial-up services, modems or other sources connect into analog switching systems, and then are A/D-converted to reach remote switching offices. There, the digitized signal is converted back to analog, only to be re-digitized for T1 transmission. In the invention, the sample points used to re-digitize the signal are synchronized with those used to originally sample the signal, nearly eliminating second quantization noise.

Abstract: An asynchronous digital system, an asynchronous data path circuit, an asynchronous digital signal processing circuit and an synchronous digital signal processing method, which enables improved processing speed while maintaining high reliability are provided by dividing the overall chip into blocks with a specified area, forming the connection between the blocks by applying thereto a delay insensitive (DI) model or a quasi delay insensitive (QDI) model, while forming each block by applying thereto a scalable delay insensitive (SDI) model. In the SDI model, the system is configured using circuit components having a delay assumed during design in which if the specification states that a signal transition (b) in a subcircuit 7 precedes a signal transition (c) in a subcircuit 8, k.multidot.

Abstract: An electronic memory system and method for use in a system for reducing quantizing noise is presented. In some Internet access and other dial-up services, modems or other sources connect into analog switching systems, and then are A/D-converted to reach remote switching offices. There, the digitized signal is converted back to analog, only to be re-digitized for T1 transmission. In the invention, the sample points used to re-digitize the signal are encoded using a dedicated electronic memory structure which permits efficient determination of valid data points.

Abstract: A method and device for generating a constellation of Pulse Code Modulation (PCM) levels to represent digital data in a communication system. The constellation of Pulse Code Modulation levels are selected to have a total number of PCM levels and an upper power limit that optimizes noise performance. PCM levels are selected to increase the minimum spacing between PCM levels as great as possible while staying within the upper power limit for a constellation having a desired number of PCM levels. To minimize the power level of the selected constellation, constellation points are selected starting from PCM levels at linear values near zero to PCM levels at higher linear values. To improve noise performance, the number of occurrences of the minimum distance between PCM levels is also reduced without exceeding the upper power level. To reduce occurrences of the minimum distance between constellation points, PCM levels at the minimum distance from the previously selected PCM level may be skipped.

Abstract: The present invention provides a method of generating a signal that may be used to determine the characteristic response of a communication channel that utilizes the public Digital Telephone Network (DTN). The channel includes the DTN, which may have Network Digital Attenuators (NDA) and/or Robbed Bit Signalling (RBS), and a Digital-to-Analog Converter (DAC), (also known as a codec), as well as the analog characteristics of the local loop, typically a twisted pair of copper wires. The present invention provides a method and apparatus to determine the optimal sampling instant of the received data stream. The present invention provides a probing signal that is well-suited for use in determining the channel's response to a known sequence of PCM codes used as data symbols. This is especially useful in so-called PCM modulation schemes that utilize the DTN, where knowledge of network and DAC distortion predicates the selection of available PCM codes used to represent data.

Abstract: The apparatus for encoding a plurality of digital information signals, having at least a first input unit for receiving a first digital information signal and a second input unit for receiving a second digital information signal. A signal combination unit combines the first and second digital information signal to generate a first combination signal. First and second data compression units compress the first and second digital information signal so as to obtain first and second data reduced digital information signals. First, second, and third masked threshold determining units determine first, second and third masked thresholds respectively from the first, second digital information signals and the combination signal respectively. A selection unit selects one masked threshold from the first and third masked threshold so as to obtain a first selected masked threshold. Preferably, the first selected masked threshold is the smallest of the first and third masked thresholds.

Abstract: Disclosed is a system for selectively enabling intermediate data processing of digital signals external to an integrated circuit (IC). The system comprises a transcoder, a codec connected to the transcoder, and a data processor located externally to the IC. In response to a strobe signal, a programmable switch diverts digital signals between the transcoder and the codec to a data path from the IC to the external processor. The digital signals are formatted within the IC for processing by the external processor. The externally processed digital signals are returned to the IC via a data path from the external processor. The digital signals are then reformatted within the IC for further processing by the IC.

Abstract: A transmission system wherein a plurality of digital signals are coded and the coded signals are used to amplitude modulate respective carrier waves. The amplitude modulated coded signals are then combined for conveyance over a non-linear transmission medium. Due to the transmission non-linearity there will be distortion of the received combined signal, thereby reducing the transmission capacity of the system. In order to increase the transmission capacity, coding of the digital signals is carried out in accordance with a code such that the amplitude of the modulated carrier for digital symbols of higher probability is reduced in relation to the amplitude thereof for digital symbols of lower probability of occurrence.

Abstract: Complex coding of speech signals, as this is used, for example, in the GSM half rate speech coder, requires the use of a signal processor. On the other hand, a hands-free facility requires an arithmetic unit which adaptively calculates filter coefficients. This is only effected during speech pauses in which the speech coder is switched to a mode in which it generates codes for background noise with only little processing circuitry and expense. According to the invention, the signal processor for the coding device is therefore used for calculating the coefficients for the echo cancelling filter during speech pauses. This requires the use of not more than a single signal processor which can thus be used twice.

Abstract: An apparatus for estimating channel tap coefficients of a communication channel is provided which calculates channel tap coefficient estimates as a function of a received sample stream corresponding to the channel whose channel taps are being estimated and as a function of at least one other received sample stream corresponding to another receive channel. The apparatus receives digital sample streams from a plurality of receive channels and an impairment correlation matrix containing impairment correlation information among multiple receive channels. In one embodiment, the apparatus includes a precombiner which forms linear combinations of initial error samples from a first received sample stream and a second received sample stream to generate refined error samples for generating channel tap coefficient estimates. In one embodiment, an impairment correlation estimator receives the complex samples and the channel tap coefficients and generates an impairment correlation matrix for multiple receive channels.

Abstract: As SCSI systems increase in size, especially where there are a plurality of physically separate components, cables are often used to connect various components of the system, such as drives, to one another via repeaters and convertors. Unfortunately, as the length of the connecting cable increases, the cable in combination with the repeaters and convertors skew the signals and data passed between the components. A method and system is presented that reclocks the skewed data and signals so as to compensate for the skewing. The method and system accomplishes this task dynamically for Fast and Fast 20 modes of operation of the SCSI bus.

Abstract: A communications system (30) includes a transceiver (42) for transmitting a plurality of bins. Individual bin BERs (bit-error-rates) are iteratively equalized. This is accomplished by applying a fine gains adjustment to the transmit power of the transceiver (42). Specifically, the BER of the bins is equalized by calculating an adjustment gain and subtracting the adjustment gain from the bin currently having the maximum margin and adding the same adjustment gain to the bin currently having the minimum margin. The iterations continue until all of the bin margins are within a predetermined threshold. This changes the individual bin transmit power levels without substantially changing the aggregate transmit power level. The BER is substantially equalized, without causing the individual transmit power levels to exceed a predetermined range, such as that required by ANSI T1E1.4.

Abstract: The signal received in a CDMA system is sampled at a rate greater than the chip rate of the spreading sequences. In order to assign the propagation delays to the arms of a rake receiver, consecutive delays are tested within a time window by evaluating, for each delay tested, an energy of the correlation between the signal received and a reference spreading sequence to which said tested delay is applied. A selection threshold is determined from the largest of the energies evaluated. A first list is compiled containing delays from the window for which the histogram of the evaluated energies exhibits a local maximum greater than the selection threshold, and a second list is compiled containing delays neighboring the delays of the first list. The delays assigned to the arms of the rake receiver are selected initially from the first list and then, if this is insufficient, from the second list.

Abstract: In a fiber optics transmission system in which a plurality of access units are cascade-connected via up-link and down-link optical fiber lines and connected to a base unit, the base unit has a beat noise detector and a noise information generator and, upon detection of beat noise, noise information is sent to each access unit over the optical fiber lines. Each access unit has a random signal generator for generating a random value in response to the noise information and a wavelength control circuit for randomly selecting and controlling the oscillation wavelength of a laser diode, and each access unit repeats random wavelength selection until the noise information is reduced.

Abstract: A power line communications analyzer (PLCA) provides a signal strength metering system and selectable signal attenuation functions for adjusting the transmitting unit's attenuation and measuring error rate without the need for a user to be present at both the sending and receiving locations under test. Each PLCA is coupled to a power line communication network via an electrical outlet and power lines. In actual operation, one of the PLCAs acts as a data transmitter and the other PLCA acts as a data receiver. The mode of operation of the PLCA can be dynamically altered during operation of the system. The PLCA comprises control logic that receives command inputs from a keypad, generates data packets for transmission on the power line communication network, receives and analyzes data packets received from the power line communication network, detects the power line signal and/or noise level and drives the LED display accordingly, and displays packet transmission information on an LCD display.

Abstract: A system and method for transmitting digital data over a wireless channel uses periodically-transmitted, differentially encoded pilot words to allow a receiver to rapidly synchronize with a transmitter. Each differentially encoded pilot word comprises at least two pilot symbols that are separated by a fixed difference. The receiver monitors differences between symbols to locate the differentially encoded pilot words, and to thereby become synchronized with the transmitter. Once the receiver becomes synchronized with the transmitter, the receiver uses the pilot symbols of the differentially encoded pilot words to perform channel estimation and compensation.

Abstract: A new data communication system is described which enables data transmission over existing telephone lines at rates higher than possible with existing methods, including conventional modems. By using a novel asymmetric configuration of the two communication endpoints, the currently-accepted maximum theoretical limits on the data rate are no longer applicable. One endpoint is connected directly to a digital telephone network, whereas the other endpoint uses a conventional telephone connection. This reduces the transmission problem to compensation for a single telephone line interface and a single analog local loop. Device for providing this compensation and the required clock synchronization were also created, enabling a practical implementation of the system. The new system can achieve rates up to 64,000 bits-per-second and has broad utility in several active areas including wide-band audio transmission, video transmission, networking, facsimile transmission, and remote computer access.

Abstract: Reception signal strength of ADPCM voice data in a transmission frame is detected every ADPCM voice data in an RSSI detecting section (2), and when a base band processing section (3) effects decision of existence of error regarding ADPCM voice data of a reception frame, only for the ADPCM voice data with the reception signal strength less than the predetermined value, the difference value is made small or made zero in an ADPCM voice data conversion processing unit (7) and then the value is decoded in an ADPCM decoding processing unit (8). ADPCM data without error are decoded as it is.

Abstract: In a data transmission system having a first and a second data transmission device and a transmission line that is not terminated, a method of minimizing signal reflection along the transmission line is described that includes the step of connecting the first and second data transmission devices to the transmission line such that the distance between the first and second data transmission devices is substantially proportional to a wavelength of a predetermined frequency. Data to be transmitted from the first data transmission device to the second data transmission device via the transmission line are encoded in the first data transmission device such that energy of the encoded data is substantially concentrated around the predetermined frequency to minimize the signal reflection without terminating the transmission line. A data transmission system with minimized signal reflection is also described.

Abstract: An apparatus and method for interpolating a digital signal where an input data signal from a data input terminal 1 is supplied through a memory 2 to an FIR low-pass digital filter 4. The memory 2 is controlled by a control circuit 6 to supply data preceding and following data which have suffered an error to the digital filter 4 to extract values A, and B. The values A, and B are calculated by an interpolation data generating circuit 7 to generate a value Y0. The data preceding and following data which have suffered an error is supplied to the digital filter 4 to extract a value A0. To the value A0, there is added values Bn calculated in advance by supplying all interpolation data which the error portion can take to an arbitrary position of the digital filter 4. The sum is compared with the value Y0. Interpolation data is generated according to a value Bn at which the sum is closest to the value Y0, and interpolation data is delivered from a data selector 3 to a data output terminal 8.

Abstract: A digital mixing to baseband decimation filter includes a mixer section and a low-pass filter section. The mixer section generates in-phase and quadrature signal components at baseband from low-resolution, high-rate digital data lying within a signal passband offset from baseband. The low-pass filter section transforms an incoming data stream to high-resolution, low-rate digital data. The mixer section can precede or follow the filter section.

Abstract: An oversampling technique that improves signal-to-quantization noise ratio without requiring substantial amounts of processing power is disclosed. The oversampling technique determines the oversample values by successively scaling the previous value using a scaling factor. The effect of the invention is the randomize the quantization error for each of the oversample values so that the quantization error becomes uncorrelated from one oversample value to the next. When such decorrelation occurs, the quantization noise is spread over the enlarged frequency band associated with the oversampling. The invention is applicable to any transmission scheme which utilizes oversampling and digital-to-analog conversion.

Abstract: An apparatus is disclosed which is adaptable for indicating communication failures and eliminating an undesired pattern of physical values of a generated signal, the signal having physical values responsive to a sequence of information. The apparatus includes a protocol sequencer, responsive to the sequence of information and to the communication failure indications, and providing RESET and RETRY control signals; a framing flag generator, responsive to the RESET signal and providing a variable length framing flag, the length of the framing flag being dependent on the RETRY signal; a multiplex switch, responsive to the RETRY and RESET signals for generating a concatenation of the framing flag and sequence of information; a whitener generator, responsive to the RESET signal, for providing a known sequence of values; and a multiplier, for multiplying said concatenation of said framing flag and the sequence of information with the known sequence of whitener values.

Abstract: A multiple access communications data and information transport system and methodology for employing the noise present in any conduit as a carrier form using wireless chaotic waveform signals supported by nonlinear noise sustaining structures. The communications system utilizes digital devices and software embodying various algorithms designed to manipulate the nonlinear perturbations of any given conduit to establish a high bandwidth bidirectional, isochronous noise immune communications signal. Additionally, the system can be combined with any of a number of wireless or wireline communication methodologies to further enhance performance in terms of bandwidth, throughput, noise immunity, power consumption, cell size, and number of users without disturbing any concurrent signal traffic.

Abstract: A receiving circuit has an automatic level setting function for a background noise on the receiver side to generate a background noise free from incongruous aural sensation.

Abstract: The noise removing device is used in a data communication system utilizing a transmitting signal modulated by a data signal to be transmitted. The noise removing device includes a receiving section for receiving the transmitting signal, and converting it into an electric signal, so as to supply an input signal, an analog signal processing circuit for selectively passing the input signal in a first frequency range including a carrier frequency of the transmitting signal, thereby removing a first noise, for converting the passed signal into a digital signal, and for outputting the digital signal, and a digital signal processing circuit for sampling the digital signal, and for selectively passing the digital signal in a second frequency range including the carrier frequency based on a result of comparison of a pattern obtained by the sampling with a predetermined pattern, thereby removing a second noise.

Abstract: In a digital FM receiver having a demodulator that receives a radio signal and generates therefrom a phase sample whose behavior determines an output of the demodulator, an apparatus for removing a low-frequency interference signal from the phase sample converts the phase sample into a frequency sample which is then high-pass filtered to produce a filtered frequency sample. The filtered frequency sample is then converted into a filtered phase sample, which may be further demodulated in accordance with known techniques. Converting the phase sample into a frequency sample may be performed by a first order difference circuit. Converting the filtered frequency sample into the filtered phase sample may be performed by an integrator.

Abstract: A received signal is sampled and a mean is calculated. A zero crossing sample is then located and used to locate a starting sample and the samples are read from that point forward, using sampling intervals and mean crossings, to decode the bit stream. Alternatively, first and second derivatives of the signal are calculated. A sign (.+-.) of the first derivative sample at the first zero crossing of the second derivative is used to determine the sign of a sample. The second derivative of the signal is then read, using sampling intervals and the sign of the first derivative, to decode the bit stream. In another alternative, samples are used to calculate a rolling average of the samples. The rolling average of the sample is compared to the sample to determine the logic value of the sample based upon whether the sample is above or below the rolling average.

Abstract: Disclosed is a system for selectively enabling intermediate data processing of digital signals external to an integrated circuit (IC). The system comprises a transcoder, a codec connected to the transcoder, and a data processor located externally to the IC. In response to a strobe signal, a programmable switch diverts digital signals between the transcoder and the codec to a data path from the IC to the external processor. The digital signals are formatted within the IC for processing by the external processor. The externally processed digital signals are returned to the IC via a data path from the external processor. The digital signals are then reformatted within the IC for further processing by the IC.

Abstract: A fiber data distributed interface (FDDI) system uses an MLT3A encoding scheme in order to reduce DC bias or baseline wander. The MLT3A encoding scheme is a scheme wherein logical zeros are transmitted as ground voltages and logical ones are transmitted as one of either positive voltages or negative voltages depending upon past transmission history (FIG. 4 ) wherein the past history is recorded by a counter C or an analog circuit. MLT3A ensures that the baseline wander or DC bias returns to zero or is maintained as close to zero as possible in a timely manner so that no FDDI systems will fail from baseline wander.

Abstract: Described herein is a fourth-order sigma-delta modulator which utilizes two second-order sigma-delta modulators connected together. Each second-order sigma-delta modulator is characterized as including integrators having a 1/2 sample period delay from input to output. A second-order sigma-delta modulator, including such integrators, exhibits a single sample period delay from input to output. A fourth-order sigma-delta modulator, which includes two such second-order sigma-delta modulators, exhibits a delay of two sample periods from input to output. The present sigma-delta modulator can be fabricated using switched capacitor circuitry to form an A/D convertor, and in another embodiment can be used as a digital noise shaper for a D/C convertor circuit. The 1/2 unit delay is implemented without requiring two D-flip flops in series, which results in a design and manufacturing advantage.