Abstract: Events that occur in a number of in domain communication channels where each channel is used by a communication service, are detected. A Bayesian Belief Network (BBN) defines a probabilistic cause-effect relationship between each cause and each effect on a victim channel. The probability of each of a number of possible causes as being a cause of interference in the victim channel is determined, by propagating observations of the interference backwards through the BBN.

Abstract: A method and apparatus for identification of interference sources are disclosed.

Abstract: The specification discloses algorithms for error recovery in pulse position modulation-based mud pulse telemetry. More particularly, the specification discloses detection and attempted correction of at least five possible error mechanisms: the missed detection of a pulse that creates an interval greater than maximum; the missed detection of a pulse that results in an interval still within acceptable boundaries; detection of an extra pulse; a pulse shift that results in data in contiguous intervals being affected; and a pulse shift resulting in a single interval data corruption.

Abstract: Interference reduction with a current-mode transversal filter having taps including binary current sources is provided. Each binary current source provides an output current having either of two distinct values, depending on a binary input. The product of a tap coefficient and an interference data signal value is obtained by independently generating the contribution from each interference data bit using a binary current source and providing these contributions to a current summing junction. Binary current sources can be implemented in analog, digital, or mixed-mode circuitry. Echo, near end cross talk, and far end cross talk are examples of interference that can be reduced in this manner. The use of binary current sources provides significant flexibility, especially in connection with multilevel modulation schemes such as pulse amplitude modulation (PAM).

Abstract: A process transmits digital data to an A/D converter via an analog channel. The process includes generating a sequence of output signals by precoding a sequence of input signal points to precompensate for ISI in the analog channel. In the precoding is matched to the ISI, at least one of the output signals would be distorted by the ISI to a signal representative of a signal point lying between quantization levels of the A/D converter. If the precoding is matched to the ISI, others of the output signals would be distorted by the ISI to signals representative of signal points that are quantization levels of the A/D converter. A majority of the input signal points are quantization levels of the A/D converter.

Abstract: A power calculating and amplitude limitation judging portion 13 calculates the power value x of the digital quadrature baseband signal I, Q from a transmission data generator 1, and compares the power value x with a power threshold value y set by a threshold value setting portion 14 to judge whether amplitude limitation is needed or not. An amplitude maximum limiting portion 12 subjects the quadrature baseband signal from the transmission data generator 1 to the amplitude maximum value limitation based on the judgment result in the power calculating and amplitude limitation judging portion 13. Thereafter, the digital quadrature baseband signal which has been subjected to the amplitude maximum value limitation by the amplitude maximum value limiting portion 12 is subjected to the distortion compensation using complex multiplication based on the distortion compensation data by a non-linear distortion compensation calculator 2.

Abstract: Provided is a device for improving voice signal in quality by discriminating a radio environment of a signal transmission path and detecting a click noise. The device for improving voice signal in quality calculates an average value for a short interval of a received ADPCM code, identifies from such average value the nature of a radio environment of a signal transmission path, and considers that a click noise is occurred when the average value exceeds a threshold value assigned radio environment. The present invention comprises a click noise detector (22) for outputting a click noise detecting signal, and a click noise suppressing circuit for suppressing the click noise by setting the velocity coefficient of an ADPCM decoder (21) as “0” for a frame in which a frame error detecting signal, which represents the detection of a frame error of the received ADPCM code, and a click noise detecting signal are both detected.

Abstract: A digital signal processor receives 1-bit audio signals acquired by sigma modulation and sent over a predetermined transmission line and makes a changeover between the audio signal and a mute pattern includes an IEEE 1394 interface. The IEEE 1394 interface receives an isochronous packet sent over an IEEE 1394-formatted transmission line, extracts music data from the packet, supplies the music data to a DRAM, monitors the amount of data stored in the DRAM, reads a 1-bit digital signal or generates a mute pattern such as a 9-6 pattern, and makes a changeover between the 1-bit digital signal read from the DRAM and mute pattern while fading them. The digital signal processor further includes a D-A converter to convert the 1-bit digital signal supplied from the IEEE 1394 interface into an analog signal.

Abstract: An impulse detector which can detect both low and high levels of impulse noise in a CDMA system is comprised of circuitry to calculate the background noise level in unused codes. Another circuit calculates the average noise power in the unused codes of each spreading interval to output the noise power per spreading interval. This average is continuously averaged over spreading intervals by another circuit which outputs the average background noise power. A comparator compares the noise power in the current spreading interval with the background noise power plus a programmable threshold and generates an erasure indication if the background noise power plus a discrimination threshold is exceeded.

Abstract: The invention relates to a modem and a method for providing bit loading computation in a multi-carrier communication system, using a plurality of simultaneous different carriers or tones by measuring the signal to noise ratios (SNRs) for carriers and by determining in dependence of the SNR a related number of bits to be allocated to the respective carrier. The method comprises the steps of providing a sliding window having a frequency width able to contain a predetermined number of the carriers, moving the sliding window over the frequency spectrum of the carriers through a plurality of positions of the window with respect to the carrier frequencies, and when the window contains the predetermined number of the carriers and there is at least one further carrier to be positioned in the window as a new carrier for bit allocation, the window is moved such that on entering a new carrier into the window at one of its borders, a carrier already in the window at the other frequency border leaves the window.

Abstract: A network interface is presented that receives packet data from a shared medium and accomplishes the signal processing required to convert the data packet to host computer formatted data separately from receiving the data packet. The network interface receives the data packet, converts the analog signal to a digitized signal, and stores the resulting sample packet in a storage queue. An off-line processor, which may be the host computer itself, performs the signal processing required to interpret the sample packet. In transmission, the off-line process converts host-formatted data to a digitized version of a transmission data packet and stores that in a transmission queue. A transmitter converts the transmission data packet format and transmits the data to the shared medium.

Abstract: The response of a receiver is calibrated using frequency-shifted stimulus signals. A source provides a stimulus signal that has a non-zero bandwidth and an adjustable spectral position. A signal path coupled between the source and the receiver introduces distortion to the stimulus signal. The receiver acquires a first digital representation of the stimulus signal at an output of the signal path with the stimulus signal adjusted to a first spectral position and acquires a second digital representation of the stimulus signal at the output of the signal path with the stimulus signal adjusted to a second spectral position that is shifted from the first spectral position by a predetermined frequency offset. A processor, designates the distortion introduced to the stimulus signal by the signal path to be equivalent at the first spectral position and the second spectral position.

Abstract: The present invention includes a method and system for compensating for cross-talk interference in communication systems. The method includes estimation of the interfering signals. Further, the method includes performing a compensation operation on at least one interfering signal.

Abstract: The present invention provides a computationally efficient technique for compression encoding of an audio signal, and further provides a technique to enhance the sound quality of the encoded audio signal. This is accomplished by including more accurate attack detection and a computationally efficient quantization technique. The improved audio coder converts the input audio signal to a digital audio signal. The audio coder then divides the digital audio signal into larger frames having a long-block frame length and partitions each of the frames into multiple short-blocks. The audio coder then computes short-block audio signal characteristics for each of the partitioned short-blocks based on changes in the input audio signal.

Abstract: A process for noise reduction during the transmission of acoustic useful signals includes the following steps of: (a) determining when a speech pause is present; (b) branching the incoming TC signal from the main signal path and utilizing a Fourier transformation to generate a frequency spectrum; (c) storing in a buffer memory (3) the last frequency spectrum recorded during the last speech pause; (d) using an inverse Fourier transformation on the respective last recorded frequency spectrum to generate a simulated noise signal; (e) subtracting the simulated noise signal in the time domain from the current incoming TC signal. As a result, the original signal is maintained uncorrupted up to the actual noise subtraction. With a simple arrangement and less computing effort than before, the process enables an overall acoustic impression to be produced, which is as agreeable as possible to the human ear and which can be matched to individual requirements.

Abstract: A startup protocol is provided for use in a communications system having a communications line with a master transceiver at a first end and a slave transceiver at a second end, each transceiver having a noise reduction system, a timing recovery system and at least one equalizer all converging at startup of the system. The operation of the startup protocol is partitioned into stages. The first stage includes the step of converging the equalizer and the timing recovery system of the slave while converging the noise reduction system of the master. Upon completion of the first stage the protocol enters a second stage which includes the step of converging the equalizer and the timing recovery system of the master, converging the noise reduction system of the slave, freezing the timing recovery system of the slave, and resetting the noise reduction system of the master.

Abstract: In a radio reception apparatus, a judgement unit judges inter-transmission users. A controller specifies the number of inter-transmission users in accordance with the judgement result. The controller calculates a power value of a received signal per one person of inter-transmission user from both of a power value of the received signal and the number of inter-transmission users. The controller estimates a power value in the case where all the members of inter-communication users come into inter-transmission users from both of the calculated power value and the number of inter-communication users.

Abstract: A high-speed analog subscriber modem operates at speeds as high as 64 kbps in both the downlink and uplink directions using a standard POTS line augmented with an enhanced codec. This enables increased upload speeds and supports 56 kbps peer-to-peer analog subscriber connections. An enhanced network codec according to the present invention supports on a POTS line both high-speed modem communications and standard PCM speech communication. The enhanced codec of the present invention may be designed to be plug-compatible with existing codecs and may be used to upgrade POTS service to support data rates approaching 64 kbps. Similarly, the present invention may be used to increase the use of the POTS channel in a DSL system.

Abstract: A method and apparatus for receiving data over a dispersive media is disclosed. The received signal is composed of an unknown data segment preceded and followed by known data segments. A replica of each known data segment is generated by the communication apparatus. Channel characteristics existing at the time of transmission of the known data segments are estimated by comparing the known data segments with the replica. Using estimations of the channel characteristics, symbols of the unknown data segment can be determined. Decisions can be made on the various unknown symbols as they are analyzed. Channel characteristics can be re-estimated after determining each symbol, each pair of symbols or each group of symbols. Two solutions of the unknown symbols can be combined in various ways to arrive at a final determination of the unknown transmitted symbols.

Abstract: Tones that are generated by a telephone or PC (such as DTMF tones) and subject to loss or noise during generation or during transmission along a medium are detected and replaced with substantially noise-free and distortionless digital signals. In one embodiment, the replacement of the tones is done in a modem embodied in an Internet telephony Gateway/Terminal, such as in a network access server coupling a time division multiplexed telephone line to a packet-switched network. The replacement of the tones may also be performed in any suitable device that provides an interface between a time division multiplexed transmission medium and a packet switched data network, such as in the modems of a cellular telephone network to Internet network access server.

Abstract: Methods, systems and computer program products for averaging measured levels from differing frame intervals based on a digital impairment pattern associated with the frame intervals are provided. Measured levels from frame intervals with the same digital impairment pattern may be averaged together and the measured levels replaced with the average. By averaging frame intervals based on a digital impairment pattern, the present invention may reduce the impact of noise by increasing the signal to noise ratio of the levels utilized for constellation generation. The use of the digital impairment patterns for determining which frame intervals to average may reduce the likelihood that digital impairments will corrupt the average by, for example, collapsing measured levels from one code point onto another code point.

Abstract: A method and apparatus for detecting typing errors of critical fixed series of characters (digits) by using at least one redundant parity digit. The invention detects the most common typing errors that involve both permutations and incorrect striking of adjacent characters during data entry via a keyboard of an electronic apparatus. The method provides warning to the data input operator when incorrect data entry occurs and optionally automatically suspends entry thereof.

Abstract: A method of and system for detecting an interfering multi-path condition is provided. A parameter is determined from a pulse of a correlation function derived from a received signal. The parameter is examined to determine if it is inconsistent with a non-interfering multi-path hypothesis. If so, an interfering multi-path condition is detected.

Abstract: An adaptive DMT modem determines, as part of a startup sequence, subscriber loop conditions. One or more loop condition adaptation operations are then performed to reduce the complexity of the filters employed in the modem, to reduce the power dissipated in the modem, to optimize performance of the modem or to reduce the likelihood of EMI being radiated into the premises of a subscriber.

Abstract: The present invention provides a digital pre-distortion filter in arrangement with a data access arrangement (DAA) on the component side (e.g., in a modem chipset). This arrangement of the pre-distortion filter outside of the DAA allows digital processes such as digital emulation of the central office impedance to remain unaffected by the pre-distortion in the transmitted signal, allowing the dynamic range of the transmitted signal to be flattened to minimize return loss without complicating the transfer function of the digital emulation of the central office complex load. In the case of a digital emulation filter, placement of a digital pre-distortion filter outside of an analog-to-digital (A/D) digital-to-analog (D/A) loop also minimizes the noise otherwise associated with the use of a pre-distortion filter. Thus, benefits of a pre-distortion filter can be gained without interfering with emulation of impedance, and without causing a significant amount of noise in the transmitted signal.

Abstract: A noise reduction system reduces noise associated with an input signal provided to a control system, while substantially minimizing the adverse affect on the responsiveness and stability of the control system. The noise reduction system includes a processor, a memory subsystem and processor executable code. The processor executable code causes the processor to perform a number of steps. Initially, the processor determines an input signal level of an input signal at an input of a noise reduction system. Next, the processor determines an output signal level of an output signal at an output of the noise reduction system. The processor then determines a magnitude of a difference signal which is the difference between the input signal level and the output signal level. When the magnitude of the difference signal is less than a predetermined noise limit, the input signal is provided to an input of a control system.

Abstract: To reduce the distortion of digital data due to on/off switching of the ringing of a Plain Old Telephone Service (POTS) signal (RINGING1, RINGING2) transmitted simultaneously with the digital data over a single line (LINE), the telephone ringing signal generator (POTS-RING GENERATOR 1, POTS RING GENERATOR 2) produces a digital data friendly ringing signal (RINGING1, RINGING2). The telephone ringing signal generator (POTS-RING GENERATOR 1, POTS RING GENERATOR 2) thereto applies a transient smoothening window (WINDOW1, WINDOW2) to a sine or cosine shaped signal (SINE1, SINE2).

Abstract: A variable bit rate coder, and an associated method, for encoding a frame of speech, such as frames of data generated during operation of a communication station operable in a cellular communication system. Selection of the coding rate is made responsive to indicia of actual coding performance of a coder at more than one coding rate.

Abstract: An asynchronous digital system, an asynchronous data path circuit, an asynchronous digital signal processing circuit and an asynchronous 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 (Dl) model or a quasi delay insensitive (QDI) model, while forming each block by applying thereto a scalable delay insensitive (SDI) model.

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: A system to analyze and compensate for noise on a transmission line (35) is provided. The system comprises a transmission card which includes a transceiver (32), and a microprocessor (34) attached to the transceiver (32). The system also comprises a transmission line (35) coupled to the transceiver (32) and at least one customer premise equipment (38) coupled to the transceiver (32) by the transmission line (35). The microprocessor (34) is operable to monitor the transmission line (32) and determine the signal-to-noise ratio on the line as a function of frequency, and adjust the rate of transmission in terms of bandwidth in terms of bits as a function of frequency.

Abstract: Method processor and processor-readable medium for reducing the level of feedback-induced noise in the output signal from an audio automatic gain control circuit are provided. The method allows for receiving a stream of pulses comprising an input signal to the automatic gain control circuit. The method further allows for receiving estimates of respective high .and low frequency energy components of the input signal. The respective high and low frequency components are averaged, e.g., over a respective sliding window. An energy scalar is calculated based on the ratio of a predefined target energy level over a combined value of the high and low frequency components. A relating action allows to compare the target energy level to the combined value of the high and low frequency components. Based on the comparison results, the calculated energy scalar is limited to within the two limit values.

Abstract: A system for detecting and correcting impulse noise present on an input data signal includes an impulse detector module receiving an input data signal and producing as output an correction enable signal indicating when an impulse correction is required. An impulse corrector module receives the input data signal and a correction enable signal and produces a corrected data signal, e.g., having the impulse canceled or blanked, as output. A reliability estimator and selector module receives the corrected data signal and the input data signal and selects as output the input signal which is more reliable. In one embodiment, the impulse detector includes first and second complementary impulse detectors, the outputs of which are analyzed by an enable and correction module to produce an impulse detection signal with improved accuracy. Preferably, the enable and correction module also indicates the most appropriate type of impulse correction in accordance with the detection signals from the complementary detectors.

Abstract: Disturbances in bipolar signals transmitted on a transmission line are suppressed by adjusting the amplitude of the signals with coarse adjustment steps and with fine adjustment steps. The amplitude adjusted signals are compared to at least a lower reference level and an upper reference level for determining a first percentage of the amplitude adjusted signals violating the lower reference level and a second percentage of the amplitude adjusted signals violating the upper reference level. The coarse adjustment steps and the fine adjustment steps are selected in accordance with an adjustment characteristic which evaluates at least the first percentage and the second percentage. The adjustment characteristic is changed if a tendency of reference level violation of the amplitude adjusted signals is detected and a check is performed if the step of changing the adjustment characteristic improves the disturbance suppression. A device for suppressing disturbances is also provided.

Abstract: The present invention relates to burst mode digital communication systems where data transmission is preceded by a preamble for acquisition of carrier and clock synchronization using maximum likelihood (ML) principle. The preamble is sampled and the set of samples obtained is processed using the optimization algorithms of the present invention to provide the ML timing estimate. The optimization algorithms in the present invention consist of three parts. The first part deals with isolating the desired optimal estimate from non-optimal extremes satisfying the same necessary condition, and provides the initial conditions for activating the binary search schemes. The second part performs the binary search for the optimal timing estimate that guarantees the convergence. On the basis of the obtained iterative sequence, the last part constructs a more rapidly convergent sequence to obtain the ML symbol timing estimate.

Abstract: A digital communication transmitter (30) implements a phase constellation (40) which defines a digital communication signal stream that is subsequently compressed (50), thereby introducing distortion into a communication signal (56) transmitted to a complementary receiver (30). The receiver (30) includes a magnitude adjuster (80) which increases the magnitude component of selected phase estimates to at least partially compensate for the compression distortion. The receiver also includes a branch metrics generator (90) having a segment (138) in which branch metric transfer function peaks and valleys are not positioned in receiver phase space to coincide with ideal phase points. The branch metric transfer functions are generated in accordance with a process (102) which bases branch metric calculations upon empirically determined probabilities that characterize system-induced distortions.

Abstract: For purposes of noise suppression, spectral subtraction filtering is performed in sample-wise fashion in the time domain using a time-domain representation of a spectral subtraction gain function computed in block-wise fashion in the frequency domain. By continuously performing time-domain filtering on a sample by sample basis, the disclosed methods and apparatus avoid block-processing delays associated with frequency-domain based spectral subtraction systems. Consequently, the disclosed methods and apparatus are particularly well suited for applications requiring very short processing delays. Moreover, since the spectral subtraction gain function is computed in a block-wise fashion in the frequency domain, high quality performance in terms of reduced tonal artifacts and low signal distortion is retained.

Abstract: Amplitudes of signals produced by a digital modulator can be clipped so that instead of processing the output vector of the digital modulator or it's component vectors, processing is applied to the base band I and Q signals prior to modulation. The I- and Q signals are applied to an input of a clipping circuit. Predetermined limit values are also applied to another input of the clipping circuit. The clipping circuit rotates on the I-Q vector towards the limit vector. At the same time the limit vector is rotated towards the I-Q vector. Rotations are performed by using Cordic algorithm. After rotations have been completed, the rotated I-Q vector is aligned with the original limit vector, and the rotated limit is aligned with the original I-Q vector. Then the length of the rotated I-Q vector will be compared with the original limit vector. If the rotated I-Q vector is shorter than the limit vector, then the original I-Q signals are output signals from the clipping circuit.

Abstract: A method and apparatus for improving an S/N ratio in a digital-to-analog conversion of a PDM signal are provided. A digital-to-analog conversion system comprises an S/N ratio improving section. The S/N ratio improving section has a signal component extractor which extracts a signal component included in the PDM signal, and outputs a digitally filtered output signal. The digitally filtered output signal has a second full scale smaller than a first full scale of the PDM signal. The S/N ratio improving section also comprises a full-scale matching unit which matches the second full scale of the digitally filtered output signal with a third full scale of digital-to-analog conversion.

Abstract: A method of excising one or more narrow-band interfering signals in a direct sequence spread spectrum receiver that performs a magnitude spectral analysis on baseband signals in a detection channel to determine which frequency bins in the spectral analysis output contain the interfering narrow-band signals by comparing the magnitude of the signals in the frequency bins to a threshold. The corrupted frequency bins are determined by the signal magnitudes exceeding the threshold and band reject filters excise the narrow-band signals in the signal channel in accordance with the corrupted frequency bins.

Abstract: A digital preemphasizer that provides a means for high fidelity reconstruction, i.e. low, acceptable bit error-rate, of an original pulse code modulation (PCM) serial stream binary data signal which has suffered degradation of fidelity, and consequent increase in bit-error-rate, during transmission on a single channel from a transmitter to a receiver, connected to each other by a transmission line exhibiting both resistive and frequency dependent loss. This is accomplished prior to transmission by amplitude encoded digital pre-emphasis of each bit of the original binary data signal to be transmitted in such manner as to mitigate or remedy the signal degrading frequency dependent losses concomitant with the signal transfer network characteristics of the fixed transmission line. Subsequent to this pre-emphasis process, the amplitude encoded signal is transmitted to the receiver connected to the other end of the fixed transmission line.

Abstract: Provided is a device for improving voice signal in quality by discriminating a radio environment of a signal transmission path and detecting a click noise. The device for improving voice signal in quality calculates an average value for a short interval of a received ADPCM code, identifies from such average value the nature of a radio environment of a signal transmission path, and considers that a click noise is occurred when the average value exceeds a threshold value assigned radio environment. The present invention comprises a click noise detector (22) for outputting a click noise detecting signal, and a click noise suppressing circuit for suppressing the click noise by setting the velocity coefficient of an ADPCM decoder (21) as “0” for a frame in which a frame error detecting signal, which represents the detection of a frame error of the received ADPCM code, and a click noise detecting signal are both detected.

Abstract: A system and method for reducing distortion in a pulse width modulated digital system by selectively implementing linearization. The linearization is implemented computationally efficient, using no division operations. A frequency detector is implemented which causes signals to bypass the linearization algorithm at frequencies where the new modes of distortion become significant as a result of the linearization when used in combination with an interpolation filter.

Abstract: Improved data communication for modems having digital identification information capability.

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: A broadband communication system (60), such as a cable television system, includes a forward path for transmitting information from headend equipment to subscribers and a reverse path for transmitting information from the subscribers to the headend equipment. The reverse path includes a transmitting station (65) that receives a first analog signal, converts the first analog signal to a first digital signal having a first number of bits, encodes the first digital signal to generate a second digital signal having a second number of bits less than the first number of bits, then optically transmits the second digital signal. A receiving station (70), also located in the reverse path, receives the second digital signal, decodes the second digital signal to generate a third digital signal having a third number of bits greater than the second number of bits, then converts the third digital signal to a second analog signal that approximates the first analog signal.

Abstract: A control system comprising an electrical circuit that produces distortion where the electrical circuit has a frequency band of operation. A pilot modulated carrier signal is frequency hopped about the frequency band of operation. The hopping pilot modulated carrier signal does not interfere with any signal applied to or generated by the electrical circuit as it is not at any time located within the frequency band of operation. Information obtained from the hopping pilot modulated carrier is provided to the control system which uses such information to cancel the distortion produced by the electrical circuit.

Abstract: A radio transmitter has a quadrature modulating portion and an RF power amplifier. The RF power amplifier follows the quadrature modulating portion. A nonlinearity-caused distortion compensating system provided in the radio transmitter includes a power calculating portion for calculating a value related to a power of a transmission signal from first baseband signals quadrature with respect to each other. First distortion compensating coefficients are determined in response to the calculated power-related value. The first distortion compensating coefficients are designed to compensate for a signal distortion caused by a nonlinearity of the quadrature modulating portion. Second distortion compensating coefficients are determined in response to the calculated power-related value. The second distortion compensating coefficients are designed to compensate for a signal distortion caused by a nonlinearity of the RF power amplifier.

Abstract: An impulse noise detector detects impulse noise in an audio signal by filtering out the audio components of the signal, then comparing an envelope of a filtered signal with a threshold obtained as a moving average of the envelope. Short-duration impulse noise is thereby detected without detecting tone bursts and other fast-rising signals of longer duration. The envelope may be taken directly from the filtered signal, or from a difference signal obtained from a preliminary envelope of the filtered signal, to emphasize impulse noise when distorted audio components leak into the filtered signal.

Abstract: A frequency mixing system provides an expanded dynamic range when compared to the dynamic range(s) of an individual mixer(s) that make up the arrangement. The frequency mixing system uses a feed-forward arrangement to reduce the distortion emanating from a single mixer due to a signal power level which would result in a frequency converted signal outside the dynamic range of the mixer. For example, the frequency mixing system splits an input signal onto a first path and a second path. On the first path, a first mixer frequency mixes the signal to produce a frequency converted signal with distortion, such as intermodulation distortion. On the second path, the amplitude of the signal is attenuated then frequency mixed by a second mixer to produce a frequency converted signal with a low and/or insignificant level of distortion.

Abstract: A coil arrangement with three windings arranged coaxially, where the outer coils are connected in series. The inductances of the outer coils are of equal magnitude and their magnetic fields have opposing polarities.The center winding induces currents in the series circuit of the outer windings which cancel each other out. The outer windings with their stray inductances form a coil which is separated from the center one and can be applied for other purposes.