Abstract: Channel estimates respectively associated with OFDM pilot symbols are used to estimate additional parameters such as change in channel phase over time, change in channel phase over frequency, and frequency selectivity.

Abstract: Channel estimates respectively associated with OFDM pilot symbols are used to estimate additional parameters such as change in channel phase over time, change in channel phase over frequency, and frequency selectivity.

Abstract: Methods and apparatus for reducing phase jitter in a multi-tone, e.g., OFDM, receiver are described. A jitter compensation filter is used to process a received signal following timing recovery and/or channel compensation to reduce and/or eliminate the effect of phase jitter. Jitter compensation filter tap weights are updated after filtering the received signal based on one or more signal error measurements. The same received signal is filtered using the updated filter, and error measurements generated from the filtered signal are used to once again update the filter's tap weights. After a fixed number of filter update cycles and/or some other filter updating stop criterion being satisfied, the filter updating process is stopped and the filtered signal is used, e.g., supplied to additional receiver circuitry.

Abstract: Methods and apparatus for implementing digital resampling circuits which create one or more bitstreams which include samples at desired rates, from an input bitstream having a fixed sample rate, are described. The resampling circuits of the present invention achieve the desired sample rates by performing digital interpolation on samples included in the input signal. The interpolation is performed using a filter, e.g., an all-pass infinite impulse response filter which produces an output as a function of a controllable signal delay.

Abstract: Methods and apparatus for reducing phase jitter in a multi-tone, e.g., OFDM, receiver are described. A jitter compensation filter is used to process a received signal following timing recovery and/or channel compensation to reduce and/or eliminate the effect of phase jitter. Jitter compensation filter tap weights are updated after filtering the received signal based on one or more signal error measurements. The same received signal is filtered using the updated filter, and error measurements generated from the filtered signal are used to once again update the filter's tap weights. After a fixed number of filter update cycles and/or some other filter updating stop criterion being satisfied, the filter updating process is stopped and the filtered signal is used, e.g., supplied to additional receiver circuitry. The filter tap values may be reset for each block of data, with the filter update process being repeated starting from preselected initial tap weight values.

Abstract: Methods and apparatus for automatically generating a set partition adjustment renormalization rate (SPARR) threshold used to determine correct or incorrect set partition synchronization in a trellis decoder as a function of a renormalization rate are described. The invention makes use of the inventor's observation that when the system is properly synchronized, the rate of growth of cumulative error sums closely corresponds to an accumulation of minimum set partition errors. In accordance with the present invention a dummy accumulator is set up to accumulate the minimum set partition error for each symbol. If decoder set partition selection is correct the renormalization rate for the dummy accumulator will be approximately the same as an accumulator set up for the normal operation.

Abstract: A sample stream having a fixed sampling rate, representing a filtered version of an input symbol stream is produced by a pulse shaping and resampling device of the present invention. The pulse shaping/resampling device can be used as part of a digital modulator. In order to accommodate a wide range of (“variable”) input baud rates, as part of the pulse shaping/resampling device, a filter having an integral upsampling ratio is used, followed by a resampler circuit having a finely adjustable resampling ratio. The resampler provides an average output rate equal to the desired fixed sampling rate. In various embodiments it is followed by a buffer, which smoothes the output to provide a uniform output rate equal to the desired fixed sampling rate. The pulse shaping/resampling circuit of the present invention may be used in place of a known pulse shaping circuit in a modulator to produce a modulator capable of supporting a wide range of input signal rates.

Abstract: Methods and apparatus for converting a relatively low frequency signal, e.g., a 1.5 MHz signal, to a high frequency signal, e.g., a 30-100 MHz signal, in the digital domain without the need for a digital mixer operating at the high frequency are described. The high frequency represents, e.g., the ultimate digital to analog conversion frequency. In accordance with the present invention an interpolation technique is used to convert the low rate digital signal to a high rate signal and to shift the carrier to a desired frequency. This is accomplished, by first positioning the information signal, e.g., the digital waveform to be modulated on a carrier at a relatively low rate using a digital mixer operating at a fraction of the ultimate digital to analog conversion frequency. The relatively low rate signal generated by the mixing operation is then converted to a high rate signal by one or more interpolator stages. An adjustable passband filter circuit is included in each interpolation stage.

Abstract: Methods and apparatus for detecting and correcting phase and amplitude imbalances existing between I (in-phase) and Q (quadrature phase) signal components of a complex signal, e.g., QAM or OPSK signal, that is being demodulated are described. The phase and amplitude imbalance and correction circuitry of the present invention are implemented as decision directed control loops which can be used in conjunction with an overall decision directed gain control loop. Amplitude imbalance is corrected by adjusting the gain of one of the I and Q signal components. Phase imbalance is corrected by adding a portion of one of the I and Q signal components to the other one of the I and Q signal components. Overall amplitude control is achieved by adjusting the gain of both the I and Q signal components by the same amount.

Abstract: Methods and apparatus for converting a relatively low frequency signal, e.g., a 1.5 MHz signal, to a high frequency signal, e.g., a 30-100 MHz signal, in the digital domain without the need for a digital mixer operating at the high frequency are described. The high frequency represents, e.g., the ultimate digital to analog conversion frequency. In accordance with the present invention an interpolation technique is used to convert the low rate digital signal to a high rate signal and to shift the carrier to a desired frequency. This is accomplished, by first positioning the information signal, e.g., the digital waveform to be modulated on a carrier at a relatively low rate using a digital mixer operating at a fraction of the ultimate digital to analog conversion frequency. The relatively low rate signal generated by the mixing operation is then converted to a high rate signal by one or more interpolator stages. An adjustable passband filter circuit is included in each interpolation stage.

Abstract: In accordance with the present invention an interpolation technique is used to convert a low rate digital signal to a high rate signal and to shift the carrier to a desired frequency. This is accomplished, by first positioning the information signal, e.g., the digital waveform to be modulated on a carrier at a relatively low rate using a digital mixer operating at a fraction of the ultimate digital to analog conversion frequency. The relatively low rate signal generated by the mixing operation is then converted to a high rate signal by one or more interpolator stages. An adjustable passband filter circuit is included in each interpolation stage. One feature of the present invention is directed to a control circuit which is response to an H bit frequency control word representing a desired output carrier frequency. The control circuit generates individual filter control signals for each adjustable filter circuit from the single H bit frequency control word.

Abstract: Improved carrier recovery methods and apparatus suitable for use with QAM, QPSK and a wide variety of other modulation formats is described. In accordance with the invention, the phase error between received symbols, representing a frequency error, is determined using one of a plurality of techniques. The estimated frequency error is used to adjust the phase and/or frequency of a received carrier signal to achieve a frequency lock. The methods and apparatus of the present invention can be easily integrated into existing carrier recovery designs to supplement known frequency In accordance with a first embodiment of the present invention, the receipt of pairs of consecutive outer symbols is detected, a frequency error associated with each pair of consecutive symbols is generated, and the frequency error is compared to a selected threshold value to determine if it is a non-ambiguous estimate of the frequency error.

Abstract: Methods and apparatus for detecting the presence of impulse noise in a signal representing digital data and for performing a limiting or blanking operation in a receiver in response thereto are disclosed. In accordance with the present invention, to detect the presence of impulse noise, in one embodiment, the instantaneous sliced error energy for each received symbol is estimated and compared to either a predetermined threshold level indicative of impulse noise or to an adaptive impulse noise threshold level. In an adaptive impulse noise threshold embodiment, an adaptive impulse noise threshold value is generated by averaging the instantaneous sliced error energy values over a period of many symbols and by then applying a preselected gain to the averaged sliced error energy value to obtain an impulse noise threshold value.

Abstract: Methods and apparatus for converting a relatively low frequency signal, e.g., a 1.5 MHz signal, to a high frequency signal, e.g., a 30-100 MHz signal, in the digital domain. An interpolation technique is used to convert the low rate digital signal to a high rate signal and to shift the carrier to a desired frequency. This is accomplished, by first positioning the information signal, e.g., the digital wave form to be modulated on a carrier at a relatively low rate using a digital mixer operating at a fraction of the ultimate digital to analog conversion frequency. The relatively low rate signal generated by the mixing operation is then converted to a high rate signal by one or more interpolation stages. An adjustable passband filter circuit is included in each interpolation stage for selectively filtering out the signal or images created as a result of a signal padding operation performed as part of each interpolation stage.

Abstract: Method and apparatus for providing a QAM, a VSB, and a joint QAM/VSB demodulator are described. The describe demodulators are designed to minimize the amount of duplicated circuitry required to implement a joint QAM/VSB demodulator capable of performing the functions of the disclosed individual QAM and VSB demodulators. All digital architectures are used for each of the described demodulators to facilitate their combination into the described joint VSB/QAM demodulator. The described demodulators are suitable for demodulating, e.g, advanced or high definition television signals modulated using QAM or VSB modulation techniques.

Abstract: A Nyquist filter implemented as a FIR filter comprising a series of N filter cells, where N is an integer is disclosed. The Nyquist filter is especially well suited for use in demodulators capable of demodulating both VSB and QAM signals. During VSB mode operation, each filter cell acts as a single tap of an N tap Nyquist filter. During QAM mode operation, an additional unit delay element, not used during VSB mode operation, is inserted into the signal path of each cell of the Nyquist filter. The introduction of this second unit delay element into the signal path of each filter cell effectively serves to convert each filter cell into two filter taps with the coefficient of the second filter tap being zero. Thus, during QAM mode operation, the Nyquist filter of the present invention operates as a 2N-tap FIR filter which has coefficients of zero for half of the 2N taps.

Abstract: A carrier recovery system for use in illustratively a passband QAM (quadrature amplitude modulation) demodulator (10) and specifically such a system, including apparatus and accompanying methods, that employs separate acquisition and tracking modes and automatic carrier-to-noise estimation. In particular, a carrier recovery system (200) is operated in two modes: an acquisition mode to first attain an initial carrier lock, during which simplified reduced constellation slicing is used to determine a quadrant of each incoming symbol; and a tracking (or lock) mode, during which full slicing is used, to accurately track variations in frequency and phase that may occur to a carrier while the system remains locked. The mode changes based upon whether, e.g., the average error energy associated with fully sliced symbols is greater than or less than a pre-defined switching threshold value.

Abstract: In a widescreen television signal processing system, side panel video information is scrambled prior to modulating an auxiliary subcarrier which is ultimately combined with main panel video information. In a preferred embodiment the scrambling process involves rotation of image line information about a pseudorandomly chosen cut-point.

Abstract: In a transmitter/encoder system for processing a widescreen television signal having main panel and side panel information, side panel luminance information and side panel "Q" color difference information modulate one phase of a quadrature modulated 3.58 MHz auxiliary subcarrier, other than the standard NTSC chrominanced subcarrier, having a field reversing phase characteristics. the other (quadrature) phase of the auxiliary subcarrier is modulated with side panel wideband "I" color difference information. The "I" modulated component is filtered by an inverse Nyquist slope filter which mirrors the filter characteristics of Nyquist slop filter used at a receiver/decoder to achieve a desired flat amplitude response when the signal component containing the side panel luminannce and "Q" color difference information is demodulated.

Abstract: A video signal transmission system includes adaptive compansion circuitry to improve the signal-to-noise ratio of transmitted signals. A compandor at the transmitter, and corresponding inverse compandor at the receiver, are arranged to have a plurality of selectable compansion transfer characteristics. Selection of particular ones of the transfer characteristics is responsive to, e.g., predicted values of current signal.