Adaptive linear and decision feedback receiver structures for coherent demodulation in asynchronous code division multiple access (CDMA) systems are considered. It is assumed that the adaptive receiver has no knowledge of the signature waveforms and timing of other users. The receiver is trained by a known training sequence prior to data transmission and continuously adjusted by an adaptive algorithm during data transmission. The proposed linear receiver is as simple as a standard single-user detector receiver consisting of a matched filter with constant coefficients, but achieves essential advantages with respect to timing recovery, multiple access interference elimination, near/far effect, narrowband and frequency-selective fading interference suppression, and user privacy

Adaptive receiver structures for asynchronous CDMA systems

Adaptive linear and decision feedback receiver structures for coherent demodulation in asynchronous code division multiple access (CDMA) systems are considered. It is assumed that the adaptive receiver has no knowledge of the signature waveforms and timing of other users. The receiver is trained by a known training sequence prior to data transmission and continuously adjusted by an adaptive algorithm during data transmission. The proposed linear receiver is as simple as a standard single-user detector receiver consisting of a matched filter with constant coefficients, but achieves essential advantages with respect to timing recovery, multiple access interference elimination, near/far effect, narrowband and frequency-selective fading interference suppression, and user privacy. An adaptive centralized decision feedback receiver has the same advantages of the linear receiver but, in addition, achieves a further improvement in multiple access interference cancellation at the expense of higher complexity. The proposed receiver structures are tested by simulation over a channel with multipath propagation, multiple access interference, narrowband interference, and additive white Gaussian noise. >

A digital radio receiver is described. The digital receiver of the present invention contemplates a digital radio receiver which operates on a received analog signal which has been converted to a digital form after preselection at the output of the antenna. The digital receiver of the present invention comprises a preselector, a high-speed analog-to-digital (A/D) converter, a digitally implemented intermediate-frequency (IF) selectivity section having an output signal at substantially baseband frequencies, and digital signal processor (DSP) circuit performing demodulation and audio filtering. The radio architecture of the present invention is programmably adaptable to virtually every known modulation scheme and is particularly suitable for implementation on integrated circuits.

Digital radio frequency receiver

Two existing blind equalization tap update recursions for 64-point and greater QAM (quadrature amplitude modulation) signal constellations are studied, along with existing and novel carrier and timing recovery techniques. It is determined that the superior tap update recursion is the one known as the constant modulus algorithm. Carrier recovery requires a modified second-order decision-directed digital phase-locked loop. An all-digital implementation of band-edge timing recovery is used. With 14.4 kb/s outbound transmission using CCITT V.33 trellis-coded 128-QAM signals having 12.5% excess bandwidth, a prototype blind retrain procedure is developed to demonstrate the feasibility of the techniques for high-speed multipoint modems. A WE DSP32-based real-time digital signal processor was employed to test the retrain over a set of severely impaired channels. For each channel in the set, the retrain succeeded at least 90% of the time. >

Joint blind equalization, carrier recovery and timing recovery for high-order QAM signal constellations

Apparatus and associated methods are provided which allow a remote user to maintain a virtual session with a server. A virtual session allows a remote and possibly mobile user to maintain a virtual presence in an office environment without actually being present. Using the present invention, a remote user can access a central application program such as an Internet service provider, a database system, an inventory system or billing system. Likewise, the remote user can receive calls and other forms of communications as though he or she were present in an office environment. A virtual session does not require a physical connection to be continuously present in order to provide a virtual connectivity. This is especially important for mobile applications where the remote user may incur long distance and/or wireless toll charges. Also, methods are presented to allow a remote unit to rapidly reconnect in a transparent and seamless way without burdening the user with the need to connect and reconnect or to upload and download information. Related methods are provided to allow the virtual session to be established, operated and maintained.

Virtual connection of a remote unit to a server

In a multipoint data communication system using quadrature-amplitude modulation, a master modem (20) and a plurality of tributary modems (11a, 11b . . . 11n) are interconnected via respective transmission channels (13a, 13b . . . 13n, 16). Adaptive equalizer circuitry (55, 56) in the master modem equalizes the channel from a particular tributary by multiplying samples of signals received from the tributary by an ensemble of tap coefficients associated with the tributary. The tap coefficient ensembles for each tributary are stored in a memory (91) from which they are retrieved at the start of transmission from that tributary. Timing-acquisition circuitry (29) within the master modem adjusts the phase of the latter's sampling circuitry (23, 27) at the start of transmission from a given tributary so that the received signals are sampled at the correct time points. In particular, a timing acquisition signal having spectral components only within the non-rolloff region of the equalized baseband-equivalent transfer function is transmitted by the tributary. The master samples and equalizes the received timing acquisition signal to form a succession of timing acquisiton equalizer outputs. The time by which the operation of the sampling circuitry is to be advanced or retarded is determined as a trigonometric function of two successive ones of the timing acquisition equalizer outputs. The timing acquisition signal is illustratively a double-dotting pattern having a four-symbol-interval period. That period is integrally related to the carrier frequency. This allows the samples needed in order to begin forming the timing acquisition equalizer outputs to be generated by replicating the samples taken over only four symbol intervals.

Timing acquisition in voiceband data sets

A quadrature amplitude modulated (QAM) data signal transmitted at T symbols per second is sampled in a data receiver at a rate of 2/T samples per second and applied to a transversal-type equalizer structure (25, 46, 34, 35) having taps spaced at T/2 second intervals. A demodulated equalized signal (aj, bj), generated once every T seconds, is quantized to form a decision (A(Alpha)j*, B(Alpha)j*) as to the value of the original modulating data symbol. An error signal (ej, ej) is formed in response to the pre- and post-quantized values of the demodulated equalized signal. Tap coefficients (ci(j), (Alpha)i(j)) used in generating the equalized signals are updated in response to a) a correction term which is a function of the error signal and b) a predetermined tap leakage term which has a constant magnitude. The introduction of the tap leakage term maintains the coefficient values at minimum levels.

Coefficient tap leakage for fractionally-spaced equalizers

In a first data receiver (10), sampling circuitry (20, 25, 35) forms samples of a received data signal representing a succession of data symbols. The samples are formed at twice the symbol rate. Transversal filter circuitry (251, 261) operates on the samples by multiplying them by respective ones of a queue of coefficients. Further circuitry (55, 60, 65, 70, 252, 262), operative in response to the resulting products, forms decisions as to the values of the transmitted symbols and updates the values of the coefficients. Timing recovery circuitry (40) periodically identifies the largest of the coefficients in magnitude and either advances or retards the operation of the sampling circuitry by a fixed step size depending on whether that coefficient is or is not within a predetermined portion of the queue. Timing recovery circuitry (830) within a second data receiver (100) operates in a similar manner, but employs a step size whose magnitude is determined by the position of the largest coefficient relative to the center of the coefficient queue.

Timing recovery technique

Fractionally spaced adaptive equalizers can readily compensate for timing-phase variations between the transmitter and receiver clocks that are on the order of several symbol intervals. When there is a frequency difference between these clocks, the equalizer compensates for the accumulated timing-phase difference by shifting the equalizer tap weights in the appropriate direction along the delay line. Performance, as measured by the mean-squared error, is not degraded until the center (largest) tap migrates toward one end of the equalizer. In this paper, we demonstrate how the spatial distribution of the equalizer coefficients can be used to lock the receiver sampling frequency to that of the transmitter. Replacing the standard envelope-derived timing recovery system with a closed-loop center-tap tracking algorithm offers significantly improved performance for narrow roll-off or severely-attenuated systems, reduced complexity, and a structure appropriate for all-digital receivers.

http://ieeexplore.ieee.org/iel7/6731003/6772018/06772024.pdf

Center-tap tracking algorithms for timing recovery

An all digital multitap transversal filter equalizer system is disclosed in which a received data signal is sampled at a predetermined rate. Each sample of the data signal is converted into a 10-bit serial digital word. The 10-bit word is multiplied by a second digital word successively at each tap to provide data for composite output data words. It has been found that by converting the composite output data word into one''s complement format, error magnitude and error polarity signals can be derived for adaptive adjustment without additional equipment.

Howard Clarence Meadors

Papers

Timing acquisition in voiceband data sets

Coefficient tap leakage for fractionally-spaced equalizers

Timing recovery technique

Center-tap tracking algorithms for timing recovery

Digital equalizer in which tap adjusting signals are derived by modifying the signal code format