Abstract: A high speed data transfer system includes a WAU (201) which is utilized to provide high speed access to satellite transferred data. The system is configured such that a plurality of data utilization (205) may access the high speed data via wireless links to the WAU (201). Advantageously, high speed data services may be provided to users without the users requiring individual satellite antennas.

Abstract: Apparatus are provided for a communication signal processing apparatus having a channel estimator configured to generate a first channel estimation at a first time point, a linear predictor coupled to the channel estimator, and an adaptive filter coupled with the linear predictor. The linear predictor is configured to predict a second channel estimation based on a second time point and a channel frequency. The second time point is concurrent or subsequent to the first time point. The second channel estimation includes a first coefficient. The linear predictor includes a first predictor having a sample time point. The first predictor is configured to generate the first coefficient of the second channel estimation based on the sample time point and the second time point. The sample time point is prior to the second time point. The adaptive filter is configured to recursively determine the second channel estimation.

Abstract: An embodiment of a method for parameterizing an input signal includes non-linearly transforming the input signal, removing higher order terms by passing the input signal through a low pass filter to produce a linear combination of data symbols and DC components, solving for the DC components, separating a plurality of mixed baseband signals from the input signal, and coordinate transforming each of the separated signals from polar coordinates to Cartesian coordinates. An embodiment of a receiver includes a non-linear transform operator receiving a digitized input signal and adapted to produce a linear combination of data symbols, DC components, and carrier effects; a low-pass filter for removing higher order terms; a separator unit for separating the plurality of mixed baseband signals; and a coordinate transform for converting the separated signals from polar coordinates to Cartesian coordinates.

Abstract: Methods and apparatus are provided for distributed polyphase spread spectrum communication and non-contiguous spectrum underlay. The method includes polyphase channelizing a direct sequence spread spectrum (DSSS) signal, determining available spectrum in an observed frequency range having a noise floor based on detection of polyphase users below the noise floor, distributing the polyphase channelized signal among the available spectrum, and interference mitigation using a combination of standard signal detection and cyclostationary feature detection methods.

Abstract: A high speed data transfer system includes a WAU (201) which is utilized to provide high speed data access to satellite transferred data. The system is configured such that a plurality of data utilization devices (205) may access the high speed data via wireless links to the WAU (201). Advantageously, high speed data services may be provided to users without the users requiring individual satellite antennas.

Abstract: A high speed data transfer system includes a WAU (201) which is utilized to provide high speed data access to satellite transferred data. The system is configured such that a plurality of data utilization devices (205) may access the high speed data via wireless links to the WAU (201). Advantageously, high speed data services may be provided to users without the users requiring individual satellite antennas.

Abstract: An orthogonal frequency division multiplexed wideband communication system provides improved time and frequency synchronization by inserting an unevenly spaced pilot sequence within the constellation data. A receive correlates the received data using the unevenly spaced pilot sequence. The pilot sequence is generated with a maximum length pseudo random noise code and inserted into frequency bins having prime numbers.

Abstract: An apparatus is disclosed for continuous load optimization of wireless communication networks by making use of statistical data about both the communication medium characteristics and the characteristics of messages that users desire to send over the network. The network comprises a network control base station (20) and a plurality of slave stations (22). One or more applications (72, 104) are associated with each station which are used to generate the messages in digital format. Each application is coupled to a protocol stack (88) that defines the required characteristics of the data and changes to the data that occur during the communication process. An interface (90) is coupled to the layers of each protocol stack (88) to allow tuning of the message characteristics to maximize successful transmission with minimum message latency.

Abstract: A non-linear signal classifier (26) includes a polynomial expansion unit for expanding signal feature vectors determined by a feature extraction unit (25) for a received signal. The expanded signal feature vectors are each combined with a plurality of signal classification models that are stored in a model memory (76). The signal classification models are each associated with a particular signal type that is recognized by the non-linear signal classifier. A scoring unit (72) generates a score for each of the signal classification models based on the result of the combination. The scores are analyzed by a selection unit (74) which determines which of the signal classification models (i.e., which of the signal types) most likely represents the received signal. Training equipment (60) is also provided for training the non-linear signal classifier (26) to recognize new signal types.

Abstract: The invention relates to a communication system (300) having a receiver (304) that is capable of performing targeted interference suppression. An interference classifier (314) within the receiver (304) analyzes a signal received from a channel (306) and identifies and classifies interference components within the signal. An interference suppressor (316) then suppresses the interference components in the signal based on interference type. In one embodiment, the interference suppressor (316) includes a plurality of interference suppression modules that are each optimal for suppressing certain interference types. The interference suppressor (316) selects one of the interference suppression modules based on the type of interference present in the received signal. In another embodiment, a hybrid interference mitigation system (10) is provided by combining targeted interference suppression, frequency hopping adaptation, and processing gain adaptation.

Abstract: The invention relates to a communication system (300) having a receiver (304) that is capable of performing targeted interference suppression. An interference classifier (314) within the receiver (304) analyzes a signal received from a channel (306) and identifies and classifies interference components within the signal. An interference suppressor (316) then suppresses the interference components in the signal based on interference type. The interference suppressor (316) includes a plurality of interference suppression modules that are each optimal for suppressing certain interference types. In one embodiment, one or more of the interference suppression modules perform interference suppression using modal moment estimates to identify different modes within the signal.

Abstract: The invention relates to a communication system (300) having a receiver (304) that is capable of performing targeted interference suppression. An interference classifier (314) within the receiver (304) analyzes a signal received from a channel (306) and identifies and classifies interference components within the signal. An interference suppressor (316) then suppresses the interference components in the signal based on interference type. In one embodiment, the interference suppressor (316) includes a plurality of interference suppression modules that are each optimal for suppressing certain interference types. The interference suppressor (316) selects one of the interference suppression modules based on the type of interference present in the received signal. In another embodiment, a hybrid interference mitigation system (10) is provided by combining targeted interference suppression, frequency hopping adaptation, and processing gain adaptation.

Abstract: The present invention relates to a communications system (100) that is capable of adapting to an unknown or varying spectral environment in a channel (14) between two communications units (12, 16). The system (100) maintains a spectral profile of the channel (14) and uses the spectral profile to determine appropriate transmit parameters for the system (100). The system (100) can be programmed to provide an optimal transmit signal for achieving a predetermined performance goal (such as, for example, maximum data rate at a given bit error rate (BER)) in light of the spectral environment in the channel (14). In one embodiment, the spectral profile is maintained in a spectrum table memory (27) that is periodically (or continuously) updated.

Abstract: A method and apparatus (100) for pitch-epoch-synchronous source-filter speech encoding by means of error component modeling methods (310) which capture fundamental orthogonal (uncorrelated) basis elements of an excitation source waveform. A periodic waveform model (318) along with four orthogonal error waveforms, desirably including phase error (319), ensemble error (321), standard deviation error (323), and mean error (324) waveforms, are incorporated together to form a complete description of the excitation. These error waveforms (319,321, 323, 324) represent those portions of the excitation that are not represented by the purely periodic model. By thus orthogonalizing the error components, the perceptual effect of each element is isolated from the composite set, and can thus be encoded separately.

Abstract: A speech coder (100) computes scalar statistics (180), ensemble statistics (190), spectral parameters (150), and a normalized excitation waveform (270) which describe a frame of speech samples. The coder (100) encodes the statistics (220, 230), spectral parameters (155), and the normalized waveform (290) for later decoding and synthesis. A speech synthesizer (900) decodes the encoded scalar statistics (570), encoded ensemble statistics (560), encoded spectral parameters (490), and encoded normalized excitation waveform (550). The synthesizer (900) then denormalizes (670) the normalized excitation waveform using the scalar statistics and the ensemble statistics, resulting in a decoded excitation waveform. Speech is synthesized (710) from the decoded excitation waveform and the decoded spectral parameters.

Abstract: A vocoder device and corresponding method characterizes and reconstructs speech excitation. An excitation analysis portion performs a cyclic excitation transformation process on a target excitation segment by rotating a peak amplitude to a beginning buffer location. The excitation phase representation is dealiased using multiple dealiasing passes based on the phase slope variance. Both primary and secondary excitation components are characterized, where the secondary excitation is characterized based on a computation of the error between the characterized primary excitation and the original excitation. Alternatively, an excitation pulse compression filter is applied to the target, resulting in a symmetric target. The symmetric target is characterized by normalizing half the symmetric target. The synthesis portion performs reconstruction and synthesis of the characterized excitation based on the characterization method employed by the analysis portion.

Abstract: A low-rate voice coding method and apparatus uses vocoder-embedded neural network techniques. A neural network controlled speech analysis processor includes a neural network which manages speech characterization, encoding , decoding, and reconstruction methodologies. The voice coding method and apparatus uses multi-layer perceptron (MLP) based neural network structures in single or multi-stage arrangements.

Abstract: A speech vocoder device and corresponding method synthesizes speech excitation waveforms. The method entails reconstructing (216) an excitation target from decoded speech data, creating (220) aligned excitation segments by normalizing (296), correlating (298), and aligning (300) a source segment and a target segment, reconstructing normalized intervening segments by ensemble interpolating (318) between the source segment and the target segment, denormalizing (320) the normalized intervening segments, and reconstructing (322) an excitation waveform from the denormalized intervening segments, the source segment, and the target segment.