Abstract: A predistortion technique for high power amplifiers includes an adaptive predistortion algorithm that operates independently of data samples to write a set of complex gain values, or predistortion parameters, to a lookup table. The algorithm may be processor-driven. The gain values are taken from the lookup table and multiplied by a complex digital baseband waveform. The gain values may first be subjected to interpolation. The downconverted output of the amplifier is measured to gauge the efficacy of the predistortion. Based on the effect of the predistortion upon the ratio of in-band power to out-of-band power, decisions are made on the set of predistortion parameters for the next iteration of the algorithm. The algorithm runs continuously, perturbing parameters and adapting the predistortion functions accordingly in an effort to continually reflect instantaneous amplitude-modulation and phase-modulation relationships that may change over time with temperature variation or component aging.

Abstract: A method and apparatus for computing soft decision input metrics to a turbo decoder includes circuits associated with eight-ary phase shift keyed (8PSK) modulation and sixteen-ary quadrature amplitude modulation (16QAM). In both implementations log-likelihood ratio (LLR) metrics on code symbols are estimated as products of various constant values and various combinations of the in-phase and quadrature components of a demodulated soft decision. In the implementation associated with the 16QAM modulation scheme, an estimate of the carrier-signal-to-interference (C/I) ratio is also used to estimate some of the LLR metrics. Estimates of the LLR metrics may also be obtained in association with generalized square QAM and M-ary PSK modulation schemes including, e.g., 64QAM, 256QAM, and 16PSK.

Abstract: A frame erasure compensation method in a variable-rate speech coder includes quantizing, with a first encoder, a pitch lag value for a current frame and a first delta pitch lag value equal to the difference between the pitch lag value for the current frame and the pitch lag value for the previous frame. A second, predictive encoder quantizes only a second delta pitch lag value for the previous frame (equal to the difference between the pitch lag value for the previous frame and the pitch lag value for the frame prior to that frame). If the frame prior to the previous frame is processed as a frame erasure, the pitch lag value for the previous frame is obtained by subtracting the first delta pitch lag value from the pitch lag value for the current frame. The pitch lag value for the erasure frame is then obtained by subtracting the second delta pitch lag value from the pitch lag value for the previous frame.

Abstract: An apparatus and method for echo cancellation is presented. The echo canceller comprises an adaptive filter that tracks the impulse response of the echo path and produces an estimate of the echo. Filter adaptation is controlled by a controller based on the rate of the far-end speech signal, the rate of the near-end signal, an acoustic loss measure, and a double talk hangover indicator. The controller may also comprise a step size adaptation unit for determining the adaptation step size of the adaptive filter. In addition, the controller may comprise a noise replacement unit, which controls replacement of the echo residual signal with comfort noise to ensure echo is completely rejected when only the far-end speaker is talking.

Abstract: Techniques to test a wireless communication link. A traffic channel is tested via a test data service option (TDSO) that may be negotiated and connected similar to other services. Test parameters values may be proposed, accepted or rejected, and negotiated. Test data for a channel is generated based on a defined data pattern or a pseudo-random number generator. Sufficient test data may be generated based on the generator for a test interval, stored to a buffer, and thereafter retrieved from a particular section of the buffer to form data block(s) for each “active” frame. The traffic channel may be tested using discontinuous transmission. A two-state Markov chain determines whether or not to transmit test data for each frame. The average frame activity and average burst length are defined by selecting the probabilities for transitioning between the ON/OFF states of the Markov chain, which may be driven by a second generator.

Abstract: The present invention is directed to a digital wireless telecommunications system that includes a plurality of base station of differing spectral capabilities, and a plurality of remote stations capable of transmitting data to and receiving transmissions from the plurality of base stations. The invention herein provides a method for remote station hand-off between base stations of a narrower spectral capacity and base stations of a wider spectral capacity. A method is provided for a wireless telecommunications infrastructure to facilitate a remote station hand-off from a set of narrowband compliant base stations to at least one wideband compliant base station while a remote station is in the coverage area of both types of base stations. Additionally, the invention herein provides remote station apparatus, base station apparatus, and base station controller apparatus for performing the handoff methodology of the present invention.

Abstract: A system provides wireless data service. The system includes a wireless modem configured to be coupled to a terminal equipment unit over a wireline broadcast medium. A network unit is coupled to the wireless modem over a wireless link connection. A network remote server is coupled to the network unit. The network remote server provides IP address information in response to terminal equipment unit address request messages for terminal equipment units coupled to the network unit. In addition, a local server is coupled to the wireless modem. The local server provides IP address information in response to terminal equipment unit address request messages from terminal equipment units coupled to the wireline broadcast medium.

Abstract: A method and apparatus for dynamically adjusting the signal-to-interference ratio (SIR) threshold used to control signal transmission power in closed loop communication systems. The closed loop power control comprises an inner loop that provides up/down transmission power control commands to a transmitter based upon a comparison of the SIR for a received signal to the threshold SIR. The outer loop dynamically adjusts the threshold SIR using historic inner loop power control information. When it is detected that the inner loop is not acceptably performing power control, the target SIR is not changed, or may increased or decreased more or less than 0.5 dB or 0.5 dB/99, respectively.

Abstract: In some aspects, each cell in the communications system can be designed to operate in accordance with a set of back-off factors that identify the reductions in peak transmit power levels for the channels associated with the back-off factors. The back-off factors are defined to provide the required power to a large percentage of the users while reducing the amount of interference. In some other aspects, the cells operate using an adaptive reuse scheme that allows the cells to efficiently allocate and reallocate the system resources to reflect changes in the system. A reuse scheme is initially defined and resources are allocated to the cells. During operation, changes in the operating conditions of the system are detected and the reuse scheme is redefined as necessary based on the detected changes. For example, the loading conditions of the cells can be detected, and the resources can be reallocated and/or the reuse scheme can be redefined.

Abstract: An optimum combiner that reduces the amount of interference imposed upon a first base station by transmissions of other base stations within the same communication system. Two antennas are used to receive transmissions within a receiving station. A rake receiver is coupled to each antenna. By optimally combining the signals that are received by each independent finger of the rake receiver, interference that is correlated between a finger associated with the first antenna and a finger associated with the second antenna can be minimized with respect to the desired signal. Optimum combining requires determination of optimum combining coefficients. A max-ratio combining algorithm may be substituted for the optimal combining algorithm depending upon interference characteristics.

Abstract: A low-bit-rate coding technique for unvoiced segments of speech includes the steps of extracting high-time-resolution energy coefficients from a frame of speech, quantizing the energy coefficients, generating a high-time-resolution energy envelope from the quantized energy coefficients, and reconstituting a residue signal by shaping a randomly generated noise vector with quantized values of the energy envelope. The energy envelope may be generated with linear interpolation technique. A post-processing measure may be obtained and compared with a predefined threshold to determine whether the coding algorithm is performing adequately.

Abstract: A system for selectively blocking or dropping calls in a telecommunications network. The system includes a first mechanism that monitors usage of the telecommunications network and provides a gauging signal in response thereto when the network usage exceeds a predetermined threshold. A second mechanism determines a percentage of calls to block by the network in response to the gauging signal. The percentage of calls to block corresponds to a desired number of blocked calls out of a certain number of calls. A third mechanism automatically blocks the percentage of calls out of the certain number of calls. In a specific embodiment, the third mechanism includes an additional mechanism that accounts for values of the percentage that do not evenly divide into the certain number of calls. The additional mechanism includes another mechanism that accounts for a remainder resulting from a division of the certain number of calls by the desired number of calls to be blocked.

Abstract: A method for tracking the phase of a quasi-periodic signal includes the steps of estimating the phase of the signal for frames during which the signal is periodic, monitoring the performance of the estimated phase with a closed-loop performance measure, and measuring the phase of the signal for frames during which the signal is periodic and performance of the estimated phase falls below a predefined threshold level. In estimating the phase, the initial phase value is set equal to the estimated final phase value of the previous frame if the previous frame was periodic. The initial phase value is set equal to a measured phase value of the previous frame if the previous frame was nonperiodic, or if the previous frame was periodic and performance of the estimated phase for the previous frame fell below the predefined threshold level. For frames during which the signal is nonperiodic, the phase of the signal is measured.

Abstract: A method and apparatus for identifying frequency bands to compute linear phase shifts between frame prototypes in a speech coder includes partitioning the frequency spectrum of a prototype of a frame by dividing the frequency spectrum into segments, assigning one or more bands to each segment, and establishing, for each segment, a set of bandwidths for the bands. The bandwidths may be fixed and uniformly distributed in any given segment. The bandwidths may be fixed and non-uniformly distributed in any segment. The bandwidths may be variable and non-uniformly distributed in any given segment.

Abstract: A system and method for determining the data rate of a frame of data at a receiver of a variable rate communications system. A vocoder at a transmitter encodes a frame of data at one of the rates of a predetermined set of rates. The data rate is dependent on the speech activity during the time frame of the data. The data frame is also formatted with overhead bits, including bits for error detection and detection. At the receiver, the data rate for the frame is determined based on hypothesis testing. Because the data rate is based on speech activity, a hypothesis test may be designed based on the statistics of speech activity. The received data frame is first decoded into information bits at the most probable rate as provided by the hypothesis test. Error metrics are generated for the decoded information bits. If the error metrics indicate that the information bits are of good quality, then the information bits are presented to a vocoder at the receiver to be processed for interface with the user.

Abstract: A distributed voice recognition system includes a digital signal processor (DSP), a nonvolatile storage medium, and a microprocessor. The DSP is configured to extract parameters from digitized input speech samples and provide the extracted parameters to the microprocessor. The nonvolatile storage medium contains a database of speech templates. The microprocessor is configured to read the contents of the nonvolatile storage medium, compare the parameters with the contents, and select a speech template based upon the comparison. The nonvolatile storage medium may be a flash memory. The DSP may be a vocoder. If the DSP is a vocoder, the parameters may be diagnostic data generated by the vocoder. The distributed voice recognition system may reside on an application specific integrated circuit (ASIC).

Abstract: A method and apparatus for subsampling phase spectrum information includes a speech coder for analyzing and reconstructing a prototype of a frame by using intelligent subsampling of phase spectrum information of the prototype. To analyze the prototype, the speech coder produces a phase parameters of a reference prototype, generates phase parameters of a current prototype, and correlates the phase parameters of the current prototype with the phase parameters of the reference prototype in multiple frequency bands. To reconstruct the prototype using linear phase shift values, the speech coder produces a phase parameters of the reference prototype, generates a set of linear phase shift values associated with the prototype, and composes a phase vector from the phase parameters and the linear phase shift values across multiple frequency bands.

Abstract: A method and apparatus for interleaving line spectral information quantization methods in a speech coder includes quantizing line spectral information with two vector quantization techniques, the first technique being a non-moving-average prediction-based technique, and the second technique being a moving-average prediction-based technique. A line spectral information vector is vector quantized with the first technique. Equivalent moving average codevectors for the first technique are computed. A memory of a moving average codebook of codevectors is updated with the equivalent moving average codevectors for a predefined number of frames that were previously processed by the speech coder. A target quantization vector for the second technique is calculated based on the updated moving average codebook memory. The target quantization vector is vector quantized with the second technique to generate a quantized target codevector.

Abstract: A method and corresponding apparatus separately estimates signal amplitude and noise in received signals, particularly symbols encoded using turbo codes. The method and apparatus, however, may be applied to such estimation in power control wireless systems (CDMA systems), other concatenated coding schemes, etc. A method and apparatus initially estimate the signal and noise, separately, in the received signal, such as by using curve fitting techniques. Thereafter, a finer estimate is provided by using the amplitude of received pilot symbols or using an estimate of the noise from the received pilot symbols.

Abstract: In a communications system, a method of transforming a set of message signals representing a message comprising the steps of first encoding one of the set of message signals in accordance with a first keyed transformation, a second encoding of the one of the set of message signals in accordance with at least one additional keyed transformation, a third encoding of the one of the set of message signals in accordance with a self inverting transformation in which at least one of the set of message signals is altered, a fourth encoding of the one of the set of message signals in accordance with at least one additional inverse keyed transformation wherein each of the at least one additional inverse keyed transformation is a corresponding inverse of at least one additional keyed transformation, and fifth encoding the one of the set of message signals in accordance with first inverse keyed transformation wherein the first inverse keyed transformation is the inverse of the first keyed transformation.