Abstract: A method for providing digital compensation and vector calibration for a base station in a wireless network is provided. The method includes performing a single process that is operable to provide simultaneously both digital compensation and vector calibration for the base station. According to one embodiment of the present disclosure, the single process is performed by receiving a transmitted signal based on a compensation/calibration (C/C) signal, processing the received signal, generating a C/C weight based on the processed signal, and using the C/C weight in a normal operational mode for the base station.

Abstract: A method of calibrating a base station that comprises an adaptive antenna array is provided. The method includes selecting one operational mode from a plurality of operational modes for the base station. An operational mode component set corresponding to the selected operational mode is selected from a plurality of operational mode component sets in the base station. The base station is calibrated using the selected operational mode component set.

Abstract: A method of calibrating multiple types of base stations is provided. The method includes selecting one of a plurality of technology types for the wireless base station. The wireless base station has an adaptive antenna array. A calibration is performed for the wireless base station based on the selected technology type.

Abstract: A method for performing a soft handoff for a mobile station from a source base station to a target base station in an Orthogonal Frequency-Division Multiple Access (OFDMA) wireless network that is capable of communicating with a plurality of mobile stations in a coverage area of the OFDMA wireless network is provided. The method includes determining whether a relative delay between receipt of a source pilot signal from a source base station and receipt of a target pilot signal from a target base station is less than a downlink coherence time. When the relative delay is less than the downlink coherence time, the soft handoff is performed.

Abstract: A method of channel estimation and interference cancellation in an Orthogonal Frequency Division Multiple Access network is provided. The method includes determining an estimate of a transmitted signal based on an initial channel estimate and a received signal. An extracted transmitted signal is generated based on the estimate of the transmitted signal using a data integrity checking method. A subsequent channel estimate is generated based on the received signal and the extracted transmitted signal. If a subsequent iteration criterion is met, a subsequent estimate of the transmitted signal is determined based on the subsequent channel estimate and the received signal. A subsequent extracted transmitted signal is generated based on the subsequent estimate of the transmitted signal using a data integrity checking method. An additional subsequent channel estimate is generated based on the received signal and the subsequent extracted transmitted signal.

Abstract: A method of processing multiple-input/multiple-output pilot signals in an Orthogonal Frequency Division Multiplexing network is provided. The method includes, for each of a plurality of transmit antennas in a transmit antenna array, generating a processed pilot signal using a gain unique to the transmit antenna and transmitting the processed pilot signal.

Abstract: A flexible and scalable deployment of orthogonal frequency division multiplexing (OFDM) technology in a mobile cellular system, including a system that deploys multiple channels within a common base station, in which different modulation techniques are used to compensate for adjacent-channel interference.

Abstract: A system and method for channel estimation and echo cancellation in wireless system repeaters. A repeater with improved echo cancellation comprises an input radio control element having a receiving antenna and an output radio control element having a transmitting antenna; a power amplifier connected between the input radio control element and the output radio control element; and a signal processing unit connected between the input radio control element and the output radio control element. The signal processing unit is configured to estimate the communication channel from the transmitting antenna to the receiving antenna to produce a channel estimation. The signal processing unit is further configured to perform echo cancellation based on the channel estimation.

Abstract: This patent disclosure describes an effective synchronization technique for an embedded signaling system based on direct sequence spread spectrum technology. The technique provides an approach to robust detection of information codes embedded in an audio signal which is subject to undergoing such processes as re-recording of the material onto a different media, and conversions from digital to analog form and from analog to digital form. The technique is useful in preventing the embedded information from being corrupted or obscured by pirates via signal distortion operations including D/A and A/D conversions or by legitimate users in artistic processing of the material. The technique according to one embodiment of the present invention includes obtaining a timing error estimate in the process of synchronization by estimating the initial sampling rate/timing error by peak picking and by restoring the correct timing by sample interpolation after synchronization.

Abstract: A quadratic phase interpolation method for synthesis of musical tones incorporates both phase and frequency measurements at the boundaries of a data frame using a weighted least square algorithm approach. The approach assumes that the true frequency and phase at the two ends of a data frame conform to a quadratic phase model and that exact match between measured phase and frequency with the quadratic model is not necessary because of the noise in the measurements.

Abstract: A coprocessor (15) for synthesizing a signal from the sum of sinusoids preferably includes an electronic system (20) having a host processor (12) that forwards frame boundary parameters to the coprocessor (15). Parameter registers (26) in coprocessor (15) store synthesis parameters for iteratively deriving amplitude and phase values for each sample point within a data frame. Adders (28, 30, 32) generate current amplitude from one addition, and current phase value from two additions, with the results stored back into parameter registers (26). A sine function calculator circuit (34), which may use a CORDIC technique, receives the current amplitude and phase values, and generates a digital component signal for the current sample point for one of the sinusoids. Digital component signals are accumulated at the sample point in a data sample buffer (40) and output at an output (44).

Abstract: This application describes a method for musical signal analysis and synthesis using global waveform fitting (GWF). GWF uses a sinusoidal model with quadratic phase. GWF tries to fit the entire data record (signal waveform) to the assumed signal model. The model parameters obtained by GWF have clear physical interpretations or meanings.

Abstract: An improved method of providing a pitch shifted or frequency transformed signal includes frequency scaling the original signal (12) and generating a desired spectrum envelope of the frequency transformed signal, As(z) by LPC analysis of the original signal (11). Further the method includes producing an approximation of the spectrum envelope of the frequency scaled signal As(z, &bgr;) by performing LPC analysis on the original signal (11), obtaining LSFs (13), scaling (15) and transforming the scaled LSFs back to LPC (17).

Abstract: A method is disclosed for synthesizing acoustic waveforms, especially musical instrument sounds. The acoustic waveforms are characterized by time-varying amplitudes, frequencies and phases of sinusoidal components. These time-varying parameters, at each analysis frame, are obtained in one embodiment by short term Fourier transforms (STFT). The spectrum envelope at each frame is parameterized with an autoregressive moving average model and applied to a waveform consisting of unit amplitude sinusoids via time-domain filtering. The resulting synthetic waveform preserves the time-varying frequency and phase information and has the same relative energy distribution among different sinusoidal components as that of the original signal. Finally, a general waveform shape for the type of acoustic signal being synthesized is applied.