Abstract: A method and apparatus for adaptive precompensation is disclosed. In one embodiment, the method comprises modifying operation of a predistortion filter in response to previous predistorted values and an original input signal, determining a precompensation error between the original input samples and the predicted loudspeaker output, and substantially reducing the precompensation error by using the exact inverse of a loudspeaker model that is a cascaded arrangement of at least one linear system with a non-linear system.

Abstract: A method and apparatus for loudpeaker equalization are described. In one embodiment, the method comprising generating a set of parameters using an invertible, non-linear system based on input audio data and output data corresponding to a prediction of an output of a loudspeaker in response to the input data, and controlling an exact non-linear inverse of the non-linear system using the set of parameters to output a predistorted version of the input data.

Abstract: A method, apparatus and system are disclosed herein for loudspeaker equalization. In one embodiment, the system comprises an input for receiving samples of an input signal, a pre-compensator to produce a pre-compensated output in response to the samples of an input signal, parameters of a loudspeaker model, and previously predistorted samples of the input signal, and a loudspeaker, corresponding to the loudspeaker model, to produce an audio output in response to the pre-compensated output.

Abstract: A speech synthesis system is provided that optimizes a synthesis filter. Optimization is achieved by minimizing a synthesis error between the original speech sample and a synthesized speech sample. A gradient search algorithm in the root domain is also provided to aid minimization of the synthesis error.

Abstract: A method and apparatus for generating excitation and model parameters in source filter models are described. In one embodiment, the method comprises generating synthesized speech samples, using a synthesis filter, in response to an excitation signal, determining a synthesis error between original speech samples and the synthesized speech sample and substantially reducing the synthesis error by computing both the excitation signal and filter parameters for the synthesis filter. The substantial reduction in the synthesis error is performed by applying a gradient descent algorithm to roots or LSPs of the polynomial representing the synthesis error over a series of iterations, and includes computing a gradient of the synthesis error in terms of gradient vectors of the synthesized speech samples by generating partial derivatives, using a recursive algorithm, for terms of a polynomial representing the synthesized speech samples over a series of iterations.

Abstract: A coding system and method for a terminal including a multi-rate codec is disclosed. The terminal includes a multi-rate adaptive coder that is capable of transmitting a continuous voice stream transmission at a source code bit rate and a channel code bit rate. A quality of service probing module probes an end-to-end network path of the continuous voice stream transmission to obtain at least one quality of service parameter. A quality of service management module determines at least one constraint associated with the continuous voice stream transmission. An adaptive bit rate algorithm module dynamically adjusts the source code bit rate and the channel code bit rate as a function of the quality of service parameter and the constraint to obtain a maximum value of perceived user performance during the continuous voice stream transmission.

Abstract: A method for energy based, non-uniform time-scale compression of audio signals includes receiving a frame of data corresponding to an input audio signal and segmenting the data into a plurality of segments. The method further includes estimating a value related to energy of the frame of data, determining a peak energy estimate for the frame, determining an energy threshold based on the peak energy estimate of the frame and comparing the value related to energy of the frame of the data with the energy threshold to control time-scale compression of the audio data.

Abstract: When copies of digital are made or after use of the digital data, the quality of the digital data is reduced or degraded. The degradation may be in any way suitable to the nature of the digital data. In one embodiment, the content provider which originates the digital data may specify a degradation policy or degradation specification model for the digital data. When the digital data is copied or moved, the copy is degraded according to this specified policy or model. In this manner, the content provider can control the extent to which the end user can copy the material. The end user can make copies limited in number only by the degradation of the digital data.

Abstract: An optimization algorithm is provided for linear prediction based speech coding systems. The optimization algorithm minimizes the error between original speech samples and synthesized speech samples. Optimized linear prediction coefficients are computed directly from a system difference equation without converting the coefficients into the root-domain.

Abstract: An efficient optimization algorithm is provided for multipulse speech coding systems. The efficient algorithm performs computations using the contribution of the non-zero pulses of the excitation function and not the zeroes of the excitation function. Accordingly, efficiency improvements of 87% to 99% are possible with the efficient optimization algorithm.

Abstract: A gradient search algorithm is provided for speech coding systems. The gradient search algorithm calculates the gradient of a speech synthesis polynomial using the contribution of decomposition coefficients. The contribution of the decomposition coefficients is then recalculated at successive iterations.

Abstract: A speech synthesis system is provided that optimizes a synthesis filter. Optimization is achieved by minimizing a synthesis error between the original speech sample and a synthesized speech sample. A gradient search algorithm in the root domain is also provided to aid minimization of the synthesis error.

Abstract: An inductive energization system for moving vehicles includes wayside inductors under the roadway and pickup inductor circuits in electrically powered vehicles. A pickup power controller has a switching circuit, including a zero-crossing trigger circuit, a current limiting inductor, and a bleed resistor. The controller provides for fast switching, desirable for closed loop control of the inductive energy transfer system, as well as low harmonic distortion of waveforms, low acoustic noise, and low maintenance requirements. The pickup inductor of the preferred embodiment has rigid metal conductors bonded together into a single member, allowing this element to serve as both a current carrying element as well as a primary structural member of the pickup inductor. The roadway inductor is split into many segments.