Ntt Docomo, Inc.

University of California

Khosrow Lashkari

Wai Chung Chu

Toshio Miki

Ravi Kumar Jain

Christine Pepin

Johnny M Matta

Daniel M Empey

Steven E Shladover

Edward H Lechner

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.

Modified volterra-wiener-hammerstein (MVWH) method for loudspeaker modeling and equalization

A method and apparatus for loudspeaker 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.

Method and apparatus for frame-based loudspeaker equalization

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.

Energy-based nonuniform time-scale modification of audio signals

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.

Source and channel rate adaptation for VoIP

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.

Joint optimization of speech excitation and filter parameters

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.

Gradient descent optimization of linear prediction coefficients for speech coders

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.

Joint optimization of excitation and model parameters in parametric speech coders

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. Sensors in the roadway detect when vehicles requiring power are present, and a wayside inductor segment controller responds to the sensory signals by energizing only those wayside inductor segments needed to transfer power to such vehicles. This control methodology improves the energy efficiency of the system. In addition, the roadway sensors can be designed to detect identification signals broadcast by vehicle identification transmitters, thereby enabling the system to charge for energy usage by each vehicle.

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Fremont, CA, United States of America

Khosrow Lashkari