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Khosrow L Lashkari

from Palo Alto, CA
Age ~76
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Also known as:
Khosrow R Lashkari, Khosro Lashkari, Khoshow Lashkari, Khoshrow Lashkari, Khadijeh Lashkari, Shkari L Khosrow
Phone and address:
749 Montrose Ave, Palo Alto, CA 94303
(650) 855-9351
Related to:
Zarmina Ashufta, 60Hassan Lashkari, 43Nassim Lashkari, 38Koshrow LashkariKhadijeh Lashkari, 68
Connected to:
Amir Lashkari, 79Atesa P Lashkari, 60Lauren Lashkoff, 38Amy E Lashlee, 62Cheryl D Lashlee, 73David C Lashlee, 56Kathleen A Lashlee, 61Rita S Lashlee, 71Robert W Lashlee, 67Adrienne L Lashley, 55

Khosrow Lashkari Phones & Addresses

  • 749 Montrose Ave, Palo Alto, CA 94303 (650) 855-9351
  • 2116 Pappas Pl, Hayward, CA 94542 (510) 583-0643
  • San Jose, CA
  • 1525 Salamanca Ct, Fremont, CA 94539 (510) 490-8049 (510) 687-1791
  • Los Altos Hills, CA
  • Newark, CA
  • Santa Clara, CA

Resumes

Resumes

Khosrow Lashkari Photo 1

Khosrow Lashkari

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Location:
San Francisco Bay Area
Industry:
Telecommunications
Work:
DeVry University 2009 - 2011
Program Dean
Silicon Valley Technical Institute (www.svtii.com) Jan 2006 - Oct 2009
President
DoCoMo USA Labs Jan 2006 - Mar 2007
Consultant
NTT DoCoMo USA Labs 2000 - 2005
Executive research engineer
DoCoMo USA Labs Jan 2001 - Dec 2004
Executive Research Engineer
Education:
Stanford University 1975 - 1982
Khosrow Lashkari Photo 2

Khosrow Lashkari

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Position:
Senior Staff Engineer at Acoustic Technology Inc.
Location:
San Francisco Bay Area
Industry:
Telecommunications
Work:
Acoustic Technology Inc. since Oct 2011
Senior Staff Engineer
DeVry University - Fremont, CA Oct 2009 - Sep 2011
Dean of Engineering
Silicon Valley Technical Institute - San Jose, CA Jan 2007 - Oct 2009
Senior Faculty & Project Leader
DoCoMo USA Labs Jul 2000 - Jan 2007
Executive Research Engineer
Ultratech 1999 - 2000
Senior Researc Engineer
Education:
Stanford University 1975 - 1982
Ph.D., Electrical Engineering

Publications

Us Patents

Joint Optimization Of Excitation And Model Parameters In Parametric Speech Coders

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US Patent:
6859775, Feb 22, 2005
Filed:
Mar 6, 2001
Appl. No.:
09/800071
Inventors:
Khosrow Lashkari - Fremont CA, US
Toshio Miki - Cupertino CA, US
Assignee:
NTT Docomo, Inc. - Tokyo
International Classification:
G01L019/04
US Classification:
704264, 704258, 704262
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.

Source And Channel Rate Adaptation For Voip

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US Patent:
7295549, Nov 13, 2007
Filed:
Feb 14, 2003
Appl. No.:
10/366944
Inventors:
Christine Pepin - Menlo Park CA, US
Johnny Matta - San Jose CA, US
Khosrow Lashkari - Fremont CA, US
Ravi Jain - Mountain View CA, US
Assignee:
NTT DoCoMo, Inc. - Tokyo
International Classification:
H04L 12/56
US Classification:
370352, 370356
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.

Energy-Based Nonuniform Time-Scale Modification Of Audio Signals

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US Patent:
7426470, Sep 16, 2008
Filed:
Oct 3, 2002
Appl. No.:
10/264042
Inventors:
Wai C. Chu - San Jose CA, US
Khosrow Lashkari - Fremont CA, US
Assignee:
NTT Docomo, Inc. - Tokyo
International Classification:
G10L 21/04
US Classification:
704503, 704500, 370521
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.

Method And Apparatus For Frame-Based Loudspeaker Equalization

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US Patent:
7826625, Nov 2, 2010
Filed:
Dec 19, 2005
Appl. No.:
11/312009
Inventors:
Khosrow Lashkari - Hayward CA, US
Assignee:
NTT DoCoMo, Inc. - Tokyo
International Classification:
H04R 3/00
US Classification:
381 96, 381 59, 381 949, 381103
Abstract:
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.

Modified Volterra-Wiener-Hammerstein (Mvwh) Method For Loudspeaker Modeling And Equalization

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US Patent:
7873172, Jan 18, 2011
Filed:
May 31, 2006
Appl. No.:
11/446085
Inventors:
Khosrow Lashkari - Hayward CA, US
Assignee:
NTT DoCoMo, Inc. - Tokyo
International Classification:
H04R 29/00
US Classification:
381 59, 381 98, 700 94
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.

Method And Apparatus For Self-Degrading Digital Data

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US Patent:
20030216824, Nov 20, 2003
Filed:
May 14, 2002
Appl. No.:
10/146404
Inventors:
Hao-Hua Chu - Mountain View CA, US
Khosrow Lashkari - Fremont CA, US
Ged Powell - San Jose CA, US
Assignee:
DoCoMo Communications Laboratories USA, Inc.
International Classification:
G06F017/00
H04R029/00
US Classification:
700/094000, 381/056000
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.

Efficient Implementation For Joint Optimization Of Excitation And Model Parameters With A General Excitation Function

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US Patent:
20040210440, Oct 21, 2004
Filed:
Oct 3, 2003
Appl. No.:
10/678247
Inventors:
Khosrow Lashkari - Fremont CA, US
Toshio Miki - Yokohama, JP
International Classification:
G10L013/00
US Classification:
704/264000
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.

Method And Apparatus For Loudspeaker Equalization

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US Patent:
20050271216, Dec 8, 2005
Filed:
Jun 3, 2005
Appl. No.:
11/145411
Inventors:
Khosrow Lashkari - Fremont CA, US
International Classification:
H04R029/00
H03G003/00
US Classification:
381059000, 381061000
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.
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Khosrow L Lashkari from Palo Alto, CA, age ~76 Get Report