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Abhijit Kulkarni

Abhijit Kulkarni, Newbury Park, CA US

Patent application numberDescriptionPublished
20090024185SPECTRAL CONTRAST ENHANCEMENT IN A COCHLEAR IMPLANT SPEECH PROCESSOR - Among other things, enhancing spectral contrast for a cochlear implant listener includes detecting a time domain signal. A first transformation is applied to the detected time domain signal to convert the time domain signal to a frequency domain signal. A second transformation is applied to the frequency domain signal to express the frequency domain signal as a sum of two or more components. A sensitivity of the cochlear implant listener to detect modulation of each component is obtained.01-22-2009
20090264960Tonality-Based Optimization of Sound Sensation for a Cochlear Implant Patient - Methods and systems of optimizing sound sensation of a cochlear implant patient include dividing an audio signal into a plurality of analysis channels, generating one or more tonality indices each representing a tonality of one of the analysis channels, generating one or more stimulation pulses configured to represent the audio signal in accordance with one or more stimulation parameters, and adjusting at least one of the stimulation parameters based on at least one of the tonality indices.10-22-2009
20100198303MODULAR COCHLEAR IMPLANT SYSTEMS INCLUDING IMPLANTABLE SOUND PROCESSORS - Exemplary cochlear implant systems include an implantable head module configured to be implanted within a head of a patient. The implantable head module includes a cochlear stimulator configured to be coupled to an electrode lead, the electrode lead including one or more electrodes configured to be in communication with one or more stimulation sites within the patient. The implantable head module also includes a signal receiver configured to receive a telemetry signal representative of an audio signal from a signal transmitter located external to the patient, a sound processor configured to process the telemetry signal and direct the cochlear stimulator to generate and apply electrical stimulation representative of the audio signal to the one or more stimulation sites via the electrode lead, and a power receiver configured to receive power for operating the implantable head module from a power transmitter located external to the patient.08-05-2010
20110064241Methods and Systems for Reducing an Effect of Ambient Noise Within an Auditory Prosthesis System - An exemplary method of reducing an effect of ambient noise within an auditory prosthesis system includes dividing an audio signal presented to an auditory prosthesis patient into a plurality of analysis channels each containing a frequency domain signal representative of a distinct frequency portion of the audio signal, determining a signal-to-noise ratio and a noise reduction gain parameter based on the signal-to-noise ratio for each of the frequency domain signals, applying noise reduction to the frequency domain signals in accordance with the determined noise reduction gain parameters to generate a noise reduced frequency domain signal corresponding to each of the analysis channels, and generating one or more stimulation parameters based on the noise reduced frequency domain signals and in accordance with at least one of a current steering stimulation strategy and an N-of-M stimulation strategy. Corresponding methods and systems are also disclosed.03-17-2011
20110069853Auditory Front End Customization - A method and system for implementing an acoustical front end customization are disclosed. The customization is implemented to optimize the sound level for each individual cochlear implant user. A known audio signal is generated using a sound source and captured by a microphone system. The captured sound signal is sampled at one or more locations along the signal processing pathway, and a spectrum is determined for the sampled signal and the known signal. A ratio of the two spectrums is related to the undesired transformation of the sampled signal, and a digital filter is designed based on the ratio to filter out the undesired transformation.03-24-2011
20110098786Remote Audio Processor Module for Auditory Prosthesis Systems - An exemplary auditory prosthesis system includes an auditory prosthesis configured to be implanted within a head of a patient and to apply electrical stimulation representative of an audio signal to one or more stimulation sites within the patient in accordance with one or more stimulation parameters, a behind-the-ear sound processing unit configured to be secured to an ear of the patient and to transmit the one or more stimulation parameters to the auditory prosthesis, and a remote audio processor module separate from the behind-the-ear sound processing unit and communicatively coupled to the behind-the-ear sound processing unit via a communication link, the remote audio processor module configured to perform at least a portion of a signal processing heuristic on the audio signal in order to facilitate generation of the one or more stimulation parameters. Corresponding systems and methods are also disclosed.04-28-2011
20110255725Beamforming Microphone System - A system and method for generating a beamforming signal is disclosed. A beam forming signal is generated by disposing a first microphone and a second microphone in horizontal coplanar alignment. The first and second microphones are used to detect a known signal to generate a first response and a second response. The first response is processed along a first signal path communicatively linked to the first microphone, and the second response is processed along a second signal path communicatively linked to the second microphone. The first and second responses are matched, and the matched responses are combined to generate the beamforming signal on a combined signal path.10-20-2011

Patent applications by Abhijit Kulkarni, Newbury Park, CA US

Abhijit Kulkarni, Thousand Oaks, CA US

Patent application numberDescriptionPublished
20080298612Audio Signal Processing - An audio system for processing two channels of audio input to provide more than two output channels. The input may be conventional stereo material or compressed audio signal data. The audio processing includes separating the input signals into frequency bands and processing the frequency bands according to processes which may differ from band to band. The audio processing includes no processing of L−R signals.12-04-2008
20080304671Audio Signal Processing - An audio system for processing two channels of audio input to provide more than two output channels. The input may be conventional stereo material or compressed audio signal data. The audio processing includes separating the input signals into frequency bands and processing the frequency bands according to processes which may differ from band to band. The audio processing includes no processing of L-R signals.12-11-2008

Abhijit Kulkarni, Newton, MA US

Patent application numberDescriptionPublished
20120057713AUTOMATIC AUDIO SYSTEM EQUALIZING - An automated process for equalizing an audio system and an apparatus for implementing the process. An audio system includes a microphone unit, for receiving the sound waves radiated from a plurality of speakers, acoustic measuring circuitry, for calculating frequency response measurements; a memory, for storing characteristic data of the loudspeaker units and further for storing the frequency response measurements; and equalization calculation circuitry, for calculating an equalization pattern responsive to the digital data and responsive to the characteristic data of the plurality of loudspeaker units. Also described is an automated equalizing system including a acoustic measuring circuitry including a microphone for measuring frequency response at a plurality of locations; a memory, for storing the frequency responses at the plurality of locations; and equalization calculation circuitry, for calculating, from the frequency responses, an optimized equalization pattern.03-08-2012
20120057724AUTOMATIC AUDIO SYSTEM EQUALIZING - An automated process for equalizing an audio system and an apparatus for implementing the process. An audio system includes a microphone unit, for receiving the sound waves radiated from a plurality of speakers, acoustic measuring circuitry, for calculating frequency response measurements; a memory, for storing characteristic data of the loudspeaker units and further for storing the frequency response measurements; and equalization calculation circuitry, for calculating an equalization pattern responsive to the digital data and responsive to the characteristic data of the plurality of loudspeaker units. Also described is an automated equalizing system including a acoustic measuring circuitry including a microphone for measuring frequency response at a plurality of locations; a memory, for storing the frequency responses at the plurality of locations; and equalization calculation circuitry, for calculating, from the frequency responses, an optimized equalization pattern.03-08-2012