Texas Instruments Corporation

Edwin Randolph Cole

Nitish Krishna Murthy

David R Thomas

Jonathan D Courtney

Yeshwant K Muthusamy

Anand Ganesh Dabak

Baboo V Gowreesunker

Yinong Ding

Sourabh Ravindran

Takahiro Unno

Supriyo Palit

Keya R Pandia

Jorge Francisco Arbona Miskimen

From a first microphone, first microphone signals are received that represent first sound waves. From a second microphone, second microphone signals are received that represent second sound waves. In response to the first microphone signals, analog processing is performed to estimate noise in the first sound waves, and first analog signals are generated for cancelling at least some of the estimated noise in the first sound waves. In response to the second microphone signals, digital processing is performed to estimate noise in the second sound waves, and digital information is generated for cancelling at least some of the estimated noise in the second sound waves. The digital information is converted into second analog signals that represent the digital information. The first and second analog signals are combined into third analog signals for cancelling at least some of the estimated noise in the first and second sound waves.

Method and system for hybrid noise cancellation

A method of environmental noise compensation a speech audio signal is provided that includes estimating a fast audio energy level and a slow audio energy level in an audio environment, wherein the speech audio signal is not part of the audio environment, and applying a gain to the speech audio signal to generate an environment compensated speech audio signal, wherein the gain is updated based on the estimated slow audio energy level when the estimated fast audio energy level is not indicative of an audio event in the audio environment and the estimated gain is not updated when the estimated fast audio energy level is indicative an audio event in the audio environment.

Method and apparatus for environmental noise compensation by determining a presence or an absence of an audio event

A heart monitor includes a single chest accelerometer (), an analog signal conditioning and sampling section () responsive to said accelerometer to produce a digital signal substantially representing acceleration, and a digital processor () operable to filter the acceleration signal into a signal affected by body motion and to cancel the body motion signal from the acceleration signal, thereby to produce an acceleration-based cardiac-related signal. Other processes and electronic systems are also disclosed.

Calculating heart rate from acceleration signals containing cardiac activity signals

Microphone signals are received from a microphone. The microphone signals represent first sound waves. A determination is made about a type of noise that likely exists in the first sound waves. In response to the type of noise, cancellation signals are generated by filtering the microphone signals with at least one of: a first filter in response to the type of noise indicating that a first type of noise likely exists in the first sound waves; and a second filter in response to the type of noise indicating that a second type of noise likely exists in the first sound waves. In response to the cancellation signals, second sound waves are output from a speaker for cancelling at least some noise in the first sound waves.

Method and system for active noise cancellation according to a type of noise

A method of dominant speech extraction is provided that includes acquiring a primary audio signal from a microphone and at least one additional audio signal from at least one additional microphone, wherein the acquired audio signals include speech and noise, decomposing each acquired audio signal into a low frequency sub-band signal and a high frequency sub-band signal, applying speech suppression beamforming to the low frequency sub-band signals to generate a reference channel having an estimate of noise in the low frequency sub-band signals, applying noise cancellation to the low frequency sub-band signal of the primary audio signal using the reference channel to generate a first signal having a low frequency estimate of the speech, applying noise suppression beamforming to the high frequency sub-band signals to generate a second signal having a high frequency estimate of the speech, and combining the first and second signals to generate a full-band audio signal.

Dominant speech extraction in the presence of diffused and directional noise sources

A method and apparatus for active noise canceling. The method includes retrieving an input sample from at least one of a feedback or feedforward microphone digitized through the sigma-delta converter, retrieving the input sample and a related filter, wherein the filter is customized to the particular headset, outputting a filtered signal through a speaker without any interpolation and reducing order of CIC filters, and outputting a response sharply tapered down.

Method and apparatus for a configurable active noise canceller

A server for a merchant computer system, the server comprising: a file store for storing a range of audio/video products in respective product files; a dialogue unit having a network connection and operable to invite and receive a client selection from among the products via the network connection; a product reader for reading the product files to generate a digital audio/video signal; a digital signal processing unit having an input connectable to receive the digital audio/video signal from the product reader, a processing core operable to apply a defined level of content degradation to the digital audio/video signal, and an output connected to output the degraded digital audio/video signal from the processing core to the network connection. It is therefore possible for a content provider to change the characteristics of an audio or video data stream supplied over a network to a potential purchaser in a controlled and variable manner. The amount of degradation is sufficient to enable a potential purchaser to appreciate the characteristics of the audio or video product, whilst reducing the perceived quality.

Supply of digital audio and video products

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.

Robust detection for embedded signaling

A system and method for providing a controllable virtual environment includes a computer ( ) with processor and a display coupled to the processor to display 2-D or 3-D virtual environment objects. Speech grammars are stored as attributes of the virtual environment objects. Voice commands are recognized by a speech recognizer ( ) and microphone ( ) coupled to the processor whereby the voice commands are used to manipulate the virtual environment objects on the display. The system is further made role-dependent whereby the display of virtual environment objects and grammar is dependent on the role of the user.

System and method for advanced interfaces for virtual environments

A system and method for providing a controllable virtual environment includes a computer (,) with processor and a display coupled to the processor to display 2-D or 3-D virtual environment objects. Speech grammars are stored as attributes of the virtual environment objects. Voice commands are recognized by a speech recognizer (,) and microphone (,) coupled to the processor whereby the voice commands are used to manipulate the virtual environment objects on the display. The system is further made role-dependent whereby the display of virtual environment objects and grammar is dependent on the role of the user.

Growing community of inventors

Dallas, TX, United States of America

Edwin Randolph Cole