Various embodiments reduce noise within a particular environment, while isolating and capturing speech in a manner that allows operation within an otherwise noisy environment. In one embodiment, an array of one or more microphones is used to selectively eliminate noise emanating from known, generally fixed locations, and pass signals from a pre-specified region or regions with reduced distortion.

Noise Reduction Systems and Methods for Voice Applications

Techniques to reduce distortion in acoustic signals in mobile computing devices are described. For example, a mobile computing device may comprise a speaker operative to receive a first signal and output a second signal. The mobile computing device may further comprise a first microphone operative to receive the second signal and a second microphone operative to receive a third signal. An echo canceller may be coupled to the first microphone and the second microphone and may be operative to compare the second signal and the third signal and reduce distortion in the third signal based on the comparison. Other embodiments are described and claimed.

Method and apparatus to reduce non-linear distortion

Systems and methods for providing single microphone noise suppression fallback are provided. In exemplary embodiments, primary and secondary acoustic signals are received. A single microphone noise estimate may be generated based on the primary acoustic signal, while a dual microphone noise estimate may be generated based on the primary and secondary acoustic signals. A combined noise estimate based on the single and dual microphone noise estimates is then determined. Using the combined noise estimate, a gain mask may be generated and applied to the primary acoustic signal to generate a noise suppressed signal. Subsequently, the noise suppressed signal may be output.

System and method for providing single microphone noise suppression fallback

Systems and methods for utilizing inter-microphone level differences (ILD) to attenuate noise and enhance speech are provided. In exemplary embodiments, primary and secondary acoustic signals are received by omni-directional microphones, and converted into primary and secondary electric signals. A differential microphone array module processes the electric signals to determine a cardioid primary signal and a cardioid secondary signal. The cardioid signals are filtered through a frequency analysis module which takes the signals and mimics a cochlea implementation (i.e., cochlear domain). Energy levels of the signals are then computed, and the results are processed by an ILD module using a non-linear combination to obtain the ILD. In exemplary embodiments, the non-linear combination comprises dividing the energy level associated with the primary microphone by the energy level associated with the secondary microphone. The ILD is utilized by a noise reduction system to enhance the speech of the primary acoustic signal.

System and method for utilizing omni-directional microphones for speech enhancement

A spatial processing stereo system (“SPSS”) that receives audio signals and a limited number of user input parameters associated with the spatial attributes of a room, such as “room size”, “stage distance”, and “stage width”. The input parameters are used to define a listening room and generate coefficients, room impulse responses, and scaling factors that are used generate additional surround signals.

Spatial processing stereo system

A microphone array apparatus includes a microphone array including microphones, one of the microphones being a reference microphone, filters receiving output signals of the microphones, and a filter coefficient calculator which receives the output signals of the microphones, a noise and a residual signal obtained by subtracting filtered output signals of the microphones other than the reference microphone from a filtered output signal of the reference microphone and which obtain filter coefficients of the filters in accordance with an evaluation function based on the residual signal.

Microphone array apparatus

To provide a three-dimensional acoustic effect to a listener in a reproduction field, via a headphone in particular, a three-dimensional acoustic apparatus is formed by a linear synthesis filter having filter coefficients that are the linear predictive coefficients obtained by performing a linear predictive analysis on an impulse response which represents the acoustic characteristics to be added to the original signal to achieve this effect. By passing the signal through this acoustic characteristics adding filter, the desired acoustic characteristics are added to the original signal, and by dividing the power spectrum of the impulse response of these acoustic characteristics into critical bandwidths and performing this linear predictive analysis based on impulse signal determined based from power spectrum signals representing the signal sound of each of these critical bandwidths, the filter coefficients of the linear synthesis filter are determined.

Naoshi Matsuo

Papers

Microphone array apparatus

Three-dimensional acoustic processor which uses linear predictive coefficients