| Patent application number | Description | Published |
|---|
| 20090135976 | RESOLVING BUFFER UNDERFLOW/OVERFLOW IN A DIGITAL SYSTEM - In a digital system with more than one clock source, lack of synchronization between the clock sources may cause overflow or underflow in sample buffers, also called sample slipping. Sample slipping may lead to undesirable artifacts in the processed signal due to discontinuities introduced by the addition or removal of extra samples. To smooth out discontinuities caused by sample slipping, samples are filtered to when a buffer overflow condition occurs, and the samples are interpolated to produce additional samples when a buffer underflow condition occurs. The interpolated samples may also be filtered. The filtering and interpolation operations can be readily implemented without adding significant burden to the computational complexity of a real-time digital system. | 05-28-2009 |
| 20090190769 | SOUND QUALITY BY INTELLIGENTLY SELECTING BETWEEN SIGNALS FROM A PLURALITY OF MICROPHONES - Sound signal reception is improved by utilizing a plurality of microphones to capture sound signals which are then weighed to dynamically adjust signal quality. A first sound signal and a second sound signal are obtained from first and second microphones, respectively, where the first and second sound signals originate from one or more sound sources. A first signal characteristic (e.g., signal power, signal signal-to-noise ratio, etc.) is obtained for the first sound signal and a second signal characteristic is obtained for the second sound signal. The first and second sound signals are weighed or scaled based on their respective first and second signal characteristics. The weighed first and second sound signals are then combined to obtain an output sound signal. | 07-30-2009 |
| 20090190774 | ENHANCED BLIND SOURCE SEPARATION ALGORITHM FOR HIGHLY CORRELATED MIXTURES - An enhanced blind source separation technique is provided to improve separation of highly correlated signal mixtures. A beamforming algorithm is used to precondition correlated first and second input signals in order to avoid indeterminacy problems typically associated with blind source separation. The beamforming algorithm may apply spatial filters to the first signal and second signal in order to amplify signals from a first direction while attenuating signals from other directions. Such directionality may serve to amplify a desired speech signal in the first signal and attenuate the desired speech signal from the second signal. Blind source separation is then performed on the beamformer output signals to separate the desired speech signal and the ambient noise and reconstruct an estimate of the desired speech signal. To enhance the operation of the beamformer and/or blind source separation, calibration may be performed at one or more stages. | 07-30-2009 |
| 20090196429 | SIGNALING MICROPHONE COVERING TO THE USER - A mechanism is provided that monitors secondary microphone signals, in a multi-microphone mobile device, to warn the user if one or more secondary microphones are covered while the mobile device is in use. In one example, smoothly averaged power estimates of the secondary microphones may be computed and compared against the noise floor estimate of a primary microphone. Microphone covering detection may be made by comparing the secondary microphone smooth power estimates to the noise floor estimate for the primary microphone. In another example, the noise floor estimates for the primary and secondary microphone signals may be compared to the difference in the sensitivity of the first and second microphones to determine if the secondary microphone is covered. Once detection is made, a warning signal may be generated and issued to the user. | 08-06-2009 |
| 20090238369 | SYSTEMS AND METHODS FOR DETECTING WIND NOISE USING MULTIPLE AUDIO SOURCES - A method for detecting wind noise is described. At least two audio signals are received. The at least two audio signals are filtered to reduce higher frequencies and to reduce lower frequencies to provide at least two filtered audio signals. The cross correlation of the at least two filtered audio signals is computed for multiple delays. A maximum cross correlation is determined from the cross correlations computed for the multiple delays. Wind noise is detected by comparing the maximum cross correlation with a threshold. | 09-24-2009 |
| 20090238377 | SPEECH ENHANCEMENT USING MULTIPLE MICROPHONES ON MULTIPLE DEVICES - Signal processing solutions take advantage of microphones located on different devices and improve the quality of transmitted voice signals in a communication system. With usage of various devices such as Bluetooth headsets, wired headsets and the like in conjunction with mobile handsets, multiple microphones located on different devices are exploited for improving performance and/or voice quality in a communication system. Audio signals are recorded by microphones on different devices and processed to produce various benefits, such as improved voice quality, background noise reduction, voice activity detection and the like. | 09-24-2009 |
| 20090240495 | METHODS AND APPARATUS FOR SUPPRESSING AMBIENT NOISE USING MULTIPLE AUDIO SIGNALS - A method for suppressing ambient noise using multiple audio signals may include providing at least two audio signals captured by at least two electro-acoustic transducers. The at least two audio signals may include desired audio and ambient noise. The method may also include performing beamforming on the at least two audio signals in order to obtain a desired audio reference signal that is separate from a noise reference signal. | 09-24-2009 |
| 20100094625 | METHODS AND APPARATUS FOR NOISE ESTIMATION - A system and method are disclosed for noise level/spectrum estimation and speech activity detection. Some embodiments include a probabilistic model to estimate noise level and subsequently detect the presence of speech. These embodiments outperform standard voice activity detectors (VADs), producing improved detection in a variety of noisy environments. | 04-15-2010 |
| 20110081026 | SUPPRESSING NOISE IN AN AUDIO SIGNAL - An electronic device for suppressing noise in an audio signal is described. The electronic device includes a processor and instructions stored in memory. The electronic device receives an input audio signal and computes an overall noise estimate based on a stationary noise estimate, a non-stationary noise estimate and an excess noise estimate. The electronic device also computes an adaptive factor based on an input Signal-to-Noise Ratio (SNR) and one or more SNR limits. A set of gains is also computed using a spectral expansion gain function. The spectral expansion gain function is based on the overall noise estimate and the adaptive factor. The electronic device also applies the set of gains to the input audio signal to produce a noise-suppressed audio signal and provides the noise-suppressed audio signal. | 04-07-2011 |
