Abstract: The force on the electrodes of an electrostatic field is used so that lateral tensile or compressive forces result which can deform a deformable element or can strongly deflect a deflectable structure. For this, a micromechanical device includes, apart from an electrode and a deformable element, an insulating spacer layer via which the electrode is fixed to the deformable element, wherein the insulating spacer layer is structured into several spaced-apart segments along a lateral direction, so that by applying an electric voltage between the electrode and the deformable element lateral tensile or compressive forces bending the deformable element along the lateral direction result. Thereby, the problem that normally accompanies electrostatic drives, namely the problem of the pull-in effect, is overcome. The deflection of the deformable element can be much larger than the gaps of the two electrodes, i.e. the above-mentioned electrode and the deformable element. A usage as a sensor is also possible.

Abstract: A signal processing system enhances communication. When an audio signal is detected, an echo component of the detected audio signal may be estimated. A near party communication device may receive an audio signal from a remote party communication device. A characteristic of the received audio signal may be adjusted based on the echo component of the detected audio signal.

Abstract: A vehicle communication system detects the presence of a passenger wearable communication device. The system receives audio signals from multiple sources inside or outside of a vehicle. The system processes the signals before routing the signals to multiple destinations. The destinations may include wearable personal communication devices, front and/or rear speakers, and/or a remote mobile device.

Abstract: A calibration device automatically calibrates a telephone hands free system. The device includes a signal processor that processes a received telephone signal that may be converted into audible sound through the hands free system. A first telephone and a second telephone communicate with the signal processor. The first telephone is programmed to transmit a predetermined test signal to the second telephone through a wireless or physical medium. A comparator compares the received test signal to a predetermined reference signal at a hands free interface. The comparison determines a deviation from the reference signal. An adaptor modifies the processing of the digitized aural information based on the deviation.

Abstract: A system estimates the spectral noise power density of an audio signal includes a spectral noise power density estimation unit, a correction term processor, and a combination processor. The spectral noise power density estimation unit may provide a first estimate of the spectral noise power density of the audio signal. The correction term processor may provide a time dependent correction term based, at least in part, on a spectral noise power density estimation error of the actual spectral noise power density. The correction term may be determined so that the spectral noise power density estimation error is reduced. The combination processor may combine the first estimate with the correction term to obtain a second estimate of the spectral noise power density that may be used for subsequent signal processing to enhance a desired signal component of the audio signal.

Abstract: A signal processing system enhances an audio signal. The audio signal is divided into audio sub-band signals. Some audio sub-band signals are excised. Other audio sub-band signals are processed to obtain enhanced audio sub-band signals. At least a portion of the excised audio sub-band signals are reconstructed. The reconstructed audio sub-band signals are synthesized with the enhanced audio sub-band signals to form an enhanced audio signal.

Abstract: A signal processing system enhances an audio signal. The audio signal is divided into audio sub-band signals. Some audio sub-band signals are excised. Other audio sub-band signals are processed to obtain enhanced audio sub-band signals. At least a portion of the excised audio sub-band signals are reconstructed. The reconstructed audio sub-band signals are synthesized with the enhanced audio sub-band signals to form an enhanced audio signal.

Abstract: A system locates a speaker in a room containing a loudspeaker and a microphone array. The loudspeaker transmits a sound that is partly reflected by a speaker. The microphone array detects the reflected sound and converts the sound into a microphone array, the speaker's distance from the microphone array, or both, based on the characteristics of the microphone signals.

Abstract: A speech dialog system interfaces a user to a computer. The system includes a signal pre-processor that processes a speech input to generate an enhanced signal and an analysis signal. A speech recognition unit may generate a recognition result based on the enhanced signal. A control unit may manage an output unit or an external device based on the information within the analysis signal.

Abstract: The present invention relates to a vehicle communication system comprising a plurality of microphones adapted to detect speech signals of different vehicle passengers, a mixer combining the audio signal components of the different microphones to a resulting speech output signal, a weighting unit determining the weighting of the audio signal components for the resulting speech output signal, where the weighting unit determines the weighting of the signal components based upon non-acoustical information about the presence of a vehicle passenger.

Abstract: A system and method for compensating audio signal components is disclosed. The method includes detecting, by a microphone, a sound signal. The sound signal comprises audio signal components resulting from reproducing an audio signal of an audio source and speech signal components corresponding to a speech signal from a person. The sound signal is filtered to whiten the sound signal. The audio signal components in the whitened sound signal are then compensated. The whitening of the compensated sound signal is removed. The filtering of the audio signal is performed using at least two filters in an alternating way, each filter using time-dependent filter coefficients.

Abstract: A filter arrangement, for cleaning gases contaminated with particles, has at least one filter unit with an exterior and an interior filter (3.1, 3.2), wherein the exterior and interior filters can be moved relative to each other, thereby exposing an opening (3.4) in the bottom regions thereof. This allows cleaning of filter units at the site of the filter units without disassembly of the same (cleaning in place—CIP). A method for cleaning is provided in that the region of the filter unit on the clean gas side including introducing a fluid sprayed via a cleaning lance protruding into the clean gas region while exposing a gap (3.4) in the bottom region of the exterior and interior filters.

Abstract: A system locates a speaker in a room containing a loudspeaker and a microphone array. The loudspeaker transmits a sound that is partly reflected by a speaker. The microphone array detects the reflected sound and converts the sound into a microphone signal. A processor determines the speaker's direction relative to the microphone array, the speaker's distance from the microphone array, or both, based on the characteristics of the microphone signals.

Abstract: An apparatus processes an acoustic input signal to provide an output signal with reduced noise. The apparatus weights the input signal based on a frequency-dependent weighting function. A frequency-dependent threshold function bounds the weighting function from below.

Abstract: An echo reduction system includes a signal analysis filter that converts an input into sub-band signals. A signal down-sampling circuit down-samples the sub-band signals at a first down-sampling rate. An echo analysis filter converts a loudspeaker signal into echo sub-band signal that are further processed by an echo down-sampling circuit. The circuit down-samples the echo sub-band signals at a second down-sampling rate to generate down-sampled echo sub-band signals. An echo compensation filter folds the down-sampled echo sub-band signals with an estimated impulse response of a loudspeaker-room-input system. A second echo down-sampling circuit down-samples the folded down-sampled echo sub-band signals at a third down-sampling rate to generate estimated echo sub-band signals. The first down-sampling rate is equal to the product of the second and third down-sampling rates.

Abstract: A codebook spectral envelope may be used to extend the bandwidth of a bandwidth limited signal. A system includes codebooks that list codebook spectral envelopes. A codebook spectral envelope may be selected based on a characteristic of the spectral envelope of the bandwidth limited signal. Modifications of selected codebook spectral envelopes may generate a bandwidth extension signal that may be added to the bandwidth limited signal to improve the quality of the signal.

Abstract: An audio enhancement refines a short-time spectrum. The refinement may reduce overlap between audio sub-bands. The sub-bands are transformed into sub-band short-time spectra. A portion of the spectra are time-delayed. The sub-band short-time spectrum and the time-delayed portion are filtered to obtain a refined sub-band short-time spectrum. The refined spectrum improves audio processing.

Abstract: A feedback reduction system adaptively processes a microphone signal. An adaptive feedback suppression filter circuit processes the microphone signal, and an adaptive feedback compensation filter circuit further processes the microphone signal. The adaptive feedback suppression filter circuit processes the signal based upon output from the adaptive filter compensation filter circuit.

Abstract: A system enhances speech by detecting a speaker's utterance through a first microphone positioned a first distance from a source of interference. A second microphone may detect the speaker's utterance at a different position. A monitoring device may estimate the power level of a first microphone signal. A synthesizer may synthesize part of the first microphone signal by processing the second microphone signal. The synthesis may occur when power level is below a predetermined level.

Abstract: A dereverberation and feedback compensation system reduces the echo received by a first audio device while reducing the speech feedback received from a second audio device. A decorrelation logic decorrelates audio signals from the first audio device. A first processor generates a noise compensation signal based on the decorrelated audio signals and system determined filter coefficients. The second processor generates an enhanced noise correlation signal based on speech signals of a second audio device and the filter coefficients used by the first processor.