Abstract: A noise suppression device receives data representative of a noise-corrupted signal which contains a speech signal and a noise signal, divides the received data into data frames, and then passes the data frames through a pre-filter to remove a dc-component and the minimum phase aspect of the noise-corrupted signal. The noise suppression device appends adjacent data frames to eliminate boundary discontinuities, and applies fast Fourier transform to the appended data frames. A voice activity detector of the noise suppression device determines if the noise-corrupted signal contains the speech signal based on components in the time domain and the frequency domain. A smoothed Wiener filter of the noise suppression device filters the data frames in the frequency domain using different sizes of a window based on the existence of the speech signal. Filter coefficients used for Wiener filter are smoothed before filtering.

Abstract: A system that reconstructs independent signals from degenerate mixtures estimates independent Auto Regressive (AR) processes from their sum. The system includes an identification system and an estimator. A mixture of two signals is inputted into the system and through the identifying and filtering processes, two estimates of the original signals are outputted. The identification system includes an ARMA identifier, a computation of autocovariance coefficients, an initializer and a gradient descent system. The estimator includes filtering.

Abstract: With a view toward properly combining a fundamental echo with a harmonics echo according to the quality of a signal, the ratio between two frequency components of the fundamental echo is determined (704), and the ratio between two frequency components of the harmonics echo is determined (702). Further, a component ratio between a fundamental component and a harmonics component in an echo receive signal is adjusted based on these two ratios (706, 708).

Abstract: The invention relates to a method of providing a user with information on the operation of the portable device and to a portable device. In the device, such a tone is produced that, due to a tone feature, can be distinguished from background noise. This feature can be tone frequency, duration, volume or moment of time. The device can analyse background noise automatically, and based on this, it adjusts at least one feature of the tone automatically such that the tone can be distinguished from background noise, and the background noise does not mask out the tone. Alternatively, the user himself can adjust the tone frequency or duration in a desired way so that it would be distinguished from background noise more clearly.

Abstract: A method of reducing undesired components from a signal that includes a desired component and undesirable components utilizes an autoregressive model technique. An autoregressive module determines a power spectral density approximation of the signal. An error component of the power spectral density approximation includes the desired component. Portions of the error component having frequencies outside of the expected range of the desired component preferably are filtered so that the result is the desired component with the undesired component removed. The invention is useful, for example, for reducing undesirable noise components from sound signals.

Abstract: The circuit for the adaptive suppression of noise is a component part of a digital hearing aid, consisting of two microphones (1, 2), two AD-converters (3, 4), two compensating filters (5, 6), two retarding elements (7, 8), two subtractors (9, 10), a processing unit (11), a DA-converter (13), an earphone (15) as well as the two filters (17, 18). The method for the adaptive suppression of noise can be implemented with the indicated circuit. The two microphones (1, 2), dependent on their spatial arrangement or their directional characteristics and dependent on the location of the acoustic signal sources, provide two differing electric signals (d1(t), d2(t)), which are digitalized in the two AD-converters (3, 4) and pre-processed together with the two fixed compensation filters (5, 6). Following subsequently are the two filters (17, 18) arranged symmetrically crosswise in forward direction with the adaptive filter coefficients (w1, w2).

Abstract: An audio signal processing device comprises signal supply means to supply over more than one input channel and per input channel over separate frequency subbands domain subchannels coded audio signals.

Abstract: An active noise cancellation aircraft headset system. A speaker is mounted within each earcup of a headset for receiving and acoustically transducing a composite noise cancellation signal. A microphone is also mounted within each earcup for transducing acoustic pressure within the earcup to a corresponding analog error signal. An analog filter receives the analog error signal and inverts it to generate an analog broadband noise cancellation signal. The analog error signal is also provided to an analog to digital converter, which receives the analog microphone error signal and converts it to a digital error signal. A DSP takes the digital error signal and, using an adaptive digital feedback filter, generates a digital tonal noise cancellation signal. A digital to analog converter then converts the digital tonal noise cancellation signal to an analog tonal noise cancellation signal so that it can be combined with the analog broadband noise cancellation signal.

Abstract: A desired acoustic signal is extracted from a noisy environment by generating a signal representative of the desired signal with a processor for a hearing aid device. The processor receives binaural signals from two microphones at different locations. The binaural inputs to the processor are converted from analog to digital format and then submitted to a discrete Fourier transform process to generate discrete spectral signal representations. The spectral signals are delayed by a number of time intervals in a dual delay line to provide a number of intermediate signals, each corresponding to a different position relative to a desired signal source. Location of the noise source is determined and the spectral content of the desired signal is determined from the intermediate signal corresponding to the noise source location. Inverse transformation of the selected intermediate signal followed by digital to analog conversion provides an output signal representative of the desired signal.

Abstract: A method and apparatus for separating signals from instantaneous and convolutive mixtures of signals. A plurality of sensors or detectors detect signals generated by a plurality of signal generating sources. The detected signals are processed in time blocks to find a separating filter, which when applied to the detected signals produces output signals that are statistically independent.

Abstract: A dual-processing interference cancelling system and method for processing a broadband input in a computationally efficient manner. Dual processing divides the input into higher and lower frequency bands and applies adaptive filter processing to the lower frequency band while applying non-adaptive filter processing to the higher frequency band. Various embodiments are shown including those based on sub-bands, broadband processing with band-limited adaptation, and broadband processing with an external main-channel generator.