Abstract: Operation of an adaptive filter includes determining a first filter coefficient and determining a second filter coefficient based on the first filter coefficient, in one embodiment, and in another embodiment includes comparing pseudo-error counts generated for each of several time intervals and setting filter coefficients based on the comparison. Adaptive filter coefficient generating apparatus includes a pseudo-error count generator adapted to receive the output of the adaptive filter and to generate counts of pseudo-errors occurring during successive time intervals, and a filter coefficient generator generating filter coefficients for the adaptive filter in response to the pseudo-error counts.

Abstract: A system for reducing noise in an acoustical signal comprises a sampler (104) for obtaining discrete samples of the acoustical signal, an analog to digital converter (106), and a noise suppression circuit (108). The noise suppression circuit (108) selects a fixed number of samples. These samples are multiplied by a windowing function and the fast Fourier transform is computed to yield transformed windowed signals. A smoothed power estimate and a noise estimate are calculated. The noise estimate and the smoothed power estimate is used to calculate a gain function. A transformed speech signal is obtained by multiplying the gain function with the transformed windowed signal. Then, the inversed fast Fourier transform of the transformed speech signal is added to a portion of the speech signal of a previous frame.

Abstract: Operation of an adaptive filter includes determining a first filter coefficient and determining a second filter coefficient based on the first filter coefficient, in one embodiment, and in another embodiment includes comparing pseudo-error counts generated for each of several time intervals and setting filter coefficients based on the comparison. Adaptive filter coefficient generating apparatus includes a pseudo-error count generator adapted to receive the output of the adaptive filter and to generate counts of pseudo-errors occurring during successive time intervals, and a filter coefficient generator generating filter coefficients for the adaptive filter in response to the pseudo-error counts.

Abstract: A parallel decision feedback equalizer (10) using residue number system (RNS) is provided. The binary input data (12, 22) is converted to the residue number system, in particular the quadratic residue number system for digital video systems, prior to filtering by a plurality of parallel adaptive filters (26). The filtered data is then converted back to the binary system by a converter (18, 34) prior to the decision making operation by a slicer (36). The output from the slicer (36) is also provided as a feedback in the residue number system to be summed with the output from the parallel adaptive filters (26). When the DFE implementation for a system requires a wide input data width and high sampling rates, the parallel decision feedback equalizer (10) provides a speed advantage over conventional approaches.

Abstract: This invention (10, 50) uses an interface (12, 52) to receive a voice input from a user, and a speech recognition unit (18, 54) coupled to the interface (12, 52) to monitor the voice input and recognize a predetermined set of voice commands from the voice input. The speech recognition unit (18, 54) generates a command signal that corresponds to the recognized voice command, which is received by a controller unit (20, 58). The controller unit (20, 58) activates a speech acquisition unit (16, 56) coupled to the controller unit (20, 58) to collect and stop collecting the voice input in response to a control signal generated by the controller unit (20, 58). a memory (24, 56) is provided to store the collected voice input.

Abstract: This invention is a method and apparatus for acoustic echo cancellation. The acoustic echo cancellation employs a modified fast affine projection filter algorithm. The projection order of the affine projection filter algorithm is selected to be small relative to the delay parameter of the transversal filter. This selection of the projection order permits a simplifying approximation in the auto-correlation matrix (the simplified matrix is called a Toeplitz matrix) of the fast affine projection filter algorithm. This simplifying approximation reduces the computational complexity of the filter algorithm without great adverse change in the convergence rate or in the residual. This modified fast affine projection filter algorithm may be practiced on a programmed digital signal processor.

Abstract: A residual echo suppression system suppresses the remaining echo signal not cancelled by an echo canceller. The residual echo suppression system replaces the remaining echo signal by reshaping the spectrum of the signal so that the spectrum shape matches the background noise spectrum. The resultant system is useful in hands-free telephones and especially in hands-free cellular telephones for use in automobiles.

Abstract: A conditioning circuit for use with a microphone 224 placed near a loudspeaker 206 has microphone-in 225 and speaker-line 208 input terminals, an echo canceller circuit (212, 216, 220, 222), a noise-suppresser circuit 230, and a synthesis filter 234 coupling noise-suppresser circuit 230 to a microphone-out output terminal 236. Microphone-in 225 and speaker-line 108 input terminals respectively receive microphone and speaker signals. The echo canceller circuit is coupled between microphone-in 225 and speaker-line 108 input terminals and produces a subband reduced-echo microphone signal by (i) transforming the microphone signal into a subband microphone signal and the speaker signal into a filtered subband speaker signal, and (ii) subband subtracting the filtered subband speaker signal from the subband microphone signal.

Abstract: An acoustic echo canceler (10) includes an adaptive filter circuit (12) and a near-end speech detection circuit (14). The adaptive filter circuit (12) includes an adaptive filter that may be of a single or dual filter structure for generating an e(t) acoustic echo canceler output signal (26) in response to an s(t) far-end speech signal (20) and an x(t) send-in signal (28). The near-end detection circuit (14) receives the s(t) far-end speech signal (20) and the x(t) send-in signal (28) at a high pass filter (32) and a high pass filter (36), respectively. The filtered s(t) far-end speech signal (20) and the filtered x(t) send-in signal (28) are averaged over a selected sampling period by a far-end sampling circuit (34) and a send-in sampling circuit (38). A divider circuit (40) generates a .gamma.(t) acoustic echo path gain/loss signal (42) in response to the filtered and averaged s(t) far-end speech signal (20) and x(t) send-in signal (28). A near-end detector (44) compares the .gamma.