Some adaptive signal processing applications, such as wideband active noise control and acoustic echo cancellation, involve adaptive filters with hundreds of taps. The computational burden associated with these long adaptive filters precludes their use for many low-cost applications. In addition, adaptive filters with many taps may also suffer from slow convergence, especially if the reference signal spectrum has a large dynamic range. Subband techniques have been previously developed for adaptive filters to solve these problems. However, the conventional approach is ruled out for many applications because delay is introduced into the signal path. The paper presents a new type of subband adaptive filter architecture in which the adaptive weights are computed in subbands, but collectively transformed into an equivalent set of wideband filter coefficients. In this manner, signal path delay is avoided while retaining the computational and convergence speed advantages of subband processing. An additional benefit accrues through a significant reduction of aliasing effects. An example of the general technique is presented for a 32-subband design using a polyphase FFT implementation. For this example, the number of multiplies required are only about one-third that of a conventional full band design with zero delay, and only slightly greater than that of a conventional subband design with 16 ms delay. >

A delayless subband adaptive filter architecture

An echo canceller and method for cancelling in a return channel signal an echoed receive channel signal where the echoed receive channel signal is combined by an echo channel with an input return channel signal. The echo canceller has a first filter which generates first filter coefficients, generates a first echo estimate signal with the first filter coefficients, and updates the first filter coefficients in response to a first filter control signal. A first summer subtracts the first echo estimate signal from a combined return channel and echo receive channel signal to generate a first echo residual signal. A second filter generates second filter coefficients, generates a second echo estimate signal with the second filter coefficients, and updates the second filter coefficients in response to a second filter control signal. A second summer subtracts the second echo estimate signal from the combined signal to generate a second echo residual signal, and provides upon the return channel the second echo residual signal. A control unit determines from the receive channel signal, the combined signal, and the first and second echo residual signals, one of a plurality of control states wherein a first control state is indicative of a receive channel signal above a first predetermined energy level, wherein when the control unit is in the first control state it generates the first control signal and generates the second control signal when at least one of a first energy ratio of the first echo residual signal and the combined signal and a second energy ratio of the second echo residual signal and the combined signal exceed a predetermined level.

Network echo canceller

An active vibration isolation system (10) includes a plurality of stiff actuators, such as piezoelectric motors (12a-c; 284, 350) which support a small mass (18) inside of a case (300). A passive isolator (20) is interposed between the small mass (18) and the payload mass (Mp). In the instance where the stiff actuators are piezoelectric motors, the sidewalls of the case (300) are used to provide compressive force along the variable length of each of the horizontal piezoelectric motor elements (284, 350) to prevent damage to same. Compensation circuitry (24) is connected between velocity sensors (17) and the stiff actuators to control the variable length thereof in this move-out-of-the-way system (10) and to compensate for resonating modes. Optionally, a payload mass velocity sensor (26) and associated circuitry may be used to provide additional control. Shear decouplers (282a-c; 352, 360) are used in conjunction with the piezoelectric motor elements (12a-c; 284, 350) to minimize the amount of shear force acting on the motor elements.

Stiff actuator active vibration isolation system

A telephone system includes two or more cardioid microphones held together and directed outwardly from a central point. Mixing circuitry and control circuitry combines and analyzes signals from the microphones and selects the signal from one of the microphones or from one of one or more predetermined combinations of microphone signals in order to track a speaker as the speaker moves about a room or as various speakers situated about the room speak then fall silent. Visual indicators, in the form of light emitting diodes (LEDs) are evenly spaced around the perimeter of a circle concentric with the microphone array. Mixing circuitry produces ten combination signals, A+B, A+C, B+C, A+B+C, A-B, B-C, A-C, A-0.5(B+C), B-0.5(A+C), and C-0.5(B+A), with the "listening beam" formed by combinations, such as A-0.5(B+C), that involve the subtraction of signals, generally being more narrowly directed than beams formed by combinations, such as A+B, that involve only the addition of signals. An omnidirectional combination A+B+C is employed when active speakers are widely scattered throughout the room. Weighting factors are employed in a known manner to provide unity gain output. Control circuitry selects the signal from the microphone or from one of the predetermined microphone combinations, based generally on the energy level of the signal, and employs the selected signal as the output signal. The control circuitry also operates to limit dithering between microphones and, by analyzing the beam selection pattern, may switch to a broader coverage pattern, rather than switching between two narrower beams that each covers one of the speakers.

Teleconferencing system which selects and optimal acoustic reception pattern from multiple patterns having different shapes and sizes

Method and apparatus for time-scaling and/or pitch shifting by discarding and/or repeating segments of a signal (302) The signal (302) is stored as a series of samples on a memory (124) where it is readable by one or more read pointers (304, 306) Periodicity of segments of the signal (302) is determined by evaluating normalized cross-correlation over a range of possible periods Transients are detected by monitoring changes in rms signal value To achieve time compression or time stretching, a segment is skipped/discarded whenever a maximum time-discrepancy between the current output and an ideal output is reached or a high periodicity is detected, a jump of the optimal length would not make this time discrepancy too high, and no transient is present in the segment to be skipped/discarded

Time-domain time/pitch scaling of speech or audio signals, with transient handling

A digital echo canceller [1] with a good protection against double-talk comprises a first filter [9] for the echo cancellation proper, a second adaptive filter [11] for making continuously adapted filter coefficients available to the first filter [9] and gate means [15] which will only apply the adapted filter coefficients to the first filter [9] if certain conditions are satisfied. In the echo canceller [1] a specific filter combination is used, in which the first filter [9] is a time-domain programmable filter (TDPF) and the second filter [11] is a frequency-domain block-adaptive (FDAF), and further means [20,20(1)] are provided that are arranged in cascade with the gate means [15] for transforming the frequency-domain filter coefficients [W(p;m)] of the FDAF [11] into time-domain filter coefficients [w(i;m)] for the TDPF [9]. These measures result in the advantages of a negligible delay in the echo cancellation by the TDPF [9], and of the possibility to improve in a simple manner the convergence behavior of the FDAF [11] and considerably reduce its computational complexity.

De Laar Franciscus Antonius Maria Van

Papers

Digital echo canceller