Abstract: A polyphonic (e.g. stereo) audioconferencing system, in which input left and right channels are time-aligned by variable delay stages (10a, 10b), controlled by a delay calculator (9) (e.g. by deriving the maximum cross-correlation value), and then summed in an adder (2) and subtracted in subtracter (3) to form sum and difference signals. The sum signal is transmitted in relatively high quality; the difference signal is reconstructed at the decoder by prediction from the sum signal using an adaptive filter (5). The decoder adaptive filter (5) is configured either by received filter coefficients or, using backwards adaptation, from a received residual signal produced by a corresponding adaptive filter (4) in the coder, or both. Preferably, the adaptive filter (4) is a lattice filter, employing a gradient algorithm for coefficient update.

Abstract: A stereo coding system is described in which 7 kHZ bandwidth sum (L+R) and difference (L-R) signals are converted to digital form (11 and 12). The difference signal is then bandlimited (13) to 2 kHZ and downsampled (14) to 4K samples/sec prior to being encoded (16), and the sum signal is encoded (15); both are then multiplexed together (17) prior to transmission. After reception, the signal is demultiplexed (19) and the difference signal is decoded (20) upsampled (21) to its original bit rate and filtered to prevent aliasing (22), and the sum signal is delayed in a digital buffer (28) and decoded (24) prior to both being converted back to analogue form (23 and 25). The delay introduced by the digital buffer (28) equalizes the excess delay introduced into the difference signal.