Signal matrixing for directional reproduction of sound
Adaptive predictive processing system
Signal processing and synthesizing method and apparatus
Multi-state speech encoder and decoder
Method for enhancing the quality of coded speech
Improving sub-band coding of speech at low bit rates by adding residual speech energy signals to sub-bands
Method for improving speech quality in code excited linear predictive speech coding
Low-bit-rate speech coder using LPC data reduction processing
Perceptual coding of audio signals
Methods and apparatus for reconstructing non-quantized adaptively transformed voice signals
ApplicationNo. 109479 filed on 08/20/1993
US Classes:704/220, Analysis by synthesis704/201, For storage or transmission704/258Synthesis
ExaminersPrimary: MacDonald, Allen R.
Assistant: Hafiz, Tariq R.
Attorney, Agent or Firm
International ClassG10L 009/00
Foreign Application Priority Data1989-06-15 GB
AbstractA 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. The complexity of the adaptive filter (4) is reduced by pre-whitening, in the encoder, both the sum and difference signals using corresponding whitening filters (14a, 14b) derived from the sum channel.