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Abstract

An efficient method for compressing audio and other sampled data signals without loss, or with a controlled amount of loss, is described. The compression apparatus contains a subset selector, an approximator, an adder, two derivative encoders, a header encoder, and a compressed block formatter. The decompression apparatus contains a compressed block parser, a header decoder, two integration decoders, an approximator, and an adder. The compressor first divides each block of input samples into a first subset and a second subset. The approximator uses the first subset samples to approximate the second subset samples. An error signal is created by subtracting the approximated second subset samples from the actual second subset samples. The first subset samples and error signal are separately encoded by the derivative encoders, which select the signal's derivative that requires the least amount of storage for a block floating point representation. A compressed block formatter combines the compression control parameters, encoded subset array, and encoded error array into a compressed block. The decompression apparatus first parses the compressed block into a header, an encoded first subset array, and an encoded error array. The header decoder recovers the compression control parameters from the header. Using the compression control parameters, the integration decoders reconstruct the first subset and error arrays from their block floating point representations. The approximator uses the first subset samples to approximate the original second subset samples. The adder combines the subset samples, the error samples, and the approximated second subset samples to identically re-create the original, uncompressed signal. An indexing method is described which allows random access to specific uncompressed samples within the stream of compressed blocks.

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