TI window compression/expansion method

Data processing: speech signal processing – linguistics – language – Audio signal bandwidth compression or expansion

Reexamination Certificate

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Details

C704S200100, C704S229000

Reexamination Certificate

active

06748363

ABSTRACT:

TECHNICAL FIELD OF INVENTION
The present invention relates to a technology for compression/expansion methods that can be applied to all transform based audio coders, or any audio coder using a windowing stage.
BACKGROUND OF THE INVENTION
In 1991, the Motion Pictures Experts Group (MPEG), a group developed under the International Standards Organization(ISO), created an audio video system standard MPEG-1. MPEG-1 had three ‘layers’, the first two layers 1 and 2 were more simple audio coding and decoding algorithms, whereas the third layer, named MP3, was a much more complex audio coding and decoding system which just recently, has received alot of notoriety. MPEG-1 is a mono channel, stereo standard which operates at a 32-48 khz sampling rate. Around 1994, MPEG-2 was created which comprised of the same three layers, but this time was multichannel, or otherwise named (5.1) for 5 directions and one sub-woofer: Center, Left, Right, Left Surroundsound, Right Surroundsound and Low Frequency Exciter(LFE). MPEG-2 also operated at a much lower sampling rate, 12-32 kHz versus the 32-48 kHz of MPEG-1. In addition, MPEG-2 was backward compatible (BAC) with MPEG-1, which meant that MPEG-2 could play all MPEG-1 data streams.
More recently, around 1997, the thinking was that the audio coding and decoding standard could be made much more optimum if the standard did not have to be backward compatible (BAC). As a result, the audio coding and decoding standard, MPEG-2 non-backward compatible (NBC) was developed and it, as the name implies, was not backward compatible with the previous standards, MPEG-1 and MPEG-2. This standard was not commercially desirable(because of the ‘non-backward compatible’ in the name) and so was changed to MPEG-2 Advanced Audio Coding (MC). MPEG-2 AAC is a multichannel system of up to 48 channels(foreign language applications are now enabled) and has a mono equivalence, if comparing against a mono standard like MP3(the third layer of MPEG-1) of 64 kbps versus MP3@64 kbps.
In any transform based audio decoder, there is a final “window-overlap-add” stage that converts the decompressed data into time domain output samples. The main data requirements to implement this stage are an input buffer containing the current decompressed data, a state buffer containing the previous decompressed data, and a constant table storing the “window” coefficients. These window-tables directly effect the quality of the output signal, and in order to keep this quality high, the tables require a significant amount of storage, about 2-4 k. In addition many of the audio compression algorithms provide support for multiple window shapes, so the storage requirements can increase to 4-8 k or more. In embedded applications, where memory is very limited, reducing the size of these tables is a necessity.
SUMMARY OF THE INVENTION
According to the present invention, there is developed a proprietary technology for compressing the window tables of audio coders to ⅛ their original size (or less) without any loss of quality. This technology can be applied to all transform based audio coders, or any audio coder that uses a windowing stage. The novel technique for reducing storage requirements for the window tables of audio coders is based on multiple differentiation. Since the difference between any two adjacent window samples is relatively small, it is more efficient to store this difference. This technique can be carried out several more times, until the returns get smaller, and the computational requirements to “undo” the compression go up. The optimum number of times to differentiate is dependent on the particular application and the window shape.


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patent: 6487535 (2002-11-01), Smyth et al.

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