[Flac-dev] Re: Lossless AMI ADPCM

[Juhana Sadeharju]

>average for audio is 2.75, and average for audio/speech is 3.17.

There were another paper on audio compression:
 ftp://ftp.funet.fi/pub/sci/audio/devel/newpapers/2283-2294.pdf

It compares to Shorten and seems to give more meaningful results.
But the audio samples there too are quite different from music;
they are instrument samples.

It now looks like the first paper gave too strong hopes to me.

 -*-

Some time ago I suggested that FLAC could include a codebook based
compressor: the codebook could be as large as 650 Mbytes (one CD)
and would be used in compressing many audiofiles.

For Vorbis I suggested the following simple method: every 16th sample
would be stored to compressed file without modification. The 16 samples
residue blocks are then vector quantized to the codebook. The residue
blocks would start from the value 0 and end to 0. The codes would have
15 samples. The reference to the codebook would be 32 bit integer which
in total would mean 1:4 compression. I have no idea how well it would
perform. The algorithm is linear but would some kind of high order
system work better?

Back to FLAC: would the final error signal squeeze enough so that
at least 1:3 total compression ratio is obtained?

Well, at least I believe that using 650 MBytes external data would
help to compress the audio further, but I have no idea how much.

Regards,

Juhana