Re: [LAD] Ardour session archiver (Was: new lossless/lossy audio compressor)

On Wed, Aug 01, 2007 at 02:12:41PM -0400, Paul Winkler wrote: seems to I reckon there is more to it than that. I'd say any AC signal centered on 0 uses at least all the bits available in the mantissa. Clearly one can not be using all 32 bits if one's signal does not have 2**32 possible values. I can't tell you off the top of my head how many distinct values can represented within -1 to +1 with a 32 bit float, but since it's less than the full range I can tell you it's more than 2**24 and less than 2**32. True, a bigger range may not result in a higher quality signal, but in the context of compression, it may be important. No value between -1 and +1 has a biased exponent greater than 0x7f. Consequently, there are at best 2**31 values an audio signal could use, and one bit that will be always zero. Of those 2**31 possible values, the vast majority of them lie around zero. Cutting the range in half (to -.5 to +.5) excludes only 1/127th of the possible values. After cutting the range in half 64 times to about +/- .00000000000000000005421, roughly half of the 2**31 previously possible values still lie within this range. Would you say half of a typical audio signal lies within this range? I'm not about to go write a program to find out, but I hope you will accept my guess of "no". The point is this: floating point numbers have a lot of potential values around zero that are unlikely to be used. But wait, it gets worse! When was the last time you saw an audio interface with a 31 bit analog to digtal converter? Maybe they exist, but I consider myself lucky to get 24 bits. No matter how you cram 24 bits in to a 32 bit float, you still have 8 bits that mean nothing, even if they have been shuffled around a bit. I would suspect that many ardour sessions contain WAVs with floats which came from a 24 or 16 bit source. I haven't looked at the jack source, but it seems to me that the conversion between 32 bit floats in +/- 1 and 24 bit integers can be lossless. A trivial archiver could do this conversion and make the session roughly 75% of the original size. Of course, if this isn't the case for some sessions, this trivial archiver can't be useful and lossless anymore, which raises another point worth considering: if testing a lossless compression for float signals, consider testing with something that didn't come straight from a digital integer source. I'd suspect that IIR filters and reverbs in the right circumstances will fill in a good many of the values near zero that may typically be unused. _______________________________________________ Linux-audio-dev mailing list Linux-audio-dev(a)lists.linuxaudio.org http://lists.linuxaudio.org/mailman/listinfo.cgi/linux-audio-dev