On Wed, 2006-11-22 at 11:59 -0500, Rick Wright wrote: > Hi List, > > I'm currently trying to convert raw audio data from 64bit float (ieee > double precision) Big Endian to other formats (i.e. WAV, OGG, etc.) for > output using some linux audio player. The problem I'm running into is > that when importing this data into various programs (see below), the > audio clips outside of +/- magnitude=1. I don't understand why this is > and, more importantly, how to avoid/workaround this limitation. it is a widely adopted convention that any floating point format normalizes the sample data to a -1.0 .. +1.0 range. doing so loses no precision or resolution or dynamic range. this applies to 32 bit and 80 bit float (there is no 64 bit floating point format, no matter what various windows audio s/w makers may claim in their advertising). if you violate this convention, you will get the results you are seeing. --p
