22 Nov
2006
22 Nov
'06
1:05 p.m.
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