22 Nov
2006
22 Nov
'06
1:57 p.m.
paul,
we recently ported the buzzhost to linux ( = aldrin). a default
convention in this program is to mix into float buffers in the -32768 to
32767 range. only recently we changed all the code to keep to -1 .. 1.
it has been claimed numerous times that buzz' sound quality was worse
than what you are used to. i couldn't hear that and i blame it on lack
of talent and mastering skills on the artists side, but i'm still
curious whether an implementation such as above can significantly impact
on the dynamic quality of digital music.
what do you say?
On Wed, 2006-11-22 at 12:05 -0500, Paul Davis wrote:
On Wed, 2006-11-22 at 11:59 -0500, Rick Wright wrote:
--
-- Leonard Ritter
-- http://www.leonard-ritter.com
-- http://www.paniq.org
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