Klas wrote:
>Hi folks!
>
>As everybody know by now, the digital area is not what I know best.
>
>I have been trying to get the "best compromise" between filesize and
>"subjective listening quality" when converting .wav to .MP3.
>
>I found that if my old 48k files are converted into 44.1 files, then peak
>level set to -12 or less - I could make reasonably good MP3 files at 56 kb=
ps
>and 22.050
>
>Does MP3 conversion work better with 44.1k sampling rate than 48k? Or is i=
t
>my own software making trouble?
>
>Does it make sense to you "digitknowers" that one must lower the peak outp=
ut
>to -12 db or less?
>
>Klas.
>Does it make sense to you "digitknowers" that one must lower the peak outp=
ut
to -12 db or less?
I guess we're saying that mp3 is a " here,.. get a sense of it"
web/email distribution application. Makes sense that file saturation
is right when hits the space bar and the material shines. I play some
music first to get a normal reference level. Most folks listen
though computer speakers, some on headphones. Some of the later group
may be using decent monitoring. Any of you check email and browse
monitoring sound on a reference speaker system? -12 to -15 dB peaks
seem about right to me for percussive tracks like birds. Ambient
tracks have greater bandwidth which folks like to monitor louder,
like music. As for quality in other applications such as sound file
processing and sub-mixing, the sound will retain more "body" if all
the bits are used and the monitoring system volume is lowered. My
experience is that equalization (tonal adjustment) is more accurate
and it will mix and master more faithfully down stream when most of
the bits are used. But wait a minute,.. I'm advising you on playback
level? Rob D.
--
Rob Danielson
Film Department
University of Wisconsin-Milwaukee
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