[BC] Balancing the processing from analog to digital
Kent Winrich
kwinrich
Thu Dec 21 11:59:46 CST 2006
Agreed! One of the local NPR stations is so proud of how little
processing they run. But when you listen to a show like "Whatya
Know?" you are always reaching for the volume control. Quiet
passages then all of a sudden thundering applause. What a PIA while driving!
There must be a happy medium. But of course you will never get a
group like this to agree what that medium is! ;-)
On 12/21/06, Robert Orban
<<mailto:rorban at earthlink.net>rorban at earthlink.net> wrote:
At 10:03 PM 12/20/2006, Dana Puopolo wrote:
>Bob,
>
>
>The notion that someone should take a medium that has perhaps 80 db
of dynamic
>range and compress and limit it until is runs within a couple db of the
>CRUSHED main channel is preposterous! Why bother HAVING all that range if
>you're not going to take some advantage of it? The one GOOD thing
I've heard
>from FM HD is the lack of noise. To me at least, peak limting (and clipping)
>does the most damage to audio of anything.
You missed my point. At least in mobile reception, there will be areas
where a significant amount of crossfading occurs between the analog FM and
HD1 digital streams. If the volume drops 10 dB each time the radio
crossfades from analog to digital, this will be a HUGE irritant -- far
worse than excessive compression or peak limiting. Indeed, I believe that
this would cause 95%+ of the audience to tune out after it happened a few
times.
If a station wants to broadcast a wide dynamic range signal, they might as
well do it on a stream. And even if they do, they need to process for
source-to-source consistency. Schulke's Beautiful Music formats maintained
the relative levels between sources well enough to allow minimal processing
while still achieving source-to-source consistency. But that was then.
These days, who has the time or budget to carefully adjust the loudness
level of every element (including each commercial) before it get entered
into a playout system? In theory, this could be done automatically by
analyzing each file's subjective loudness level and adjusting the file's
level appropriately (which is NOT the same as peak-normalizing the file;
peak normalization has nothing to do with subjective loudness). But in
practice, this isn't what happens at most stations.
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