"David Morgan (MAMS)"

I never did respond to you. (Take me with a grain of salt, here). You
just
kinda jumped in and said to Lorin that we were doing it all wrong.

Your responses that I read just listed a bunch of different gain control
products without much reference to your rational or reason why you needed so
many gain control stages.

That's the wonderful thing about Google... a quick search on my name
and 'compression' would have shown that I've been preaching against
it's over use around here for at least 8 years.

I too have crusaded against too much compression especially in the broadcast
field. I think we find more and more to agree on as we talk.

Not to initiate another debate, but limiting in most cases is just lopping
off
the tops of otherwise smooth waveforms, creating near squares. There
aren't too many inexpensive limiters that can do much more than that.
I'd use higher ratio compression if at all possible, except in the event
of seriously high transient peak source material. Whatever I do, I try
not to ruin the source material before it ever gets well into the chain
by creating near square waves from the git-go. There's nothing that
can fix it after that happens.

Just a unclarity in semantics. We never discussed what you mean as limiting
as to what I mean. I think we both do this part the much the same. I never
use extremely high limit ratios either where I can control them. Where I
can't set the ratio, I use very little or none of it.

(EG: I prefer fast attack times and modest release
times on the majority of my applications, and that comes from 30 years of
recording and live sound... different, I think, from your experience).
But we're
here to share what we do know, so don't let this put you off... just be
up for
justifying your recommendations with a little more content.

Me too until I attended a mastering session where the engineer used
extremely slow attack and release. I mean 500 ms attack and 4 sec release.
Not that I advocate that for every or even many situations, but it opened my
eyes to new possibilities. For example, try slow attack and FAST release
for a nice tight sound with lots of definition on a bass instrument.

The only problem was that in the basis of your disagreement, you really
provided no specific facts or examples... only that you knew, 'better'.

I just said I disagreed with the statement "compress early and compress
often". I don't think it takes a lot of rhetoric to claim that to be an
oversimplification.

A little advice from having hung around here for a while.... If you're
going
to offer some advice or lodge a disagreement, be prepared to say more
than, "I think that's wrong and I'd do it differently".... *especially* if
you
haven't read the entire thread. ;-)

First of all, it isn't always possible to tell if the entire thread is there
or not. I thought I DID have the entire thread until you pointed out some
info I didn't have.

Secondly I thought I did say some of the reasons WHY I did things the things
I did.

Thirdly, what I "respectively disagreed" with was the statement "compress
early and compress often". And even then I only disagreed in that I claim
there is more to it than just that. I thought a few basic examples was all
that were needed to deny such a wide sweeping generalization. Compress
early and compress often, maybe yes that can work, but more important is
your settings, not simply doing a lot of it and doing it often. Give me a
break here pal! I too have been making a living in audio for 29 years now
and I don't think a full thesis on my compression philosophy is needed to
argue there is more to compression than "compress early and compress often."
I was prepared to discuss specifics, but most others (you excepted) were
more interested in blowing off steam than discussing specifics.

Julian

"David Morgan (MAMS)" wrote

I've definitely heard plenty of frequency related changes in my recent
experiments. The phasey, beating sort of thing definitely seems to
be louder that the basic levels using 44.1 encryption, but it lessens
dramatically when going to lower encryption rates like 16K as Carey
had me try earlier (for the tiny baud rate stuff).

You'll hear even less if you go to mono.

Julian

"Julian Adamaitis" wrote in message ...

"David Morgan (MAMS)" wrote

I've definitely heard plenty of frequency related changes in my recent
experiments. The phasey, beating sort of thing definitely seems to
be louder that the basic levels using 44.1 encryption, but it lessens
dramatically when going to lower encryption rates like 16K as Carey
had me try earlier (for the tiny baud rate stuff).

You'll hear even less if you go to mono.

Julian


I'm giving that some serious thought... but I had some non-pannable
stereo efx returns going and a couple of stereo miked sources panned
in the mix, which might go south on me if collapsed to mono. I'll be
experimenting, thanks....

DM

"Don Pearce" wrote

I'm not sure what compelling argument can counter "sounds like ****".
Virtually very high budget vocal album on the market has compression on
the
lead artist. Does every vocal album ever released sound like **** to you
too?

No but some are more objectionable than others. I'm particularly
addressing talk radio, where the voice just pumps non stop.

I agree. Talk radio often sounds unacceptably compressed.

There is nothing to stop car radio manufacturers including compression
- it could be done for pennies. And of course DAB has that facility
built into the spec. But guess what? The radio engineers still
compress at source.

I don't know how it works in the UK. Here in the US, I understand that it
will be required to get a different modulation processor for my digital
signal on the 2 stations I will be converting to HD. The analog radio
processor is not adequate for reasons I don't completely understand.
Perhaps if Mr. Orban is reading he can explain. I for one am not too happy
about having to spend up to $10,000 for a second processor.

Peak limiting is another matter. Certainly in the UK it is a
requirement for any FM transmitter, so overmodulation is not possible.
But what sort of engineer can't maintain headroom on his feed? A poor
one, obviously.

It comes back to the loudness issue again.


4. Louder is not better.

I agree, but most station managers complain, no, not most, but EVERY
station
manager I have ever worked for does complain if he doesn't perceive his
station as loud as the other guy's.

Glad you agree - but the fact that it is the station manager who is
cracking the whip doesn't make it any better.

It's a general consensus in radio that louder is better. I agree only to a
certain point. My job is to make sure I don't exceed peak modulation and to
follow my GM's requests as to our "sound". My challenge is to make it loud
enough to please management without making it sound weird. I again invite
you to listen to the web streams I processed for kexp.org with an Orban
6200.

Julian

"David Morgan (MAMS)" wrote

I'm giving that some serious thought... but I had some non-pannable
stereo efx returns going and a couple of stereo miked sources panned
in the mix, which might go south on me if collapsed to mono. I'll be
experimenting, thanks....

I figured you probably had some reason for rejecting mono. Let us know.

Julian

"Julian Adamaitis" wrote in message ...

I too have crusaded against too much compression especially in the broadcast
field. I think we find more and more to agree on as we talk.

Indeed... it's just a magnitude or three higher in difficulty to discuss when
the topic of compression is the subject matter. There probably isn't any
other facet of audio that's more susceptible to the minute specifics of the
source material (which is constantly changing) than compression. I have
to say that I dislike most of the 'primers' and the 'recommended settings'
data that's out there because when dealing with compression, you just
*have* to be there, to understand in even the slightest fashion, what
needs to be done, if anything, to the source material.

Just a unclarity in semantics. We never discussed what you mean as
limiting as to what I mean.

Agreed. It's unfortunate that we'd both have to write a couple of books
here on the subject, before we could remotely understand each other's
philosophies and applications.

I caught your last reply to Don, and I think we're probably dealing with our
terminology from two widely diverse perspectives. I'm not worried about
overmodulation, but you have to be. You may not have to be worried
about singers that suddenly start channeling Aretha Franklin in the middle
of a live mix, but I do.

Me too until I attended a mastering session where the engineer used
extremely slow attack and release. I mean 500 ms attack and 4 sec release.
Not that I advocate that for every or even many situations, but it opened my
eyes to new possibilities.

I'd be willing to wager that this was followed by some serious peak limiting
and volume maximization processing. The long release doesn't sound
appealing to me... but I wasn't there to understand *why* it worked on
your source material.

Give me a break here pal!

Done... ;-)

I was prepared to discuss specifics, but most others (you excepted) were
more interested in blowing off steam than discussing specifics.

Like I said g, you gotta' be there in *every situation* to get a grip on the
specifics... or as you said, we'd be writing thesis' and books on compression
for years to come before we'd have the slightest idea what the other is actually
experiencing and why we might do what we do.


Cheers,

--
David Morgan (MAMS)
http://www.m-a-m-s DOT com
Morgan Audio Media Service
Dallas, Texas (214) 662-9901
_______________________________________
http://www.artisan-recordingstudio.com

"David Morgan (MAMS)" wrote (responding to
Julian):

I respect your opinion. I understand it's based on experience and
preference.


Some of the disparity in approach may be the result of the difference in
applications. In radio, the music content is already mixed and
mastered, so some of the dynamics control is already done.

--
"It CAN'T be too loud... some of the red lights aren't even on yet!"
- Lorin David Schultz
in the control room
making even bad news sound good

(Remove spamblock to reply)

"Julian Adamaitis" wrote:

Me too until I attended a mastering session where the engineer used
extremely slow attack and release.


I think you and I are trying to accomplish very different things with
our dynamics control devices. I find very little use for slow attack on
a limiter. If I'm using a limiter at all, it's to catch a transient. A
slow attack won't catch that, so it defeats the purpose.

I'd rather use a couple stages of compression with progessively higher
thresholds and ratios to even things out without clipping or chopping.
Kinda like how the SuperNice Mode on the RNC works.



try slow attack and FAST release for a nice tight sound with lots of
definition on a bass instrument.

Have you not found that to be an ideal recipe for distorting the bass
instrument? Or maybe I should ask what you mean by "fast release?"



I just said I disagreed with the statement "compress early and
compress often". I don't think it takes a lot of rhetoric to claim
that to be an oversimplification.

Uh, yeah, it would be safe to say that one cannot summarize a dynamics
control regime in five words without skipping a few details! g. It
was, however, written in the context of a specific application, and
there was some discussion of how it was being applied. Besides, it was
intended to be a semi-humourous "rule-of-thumb" kind of remark, not a
mantra. And it works for me more often than not (though there have
obviously been more than a couple "nots").

--
"It CAN'T be too loud... some of the red lights aren't even on yet!"
- Lorin David Schultz
in the control room
making even bad news sound good

(Remove spamblock to reply)

"Don Pearce" wrote:

1. Compression on voice sounds like ****. When I hear it it gives me
a feeling not unlike seasickness. I like the sound of the human voice
without compression.

Of course, who doesn't? But *where* are you listening, and what are you
doing while you do? Most people are working, cooking or driving while
listening. They are NOT sitting in a quiet room, listening attentively.
On top of that, variations in playback systems have a dramatic affect on
what the listener hears.

Sure, an unprocessed voice sounds *prettier*, but intelligibility
increases dramatically with judicious use of EQ and compression.

Given a choice between appealing to purist aesthetic sensibilities and
making sure listeners actually hear and can understand what's being
said, I'll compromise my artistic vision.



2. Radio, like every other modern medium, has more than enough
dynamic range by far to accommodate the natural variations of level
from the human voice.

Sure, but so what? The world past the antenna doesn't. See above.



3. If an engineer would simply take the trouble to set the level
properly rather than just leave it to some pump-suck machine, there
would be no need for compression.

You OBVIOUSLY haven't worked with some of our news anchors. Blaming the
operator for the ENORMOUS variability in the level and enunciation of
what comes out of their mouths suggests that either you could stand some
time on my side of the glass or that you have the distinct priviledge of
working with much more controlled voices than do I.

Listen, if you're okay with missing a word here and there, or
occasionally mishearing what's been said, I won't fault you for it. I
just hope that you'll forgive me for knuckling under to pressure from
the VAST majority of listeners who want to hear every word with having
to concentrate too hard. In my line of work, Enemy Number One is
"Whaddesay?"



4. Louder is not better.

I agree. I think that's why playback devices have volume controls.



5. Louder is not better.

I agree, but I have to concede certain practical, competitive realities.
If every other channel on the dial is 10dB louder than mine, no one will
notice mine as they flip through the channels. I gotta at least be in
the ballpark to be in the game.

--
"It CAN'T be too loud... some of the red lights aren't even on yet!"
- Lorin David Schultz
in the control room
making even bad news sound good

(Remove spamblock to reply)

Lorin David Schultz wrote:
"Don Pearce" wrote:

1. Compression on voice sounds like ****. When I hear it it gives me
a feeling not unlike seasickness. I like the sound of the human voice
without compression.


Of course, who doesn't? But *where* are you listening, and what are you
doing while you do? Most people are working, cooking or driving while
listening. They are NOT sitting in a quiet room, listening attentively.
On top of that, variations in playback systems have a dramatic affect on
what the listener hears.

Sure, an unprocessed voice sounds *prettier*, but intelligibility
increases dramatically with judicious use of EQ and compression.

Given a choice between appealing to purist aesthetic sensibilities and
making sure listeners actually hear and can understand what's being
said, I'll compromise my artistic vision.


Since when was intelligibility important? If it was the sations wouldn't
employ the half-witted broadcasters mos of them currently do. They would
also abandon the phone-in and make contributors come to the studio.



2. Radio, like every other modern medium, has more than enough
dynamic range by far to accommodate the natural variations of level
from the human voice.


Sure, but so what? The world past the antenna doesn't. See above.


It most certainly does.




3. If an engineer would simply take the trouble to set the level
properly rather than just leave it to some pump-suck machine, there
would be no need for compression.


You OBVIOUSLY haven't worked with some of our news anchors. Blaming the
operator for the ENORMOUS variability in the level and enunciation of
what comes out of their mouths suggests that either you could stand some
time on my side of the glass or that you have the distinct priviledge of
working with much more controlled voices than do I.

So sack them. Keep on sacking until you find a real broadcaster. News
anchors on the BBC can stay within a dB or so whatever the circumstances
because they understand their jobs.

Listen, if you're okay with missing a word here and there, or
occasionally mishearing what's been said, I won't fault you for it. I
just hope that you'll forgive me for knuckling under to pressure from
the VAST majority of listeners who want to hear every word with having
to concentrate too hard. In my line of work, Enemy Number One is
"Whaddesay?"


I find it a great deal easier to understand what is said by a natural
voice than a compresed one. With compression all I get is the sound, not
the meaning.



4. Louder is not better.


I agree. I think that's why playback devices have volume controls.


How many broadcast engineers understand that little fact?


5. Louder is not better.


I agree, but I have to concede certain practical, competitive realities.
If every other channel on the dial is 10dB louder than mine, no one will
notice mine as they flip through the channels. I gotta at least be in
the ballpark to be in the game.

Talking of channel flipping, an overly loud station is one which causes
me to hit the "next" button every time.

d

"Lorin David Schultz" wrote in message news:EQCae.89805$7Q4.77732@clgrps13...
"David Morgan (MAMS)" wrote (responding to
Julian):

I respect your opinion. I understand it's based on experience and
preference.


Some of the disparity in approach may be the result of the difference in
applications. In radio, the music content is already mixed and
mastered, so some of the dynamics control is already done.

Have you had any requests for CD versions and Radio versions of
mixes? I recently had my first. I'm all too familiar with the vocal up
and vocal down versions, but now there seems to be a new need for
restoring some of the dynamic range people are removing by over-
compressing... apparently because it's screwing with the way in which the
high-dollar broadcast levelling/limiting gear does it's job. Hypercompressed
material doesn't sound as good going through the broadcast processors
as does the same material with a little more of the dynamic range left in
so that the broadcast devices can do what they were designed to do.

Comments... either of you guys ??

"Lorin David Schultz" wrote

I think you and I are trying to accomplish very different things with our
dynamics control devices. I find very little use for slow attack on a
limiter. If I'm using a limiter at all, it's to catch a transient. A
slow attack won't catch that, so it defeats the purpose.

As I said, the application in question was for mastering a session that was
already mixed and individual tracks has compression, limiting etc. If I
understand what and why the engineer did what he did (which is doubtful!) my
impression was he was using this to merely even out volume differences
between tracks and remove some of the dynamic range in individual cuts. He
would re-adjust the threshold for each cut as he went. It was more of an
AGC than compression or limiting.

I'd rather use a couple stages of compression with progessively higher
thresholds and ratios to even things out without clipping or chopping.
Kinda like how the SuperNice Mode on the RNC works.

That's pretty much what I said a couple of posts back.

try slow attack and FAST release for a nice tight sound with lots of
definition on a bass instrument.

Have you not found that to be an ideal recipe for distorting the bass
instrument? Or maybe I should ask what you mean by "fast release?"

Fast enough release that it doesn't boom and sustain forever. I'd use to
ear to judge what poitn that is. It sounds tighter sometimes if the release
underemphasizes the sustain slightly. If you have a player with a lot of
string attack, you may want your attack time to be slow enough to let it all
through, because if it is too fast, it will be lost. Or if you
intentionally want to loose the pops then set it for a fast attack.

Uh, yeah, it would be safe to say that one cannot summarize a dynamics
control regime in five words without skipping a few details! g. It was,
however, written in the context of a specific application, and there was
some discussion of how it was being applied. Besides, it was intended to
be a semi-humourous "rule-of-thumb" kind of remark, not a mantra. And it
works for me more often than not (though there have obviously been more
than a couple "nots").

Sorry. I don't know most of you guys well enough to tell when you're
kidding yet!

Julian

"Lorin David Schultz" wrote

I think you and I are trying to accomplish very different things with our
dynamics control devices. I find very little use for slow attack on a
limiter. If I'm using a limiter at all, it's to catch a transient. A
slow attack won't catch that, so it defeats the purpose.

I also agree. A limiter is of very limited value with a slow attack. It
becomes more of an AGC in that case.

Julian

"Julian Adamaitis" wrote in message...

I don't know how it works in the UK. Here in the US, I understand that it
will be required to get a different modulation processor for my digital
signal on the 2 stations I will be converting to HD. The analog radio
processor is not adequate for reasons I don't completely understand.
Perhaps if Mr. Orban is reading he can explain. I for one am not too happy
about having to spend up to $10,000 for a second processor.

I'd definitely check into that... and I'd love to have a shot at hearing the
reasoning behind it. Honestly, it sounds to me like just another push
for even tighter control over potential overmodulation in order to step
up broadcast output (volume levels) to yet another plateau. I'll bet
they're thinking that more expensive, read-ahead delay type limiting
will keep a tighter reign on over-modulation while increasing levels
to a greater degree than traditional analogue devices.

Maybe you have a chance to speak out here... quality versus loudness
could definitely come into play, and somehow I just don't believe this
has anything at all to do with the fact that station output programming
will be derived from HD. Then again, maybe there are some side-band
issues, or something, in HD that have to be addressed seperately.

Scott D. ?

I again invite you to listen to the web streams I processed for kexp.org
with an Orban 6200.

I wanna' hear something done with the Behringer... ;-)

Just kidding... I've got a road trip coming up, so maybe we'll talk again
on Wednesday night.

Peace,

DM

"Bob Cain" wrote in message ...


David Morgan (MAMS) wrote:
Hey all,

I have, at the advisement of several, put Win Lame on a couple of PCs
to use for converting .wav files to higher rate MP3s and have been happy.

Lame has not put a lot of effort into the low bit rates. I
think the winner there lies with the dark force, WMA.


Possibly... but I'm trying to cook both formulae on the same stove. ;-)

Honest to gosh... I didn't realize that I could leave 44.1 behind as an encryption
rate!! Taking the same encode function in LAME down to a 16Khz output
renders a pretty acceptable stereo, 32K baud rate.

Like I said, I'm new at this. I've had LAME for around a year now, but this
church web archive thing is the first time that I've had to give consideration
to creating anything less than a 160K MP3.

DM

"David Morgan (MAMS)" wrote

Have you had any requests for CD versions and Radio versions of
mixes? I recently had my first. I'm all too familiar with the vocal up
and vocal down versions, but now there seems to be a new need for
restoring some of the dynamic range people are removing by over-
compressing... apparently because it's screwing with the way in which the
high-dollar broadcast levelling/limiting gear does it's job.
Hypercompressed
material doesn't sound as good going through the broadcast processors
as does the same material with a little more of the dynamic range left in
so that the broadcast devices can do what they were designed to do.

Comments... either of you guys ??

Interesting idea. I've heard it mentioned over the years, but never had any
first hand experience with it. You'd almost need one of those broadcast
processors to tell what effect if any your mastering has. With certain
material, like techno with extremely high transients, it might sound better
if you limit more rather than less, so you can control that part yourself
instead of giving it to the downstream broadcast device to mangle. Other
places you might want to compress less to accommodate the extra compression
it will get later. There are a vast range of settings on Broadcast
processors from "classical" to "open pop" (my favorite) to "rock" to
"grunge". It seems you'd have to make 3 or 4 masters for which degree of
processing the station in question uses. The problem with these devices is
that they have so many settings and the user has so much control to modify
the settings some really awful results are possible by people who are merely
trying to be loud without having enough experience to hear when they have
gone too far.

Julian

"Lorin David Schultz" wrote in message news:T5Dae.89808$7Q4.63865@clgrps13...

"Julian Adamaitis" wrote:

Me too until I attended a mastering session where the engineer used
extremely slow attack and release.

I'd really love to have been present to see under what set of circumstances
that a four second release time would have been employed. If there was
any density at all to the program material, it seems like the compressor
would have entered a state of steady compression and never come out.
At first thought, this would drastically change the 'appearance' of the original.

I think you and I are trying to accomplish very different things with
our dynamics control devices. I find very little use for slow attack on
a limiter. If I'm using a limiter at all, it's to catch a transient. A
slow attack won't catch that, so it defeats the purpose.

I didn't want to go there, but I didn't understand that such an option existed
on a peak limiter. If the objective is to catch and stop the peaks, which
are predominately the result of transients, then a slow attack is utterly
worthless. Julian is confusing me a little as to whether he's being a
purist on me or whether he's trying to avoid station overmodulation.
I don't think you can successfully be both in that situation. ;-)

try slow attack and FAST release for a nice tight sound with lots of
definition on a bass instrument.

Have you not found that to be an ideal recipe for distorting the bass
instrument? Or maybe I should ask what you mean by "fast release?"

Anything less than a couple of hundred milliseconds, and my experience
has been that distortion occurs when the following note strikes while the
compressor is still in the fast release stage... a sure fire recipe for erratic
pumping as well (depending on another dozen factors, of course). The
same tends to also apply on low frequency material if the attack side of
the equation is too fast, at least in my experience.

If I were letting the peaks get by, but wanted to set a limited RMS output
level, I'd just use a higher compression ratio (on a compressor that will
handle the load as transparently as possible). One reason I still like the
old ASHLY CL-52E so much is that it's very nearly an RMS limiter - even
at modest ratios it will practically brick wall it's output, and very little in the
way of transients gets by with a faster attack.

Oh well, it's late and I'm rambling... just wanted to say hello before slipping
down to Austin for a couple of days (after I finish my newly acquired MP3
archiving task for the church) tomorrow night.

DM

"David Morgan (MAMS)" wrote

I'd definitely check into that... and I'd love to have a shot at hearing
the
reasoning behind it. Honestly, it sounds to me like just another push
for even tighter control over potential overmodulation in order to step
up broadcast output (volume levels) to yet another plateau. I'll bet
they're thinking that more expensive, read-ahead delay type limiting
will keep a tighter reign on over-modulation while increasing levels
to a greater degree than traditional analogue devices.

The bandwidth is tightly limited to what is left over from existing analog
FM to 96kbps which is then divided up between sometimes 3 programs! Talk
about low bandwidth challenges.

Maybe you have a chance to speak out here... quality versus loudness
could definitely come into play, and somehow I just don't believe this
has anything at all to do with the fact that station output programming
will be derived from HD. Then again, maybe there are some side-band
issues, or something, in HD that have to be addressed separately.

I wish I could express an opinion, but instead I was "informed" of the
standard while at the NAB last week.

Julian

"David Morgan (MAMS)" wrote

I'd really love to have been present to see under what set of
circumstances
that a four second release time would have been employed. If there was
any density at all to the program material, it seems like the compressor
would have entered a state of steady compression and never come out.
At first thought, this would drastically change the 'appearance' of the
original.

Possibly true depending on the threshold setting. As I said, he spent a
long time readjusting the threshold for every cut.

I think you and I are trying to accomplish very different things with
our dynamics control devices. I find very little use for slow attack on
a limiter. If I'm using a limiter at all, it's to catch a transient. A
slow attack won't catch that, so it defeats the purpose.

I didn't want to go there, but I didn't understand that such an option
existed
on a peak limiter. If the objective is to catch and stop the peaks, which
are predominately the result of transients, then a slow attack is utterly
worthless. Julian is confusing me a little as to whether he's being a
purist on me or whether he's trying to avoid station overmodulation.
I don't think you can successfully be both in that situation. ;-)

I've been in situations of studio mixing and broadcast engineering and I may
lapse between the 2 without acknowledging it. In general, my experience is
a sum total of both professions as well as some live sound and semi-pro
mastering.

Also when I say limiter, I could mean anything with a ratio of 10:1 or more.
At 8:1 or 10:1 a slow attack can be useful. When you say peak limiter, I
assume you mean something that catches peaks, which can't be done with slow
attacks. It seems most combinations can be useful somewhere.

Julian

"David Morgan (MAMS)" wrote:

Have you had any requests for CD versions and Radio versions of
mixes? I recently had my first. I'm all too familiar with the
vocal up and vocal down versions, but now there seems to be a new
need for restoring some of the dynamic range people are removing by
over-compressing... apparently because it's screwing with the way in
which the high-dollar broadcast levelling/limiting gear does it's
job. Hypercompressed material doesn't sound as good going through the
broadcast processors as does the same material with a little more of
the dynamic range left in so that the broadcast devices can do what
they were designed to do.

Comments... either of you guys ??


All I know about that is what Bob Orban has written here. Given the
source, I tend think there may be some validity to that claim! g

I have no control over our "final stage" processing, nor have I ever
seen how it's set. I don't even know which box we're using.

I have, however, frequently compared recordings from Master Control to
what I recorded in the control room. Based on those, I got the
impression that our processing seems to be set to "fairly benign."
That's not all that surprising, as our Chief Engineer leans towards
erring on the side of safety (he thinks anything that *peaks*
over -20dBFS is too hot). Since I was once called into the Principal's
office for putting the transmitter in jeopardy with excessive levels, I
guess we're not doing much peak limiting either!

I don't have much occasion to put commercially mastered material to air,
so it's hard for me to judge. All the music I deal with is either live
or from our network library. The network stuff isn't mashed like a
commercial CD, so I don't know what pre-crushed material sounds like
through our chain.

--
"It CAN'T be too loud... some of the red lights aren't even on yet!"
- Lorin David Schultz
in the control room
making even bad news sound good

(Remove spamblock to reply)

"David Morgan (MAMS)" wrote:

Oh well, it's late and I'm rambling... just wanted to say hello
before slipping down to Austin for a couple of days (after I finish
my newly acquired MP3 archiving task for the church) tomorrow night.



Speaking of that task, where are you at with it? Are you making any
headway? Have you found a way to make a 20kb/s file that doesn't sound
like it's playing underwater?

--
"It CAN'T be too loud... some of the red lights aren't even on yet!"
- Lorin David Schultz
in the control room
making even bad news sound good

(Remove spamblock to reply)

"Lorin David Schultz" wrote in message news:2HNae.2128$tg1.217@edtnps84...
"David Morgan (MAMS)" wrote:

Oh well, it's late and I'm rambling... just wanted to say hello
before slipping down to Austin for a couple of days (after I finish
my newly acquired MP3 archiving task for the church) tomorrow night.



Speaking of that task, where are you at with it? Are you making any
headway? Have you found a way to make a 20kb/s file that doesn't sound
like it's playing underwater?

Not great, but good enough at 32Kbps

I did three services worth of CD editing and MP3 encoding at higher baud
rates as experiments; with 80, 128, and 160. I stayed at 44.1k encryption
the whole time though, so I may pull that back to 32khz next week and do
some more listening. I let a friend take the MP3 files that I'd made (above)
for each week and do the 20K stuff because I just hadn't yet figured this gig
out well enough to get usable results. So this afternoon, I'll be dropping to
a 16k encryption rate (thanks again to Carey Carlan) and have decided to
stick with Win Lame. I've also decided to go with 32Kb files over 20K for the
dial-up version. It's more closely resembling being under a pillow than under
water, and I can breath better under a pillow. The church just wants it somewhere
below 40kb to help stop the buffering for those on a dial-up. As a few folks have
mentioned... I may try going mono as well - - but not this week... I'm headed for
a couple of vacation days in Austin, and I'm not up for more lengthy experimenting
and listening right now. ;-) I'll get the hang of it sooner or later, with a little help
from my friends.

Thanks and cheers to ya',

DM

On Sun, 24 Apr 2005 15:40:48 -0400, David Morgan \(MAMS\) wrote
(in article 4DSae.2887$yc.2535@trnddc04):


"Lorin David Schultz" wrote in message
news:2HNae.2128$tg1.217@edtnps84...
"David Morgan (MAMS)" wrote:

Oh well, it's late and I'm rambling... just wanted to say hello
before slipping down to Austin for a couple of days (after I finish
my newly acquired MP3 archiving task for the church) tomorrow night.



Speaking of that task, where are you at with it? Are you making any
headway? Have you found a way to make a 20kb/s file that doesn't sound
like it's playing underwater?

Not great, but good enough at 32Kbps

I did three services worth of CD editing and MP3 encoding at higher baud
rates as experiments; with 80, 128, and 160. I stayed at 44.1k encryption
the whole time though, so I may pull that back to 32khz next week and do
some more listening. I let a friend take the MP3 files that I'd made
(above)
for each week and do the 20K stuff because I just hadn't yet figured this gig
out well enough to get usable results. So this afternoon, I'll be dropping
to
a 16k encryption rate (thanks again to Carey Carlan) and have decided to
stick with Win Lame. I've also decided to go with 32Kb files over 20K for
the
dial-up version. It's more closely resembling being under a pillow than
under
water, and I can breath better under a pillow. The church just wants it
somewhere
below 40kb to help stop the buffering for those on a dial-up. As a few folks

have
mentioned... I may try going mono as well - - but not this week... I'm
headed for
a couple of vacation days in Austin, and I'm not up for more lengthy
experimenting
and listening right now. ;-) I'll get the hang of it sooner or later,
with a little help
from my friends.

Thanks and cheers to ya',

DM

Mono helps. Real audio actually sounds better at low bit rates over here.

Regards,

Ty Ford



-- Ty Ford's equipment reviews, audio samples, rates and other audiocentric
stuff are at www.tyford.com

Bob Cain wrote in
:

snip.....snip...

I think the winner there lies with the dark force....


You, as someone who, for whatever reason, has turned his soul over to the
dark force, certainly ought to know.

In article ,
The Ghost wrote:

Bob Cain wrote in
:

snip.....snip...

I think the winner there lies with the dark force....


You, as someone who, for whatever reason, has turned his soul over to the
dark force, certainly ought to know.


Oh, for Christ's sake, "ghost", put a sock in it, will ya? IF and WHEN
you actually have something useful and informative to say (an occurrence
I have yet to witness), please speak up and share your constructive
insight. Until then, ZIP IT, pinhead - you and Phil and all the rest of
you emotional 4-year-olds who've inexplicably wandered over from the
rec.audio.opinion.brain-dead sandbox! Sheesh! Enough already!!

--
Brendan Doyle

David Morgan (MAMS) wrote:

Win Lame is just great for the 128K stream, but they need a 24 or 32K
stream for the dial-up people and after much manual experimentation
with Lame, the results are seriously sub-par.

As others have mentioned, the Fraunhofer encoder usually beats out the
competition at lower bitrates.



One friend from another church has been using one of the Nero products
to do a 20K encode that doesn't sound bad at all, but he has a huge suite
of tools that were loaded along with the MP3 conversion stuff. I don't want
to stink up either my personal PC or any of my workstations with unnecessary
gobbledygook that might cause conflicts.

If you only install the first of the three Nero files (Update-Package 1
in their terms) you avoid the clutter. The stock MP3 encoder they ship
is LAME, BTW. Their mp3PRO encoder ($19.99, works with either version
5.5 or 6) comes from Fraunhofer. Just be sure to pick a standard MP3
encoding option.

Nero 6 also includes AAC which sounds far better than MP3 at low
bitrates, but you sacrifice compatibility with most portable players and
some older software players.

david wrote:

Couple years ago I tried everything available on the Mac and found to
my surprise iTunes had the best sound at tiny sampling rates.

Of course--it uses AAC.

Bill Ruys wrote:
I've just tried encoding at 32 Kbps with WMA, LAME (mp3) and Fraunhofer
(mp3) CODECs. In all cases, the Fraunhofer CODEC produced the best results.

Agreed.



I don't have an Ogg player installed right now so I couldn't test Ogg files.

They sound great at higher bitrates, but I wouldn't pick them for this
application.

Brendan Doyle wrote in
:


Oh, for Christ's sake, "ghost", put a sock in it, will ya?

Send your complaints to Bob Cain. He started this four years ago and it
isn't going to end until either he or I reach room temperature.


IF and WHEN
you actually have something useful and informative to say (an
occurrence I have yet to witness), please speak up and share your
constructive insight.

Assuming that are capable of understanding the technical content, visit
alt.sci.physics.acoustics sometime and join in the discussions. So far no
one from any of he audio groups has demonstrated the requisite technical
expertise. Perhaps you will be the first.



Until then, ZIP IT, pinhead - you and Phil and
all the rest of you emotional 4-year-olds who've inexplicably wandered
over from the rec.audio.opinion.brain-dead sandbox! Sheesh! Enough
already!!

Given that you are complaining to the effect rather than to the cause, you
are the one who is the pinhead, not I.

On Tue, 26 Apr 2005 12:24:49 -0400, Kurt Albershardt wrote
(in article ):

David Morgan (MAMS) wrote:

Win Lame is just great for the 128K stream, but they need a 24 or 32K
stream for the dial-up people and after much manual experimentation
with Lame, the results are seriously sub-par.

As others have mentioned, the Fraunhofer encoder usually beats out the
competition at lower bitrates.

my experience is that realaudio beats out any low bitrate mp3, frau or not.

Ty Ford





-- Ty Ford's equipment reviews, audio samples, rates and other audiocentric
stuff are at www.tyford.com

In article ,
The Ghost wrote:

Brendan Doyle wrote in
:

Oh, for Christ's sake, "ghost", put a sock in it, will ya?

Send your complaints to Bob Cain. He started this four years ago and it
isn't going to end until either he or I reach room temperature.

"But, mommy, HE started it!" Grow up already, will you? What the hell's
the matter with you? Nobody over here gives a flying **** about your
pointless little never-ending ****ing match about "Doppler distortion"
or whatever your technical snit-du-jour is about today. Take it out back
already where the rest of us don't have to hear about it day-in and
day-out, and come back when you're ready for adult conversation, if ever.

I don't know Bob Cain, but at least he DOES contribute with some
regularity to discussions here actually related to the purposes of the
group. On the other hand, I have NEVER seen you contribute
constructively to ANY thread in rec.audio.pro. As far as I can tell,
your SOLE reason for posting to this newsgroup is to throw rocks at Bob
Cain every time he sticks up his head. In the real world, that would be
grounds for a restraining order at the very least. Your kind of
anonymous antisocial behavior is, unfortunately, the scourge of the
Internet, it being the one place where cowardly people can behave like
you do without the threat of being whacked upside the head.


IF and WHEN you actually have something useful and informative to
say (an occurrence I have yet to witness), please speak up and share
your constructive insight.

Assuming that are capable of understanding the technical content, visit
alt.sci.physics.acoustics sometime and join in the discussions. So far no
one from any of he audio groups has demonstrated the requisite technical
expertise. Perhaps you will be the first.

Let me rephrase that: "So far no one from THIS audio group has
demonstrated the desire to visit alt.sci.physics.acoustic and be
subjected to continual pointless abuse from anonymous immature
sociopaths like yourself." There are several contributors to this group
that have more "requisite technical expertise" to discuss acoustic
physics in their little fingers than guys like you will ever have in
your lives. The difference between you and them is that they are adults
who aren't constantly blowing their own horns in order to bolster their
fragile little insecure egos by demonstrating how superior they are to
everyone else around them.

Believe me, when I want to research any acoustic physics topics,
alt.sci.physics.acoustic will be the LAST place I'll look! There's a
reason that it's an "alt" group. I generally like to discuss technical
topics with intelligent, informed adults who identify themselves, rather
than with argumentative, emotionally-stunted trolls who are too cowardly
to post under their own names.


Until then, ZIP IT, pinhead - you and Phil and all the rest of you
emotional 4-year-olds who've inexplicably wandered over from the
rec.audio.opinion.brain-dead sandbox! Sheesh! Enough already!!

Given that you are complaining to the effect rather than to the cause, you
are the one who is the pinhead, not I.

Gee, it's all Bob's fault, isn't it? "HE started it four years ago, so
now I'm obliged to devote the rest of my sorry little life to following
him around the Internet heaping abuse on him wherever he shows up." How
pathetic. Now I understand what you meant by your comment in your first
sentence:

...it isn't going to end until either he or I reach room temperature.

Room temperature is the temperature of corpses, isn't it? It does seem
like one solution for you, though a bit drastic. Maybe you need a new
hobby or a nice doggy or something. I sure hope you find some way to
restart your long-arrested emotional development, and start to grow up
before you do finally reach room temperature. Who knows - you might be
happy some day!

--
Brendan Doyle

Ty Ford wrote:
On Tue, 26 Apr 2005 12:24:49 -0400, Kurt Albershardt wrote
(in article ):


David Morgan (MAMS) wrote:

Win Lame is just great for the 128K stream, but they need a 24 or 32K
stream for the dial-up people and after much manual experimentation
with Lame, the results are seriously sub-par.

As others have mentioned, the Fraunhofer encoder usually beats out the
competition at lower bitrates.


my experience is that realaudio beats out any low bitrate mp3, frau or not.


As do Quicktime, WMA, and AAC. But (like RealAudio) these are not MP3
codecs.

On 4/27/05 12:29 PM, in article , "Kurt
Albershardt" wrote:

Ty Ford wrote:
On Tue, 26 Apr 2005 12:24:49 -0400, Kurt Albershardt wrote
(in article ):


David Morgan (MAMS) wrote:

Win Lame is just great for the 128K stream, but they need a 24 or 32K
stream for the dial-up people and after much manual experimentation
with Lame, the results are seriously sub-par.

As others have mentioned, the Fraunhofer encoder usually beats out the
competition at lower bitrates.


my experience is that realaudio beats out any low bitrate mp3, frau or not.


As do Quicktime, WMA, and AAC. But (like RealAudio) these are not MP3
codecs.

OK I give...
I'm lost...
Where might I go for a primer on digital audio compression systems and how
they relate to (what I thought was a standard) mp3?

SSJVCmag wrote:
On 4/27/05 12:29 PM, in article , "Kurt
Albershardt" wrote:


Ty Ford wrote:

On Tue, 26 Apr 2005 12:24:49 -0400, Kurt Albershardt wrote
(in article ):

David Morgan (MAMS) wrote:

Win Lame is just great for the 128K stream, but they need a 24 or 32K
stream for the dial-up people and after much manual experimentation
with Lame, the results are seriously sub-par.


As others have mentioned, the Fraunhofer encoder usually beats out the
competition at lower bitrates.


my experience is that realaudio beats out any low bitrate mp3, frau or not.


As do Quicktime, WMA, and AAC. But (like RealAudio) these are not MP3
codecs.


Where might I go for a primer on digital audio compression systems and how
they relate to (what I thought was a standard) mp3?


MP3 (properly called MPEG-1 layer 3) is a standard, at least for
playback. All of them are standards of a sort. None of them are
compatible with each other. All of them come with snarly royalty and
patent issues attached.

http://wiki.hydrogenaudio.org/index.php?title=Lossy

Kurt Albershardt wrote:
MP3 (properly called MPEG-1 layer 3) is a standard, at least for
playback. All of them are standards of a sort. None of them are
compatible with each other. All of them come with snarly royalty and
patent issues attached.

With the exception of Ogg Vorbis of course.

Johann
--
Darf jetzt schon der mieseste Abschaum des Usenet Wahlen abhalten? Aus
Protest gegen dieses asoziale begin/end-Arschloch
werde ich fortan ohne Morver posten. (Klaus "Diego Alfredo Unada"
Ketelaer in )

Johann Burkard wrote:
Kurt Albershardt wrote:

MP3 (properly called MPEG-1 layer 3) is a standard, at least for
playback. All of them are standards of a sort. None of them are
compatible with each other. All of them come with snarly royalty and
patent issues attached.


With the exception of Ogg Vorbis of course.


Of course.

....but Ogg is not so great at low bitrates.

"SSJVCmag" wrote

OK I give...
I'm lost...
Where might I go for a primer on digital audio compression systems and how
they relate to (what I thought was a standard) mp3?

How they relate is purely subjective, but there is some consensus as to
which ones sound better for what bit rates and music types.

Julian

"Kurt Albershardt" wrote in message ...
SSJVCmag wrote:
On 4/27/05 12:29 PM, in article , "Kurt
Albershardt" wrote:


Ty Ford wrote:

On Tue, 26 Apr 2005 12:24:49 -0400, Kurt Albershardt wrote
(in article ):

David Morgan (MAMS) wrote:

Win Lame is just great for the 128K stream, but they need a 24 or 32K
stream for the dial-up people and after much manual experimentation
with Lame, the results are seriously sub-par.


As others have mentioned, the Fraunhofer encoder usually beats out the
competition at lower bitrates.


my experience is that realaudio beats out any low bitrate mp3, frau or not.


As do Quicktime, WMA, and AAC. But (like RealAudio) these are not MP3
codecs.


Where might I go for a primer on digital audio compression systems and how
they relate to (what I thought was a standard) mp3?


MP3 (properly called MPEG-1 layer 3) is a standard, at least for
playback. All of them are standards of a sort. None of them are
compatible with each other. All of them come with snarly royalty and
patent issues attached.

http://wiki.hydrogenaudio.org/index.php?title=Lossy


I appreciate the link... as you know I'm a newbie at the whole MP3 thing.

At first, due to my lack of experience, I was afraid to encode at 16Khz,
believing that I would not be making a universally compatible MP3, since
this was considered MPEG-2.

I'm personally staying with Win LAME now that I understand a little bit
more. I need a very typical MP3 that can be read by any ordinary player
with the file association in place. I simply can't use the 'player specific'
schemes like REAL or WMA, etc... I will do *nothing* that requires the
end user to download, install and use a specific piece of software. The
church has already been through the REAL nightmare with mountains
of complaints... everything from not having the software, installing the
software causing changes in other software (usually just file associations
lost due to lack of understanding), to the spyware issue, and it was
decided many months ago that REAL would no longer be a part of
the church's web archiving process.

--
David Morgan (MAMS)
http://www.m-a-m-s DOT com
Morgan Audio Media Service
Dallas, Texas (214) 662-9901
_______________________________________
http://www.artisan-recordingstudio.com

MP3 (properly called MPEG-1 layer 3) is a standard, at least for
playback. All of them are standards of a sort. None of them are
compatible with each other.

That's what I like about standards... There're so MENY of them...

Brendan Doyle wrote in
:

In article ,
The Ghost wrote:

Brendan Doyle wrote in
:

Oh, for Christ's sake, "ghost", put a sock in it, will ya?

Send your complaints to Bob Cain. He started this four years ago and
it isn't going to end until either he or I reach room temperature.

"But, mommy, HE started it!" Grow up already, will you? What the
hell's the matter with you? Nobody over here gives a flying **** about
your pointless little never-ending ****ing match about "Doppler
distortion" or whatever your technical snit-du-jour is about today.
Take it out back already where the rest of us don't have to hear about
it day-in and day-out, and come back when you're ready for adult
conversation, if ever.

That which constitutes a so-called "adult" conversation is a matter of
opinion. The (pathetic) technical discussion has already occurred, and I
don't recall your contribution to it.


I don't know Bob Cain, but at least he DOES contribute with some
regularity to discussions here actually related to the purposes of the
group. On the other hand, I have NEVER seen you contribute
constructively to ANY thread in rec.audio.pro.

Either you have been asleep or you lack the ability to recognize that which
constitutes a constructive contribution. In the event that you were
asleep, let me suggest that you do a google search for "Experimental
Evidence for Dynamic Doppler Shift" which I posted on 8/14/2004. I went to
the trouble to set up and perform an experiment to test a hypothesis. What
exactly did YOU do?

As far as I can tell,
your SOLE reason for posting to this newsgroup is to throw rocks at
Bob Cain every time he sticks up his head. In the real world, that
would be grounds for a restraining order at the very least. Your kind
of anonymous antisocial behavior is, unfortunately, the scourge of the
Internet, it being the one place where cowardly people can behave like
you do without the threat of being whacked upside the head.

You are an ignoramus. If you check the record, you will find that Bob Cain
is the stalker and the coward, not I. Nonetheless, if you think otherwise
and if you would like a personal one-on-one physical confrontation, let me
know and I will accommodate you.


snip.........snip


Let me rephrase that: "So far no one from THIS audio group has
demonstrated the desire to visit alt.sci.physics.acoustic and be
subjected to continual pointless abuse from anonymous immature
sociopaths like yourself."

Dream on, moron, or do a google search for Bob Cain's posts to
alt.sci.physics.acoustics and be prepared for a reality check. The record
clearly demonstrates that Bob Cain is the immature sociopath and is the one
who is the perpetrator of the so-called continual pointless abuse.



There are several contributors to this
group that have more "requisite technical expertise" to discuss
acoustic physics in their little fingers than guys like you will ever
have in your lives.

Perhaps, but so far I haven't seen any eveidence to support such an
assertion.


The difference between you and them is that they
are adults who aren't constantly blowing their own horns in order to
bolster their fragile little insecure egos by demonstrating how
superior they are to everyone else around them.

If you really believe that, you are in serious need of a reality check.
Rec.audio.pro is littered by exactly the sort of individuals that you
describe. In fact, you appear to be one of them.


Believe me, when I want to research any acoustic physics topics,
alt.sci.physics.acoustic will be the LAST place I'll look!

An admission of your ingorance, stupidity and obvious lack of discernment.


There's a
reason that it's an "alt" group. I generally like to discuss technical
topics with intelligent, informed adults who identify themselves,
rather than with argumentative, emotionally-stunted trolls who are too
cowardly to post under their own names.

I seriously doubt that you are capable of having an intelligent technical
discussion with anyone. Your comments certainly suggest that you are much
more emotional than intelligent.


snip.....snip

Gee, it's all Bob's fault, isn't it? "HE started it four years ago, so
now I'm obliged to devote the rest of my sorry little life to
following him around the Internet heaping abuse on him wherever he
shows up." How pathetic. Now I understand what you meant by your
comment in your first sentence:

Believe what you will. The fact of the matter is that I am hardly devoting
my life to following Bob Cain around the Internet. Nonetheless, if/when I
periodically check his posts, I will continue to point out his lies and his
hypocrisy.


Room temperature is the temperature of corpses, isn't it? It does seem
like one solution for you, though a bit drastic. Maybe you need a new
hobby or a nice doggy or something. I sure hope you find some way to
restart your long-arrested emotional development, and start to grow up
before you do finally reach room temperature. Who knows - you might be
happy some day!

Thank you for that diagnosis, Dr.Doyle. Too bad that you don't have the
credentials to back it up. In that regard you and Bob Cain have at least
one thing in common.