Home » The PARIS Forums » PARIS: Main » The nut we have to crack
| Re: The nut we have to crack [message #74824 is a reply to message #74810] |
Wed, 25 October 2006 12:59   |
Mic Cross
Messages: 30
Registered: June 2005
|
Member |
|
|
Very interesting stuff! Question: does lowering the amplitude
reduce bit depth/resolution? Or does this not apply here? I remember one
discussion where digital amplitude was related to resolution.
Mic.
"Tony Benson" <tony@standinghampton.com> wrote:
>Neil,
>
>Thanks for posting this. Last night I recorded a test song, drums, guitar,
>and bass in DP. I dropped the individual channel faders to -6.0 and added
a
>limiter (for "make-up" gain and almost no limiting) to the DP main out.
I
>didn't raise any channel fader above -6.0. Only lowered channels to balance
>levels. Even though I only had about 20 channels going, I could already
tell
>it was one of the better sounding mixes I've been able to get out of DP.
Can
>it really be this simple? I was so used to maximizing the levels in PARIS
>that I took that methodology over to DP and my mixes in DP always sounded
>"smaller". Now I'm jazzed about doing some more experimentation in DP.
>Thanks again.
>
>Tony
>
>"Neil" <IOUIU@OIU.com> wrote in message news:453d8006$1@linux...
>>
>> "chuck duffy" <c@c.com> wrote:
>>
>>>If we can't get decent mixes out of a native daw then something is wrong.
>>> Let's find the thing that's wrong, and make it right.
>>
>> (Long, but thought-provoking, and hopefully helpful, rant
>> follows):
>>
>> I think the thing that's wrong is that some people just can't
>> get their heads around the differences between analog & digital.
>> With analog, "big" = hotter, and so hotter is better. When you
>> overbias your tape machines & smack the hell out of the tape,
>> you're getting compression right off the bat on every track you
>> do that with, so one gets used to hearing most tracks with some
>> degree of tape compression already... and we all know that
>> compression can make things sound "bigger". Or, you use a
>> compressor on the way in to the tape so that you get a better
>> SNR, but since that's not an issue with digital (unless you're
>> recording at levels so low that you just simply get poor
>> resolution, but that's a slightly different scenario), people
>> quit using compressors on the way in to digital since SNR isn't
>> an issue there.... you also can't smack an AD convertor hard &
>> expect it to like it - unlike tape. So right off the bat we've
>> got a whole different set of dynamics action going on from one
>> world to the other - then, when you've already got that
>> compressed kick or bassline on tape, you compress it more, and
>> you're compressing an already-compressed signal, so when you
>> apply compression to your uncompressed kick on your DAW you're
>> thinking "nah, that CAN'T be right, it can't need THAT much
>> compression! I'd better back that off a bit!" (because you're
>> looking at the ratios & the threshhold, etc, instead of using
>> your ears). EQ reacts differently with digital, too... if you're
>> used to mixing on a console, you might be used to boosting or
>> cutting something by 3, 4, 6db & getting an audible
>> difference... with digital/plugin EQ's, sometimes you gotta
>> boost or cut HUGE swaths of that frequency to really make a
>> difference... why? I think it's a phase thing... you get more
>> phase shift with analog filters, and so the change is more
>> apparent at smaller degrees of boost & cut. That also helps to
>> isolate things to have their own place in the mix at the same
>> time... considering that phase is the reason we have two ears -
>> it's the thing that makes it possible for us to tell which
>> direction a sound is coming from - this makes perfect sense.
>>
>> So, those of us (and I think that's "most of us here") who cut
>> our teeth in the analog world first, and are used to all the
>> things mentioned above - and who have not changed that style of
>> mixing - could be disappointed in Native systems - not because
>> they fall short of analog or Paris, but because they are
>> actually much more accurate (assuming good quality convertors)
>> & as a result do not impart certain types of coloration that we
>> might interpret as "pleasing". If you could go back to a great
>> mix you did on analog & a console & take out half of the amount
>> of dynamics processing & half of the amount of EQ'ing you did,
>> what would you get? A mix that sounded flatter & more colorless
>> & with less dimension than the one you ended up with. Want
>> proof? Here it is: If you didn't need the amount of EQ &
>> dynamics you applied, you wouldn't have done so! If half the
>> amounts/degrees of those things would have sufficed, that's
>> what you would have used! So Paris sounds & acts kinda like
>> analog, and people who like Paris like that aspect of it... how
>> do we know there's not a few lines of code in there somewhere
>> that adds graduated degrees of even-harmonic distortion when
>> you push the faders or saturate the mix buss to whatever
>> degree? I personally don't think it's strictly a DSP thing,
>> because let's face it.. a plugin is basically doing the same
>> thing to your mix whether it's running of a processor on it's
>> own card or off your CPU; the difference being how well a
>> particular VST or Direct-X compressor or reverb is written (and
>> what it's designed to do in terms of treating the sound) vs.
>> whatever DSP compressor or reverb plugin you're talking about.
>> Can I get an "Amen, brutha!" on that?
>>
>> Chuck's nailed the Paris mix buss thing, it seems, with that
>> -22db at the channel & +22db at the mix buss, but WHY does that
>> make a difference? Well, here's why gang... it's just as I said
>> earlier in another thread - you've got to give yourself some
>> headroom, dammit! Paris apparently does this for you. Want to
>> prove me wrong? Open up a Paris mix and drag the mix buss
>> master fader down 22db from wherever you have it, then insert
>> any plugin that has an output level control on each individual
>> channel of that mix - if the plugin is a compressor, for
>> example, don't use any compression, just use the output
>> control - now boost every channel by 22db using that output
>> control... if it only goes up 10 db, then insert that plugin
>> twice in a row & max out the output on each insertion...
>> that'll be close enough... how's that sound? I'll bet it won't
>> sound all that good! Are you hearing that "overstuffed" mix
>> buss sound? Is it smaller, with less dimension? I'd be curious
>> to see what you guys think if you try this. Now that we know
>> what Chuck told us he discovered, this is the best way to see
>> if that makes a difference or not (my guess - it DOES make a
>> difference, otherwise, they wouldn't have written the code that
>> way!).
>>
>>
>> So how can you get "big" in Native? Give yourself what Paris
>> apparently already gives you... some headroom - think "clean",
>> then dirty it up if you have to later... hell, just mash the
>> mix with a comp & limiter or an L2 or something equivalent -
>> you'll get all the harmonic distortion you want. I wasn't
>> kidding the other day when I said: "Think zen when mixing in
>> Cubase" it's all gotta flow without clips, gang... think about
>> it... if you have one channel getting "overs" in a 32-bit float-
>> point system, you may not notice it... heck you can't notice
>> each sample in a given sound file can you? Of course not. But
>> if you start adding more channels, and each of those channels
>> is running hot... let's say 32 channels - as a comparison
>> for you guys running two-card paris systems & no native mixes.
>> and let's say you're running hot (over zero) about 25% of the
>> time on each channel - that's 352,000 errors PER SECOND across
>> the 32 tracks. That's a lot of floating-point math going on
>> there, isn't it? And in this scenario, I want you to think of
>> each error as a mistake, because that's what it is... in this
>> style of mixing, it's a mistake. How can you expect something
>> that's got 352,000 mistakes per second going on, to sound good?
>>
>> Are you still not convinced? Then you should also definitely
>> investigate running stems (submixes) & reimporting. When I've
>> done this I definitely can hear a difference, and I suspect you
>> most likely will be able to as well.. it is NOT a huge
>> difference, but it's audible. In fact, some months ago I posted
>> a stems mix vs. a non-stems mix & a number of you said you
>> could hear a difference. Now, if you think "aww, this is just
>> another pain-in-the-ass procedure I have to go through if I mix
>> in Native", keep in mind that you can run 90 Million stems
>> mixes in the time it will take Deej to set up his first Pulsar
>> card, and another 900 million in the time that it takes Chuck
>> to research & write that plugin (OK, just giving hell to Deej
>> there, and no really no offense intended to Chucks coding
>> capability, but I'm just saying this is something you can do
>> RIGHT NOW, TONIGHT if you want to if you have a Native system,
>> without having to wait for anything new). Now, if you have a
>> small project - one acoustic guitar, piano, & a vocal - with
>> just a few tracks, running stems won't make a difference, but
>> if you have a large project, give it a shot... you may not hear
>> enough of a difference to make it worth doing in any given
>> instance, but then again, you might.
>>
>> So, now that I hope I've made my case, here's my own personal
>> guidelines for Native mixing - try it out & see wat you think:
>>
>> 1.) Do NOT bring down your Master Fader. It stays at zero
>> (unless you're doing a fade).
>>
>> 2.) On your Master inserts, use a peakstop/brickwall limiter
>> set anywhere from -.03 to -3db, depending on how much headroom
>> you want to give your mastering engineer. Settings for volume
>> maximization & other parameters will, of course, depend on the
>> program material.
>>
>> 3.) Record at 24-bit 88.2k or higher (Dan Lavry has a white
>> paper that makes a good case for a 60k sample rate - in order
>> to get the ringing from the convertors' FIR filters out of the
>> top range of our hearing - but since there is no standard 60k
>> sample rate, 88.2 is the next one up). Also, 16-bit may have
>> worked with Paris for whatever reason (maybe it just enhanced
>> the harmonic distortion you're hearing?), but let's face it,
>> everybody knows that more bits = greater "truth", especially
>> when combined with higher resolutions.
>>
>> 4.) Default your individual channel settings to -6db or lower...
>> I find that -6 is a good place to start because you can load up
>> a decent amount of tracks without overloading the mix buss &
>> hitting your limiter too hard at that level. Consider setting
>> it lower as a starting point if you plan on getting into the
>> range of 40+ tracks. HERE'S THE KEY... if you've got your mix
>> roughed out & you can pull out that peakstop limiter I
>> mentioned in #2 & NOT go over zero on the Master - you're
>> golden. Fuck it, set 'em all at -15 as a starting point if you
>> want, Paris is already setting them for you at -22, right? If
>> you're getting a few scant overs without the limiter, you're
>> still ok, really... the idea is not to overstuff the mix buss
>> so heavily that if you pull the limiter off you're going into
>> the +5, +6 range without it.
>>
>> Think "clean" people = think "no clips" (or as few as
>> possible), you get 30-40 channels of "overs" constantly (like
>> the 352,000 of 'em per second in the example I gave earlier),
>> and it's going to get harsh & thin.... it's a cumulative effect.
>>
>> That's it, really... it's just like any other tool - you can't
>> use an allen wrench to properly drive a nail, and you can't use
>> a hammer to trim your nose hair.
>>
>> Happy Native mixing!
>>
>> (think "zen"!)
>>
>> Neil
>
>
|
|
|
|
|
|
| Re: The nut we have to crack [message #74828 is a reply to message #74824] |
Wed, 25 October 2006 13:44 |
Tony Benson
Messages: 453
Registered: June 2006
|
Senior Member |
|
|
I assume it does Mic, but going by ears, things sounded good. I guess I
don't know for sure how lowering the fader level affects the bit depth in
DP. It is a question I was wondering about also.
Tony
"Mic Cross" <crzymnmchl@comcast.net> wrote in message
news:453fc211$1@linux...
>
> Very interesting stuff! Question: does lowering the amplitude
> reduce bit depth/resolution? Or does this not apply here? I remember one
> discussion where digital amplitude was related to resolution.
>
> Mic.
>
>
> "Tony Benson" <tony@standinghampton.com> wrote:
>>Neil,
>>
>>Thanks for posting this. Last night I recorded a test song, drums, guitar,
>
>>and bass in DP. I dropped the individual channel faders to -6.0 and added
> a
>>limiter (for "make-up" gain and almost no limiting) to the DP main out.
> I
>>didn't raise any channel fader above -6.0. Only lowered channels to
>>balance
>
>>levels. Even though I only had about 20 channels going, I could already
> tell
>>it was one of the better sounding mixes I've been able to get out of DP.
> Can
>>it really be this simple? I was so used to maximizing the levels in PARIS
>
>>that I took that methodology over to DP and my mixes in DP always sounded
>
>>"smaller". Now I'm jazzed about doing some more experimentation in DP.
>>Thanks again.
>>
>>Tony
>>
>>"Neil" <IOUIU@OIU.com> wrote in message news:453d8006$1@linux...
>>>
>>> "chuck duffy" <c@c.com> wrote:
>>>
>>>>If we can't get decent mixes out of a native daw then something is
>>>>wrong.
>>>> Let's find the thing that's wrong, and make it right.
>>>
>>> (Long, but thought-provoking, and hopefully helpful, rant
>>> follows):
>>>
>>> I think the thing that's wrong is that some people just can't
>>> get their heads around the differences between analog & digital.
>>> With analog, "big" = hotter, and so hotter is better. When you
>>> overbias your tape machines & smack the hell out of the tape,
>>> you're getting compression right off the bat on every track you
>>> do that with, so one gets used to hearing most tracks with some
>>> degree of tape compression already... and we all know that
>>> compression can make things sound "bigger". Or, you use a
>>> compressor on the way in to the tape so that you get a better
>>> SNR, but since that's not an issue with digital (unless you're
>>> recording at levels so low that you just simply get poor
>>> resolution, but that's a slightly different scenario), people
>>> quit using compressors on the way in to digital since SNR isn't
>>> an issue there.... you also can't smack an AD convertor hard &
>>> expect it to like it - unlike tape. So right off the bat we've
>>> got a whole different set of dynamics action going on from one
>>> world to the other - then, when you've already got that
>>> compressed kick or bassline on tape, you compress it more, and
>>> you're compressing an already-compressed signal, so when you
>>> apply compression to your uncompressed kick on your DAW you're
>>> thinking "nah, that CAN'T be right, it can't need THAT much
>>> compression! I'd better back that off a bit!" (because you're
>>> looking at the ratios & the threshhold, etc, instead of using
>>> your ears). EQ reacts differently with digital, too... if you're
>>> used to mixing on a console, you might be used to boosting or
>>> cutting something by 3, 4, 6db & getting an audible
>>> difference... with digital/plugin EQ's, sometimes you gotta
>>> boost or cut HUGE swaths of that frequency to really make a
>>> difference... why? I think it's a phase thing... you get more
>>> phase shift with analog filters, and so the change is more
>>> apparent at smaller degrees of boost & cut. That also helps to
>>> isolate things to have their own place in the mix at the same
>>> time... considering that phase is the reason we have two ears -
>>> it's the thing that makes it possible for us to tell which
>>> direction a sound is coming from - this makes perfect sense.
>>>
>>> So, those of us (and I think that's "most of us here") who cut
>>> our teeth in the analog world first, and are used to all the
>>> things mentioned above - and who have not changed that style of
>>> mixing - could be disappointed in Native systems - not because
>>> they fall short of analog or Paris, but because they are
>>> actually much more accurate (assuming good quality convertors)
>>> & as a result do not impart certain types of coloration that we
>>> might interpret as "pleasing". If you could go back to a great
>>> mix you did on analog & a console & take out half of the amount
>>> of dynamics processing & half of the amount of EQ'ing you did,
>>> what would you get? A mix that sounded flatter & more colorless
>>> & with less dimension than the one you ended up with. Want
>>> proof? Here it is: If you didn't need the amount of EQ &
>>> dynamics you applied, you wouldn't have done so! If half the
>>> amounts/degrees of those things would have sufficed, that's
>>> what you would have used! So Paris sounds & acts kinda like
>>> analog, and people who like Paris like that aspect of it... how
>>> do we know there's not a few lines of code in there somewhere
>>> that adds graduated degrees of even-harmonic distortion when
>>> you push the faders or saturate the mix buss to whatever
>>> degree? I personally don't think it's strictly a DSP thing,
>>> because let's face it.. a plugin is basically doing the same
>>> thing to your mix whether it's running of a processor on it's
>>> own card or off your CPU; the difference being how well a
>>> particular VST or Direct-X compressor or reverb is written (and
>>> what it's designed to do in terms of treating the sound) vs.
>>> whatever DSP compressor or reverb plugin you're talking about.
>>> Can I get an "Amen, brutha!" on that?
>>>
>>> Chuck's nailed the Paris mix buss thing, it seems, with that
>>> -22db at the channel & +22db at the mix buss, but WHY does that
>>> make a difference? Well, here's why gang... it's just as I said
>>> earlier in another thread - you've got to give yourself some
>>> headroom, dammit! Paris apparently does this for you. Want to
>>> prove me wrong? Open up a Paris mix and drag the mix buss
>>> master fader down 22db from wherever you have it, then insert
>>> any plugin that has an output level control on each individual
>>> channel of that mix - if the plugin is a compressor, for
>>> example, don't use any compression, just use the output
>>> control - now boost every channel by 22db using that output
>>> control... if it only goes up 10 db, then insert that plugin
>>> twice in a row & max out the output on each insertion...
>>> that'll be close enough... how's that sound? I'll bet it won't
>>> sound all that good! Are you hearing that "overstuffed" mix
>>> buss sound? Is it smaller, with less dimension? I'd be curious
>>> to see what you guys think if you try this. Now that we know
>>> what Chuck told us he discovered, this is the best way to see
>>> if that makes a difference or not (my guess - it DOES make a
>>> difference, otherwise, they wouldn't have written the code that
>>> way!).
>>>
>>>
>>> So how can you get "big" in Native? Give yourself what Paris
>>> apparently already gives you... some headroom - think "clean",
>>> then dirty it up if you have to later... hell, just mash the
>>> mix with a comp & limiter or an L2 or something equivalent -
>>> you'll get all the harmonic distortion you want. I wasn't
>>> kidding the other day when I said: "Think zen when mixing in
>>> Cubase" it's all gotta flow without clips, gang... think about
>>> it... if you have one channel getting "overs" in a 32-bit float-
>>> point system, you may not notice it... heck you can't notice
>>> each sample in a given sound file can you? Of course not. But
>>> if you start adding more channels, and each of those channels
>>> is running hot... let's say 32 channels - as a comparison
>>> for you guys running two-card paris systems & no native mixes.
>>> and let's say you're running hot (over zero) about 25% of the
>>> time on each channel - that's 352,000 errors PER SECOND across
>>> the 32 tracks. That's a lot of floating-point math going on
>>> there, isn't it? And in this scenario, I want you to think of
>>> each error as a mistake, because that's what it is... in this
>>> style of mixing, it's a mistake. How can you expect something
>>> that's got 352,000 mistakes per second going on, to sound good?
>>>
>>> Are you still not convinced? Then you should also definitely
>>> investigate running stems (submixes) & reimporting. When I've
>>> done this I definitely can hear a difference, and I suspect you
>>> most likely will be able to as well.. it is NOT a huge
>>> difference, but it's audible. In fact, some months ago I posted
>>> a stems mix vs. a non-stems mix & a number of you said you
>>> could hear a difference. Now, if you think "aww, this is just
>>> another pain-in-the-ass procedure I have to go through if I mix
>>> in Native", keep in mind that you can run 90 Million stems
>>> mixes in the time it will take Deej to set up his first Pulsar
>>> card, and another 900 million in the time that it takes Chuck
>>> to research & write that plugin (OK, just giving hell to Deej
>>> there, and no really no offense intended to Chucks coding
>>> capability, but I'm just saying this is something you can do
>>> RIGHT NOW, TONIGHT if you want to if you have a Native system,
>>> without having to wait for anything new). Now, if you have a
>>> small project - one acoustic guitar, piano, & a vocal - with
>>> just a few tracks, running stems won't make a difference, but
>>> if you have a large project, give it a shot... you may not hear
>>> enough of a difference to make it worth doing in any given
>>> instance, but then again, you might.
>>>
>>> So, now that I hope I've made my case, here's my own personal
>>> guidelines for Native mixing - try it out & see wat you think:
>>>
>>> 1.) Do NOT bring down your Master Fader. It stays at zero
>>> (unless you're doing a fade).
>>>
>>> 2.) On your Master inserts, use a peakstop/brickwall limiter
>>> set anywhere from -.03 to -3db, depending on how much headroom
>>> you want to give your mastering engineer. Settings for volume
>>> maximization & other parameters will, of course, depend on the
>>> program material.
>>>
>>> 3.) Record at 24-bit 88.2k or higher (Dan Lavry has a white
>>> paper that makes a good case for a 60k sample rate - in order
>>> to get the ringing from the convertors' FIR filters out of the
>>> top range of our hearing - but since there is no standard 60k
>>> sample rate, 88.2 is the next one up). Also, 16-bit may have
>>> worked with Paris for whatever reason (maybe it just enhanced
>>> the harmonic distortion you're hearing?), but let's face it,
>>> everybody knows that more bits = greater "truth", especially
>>> when combined with higher resolutions.
>>>
>>> 4.) Default your individual channel settings to -6db or lower...
>>> I find that -6 is a good place to start because you can load up
>>> a decent amount of tracks without overloading the mix buss &
>>> hitting your limiter too hard at that level. Consider setting
>>> it lower as a starting point if you plan on getting into the
>>> range of 40+ tracks. HERE'S THE KEY... if you've got your mix
>>> roughed out & you can pull out that peakstop limiter I
>>> mentioned in #2 & NOT go over zero on the Master - you're
>>> golden. Fuck it, set 'em all at -15 as a starting point if you
>>> want, Paris is already setting them for you at -22, right? If
>>> you're getting a few scant overs without the limiter, you're
>>> still ok, really... the idea is not to overstuff the mix buss
>>> so heavily that if you pull the limiter off you're going into
>>> the +5, +6 range without it.
>>>
>>> Think "clean" people = think "no clips" (or as few as
>>> possible), you get 30-40 channels of "overs" constantly (like
>>> the 352,000 of 'em per second in the example I gave earlier),
>>> and it's going to get harsh & thin.... it's a cumulative effect.
>>>
>>> That's it, really... it's just like any other tool - you can't
>>> use an allen wrench to properly drive a nail, and you can't use
>>> a hammer to trim your nose hair.
>>>
>>> Happy Native mixing!
>>>
>>> (think "zen"!)
>>>
>>> Neil
>>
>>
>
|
|
|
|
| Re: The nut we have to crack [message #74831 is a reply to message #74828] |
Wed, 25 October 2006 14:31 |
Mic Cross
Messages: 30
Registered: June 2005
|
Member |
|
|
Quote from Dedric a little further down:
"I always thought Paris was harder to get a clear top end out of. Nuendo
sounded clearer to me immediately. Some of that was Paris' converters, some
wasn't. If tracks are being cut by 22dB before you even start processing
you are losing 3.5 bits of resolution from 24-bit files (depending on how
Paris transfers to larger bit depths for processing, and where it lops them
off in the end)."
The 22db cut is at mix stage rather than tracking, right? So I think (would
love to be corrected!) that Dedric is talking about a 3.5 bit loss as Paris
works its magic. Is this right?
Mic.
"Tony Benson" <tony@standinghampton.com> wrote:
>I assume it does Mic, but going by ears, things sounded good. I guess I
>don't know for sure how lowering the fader level affects the bit depth in
>DP. It is a question I was wondering about also.
>
>Tony
>
>
>"Mic Cross" <crzymnmchl@comcast.net> wrote in message
>news:453fc211$1@linux...
>>
>> Very interesting stuff! Question: does lowering the amplitude
>> reduce bit depth/resolution? Or does this not apply here? I remember one
>> discussion where digital amplitude was related to resolution.
>>
>> Mic.
>>
>>
>> "Tony Benson" <tony@standinghampton.com> wrote:
>>>Neil,
>>>
>>>Thanks for posting this. Last night I recorded a test song, drums, guitar,
>>
>>>and bass in DP. I dropped the individual channel faders to -6.0 and added
>> a
>>>limiter (for "make-up" gain and almost no limiting) to the DP main out.
>> I
>>>didn't raise any channel fader above -6.0. Only lowered channels to
>>>balance
>>
>>>levels. Even though I only had about 20 channels going, I could already
>> tell
>>>it was one of the better sounding mixes I've been able to get out of DP.
>> Can
>>>it really be this simple? I was so used to maximizing the levels in PARIS
>>
>>>that I took that methodology over to DP and my mixes in DP always sounded
>>
>>>"smaller". Now I'm jazzed about doing some more experimentation in DP.
>>>Thanks again.
>>>
>>>Tony
>>>
>>>"Neil" <IOUIU@OIU.com> wrote in message news:453d8006$1@linux...
>>>>
>>>> "chuck duffy" <c@c.com> wrote:
>>>>
>>>>>If we can't get decent mixes out of a native daw then something is
>>>>>wrong.
>>>>> Let's find the thing that's wrong, and make it right.
>>>>
>>>> (Long, but thought-provoking, and hopefully helpful, rant
>>>> follows):
>>>>
>>>> I think the thing that's wrong is that some people just can't
>>>> get their heads around the differences between analog & digital.
>>>> With analog, "big" = hotter, and so hotter is better. When you
>>>> overbias your tape machines & smack the hell out of the tape,
>>>> you're getting compression right off the bat on every track you
>>>> do that with, so one gets used to hearing most tracks with some
>>>> degree of tape compression already... and we all know that
>>>> compression can make things sound "bigger". Or, you use a
>>>> compressor on the way in to the tape so that you get a better
>>>> SNR, but since that's not an issue with digital (unless you're
>>>> recording at levels so low that you just simply get poor
>>>> resolution, but that's a slightly different scenario), people
>>>> quit using compressors on the way in to digital since SNR isn't
>>>> an issue there.... you also can't smack an AD convertor hard &
>>>> expect it to like it - unlike tape. So right off the bat we've
>>>> got a whole different set of dynamics action going on from one
>>>> world to the other - then, when you've already got that
>>>> compressed kick or bassline on tape, you compress it more, and
>>>> you're compressing an already-compressed signal, so when you
>>>> apply compression to your uncompressed kick on your DAW you're
>>>> thinking "nah, that CAN'T be right, it can't need THAT much
>>>> compression! I'd better back that off a bit!" (because you're
>>>> looking at the ratios & the threshhold, etc, instead of using
>>>> your ears). EQ reacts differently with digital, too... if you're
>>>> used to mixing on a console, you might be used to boosting or
>>>> cutting something by 3, 4, 6db & getting an audible
>>>> difference... with digital/plugin EQ's, sometimes you gotta
>>>> boost or cut HUGE swaths of that frequency to really make a
>>>> difference... why? I think it's a phase thing... you get more
>>>> phase shift with analog filters, and so the change is more
>>>> apparent at smaller degrees of boost & cut. That also helps to
>>>> isolate things to have their own place in the mix at the same
>>>> time... considering that phase is the reason we have two ears -
>>>> it's the thing that makes it possible for us to tell which
>>>> direction a sound is coming from - this makes perfect sense.
>>>>
>>>> So, those of us (and I think that's "most of us here") who cut
>>>> our teeth in the analog world first, and are used to all the
>>>> things mentioned above - and who have not changed that style of
>>>> mixing - could be disappointed in Native systems - not because
>>>> they fall short of analog or Paris, but because they are
>>>> actually much more accurate (assuming good quality convertors)
>>>> & as a result do not impart certain types of coloration that we
>>>> might interpret as "pleasing". If you could go back to a great
>>>> mix you did on analog & a console & take out half of the amount
>>>> of dynamics processing & half of the amount of EQ'ing you did,
>>>> what would you get? A mix that sounded flatter & more colorless
>>>> & with less dimension than the one you ended up with. Want
>>>> proof? Here it is: If you didn't need the amount of EQ &
>>>> dynamics you applied, you wouldn't have done so! If half the
>>>> amounts/degrees of those things would have sufficed, that's
>>>> what you would have used! So Paris sounds & acts kinda like
>>>> analog, and people who like Paris like that aspect of it... how
>>>> do we know there's not a few lines of code in there somewhere
>>>> that adds graduated degrees of even-harmonic distortion when
>>>> you push the faders or saturate the mix buss to whatever
>>>> degree? I personally don't think it's strictly a DSP thing,
>>>> because let's face it.. a plugin is basically doing the same
>>>> thing to your mix whether it's running of a processor on it's
>>>> own card or off your CPU; the difference being how well a
>>>> particular VST or Direct-X compressor or reverb is written (and
>>>> what it's designed to do in terms of treating the sound) vs.
>>>> whatever DSP compressor or reverb plugin you're talking about.
>>>> Can I get an "Amen, brutha!" on that?
>>>>
>>>> Chuck's nailed the Paris mix buss thing, it seems, with that
>>>> -22db at the channel & +22db at the mix buss, but WHY does that
>>>> make a difference? Well, here's why gang... it's just as I said
>>>> earlier in another thread - you've got to give yourself some
>>>> headroom, dammit! Paris apparently does this for you. Want to
>>>> prove me wrong? Open up a Paris mix and drag the mix buss
>>>> master fader down 22db from wherever you have it, then insert
>>>> any plugin that has an output level control on each individual
>>>> channel of that mix - if the plugin is a compressor, for
>>>> example, don't use any compression, just use the output
>>>> control - now boost every channel by 22db using that output
>>>> control... if it only goes up 10 db, then insert that plugin
>>>> twice in a row & max out the output on each insertion...
>>>> that'll be close enough... how's that sound? I'll bet it won't
>>>> sound all that good! Are you hearing that "overstuffed" mix
>>>> buss sound? Is it smaller, with less dimension? I'd be curious
>>>> to see what you guys think if you try this. Now that we know
>>>> what Chuck told us he discovered, this is the best way to see
>>>> if that makes a difference or not (my guess - it DOES make a
>>>> difference, otherwise, they wouldn't have written the code that
>>>> way!).
>>>>
>>>>
>>>> So how can you get "big" in Native? Give yourself what Paris
>>>> apparently already gives you... some headroom - think "clean",
>>>> then dirty it up if you have to later... hell, just mash the
>>>> mix with a comp & limiter or an L2 or something equivalent -
>>>> you'll get all the harmonic distortion you want. I wasn't
>>>> kidding the other day when I said: "Think zen when mixing in
>>>> Cubase" it's all gotta flow without clips, gang... think about
>>>> it... if you have one channel getting "overs" in a 32-bit float-
>>>> point system, you may not notice it... heck you can't notice
>>>> each sample in a given sound file can you? Of course not. But
>>>> if you start adding more channels, and each of those channels
>>>> is running hot... let's say 32 channels - as a comparison
>>>> for you guys running two-card paris systems & no native mixes.
>>>> and let's say you're running hot (over zero) about 25% of the
>>>> time on each channel - that's 352,000 errors PER SECOND across
>>>> the 32 tracks. That's a lot of floating-point math going on
>>>> there, isn't it? And in this scenario, I want you to think of
>>>> each error as a mistake, because that's what it is... in this
>>>> style of mixing, it's a mistake. How can you expect something
>>>> that's got 352,000 mistakes per second going on, to sound good?
>>>>
>>>> Are you still not convinced? Then you should also definitely
>>>> investigate running stems (submixes) & reimporting. When I've
>>>> done this I definitely can hear a difference, and I suspect you
>>>> most likely will be able to as well.. it is NOT a huge
>>>> difference, but it's audible. In fact, some months ago I posted
>>>> a stems mix vs. a non-stems mix & a number of you said you
>>>> could hear a difference. Now, if you think "aww, this is just
>>>> another pain-in-the-ass procedure I have to go through if I mix
>>>> in Native", keep in mind that you can run 90 Million stems
>>>> mixes in the time it will take Deej to set up his first Pulsar
>>>> card, and another 900 million in the time that it takes Chuck
>>>> to research & write that plugin (OK, just giving hell to Deej
>>>> there, and no really no offense intended to Chucks coding
>>>> capability, but I'm just saying this is something you can do
>>>> RIGHT NOW, TONIGHT if you want to if you have a Native system,
>>>> without having to wait for anything new). Now, if you have a
>>>> small project - one acoustic guitar, piano, & a vocal - with
>>>> just a few tracks, running stems won't make a difference, but
>>>> if you have a large project, give it a shot... you may not hear
>>>> enough of a difference to make it worth doing in any given
>>>> instance, but then again, you might.
>>>>
>>>> So, now that I hope I've made my case, here's my own personal
>>>> guidelines for Native mixing - try it out & see wat you think:
>>>>
>>>> 1.) Do NOT bring down your Master Fader. It stays at zero
>>>> (unless you're doing a fade).
>>>>
>>>> 2.) On your Master inserts, use a peakstop/brickwall limiter
>>>> set anywhere from -.03 to -3db, depending on how much headroom
>>>> you want to give your mastering engineer. Settings for volume
>>>> maximization & other parameters will, of course, depend on the
>>>> program material.
>>>>
>>>> 3.) Record at 24-bit 88.2k or higher (Dan Lavry has a white
>>>> paper that makes a good case for a 60k sample rate - in order
>>>> to get the ringing from the convertors' FIR filters out of the
>>>> top range of our hearing - but since there is no standard 60k
>>>> sample rate, 88.2 is the next one up). Also, 16-bit may have
>>>> worked with Paris for whatever reason (maybe it just enhanced
>>>> the harmonic distortion you're hearing?), but let's face it,
>>>> everybody knows that more bits = greater "truth", especially
>>>> when combined with higher resolutions.
>>>>
>>>> 4.) Default your individual channel settings to -6db or lower...
>>>> I find that -6 is a good place to start because you can load up
>>>> a decent amount of tracks without overloading the mix buss &
>>>> hitting your limiter too hard at that level. Consider setting
>>>> it lower as a starting point if you plan on getting into the
>>>> range of 40+ tracks. HERE'S THE KEY... if you've got your mix
>>>> roughed out & you can pull out that peakstop limiter I
>>>> mentioned in #2 & NOT go over zero on the Master - you're
>>>> golden. Fuck it, set 'em all at -15 as a starting point if you
>>>> want, Paris is already setting them for you at -22, right? If
>>>> you're getting a few scant overs without the limiter, you're
>>>> still ok, really... the idea is not to overstuff the mix buss
>>>> so heavily that if you pull the limiter off you're going into
>>>> the +5, +6 range without it.
>>>>
>>>> Think "clean" people = think "no clips" (or as few as
>>>> possible), you get 30-40 channels of "overs" constantly (like
>>>> the 352,000 of 'em per second in the example I gave earlier),
>>>> and it's going to get harsh & thin.... it's a cumulative effect.
>>>>
>>>> That's it, really... it's just like any other tool - you can't
>>>> use an allen wrench to properly drive a nail, and you can't use
>>>> a hammer to trim your nose hair.
>>>>
>>>> Happy Native mixing!
>>>>
>>>> (think "zen"!)
>>>>
>>>> Neil
>>>
>>>
>>
>
>
|
|
|
|
| Re: The nut we have to crack [message #74832 is a reply to message #74831] |
Wed, 25 October 2006 14:53 |
Tony Benson
Messages: 453
Registered: June 2006
|
Senior Member |
|
|
That's what I understand, but I'm not a tech geek (no offense to the tech
geeks of course) on how different DAW's handle the math involved in changing
gain at the per track (channel) level. Maybe since the math involved is
handled at a higher level (32 bit floating? whatever Integer?) the actual
bit reduction isn't an issue.
I can say that I didn't notice anything strange going on with my little test
recording as far as "graininess" or anything else I would call "low bit"
sounding. It was actually the opposite. I was able to hear more separation
and nuance than mixing at higher channel levels. I know it sounds cliché,
bit I could hear more space around each track. It was much easier to get
things to "sit right" in the mix. I'm going to try this on some higher track
counts and see if it still holds true.
Tony
"Mic Cross" <crzymnmchl@cocmast.net> wrote in message
news:453fd7ae$1@linux...
>
> Quote from Dedric a little further down:
>
> "I always thought Paris was harder to get a clear top end out of. Nuendo
> sounded clearer to me immediately. Some of that was Paris' converters,
> some
> wasn't. If tracks are being cut by 22dB before you even start processing
> you are losing 3.5 bits of resolution from 24-bit files (depending on how
> Paris transfers to larger bit depths for processing, and where it lops
> them
> off in the end)."
>
> The 22db cut is at mix stage rather than tracking, right? So I think
> (would
> love to be corrected!) that Dedric is talking about a 3.5 bit loss as
> Paris
> works its magic. Is this right?
>
> Mic.
>
>
>
> "Tony Benson" <tony@standinghampton.com> wrote:
>>I assume it does Mic, but going by ears, things sounded good. I guess I
>
>>don't know for sure how lowering the fader level affects the bit depth in
>
>>DP. It is a question I was wondering about also.
>>
>>Tony
>>
>>
>>"Mic Cross" <crzymnmchl@comcast.net> wrote in message
>>news:453fc211$1@linux...
>>>
>>> Very interesting stuff! Question: does lowering the amplitude
>>> reduce bit depth/resolution? Or does this not apply here? I remember one
>>> discussion where digital amplitude was related to resolution.
>>>
>>> Mic.
>>>
>>>
>>> "Tony Benson" <tony@standinghampton.com> wrote:
>>>>Neil,
>>>>
>>>>Thanks for posting this. Last night I recorded a test song, drums,
>>>>guitar,
>>>
>>>>and bass in DP. I dropped the individual channel faders to -6.0 and
>>>>added
>>> a
>>>>limiter (for "make-up" gain and almost no limiting) to the DP main out.
>>> I
>>>>didn't raise any channel fader above -6.0. Only lowered channels to
>>>>balance
>>>
>>>>levels. Even though I only had about 20 channels going, I could already
>>> tell
>>>>it was one of the better sounding mixes I've been able to get out of DP.
>>> Can
>>>>it really be this simple? I was so used to maximizing the levels in
>>>>PARIS
>>>
>>>>that I took that methodology over to DP and my mixes in DP always
>>>>sounded
>>>
>>>>"smaller". Now I'm jazzed about doing some more experimentation in DP.
>>>>Thanks again.
>>>>
>>>>Tony
>>>>
>>>>"Neil" <IOUIU@OIU.com> wrote in message news:453d8006$1@linux...
>>>>>
>>>>> "chuck duffy" <c@c.com> wrote:
>>>>>
>>>>>>If we can't get decent mixes out of a native daw then something is
>>>>>>wrong.
>>>>>> Let's find the thing that's wrong, and make it right.
>>>>>
>>>>> (Long, but thought-provoking, and hopefully helpful, rant
>>>>> follows):
>>>>>
>>>>> I think the thing that's wrong is that some people just can't
>>>>> get their heads around the differences between analog & digital.
>>>>> With analog, "big" = hotter, and so hotter is better. When you
>>>>> overbias your tape machines & smack the hell out of the tape,
>>>>> you're getting compression right off the bat on every track you
>>>>> do that with, so one gets used to hearing most tracks with some
>>>>> degree of tape compression already... and we all know that
>>>>> compression can make things sound "bigger". Or, you use a
>>>>> compressor on the way in to the tape so that you get a better
>>>>> SNR, but since that's not an issue with digital (unless you're
>>>>> recording at levels so low that you just simply get poor
>>>>> resolution, but that's a slightly different scenario), people
>>>>> quit using compressors on the way in to digital since SNR isn't
>>>>> an issue there.... you also can't smack an AD convertor hard &
>>>>> expect it to like it - unlike tape. So right off the bat we've
>>>>> got a whole different set of dynamics action going on from one
>>>>> world to the other - then, when you've already got that
>>>>> compressed kick or bassline on tape, you compress it more, and
>>>>> you're compressing an already-compressed signal, so when you
>>>>> apply compression to your uncompressed kick on your DAW you're
>>>>> thinking "nah, that CAN'T be right, it can't need THAT much
>>>>> compression! I'd better back that off a bit!" (because you're
>>>>> looking at the ratios & the threshhold, etc, instead of using
>>>>> your ears). EQ reacts differently with digital, too... if you're
>>>>> used to mixing on a console, you might be used to boosting or
>>>>> cutting something by 3, 4, 6db & getting an audible
>>>>> difference... with digital/plugin EQ's, sometimes you gotta
>>>>> boost or cut HUGE swaths of that frequency to really make a
>>>>> difference... why? I think it's a phase thing... you get more
>>>>> phase shift with analog filters, and so the change is more
>>>>> apparent at smaller degrees of boost & cut. That also helps to
>>>>> isolate things to have their own place in the mix at the same
>>>>> time... considering that phase is the reason we have two ears -
>>>>> it's the thing that makes it possible for us to tell which
>>>>> direction a sound is coming from - this makes perfect sense.
>>>>>
>>>>> So, those of us (and I think that's "most of us here") who cut
>>>>> our teeth in the analog world first, and are used to all the
>>>>> things mentioned above - and who have not changed that style of
>>>>> mixing - could be disappointed in Native systems - not because
>>>>> they fall short of analog or Paris, but because they are
>>>>> actually much more accurate (assuming good quality convertors)
>>>>> & as a result do not impart certain types of coloration that we
>>>>> might interpret as "pleasing". If you could go back to a great
>>>>> mix you did on analog & a console & take out half of the amount
>>>>> of dynamics processing & half of the amount of EQ'ing you did,
>>>>> what would you get? A mix that sounded flatter & more colorless
>>>>> & with less dimension than the one you ended up with. Want
>>>>> proof? Here it is: If you didn't need the amount of EQ &
>>>>> dynamics you applied, you wouldn't have done so! If half the
>>>>> amounts/degrees of those things would have sufficed, that's
>>>>> what you would have used! So Paris sounds & acts kinda like
>>>>> analog, and people who like Paris like that aspect of it... how
>>>>> do we know there's not a few lines of code in there somewhere
>>>>> that adds graduated degrees of even-harmonic distortion when
>>>>> you push the faders or saturate the mix buss to whatever
>>>>> degree? I personally don't think it's strictly a DSP thing,
>>>>> because let's face it.. a plugin is basically doing the same
>>>>> thing to your mix whether it's running of a processor on it's
>>>>> own card or off your CPU; the difference being how well a
>>>>> particular VST or Direct-X compressor or reverb is written (and
>>>>> what it's designed to do in terms of treating the sound) vs.
>>>>> whatever DSP compressor or reverb plugin you're talking about.
>>>>> Can I get an "Amen, brutha!" on that?
>>>>>
>>>>> Chuck's nailed the Paris mix buss thing, it seems, with that
>>>>> -22db at the channel & +22db at the mix buss, but WHY does that
>>>>> make a difference? Well, here's why gang... it's just as I said
>>>>> earlier in another thread - you've got to give yourself some
>>>>> headroom, dammit! Paris apparently does this for you. Want to
>>>>> prove me wrong? Open up a Paris mix and drag the mix buss
>>>>> master fader down 22db from wherever you have it, then insert
>>>>> any plugin that has an output level control on each individual
>>>>> channel of that mix - if the plugin is a compressor, for
>>>>> example, don't use any compression, just use the output
>>>>> control - now boost every channel by 22db using that output
>>>>> control... if it only goes up 10 db, then insert that plugin
>>>>> twice in a row & max out the output on each insertion...
>>>>> that'll be close enough... how's that sound? I'll bet it won't
>>>>> sound all that good! Are you hearing that "overstuffed" mix
>>>>> buss sound? Is it smaller, with less dimension? I'd be curious
>>>>> to see what you guys think if you try this. Now that we know
>>>>> what Chuck told us he discovered, this is the best way to see
>>>>> if that makes a difference or not (my guess - it DOES make a
>>>>> difference, otherwise, they wouldn't have written the code that
>>>>> way!).
>>>>>
>>>>>
>>>>> So how can you get "big" in Native? Give yourself what Paris
>>>>> apparently already gives you... some headroom - think "clean",
>>>>> then dirty it up if you have to later... hell, just mash the
>>>>> mix with a comp & limiter or an L2 or something equivalent -
>>>>> you'll get all the harmonic distortion you want. I wasn't
>>>>> kidding the other day when I said: "Think zen when mixing in
>>>>> Cubase" it's all gotta flow without clips, gang... think about
>>>>> it... if you have one channel getting "overs" in a 32-bit float-
>>>>> point system, you may not notice it... heck you can't notice
>>>>> each sample in a given sound file can you? Of course not. But
>>>>> if you start adding more channels, and each of those channels
>>>>> is running hot... let's say 32 channels - as a comparison
>>>>> for you guys running two-card paris systems & no native mixes.
>>>>> and let's say you're running hot (over zero) about 25% of the
>>>>> time on each channel - that's 352,000 errors PER SECOND across
>>>>> the 32 tracks. That's a lot of floating-point math going on
>>>>> there, isn't it? And in this scenario, I want you to think of
>>>>> each error as a mistake, because that's what it is... in this
>>>>> style of mixing, it's a mistake. How can you expect something
>>>>> that's got 352,000 mistakes per second going on, to sound good?
>>>>>
>>>>> Are you still not convinced? Then you should also definitely
>>>>> investigate running stems (submixes) & reimporting. When I've
>>>>> done this I definitely can hear a difference, and I suspect you
>>>>> most likely will be able to as well.. it is NOT a huge
>>>>> difference, but it's audible. In fact, some months ago I posted
>>>>> a stems mix vs. a non-stems mix & a number of you said you
>>>>> could hear a difference. Now, if you think "aww, this is just
>>>>> another pain-in-the-ass procedure I have to go through if I mix
>>>>> in Native", keep in mind that you can run 90 Million stems
>>>>> mixes in the time it will take Deej to set up his first Pulsar
>>>>> card, and another 900 million in the time that it takes Chuck
>>>>> to research & write that plugin (OK, just giving hell to Deej
>>>>> there, and no really no offense intended to Chucks coding
>>>>> capability, but I'm just saying this is something you can do
>>>>> RIGHT NOW, TONIGHT if you want to if you have a Native system,
>>>>> without having to wait for anything new). Now, if you have a
>>>>> small project - one acoustic guitar, piano, & a vocal - with
>>>>> just a few tracks, running stems won't make a difference, but
>>>>> if you have a large project, give it a shot... you may not hear
>>>>> enough of a difference to make it worth doing in any given
>>>>> instance, but then again, you might.
>>>>>
>>>>> So, now that I hope I've made my case, here's my own personal
>>>>> guidelines for Native mixing - try it out & see wat you think:
>>>>>
>>>>> 1.) Do NOT bring down your Master Fader. It stays at zero
>>>>> (unless you're doing a fade).
>>>>>
>>>>> 2.) On your Master inserts, use a peakstop/brickwall limiter
>>>>> set anywhere from -.03 to -3db, depending on how much headroom
>>>>> you want to give your mastering engineer. Settings for volume
>>>>> maximization & other parameters will, of course, depend on the
>>>>> program material.
>>>>>
>>>>> 3.) Record at 24-bit 88.2k or higher (Dan Lavry has a white
>>>>> paper that makes a good case for a 60k sample rate - in order
>>>>> to get the ringing from the convertors' FIR filters out of the
>>>>> top range of our hearing - but since there is no standard 60k
>>>>> sample rate, 88.2 is the next one up). Also, 16-bit may have
>>>>> worked with Paris for whatever reason (maybe it just enhanced
>>>>> the harmonic distortion you're hearing?), but let's face it,
>>>>> everybody knows that more bits = greater "truth", especially
>>>>> when combined with higher resolutions.
>>>>>
>>>>> 4.) Default your individual channel settings to -6db or lower...
>>>>> I find that -6 is a good place to start because you can load up
>>>>> a decent amount of tracks without overloading the mix buss &
>>>>> hitting your limiter too hard at that level. Consider setting
>>>>> it lower as a starting point if you plan on getting into the
>>>>> range of 40+ tracks. HERE'S THE KEY... if you've got your mix
>>>>> roughed out & you can pull out that peakstop limiter I
>>>>> mentioned in #2 & NOT go over zero on the Master - you're
>>>>> golden. Fuck it, set 'em all at -15 as a starting point if you
>>>>> want, Paris is already setting them for you at -22, right? If
>>>>> you're getting a few scant overs without the limiter, you're
>>>>> still ok, really... the idea is not to overstuff the mix buss
>>>>> so heavily that if you pull the limiter off you're going into
>>>>> the +5, +6 range without it.
>>>>>
>>>>> Think "clean" people = think "no clips" (or as few as
>>>>> possible), you get 30-40 channels of "overs" constantly (like
>>>>> the 352,000 of 'em per second in the example I gave earlier),
>>>>> and it's going to get harsh & thin.... it's a cumulative effect.
>>>>>
>>>>> That's it, really... it's just like any other tool - you can't
>>>>> use an allen wrench to properly drive a nail, and you can't use
>>>>> a hammer to trim your nose hair.
>>>>>
>>>>> Happy Native mixing!
>>>>>
>>>>> (think "zen"!)
>>>>>
>>>>> Neil
>>>>
>>>>
>>>
>>
>>
>
|
|
|
|
| Re: The nut we have to crack [message #74867 is a reply to message #74832] |
Thu, 26 October 2006 07:39 |
Dedric Terry
Messages: 788
Registered: June 2007
|
Senior Member |
|
|
Actually the 3.5 bit loss assumes either of two situations:
1) I'm guessing Chuck was referring to the EDS code, so if the gain
reduction happens (for some unknown reason) after reading the file off of
disk, and before pushing it into the higher bit depth processing section,
then it would pad 0's for any extra bits beyond 24.
2) More likely, if you reduce *all* of your tracks by 22dB, sum them, reduce
the master fader (as many might), then you could effectively have some
tracks lose their original lower bits simply because that is all pushed down
below 23-bits in the sum before sent back out as a 24-bit stream.
Next assumption - we can actually hear -122dB. :-) That's where this is
happening.
Really this isn't a big deal - what bothers me about the concept of every
track being reduced by 22dB without the express written consent of the
engineer/mixer is that it is misleading and presupposing you need to reduce
track gain to get the mix to work.
In any given large mix, I may actually end up with many tracks down by
15-25dB in order to keep the master in the right range, but it's easier to
make that choice based on what the song, the tracks and the mix need. For
sure it seems to work in Paris to some degree. But if you start knowing how
your mix should sound (hearing it mentally), and how each track should fit
into that, it's easy to create that sonic space with most any mixing medium.
That's where the argument about one DAW mixing better than another falls
down for me - it says the engineer is letting the medium dictate the mix
rather than the engineer. That isn't engineering.
Regardless of what you mix in, there is only 40Hz to about 17kHz of actual
human listening/hearing range in the final product, and only 0dBFS of max
level, and -96dB of min level. That's the space we have to work with, and
only so much can fit in there. DAWs don't prevent music from fitting in
that comparatively small range, people do.
The point really is that this technical discovery about Paris says one and
only one thing:
* When you mix digitally, control the levels of your tracks to fit the mix
rather than assuming you can just push up faders and have each track find
it's own space automatically *
Just my opinion,
Dedric
PS: We are in the midst of a blizzard here - about 10" on the ground now
with winds up to and over 40mph.
On 10/25/06 3:53 PM, in article 453fdae4@linux, "Tony Benson"
<tony@standinghampton.com> wrote:
> That's what I understand, but I'm not a tech geek (no offense to the tech
> geeks of course) on how different DAW's handle the math involved in changing
> gain at the per track (channel) level. Maybe since the math involved is
> handled at a higher level (32 bit floating? whatever Integer?) the actual
> bit reduction isn't an issue.
>
> I can say that I didn't notice anything strange going on with my little test
> recording as far as "graininess" or anything else I would call "low bit"
> sounding. It was actually the opposite. I was able to hear more separation
> and nuance than mixing at higher channel levels. I know it sounds cliché,
> bit I could hear more space around each track. It was much easier to get
> things to "sit right" in the mix. I'm going to try this on some higher track
> counts and see if it still holds true.
>
> Tony
>
>
> "Mic Cross" <crzymnmchl@cocmast.net> wrote in message
> news:453fd7ae$1@linux...
>>
>> Quote from Dedric a little further down:
>>
>> "I always thought Paris was harder to get a clear top end out of. Nuendo
>> sounded clearer to me immediately. Some of that was Paris' converters,
>> some
>> wasn't. If tracks are being cut by 22dB before you even start processing
>> you are losing 3.5 bits of resolution from 24-bit files (depending on how
>> Paris transfers to larger bit depths for processing, and where it lops
>> them
>> off in the end)."
>>
>> The 22db cut is at mix stage rather than tracking, right? So I think
>> (would
>> love to be corrected!) that Dedric is talking about a 3.5 bit loss as
>> Paris
>> works its magic. Is this right?
>>
>> Mic.
>>
>
|
|
|
|
| Re: The nut we have to crack [message #74871 is a reply to message #74804] |
Thu, 26 October 2006 08:19 |
Dedric Terry
Messages: 788
Registered: June 2007
|
Senior Member |
|
|
Hi Lamont,
Yes, there is a difference between how analog handles "headroom", and how
digital does, but there is no difference between how digital desks and DAWs
handle it - they are all limited to 0dBFS for the actual digital data that
passes through. There may certainly be differences in how a digital desk
manages the digital path, or how you mix on it, but that doesn't necessarily
mean it's more than 24-bit all the way through. Unless a desk is completely
mixing within a cpu to maintain full floating point math, it will be fixed
point, either 24 or 48 bit for most of the path - the same as TC Powercore
or ProTools.
As far as comparing stereo wav files, if there is a difference with one DAW
vs. others, the one isn't representing the stereo track correctly. I can
open up any stereo wav file in SX, Nuendo, Sequoia, Vegas, and even iTunes
(since all of my systems are piped to the same playback converters), and all
sound identical, regardless if the original track was mixed on an SSL,
analog, ProTools, or any other DAW.
Now, if a DAW only supports mono files (e.g. Paris and ProTools), converting
an interleaved stereo wav file could sound different when played back in
dual mono, but that would likely be to a difference in pan law between the
source and the playback DAW, or alignment issues. I have heard this happen
(in Paris I believe) - the two channels sound wider, but the middle sounds
disconnected and almost "missing" with dual mono files for a stereo track,
where it sounds a little less wide but coherent across the middle as
interleaved stereo. I wouldn't say this *should* be the case with
interleaved stereo vs. dual mono, but it could be.
Obviously your workflow works great for you. Mine works great for me.
While a lot of engineering has a consistent basic technical methodology,
personal preference still plays a significant role. Now if we could just
get the "engineers" (aka artists' brothers in law, best friends, etc) that
are putting bad mixes on the radio to connect the technical basics with
inexperienced preference, we might be able to tune into listenable music
again some day...
Regards,
Dedric
On 10/25/06 8:40 AM, in article 453f775c$1@linux, "LaMont"
<jjdpro@ameritech.net> wrote:
>
> Dedric,
> My point has moe to do with 'head-room' of ITB mixes versus, using a analog
> or digital mixer for summing.
>
> There is a difference. Also, I challege anyone to open up say SX, DP, Logic
> and play a stereo way file @ unity gain ..then, If you have copy of say
> Pr-Tools
> LE M-powered, import that same file.. Then listen.
> You can here the difference, even using the same audio interface..
>
> I agree with you that you have to mix differently using the natives, but
> Soft ware has a sound.. To me and others, to get make SX/Nuendo slam at it's
> best, is to using a outboard summing mixer.
>
> These days, my work flow is to record,edit,then bounce stems from Nuendo.
> Simply put, there is no better workflow DAW on the planet for such tasks.
> Then, I either mix in Pro-Tools or Paris depending on the color I'm going
> for.
>
> Dedric Terry <dterry@keyofd.net> wrote:
>> Lamont - if your D-A converters affect the way you mix inside a DAW, you
>> aren't mixing what you think you are. Certainly converters can sound
>> different, but the differences at the RME/Apogee level aren't in siginficant
>> areas (mainly a slight difference in sound of the top end - yes I've heard
>> all of these, along with Myteks, Cranesong, DCS, and others side by side
> -
>> Cranesong is my favorite - Myteks are great, but a little sterile. DCS is
>> just too expensive).
>>
>> 1) If you are saying you mix differently on a console because you are using
>> Apogee converters from the DAW with soft limit vs. RME converters, also
> into
>> the same desk (no mixing in SX, just playback), you are simply using the
>> converters to color the signal (albeit only slightly), in different ways
> -
>> Softlimit just limiting. Nothing wrong with that, but that is altering
> the
>> tracks going in, not the mixing platform itself. Saying RME converters
>> limit you because they don't have a limiter built in says you aren't mixing
>> the way most of us do - you are trying to get analog saturation out of
>> digital - ain't gonna happen.
>>
>> 2) If you are mixing inside SX and change your approach depending on which
>> converter you monitor through, then that's a problem since your bounces
>> aren't going to be the same, and your decisions aren't going to be
>> consistent. The idea is to have your monitoring chain *not* affect your
> mix
>> decisions, but enable more accurate ones.
>>
>> If you mixdown to a 2-track of some sort (Masterlink, etc), then you are
>> using SoftLimit as a limiter on the output. You could achieve the same
>> think in a multitude of different ways.
>>
>> Regards,
>> Dedric
>>
>> On 10/24/06 1:44 PM, in article 453e6d24$1@linux, "LaMont"
>> <jjdpro@ameritech.net> wrote:
>>
>>>
>>> Neil I do mix follow the native mix rules. No overs, faders around -5db
> ect,
>>> and I can make it sound good..
>>>
>>> However, when I add in a mixer for summing, all of those native mixing
> rules
>>> are out the window. The whole mix "sonically" opens up..
>>>
>>> As well as, If I'm using Apogees AD16x/DA16x with soft-limiter set on,
> I
>>> can mix like I want to in SX. With RME interface's and converters, I have
>>> to abide by the rules.
>>>
>>> Lastly, when i have to mix (In the Box) using SX/Nuendo, I refer to the
>>> Charles
>>> dye method and add in Harmonic distortion via plugs in (namely) antares
> Mic
>>> modler(tube) on the inserts. This gives a different texture to the faders.
>>> These days,I just use the SSL plugs which have that harmonic distort color
>>> that helps a native mix...
>>> "Neil" <OIUOIU@OIU.com> wrote:
>>>>
>>>> "LaMont" <jjdpro@ameritech.net> wrote:
>>>>>
>>>>> My Point exactly.. If all of you who use Nuendo or Cubase cannot hear
> that
>>>>> there is something going on (software-wise) in Cubase or Nuendo that's
>>> not
>>>>> bringing "Full-life" to our wav files, then,I'm sorry, your ears are
> not
>>>>> as good as you may think..
>>>>
>>>> There IS something going on... IME, I think that a lot of people
>>>> are using the tool in a manner in which it was not designed for.
>>>> It's not designed to accomodate 50 tracks worth of clips/overs
>>>> resulting in hundreds of thousands of errors per second... it's
>>>> as simple as that.
>>>> I don't think anyone who's said you can get good mixes out of
>>>> Native suystems has insisted that it sounds exactly like Paris
>>>> (or PT, or analog, or anything else), so is something different
>>>> going on? Yeah... it's different - doesn't mean that it can't be
>>>> good.
>>>
>>
>
|
|
|
|
| Re: The nut we have to crack [message #74887 is a reply to message #74867] |
Thu, 26 October 2006 11:01 |
Tony Benson
Messages: 453
Registered: June 2006
|
Senior Member |
|
|
Thanks Dedric. I really don't consider myself to be an "engineer". I'm a
songwriter/musician who wants record professional sounding tracks in my home
studio. I've been into mixing both live and in the studio for 20 years or
so, but am completely self taught. Any "real" knowledge is always
appreciated. Anyway, this new approach (for me anyway) might just let me
actually get the kind of mixes I hear in my head out of DP.
Tony
"Dedric Terry" <dterry@keyofd.net> wrote in message
news:C16624A7.4B4C%dterry@keyofd.net...
> Actually the 3.5 bit loss assumes either of two situations:
>
> 1) I'm guessing Chuck was referring to the EDS code, so if the gain
> reduction happens (for some unknown reason) after reading the file off of
> disk, and before pushing it into the higher bit depth processing section,
> then it would pad 0's for any extra bits beyond 24.
>
> 2) More likely, if you reduce *all* of your tracks by 22dB, sum them,
> reduce
> the master fader (as many might), then you could effectively have some
> tracks lose their original lower bits simply because that is all pushed
> down
> below 23-bits in the sum before sent back out as a 24-bit stream.
>
> Next assumption - we can actually hear -122dB. :-) That's where this is
> happening.
>
> Really this isn't a big deal - what bothers me about the concept of every
> track being reduced by 22dB without the express written consent of the
> engineer/mixer is that it is misleading and presupposing you need to
> reduce
> track gain to get the mix to work.
>
> In any given large mix, I may actually end up with many tracks down by
> 15-25dB in order to keep the master in the right range, but it's easier to
> make that choice based on what the song, the tracks and the mix need. For
> sure it seems to work in Paris to some degree. But if you start knowing
> how
> your mix should sound (hearing it mentally), and how each track should fit
> into that, it's easy to create that sonic space with most any mixing
> medium.
> That's where the argument about one DAW mixing better than another falls
> down for me - it says the engineer is letting the medium dictate the mix
> rather than the engineer. That isn't engineering.
>
> Regardless of what you mix in, there is only 40Hz to about 17kHz of actual
> human listening/hearing range in the final product, and only 0dBFS of max
> level, and -96dB of min level. That's the space we have to work with, and
> only so much can fit in there. DAWs don't prevent music from fitting in
> that comparatively small range, people do.
>
> The point really is that this technical discovery about Paris says one and
> only one thing:
>
> * When you mix digitally, control the levels of your tracks to fit the mix
> rather than assuming you can just push up faders and have each track find
> it's own space automatically *
>
> Just my opinion,
> Dedric
>
> PS: We are in the midst of a blizzard here - about 10" on the ground now
> with winds up to and over 40mph.
>
> On 10/25/06 3:53 PM, in article 453fdae4@linux, "Tony Benson"
> <tony@standinghampton.com> wrote:
>
>> That's what I understand, but I'm not a tech geek (no offense to the tech
>> geeks of course) on how different DAW's handle the math involved in
>> changing
>> gain at the per track (channel) level. Maybe since the math involved is
>> handled at a higher level (32 bit floating? whatever Integer?) the actual
>> bit reduction isn't an issue.
>>
>> I can say that I didn't notice anything strange going on with my little
>> test
>> recording as far as "graininess" or anything else I would call "low bit"
>> sounding. It was actually the opposite. I was able to hear more
>> separation
>> and nuance than mixing at higher channel levels. I know it sounds cliché,
>> bit I could hear more space around each track. It was much easier to get
>> things to "sit right" in the mix. I'm going to try this on some higher
>> track
>> counts and see if it still holds true.
>>
>> Tony
>>
>>
>> "Mic Cross" <crzymnmchl@cocmast.net> wrote in message
>> news:453fd7ae$1@linux...
>>>
>>> Quote from Dedric a little further down:
>>>
>>> "I always thought Paris was harder to get a clear top end out of.
>>> Nuendo
>>> sounded clearer to me immediately. Some of that was Paris' converters,
>>> some
>>> wasn't. If tracks are being cut by 22dB before you even start
>>> processing
>>> you are losing 3.5 bits of resolution from 24-bit files (depending on
>>> how
>>> Paris transfers to larger bit depths for processing, and where it lops
>>> them
>>> off in the end)."
>>>
>>> The 22db cut is at mix stage rather than tracking, right? So I think
>>> (would
>>> love to be corrected!) that Dedric is talking about a 3.5 bit loss as
>>> Paris
>>> works its magic. Is this right?
>>>
>>> Mic.
>>>
>>
>
|
|
|
|
|
|
| Re: The nut we have to crack [message #74991 is a reply to message #74713] |
Fri, 27 October 2006 21:19 |
Martin Harrington
Messages: 560
Registered: September 2005
|
Senior Member |
|
|
Thanks for that well thought out "rant", Neil.
Thats pretty much the way I feel with Nuendo, although I don't put an
abitary setting on each channel, (probably because I''m only recording a
couple of tracks at a time,) but I certainly record to within about 8dbfs
from 0, and always put a limiter on the output,
--
Martin Harrington
www.lendanear-sound.com
"Neil" <IOUIU@OIU.com> wrote in message news:453d8006$1@linux...
>
> "chuck duffy" <c@c.com> wrote:
>
>>If we can't get decent mixes out of a native daw then something is wrong.
>> Let's find the thing that's wrong, and make it right.
>
> (Long, but thought-provoking, and hopefully helpful, rant
> follows):
>
> I think the thing that's wrong is that some people just can't
> get their heads around the differences between analog & digital.
> With analog, "big" = hotter, and so hotter is better. When you
> overbias your tape machines & smack the hell out of the tape,
> you're getting compression right off the bat on every track you
> do that with, so one gets used to hearing most tracks with some
> degree of tape compression already... and we all know that
> compression can make things sound "bigger". Or, you use a
> compressor on the way in to the tape so that you get a better
> SNR, but since that's not an issue with digital (unless you're
> recording at levels so low that you just simply get poor
> resolution, but that's a slightly different scenario), people
> quit using compressors on the way in to digital since SNR isn't
> an issue there.... you also can't smack an AD convertor hard &
> expect it to like it - unlike tape. So right off the bat we've
> got a whole different set of dynamics action going on from one
> world to the other - then, when you've already got that
> compressed kick or bassline on tape, you compress it more, and
> you're compressing an already-compressed signal, so when you
> apply compression to your uncompressed kick on your DAW you're
> thinking "nah, that CAN'T be right, it can't need THAT much
> compression! I'd better back that off a bit!" (because you're
> looking at the ratios & the threshhold, etc, instead of using
> your ears). EQ reacts differently with digital, too... if you're
> used to mixing on a console, you might be used to boosting or
> cutting something by 3, 4, 6db & getting an audible
> difference... with digital/plugin EQ's, sometimes you gotta
> boost or cut HUGE swaths of that frequency to really make a
> difference... why? I think it's a phase thing... you get more
> phase shift with analog filters, and so the change is more
> apparent at smaller degrees of boost & cut. That also helps to
> isolate things to have their own place in the mix at the same
> time... considering that phase is the reason we have two ears -
> it's the thing that makes it possible for us to tell which
> direction a sound is coming from - this makes perfect sense.
>
> So, those of us (and I think that's "most of us here") who cut
> our teeth in the analog world first, and are used to all the
> things mentioned above - and who have not changed that style of
> mixing - could be disappointed in Native systems - not because
> they fall short of analog or Paris, but because they are
> actually much more accurate (assuming good quality convertors)
> & as a result do not impart certain types of coloration that we
> might interpret as "pleasing". If you could go back to a great
> mix you did on analog & a console & take out half of the amount
> of dynamics processing & half of the amount of EQ'ing you did,
> what would you get? A mix that sounded flatter & more colorless
> & with less dimension than the one you ended up with. Want
> proof? Here it is: If you didn't need the amount of EQ &
> dynamics you applied, you wouldn't have done so! If half the
> amounts/degrees of those things would have sufficed, that's
> what you would have used! So Paris sounds & acts kinda like
> analog, and people who like Paris like that aspect of it... how
> do we know there's not a few lines of code in there somewhere
> that adds graduated degrees of even-harmonic distortion when
> you push the faders or saturate the mix buss to whatever
> degree? I personally don't think it's strictly a DSP thing,
> because let's face it.. a plugin is basically doing the same
> thing to your mix whether it's running of a processor on it's
> own card or off your CPU; the difference being how well a
> particular VST or Direct-X compressor or reverb is written (and
> what it's designed to do in terms of treating the sound) vs.
> whatever DSP compressor or reverb plugin you're talking about.
> Can I get an "Amen, brutha!" on that?
>
> Chuck's nailed the Paris mix buss thing, it seems, with that
> -22db at the channel & +22db at the mix buss, but WHY does that
> make a difference? Well, here's why gang... it's just as I said
> earlier in another thread - you've got to give yourself some
> headroom, dammit! Paris apparently does this for you. Want to
> prove me wrong? Open up a Paris mix and drag the mix buss
> master fader down 22db from wherever you have it, then insert
> any plugin that has an output level control on each individual
> channel of that mix - if the plugin is a compressor, for
> example, don't use any compression, just use the output
> control - now boost every channel by 22db using that output
> control... if it only goes up 10 db, then insert that plugin
> twice in a row & max out the output on each insertion...
> that'll be close enough... how's that sound? I'll bet it won't
> sound all that good! Are you hearing that "overstuffed" mix
> buss sound? Is it smaller, with less dimension? I'd be curious
> to see what you guys think if you try this. Now that we know
> what Chuck told us he discovered, this is the best way to see
> if that makes a difference or not (my guess - it DOES make a
> difference, otherwise, they wouldn't have written the code that
> way!).
>
>
> So how can you get "big" in Native? Give yourself what Paris
> apparently already gives you... some headroom - think "clean",
> then dirty it up if you have to later... hell, just mash the
> mix with a comp & limiter or an L2 or something equivalent -
> you'll get all the harmonic distortion you want. I wasn't
> kidding the other day when I said: "Think zen when mixing in
> Cubase" it's all gotta flow without clips, gang... think about
> it... if you have one channel getting "overs" in a 32-bit float-
> point system, you may not notice it... heck you can't notice
> each sample in a given sound file can you? Of course not. But
> if you start adding more channels, and each of those channels
> is running hot... let's say 32 channels - as a comparison
> for you guys running two-card paris systems & no native mixes.
> and let's say you're running hot (over zero) about 25% of the
> time on each channel - that's 352,000 errors PER SECOND across
> the 32 tracks. That's a lot of floating-point math going on
> there, isn't it? And in this scenario, I want you to think of
> each error as a mistake, because that's what it is... in this
> style of mixing, it's a mistake. How can you expect something
> that's got 352,000 mistakes per second going on, to sound good?
>
> Are you still not convinced? Then you should also definitely
> investigate running stems (submixes) & reimporting. When I've
> done this I definitely can hear a difference, and I suspect you
> most likely will be able to as well.. it is NOT a huge
> difference, but it's audible. In fact, some months ago I posted
> a stems mix vs. a non-stems mix & a number of you said you
> could hear a difference. Now, if you think "aww, this is just
> another pain-in-the-ass procedure I have to go through if I mix
> in Native", keep in mind that you can run 90 Million stems
> mixes in the time it will take Deej to set up his first Pulsar
> card, and another 900 million in the time that it takes Chuck
> to research & write that plugin (OK, just giving hell to Deej
> there, and no really no offense intended to Chucks coding
> capability, but I'm just saying this is something you can do
> RIGHT NOW, TONIGHT if you want to if you have a Native system,
> without having to wait for anything new). Now, if you have a
> small project - one acoustic guitar, piano, & a vocal - with
> just a few tracks, running stems won't make a difference, but
> if you have a large project, give it a shot... you may not hear
> enough of a difference to make it worth doing in any given
> instance, but then again, you might.
>
> So, now that I hope I've made my case, here's my own personal
> guidelines for Native mixing - try it out & see wat you think:
>
> 1.) Do NOT bring down your Master Fader. It stays at zero
> (unless you're doing a fade).
>
> 2.) On your Master inserts, use a peakstop/brickwall limiter
> set anywhere from -.03 to -3db, depending on how much headroom
> you want to give your mastering engineer. Settings for volume
> maximization & other parameters will, of course, depend on the
> program material.
>
> 3.) Record at 24-bit 88.2k or higher (Dan Lavry has a white
> paper that makes a good case for a 60k sample rate - in order
> to get the ringing from the convertors' FIR filters out of the
> top range of our hearing - but since there is no standard 60k
> sample rate, 88.2 is the next one up). Also, 16-bit may have
> worked with Paris for whatever reason (maybe it just enhanced
> the harmonic distortion you're hearing?), but let's face it,
> everybody knows that more bits = greater "truth", especially
> when combined with higher resolutions.
>
> 4.) Default your individual channel settings to -6db or lower...
> I find that -6 is a good place to start because you can load up
> a decent amount of tracks without overloading the mix buss &
> hitting your limiter too hard at that level. Consider setting
> it lower as a starting point if you plan on getting into the
> range of 40+ tracks. HERE'S THE KEY... if you've got your mix
> roughed out & you can pull out that peakstop limiter I
> mentioned in #2 & NOT go over zero on the Master - you're
> golden. Fuck it, set 'em all at -15 as a starting point if you
> want, Paris is already setting them for you at -22, right? If
> you're getting a few scant overs without the limiter, you're
> still ok, really... the idea is not to overstuff the mix buss
> so heavily that if you pull the limiter off you're going into
> the +5, +6 range without it.
>
> Think "clean" people = think "no clips" (or as few as
> possible), you get 30-40 channels of "overs" constantly (like
> the 352,000 of 'em per second in the example I gave earlier),
> and it's going to get harsh & thin.... it's a cumulative effect.
>
> That's it, really... it's just like any other tool - you can't
> use an allen wrench to properly drive a nail, and you can't use
> a hammer to trim your nose hair.
>
> Happy Native mixing!
>
> (think "zen"!)
>
> Neil
|
|
|
|
| Re: The nut we have to crack [message #74993 is a reply to message #74728] |
Fri, 27 October 2006 21:32 |
Martin Harrington
Messages: 560
Registered: September 2005
|
Senior Member |
|
|
Hmmm
So you're happy to truncate 24 bit files to 20 bit and then convert back to
24 bit for the DVD master....how does it sound?
--
Martin Harrington
www.lendanear-sound.com
"espresso" <audio@espressodigital.com> wrote in message
news:453dc969$1@linux...
>I posted this a while ago.. I now use Nuendo with a Layla 3G to output 8 x
> analog channels and 8 x ADAT stems to Paris ie. Paris is now my mixer, its
> in the same computer, its easy....hardly any overhead as all Paris is
> doing
> is sitting in 'live' mode. I've been doing a bunch of live concert DVDs
> with
> 50 odd channels - 2 hour files - no chance I'd be wanting to convert all
> those puppies to .pafs... or even the stems for that matter. The proof is
> in
> the sound - the files played through Paris are alive and have depth. Same
> mix in Nuendo...urrrgghh. I know that I'm getting a double belt of DA-AD
> plus losing 4 bits of info through the Paris ADAT, but honestly the end
> justifies the means. All i'm trying to add to the discussion is - if you
> want the functionality of the native program plus the Paris sound its
> readily achievable without having to jump through the '2nd computer as FX
> buss' hoops.
>
> Cheers,
>
> David.
>
>
> "Kim" <hiddensounds@hotmail.com> wrote in message news:453d9ba4$1@linux...
>>
>>
>> Chuck,
>>
>> There was talk some time ago (oh how the years wander on...) of somebody
>> making an EDS chip emulator, which would then allow various
>> possibilities,
>> which one would assume would include:
>>
>> 1) a "Virtual" EDS card driver which emulates all the functionality of an
>> EDS card down to the last bit, and hence plugs right into Paris allowing
>> more submixes, natively, but with the same sound characteristics as the
> EDS
>> subs, or...
>> 2) using the same technology, a virtual Paris mix bus, which uses the
> emaulation
>> of the EDS alongside the code from the Paris OS to basically allow a
>> Paris
>> mix bus, using something like rewire, to plug in to a native app.
>>
>> I believe the talk was inspired by Matthew Craig's efforts in creating
>> the
>> VST Paris EQ, which does basically this same thing, emulating the EDS
> functionality
>> and hence generating pretty much identical output to the same audio going
>> through the card itself.
>>
>> This would sure sort out the issues if anybody with enough knowhow and
> dedication
>> got on board. Suddenly any app could have the Paris mix bus, not to
> mention
>> the paris EQ... that would pretty much put an end to all this
>> shennigans
>> i would think.
>>
>> Cheers,
>> Kim.
>>
>> "chuck duffy" <c@c.com> wrote:
>> >
>> >DJ,
>> >
>> >Listen I know you love messing with this stuff, but I think we need to
> focus
>> >on how to get the mixes we want out of an all native system.
>> >
>> >It just doesn't make any sense to me to get onboard with another weird,
>> proprietary
>> >dsp system. Creamware is as weird, oddball nad proprietary as it gets.
>>
>> >Why bother with it? Why bother with UAD or anything else. It just
> doesn't
>> >make sense to me.
>> >
>> >If we can't get decent mixes out of a native daw then something is
>> >wrong.
>> > Let's find the thing that's wrong, and make it right.
>> >
>> >Chuck
>>
>
>
|
|
|
|
| Re: The nut we have to crack [message #74999 is a reply to message #74993] |
Sat, 28 October 2006 02:58 |
Ted Gerber
Messages: 705
Registered: January 2009
|
Senior Member |
|
|
Hi Martin-
I specifically remember others in the past using the Paris Adat In from
24 bit sources, being thrilled with the sound, happliy knowing truncation
was happening. As David says below:
"honestly, the end justifies the means"
FWIW
Ted
"Martin Harrington" <lendan@bigpond.net.au> wrote:
>Hmmm
>So you're happy to truncate 24 bit files to 20 bit and then convert back
to
>24 bit for the DVD master....how does it sound?
>--
>Martin Harrington
>www.lendanear-sound.com
>
>"espresso" <audio@espressodigital.com> wrote in message
>news:453dc969$1@linux...
>>I posted this a while ago.. I now use Nuendo with a Layla 3G to output
8 x
>> analog channels and 8 x ADAT stems to Paris ie. Paris is now my mixer,
its
>> in the same computer, its easy....hardly any overhead as all Paris is
>> doing
>> is sitting in 'live' mode. I've been doing a bunch of live concert DVDs
>> with
>> 50 odd channels - 2 hour files - no chance I'd be wanting to convert all
>> those puppies to .pafs... or even the stems for that matter. The proof
is
>> in
>> the sound - the files played through Paris are alive and have depth. Same
>> mix in Nuendo...urrrgghh. I know that I'm getting a double belt of DA-AD
>> plus losing 4 bits of info through the Paris ADAT, but honestly the end
>> justifies the means. All i'm trying to add to the discussion is - if you
>> want the functionality of the native program plus the Paris sound its
>> readily achievable without having to jump through the '2nd computer as
FX
>> buss' hoops.
>>
>> Cheers,
>>
>> David.
>>
>>
>> "Kim" <hiddensounds@hotmail.com> wrote in message news:453d9ba4$1@linux...
>>>
>>>
>>> Chuck,
>>>
>>> There was talk some time ago (oh how the years wander on...) of somebody
>>> making an EDS chip emulator, which would then allow various
>>> possibilities,
>>> which one would assume would include:
>>>
>>> 1) a "Virtual" EDS card driver which emulates all the functionality of
an
>>> EDS card down to the last bit, and hence plugs right into Paris allowing
>>> more submixes, natively, but with the same sound characteristics as the
>> EDS
>>> subs, or...
>>> 2) using the same technology, a virtual Paris mix bus, which uses the
>> emaulation
>>> of the EDS alongside the code from the Paris OS to basically allow a
>>> Paris
>>> mix bus, using something like rewire, to plug in to a native app.
>>>
>>> I believe the talk was inspired by Matthew Craig's efforts in creating
>>> the
>>> VST Paris EQ, which does basically this same thing, emulating the EDS
>> functionality
>>> and hence generating pretty much identical output to the same audio going
>>> through the card itself.
>>>
>>> This would sure sort out the issues if anybody with enough knowhow and
>> dedication
>>> got on board. Suddenly any app could have the Paris mix bus, not to
>> mention
>>> the paris EQ... that would pretty much put an end to all this
>>> shennigans
>>> i would think.
>>>
>>> Cheers,
>>> Kim.
>>>
>>> "chuck duffy" <c@c.com> wrote:
>>> >
>>> >DJ,
>>> >
>>> >Listen I know you love messing with this stuff, but I think we need
to
>> focus
>>> >on how to get the mixes we want out of an all native system.
>>> >
>>> >It just doesn't make any sense to me to get onboard with another weird,
>>> proprietary
>>> >dsp system. Creamware is as weird, oddball nad proprietary as it gets.
>>>
>>> >Why bother with it? Why bother with UAD or anything else. It just
>> doesn't
>>> >make sense to me.
>>> >
>>> >If we can't get decent mixes out of a native daw then something is
>>> >wrong.
>>> > Let's find the thing that's wrong, and make it right.
>>> >
>>> >Chuck
>>>
>>
>>
>
>
|
|
|
|
| Re: The nut we have to crack [message #75254 is a reply to message #74993] |
Wed, 01 November 2006 10:13 |
animix
Messages: 356
Registered: September 2006
|
Senior Member |
|
|
I do it oftten (as in almost always) for Cd's. Dithering at the final stage
seems to cover any quantization noise created by the truncation....but this
is for CD's. I'm not doing DVD's here, but I would definitely apply some
sort of dither or noise shaping (as in using Waves IDR) to the audio, even
if not going to 16 bit.
Deej
"Martin Harrington" <lendan@bigpond.net.au> wrote in message
news:4542db33$1@linux...
> Hmmm
> So you're happy to truncate 24 bit files to 20 bit and then convert back
to
> 24 bit for the DVD master....how does it sound?
> --
> Martin Harrington
> www.lendanear-sound.com
>
> "espresso" <audio@espressodigital.com> wrote in message
> news:453dc969$1@linux...
> >I posted this a while ago.. I now use Nuendo with a Layla 3G to output 8
x
> > analog channels and 8 x ADAT stems to Paris ie. Paris is now my mixer,
its
> > in the same computer, its easy....hardly any overhead as all Paris is
> > doing
> > is sitting in 'live' mode. I've been doing a bunch of live concert DVDs
> > with
> > 50 odd channels - 2 hour files - no chance I'd be wanting to convert all
> > those puppies to .pafs... or even the stems for that matter. The proof
is
> > in
> > the sound - the files played through Paris are alive and have depth.
Same
> > mix in Nuendo...urrrgghh. I know that I'm getting a double belt of DA-AD
> > plus losing 4 bits of info through the Paris ADAT, but honestly the end
> > justifies the means. All i'm trying to add to the discussion is - if you
> > want the functionality of the native program plus the Paris sound its
> > readily achievable without having to jump through the '2nd computer as
FX
> > buss' hoops.
> >
> > Cheers,
> >
> > David.
> >
> >
> > "Kim" <hiddensounds@hotmail.com> wrote in message
news:453d9ba4$1@linux...
> >>
> >>
> >> Chuck,
> >>
> >> There was talk some time ago (oh how the years wander on...) of
somebody
> >> making an EDS chip emulator, which would then allow various
> >> possibilities,
> >> which one would assume would include:
> >>
> >> 1) a "Virtual" EDS card driver which emulates all the functionality of
an
> >> EDS card down to the last bit, and hence plugs right into Paris
allowing
> >> more submixes, natively, but with the same sound characteristics as the
> > EDS
> >> subs, or...
> >> 2) using the same technology, a virtual Paris mix bus, which uses the
> > emaulation
> >> of the EDS alongside the code from the Paris OS to basically allow a
> >> Paris
> >> mix bus, using something like rewire, to plug in to a native app.
> >>
> >> I believe the talk was inspired by Matthew Craig's efforts in creating
> >> the
> >> VST Paris EQ, which does basically this same thing, emulating the EDS
> > functionality
> >> and hence generating pretty much identical output to the same audio
going
> >> through the card itself.
> >>
> >> This would sure sort out the issues if anybody with enough knowhow and
> > dedication
> >> got on board. Suddenly any app could have the Paris mix bus, not to
> > mention
> >> the paris EQ... that would pretty much put an end to all this
> >> shennigans
> >> i would think.
> >>
> >> Cheers,
> >> Kim.
> >>
> >> "chuck duffy" <c@c.com> wrote:
> >> >
> >> >DJ,
> >> >
> >> >Listen I know you love messing with this stuff, but I think we need to
> > focus
> >> >on how to get the mixes we want out of an all native system.
> >> >
> >> >It just doesn't make any sense to me to get onboard with another
weird,
> >> proprietary
> >> >dsp system. Creamware is as weird, oddball nad proprietary as it
gets.
> >>
> >> >Why bother with it? Why bother with UAD or anything else. It just
> > doesn't
> >> >make sense to me.
> >> >
> >> >If we can't get decent mixes out of a native daw then something is
> >> >wrong.
> >> > Let's find the thing that's wrong, and make it right.
> >> >
> >> >Chuck
> >>
> >
> >
>
>
|
|
|
|
| Re: The nut we have to crack [message #75268 is a reply to message #75254] |
Wed, 01 November 2006 20:31 |
Martin Harrington
Messages: 560
Registered: September 2005
|
Senior Member |
|
|
What you do is sound, (thinking), but to go back to full spec DVD,that
doesn't make much sense to me.
--
Martin Harrington
www.lendanear-sound.com
"DJ" <notachance@net.net> wrote in message news:4548e372@linux...
>I do it oftten (as in almost always) for Cd's. Dithering at the final stage
> seems to cover any quantization noise created by the truncation....but
> this
> is for CD's. I'm not doing DVD's here, but I would definitely apply some
> sort of dither or noise shaping (as in using Waves IDR) to the audio, even
> if not going to 16 bit.
>
> Deej
>
> "Martin Harrington" <lendan@bigpond.net.au> wrote in message
> news:4542db33$1@linux...
>> Hmmm
>> So you're happy to truncate 24 bit files to 20 bit and then convert back
> to
>> 24 bit for the DVD master....how does it sound?
>> --
>> Martin Harrington
>> www.lendanear-sound.com
>>
>> "espresso" <audio@espressodigital.com> wrote in message
>> news:453dc969$1@linux...
>> >I posted this a while ago.. I now use Nuendo with a Layla 3G to output
>> >8
> x
>> > analog channels and 8 x ADAT stems to Paris ie. Paris is now my mixer,
> its
>> > in the same computer, its easy....hardly any overhead as all Paris is
>> > doing
>> > is sitting in 'live' mode. I've been doing a bunch of live concert DVDs
>> > with
>> > 50 odd channels - 2 hour files - no chance I'd be wanting to convert
>> > all
>> > those puppies to .pafs... or even the stems for that matter. The proof
> is
>> > in
>> > the sound - the files played through Paris are alive and have depth.
> Same
>> > mix in Nuendo...urrrgghh. I know that I'm getting a double belt of
>> > DA-AD
>> > plus losing 4 bits of info through the Paris ADAT, but honestly the end
>> > justifies the means. All i'm trying to add to the discussion is - if
>> > you
>> > want the functionality of the native program plus the Paris sound its
>> > readily achievable without having to jump through the '2nd computer as
> FX
>> > buss' hoops.
>> >
>> > Cheers,
>> >
>> > David.
>> >
>> >
>> > "Kim" <hiddensounds@hotmail.com> wrote in message
> news:453d9ba4$1@linux...
>> >>
>> >>
>> >> Chuck,
>> >>
>> >> There was talk some time ago (oh how the years wander on...) of
> somebody
>> >> making an EDS chip emulator, which would then allow various
>> >> possibilities,
>> >> which one would assume would include:
>> >>
>> >> 1) a "Virtual" EDS card driver which emulates all the functionality of
> an
>> >> EDS card down to the last bit, and hence plugs right into Paris
> allowing
>> >> more submixes, natively, but with the same sound characteristics as
>> >> the
>> > EDS
>> >> subs, or...
>> >> 2) using the same technology, a virtual Paris mix bus, which uses the
>> > emaulation
>> >> of the EDS alongside the code from the Paris OS to basically allow a
>> >> Paris
>> >> mix bus, using something like rewire, to plug in to a native app.
>> >>
>> >> I believe the talk was inspired by Matthew Craig's efforts in creating
>> >> the
>> >> VST Paris EQ, which does basically this same thing, emulating the EDS
>> > functionality
>> >> and hence generating pretty much identical output to the same audio
> going
>> >> through the card itself.
>> >>
>> >> This would sure sort out the issues if anybody with enough knowhow and
>> > dedication
>> >> got on board. Suddenly any app could have the Paris mix bus, not to
>> > mention
>> >> the paris EQ... that would pretty much put an end to all this
>> >> shennigans
>> >> i would think.
>> >>
>> >> Cheers,
>> >> Kim.
>> >>
>> >> "chuck duffy" <c@c.com> wrote:
>> >> >
>> >> >DJ,
>> >> >
>> >> >Listen I know you love messing with this stuff, but I think we need
>> >> >to
>> > focus
>> >> >on how to get the mixes we want out of an all native system.
>> >> >
>> >> >It just doesn't make any sense to me to get onboard with another
> weird,
>> >> proprietary
>> >> >dsp system. Creamware is as weird, oddball nad proprietary as it
> gets.
>> >>
>> >> >Why bother with it? Why bother with UAD or anything else. It just
>> > doesn't
>> >> >make sense to me.
>> >> >
>> >> >If we can't get decent mixes out of a native daw then something is
>> >> >wrong.
>> >> > Let's find the thing that's wrong, and make it right.
>> >> >
>> >> >Chuck
>> >>
>> >
>> >
>>
>>
>
>
|
|
|
|
| Re: The nut we have to crack [message #75296 is a reply to message #75268] |
Thu, 02 November 2006 14:47 |
animix
Messages: 356
Registered: September 2006
|
Senior Member |
|
|
Yeah.....if full spec DVD (as in 96k/24bit) was my destination format, I'd
not be using Paris for mixing anyway.
;o)
"Martin Harrington" <lendan@bigpond.net.au> wrote in message
news:45497267$1@linux...
> What you do is sound, (thinking), but to go back to full spec DVD,that
> doesn't make much sense to me.
> --
> Martin Harrington
> www.lendanear-sound.com
>
> "DJ" <notachance@net.net> wrote in message news:4548e372@linux...
> >I do it oftten (as in almost always) for Cd's. Dithering at the final
stage
> > seems to cover any quantization noise created by the truncation....but
> > this
> > is for CD's. I'm not doing DVD's here, but I would definitely apply some
> > sort of dither or noise shaping (as in using Waves IDR) to the audio,
even
> > if not going to 16 bit.
> >
> > Deej
> >
> > "Martin Harrington" <lendan@bigpond.net.au> wrote in message
> > news:4542db33$1@linux...
> >> Hmmm
> >> So you're happy to truncate 24 bit files to 20 bit and then convert
back
> > to
> >> 24 bit for the DVD master....how does it sound?
> >> --
> >> Martin Harrington
> >> www.lendanear-sound.com
> >>
> >> "espresso" <audio@espressodigital.com> wrote in message
> >> news:453dc969$1@linux...
> >> >I posted this a while ago.. I now use Nuendo with a Layla 3G to
output
> >> >8
> > x
> >> > analog channels and 8 x ADAT stems to Paris ie. Paris is now my
mixer,
> > its
> >> > in the same computer, its easy....hardly any overhead as all Paris is
> >> > doing
> >> > is sitting in 'live' mode. I've been doing a bunch of live concert
DVDs
> >> > with
> >> > 50 odd channels - 2 hour files - no chance I'd be wanting to convert
> >> > all
> >> > those puppies to .pafs... or even the stems for that matter. The
proof
> > is
> >> > in
> >> > the sound - the files played through Paris are alive and have depth.
> > Same
> >> > mix in Nuendo...urrrgghh. I know that I'm getting a double belt of
> >> > DA-AD
> >> > plus losing 4 bits of info through the Paris ADAT, but honestly the
end
> >> > justifies the means. All i'm trying to add to the discussion is - if
> >> > you
> >> > want the functionality of the native program plus the Paris sound its
> >> > readily achievable without having to jump through the '2nd computer
as
> > FX
> >> > buss' hoops.
> >> >
> >> > Cheers,
> >> >
> >> > David.
> >> >
> >> >
> >> > "Kim" <hiddensounds@hotmail.com> wrote in message
> > news:453d9ba4$1@linux...
> >> >>
> >> >>
> >> >> Chuck,
> >> >>
> >> >> There was talk some time ago (oh how the years wander on...) of
> > somebody
> >> >> making an EDS chip emulator, which would then allow various
> >> >> possibilities,
> >> >> which one would assume would include:
> >> >>
> >> >> 1) a "Virtual" EDS card driver which emulates all the functionality
of
> > an
> >> >> EDS card down to the last bit, and hence plugs right into Paris
> > allowing
> >> >> more submixes, natively, but with the same sound characteristics as
> >> >> the
> >> > EDS
> >> >> subs, or...
> >> >> 2) using the same technology, a virtual Paris mix bus, which uses
the
> >> > emaulation
> >> >> of the EDS alongside the code from the Paris OS to basically allow a
> >> >> Paris
> >> >> mix bus, using something like rewire, to plug in to a native app.
> >> >>
> >> >> I believe the talk was inspired by Matthew Craig's efforts in
creating
> >> >> the
> >> >> VST Paris EQ, which does basically this same thing, emulating the
EDS
> >> > functionality
> >> >> and hence generating pretty much identical output to the same audio
> > going
> >> >> through the card itself.
> >> >>
> >> >> This would sure sort out the issues if anybody with enough knowhow
and
> >> > dedication
> >> >> got on board. Suddenly any app could have the Paris mix bus, not to
> >> > mention
> >> >> the paris EQ... that would pretty much put an end to all this
> >> >> shennigans
> >> >> i would think.
> >> >>
> >> >> Cheers,
> >> >> Kim.
> >> >>
> >> >> "chuck duffy" <c@c.com> wrote:
> >> >> >
> >> >> >DJ,
> >> >> >
> >> >> >Listen I know you love messing with this stuff, but I think we need
> >> >> >to
> >> > focus
> >> >> >on how to get the mixes we want out of an all native system.
> >> >> >
> >> >> >It just doesn't make any sense to me to get onboard with another
> > weird,
> >> >> proprietary
> >> >> >dsp system. Creamware is as weird, oddball nad proprietary as it
> > gets.
> >> >>
> >> >> >Why bother with it? Why bother with UAD or anything else. It just
> >> > doesn't
> >> >> >make sense to me.
> >> >> >
> >> >> >If we can't get decent mixes out of a native daw then something is
> >> >> >wrong.
> >> >> > Let's find the thing that's wrong, and make it right.
> >> >> >
> >> >> >Chuck
> >> >>
> >> >
> >> >
> >>
> >>
> >
> >
>
>
|
|
|
|
| Re: The nut we have to crack [message #75298 is a reply to message #75296] |
Thu, 02 November 2006 15:33 |
Martin Harrington
Messages: 560
Registered: September 2005
|
Senior Member |
|
|
Good Point
--
Martin Harrington
www.lendanear-sound.com
"DJ" <notachance@net.net> wrote in message news:454a751e@linux...
> Yeah.....if full spec DVD (as in 96k/24bit) was my destination format, I'd
> not be using Paris for mixing anyway.
>
> ;o)
>
> "Martin Harrington" <lendan@bigpond.net.au> wrote in message
> news:45497267$1@linux...
>> What you do is sound, (thinking), but to go back to full spec DVD,that
>> doesn't make much sense to me.
>> --
>> Martin Harrington
>> www.lendanear-sound.com
>>
>> "DJ" <notachance@net.net> wrote in message news:4548e372@linux...
>> >I do it oftten (as in almost always) for Cd's. Dithering at the final
> stage
>> > seems to cover any quantization noise created by the truncation....but
>> > this
>> > is for CD's. I'm not doing DVD's here, but I would definitely apply
>> > some
>> > sort of dither or noise shaping (as in using Waves IDR) to the audio,
> even
>> > if not going to 16 bit.
>> >
>> > Deej
>> >
>> > "Martin Harrington" <lendan@bigpond.net.au> wrote in message
>> > news:4542db33$1@linux...
>> >> Hmmm
>> >> So you're happy to truncate 24 bit files to 20 bit and then convert
> back
>> > to
>> >> 24 bit for the DVD master....how does it sound?
>> >> --
>> >> Martin Harrington
>> >> www.lendanear-sound.com
>> >>
>> >> "espresso" <audio@espressodigital.com> wrote in message
>> >> news:453dc969$1@linux...
>> >> >I posted this a while ago.. I now use Nuendo with a Layla 3G to
> output
>> >> >8
>> > x
>> >> > analog channels and 8 x ADAT stems to Paris ie. Paris is now my
> mixer,
>> > its
>> >> > in the same computer, its easy....hardly any overhead as all Paris
>> >> > is
>> >> > doing
>> >> > is sitting in 'live' mode. I've been doing a bunch of live concert
> DVDs
>> >> > with
>> >> > 50 odd channels - 2 hour files - no chance I'd be wanting to convert
>> >> > all
>> >> > those puppies to .pafs... or even the stems for that matter. The
> proof
>> > is
>> >> > in
>> >> > the sound - the files played through Paris are alive and have depth.
>> > Same
>> >> > mix in Nuendo...urrrgghh. I know that I'm getting a double belt of
>> >> > DA-AD
>> >> > plus losing 4 bits of info through the Paris ADAT, but honestly the
> end
>> >> > justifies the means. All i'm trying to add to the discussion is - if
>> >> > you
>> >> > want the functionality of the native program plus the Paris sound
>> >> > its
>> >> > readily achievable without having to jump through the '2nd computer
> as
>> > FX
>> >> > buss' hoops.
>> >> >
>> >> > Cheers,
>> >> >
>> >> > David.
>> >> >
>> >> >
>> >> > "Kim" <hiddensounds@hotmail.com> wrote in message
>> > news:453d9ba4$1@linux...
>> >> >>
>> >> >>
>> >> >> Chuck,
>> >> >>
>> >> >> There was talk some time ago (oh how the years wander on...) of
>> > somebody
>> >> >> making an EDS chip emulator, which would then allow various
>> >> >> possibilities,
>> >> >> which one would assume would include:
>> >> >>
>> >> >> 1) a "Virtual" EDS card driver which emulates all the functionality
> of
>> > an
>> >> >> EDS card down to the last bit, and hence plugs right into Paris
>> > allowing
>> >> >> more submixes, natively, but with the same sound characteristics as
>> >> >> the
>> >> > EDS
>> >> >> subs, or...
>> >> >> 2) using the same technology, a virtual Paris mix bus, which uses
> the
>> >> > emaulation
>> >> >> of the EDS alongside the code from the Paris OS to basically allow
>> >> >> a
>> >> >> Paris
>> >> >> mix bus, using something like rewire, to plug in to a native app.
>> >> >>
>> >> >> I believe the talk was inspired by Matthew Craig's efforts in
> creating
>> >> >> the
>> >> >> VST Paris EQ, which does basically this same thing, emulating the
> EDS
>> >> > functionality
>> >> >> and hence generating pretty much identical output to the same audio
>> > going
>> >> >> through the card itself.
>> >> >>
>> >> >> This would sure sort out the issues if anybody with enough knowhow
> and
>> >> > dedication
>> >> >> got on board. Suddenly any app could have the Paris mix bus, not to
>> >> > mention
>> >> >> the paris EQ... that would pretty much put an end to all this
>> >> >> shennigans
>> >> >> i would think.
>> >> >>
>> >> >> Cheers,
>> >> >> Kim.
>> >> >>
>> >> >> "chuck duffy" <c@c.com> wrote:
>> >> >> >
>> >> >> >DJ,
>> >> >> >
>> >> >> >Listen I know you love messing with this stuff, but I think we
>> >> >> >need
>> >> >> >to
>> >> > focus
>> >> >> >on how to get the mixes we want out of an all native system.
>> >> >> >
>> >> >> >It just doesn't make any sense to me to get onboard with another
>> > weird,
>> >> >> proprietary
>> >> >> >dsp system. Creamware is as weird, oddball nad proprietary as it
>> > gets.
>> >> >>
>> >> >> >Why bother with it? Why bother with UAD or anything else. It
>> >> >> >just
>> >> > doesn't
>> >> >> >make sense to me.
>> >> >> >
>> >> >> >If we can't get decent mixes out of a native daw then something is
>> >> >> >wrong.
>> >> >> > Let's find the thing that's wrong, and make it right.
>> >> >> >
>> >> >> >Chuck
>> >> >>
>> >> >
>> >> >
>> >>
>> >>
>> >
>> >
>>
>>
>
>
|
|
|
|
Goto Forum:
Current Time: Mon Nov 18 21:48:18 PST 2024
Total time taken to generate the page: 0.23094 seconds
|
The PARIS Forums
Search
Help
Members
Register
Login
Home
