Home » The PARIS Forums » PARIS: Main » Paris Skins
| Paris Skins [message #76033] |
Wed, 15 November 2006 21:36  |
highmtn
Messages: 13
Registered: November 2006
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Junior Member |
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ownscale" there. Digital Mixer? Dunno if that'd do it -
> I'd have to hear it, try it, etc. Passive summing mixer?
> I dunno how much of that is really snake-oil: I'd have to buy-
> it-try-it, a/b mixes, etc, etc, sell it if I hate it, try
> something else, shitcan it if I hate it.... and I'm kinda tired
> of the whole cycle that I've gotten into lately of tweak,
> remix, find a workaround
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| Re: Paris Skins [message #76034 is a reply to message #76033] |
Wed, 15 November 2006 21:51  |
Kim
Messages: 1246
Registered: October 2005
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Senior Member |
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, tweak, remix, try something else,
> tweak/remix, ad nauseum.
>
> I'm NOT being productive, is the point.
>
> So, I have no fucking clue as to where to go next with this, or
> if there's some affordable solution out there somewhere that'll
> get me what I want, which is simply a streamlined, killer-
> sounding, one-box solution. I'm tending to think "not".
>
> NeilThis is a multi-part message in MIME format.
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Strange....!!=20
Does the driver show up with a "!" or a "?" in the device manager ? What =
OS you running?
Rob
"Mike R." <emarenot@yahoo.com> wrote in message =
ne
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| Re: Paris Skins / examples [message #76035 is a reply to message #76034] |
Wed, 15 November 2006 22:15 |
Yanoska
Messages: 32
Registered: January 2007
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Member |
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ws:45840d3c$1@linux...
Any ideas on why I have to reload my Paris drivers everytime I boot =
the program? I'm running the EDS card on an Asus A7S333, PCI slot 3. =
Vid card is slot one, slot two is blank to make room for the EDS =
daughter card. I did try to reinstall Paris (2.0), but that did not =
seem to fix the problem. =20
Thanks gents.
MR
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<DIV><FONT size=3D2>Strange....!! </FONT></DIV>
<DIV><FONT size=3D2>Does the driver show up with a "!" or a "?" in =
the device=20
manager ? What OS you running?</FONT></DIV>
<DIV><FONT size=3D2></FONT> </DIV>
<DIV><FONT size=3D2>Rob</FONT></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Mike R." <<A=20
href=3D"mailto:emarenot@yahoo.com">emarenot@yahoo.com</A>> wrote in =
message=20
<A href=3D"news:45840d3c$1@linux">news:45840d3c$1@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>Any ideas on why I have to reload my =
Paris=20
drivers everytime I boot the program? I'm running the EDS card =
on an=20
Asus A7S333, PCI slot 3. Vid card is slot one, slot two is blank =
to make=20
room for the EDS daughter card. I did try to reinstall Paris =
(2.0), but=20
that did not seem to fix the problem. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Thanks gents.</FONT></DIV>
<DIV><FONT face=3DArial =
size=3D2>MR</FONT></DIV></BLOCKQUOTE></BODY></HTML>
------=_NextPart_000_001A_01C72138.9B863EA0--Worst case scenario - strategic surgical editing - to just get the really
noticable squeeks attenuated or removed...wouldn't even dream of a full song
edit as squeaks are part of the playing and I personally think they add
character to the track
DOn
"alex plasko" <alex.plasko@snet.net> wrote in message
news:45842fa9$1@linux...
> neil,I dont know man .If were not putting fret noise in, to make a sampler
> sound more realistic, were trying to remove it from the real deal for a
> cleaner sound. does it ever end?
> "Neil" <IUOIU@OIU.com> wrote in message news:458374eb$1@linux...
>>
>> I don't know - maybe that'd work... problem with trying to
>> grab "a" sample of the noise is that that kind of noise isn't
>> consistent at all - it varies in pitch & harmonic content quite
>> a bit from one "squeek" to the next.
>>
>> Neil
>>
>>
>> "alex plasko" <alex.plasko@snet.net> wrote:
>>>neil, how about something like denoiser? if you can get a clean sample of
>>
>>>the fret noise denoiser will wipe it and interpolate signal befor and
>>>after
>>
>>>that spot.sound forge denoiser works pretty good for things of that
>>>nature
>>
>>>. never tried it for fret noise though.
>>>
>>>"Neil" <OIUOIU@OIU.com> wrote in message news:45835bb9$1@linux...
>>>>
>>>> Manully editing might be the best bet... Don, if you're trying
>>>> a de-esser for that, you might need like 3 or 4 of 'em in a
>>>> row, each centered on different frequencies. Fortunately,
>>>> Spitfish doesn't take up much CPU power (if that's what you're
>>>> using).
>>>>
>>>> Also, you could try re-micing it through your monitors (muting
>>>> the playback on the recording channel) with a dynamic mic
>>>> backed-off a bit from the speakers... whatever you do, don't
>>>> compress when playing back to re-mic if you try this. That'll
>>>> just make the squeak stand out more.
>>>>
>>>> Wanna send me a sample of the wav file & I'll see if I can come
>>>> up with an idea for you?
>>>>
>>>> Neil
>>>>
>>>>
>>>> "Sarah" <sarahjane@sarahtonin.com> wrote:
>>>>> Are you talking about strings buzzing against frets, or fingers
>>>>>chirping and squeaking on the string windings? Cuz I've solved the
>>>>>latter
>>>>
>>>>>to my satisfaction by switching to Elixirs. To fix the noise that's
>>>>>already
>>>>
>>>>>there, I'd be inclined to just go in there with a wave editor and
>>>>>manually
>>>>
>>>>>fix the noises one at a time, but then, I'm obsessive. :)
>>>>>
>>>>>Sarah
>>>>>
>>>>>www.sarahtonin.com
>>>>>
>>>>>"Don Nafe" <dnafe@magma.ca> wrote in message news:45834c91@linux...
>>>>>> Hi all
>>>>>>
>>>>>> Actually it isn't bad at all but several inital comments on this
>>>>>> particular track I'm mixing is there's too much fret noise on the
>>>>>> acoustic.
>>>>>>
>>>>>> Most of it is between chords (picking)
>>>>>>
>>>>>> One thought was using a de-esser...all other's are more than welcome
>>>>>>
>>>>>> Don
>>>>>>
>>>>>
>>>>>
>>>>
>>>
>>>
>>
>
>This is a multi-part message in MIME format.
------=_NextPart_000_000A_01C72136.265C9220
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charset="iso-8859-1"
Content-Transfer-Encoding: 7bit
Hey Rob,
Thanks for responding. I should have mentioned: Win98SE... and no, no "!"
or "?" shows up in the device manager. In fact, the driver is listed. But,
I have to "remove" the bugger before shutting down, so that, when I reboot
the computer it see's the driver and loads it. If I don't do this, Paris
will not load. I get an error message "error 61d/something." The EDS card
is on IRQ3 and shares it only with the APCI thingy. Once I go through this
reloading-the-driver-procedure, all is well, Paris runs and is happy.
MR
"Rob Arsenault" <mani1147 AT hotmail.com> wrote in message
news:4584659e@linux...
Strange....!!
Does the driver show up with a "!" or a "?" in the device manager ? What
OS you running?
Rob
"Mike R." <emarenot@yahoo.com> wrote in message news:45840d3c$1@linux...
Any ideas on why I have to reload my Paris drivers everytime I boot the
program? I'm running the EDS card on an Asus A7S333, PCI slot 3. Vid card
is slot one, slot two is blank to make room for the EDS daughter card. I
did try to reinstall Paris (2.0), but that did not seem to fix the problem.
Thanks gents.
MR
------=_NextPart_000_000A_01C72136.265C9220
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charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
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<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1555" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hey Rob,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Thanks for responding. I should =
have=20
mentioned: Win98SE... and no, no "!" or "?" shows up in the device=20
manager. In fact, the driver is listed. But, I have to =
"remove" the=20
bugger before shutting down, so that, when I reboot the computer it =
see's the=20
driver and loads it. If I don't do this, Paris will not =
load. I get=20
an error message "error 61d/something." The EDS card is on IRQ3 =
and shares=20
it only with the APCI thingy. Once I go through this=20
reloading-the-driver-procedure, all is well, Paris runs and is=20
happy.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>MR</FONT></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Rob Arsenault" <mani1147 AT hotmail.com> wrote in message =
<A=20
href=3D"news:4584659e@linux">news:4584659e@linux</A>...</DIV>
<DIV><FONT size=3D2>Strange....!! </FONT></DIV>
<DIV><FONT size=3D2>Does the driver show up with a "!" or a "?" =
in the=20
device manager ? What OS you running?</FONT></DIV>
<DIV><FONT size=3D2></FONT> </DIV>
<DIV><FONT size=3D2>Rob</FONT></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Mike R." <<A=20
href=3D"mailto:emarenot@yahoo.com">emarenot@yahoo.com</A>> wrote =
in message=20
<A href=3D"news:45840d3c$1@linux">news:45840d3c$1@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>Any ideas on why I have to reload =
my Paris=20
drivers everytime I boot the program? I'm running the EDS card =
on an=20
Asus A7S333, PCI slot 3. Vid card is slot one, slot two is =
blank to=20
make room for the EDS daughter card. I did try to reinstall =
Paris=20
(2.0), but that did not seem to fix the problem. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Thanks gents.</FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2>MR</FONT></DIV></BLOCKQUOTE></BLOCKQUOTE></BODY ></HTML>
------=_NextPart_000_000A_01C72136.265C9220--how about autotune on those offending sqweaks.hehehe
"Don Nafe" <dnafe@magma.ca> wrote in message news:45846a94@linux...
> Worst case scenario - strategic surgical editing - to just get the really
> noticable squeeks attenuated or removed...wouldn't even dream of a full
> song edit as squeaks are part of the playing and I personally think they
> add character to the track
>
> DOn
>
>
> "alex plasko" <alex.plasko@snet.net> wrote in message
> news:45842fa9$1@linux...
>> neil,I dont know man .If were not putting fret noise in, to make a
>> sampler sound more realistic, were trying to remove it from the real deal
>> for a cleaner sound. does it ever end?
>> "Neil" <IUOIU@OIU.com> wrote in message news:458374eb$1@linux...
>>>
>>> I don't know - maybe that'd work... problem with trying to
>>> grab "a" sample of the noise is that that kind of noise isn't
>>> consistent at all - it varies in pitch & harmonic content quite
>>> a bit from one "squeek" to the next.
>>>
>>> Neil
>>>
>>>
>>> "alex plasko" <alex.plasko@snet.net> wrote:
>>>>neil, how about something like denoiser? if you can get a clean sample
>>>>of
>>>
>>>>the fret noise denoiser will wipe it and interpolate signal befor and
>>>>after
>>>
>>>>that spot.sound forge denoiser works pretty good for things of that
>>>>nature
>>>
>>>>. never tried it for fret noise though.
>>>>
>>>>"Neil" <OIUOIU@OIU.com> wrote in message news:45835bb9$1@linux...
>>>>>
>>>>> Manully editing might be the best bet... Don, if you're trying
>>>>> a de-esser for that, you might need like 3 or 4 of 'em in a
>>>>> row, each centered on different frequencies. Fortunately,
>>>>> Spitfish doesn't take up much CPU power (if that's what you're
>>>>> using).
>>>>>
>>>>> Also, you could try re-micing it through your monitors (muting
>>>>> the playback on the recording channel) with a dynamic mic
>>>>> backed-off a bit from the speakers... whatever you do, don't
>>>>> compress when playing back to re-mic if you try this. That'll
>>>>> just make the squeak stand out more.
>>>>>
>>>>> Wanna send me a sample of the wav file & I'll see if I can come
>>>>> up with an idea for you?
>>>>>
>>>>> Neil
>>>>>
>>>>>
>>>>> "Sarah" <sarahjane@sarahtonin.com> wrote:
>>>>>> Are you talking about strings buzzing against frets, or fingers
>>>>>>chirping and squeaking on the string windings? Cuz I've solved the
>>>>>>latter
>>>>>
>>>>>>to my satisfaction by switching to Elixirs. To fix the noise that's
>>>>>>already
>>>>>
>>>>>>there, I'd be inclined to just go in there with a wave editor and
>>>>>>manually
>>>>>
>>>>>>fix the noises one at a time, but then, I'm obsessive. :)
>>>>>>
>>>>>>Sarah
>>>>>>
>>>>>>www.sarahtonin.com
>>>>>>
>>>>>>"Don Nafe" <dnafe@magma.ca> wrote in message news:45834c91@linux...
>>>>>>> Hi all
>>>>>>>
>>>>>>> Actually it isn't bad at all but several inital comments on this
>>>>>>> particular track I'm mixing is there's too much fret noise on the
>>>>>>> acoustic.
>>>>>>>
>>>>>>> Most of it is between chords (picking)
>>>>>>>
>>>>>>> One thought was using a de-esser...all other's are more than welcome
>>>>>>>
>>>>>>> Don
>>>>>>>
>>>>>>
>>>>>>
>>>>>
>>>>
>>>>
>>>
>>
>>
>
>Jamie K <Meta@Dimensional.com> wrote:
>
>What if you start at the end and work backwards?
Then that means I have to start each mix with a preconceived
notion, instead of letting the song "mix itself", as it
were.... IOW, to me a song should just come together - not
necessarily simply/easily, but the song should tell you which
direction to go in, not the other way around.
>Instead of trying this or that voodoo piece of gear, hearing
>what it does, and ditching it if it doesn't do what you want...
I haven't exactly done that, though... I've tried things that
SHOULD get me closer to where I want to be, based on what I've
heard other people say about them.
>Start with what you want. Hear it in your head. Then analyze
>what it is about your final desired sound that equals "killer"
>for you. Is it clean, non-distorted crispness? Is it a type of
distortion? Is it an eq curve, certain frequencies you need to
hear? Is it combinations of instruments that don't step on each
other? Is it stereo wideness?
It's hard to say exactly - it's almost one of those: "I'll
know it when I hear it" kinda things... you know where you dial
something in & just jump out of your chair & say "YEAH! THAT'S
IT RIGHT THERE!" I never have those moments anymore... it's
always: "eh, that's not too bad" or "That's pretty close." or
something like that.
Ultimately, though, what I'm short of mix-wise is a combination
of clarity/power/size. And by "power", I don't mean RMS;
by "size" I don't mean depth or width alone, but a combination
of both. In Paris, I could get the power & size, but not the
clarity; in Cubase I can get the clarity, but I'm coming up
short on the power & size. Summing in Paris helped, but then I
lose some clarity & transparency. Running stems out of Cubase,
then reimporting into a new project to get the final 2-mix
helps too... that way you get a bit more well-defined
soundstage & get to keep the clarity, but this is just another
workaround, really... I still would like to be able to hear the
FINAL 2-track product as I'm mixing, without another step in
between. I was hoping Pulsar could keep me in the digital
domain all the way through the process, but it can't at my
chosen samplerate. Convert everything to 96k then use Pulsar?
Nope, this puts my PC over the top - now I'd be looking at more
money for another upgrade just to see if the experiment would
work. Convert everything to 44.1? Maybe I should... maybe my hi-
rez quest is nothing more than tilting at windmills.
>Record something very simple. For example, just your voice, or
>just your guitar. Can you make just that one track sound like
>you envision
Yes, one track is fine, two is fine, a few is fine, a lot is
even fine... it's the entire mix that I'm not able to get where
I want... maybe it's VST EFX vs hardware, maybe it's ITB mixing
of any kind vs a console, maybe I really like phasey/smeary
analog EQ & just don't know it. Point is, at this stage, I
don't KNOW what would get me there! If I did, I'd implement
that & be done with dicking around.
I appreciate your input, Jamie... it's just one of those things
where I'm 98%-99% there, but that last 1 or 2 percent are the
most important ones - like the keystone at the top of an arch...
it's only one stone, but without it, all the rest collapse.
Neil"Neil" OIUOIU!OIU.com wrote:
>Convert everything to 96k then use Pulsar?
Forgot to mention: I also don't like the math on this option,
either... converting from 88.2k to 96k is like asking the
machine: "Please fuck up every 11th sample on each & every one
of these pristine, hi-rez tracks I recorded.".
NeilI found the problem with the Link Send Panning. I had set my pans exactly
where I wanted them and THEN clicked on "Link Send Routing Panners to Channel
Panner". Guess what? The Send stays CENTERED U N T I L you move the panner.
As soon as you move the pan on the audio channel the send pan follows perfectly.
So the lesson for me is to enable the LSRPtCP before I set my pans.
I have a lot more to read apparently. That's GOOD. Better that then have
a very buggy app. Now I'm back on track.
John
"Neil" <IUI@OIU.com> wrote:
>
>Nah, that # just means 944 more lines of code than v3.1.1.000.
>
>:D
>
>
>"John" <no@no.com> wrote:
>>
>>So I have to wait for 944 upgrades before it gets good? Ouch.
>>
>>"DJ" <nowayjose@dude.net> wrote:
>>>
>>>"Neil" <IUOIU@OIU.com> wrote in message news:4582a485$1@linux...
>>>> This is why I never get version "point-zero" of ANYTHING. Or at
>>>> least I try to avoid it until I find out from other people if
>>>> it's relatively bug-free.
>>>
>>>AMEN!!!.......v3.1.1.944 here. I'm not going anywhere. Everything works
>>(or
>>>at least everything that I use works)
>>>>
>>>> Sorry to hear this, John.
>>>
>>>Me too.
>>>;o(
>>>
>>>
>>
>Can you reserve an IRQ in the BIOS as a Legacy Device and Turn off PNP in
the BIOS too? Does the manual say if Slot 3 shares with anything else?
"Mike R." <emarenot@yahoo.com> wrote:
>
>
>Hey Rob,
>Thanks for responding. I should have mentioned: Win98SE... and no, no "!"
>or "?" shows up in the device manager. In fact, the driver is listed.
But,
>I have to "remove" the bugger before shutting down, so that, when I reboot
>the computer it see's the driver and loads it. If I don't do this, Paris
>will not load. I get an error message "error 61d/something." The EDS card
>is on IRQ3 and shares it only with the APCI thingy. Once I go through this
>reloading-the-driver-procedure, all is well, Paris runs and is happy.
>MR
> "Rob Arsenault" <mani1147 AT hotmail.com> wrote in message
>news:4584659e@linux...
> Strange....!!
> Does the driver show up with a "!" or a "?" in the device manager ? What
>OS you running?
>
> Rob
> "Mike R." <emarenot@yahoo.com> wrote in message news:45840d3c$1@linux...
> Any ideas on why I have to reload my Paris drivers everytime I boot
the
>program? I'm running the EDS card on an Asus A7S333, PCI slot 3. Vid card
>is slot one, slot two is blank to make room for the EDS daughter card.
I
>did try to reinstall Paris (2.0), but that did not seem to fix the problem.
> Thanks gents.
> MR
>
>
><!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
><HTML><HEAD>
><META http-equiv=3DContent-Type content=3D"text/html; =
>charset=3Diso-8859-1">
><META content=3D"MSHTML 6.00.2800.1555" name=3DGENERATOR>
><STYLE></STYLE>
></HEAD>
><BODY bgColor=3D#ffffff>
><DIV><FONT face=3DArial size=3D2>Hey Rob,</FONT></DIV>
><DIV><FONT face=3DArial size=3D2>Thanks for responding. I should =
>have=20
>mentioned: Win98SE... and no, no "!" or "?" shows up in the device=20
>manager. In fact, the driver is listed. But, I have to =
>"remove" the=20
>bugger before shutting down, so that, when I reboot the computer it =
>see's the=20
>driver and loads it. If I don't do this, Paris will not =
>load. I get=20
>an error message "error 61d/something." The EDS card is on IRQ3 =
>and shares=20
>it only with the APCI thingy. Once I go through this=20
>reloading-the-driver-procedure, all is well, Paris runs and is=20
>happy.</FONT></DIV>
><DIV><FONT face=3DArial size=3D2>MR</FONT></DIV>
><BLOCKQUOTE dir=3Dltr=20
>style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
>BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
> <DIV>"Rob Arsenault" <mani1147 AT hotmail.com> wrote in message =
><A=20
> href=3D"news:4584659e@linux">news:4584659e@linux</A>...</DIV>
> <DIV><FONT size=3D2>Strange....!! </FONT></DIV>
> <DIV><FONT size=3D2>Does the driver show up with a "!" or a "?" =
>in the=20
> device manager ? What OS you running?</FONT></DIV>
> <DIV><FONT size=3D2></FONT> </DIV>
> <DIV><FONT size=3D2>Rob</FONT></DIV>
> <BLOCKQUOTE dir=3Dltr=20
> style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
>BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
> <DIV>"Mike R." <<A=20
> href=3D"mailto:emarenot@yahoo.com">emarenot@yahoo.com</A>> wrote =
>in message=20
> <A href=3D"news:45840d3c$1@linux">news:45840d3c$1@linux</A>...</DIV>
> <DIV><FONT face=3DArial size=3D2>Any ideas on why I have to reload =
>my Paris=20
> drivers everytime I boot the program? I'm running the EDS card =
>on an=20
> Asus A7S333, PCI slot 3. Vid card is slot one, slot two is =
>blank to=20
> make room for the EDS daughter card. I did try to reinstall =
>Paris=20
> (2.0), but that did not seem to fix the problem. </FONT></DIV>
> <DIV><FONT face=3DArial size=3D2>Thanks gents.</FONT></DIV>
> <DIV><FONT face=3DArial=20
>size=3D2>MR</FONT></DIV></BLOCKQUOTE></BLOCKQUOTE></BODY ></HTML>
>
>"Neil" <IUOI@OIU.com> wrote in message news:45842218$1@linux...
>
> "DJ" <nowayjose@dude.net> wrote:
>
>>but after Neil's experience with Pulsar, I'm gonna have to add
>>an emphatic "IMHO" and also a big, fat "YMMV" to everything I
>>recommend from now on..
>
> After my experience with Pulsar, I'm tempted to sell everything
> except my guitars & Fantom & simply pay someone else to record
> me when I feel like recording my stuff, and just forget all
> about recording any other people. I'm serious - I'm totally in
> a quandary as to what to do right now... I am SO sick &
> tired of trying this & trying that & still NEVER being able to
> get exactly what I want final mix-wise. "Close" isn't cutting
> it for me anymore.
>
> I really don't know where to go at this point, honestly.
> PTHD? Simple one-shot solution... can't really justify the
> expense, though. Analog Mixer? Same thing as PTHD with regard
> to the expense for a good one & I don't see the benefit of
> going "downscale" there. Digital Mixer? Dunno if that'd do it -
> I'd have to hear it, try it, etc. Passive summing mixer?
> I dunno how much of that is really snake-oil: I'd have to buy-
> it-try-it, a/b mixes, etc, etc, sell it if I hate it, try
> something else, shitcan it if I hate it.... and I'm kinda tired
> of the whole cycle that I've gotten into lately of tweak,
> remix, find a workaround, tweak, remix, try something else,
> tweak/remix, ad nauseum.
>
> I'm NOT being productive, is the point.
>
> So, I have no fucking clue as to where to go next with this, or
> if there's some affordable solution out there somewhere that'll
> get me what I want, which is simply a streamlined, killer-
> sounding, one-box solution. I'm tending to think "not".
>
> Neil
Neil,
I'm very happy with the sonic footprintof my DAW. The hoops I jump through
are ridiculous and I doubt that at the end of the day, anyone but me would
GAF about the sublety of the differences. There are lots of disadvantages to
a multi modular system. I've got so many different things interfacing
digitally that if something starts to act up, the probabality of me finding
it within less than an hour or two are practically nil. I know the system
well though and can pretty much troubleshoot it efficiently, but there are
lots of potential Gremlins. It takes me about 10 minutes to boot from a dead
standstill and get my templates loaded and ready to import files to mix.
That's a little slow, but tolerable.......but if something goes wrong, then
it's 10 more minutes.etc. .........that's when I start going postal.
Tonight I started having some stability problems with a project. I got
kinda' fatigued and sloppy and made some "not so optimal" edits in Cubase. I
think this may have caused some problems when inserting UAD-1
plugins.........I think.............. then again, it might be heat. I can
walk into my machine room and the temperature is 15 -20 degrees above room
temp with the exhaust fans blowing out of 3 x Magmas with 3 x Pulsars, 4 x
UAD-1's and 4 x EDS X cards. I had three glitches within about 5 minutes so
and these cards are working hard so it could be that after 7 hours, they
need a rest. I know I do so we'll see if things behave normally in the
morning when the Magma's are cooled down and I am seeing straight.
Cheers,
Deej
.."John" <no@no.com> wrote:
>
>Does the manual say if Slot 3 shares with anything else?
I can't recall - does Win98 have the "system information"
applet like XP does? I don't mean the "System" icon where you
can get to the Device Manager & whatnot, but the one where you
go to your main menu & click on Programs/Accesssories/System
Tools/ System Information? If it does, then he can open that up
& look under "Hardware Resources"/"IRQ's" and it'll give him a
list of everything & he can see if anything's
conflicting/sharing with Slot 3.
NeilThis is a multi-part message in MIME format.
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Content-Type: text/plain;
charset="iso-8859-1"
Content-Transfer-Encoding: 7bit
http://www.gearslutz.com/board/showthread.php?t=93177
Found this on gearslutz. Weiss has some interesting comments on sampling
rate. He seems to feel that 60khz would be the "ideal" rate. His comments
on 96k vs 88.2 are also interesting. Apparently, his latest product is an
ITB processing unit, like Paris, like the new Fairlight thingy, like
Receptor. I never thought of it this way, but he mentions that his ITB
product (presumably versus some "native" solution) frees up the developer
from being so circumscribed byt the demands of the computer's hardware and
softward. That sounds good, but it seems like six of one, half dozen of the
other. Whenever I hear about scalable architecture I think Roland or Korg,
and how future proof really means one or two years of support/development
after then sale (if that) and on to the next thing.
MR
------=_NextPart_000_0009_01C721CC.A4CEA6C0
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charset="iso-8859-1"
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<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
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</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2><A=20
href=3D"http://www.gearslutz.com/board/showthread.php?t=3D93177">http://w=
ww.gearslutz.com/board/showthread.php?t=3D93177</A></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Found this on gearslutz. Weiss =
has some=20
interesting comments on sampling rate. He seems to feel that 60khz =
would=20
be the "ideal" rate. His comments on 96k vs 88.2 are also=20
interesting. Apparently, his latest product is an ITB processing =
unit,=20
like Paris, like the new Fairlight thingy, like Receptor. I never =
thought=20
of it this way, but he mentions that his =
ITB product (presumably=20
versus some "native" solution) frees up the developer from being so=20
circumscribed byt the demands of the computer's hardware and softward. =
That=20
sounds good, but it seems like six of one, half dozen of the =
other. =20
Whenever I hear about scalable architecture I think Roland or Korg, and =
how=20
future proof really means one or two years of support/development after =
then=20
sale (if that) and on to the next thing. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>MR</FONT></DIV></BODY></HTML>
------=_NextPart_000_0009_01C721CC.A4CEA6C0--Thanks Mike, great read. Gonna go post this link on the 88.2k
thread that DJ started on the Pulsar forum now.
Neil
"Mike R." <emarenot@yahoo.com> wrote:
>
>
>http://www.gearslutz.com/board/showthread.php?t=93177
>
>Found this on gearslutz. Weiss has some interesting comments on sampling
>rate. He seems to feel that 60khz would be the "ideal" rate. His comments
>on 96k vs 88.2 are also interesting. Apparently, his latest product is
an
>ITB processing unit, like Paris, like the new Fairlight thingy, like
>Receptor. I never thought of it this way, but he mentions that his ITB
>product (presumably versus some "native" solution) frees up the developer
>from being so circumscribed byt the demands of the computer's hardware and
>softward. That sounds good, but it seems like six of one, half dozen of
the
>other. Whenever I hear about scalable architecture I think Roland or Korg,
>and how future proof really means one or two years of support/development
>after then sale (if that) and on to the next thing.
>MR
>
>
><!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
><HTML><HEAD>
><META http-equiv=3DContent-Type content=3D"text/html; =
>charset=3Diso-8859-1">
><META content=3D"MSHTML 6.00.2800.1555" name=3DGENERATOR>
><STYLE></STYLE>
></HEAD>
><BODY bgColor=3D#ffffff>
><DIV><FONT face=3DArial size=3D2><A=20
>href=3D"http://www.gearslutz.com/board/showthread.php?t=3D93177">http://w=
>ww.gearslutz.com/board/showthread.php?t=3D93177</A></FONT ></DIV>
><DIV><FONT face=3DArial size=3D2></FONT> </DIV>
><DIV><FONT face=3DArial size=3D2>Found this on gearslutz. Weiss =
>has some=20
>interesting comments on sampling rate. He seems to feel that 60khz =
>would=20
>be the "ideal" rate. His comments on 96k vs 88.2 are also=20
>interesting. Apparently, his latest product is an ITB processing =
>unit,=20
>like Paris, like the new Fairlight thingy, like Receptor. I never =
>thought=20
>of it this way, but he mentions that his =
>ITB product (presumably=20
>versus some "native" solution) frees up the developer from being so=20
>circumscribed byt the demands of the computer's hardware and softward. =
>That=20
>sounds good, but it seems like six of one, half dozen of the =
>other. =20
>Whenever I hear about scalable architecture I think Roland or Korg, and
=
>how=20
>future proof really means one or two years of support/development after
=
>then=20
>sale (if that) and on to the next thing. </FONT></DIV>
><DIV><FONT face=3DArial size=3D2>MR</FONT></DIV></BODY></HTML>
>
>Neil -
Knowing your music, studio setup, and experience/ability, I think you
might consider this advice. You work in a genre that really stresses
the core recording resources of a studio, and have been, to an extent,
patching together a studio to achieve that sound. But things are not
working the way you would like, optimally.
So why not start with a blank slate. Find out what the studios that
produce your reference sounds are using to get those sounds and see if
you can get a similar subset of gear together to get you in the game.
You might say, "Well, they are using large SSL consoles and racks of
vintage preamps and processors and a closet full of expensive mics,
and I can't afford that". However, these days, we can buy single
channels of Neve, SSL, API, etc., consoles, and software/dsp
emulations of virtually every piece of outboard gear ever made. There
are a staggering array of mics that are excellent performers for a
fraction of the cost of classic German models. And finally, sampling
technology has turned live performers into an option and not
necessity.
So, I believe the technology is out there to allow you to make your
music as you envision it. Just keep in mind that if your current tools
aren't cutting it, then its time to move on to new ones. That's how I
arrived to the Paris scene over 5 year ago, and may, someday, move on
to who-knows-what.
- Paul Artola
Ellicott City, MarylandIsn't that the one where you attempt a ground invasion of Russia from
the west?
- Napoleon Artola
Elba, Maryland
On Sat, 16 Dec 2006 20:50:31 -0800, "J.S."
<sonicartproductions@hotmail.com> wrote:
>Neil,
Have you blind tested your 88.2 files vs. 44.1 on your gear? Have you listened
to Voxengo's r8brain (or pro)?
Also, I think you should look up a producer or engineer you think is the
best in your genre and pick their brains, perhaps even hire them to track/mix
a few tunes. You'll probably learn what you want in that scenario. Or,
it's possible that they won't get you closer either, in which case your time
searching for what works for you won't be wasted in any sense (and I don't
think it's wasted in any case). But I think you're close and maybe just
need a little time away to clear the air.
Good luck!
Graham
"Neil" <IUOI@OIU.com> wrote:
>
>"Neil" OIUOIU!OIU.com wrote:
>>Convert everything to 96k then use Pulsar?
>
>Forgot to mention: I also don't like the math on this option,
>either... converting from 88.2k to 96k is like asking the
>machine: "Please fuck up every 11th sample on each & every one
>of these pristine, hi-rez tracks I recorded.".
>
>Neil http://www.roughlydrafted.com/RD/Q4.06/6565C879-3C4C-4014-95 F0-EF1DF464D378.html
What do you guys think?
I've never had a problem burning CDs of music I've purchased with iTunes,
or copying those CDs I've burned. I've never had problems converting files
and dragging and dropping copies to my non apple mp3 player.
JamesI think its much ado about nothing.I have a toshiba mp3 player and you cant
transfer files to another toshiba.they all have a device specific code tag
on every file you load.
doesnt mean a thing if you are dealing with mp3 files to start with.I
wouldnt touch a zune either way
"James McCloskey" <excelsm@hotmail.com> wrote in message
news:4585a8ff$1@linux...
>
> http://www.roughlydrafted.com/RD/Q4.06/6565C879-3C4C-4014-95 F0-EF1DF464D378.html
>
>
> What do you guys think?
>
> I've never had a problem burning CDs of music I've purchased with iTunes,
> or copying those CDs I've burned. I've never had problems converting
> files
> and dragging and dropping copies to my non apple mp3 player.
>
> James
>I have that mobo, and it seems like that slot shares IRQ with another
onboard resource.. perhaps USB or onboard sound card. I don't recall, WinXP
doesn't care and I use every single slot on my A7s333. I'd move the EDS card
down one slot to #4 and see what happens, or if you have good enough airflow
and a normal (not stacked/giant fan&heatsink) video card, It should not be
an issue at all running it in the 2nd PCI slot.
FWIW, I used 3 EDS cards on that mobo at one time using WinME.
AA
"John" <no@no.com> wrote in message news:4584ae53$1@linux...
>
> Can you reserve an IRQ in the BIOS as a Legacy Device and Turn off PNP in
> the BIOS too? Does the manual say if Slot 3 shares with anything else?
>
>
>
> "Mike R." <emarenot@yahoo.com> wrote:
>>
>>
>>Hey Rob,
>>Thanks for responding. I should have mentioned: Win98SE... and no, no "!"
>>or "?" shows up in the device manager. In fact, the driver is listed.
> But,
>>I have to "remove" the bugger before shutting down, so that, when I reboot
>>the computer it see's the driver and loads it. If I don't do this, Paris
>>will not load. I get an error message "error 61d/something." The EDS
>>card
>>is on IRQ3 and shares it only with the APCI thingy. Once I go through
>>this
>>reloading-the-driver-procedure, all is well, Paris runs and is happy.
>>MR
>> "Rob Arsenault" <mani1147 AT hotmail.com> wrote in message
>>news:4584659e@linux...
>> Strange....!!
>> Does the driver show up with a "!" or a "?" in the device manager ? What
>>OS you running?
>>
>> Rob
>> "Mike R." <emarenot@yahoo.com> wrote in message
>> news:45840d3c$1@linux...
>> Any ideas on why I have to reload my Paris drivers everytime I boot
> the
>>program? I'm running the EDS card on an Asus A7S333, PCI slot 3. Vid
>>card
>>is slot one, slot two is blank to make room for the EDS daughter card.
> I
>>did try to reinstall Paris (2.0), but that did not seem to fix the
>>problem.
>> Thanks gents.
>> MR
>>
>>
>><!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
>><HTML><HEAD>
>><META http-equiv=3DContent-Type content=3D"text/html; =
>>charset=3Diso-8859-1">
>><META content=3D"MSHTML 6.00.2800.1555" name=3DGENERATOR>
>><STYLE></STYLE>
>></HEAD>
>><BODY bgColor=3D#ffffff>
>><DIV><FONT face=3DArial size=3D2>Hey Rob,</FONT></DIV>
>><DIV><FONT face=3DArial size=3D2>Thanks for responding. I should =
>>have=20
>>mentioned: Win98SE... and no, no "!" or "?" shows up in the device=20
>>manager. In fact, the driver is listed. But, I have to =
>>"remove" the=20
>>bugger before shutting down, so that, when I reboot the computer it =
>>see's the=20
>>driver and loads it. If I don't do this, Paris will not =
>>load. I get=20
>>an error message "error 61d/something." The EDS card is on IRQ3 =
>>and shares=20
>>it only with the APCI thingy. Once I go through this=20
>>reloading-the-driver-procedure, all is well, Paris runs and is=20
>>happy.</FONT></DIV>
>><DIV><FONT face=3DArial size=3D2>MR</FONT></DIV>
>><BLOCKQUOTE dir=3Dltr=20
>>style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
>>BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
>> <DIV>"Rob Arsenault" <mani1147 AT hotmail.com> wrote in message =
>><A=20
>> href=3D"news:4584659e@linux">news:4584659e@linux</A>...</DIV>
>> <DIV><FONT size=3D2>Strange....!! </FONT></DIV>
>> <DIV><FONT size=3D2>Does the driver show up with a "!" or a "?" =
>>in the=20
>> device manager ? What OS you running?</FONT></DIV>
>> <DIV><FONT size=3D2></FONT> </DIV>
>> <DIV><FONT size=3D2>Rob</FONT></DIV>
>> <BLOCKQUOTE dir=3Dltr=20
>> style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
>>BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
>> <DIV>"Mike R." <<A=20
>> href=3D"mailto:emarenot@yahoo.com">emarenot@yahoo.com</A>> wrote =
>>in message=20
>> <A href=3D"news:45840d3c$1@linux">news:45840d3c$1@linux</A>...</DIV>
>> <DIV><FONT face=3DArial size=3D2>Any ideas on why I have to reload =
>>my Paris=20
>> drivers everytime I boot the program? I'm running the EDS card =
>>on an=20
>> Asus A7S333, PCI slot 3. Vid card is slot one, slot two is =
>>blank to=20
>> make room for the EDS daughter card. I did try to reinstall =
>>Paris=20
>> (2.0), but that did not seem to fix the problem. </FONT></DIV>
>> <DIV><FONT face=3DArial size=3D2>Thanks gents.</FONT></DIV>
>> <DIV><FONT face=3DArial=20
>>size=3D2>MR</FONT></DIV></BLOCKQUOTE></BLOCKQUOTE></BODY ></HTML>
>>
>>
>Paul Artola <artola@comcast.net> wrote:
>So why not start with a blank slate. Find out what the studios
>that produce your reference sounds are using to get those
>sounds and see if you can get a similar subset of gear
>together to get you in the game.
Well, there ARE no reference sounds, really - I mean, yeah it's
Prog, but I'm not necessarily going for a Rush sound or a Dream
Theater sound or a Yes sound, per se'
>You might say, "Well, they are using large SSL consoles and
>racks of vintage preamps and processors and a closet full of
>expensive mics, and I can't afford that".
I have good signal chain compliment, really. Let me show you
something... here's a hi-rez mp3 file of 4 comparison clips from
two projects that I did in the same genre... one was done on
PTHD, tracked with nothing but the best (by most people's
estimation) killer mics & pres - vintage Neve's, Focusrite ISA
110's, API's, Summit tube pres & comps, LA2A's) and mixed
through an SSL 4000 series console using EFX like two Lexi
480L's, a couple of AMS reverbs & delays, you get the picture.
The other was done here with only my own gear (except their
instruments & amps, I mean) & tracked & mixed in CubaseSX.
Both projects were done at the same samplerate (88.2k), and
the four song clips (two clips from each project) are all final
mixes, prior to mastering. Check it out...
http://www.saqqararecords.com/MiscAudio/PTHD-SSL%20vs%20Cuba seSX-ITB.mp3
Now, it's not the same songs mixed in different platforms in an
a/b comparson, but it's close enough in style, and one of the
guitar players & the singer appear on both projects... the 1st
& 3rd song clips are the PTHD/SSL mixes; two & four are the
Cubase/ITB mixes... dare I say the Cubase mixes sound better?
Fuller/more well-defined/bigger.
What do you guys think? Yes? No? Am I nuts?
Problem is, I can't get my OWN music to the right degree of
this bigger/more well-defined (not exactly like this other
stuff in the clips - it's a different style, but you know what
I'm getting at, yes?) place of existence in it's own right.
>So, I believe the technology is out there to allow you to make
>your music as you envision it. Just keep in mind that if your
>current tools aren't cutting it, then its time to move on to
>new ones.
EXACTLY! But, to WHAT?????!!!!!?????
NeilI think they should leave the beta tag on them.
:O
"John" <no@no.com> wrote in message news:4582c8fa$1@linux...
>
> You think they should all rename the .0 releases "The quest for cash"?
>
> hehe"Graham Duncan" <graham@grahamduncan.com> wrote:
>
>Neil,
>
>Have you blind tested your 88.2 files vs. 44.1 on your gear?
Not in the context of individual tracks within an entire song -
that would involve retracking an entire piece track by track...
something I'm not interested in doing... YET! Simply converting
88.2k tracks to 44.1 wouldn't be a fair test becasue one would
have the extra step of conversion & the other one wouldn't.
I have, however, a/b'd 44.1k mixes to 88.2k mixes of the same
song & yes, I can hear a difference there if that's what you
meant.
>Have you listened to Voxengo's r8brain (or pro)?
Yep, I have r8Brain... I think it's a VERY good samplerate
convertor. Don't have the "pro" version, though.
>Also, I think you should look up a producer or engineer you
>think is the best in your genre and pick their brains, perhaps
>even hire them to track/mix a few tunes.
Not a bad idea, but WHO? Maybe it's not the gear, maybe it's
the operator (me).
NeilHey Neil
I'm hearing a definite depth / width issue for sure...sonically you've
pretty well nailed it but there are some big differences in the soundscape.
Also the other mixes sound a tad warmer for some reason.
Don
"Neil" <IUOIU@OIU.com> wrote in message news:4585cae3$1@linux...
>
> Paul Artola <artola@comcast.net> wrote:
>>So why not start with a blank slate. Find out what the studios
>>that produce your reference sounds are using to get those
>>sounds and see if you can get a similar subset of gear
>>together to get you in the game.
>
> Well, there ARE no reference sounds, really - I mean, yeah it's
> Prog, but I'm not necessarily going for a Rush sound or a Dream
> Theater sound or a Yes sound, per se'
>
>>You might say, "Well, they are using large SSL consoles and
>>racks of vintage preamps and processors and a closet full of
>>expensive mics, and I can't afford that".
>
> I have good signal chain compliment, really. Let me show you
> something... here's a hi-rez mp3 file of 4 comparison clips from
> two projects that I did in the same genre... one was done on
> PTHD, tracked with nothing but the best (by most people's
> estimation) killer mics & pres - vintage Neve's, Focusrite ISA
> 110's, API's, Summit tube pres & comps, LA2A's) and mixed
> through an SSL 4000 series console using EFX like two Lexi
> 480L's, a couple of AMS reverbs & delays, you get the picture.
> The other was done here with only my own gear (except their
> instruments & amps, I mean) & tracked & mixed in CubaseSX.
> Both projects were done at the same samplerate (88.2k), and
> the four song clips (two clips from each project) are all final
> mixes, prior to mastering. Check it out...
>
> http://www.saqqararecords.com/MiscAudio/PTHD-SSL%20vs%20Cuba seSX-ITB.mp3
>
>
> Now, it's not the same songs mixed in different platforms in an
> a/b comparson, but it's close enough in style, and one of the
> guitar players & the singer appear on both projects... the 1st
> & 3rd song clips are the PTHD/SSL mixes; two & four are the
> Cubase/ITB mixes... dare I say the Cubase mixes sound better?
> Fuller/more well-defined/bigger.
>
> What do you guys think? Yes? No? Am I nuts?
>
> Problem is, I can't get my OWN music to the right degree of
> this bigger/more well-defined (not exactly like this other
> stuff in the clips - it's a different style, but you know what
> I'm getting at, yes?) place of existence in it's own right.
>
>
>>So, I believe the technology is out there to allow you to make
>>your music as you envision it. Just keep in mind that if your
>>current tools aren't cutting it, then its time to move on to
>>new ones.
>
> EXACTLY! But, to WHAT?????!!!!!?????
>
> Neil"Don Nafe" <dnafe@magma.ca> wrote:
>Hey Neil
>
>I'm hearing a definite depth / width issue for sure...sonically you've
>pretty well nailed it but there are some big differences in the soundscape.
Elaborate, please... which ones do you like better - clips 1 &
3 or 2 & 4? Which ones have the better soundscape?
>Also the other mixes sound a tad warmer for some reason.
The PTHD/SSL mixes, you mean?
Neil>Neil" <OUIOUI@OI.com> wrote in message news:4585db10$1@linux...
>
> "Don Nafe" <dnafe@magma.ca> wrote:
>>Hey Neil
>>
>>I'm hearing a definite depth / width issue for sure...sonically you've
>>pretty well nailed it but there are some big differences in the
>>soundscape.
>
> Elaborate, please... which ones do you like better - clips 1 &
> 3 or 2 & 4? Which ones have the better soundscape?
>
>>Also the other mixes sound a tad warmer for some reason.
>
> The PTHD/SSL mixes, you mean?
>
> Neil
Hey Neil
1 & 2 if I remember were the better mixes at least on my laptop and
headphones (buds)
Sonically 3 & 4 mimic 1 & 2 - the sounds are very similar respectively (1 &
3 / 2 & 4) but there seems to be that " something" that's missing
Now to be fair I too am having a hard time with that "something" and believe
it or not the idea of sitting in with a reputable AE during mixdown is
looking like a necessity and if it's at all possible I will do it this year.
Anyway back to the differences..the instruments in 1 & 2 seem to have more
depth than 3 & 4...and with the additional front to back info I believe you
are able to extend things outward more, which is another area 3 & 4 seem to
be lacking
does any of this make sense?
Don"Don Nafe" <dnafe@magma.ca> wrote:
>Hey Neil
>
>1 & 2 if I remember were the better mixes at least on my
>laptop and headphones (buds).
OK, well #1 is a PTHD/SSL console mix and #2 is a CubaseSX/In-
The-Box Native mix.
>Sonically 3 & 4 mimic 1 & 2 - the sounds are very similar
>respectively (1 & 3 / 2 & 4) but there seems to be
>that "something" that's missing.
OK, but here's the BIG difference, IMO... first of all, the
reason I put them in the order I did is that the first two are
basically rip-roarers all the way through those sections of
each song - good comparison there from one to the other...
3 & 4 are back-to-back for the same kind of reason - both of
them have softer parts, and I placed the softer parts leading
out of one & into the other. Listen to the section from 1:43 in
the file to about 2:27... now, the part from 1:43 'til 2:12 is
definitely warmer-sounding... almost has a "70's" feel to it
in terms of tonality & the way the delay works with the vocals
(that was an AMS delay, BTW); I think it has a great feel, but
you can also hear the bandwidth limitations of going out of
PTHD via analog into the SSL, because as soon as the next clip
comes in at 2:13, listen to the "vertical" expand... what I
mean is the bandwidth - Don, if you're listening on a laptop &
earbuds, I don't know if you can really tell what I'm talking
about - listen on your monitors, burn it to a CD & playback
over them or something if you really want to see what I'm
saying... even on the hi-rez mp3 you can hear the "vertical"
expand as the next clip come in at 2:13.
You guys hearing what I'm talking about? Better defined lows
& highs... plus cleaner/better resolution all-around?
Anyone? Bueller? lol
Neil
>Now to be fair I too am having a hard time with that "something" and believe
>it or not the idea of sitting in with a reputable AE during mixdown is
>looking like a necessity and if it's at all possible I will do it this year.
>
>Anyway back to the differences..the instruments in 1 & 2 seem to have more
>depth than 3 & 4...and with the additional front to back info I believe
you
>are able to extend things outward more, which is another area 3 & 4 seem
to
>be lacking
>
>does any of this make sense?
>
>Don
>
>Thanks for the responses.
Yah, Win 98 does have a system info applet. My card shares an IRQ with
something called "ACPI IRQ holder for PCI IRQ steering" whatever the F that
means/is. Will shifting the EDS card to a new PCI slot shift the IRQ
assignment? Part of the reason I've not tried a move is that I'm afraid it
will and I'll have to go through a song and dance routine to get the EDS
card on an unpopulated IRQ. Aaron -you've got three EDS cards on that
A7S -pretty amazing. How do you keep it cool in there?
Thanks again for the help all.
MR
"Aaron Allen" <know-spam@not_here.dude> wrote in message
news:4585ca22$1@linux...
> I have that mobo, and it seems like that slot shares IRQ with another
> onboard resource.. perhaps USB or onboard sound card. I don't recall,
WinXP
> doesn't care and I use every single slot on my A7s333. I'd move the EDS
card
> down one slot to #4 and see what happens, or if you have good enough
airflow
> and a normal (not stacked/giant fan&heatsink) video card, It should not be
> an issue at all running it in the 2nd PCI slot.
> FWIW, I used 3 EDS cards on that mobo at one time using WinME.
>
> AA
>
>
> "John" <no@no.com> wrote in message news:4584ae53$1@linux...
> >
> > Can you reserve an IRQ in the BIOS as a Legacy Device and Turn off PNP
in
> > the BIOS too? Does the manual say if Slot 3 shares with anything else?
> >
> >
> >
> > "Mike R." <emarenot@yahoo.com> wrote:
> >>
> >>
> >>Hey Rob,
> >>Thanks for responding. I should have mentioned: Win98SE... and no, no
"!"
> >>or "?" shows up in the device manager. In fact, the driver is listed.
> > But,
> >>I have to "remove" the bugger before shutting down, so that, when I
reboot
> >>the computer it see's the driver and loads it. If I don't do this,
Paris
> >>will not load. I get an error message "error 61d/something." The EDS
> >>card
> >>is on IRQ3 and shares it only with the APCI thingy. Once I go through
> >>this
> >>reloading-the-driver-procedure, all is well, Paris runs and is happy.
> >>MR
> >> "Rob Arsenault" <mani1147 AT hotmail.com> wrote in message
> >>news:4584659e@linux...
> >> Strange....!!
> >> Does the driver show up with a "!" or a "?" in the device manager ?
What
> >>OS you running?
> >>
> >> Rob
> >> "Mike R." <emarenot@yahoo.com> wrote in message
> >> news:45840d3c$1@linux...
> >> Any ideas on why I have to reload my Paris drivers everytime I boot
> > the
> >>program? I'm running the EDS card on an Asus A7S333, PCI slot 3. Vid
> >>card
> >>is slot one, slot two is blank to make room for the EDS daughter card.
> > I
> >>did try to reinstall Paris (2.0), but that did not seem to fix the
> >>problem.
> >> Thanks gents.
> >> MR
> >>
> >>
> >><!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
> >><HTML><HEAD>
> >><META http-equiv=3DContent-Type content=3D"text/html; =
> >>charset=3Diso-8859-1">
> >><META content=3D"MSHTML 6.00.2800.1555" name=3DGENERATOR>
> >><STYLE></STYLE>
> >></HEAD>
> >><BODY bgColor=3D#ffffff>
> >><DIV><FONT face=3DArial size=3D2>Hey Rob,</FONT></DIV>
> >><DIV><FONT face=3DArial size=3D2>Thanks for responding. I should =
> >>have=20
> >>mentioned: Win98SE... and no, no "!" or "?" shows up in the device=20
> >>manager. In fact, the driver is listed. But, I have to =
> >>"remove" the=20
> >>bugger before shutting down, so that, when I reboot the computer it =
> >>see's the=20
> >>driver and loads it. If I don't do this, Paris will not =
> >>load. I get=20
> >>an error message "error 61d/something." The EDS card is on IRQ3 =
> >>and shares=20
> >>it only with the APCI thingy. Once I go through this=20
> >>reloading-the-driver-procedure, all is well, Paris runs and is=20
> >>happy.</FONT></DIV>
> >><DIV><FONT face=3DArial size=3D2>MR</FONT></DIV>
> >><BLOCKQUOTE dir=3Dltr=20
> >>style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
> >>BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
> >> <DIV>"Rob Arsenault" <mani1147 AT hotmail.com> wrote in message =
> >><A=20
> >> href=3D"news:4584659e@linux">news:4584659e@linux</A>...</DIV>
> >> <DIV><FONT size=3D2>Strange....!! </FONT></DIV>
> >> <DIV><FONT size=3D2>Does the driver show up with a "!" or a "?" =
> >>in the=20
> >> device manager ? What OS you running?</FONT></DIV>
> >> <DIV><FONT size=3D2></FONT> </DIV>
> >> <DIV><FONT size=3D2>Rob</FONT></DIV>
> >> <BLOCKQUOTE dir=3Dltr=20
> >> style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
> >>BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
> >> <DIV>"Mike R." <<A=20
> >> href=3D"mailto:emarenot@yahoo.com">emarenot@yahoo.com</A>> wrote =
> >>in message=20
> >> <A href=3D"news:45840d3c$1@linux">news:45840d3c$1@linux</A>...</DIV>
> >> <DIV><FONT face=3DArial size=3D2>Any ideas on why I have to reload =
> >>my Paris=20
> >> drivers everytime I boot the program? I'm running the EDS card =
> >>on an=20
> >> Asus A7S333, PCI slot 3. Vid card is slot one, slot two is =
> >>blank to=20
> >> make room for the EDS daughter card. I did try to reinstall =
> >>Paris=20
> >> (2.0), but that did not seem to fix the problem. </FONT></DIV>
> >> <DIV><FONT face=3DArial size=3D2>Thanks gents.</FONT></DIV>
> >> <DIV><FONT face=3DArial=20
> >>size=3D2>MR</FONT></DIV></BLOCKQUOTE></BLOCKQUOTE></BODY ></HTML>
> >>
> >>
> >
>
>Hey Neil
I think I'm hearing what you're referring to. Clip 3 from about 1:43 to to
2:12 and the first clip both definately have that 70's sound to
'em -warmer -the sound is "tighter" also. (BTW I love the AMS effect -just
the exact right amount -and that guy's got great intonation too...) Clips
two and four do seem to expand as you mentioned. I'm hearing lots more high
end detail along with a nice rich bottom.
If your comments are meant to suggest, in part, that ITB solutions can
generate wonderful stuff relative to great outboard gear (SSL, AMS etc..) I
agree. I've not got golden ears, and someone will always be able to hear
with more sophistication than I, but I think that at the level you're
working the differences in the mixes are a matter of taste. Each one of
those mixes sounds great, perhaps for different reasons, but they all sit
real well on my eardrums.
Thanks for taking the time to upload them for the comparison. I'm gonna try
and put up on my studio monitors tomorrow.
MR
"Neil" <OIUOIU@OIU.com> wrote in message news:4585f40b$1@linux...
>
> "Don Nafe" <dnafe@magma.ca> wrote:
> >Hey Neil
> >
> >1 & 2 if I remember were the better mixes at least on my
> >laptop and headphones (buds).
>
> OK, well #1 is a PTHD/SSL console mix and #2 is a CubaseSX/In-
> The-Box Native mix.
>
>
> >Sonically 3 & 4 mimic 1 & 2 - the sounds are very similar
> >respectively (1 & 3 / 2 & 4) but there seems to be
> >that "something" that's missing.
>
> OK, but here's the BIG difference, IMO... first of all, the
> reason I put them in the order I did is that the first two are
> basically rip-roarers all the way through those sections of
> each song - good comparison there from one to the other...
> 3 & 4 are back-to-back for the same kind of reason - both of
> them have softer parts, and I placed the softer parts leading
> out of one & into the other. Listen to the section from 1:43 in
> the file to about 2:27... now, the part from 1:43 'til 2:12 is
> definitely warmer-sounding... almost has a "70's" feel to it
> in terms of tonality & the way the delay works with the vocals
> (that was an AMS delay, BTW); I think it has a great feel, but
> you can also hear the bandwidth limitations of going out of
> PTHD via analog into the SSL, because as soon as the next clip
> comes in at 2:13, listen to the "vertical" expand... what I
> mean is the bandwidth - Don, if you're listening on a laptop &
> earbuds, I don't know if you can really tell what I'm talking
> about - listen on your monitors, burn it to a CD & playback
> over them or something if you really want to see what I'm
> saying... even on the hi-rez mp3 you can hear the "vertical"
> expand as the next clip come in at 2:13.
>
> You guys hearing what I'm talking about? Better defined lows
> & highs... plus cleaner/better resolution all-around?
>
> Anyone? Bueller? lol
>
> Neil
>
>
>
>
> >Now to be fair I too am having a hard time with that "something" and
believe
>
> >it or not the idea of sitting in with a reputable AE during mixdown is
> >looking like a necessity and if it's at all possible I will do it this
year.
> >
> >Anyway back to the differences..the instruments in 1 & 2 seem to have
more
>
> >depth than 3 & 4...and with the additional front to back info I believe
> you
> >are able to extend things outward more, which is another area 3 & 4 seem
> to
> >be lacking
> >
> >does any of this make sense?
> >
> >Don
> >
> >
>I like your mixes better...........much better. I'd say you're there amigo.
Stop sweatin' this and go back to making music. For the genre you're working
with, you've nailed it. You can relax now and stop chasing this (and
spending money.)
;o)
"Neil" <IUOIU@OIU.com> wrote in message news:4585cae3$1@linux...
>
> Paul Artola <artola@comcast.net> wrote:
>>So why not start with a blank slate. Find out what the studios
>>that produce your reference sounds are using to get those
>>sounds and see if you can get a similar subset of gear
>>together to get you in the game.
>
> Well, there ARE no reference sounds, really - I mean, yeah it's
> Prog, but I'm not necessarily going for a Rush sound or a Dream
> Theater sound or a Yes sound, per se'
>
>>You might say, "Well, they are using large SSL consoles and
>>racks of vintage preamps and processors and a closet full of
>>expensive mics, and I can't afford that".
>
> I have good signal chain compliment, really. Let me show you
> something... here's a hi-rez mp3 file of 4 comparison clips from
> two projects that I did in the same genre... one was done on
> PTHD, tracked with nothing but the best (by most people's
> estimation) killer mics & pres - vintage Neve's, Focusrite ISA
> 110's, API's, Summit tube pres & comps, LA2A's) and mixed
> through an SSL 4000 series console using EFX like two Lexi
> 480L's, a couple of AMS reverbs & delays, you get the picture.
> The other was done here with only my own gear (except their
> instruments & amps, I mean) & tracked & mixed in CubaseSX.
> Both projects were done at the same samplerate (88.2k), and
> the four song clips (two clips from each project) are all final
> mixes, prior to mastering. Check it out...
>
> http://www.saqqararecords.com/MiscAudio/PTHD-SSL%20vs%20Cuba seSX-ITB.mp3
>
>
> Now, it's not the same songs mixed in different platforms in an
> a/b comparson, but it's close enough in style, and one of the
> guitar players & the singer appear on both projects... the 1st
> & 3rd song clips are the PTHD/SSL mixes; two & four are the
> Cubase/ITB mixes... dare I say the Cubase mixes sound better?
> Fuller/more well-defined/bigger.
>
> What do you guys think? Yes? No? Am I nuts?
>
> Problem is, I can't get my OWN music to the right degree of
> this bigger/more well-defined (not exactly like this other
> stuff in the clips - it's a different style, but you know what
> I'm getting at, yes?) place of existence in it's own right.
>
>
>>So, I believe the technology is out there to allow you to make
>>your music as you envision it. Just keep in mind that if your
>>current tools aren't cutting it, then its time to move on to
>>new ones.
>
> EXACTLY! But, to WHAT?????!!!!!?????
>
> Neil"DJ" <nowayjose@dude.net> wrote:
>I like your mixes better...........much better. I'd say you're
>there amigo.
OK, well they were ALL "my" mixes... the differences were just
me doing it on PTHD and an SSL out at Sonic Ranch , vs. me
doing it on CubaseSX & mixing ITB here.
>Stop sweatin' this and go back to making music. For the genre
>you're working with, you've nailed it. You can relax now and
>stop chasing this (and spending money.)
OK, you say so; but check this out... here's another set of
comparsion clips, back-to-back in one file. Same segment of the
same song (my stuff this time) - 3 slightly different versions,
neither of which I am 100% happy with (and I'll tell you why
later after I hear your feedback) which of the three do you
like better, or what kind of comments do you have about each?
http://saqqararecords.com/MiscAudio/DracoClip-3versions.mp3
Neil"Mike R." <emarenot@yahoo.com> wrote:
>Hey Neil
>I think I'm hearing what you're referring to. Clip 3 from about 1:43 to
to
>2:12 and the first clip both definately have that 70's sound to
>'em -warmer -the sound is "tighter" also. (BTW I love the AMS effect -just
>the exact right amount -and that guy's got great intonation too...)
Yeah, he did good in that part... he's better with intonation
on the softer stuff - it's when he hits it hard that I need to
use some Auto-Tune on him... interestingly, though, he doesn't
like everything tuned perfectly - he likes a few things to
remain slightly off here & there, and he'll actually pick
certain specific notes that he likes to remain "off" in pitch
sometimes.
>Clips two and four do seem to expand as you mentioned. I'm
>hearing lots more high end detail along with a nice rich
>bottom.
YEPPERS!
>If your comments are meant to suggest, in part, that ITB
>solutions can generate wonderful stuff relative to great
>outboard gear (SSL, AMS etc..) I agree.
Partly, yes, that's part of my intent.
>I think that at the level you're working the differences in
>the mixes are a matter of taste.
The "level" at which I'm working is miles below the level of
any number of perhaps a dozen or so guys on this newsgroup,
but I get what you're saying... i.e: none of the mixes
completely suck, and the differences might be either personal
taste preferences, or choices made at the time that happen to be
more appropriate to the task at hand. HOWEVER... when I point
out what to listen for, you CAN hear the differences I am
referring to... THIS was the point - these are the things I'm
noticing.
>Thanks for taking the time to upload them for the comparison.
>I'm gonna try and put up on my studio monitors tomorrow.
No Prob.. enjoy! If for no other reason than to learn
what "NOT" to do! lol
Neilsoftware can't find hardware.
AA
"J.S." <sonicartproductions@hotmail.com> wrote in message
news:4584ca06$1@linux...
>
>Hmmmm............mix #2 was my choice. It had more depth and breadth to my
ears. ......followed by #3 which I preferred to #1.#1 seemed a bit less 3D
and harder around the edges than 2 or 3. All of the were good mixes. None of
them sounded harsh or unpleasant.
Deej
.
"Neil" <OUOIU@OIU.om> wrote in message news:45860fcc$1@linux...
>
> "DJ" <nowayjose@dude.net> wrote:
>>I like your mixes better...........much better. I'd say you're
>>there amigo.
>
> OK, well they were ALL "my" mixes... the differences were just
> me doing it on PTHD and an SSL out at Sonic Ranch , vs. me
> doing it on CubaseSX & mixing ITB here.
>
>>Stop sweatin' this and go back to making music. For the genre
>>you're working with, you've nailed it. You can relax now and
>>stop chasing this (and spending money.)
>
> OK, you say so; but check this out... here's another set of
> comparsion clips, back-to-back in one file. Same segment of the
> same song (my stuff this time) - 3 slightly different versions,
> neither of which I am 100% happy with (and I'll tell you why
> later after I hear your feedback) which of the three do you
> like better, or what kind of comments do you have about each?
>
> http://saqqararecords.com/MiscAudio/DracoClip-3versions.mp3
>
>
> Neil"DJ" <nowayjose@dude.net> wrote:
>Hmmmm............mix #2 was my choice. It had more depth and
>breadth to my ears. ......followed by #3 which I preferred to
>#1. #1 seemed a bit less 3D and harder around the edges than 2
>or 3. All of the were good mixes. None of them sounded harsh
>or unpleasant.
OK, I'll tell you the differences shortly - I'm hoping to get a
few more opinions if anyone's reading. There's no tricks or
anything like that... it's just one method vs. another.
Chime in, guys!
NeilANYONE? NO ONE?
"Goran Stojiljkovic" <goran.stojiljkovic@os.t-com.hr> wrote in message
news:45813156@linux...
>I would like to run my 3 eds sytem with xp and waves ssl bundle...
> anyone use that combination with no crashes and blue screen.....
>
> please help me to decide..yes or no..it has to be stable because I work
> for money with my Pasis system..
> Splico
>We all,(well nearly all) worked or have worked for Money with our Paris
systems....most on XP.
When I was using Paris, (on XP), it worked flawlessly with the Waves
platinum bundle....SSl wasn't out then.
--
Martin Harrington
www.lendanear-sound.com
"Goran Stojiljkovic" <goran.stojiljkovic@os.t-com.hr> wrote in message
news:45865732@linux...
> ANYONE? NO ONE?
> "Goran Stojiljkovic" <goran.stojiljkovic@os.t-com.hr> wrote in message
> news:45813156@linux...
>>I would like to run my 3 eds sytem with xp and waves ssl bundle...
>> anyone use that combination with no crashes and blue screen.....
>>
>> please help me to decide..yes or no..it has to be stable because I work
>> for money with my Pasis system..
>> Splico
>>
>
>Hmm Neil.... I prefered nr.2, as Deej. But my ears have no problems with any
of the mixes. All 3 sounded good but in a little different ways. Nr.1 have a
little more "tube-old" sound(in my old ears:) but loose something in the
bottom, as nr.2. I'm only listening with headphones on my internet machine
for the moment, but as allways, it will allways be subjective when we're
talking about listening to music.
Erling
"Neil" <OIUIU@OIU.com> skrev i melding news:45863e69$1@linux...
>
> "DJ" <nowayjose@dude.net> wrote:
>>Hmmmm............mix #2 was my choice. It had more depth and
>>breadth to my ears. ......followed by #3 which I preferred to
>>#1. #1 seemed a bit less 3D and harder around the edges than 2
>>or 3. All of the were good mixes. None of them sounded harsh
>>or unpleasant.
>
> OK, I'll tell you the differences shortly - I'm hoping to get a
> few more opinions if anyone's reading. There's no tricks or
> anything like that... it's just one method vs. another.
>
> Chime in, guys!
>
> Neil#2 is my favourite too. #3 got more detailed keyboards. It lifts the
arrangement, but fights a little with the vocal. (At least on my
computer-speakers :-).
Bjorn R
"erlilo" <erlilo@nospamonline.no> wrote in message news:45866fcc@linux...
> Hmm Neil.... I prefered nr.2, as Deej. But my ears have no problems with
any
> of the mixes. All 3 sounded good but in a little different ways. Nr.1 have
a
> little more "tube-old" sound(in my old ears:) but loose something in the
> bottom, as nr.2. I'm only listening with headphones on my internet
machine
> for the moment, but as allways, it will allways be subjective when we're
> talking about listening to music.
>
> Erling
>
>
> "Neil" <OIUIU@OIU.com> skrev i melding news:45863e69$1@linux...
> >
> > "DJ" <nowayjose@dude.net> wrote:
> >>Hmmmm............mix #2 was my choice. It had more depth and
> >>breadth to my ears. ......followed by #3 which I preferred to
> >>#1. #1 seemed a bit less 3D and harder around the edges than 2
> >>or 3. All of the were good mixes. None of them sounded harsh
> >>or unpleasant.
> >
> > OK, I'll tell you the differences shortly - I'm hoping to get a
> > few more opinions if anyone's reading. There's no tricks or
> > anything like that... it's just one method vs. another.
> >
> > Chime in, guys!
> >
> > Neil
>
>"DJ" <nowayjose@dude.net> wrote in message news:45863b07$1@linux...
> Hmmmm............mix #2 was my choice. It had more depth and breadth to my
> ears. ......followed by #3 which I preferred to #1.#1 seemed a bit less 3D
> and harder around the edges than 2 or 3. All of the were good mixes. None
> of them sounded harsh or unpleasant.
>
> Deej
>
I agree with DJ's accessment of the mixes but I prefer # ...just a personal
taste thing here #1 just seemed to fit what my ears like to hear.
DOn
..
> "Neil" <OUOIU@OIU.om> wrote in message news:45860fcc$1@linux...
>>
>> "DJ" <nowayjose@dude.net> wrote:
>>>I like your mixes better...........much better. I'd say you're
>>>there amigo.
>>
>> OK, well they were ALL "my" mixes... the differences were just
>> me doing it on PTHD and an SSL out at Sonic Ranch , vs. me
>> doing it on CubaseSX & mixing ITB here.
>>
>>>Stop sweatin' this and go back to making music. For the genre
>>>you're working with, you've nailed it. You can relax now and
>>>stop chasing this (and spending money.)
>>
>> OK, you say so; but check this out... here's another set of
>> comparsion clips, back-to-back in one file. Same segment of the
>> same song (my stuff this time) - 3 slightly different versions,
>> neither of which I am 100% happy with (and I'll tell you why
>> later after I hear your feedback) which of the three do you
>> like better, or what kind of comments do you have about each?
>>
>> http://saqqararecords.com/MiscAudio/DracoClip-3versions.mp3
>>
>>
>> Neil
>
>You're right Neil
I can definitely hear a difference. in the back to back files now that I
have my ears back.
Don
"Neil" <OIUOIU@OIU.com> wrote in message news:4585f40b$1@linux...
>
> "Don Nafe" <dnafe@magma.ca> wrote:
>>Hey Neil
>>
>>1 & 2 if I remember were the better mixes at least on my
>>laptop and headphones (buds).
>
> OK, well #1 is a PTHD/SSL console mix and #2 is a CubaseSX/In-
> The-Box Native mix.
>
>
>>Sonically 3 & 4 mimic 1 & 2 - the sounds are very similar
>>respectively (1 & 3 / 2 & 4) but there seems to be
>>that "something" that's missing.
>
> OK, but here's the BIG difference, IMO... first of all, the
> reason I put them in the order I did is that the first two are
> basically rip-roarers all the way through those sections of
> each song - good comparison there from one to the other...
> 3 & 4 are back-to-back for the same kind of reason - both of
> them have softer parts, and I placed the softer parts leading
> out of one & into the other. Listen to the section from 1:43 in
> the file to about 2:27... now, the part from 1:43 'til 2:12 is
> definitely warmer-sounding... almost has a "70's" feel to it
> in terms of tonality & the way the delay works with the vocals
> (that was an AMS delay, BTW); I think it has a great feel, but
> you can also hear the bandwidth limitations of going out of
> PTHD via analog into the SSL, because as soon as the next clip
> comes in at 2:13, listen to the "vertical" expand... what I
> mean is the bandwidth - Don, if you're listening on a laptop &
> earbuds, I don't know if you can really tell what I'm talking
> about - listen on your monitors, burn it to a CD & playback
> over them or something if you really want to see what I'm
> saying... even on the hi-rez mp3 you can hear the "vertical"
> expand as the next clip come in at 2:13.
>
> You guys hearing what I'm talking about? Better defined lows
> & highs... plus cleaner/better resolution all-around?
>
> Anyone? Bueller? lol
>
> Neil
>
>
>
>
>>Now to be fair I too am having a hard time with that "something" and
>>believe
>
>>it or not the idea of sitting in with a reputable AE during mixdown is
>>looking like a necessity and if it's at all possible I will do it this
>>year.
>>
>>Anyway back to the differences..the instruments in 1 & 2 seem to have more
>
>>depth than 3 & 4...and with the additional front to back info I believe
> you
>>are able to extend things outward more, which is another area 3 & 4 seem
> to
>>be lacking
>>
>>does any of this make sense?
>>
>>Don
>>
>>
>Interesting... OK, here's the difference between the three
versions - first of all, all three were ITB Native mixes, NOT
using stems or anything like that:
* #1 is mixed with a BuzzMaxi limiter across the 2-buss, set to
-3db threshhold. This one to me is the cleanest of the three
versions. If I go much beyond -3 on this particular piece, you
can really hear the limiting, so -3 is where it remained.
I used UV22hr for dither on this one.
* #2 is the exact same mix, albeit with not one, but THREE
BuzzMaxi's in a row strapped across the 2-buss, each set at
-1db threshhold. This one's still pretty clean, and IMO it
provides a bit more power & impact than does mix #1. Again,
UV22hr provided the dither algo on this version.
* #3 is with Izotope Ozone strapped across the 2-buss instead
of the BuzzMaxi's... in this case, I used a little bit of
compression, a little bit of limiting/volume maximization, and
a little bit of the stereo spread section (no exciter or
anything else), and used the dither algo in Ozone. Now THIS mix
to me has more power, impact & depth, but ALSO - you can hear
some distortion or distortion-like artifacts... hear it?
IOW, for my taste, I like mix 3 better EXCEPT for the
distortion - now, I was being very conservative on compression,
limiting & stereo spread, and although Ozone takes a big chunk
of CPU cycles, my CPU meter still had some headroom (this
project went from 60% to 80% once I strapped Ozone across the
mix) so we know it's not glitching artifacts from overload.
I don't know why I'm getting the distortion, but if I could get
mix 3 without that, THAT'S what I'd be going for.
It's kinda like when I did the summing comparison files on that
other song with Native 2-buss, Native Stems & "Summed in Paris"
mixes... I liked a combination of certain aspects of the Native
Stems mix & the Summed-in-Paris mix, but I can't get all things
hitting at once!
NeilThis is what I do... the Fade handles in Nuendo ROCK for this.
David.
Don Nafe wrote:
> Worst case scenario - strategic surgical editing - to just get the really
> noticable squeeks attenuated or removed...wouldn't even dream of a full song
> edit as squeaks are part of the playing and I personally think they add
> character to the track
>
> DOn
>
>
> "alex plasko" <alex.plasko@snet.net> wrote in message
> news:45842fa9$1@linux...
>
>>neil,I dont know man .If were not putting fret noise in, to make a sampler
>>sound more realistic, were trying to remove it from the real deal for a
>>cleaner sound. does it ever end?
>>"Neil" <IUOIU@OIU.com> wrote in message news:458374eb$1@linux...
>>
>>>I don't know - maybe that'd work... problem with trying to
>>>grab "a" sample of the noise is that that kind of noise isn't
>>>consistent at all - it varies in pitch & harmonic content quite
>>>a bit from one "squeek" to the next.
>>>
>>>Neil
>>>
>>>
>>>"alex plasko" <alex.plasko@snet.net> wrote:
>>>
>>>>neil, how about something like denoiser? if you can get a clean sample of
>>>
>>>>the fret noise denoiser will wipe it and interpolate signal befor and
>>>>after
>>>
>>>>that spot.sound forge denoiser works pretty good for things of that
>>>>nature
>>>
>>>>. never tried it for fret noise though.
>>>>
>>>>"Neil" <OIUOIU@OIU.com> wrote in message news:45835bb9$1@linux...
>>>>
>>>>>Manully editing might be the best bet... Don, if you're trying
>>>>>a de-esser for that, you might need like 3 or 4 of 'em in a
>>>>>row, each centered on different frequencies. Fortunately,
>>>>>Spitfish doesn't take up much CPU power (if that's what you're
>>>>>using).
>>>>>
>>>>>Also, you could try re-micing it through your monitors (muting
>>>>>the playback on the recording channel) with a dynamic mic
>>>>>backed-off a bit from the speakers... whatever you do, don't
>>>>>compress when playing back to re-mic if you try this. That'll
>>>>>just make the squeak stand out more.
>>>>>
>>>>>Wanna send me a sample of the wav file & I'll see if I can come
>>>>>up with an idea for you?
>>>>>
>>>>>Neil
>>>>>
>>>>>
>>>>>"Sarah" <sarahjane@sarahtonin.com> wrote:
>>>>>
>>>>>> Are you talking about strings buzzing against frets, or fingers
>>>>>>chirping and squeaking on the string windings? Cuz I've solved the
>>>>>>latter
>>>>>
>>>>>>to my satisfaction by switching to Elixirs. To fix the noise that's
>>>>>>already
>>>>>
>>>>>>there, I'd be inclined to just go in there with a wave editor and
>>>>>>manually
>>>>>
>>>>>>fix the noises one at a time, but then, I'm obsessive. :)
>>>>>>
>>>>>>Sarah
>>>>>>
>>>>>>www.sarahtonin.com
>>>>>>
>>>>>>"Don Nafe" <dnafe@magma.ca> wrote in message news:45834c91@linux...
>>>>>>
>>>>>>>Hi all
>>>>>>>
>>>>>>>Actually it isn't bad at all but several inital comments on this
>>>>>>>particular track I'm mixing is there's too much fret noise on the
>>>>>>>acoustic.
>>>>>>>
>>>>>>>Most of it is between chords (picking)
>>>>>>>
>>>>>>>One thought was using a de-esser...all other's are more than welcome
>>>>>>>
>>>>>>>Don
>>>>>>>
>>>>>>
>>>>>>
>>>>
>>
>
>Hey all,
Does anyone know if a new "Journaled" MAC Pro can write to a "Non
Journaled" Firewire drive from a G4 running Tiger?
I know for a fact it doesn't work the other way around.
Thanks,
David.EK Sound <askme@nospam.com> wrote:
>Hey all,
>
>Does anyone know if a new "Journaled" MAC Pro can write to a "Non
>Journaled" Firewire drive from a G4 running Tiger?
>
>I know for a fact it doesn't work the other way around.
>
>Thanks,
>
>David.
I can't give you a definitive answer, because I don't really know. Mac OSX
has had Journaling since 10.3, and it runs on a G4. Here is some info on
the subject:
http://docs.info.apple.com/article.html?artnum=107249
http://www.hmug.org/HowTos/JFS.html
http://www.kernelthread.com/mac/osx/arch_fs.html
http://www.macosxhints.com/article.php?story=200211120022008 6
JamesTry this link:
http://search.info.apple.com/?search=Go&lr=lang_en&k word=&type=&newstype=&q=Journaling
"James McCloskey" <excelsm@hotmail.com> wrote:
>
>EK Sound <askme@nospam.com> wrote:
>>Hey all,
>>
>>Does anyone know if a new "Journaled" MAC Pro can write to a "Non
>>Journaled" Firewire drive from a G4 running Tiger?
>>
>>I know for a fact it doesn't work the other way around.
>>
>>Thanks,
>>
>>David.
>
>I can't give you a definitive answer, because I don't really know. Mac
OSX
>has had Journaling since 10.3, and it runs on a G4. Here is some info on
>the subject:
>
>http://docs.info.apple.com/article.html?artnum=107249
>
>http://www.hmug.org/HowTos/JFS.html
>
>http://www.kernelthread.com/mac/osx/arch_fs.html
>
> http://www.macosxhints.com/article.php?story=200211120022008 6
>
>James
>What exactly IS "journaling"? Not familiar with that term at all.
Neil
"James McCloskey" <excelsm@hotmail.com> wrote:
>
>Try this link:
>
> http://search.info.apple.com/?search=Go&lr=lang_en&k word=&type=&newstype=&q=Journaling
>
>
>"James McCloskey" <excelsm@hotmail.com> wrote:
>>
>>EK Sound <askme@nospam.com> wrote:
>>>Hey all,
>>>
>>>Does anyone know if a new "Journaled" MAC Pro can write to a "Non
>>>Journaled" Firewire drive from a G4 running Tiger?
>>>
>>>I know for a fact it doesn't work the other way around.
>>>
>>>Thanks,
>>>
>>>David.
>>
>>I can't give you a definitive answer, because I don't really know. Mac
>OSX
>>has had Journaling since 10.3, and it runs on a G4. Here is some info
on
>>the subject:
>>
>>http://docs.info.apple.com/article.html?artnum=107249
>>
>>http://www.hmug.org/HowTos/JFS.html
>>
>>http://www.kernelthread.com/mac/osx/arch_fs.html
>>
>> http://www.macosxhints.com/article.php?story=200211120022008 6
>>
>>James
>>
>This is a multi-part message in MIME format.
------=_NextPart_000_002D_01C722AD.4E884AF0
Content-Type: text/plain;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
Neil,
I haven't had time to listen but just read all the words here.
It sounds to me like you need to let a great mastering engineer
do their magic. That will probably take you to the next level.
Pick the mix that comes closest to your liking. Master to add/subtract
the qualities that aren't quite there. The big boys will make that =
happen.
This happens to me almost every time. =20
Tom
"Don Nafe" <dnafe@magma.ca> wrote in message news:45868c0f@linux...
"DJ" <nowayjose@dude.net> wrote in message news:45863b07$1@linux...
> Hmmmm............mix #2 was my choice. It had more depth and breadth =
to my=20
> ears. ......followed by #3 which I preferred to #1.#1 seemed a bit =
less 3D=20
> and harder around the edges than 2 or 3. All of the were good mixes. =
None=20
> of them sounded harsh or unpleasant.
>
> Deej
>
I agree with DJ's accessment of the mixes but I prefer # ...just a =
personal=20
taste thing here #1 just seemed to fit what my ears like to hear.
DOn
.
> "Neil" <OUOIU@OIU.om> wrote in message news:45860fcc$1@linux...
>>
>> "DJ" <nowayjose@dude.net> wrote:
>>>I like your mixes better...........much better. I'd say you're
>>>there amigo.
>>
>> OK, well they were ALL "my" mixes... the differences were just
>> me doing it on PTHD and an SSL out at Sonic Ranch , vs. me
>> doing it on CubaseSX & mixing ITB here.
>>
>>>Stop sweatin' this and go back to making music. For the genre
>>>you're working with, you've nailed it. You can relax now and
>>>stop chasing this (and spending money.)
>>
>> OK, you say so; but check this out... here's another set of
>> comparsion clips, back-to-back in one file. Same segment of the
>> same song (my stuff this time) - 3 slightly different versions,
>> neither of which I am 100% happy with (and I'll tell you why
>> later after I hear your feedback) which of the three do you
>> like better, or what kind of comments do you have about each?
>>
>> http://saqqararecords.com/MiscAudio/DracoClip-3versions.mp3
>>
>>
>> Neil
>
>=20
I choose Polesoft Lockspam to fight spam, and you?
http://www.polesoft.com/refer.html
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<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Neil,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I haven't had time to listen but just =
read all the=20
words here.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>It sounds to me like you need to let =
a great=20
mastering engineer</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>do their magic. That will =
probably take you=20
to the next level.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Pick the mix that comes closest to your =
liking. Master to add/subtract</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>the qualities that aren't quite =
there. The=20
big boys will make that happen.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>This happens to me almost every =
time. =20
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Tom</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV>"Don Nafe" <<A =
href=3D"mailto:dnafe@magma.ca">dnafe@magma.ca</A>> wrote=20
in message <A =
href=3D"news:45868c0f@linux">news:45868c0f@linux</A>...</DIV>
<BLOCKQUOTE=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">"DJ"=20
<<A href=3D"mailto:nowayjose@dude.net">nowayjose@dude.net</A>> =
wrote in=20
message <A =
href=3D"news:45863b07$1@linux">news:45863b07$1@linux</A>...<BR>>=20
Hmmmm............mix #2 was my choice. It had more depth and breadth =
to my=20
<BR>> ears. ......followed by #3 which I preferred to #1.#1 seemed =
a bit=20
less 3D <BR>> and harder around the edges than 2 or 3. All of the =
were good=20
mixes. None <BR>> of them sounded harsh or =
unpleasant.<BR>><BR>>=20
Deej<BR>><BR><BR>I agree with DJ's accessment of the mixes but I =
prefer #=20
...just a personal <BR>taste thing here #1 just seemed to fit what my =
ears=20
like to hear.<BR><BR>DOn<BR><BR><BR><BR><BR>.<BR>> "Neil" <<A=20
href=3D"mailto:OUOIU@OIU.om">OUOIU@OIU.om</A>> wrote in message <A=20
=
href=3D"news:45860fcc$1@linux">news:45860fcc$1@linux</A>...<BR>>><B=
R>>>=20
"DJ" <<A =
href=3D"mailto:nowayjose@dude.net">nowayjose@dude.net</A>>=20
wrote:<BR>>>>I like your mixes better...........much better. =
I'd say=20
you're<BR>>>>there amigo.<BR>>><BR>>> OK, well =
they were=20
ALL "my" mixes... the differences were just<BR>>> me doing it on =
PTHD=20
and an SSL out at Sonic Ranch , vs. me<BR>>> doing it on =
CubaseSX &=20
mixing ITB here.<BR>>><BR>>>>Stop sweatin' this and go =
back to=20
making music. For the genre<BR>>>>you're working with, you've =
nailed=20
it. You can relax now and<BR>>>>stop chasing this (and =
spending=20
money.)<BR>>><BR>>> OK, you say so; but check this out... =
here's=20
another set of<BR>>> comparsion clips, back-to-back in one file. =
Same=20
segment of the<BR>>> same song (my stuff this time) - 3 slightly =
different versions,<BR>>> neither of which I am 100% happy with =
(and=20
I'll tell you why<BR>>> later after I hear your feedback) which =
of the=20
three do you<BR>>> like better, or what kind of comments do you =
have=20
about each?<BR>>><BR>>> <A=20
=
href=3D"http://saqqararecords.com/MiscAudio/DracoClip-3versions.mp3">http=
://saqqararecords.com/MiscAudio/DracoClip-3versions.mp3</A><BR>>><B=
R>>><BR>>>=20
Neil<BR>><BR>> <BR><BR></BLOCKQUOTE>
<DIV><FONT size=3D2><BR><BR>I choose Polesoft Lockspam to fight spam, =
and=20
you?<BR><A=20
href=3D"http://www.polesoft.com/refer.html">http://www.polesoft.com/refer=
..html</A> </FONT></DIV></BODY ></HTML>
------=_NextPart_000_002D_01C722AD.4E884AF0--"Neil" <IUOIU@OIU.com> wrote:
>
>What exactly IS "journaling"? Not familiar with that term at all.
>
>Neil
>
From Apple:
"When you enable journaling on a disk, a continuous record of changes to
files on the disk is maintained in the journal. If your computer stops because
of a power failure or some other issue, the journal is used to restore the
disk to a known-good state when the server restarts.
"With journaling turned on, the file system logs transactions as they occur.
If the server fails in the middle of an operation, the file system can 'replay'
the information in its log and complete the operation when the server restarts."
GeneAlso from Apple:
Backward Compatible
Journaled file system is part of a set of incremental enhancements to the
Mac OS Extended file system, and it is backward compatible with the Mac OS
Extended file system. Users can read, write, and access journaled Mac OS
Extended volumes on computers that do not have a journaling feature.
Most disk utilities designed to work with the Mac OS Extended file system
can also be used when journaling is turned on. You should check with your
disk utility vendor before using earlier disk utilities with a journaled
file system.
James
EK Sound <askme@nospam.com> wrote:
>Hey all,
>
>Does anyone know if a new "Journaled" MAC Pro can write to a "Non
>Journaled" Firewire drive from a G4 running Tiger?
>
>I know for a fact it doesn't work the other way around.
>
>Thanks,
>
>David.I don't know if this will help you or not, there is also Target Disk Mode.
How to use FireWire target disk mode
http://docs.info.apple.com/article.html?artnum=58583
More info on Target Disk Mode:
http://search.info.apple.com/?search=Go&lr=lang_en&k word=&type=&newstype=&q=Firewire%20target%20disk
James
EK Sound <askme@nospam.com> wrote:
>Hey all,
>
>Does anyone know if a new "Journaled" MAC Pro can write to a "Non
>Journaled" Firewire drive from a G4 running Tiger?
>
>I know for a fact it doesn't work the other way around.
>
>Thanks,
>
>David.Neil,
You test confuses the hell out of me. Reason: You have so many variables(plugins
on this) another on that ..What gives??
On of all of your test, none have been on equal footing.
How about NO plugins on the master bus? Nothing..Then, render the stereo
file and see which sounds best. Your test always seems like you're hiding
soemthing from us the listeners and it seems that you;re always rigging the
test towards Cubase SX.
Being a Neundo owner and user, there is no way in hell that you, me or anybody
is going to get a better mix (ITB) over and SSL summed mix .. The Summing
bus in SX/ Nuendo is it's achilles heel and needs to improve big time, especialy
for mixing agressive music (Hard Rock /Hip Hop R & B..) You have to really
play it safe with the faders (ITB) mixing those genres..
But, when you're mixing a project over 60 tracks with lots of plugs, the
mixing buss starts to to suffer. Unlike Paris and Pro Tools, which mix summing
busses hold's up very well, Neundo /SX sounds losses it's dept..
What are you going for??
"Neil" <IUOIU@OIU.com> wrote:
>
>Interesting... OK, here's the difference between the three
>versions - first of all, all three were ITB Native mixes, NOT
>using stems or anything like that:
>
>* #1 is mixed with a BuzzMaxi limiter across the 2-buss, set to
>-3db threshhold. This one to me is the cleanest of the three
>versions. If I go much beyond -3 on this particular piece, you
>can really hear the limiting, so -3 is where it remained.
>I used UV22hr for dither on this one.
>
>* #2 is the exact same mix, albeit with not one, but THREE
>BuzzMaxi's in a row strapped across the 2-buss, each set at
>-1db threshhold. This one's still pretty clean, and IMO it
>provides a bit more power & impact than does mix #1. Again,
>UV22hr provided the dither algo on this version.
>
>* #3 is with Izotope Ozone strapped across the 2-buss instead
>of the BuzzMaxi's... in this case, I used a little bit of
>compression, a little bit of limiting/volume maximization, and
>a little bit of the stereo spread section (no exciter or
>anything else), and used the dither algo in Ozone. Now THIS mix
>to me has more power, impact & depth, but ALSO - you can hear
>some distortion or distortion-like artifacts... hear it?
>
>IOW, for my taste, I like mix 3 better EXCEPT for the
>distortion - now, I was being very conservative on compression,
>limiting & stereo spread, and although Ozone takes a big chunk
>of CPU cycles, my CPU meter still had some headroom (this
>project went from 60% to 80% once I strapped Ozone across the
>mix) so we know it's not glitching artifacts from overload.
>I don't know why I'm getting the distortion, but if I could get
>mix 3 without that, THAT'S what I'd be going for.
>
>It's kinda like when I did the summing comparison files on that
>other song with Native 2-buss, Native Stems & "Summed in Paris"
>mixes... I liked a combination of certain aspects of the Native
>Stems mix & the Summed-in-Paris mix, but I can't get all things
>hitting at once!
>
>NeilThis is a multi-part message in MIME format.
------=_NextPart_000_003D_01C722A8.F92944F0
Content-Type: text/plain;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
Good point Tom.
;o)
"Tom Bruhl" <arpegio@comcast.net> wrote in message =
news:4586e364@linux...
Neil,
I haven't had time to listen but just read all the words here.
It sounds to me like you need to let a great mastering engineer
do their magic. That will probably take you to the next level.
Pick the mix that comes closest to your liking. Master to =
add/subtract
the qualities that aren't quite there. The big boys will make that =
happen.
This happens to me almost every time. =20
Tom
"Don Nafe" <dnafe@magma.ca> wrote in message news:45868c0f@linux...
"DJ" <nowayjose@dude.net> wrote in message news:45863b07$1@linux...
> Hmmmm............mix #2 was my choice. It had more depth and =
breadth to my=20
> ears. ......followed by #3 which I preferred to #1.#1 seemed a bit =
less 3D=20
> and harder around the edges than 2 or 3. All of the were good =
mixes. None=20
> of them sounded harsh or unpleasant.
>
> Deej
>
I agree with DJ's accessment of the mixes but I prefer # ...just a =
personal=20
taste thing here #1 just seemed to fit what my ears like to hear.
DOn
.
> "Neil" <OUOIU@OIU.om> wrote in message news:45860fcc$1@linux...
>>
>> "DJ" <nowayjose@dude.net> wrote:
>>>I like your mixes better...........much better. I'd say you're
>>>there amigo.
>>
>> OK, well they were ALL "my" mixes... the differences were just
>> me doing it on PTHD and an SSL out at Sonic Ranch , vs. me
>> doing it on CubaseSX & mixing ITB here.
>>
>>>Stop sweatin' this and go back to making music. For the genre
>>>you're working with, you've nailed it. You can relax now and
>>>stop chasing this (and spending money.)
>>
>> OK, you say so; but check this out... here's another set of
>> comparsion clips, back-to-back in one file. Same segment of the
>> same song (my stuff this time) - 3 slightly different versions,
>> neither of which I am 100% happy with (and I'll tell you why
>> later after I hear your feedback) which of the three do you
>> like better, or what kind of comments do you have about each?
>>
>> http://saqqararecords.com/MiscAudio/DracoClip-3versions.mp3
>>
>>
>> Neil
>
>=20
I choose Polesoft Lockspam to fight spam, and you?
http://www.polesoft.com/refer.html
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<DIV><FONT face=3DArial size=3D2>Good point Tom.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>;o)</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Tom Bruhl" <<A=20
href=3D"mailto:arpegio@comcast.net">arpegio@comcast.net</A>> wrote =
in message=20
<A href=3D"news:4586e364@linux">news:4586e364@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>Neil,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I haven't had time to listen but just =
read all=20
the words here.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>It sounds to me like you need to let =
a great=20
mastering engineer</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>do their magic. That will =
probably take you=20
to the next level.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Pick the mix that comes closest to =
your=20
liking. Master to add/subtract</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>the qualities that aren't quite =
there. The=20
big boys will make that happen.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>This happens to me almost every =
time. =20
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Tom</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV>"Don Nafe" <<A =
href=3D"mailto:dnafe@magma.ca">dnafe@magma.ca</A>>=20
wrote in message <A=20
href=3D"news:45868c0f@linux">news:45868c0f@linux</A>...</DIV>
<BLOCKQUOTE=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">"DJ"=20
<<A href=3D"mailto:nowayjose@dude.net">nowayjose@dude.net</A>> =
wrote in=20
message <A =
href=3D"news:45863b07$1@linux">news:45863b07$1@linux</A>...<BR>>=20
Hmmmm............mix #2 was my choice. It had more depth and breadth =
to my=20
<BR>> ears. ......followed by #3 which I preferred to #1.#1 =
seemed a bit=20
less 3D <BR>> and harder around the edges than 2 or 3. All of the =
were=20
good mixes. None <BR>> of them sounded harsh or=20
unpleasant.<BR>><BR>> Deej<BR>><BR><BR>I agree with DJ's =
accessment=20
of the mixes but I prefer # ...just a personal <BR>taste thing here =
#1 just=20
seemed to fit what my ears like to=20
hear.<BR><BR>DOn<BR><BR><BR><BR><BR>.<BR>> "Neil" <<A=20
href=3D"mailto:OUOIU@OIU.om">OUOIU@OIU.om</A>> wrote in message =
<A=20
=
href=3D"news:45860fcc$1@linux">news:45860fcc$1@linux</A>...<BR>>><B=
R>>>=20
"DJ" <<A =
href=3D"mailto:nowayjose@dude.net">nowayjose@dude.net</A>>=20
wrote:<BR>>>>I like your mixes better...........much =
better. I'd=20
say you're<BR>>>>there amigo.<BR>>><BR>>> OK, =
well they=20
were ALL "my" mixes... the differences were just<BR>>> me =
doing it on=20
PTHD and an SSL out at Sonic Ranch , vs. me<BR>>> doing it on =
CubaseSX=20
& mixing ITB here.<BR>>><BR>>>>Stop sweatin' this =
and go=20
back to making music. For the genre<BR>>>>you're working =
with,=20
you've nailed it. You can relax now and<BR>>>>stop chasing =
this=20
(and spending money.)<BR>>><BR>>> OK, you say so; =
but=20
check this out... here's another set of<BR>>> comparsion =
clips,=20
back-to-back in one file. Same segment of the<BR>>> same song =
(my=20
stuff this time) - 3 slightly different versions,<BR>>> =
neither of=20
which I am 100% happy with (and I'll tell you why<BR>>> later =
after I=20
hear your feedback) which of the three do you<BR>>> like =
better, or=20
what kind of comments do you have about =
each?<BR>>><BR>>> <A=20
=
href=3D"http://saqqararecords.com/MiscAudio/DracoClip-3versions.mp3">http=
://saqqararecords.com/MiscAudio/DracoClip-3versions.mp3</A><BR>>><B=
R>>><BR>>>=20
Neil<BR>><BR>> <BR><BR></BLOCKQUOTE>
<DIV><FONT size=3D2><BR><BR>I choose Polesoft Lockspam to fight spam, =
and=20
you?<BR><A=20
=
href=3D"http://www.polesoft.com/refer.html">http://www.polesoft.com/refer=
..html</A> </FONT></DIV></BLOCKQUOTE ></BODY></HTML>
------=_NextPart_000_003D_01C722A8.F92944F0--"LaMont" <jjdpro@ameritech.net> wrote:
>
>Neil,
>You test confuses the hell out of me. Reason: You have so many variables(plugins
>on this) another on that ..What gives??
I'm just trying different things on them, Lamont - this wasn't
intended to be a blind a/b test, it's simply a matter of: "here's
what I did, here's what I think about the sound of each one;
what do you guys think?"
>On of all of your test, none have been on equal footing.
>How about NO plugins on the master bus? Nothing..Then, render the stereo
>file and see which sounds best. Your test always seems like you're hiding
>soemthing from us the listeners and it seems that you;re always rigging
the
>test towards Cubase SX.
????? How do you figure? When I did the summing comparisons
They were all the exact same mix, just summed in different ways -
I even said that I liked certain aspects of the summed-in-Paris
mix myself! When I posted the file that had 2 PTHD/SSL mixes & 2
Cubase Mixes, I would've had to have planned on rigging that
comparison well over a year ago & would've had to PURPOSELY do
less than my best efforts on the SSL mixes at the time.
>Being a Neundo owner and user, there is no way in hell that you, me or anybody
>is going to get a better mix (ITB) over and SSL summed mix ..
So what are you saying... that you think the Cubase mixes sound
better and that I must've purposely modified the SSL files to
make them sound worse? Or do you think the Cubase mixes sound
better and you simply refuse to believe what you're hearing?
Or do you disagree & think that the SSL mixes sound better... in
which case, I have no issue with your choice - it would simply
be a matter of personal preference. I'd disagree with you in
this instance because I happen to think the Cubase ones DO sound
better.
Lamont, I'm not "rigging" anything - I have no reason to pimp for
Steinberg or anything like that, nor am I trying to say "my kung-
fu is better" just to defend my gear choice. I'm trying to find
out how to get a certain sound out of it, is all.
Neilneil, they all sound good to me .id be happy to work with that caliber of
talent on a regular basis.the best advice in this thread is to send it out
to a mastering house.different set of ears /perspective,and not so attached
to it, etc.
>>Neil,
>>You test confuses the hell out of me. Reason: You have so many
>>variables(plugins
>>on this) another on that ..What gives??
>
> I'm just trying different things on them, Lamont - this wasn't
> intended to be a blind a/b test, it's simply a matter of: "here's
> what I did, here's what I think about the sound of each one;
> what do you guys think?"
>
>
>>On of all of your test, none have been on equal footing.
>>How about NO plugins on the master bus? Nothing..Then, render the stereo
>>file and see which sounds best. Your test always seems like you're hiding
>>soemthing from us the listeners and it seems that you;re always rigging
> the
>>test towards Cubase SX.
>
> ????? How do you figure? When I did the summing comparisons
> They were all the exact same mix, just summed in different ways -
> I even said that I liked certain aspects of the summed-in-Paris
> mix myself! When I posted the file that had 2 PTHD/SSL mixes & 2
> Cubase Mixes, I would've had to have planned on rigging that
> comparison well over a year ago & would've had to PURPOSELY do
> less than my best efforts on the SSL mixes at the time.
>
>>Being a Neundo owner and user, there is no way in hell that you, me or
>>anybody
>>is going to get a better mix (ITB) over and SSL summed mix ..
>
> So what are you saying... that you think the Cubase mixes sound
> better and that I must've purposely modified the SSL files to
> make them sound worse? Or do you think the Cubase mixes sound
> better and you simply refuse to believe what you're hearing?
>
> Or do you disagree & think that the SSL mixes sound better... in
> which case, I have no issue with your choice - it would simply
> be a matter of personal preference. I'd disagree with you in
> this instance because I happen to think the Cubase ones DO sound
> better.
>
> Lamont, I'm not "rigging" anything - I have no reason to pimp for
> Steinberg or anything like that, nor am I trying to say "my kung-
> fu is better" just to defend my gear choice. I'm trying to find
> out how to get a certain sound out of it, is all.
>
> Neil"I'd disagree with you in this instance because I happen to think the Cubase
ones DO sound better."
Then that SSL Engineer does not know what they are doing with board. There's
no way a mix coming off of that board SSL should sound better than a ITB
Cubase SX mix..
Sorry, that just does not jive. That engineer does not know how to push he
SSL or just not familiar with it.
"Neil" <oiUOIU@OIU.com> wrote:
>
>"LaMont" <jjdpro@ameritech.net> wrote:
>>
>>Neil,
>>You test confuses the hell out of me. Reason: You have so many variables(plugins
>>on this) another on that ..What gives??
>
>I'm just trying different things on them, Lamont - this wasn't
>intended to be a blind a/b test, it's simply a matter of: "here's
>what I did, here's what I think about the sound of each one;
>what do you guys think?"
>
>
>>On of all of your test, none have been on equal footing.
>>How about NO plugins on the master bus? Nothing..Then, render the stereo
>>file and see which sounds best. Your test always seems like you're hiding
>>soemthing from us the listeners and it seems that you;re always rigging
>the
>>test towards Cubase SX.
>
>????? How do you figure? When I did the summing comparisons
>They were all the exact same mix, just summed in different ways -
>I even said that I liked certain aspects of the summed-in-Paris
>mix myself! When I posted the file that had 2 PTHD/SSL mixes & 2
>Cubase Mixes, I would've had to have planned on rigging that
>comparison well over a year ago & would've had to PURPOSELY do
>less than my best efforts on the SSL mixes at the time.
>
>>Being a Neundo owner and user, there is no way in hell that you, me or
anybody
>>is going to get a better mix (ITB) over and SSL summed mix ..
>
>So what are you saying... that you think the Cubase mixes sound
>better and that I must've purposely modified the SSL files to
>make them sound worse? Or do you think the Cubase mixes sound
>better and you simply refuse to believe what you're hearing?
>
>Or do you disagree & think that the SSL mixes sound better... in
>which case, I have no issue with your choice - it would simply
>be a matter of personal preference. I'd disagree with you in
>this instance because I happen to think the Cubase ones DO sound
>better.
>
>Lamont, I'm not "rigging" anything - I have no reason to pimp for
>Steinberg or anything like that, nor am I trying to say "my kung-
>fu is better" just to defend my gear choice. I'm trying to find
>out how to get a certain sound out of it, is all.
>
>Neil"LaMont" <jjdpro@ameritech.net> wrote:
>
>"I'd disagree with you in this instance because I happen to think the Cubase
>ones DO sound better."
>
>Then that SSL Engineer does not know what they are doing with board. There's
>no way a mix coming off of that board SSL should sound better than a ITB
>Cubase SX mix..
>
>Sorry, that just does not jive. That engineer does not know how to push
he
>SSL or just not familiar with it.
You're not really paying attention, are you? It was the same
engineer (me). And as far as whether or not I know how to use
that particular board, I guess that would be a matter of
your opinion. I don't think the SSL mixes are bad ones, I think
they came out good; I just think that you can hear more detail
in the ITB mixes in the examples I gave, and they have more
wideband frequency content from top to bottom.
Anyway, my point of that particular comparison wasn't to say
"ITB mixes are better than using a large-format console that
costs somewhere in the six-figure range", the point of it was to
address a signal-chain suggestion that Paul had... he had
suggested perhaps that I needed to pick up a few pieces of
killer vintage gear, and I was just demonstrating that I think
the various signal chain components that I have here are on par
with most anything that can be found in heavy-hitter studios...
we used probably around $100k's worth of mics & pre's on the
PTHD/SSL mixes, plus obviously you're looking at another
roughly $100k for that particular console (40-channel E-series,
black EQ's, w/G-series Computer & Total Recall package), add in
the PTHD, outboard gear & whatnot, and you end up with
somewhere around a quarter-mil's worth of equipment involved in
that project. The project done at my place was done with my
gear, which certainly doesn't tally up to anywhere remotely
close to that cost & none of it bears a "vintage" stamp, but it
sounds competitive with the project that used all the heavy-
hitter stuff.
NeilThis article is essentially anti-MSoft polemic, which is all fine by me. However,
I can confirm that people have brought me files ripped via iTunes that I
had to go through all kinds of gyrations to burn to a CD. Usually it was
someone at work who wanted a mix of songs they had purchased and songs they
had ripped. For me, obviously, any format that won't play on a Debian machine
is a non-starter, and iTunes ripped songs are nearly impossible to play on
a linux machine. I find any system that refuses to allow me to choose an
operating system onerous.
In addition, the article is (typically) wrong about what 'open source' advocates
want. There is a wide range of opinions about DRM and free software. Mostly
the free software community has no opinion about DRM unless it refuses to
allow an opt out option. A user should have the right to get information
and play back content on free systems. For example, it was impossible to
play DVDs on linux boxes for some time, which free software people found
unfair and in the end circumvented. I hope all of the DRM systems that prevent
me from choosing how and where I play my chosen music meet a similar fate.
There is a long running discussion about the GPL v. 3 and DRM that is raging
in free software land. Unsurprisingly Linus Torvalds and Richard Stallman
have been firing back and forth about it. There are a couple of very good
articles at Fortune magazine (of all things) about Linus and Richard. Far
better than this article.
TCB
"James McCloskey" <excelsm@hotmail.com> wrote:
>
> http://www.roughlydrafted.com/RD/Q4.06/6565C879-3C4C-4014-95 F0-EF1DF464D378.html
>
>
>What do you guys think?
>
>I've never had a problem burning CDs of music I've purchased with iTunes,
>or copying those CDs I've burned. I've never had problems converting files
>and dragging and dropping copies to my non apple mp3 player.
>
>James
>Sort of like "Dr Watson" was like on the PC
--
Martin Harrington
www.lendanear-sound.com
"Gene Lennon" <glennon@nospMyrealbox.com> wrote in message
news:4586e48b$1@linux...
>
> "Neil" <IUOIU@OIU.com> wrote:
>>
>>What exactly IS "journaling"? Not familiar with that term at all.
>>
>>Neil
>>
> From Apple:
>
> "When you enable journaling on a disk, a continuous record of changes to
> files on the disk is maintained in the journal. If your computer stops
> because
> of a power failure or some other issue, the journal is used to restore the
> disk to a known-good state when the server restarts.
>
> "With journaling turned on, the file system logs transactions as they
> occur.
> If the server fails in the middle of an operation, the file system can
> 'replay'
> the information in its log and complete the operation when the server
> restarts."
>
> Genesleeping with fists and i are leaving for redondo beach (and my crazy
sister in a couple of hour; since i won't be near a
computer...okay...my computer with my news reader i'll be off line.
so Merry Christmas to one and all who celebrate this time of the year
for any reason of their own.
ciaoAnother session full of clicking, popping, whistling and hooting from my UAD-1
plugins. Which makes me wonder - If I were to switch to a PC for my Paris
rig, what issues would I have to deal with? Would I have the new UAD-1 FX,
like the Neve EQs? How stable is Paris on a properly set up PC? Does having
a dual processor machine affect Paris performance?
And then, taking it to it's (logical?) extreme - Like switching to Nuendo
- would I be able to stick my Mac? Or would I do better w/ PC? Does Nuendo
have automatic latency compensation? Are you Nuendo folks happy w/ the sound?
Are any of you summing externally?
So many questions.
Thanks! And Happy Holidays to you all!!!
GanttMy Paris rig is pretty darn stable. AMD 2 gig single core Asus S-733 Mobo.
I'm running 3 Uad cards and 6 eds cards okn a Magma.
Never heard a click, whistle, or pop (except the pops that the eds cards
make at boot up)
Rod
"Gantt Kushner" <ganttmann@comcast.net> wrote:
>
>Another session full of clicking, popping, whistling and hooting from my
UAD-1
>plugins. Which makes me wonder - If I were to switch to a PC for my Paris
>rig, what issues would I have to deal with? Would I have the new UAD-1
FX,
>like the Neve EQs? How stable is Paris on a properly set up PC? Does having
>a dual processor machine affect Paris performance?
>
>And then, taking it to it's (logical?) extreme - Like switching to Nuendo
>- would I be able to stick my Mac? Or would I do better w/ PC? Does Nuendo
>have automatic latency compensation? Are you Nuendo folks happy w/ the
sound?
> Are any of you summing externally?
>
>So many questions.
>
>Thanks! And Happy Holidays to you all!!!
>
>GanttGannt, is it possible you have an underpowered power supply for
the task of running the Paris cards plus the UAD's (or that the
PSU may be going bad)? I mean, that's generally the most common
cause of popping & clicking, etc.
OR... is it only popping & clicking at the start of waveforms &
on edits/punches? I recall having a couple of Waves plugins that
did that in Paris - wonder if it might be the same kind of thing
going on? If so, I remedied that by recording a couple seconds
of empty space right before the original waveform actually
started, then automating the mute function so what would happen
is that the pulgin would trigger and pop before the waveform came
in in the silent (and muted) portion & then it would un-mute
right before the waveform actually started. I can't recall what
I did on the edits/punches, though - it may have been a
crossfade thing or I may have had to render/bounce those
portions as a single trackfile.
Just tossing out ideas here... sorry for the redundancy if
you've already checked these things.
Neil
"Gantt Kushner" <ganttmann@comcast.net> wrote:
>
>Another session full of clicking, popping, whistling and hooting from my
UAD-1
>plugins. Which makes me wonder - If I were to switch to a PC for my Paris
>rig, what issues would I have to deal with? Would I have the new UAD-1
FX,
>like the Neve EQs? How stable is Paris on a properly set up PC? Does having
>a dual processor machine affect Paris performance?
>
>And then, taking it to it's (logical?) extreme - Like switching to Nuendo
>- would I be able to stick my Mac? Or would I do better w/ PC? Does Nuendo
>have automatic latency compensation? Are you Nuendo folks happy w/ the
sound?
> Are any of you summing externally?
>
>So many questions.
>
>Thanks! And Happy Holidays to you all!!!
>
>GanttOh, BTW, having said all that - I think you're a perfect
candidate for "conversion", my son lol
Yes, Nuendo (and SX) have latency compensation, and as far as
your question about summing externally goes, I've tried this
combination recently (mixed in SX, summed in Paris via analog
outs - 4 stereo submixes into an 8-in module), and the results
don't suck. Not sure if it's exactly the sound I'm looking for,
but I did like certain aspects of it - tends to "warm up" the
mix a bit & kinda sounded a little smoother than a pure ITB
Native mix.
Neil
"Neil" <OIUOIU@OIU.com> wrote:
>
>Gannt, is it possible you have an underpowered power supply for
>the task of running the Paris cards plus the UAD's (or that the
>PSU may be going bad)? I mean, that's generally the most common
>cause of popping & clicking, etc.
>
>OR... is it only popping & clicking at the start of waveforms &
>on edits/punches? I recall having a couple of Waves plugins that
>did that in Paris - wonder if it might be the same kind of thing
>going on? If so, I remedied that by recording a couple seconds
>of empty space right before the original waveform actually
>started, then automating the mute function so what would happen
>is that the pulgin would trigger and pop before the waveform came
>in in the silent (and muted) portion & then it would un-mute
>right before the waveform actually started. I can't recall what
>I did on the edits/punches, though - it may have been a
>crossfade thing or I may have had to render/bounce those
>portions as a single trackfile.
>
>Just tossing out ideas here... sorry for the redundancy if
>you've already checked these things.
>
>Neil
>
>
>
>
>"Gantt Kushner" <ganttmann@comcast.net> wrote:
>>
>>Another session full of clicking, popping, whistling and hooting from my
>UAD-1
>>plugins. Which makes me wonder - If I were to switch to a PC for my Paris
>>rig, what issues would I have to deal with? Would I have the new UAD-1
>FX,
>>like the Neve EQs? How stable is Paris on a properly set up PC? Does having
>>a dual processor machine affect Paris performance?
>>
>>And then, taking it to it's (logical?) extreme - Like switching to Nuendo
>>- would I be able to stick my Mac? Or would I do better w/ PC? Does Nuendo
>>have automatic latency compensation? Are you Nuendo folks happy w/ the
>sound?
>> Are any of you summing externally?
>>
>>So many questions.
>>
>>Thanks! And Happy Holidays to you all!!!
>>
>>Gantt
>Hey Neil,
All I'm saying is: All DAW software have their own unique sound. Despite
what those lame summing test shows..
PT-HD has a very distinct sound. A very polished sound, with a nice top end,
but with full audio spectrum represented. Mixer/Summing buss can be pushed,
but you have to watch it.
Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype the top
nor bottom end.
Logic Audio: Very Broad- Aggressive sound, that really works for Rock and
R & B/Gospel mixes.
Digital Performer: With their hardware, superb audio quality. Full bodied
sound .
Sonar: Very flat sounding. I would say that Sonar is your most vanilla sound
DW on the market..
Samplitude : A little less top end than Pro Tools. Full bodied 3d sound..
Paris: Dark sounding in comparison to the the other DAWs. But, has a 3d sound
quality that's full bodied.
I feel that you asking SX to be something it's not with some analog summing.
Especialy for your genre of music..
"Neil" <IUOIU@OIU.com> wrote:
>
>"Lamont" <jjdpro@ameritech.net> wrote:
>>
>>"I'd disagree with you in this instance because I happen to think the Cubase
>>ones DO sound better."
>>
>>Then that SSL Engineer does not know what they are doing with board. There's
>>no way a mix coming off of that board SSL should sound better than a ITB
>>Cubase SX mix..
>>
>>Sorry, that just does not jive. That engineer does not know how to push
>he
>>SSL or just not familiar with it.
>
>You're not really paying attention, are you? It was the same
>engineer (me). And as far as whether or not I know how to use
>that particular board, I guess that would be a matter of
>your opinion. I don't think the SSL mixes are bad ones, I think
>they came out good; I just think that you can hear more detail
>in the ITB mixes in the examples I gave, and they have more
>wideband frequency content from top to bottom.
>
>Anyway, my point of that particular comparison wasn't to say
>"ITB mixes are better than using a large-format console that
>costs somewhere in the six-figure range", the point of it was to
>address a signal-chain suggestion that Paul had... he had
>suggested perhaps that I needed to pick up a few pieces of
>killer vintage gear, and I was just demonstrating that I think
>the various signal chain components that I have here are on par
>with most anything that can be found in heavy-hitter studios...
>we used probably around $100k's worth of mics & pre's on the
>PTHD/SSL mixes, plus obviously you're looking at another
>roughly $100k for that particular console (40-channel E-series,
>black EQ's, w/G-series Computer & Total Recall package), add in
>the PTHD, outboard gear & whatnot, and you end up with
>somewhere around a quarter-mil's worth of equipment involved in
>that project. The project done at my place was done with my
>gear, which certainly doesn't tally up to anywhere remotely
>close to that cost & none of it bears a "vintage" stamp, but it
>sounds competitive with the project that used all the heavy-
>hitter stuff.
>
>NeilI too have experienced the pops as a result of plugins...my solution to this
is to take a full song length track, copy it, apply a gain reduction of 90db
and rename it empty audio...then make a copy of it, put it on an open track
and cut and paste all the edits from the offending track onto this track
then move it back to the track where the edits came from...works like a
charm.
That being said I do believe we're probably looking at a hardware issue with
Gantt's set up...and I'm sorry having played with Nuendo and SX they suck
donkey balls...PT Rules.....nuff said!
:-) <- pointing out smiley
DOn
"Neil" <OIUOIU@OIU.com> wrote in message news:4588080a$1@linux...
>
> Oh, BTW, having said all that - I think you're a perfect
> candidate for "conversion", my son lol
> Yes, Nuendo (and SX) have latency compensation, and as far as
> your question about summing externally goes, I've tried this
> combination recently (mixed in SX, summed in Paris via analog
> outs - 4 stereo submixes into an 8-in module), and the results
> don't suck. Not sure if it's exactly the sound I'm looking for,
> but I did like certain aspects of it - tends to "warm up" the
> mix a bit & kinda sounded a little smoother than a pure ITB
> Native mix.
>
> Neil
>
>
>
> "Neil" <OIUOIU@OIU.com> wrote:
>>
>>Gannt, is it possible you have an underpowered power supply for
>>the task of running the Paris cards plus the UAD's (or that the
>>PSU may be going bad)? I mean, that's generally the most common
>>cause of popping & clicking, etc.
>>
>>OR... is it only popping & clicking at the start of waveforms &
>>on edits/punches? I recall having a couple of Waves plugins that
>>did that in Paris - wonder if it might be the same kind of thing
>>going on? If so, I remedied that by recording a couple seconds
>>of empty space right before the original waveform actually
>>started, then automating the mute function so what would happen
>>is that the pulgin would trigger and pop before the waveform came
>>in in the silent (and muted) portion & then it would un-mute
>>right before the waveform actually started. I can't recall what
>>I did on the edits/punches, though - it may have been a
>>crossfade thing or I may have had to render/bounce those
>>portions as a single trackfile.
>>
>>Just tossing out ideas here... sorry for the redundancy if
>>you've already checked these things.
>>
>>Neil
>>
>>
>>
>>
>>"Gantt Kushner" <ganttmann@comcast.net> wrote:
>>>
>>>Another session full of clicking, popping, whistling and hooting from my
>>UAD-1
>>>plugins. Which makes me wonder - If I were to switch to a PC for my
>>>Paris
>>>rig, what issues would I have to deal with? Would I have the new UAD-1
>>FX,
>>>like the Neve EQs? How stable is Paris on a properly set up PC? Does
>>>having
>>>a dual processor machine affect Paris performance?
>>>
>>>And then, taking it to it's (logical?) extreme - Like switching to Nuendo
>>>- would I be able to stick my Mac? Or would I do better w/ PC? Does
>>>Nuendo
>>>have automatic latency compensation? Are you Nuendo folks happy w/ the
>>sound?
>>> Are any of you summing externally?
>>>
>>>So many questions.
>>>
>>>Thanks! And Happy Holidays to you all!!!
>>>
>>>Gantt
>>
>Hey Rod,
Side question. I'm running the same mobo and I've been thinking about
upping the RAM -what brand are you using?
MR
"Rod Lincoln" <rlincoln@nospam.kc.rr.com> wrote in message
news:4587eb20$1@linux...
>
> My Paris rig is pretty darn stable. AMD 2 gig single core Asus S-733 Mobo.
> I'm running 3 Uad cards and 6 eds cards okn a Magma.
> Never heard a click, whistle, or pop (except the pops that the eds cards
> make at boot up)
> Rod
> "Gantt Kushner" <ganttmann@comcast.net> wrote:
> >
> >Another session full of clicking, popping, whistling and hooting from my
> UAD-1
> >plugins. Which makes me wonder - If I were to switch to a PC for my
Paris
> >rig, what issues would I have to deal with? Would I have the new UAD-1
> FX,
> >like the Neve EQs? How stable is Paris on a properly set up PC? Does
having
> >a dual processor machine affect Paris performance?
> >
> >And then, taking it to it's (logical?) extreme - Like switching to Nuendo
> >- would I be able to stick my Mac? Or would I do better w/ PC? Does
Nuendo
> >have automatic latency compensation? Are you Nuendo folks happy w/ the
> sound?
> > Are any of you summing externally?
> >
> >So many questions.
> >
> >Thanks! And Happy Holidays to you all!!!
> >
> >Gantt
>Gantt
I'd also suggest checking out the power supply.
But if you're considering a switch to a PC. I've been running Paris, like
Rod, on an Asus A7s333 for a few years now. I'm currently having a problem
with the Paris driver, but other than that, that rig has been stable.
Though, I don't have an UDA plugs, just Waves stuff, and freebie VST's.
Good luck
MR
"Gantt Kushner" <ganttmann@comcast.net> wrote in message
news:4587de86$1@linux...
>
> Another session full of clicking, popping, whistling and hooting from my
UAD-1
> plugins. Which makes me wonder - If I were to switch to a PC for my Paris
> rig, what issues would I have to deal with? Would I have the new UAD-1
FX,
> like the Neve EQs? How stable is Paris on a properly set up PC? Does
having
> a dual processor machine affect Paris performance?
>
> And then, taking it to it's (logical?) extreme - Like switching to Nuendo
> - would I be able to stick my Mac? Or would I do better w/ PC? Does
Nuendo
> have automatic latency compensation? Are you Nuendo folks happy w/ the
sound?
> Are any of you summing externally?
>
> So many questions.
>
> Thanks! And Happy Holidays to you all!!!
>
> GanttThat's interesting - all those DAW sonic interpretations, I
mean... I haven't had a chance to usee all of those, so it's
good information.
I still don't understand why you consider my summing
comparisons "lame", however - it was a fair set of tests;
the same mix summed in different ways. Not trying to prove a
point or to rig it so one sounded any better than the other - in
fact, if you recall the thread, different people liked different
summed versions for different reasons... there wasn't any one
version that stood out as being "the one" that everyone felt
sounded better. The only reason I didn't come right out & say
right away which version was which is so that I didn't bias
anyone's opinion beforehand by mentioning that... NOT to try
& "hide" anything or "trick" anyone, as you accused me of
Sheesh!
Neil
"Lamont" <jjdpro@ameritech.net> wrote:
>
>Hey Neil,
>
>All I'm saying is: All DAW software have their own unique sound. Despite
>what those lame summing test shows..
>
>PT-HD has a very distinct sound. A very polished sound, with a nice top
end,
>but with full audio spectrum represented. Mixer/Summing buss can be pushed,
>but you have to watch it.
>
>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype the top
>nor bottom end.
>
>Logic Audio: Very Broad- Aggressive sound, that really works for Rock and
>R & B/Gospel mixes.
>
>Digital Performer: With their hardware, superb audio quality. Full bodied
>sound .
>
>Sonar: Very flat sounding. I would say that Sonar is your most vanilla sound
>DW on the market..
>
>Samplitude : A little less top end than Pro Tools. Full bodied 3d sound..
>
>Paris: Dark sounding in comparison to the the other DAWs. But, has a 3d
sound
>quality that's full bodied.
>
>I feel that you asking SX to be something it's not with some analog summing.
>Especialy for your genre of music..
>
>
>
>
>"Neil" <IUOIU@OIU.com> wrote:
>>
>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>
>>>"I'd disagree with you in this instance because I happen to think the
Cubase
>>>ones DO sound better."
>>>
>>>Then that SSL Engineer does not know what they are doing with board. There's
>>>no way a mix coming off of that board SSL should sound better than a ITB
>>>Cubase SX mix..
>>>
>>>Sorry, that just does not jive. That engineer does not know how to push
>>he
>>>SSL or just not familiar with it.
>>
>>You're not really paying attention, are you? It was the same
>>engineer (me). And as far as whether or not I know how to use
>>that particular board, I guess that would be a matter of
>>your opinion. I don't think the SSL mixes are bad ones, I think
>>they came out good; I just think that you can hear more detail
>>in the ITB mixes in the examples I gave, and they have more
>>wideband frequency content from top to bottom.
>>
>>Anyway, my point of that particular comparison wasn't to say
>>"ITB mixes are better than using a large-format console that
>>costs somewhere in the six-figure range", the point of it was to
>>address a signal-chain suggestion that Paul had... he had
>>suggested perhaps that I needed to pick up a few pieces of
>>killer vintage gear, and I was just demonstrating that I think
>>the various signal chain components that I have here are on par
>>with most anything that can be found in heavy-hitter studios...
>>we used probably around $100k's worth of mics & pre's on the
>>PTHD/SSL mixes, plus obviously you're looking at another
>>roughly $100k for that particular console (40-channel E-series,
>>black EQ's, w/G-series Computer & Total Recall package), add in
>>the PTHD, outboard gear & whatnot, and you end up with
>>somewhere around a quarter-mil's worth of equipment involved in
>>that project. The project done at my place was done with my
>>gear, which certainly doesn't tally up to anywhere remotely
>>close to that cost & none of it bears a "vintage" stamp, but it
>>sounds competitive with the project that used all the heavy-
>>hitter stuff.
>>
>>Neil
>I have sent you round trip tickets to Tijuana. See you in a couple of days.
;oP
"rick" <parnell68@hotmail.com> wrote in message
news:pcafo25786dlglphvtv8b30fu5ub9ogbbd@4ax.com...
> sleeping with fists and i are leaving for redondo beach (and my crazy
> sister in a couple of hour; since i won't be near a
> computer...okay...my computer with my news reader i'll be off line.
> so Merry Christmas to one and all who celebrate this time of the year
> for any reason of their own.
>
> ciaoHi Gantt,
You really should find a way to test the Paris and UAD cards you have in
another MAC before you go crazy with upgrades.
Gantt Kushner wrote:
> Another session full of clicking, popping, whistling and hooting from my UAD-1
> plugins. Which makes me wonder - If I were to switch to a PC for my Paris
> rig, what issues would I have to deal with?
Learning a completely different GUI. A separate set of issues than the
MAC has.
Would I have the new UAD-1 FX,
> like the Neve EQs?
Don't know if the new ones will work in Paris on the PC side or not.
They do work in all the other DAWs out there.
How stable is Paris on a properly set up PC?
Rock Solid if it is on older PC hardware from 4 to 6 years ago using
Windows 98SE. It will work pretty good with some tweaking on newer
hardware on XP.
Does having
> a dual processor machine affect Paris performance?
No. The current faster processors will allow for Paris to run more
native plug ins even if Paris is not using both CPUs.
>
> And then, taking it to it's (logical?) extreme - Like switching to Nuendo
> - would I be able to stick my Mac?
For MAC you would new a pre-Intel G5 and a high quality sound interface
if you want to use Nuendo. You can use a new Intel based G5 with Cubase
4, Logic and I believe DP also. You would have to sell you UAD cards and
get the PCI-e versions or get a PCI to PCI-e expansions chassis for the
Intel G5. It does not have any PCI slots. You will also need to check to
see if you use any other native plug-ins with Paris to make sure they
have Intel Mac support.
Or would I do better w/ PC?
In order for Apple to progress they had to go to PC hardware platform.
To me the main differences are you have a different OS GUI and on the PC
side you have better choices with Motherboard and Case configurations
which allow for more internal drive and hardware configurations.
otherwise there is very little difference.
Does Nuendo
> have automatic latency compensation?
They were the first and still do it he best.
Are you Nuendo folks happy w/ the sound?
The "sound" is effected more by the audio hardware you have and by how
you mix in the program. You have to mix very differently in native
systems than you do in Paris.
> Are any of you summing externally?
Allot on people mix on the console like people do with Pro Tools. They
are comfortable with it that way and the analog/digital mixer adds it
sound to the mix in a way that they like.
Cubase/Nuendo offers allot of flexibility in running ITB only or
integrating with a analog outboard system.
>
> So many questions.
>
> Thanks! And Happy Holidays to you all!!!
>
> Gantt
If you only really need what Paris offers along with the UAD then I
would just conform the functionality of the Paris and UAD hardware. If
they work fine then check into picking up a used G4 and continue with
the way you are working now. Anything else will potentially be a large
investment of time and money whether it be PC or MAC.
--
Chris Ludwig
ADK Pro Audio
(859) 635-5762
www.adkproaudio.com
chrisl@adkproaudio.comHey Neil,
I was not reffering to your test as Lame, but rather that popular summing
test cd that was sold, that really proved nothing (imho) and BrianT as well.
Due to that fact fact that all softwafre behave differently under different
circumstances..
"Neil" <IUOIU@OIU.com> wrote:
>
>That's interesting - all those DAW sonic interpretations, I
>mean... I haven't had a chance to usee all of those, so it's
>good information.
>
>I still don't understand why you consider my summing
>comparisons "lame", however - it was a fair set of tests;
>the same mix summed in different ways. Not trying to prove a
>point or to rig it so one sounded any better than the other - in
>fact, if you recall the thread, different people liked different
>summed versions for different reasons... there wasn't any one
>version that stood out as being "the one" that everyone felt
>sounded better. The only reason I didn't come right out & say
>right away which version was which is so that I didn't bias
>anyone's opinion beforehand by mentioning that... NOT to try
>& "hide" anything or "trick" anyone, as you accused me of
>
>Sheesh!
>
>Neil
>
>
>"Lamont" <jjdpro@ameritech.net> wrote:
>>
>>Hey Neil,
>>
>>All I'm saying is: All DAW software have their own unique sound. Despite
>>what those lame summing test shows..
>>
>>PT-HD has a very distinct sound. A very polished sound, with a nice top
>end,
>>but with full audio spectrum represented. Mixer/Summing buss can be pushed,
>>but you have to watch it.
>>
>>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype the
top
>>nor bottom end.
>>
>>Logic Audio: Very Broad- Aggressive sound, that really works for Rock and
>>R & B/Gospel mixes.
>>
>>Digital Performer: With their hardware, superb audio quality. Full bodied
>>sound .
>>
>>Sonar: Very flat sounding. I would say that Sonar is your most vanilla
sound
>>DW on the market..
>>
>>Samplitude : A little less top end than Pro Tools. Full bodied 3d sound..
>>
>>Paris: Dark sounding in comparison to the the other DAWs. But, has a 3d
>sound
>>quality that's full bodied.
>>
>>I feel that you asking SX to be something it's not with some analog summing.
>>Especialy for your genre of music..
>>
>>
>>
>>
>>"Neil" <IUOIU@OIU.com> wrote:
>>>
>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>
>>>>"I'd disagree with you in this instance because I happen to think the
>Cubase
>>>>ones DO sound better."
>>>>
>>>>Then that SSL Engineer does not know what they are doing with board.
There's
>>>>no way a mix coming off of that board SSL should sound better than a
ITB
>>>>Cubase SX mix..
>>>>
>>>>Sorry, that just does not jive. That engineer does not know how to push
>>>he
>>>>SSL or just not familiar with it.
>>>
>>>You're not really paying attention, are you? It was the same
>>>engineer (me). And as far as whether or not I know how to use
>>>that particular board, I guess that would be a matter of
>>>your opinion. I don't think the SSL mixes are bad ones, I think
>>>they came out good; I just think that you can hear more detail
>>>in the ITB mixes in the examples I gave, and they have more
>>>wideband frequency content from top to bottom.
>>>
>>>Anyway, my point of that particular comparison wasn't to say
>>>"ITB mixes are better than using a large-format console that
>>>costs somewhere in the six-figure range", the point of it was to
>>>address a signal-chain suggestion that Paul had... he had
>>>suggested perhaps that I needed to pick up a few pieces of
>>>killer vintage gear, and I was just demonstrating that I think
>>>the various signal chain components that I have here are on par
>>>with most anything that can be found in heavy-hitter studios...
>>>we used probably around $100k's worth of mics & pre's on the
>>>PTHD/SSL mixes, plus obviously you're looking at another
>>>roughly $100k for that particular console (40-channel E-series,
>>>black EQ's, w/G-series Computer & Total Recall package), add in
>>>the PTHD, outboard gear & whatnot, and you end up with
>>>somewhere around a quarter-mil's worth of equipment involved in
>>>that project. The project done at my place was done with my
>>>gear, which certainly doesn't tally up to anywhere remotely
>>>close to that cost & none of it bears a "vintage" stamp, but it
>>>sounds competitive with the project that used all the heavy-
>>>hitter stuff.
>>>
>>>Neil
>>
>Samsung is the recommended, but I'm using Crucial w/o any trouble on two of
these Sis Asus mobos.
www.crucial.com
AA
"Mike R." <emarenot@yahoo.com> wrote in message news:45881dcc@linux...
> Hey Rod,
> Side question. I'm running the same mobo and I've been thinking about
> upping the RAM -what brand are you using?
> MR
> "Rod Lincoln" <rlincoln@nospam.kc.rr.com> wrote in message
> news:4587eb20$1@linux...
>>
>> My Paris rig is pretty darn stable. AMD 2 gig single core Asus S-733
>> Mobo.
>> I'm running 3 Uad cards and 6 eds cards okn a Magma.
>> Never heard a click, whistle, or pop (except the pops that the eds cards
>> make at boot up)
>> Rod
>> "Gantt Kushner" <ganttmann@comcast.net> wrote:
>> >
>> >Another session full of clicking, popping, whistling and hooting from my
>> UAD-1
>> >plugins. Which makes me wonder - If I were to switch to a PC for my
> Paris
>> >rig, what issues would I have to deal with? Would I have the new UAD-1
>> FX,
>> >like the Neve EQs? How stable is Paris on a properly set up PC? Does
> having
>> >a dual processor machine affect Paris performance?
>> >
>> >And then, taking it to it's (logical?) extreme - Like switching to
>> >Nuendo
>> >- would I be able to stick my Mac? Or would I do better w/ PC? Does
> Nuendo
>> >have automatic latency compensation? Are you Nuendo folks happy w/ the
>> sound?
>> > Are any of you summing externally?
>> >
>> >So many questions.
>> >
>> >Thanks! And Happy Holidays to you all!!!
>> >
>> >Gantt
>>
>
>Merry Christmas to you too Rick! You can always access the web page version
of the paris news group at: http://webnews.parisnewsgroup.com/
James
rick <parnell68@hotmail.com> wrote:
>sleeping with fists and i are leaving for redondo beach (and my crazy
>sister in a couple of hour; since i won't be near a
>computer...okay...my computer with my news reader i'll be off line.
>so Merry Christmas to one and all who celebrate this time of the year
>for any reason of their own.
>
>ciaoI usually render tracks that have lots of edits before mixing. I've found
the UAD-1 FX to be pretty glitchy around edits. I also always create a second
or two of silence before and after every track that has plugins inserted.
In fact, I discovered that the Lexicon PCM 42 delay plugin needs a full
delays worth of track after the event to function properly.
Anyway, even w/ rendered tracks I'm having problems...
Gantt
"Don Nafe" <dnafe@magma.ca> wrote:
>I too have experienced the pops as a result of plugins...my solution to
this
>is to take a full song length track, copy it, apply a gain reduction of
90db
>and rename it empty audio...then make a copy of it, put it on an open track
>and cut and paste all the edits from the offending track onto this track
>then move it back to the track where the edits came from...works like a
>charm.
>
>That being said I do believe we're probably looking at a hardware issue
with
>Gantt's set up...and I'm sorry having played with Nuendo and SX they suck
>donkey balls...PT Rules.....nuff said!
>
>:-) <- pointing out smiley
>
>DOn
>
>
>"Neil" <OIUOIU@OIU.com> wrote in message news:4588080a$1@linux...
>>
>> Oh, BTW, having said all that - I think you're a perfect
>> candidate for "conversion", my son lol
>> Yes, Nuendo (and SX) have latency compensation, and as far as
>> your question about summing externally goes, I've tried this
>> combination recently (mixed in SX, summed in Paris via analog
>> outs - 4 stereo submixes into an 8-in module), and the results
>> don't suck. Not sure if it's exactly the sound I'm looking for,
>> but I did like certain aspects of it - tends to "warm up" the
>> mix a bit & kinda sounded a little smoother than a pure ITB
>> Native mix.
>>
>> Neil
>>
>>
>>
>> "Neil" <OIUOIU@OIU.com> wrote:
>>>
>>>Gannt, is it possible you have an underpowered power supply for
>>>the task of running the Paris cards plus the UAD's (or that the
>>>PSU may be going bad)? I mean, that's generally the most common
>>>cause of popping & clicking, etc.
>>>
>>>OR... is it only popping & clicking at the start of waveforms &
>>>on edits/punches? I recall having a couple of Waves plugins that
>>>did that in Paris - wonder if it might be the same kind of thing
>>>going on? If so, I remedied that by recording a couple seconds
>>>of empty space right before the original waveform actually
>>>started, then automating the mute function so what would happen
>>>is that the pulgin would trigger and pop before the waveform came
>>>in in the silent (and muted) portion & then it would un-mute
>>>right before the waveform actually started. I can't recall what
>>>I did on the edits/punches, though - it may have been a
>>>crossfade thing or I may have had to render/bounce those
>>>portions as a single trackfile.
>>>
>>>Just tossing out ideas here... sorry for the redundancy if
>>>you've already checked these things.
>>>
>>>Neil
>>>
>>>
>>>
>>>
>>>"Gantt Kushner" <ganttmann@comcast.net> wrote:
>>>>
>>>>Another session full of clicking, popping, whistling and hooting from
my
>>>UAD-1
>>>>plugins. Which makes me wonder - If I were to switch to a PC for my
>>>>Paris
>>>>rig, what issues would I have to deal with? Would I have the new UAD-1
>>>FX,
>>>>like the Neve EQs? How stable is Paris on a properly set up PC? Does
>>>>having
>>>>a dual processor machine affect Paris performance?
>>>>
>>>>And then, taking it to it's (logical?) extreme - Like switching to Nuendo
>>>>- would I be able to stick my Mac? Or would I do better w/ PC? Does
>>>>Nuendo
>>>>have automatic latency compensation? Are you Nuendo folks happy w/ the
>>>sound?
>>>> Are any of you summing externally?
>>>>
>>>>So many questions.
>>>>
>>>>Thanks! And Happy Holidays to you all!!!
>>>>
>>>>Gantt
>>>
>>
>
>Both of my UAD-1 cards live in my G4. The EDS cards all live in my Magma
13 slot expansion chassis. Is it possible that my G4 doesn't have enough
juice to run 2 UAD-1 cards?
Gantt
"Neil" <OIUOIU@OIU.com> wrote:
>
>Gannt, is it possible you have an underpowered power supply for
>the task of running the Paris cards plus the UAD's (or that the
>PSU may be going bad)? I mean, that's generally the most common
>cause of popping & clicking, etc.
>
>OR... is it only popping & clicking at the start of waveforms &
>on edits/punches? I recall having a couple of Waves plugins that
>did that in Paris - wonder if it might be the same kind of thing
>going on? If so, I remedied that by recording a couple seconds
>of empty space right before the original waveform actually
>started, then automating the mute function so what would happen
>is that the pulgin would trigger and pop before the waveform came
>in in the silent (and muted) portion & then it would un-mute
>right before the waveform actually started. I can't recall what
>I did on the edits/punches, though - it may have been a
>crossfade thing or I may have had to render/bounce those
>portions as a single trackfile.
>
>Just tossing out ideas here... sorry for the redundancy if
>you've already checked these things.
>
>Neil
>
>
>
>
>"Gantt Kushner" <ganttmann@comcast.net> wrote:
>>
>>Another session full of clicking, popping, whistling and hooting from my
>UAD-1
>>plugins. Which makes me wonder - If I were to switch to a PC for my Paris
>>rig, what issues would I have to deal with? Would I have the new UAD-1
>FX,
>>like the Neve EQs? How stable is Paris on a properly set up PC? Does having
>>a dual processor machine affect Paris performance?
>>
>>And then, taking it to it's (logical?) extreme - Like switching to Nuendo
>>- would I be able to stick my Mac? Or would I do better w/ PC? Does Nuendo
>>have automatic latency compensation? Are you Nuendo folks happy w/ the
>sound?
>> Are any of you summing externally?
>>
>>So many questions.
>>
>>Thanks! And Happy Holidays to you all!!!
>>
>>Gantt
>Either Samsung or Crucial...I don't remember which. I have 1 gig
Rod
"Mike R." <emarenot@yahoo.com> wrote:
>Hey Rod,
>Side question. I'm running the same mobo and I've been thinking about
>upping the RAM -what brand are you using?
>MR
>"Rod Lincoln" <rlincoln@nospam.kc.rr.com> wrote in message
>news:4587eb20$1@linux...
>>
>> My Paris rig is pretty darn stable. AMD 2 gig single core Asus S-733 Mobo.
>> I'm running 3 Uad cards and 6 eds cards okn a Magma.
>> Never heard a click, whistle, or pop (except the pops that the eds cards
>> make at boot up)
>> Rod
>> "Gantt Kushner" <ganttmann@comcast.net> wrote:
>> >
>> >Another session full of clicking, popping, whistling and hooting from
my
>> UAD-1
>> >plugins. Which makes me wonder - If I were to switch to a PC for my
>Paris
>> >rig, what issues would I have to deal with? Would I have the new UAD-1
>> FX,
>> >like the Neve EQs? How stable is Paris on a properly set up PC? Does
>having
>> >a dual processor machine affect Paris performance?
>> >
>> >And then, taking it to it's (logical?) extreme - Like switching to Nuendo
>> >- would I be able to stick my Mac? Or would I do better w/ PC? Does
>Nuendo
>> >have automatic latency compensation? Are you Nuendo folks happy w/ the
>> sound?
>> > Are any of you summing externally?
>> >
>> >So many questions.
>> >
>> >Thanks! And Happy Holidays to you all!!!
>> >
>> >Gantt
>>
>
>"Gantt Kushner" <ganttmann@comcast.net> wrote in message
news:45884918$1@linux...
>
> I usually render tracks that have lots of edits before mixing. I've found
> the UAD-1 FX to be pretty glitchy around edits. I also always create a
> second
> or two of silence before and after every track that has plugins inserted.
> In fact, I discovered that the Lexicon PCM 42 delay plugin needs a full
> delays worth of track after the event to function properly.
>
> Anyway, even w/ rendered tracks I'm having problems...
>
> Gantt
That's the final step after all is said and done because I've been know to
change things right up until the mix is approverd for mastering
Don
>
> "Don Nafe" <dnafe@magma.ca> wrote:
>>I too have experienced the pops as a result of plugins...my solution to
> this
>>is to take a full song length track, copy it, apply a gain reduction of
> 90db
>>and rename it empty audio...then make a copy of it, put it on an open
>>track
>
>>and cut and paste all the edits from the offending track onto this track
>
>>then move it back to the track where the edits came from...works like a
>
>>charm.
>>
>>That being said I do believe we're probably looking at a hardware issue
> with
>>Gantt's set up...and I'm sorry having played with Nuendo and SX they suck
>
>>donkey balls...PT Rules.....nuff said!
>>
>>:-) <- pointing out smiley
>>
>>DOn
>>
>>
>>"Neil" <OIUOIU@OIU.com> wrote in message news:4588080a$1@linux...
>>>
>>> Oh, BTW, having said all that - I think you're a perfect
>>> candidate for "conversion", my son lol
>>> Yes, Nuendo (and SX) have latency compensation, and as far as
>>> your question about summing externally goes, I've tried this
>>> combination recently (mixed in SX, summed in Paris via analog
>>> outs - 4 stereo submixes into an 8-in module), and the results
>>> don't suck. Not sure if it's exactly the sound I'm looking for,
>>> but I did like certain aspects of it - tends to "warm up" the
>>> mix a bit & kinda sounded a little smoother than a pure ITB
>>> Native mix.
>>>
>>> Neil
>>>
>>>
>>>
>>> "Neil" <OIUOIU@OIU.com> wrote:
>>>>
>>>>Gannt, is it possible you have an underpowered power supply for
>>>>the task of running the Paris cards plus the UAD's (or that the
>>>>PSU may be going bad)? I mean, that's generally the most common
>>>>cause of popping & clicking, etc.
>>>>
>>>>OR... is it only popping & clicking at the start of waveforms &
>>>>on edits/punches? I recall having a couple of Waves plugins that
>>>>did that in Paris - wonder if it might be the same kind of thing
>>>>going on? If so, I remedied that by recording a couple seconds
>>>>of empty space right before the original waveform actually
>>>>started, then automating the mute function so what would happen
>>>>is that the pulgin would trigger and pop before the waveform came
>>>>in in the silent (and muted) portion & then it would un-mute
>>>>right before the waveform actually started. I can't recall what
>>>>I did on the edits/punches, though - it may have been a
>>>>crossfade thing or I may have had to render/bounce those
>>>>portions as a single trackfile.
>>>>
>>>>Just tossing out ideas here... sorry for the redundancy if
>>>>you've already checked these things.
>>>>
>>>>Neil
>>>>
>>>>
>>>>
>>>>
>>>>"Gantt Kushner" <ganttmann@comcast.net> wrote:
>>>>>
>>>>>Another session full of clicking, popping, whistling and hooting from
> my
>>>>UAD-1
>>>>>plugins. Which makes me wonder - If I were to switch to a PC for my
>
>>>>>Paris
>>>>>rig, what issues would I have to deal with? Would I have the new UAD-1
>>>>FX,
>>>>>like the Neve EQs? How stable is Paris on a properly set up PC? Does
>
>>>>>having
>>>>>a dual processor machine affect Paris performance?
>>>>>
>>>>>And then, taking it to it's (logical?) extreme - Like switching to
>>>>>Nuendo
>>>>>- would I be able to stick my Mac? Or would I do better w/ PC? Does
>
>>>>>Nuendo
>>>>>have automatic latency compensation? Are you Nuendo folks happy w/ the
>>>>sound?
>>>>> Are any of you summing externally?
>>>>>
>>>>>So many questions.
>>>>>
>>>>>Thanks! And Happy Holidays to you all!!!
>>>>>
>>>>>Gantt
>>>>
>>>
>>
>>
>I'd like to see this proven in a controlled ABY test.
"Lamont" <jjdpro@ameritech.net> wrote:
>
>Hey Neil,
>
>All I'm saying is: All DAW software have their own unique sound. Despite
>what those lame summing test shows..
>
>PT-HD has a very distinct sound. A very polished sound, with a nice top
end,
>but with full audio spectrum represented. Mixer/Summing buss can be pushed,
>but you have to watch it.
>
>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype the top
>nor bottom end.
>
>Logic Audio: Very Broad- Aggressive sound, that really works for Rock and
>R & B/Gospel mixes.
>
>Digital Performer: With their hardware, superb audio quality. Full bodied
>sound .
>
>Sonar: Very flat sounding. I would say that Sonar is your most vanilla sound
>DW on the market..
>
>Samplitude : A little less top end than Pro Tools. Full bodied 3d sound..
>
>Paris: Dark sounding in comparison to the the other DAWs. But, has a 3d
sound
>quality that's full bodied.
>
>I feel that you asking SX to be something it's not with some analog summing.
>Especialy for your genre of music..
>
>
>
>
>"Neil" <IUOIU@OIU.com> wrote:
>>
>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>
>>>"I'd disagree with you in this instance because I happen to think the
Cubase
>>>ones DO sound better."
>>>
>>>Then that SSL Engineer does not know what they are doing with board. There's
>>>no way a mix coming off of that board SSL should sound better than a ITB
>>>Cubase SX mix..
>>>
>>>Sorry, that just does not jive. That engineer does not know how to push
>>he
>>>SSL or just not familiar with it.
>>
>>You're not really paying attention, are you? It was the same
>>engineer (me). And as far as whether or not I know how to use
>>that particular board, I guess that would be a matter of
>>your opinion. I don't think the SSL mixes are bad ones, I think
>>they came out good; I just think that you can hear more detail
>>in the ITB mixes in the examples I gave, and they have more
>>wideband frequency content from top to bottom.
>>
>>Anyway, my point of that particular comparison wasn't to say
>>"ITB mixes are better than using a large-format console that
>>costs somewhere in the six-figure range", the point of it was to
>>address a signal-chain suggestion that Paul had... he had
>>suggested perhaps that I needed to pick up a few pieces of
>>killer vintage gear, and I was just demonstrating that I think
>>the various signal chain components that I have here are on par
>>with most anything that can be found in heavy-hitter studios...
>>we used probably around $100k's worth of mics & pre's on the
>>PTHD/SSL mixes, plus obviously you're looking at another
>>roughly $100k for that particular console (40-channel E-series,
>>black EQ's, w/G-series Computer & Total Recall package), add in
>>the PTHD, outboard gear & whatnot, and you end up with
>>somewhere around a quarter-mil's worth of equipment involved in
>>that project. The project done at my place was done with my
>>gear, which certainly doesn't tally up to anywhere remotely
>>close to that cost & none of it bears a "vintage" stamp, but it
>>sounds competitive with the project that used all the heavy-
>>hitter stuff.
>>
>>Neil
>Hi Gantt....running XP...I don't get clicks or pops with UAD,just weirdness
at crossfade edits....and LATENCY...which I hate dealing with...my one card
rig has been working OK for 6 years,3 years on XP...I am getting Nuendo,just
to see what it's like on PC....zan
"Gantt Kushner" <ganttmann@comcast.net> wrote in message
news:4587de86$1@linux...
>
> Another session full of clicking, popping, whistling and hooting from my
UAD-1
> plugins. Which makes me wonder - If I were to switch to a PC for my Paris
> rig, what issues would I have to deal with? Would I have the new UAD-1
FX,
> like the Neve EQs? How stable is Paris on a properly set up PC? Does
having
> a dual processor machine affect Paris performance?
>
> And then, taking it to it's (logical?) extreme - Like switching to Nuendo
> - would I be able to stick my Mac? Or would I do better w/ PC? Does
Nuendo
> have automatic latency compensation? Are you Nuendo folks happy w/ the
sound?
> Are any of you summing externally?
>
> So many questions.
>
> Thanks! And Happy Holidays to you all!!!
>
> GanttThe only real test is with the ears and not scopes and graphs.
"TCB" <nobody@ishere.com> wrote:
>
>I'd like to see this proven in a controlled ABY test.
>
>"Lamont" <jjdpro@ameritech.net> wrote:
>>
>>Hey Neil,
>>
>>All I'm saying is: All DAW software have their own unique sound. Despite
>>what those lame summing test shows..
>>
>>PT-HD has a very distinct sound. A very polished sound, with a nice top
>end,
>>but with full audio spectrum represented. Mixer/Summing buss can be pushed,
>>but you have to watch it.
>>
>>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype the
top
>>nor bottom end.
>>
>>Logic Audio: Very Broad- Aggressive sound, that really works for Rock and
>>R & B/Gospel mixes.
>>
>>Digital Performer: With their hardware, superb audio quality. Full bodied
>>sound .
>>
>>Sonar: Very flat sounding. I would say that Sonar is your most vanilla
sound
>>DW on the market..
>>
>>Samplitude : A little less top end than Pro Tools. Full bodied 3d sound..
>>
>>Paris: Dark sounding in comparison to the the other DAWs. But, has a 3d
>sound
>>quality that's full bodied.
>>
>>I feel that you asking SX to be something it's not with some analog summing.
>>Especialy for your genre of music..
>>
>>
>>
>>
>>"Neil" <IUOIU@OIU.com> wrote:
>>>
>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>
>>>>"I'd disagree with you in this instance because I happen to think the
>Cubase
>>>>ones DO sound better."
>>>>
>>>>Then that SSL Engineer does not know what they are doing with board.
There's
>>>>no way a mix coming off of that board SSL should sound better than a
ITB
>>>>Cubase SX mix..
>>>>
>>>>Sorry, that just does not jive. That engineer does not know how to push
>>>he
>>>>SSL or just not familiar with it.
>>>
>>>You're not really paying attention, are you? It was the same
>>>engineer (me). And as far as whether or not I know how to use
>>>that particular board, I guess that would be a matter of
>>>your opinion. I don't think the SSL mixes are bad ones, I think
>>>they came out good; I just think that you can hear more detail
>>>in the ITB mixes in the examples I gave, and they have more
>>>wideband frequency content from top to bottom.
>>>
>>>Anyway, my point of that particular comparison wasn't to say
>>>"ITB mixes are better than using a large-format console that
>>>costs somewhere in the six-figure range", the point of it was to
>>>address a signal-chain suggestion that Paul had... he had
>>>suggested perhaps that I needed to pick up a few pieces of
>>>killer vintage gear, and I was just demonstrating that I think
>>>the various signal chain components that I have here are on par
>>>with most anything that can be found in heavy-hitter studios...
>>>we used probably around $100k's worth of mics & pre's on the
>>>PTHD/SSL mixes, plus obviously you're looking at another
>>>roughly $100k for that particular console (40-channel E-series,
>>>black EQ's, w/G-series Computer & Total Recall package), add in
>>>the PTHD, outboard gear & whatnot, and you end up with
>>>somewhere around a quarter-mil's worth of equipment involved in
>>>that project. The project done at my place was done with my
>>>gear, which certainly doesn't tally up to anywhere remotely
>>>close to that cost & none of it bears a "vintage" stamp, but it
>>>sounds competitive with the project that used all the heavy-
>>>hitter stuff.
>>>
>>>Neil
>>
>Sony BMG Settles Suit Over CDs
http://apnews.excite.com/article/20061219/D8M44CE00.htmlWhich is why ABY testing uses expert listeners instead of scopes and graphs.
I'm not saying you're wrong, esp. about ITB vs external summing. One would
expect that to sound at least slightly different. But I would be absolutely
shocked if anyone could tell in a controlled ABY test whether they were listening
to SX, Performer, of Sonar.
"LaMont" <jjdpro@ameritech.net> wrote:
>
>The only real test is with the ears and not scopes and graphs.
>
>"TCB" <nobody@ishere.com> wrote:
>>
>>I'd like to see this proven in a controlled ABY test.
>>
>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>
>>>Hey Neil,
>>>
>>>All I'm saying is: All DAW software have their own unique sound. Despite
>>>what those lame summing test shows..
>>>
>>>PT-HD has a very distinct sound. A very polished sound, with a nice top
>>end,
>>>but with full audio spectrum represented. Mixer/Summing buss can be pushed,
>>>but you have to watch it.
>>>
>>>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype the
>top
>>>nor bottom end.
>>>
>>>Logic Audio: Very Broad- Aggressive sound, that really works for Rock
and
>>>R & B/Gospel mixes.
>>>
>>>Digital Performer: With their hardware, superb audio quality. Full bodied
>>>sound .
>>>
>>>Sonar: Very flat sounding. I would say that Sonar is your most vanilla
>sound
>>>DW on the market..
>>>
>>>Samplitude : A little less top end than Pro Tools. Full bodied 3d sound..
>>>
>>>Paris: Dark sounding in comparison to the the other DAWs. But, has a 3d
>>sound
>>>quality that's full bodied.
>>>
>>>I feel that you asking SX to be something it's not with some analog summing.
>>>Especialy for your genre of music..
>>>
>>>
>>>
>>>
>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>
>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>
>>>>>"I'd disagree with you in this instance because I happen to think the
>>Cubase
>>>>>ones DO sound better."
>>>>>
>>>>>Then that SSL Engineer does not know what they are doing with board.
>There's
>>>>>no way a mix coming off of that board SSL should sound better than a
>ITB
>>>>>Cubase SX mix..
>>>>>
>>>>>Sorry, that just does not jive. That engineer does not know how to push
>>>>he
>>>>>SSL or just not familiar with it.
>>>>
>>>>You're not really paying attention, are you? It was the same
>>>>engineer (me). And as far as whether or not I know how to use
>>>>that particular board, I guess that would be a matter of
>>>>your opinion. I don't think the SSL mixes are bad ones, I think
>>>>they came out good; I just think that you can hear more detail
>>>>in the ITB mixes in the examples I gave, and they have more
>>>>wideband frequency content from top to bottom.
>>>>
>>>>Anyway, my point of that particular comparison wasn't to say
>>>>"ITB mixes are better than using a large-format console that
>>>>costs somewhere in the six-figure range", the point of it was to
>>>>address a signal-chain suggestion that Paul had... he had
>>>>suggested perhaps that I needed to pick up a few pieces of
>>>>killer vintage gear, and I was just demonstrating that I think
>>>>the various signal chain components that I have here are on par
>>>>with most anything that can be found in heavy-hitter studios...
>>>>we used probably around $100k's worth of mics & pre's on the
>>>>PTHD/SSL mixes, plus obviously you're looking at another
>>>>roughly $100k for that particular console (40-channel E-series,
>>>>black EQ's, w/G-series Computer & Total Recall package), add in
>>>>the PTHD, outboard gear & whatnot, and you end up with
>>>>somewhere around a quarter-mil's worth of equipment involved in
>>>>that project. The project done at my place was done with my
>>>>gear, which certainly doesn't tally up to anywhere remotely
>>>>close to that cost & none of it bears a "vintage" stamp, but it
>>>>sounds competitive with the project that used all the heavy-
>>>>hitter stuff.
>>>>
>>>>Neil
>>>
>>
>"Companies that want to load their CDs with software that limits the ability
to copy music should fully inform consumers about it, not hide it, and make
sure it doesn't inflict security vulnerabilities on computers," California
Attorney General Bill Lockyer said in a statement.
Bizarrely sane and well said for a politician.
TCB
"James McCloskey" <excelsm@hotmail.com> wrote:
>
>Sony BMG Settles Suit Over CDs
>
>http://apnews.excite.com/article/20061219/D8M44CE00.htmlIf only XP had a journaling fs. In the free software world, there's reiserfs
that is designed ground up for journaling and ext3 can be set up for journaling.
I've always wanted to mess around with rfs and the file system that SGI released
under the GPL (xfs) a while back, but when I'm installing I just say, 'Oh,
whatever, I'm in a rush and I'll just use ext3.'
Also, *BSD and Solaris both have very powerful file systems that have all
kinds of stuff built in but are a massive PITA to manage. Ironically enough,
Exchange 2003 essentially uses a 'journal' like method for maintaining mailboxes
and logs, after a crash they can be played back if they are on the disk,
but by default installs on (cannot be journaled) NTFS drives. At least they're
getting halrway there.
TCB
"Martin Harrington" <lendan@bigpond.net.au> wrote:
>Sort of like "Dr Watson" was like on the PC
>--
>Martin Harrington
>www.lendanear-sound.com
>
>"Gene Lennon" <glennon@nospMyrealbox.com> wrote in message
>news:4586e48b$1@linux...
>>
>> "Neil" <IUOIU@OIU.com> wrote:
>>>
>>>What exactly IS "journaling"? Not familiar with that term at all.
>>>
>>>Neil
>>>
>> From Apple:
>>
>> "When you enable journaling on a disk, a continuous record of changes
to
>> files on the disk is maintained in the journal. If your computer stops
>> because
>> of a power failure or some other issue, the journal is used to restore
the
>> disk to a known-good state when the server restarts.
>>
>> "With journaling turned on, the file system logs transactions as they
>> occur.
>> If the server fails in the middle of an operation, the file system can
>> 'replay'
>> the information in its log and complete the operation when the server
>> restarts."
>>
>> Gene
>
>Thad, your gonna love this one. Sorry I couldn't resist.
http://www.chron.com/disp/story.mpl/chronicle/4411680.html
Actually, I'm not sure I believe the assumption of this article.
JamesThad, you really can't hear the difference?? Maybe I own too many software
DAWS thru the years.
Starting on Logic Audio 3.0, then to Cakwalk, Pro Tools, DP, Paris,Acid,
Neundo, Sonar, samplitude..
I can hear the diference with the same audio interface with the same wav
file(s)as oon as I import the file or files.
These days, depending on the genre I'm mising determines which DAW software
I'll use.
My circle of engineer and producer buddies all can hear the difference in
a second. Just the other day, we were mixing this R&B(ish)Gospel track and
somebody said, 'Mont, this is begging for Paris. Another track, the call
was for Pro Tools. And another,Nuendo..
I know BrianT feels and hears the same way in different DAW software. It's
really obvious..
"TCB" <nobody@ishere.com> wrote:
>
>Which is why ABY testing uses expert listeners instead of scopes and graphs.
>
>
>I'm not saying you're wrong, esp. about ITB vs external summing. One would
>expect that to sound at least slightly different. But I would be absolutely
>shocked if anyone could tell in a controlled ABY test whether they were
listening
>to SX, Performer, of Sonar.
>
>"LaMont" <jjdpro@ameritech.net> wrote:
>>
>>The only real test is with the ears and not scopes and graphs.
>>
>>"TCB" <nobody@ishere.com> wrote:
>>>
>>>I'd like to see this proven in a controlled ABY test.
>>>
>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>
>>>>Hey Neil,
>>>>
>>>>All I'm saying is: All DAW software have their own unique sound. Despite
>>>>what those lame summing test shows..
>>>>
>>>>PT-HD has a very distinct sound. A very polished sound, with a nice top
>>>end,
>>>>but with full audio spectrum represented. Mixer/Summing buss can be pushed,
>>>>but you have to watch it.
>>>>
>>>>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype the
>>top
>>>>nor bottom end.
>>>>
>>>>Logic Audio: Very Broad- Aggressive sound, that really works for Rock
>and
>>>>R & B/Gospel mixes.
>>>>
>>>>Digital Performer: With their hardware, superb audio quality. Full bodied
>>>>sound .
>>>>
>>>>Sonar: Very flat sounding. I would say that Sonar is your most vanilla
>>sound
>>>>DW on the market..
>>>>
>>>>Samplitude : A little less top end than Pro Tools. Full bodied 3d sound..
>>>>
>>>>Paris: Dark sounding in comparison to the the other DAWs. But, has a
3d
>>>sound
>>>>quality that's full bodied.
>>>>
>>>>I feel that you asking SX to be something it's not with some analog summing.
>>>>Especialy for your genre of music..
>>>>
>>>>
>>>>
>>>>
>>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>>
>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>
>>>>>>"I'd disagree with you in this instance because I happen to think the
>>>Cubase
>>>>>>ones DO sound better."
>>>>>>
>>>>>>Then that SSL Engineer does not know what they are doing with board.
>>There's
>>>>>>no way a mix coming off of that board SSL should sound better than
a
>>ITB
>>>>>>Cubase SX mix..
>>>>>>
>>>>>>Sorry, that just does not jive. That engineer does not know how to
push
>>>>>he
>>>>>>SSL or just not familiar with it.
>>>>>
>>>>>You're not really paying attention, are you? It was the same
>>>>>engineer (me). And as far as whether or not I know how to use
>>>>>that particular board, I guess that would be a matter of
>>>>>your opinion. I don't think the SSL mixes are bad ones, I think
>>>>>they came out good; I just think that you can hear more detail
>>>>>in the ITB mixes in the examples I gave, and they have more
>>>>>wideband frequency content from top to bottom.
>>>>>
>>>>>Anyway, my point of that particular comparison wasn't to say
>>>>>"ITB mixes are better than using a large-format console that
>>>>>costs somewhere in the six-figure range", the point of it was to
>>>>>address a signal-chain suggestion that Paul had... he had
>>>>>suggested perhaps that I needed to pick up a few pieces of
>>>>>killer vintage gear, and I was just demonstrating that I think
>>>>>the various signal chain components that I have here are on par
>>>>>with most anything that can be found in heavy-hitter studios...
>>>>>we used probably around $100k's worth of mics & pre's on the
>>>>>PTHD/SSL mixes, plus obviously you're looking at another
>>>>>roughly $100k for that particular console (40-channel E-series,
>>>>>black EQ's, w/G-series Computer & Total Recall package), add in
>>>>>the PTHD, outboard gear & whatnot, and you end up with
>>>>>somewhere around a quarter-mil's worth of equipment involved in
>>>>>that project. The project done at my place was done with my
>>>>>gear, which certainly doesn't tally up to anywhere remotely
>>>>>close to that cost & none of it bears a "vintage" stamp, but it
>>>>>sounds competitive with the project that used all the heavy-
>>>>>hitter stuff.
>>>>>
>>>>>Neil
>>>>
>>>
>>
>Deej, you should ask Santa for one of these. You could tell him you'll die
if you don't get one; )
Peerless LCT-104 Multi-Screen LCD Desktop Mount
http://www.cdw.com/shop/products/default.aspx?edc=849750& ;cm_ven=RKG&cm_cat=adwords&cm_pla=Video&cm_ite=p eerless_lct-104They say imitation is the best form of flattery, in this case, this sure sucks
for Apple.
Apples intentions have been well established in the industry for some time
now. I guess the innovator is always copied. I'm sure in the end they will
do fine. They have lost iTV and now iPhone.
http://blogs.zdnet.com/Berlind/?p=233
http://blogs.zdnet.com/BTL/?p=4160This is a multi-part message in MIME format.
------=_NextPart_000_0075_01C723A5.F852CB20
Content-Type: text/plain;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
I am using a NVidia GeForce 7300GT in a AMD 64X2, 4800.
I am seeing screen resolutions of 1280x1024, 1600x 1024 and 1600x1200
as options on the video card. Do those change when a different monitor =
is attached?
Can I use two 20 or 22" widescreen LCDs with native 1680x1050?
http://www.tigerdirect.com/applications/SearchTools/item-det ails.asp?EdpN=
o=3D2549197&CatId=3D0
If not what are my options in the 19" and up flatcsreen LCDs these days? =
I want to use two
but will deal with one at first.
Any deals from those who have been looking would be great!
Thanks,
Tom
I choose Polesoft Lockspam to fight spam, and you?
http://www.polesoft.com/refer.html
------=_NextPart_000_0075_01C723A5.F852CB20
Content-Type: text/html;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1400" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>I am using a NVidia GeForce 7300GT in a =
AMD 64X2,=20
4800.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>I am seeing screen resolutions of =
1280x1024, 1600x=20
1024 and 1600x1200</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>as options on the video card. Do =
those change=20
when a different monitor is attached?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Can I use two 20 or 22" widescreen LCDs =
with native=20
1680x1050?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2><A=20
href=3D" http://www.tigerdirect.com/applications/SearchTools/item-det ails.=
asp?EdpNo=3D2549197&CatId=3D0">http://www.tigerdirect.com/application=
s/SearchTools/item-details.asp?EdpNo=3D2549197&CatId =3D0 </A></FONT></=
DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>If not what are my options in the =
19" and up=20
flatcsreen LCDs these days? I want to use two</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>but will deal with one at =
first.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Any deals from those who have been =
looking would be=20
great!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Thanks,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Tom</FONT></DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT size=3D2><BR><BR>I choose Polesoft Lockspam to fight spam, =
and=20
you?<BR><A=20
href=3D"http://www.polesoft.com/refer.html">http://www.polesoft.com/refer=
..html</A> </FONT></DIV></BODY ></HTML>
------=_NextPart_000_0075_01C723A5.F852CB20--"Don Nafe" <dnafe@magma.ca> wrote:
>
>"Gantt Kushner" <ganttmann@comcast.net> wrote in message
>news:45884918$1@linux...
>>
>> I usually render tracks that have lots of edits before mixing. I've found
>> the UAD-1 FX to be pretty glitchy around edits. I also always create
a
>> second
>> or two of silence before and after every track that has plugins inserted.
>> In fact, I discovered that the Lexicon PCM 42 delay plugin needs a full
>> delays worth of track after the event to function properly.
>>
>> Anyway, even w/ rendered tracks I'm having problems...
>>
>> Gantt
>
>That's the final step after all is said and done because I've been know
to
>change things right up until the mix is approverd for mastering
>
>Don
You can always clone a track, render one & keep the edits
without rendering on the first track just in case (assuming you
have enough track capacity available to do this with - in SX,
and I presume also in Nuendo, you can disable tracks to conserve
CPU usage & that's the prefect time to use that function - on
the "safety" track. copy(ies).
NeilBelieve it or not I think Apple shoud sue them for trademark infringement
I love litigation during the Christmas season
I think they'd have a great case...when was the last time cisco had an "i"
anything
DOn
"James McCloskey" <excelsm@hotmail.com> wrote in message
news:45887b89$1@linux...
>
> They say imitation is the best form of flattery, in this case, this sure
> sucks
> for Apple.
>
> Apples intentions have been well established in the industry for some time
> now. I guess the innovator is always copied. I'm sure in the end they
> will
> do fine. They have lost iTV and now iPhone.
>
> http://blogs.zdnet.com/Berlind/?p=233
>
> http://blogs.zdnet.com/BTL/?p=4160Finally!! After al the speculation..Now, If Apple comes out with Logic pro
8 (Aka) Fianl Cut /Soundtrack Pro 2.0.. Yahooo
"James McCloskey" <excelsm@hotmail.com> wrote:
>
>They say imitation is the best form of flattery, in this case, this sure
sucks
>for Apple.
>
>Apples intentions have been well established in the industry for some time
>now. I guess the innovator is always copied. I'm sure in the end they
will
>do fine. They have lost iTV and now iPhone.
>
>http://blogs.zdnet.com/Berlind/?p=233
>
>http://blogs.zdnet.com/BTL/?p=4160"Gantt Kushner" <ganttmann@comcast.net> wrote:
>
>Both of my UAD-1 cards live in my G4. The EDS cards all live in my Magma
>13 slot expansion chassis. Is it possible that my G4 doesn't have enough
>juice to run 2 UAD-1 cards?
Gantt... I'm not very knowledgeable about Macs, so I can't
really say for sure, but it would seem weird if it doesn't have
enough wattage capacity for just those two cards - if that's
all there are in the 'puter itself.
This would be a moot point if the PSU is going bad, of course.
A couple of thoughts:
1.) Does the Magma leach any power away from the computer at
all? IOW, apart from it's own power supply, does it suck any
from the PCI bus just by virtue of the fact that it interfaces
with that?
2.) Have you tried using the rig with only ONE UAD card... not
necessarily as a practical application since if you have two,
you have two for a reason, but I mean just to see if you get
rid of the popping, etc. by removing either of the cards? If
it's a power-related issue this could be helpful to try in
order to troubleshoot.
3.) Do you think it could be ONE of the UAD cards doing this?
IOW, could one of 'em be bad? Have you been able to establish
that it's, in fact, BOTH of them that are vomiting artifacts?
4.) Have you been able to establish if either your disk
streaming capacity is being exceeded, or if your CPU is maxed
out? Either one could cause popping/clicking (although the
former would more likely cause stuttering & dropouts).
Just trying to maybe narrow things down.
NeilLaMont wrote:
>>Finally!! After al the speculation..Now, If Apple comes out with Logic
pro
8 (Aka) Fianl Cut /Soundtrack Pro 2.0.. Yahooo<<
Well who knows LaMont, but I kind of doubt it. Mac World has never been
a pro Audio show for Apple. They used to have a booth at NAMM, but they
usually just show what they currently have at the time. It will likely be
GarageBand 07 at MacWorld. Did you hear something?
The year kicks off with CES in Las Vegas, followed by MacWorld then the NAMM
Show. Apple never exhibits at CES, it's kind of a Bill Gates show. Apple
usually shows off their new consumer electronics at MW. The new pro audio
and pro video products are shown off at NAB in april. However, things are
changing at Apple, so you never know.
I think Apple sees SoundTrack as the integrated audio part of FCP, not part
of Logic. I think their strategy is, if you want the features of SoundTrack
you'll ante up for FCP. Then you'll become an FCP user, or vice versa.
I think they will continue to build a tighter integration between Logic and
Sound Track. I think your right, from a users point of view, it would be
a good thing if they integrated all of SoundTrack features in to logic and
improved the usability through the user interface.
Apple needs to make Logic intuitive and easy to use. The Logic, build your
own DAW the way you want, has to be in the back ground. It's great that
you can tweak Logic, but it needs to be easy out of the box. It needs to
be, within fifteen minutes out of the box, your recording, editing, MIDI
and mixing. Nobody wants to read a thick telephone book style manual anymore.
They also need to catch up a bit on some of the whiz bang features other
DAW softwares have now, like Live and SX 4. They should copy others for
once, and also innovate. They really need to make the features so universal
that anybody can painlessly switch to logic, or at least be able to jump
on it in a pinch and be OK.
A huge mistake was kicking VSTs out. they should have kept VST compatibility.
Somebody please correct me if I'm wrong on this one.
In the end, I think Apple will improve Logic and it will be a serious contender
in 07.
JamesI know we disagree here Lamont and that's totally cool, so I won't take this
beyond this one response, and this isn't really directed to you, but my general
thoughts on the matter.
In Neil's "defense" (not that he needs it), I and others have done this comparison
to death and the conclusion I've come to is that people are 80% influenced
by a change in environment (e.g. software interface) and 20% ears. Sorry
to say it, but the difference in sound between floating point DAWs is far
from real. It's just good, albeit unintentional marketing created by users
and capitolized by manufacturers. Perceiving a "sound" in DAWs that in actuality
process data identically, is a bad reason to pick a DAW, but of course there
is nothing wrong with thinking you hear a difference as long as it doesn't
become an unwritten law of engineering at large. Preferring to work with
one or the other, and "feeling" better about it for whatever reason is a
great reason to pick one DAW over another.
There was a recent thread that Nuendo handled gain through groups differently,
so I put Nuendo, Sonar 6 (both 32 and 64-bit engines) and Sequoia 8.3 to
the test - identical tests, setup to the 1/100th of a dB identically and
came up with absolutely no difference, either audible or scientific. To
be honest, this was the one test where I could have said, yes there is an
understandable difference between DAWs in a simple math function, and the
only one in the DAW that actually might make sense, yet even that did not
exist. The reason - math is math. You can paint it red, blue, silver or
dull grey, but it's still the same math unless the programmer was high or
completely incompetent when they wrote the code.
I thought it was entirely possible the original poster had found something
different in Nuendo, but when it came down to really understanding and reproducing
what happens in DAW summing and gain structures accurately between each DAW,
there was none, nada, nil. The assertion was completely squashed. This also
showed me how easy it is for a wide range of professionals to misinterpret
digital audio - whether hearing things, or just setting up a test with a
single missed variable that completely invalidates the whole process.
If you hear a difference, great. I've thought I heard a difference doing
similar comparisons, then changed my perspective (nothing else - not converters,
nothing - just reset my expectations, and switched back and forth) and could
hear no difference.
Just leave some room for other opinions when you post yours on this subject
since it is very obvious that hearing is not as universally objective and
identically referenced as everyone might like to believe, and is highly visually
and environmentally affected. Some will hear differences in DAWs. There
are Cubase SX 3 users claiming Cubase 4 sounds different. Sigh. Then they
realize they aren't even using the same project... or at least different
EQs, or etc, etc....
Say what you want about published summing tests, but Lynn's tests are as
accurate as it gets, and that bears out in the results (all floating point
DAWs cancel and sound identical - if you are hearing a difference, you are
hearing things that aren't there, or you forgot to align their gain and placement).
I've worked with Lynn at least briefly enough to know his attention to detail.
In the same way people will disagree about PCs and Macs until neither exists,
so will audio engineers disagree about DAWs. This is one debate that will
always exist as long as we have different ears, eyes, brains,... and opinions.
What Neil has done is to prove that opinions are always going to differ (i.e.
no consensus on the "best" mix of the ones posted). And in truth everyone
has a different perception of sound in general - not everyone wants to hear
things the same way, so we judge "best" from very different perspectives.
There is no single gold standard. There are variations and mutated combinations,
but all are subjective. That in and of itself implies very distinctly that
people can and will even perceive the exact same sound differently if presented
with any variable that changes the brain's interpretation, even if just a
visual distraction. Just change the lights in the room and see if you perceive
a song differently played back exactly the same way. Or have a cat run across
a desk while listening. Whether you care to admit it or not, it is there,
and that is actually the beauty of how our sense interact to create perception.
That may be our undoing with DAW comparison tests, but it's also what keeps
music fresh and creative, when we allow it to.
So my suggestion is to use what makes you most creative, even if it's just
a "feeling" working with that DAW gives you - be it the workflow, the GUI,
or even the name brand reputation. But, as we all know, if you can't make
most any material sound good on whatever DAW you choose, the DAW isn't the
problem.
Regards,
Dedric
"Neil" <IUOIU@OIU.com> wrote:
>
>That's interesting - all those DAW sonic interpretations, I
>mean... I haven't had a chance to usee all of those, so it's
>good information.
>
>I still don't understand why you consider my summing
>comparisons "lame", however - it was a fair set of tests;
>the same mix summed in different ways. Not trying to prove a
>point or to rig it so one sounded any better than the other - in
>fact, if you recall the thread, different people liked different
>summed versions for different reasons... there wasn't any one
>version that stood out as being "the one" that everyone felt
>sounded better. The only reason I didn't come right out & say
>right away which version was which is so that I didn't bias
>anyone's opinion beforehand by mentioning that... NOT to try
>& "hide" anything or "trick" anyone, as you accused me of
>
>Sheesh!
>
>Neil
>
>
>"Lamont" <jjdpro@ameritech.net> wrote:
>>
>>Hey Neil,
>>
>>All I'm saying is: All DAW software have their own unique sound. Despite
>>what those lame summing test shows..
>>
>>PT-HD has a very distinct sound. A very polished sound, with a nice top
>end,
>>but with full audio spectrum represented. Mixer/Summing buss can be pushed,
>>but you have to watch it.
>>
>>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype the
top
>>nor bottom end.
>>
>>Logic Audio: Very Broad- Aggressive sound, that really works for Rock and
>>R & B/Gospel mixes.
>>
>>Digital Performer: With their hardware, superb audio quality. Full bodied
>>sound .
>>
>>Sonar: Very flat sounding. I would say that Sonar is your most vanilla
sound
>>DW on the market..
>>
>>Samplitude : A little less top end than Pro Tools. Full bodied 3d sound..
>>
>>Paris: Dark sounding in comparison to the the other DAWs. But, has a 3d
>sound
>>quality that's full bodied.
>>
>>I feel that you asking SX to be something it's not with some analog summing.
>>Especialy for your genre of music..
>>
>>
>>
>>
>>"Neil" <IUOIU@OIU.com> wrote:
>>>
>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>
>>>>"I'd disagree with you in this instance because I happen to think the
>Cubase
>>>>ones DO sound better."
>>>>
>>>>Then that SSL Engineer does not know what they are doing with board.
There's
>>>>no way a mix coming off of that board SSL should sound better than a
ITB
>>>>Cubase SX mix..
>>>>
>>>>Sorry, that just does not jive. That engineer does not know how to push
>>>he
>>>>SSL or just not familiar with it.
>>>
>>>You're not really paying attention, are you? It was the same
>>>engineer (me). And as far as whether or not I know how to use
>>>that particular board, I guess that would be a matter of
>>>your opinion. I don't think the SSL mixes are bad ones, I think
>>>they came out good; I just think that you can hear more detail
>>>in the ITB mixes in the examples I gave, and they have more
>>>wideband frequency content from top to bottom.
>>>
>>>Anyway, my point of that particular comparison wasn't to say
>>>"ITB mixes are better than using a large-format console that
>>>costs somewhere in the six-figure range", the point of it was to
>>>address a signal-chain suggestion that Paul had... he had
>>>suggested perhaps that I needed to pick up a few pieces of
>>>killer vintage gear, and I was just demonstrating that I think
>>>the various signal chain components that I have here are on par
>>>with most anything that can be found in heavy-hitter studios...
>>>we used probably around $100k's worth of mics & pre's on the
>>>PTHD/SSL mixes, plus obviously you're looking at another
>>>roughly $100k for that particular console (40-channel E-series,
>>>black EQ's, w/G-series Computer & Total Recall package), add in
>>>the PTHD, outboard gear & whatnot, and you end up with
>>>somewhere around a quarter-mil's worth of equipment involved in
>>>that project. The project done at my place was done with my
>>>gear, which certainly doesn't tally up to anywhere remotely
>>>close to that cost & none of it bears a "vintage" stamp, but it
>>>sounds competitive with the project that used all the heavy-
>>>hitter stuff.
>>>
>>>Neil
>>
>Hi Neil - responses interleaved below...
"Neil" <OIUIU@OIU.com> wrote:
>1.) Does the Magma leach any power away from the computer at
>all? IOW, apart from it's own power supply, does it suck any
>from the PCI bus just by virtue of the fact that it interfaces
>with that?
I don't know the answer to this one. The Magma has a PCI card that interfaces
it w/ the computer. I'm sure it draws some power. The Magma itself has
it's own power supply.
>
>2.) Have you tried using the rig with only ONE UAD card... not
>necessarily as a practical application since if you have two,
>you have two for a reason, but I mean just to see if you get
>rid of the popping, etc. by removing either of the cards? If
>it's a power-related issue this could be helpful to try in
>order to troubleshoot.
I tried popping one card - the older one - out. It didn't make any strange
noises, but the problem is intermittent. I can go hours, even days, without
any noises.
>
>3.) Do you think it could be ONE of the UAD cards doing this?
>IOW, could one of 'em be bad? Have you been able to establish
>that it's, in fact, BOTH of them that are vomiting artifacts?
Yes, I do think it might be one card going bad. I ordered a new one from
Morgan, but it hasn't arrived yet.
>
>4.) Have you been able to establish if either your disk
>streaming capacity is being exceeded, or if your CPU is maxed
>out? Either one could cause popping/clicking (although the
>former would more likely cause stuttering & dropouts).
Don't know the answer to this either. The problems can occur w/ the UAD-1
CPU meter as low as 40% - 50%. I used to be able to push it to over 70%
with no noises. The screen re-draw would slow down a bit, but it would play.
>
>Just trying to maybe narrow things down.
I appreciate any and all ideas!
Thanks,
GanttThis is a multi-part message in MIME format.
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The answers you seek probably lie in here:
http://www.google.com/search?q=3DNVidia+GeForce+7300GT+& sourceid=3Die7&rl=
s=3Dcom.microsoft:en-US&ie=3Dutf8&oe=3Dutf8
AA
"Tom Bruhl" <arpegio@comcast.net> wrote in message =
news:45888485@linux...
I am using a NVidia GeForce 7300GT in a AMD 64X2, 4800.
I am seeing screen resolutions of 1280x1024, 1600x 1024 and 1600x1200
as options on the video card. Do those change when a different =
monitor is attached?
Can I use two 20 or 22" widescreen LCDs with native 1680x1050?
=
http://www.tigerdirect.com/applications/SearchTools/item-det ails.asp?EdpN=
o=3D2549197&CatId=3D0
If not what are my options in the 19" and up flatcsreen LCDs these =
days? I want to use two
but will deal with one at first.
Any deals from those who have been looking would be great!
Thanks,
Tom
I choose Polesoft Lockspam to fight spam, and you?
http://www.polesoft.com/refer.html
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charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.5730.11" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>The answers you seek probably lie in=20
here:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2><A=20
href=3D" http://www.google.com/search?q=3DNVidia+GeForce+7300GT+& amp;sourc=
eid=3Die7&rls=3Dcom.microsoft:en-US&ie=3Dutf 8&oe=3Dutf8 ">http=
://www.google.com/search?q=3DNVidia+GeForce+7300GT+&sourceid=3Die7&am=
p;rls=3Dcom.microsoft:en-US&ie=3Dutf8&oe=3Du tf8 </A></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>AA</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Tom Bruhl" <<A=20
href=3D"mailto:arpegio@comcast.net">arpegio@comcast.net</A>> wrote =
in message=20
<A href=3D"news:45888485@linux">news:45888485@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>I am using a NVidia GeForce 7300GT in =
a AMD 64X2,=20
4800.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>I am seeing screen resolutions of =
1280x1024,=20
1600x 1024 and 1600x1200</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>as options on the video card. =
Do those=20
change when a different monitor is attached?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Can I use two 20 or 22" widescreen =
LCDs with=20
native 1680x1050?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2><A=20
=
href=3D" http://www.tigerdirect.com/applications/SearchTools/item-det ails.=
asp?EdpNo=3D2549197&CatId=3D0">http://www.tigerdirect.com/application=
s/SearchTools/item-details.asp?EdpNo=3D2549197&CatId =3D0 </A></FONT></=
DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>If not what are my options =
in the 19" and up=20
flatcsreen LCDs these days? I want to use two</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>but will deal with one at =
first.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Any deals from those who have been =
looking would=20
be great!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Thanks,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Tom</FONT></DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT size=3D2><BR><BR>I choose Polesoft Lockspam to fight spam, =
and=20
you?<BR><A=20
=
href=3D"http://www.polesoft.com/refer.html">http://www.polesoft.com/refer=
..html</A> </FONT></DIV></BLOCKQUOTE ></BODY></HTML>
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Thanks Aaron!
This is what I found:
"Dual-Link DVI Support
Able to drive the industry's largest and highest resolution flat-panel =
displays up to 2560x1600. Feature on GeForce 7300 GT GPUs only.
Single-Link DVI Support
Able to drive the industry's largest and highest resolution flat-panel =
displays up to and including 1900x1200. May vary by model."
nView Multi-Display Technology
The NVIDIA=AE nView=AE hardware and software technology combination =
delivers maximum flexibility for multi-display options, and provides =
unprecedented end-user control of the desktop experience. NVIDIA GPUs =
are enabled to support multi-displays, but graphics cards vary. Please =
verify multi-display support in the graphics card before purchasing.
Advanced Display Functionality=20
a.. Dual integrated 400MHz RAMDACs for analog display resolutions up =
to and including 2048x1536 at 85Hz=20
b.. Dual-link DVI capability to drive the industry's largest and =
highest resolution digital flat panel displays up to 2560x16001 *
c.. * 1. SLI support for GeForce 7950 GX2 will be provided through a =
future ForceWare driver release.
d.. Integrated HDTV encoder provides analog TV-output =
(Component/Composite/S-Video) up to 1080i resolution=20
e.. Full NVIDIA=AE nView=AE multi-display technology capability=20
I do have 512 Meg of memory on the 7300GT. Should I expect this card to =
support all widescreen resolutions within 2560x1600?
Man that would be great! When using a 19" VGA and only offered =
1600x1200 max out the VGA output.
Do you think the 2560x1600 is split between two monitors?
Idunno?
Tom
"Aaron Allen" <know-spam@not_here.dude> wrote in message =
news:4588e0f1$1@linux...
The answers you seek probably lie in here:
=
http://www.google.com/search?q=3DNVidia+GeForce+7300GT+& sourceid=3Die7&rl=
s=3Dcom.microsoft:en-US&ie=3Dutf8&oe=3Dutf8
AA
"Tom Bruhl" <arpegio@comcast.net> wrote in message =
news:45888485@linux...
I am using a NVidia GeForce 7300GT in a AMD 64X2, 4800.
I am seeing screen resolutions of 1280x1024, 1600x 1024 and =
1600x1200
as options on the video card. Do those change when a different =
monitor is attached?
Can I use two 20 or 22" widescreen LCDs with native 1680x1050?
=
http://www.tigerdirect.com/applications/SearchTools/item-det ails.asp?EdpN=
o=3D2549197&CatId=3D0
If not what are my options in the 19" and up flatcsreen LCDs these =
days? I want to use two
but will deal with one at first.
Any deals from those who have been looking would be great!
Thanks,
Tom
I choose Polesoft Lockspam to fight spam, and you?
http://www.polesoft.com/refer.html
------=_NextPart_000_0015_01C723E2.6B1A5F60
Content-Type: text/html;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1400" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Thanks Aaron!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>This is what I found:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>"<STRONG><SPAN =
class=3DadditionalInfoHeader>Dual-Link=20
DVI Support</SPAN><BR></STRONG>Able to drive the industry's largest and =
highest=20
resolution flat-panel displays up to 2560x1600. <I>Feature on GeForce =
7300 GT=20
GPUs only</I>.<BR><BR><STRONG><SPAN =
class=3DadditionalInfoHeader>Single-Link DVI=20
Support</SPAN><BR></STRONG>Able to drive the industry=92s largest and =
highest=20
resolution flat-panel displays up to and including 1900x1200. <I>May =
vary by=20
model</I>."</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><A class=3DadditionalInfoHeader=20
href=3D"http://www.nvidia.com/object/feature_nview.html"><STRONG><FONT=20
color=3D#000000>nView Multi-Display =
Technology</FONT></STRONG></A><BR>The NVIDIA=AE=20
nView=AE hardware and software technology combination delivers maximum =
flexibility=20
for multi-display options, and provides unprecedented end-user control =
of the=20
desktop experience. <I>NVIDIA GPUs are enabled to support =
multi-displays, but=20
graphics cards vary. Please verify multi-display support in the graphics =
card=20
before purchasing</I>.</DIV>
<DIV> </DIV>
<DIV><STRONG>Advanced Display Functionality</STRONG>=20
<UL>
<LI>Dual integrated 400MHz RAMDACs for analog display resolutions up =
to and=20
including 2048x1536 at 85Hz=20
<LI>Dual-link DVI capability to drive the industry's largest and =
highest=20
resolution digital flat panel displays up to 2560x1600<SUP>1</SUP>=20
<SUP>*</SUP>
<LI><SUP>* </SUP>1. <I>SLI support for GeForce 7950 GX2 will be =
provided=20
through a future ForceWare driver release.</I><BR>
<LI>Integrated HDTV encoder provides analog TV-output=20
(Component/Composite/S-Video) up to 1080i resolution=20
<LI>Full NVIDIA=AE nView=AE multi-display technology capability =
</LI></UL></DIV>
<DIV><BR> </DIV>
<DIV><FONT face=3DArial size=3D2>I do have 512 Meg of memory on the =
7300GT. Should I expect this card to support all widescreen=20
resolutions within 2560x1600?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Man that would be =
great! When using a=20
19" VGA and only offered 1600x1200 max out the VGA output.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Do you think the 2560x1600 =
is split between=20
two monitors?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Idunno?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Tom</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT><FONT face=3DArial=20
size=3D2></FONT><BR> </DIV>
<DIV><FONT face=3DArial size=3D2><BR> </DIV></FONT>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Aaron Allen" <<A=20
=
href=3D"mailto:know-spam@not_here.dude">know-spam@not_here.dude</A>> =
wrote in=20
message <A =
href=3D"news:4588e0f1$1@linux">news:4588e0f1$1@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>The answers you seek probably lie in=20
here:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2><A=20
=
href=3D" http://www.google.com/search?q=3DNVidia+GeForce+7300GT+& amp;sourc=
eid=3Die7&rls=3Dcom.microsoft:en-US&ie=3Dutf 8&oe=3Dutf8 ">http=
://www.google.com/search?q=3DNVidia+GeForce+7300GT+&sourceid=3Die7&am=
p;rls=3Dcom.microsoft:en-US&ie=3Dutf8&oe=3Du tf8 </A></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>AA</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Tom Bruhl" <<A=20
href=3D"mailto:arpegio@comcast.net">arpegio@comcast.net</A>> =
wrote in=20
message <A =
href=3D"news:45888485@linux">news:45888485@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>I am using a NVidia GeForce 7300GT =
in a AMD=20
64X2, 4800.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>I am seeing screen resolutions of =
1280x1024,=20
1600x 1024 and 1600x1200</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>as options on the video card. =
Do those=20
change when a different monitor is attached?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Can I use two 20 or 22" widescreen =
LCDs with=20
native 1680x1050?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2><A=20
=
href=3D" http://www.tigerdirect.com/applications/SearchTools/item-det ails.=
asp?EdpNo=3D2549197&CatId=3D0">http://www.tigerdirect.com/application=
s/SearchTools/item-details.asp?EdpNo=3D2549197&CatId =3D0 </A></FONT></=
DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>If not what are my options =
in the 19" and=20
up flatcsreen LCDs these days? I want to use =
two</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>but will deal with one at =
first.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Any deals from those who have been =
looking=20
would be great!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Thanks,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Tom</FONT></DIV>
<DIV> </DIV>
<DIV> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT size=3D2><BR><BR>I choose Polesoft Lockspam to fight =
spam, and=20
you?<BR><A=20
=
href=3D"http://www.polesoft.com/refer.html">http://www.polesoft.com/refer=
..html</A> </FONT></DIV></BLOCKQUOTE ></BLOCKQUOTE></BODY>=
</HTML>
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charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
My card has one VGA, one DVI and one S Video connector on it.
I choose Polesoft Lockspam to fight spam, and you?
http://www.polesoft.com/refer.html
------=_NextPart_000_000B_01C723E3.40E89080
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<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>
<DIV>My card has one VGA, one DVI and one S Video connector on =
it.</FONT></DIV></DIV>
<DIV><FONT size=3D2><BR><BR>I choose Polesoft Lockspam to fight spam, =
and=20
you?<BR><A=20
href=3D"http://www.polesoft.com/refer.html">http://www.polesoft.com/refer=
..html</A> </FONT></DIV></BODY ></HTML>
------=_NextPart_000_000B_01C723E3.40E89080--"Neil" <IUOIU@OIU.com> wrote in message news:45888685$1@linux...
>
> "Don Nafe" <dnafe@magma.ca> wrote:
>>
>>"Gantt Kushner" <ganttmann@comcast.net> wrote in message
>>news:45884918$1@linux...
>>>
>>> I usually render tracks that have lots of edits before mixing. I've
>>> found
>>> the UAD-1 FX to be pretty glitchy around edits. I also always create
> a
>>> second
>>> or two of silence before and after every track that has plugins
>>> inserted.
>>> In fact, I discovered that the Lexicon PCM 42 delay plugin needs a full
>>> delays worth of track after the event to function properly.
>>>
>>> Anyway, even w/ rendered tracks I'm having problems...
>>>
>>> Gantt
>>
>>That's the final step after all is said and done because I've been know
> to
>>change things right up until the mix is approverd for mastering
>>
>>Don
>
> You can always clone a track, render one & keep the edits
> without rendering on the first track just in case (assuming you
> have enough track capacity available to do this with - in SX,
> and I presume also in Nuendo, you can disable tracks to conserve
> CPU usage & that's the prefect time to use that function - on
> the "safety" track. copy(ies).
>
> Neil
I'll also use jails for the uncloned / unrendered version of the tracks on
ocassion...that's come in handy a number of timesHi Mike,
I have been asked to build another PARIS rig and would love
to know more about your Dual core , what you did and how you got there.
Please let us know how in every detail you were able to get this going.
I will find a way to reward you :)
Thanks Mike :)
Morgan
Eastcoast Music Mall
morganp@ntplx.net
Hi Mike,
Congrats on your hard work . Today I was
doing homework with my son Logan.
His assignment was to define " Hero "
One that acts with courage and bravery despite
overwhelming odds.
Mike - you are now My Hero !!
Respectfully Yours ,
Morgan
Mike P wrote:
> Ok, so it took ten days of total frustration, but I have succeeded!
> My HP dual core is running Paris flawlessly, and it is awesome. This
computer
> is blazingly fast. I was processing files in wavelab that would have
taken
> 4-5 min on my old cpu, in 52secs. As I have faced every error known
to man during this install, and fixed them,
> if anyone has questions, please ask. Thanks to everyone who helped
along the way.Gantt...somemore responses below......
"Gantt Kushner" <ganttmann@comcast.net> wrote in message
news:4588d307$1@linux...
>
> Hi Neil - responses interleaved below...
>
> "Neil" <OIUIU@OIU.com> wrote:
>
>>1.) Does the Magma leach any power away from the computer at
>>all? IOW, apart from it's own power supply, does it suck any
>>from the PCI bus just by virtue of the fact that it interfaces
>>with that?
>
> I don't know the answer to this one. The Magma has a PCI card that
> interfaces
> it w/ the computer. I'm sure it draws some power. The Magma itself has
> it's own power supply.
I doubt it. Magma has it's own PSU. It'ws possible though that the Magma PSU
could be going south and causing trouble.
>>
>>2.) Have you tried using the rig with only ONE UAD card... not
>>necessarily as a practical application since if you have two,
>>you have two for a reason, but I mean just to see if you get
>>rid of the popping, etc. by removing either of the cards? If
>>it's a power-related issue this could be helpful to try in
>>order to troubleshoot.
>
> I tried popping one card - the older one - out. It didn't make any
> strange
> noises, but the problem is intermittent. I can go hours, even days,
> without
> any noises.
>
>>
>>3.) Do you think it could be ONE of the UAD cards doing this?
>>IOW, could one of 'em be bad? Have you been able to establish
>>that it's, in fact, BOTH of them that are vomiting artifacts?
>
> Yes, I do think it might be one card going bad. I ordered a new one from
> Morgan, but it hasn't arrived yet.
I have had two UAD-1 cards die on me since I bought my first one (which
still works) in September 2001.
>
>>
>>4.) Have you been able to establish if either your disk
>>streaming capacity is being exceeded, or if your CPU is maxed
>>out? Either one could cause popping/clicking (although the
>>former would more likely cause stuttering & dropouts).
>
> Don't know the answer to this either. The problems can occur w/ the UAD-1
> CPU meter as low as 40% - 50%. I used to be able to push it to over 70%
> with no noises. The screen re-draw would slow down a bit, but it would
> play.
This is sounding more and more like a PCI bus overload, wherein you're just
exceeding it's bandwidth. I had this problem in spades when I was mixing RME
and UAD-1 PCI cards in a Magma. Problem was solved by using *only* UAD-1
cards in the Magma and getting another Magma for the RME cards.
Deej
>
>>
>>Just trying to maybe narrow things down.
>
> I appreciate any and all ideas!
>
> Thanks,
>
> Gantt
>I'm not convinced I can hear any difference between native systems, with the
exception of the last version of ACID I used (way back, I think v. 3) which
did sound truly awful. The real test on that one for me was the DAWSUM CD
(which I purchased and dutifully scored because I was convinced 'summing'
was the real reason PARIS sounded so good) wherein I discovered that I could
barely tell one mix from the next even when hearing vastly different systems.
Since then I am a skeptic, as opposed to a disbeliever, when I hear that
one piece of software sounds greatly better, or even different, than another.
I'm not saying some people can't tell some pieces of software from other
pieces of software, I'm just saying I'm skeptical one system is 'bright'
or 'sharp' or anything else until someone can produce statistically meaningful
results in an ABY test.
One of the great things about that DAWSUM CD is it has let me use the software
that I like the most, without worrying too much about the sound. That would
be Ableton Live most of the time, with SX as a backup if the editing gets
more intense. For me that alone was worth the time I spent on the DAWSUM
CD.
TCB
"LaMont" <jjdpro@ameritech.net> wrote:
>
>Thad, you really can't hear the difference?? Maybe I own too many software
>DAWS thru the years.
>
>Starting on Logic Audio 3.0, then to Cakwalk, Pro Tools, DP, Paris,Acid,
>Neundo, Sonar, samplitude..
>
>I can hear the diference with the same audio interface with the same wav
>file(s)as oon as I import the file or files.
>
>These days, depending on the genre I'm mising determines which DAW software
>I'll use.
>My circle of engineer and producer buddies all can hear the difference in
>a second. Just the other day, we were mixing this R&B(ish)Gospel track and
>somebody said, 'Mont, this is begging for Paris. Another track, the call
>was for Pro Tools. And another,Nuendo..
>I know BrianT feels and hears the same way in different DAW software. It's
>really obvious..
>
>
>"TCB" <nobody@ishere.com> wrote:
>>
>>Which is why ABY testing uses expert listeners instead of scopes and graphs.
>>
>>
>>I'm not saying you're wrong, esp. about ITB vs external summing. One would
>>expect that to sound at least slightly different. But I would be absolutely
>>shocked if anyone could tell in a controlled ABY test whether they were
>listening
>>to SX, Performer, of Sonar.
>>
>>"LaMont" <jjdpro@ameritech.net> wrote:
>>>
>>>The only real test is with the ears and not scopes and graphs.
>>>
>>>"TCB" <nobody@ishere.com> wrote:
>>>>
>>>>I'd like to see this proven in a controlled ABY test.
>>>>
>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>
>>>>>Hey Neil,
>>>>>
>>>>>All I'm saying is: All DAW software have their own unique sound. Despite
>>>>>what those lame summing test shows..
>>>>>
>>>>>PT-HD has a very distinct sound. A very polished sound, with a nice
top
>>>>end,
>>>>>but with full audio spectrum represented. Mixer/Summing buss can be
pushed,
>>>>>but you have to watch it.
>>>>>
>>>>>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype the
>>>top
>>>>>nor bottom end.
>>>>>
>>>>>Logic Audio: Very Broad- Aggressive sound, that really works for Rock
>>and
>>>>>R & B/Gospel mixes.
>>>>>
>>>>>Digital Performer: With their hardware, superb audio quality. Full bodied
>>>>>sound .
>>>>>
>>>>>Sonar: Very flat sounding. I would say that Sonar is your most vanilla
>>>sound
>>>>>DW on the market..
>>>>>
>>>>>Samplitude : A little less top end than Pro Tools. Full bodied 3d sound..
>>>>>
>>>>>Paris: Dark sounding in comparison to the the other DAWs. But, has a
>3d
>>>>sound
>>>>>quality that's full bodied.
>>>>>
>>>>>I feel that you asking SX to be something it's not with some analog
summing.
>>>>>Especialy for your genre of music..
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>>>
>>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>>
>>>>>>>"I'd disagree with you in this instance because I happen to think
the
>>>>Cubase
>>>>>>>ones DO sound better."
>>>>>>>
>>>>>>>Then that SSL Engineer does not know what they are doing with board.
>>>There's
>>>>>>>no way a mix coming off of that board SSL should sound better than
>a
>>>ITB
>>>>>>>Cubase SX mix..
>>>>>>>
>>>>>>>Sorry, that just does not jive. That engineer does not know how to
>push
>>>>>>he
>>>>>>>SSL or just not familiar with it.
>>>>>>
>>>>>>You're not really paying attention, are you? It was the same
>>>>>>engineer (me). And as far as whether or not I know how to use
>>>>>>that particular board, I guess that would be a matter of
>>>>>>your opinion. I don't think the SSL mixes are bad ones, I think
>>>>>>they came out good; I just think that you can hear more detail
>>>>>>in the ITB mixes in the examples I gave, and they have more
>>>>>>wideband frequency content from top to bottom.
>>>>>>
>>>>>>Anyway, my point of that particular comparison wasn't to say
>>>>>>"ITB mixes are better than using a large-format console that
>>>>>>costs somewhere in the six-figure range", the point of it was to
>>>>>>address a signal-chain suggestion that Paul had... he had
>>>>>>suggested perhaps that I needed to pick up a few pieces of
>>>>>>killer vintage gear, and I was just demonstrating that I think
>>>>>>the various signal chain components that I have here are on par
>>>>>>with most anything that can be found in heavy-hitter studios...
>>>>>>we used probably around $100k's worth of mics & pre's on the
>>>>>>PTHD/SSL mixes, plus obviously you're looking at another
>>>>>>roughly $100k for that particular console (40-channel E-series,
>>>>>>black EQ's, w/G-series Computer & Total Recall package), add in
>>>>>>the PTHD, outboard gear & whatnot, and you end up with
>>>>>>somewhere around a quarter-mil's worth of equipment involved in
>>>>>>that project. The project done at my place was done with my
>>>>>>gear, which certainly doesn't tally up to anywhere remotely
>>>>>>close to that cost & none of it bears a "vintage" stamp, but it
>>>>>>sounds competitive with the project that used all the heavy-
>>>>>>hitter stuff.
>>>>>>
>>>>>>Neil
>>>>>
>>>>
>>>
>>
>HI Tom,
These monitors will be fine with your card. The 7300GT supports that
resolution on both the VGA and DVI connections.I would run both with VGA
connections. Having one hooked up DVI and one VGA may make the 2 look
very different in brightness and such.
Chris
Tom Bruhl wrote:
> My card has one VGA, one DVI and one S Video connector on it.
>
>
> I choose Polesoft Lockspam to fight spam, and you?
> http://www.polesoft.com/refer.html
--
Chris Ludwig
ADK
chrisl@adkproaudio.com <mailto:chrisl@adkproaudio.com>
www.adkproaudio.com <http://www.adkproaudio.com/>
(859) 635-5762Just a head's up. I've always avoided using the nVidia display manager software,
just on the theory that any such stuff would be definition be hinky. On the
new machine I have hooked up to my 24" 1920x1600 monitor I decided to try
it out. It seems to work very well, with zero processor overhead that I can
see. 32 virtual desktops can be defined as well, application constrained
or not, and accessed via hotkeys. Nice to see that fancy XP boxes are finally
getting a level of window management as good as linux boxes have had for
years ;-)
TCBOK, so I've figured out that I can use my Dell keyboard on a Mac G4 Dual system....
what I can't figure out is what keystrokes to get the CD tray to eject, LOL...
so Mac gurus.. how's that happen w/o the eject key (non-Mac keyboard)? using
a mac keyboard isn't really optional as I need, for space reasons, to use
a KVM and only one keyboard and mouse. This model Mac has no eject on the
computer itself.
AAAaron,
If your using OSX,go to the finder under file,you should see eject.
Or you can just drag the icon to the trash and it should eject.
respect
Nappy
"Aaron Allen" <AA@nospam.net> wrote:
>
>OK, so I've figured out that I can use my Dell keyboard on a Mac G4 Dual
system....
>what I can't figure out is what keystrokes to get the CD tray to eject,
LOL...
>
>so Mac gurus.. how's that happen w/o the eject key (non-Mac keyboard)? using
>a mac keyboard isn't really optional as I need, for space reasons, to use
>a KVM and only one keyboard and mouse. This model Mac has no eject on the
>computer itself.
>
>AAGood morning gents,
I finally took another leap and purchased a Lucid WC generator. Have three
EDS cards in a 7-slot Magma (with one UAD-1). MEC connected to card A, MEC
connected to card B, and 442 connected to card C. Edited the paris.cfg file
to use house sync and the two MECs lock up fine to the Lucid. However, the
442 acts as if the WC cable isn't even plugged in. I'm using three separate
75-ohm BNC-BNC cables.
Any thoughts?
Thanks & happy holidays to all!
MarkTry the 442 on card A...
David.
Mark McDermott wrote:
> Good morning gents,
>
> I finally took another leap and purchased a Lucid WC generator. Have three
> EDS cards in a 7-slot Magma (with one UAD-1). MEC connected to card A, MEC
> connected to card B, and 442 connected to card C. Edited the paris.cfg file
> to use house sync and the two MECs lock up fine to the Lucid. However, the
> 442 acts as if the WC cable isn't even plugged in. I'm using three separate
> 75-ohm BNC-BNC cables.
>
> Any thoughts?
>
> Thanks & happy holidays to all!
>
> MarkHey,
Yes the NView stuff works great. just did a 2 card 4 head system hat
works great. Much less clunky and best of all fan less not like the
Matrox crap. The new 8000 series card I setup is real quiet for a gamer
card. The matrox PCI-e card are load as hell.
Chris
TCB wrote:
> Just a head's up. I've always avoided using the nVidia display manager software,
> just on the theory that any such stuff would be definition be hinky. On the
> new machine I have hooked up to my 24" 1920x1600 monitor I decided to try
> it out. It seems to work very well, with zero processor overhead that I can
> see. 32 virtual desktops can be defined as well, application constrained
> or not, and accessed via hotkeys. Nice to see that fancy XP boxes are finally
> getting a level of window management as good as linux boxes have had for
> years ;-)
>
> TCB
--
Chris Ludwig
ADK Pro Audio
(859) 635-5762
www.adkproaudio.com
chrisl@adkproaudio.com>with the exception of the last version of ACID I used (way back, I think v.
>3) which did sound truly awful.<
Don't hold your breath hoping ACID will sound any better. I DL'ed V6.0 and
it sounds pretty awful too. I'm going to run it using Rewire in cubase SX 3
and lightpipe it to Paris and see if it's the actual ACID audio engine or
the summing. I've got a feeling it's the audio engine..just a
feeling......because I've also got Vegas here and it sucks too.
;o)
"TCB" <nobody@ishere.com> wrote in message news:45894947$1@linux...
>
> I'm not convinced I can hear any difference between native systems, with
> the
> exception of the last version of ACID I used (way back, I think v. 3)
> which
> did sound truly awful. The real test on that one for me was the DAWSUM CD
> (which I purchased and dutifully scored because I was convinced 'summing'
> was the real reason PARIS sounded so good) wherein I discovered that I
> could
> barely tell one mix from the next even when hearing vastly different
> systems.
> Since then I am a skeptic, as opposed to a disbeliever, when I hear that
> one piece of software sounds greatly better, or even different, than
> another.
> I'm not saying some people can't tell some pieces of software from other
> pieces of software, I'm just saying I'm skeptical one system is 'bright'
> or 'sharp' or anything else until someone can produce statistically
> meaningful
> results in an ABY test.
>
> One of the great things about that DAWSUM CD is it has let me use the
> software
> that I like the most, without worrying too much about the sound. That
> would
> be Ableton Live most of the time, with SX as a backup if the editing gets
> more intense. For me that alone was worth the time I spent on the DAWSUM
> CD.
>
> TCB
>
> "LaMont" <jjdpro@ameritech.net> wrote:
>>
>>Thad, you really can't hear the difference?? Maybe I own too many software
>>DAWS thru the years.
>>
>>Starting on Logic Audio 3.0, then to Cakwalk, Pro Tools, DP, Paris,Acid,
>>Neundo, Sonar, samplitude..
>>
>>I can hear the diference with the same audio interface with the same wav
>>file(s)as oon as I import the file or files.
>>
>>These days, depending on the genre I'm mising determines which DAW
>>software
>>I'll use.
>>My circle of engineer and producer buddies all can hear the difference in
>>a second. Just the other day, we were mixing this R&B(ish)Gospel track and
>>somebody said, 'Mont, this is begging for Paris. Another track, the call
>>was for Pro Tools. And another,Nuendo..
>>I know BrianT feels and hears the same way in different DAW software. It's
>>really obvious..
>>
>>
>>"TCB" <nobody@ishere.com> wrote:
>>>
>>>Which is why ABY testing uses expert listeners instead of scopes and
>>>graphs.
>>>
>>>
>>>I'm not saying you're wrong, esp. about ITB vs external summing. One
>>>would
>>>expect that to sound at least slightly different. But I would be
>>>absolutely
>>>shocked if anyone could tell in a controlled ABY test whether they were
>>listening
>>>to SX, Performer, of Sonar.
>>>
>>>"LaMont" <jjdpro@ameritech.net> wrote:
>>>>
>>>>The only real test is with the ears and not scopes and graphs.
>>>>
>>>>"TCB" <nobody@ishere.com> wrote:
>>>>>
>>>>>I'd like to see this proven in a controlled ABY test.
>>>>>
>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>
>>>>>>Hey Neil,
>>>>>>
>>>>>>All I'm saying is: All DAW software have their own unique sound.
>>>>>>Despite
>>>>>>what those lame summing test shows..
>>>>>>
>>>>>>PT-HD has a very distinct sound. A very polished sound, with a nice
> top
>>>>>end,
>>>>>>but with full audio spectrum represented. Mixer/Summing buss can be
> pushed,
>>>>>>but you have to watch it.
>>>>>>
>>>>>>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype the
>>>>top
>>>>>>nor bottom end.
>>>>>>
>>>>>>Logic Audio: Very Broad- Aggressive sound, that really works for Rock
>>>and
>>>>>>R & B/Gospel mixes.
>>>>>>
>>>>>>Digital Performer: With their hardware, superb audio quality. Full
>>>>>>bodied
>>>>>>sound .
>>>>>>
>>>>>>Sonar: Very flat sounding. I would say that Sonar is your most vanilla
>>>>sound
>>>>>>DW on the market..
>>>>>>
>>>>>>Samplitude : A little less top end than Pro Tools. Full bodied 3d
>>>>>>sound..
>>>>>>
>>>>>>Paris: Dark sounding in comparison to the the other DAWs. But, has a
>>3d
>>>>>sound
>>>>>>quality that's full bodied.
>>>>>>
>>>>>>I feel that you asking SX to be something it's not with some analog
> summing.
>>>>>>Especialy for your genre of music..
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>>>>
>>>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>>>
>>>>>>>>"I'd disagree with you in this instance because I happen to think
> the
>>>>>Cubase
>>>>>>>>ones DO sound better."
>>>>>>>>
>>>>>>>>Then that SSL Engineer does not know what they are doing with board.
>>>>There's
>>>>>>>>no way a mix coming off of that board SSL should sound better than
>>a
>>>>ITB
>>>>>>>>Cubase SX mix..
>>>>>>>>
>>>>>>>>Sorry, that just does not jive. That engineer does not know how to
>>push
>>>>>>>he
>>>>>>>>SSL or just not familiar with it.
>>>>>>>
>>>>>>>You're not really paying attention, are you? It was the same
>>>>>>>engineer (me). And as far as whether or not I know how to use
>>>>>>>that particular board, I guess that would be a matter of
>>>>>>>your opinion. I don't think the SSL mixes are bad ones, I think
>>>>>>>they came out good; I just think that you can hear more detail
>>>>>>>in the ITB mixes in the examples I gave, and they have more
>>>>>>>wideband frequency content from top to bottom.
>>>>>>>
>>>>>>>Anyway, my point of that particular comparison wasn't to say
>>>>>>>"ITB mixes are better than using a large-format console that
>>>>>>>costs somewhere in the six-figure range", the point of it was to
>>>>>>>address a signal-chain suggestion that Paul had... he had
>>>>>>>suggested perhaps that I needed to pick up a few pieces of
>>>>>>>killer vintage gear, and I was just demonstrating that I think
>>>>>>>the various signal chain components that I have here are on par
>>>>>>>with most anything that can be found in heavy-hitter studios...
>>>>>>>we used probably around $100k's worth of mics & pre's on the
>>>>>>>PTHD/SSL mixes, plus obviously you're looking at another
>>>>>>>roughly $100k for that particular console (40-channel E-series,
>>>>>>>black EQ's, w/G-series Computer & Total Recall package), add in
>>>>>>>the PTHD, outboard gear & whatnot, and you end up with
>>>>>>>somewhere around a quarter-mil's worth of equipment involved in
>>>>>>>that project. The project done at my place was done with my
>>>>>>>gear, which certainly doesn't tally up to anywhere remotely
>>>>>>>close to that cost & none of it bears a "vintage" stamp, but it
>>>>>>>sounds competitive with the project that used all the heavy-
>>>>>>>hitter stuff.
>>>>>>>
>>>>>>>Neil
>>>>>>
>>>>>
>>>>
>>>
>>
>I'm rooting for Logic audio. I really am. But, that interface and the way
it handles audio,editing audio has to be changed. It need Soundtrac pro's
look. It needs a polished lookink mixer..Not an apogee symphony mixer add-on..
I hae to say this, but they need to the asthestics of Pro Tools & Nuendo/SX.
Simple, elegant, but powerful.
If not, then Logic & DP will be small players for film composers only. Meanwhile,
Pro Tools/SX, Sonar, Samplitude keeps a great divide .. Just having great
midi is not going to cut anymore these days. Nuendo/SX & even Protools does
more than most user need to compose tier music.
I really do hope somebody at emagic/Apple comes to term with their delima,,
and re-write the entire Logic App from group up like Steingberg did back
in 2000 with Neundo, then SX..
"James McCloskey" <excelsm@hotmail.com> wrote:
>
>LaMont wrote:
>
>>>Finally!! After al the speculation..Now, If Apple comes out with Logic
>pro
>8 (Aka) Fianl Cut /Soundtrack Pro 2.0.. Yahooo<<
>
>Well who knows LaMont, but I kind of doubt it. Mac World has never been
>a pro Audio show for Apple. They used to have a booth at NAMM, but they
>usually just show what they currently have at the time. It will likely
be
>GarageBand 07 at MacWorld. Did you hear something?
>
>The year kicks off with CES in Las Vegas, followed by MacWorld then the
NAMM
>Show. Apple never exhibits at CES, it's kind of a Bill Gates show. Apple
>usually shows off their new consumer electronics at MW. The new pro audio
>and pro video products are shown off at NAB in april. However, things are
>changing at Apple, so you never know.
>
>I think Apple sees SoundTrack as the integrated audio part of FCP, not part
>of Logic. I think their strategy is, if you want the features of SoundTrack
>you'll ante up for FCP. Then you'll become an FCP user, or vice versa.
>I think they will continue to build a tighter integration between Logic
and
>Sound Track. I think your right, from a users point of view, it would be
>a good thing if they integrated all of SoundTrack features in to logic and
>improved the usability through the user interface.
>
>Apple needs to make Logic intuitive and easy to use. The Logic, build your
>own DAW the way you want, has to be in the back ground. It's great that
>you can tweak Logic, but it needs to be easy out of the box. It needs to
>be, within fifteen minutes out of the box, your recording, editing, MIDI
>and mixing. Nobody wants to read a thick telephone book style manual anymore.
> They also need to catch up a bit on some of the whiz bang features other
>DAW softwares have now, like Live and SX 4. They should copy others for
>once, and also innovate. They really need to make the features so universal
>that anybody can painlessly switch to logic, or at least be able to jump
>on it in a pinch and be OK.
>
>A huge mistake was kicking VSTs out. they should have kept VST compatibility.
> Somebody please correct me if I'm wrong on this one.
>
>In the end, I think Apple will improve Logic and it will be a serious contender
>in 07.
>
>James....or card C
;o)
"EK Sound" <askme@nospam.com> wrote in message news:458973cb$1@linux...
> Try the 442 on card A...
>
> David.
>
> Mark McDermott wrote:
>> Good morning gents,
>>
>> I finally took another leap and purchased a Lucid WC generator. Have
>> three
>> EDS cards in a 7-slot Magma (with one UAD-1). MEC connected to card A,
>> MEC
>> connected to card B, and 442 connected to card C. Edited the paris.cfg
>> file
>> to use house sync and the two MECs lock up fine to the Lucid. However,
>> the
>> 442 acts as if the WC cable isn't even plugged in. I'm using three
>> separate
>> 75-ohm BNC-BNC cables.
>>
>> Any thoughts?
>>
>> Thanks & happy holidays to all!
>>
>> MarkI've been pleasantly surprised. I also wonder if I really need a second big
monitor with the 24" I have now. I might put a 19" to the right of it that
I can switch to my linux box but it might be overkill, esp with where my
speakers are placed.
Anyway, it's worth a look if people are on nVidia cards.
TCB
Chris Ludwig <chrisl@adkproaudio.com> wrote:
>Hey,
>Yes the NView stuff works great. just did a 2 card 4 head system hat
>works great. Much less clunky and best of all fan less not like the
>Matrox crap. The new 8000 series card I setup is real quiet for a gamer
>card. The matrox PCI-e card are load as hell.
>
>Chris
>
>
>TCB wrote:
>> Just a head's up. I've always avoided using the nVidia display manager
software,
>> just on the theory that any such stuff would be definition be hinky. On
the
>> new machine I have hooked up to my 24" 1920x1600 monitor I decided to
try
>> it out. It seems to work very well, with zero processor overhead that
I can
>> see. 32 virtual desktops can be defined as well, application constrained
>> or not, and accessed via hotkeys. Nice to see that fancy XP boxes are
finally
>> getting a level of window management as good as linux boxes have had for
>> years ;-)
>>
>> TCB
>
>--
>Chris Ludwig
>
>ADK Pro Audio
>(859) 635-5762
>www.adkproaudio.com
>chrisl@adkproaudio.comHey Dedric and Neil,
I reason I think that the Summing CD test(good intentions) were lame was
because.. If a person can;t hear the difference btw a stereo wav file that's
in Acid vs Sonar really needs a hearing test.
For reason of my music work, I have to work with different DAWs, so I'm very
familiar with their sound qualities. My circle of producers and engineers
talk about the daw sonics all the time. It's really no big deal anymore..
The same logic applies when Roger Nichols (a few) years back in his article
about master CD's and that he found out that 4 differnt CD burners yeilded
differnt sonic results. Sure, he sated that Math is the Math :) but, his
and the masering engineers Ears told them soemthing was different. Hummm???
Now, back to DAW sonics. I can hear the difference btw Paris and Nuendo vs
Pro Tools, Logic audio.. There is no math to this, this is an ear thing..You
either hear or you don't.. Simple.
But, good ears can hear it. .
I really think the problem is, noone want to no that their money that they've
spent on a given DAW, has sonic limitations or shall we say, just different..
I like that they all sound different. It's good to have choice when mixing
a song. Some DAWs, depending on the genre will yield better or the desired
results and than another.
EX. I would not mix a Acoustic jazz record today with Paris..reason, I'm
going for clarity at it's highest level.. For that project, It's either Neundo
or Pro Tools and may Samplitude..Why should I fight with Paris's thick, gooy
sonics, when I'm going for clarity. Well, Pro Tools and Nuendo/SX has that
sound right out the gate.. Which makes my job a lot easier. simple. This
is not tosay that I could not get the job done in Paris..i could..But, for
that Acoutic Jazz project , the other 2 daws gives me what I'm looking for
without even touching an eq..
This is not all about math. As BrianT states: Use you ears..Forget the math..What
does knowing the math do for you anyway? Nothing, it just proves that you
know the math. Does not tell you diddly about the sonics.. Just ask Roger
Nichols..
"Dedric Terry" <d@nospam.net> wrote:
>
>I know we disagree here Lamont and that's totally cool, so I won't take
this
>beyond this one response, and this isn't really directed to you, but my
general
>thoughts on the matter.
>
>In Neil's "defense" (not that he needs it), I and others have done this
comparison
>to death and the conclusion I've come to is that people are 80% influenced
>by a change in environment (e.g. software interface) and 20% ears. Sorry
>to say it, but the difference in sound between floating point DAWs is far
>from real. It's just good, albeit unintentional marketing created by users
>and capitolized by manufacturers. Perceiving a "sound" in DAWs that in
actuality
>process data identically, is a bad reason to pick a DAW, but of course there
>is nothing wrong with thinking you hear a difference as long as it doesn't
>become an unwritten law of engineering at large. Preferring to work with
>one or the other, and "feeling" better about it for whatever reason is a
>great reason to pick one DAW over another.
>
>There was a recent thread that Nuendo handled gain through groups differently,
>so I put Nuendo, Sonar 6 (both 32 and 64-bit engines) and Sequoia 8.3 to
>the test - identical tests, setup to the 1/100th of a dB identically and
>came up with absolutely no difference, either audible or scientific. To
>be honest, this was the one test where I could have said, yes there is an
>understandable difference between DAWs in a simple math function, and the
>only one in the DAW that actually might make sense, yet even that did not
>exist. The reason - math is math. You can paint it red, blue, silver or
>dull grey, but it's still the same math unless the programmer was high or
>completely incompetent when they wrote the code.
>
>I thought it was entirely possible the original poster had found something
>different in Nuendo, but when it came down to really understanding and reproducing
>what happens in DAW summing and gain structures accurately between each
DAW,
>there was none, nada, nil. The assertion was completely squashed. This also
>showed me how easy it is for a wide range of professionals to misinterpret
>digital audio - whether hearing things, or just setting up a test with a
>single missed variable that completely invalidates the whole process.
>
>If you hear a difference, great. I've thought I heard a difference doing
>similar comparisons, then changed my perspective (nothing else - not converters,
>nothing - just reset my expectations, and switched back and forth) and could
>hear no difference.
>
>Just leave some room for other opinions when you post yours on this subject
>since it is very obvious that hearing is not as universally objective and
>identically referenced as everyone might like to believe, and is highly
visually
>and environmentally affected. Some will hear differences in DAWs. There
>are Cubase SX 3 users claiming Cubase 4 sounds different. Sigh. Then they
>realize they aren't even using the same project... or at least different
>EQs, or etc, etc....
>
>Say what you want about published summing tests, but Lynn's tests are as
>accurate as it gets, and that bears out in the results (all floating point
>DAWs cancel and sound identical - if you are hearing a difference, you are
>hearing things that aren't there, or you forgot to align their gain and
placement).
> I've worked with Lynn at least briefly enough to know his attention to
detail.
> In the same way people will disagree about PCs and Macs until neither exists,
>so will audio engineers disagree about DAWs. This is one debate that will
>always exist as long as we have different ears, eyes, brains,... and opinions.
>
>
>What Neil has done is to prove that opinions are always going to differ
(i.e.
>no consensus on the "best" mix of the ones posted). And in truth everyone
>has a different perception of sound in general - not everyone wants to hear
>things the same way, so we judge "best" from very different perspectives.
> There is no single gold standard. There are variations and mutated combinations,
>but all are subjective. That in and of itself implies very distinctly that
>people can and will even perceive the exact same sound differently if presented
>with any variable that changes the brain's interpretation, even if just
a
>visual distraction. Just change the lights in the room and see if you perceive
>a song differently played back exactly the same way. Or have a cat run
across
>a desk while listening. Whether you care to admit it or not, it is there,
>and that is actually the beauty of how our sense interact to create perception.
> That may be our undoing with DAW comparison tests, but it's also what keeps
>music fresh and creative, when we allow it to.
>
>So my suggestion is to use what makes you most creative, even if it's just
>a "feeling" working with that DAW gives you - be it the workflow, the GUI,
>or even the name brand reputation. But, as we all know, if you can't make
>most any material sound good on whatever DAW you choose, the DAW isn't the
>problem.
>
>Regards,
>Dedric
>
>"Neil" <IUOIU@OIU.com> wrote:
>>
>>That's interesting - all those DAW sonic interpretations, I
>>mean... I haven't had a chance to usee all of those, so it's
>>good information.
>>
>>I still don't understand why you consider my summing
>>comparisons "lame", however - it was a fair set of tests;
>>the same mix summed in different ways. Not trying to prove a
>>point or to rig it so one sounded any better than the other - in
>>fact, if you recall the thread, different people liked different
>>summed versions for different reasons... there wasn't any one
>>version that stood out as being "the one" that everyone felt
>>sounded better. The only reason I didn't come right out & say
>>right away which version was which is so that I didn't bias
>>anyone's opinion beforehand by mentioning that... NOT to try
>>& "hide" anything or "trick" anyone, as you accused me of
>>
>>Sheesh!
>>
>>Neil
>>
>>
>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>
>>>Hey Neil,
>>>
>>>All I'm saying is: All DAW software have their own unique sound. Despite
>>>what those lame summing test shows..
>>>
>>>PT-HD has a very distinct sound. A very polished sound, with a nice top
>>end,
>>>but with full audio spectrum represented. Mixer/Summing buss can be pushed,
>>>but you have to watch it.
>>>
>>>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype the
>top
>>>nor bottom end.
>>>
>>>Logic Audio: Very Broad- Aggressive sound, that really works for Rock
and
>>>R & B/Gospel mixes.
>>>
>>>Digital Performer: With their hardware, superb audio quality. Full bodied
>>>sound .
>>>
>>>Sonar: Very flat sounding. I would say that Sonar is your most vanilla
>sound
>>>DW on the market..
>>>
>>>Samplitude : A little less top end than Pro Tools. Full bodied 3d sound..
>>>
>>>Paris: Dark sounding in comparison to the the other DAWs. But, has a 3d
>>sound
>>>quality that's full bodied.
>>>
>>>I feel that you asking SX to be something it's not with some analog summing.
>>>Especialy for your genre of music..
>>>
>>>
>>>
>>>
>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>
>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>
>>>>>"I'd disagree with you in this instance because I happen to think the
>>Cubase
>>>>>ones DO sound better."
>>>>>
>>>>>Then that SSL Engineer does not know what they are doing with board.
>There's
>>>>>no way a mix coming off of that board SSL should sound better than a
>ITB
>>>>>Cubase SX mix..
>>>>>
>>>>>Sorry, that just does not jive. That engineer does not know how to push
>>>>he
>>>>>SSL or just not familiar with it.
>>>>
>>>>You're not really paying attention, are you? It was the same
>>>>engineer (me). And as far as whether or not I know how to use
>>>>that particular board, I guess that would be a matter of
>>>>your opinion. I don't think the SSL mixes are bad ones, I think
>>>>they came out good; I just think that you can hear more detail
>>>>in the ITB mixes in the examples I gave, and they have more
>>>>wideband frequency content from top to bottom.
>>>>
>>>>Anyway, my point of that particular comparison wasn't to say
>>>>"ITB mixes are better than using a large-format console that
>>>>costs somewhere in the six-figure range", the point of it was to
>>>>address a signal-chain suggestion that Paul had... he had
>>>>suggested perhaps that I needed to pick up a few pieces of
>>>>killer vintage gear, and I was just demonstrating that I think
>>>>the various signal chain components that I have here are on par
>>>>with most anything that can be found in heavy-hitter studios...
>>>>we used probably around $100k's worth of mics & pre's on the
>>>>PTHD/SSL mixes, plus obviously you're looking at another
>>>>roughly $100k for that particular console (40-channel E-series,
>>>>black EQ's, w/G-series Computer & Total Recall package), add in
>>>>the PTHD, outboard gear & whatnot, and you end up with
>>>>somewhere around a quarter-mil's worth of equipment involved in
>>>>that project. The project done at my place was done with my
>>>>gear, which certainly doesn't tally up to anywhere remotely
>>>>close to that cost & none of it bears a "vintage" stamp, but it
>>>>sounds competitive with the project that used all the heavy-
>>>>hitter stuff.
>>>>
>>>>Neil
>>>
>>
>BrianT use to say all the time..Software has a sound. And I agree with him..
"DJ" <nowayjose@dude.net> wrote:
>>with the exception of the last version of ACID I used (way back, I think
v.
>>3) which did sound truly awful.<
>Don't hold your breath hoping ACID will sound any better. I DL'ed V6.0
and
>it sounds pretty awful too. I'm going to run it using Rewire in cubase SX
3
>and lightpipe it to Paris and see if it's the actual ACID audio engine or
>the summing. I've got a feeling it's the audio engine..just a
>feeling......because I've also got Vegas here and it sucks too.
>
>;o)
>
>"TCB" <nobody@ishere.com> wrote in message news:45894947$1@linux...
>>
>> I'm not convinced I can hear any difference between native systems, with
>> the
>> exception of the last version of ACID I used (way back, I think v. 3)
>> which
>> did sound truly awful. The real test on that one for me was the DAWSUM
CD
>> (which I purchased and dutifully scored because I was convinced 'summing'
>> was the real reason PARIS sounded so good) wherein I discovered that I
>> could
>> barely tell one mix from the next even when hearing vastly different
>> systems.
>> Since then I am a skeptic, as opposed to a disbeliever, when I hear that
>> one piece of software sounds greatly better, or even different, than
>> another.
>> I'm not saying some people can't tell some pieces of software from other
>> pieces of software, I'm just saying I'm skeptical one system is 'bright'
>> or 'sharp' or anything else until someone can produce statistically
>> meaningful
>> results in an ABY test.
>>
>> One of the great things about that DAWSUM CD is it has let me use the
>> software
>> that I like the most, without worrying too much about the sound. That
>> would
>> be Ableton Live most of the time, with SX as a backup if the editing gets
>> more intense. For me that alone was worth the time I spent on the DAWSUM
>> CD.
>>
>> TCB
>>
>> "LaMont" <jjdpro@ameritech.net> wrote:
>>>
>>>Thad, you really can't hear the difference?? Maybe I own too many software
>>>DAWS thru the years.
>>>
>>>Starting on Logic Audio 3.0, then to Cakwalk, Pro Tools, DP, Paris,Acid,
>>>Neundo, Sonar, samplitude..
>>>
>>>I can hear the diference with the same audio interface with the same wav
>>>file(s)as oon as I import the file or files.
>>>
>>>These days, depending on the genre I'm mising determines which DAW
>>>software
>>>I'll use.
>>>My circle of engineer and producer buddies all can hear the difference
in
>>>a second. Just the other day, we were mixing this R&B(ish)Gospel track
and
>>>somebody said, 'Mont, this is begging for Paris. Another track, the call
>>>was for Pro Tools. And another,Nuendo..
>>>I know BrianT feels and hears the same way in different DAW software.
It's
>>>really obvious..
>>>
>>>
>>>"TCB" <nobody@ishere.com> wrote:
>>>>
>>>>Which is why ABY testing uses expert listeners instead of scopes and
>>>>graphs.
>>>>
>>>>
>>>>I'm not saying you're wrong, esp. about ITB vs external summing. One
>>>>would
>>>>expect that to sound at least slightly different. But I would be
>>>>absolutely
>>>>shocked if anyone could tell in a controlled ABY test whether they were
>>>listening
>>>>to SX, Performer, of Sonar.
>>>>
>>>>"LaMont" <jjdpro@ameritech.net> wrote:
>>>>>
>>>>>The only real test is with the ears and not scopes and graphs.
>>>>>
>>>>>"TCB" <nobody@ishere.com> wrote:
>>>>>>
>>>>>>I'd like to see this proven in a controlled ABY test.
>>>>>>
>>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>>
>>>>>>>Hey Neil,
>>>>>>>
>>>>>>>All I'm saying is: All DAW software have their own unique sound.
>>>>>>>Despite
>>>>>>>what those lame summing test shows..
>>>>>>>
>>>>>>>PT-HD has a very distinct sound. A very polished sound, with a nice
>> top
>>>>>>end,
>>>>>>>but with full audio spectrum represented. Mixer/Summing buss can be
>> pushed,
>>>>>>>but you have to watch it.
>>>>>>>
>>>>>>>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype
the
>>>>>top
>>>>>>>nor bottom end.
>>>>>>>
>>>>>>>Logic Audio: Very Broad- Aggressive sound, that really works for Rock
>>>>and
>>>>>>>R & B/Gospel mixes.
>>>>>>>
>>>>>>>Digital Performer: With their hardware, superb audio quality. Full
>>>>>>>bodied
>>>>>>>sound .
>>>>>>>
>>>>>>>Sonar: Very flat sounding. I would say that Sonar is your most vanilla
>>>>>sound
>>>>>>>DW on the market..
>>>>>>>
>>>>>>>Samplitude : A little less top end than Pro Tools. Full bodied 3d
>>>>>>>sound..
>>>>>>>
>>>>>>>Paris: Dark sounding in comparison to the the other DAWs. But, has
a
>>>3d
>>>>>>sound
>>>>>>>quality that's full bodied.
>>>>>>>
>>>>>>>I feel that you asking SX to be something it's not with some analog
>> summing.
>>>>>>>Especialy for your genre of music..
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>>>>>
>>>>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>>>>
>>>>>>>>>"I'd disagree with you in this instance because I happen to think
>> the
>>>>>>Cubase
>>>>>>>>>ones DO sound better."
>>>>>>>>>
>>>>>>>>>Then that SSL Engineer does not know what they are doing with board.
>>>>>There's
>>>>>>>>>no way a mix coming off of that board SSL should sound better than
>>>a
>>>>>ITB
>>>>>>>>>Cubase SX mix..
>>>>>>>>>
>>>>>>>>>Sorry, that just does not jive. That engineer does not know how
to
>>>push
>>>>>>>>he
>>>>>>>>>SSL or just not familiar with it.
>>>>>>>>
>>>>>>>>You're not really paying attention, are you? It was the same
>>>>>>>>engineer (me). And as far as whether or not I know how to use
>>>>>>>>that particular board, I guess that would be a matter of
>>>>>>>>your opinion. I don't think the SSL mixes are bad ones, I think
>>>>>>>>they came out good; I just think that you can hear more detail
>>>>>>>>in the ITB mixes in the examples I gave, and they have more
>>>>>>>>wideband frequency content from top to bottom.
>>>>>>>>
>>>>>>>>Anyway, my point of that particular comparison wasn't to say
>>>>>>>>"ITB mixes are better than using a large-format console that
>>>>>>>>costs somewhere in the six-figure range", the point of it was to
>>>>>>>>address a signal-chain suggestion that Paul had... he had
>>>>>>>>suggested perhaps that I needed to pick up a few pieces of
>>>>>>>>killer vintage gear, and I was just demonstrating that I think
>>>>>>>>the various signal chain components that I have here are on par
>>>>>>>>with most anything that can be found in heavy-hitter studios...
>>>>>>>>we used probably around $100k's worth of mics & pre's on the
>>>>>>>>PTHD/SSL mixes, plus obviously you're looking at another
>>>>>>>>roughly $100k for that particular console (40-channel E-series,
>>>>>>>>black EQ's, w/G-series Computer & Total Recall package), add in
>>>>>>>>the PTHD, outboard gear & whatnot, and you end up with
>>>>>>>>somewhere around a quarter-mil's worth of equipment involved in
>>>>>>>>that project. The project done at my place was done with my
>>>>>>>>gear, which certainly doesn't tally up to anywhere remotely
>>>>>>>>close to that cost & none of it bears a "vintage" stamp, but it
>>>>>>>>sounds competitive with the project that used all the heavy-
>>>>>>>>hitter stuff.
>>>>>>>>
>>>>>>>>Neil
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>
>Hi Aaron,
Email me direct and I'll send you a little desktop app
called "eject". It works great and can be placed
to work via the function keys on a Mac.
cheers,
Mike
"Aaron Allen" <AA@nospam.net> wrote:
>
>OK, so I've figured out that I can use my Dell keyboard on a Mac G4 Dual
system....
>what I can't figure out is what keystrokes to get the CD tray to eject,
LOL...
>
>so Mac gurus.. how's that happen w/o the eject key (non-Mac keyboard)? using
>a mac keyboard isn't really optional as I need, for space reasons, to use
>a KVM and only one keyboard and mouse. This model Mac has no eject on the
>computer itself.
>
>AAI picked up a Colorgraphics Xentera 4 x head the other day. It's an old AGP
model and it's rockin' along in my native rig, even sharing IRQ's with the
Pulsar cards with no issues at all. I was very surprised at this.......and
so now it has been "Borg'ed" into the DAW'ism.
;o)
"TCB" <nobody@ishere.com> wrote in message news:45897d13$1@linux...
>
> I've been pleasantly surprised. I also wonder if I really need a second
> big
> monitor with the 24" I have now. I might put a 19" to the right of it that
> I can switch to my linux box but it might be overkill, esp with where my
> speakers are placed.
>
> Anyway, it's worth a look if people are on nVidia cards.
>
> TCB
>
> Chris Ludwig <chrisl@adkproaudio.com> wrote:
>>Hey,
>>Yes the NView stuff works great. just did a 2 card 4 head system hat
>>works great. Much less clunky and best of all fan less not like the
>>Matrox crap. The new 8000 series card I setup is real quiet for a gamer
>
>>card. The matrox PCI-e card are load as hell.
>>
>>Chris
>>
>>
>>TCB wrote:
>>> Just a head's up. I've always avoided using the nVidia display manager
> software,
>>> just on the theory that any such stuff would be definition be hinky. On
> the
>>> new machine I have hooked up to my 24" 1920x1600 monitor I decided to
> try
>>> it out. It seems to work very well, with zero processor overhead that
> I can
>>> see. 32 virtual desktops can be defined as well, application constrained
>>> or not, and accessed via hotkeys. Nice to see that fancy XP boxes are
> finally
>>> getting a level of window management as good as linux boxes have had for
>>> years ;-)
>>>
>>> TCB
>>
>>--
>>Chris Ludwig
>>
>>ADK Pro Audio
>>(859) 635-5762
>>www.adkproaudio.com
>>chrisl@adkproaudio.com
>..O'yeah.....and it's got some software that seems to allow you do do just
about anything.
"DJ" <nowayjose@dude.net> wrote in message news:45898ed9@linux...
>I picked up a Colorgraphics Xentera 4 x head the other day. It's an old AGP
>model and it's rockin' along in my native rig, even sharing IRQ's with the
>Pulsar cards with no issues at all. I was very surprised at this.......and
>so now it has been "Borg'ed" into the DAW'ism.
>
> ;o)
>
> "TCB" <nobody@ishere.com> wrote in message news:45897d13$1@linux...
>>
>> I've been pleasantly surprised. I also wonder if I really need a second
>> big
>> monitor with the 24" I have now. I might put a 19" to the right of it
>> that
>> I can switch to my linux box but it might be overkill, esp with where my
>> speakers are placed.
>>
>> Anyway, it's worth a look if people are on nVidia cards.
>>
>> TCB
>>
>> Chris Ludwig <chrisl@adkproaudio.com> wrote:
>>>Hey,
>>>Yes the NView stuff works great. just did a 2 card 4 head system hat
>>>works great. Much less clunky and best of all fan less not like the
>>>Matrox crap. The new 8000 series card I setup is real quiet for a gamer
>>
>>>card. The matrox PCI-e card are load as hell.
>>>
>>>Chris
>>>
>>>
>>>TCB wrote:
>>>> Just a head's up. I've always avoided using the nVidia display manager
>> software,
>>>> just on the theory that any such stuff would be definition be hinky. On
>> the
>>>> new machine I have hooked up to my 24" 1920x1600 monitor I decided to
>> try
>>>> it out. It seems to work very well, with zero processor overhead that
>> I can
>>>> see. 32 virtual desktops can be defined as well, application
>>>> constrained
>>>> or not, and accessed via hotkeys. Nice to see that fancy XP boxes are
>> finally
>>>> getting a level of window management as good as linux boxes have had
>>>> for
>>>> years ;-)
>>>>
>>>> TCB
>>>
>>>--
>>>Chris Ludwig
>>>
>>>ADK Pro Audio
>>>(859) 635-5762
>>>www.adkproaudio.com
>>>chrisl@adkproaudio.com
>>
>
>I'm so used to putting things in jail in Paris, but haven't found an equivalent
in Logic
- is there one?
George AxonThis is a multi-part message in MIME format.
---=_linux45899547
Content-Type: text/plain
Content-Transfer-Encoding: 7bit
"Mike Mullin" <lunarlordship@sympatico.ca> wrote:
>
>Hi Aaron,
>Email me direct and I'll send you a little desktop app
>called "eject". It works great and can be placed
>to work via the function keys on a Mac.
Is it one of these...?
:D
---=_linux45899547
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---=_linux45899547--Gene and Jamie both moved to Logic.
"George Axon" <georgeaxon@rogers.com> wrote:
>
>I'm so used to putting things in jail in Paris, but haven't found an equivalent
>in Logic
>- is there one?
>
>George AxonThat's too bad. I think people have an instinctive thing against the sound
of Live as well, just because it also loops like ACID does. Live sounds like
a properly written native DAW when working with non time stretched tracks.
The sound quality on the stretched audio is amazing, all things considered,
but the non-stretched sound is indistinguishable from SX. Too bad the only
really truly awful sounding app has to bring down a perfectly nice sounding
one.
TCB
"DJ" <nowayjose@dude.net> wrote:
>>with the exception of the last version of ACID I used (way back, I think
v.
>>3) which did sound truly awful.<
>Don't hold your breath hoping ACID will sound any better. I DL'ed V6.0
and
>it sounds pretty awful too. I'm going to run it using Rewire in cubase SX
3
>and lightpipe it to Paris and see if it's the actual ACID audio engine or
>the summing. I've got a feeling it's the audio engine..just a
>feeling......because I've also got Vegas here and it sucks too.
>
>;o)
>
>"TCB" <nobody@ishere.com> wrote in message news:45894947$1@linux...
>>
>> I'm not convinced I can hear any difference between native systems, with
>> the
>> exception of the last version of ACID I used (way back, I think v. 3)
>> which
>> did sound truly awful. The real test on that one for me was the DAWSUM
CD
>> (which I purchased and dutifully scored because I was convinced 'summing'
>> was the real reason PARIS sounded so good) wherein I discovered that I
>> could
>> barely tell one mix from the next even when hearing vastly different
>> systems.
>> Since then I am a skeptic, as opposed to a disbeliever, when I hear that
>> one piece of software sounds greatly better, or even different, than
>> another.
>> I'm not saying some people can't tell some pieces of software from other
>> pieces of software, I'm just saying I'm skeptical one system is 'bright'
>> or 'sharp' or anything else until someone can produce statistically
>> meaningful
>> results in an ABY test.
>>
>> One of the great things about that DAWSUM CD is it has let me use the
>> software
>> that I like the most, without worrying too much about the sound. That
>> would
>> be Ableton Live most of the time, with SX as a backup if the editing gets
>> more intense. For me that alone was worth the time I spent on the DAWSUM
>> CD.
>>
>> TCB
>>
>> "LaMont" <jjdpro@ameritech.net> wrote:
>>>
>>>Thad, you really can't hear the difference?? Maybe I own too many software
>>>DAWS thru the years.
>>>
>>>Starting on Logic Audio 3.0, then to Cakwalk, Pro Tools, DP, Paris,Acid,
>>>Neundo, Sonar, samplitude..
>>>
>>>I can hear the diference with the same audio interface with the same wav
>>>file(s)as oon as I import the file or files.
>>>
>>>These days, depending on the genre I'm mising determines which DAW
>>>software
>>>I'll use.
>>>My circle of engineer and producer buddies all can hear the difference
in
>>>a second. Just the other day, we were mixing this R&B(ish)Gospel track
and
>>>somebody said, 'Mont, this is begging for Paris. Another track, the call
>>>was for Pro Tools. And another,Nuendo..
>>>I know BrianT feels and hears the same way in different DAW software.
It's
>>>really obvious..
>>>
>>>
>>>"TCB" <nobody@ishere.com> wrote:
>>>>
>>>>Which is why ABY testing uses expert listeners instead of scopes and
>>>>graphs.
>>>>
>>>>
>>>>I'm not saying you're wrong, esp. about ITB vs external summing. One
>>>>would
>>>>expect that to sound at least slightly different. But I would be
>>>>absolutely
>>>>shocked if anyone could tell in a controlled ABY test whether they were
>>>listening
>>>>to SX, Performer, of Sonar.
>>>>
>>>>"LaMont" <jjdpro@ameritech.net> wrote:
>>>>>
>>>>>The only real test is with the ears and not scopes and graphs.
>>>>>
>>>>>"TCB" <nobody@ishere.com> wrote:
>>>>>>
>>>>>>I'd like to see this proven in a controlled ABY test.
>>>>>>
>>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>>
>>>>>>>Hey Neil,
>>>>>>>
>>>>>>>All I'm saying is: All DAW software have their own unique sound.
>>>>>>>Despite
>>>>>>>what those lame summing test shows..
>>>>>>>
>>>>>>>PT-HD has a very distinct sound. A very polished sound, with a nice
>> top
>>>>>>end,
>>>>>>>but with full audio spectrum represented. Mixer/Summing buss can be
>> pushed,
>>>>>>>but you have to watch it.
>>>>>>>
>>>>>>>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype
the
>>>>>top
>>>>>>>nor bottom end.
>>>>>>>
>>>>>>>Logic Audio: Very Broad- Aggressive sound, that really works for Rock
>>>>and
>>>>>>>R & B/Gospel mixes.
>>>>>>>
>>>>>>>Digital Performer: With their hardware, superb audio quality. Full
>>>>>>>bodied
>>>>>>>sound .
>>>>>>>
>>>>>>>Sonar: Very flat sounding. I would say that Sonar is your most vanilla
>>>>>sound
>>>>>>>DW on the market..
>>>>>>>
>>>>>>>Samplitude : A little less top end than Pro Tools. Full bodied 3d
>>>>>>>sound..
>>>>>>>
>>>>>>>Paris: Dark sounding in comparison to the the other DAWs. But, has
a
>>>3d
>>>>>>sound
>>>>>>>quality that's full bodied.
>>>>>>>
>>>>>>>I feel that you asking SX to be something it's not with some analog
>> summing.
>>>>>>>Especialy for your genre of music..
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>>>>>
>>>>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>>>>
>>>>>>>>>"I'd disagree with you in this instance because I happen to think
>> the
>>>>>>Cubase
>>>>>>>>>ones DO sound better."
>>>>>>>>>
>>>>>>>>>Then that SSL Engineer does not know what they are doing with board.
>>>>>There's
>>>>>>>>>no way a mix coming off of that board SSL should sound better than
>>>a
>>>>>ITB
>>>>>>>>>Cubase SX mix..
>>>>>>>>>
>>>>>>>>>Sorry, that just does not jive. That engineer does not know how
to
>>>push
>>>>>>>>he
>>>>>>>>>SSL or just not familiar with it.
>>>>>>>>
>>>>>>>>You're not really paying attention, are you? It was the same
>>>>>>>>engineer (me). And as far as whether or not I know how to use
>>>>>>>>that particular board, I guess that would be a matter of
>>>>>>>>your opinion. I don't think the SSL mixes are bad ones, I think
>>>>>>>>they came out good; I just think that you can hear more detail
>>>>>>>>in the ITB mixes in the examples I gave, and they have more
>>>>>>>>wideband frequency content from top to bottom.
>>>>>>>>
>>>>>>>>Anyway, my point of that particular comparison wasn't to say
>>>>>>>>"ITB mixes are better than using a large-format console that
>>>>>>>>costs somewhere in the six-figure range", the point of it was to
>>>>>>>>address a signal-chain suggestion that Paul had... he had
>>>>>>>>suggested perhaps that I needed to pick up a few pieces of
>>>>>>>>killer vintage gear, and I was just demonstrating that I think
>>>>>>>>the various signal chain components that I have here are on par
>>>>>>>>with most anything that can be found in heavy-hitter studios...
>>>>>>>>we used probably around $100k's worth of mics & pre's on the
>>>>>>>>PTHD/SSL mixes, plus obviously you're looking at another
>>>>>>>>roughly $100k for that particular console (40-channel E-series,
>>>>>>>>black EQ's, w/G-series Computer & Total Recall package), add in
>>>>>>>>the PTHD, outboard gear & whatnot, and you end up with
>>>>>>>>somewhere around a quarter-mil's worth of equipment involved in
>>>>>>>>that project. The project done at my place was done with my
>>>>>>>>gear, which certainly doesn't tally up to anywhere remotely
>>>>>>>>close to that cost & none of it bears a "vintage" stamp, but it
>>>>>>>>sounds competitive with the project that used all the heavy-
>>>>>>>>hitter stuff.
>>>>>>>>
>>>>>>>>Neil
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>
>Of course Paris sounds different on Lynn's sampler, that was audible, and
there are technical reasons why Paris will always sound different, but I
didn't like it better on the sampler CD, to be honest, though the
differences were subtle. Also, we weren't talking about Acid vs. Sonar
specifically. I don't even bother with Acid as a DAW example - it's a loop
app. Vegas is a video app that has had life as an audio app to some degree,
but iMovie does audio as well, yet that doesn't really put it in the same
category as professional DAW apps like Nuendo, PTHD, Sequoia, etc. I use
Vegas for video, but not audio.
On Lynn's sampler, Samplitude, Nuendo, Fairlight and the other natives don't
sound different and aren't different in the unity gain examples
(even the PTHD mix cancels with these). If you hear two files sounding
differently that cancel to complete null, an audio difference isn't what you
are hearing. When there are differences in non-unity gain mix summing
tests, you have an extra variable to account for - how is the gain
calculated? Gain
is non-linear (power), not adding two numbers together. So how is pan law
factored in, and where? Are your faders exactly the same, or 0.001dB
variant?
Also if you drop the same stereo file in two different pro audio apps and
hear a difference, one of the two apps is defective. There is nothing
happening with a stereo file playback when no gain change or plugins are
active - just audio streaming to the driver from disk. If you hear a
difference there, I would be quickly trying to find out why. Something is
wrong.
The point I am making is that these arguments usually come up as blanket
statements with no qualification of what exactly sounds
different, why it might, or solid well reasoned attempts to find out why, or
if there could be a real difference, or just a perceived one.
Usually the "use your ears" comment comes up when there is no technical
rebuttal for when the science and good
ears agree. Of course "use your ears" first from a creative perspective,
but if you are making a technical, scientific statement, then such comments
aren't a good foundation to work from. It's a great motto, but a bit of a
cop out in a technical discussion.
Regards,
Dedric
"LaMont" <jjdpro@ameriech.net> wrote in message news:45897f73$1@linux...
>
> Hey Dedric and Neil,
>
> I reason I think that the Summing CD test(good intentions) were lame was
> because.. If a person can;t hear the difference btw a stereo wav file
> that's
> in Acid vs Sonar really needs a hearing test.
>
> For reason of my music work, I have to work with different DAWs, so I'm
> very
> familiar with their sound qualities. My circle of producers and engineers
> talk about the daw sonics all the time. It's really no big deal anymore..
>
> The same logic applies when Roger Nichols (a few) years back in his
> article
> about master CD's and that he found out that 4 differnt CD burners yeilded
> differnt sonic results. Sure, he sated that Math is the Math :) but, his
> and the masering engineers Ears told them soemthing was different.
> Hummm???
>
> Now, back to DAW sonics. I can hear the difference btw Paris and Nuendo vs
> Pro Tools, Logic audio.. There is no math to this, this is an ear
> thing..You
> either hear or you don't.. Simple.
> But, good ears can hear it. .
>
> I really think the problem is, noone want to no that their money that
> they've
> spent on a given DAW, has sonic limitations or shall we say, just
> different..
>
> I like that they all sound different. It's good to have choice when mixing
> a song. Some DAWs, depending on the genre will yield better or the desired
> results and than another.
> EX. I would not mix a Acoustic jazz record today with Paris..reason, I'm
> going for clarity at it's highest level.. For that project, It's either
> Neundo
> or Pro Tools and may Samplitude..Why should I fight with Paris's thick,
> gooy
> sonics, when I'm going for clarity. Well, Pro Tools and Nuendo/SX has that
> sound right out the gate.. Which makes my job a lot easier. simple. This
> is not tosay that I could not get the job done in Paris..i could..But, for
> that Acoutic Jazz project , the other 2 daws gives me what I'm looking
> for
> without even touching an eq..
>
> This is not all about math. As BrianT states: Use you ears..Forget the
> math..What
> does knowing the math do for you anyway? Nothing, it just proves that you
> know the math. Does not tell you diddly about the sonics.. Just ask Roger
> Nichols..
>
>
> "Dedric Terry" <d@nospam.net> wrote:
>>
>>I know we disagree here Lamont and that's totally cool, so I won't take
> this
>>beyond this one response, and this isn't really directed to you, but my
> general
>>thoughts on the matter.
>>
>>In Neil's "defense" (not that he needs it), I and others have done this
> comparison
>>to death and the conclusion I've come to is that people are 80% influenced
>>by a change in environment (e.g. software interface) and 20% ears. Sorry
>>to say it, but the difference in sound between floating point DAWs is far
>>from real. It's just good, albeit unintentional marketing created by
>>users
>>and capitolized by manufacturers. Perceiving a "sound" in DAWs that in
> actuality
>>process data identically, is a bad reason to pick a DAW, but of course
>>there
>>is nothing wrong with thinking you hear a difference as long as it doesn't
>>become an unwritten law of engineering at large. Preferring to work with
>>one or the other, and "feeling" better about it for whatever reason is a
>>great reason to pick one DAW over another.
>>
>>There was a recent thread that Nuendo handled gain through groups
>>differently,
>>so I put Nuendo, Sonar 6 (both 32 and 64-bit engines) and Sequoia 8.3 to
>>the test - identical tests, setup to the 1/100th of a dB identically and
>>came up with absolutely no difference, either audible or scientific. To
>>be honest, this was the one test where I could have said, yes there is an
>>understandable difference between DAWs in a simple math function, and the
>>only one in the DAW that actually might make sense, yet even that did not
>>exist. The reason - math is math. You can paint it red, blue, silver or
>>dull grey, but it's still the same math unless the programmer was high or
>>completely incompetent when they wrote the code.
>>
>>I thought it was entirely possible the original poster had found something
>>different in Nuendo, but when it came down to really understanding and
>>reproducing
>>what happens in DAW summing and gain structures accurately between each
> DAW,
>>there was none, nada, nil. The assertion was completely squashed. This
>>also
>>showed me how easy it is for a wide range of professionals to misinterpret
>>digital audio - whether hearing things, or just setting up a test with a
>>single missed variable that completely invalidates the whole process.
>>
>>If you hear a difference, great. I've thought I heard a difference doing
>>similar comparisons, then changed my perspective (nothing else - not
>>converters,
>>nothing - just reset my expectations, and switched back and forth) and
>>could
>>hear no difference.
>>
>>Just leave some room for other opinions when you post yours on this
>>subject
>>since it is very obvious that hearing is not as universally objective and
>>identically referenced as everyone might like to believe, and is highly
> visually
>>and environmentally affected. Some will hear differences in DAWs. There
>>are Cubase SX 3 users claiming Cubase 4 sounds different. Sigh. Then
>>they
>>realize they aren't even using the same project... or at least different
>>EQs, or etc, etc....
>>
>>Say what you want about published summing tests, but Lynn's tests are as
>>accurate as it gets, and that bears out in the results (all floating point
>>DAWs cancel and sound identical - if you are hearing a difference, you are
>>hearing things that aren't there, or you forgot to align their gain and
> placement).
>> I've worked with Lynn at least briefly enough to know his attention to
> detail.
>> In the same way people will disagree about PCs and Macs until neither
>> exists,
>>so will audio engineers disagree about DAWs. This is one debate that will
>>always exist as long as we have different ears, eyes, brains,... and
>>opinions.
>>
>>
>>What Neil has done is to prove that opinions are always going to differ
> (i.e.
>>no consensus on the "best" mix of the ones posted). And in truth everyone
>>has a different perception of sound in general - not everyone wants to
>>hear
>>things the same way, so we judge "best" from very different perspectives.
>> There is no single gold standard. There are variations and mutated
>> combinations,
>>but all are subjective. That in and of itself implies very distinctly
>>that
>>people can and will even perceive the exact same sound differently if
>>presented
>>with any variable that changes the brain's interpretation, even if just
> a
>>visual distraction. Just change the lights in the room and see if you
>>perceive
>>a song differently played back exactly the same way. Or have a cat run
> across
>>a desk while listening. Whether you care to admit it or not, it is there,
>>and that is actually the beauty of how our sense interact to create
>>perception.
>> That may be our undoing with DAW comparison tests, but it's also what
>> keeps
>>music fresh and creative, when we allow it to.
>>
>>So my suggestion is to use what makes you most creative, even if it's just
>>a "feeling" working with that DAW gives you - be it the workflow, the GUI,
>>or even the name brand reputation. But, as we all know, if you can't make
>>most any material sound good on whatever DAW you choose, the DAW isn't the
>>problem.
>>
>>Regards,
>>Dedric
>>
>>"Neil" <IUOIU@OIU.com> wrote:
>>>
>>>That's interesting - all those DAW sonic interpretations, I
>>>mean... I haven't had a chance to usee all of those, so it's
>>>good information.
>>>
>>>I still don't understand why you consider my summing
>>>comparisons "lame", however - it was a fair set of tests;
>>>the same mix summed in different ways. Not trying to prove a
>>>point or to rig it so one sounded any better than the other - in
>>>fact, if you recall the thread, different people liked different
>>>summed versions for different reasons... there wasn't any one
>>>version that stood out as being "the one" that everyone felt
>>>sounded better. The only reason I didn't come right out & say
>>>right away which version was which is so that I didn't bias
>>>anyone's opinion beforehand by mentioning that... NOT to try
>>>& "hide" anything or "trick" anyone, as you accused me of
>>>
>>>Sheesh!
>>>
>>>Neil
>>>
>>>
>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>
>>>>Hey Neil,
>>>>
>>>>All I'm saying is: All DAW software have their own unique sound.
>>>>Despite
>>>>what those lame summing test shows..
>>>>
>>>>PT-HD has a very distinct sound. A very polished sound, with a nice top
>>>end,
>>>>but with full audio spectrum represented. Mixer/Summing buss can be
>>>>pushed,
>>>>but you have to watch it.
>>>>
>>>>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype the
>>top
>>>>nor bottom end.
>>>>
>>>>Logic Audio: Very Broad- Aggressive sound, that really works for Rock
> and
>>>>R & B/Gospel mixes.
>>>>
>>>>Digital Performer: With their hardware, superb audio quality. Full
>>>>bodied
>>>>sound .
>>>>
>>>>Sonar: Very flat sounding. I would say that Sonar is your most vanilla
>>sound
>>>>DW on the market..
>>>>
>>>>Samplitude : A little less top end than Pro Tools. Full bodied 3d
>>>>sound..
>>>>
>>>>Paris: Dark sounding in comparison to the the other DAWs. But, has a 3d
>>>sound
>>>>quality that's full bodied.
>>>>
>>>>I feel that you asking SX to be something it's not with some analog
>>>>summing.
>>>>Especialy for your genre of music..
>>>>
>>>>
>>>>
>>>>
>>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>>
>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>
>>>>>>"I'd disagree with you in this instance because I happen to think the
>>>Cubase
>>>>>>ones DO sound better."
>>>>>>
>>>>>>Then that SSL Engineer does not know what they are doing with board.
>>There's
>>>>>>no way a mix coming off of that board SSL should sound better than a
>>ITB
>>>>>>Cubase SX mix..
>>>>>>
>>>>>>Sorry, that just does not jive. That engineer does not know how to
>>>>>>push
>>>>>he
>>>>>>SSL or just not familiar with it.
>>>>>
>>>>>You're not really paying attention, are you? It was the same
>>>>>engineer (me). And as far as whether or not I know how to use
>>>>>that particular board, I guess that would be a matter of
>>>>>your opinion. I don't think the SSL mixes are bad ones, I think
>>>>>they came out good; I just think that you can hear more detail
>>>>>in the ITB mixes in the examples I gave, and they have more
>>>>>wideband frequency content from top to bottom.
>>>>>
>>>>>Anyway, my point of that particular comparison wasn't to say
>>>>>"ITB mixes are better than using a large-format console that
>>>>>costs somewhere in the six-figure range", the point of it was to
>>>>>address a signal-chain suggestion that Paul had... he had
>>>>>suggested perhaps that I needed to pick up a few pieces of
>>>>>killer vintage gear, and I was just demonstrating that I think
>>>>>the various signal chain components that I have here are on par
>>>>>with most anything that can be found in heavy-hitter studios...
>>>>>we used probably around $100k's worth of mics & pre's on the
>>>>>PTHD/SSL mixes, plus obviously you're looking at another
>>>>>roughly $100k for that particular console (40-channel E-series,
>>>>>black EQ's, w/G-series Computer & Total Recall package), add in
>>>>>the PTHD, outboard gear & whatnot, and you end up with
>>>>>somewhere around a quarter-mil's worth of equipment involved in
>>>>>that project. The project done at my place was done with my
>>>>>gear, which certainly doesn't tally up to anywhere remotely
>>>>>close to that cost & none of it bears a "vintage" stamp, but it
>>>>>sounds competitive with the project that used all the heavy-
>>>>>hitter stuff.
>>>>>
>>>>>Neil
>>>>
>>>
>>
>This is a multi-part message in MIME format.
------=_NextPart_000_0009_01C72452.A2714940
Content-Type: text/plain;
charset="iso-8859-1"
Content-Transfer-Encoding: 7bit
I've been using ITB verbs and Paris verbs for most of my work. But recently
I hung a TC reverb off on of Paris' external outs. I set up one of the Aux
send/returns to handle the TC. How do I record the reverberated audio into
the submix? Does this happen "automatically" when I create an intelligent
submix or do I have to return the verb to two of its own tracks? From what
I've read there, the manual doesn't have a real clear answer to the
question -though I could have missed something.
Thanks
MR
------=_NextPart_000_0009_01C72452.A2714940
Content-Type: text/html;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1555" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>I've been using ITB verbs and Paris =
verbs for most=20
of my work. But recently I hung a TC reverb off on of Paris' =
external=20
outs. I set up one of the Aux send/returns to handle the TC. =
How do=20
I record the reverberated audio into the submix? Does this happen=20
"automatically" when I create an intelligent submix or do I have to =
return the=20
verb to two of its own tracks? From what I've read there, the =
manual=20
doesn't have a real clear answer to the question -though I could have =
missed=20
something.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Thanks</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>MR</FONT></DIV></BODY></HTML>
------=_NextPart_000_0009_01C72452.A2714940--If you want to print the verb you have to return it to it's own tracks. Instead
of connecting the returns to the aux, connect them to 2 inupts, just like
you were were recording something. Hit record, and let the song play out.
Rod
"Mike R." <emarenot@yahoo.com> wrote:
>
>
>I've been using ITB verbs and Paris verbs for most of my work. But recently
>I hung a TC reverb off on of Paris' external outs. I set up one of the
Aux
>send/returns to handle the TC. How do I record the reverberated audio into
>the submix? Does this happen "automatically" when I create an intelligent
>submix or do I have to return the verb to two of its own tracks? From what
>I've read there, the manual doesn't have a real clear answer to the
>question -though I could have missed something.
>Thanks
>MR
>
>
><!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
><HTML><HEAD>
><META http-equiv=3DContent-Type content=3D"text/html; =
>charset=3Diso-8859-1">
><META content=3D"MSHTML 6.00.2800.1555" name=3DGENERATOR>
><STYLE></STYLE>
></HEAD>
><BODY bgColor=3D#ffffff>
><DIV><FONT face=3DArial size=3D2>I've been using ITB verbs and Paris =
>verbs for most=20
>of my work. But recently I hung a TC reverb off on of Paris' =
>external=20
>outs. I set up one of the Aux send/returns to handle the TC. =
>How do=20
>I record the reverberated audio into the submix? Does this happen=20
>"automatically" when I create an intelligent submix or do I have to =
>return the=20
>verb to two of its own tracks? From what I've read there, the =
>manual=20
>doesn't have a real clear answer to the question -though I could have =
>missed=20
>something.</FONT></DIV>
><DIV><FONT face=3DArial size=3D2>Thanks</FONT></DIV>
><DIV><FONT face=3DArial size=3D2>MR</FONT></DIV></BODY></HTML>
>
>Agreed..
"TCB" <nobody@ishere.com> wrote:
>
>That's too bad. I think people have an instinctive thing against the sound
>of Live as well, just because it also loops like ACID does. Live sounds
like
>a properly written native DAW when working with non time stretched tracks.
>The sound quality on the stretched audio is amazing, all things considered,
>but the non-stretched sound is indistinguishable from SX. Too bad the only
>really truly awful sounding app has to bring down a perfectly nice sounding
>one.
>
>TCB
>
>"DJ" <nowayjose@dude.net> wrote:
>>>with the exception of the last version of ACID I used (way back, I think
>v.
>>>3) which did sound truly awful.<
>>Don't hold your breath hoping ACID will sound any better. I DL'ed V6.0
>and
>>it sounds pretty awful too. I'm going to run it using Rewire in cubase
SX
>3
>>and lightpipe it to Paris and see if it's the actual ACID audio engine
or
>
>>the summing. I've got a feeling it's the audio engine..just a
>>feeling......because I've also got Vegas here and it sucks too.
>>
>>;o)
>>
>>"TCB" <nobody@ishere.com> wrote in message news:45894947$1@linux...
>>>
>>> I'm not convinced I can hear any difference between native systems, with
>
>>> the
>>> exception of the last version of ACID I used (way back, I think v. 3)
>
>>> which
>>> did sound truly awful. The real test on that one for me was the DAWSUM
>CD
>>> (which I purchased and dutifully scored because I was convinced 'summing'
>>> was the real reason PARIS sounded so good) wherein I discovered that
I
>
>>> could
>>> barely tell one mix from the next even when hearing vastly different
>>> systems.
>>> Since then I am a skeptic, as opposed to a disbeliever, when I hear that
>>> one piece of software sounds greatly better, or even different, than
>>> another.
>>> I'm not saying some people can't tell some pieces of software from other
>>> pieces of software, I'm just saying I'm skeptical one system is 'bright'
>>> or 'sharp' or anything else until someone can produce statistically
>>> meaningful
>>> results in an ABY test.
>>>
>>> One of the great things about that DAWSUM CD is it has let me use the
>
>>> software
>>> that I like the most, without worrying too much about the sound. That
>
>>> would
>>> be Ableton Live most of the time, with SX as a backup if the editing
gets
>>> more intense. For me that alone was worth the time I spent on the DAWSUM
>>> CD.
>>>
>>> TCB
>>>
>>> "LaMont" <jjdpro@ameritech.net> wrote:
>>>>
>>>>Thad, you really can't hear the difference?? Maybe I own too many software
>>>>DAWS thru the years.
>>>>
>>>>Starting on Logic Audio 3.0, then to Cakwalk, Pro Tools, DP, Paris,Acid,
>>>>Neundo, Sonar, samplitude..
>>>>
>>>>I can hear the diference with the same audio interface with the same
wav
>>>>file(s)as oon as I import the file or files.
>>>>
>>>>These days, depending on the genre I'm mising determines which DAW
>>>>software
>>>>I'll use.
>>>>My circle of engineer and producer buddies all can hear the difference
>in
>>>>a second. Just the other day, we were mixing this R&B(ish)Gospel track
>and
>>>>somebody said, 'Mont, this is begging for Paris. Another track, the call
>>>>was for Pro Tools. And another,Nuendo..
>>>>I know BrianT feels and hears the same way in different DAW software.
>It's
>>>>really obvious..
>>>>
>>>>
>>>>"TCB" <nobody@ishere.com> wrote:
>>>>>
>>>>>Which is why ABY testing uses expert listeners instead of scopes and
>
>>>>>graphs.
>>>>>
>>>>>
>>>>>I'm not saying you're wrong, esp. about ITB vs external summing. One
>
>>>>>would
>>>>>expect that to sound at least slightly different. But I would be
>>>>>absolutely
>>>>>shocked if anyone could tell in a controlled ABY test whether they were
>>>>listening
>>>>>to SX, Performer, of Sonar.
>>>>>
>>>>>"LaMont" <jjdpro@ameritech.net> wrote:
>>>>>>
>>>>>>The only real test is with the ears and not scopes and graphs.
>>>>>>
>>>>>>"TCB" <nobody@ishere.com> wrote:
>>>>>>>
>>>>>>>I'd like to see this proven in a controlled ABY test.
>>>>>>>
>>>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>>>
>>>>>>>>Hey Neil,
>>>>>>>>
>>>>>>>>All I'm saying is: All DAW software have their own unique sound.
>>>>>>>>Despite
>>>>>>>>what those lame summing test shows..
>>>>>>>>
>>>>>>>>PT-HD has a very distinct sound. A very polished sound, with a nice
>>> top
>>>>>>>end,
>>>>>>>>but with full audio spectrum represented. Mixer/Summing buss can
be
>>> pushed,
>>>>>>>>but you have to watch it.
>>>>>>>>
>>>>>>>>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype
>the
>>>>>>top
>>>>>>>>nor bottom end.
>>>>>>>>
>>>>>>>>Logic Audio: Very Broad- Aggressive sound, that really works for
Rock
>>>>>and
>>>>>>>>R & B/Gospel mixes.
>>>>>>>>
>>>>>>>>Digital Performer: With their hardware, superb audio quality. Full
>
>>>>>>>>bodied
>>>>>>>>sound .
>>>>>>>>
>>>>>>>>Sonar: Very flat sounding. I would say that Sonar is your most vanilla
>>>>>>sound
>>>>>>>>DW on the market..
>>>>>>>>
>>>>>>>>Samplitude : A little less top end than Pro Tools. Full bodied 3d
>
>>>>>>>>sound..
>>>>>>>>
>>>>>>>>Paris: Dark sounding in comparison to the the other DAWs. But, has
>a
>>>>3d
>>>>>>>sound
>>>>>>>>quality that's full bodied.
>>>>>>>>
>>>>>>>>I feel that you asking SX to be something it's not with some analog
>>> summing.
>>>>>>>>Especialy for your genre of music..
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>
>>>>>>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>>>>>>
>>>>>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>>>>>
>>>>>>>>>>"I'd disagree with you in this instance because I happen to think
>>> the
>>>>>>>Cubase
>>>>>>>>>>ones DO sound better."
>>>>>>>>>>
>>>>>>>
|
|
|
|
| Re: Paris Skins / examples 2 [message #76036 is a reply to message #76035] |
Wed, 15 November 2006 22:17 |
Yanoska
Messages: 32
Registered: January 2007
|
Member |
|
|
>>>Then that SSL Engineer does not know what they are doing with board.
>>>>>>There's
>>>>>>>>>>no way a mix coming off of that board SSL should sound better than
>>>>a
>>>>>>ITB
>>>>>>>>>>Cubase SX mix..
>>>>>>>>>>
>>>>>>>>>>Sorry, that just does not jive. That engineer does not know how
>to
>>>>push
>>>>>>>>>he
>>>>>>>>>>SSL or just not familiar with it.
>>>>>>>>>
>>>>>>>>>You're not really paying attention, are you? It was the same
>>>>>>>>>engineer (me). And as far as whether or not I know how to use
>>>>>>>>>that particular board, I guess that would be a matter of
>>>>>>>>>your opinion. I don't think the SSL mixes are bad ones, I think
>>>>>>>>>they came out good; I just think that you can hear more detail
>>>>>>>>>in the ITB mixes in the examples I gave, and they have more
>>>>>>>>>wideband frequency content from top to bottom.
>>>>>>>>>
>>>>>>>>>Anyway, my point of that particular comparison wasn't to say
>>>>>>>>>"ITB mixes are better than using a large-format console that
>>>>>>>>>costs somewhere in the six-figure range", the point of it was to
>>>>>>>>>address a signal-chain suggestion that Paul had... he had
>>>>>>>>>suggested perhaps that I needed to pick up a few pieces of
>>>>>>>>>killer vintage gear, and I was just demonstrating that I think
>>>>>>>>>the various signal chain components that I have here are on par
>>>>>>>>>with most anything that can be found in heavy-hitter studios...
>>>>>>>>>we used probably around $100k's worth of mics & pre's on the
>>>>>>>>>PTHD/SSL mixes, plus obviously you're looking at another
>>>>>>>>>roughly $100k for that particular console (40-channel E-series,
>>>>>>>>>black EQ's, w/G-series Computer & Total Recall package), add in
>>>>>>>>>the PTHD, outboard gear & whatnot, and you end up with
>>>>>>>>>somewhere around a quarter-mil's worth of equipment involved in
>>>>>>>>>that project. The project done at my place was done with my
>>>>>>>>>gear, which certainly doesn't tally up to anywhere remotely
>>>>>>>>>close to that cost & none of it bears a "vintage" stamp, but it
>>>>>>>>>sounds competitive with the project that used all the heavy-
>>>>>>>>>hitter stuff.
>>>>>>>>>
>>>>>>>>>Neil
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>
>>
>Thanks Rod.
Damn, I was living in this fantasy that somehow the aux returns would simply
get "mixed in" when I made a submix.
Cheers,
MR
"Rod Lincoln" <rlincoln@nosopam.kc.rr.com> wrote in message
news:4589a8db$1@linux...
>
> If you want to print the verb you have to return it to it's own tracks.
Instead
> of connecting the returns to the aux, connect them to 2 inupts, just like
> you were were recording something. Hit record, and let the song play out.
> Rod
> "Mike R." <emarenot@yahoo.com> wrote:
> >
> >
> >I've been using ITB verbs and Paris verbs for most of my work. But
recently
> >I hung a TC reverb off on of Paris' external outs. I set up one of the
> Aux
> >send/returns to handle the TC. How do I record the reverberated audio
into
> >the submix? Does this happen "automatically" when I create an intelligent
> >submix or do I have to return the verb to two of its own tracks? From
what
> >I've read there, the manual doesn't have a real clear answer to the
> >question -though I could have missed something.
> >Thanks
> >MR
> >
> >
> ><!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
> ><HTML><HEAD>
> ><META http-equiv=3DContent-Type content=3D"text/html; =
> >charset=3Diso-8859-1">
> ><META content=3D"MSHTML 6.00.2800.1555" name=3DGENERATOR>
> ><STYLE></STYLE>
> ></HEAD>
> ><BODY bgColor=3D#ffffff>
> ><DIV><FONT face=3DArial size=3D2>I've been using ITB verbs and Paris =
> >verbs for most=20
> >of my work. But recently I hung a TC reverb off on of Paris' =
> >external=20
> >outs. I set up one of the Aux send/returns to handle the TC. =
> >How do=20
> >I record the reverberated audio into the submix? Does this happen=20
> >"automatically" when I create an intelligent submix or do I have to =
> >return the=20
> >verb to two of its own tracks? From what I've read there, the =
> >manual=20
> >doesn't have a real clear answer to the question -though I could have =
> >missed=20
> >something.</FONT></DIV>
> ><DIV><FONT face=3DArial size=3D2>Thanks</FONT></DIV>
> ><DIV><FONT face=3DArial size=3D2>MR</FONT></DIV></BODY></HTML>
> >
> >
>Well, hey, at least we can agree that Live sounds good.
I'm really psyched for my Live/Scope setup. The Core Duo desktop is set up
and running nicely (what a change from a three year old Athlon) so I'll have
gobs of native f/x and instruments, UAD-1 plugs, and Scope synths/effects/mixing.
Then I will officially be 100% at fault if I suck ;-)
TCB
"LaMont" <jjdpro@gmail.com> wrote:
>
>Agreed..
>"TCB" <nobody@ishere.com> wrote:
>>
>>That's too bad. I think people have an instinctive thing against the sound
>>of Live as well, just because it also loops like ACID does. Live sounds
>like
>>a properly written native DAW when working with non time stretched tracks.
>>The sound quality on the stretched audio is amazing, all things considered,
>>but the non-stretched sound is indistinguishable from SX. Too bad the only
>>really truly awful sounding app has to bring down a perfectly nice sounding
>>one.
>>
>>TCB
>>
>>"DJ" <nowayjose@dude.net> wrote:
>>>>with the exception of the last version of ACID I used (way back, I think
>>v.
>>>>3) which did sound truly awful.<
>>>Don't hold your breath hoping ACID will sound any better. I DL'ed V6.0
>>and
>>>it sounds pretty awful too. I'm going to run it using Rewire in cubase
>SX
>>3
>>>and lightpipe it to Paris and see if it's the actual ACID audio engine
>or
>>
>>>the summing. I've got a feeling it's the audio engine..just a
>>>feeling......because I've also got Vegas here and it sucks too.
>>>
>>>;o)
>>>
>>>"TCB" <nobody@ishere.com> wrote in message news:45894947$1@linux...
>>>>
>>>> I'm not convinced I can hear any difference between native systems,
with
>>
>>>> the
>>>> exception of the last version of ACID I used (way back, I think v. 3)
>>
>>>> which
>>>> did sound truly awful. The real test on that one for me was the DAWSUM
>>CD
>>>> (which I purchased and dutifully scored because I was convinced 'summing'
>>>> was the real reason PARIS sounded so good) wherein I discovered that
>I
>>
>>>> could
>>>> barely tell one mix from the next even when hearing vastly different
>
>>>> systems.
>>>> Since then I am a skeptic, as opposed to a disbeliever, when I hear
that
>>>> one piece of software sounds greatly better, or even different, than
>
>>>> another.
>>>> I'm not saying some people can't tell some pieces of software from other
>>>> pieces of software, I'm just saying I'm skeptical one system is 'bright'
>>>> or 'sharp' or anything else until someone can produce statistically
>>>> meaningful
>>>> results in an ABY test.
>>>>
>>>> One of the great things about that DAWSUM CD is it has let me use the
>>
>>>> software
>>>> that I like the most, without worrying too much about the sound. That
>>
>>>> would
>>>> be Ableton Live most of the time, with SX as a backup if the editing
>gets
>>>> more intense. For me that alone was worth the time I spent on the DAWSUM
>>>> CD.
>>>>
>>>> TCB
>>>>
>>>> "LaMont" <jjdpro@ameritech.net> wrote:
>>>>>
>>>>>Thad, you really can't hear the difference?? Maybe I own too many software
>>>>>DAWS thru the years.
>>>>>
>>>>>Starting on Logic Audio 3.0, then to Cakwalk, Pro Tools, DP, Paris,Acid,
>>>>>Neundo, Sonar, samplitude..
>>>>>
>>>>>I can hear the diference with the same audio interface with the same
>wav
>>>>>file(s)as oon as I import the file or files.
>>>>>
>>>>>These days, depending on the genre I'm mising determines which DAW
>>>>>software
>>>>>I'll use.
>>>>>My circle of engineer and producer buddies all can hear the difference
>>in
>>>>>a second. Just the other day, we were mixing this R&B(ish)Gospel track
>>and
>>>>>somebody said, 'Mont, this is begging for Paris. Another track, the
call
>>>>>was for Pro Tools. And another,Nuendo..
>>>>>I know BrianT feels and hears the same way in different DAW software.
>>It's
>>>>>really obvious..
>>>>>
>>>>>
>>>>>"TCB" <nobody@ishere.com> wrote:
>>>>>>
>>>>>>Which is why ABY testing uses expert listeners instead of scopes and
>>
>>>>>>graphs.
>>>>>>
>>>>>>
>>>>>>I'm not saying you're wrong, esp. about ITB vs external summing. One
>>
>>>>>>would
>>>>>>expect that to sound at least slightly different. But I would be
>>>>>>absolutely
>>>>>>shocked if anyone could tell in a controlled ABY test whether they
were
>>>>>listening
>>>>>>to SX, Performer, of Sonar.
>>>>>>
>>>>>>"LaMont" <jjdpro@ameritech.net> wrote:
>>>>>>>
>>>>>>>The only real test is with the ears and not scopes and graphs.
>>>>>>>
>>>>>>>"TCB" <nobody@ishere.com> wrote:
>>>>>>>>
>>>>>>>>I'd like to see this proven in a controlled ABY test.
>>>>>>>>
>>>>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>>>>
>>>>>>>>>Hey Neil,
>>>>>>>>>
>>>>>>>>>All I'm saying is: All DAW software have their own unique sound.
>
>>>>>>>>>Despite
>>>>>>>>>what those lame summing test shows..
>>>>>>>>>
>>>>>>>>>PT-HD has a very distinct sound. A very polished sound, with a nice
>>>> top
>>>>>>>>end,
>>>>>>>>>but with full audio spectrum represented. Mixer/Summing buss can
>be
>>>> pushed,
>>>>>>>>>but you have to watch it.
>>>>>>>>>
>>>>>>>>>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype
>>the
>>>>>>>top
>>>>>>>>>nor bottom end.
>>>>>>>>>
>>>>>>>>>Logic Audio: Very Broad- Aggressive sound, that really works for
>Rock
>>>>>>and
>>>>>>>>>R & B/Gospel mixes.
>>>>>>>>>
>>>>>>>>>Digital Performer: With their hardware, superb audio quality. Full
>>
>>>>>>>>>bodied
>>>>>>>>>sound .
>>>>>>>>>
>>>>>>>>>Sonar: Very flat sounding. I would say that Sonar is your most vanilla
>>>>>>>sound
>>>>>>>>>DW on the market..
>>>>>>>>>
>>>>>>>>>Samplitude : A little less top end than Pro Tools. Full bodied 3d
>>
>>>>>>>>>sound..
>>>>>>>>>
>>>>>>>>>Paris: Dark sounding in comparison to the the other DAWs. But, has
>>a
>>>>>3d
>>>>>>>>sound
>>>>>>>>>quality that's full bodied.
>>>>>>>>>
>>>>>>>>>I feel that you asking SX to be something it's not with some analog
>>>> summing.
>>>>>>>>>Especialy for your genre of music..
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>
>>>>>>>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>>>>>>>
>>>>>>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>>>>>>
>>>>>>>>>>>"I'd disagree with you in this instance because I happen to think
>>>> the
>>>>>>>>Cubase
>>>>>>>>>>>ones DO sound better."
>>>>>>>>>>>
>>>>>>>>>>>Then that SSL Engineer does not know what they are doing with
board.
>>>>>>>There's
>>>>>>>>>>>no way a mix coming off of that board SSL should sound better
than
>>>>>a
>>>>>>>ITB
>>>>>>>>>>>Cubase SX mix..
>>>>>>>>>>>
>>>>>>>>>>>Sorry, that just does not jive. That engineer does not know how
>>to
>>>>>push
>>>>>>>>>>he
>>>>>>>>>>>SSL or just not familiar with it.
>>>>>>>>>>
>>>>>>>>>>You're not really paying attention, are you? It was the same
>>>>>>>>>>engineer (me). And as far as whether or not I know how to use
>>>>>>>>>>that particular board, I guess that would be a matter of
>>>>>>>>>>your opinion. I don't think the SSL mixes are bad ones, I think
>>>>>>>>>>they came out good; I just think that you can hear more detail
>>>>>>>>>>in the ITB mixes in the examples I gave, and they have more
>>>>>>>>>>wideband frequency content from top to bottom.
>>>>>>>>>>
>>>>>>>>>>Anyway, my point of that particular comparison wasn't to say
>>>>>>>>>>"ITB mixes are better than using a large-format console that
>>>>>>>>>>costs somewhere in the six-figure range", the point of it was to
>>>>>>>>>>address a signal-chain suggestion that Paul had... he had
>>>>>>>>>>suggested perhaps that I needed to pick up a few pieces of
>>>>>>>>>>killer vintage gear, and I was just demonstrating that I think
>>>>>>>>>>the various signal chain components that I have here are on par
>>>>>>>>>>with most anything that can be found in heavy-hitter studios...
>>>>>>>>>>we used probably around $100k's worth of mics & pre's on the
>>>>>>>>>>PTHD/SSL mixes, plus obviously you're looking at another
>>>>>>>>>>roughly $100k for that particular console (40-channel E-series,
>>>>>>>>>>black EQ's, w/G-series Computer & Total Recall package), add in
>>>>>>>>>>the PTHD, outboard gear & whatnot, and you end up with
>>>>>>>>>>somewhere around a quarter-mil's worth of equipment involved in
>>>>>>>>>>that project. The project done at my place was done with my
>>>>>>>>>>gear, which certainly doesn't tally up to anywhere remotely
>>>>>>>>>>close to that cost & none of it bears a "vintage" stamp, but it
>>>>>>>>>>sounds competitive with the project that used all the heavy-
>>>>>>>>>>hitter stuff.
>>>>>>>>>>
>>>>>>>>>>Neil
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>
>>>
>>
>"Mike R." <emarenot@yahoo.com> wrote in message news:4589a9b9@linux...
> Thanks Rod.
> Damn, I was living in this fantasy that somehow the aux returns would
> simply
> get "mixed in" when I made a submix.
> Cheers,
> MR
>
For a second I thought you were going to say you had a fantasy about ID
releasing a new version of PAris
Don
> "Rod Lincoln" <rlincoln@nosopam.kc.rr.com> wrote in message
> news:4589a8db$1@linux...
>>
>> If you want to print the verb you have to return it to it's own tracks.
> Instead
>> of connecting the returns to the aux, connect them to 2 inupts, just like
>> you were were recording something. Hit record, and let the song play out.
>> Rod
>> "Mike R." <emarenot@yahoo.com> wrote:
>> >
>> >
>> >I've been using ITB verbs and Paris verbs for most of my work. But
> recently
>> >I hung a TC reverb off on of Paris' external outs. I set up one of the
>> Aux
>> >send/returns to handle the TC. How do I record the reverberated audio
> into
>> >the submix? Does this happen "automatically" when I create an
>> >intelligent
>> >submix or do I have to return the verb to two of its own tracks? From
> what
>> >I've read there, the manual doesn't have a real clear answer to the
>> >question -though I could have missed something.
>> >Thanks
>> >MR
>> >
>> >
>> ><!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
>> ><HTML><HEAD>
>> ><META http-equiv=3DContent-Type content=3D"text/html; =
>> >charset=3Diso-8859-1">
>> ><META content=3D"MSHTML 6.00.2800.1555" name=3DGENERATOR>
>> ><STYLE></STYLE>
>> ></HEAD>
>> ><BODY bgColor=3D#ffffff>
>> ><DIV><FONT face=3DArial size=3D2>I've been using ITB verbs and Paris =
>> >verbs for most=20
>> >of my work. But recently I hung a TC reverb off on of Paris' =
>> >external=20
>> >outs. I set up one of the Aux send/returns to handle the TC. =
>> >How do=20
>> >I record the reverberated audio into the submix? Does this happen=20
>> >"automatically" when I create an intelligent submix or do I have to =
>> >return the=20
>> >verb to two of its own tracks? From what I've read there, the =
>> >manual=20
>> >doesn't have a real clear answer to the question -though I could have =
>> >missed=20
>> >something.</FONT></DIV>
>> ><DIV><FONT face=3DArial size=3D2>Thanks</FONT></DIV>
>> ><DIV><FONT face=3DArial size=3D2>MR</FONT></DIV></BODY></HTML>
>> >
>> >
>>
>
>This is a multi-part message in MIME format.
------=_NextPart_000_0064_01C7245E.7BDB16B0
Content-Type: text/plain;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
Thanks Chris.=20
This graphics stuff is confusing to me.
I have to convert the DVI to VGA because of the 20' length
as I understand. I was going to split it with a Y cable too.
They'll split to a 21"VGA and a 22"LCD. Will that work on=20
the far side of a KVM? I know the resolutions won't match
but I'll only need the VGAs when mixing so I'll adjust the=20
7300 differently for that application. Sooner or later I'll get
4 matching 22" LCDs and quad cards for both comps.
Will this work?
Tom
"Chris Ludwig" <chrisl@adkproaudio.com> wrote in message =
news:45894ef0$1@linux...
HI Tom,
These monitors will be fine with your card. The 7300GT supports that=20
resolution on both the VGA and DVI connections.I would run both with =
VGA=20
connections. Having one hooked up DVI and one VGA may make the 2 look=20
very different in brightness and such.
Chris
Tom Bruhl wrote:
> My card has one VGA, one DVI and one S Video connector on it.
>
>
> I choose Polesoft Lockspam to fight spam, and you?
> http://www.polesoft.com/refer.html =20
--=20
Chris Ludwig
ADK
chrisl@adkproaudio.com <mailto:chrisl@adkproaudio.com>
www.adkproaudio.com <http://www.adkproaudio.com/>
(859) 635-5762
------=_NextPart_000_0064_01C7245E.7BDB16B0
Content-Type: text/html;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1400" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Thanks Chris. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>This graphics stuff is confusing to=20
me.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>I have to convert the DVI to VGA =
because of the 20'=20
length</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>as I understand. I was going to =
split it with=20
a Y cable too.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>They'll split to </FONT><FONT =
face=3DArial size=3D2>a=20
21"VGA and a 22"LCD. Will that work on </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>the far side of a KVM? I know the =
resolutions=20
won't match</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>but I'll only need the VGAs when mixing =
so I'll=20
adjust the </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>7300 differently for that =
application. Sooner=20
or later I'll get</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>4 matching 22" LCDs and quad cards for =
both=20
comps.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Will this work?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Tom</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<BLOCKQUOTE=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Chris Ludwig" <<A=20
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A>> =
wrote in=20
message <A =
href=3D"news:45894ef0$1@linux">news:45894ef0$1@linux</A>...</DIV>HI=20
Tom,<BR>These monitors will be fine with your card. The 7300GT =
supports that=20
<BR>resolution on both the VGA and DVI connections.I would run both =
with VGA=20
<BR>connections. Having one hooked up DVI and one VGA may make the 2 =
look=20
<BR>very different in brightness and such.<BR>Chris<BR><BR><BR>Tom =
Bruhl=20
wrote:<BR><BR>> My card has one VGA, one DVI and one S Video =
connector on=20
it.<BR>><BR>><BR>> I choose Polesoft Lockspam to fight spam, =
and=20
you?<BR>> <A=20
=
href=3D"http://www.polesoft.com/refer.html">http://www.polesoft.com/refer=
..html</A> =20
<BR><BR><BR>-- <BR>Chris Ludwig<BR>ADK<BR><A=20
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A> =
<<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">mailto:chrisl@adkproaudio.com</A>&=
gt;<BR><A=20
href=3D"http://www.adkproaudio.com">www.adkproaudio.com</A> <<A=20
=
href=3D"http://www.adkproaudio.com/">http://www.adkproaudio.com/</A>><=
BR>(859)=20
635-5762</BLOCKQUOTE></BODY></HTML>
------=_NextPart_000_0064_01C7245E.7BDB16B0--Do you just want to keep some tracks around without hearing them in the
project?
Or describe what you're after, exactly and I'll see if I have any
suggestions...
Cheers,
-Jamie
www.JamieKrutz.com
George Axon wrote:
> I'm so used to putting things in jail in Paris, but haven't found an equivalent
> in Logic
> - is there one?
>
> George AxonThanks Mike :)
I'll keep you posted on our progress :)
Morgan wrote:
> Hi Mike,
>
> I have been asked to build another PARIS rig and would love
> to know more about your Dual core , what you did and how you got there.
> Please let us know how in every detail you were able to get this going.
> I will find a way to reward you :)
>
> Thanks Mike :)
> Morgan
> Eastcoast Music Mall
> morganp@ntplx.net
>
>
> Hi Mike,
>
> Congrats on your hard work . Today I was
> doing homework with my son Logan.
> His assignment was to define " Hero "
> One that acts with courage and bravery despite
> overwhelming odds.
>
> Mike - you are now My Hero !!
>
> Respectfully Yours ,
>
> Morgan
>
>
> Mike P wrote:
>
> > Ok, so it took ten days of total frustration, but I have succeeded!
> > My HP dual core is running Paris flawlessly, and it is awesome. This
> computer
> > is blazingly fast. I was processing files in wavelab that would have
> taken
> > 4-5 min on my old cpu, in 52secs. As I have faced every error known
> to man during this install, and fixed them,
> > if anyone has questions, please ask. Thanks to everyone who helped
> along the way.
>They will get mixed in. Rod was posting about if you want to print the verbs
to individual tracks. Lots of folks do that to be able to change the verb
level later.
Chuck
"Mike R." <emarenot@yahoo.com> wrote:
>Thanks Rod.
>Damn, I was living in this fantasy that somehow the aux returns would simply
>get "mixed in" when I made a submix.
>Cheers,
>MR
>
>"Rod Lincoln" <rlincoln@nosopam.kc.rr.com> wrote in message
>news:4589a8db$1@linux...
>>
>> If you want to print the verb you have to return it to it's own tracks.
>Instead
>> of connecting the returns to the aux, connect them to 2 inupts, just like
>> you were were recording something. Hit record, and let the song play out.
>> Rod
>> "Mike R." <emarenot@yahoo.com> wrote:
>> >
>> >
>> >I've been using ITB verbs and Paris verbs for most of my work. But
>recently
>> >I hung a TC reverb off on of Paris' external outs. I set up one of the
>> Aux
>> >send/returns to handle the TC. How do I record the reverberated audio
>into
>> >the submix? Does this happen "automatically" when I create an intelligent
>> >submix or do I have to return the verb to two of its own tracks? From
>what
>> >I've read there, the manual doesn't have a real clear answer to the
>> >question -though I could have missed something.
>> >Thanks
>> >MR
>> >
>> >
>> ><!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
>> ><HTML><HEAD>
>> ><META http-equiv=3DContent-Type content=3D"text/html; =
>> >charset=3Diso-8859-1">
>> ><META content=3D"MSHTML 6.00.2800.1555" name=3DGENERATOR>
>> ><STYLE></STYLE>
>> ></HEAD>
>> ><BODY bgColor=3D#ffffff>
>> ><DIV><FONT face=3DArial size=3D2>I've been using ITB verbs and Paris
=
>> >verbs for most=20
>> >of my work. But recently I hung a TC reverb off on of Paris' =
>> >external=20
>> >outs. I set up one of the Aux send/returns to handle the TC. =
>> >How do=20
>> >I record the reverberated audio into the submix? Does this happen=20
>> >"automatically" when I create an intelligent submix or do I have to =
>> >return the=20
>> >verb to two of its own tracks? From what I've read there, the =
>> >manual=20
>> >doesn't have a real clear answer to the question -though I could have
=
>> >missed=20
>> >something.</FONT></DIV>
>> ><DIV><FONT face=3DArial size=3D2>Thanks</FONT></DIV>
>> ><DIV><FONT face=3DArial size=3D2>MR</FONT></DIV></BODY></HTML>
>> >
>> >
>>
>
>"Mike R." <emarenot@yahoo.com> wrote:
>Damn, I was living in this fantasy that somehow the aux returns would simply
>get "mixed in" when I made a submix.
Anything on the Auxes that is active when you bounce or render will be
mixed in, just as it sounds on playback. If you want further control of
the verbs later in the mix process (EQs, levels etc) then you return the
verbs (sent via Aux) to 2 channels and record the input. Can't remember
what, if any, muting of other channels has to happen in order for you to
only receive back the wet verb signal...
TedNope, no jails in LogicPro. I just stash clumps of stuff after the song end,
though this is clearly not as easy as PARIS jails.
>I'm so used to putting things in jail in Paris, but haven't found an equivalent
>in Logic
>- is there one?
>
>George AxonSee Dedric, everthing is this life is not expalinable. Although we would like
it to be 2+2 =4, the rality is that sometimes 2+2=4.40..Why, becuase the
the math is flawed. Why is the math flawed? Becuase we as humans are flawed.
Say what you will about the metric system, which is a great tool.But, sometimes
working in inches and 16ths, 3/4s works better.
When a guy like Roger Nichols bangs his preverbial head around this issue
as to why his mix sound different being rendered from different and sometimes
the same cd mastering devices is expalinable, however the explanation does
not jive with the science.
Are we to believe that the Science we have today about digital audio is
the the Last word?? No.. In the future, some new science will come along
and either rebuff our current science or enhance it.
We I and other say.. We drop a stereo wav file in a given daw)(unity gain)
using the same audio converter...We can hear the diference. And it's sonically
obvious..
Lynns test is flawed because of the Roger Nicohls CD mastering problem. Things
change when going to render to CD.
Hey some peole on this earth can hear and see better than others..That's
just a fact
"Dedric Terry" <dedric@echomg.com> wrote:
>Of course Paris sounds different on Lynn's sampler, that was audible, and
>there are technical reasons why Paris will always sound different, but I
>didn't like it better on the sampler CD, to be honest, though the
>differences were subtle. Also, we weren't talking about Acid vs. Sonar
>specifically. I don't even bother with Acid as a DAW example - it's a loop
>app. Vegas is a video app that has had life as an audio app to some degree,
>but iMovie does audio as well, yet that doesn't really put it in the same
>category as professional DAW apps like Nuendo, PTHD, Sequoia, etc. I use
>Vegas for video, but not audio.
>
>On Lynn's sampler, Samplitude, Nuendo, Fairlight and the other natives don't
>sound different and aren't different in the unity gain examples
>(even the PTHD mix cancels with these). If you hear two files sounding
>differently that cancel to complete null, an audio difference isn't what
you
>are hearing. When there are differences in non-unity gain mix summing
>tests, you have an extra variable to account for - how is the gain
>calculated? Gain
>is non-linear (power), not adding two numbers together. So how is pan law
>factored in, and where? Are your faders exactly the same, or 0.001dB
>variant?
>
>Also if you drop the same stereo file in two different pro audio apps and
>hear a difference, one of the two apps is defective. There is nothing
>happening with a stereo file playback when no gain change or plugins are
>active - just audio streaming to the driver from disk. If you hear a
>difference there, I would be quickly trying to find out why. Something
is
>wrong.
>
>The point I am making is that these arguments usually come up as blanket
>statements with no qualification of what exactly sounds
>different, why it might, or solid well reasoned attempts to find out why,
or
>if there could be a real difference, or just a perceived one.
>
>Usually the "use your ears" comment comes up when there is no technical
>rebuttal for when the science and good
>ears agree. Of course "use your ears" first from a creative perspective,
>but if you are making a technical, scientific statement, then such comments
>aren't a good foundation to work from. It's a great motto, but a bit of
a
>cop out in a technical discussion.
>
>Regards,
>Dedric
>
>"LaMont" <jjdpro@ameriech.net> wrote in message news:45897f73$1@linux...
>>
>> Hey Dedric and Neil,
>>
>> I reason I think that the Summing CD test(good intentions) were lame was
>> because.. If a person can;t hear the difference btw a stereo wav file
>> that's
>> in Acid vs Sonar really needs a hearing test.
>>
>> For reason of my music work, I have to work with different DAWs, so I'm
>> very
>> familiar with their sound qualities. My circle of producers and engineers
>> talk about the daw sonics all the time. It's really no big deal anymore..
>>
>> The same logic applies when Roger Nichols (a few) years back in his
>> article
>> about master CD's and that he found out that 4 differnt CD burners yeilded
>> differnt sonic results. Sure, he sated that Math is the Math :) but, his
>> and the masering engineers Ears told them soemthing was different.
>> Hummm???
>>
>> Now, back to DAW sonics. I can hear the difference btw Paris and Nuendo
vs
>> Pro Tools, Logic audio.. There is no math to this, this is an ear
>> thing..You
>> either hear or you don't.. Simple.
>> But, good ears can hear it. .
>>
>> I really think the problem is, noone want to no that their money that
>> they've
>> spent on a given DAW, has sonic limitations or shall we say, just
>> different..
>>
>> I like that they all sound different. It's good to have choice when mixing
>> a song. Some DAWs, depending on the genre will yield better or the desired
>> results and than another.
>> EX. I would not mix a Acoustic jazz record today with Paris..reason, I'm
>> going for clarity at it's highest level.. For that project, It's either
>> Neundo
>> or Pro Tools and may Samplitude..Why should I fight with Paris's thick,
>> gooy
>> sonics, when I'm going for clarity. Well, Pro Tools and Nuendo/SX has
that
>> sound right out the gate.. Which makes my job a lot easier. simple. This
>> is not tosay that I could not get the job done in Paris..i could..But,
for
>> that Acoutic Jazz project , the other 2 daws gives me what I'm looking
>> for
>> without even touching an eq..
>>
>> This is not all about math. As BrianT states: Use you ears..Forget the
>> math..What
>> does knowing the math do for you anyway? Nothing, it just proves that
you
>> know the math. Does not tell you diddly about the sonics.. Just ask Roger
>> Nichols..
>>
>>
>> "Dedric Terry" <d@nospam.net> wrote:
>>>
>>>I know we disagree here Lamont and that's totally cool, so I won't take
>> this
>>>beyond this one response, and this isn't really directed to you, but my
>> general
>>>thoughts on the matter.
>>>
>>>In Neil's "defense" (not that he needs it), I and others have done this
>> comparison
>>>to death and the conclusion I've come to is that people are 80% influenced
>>>by a change in environment (e.g. software interface) and 20% ears. Sorry
>>>to say it, but the difference in sound between floating point DAWs is
far
>>>from real. It's just good, albeit unintentional marketing created by
>>>users
>>>and capitolized by manufacturers. Perceiving a "sound" in DAWs that in
>> actuality
>>>process data identically, is a bad reason to pick a DAW, but of course
>>>there
>>>is nothing wrong with thinking you hear a difference as long as it doesn't
>>>become an unwritten law of engineering at large. Preferring to work with
>>>one or the other, and "feeling" better about it for whatever reason is
a
>>>great reason to pick one DAW over another.
>>>
>>>There was a recent thread that Nuendo handled gain through groups
>>>differently,
>>>so I put Nuendo, Sonar 6 (both 32 and 64-bit engines) and Sequoia 8.3
to
>>>the test - identical tests, setup to the 1/100th of a dB identically and
>>>came up with absolutely no difference, either audible or scientific.
To
>>>be honest, this was the one test where I could have said, yes there is
an
>>>understandable difference between DAWs in a simple math function, and
the
>>>only one in the DAW that actually might make sense, yet even that did
not
>>>exist. The reason - math is math. You can paint it red, blue, silver
or
>>>dull grey, but it's still the same math unless the programmer was high
or
>>>completely incompetent when they wrote the code.
>>>
>>>I thought it was entirely possible the original poster had found something
>>>different in Nuendo, but when it came down to really understanding and
>>>reproducing
>>>what happens in DAW summing and gain structures accurately between each
>> DAW,
>>>there was none, nada, nil. The assertion was completely squashed. This
>>>also
>>>showed me how easy it is for a wide range of professionals to misinterpret
>>>digital audio - whether hearing things, or just setting up a test with
a
>>>single missed variable that completely invalidates the whole process.
>>>
>>>If you hear a difference, great. I've thought I heard a difference doing
>>>similar comparisons, then changed my perspective (nothing else - not
>>>converters,
>>>nothing - just reset my expectations, and switched back and forth) and
>>>could
>>>hear no difference.
>>>
>>>Just leave some room for other opinions when you post yours on this
>>>subject
>>>since it is very obvious that hearing is not as universally objective
and
>>>identically referenced as everyone might like to believe, and is highly
>> visually
>>>and environmentally affected. Some will hear differences in DAWs. There
>>>are Cubase SX 3 users claiming Cubase 4 sounds different. Sigh. Then
>>>they
>>>realize they aren't even using the same project... or at least different
>>>EQs, or etc, etc....
>>>
>>>Say what you want about published summing tests, but Lynn's tests are
as
>>>accurate as it gets, and that bears out in the results (all floating point
>>>DAWs cancel and sound identical - if you are hearing a difference, you
are
>>>hearing things that aren't there, or you forgot to align their gain and
>> placement).
>>> I've worked with Lynn at least briefly enough to know his attention to
>> detail.
>>> In the same way people will disagree about PCs and Macs until neither
>>> exists,
>>>so will audio engineers disagree about DAWs. This is one debate that
will
>>>always exist as long as we have different ears, eyes, brains,... and
>>>opinions.
>>>
>>>
>>>What Neil has done is to prove that opinions are always going to differ
>> (i.e.
>>>no consensus on the "best" mix of the ones posted). And in truth everyone
>>>has a different perception of sound in general - not everyone wants to
>>>hear
>>>things the same way, so we judge "best" from very different perspectives.
>>> There is no single gold standard. There are variations and mutated
>>> combinations,
>>>but all are subjective. That in and of itself implies very distinctly
>>>that
>>>people can and will even perceive the exact same sound differently if
>>>presented
>>>with any variable that changes the brain's interpretation, even if just
>> a
>>>visual distraction. Just change the lights in the room and see if you
>>>perceive
>>>a song differently played back exactly the same way. Or have a cat run
>> across
>>>a desk while listening. Whether you care to admit it or not, it is there,
>>>and that is actually the beauty of how our sense interact to create
>>>perception.
>>> That may be our undoing with DAW comparison tests, but it's also what
>>> keeps
>>>music fresh and creative, when we allow it to.
>>>
>>>So my suggestion is to use what makes you most creative, even if it's
just
>>>a "feeling" working with that DAW gives you - be it the workflow, the
GUI,
>>>or even the name brand reputation. But, as we all know, if you can't
make
>>>most any material sound good on whatever DAW you choose, the DAW isn't
the
>>>problem.
>>>
>>>Regards,
>>>Dedric
>>>
>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>
>>>>That's interesting - all those DAW sonic interpretations, I
>>>>mean... I haven't had a chance to usee all of those, so it's
>>>>good information.
>>>>
>>>>I still don't understand why you consider my summing
>>>>comparisons "lame", however - it was a fair set of tests;
>>>>the same mix summed in different ways. Not trying to prove a
>>>>point or to rig it so one sounded any better than the other - in
>>>>fact, if you recall the thread, different people liked different
>>>>summed versions for different reasons... there wasn't any one
>>>>version that stood out as being "the one" that everyone felt
>>>>sounded better. The only reason I didn't come right out & say
>>>>right away which version was which is so that I didn't bias
>>>>anyone's opinion beforehand by mentioning that... NOT to try
>>>>& "hide" anything or "trick" anyone, as you accused me of
>>>>
>>>>Sheesh!
>>>>
>>>>Neil
>>>>
>>>>
>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>
>>>>>Hey Neil,
>>>>>
>>>>>All I'm saying is: All DAW software have their own unique sound.
>>>>>Despite
>>>>>what those lame summing test shows..
>>>>>
>>>>>PT-HD has a very distinct sound. A very polished sound, with a nice
top
>>>>end,
>>>>>but with full audio spectrum represented. Mixer/Summing buss can be
>>>>>pushed,
>>>>>but you have to watch it.
>>>>>
>>>>>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype the
>>>top
>>>>>nor bottom end.
>>>>>
>>>>>Logic Audio: Very Broad- Aggressive sound, that really works for Rock
>> and
>>>>>R & B/Gospel mixes.
>>>>>
>>>>>Digital Performer: With their hardware, superb audio quality. Full
>>>>>bodied
>>>>>sound .
>>>>>
>>>>>Sonar: Very flat sounding. I would say that Sonar is your most vanilla
>>>sound
>>>>>DW on the market..
>>>>>
>>>>>Samplitude : A little less top end than Pro Tools. Full bodied 3d
>>>>>sound..
>>>>>
>>>>>Paris: Dark sounding in comparison to the the other DAWs. But, has a
3d
>>>>sound
>>>>>quality that's full bodied.
>>>>>
>>>>>I feel that you asking SX to be something it's not with some analog
>>>>>summing.
>>>>>Especialy for your genre of music..
>>>>>
>>>>>
>>>>>
>>>>>
>>>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>>>
>>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>>
>>>>>>>"I'd disagree with you in this instance because I happen to think
the
>>>>Cubase
>>>>>>>ones DO sound better."
>>>>>>>
>>>>>>>Then that SSL Engineer does not know what they are doing with board.
>>>There's
>>>>>>>no way a mix coming off of that board SSL should sound better than
a
>>>ITB
>>>>>>>Cubase SX mix..
>>>>>>>
>>>>>>>Sorry, that just does not jive. That engineer does not know how to
>>>>>>>push
>>>>>>he
>>>>>>>SSL or just not familiar with it.
>>>>>>
>>>>>>You're not really paying attention, are you? It was the same
>>>>>>engineer (me). And as far as whether or not I know how to use
>>>>>>that particular board, I guess that would be a matter of
>>>>>>your opinion. I don't think the SSL mixes are bad ones, I think
>>>>>>they came out good; I just think that you can hear more detail
>>>>>>in the ITB mixes in the examples I gave, and they have more
>>>>>>wideband frequency content from top to bottom.
>>>>>>
>>>>>>Anyway, my point of that particular comparison wasn't to say
>>>>>>"ITB mixes are better than using a large-format console that
>>>>>>costs somewhere in the six-figure range", the point of it was to
>>>>>>address a signal-chain suggestion that Paul had... he had
>>>>>>suggested perhaps that I needed to pick up a few pieces of
>>>>>>killer vintage gear, and I was just demonstrating that I think
>>>>>>the various signal chain components that I have here are on par
>>>>>>with most anything that can be found in heavy-hitter studios...
>>>>>>we used probably around $100k's worth of mics & pre's on the
>>>>>>PTHD/SSL mixes, plus obviously you're looking at another
>>>>>>roughly $100k for that particular console (40-channel E-series,
>>>>>>black EQ's, w/G-series Computer & Total Recall package), add in
>>>>>>the PTHD, outboard gear & whatnot, and you end up with
>>>>>>somewhere around a quarter-mil's worth of equipment involved in
>>>>>>that project. The project done at my place was done with my
>>>>>>gear, which certainly doesn't tally up to anywhere remotely
>>>>>>close to that cost & none of it bears a "vintage" stamp, but it
>>>>>>sounds competitive with the project that used all the heavy-
>>>>>>hitter stuff.
>>>>>>
>>>>>>Neil
>>>>>
>>>>
>>>
>>
>
>Lol!!! :)
"TCB" <nobody@ishere.com> wrote:
>
>Well, hey, at least we can agree that Live sounds good.
>
>I'm really psyched for my Live/Scope setup. The Core Duo desktop is set
up
>and running nicely (what a change from a three year old Athlon) so I'll
have
>gobs of native f/x and instruments, UAD-1 plugs, and Scope synths/effects/mixing.
>Then I will officially be 100% at fault if I suck ;-)
>
>TCB
>
>"LaMont" <jjdpro@gmail.com> wrote:
>>
>>Agreed..
>>"TCB" <nobody@ishere.com> wrote:
>>>
>>>That's too bad. I think people have an instinctive thing against the sound
>>>of Live as well, just because it also loops like ACID does. Live sounds
>>like
>>>a properly written native DAW when working with non time stretched tracks.
>>>The sound quality on the stretched audio is amazing, all things considered,
>>>but the non-stretched sound is indistinguishable from SX. Too bad the
only
>>>really truly awful sounding app has to bring down a perfectly nice sounding
>>>one.
>>>
>>>TCB
>>>
>>>"DJ" <nowayjose@dude.net> wrote:
>>>>>with the exception of the last version of ACID I used (way back, I think
>>>v.
>>>>>3) which did sound truly awful.<
>>>>Don't hold your breath hoping ACID will sound any better. I DL'ed V6.0
>>>and
>>>>it sounds pretty awful too. I'm going to run it using Rewire in cubase
>>SX
>>>3
>>>>and lightpipe it to Paris and see if it's the actual ACID audio engine
>>or
>>>
>>>>the summing. I've got a feeling it's the audio engine..just a
>>>>feeling......because I've also got Vegas here and it sucks too.
>>>>
>>>>;o)
>>>>
>>>>"TCB" <nobody@ishere.com> wrote in message news:45894947$1@linux...
>>>>>
>>>>> I'm not convinced I can hear any difference between native systems,
>with
>>>
>>>>> the
>>>>> exception of the last version of ACID I used (way back, I think v.
3)
>>>
>>>>> which
>>>>> did sound truly awful. The real test on that one for me was the DAWSUM
>>>CD
>>>>> (which I purchased and dutifully scored because I was convinced 'summing'
>>>>> was the real reason PARIS sounded so good) wherein I discovered that
>>I
>>>
>>>>> could
>>>>> barely tell one mix from the next even when hearing vastly different
>>
>>>>> systems.
>>>>> Since then I am a skeptic, as opposed to a disbeliever, when I hear
>that
>>>>> one piece of software sounds greatly better, or even different, than
>>
>>>>> another.
>>>>> I'm not saying some people can't tell some pieces of software from
other
>>>>> pieces of software, I'm just saying I'm skeptical one system is 'bright'
>>>>> or 'sharp' or anything else until someone can produce statistically
>
>>>>> meaningful
>>>>> results in an ABY test.
>>>>>
>>>>> One of the great things about that DAWSUM CD is it has let me use the
>>>
>>>>> software
>>>>> that I like the most, without worrying too much about the sound. That
>>>
>>>>> would
>>>>> be Ableton Live most of the time, with SX as a backup if the editing
>>gets
>>>>> more intense. For me that alone was worth the time I spent on the DAWSUM
>>>>> CD.
>>>>>
>>>>> TCB
>>>>>
>>>>> "LaMont" <jjdpro@ameritech.net> wrote:
>>>>>>
>>>>>>Thad, you really can't hear the difference?? Maybe I own too many software
>>>>>>DAWS thru the years.
>>>>>>
>>>>>>Starting on Logic Audio 3.0, then to Cakwalk, Pro Tools, DP, Paris,Acid,
>>>>>>Neundo, Sonar, samplitude..
>>>>>>
>>>>>>I can hear the diference with the same audio interface with the same
>>wav
>>>>>>file(s)as oon as I import the file or files.
>>>>>>
>>>>>>These days, depending on the genre I'm mising determines which DAW
>>>>>>software
>>>>>>I'll use.
>>>>>>My circle of engineer and producer buddies all can hear the difference
>>>in
>>>>>>a second. Just the other day, we were mixing this R&B(ish)Gospel track
>>>and
>>>>>>somebody said, 'Mont, this is begging for Paris. Another track, the
>call
>>>>>>was for Pro Tools. And another,Nuendo..
>>>>>>I know BrianT feels and hears the same way in different DAW software.
>>>It's
>>>>>>really obvious..
>>>>>>
>>>>>>
>>>>>>"TCB" <nobody@ishere.com> wrote:
>>>>>>>
>>>>>>>Which is why ABY testing uses expert listeners instead of scopes and
>>>
>>>>>>>graphs.
>>>>>>>
>>>>>>>
>>>>>>>I'm not saying you're wrong, esp. about ITB vs external summing. One
>>>
>>>>>>>would
>>>>>>>expect that to sound at least slightly different. But I would be
>>>>>>>absolutely
>>>>>>>shocked if anyone could tell in a controlled ABY test whether they
>were
>>>>>>listening
>>>>>>>to SX, Performer, of Sonar.
>>>>>>>
>>>>>>>"LaMont" <jjdpro@ameritech.net> wrote:
>>>>>>>>
>>>>>>>>The only real test is with the ears and not scopes and graphs.
>>>>>>>>
>>>>>>>>"TCB" <nobody@ishere.com> wrote:
>>>>>>>>>
>>>>>>>>>I'd like to see this proven in a controlled ABY test.
>>>>>>>>>
>>>>>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>>>>>
>>>>>>>>>>Hey Neil,
>>>>>>>>>>
>>>>>>>>>>All I'm saying is: All DAW software have their own unique sound.
>>
>>>>>>>>>>Despite
>>>>>>>>>>what those lame summing test shows..
>>>>>>>>>>
>>>>>>>>>>PT-HD has a very distinct sound. A very polished sound, with a
nice
>>>>> top
>>>>>>>>>end,
>>>>>>>>>>but with full audio spectrum represented. Mixer/Summing buss can
>>be
>>>>> pushed,
>>>>>>>>>>but you have to watch it.
>>>>>>>>>>
>>>>>>>>>>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype
>>>the
>>>>>>>>top
>>>>>>>>>>nor bottom end.
>>>>>>>>>>
>>>>>>>>>>Logic Audio: Very Broad- Aggressive sound, that really works for
>>Rock
>>>>>>>and
>>>>>>>>>>R & B/Gospel mixes.
>>>>>>>>>>
>>>>>>>>>>Digital Performer: With their hardware, superb audio quality. Full
>>>
>>>>>>>>>>bodied
>>>>>>>>>>sound .
>>>>>>>>>>
>>>>>>>>>>Sonar: Very flat sounding. I would say that Sonar is your most
vanilla
>>>>>>>>sound
>>>>>>>>>>DW on the market..
>>>>>>>>>>
>>>>>>>>>>Samplitude : A little less top end than Pro Tools. Full bodied
3d
>>>
>>>>>>>>>>sound..
>>>>>>>>>>
>>>>>>>>>>Paris: Dark sounding in comparison to the the other DAWs. But,
has
>>>a
>>>>>>3d
>>>>>>>>>sound
>>>>>>>>>>quality that's full bodied.
>>>>>>>>>>
>>>>>>>>>>I feel that you asking SX to be something it's not with some analog
>>>>> summing.
>>>>>>>>>>Especialy for your genre of music..
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>
>>>>>>>>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>>>>>>>>
>>>>>>>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>>>>>>>
>>>>>>>>>>>>"I'd disagree with you in this instance because I happen to think
>>>>> the
>>>>>>>>>Cubase
>>>>>>>>>>>>ones DO sound better."
>>>>>>>>>>>>
>>>>>>>>>>>>Then that SSL Engineer does not know what they are doing with
>board.
>>>>>>>>There's
>>>>>>>>>>>>no way a mix coming off of that board SSL should sound better
>than
>>>>>>a
>>>>>>>>ITB
>>>>>>>>>>>>Cubase SX mix..
>>>>>>>>>>>>
>>>>>>>>>>>>Sorry, that just does not jive. That engineer does not know how
>>>to
>>>>>>push
>>>>>>>>>>>he
>>>>>>>>>>>>SSL or just not familiar with it.
>>>>>>>>>>>
>>>>>>>>>>>You're not really paying attention, are you? It was the same
>>>>>>>>>>>engineer (me). And as far as whether or not I know how to use
>>>>>>>>>>>that particular board, I guess that would be a matter of
>>>>>>>>>>>your opinion. I don't think the SSL mixes are bad ones, I think
>>>>>>>>>>>they came out good; I just think that you can hear more detail
>>>>>>>>>>>in the ITB mixes in the examples I gave, and they have more
>>>>>>>>>>>wideband frequency content from top to bottom.
>>>>>>>>>>>
>>>>>>>>>>>Anyway, my point of that particular comparison wasn't to say
>>>>>>>>>>>"ITB mixes are better than using a large-format console that
>>>>>>>>>>>costs somewhere in the six-figure range", the point of it was
to
>>>>>>>>>>>address a signal-chain suggestion that Paul had... he had
>>>>>>>>>>>suggested perhaps that I needed to pick up a few pieces of
>>>>>>>>>>>killer vintage gear, and I was just demonstrating that I think
>>>>>>>>>>>the various signal chain components that I have here are on par
>>>>>>>>>>>with most anything that can be found in heavy-hitter studios...
>>>>>>>>>>>we used probably around $100k's worth of mics & pre's on the
>>>>>>>>>>>PTHD/SSL mixes, plus obviously you're looking at another
>>>>>>>>>>>roughly $100k for that particular console (40-channel E-series,
>>>>>>>>>>>black EQ's, w/G-series Computer & Total Recall package), add in
>>>>>>>>>>>the PTHD, outboard gear & whatnot, and you end up with
>>>>>>>>>>>somewhere around a quarter-mil's worth of equipment involved in
>>>>>>>>>>>that project. The project done at my place was done with my
>>>>>>>>>>>gear, which certainly doesn't tally up to anywhere remotely
>>>>>>>>>>>close to that cost & none of it bears a "vintage" stamp, but it
>>>>>>>>>>>sounds competitive with the project that used all the heavy-
>>>>>>>>>>>hitter stuff.
>>>>>>>>>>>
>>>>>>>>>>>Neil
>>>>>>>>>>
>>>>>>>>>
>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>
>>>>
>>>
>>
>I guess what I'm after is being able to take things ou of the audio window
and place
them back on the arrange page in the place where thyey originally came from.
To keep
track of vocal takes etc.Logic can have as many tracks as you like, you're not limited like PARIS
in regular mode. So there's plenty of room in the Arrange window even
for tracks you're not using at the moment. You can mute tracks and hide
tracks at any time. So I work in the Arrange window and don't need to
move tracks in and out. It's a big playground.
For comping takes, Logic works similarly to the PARIS freform mode but
without having to switch modes. You can use copies of the same track for
each take, cut the takes (regions) with the scissors tool and and mute
the sections you don't want to use with the mute tool (right click for
the tools).
You can use cycle record to have Logic automatically create new copies
of the same track as you record multiple takes. Time saver.
Cheers,
-Jamie
www.JamieKrutz.com
George Axon wrote:
> I guess what I'm after is being able to take things ou of the audio window
> and place
> them back on the arrange page in the place where thyey originally came from.
> To keep
> track of vocal takes etc.To settle this, we're gonna have to get you two in the same room with
multiple DAWs in a double blind test and see if you can hear a
difference between the same exact stereo file playing back through the
same monitoring chain from different DAWs.
Cheers,
-Jamie
www.JamieKrutz.com
LaMont wrote:
> See Dedric, everthing is this life is not expalinable. Although we would like
> it to be 2+2 =4, the rality is that sometimes 2+2=4.40..Why, becuase the
> the math is flawed. Why is the math flawed? Becuase we as humans are flawed.
> Say what you will about the metric system, which is a great tool.But, sometimes
> working in inches and 16ths, 3/4s works better.
>
> When a guy like Roger Nichols bangs his preverbial head around this issue
> as to why his mix sound different being rendered from different and sometimes
> the same cd mastering devices is expalinable, however the explanation does
> not jive with the science.
> Are we to believe that the Science we have today about digital audio is
> the the Last word?? No.. In the future, some new science will come along
> and either rebuff our current science or enhance it.
>
> We I and other say.. We drop a stereo wav file in a given daw)(unity gain)
> using the same audio converter...We can hear the diference. And it's sonically
> obvious..
>
> Lynns test is flawed because of the Roger Nicohls CD mastering problem. Things
> change when going to render to CD.
>
> Hey some peole on this earth can hear and see better than others..That's
> just a fact
>
> "Dedric Terry" <dedric@echomg.com> wrote:
>> Of course Paris sounds different on Lynn's sampler, that was audible, and
>
>> there are technical reasons why Paris will always sound different, but I
>
>> didn't like it better on the sampler CD, to be honest, though the
>> differences were subtle. Also, we weren't talking about Acid vs. Sonar
>
>> specifically. I don't even bother with Acid as a DAW example - it's a loop
>
>> app. Vegas is a video app that has had life as an audio app to some degree,
>
>> but iMovie does audio as well, yet that doesn't really put it in the same
>
>> category as professional DAW apps like Nuendo, PTHD, Sequoia, etc. I use
>
>> Vegas for video, but not audio.
>>
>> On Lynn's sampler, Samplitude, Nuendo, Fairlight and the other natives don't
>
>> sound different and aren't different in the unity gain examples
>> (even the PTHD mix cancels with these). If you hear two files sounding
>
>> differently that cancel to complete null, an audio difference isn't what
> you
>> are hearing. When there are differences in non-unity gain mix summing
>> tests, you have an extra variable to account for - how is the gain
>> calculated? Gain
>> is non-linear (power), not adding two numbers together. So how is pan law
>
>> factored in, and where? Are your faders exactly the same, or 0.001dB
>> variant?
>>
>> Also if you drop the same stereo file in two different pro audio apps and
>
>> hear a difference, one of the two apps is defective. There is nothing
>> happening with a stereo file playback when no gain change or plugins are
>
>> active - just audio streaming to the driver from disk. If you hear a
>> difference there, I would be quickly trying to find out why. Something
> is
>> wrong.
>>
>> The point I am making is that these arguments usually come up as blanket
>
>> statements with no qualification of what exactly sounds
>> different, why it might, or solid well reasoned attempts to find out why,
> or
>> if there could be a real difference, or just a perceived one.
>>
>> Usually the "use your ears" comment comes up when there is no technical
>
>> rebuttal for when the science and good
>> ears agree. Of course "use your ears" first from a creative perspective,
>
>> but if you are making a technical, scientific statement, then such comments
>> aren't a good foundation to work from. It's a great motto, but a bit of
> a
>> cop out in a technical discussion.
>>
>> Regards,
>> Dedric
>>
>> "LaMont" <jjdpro@ameriech.net> wrote in message news:45897f73$1@linux...
>>> Hey Dedric and Neil,
>>>
>>> I reason I think that the Summing CD test(good intentions) were lame was
>>> because.. If a person can;t hear the difference btw a stereo wav file
>
>>> that's
>>> in Acid vs Sonar really needs a hearing test.
>>>
>>> For reason of my music work, I have to work with different DAWs, so I'm
>
>>> very
>>> familiar with their sound qualities. My circle of producers and engineers
>>> talk about the daw sonics all the time. It's really no big deal anymore..
>>>
>>> The same logic applies when Roger Nichols (a few) years back in his
>>> article
>>> about master CD's and that he found out that 4 differnt CD burners yeilded
>>> differnt sonic results. Sure, he sated that Math is the Math :) but, his
>>> and the masering engineers Ears told them soemthing was different.
>>> Hummm???
>>>
>>> Now, back to DAW sonics. I can hear the difference btw Paris and Nuendo
> vs
>>> Pro Tools, Logic audio.. There is no math to this, this is an ear
>>> thing..You
>>> either hear or you don't.. Simple.
>>> But, good ears can hear it. .
>>>
>>> I really think the problem is, noone want to no that their money that
>
>>> they've
>>> spent on a given DAW, has sonic limitations or shall we say, just
>>> different..
>>>
>>> I like that they all sound different. It's good to have choice when mixing
>>> a song. Some DAWs, depending on the genre will yield better or the desired
>>> results and than another.
>>> EX. I would not mix a Acoustic jazz record today with Paris..reason, I'm
>>> going for clarity at it's highest level.. For that project, It's either
>
>>> Neundo
>>> or Pro Tools and may Samplitude..Why should I fight with Paris's thick,
>
>>> gooy
>>> sonics, when I'm going for clarity. Well, Pro Tools and Nuendo/SX has
> that
>>> sound right out the gate.. Which makes my job a lot easier. simple. This
>>> is not tosay that I could not get the job done in Paris..i could..But,
> for
>>> that Acoutic Jazz project , the other 2 daws gives me what I'm looking
>
>>> for
>>> without even touching an eq..
>>>
>>> This is not all about math. As BrianT states: Use you ears..Forget the
>
>>> math..What
>>> does knowing the math do for you anyway? Nothing, it just proves that
> you
>>> know the math. Does not tell you diddly about the sonics.. Just ask Roger
>>> Nichols..
>>>
>>>
>>> "Dedric Terry" <d@nospam.net> wrote:
>>>> I know we disagree here Lamont and that's totally cool, so I won't take
>>> this
>>>> beyond this one response, and this isn't really directed to you, but my
>>> general
>>>> thoughts on the matter.
>>>>
>>>> In Neil's "defense" (not that he needs it), I and others have done this
>>> comparison
>>>> to death and the conclusion I've come to is that people are 80% influenced
>>>> by a change in environment (e.g. software interface) and 20% ears. Sorry
>>>> to say it, but the difference in sound between floating point DAWs is
> far
>>> >from real. It's just good, albeit unintentional marketing created by
>
>>>> users
>>>> and capitolized by manufacturers. Perceiving a "sound" in DAWs that in
>>> actuality
>>>> process data identically, is a bad reason to pick a DAW, but of course
>
>>>> there
>>>> is nothing wrong with thinking you hear a difference as long as it doesn't
>>>> become an unwritten law of engineering at large. Preferring to work with
>>>> one or the other, and "feeling" better about it for whatever reason is
> a
>>>> great reason to pick one DAW over another.
>>>>
>>>> There was a recent thread that Nuendo handled gain through groups
>>>> differently,
>>>> so I put Nuendo, Sonar 6 (both 32 and 64-bit engines) and Sequoia 8.3
> to
>>>> the test - identical tests, setup to the 1/100th of a dB identically and
>>>> came up with absolutely no difference, either audible or scientific.
> To
>>>> be honest, this was the one test where I could have said, yes there is
> an
>>>> understandable difference between DAWs in a simple math function, and
> the
>>>> only one in the DAW that actually might make sense, yet even that did
> not
>>>> exist. The reason - math is math. You can paint it red, blue, silver
> or
>>>> dull grey, but it's still the same math unless the programmer was high
> or
>>>> completely incompetent when they wrote the code.
>>>>
>>>> I thought it was entirely possible the original poster had found something
>>>> different in Nuendo, but when it came down to really understanding and
>
>>>> reproducing
>>>> what happens in DAW summing and gain structures accurately between each
>>> DAW,
>>>> there was none, nada, nil. The assertion was completely squashed. This
>
>>>> also
>>>> showed me how easy it is for a wide range of professionals to misinterpret
>>>> digital audio - whether hearing things, or just setting up a test with
> a
>>>> single missed variable that completely invalidates the whole process.
>>>>
>>>> If you hear a difference, great. I've thought I heard a difference doing
>>>> similar comparisons, then changed my perspective (nothing else - not
>>>> converters,
>>>> nothing - just reset my expectations, and switched back and forth) and
>
>>>> could
>>>> hear no difference.
>>>>
>>>> Just leave some room for other opinions when you post yours on this
>>>> subject
>>>> since it is very obvious that hearing is not as universally objective
> and
>>>> identically referenced as everyone might like to believe, and is highly
>>> visually
>>>> and environmentally affected. Some will hear differences in DAWs. There
>>>> are Cubase SX 3 users claiming Cubase 4 sounds different. Sigh. Then
>
>>>> they
>>>> realize they aren't even using the same project... or at least different
>>>> EQs, or etc, etc....
>>>>
>>>> Say what you want about published summing tests, but Lynn's tests are
> as
>>>> accurate as it gets, and that bears out in the results (all floating point
>>>> DAWs cancel and sound identical - if you are hearing a difference, you
> are
>>>> hearing things that aren't there, or you forgot to align their gain and
>>> placement).
>>>> I've worked with Lynn at least briefly enough to know his attention to
>>> detail.
>>>> In the same way people will disagree about PCs and Macs until neither
>
>>>> exists,
>>>> so will audio engineers disagree about DAWs. This is one debate that
> will
>>>> always exist as long as we have different ears, eyes, brains,... and
>>>> opinions.
>>>>
>>>>
>>>> What Neil has done is to prove that opinions are always going to differ
>>> (i.e.
>>>> no consensus on the "best" mix of the ones posted). And in truth everyone
>>>> has a different perception of sound in general - not everyone wants to
>
>>>> hear
>>>> things the same way, so we judge "best" from very different perspectives.
>>>> There is no single gold standard. There are variations and mutated
>>>> combinations,
>>>> but all are subjective. That in and of itself implies very distinctly
>
>>>> that
>>>> people can and will even perceive the exact same sound differently if
>
>>>> presented
>>>> with any variable that changes the brain's interpretation, even if just
>>> a
>>>> visual distraction. Just change the lights in the room and see if you
>
>>>> perceive
>>>> a song differently played back exactly the same way. Or have a cat run
>>> across
>>>> a desk while listening. Whether you care to admit it or not, it is there,
>>>> and that is actually the beauty of how our sense interact to create
>>>> perception.
>>>> That may be our undoing with DAW comparison tests, but it's also what
>
>>>> keeps
>>>> music fresh and creative, when we allow it to.
>>>>
>>>> So my suggestion is to use what makes you most creative, even if it's
> just
>>>> a "feeling" working with that DAW gives you - be it the workflow, the
> GUI,
>>>> or even the name brand reputation. But, as we all know, if you can't
> make
>>>> most any material sound good on whatever DAW you choose, the DAW isn't
> the
>>>> problem.
>>>>
>>>> Regards,
>>>> Dedric
>>>>
>>>> "Neil" <IUOIU@OIU.com> wrote:
>>>>> That's interesting - all those DAW sonic interpretations, I
>>>>> mean... I haven't had a chance to usee all of those, so it's
>>>>> good information.
>>>>>
>>>>> I still don't understand why you consider my summing
>>>>> comparisons "lame", however - it was a fair set of tests;
>>>>> the same mix summed in different ways. Not trying to prove a
>>>>> point or to rig it so one sounded any better than the other - in
>>>>> fact, if you recall the thread, different people liked different
>>>>> summed versions for different reasons... there wasn't any one
>>>>> version that stood out as being "the one" that everyone felt
>>>>> sounded better. The only reason I didn't come right out & say
>>>>> right away which version was which is so that I didn't bias
>>>>> anyone's opinion beforehand by mentioning that... NOT to try
>>>>> & "hide" anything or "trick" anyone, as you accused me of
>>>>>
>>>>> Sheesh!
>>>>>
>>>>> Neil
>>>>>
>>>>>
>>>>> "Lamont" <jjdpro@ameritech.net> wrote:
>>>>>> Hey Neil,
>>>>>>
>>>>>> All I'm saying is: All DAW software have their own unique sound.
>>>>>> Despite
>>>>>> what those lame summing test shows..
>>>>>>
>>>>>> PT-HD has a very distinct sound. A very polished sound, with a nice
> top
>>>>> end,
>>>>>> but with full audio spectrum represented. Mixer/Summing buss can be
>
>>>>>> pushed,
>>>>>> but you have to watch it.
>>>>>>
>>>>>> Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype the
>>>> top
>>>>>> nor bottom end.
>>>>>>
>>>>>> Logic Audio: Very Broad- Aggressive sound, that really works for Rock
>>> and
>>>>>> R & B/Gospel mixes.
>>>>>>
>>>>>> Digital Performer: With their hardware, superb audio quality. Full
>>>>>> bodied
>>>>>> sound .
>>>>>>
>>>>>> Sonar: Very flat sounding. I would say that Sonar is your most vanilla
>>>> sound
>>>>>> DW on the market..
>>>>>>
>>>>>> Samplitude : A little less top end than Pro Tools. Full bodied 3d
>>>>>> sound..
>>>>>>
>>>>>> Paris: Dark sounding in comparison to the the other DAWs. But, has a
> 3d
>>>>> sound
>>>>>> quality that's full bodied.
>>>>>>
>>>>>> I feel that you asking SX to be something it's not with some analog
>
>>>>>> summing.
>>>>>> Especialy for your genre of music..
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>> "Neil" <IUOIU@OIU.com> wrote:
>>>>>>> "Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>>> "I'd disagree with you in this instance because I happen to think
> the
>>>>> Cubase
>>>>>>>> ones DO sound better."
>>>>>>>>
>>>>>>>> Then that SSL Engineer does not know what they are doing with board.
>>>> There's
>>>>>>>> no way a mix coming off of that board SSL should sound better than
> a
>>>> ITB
>>>>>>>> Cubase SX mix..
>>>>>>>>
>>>>>>>> Sorry, that just does not jive. That engineer does not know how to
>
>>>>>>>> push
>>>>>>> he
>>>>>>>> SSL or just not familiar with it.
>>>>>>> You're not really paying attention, are you? It was the same
>>>>>>> engineer (me). And as far as whether or not I know how to use
>>>>>>> that particular board, I guess that would be a matter of
>>>>>>> your opinion. I don't think the SSL mixes are bad ones, I think
>>>>>>> they came out good; I just think that you can hear more detail
>>>>>>> in the ITB mixes in the examples I gave, and they have more
>>>>>>> wideband frequency content from top to bottom.
>>>>>>>
>>>>>>> Anyway, my point of that particular comparison wasn't to say
>>>>>>> "ITB mixes are better than using a large-format console that
>>>>>>> costs somewhere in the six-figure range", the point of it was to
>>>>>>> address a signal-chain suggestion that Paul had... he had
>>>>>>> suggested perhaps that I needed to pick up a few pieces of
>>>>>>> killer vintage gear, and I was just demonstrating that I think
>>>>>>> the various signal chain components that I have here are on par
>>>>>>> with most anything that can be found in heavy-hitter studios...
>>>>>>> we used probably around $100k's worth of mics & pre's on the
>>>>>>> PTHD/SSL mixes, plus obviously you're looking at another
>>>>>>> roughly $100k for that particular console (40-channel E-series,
>>>>>>> black EQ's, w/G-series Computer & Total Recall package), add in
>>>>>>> the PTHD, outboard gear & whatnot, and you end up with
>>>>>>> somewhere around a quarter-mil's worth of equipment involved in
>>>>>>> that project. The project done at my place was done with my
>>>>>>> gear, which certainly doesn't tally up to anywhere remotely
>>>>>>> close to that cost & none of it bears a "vintage" stamp, but it
>>>>>>> sounds competitive with the project that used all the heavy-
>>>>>>> hitter stuff.
>>>>>>>
>>>>>>> Neil
>>
>"LaMont" <jjdpro@ameritech.net> wrote:
>
>Lynns test is flawed because of the Roger Nicohls CD mastering
>problem. Things change when going to render to CD.
I don't know that that makes those tests flawed - you're
talking about another step being inserted into the process.
The additional step of mastering presents a whole different
set of issues unto itself.... doesn't necessarily mean that
tests done on steps involved prior to mastering are
automatically invalid.
>Hey some peole on this earth can hear and see better than
>others..That's just a fact
Let's separate the esoteric from the mundane here... I, for
one, am not necessarily interested in only those who can hear
better, I'm interested in those with average hearing
and "untrained" ears, too. Do I have "good ears"? Yeah,
I guess. "Golden Ears"? Probably not. Yet, in the comparison
files I did, I think its all safe to say we were ALL reduced to
an equal footing with those of average hearing & "untrained
ears", simply because they were rendered to an mp3 format,
that's incapable of reproducing the frequency range & fidelity
of the original files... yet EVERYONE heard a difference
between one clip & another. Again, some liked one version or
another for different reasons, but my point is: there was
indeed a perceivable difference between them, even though we
were all reduced to a lesser set of capabilities, hearing-wise,
than if we all had the original wav files to listen to.
THIS is what I'm talking about - not something that only makes
a difference to the "Golden Ears" types, but something that
enhances the listening pleasure to everyone, even the average
Joe who may listen to it. I don't know that subtle differences
between one Native DAW & another - whether they really exist or
not - are going to make this kind of level of change in
what each contributes to a given mix. I've said before that
when it comes to tracking, it's all about the convertors, when
it comes to mixing, it's all about the summing... maybe that's
too broad of an empirical statement to be 100% accurate, but I
think there's a lot of truth to it.
I also don't know that whether two files "null" to 100% or not
is the only test of if "a" must therefore sound just
like "b"... let's face it, if something nulls, all it means is
that every peak is equal in amplitude... but what's going on
BELOW the peaks? What does it sound like at 200hz @ 5db down
from the peak at that frequency, for example? What is something
DOING to the sound at perhaps a lower (but still audible)
level, as opposed to at what amplitude is it outputting the
sound is something that a null test can't always address, IMO.
So I guess I've made some points supporting both sides of the
argument... fight on! lol
Neil"LaMont" <jjdpro@ameritech.net> wrote in message news:4589d1aa$1@linux...
>
> See Dedric, everthing is this life is not expalinable. Although we would
> like
> it to be 2+2 =4, the rality is that sometimes 2+2=4.40..Why, becuase the
> the math is flawed. Why is the math flawed? Becuase we as humans are
> flawed.
You can wax philosophical all you want, but I think you would find a strong
argument
with a lot of very knowledgeable engineers, programmers and mathematicians
on that -
my college professors in math, engineering and digital signal processing
being among those.
Saying that not everything is explanable is saying that
software is partly unpredictable and has a mind and behavior of its' own.
It doesn't in this case (no neural nets, or ai going on here). Sure, you
can program audio
processing differently when that is the goal, but you can easily determine
when that happens,
and when it doesn't.
I don't agree that just because a guy is well know that he has the last word
on the issue.
I also don't agree with your supposed assessment that we disagree because
"some people hear
better than others". You have no way of knowing what and how well I hear,
or how
well, or how baised and subjectively other engineers you quote might hear.
When I have
more time we'll put this to the test. I'll post something to listen to and
we'll find find out
what is heard and what isn't.
Also, come on, 2+2 is never 4.40. That's not even in the ballpark of being
a logical and reasonable
analogy. We might as well claim that blue sounds better than green, or the
sun only circles the earth
when the man in the moon is making cheddar cheese.
You also threw in Roger Nichols' CD mastering test as a reason to discount
Lynn's summing test,
but that has nothing to do with pulling files off of the *same* CD, hearing,
and watching them cancel
to null, so that is a poor arguement as well.
Lamont, I've read some very well respected engineers, RN being one of them,
claim some absolute
crap in this realm, and back it up with highly suspect comparisons. I'm also
not
the only one who has noticed this, but out of respect we hold our tongues
and shake our heads, quietly trying
to keep the rest of the audio community from succumbing to audio myths
rather than practicing
intelligent engineering. I don't know everything - far from it - but I do
know what I hear, and how to connect
it with what I see and what I do know.
Why do I use scientific comparisons such as phase cancellation and isolating
variables to make my comparisons?
To rule out perception, visual distraction, and preconceived ideas built by
reading newsgroups, articles
by "famous" engineers, and peer pressure. I don't disagree that DAWs can be
designed to sound differently, but
when they aren't, *and* we verify that they aren't, sorry, there is nothing
left to be different other than the GUI,
marketing and street hype.
Maybe someday I'll win a grammy or write a hit song for a ditzy pop starlet
so my opinion will carry some weight. ;-)
In the mean time, just don't believe everything you read, and only half of
what you hear.
Dedric
> Say what you will about the metric system, which is a great tool.But,
> sometimes
> working in inches and 16ths, 3/4s works better.
>
> When a guy like Roger Nichols bangs his preverbial head around this issue
> as to why his mix sound different being rendered from different and
> sometimes
> the same cd mastering devices is expalinable, however the explanation does
> not jive with the science.
> Are we to believe that the Science we have today about digital audio is
> the the Last word?? No.. In the future, some new science will come along
> and either rebuff our current science or enhance it.
>
> We I and other say.. We drop a stereo wav file in a given daw)(unity gain)
> using the same audio converter...We can hear the diference. And it's
> sonically
> obvious..
>
> Lynns test is flawed because of the Roger Nicohls CD mastering problem.
> Things
> change when going to render to CD.
>
> Hey some peole on this earth can hear and see better than others..That's
> just a fact
>
> "Dedric Terry" <dedric@echomg.com> wrote:
>>Of course Paris sounds different on Lynn's sampler, that was audible, and
>
>>there are technical reasons why Paris will always sound different, but I
>
>>didn't like it better on the sampler CD, to be honest, though the
>>differences were subtle. Also, we weren't talking about Acid vs. Sonar
>
>>specifically. I don't even bother with Acid as a DAW example - it's a
>>loop
>
>>app. Vegas is a video app that has had life as an audio app to some
>>degree,
>
>>but iMovie does audio as well, yet that doesn't really put it in the same
>
>>category as professional DAW apps like Nuendo, PTHD, Sequoia, etc. I use
>
>>Vegas for video, but not audio.
>>
>>On Lynn's sampler, Samplitude, Nuendo, Fairlight and the other natives
>>don't
>
>>sound different and aren't different in the unity gain examples
>>(even the PTHD mix cancels with these). If you hear two files sounding
>
>>differently that cancel to complete null, an audio difference isn't what
> you
>>are hearing. When there are differences in non-unity gain mix summing
>>tests, you have an extra variable to account for - how is the gain
>>calculated? Gain
>>is non-linear (power), not adding two numbers together. So how is pan law
>
>>factored in, and where? Are your faders exactly the same, or 0.001dB
>>variant?
>>
>>Also if you drop the same stereo file in two different pro audio apps and
>
>>hear a difference, one of the two apps is defective. There is nothing
>>happening with a stereo file playback when no gain change or plugins are
>
>>active - just audio streaming to the driver from disk. If you hear a
>>difference there, I would be quickly trying to find out why. Something
> is
>>wrong.
>>
>>The point I am making is that these arguments usually come up as blanket
>
>>statements with no qualification of what exactly sounds
>>different, why it might, or solid well reasoned attempts to find out why,
> or
>>if there could be a real difference, or just a perceived one.
>>
>>Usually the "use your ears" comment comes up when there is no technical
>
>>rebuttal for when the science and good
>>ears agree. Of course "use your ears" first from a creative perspective,
>
>>but if you are making a technical, scientific statement, then such
>>comments
>>aren't a good foundation to work from. It's a great motto, but a bit of
> a
>>cop out in a technical discussion.
>>
>>Regards,
>>Dedric
>>
>>"LaMont" <jjdpro@ameriech.net> wrote in message news:45897f73$1@linux...
>>>
>>> Hey Dedric and Neil,
>>>
>>> I reason I think that the Summing CD test(good intentions) were lame was
>>> because.. If a person can;t hear the difference btw a stereo wav file
>
>>> that's
>>> in Acid vs Sonar really needs a hearing test.
>>>
>>> For reason of my music work, I have to work with different DAWs, so I'm
>
>>> very
>>> familiar with their sound qualities. My circle of producers and
>>> engineers
>>> talk about the daw sonics all the time. It's really no big deal
>>> anymore..
>>>
>>> The same logic applies when Roger Nichols (a few) years back in his
>>> article
>>> about master CD's and that he found out that 4 differnt CD burners
>>> yeilded
>>> differnt sonic results. Sure, he sated that Math is the Math :) but, his
>>> and the masering engineers Ears told them soemthing was different.
>>> Hummm???
>>>
>>> Now, back to DAW sonics. I can hear the difference btw Paris and Nuendo
> vs
>>> Pro Tools, Logic audio.. There is no math to this, this is an ear
>>> thing..You
>>> either hear or you don't.. Simple.
>>> But, good ears can hear it. .
>>>
>>> I really think the problem is, noone want to no that their money that
>
>>> they've
>>> spent on a given DAW, has sonic limitations or shall we say, just
>>> different..
>>>
>>> I like that they all sound different. It's good to have choice when
>>> mixing
>>> a song. Some DAWs, depending on the genre will yield better or the
>>> desired
>>> results and than another.
>>> EX. I would not mix a Acoustic jazz record today with Paris..reason, I'm
>>> going for clarity at it's highest level.. For that project, It's either
>
>>> Neundo
>>> or Pro Tools and may Samplitude..Why should I fight with Paris's thick,
>
>>> gooy
>>> sonics, when I'm going for clarity. Well, Pro Tools and Nuendo/SX has
> that
>>> sound right out the gate.. Which makes my job a lot easier. simple. This
>>> is not tosay that I could not get the job done in Paris..i could..But,
> for
>>> that Acoutic Jazz project , the other 2 daws gives me what I'm looking
>
>>> for
>>> without even touching an eq..
>>>
>>> This is not all about math. As BrianT states: Use you ears..Forget the
>
>>> math..What
>>> does knowing the math do for you anyway? Nothing, it just proves that
> you
>>> know the math. Does not tell you diddly about the sonics.. Just ask
>>> Roger
>>> Nichols..
>>>
>>>
>>> "Dedric Terry" <d@nospam.net> wrote:
>>>>
>>>>I know we disagree here Lamont and that's totally cool, so I won't take
>>> this
>>>>beyond this one response, and this isn't really directed to you, but my
>>> general
>>>>thoughts on the matter.
>>>>
>>>>In Neil's "defense" (not that he needs it), I and others have done this
>>> comparison
>>>>to death and the conclusion I've come to is that people are 80%
>>>>influenced
>>>>by a change in environment (e.g. software interface) and 20% ears.
>>>>Sorry
>>>>to say it, but the difference in sound between floating point DAWs is
> far
>>>>from real. It's just good, albeit unintentional marketing created by
>
>>>>users
>>>>and capitolized by manufacturers. Perceiving a "sound" in DAWs that in
>>> actuality
>>>>process data identically, is a bad reason to pick a DAW, but of course
>
>>>>there
>>>>is nothing wrong with thinking you hear a difference as long as it
>>>>doesn't
>>>>become an unwritten law of engineering at large. Preferring to work
>>>>with
>>>>one or the other, and "feeling" better about it for whatever reason is
> a
>>>>great reason to pick one DAW over another.
>>>>
>>>>There was a recent thread that Nuendo handled gain through groups
>>>>differently,
>>>>so I put Nuendo, Sonar 6 (both 32 and 64-bit engines) and Sequoia 8.3
> to
>>>>the test - identical tests, setup to the 1/100th of a dB identically and
>>>>came up with absolutely no difference, either audible or scientific.
> To
>>>>be honest, this was the one test where I could have said, yes there is
> an
>>>>understandable difference between DAWs in a simple math function, and
> the
>>>>only one in the DAW that actually might make sense, yet even that did
> not
>>>>exist. The reason - math is math. You can paint it red, blue, silver
> or
>>>>dull grey, but it's still the same math unless the programmer was high
> or
>>>>completely incompetent when they wrote the code.
>>>>
>>>>I thought it was entirely possible the original poster had found
>>>>something
>>>>different in Nuendo, but when it came down to really understanding and
>
>>>>reproducing
>>>>what happens in DAW summing and gain structures accurately between each
>>> DAW,
>>>>there was none, nada, nil. The assertion was completely squashed. This
>
>>>>also
>>>>showed me how easy it is for a wide range of professionals to
>>>>misinterpret
>>>>digital audio - whether hearing things, or just setting up a test with
> a
>>>>single missed variable that completely invalidates the whole process.
>>>>
>>>>If you hear a difference, great. I've thought I heard a difference
>>>>doing
>>>>similar comparisons, then changed my perspective (nothing else - not
>>>>converters,
>>>>nothing - just reset my expectations, and switched back and forth) and
>
>>>>could
>>>>hear no difference.
>>>>
>>>>Just leave some room for other opinions when you post yours on this
>>>>subject
>>>>since it is very obvious that hearing is not as universally objective
> and
>>>>identically referenced as everyone might like to believe, and is highly
>>> visually
>>>>and environmentally affected. Some will hear differences in DAWs.
>>>>There
>>>>are Cubase SX 3 users claiming Cubase 4 sounds different. Sigh. Then
>
>>>>they
>>>>realize they aren't even using the same project... or at least different
>>>>EQs, or etc, etc....
>>>>
>>>>Say what you want about published summing tests, but Lynn's tests are
> as
>>>>accurate as it gets, and that bears out in the results (all floating
>>>>point
>>>>DAWs cancel and sound identical - if you are hearing a difference, you
> are
>>>>hearing things that aren't there, or you forgot to align their gain and
>>> placement).
>>>> I've worked with Lynn at least briefly enough to know his attention to
>>> detail.
>>>> In the same way people will disagree about PCs and Macs until neither
>
>>>> exists,
>>>>so will audio engineers disagree about DAWs. This is one debate that
> will
>>>>always exist as long as we have different ears, eyes, brains,... and
>>>>opinions.
>>>>
>>>>
>>>>What Neil has done is to prove that opinions are always going to differ
>>> (i.e.
>>>>no consensus on the "best" mix of the ones posted). And in truth
>>>>everyone
>>>>has a different perception of sound in general - not everyone wants to
>
>>>>hear
>>>>things the same way, so we judge "best" from very different
>>>>perspectives.
>>>> There is no single gold standard. There are variations and mutated
>>>> combinations,
>>>>but all are subjective. That in and of itself implies very distinctly
>
>>>>that
>>>>people can and will even perceive the exact same sound differently if
>
>>>>presented
>>>>with any variable that changes the brain's interpretation, even if just
>>> a
>>>>visual distraction. Just change the lights in the room and see if you
>
>>>>perceive
>>>>a song differently played back exactly the same way. Or have a cat run
>>> across
>>>>a desk while listening. Whether you care to admit it or not, it is
>>>>there,
>>>>and that is actually the beauty of how our sense interact to create
>>>>perception.
>>>> That may be our undoing with DAW comparison tests, but it's also what
>
>>>> keeps
>>>>music fresh and creative, when we allow it to.
>>>>
>>>>So my suggestion is to use what makes you most creative, even if it's
> just
>>>>a "feeling" working with that DAW gives you - be it the workflow, the
> GUI,
>>>>or even the name brand reputation. But, as we all know, if you can't
> make
>>>>most any material sound good on whatever DAW you choose, the DAW isn't
> the
>>>>problem.
>>>>
>>>>Regards,
>>>>Dedric
>>>>
>>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>>
>>>>>That's interesting - all those DAW sonic interpretations, I
>>>>>mean... I haven't had a chance to usee all of those, so it's
>>>>>good information.
>>>>>
>>>>>I still don't understand why you consider my summing
>>>>>comparisons "lame", however - it was a fair set of tests;
>>>>>the same mix summed in different ways. Not trying to prove a
>>>>>point or to rig it so one sounded any better than the other - in
>>>>>fact, if you recall the thread, different people liked different
>>>>>summed versions for different reasons... there wasn't any one
>>>>>version that stood out as being "the one" that everyone felt
>>>>>sounded better. The only reason I didn't come right out & say
>>>>>right away which version was which is so that I didn't bias
>>>>>anyone's opinion beforehand by mentioning that... NOT to try
>>>>>& "hide" anything or "trick" anyone, as you accused me of
>>>>>
>>>>>Sheesh!
>>>>>
>>>>>Neil
>>>>>
>>>>>
>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>
>>>>>>Hey Neil,
>>>>>>
>>>>>>All I'm saying is: All DAW software have their own unique sound.
>>>>>>Despite
>>>>>>what those lame summing test shows..
>>>>>>
>>>>>>PT-HD has a very distinct sound. A very polished sound, with a nice
> top
>>>>>end,
>>>>>>but with full audio spectrum represented. Mixer/Summing buss can be
>
>>>>>>pushed,
>>>>>>but you have to watch it.
>>>>>>
>>>>>>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype the
>>>>top
>>>>>>nor bottom end.
>>>>>>
>>>>>>Logic Audio: Very Broad- Aggressive sound, that really works for Rock
>>> and
>>>>>>R & B/Gospel mixes.
>>>>>>
>>>>>>Digital Performer: With their hardware, superb audio quality. Full
>>>>>>bodied
>>>>>>sound .
>>>>>>
>>>>>>Sonar: Very flat sounding. I would say that Sonar is your most vanilla
>>>>sound
>>>>>>DW on the market..
>>>>>>
>>>>>>Samplitude : A little less top end than Pro Tools. Full bodied 3d
>>>>>>sound..
>>>>>>
>>>>>>Paris: Dark sounding in comparison to the the other DAWs. But, has a
> 3d
>>>>>sound
>>>>>>quality that's full bodied.
>>>>>>
>>>>>>I feel that you asking SX to be something it's not with some analog
>
>>>>>>summing.
>>>>>>Especialy for your genre of music..
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>>>>
>>>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>>>
>>>>>>>>"I'd disagree with you in this instance because I happen to think
> the
>>>>>Cubase
>>>>>>>>ones DO sound better."
>>>>>>>>
>>>>>>>>Then that SSL Engineer does not know what they are doing with board.
>>>>There's
>>>>>>>>no way a mix coming off of that board SSL should sound better than
> a
>>>>ITB
>>>>>>>>Cubase SX mix..
>>>>>>>>
>>>>>>>>Sorry, that just does not jive. That engineer does not know how to
>
>>>>>>>>push
>>>>>>>he
>>>>>>>>SSL or just not familiar with it.
>>>>>>>
>>>>>>>You're not really paying attention, are you? It was the same
>>>>>>>engineer (me). And as far as whether or not I know how to use
>>>>>>>that particular board, I guess that would be a matter of
>>>>>>>your opinion. I don't think the SSL mixes are bad ones, I think
>>>>>>>they came out good; I just think that you can hear more detail
>>>>>>>in the ITB mixes in the examples I gave, and they have more
>>>>>>>wideband frequency content from top to bottom.
>>>>>>>
>>>>>>>Anyway, my point of that particular comparison wasn't to say
>>>>>>>"ITB mixes are better than using a large-format console that
>>>>>>>costs somewhere in the six-figure range", the point of it was to
>>>>>>>address a signal-chain suggestion that Paul had... he had
>>>>>>>suggested perhaps that I needed to pick up a few pieces of
>>>>>>>killer vintage gear, and I was just demonstrating that I think
>>>>>>>the various signal chain components that I have here are on par
>>>>>>>with most anything that can be found in heavy-hitter studios...
>>>>>>>we used probably around $100k's worth of mics & pre's on the
>>>>>>>PTHD/SSL mixes, plus obviously you're looking at another
>>>>>>>roughly $100k for that particular console (40-channel E-series,
>>>>>>>black EQ's, w/G-series Computer & Total Recall package), add in
>>>>>>>the PTHD, outboard gear & whatnot, and you end up with
>>>>>>>somewhere around a quarter-mil's worth of equipment involved in
>>>>>>>that project. The project done at my place was done with my
>>>>>>>gear, which certainly doesn't tally up to anywhere remotely
>>>>>>>close to that cost & none of it bears a "vintage" stamp, but it
>>>>>>>sounds competitive with the project that used all the heavy-
>>>>>>>hitter stuff.
>>>>>>>
>>>>>>>Neil
>>>>>>
>>>>>
>>>>
>>>
>>
>>
>"Neil" <IUOIU@OIU.com> wrote in message news:4589e206$1@linux...
> I also don't know that whether two files "null" to 100% or not
> is the only test of if "a" must therefore sound just
> like "b"... let's face it, if something nulls, all it means is
> that every peak is equal in amplitude... but what's going on
> BELOW the peaks? What does it sound like at 200hz @ 5db down
> from the peak at that frequency, for example? What is something
Neil, the digital signal is represented by more than peak amplitude in order
to
represent modulation, period and phase of the waveform, so phase
cancellation
has to compare more than peaks. Pull up a 44.1k, 24-bit file in Cool Edit
or Audition
and zoom - you'll see sample points all along the wave for anything below
20kHz.
All of those sample points have to cancel for a phase invert test to cancel
completely.
The point at which we lose any resolution is the last bit, and only when
that bit is change.
When comparing two copies of the same file, or identical files, neither with
any gain change,
with one inverted, we aren't changing any bits, unless the audio app is
severely flawed, so
even quantization noise is out. Even in tests in Nuendo changing the gain
of a file and making
up that gain later, and comparing it to a phase inverted copy of that file
shows that only
quantization noise below -144dB remains, and only a few peaks
reach -136dB (which is the point at which we see deviation in the last bit
due to truncation
in the gain change process).
Dedric
> DOING to the sound at perhaps a lower (but still audible)
> level, as opposed to at what amplitude is it outputting the
> sound is something that a null test can't always address, IMO.
>
> So I guess I've made some points supporting both sides of the
> argument... fight on! lol
>
> Neil"Dedric Terry" <dedric@echomg.com> wrote:
>
>"Neil" <IUOIU@OIU.com> wrote in message news:4589e206$1@linux...
>
>> I also don't know that whether two files "null" to 100% or not
>> is the only test of if "a" must therefore sound just
>> like "b"... let's face it, if something nulls, all it means is
>> that every peak is equal in amplitude... but what's going on
>> BELOW the peaks? What does it sound like at 200hz @ 5db down
>> from the peak at that frequency, for example? What is something
>
>Neil, the digital signal is represented by more than peak amplitude in order
>to
>represent modulation, period and phase of the waveform, so phase
>cancellation
>has to compare more than peaks. Pull up a 44.1k, 24-bit file in Cool Edit
>or Audition
>and zoom - you'll see sample points all along the wave for anything below
>20kHz.
>
>All of those sample points have to cancel for a phase invert test to cancel
>completely.
At 44,1000 sample points per second, I'll bet you can have
PLENNNNTY of samples that "miss" or don't cancel completely &
still get a "null".
What is the sound of one sample clapping?
How about two?
How about twenty seven samples, all roughly evenly-spaced
across the couse of a second... can you hear that if the
difference between those 27 samples is an average of couple of
db each?
How far does it go in # of samples per second & difference in
db for each one before you can hear it? We don't really know,
do we? Who's done that test? No one. Ever.
Neil>
> How far does it go in # of samples per second & difference in
> db for each one before you can hear it? We don't really know,
> do we? Who's done that test? No one. Ever.
Well, that's what the software is doing when adding a phase inverted file to
it's original - inverting
each word (24 or 16 bit) at each sample point in the audio file, and
comparing them to the same
sample position in the non-inverted file (e.g. one by one, 44,100 of them
every second).
So, yes, we've all done exactly that, everytime we do a phase invert test
and bounce the output to 32-bit
floating point, or watch it on a realtime analyzer set to read below -144dB,
which is the end of where 24-bit
can represent. A null is nothing showing up even below that level. If you
zoom in on the analyzer you will
see single sample peaks on a phase cancellation test difference file, so
even if one shows up, you can see it.
The tests I did were completely blank down to -200 dB (far below the last
bit). It's safe to say there is no difference, even in
quantization noise, which by technical rights, is considered below the level
of "cancellation" in such tests.
As far as the question of how much, or rather how little change can we
hear - that is certainly
been debated. But that's where the scientific side comes into play (when
our ears mislead us too easily).
If you cancel it in software, you probably aren't hearing a difference in
reality. If software math is that
unreliable, then you would never be able to recall a mix and run down a
"duplicate" - even without random
variables such as reverb or delay, it wouldn't sound even close to the same.
Regards,
Dedric
"Nei" <IUOIU@OIU.com> wrote in message news:4589f623$1@linux...
> At 44,1000 sample points per second, I'll bet you can have
> PLENNNNTY of samples that "miss" or don't cancel completely &
> still get a "null".
>
> What is the sound of one sample clapping?
>
> How about two?
>
> How about twenty seven samples, all roughly evenly-spaced
> across the couse of a second... can you hear that if the
> difference between those 27 samples is an average of couple of
> db each?
>
> How far does it go in # of samples per second & difference in
> db for each one before you can hear it? We don't really know,
> do we? Who's done that test? No one. Ever.
>
>
> NeilHey Deej,
What were the symptoms of your dying UAD-1 cards?
Thanks!
Gantt
"DJ" <nowayjose@dude.net> wrote:
>I have had two UAD-1 cards die on me since I bought my first one (which
>still works) in September 2001.
>"Gantt Kushner" <ganttmann@comcast.net> wrote:
>
>Hey Deej,
>
>What were the symptoms of your dying UAD-1 cards?
Pallor, gasping of breath, thready pulsewave modulation.
:DI'm sorry Dedric, but your statements below are full of hot Scientific hot
air balonga..
You should always trust your ears, and not some stupid math. This is music,
not a science project..Get the wax out and listen with your ears instead
of your scopes and graphs.. And, while your at it, how about acually working
with differnt DAW software to actually hear the difference. Your conclusions
that it's all "perception" is hoggwash..
And if you read RN's piece on CD mastering, you'd know he was stuck on math
as well, seeing as he's an Electical Engineer(EE).Software as a sound.
As far as the question of how much, or rather how little change can we hear
- that is certainly
been debated. But that's where the scientific side comes into play (when
our ears mislead us too easily).
If you cancel it in software, you probably aren't hearing a difference in
reality. If software math is that unreliable, then you would never be able
to recall a mix and run down a "duplicate" - even without random variables
such as reverb or delay, it wouldn't sound even close to the same.
"Dedric Terry" <dedric@echomg.com> wrote:
>>
>> How far does it go in # of samples per second & difference in
>> db for each one before you can hear it? We don't really know,
>> do we? Who's done that test? No one. Ever.
>
>Well, that's what the software is doing when adding a phase inverted file
to
>it's original - inverting
>each word (24 or 16 bit) at each sample point in the audio file, and
>comparing them to the same
>sample position in the non-inverted file (e.g. one by one, 44,100 of them
>every second).
>
>So, yes, we've all done exactly that, everytime we do a phase invert test
>and bounce the output to 32-bit
>floating point, or watch it on a realtime analyzer set to read below -144dB,
>which is the end of where 24-bit
>can represent. A null is nothing showing up even below that level. If
you
>zoom in on the analyzer you will
>see single sample peaks on a phase cancellation test difference file, so
>even if one shows up, you can see it.
>The tests I did were completely blank down to -200 dB (far below the last
>bit). It's safe to say there is no difference, even in
>quantization noise, which by technical rights, is considered below the level
>of "cancellation" in such tests.
>
>As far as the question of how much, or rather how little change can we
>hear - that is certainly
>been debated. But that's where the scientific side comes into play (when
>our ears mislead us too easily).
>If you cancel it in software, you probably aren't hearing a difference in
>reality. If software math is that
>unreliable, then you would never be able to recall a mix and run down a
>"duplicate" - even without random
>variables such as reverb or delay, it wouldn't sound even close to the same.
>
>Regards,
>Dedric
>
>"Nei" <IUOIU@OIU.com> wrote in message news:4589f623$1@linux...
>
>> At 44,1000 sample points per second, I'll bet you can have
>> PLENNNNTY of samples that "miss" or don't cancel completely &
>> still get a "null".
>>
>> What is the sound of one sample clapping?
>>
>> How about two?
>>
>> How about twenty seven samples, all roughly evenly-spaced
>> across the couse of a second... can you hear that if the
>> difference between those 27 samples is an average of couple of
>> db each?
>>
>> How far does it go in # of samples per second & difference in
>> db for each one before you can hear it? We don't really know,
>> do we? Who's done that test? No one. Ever.
>>
>>
>> Neil
>
>Lamont,
If you had read my post you would know that I do use my ears, and have used
pretty much every
DAW in this thread. You would also have read that I always recommend ears
first, and have
in several other threads. My ears are what I base my decisions on
exclusively. You obviously
don't understand the process of comparative testing or you would have
understood (as I stated) that the
"scopes and graphs" as you call them, come after the ears to confirm and/or
put a clearer face
on the "I think it sounds like this" perception. To be clear, since you
seem to be missing this point,
this "testing" is for the purpose of understanding what's behind the tools
we use, and sorting out
fact from myth - not for making mixing decisions.
You have offered nothing other than your opinion in broad, emotionally based
comments throughout this
thread with no specifics to back up your claim. You asserted a technical
claim with "use your ears" and
"software has a sound" instead of a firm grasp of what is actually being
discussed. I and others have proven
your blanket suppositions wrong in other situations several times, but it
isn't worth rehashing here given the
direction and lack of objectivity this discussion is taking. You hear what
you hear and that's fine, but I don't
appreciate your condescending tone here, so this will be my last post on
this topic here. Good luck!
Dedric
"LaMont" <jjdpro@gmail.com> wrote in message news:458a22bc$1@linux...
>
> I'm sorry Dedric, but your statements below are full of hot Scientific hot
> air balonga..
>
> You should always trust your ears, and not some stupid math. This is
> music,
> not a science project..Get the wax out and listen with your ears instead
> of your scopes and graphs.. And, while your at it, how about acually
> working
> with differnt DAW software to actually hear the difference. Your
> conclusions
> that it's all "perception" is hoggwash..
>
> And if you read RN's piece on CD mastering, you'd know he was stuck on
> math
> as well, seeing as he's an Electical Engineer(EE).Software as a sound.
>
>
> As far as the question of how much, or rather how little change can we
> hear
> - that is certainly
> been debated. But that's where the scientific side comes into play (when
> our ears mislead us too easily).
> If you cancel it in software, you probably aren't hearing a difference in
>
> reality. If software math is that unreliable, then you would never be able
> to recall a mix and run down a "duplicate" - even without random variables
> such as reverb or delay, it wouldn't sound even close to the same.
>
>
>
> "Dedric Terry" <dedric@echomg.com> wrote:
>>>
>>> How far does it go in # of samples per second & difference in
>>> db for each one before you can hear it? We don't really know,
>>> do we? Who's done that test? No one. Ever.
>>
>>Well, that's what the software is doing when adding a phase inverted file
> to
>>it's original - inverting
>>each word (24 or 16 bit) at each sample point in the audio file, and
>>comparing them to the same
>>sample position in the non-inverted file (e.g. one by one, 44,100 of them
>
>>every second).
>>
>>So, yes, we've all done exactly that, everytime we do a phase invert test
>
>>and bounce the output to 32-bit
>>floating point, or watch it on a realtime analyzer set to read
>>below -144dB,
>
>>which is the end of where 24-bit
>>can represent. A null is nothing showing up even below that level. If
> you
>>zoom in on the analyzer you will
>>see single sample peaks on a phase cancellation test difference file, so
>
>>even if one shows up, you can see it.
>>The tests I did were completely blank down to -200 dB (far below the last
>
>>bit). It's safe to say there is no difference, even in
>>quantization noise, which by technical rights, is considered below the
>>level
>
>>of "cancellation" in such tests.
>>
>>As far as the question of how much, or rather how little change can we
>>hear - that is certainly
>>been debated. But that's where the scientific side comes into play (when
>
>>our ears mislead us too easily).
>>If you cancel it in software, you probably aren't hearing a difference in
>
>>reality. If software math is that
>>unreliable, then you would never be able to recall a mix and run down a
>
>>"duplicate" - even without random
>>variables such as reverb or delay, it wouldn't sound even close to the
>>same.
>>
>>Regards,
>>Dedric
>>
>>"Nei" <IUOIU@OIU.com> wrote in message news:4589f623$1@linux...
>>
>>> At 44,1000 sample points per second, I'll bet you can have
>>> PLENNNNTY of samples that "miss" or don't cancel completely &
>>> still get a "null".
>>>
>>> What is the sound of one sample clapping?
>>>
>>> How about two?
>>>
>>> How about twenty seven samples, all roughly evenly-spaced
>>> across the couse of a second... can you hear that if the
>>> difference between those 27 samples is an average of couple of
>>> db each?
>>>
>>> How far does it go in # of samples per second & difference in
>>> db for each one before you can hear it? We don't really know,
>>> do we? Who's done that test? No one. Ever.
>>>
>>>
>>> Neil
>>
>>
> http://www.zdnetasia.com/news/hardware/0,39042972,61976567,0 0.htm
external drives with internal speeds !"Dedric Terry" <dedric@echomg.com> wrote:
>The tests I did were completely blank down to -200 dB (far below the last
>bit). It's safe to say there is no difference, even in
>quantization noise, which by technical rights, is considered below the level
>of "cancellation" in such tests.
I'm not necessarily talking about just the first bit or the
last bit, but also everything in between... what happens on bit
#12, for example? Everything on bit #12 should be audible, but
in an a/b test what if thre are differences in what bits #8
through #12 sound like, but the amplutide is stll the same on
both files at that point, you'll get a null, right? Extrapolate
that out somewhat & let's say there are differences in bits #8
through #12 on sample points 3, 17, 1,000, 4,523, 7,560, etc,
etc through 43,972... Now this is breaking things down well
beyond what I think can be measured, if I'm not mistaken (I
dn't know of any way we could extract JUST that information
from each file & play it back for an a/b test; but would not
that be enough to have to "null-able" files that do actually
sound somewhat different?
I guess what I'm saying is that since each sample in a musical
track or full song file doesn't represent a pure, simple set of
content like a sample of a sine wave would - there's a whole
world of harmonic structure in each sample of a song file, and
I think (although I'll admit - I can't "prove") that there is
plenty of room for some variables between the first bit & the
last bit while still allowing for a null test to be successful.
No? Am I wacked out of my mind?
NeilIf you think data can't be written consistently, down to the last bit, over
and over on a CD you better check to see that your Social Security Number,
bank balances, credit card transactions, and paychecks are changing. Because
they need to if that's what you believe.
TCB
"LaMont" <jjdpro@ameritech.net> wrote:
>
>See Dedric, everthing is this life is not expalinable. Although we would
like
>it to be 2+2 =4, the rality is that sometimes 2+2=4.40..Why, becuase the
>the math is flawed. Why is the math flawed? Becuase we as humans are flawed.
>Say what you will about the metric system, which is a great tool.But, sometimes
>working in inches and 16ths, 3/4s works better.
>
>When a guy like Roger Nichols bangs his preverbial head around this issue
>as to why his mix sound different being rendered from different and sometimes
>the same cd mastering devices is expalinable, however the explanation does
>not jive with the science.
>Are we to believe that the Science we have today about digital audio is
>the the Last word?? No.. In the future, some new science will come along
>and either rebuff our current science or enhance it.
>
>We I and other say.. We drop a stereo wav file in a given daw)(unity gain)
>using the same audio converter...We can hear the diference. And it's sonically
>obvious..
>
>Lynns test is flawed because of the Roger Nicohls CD mastering problem.
Things
>change when going to render to CD.
>
>Hey some peole on this earth can hear and see better than others..That's
>just a fact
>
>"Dedric Terry" <dedric@echomg.com> wrote:
>>Of course Paris sounds different on Lynn's sampler, that was audible, and
>
>>there are technical reasons why Paris will always sound different, but
I
>
>>didn't like it better on the sampler CD, to be honest, though the
>>differences were subtle. Also, we weren't talking about Acid vs. Sonar
>
>>specifically. I don't even bother with Acid as a DAW example - it's a
loop
>
>>app. Vegas is a video app that has had life as an audio app to some degree,
>
>>but iMovie does audio as well, yet that doesn't really put it in the same
>
>>category as professional DAW apps like Nuendo, PTHD, Sequoia, etc. I use
>
>>Vegas for video, but not audio.
>>
>>On Lynn's sampler, Samplitude, Nuendo, Fairlight and the other natives
don't
>
>>sound different and aren't different in the unity gain examples
>>(even the PTHD mix cancels with these). If you hear two files sounding
>
>>differently that cancel to complete null, an audio difference isn't what
>you
>>are hearing. When there are differences in non-unity gain mix summing
>>tests, you have an extra variable to account for - how is the gain
>>calculated? Gain
>>is non-linear (power), not adding two numbers together. So how is pan
law
>
>>factored in, and where? Are your faders exactly the same, or 0.001dB
>>variant?
>>
>>Also if you drop the same stereo file in two different pro audio apps and
>
>>hear a difference, one of the two apps is defective. There is nothing
>>happening with a stereo file playback when no gain change or plugins are
>
>>active - just audio streaming to the driver from disk. If you hear a
>>difference there, I would be quickly trying to find out why. Something
>is
>>wrong.
>>
>>The point I am making is that these arguments usually come up as blanket
>
>>statements with no qualification of what exactly sounds
>>different, why it might, or solid well reasoned attempts to find out why,
>or
>>if there could be a real difference, or just a perceived one.
>>
>>Usually the "use your ears" comment comes up when there is no technical
>
>>rebuttal for when the science and good
>>ears agree. Of course "use your ears" first from a creative perspective,
>
>>but if you are making a technical, scientific statement, then such comments
>>aren't a good foundation to work from. It's a great motto, but a bit of
>a
>>cop out in a technical discussion.
>>
>>Regards,
>>Dedric
>>
>>"LaMont" <jjdpro@ameriech.net> wrote in message news:45897f73$1@linux...
>>>
>>> Hey Dedric and Neil,
>>>
>>> I reason I think that the Summing CD test(good intentions) were lame
was
>>> because.. If a person can;t hear the difference btw a stereo wav file
>
>>> that's
>>> in Acid vs Sonar really needs a hearing test.
>>>
>>> For reason of my music work, I have to work with different DAWs, so I'm
>
>>> very
>>> familiar with their sound qualities. My circle of producers and engineers
>>> talk about the daw sonics all the time. It's really no big deal anymore..
>>>
>>> The same logic applies when Roger Nichols (a few) years back in his
>>> article
>>> about master CD's and that he found out that 4 differnt CD burners yeilded
>>> differnt sonic results. Sure, he sated that Math is the Math :) but,
his
>>> and the masering engineers Ears told them soemthing was different.
>>> Hummm???
>>>
>>> Now, back to DAW sonics. I can hear the difference btw Paris and Nuendo
>vs
>>> Pro Tools, Logic audio.. There is no math to this, this is an ear
>>> thing..You
>>> either hear or you don't.. Simple.
>>> But, good ears can hear it. .
>>>
>>> I really think the problem is, noone want to no that their money that
>
>>> they've
>>> spent on a given DAW, has sonic limitations or shall we say, just
>>> different..
>>>
>>> I like that they all sound different. It's good to have choice when mixing
>>> a song. Some DAWs, depending on the genre will yield better or the desired
>>> results and than another.
>>> EX. I would not mix a Acoustic jazz record today with Paris..reason,
I'm
>>> going for clarity at it's highest level.. For that project, It's either
>
>>> Neundo
>>> or Pro Tools and may Samplitude..Why should I fight with Paris's thick,
>
>>> gooy
>>> sonics, when I'm going for clarity. Well, Pro Tools and Nuendo/SX has
>that
>>> sound right out the gate.. Which makes my job a lot easier. simple. This
>>> is not tosay that I could not get the job done in Paris..i could..But,
>for
>>> that Acoutic Jazz project , the other 2 daws gives me what I'm looking
>
>>> for
>>> without even touching an eq..
>>>
>>> This is not all about math. As BrianT states: Use you ears..Forget the
>
>>> math..What
>>> does knowing the math do for you anyway? Nothing, it just proves that
>you
>>> know the math. Does not tell you diddly about the sonics.. Just ask Roger
>>> Nichols..
>>>
>>>
>>> "Dedric Terry" <d@nospam.net> wrote:
>>>>
>>>>I know we disagree here Lamont and that's totally cool, so I won't take
>>> this
>>>>beyond this one response, and this isn't really directed to you, but
my
>>> general
>>>>thoughts on the matter.
>>>>
>>>>In Neil's "defense" (not that he needs it), I and others have done this
>>> comparison
>>>>to death and the conclusion I've come to is that people are 80% influenced
>>>>by a change in environment (e.g. software interface) and 20% ears. Sorry
>>>>to say it, but the difference in sound between floating point DAWs is
>far
>>>>from real. It's just good, albeit unintentional marketing created by
>
>>>>users
>>>>and capitolized by manufacturers. Perceiving a "sound" in DAWs that
in
>>> actuality
>>>>process data identically, is a bad reason to pick a DAW, but of course
>
>>>>there
>>>>is nothing wrong with thinking you hear a difference as long as it doesn't
>>>>become an unwritten law of engineering at large. Preferring to work
with
>>>>one or the other, and "feeling" better about it for whatever reason is
>a
>>>>great reason to pick one DAW over another.
>>>>
>>>>There was a recent thread that Nuendo handled gain through groups
>>>>differently,
>>>>so I put Nuendo, Sonar 6 (both 32 and 64-bit engines) and Sequoia 8.3
>to
>>>>the test - identical tests, setup to the 1/100th of a dB identically
and
>>>>came up with absolutely no difference, either audible or scientific.
>To
>>>>be honest, this was the one test where I could have said, yes there is
>an
>>>>understandable difference between DAWs in a simple math function, and
>the
>>>>only one in the DAW that actually might make sense, yet even that did
>not
>>>>exist. The reason - math is math. You can paint it red, blue, silver
>or
>>>>dull grey, but it's still the same math unless the programmer was high
>or
>>>>completely incompetent when they wrote the code.
>>>>
>>>>I thought it was entirely possible the original poster had found something
>>>>different in Nuendo, but when it came down to really understanding and
>
>>>>reproducing
>>>>what happens in DAW summing and gain structures accurately between each
>>> DAW,
>>>>there was none, nada, nil. The assertion was completely squashed. This
>
>>>>also
>>>>showed me how easy it is for a wide range of professionals to misinterpret
>>>>digital audio - whether hearing things, or just setting up a test with
>a
>>>>single missed variable that completely invalidates the whole process.
>>>>
>>>>If you hear a difference, great. I've thought I heard a difference doing
>>>>similar comparisons, then changed my perspective (nothing else - not
>>>>converters,
>>>>nothing - just reset my expectations, and switched back and forth) and
>
>>>>could
>>>>hear no difference.
>>>>
>>>>Just leave some room for other opinions when you post yours on this
>>>>subject
>>>>since it is very obvious that hearing is not as universally objective
>and
>>>>identically referenced as everyone might like to believe, and is highly
>>> visually
>>>>and environmentally affected. Some will hear differences in DAWs. There
>>>>are Cubase SX 3 users claiming Cubase 4 sounds different. Sigh. Then
>
>>>>they
>>>>realize they aren't even using the same project... or at least different
>>>>EQs, or etc, etc....
>>>>
>>>>Say what you want about published summing tests, but Lynn's tests are
>as
>>>>accurate as it gets, and that bears out in the results (all floating
point
>>>>DAWs cancel and sound identical - if you are hearing a difference, you
>are
>>>>hearing things that aren't there, or you forgot to align their gain and
>>> placement).
>>>> I've worked with Lynn at least briefly enough to know his attention
to
>>> detail.
>>>> In the same way people will disagree about PCs and Macs until neither
>
>>>> exists,
>>>>so will audio engineers disagree about DAWs. This is one debate that
>will
>>>>always exist as long as we have different ears, eyes, brains,... and
>>>>opinions.
>>>>
>>>>
>>>>What Neil has done is to prove that opinions are always going to differ
>>> (i.e.
>>>>no consensus on the "best" mix of the ones posted). And in truth everyone
>>>>has a different perception of sound in general - not everyone wants to
>
>>>>hear
>>>>things the same way, so we judge "best" from very different perspectives.
>>>> There is no single gold standard. There are variations and mutated
>>>> combinations,
>>>>but all are subjective. That in and of itself implies very distinctly
>
>>>>that
>>>>people can and will even perceive the exact same sound differently if
>
>>>>presented
>>>>with any variable that changes the brain's interpretation, even if just
>>> a
>>>>visual distraction. Just change the lights in the room and see if you
>
>>>>perceive
>>>>a song differently played back exactly the same way. Or have a cat run
>>> across
>>>>a desk while listening. Whether you care to admit it or not, it is there,
>>>>and that is actually the beauty of how our sense interact to create
>>>>perception.
>>>> That may be our undoing with DAW comparison tests, but it's also what
>
>>>> keeps
>>>>music fresh and creative, when we allow it to.
>>>>
>>>>So my suggestion is to use what makes you most creative, even if it's
>just
>>>>a "feeling" working with that DAW gives you - be it the workflow, the
>GUI,
>>>>or even the name brand reputation. But, as we all know, if you can't
>make
>>>>most any material sound good on whatever DAW you choose, the DAW isn't
>the
>>>>problem.
>>>>
>>>>Regards,
>>>>Dedric
>>>>
>>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>>
>>>>>That's interesting - all those DAW sonic interpretations, I
>>>>>mean... I haven't had a chance to usee all of those, so it's
>>>>>good information.
>>>>>
>>>>>I still don't understand why you consider my summing
>>>>>comparisons "lame", however - it was a fair set of tests;
>>>>>the same mix summed in different ways. Not trying to prove a
>>>>>point or to rig it so one sounded any better than the other - in
>>>>>fact, if you recall the thread, different people liked different
>>>>>summed versions for different reasons... there wasn't any one
>>>>>version that stood out as being "the one" that everyone felt
>>>>>sounded better. The only reason I didn't come right out & say
>>>>>right away which version was which is so that I didn't bias
>>>>>anyone's opinion beforehand by mentioning that... NOT to try
>>>>>& "hide" anything or "trick" anyone, as you accused me of
>>>>>
>>>>>Sheesh!
>>>>>
>>>>>Neil
>>>>>
>>>>>
>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>
>>>>>>Hey Neil,
>>>>>>
>>>>>>All I'm saying is: All DAW software have their own unique sound.
>>>>>>Despite
>>>>>>what those lame summing test shows..
>>>>>>
>>>>>>PT-HD has a very distinct sound. A very polished sound, with a nice
>top
>>>>>end,
>>>>>>but with full audio spectrum represented. Mixer/Summing buss can be
>
>>>>>>pushed,
>>>>>>but you have to watch it.
>>>>>>
>>>>>>Nuendo/SX: Has a very Clear, 2 dimension sound, that does not hype
the
>>>>top
>>>>>>nor bottom end.
>>>>>>
>>>>>>Logic Audio: Very Broad- Aggressive sound, that really works for Rock
>>> and
>>>>>>R & B/Gospel mixes.
>>>>>>
>>>>>>Digital Performer: With their hardware, superb audio quality. Full
>>>>>>bodied
>>>>>>sound .
>>>>>>
>>>>>>Sonar: Very flat sounding. I would say that Sonar is your most vanilla
>>>>sound
>>>>>>DW on the market..
>>>>>>
>>>>>>Samplitude : A little less top end than Pro Tools. Full bodied 3d
>>>>>>sound..
>>>>>>
>>>>>>Paris: Dark sounding in comparison to the the other DAWs. But, has
a
>3d
>>>>>sound
>>>>>>quality that's full bodied.
>>>>>>
>>>>>>I feel that you asking SX to be something it's not with some analog
>
>>>>>>summing.
>>>>>>Especialy for your genre of music..
>>>>>>
>>>>>>
>>>>>>
>>>>>>
>>>>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>>>>
>>>>>>>"Lamont" <jjdpro@ameritech.net> wrote:
>>>>>>>>
>>>>>>>>"I'd disagree with you in this instance because I happen to think
>the
>>>>>Cubase
>>>>>>>>ones DO sound better."
>>>>>>>>
>>>>>>>>Then that SSL Engineer does not know what they are doing with board.
>>>>There's
>>>>>>>>no way a mix coming off of that board SSL should sound better than
>a
>>>>ITB
>>>>>>>>Cubase SX mix..
>>>>>>>>
>>>>>>>>Sorry, that just does not jive. That engineer does not know how to
>
>>>>>>>>push
>>>>>>>he
>>>>>>>>SSL or just not familiar with it.
>>>>>>>
>>>>>>>You're not really paying attention, are you? It was the same
>>>>>>>engineer (me). And as far as whether or not I know how to use
>>>>>>>that particular board, I guess that would be a matter of
>>>>>>>your opinion. I don't think the SSL mixes are bad ones, I think
>>>>>>>they came out good; I just think that you can hear more detail
>>>>>>>in the ITB mixes in the examples I gave, and they have more
>>>>>>>wideband frequency content from top to bottom.
>>>>>>>
>>>>>>>Anyway, my point of that particular comparison wasn't to say
>>>>>>>"ITB mixes are better than using a large-format console that
>>>>>>>costs somewhere in the six-figure range", the point of it was to
>>>>>>>address a signal-chain suggestion that Paul had... he had
>>>>>>>suggested perhaps that I needed to pick up a few pieces of
>>>>>>>killer vintage gear, and I was just demonstrating that I think
>>>>>>>the various signal chain components that I have here are on par
>>>>>>>with most anything that can be found in heavy-hitter studios...
>>>>>>>we used probably around $100k's worth of mics & pre's on the
>>>>>>>PTHD/SSL mixes, plus obviously you're looking at another
>>>>>>>roughly $100k for that particular console (40-channel E-series,
>>>>>>>black EQ's, w/G-series Computer & Total Recall package), add in
>>>>>>>the PTHD, outboard gear & whatnot, and you end up with
>>>>>>>somewhere around a quarter-mil's worth of equipment involved in
>>>>>>>that project. The project done at my place was done with my
>>>>>>>gear, which certainly doesn't tally up to anywhere remotely
>>>>>>>close to that cost & none of it bears a "vintage" stamp, but it
>>>>>>>sounds competitive with the project that used all the heavy-
>>>>>>>hitter stuff.
>>>>>>>
>>>>>>>Neil
>>>>>>
>>>>>
>>>>
>>>
>>
>>
>HI John,
They have been available for months now. SIIG and Adaptec both make
express and PCMCIA cards and people like Granite Digital, Lacie all make
drives and enclosures with e-sata. One of the main motherboards we use
has an E-Sata port on it. Having the ability to do 2 60mb/s+ 400GB
drives on a laptop is quite nice. Some of the controller will allow for
Raid 1and 0 setups too. They aren't much more than Firewire/USB options.
Chris
John wrote:
> http://www.zdnetasia.com/news/hardware/0,39042972,61976567,0 0.htm
>
>external drives with internal speeds !
>
>
--
Chris Ludwig
ADK
chrisl@adkproaudio.com <mailto:chrisl@adkproaudio.com>
www.adkproaudio.com <http://www.adkproaudio.com/>
(859) 635-5762One of them created annoying bursts of noise in the monitors so I had to
fire it. The other one just didn't show up for work one day.
;o)
"Neil" <IUOIU@OIU.com> wrote in message news:458a08d6$1@linux...
>
> "Gantt Kushner" <ganttmann@comcast.net> wrote:
>>
>>Hey Deej,
>>
>>What were the symptoms of your dying UAD-1 cards?
>
> Pallor, gasping of breath, thready pulsewave modulation.
>
> :D
>Because digital audio is simply individual amplitude samples taken at
regular increments, if two tracks (sets of sample numbers) cancel out
completely they are, by definition, identical at every sample point.
Simple as that.
The music we hear is recreated from the sample point numbers when they
are converted back to analog. As we know, sound carried through the air
is an analog phenomenon.
The world of harmonic structure we hear comes from the combination of
waveforms at different frequencies used in the music. The ability of a
series of simple amplitude samples to accurately recreate such combined
frequency information is determined by the frequency of the regular
increments - the sample rate - how often an amplitude measurement is
recorded.
Nyquist suggests using at least 2x the highest frequency you want to
reproduce. But as long as the systems you are comparing are using the
same sample rate, that part of the equation is removed as a variable.
The quality of the A to D and D to A converters plays a part in what we
hear, but those stages can be removed as variables depending on what and
how you test - for example by comparing the same digital file bounced
digitally from different DAWs.
Cheers,
-Jamie
www.JamieKrutz.com
Neil wrote:
> "Dedric Terry" <dedric@echomg.com> wrote:
>> The tests I did were completely blank down to -200 dB (far below the last
>
>> bit). It's safe to say there is no difference, even in
>> quantization noise, which by technical rights, is considered below the level
>
>> of "cancellation" in such tests.
>
> I'm not necessarily talking about just the first bit or the
> last bit, but also everything in between... what happens on bit
> #12, for example? Everything on bit #12 should be audible, but
> in an a/b test what if thre are differences in what bits #8
> through #12 sound like, but the amplutide is stll the same on
> both files at that point, you'll get a null, right? Extrapolate
> that out somewhat & let's say there are differences in bits #8
> through #12 on sample points 3, 17, 1,000, 4,523, 7,560, etc,
> etc through 43,972... Now this is breaking things down well
> beyond what I think can be measured, if I'm not mistaken (I
> dn't know of any way we could extract JUST that information
> from each file & play it back for an a/b test; but would not
> that be enough to have to "null-able" files that do actually
> sound somewhat different?
>
> I guess what I'm saying is that since each sample in a musical
> track or full song file doesn't represent a pure, simple set of
> content like a sample of a sine wave would - there's a whole
> world of harmonic structure in each sample of a song file, and
> I think (although I'll admit - I can't "prove") that there is
> plenty of room for some variables between the first bit & the
> last bit while still allowing for a null test to be successful.
>
> No? Am I wacked out of my mind?
>
> Neil
>
>http://www.nbc.com/Saturday_Night_Live/uncensored.shtmlAnnoying bursts of noise is exactly what's happening! How did you figure
out whci card was bad?
Gantt
"DJ" <nowayjose@dude.net> wrote:
>One of them created annoying bursts of noise in the monitors so I had to
>fire it. The other one just didn't show up for work one day.
>
>;o)
>
>
>"Neil" <IUOIU@OIU.com> wrote in message news:458a08d6$1@linux...
>>
>> "Gantt Kushner" <ganttmann@comcast.net> wrote:
>>>
>>>Hey Deej,
>>>
>>>What were the symptoms of your dying UAD-1 cards?
>>
>> Pallor, gasping of breath, thready pulsewave modulation.
>>
>> :D
>>
>
>Sounds like Justin is trying to toughen up his image a bit...or trying to be
relevant
DOn
"James McCloskey" <excelsm@hotmail.com> wrote in message
news:458aefa2$1@linux...
>
> http://www.nbc.com/Saturday_Night_Live/uncensored.shtmlI'm always the last to know. :-(
Chris Ludwig <chrisl@adkproaudio.com> wrote:
>HI John,
>They have been available for months now. SIIG and Adaptec both make
>express and PCMCIA cards and people like Granite Digital, Lacie all make
>drives and enclosures with e-sata. One of the main motherboards we use
>has an E-Sata port on it. Having the ability to do 2 60mb/s+ 400GB
>drives on a laptop is quite nice. Some of the controller will allow for
>Raid 1and 0 setups too. They aren't much more than Firewire/USB options.
>
>Chris
>
>
>John wrote:
>
>> http://www.zdnetasia.com/news/hardware/0,39042972,61976567,0 0.htm
>>
>>external drives with internal speeds !
>>
>>
>
>--
>Chris Ludwig
>ADK
>chrisl@adkproaudio.com <mailto:chrisl@adkproaudio.com>
>www.adkproaudio.com <http://www.adkproaudio.com/>
>(859) 635-5762.........this is one of the things that makes America a shining beacon of
hope in an otherwise.....awwww......nevermind.
;o)
"James McCloskey" <excelsm@hotmail.com> wrote in message
news:458aefa2$1@linux...
>
> http://www.nbc.com/Saturday_Night_Live/uncensored.shtmlThis is a multi-part message in MIME format.
------=_NextPart_000_00CC_01C7252D.8FEF7720
Content-Type: text/plain;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
I hope you're not doing a passive Y split. Look for a VGA splitter on =
google, something like=20
http://www.pcmall.com/pcmall/shop/detail~dpno~767745.asp
They're cheap and OH so worth it
AA
"Tom Bruhl" <arpegio@comcast.net> wrote in message =
news:4589ba0a$1@linux...
Thanks Chris.=20
This graphics stuff is confusing to me.
I have to convert the DVI to VGA because of the 20' length
as I understand. I was going to split it with a Y cable too.
They'll split to a 21"VGA and a 22"LCD. Will that work on=20
the far side of a KVM? I know the resolutions won't match
but I'll only need the VGAs when mixing so I'll adjust the=20
7300 differently for that application. Sooner or later I'll get
4 matching 22" LCDs and quad cards for both comps.
Will this work?
Tom
"Chris Ludwig" <chrisl@adkproaudio.com> wrote in message =
news:45894ef0$1@linux...
HI Tom,
These monitors will be fine with your card. The 7300GT supports that =
resolution on both the VGA and DVI connections.I would run both with =
VGA=20
connections. Having one hooked up DVI and one VGA may make the 2 =
look=20
very different in brightness and such.
Chris
Tom Bruhl wrote:
> My card has one VGA, one DVI and one S Video connector on it.
>
>
> I choose Polesoft Lockspam to fight spam, and you?
> http://www.polesoft.com/refer.html =20
--=20
Chris Ludwig
ADK
chrisl@adkproaudio.com <mailto:chrisl@adkproaudio.com>
www.adkproaudio.com <http://www.adkproaudio.com/>
(859) 635-5762
------=_NextPart_000_00CC_01C7252D.8FEF7720
Content-Type: text/html;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.5730.11" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>I hope you're not doing a passive Y =
split. Look=20
for a VGA splitter on google, something like </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2><A=20
href=3D"http://www.pcmall.com/pcmall/shop/detail~dpno~767745.asp">http://=
www.pcmall.com/pcmall/shop/detail~dpno~767745.asp</A></FONT ></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>They're cheap and OH so worth =
it</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>AA</FONT></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Tom Bruhl" <<A=20
href=3D"mailto:arpegio@comcast.net">arpegio@comcast.net</A>> wrote =
in message=20
<A href=3D"news:4589ba0a$1@linux">news:4589ba0a$1@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks Chris. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>This graphics stuff is confusing to=20
me.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>I have to convert the DVI to VGA =
because of the=20
20' length</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>as I understand. I was going to =
split it=20
with a Y cable too.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>They'll split to </FONT><FONT =
face=3DArial size=3D2>a=20
21"VGA and a 22"LCD. Will that work on </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>the far side of a KVM? I know =
the=20
resolutions won't match</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>but I'll only need the VGAs when =
mixing so I'll=20
adjust the </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>7300 differently for that =
application. =20
Sooner or later I'll get</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>4 matching 22" LCDs and quad cards =
for both=20
comps.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Will this work?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Tom</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<BLOCKQUOTE=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Chris Ludwig" <<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A>> =
wrote in=20
message <A =
href=3D"news:45894ef0$1@linux">news:45894ef0$1@linux</A>...</DIV>HI=20
Tom,<BR>These monitors will be fine with your card. The 7300GT =
supports that=20
<BR>resolution on both the VGA and DVI connections.I would run both =
with VGA=20
<BR>connections. Having one hooked up DVI and one VGA may make the 2 =
look=20
<BR>very different in brightness and such.<BR>Chris<BR><BR><BR>Tom =
Bruhl=20
wrote:<BR><BR>> My card has one VGA, one DVI and one S Video =
connector on=20
it.<BR>><BR>><BR>> I choose Polesoft Lockspam to fight =
spam, and=20
you?<BR>> <A=20
=
href=3D"http://www.polesoft.com/refer.html">http://www.polesoft.com/refer=
..html</A> =20
<BR><BR><BR>-- <BR>Chris Ludwig<BR>ADK<BR><A=20
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A> =
<<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">mailto:chrisl@adkproaudio.com</A>&=
gt;<BR><A=20
href=3D"http://www.adkproaudio.com">www.adkproaudio.com</A> <<A=20
=
href=3D"http://www.adkproaudio.com/">http://www.adkproaudio.com/</A>><=
BR>(859)=20
635-5762</BLOCKQUOTE></BLOCKQUOTE></BODY></HTML>
------=_NextPart_000_00CC_01C7252D.8FEF7720--I pulled the cards and then tested them one by one. It was a PITA. Sorry
you're having this grief.
;o(
"Gantt Kushner" <ganttmann@comcast.net> wrote in message
news:458af983$1@linux...
>
> Annoying bursts of noise is exactly what's happening! How did you figure
> out whci card was bad?
>
> Gantt
>
> "DJ" <nowayjose@dude.net> wrote:
>>One of them created annoying bursts of noise in the monitors so I had to
>
>>fire it. The other one just didn't show up for work one day.
>>
>>;o)
>>
>>
>>"Neil" <IUOIU@OIU.com> wrote in message news:458a08d6$1@linux...
>>>
>>> "Gantt Kushner" <ganttmann@comcast.net> wrote:
>>>>
>>>>Hey Deej,
>>>>
>>>>What were the symptoms of your dying UAD-1 cards?
>>>
>>> Pallor, gasping of breath, thready pulsewave modulation.
>>>
>>> :D
>>>
>>
>>
>I think the broadcast version is funnier...
GrahamHi Neil,
Jamie is right. And you aren't wacked out - you are thinking this through
in a reasonable manner, but coming to the wrong
conclusion - easy to do given how confusing digital audio can be. Each word
represents an amplitude
point on a single curve that is changing over time, and can vary with a
speed up to the Nyquist frequency (as Jamie described).
The complex harmonic content we hear is actually the frequency modulation of
a single waveform,
that over a small amount of time creates the sound we translate - we don't
really hear a single sample at a time,
but thousands of samples at a time (1 sample alone could at most represent a
single positive or negative peak
of a 22,050Hz waveform).
If one bit doesn't cancel, esp. if it's a higher order bit than number 24,
you may hear, and will see that easily,
and the higher the bit in the dynamic range (higher order) the more audible
the difference.
Since each bit is 6dB of dynamic range, you can extrapolate how "loud" that
bit's impact will be
if there is a variation.
Now, obviously if we are talking about 1 sample in a 44.1k rate song, then
it simply be a
click (only audible if it's a high enough order bit) instead of an obvious
musical difference, but that should never
happen in a phase cancellation test between identical files higher than bit
24, unless there are clock sync problems,
driver issues, or the DAW is an early alpha version. :-)
By definition of what DAWs do during playback and record, every audio stream
has the same point in time (judged by the timeline)
played back sample accurately, one word at a time, at whatever sample rate
we are using. A phase cancellation test uses that
fact to compare two audio files word for word (and hence bit for bit since
each bit of a 24-bit word would
be at the same bit slot in each 24-bit word). Assuming they are aligned to
the same start point, sample
accurately, and both are the same set of sample words at each sample point,
bit for bit, and one is phase inverted,
they will cancel through all 24 bits. For two files to cancel completely
for the duration of the file, each and every bit in each word
must be the exact opposite of that same bit position in a word at the same
sample point. This is why zooming in on an FFT
of the full difference file is valuable as it can show any differences in
the lower order bits that wouldn't be audible. So even if
there is no audible difference, the visual followup will show if the two
files truly cancel even a levels below hearing, or
outside of a frequency change that we will perceive.
When they don't cancel, usually there will be way more than 1 bit
difference - it's usually one or more bits in the words for
thousands of samples. From a musical standpoint this is usually in a
frequency range (low freq, or high freq most often) - that will
show up as the difference between them, and that usually happens due to some
form of processing difference between the files,
such as EQ, compression, frequency dependant gain changes, etc. That is what
I believe you are thinking through, but when
talking about straight summing with no gain change (or known equal gain
changes), we are only looking at linear, one for one
comparisons between the two files' frequency representations.
Regards,
Dedric
> Neil wrote:
>> "Dedric Terry" <dedric@echomg.com> wrote:
>>> The tests I did were completely blank down to -200 dB (far below the
>>> last
>>
>>> bit). It's safe to say there is no difference, even in
>>> quantization noise, which by technical rights, is considered below the
>>> level
>>
>>> of "cancellation" in such tests.
>>
>> I'm not necessarily talking about just the first bit or the
>> last bit, but also everything in between... what happens on bit
>> #12, for example? Everything on bit #12 should be audible, but
>> in an a/b test what if thre are differences in what bits #8
>> through #12 sound like, but the amplutide is stll the same on
>> both files at that point, you'll get a null, right? Extrapolate
>> that out somewhat & let's say there are differences in bits #8
>> through #12 on sample points 3, 17, 1,000, 4,523, 7,560, etc,
>> etc through 43,972... Now this is breaking things down well
>> beyond what I think can be measured, if I'm not mistaken (I
>> dn't know of any way we could extract JUST that information
>> from each file & play it back for an a/b test; but would not
>> that be enough to have to "null-able" files that do actually
>> sound somewhat different?
>>
>> I guess what I'm saying is that since each sample in a musical
>> track or full song file doesn't represent a pure, simple set of
>> content like a sample of a sine wave would - there's a whole
>> world of harmonic structure in each sample of a song file, and
>> I think (although I'll admit - I can't "prove") that there is
>> plenty of room for some variables between the first bit & the
>> last bit while still allowing for a null test to be successful.
>>
>> No? Am I wacked out of my mind?
>>
>> Neil
>>For those of you who will be voting for the 49th annual Grammys, please give
a listen to, and consider Guy Clark's "Workbench Songs" for Best
Contemporary Folk/Americana Album (Catagory 68 in Field 14 - Folk). Edited
and mixed entirely in PARIS.
THANK YOU,
ChrisDedric - first of all, great explanation - esp. your 2nd
paragraph. Next, let's take a look at something in the form of
the best "graph" I can do in this NG's format... let's assume
that each dot in the simple graph below is a sample point on a
segment of a waveform, and let's futher assume that each "I"
below represents four bits (I don't want to make it too
vertically large, for ease of reading) - so we're dealing with
a 16-bit wav file, with the 5th "dot" from the start point on
the left being a full-amplitude, zero-db-line 16 bit sample.
Now.... really, all I have to do to get a "null" is to have the
amplitude match at each "dot" on the waveform, yes? This, of
course, is a very simplistic graphic example, so bear with
me... but if I have each "dot" matching in amplitude &
therefore can get a null, what about the bits & content thereof
in between the extremes between the maxes & zero-line
crossings? Are you saying that there can be no variables in
sound between those sections that would still result in a null?
What about all the "I"'s that represent bits in between the
maxes & the minimums?
.
. I .
. I I I . What about the stuff in here?
. I I I I I . .....or in here????
.. I I I I I I I .
-------------------------------------
. I I I I I I I .
. I I I I I . Again, what about this region?
. I I I . ... or this region?
. I .
.
NeilThis is a multi-part message in MIME format.
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Thanks Aaron,
Will this work on the backside of a KVM? I can't put it in front.
I guess if it needed to be in front of the KVM I could just get a=20
single output jobbie and split it at the other side. Do those exist?
"Aaron Allen" <know-spam@not_here.dude> wrote in message =
news:458b23a2@linux...
I hope you're not doing a passive Y split. Look for a VGA splitter on =
google, something like=20
http://www.pcmall.com/pcmall/shop/detail~dpno~767745.asp
They're cheap and OH so worth it
AA
"Tom Bruhl" <arpegio@comcast.net> wrote in message =
news:4589ba0a$1@linux...
Thanks Chris.=20
This graphics stuff is confusing to me.
I have to convert the DVI to VGA because of the 20' length
as I understand. I was going to split it with a Y cable too.
They'll split to a 21"VGA and a 22"LCD. Will that work on=20
the far side of a KVM? I know the resolutions won't match
but I'll only need the VGAs when mixing so I'll adjust the=20
7300 differently for that application. Sooner or later I'll get
4 matching 22" LCDs and quad cards for both comps.
Will this work?
Tom
"Chris Ludwig" <chrisl@adkproaudio.com> wrote in message =
news:45894ef0$1@linux...
HI Tom,
These monitors will be fine with your card. The 7300GT supports =
that=20
resolution on both the VGA and DVI connections.I would run both =
with VGA=20
connections. Having one hooked up DVI and one VGA may make the 2 =
look=20
very different in brightness and such.
Chris
Tom Bruhl wrote:
> My card has one VGA, one DVI and one S Video connector on it.
>
>
> I choose Polesoft Lockspam to fight spam, and you?
> http://www.polesoft.com/refer.html =20
--=20
Chris Ludwig
ADK
chrisl@adkproaudio.com <mailto:chrisl@adkproaudio.com>
www.adkproaudio.com <http://www.adkproaudio.com/>
(859) 635-5762
------=_NextPart_000_005B_01C7256C.C32E5E00
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charset="iso-8859-1"
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1400" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Thanks Aaron,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Will this work on the backside of a =
KVM? I=20
can't put it in front.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I guess if it needed to be in front of =
the=20
KVM I could just get a </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>single output </FONT><FONT face=3DArial =
size=3D2>jobbie=20
and split it at the other side. Do those exist?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Aaron Allen" <<A=20
=
href=3D"mailto:know-spam@not_here.dude">know-spam@not_here.dude</A>> =
wrote in=20
message <A =
href=3D"news:458b23a2@linux">news:458b23a2@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>I hope you're not doing a passive Y =
split. Look=20
for a VGA splitter on google, something like </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2><A=20
=
href=3D"http://www.pcmall.com/pcmall/shop/detail~dpno~767745.asp">http://=
www.pcmall.com/pcmall/shop/detail~dpno~767745.asp</A></FONT ></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>They're cheap and OH so worth =
it</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>AA</FONT></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Tom Bruhl" <<A=20
href=3D"mailto:arpegio@comcast.net">arpegio@comcast.net</A>> =
wrote in=20
message <A =
href=3D"news:4589ba0a$1@linux">news:4589ba0a$1@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks Chris. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>This graphics stuff is confusing to =
me.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>I have to convert the DVI to VGA =
because of the=20
20' length</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>as I understand. I was going =
to split it=20
with a Y cable too.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>They'll split to </FONT><FONT =
face=3DArial=20
size=3D2>a 21"VGA and a 22"LCD. Will that work on =
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>the far side of a KVM? I know =
the=20
resolutions won't match</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>but I'll only need the VGAs when =
mixing so I'll=20
adjust the </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>7300 differently for that =
application. =20
Sooner or later I'll get</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>4 matching 22" LCDs and quad cards =
for both=20
comps.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Will this work?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Tom</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<BLOCKQUOTE=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Chris Ludwig" <<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A>> =
wrote=20
in message <A=20
=
href=3D"news:45894ef0$1@linux">news:45894ef0$1@linux</A>...</DIV>HI=20
Tom,<BR>These monitors will be fine with your card. The 7300GT =
supports=20
that <BR>resolution on both the VGA and DVI connections.I would =
run both=20
with VGA <BR>connections. Having one hooked up DVI and one VGA may =
make=20
the 2 look <BR>very different in brightness and=20
such.<BR>Chris<BR><BR><BR>Tom Bruhl wrote:<BR><BR>> My card has =
one=20
VGA, one DVI and one S Video connector on =
it.<BR>><BR>><BR>> I=20
choose Polesoft Lockspam to fight spam, and you?<BR>> <A=20
=
href=3D"http://www.polesoft.com/refer.html">http://www.polesoft.com/refer=
..html</A> =20
<BR><BR><BR>-- <BR>Chris Ludwig<BR>ADK<BR><A=20
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A> =
<<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">mailto:chrisl@adkproaudio.com</A>&=
gt;<BR><A=20
href=3D"http://www.adkproaudio.com">www.adkproaudio.com</A> <<A =
=
href=3D"http://www.adkproaudio.com/">http://www.adkproaudio.com/</A>><=
BR>(859)=20
635-5762</BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BODY></HTML >
------=_NextPart_000_005B_01C7256C.C32E5E00--Neil,
Actually what you are showing with the I's is the power of the waveform
(area under the curve),
not the bits. Only the curve itself is the actual amplitude of the wave.
The amplitude as represented in
16 bit words would be shown on the y-axis as 65535 steps, and only represent
the vertical dimension, or amplitude of a specific
point in time, not the area under it. The x-axis is of course time - one dot
per sample point,
each sample point is represented one 16-bit word, stored at that point in
time. Each word is used to define where
on the y-axis (how far from 0 volts amplitude) that dot appears - 0000 0000
0000 0000 of course would be the
0 amplitude point. An amplitude of 0 volts would equate to a dB power
of -infinity in the digital realm if
we have an infinite number of bits to subdivide the y axis with, but in
reality 0 is effectively -144dB for 24-bit audio
and -96dB for 16 bit.
Since we work with levels in dB in a DAW we start to think of waveforms as
being defined by that quantity, but
when the y axis is in dB, it's really telling us the power of that signal
around a point in time (area under the curve
over an average distance), not the amplitude.
There aren't additional points or content beneath the outline you drew.
It's a bit non-intuituve,
but think of audio as only existing as a function of a change in time and
the concept makes more sense.
Regardless of how complex the actual music is, there is still only a single
waveform that reaches our ears from a
single source - e.g a single amplitude point at a given point in time.
What we "hear" is how that waveform changes
up and down, over a certain time period.
Regards,
Dedric
"Neil" <IUOIU@OIU.com> wrote in message news:458b75af$1@linux...
>
> Dedric - first of all, great explanation - esp. your 2nd
> paragraph. Next, let's take a look at something in the form of
> the best "graph" I can do in this NG's format... let's assume
> that each dot in the simple graph below is a sample point on a
> segment of a waveform, and let's futher assume that each "I"
> below represents four bits (I don't want to make it too
> vertically large, for ease of reading) - so we're dealing with
> a 16-bit wav file, with the 5th "dot" from the start point on
> the left being a full-amplitude, zero-db-line 16 bit sample.
>
> Now.... really, all I have to do to get a "null" is to have the
> amplitude match at each "dot" on the waveform, yes? This, of
> course, is a very simplistic graphic example, so bear with
> me... but if I have each "dot" matching in amplitude &
> therefore can get a null, what about the bits & content thereof
> in between the extremes between the maxes & zero-line
> crossings? Are you saying that there can be no variables in
> sound between those sections that would still result in a null?
> What about all the "I"'s that represent bits in between the
> maxes & the minimums?
>
> .
> . I .
> . I I I . What about the stuff in here?
> . I I I I I . .....or in here????
> . I I I I I I I .
> -------------------------------------
> . I I I I I I I .
> . I I I I I . Again, what about this region?
> . I I I . ... or this region?
> . I .
> .
>
> NeilThis is a multi-part message in MIME format.
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.....Yeahhh, you're talking about beacon, I'm talking about =
Christmassoup....;o)
http://www.metacafe.com/watch/315622/happy_new_year/ =20
Erling
"DJ" <nowayjose@dude.net> skrev i melding news:458b1dd0@linux...
> ........this is one of the things that makes America a shining beacon =
of=20
> hope in an otherwise.....awwww......nevermind.
>=20
> ;o)
>=20
>=20
> "James McCloskey" <excelsm@hotmail.com> wrote in message=20
> news:458aefa2$1@linux...
>>
>> http://www.nbc.com/Saturday_Night_Live/uncensored.shtml=20
>=20
>
------=_NextPart_000_0029_01C725B4.83F4FBC0
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<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2900.3020" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY>
<DIV><FONT face=3DArial size=3D2>....Yeahhh, you're talking about =
beacon, I'm=20
talking about Christmassoup....;o)</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV>
<DIV><A =
href=3D"http://www.metacafe.com/watch/315622/happy_new_year/"><FONT=20
face=3DArial=20
size=3D2>http://www.metacafe.com/watch/315622/happy_new_year/</FONT></A><=
FONT=20
face=3DArial size=3D2> </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT><FONT face=3DArial =
size=3D2>Erling</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV></DIV>
<DIV><FONT face=3DArial size=3D2>"DJ" <</FONT><A=20
href=3D"mailto:nowayjose@dude.net"><FONT face=3DArial=20
size=3D2>nowayjose@dude.net</FONT></A><FONT face=3DArial size=3D2>> =
skrev i melding=20
</FONT><A href=3D"news:458b1dd0@linux"><FONT face=3DArial=20
size=3D2>news:458b1dd0@linux</FONT></A><FONT face=3DArial=20
size=3D2>...</FONT></DIV><FONT face=3DArial size=3D2>> ........this =
is one of the=20
things that makes America a shining beacon of <BR>> hope in an=20
otherwise.....awwww......nevermind.<BR>> <BR>> ;o)<BR>> =
<BR>>=20
<BR>> "James McCloskey" <</FONT><A =
href=3D"mailto:excelsm@hotmail.com"><FONT=20
face=3DArial size=3D2>excelsm@hotmail.com</FONT></A><FONT face=3DArial =
size=3D2>>=20
wrote in message <BR>> </FONT><A href=3D"news:458aefa2$1@linux"><FONT =
face=3DArial size=3D2>news:458aefa2$1@linux</FONT></A><FONT face=3DArial =
size=3D2>...<BR>>><BR>>> </FONT><A=20
href=3D"http://www.nbc.com/Saturday_Night_Live/uncensored.shtml"><FONT =
face=3DArial=20
size=3D2>http://www.nbc.com/Saturday_Night_Live/uncensored.shtml</FONT></=
A><FONT=20
face=3DArial size=3D2> <BR>> <BR>></FONT></BODY></HTML>
------=_NextPart_000_0029_01C725B4.83F4FBC0--wouldn't be so bad if it was actually funny
PS. Why do people link to stuff they find offensive? :P
"James McCloskey" <excelsm@hotmail.com> wrote in message
news:458aefa2$1@linux...
>
> http://www.nbc.com/Saturday_Night_Live/uncensored.shtmlGreetings :)
Special for the PARIS community going on now !
UAD-1 DSP cards - Ends Dec 31
https://my.uaudio.com/store/
$ 295 on project pak
$ 595 on Flexi - Pak
$ 999 on Ultra pak
$ 895 on EXPERT Pak PCI-E $ 750 voucher included
$ 395 on EXPRESS Pak PCI-E $ 100 voucher included
http://www.uaudio.com/products/digital/expertpak/demo.html
As always - trades are always welcome !
Morgan
Eastcoast Music Mall
800-901-2001
morganp@ntplx.netOkay...
I guess what I'm saying is this:
-Is it possible that diferent DAW manufactuers "code" their app differently
for sound results.
I the answer is yes, then,the real task is to discover or rather un-cover
what's say: Motu's vision of summing, versus Digidesign, versus Steinberg
and so on..
What's under the hood. To me and others,when Digi re-coded their summing
engine, it was obvious that Pro Tools has an obvious top end (8k-10k) bump.
Where as Steinberg's summing is very neutral.
"Dedric Terry" <dedric@echomg.com> wrote:
>Hi Neil,
>
>Jamie is right. And you aren't wacked out - you are thinking this through
>in a reasonable manner, but coming to the wrong
>conclusion - easy to do given how confusing digital audio can be. Each
word
>represents an amplitude
>point on a single curve that is changing over time, and can vary with a
>speed up to the Nyquist frequency (as Jamie described).
>The complex harmonic content we hear is actually the frequency modulation
of
>a single waveform,
>that over a small amount of time creates the sound we translate - we don't
>really hear a single sample at a time,
>but thousands of samples at a time (1 sample alone could at most represent
a
>single positive or negative peak
>of a 22,050Hz waveform).
>
>If one bit doesn't cancel, esp. if it's a higher order bit than number 24,
>you may hear, and will see that easily,
>and the higher the bit in the dynamic range (higher order) the more audible
>the difference.
>Since each bit is 6dB of dynamic range, you can extrapolate how "loud" that
>bit's impact will be
>if there is a variation.
>
>Now, obviously if we are talking about 1 sample in a 44.1k rate song, then
>it simply be a
>click (only audible if it's a high enough order bit) instead of an obvious
>musical difference, but that should never
>happen in a phase cancellation test between identical files higher than
bit
>24, unless there are clock sync problems,
>driver issues, or the DAW is an early alpha version. :-)
>
>By definition of what DAWs do during playback and record, every audio stream
>has the same point in time (judged by the timeline)
>played back sample accurately, one word at a time, at whatever sample rate
>we are using. A phase cancellation test uses that
>fact to compare two audio files word for word (and hence bit for bit since
>each bit of a 24-bit word would
>be at the same bit slot in each 24-bit word). Assuming they are aligned
to
>the same start point, sample
>accurately, and both are the same set of sample words at each sample point,
>bit for bit, and one is phase inverted,
>they will cancel through all 24 bits. For two files to cancel completely
>for the duration of the file, each and every bit in each word
>must be the exact opposite of that same bit position in a word at the same
>sample point. This is why zooming in on an FFT
>of the full difference file is valuable as it can show any differences in
>the lower order bits that wouldn't be audible. So even if
>there is no audible difference, the visual followup will show if the two
>files truly cancel even a levels below hearing, or
>outside of a frequency change that we will perceive.
>
>When they don't cancel, usually there will be way more than 1 bit
>difference - it's usually one or more bits in the words for
>thousands of samples. From a musical standpoint this is usually in a
>frequency range (low freq, or high freq most often) - that will
>show up as the difference between them, and that usually happens due to
some
>form of processing difference between the files,
>such as EQ, compression, frequency dependant gain changes, etc. That is
what
>I believe you are thinking through, but when
>talking about straight summing with no gain change (or known equal gain
>changes), we are only looking at linear, one for one
>comparisons between the two files' frequency representations.
>
>Regards,
>Dedric
>
>> Neil wrote:
>>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>> The tests I did were completely blank down to -200 dB (far below the
>>>> last
>>>
>>>> bit). It's safe to say there is no difference, even in
>>>> quantization noise, which by technical rights, is considered below the
>>>> level
>>>
>>>> of "cancellation" in such tests.
>>>
>>> I'm not necessarily talking about just the first bit or the
>>> last bit, but also everything in between... what happens on bit
>>> #12, for example? Everything on bit #12 should be audible, but
>>> in an a/b test what if thre are differences in what bits #8
>>> through #12 sound like, but the amplutide is stll the same on
>>> both files at that point, you'll get a null, right? Extrapolate
>>> that out somewhat & let's say there are differences in bits #8
>>> through #12 on sample points 3, 17, 1,000, 4,523, 7,560, etc,
>>> etc through 43,972... Now this is breaking things down well
>>> beyond what I think can be measured, if I'm not mistaken (I
>>> dn't know of any way we could extract JUST that information
>>> from each file & play it back for an a/b test; but would not
>>> that be enough to have to "null-able" files that do actually
>>> sound somewhat different?
>>>
>>> I guess what I'm saying is that since each sample in a musical
>>> track or full song file doesn't represent a pure, simple set of
>>> content like a sample of a sine wave would - there's a whole
>>> world of harmonic structure in each sample of a song file, and
>>> I think (although I'll admit - I can't "prove") that there is
>>> plenty of room for some variables between the first bit & the
>>> last bit while still allowing for a null test to be successful.
>>>
>>> No? Am I wacked out of my mind?
>>>
>>> Neil
>>>
>Thanks for sharing..
Actually, I thought it was funny.
You see when I happen to stop by one of these genre songs
on the radio I always put my own lyrics to them.
Usually they are very simular to what you hear on that clip.
It is the only thing that makes that style of music enjoyable for me.
humming.... it's my d*ck in a box....Mmmmm Mmmmm girl.........
--
Thanks,
BrandonNeil,
You're using an analog waveform that is leading you to think incorrectly
about sampling. This is (very roughly) how it would look if you're working
with 16 bit samples.
0101010101010101
0101010111011101
0101110111010101
0101111111010100
0101110101111101
0111011101111101
0111110101110101
0100010111000100
0100011101010101
0001011100010101
0000010111111100
0001000001010111
0100000111110101
0111011101010000
0101011101000000
0101011111000101
0101010101010101
The easiest way to think of how the sampler works is that it looks at the
incoming voltage to the converter and asks 'Is this in the top or bottom
half of the possible amplitudes I can measure.' If it's in the top half it
writes a 1, if it's in the bottom half, it writes a zero. The next bit asks,
'Now that I know which half of my measurable voltage I'm looking at, is the
voltage in the top half of that half or the bottom half?' That's bit number
two. Then it's on to, 'Now that I know what quarter it's it, is it in the
top or bottom half of that quarter?' And so on sixteen time giving it a resolution
of 2 to the sixteenth power.
In other words, asking if the bits under the sample would sound is like asking
how the road would drive if it were 30 feet underground.
Now then, to get back to the original argument, people like me (and I think
Dedric but I'll let him speak for himself) get a little hacked off when someone
says, 'you have to use your ears' when it's possible using various computer
tools to check exactly how many of those samples match in two given files.
The nulling trick is just a very easy way to get a quick read on one aspect,
which is to answer the question 'do these two files match?' But there are
others and I've used them. And the sameness between properly written (by
which I mean lacking in serious bugs) audio applications is startling and
their differences so minor that other errors (analog cables, dust on the
speaker cone, humidity and temperature in the room) are far more likely to
cause a difference.
Personally I think this all stems from romanticism about music and the purity
of art. I have yet to hear someone tell me they need financial calculations
down to 25 decimal points. They need them done to (at most) five decimal
points because the smallest commonly used financial divisor is the basis
point, or one one hundredth of a penny. So internally you calculate to five
decimal places and round up or down from there and get on with your life.
As geeky as finance guys can get, nobody ever says, 'You know, Thad, that
last basis point just isn't really punchy enough for this deal. LBO guys
need really punchy returns, so can you run that calculation out a few more
bits to get a punchier basis point?' Scientists are also extremely careful
to keep 'false precision' out of their calculations, so if one instrument
will measure to four decimal points and the others will measure to 12 they
understand that everything the higher resolution instruments measure beyond
four accurate decimal points is worthless. They usually won't even record
the data to be sure they don't claim greater precision than they have, because
that's considered a horribly embarrassing junior high school mistake. But
musicians and audio engineers think that just because the data is sound data
somehow it enters a nebulous zone where that last one hundredth of a penny
can be punchier. Hey, if it gets you through the day, that's fine by me,
but there are things about digital audio that can be proven true or false
using the data. For things that can't be proven true or false with the data
itself there is ABY testing, which is a controlled way to use the most precise
audio measuring instruments available (our ears, at least until bats will
wear headphones) to see if things sound different. When it's not in the data,
and it's not in the ABY, I say it doesn't exist.
TCB
"Neil" <IUOIU@OIU.com> wrote:
>
>Dedric - first of all, great explanation - esp. your 2nd
>paragraph. Next, let's take a look at something in the form of
>the best "graph" I can do in this NG's format... let's assume
>that each dot in the simple graph below is a sample point on a
>segment of a waveform, and let's futher assume that each "I"
>below represents four bits (I don't want to make it too
>vertically large, for ease of reading) - so we're dealing with
>a 16-bit wav file, with the 5th "dot" from the start point on
>the left being a full-amplitude, zero-db-line 16 bit sample.
>
>Now.... really, all I have to do to get a "null" is to have the
>amplitude match at each "dot" on the waveform, yes? This, of
>course, is a very simplistic graphic example, so bear with
>me... but if I have each "dot" matching in amplitude &
>therefore can get a null, what about the bits & content thereof
>in between the extremes between the maxes & zero-line
>crossings? Are you saying that there can be no variables in
>sound between those sections that would still result in a null?
>What about all the "I"'s that represent bits in between the
>maxes & the minimums?
>
> .
> . I .
> . I I I . What about the stuff in here?
> . I I I I I . .....or in here????
>. I I I I I I I .
>-------------------------------------
> . I I I I I I I .
> . I I I I I . Again, what about this region?
> . I I I . ... or this region?
> . I .
> .
>
>NeilThad, I assume that you ar ereferring to me (using your ears).
Look, I think we are talking about two differnt things here:
1) Digital data
2) Software (DAWS) coding
You and Dedric have been concentrating on the laws of Digital audio. That's
fine. But, I'm talking about the Software that we use to decode our digital
audio.
Like my previous post states, are we saying that DAW software can't be written
for certain sonic results?
"TCB" <nobody@ishere.com> wrote:
>
>Neil,
>
>You're using an analog waveform that is leading you to think incorrectly
>about sampling. This is (very roughly) how it would look if you're working
>with 16 bit samples.
>
>0101010101010101
>0101010111011101
>0101110111010101
>0101111111010100
>0101110101111101
>0111011101111101
>0111110101110101
>0100010111000100
>0100011101010101
>0001011100010101
>0000010111111100
>0001000001010111
>0100000111110101
>0111011101010000
>0101011101000000
>0101011111000101
>0101010101010101
>
>The easiest way to think of how the sampler works is that it looks at the
>incoming voltage to the converter and asks 'Is this in the top or bottom
>half of the possible amplitudes I can measure.' If it's in the top half
it
>writes a 1, if it's in the bottom half, it writes a zero. The next bit asks,
>'Now that I know which half of my measurable voltage I'm looking at, is
the
>voltage in the top half of that half or the bottom half?' That's bit number
>two. Then it's on to, 'Now that I know what quarter it's it, is it in the
>top or bottom half of that quarter?' And so on sixteen time giving it a
resolution
>of 2 to the sixteenth power.
>
>In other words, asking if the bits under the sample would sound is like
asking
>how the road would drive if it were 30 feet underground.
>
>Now then, to get back to the original argument, people like me (and I think
>Dedric but I'll let him speak for himself) get a little hacked off when
someone
>says, 'you have to use your ears' when it's possible using various computer
>tools to check exactly how many of those samples match in two given files.
>The nulling trick is just a very easy way to get a quick read on one aspect,
>which is to answer the question 'do these two files match?' But there are
>others and I've used them. And the sameness between properly written (by
>which I mean lacking in serious bugs) audio applications is startling and
>their differences so minor that other errors (analog cables, dust on the
>speaker cone, humidity and temperature in the room) are far more likely
to
>cause a difference.
>
>Personally I think this all stems from romanticism about music and the purity
>of art. I have yet to hear someone tell me they need financial calculations
>down to 25 decimal points. They need them done to (at most) five decimal
>points because the smallest commonly used financial divisor is the basis
>point, or one one hundredth of a penny. So internally you calculate to five
>decimal places and round up or down from there and get on with your life.
>As geeky as finance guys can get, nobody ever says, 'You know, Thad, that
>last basis point just isn't really punchy enough for this deal. LBO guys
>need really punchy returns, so can you run that calculation out a few more
>bits to get a punchier basis point?' Scientists are also extremely careful
>to keep 'false precision' out of their calculations, so if one instrument
>will measure to four decimal points and the others will measure to 12 they
>understand that everything the higher resolution instruments measure beyond
>four accurate decimal points is worthless. They usually won't even record
>the data to be sure they don't claim greater precision than they have, because
>that's considered a horribly embarrassing junior high school mistake. But
>musicians and audio engineers think that just because the data is sound
data
>somehow it enters a nebulous zone where that last one hundredth of a penny
>can be punchier. Hey, if it gets you through the day, that's fine by me,
>but there are things about digital audio that can be proven true or false
>using the data. For things that can't be proven true or false with the data
>itself there is ABY testing, which is a controlled way to use the most precise
>audio measuring instruments available (our ears, at least until bats will
>wear headphones) to see if things sound different. When it's not in the
data,
>and it's not in the ABY, I say it doesn't exist.
>
>TCB
>
>"Neil" <IUOIU@OIU.com> wrote:
>>
>>Dedric - first of all, great explanation - esp. your 2nd
>>paragraph. Next, let's take a look at something in the form of
>>the best "graph" I can do in this NG's format... let's assume
>>that each dot in the simple graph below is a sample point on a
>>segment of a waveform, and let's futher assume that each "I"
>>below represents four bits (I don't want to make it too
>>vertically large, for ease of reading) - so we're dealing with
>>a 16-bit wav file, with the 5th "dot" from the start point on
>>the left being a full-amplitude, zero-db-line 16 bit sample.
>>
>>Now.... really, all I have to do to get a "null" is to have the
>>amplitude match at each "dot" on the waveform, yes? This, of
>>course, is a very simplistic graphic example, so bear with
>>me... but if I have each "dot" matching in amplitude &
>>therefore can get a null, what about the bits & content thereof
>>in between the extremes between the maxes & zero-line
>>crossings? Are you saying that there can be no variables in
>>sound between those sections that would still result in a null?
>>What about all the "I"'s that represent bits in between the
>>maxes & the minimums?
>>
>> .
>> . I .
>> . I I I . What about the stuff in here?
>> . I I I I I . .....or in here????
>>. I I I I I I I .
>>-------------------------------------
>> . I I I I I I I .
>> . I I I I I . Again, what about this region?
>> . I I I . ... or this region?
>> . I .
>> .
>>
>>Neil
>"LaMont" <jjdpro@ameritech.net> wrote in message news:458be8d5$1@linux...
>
> Okay...
> I guess what I'm saying is this:
>
> -Is it possible that diferent DAW manufactuers "code" their app
> differently
> for sound results.
Of course it is *possible* to do this, but only if the DAW has a specific
sound shaping purpose
beyond normal summing/mixing. Users talk about wanting developers to add a
"Neve sound" or "API sound" option to summing engines,
but that's really impractical given the amount of dsp required to make a
decent emulation (with convolution, dynamic EQ functions,
etc). For sake of not eating up all cpu processing, that could likely only
surface as is a built in EQ, which
no one wants universally in summing, and anyone can add at will already.
So it hasn't happened yet and isn't likely to as it detours from the basic
tenant of audio recording - recreate what comes in as
accurately as possible.
What Digi did in recoding their summing engine was try to recover some
of the damage done by the 24-bit buss in Mix systems. Motorola 56k dsps are
24-bit fixed point chips and I think
the new generation (321?) still is, but they use double words now for
48-bits). And though plugins could process at 48-bit by
doubling up and using upper and lower 24-bit words for 48-bit outputs, the
buss
between chips was 24-bits, so they had to dither to 24-bits after every
plugin. The mixer (if I recall correctly) also
had a 24-bit buss, so what Digi did is to add a dither stage to the mixer to
prevent this
constant truncation of data. 24-bits isn't enough to cover summing for more
than a few tracks without
losing information in the 16-bit world, and in the 24-bit world some
information will be lost, at least at the lowest levels.
Adding a dither stage (though I think they did more than that - perhaps
implement a 48-bit double word stage as well),
simply smoothed over the truncation that was happening, but it didn't solve
the problem, so with HD
they went to a double-word path - throughout I believe, including the path
between chips. I believe the chips
are still 24-bit, but by doubling up the processing (yes at a cost of twice
the overhead), they get a 48-bit engine.
This not only provided better headroom, but greater resolution. Higher bit
depths subdivide the amplitude with greater resolution, and that's
really where we get the definition of dynamic range - by lowering the signal
to quantization noise ratio.
With DAWs that use 32-bit floating point math all the way through, the only
reason for altering the summing
is by error, and that's an error that would actually be hard to make and get
past a very basic alpha stage of testing.
There is a small difference in fixed point math and floating point math, or
at least a theoretical difference in how it affects audio
in certain cases, but not necessarily in the result for calculating gain in
either for the same audio file. Where any differences might show up is
complicated, and I believe only appear at levels below 24-bit (or in
headroom with tracks pushed beyond 0dBFS), or when/if
there areany differences in where each amplitude level is quantized.
Obviously there can be differences if the DAW has to use varying bit depths
throughout a single summing path to accomodate hardware
as well as software summing, since there may be truncation or rounding along
the way, but that impacts the lowest bit
level, and hence - spacial reproduction, reverb tails perhaps, and "depth",
not the levels most music so the differences are most
often more subtle than not. But most modern DAWs have eliminated those
"rough edges" in the math by increasing the bit depth to accomodate normal
summing required for mixing audio.
So with Lynn's unity gain summing test (A files on the CD I believe), DAWs
were never asked to sum beyond 24-bits,
at least not on the upper end of the dynamic range, so everything that could
represent 24-bits accurately would cancel. The only ones
that didn't were ones that had a different bit depth and/or gain structure
whether hybrid or native
(e.g. Paris' subtracting 20dB from tracks and adding it to the buss). In
this case, PTHD cancelled (when I tested it) with
Nuendo, Samplitude, Logic, etc because the impact of the 48-bit fixed vs.
32-bit float wasn't a factor.
When trying other tests, even when adding and subtracting gain, Nuendo,
Sequoia and Sonar cancel - both audibly and
visually at inaudible levels, which only proves that one isn't making an
error when calculating basic gain. Since a dB is well defined,
and the math to add gain is simple, they shouldn't. The fact that they all
use 32-bit float all the way through eliminates a difference
in data structure as well, and this just verifies that. There was a time
that supposedly Logic (v3, v4?) was partly 24-bit, or so the rumor went,
but it's 32-bit float all the way through now just as Sonar, Nuendo/Cubase,
Samplitude/Sequoia, DP, Audition (I presume at least).
I don't know what Acid or Live use. Saw promotes a fixed point engine, but
I don't know if it is still 24-bit, or now 48 bit.
That was an intentional choice by the developer, but he's the only one I
know of that stuck with 24-bit for summing
intentionally, esp. after the Digi Mix system mixer incident.
Long answer, but to sum up, it is certainly physically *possible* for a
developer to code something differently intentionally, but not
in reality likely since it would be breaking some basic fixed point or
floating point math rules. Where the differences really
showed up in the past is with PT Mix systems where the limitation was really
significant - e.g. 24 bit with truncation at several stages.
That really isn't such an issue anymore. Given the differences in workflow,
missing something in workflow or layout differences
is easy enough to do (e.g. Sonar doesn't have group and busses the way
Nuendo does, as it's outputs are actually driver outputs,
not software busses, so in Sonar, busses are actually outputs, and sub
busses are actually busses in Nuendo. There are no,
or at least I haven't found the equivalent of a Nuendo group in Sonar - that
affects the results of some tests (though not basic
summing) if not taken into account, but when taken into account, they work
exactly the same way).
So at least when talking about apps with 32-bit float all the way through,
it's safe to say (since it has been proven) that summing isn't different
unless
there is an error somewhere, or variation in how the user duplicates the
same mix in two different apps.
Imho, that's actually a very good thing - approaching a more consistent
basis for recording and mixing from which users can make all
of the decisions as to how the final product will sound and not be required
to decide when purchasing a pricey console, and have to
focus their business on clients who want "that sound". I believe we are
actually closer to the pure definition of recording now than
we once were.
Regards,
Dedric
>
> I the answer is yes, then,the real task is to discover or rather un-cover
> what's say: Motu's vision of summing, versus Digidesign, versus Steinberg
> and so on..
>
> What's under the hood. To me and others,when Digi re-coded their summing
> engine, it was obvious that Pro Tools has an obvious top end (8k-10k)
> bump.
> Where as Steinberg's summing is very neutral.
>
> "Dedric Terry" <dedric@echomg.com> wrote:
>>Hi Neil,
>>
>>Jamie is right. And you aren't wacked out - you are thinking this through
>
>>in a reasonable manner, but coming to the wrong
>>conclusion - easy to do given how confusing digital audio can be. Each
> word
>>represents an amplitude
>>point on a single curve that is changing over time, and can vary with a
>
>>speed up to the Nyquist frequency (as Jamie described).
>>The complex harmonic content we hear is actually the frequency modulation
> of
>>a single waveform,
>>that over a small amount of time creates the sound we translate - we don't
>
>>really hear a single sample at a time,
>>but thousands of samples at a time (1 sample alone could at most represent
> a
>>single positive or negative peak
>>of a 22,050Hz waveform).
>>
>>If one bit doesn't cancel, esp. if it's a higher order bit than number 24,
>
>>you may hear, and will see that easily,
>>and the higher the bit in the dynamic range (higher order) the more
>>audible
>
>>the difference.
>>Since each bit is 6dB of dynamic range, you can extrapolate how "loud"
>>that
>
>>bit's impact will be
>>if there is a variation.
>>
>>Now, obviously if we are talking about 1 sample in a 44.1k rate song, then
>
>>it simply be a
>>click (only audible if it's a high enough order bit) instead of an obvious
>
>>musical difference, but that should never
>>happen in a phase cancellation test between identical files higher than
> bit
>>24, unless there are clock sync problems,
>>driver issues, or the DAW is an early alpha version. :-)
>>
>>By definition of what DAWs do during playback and record, every audio
>>stream
>
>>has the same point in time (judged by the timeline)
>>played back sample accurately, one word at a time, at whatever sample
>>rate
>
>>we are using. A phase cancellation test uses that
>>fact to compare two audio files word for word (and hence bit for bit since
>
>>each bit of a 24-bit word would
>>be at the same bit slot in each 24-bit word). Assuming they are aligned
> to
>>the same start point, sample
>>accurately, and both are the same set of sample words at each sample
>>point,
>
>>bit for bit, and one is phase inverted,
>>they will cancel through all 24 bits. For two files to cancel completely
>
>>for the duration of the file, each and every bit in each word
>>must be the exact opposite of that same bit position in a word at the same
>
>>sample point. This is why zooming in on an FFT
>>of the full difference file is valuable as it can show any differences in
>
>>the lower order bits that wouldn't be audible. So even if
>>there is no audible difference, the visual followup will show if the two
>
>>files truly cancel even a levels below hearing, or
>>outside of a frequency change that we will perceive.
>>
>>When they don't cancel, usually there will be way more than 1 bit
>>difference - it's usually one or more bits in the words for
>>thousands of samples. From a musical standpoint this is usually in a
>>frequency range (low freq, or high freq most often) - that will
>>show up as the difference between them, and that usually happens due to
> some
>>form of processing difference between the files,
>>such as EQ, compression, frequency dependant gain changes, etc. That is
> what
>>I believe you are thinking through, but when
>>talking about straight summing with no gain change (or known equal gain
>
>>changes), we are only looking at linear, one for one
>>comparisons between the two files' frequency representations.
>>
>>Regards,
>>Dedric
>>
>>> Neil wrote:
>>>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>>> The tests I did were completely blank down to -200 dB (far below the
>
>>>>> last
>>>>
>>>>> bit). It's safe to say there is no difference, even in
>>>>> quantization noise, which by technical rights, is considered below the
>
>>>>> level
>>>>
>>>>> of "cancellation" in such tests.
>>>>
>>>> I'm not necessarily talking about just the first bit or the
>>>> last bit, but also everything in between... what happens on bit
>>>> #12, for example? Everything on bit #12 should be audible, but
>>>> in an a/b test what if thre are differences in what bits #8
>>>> through #12 sound like, but the amplutide is stll the same on
>>>> both files at that point, you'll get a null, right? Extrapolate
>>>> that out somewhat & let's say there are differences in bits #8
>>>> through #12 on sample points 3, 17, 1,000, 4,523, 7,560, etc,
>>>> etc through 43,972... Now this is breaking things down well
>>>> beyond what I think can be measured, if I'm not mistaken (I
>>>> dn't know of any way we could extract JUST that information
>>>> from each file & play it back for an a/b test; but would not
>>>> that be enough to have to "null-able" files that do actually
>>>> sound somewhat different?
>>>>
>>>> I guess what I'm saying is that since each sample in a musical
>>>> track or full song file doesn't represent a pure, simple set of
>>>> content like a sample of a sine wave would - there's a whole
>>>> world of harmonic structure in each sample of a song file, and
>>>> I think (although I'll admit - I can't "prove") that there is
>>>> plenty of room for some variables between the first bit & the
>>>> last bit while still allowing for a null test to be successful.
>>>>
>>>> No? Am I wacked out of my mind?
>>>>
>>>> Neil
>>>>
>>
>> You see when I happen to stop by one of these genre songs
> on the radio I always put my own lyrics to them.<
Hehehe......I do this too.............and dude, you don't even want to
know...some of the old Stones songs worked great for this since you can
never understand WTF Mick is singing anyway.
;o)
"Brandon" <a@a.com> wrote in message news:458bf54c@linux...
> Thanks for sharing..
> Actually, I thought it was funny.
> You see when I happen to stop by one of these genre songs
> on the radio I always put my own lyrics to them.
> Usually they are very simular to what you hear on that clip.
> It is the only thing that makes that style of music enjoyable for me.
>
> humming.... it's my d*ck in a box....Mmmmm Mmmmm girl.........
> --
> Thanks,
>
> Brandon
>
>Cmon...how can you resist?
One I absolutely can not resist...
Would you_____ if I asked you to _____?
Would you_____ if I asked you to _____?
Would you_____ ?
On your______?
Would you ________ my _________ tonite?
You can __________________baby?
The possibilities are endless!!!
--
Thanks,
BrandonThis is a multi-part message in MIME format.
------=_NextPart_000_0023_01C725B3.EDA78160
Content-Type: text/plain;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
I'm not getting the visual you have in mind.. diagram it?
AA
"Tom Bruhl" <arpegio@comcast.net> wrote in message =
news:458b7f6f@linux...
Thanks Aaron,
Will this work on the backside of a KVM? I can't put it in front.
I guess if it needed to be in front of the KVM I could just get a=20
single output jobbie and split it at the other side. Do those exist?
"Aaron Allen" <know-spam@not_here.dude> wrote in message =
news:458b23a2@linux...
I hope you're not doing a passive Y split. Look for a VGA splitter =
on google, something like=20
http://www.pcmall.com/pcmall/shop/detail~dpno~767745.asp
They're cheap and OH so worth it
AA
"Tom Bruhl" <arpegio@comcast.net> wrote in message =
news:4589ba0a$1@linux...
Thanks Chris.=20
This graphics stuff is confusing to me.
I have to convert the DVI to VGA because of the 20' length
as I understand. I was going to split it with a Y cable too.
They'll split to a 21"VGA and a 22"LCD. Will that work on=20
the far side of a KVM? I know the resolutions won't match
but I'll only need the VGAs when mixing so I'll adjust the=20
7300 differently for that application. Sooner or later I'll get
4 matching 22" LCDs and quad cards for both comps.
Will this work?
Tom
"Chris Ludwig" <chrisl@adkproaudio.com> wrote in message =
news:45894ef0$1@linux...
HI Tom,
These monitors will be fine with your card. The 7300GT supports =
that=20
resolution on both the VGA and DVI connections.I would run both =
with VGA=20
connections. Having one hooked up DVI and one VGA may make the 2 =
look=20
very different in brightness and such.
Chris
Tom Bruhl wrote:
> My card has one VGA, one DVI and one S Video connector on it.
>
>
> I choose Polesoft Lockspam to fight spam, and you?
> http://www.polesoft.com/refer.html =20
--=20
Chris Ludwig
ADK
chrisl@adkproaudio.com <mailto:chrisl@adkproaudio.com>
www.adkproaudio.com <http://www.adkproaudio.com/>
(859) 635-5762
------=_NextPart_000_0023_01C725B3.EDA78160
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charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.5730.11" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>I'm not getting the visual you =
have in mind..=20
diagram it?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>AA</FONT></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Tom Bruhl" <<A=20
href=3D"mailto:arpegio@comcast.net">arpegio@comcast.net</A>> wrote =
in message=20
<A href=3D"news:458b7f6f@linux">news:458b7f6f@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks Aaron,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Will this work on the backside of a =
KVM? I=20
can't put it in front.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I guess if it needed to be in front =
of the=20
KVM I could just get a </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>single output </FONT><FONT =
face=3DArial=20
size=3D2>jobbie and split it at the other side. Do those=20
exist?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Aaron Allen" <<A=20
=
href=3D"mailto:know-spam@not_here.dude">know-spam@not_here.dude</A>> =
wrote=20
in message <A =
href=3D"news:458b23a2@linux">news:458b23a2@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>I hope you're not doing a passive Y =
split. Look=20
for a VGA splitter on google, something like </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2><A=20
=
href=3D"http://www.pcmall.com/pcmall/shop/detail~dpno~767745.asp">http://=
www.pcmall.com/pcmall/shop/detail~dpno~767745.asp</A></FONT ></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>They're cheap and OH so worth =
it</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>AA</FONT></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Tom Bruhl" <<A=20
href=3D"mailto:arpegio@comcast.net">arpegio@comcast.net</A>> =
wrote in=20
message <A =
href=3D"news:4589ba0a$1@linux">news:4589ba0a$1@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks Chris. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>This graphics stuff is confusing =
to=20
me.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>I have to convert the DVI to VGA =
because of=20
the 20' length</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>as I understand. I was =
going to split=20
it with a Y cable too.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>They'll split to </FONT><FONT =
face=3DArial=20
size=3D2>a 21"VGA and a 22"LCD. Will that work on =
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>the far side of a KVM? I =
know the=20
resolutions won't match</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>but I'll only need the VGAs when =
mixing so=20
I'll adjust the </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>7300 differently for that =
application. =20
Sooner or later I'll get</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>4 matching 22" LCDs and quad =
cards for both=20
comps.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Will this work?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Tom</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<BLOCKQUOTE=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Chris Ludwig" <<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A>>=20
wrote in message <A=20
=
href=3D"news:45894ef0$1@linux">news:45894ef0$1@linux</A>...</DIV>HI=20
Tom,<BR>These monitors will be fine with your card. The 7300GT =
supports=20
that <BR>resolution on both the VGA and DVI connections.I would =
run both=20
with VGA <BR>connections. Having one hooked up DVI and one VGA =
may make=20
the 2 look <BR>very different in brightness and=20
such.<BR>Chris<BR><BR><BR>Tom Bruhl wrote:<BR><BR>> My card =
has one=20
VGA, one DVI and one S Video connector on =
it.<BR>><BR>><BR>> I=20
choose Polesoft Lockspam to fight spam, and you?<BR>> <A=20
=
href=3D"http://www.polesoft.com/refer.html">http://www.polesoft.com/refer=
..html</A> =20
<BR><BR><BR>-- <BR>Chris Ludwig<BR>ADK<BR><A=20
=
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A> <<A =
=
href=3D"mailto:chrisl@adkproaudio.com">mailto:chrisl@adkproaudio.com</A>&=
gt;<BR><A=20
href=3D"http://www.adkproaudio.com">www.adkproaudio.com</A> =
<<A=20
=
href=3D"http://www.adkproaudio.com/">http://www.adkproaudio.com/</A>><=
BR>(859)=20
=
635-5762</BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE ></BODY></HTML=
>
------=_NextPart_000_0023_01C725B3.EDA78160--Actually, I wasn't referring specifically to you, I hear similar things all
the time all over the place. The first time I went through this on this forum
a couple of years ago was the Great CD Burning Speed Debate. In that one
Derek and I came up with about a gazillion ways to show that you could rip-burn-rip-burn
over and over again at all kinds of different speeds and wind up with exactly
the same data or audio CD. And I mean the same as in I slurped the whole
audio file *as a string* into perl and checked the samples. Having done that,
and thereby proven beyond a shadow of a doubt, I was told, roughly, that
clearly I couldn't hear well enough for these esoteric discussions. 'Use
your ears, dude.'
So, it is possible to write a DAW with a filter on the master bus? Yes, of
course it is. Why anyone would want to do such a thing is beyond me since
there are conventions that are pretty much constant throughout the digital
audio world about how signals should be mixed together. So if DAW X is a
little more present in the second to the top octave (I think you mentioned
one being so) I would call that either a bug or mistaken perception. If I
could do the export file, flip polarity trick and the files didn't null I'd
say, 'Interesting, let's be sure my test is good. Is there an EQ on a track
in one mix and not the other? Is there a group track that is doubling the
guitars in one mix and not the other?' If, on the other hand, the tracks
did null I'd say, 'Hmmmmmm, maybe I'm hearing a difference where there isn't
one.'
Lastly, and this is just a quirk for me, I find it odd that musicians and
audio engineers are so disinterested in taking seriously expert opinion.
This is rampant in the audiophile world where off the record the engineers
themselves will tell you they're not sure the $3k speaker cables they used
to hook up their new speaker line makes any difference. But with Neil, for
example, his mixes are 20 times better than mine for that kind of music.
If he gave me advice and opinion I would take it very seriously. But for
some reason when people like me and Dedric, who have developed extensive
knowledge into how computers work, are very often brushed off very quickly.
Dedric isn't even a jerk about, while I'm a jerk about it only sometimes,
so I find that reaction to be, well, odd. But like I said, whatever gets
ya through the day, I'm not looking for converts and nobody is paying me
to post here.
TCB
"LaMont" <jjdpro@ameritech.net> wrote:
>
>Thad, I assume that you ar ereferring to me (using your ears).
>
>Look, I think we are talking about two differnt things here:
>
>1) Digital data
>
>2) Software (DAWS) coding
>
>You and Dedric have been concentrating on the laws of Digital audio. That's
>fine. But, I'm talking about the Software that we use to decode our digital
>audio.
>
>Like my previous post states, are we saying that DAW software can't be written
>for certain sonic results?
>
>
>"TCB" <nobody@ishere.com> wrote:
>>
>>Neil,
>>
>>You're using an analog waveform that is leading you to think incorrectly
>>about sampling. This is (very roughly) how it would look if you're working
>>with 16 bit samples.
>>
>>0101010101010101
>>0101010111011101
>>0101110111010101
>>0101111111010100
>>0101110101111101
>>0111011101111101
>>0111110101110101
>>0100010111000100
>>0100011101010101
>>0001011100010101
>>0000010111111100
>>0001000001010111
>>0100000111110101
>>0111011101010000
>>0101011101000000
>>0101011111000101
>>0101010101010101
>>
>>The easiest way to think of how the sampler works is that it looks at the
>>incoming voltage to the converter and asks 'Is this in the top or bottom
>>half of the possible amplitudes I can measure.' If it's in the top half
>it
>>writes a 1, if it's in the bottom half, it writes a zero. The next bit
asks,
>>'Now that I know which half of my measurable voltage I'm looking at, is
>the
>>voltage in the top half of that half or the bottom half?' That's bit number
>>two. Then it's on to, 'Now that I know what quarter it's it, is it in the
>>top or bottom half of that quarter?' And so on sixteen time giving it a
>resolution
>>of 2 to the sixteenth power.
>>
>>In other words, asking if the bits under the sample would sound is like
>asking
>>how the road would drive if it were 30 feet underground.
>>
>>Now then, to get back to the original argument, people like me (and I think
>>Dedric but I'll let him speak for himself) get a little hacked off when
>someone
>>says, 'you have to use your ears' when it's possible using various computer
>>tools to check exactly how many of those samples match in two given files.
>>The nulling trick is just a very easy way to get a quick read on one aspect,
>>which is to answer the question 'do these two files match?' But there are
>>others and I've used them. And the sameness between properly written (by
>>which I mean lacking in serious bugs) audio applications is startling and
>>their differences so minor that other errors (analog cables, dust on the
>>speaker cone, humidity and temperature in the room) are far more likely
>to
>>cause a difference.
>>
>>Personally I think this all stems from romanticism about music and the
purity
>>of art. I have yet to hear someone tell me they need financial calculations
>>down to 25 decimal points. They need them done to (at most) five decimal
>>points because the smallest commonly used financial divisor is the basis
>>point, or one one hundredth of a penny. So internally you calculate to
five
>>decimal places and round up or down from there and get on with your life.
>>As geeky as finance guys can get, nobody ever says, 'You know, Thad, that
>>last basis point just isn't really punchy enough for this deal. LBO guys
>>need really punchy returns, so can you run that calculation out a few more
>>bits to get a punchier basis point?' Scientists are also extremely careful
>>to keep 'false precision' out of their calculations, so if one instrument
>>will measure to four decimal points and the others will measure to 12 they
>>understand that everything the higher resolution instruments measure beyond
>>four accurate decimal points is worthless. They usually won't even record
>>the data to be sure they don't claim greater precision than they have,
because
>>that's considered a horribly embarrassing junior high school mistake. But
>>musicians and audio engineers think that just because the data is sound
>data
>>somehow it enters a nebulous zone where that last one hundredth of a penny
>>can be punchier. Hey, if it gets you through the day, that's fine by me,
>>but there are things about digital audio that can be proven true or false
>>using the data. For things that can't be proven true or false with the
data
>>itself there is ABY testing, which is a controlled way to use the most
precise
>>audio measuring instruments available (our ears, at least until bats will
>>wear headphones) to see if things sound different. When it's not in the
>data,
>>and it's not in the ABY, I say it doesn't exist.
>>
>>TCB
>>
>>"Neil" <IUOIU@OIU.com> wrote:
>>>
>>>Dedric - first of all, great explanation - esp. your 2nd
>>>paragraph. Next, let's take a look at something in the form of
>>>the best "graph" I can do in this NG's format... let's assume
>>>that each dot in the simple graph below is a sample point on a
>>>segment of a waveform, and let's futher assume that each "I"
>>>below represents four bits (I don't want to make it too
>>>vertically large, for ease of reading) - so we're dealing with
>>>a 16-bit wav file, with the 5th "dot" from the start point on
>>>the left being a full-amplitude, zero-db-line 16 bit sample.
>>>
>>>Now.... really, all I have to do to get a "null" is to have the
>>>amplitude match at each "dot" on the waveform, yes? This, of
>>>course, is a very simplistic graphic example, so bear with
>>>me... but if I have each "dot" matching in amplitude &
>>>therefore can get a null, what about the bits & content thereof
>>>in between the extremes between the maxes & zero-line
>>>crossings? Are you saying that there can be no variables in
>>>sound between those sections that would still result in a null?
>>>What about all the "I"'s that represent bits in between the
>>>maxes & the minimums?
>>>
>>> .
>>> . I .
>>> . I I I . What about the stuff in here?
>>> . I I I I I . .....or in here????
>>>. I I I I I I I .
>>>-------------------------------------
>>> . I I I I I I I .
>>> . I I I I I . Again, what about this region?
>>> . I I I . ... or this region?
>>> . I .
>>> .
>>>
>>>Neil
>>
>I'd say this is a pretty common phenomena...... or the link below wouldn't
exist.
http://www.kissthisguy.com/
FWIW, I found it funny, but a warning would've been appreciated since I had
a 5 year old running around behind me.
AA
"DJ" <nowayjose@dude.net> wrote in message news:458c0305@linux...
>> You see when I happen to stop by one of these genre songs
>> on the radio I always put my own lyrics to them.<
>
> Hehehe......I do this too.............and dude, you don't even want to
> know...some of the old Stones songs worked great for this since you can
> never understand WTF Mick is singing anyway.
>
> ;o)
>
> "Brandon" <a@a.com> wrote in message news:458bf54c@linux...
>> Thanks for sharing..
>> Actually, I thought it was funny.
>> You see when I happen to stop by one of these genre songs
>> on the radio I always put my own lyrics to them.
>> Usually they are very simular to what you hear on that clip.
>> It is the only thing that makes that style of music enjoyable for me.
>>
>> humming.... it's my d*ck in a box....Mmmmm Mmmmm girl.........
>> --
>> Thanks,
>>
>> Brandon
>>
>>
>
>This is a multi-part message in MIME format.
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Okay Aaron I hope I get this right.
The first diagram is what I think might not work due to the position of =
the splitter. I'm afraid it won't amplify the signal if
placed at the end of the line. If I'm wrong then that's great! You'll =
want to widen your view for this one.
Example 1:
=
21" VGA =
monitor & 22" LCD with VGA input
=
2 x 5' extensions
=
Video Splitter (Aarons active type)
=
20' extension
KVM #1 out =
KVM #1 In =20
VGA #1 out
Paris Comp - G550=20
VGA #2 out =20
KVM #2 In =20
KVM #2 out
=
20' extension
=
Video Splitter (Aaron's active type)
=
2 x 5' extensions
=
21" VGA =
monitor & 22" LCD with VGA input
KVM #1 out =
(Shares with above from this point on.)
KVM #3 in =
=20
VGA #1 out
Cubase Comp - 7300GT
DVI out to VGA connector out
KVM #4 in
KVM #2 out =
(Shares with above from this point on.)
=20
=20
OR:
Example 2:
=
=
21" VGA monitor & 22" LCD with VGA input
=
=
2 x 5' extensions
=
Passive Video Splitter (Y =
cable)
=
20' extension
=
KVM #1 out=20
KVM =
#1 In =20
Video amplifier (Like =
splitter in example 1 with only one output)=20
VGA #1 out
Paris Comp - G550=20
VGA #2 out=20
Video amplifier (Like =
splitter in example 1 with only one output)
KVM =
#2 In =20
=
KVM #2 out=20
=
20' extension
=
Passive Video Splitter (Y =
cable)
=
=
2 x 5' extensions
=
=
21" VGA monitor & 22" LCD with VGA input
=
KVM #1 out (Shares with above from this point on.)
KVM =
#3 in =20
Video amplifier (Like =
splitter in example 1 with only one output)
VGA #1 out
Cubase Comp - 7300GT
DVI out to VGA connector out=20
Video amplifier (Like =
splitter in example 1 with only one output)
KVM =
#4 in
=
KVM #2 out (Shares with above from this point on.)
"Aaron Allen" <know-spam@not_here.dude> wrote in message =
news:458c050f@linux...
I'm not getting the visual you have in mind.. diagram it?
AA
"Tom Bruhl" <arpegio@comcast.net> wrote in message =
news:458b7f6f@linux...
Thanks Aaron,
Will this work on the backside of a KVM? I can't put it in front.
I guess if it needed to be in front of the KVM I could just get a=20
single output jobbie and split it at the other side. Do those =
exist?
"Aaron Allen" <know-spam@not_here.dude> wrote in message =
news:458b23a2@linux...
I hope you're not doing a passive Y split. Look for a VGA splitter =
on google, something like=20
http://www.pcmall.com/pcmall/shop/detail~dpno~767745.asp
They're cheap and OH so worth it
AA
"Tom Bruhl" <arpegio@comcast.net> wrote in message =
news:4589ba0a$1@linux...
Thanks Chris.=20
This graphics stuff is confusing to me.
I have to convert the DVI to VGA because of the 20' length
as I understand. I was going to split it with a Y cable too.
They'll split to a 21"VGA and a 22"LCD. Will that work on=20
the far side of a KVM? I know the resolutions won't match
but I'll only need the VGAs when mixing so I'll adjust the=20
7300 differently for that application. Sooner or later I'll get
4 matching 22" LCDs and quad cards for both comps.
Will this work?
Tom
"Chris Ludwig" <chrisl@adkproaudio.com> wrote in message =
news:45894ef0$1@linux...
HI Tom,
These monitors will be fine with your card. The 7300GT =
supports that=20
resolution on both the VGA and DVI connections.I would run =
both with VGA=20
connections. Having one hooked up DVI and one VGA may make the =
2 look=20
very different in brightness and such.
Chris
Tom Bruhl wrote:
> My card has one VGA, one DVI and one S Video connector on =
it.
>
>
> I choose Polesoft Lockspam to fight spam, and you?
> http://www.polesoft.com/refer.html =20
--=20
Chris Ludwig
ADK
chrisl@adkproaudio.com <mailto:chrisl@adkproaudio.com>
www.adkproaudio.com <http://www.adkproaudio.com/>
(859) 635-5762
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charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1400" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Okay Aaron I hope I get this =
right.</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=3DArial size=3D2>The first diagram is what I think might =
not work=20
due to the position of the splitter. I'm afraid it won't amplify =
the=20
signal if</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>placed at the end of the line. If =
I'm wrong=20
then that's great! You'll want to widen your view for this=20
one.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2><U>Example 1:</U></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial=20
size=3D2> &nbs=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; 21"=20
VGA monitor & 22" LCD with VGA input</FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2> &nbs=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
2=20
x 5' extensions</FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2> &nbs=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; Video=20
Splitter (Aarons active type)</FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2> &nbs=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; 20'=20
extension</FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2> &nbs=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; KVM=20
#1 out </FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2> &nbs=
p; &nbs p; &nbs p=
;   ;   ;=
=20
KVM #1 In </FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2> &nbs=
p; &nbs p; &nbs p=
;   ;=20
VGA #1 out</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Paris Comp - G550 </FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2> &nbs=
p; &nbs p; &nbs p=
;   ;=20
VGA #2 out </FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2> &nbs=
p; &nbs p; &nbs p=
;   ;   ;=
=20
KVM #2 In </FONT></DIV>
<DIV><FONT size=3D2><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; KVM=20
#2 out</FONT>
<DIV><FONT face=3DArial=20
size=3D2> &nbs=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; 20'=20
extension</FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2> &nbs=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; Video=20
Splitter (Aaron's active type)</FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2> &nbs=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
2=20
x 5' extensions</FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2> &nbs=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; =20
21" VGA monitor & 22" LCD with VGA input</FONT></DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; KVM=20
#1 out (Shares with above from this point on.)</FONT></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp;KVM=20
#3 in &n bsp; &n b=
sp; &nb sp; &nb s=
p; </FONT></DIV>
<DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp;VGA=20
#1 out</FONT></DIV>
<DIV><FONT face=3DArial>Cubase Comp - 7300GT</FONT></DIV>
<DIV><FONT=20
face=3DArial> =
DVI=20
out to VGA connector out</FONT></DIV>
<DIV><FONT face=3DArial=20
size=3D2> &nbs=
p; &nbs p; &nbs p=
;   ;   ;=
KVM=20
#4 in</FONT></DIV></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; =20
KVM #2 out (Shares with above from this point on.)</FONT></DIV>
<DIV><FONT face=3DArial> </FONT></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; </FONT></DIV>
<DIV><FONT=20
face=3DArial> =
<FONT=20
size=3D5> OR:</FONT></FONT></DIV>
<DIV><U><FONT face=3DArial>Example 2:</FONT></U></DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; 21 " VGA monitor =
& 22"=20
LCD with VGA input</FONT></DIV>
<DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ; =20
2 x 5' extensions</FONT></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; =20
Passive Video Splitter (Y cable)</FONT></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; =20
20' extension</FONT></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;=20
KVM=20
#1 out </FONT></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; =20
KVM #1 In </FONT></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; =20
Video amplifier (Like splitter in example 1 with only one =
output)=20
</FONT></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp;=20
VGA #1 out</FONT></DIV>
<DIV><FONT face=3DArial>Paris Comp - G550 </FONT></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp;=20
VGA #2 out </FONT></DIV>
<DIV> <FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; =20
Video amplifier (Like splitter in example 1 with only one=20
output)</FONT></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; =20
KVM #2 =
In =20
</FONT></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ; =20
KVM #2 out</FONT>=20
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
=20
20' extension</FONT></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
=20
&=
nbsp; & nbsp; =20
Passive Video Splitter (Y cable)</FONT></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
; 2=20
x 5' extensions</FONT></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; =20
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; =20
21" VGA monitor & 22" LCD with VGA input</FONT></DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ; =20
KVM #1 out (Shares with above from this point on.)</FONT></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; =20
KVM=20
#3 in &n bsp; &n b=
sp; &nb sp; &nb s=
p; </FONT></DIV>
<DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; =20
Video amplifier (Like splitter in example 1 with only one=20
output)</FONT></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp;VGA=20
#1 out</FONT></DIV>
<DIV><FONT face=3DArial>Cubase Comp - 7300GT</FONT></DIV><FONT=20
face=3DArial> =
DVI=20
out to VGA connector out</FONT>=20
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; =20
Video amplifier (Like splitter in example 1 with only one=20
output)</FONT></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p;=20
KVM #4 in</FONT></DIV></DIV>
<DIV><FONT=20
face=3DArial> =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ; =20
KVM #2 out (Shares with above from this point =
on.)</FONT></DIV></DIV></DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV>
<DIV><FONT face=3DArial></FONT> </DIV></FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Aaron Allen" <<A=20
=
href=3D"mailto:know-spam@not_here.dude">know-spam@not_here.dude</A>> =
wrote in=20
message <A =
href=3D"news:458c050f@linux">news:458c050f@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>I'm not getting the visual you =
have in=20
mind.. diagram it?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>AA</FONT></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Tom Bruhl" <<A=20
href=3D"mailto:arpegio@comcast.net">arpegio@comcast.net</A>> =
wrote in=20
message <A =
href=3D"news:458b7f6f@linux">news:458b7f6f@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks Aaron,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Will this work on the backside of a =
KVM? =20
I can't put it in front.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I guess if it needed to be in front =
of the=20
KVM I could just get a </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>single output </FONT><FONT =
face=3DArial=20
size=3D2>jobbie and split it at the other side. Do those=20
exist?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Aaron Allen" <<A=20
=
href=3D"mailto:know-spam@not_here.dude">know-spam@not_here.dude</A>>=20
wrote in message <A=20
href=3D"news:458b23a2@linux">news:458b23a2@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>I hope you're not doing a passive =
Y split.=20
Look for a VGA splitter on google, something like =
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2><A=20
=
href=3D"http://www.pcmall.com/pcmall/shop/detail~dpno~767745.asp">http://=
www.pcmall.com/pcmall/shop/detail~dpno~767745.asp</A></FONT ></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>They're cheap and OH so worth =
it</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>AA</FONT></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Tom Bruhl" <<A=20
href=3D"mailto:arpegio@comcast.net">arpegio@comcast.net</A>> =
wrote in=20
message <A=20
=
href=3D"news:4589ba0a$1@linux">news:4589ba0a$1@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>Thanks =
Chris. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>This graphics stuff is =
confusing to=20
me.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>I have to convert the DVI to =
VGA because of=20
the 20' length</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>as I understand. I was =
going to split=20
it with a Y cable too.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>They'll split to </FONT><FONT =
face=3DArial=20
size=3D2>a 21"VGA and a 22"LCD. Will that work on =
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>the far side of a KVM? I =
know the=20
resolutions won't match</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>but I'll only need the VGAs =
when mixing so=20
I'll adjust the </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>7300 differently for that=20
application. Sooner or later I'll get</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>4 matching 22" LCDs and quad =
cards for both=20
comps.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Will this work?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Tom</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<BLOCKQUOTE=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: =
5px; BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Chris Ludwig" <<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A>>=20
wrote in message <A=20
=
href=3D"news:45894ef0$1@linux">news:45894ef0$1@linux</A>...</DIV>HI=20
Tom,<BR>These monitors will be fine with your card. The 7300GT =
supports that <BR>resolution on both the VGA and DVI =
connections.I=20
would run both with VGA <BR>connections. Having one hooked up =
DVI and=20
one VGA may make the 2 look <BR>very different in brightness =
and=20
such.<BR>Chris<BR><BR><BR>Tom Bruhl wrote:<BR><BR>> My card =
has one=20
VGA, one DVI and one S Video connector on =
it.<BR>><BR>><BR>>=20
I choose Polesoft Lockspam to fight spam, and you?<BR>> <A=20
=
href=3D"http://www.polesoft.com/refer.html">http://www.polesoft.com/refer=
..html</A> =20
<BR><BR><BR>-- <BR>Chris Ludwig<BR>ADK<BR><A=20
=
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A> <<A =
=
href=3D"mailto:chrisl@adkproaudio.com">mailto:chrisl@adkproaudio.com</A>&=
gt;<BR><A=20
href=3D"http://www.adkproaudio.com">www.adkproaudio.com</A> =
<<A=20
=
href=3D"http://www.adkproaudio.com/">http://www.adkproaudio.com/</A>><=
BR>(859)=20
=
635-5762</BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE></BLOCKQUOTE ></BLOCKQUOTE>=
</BODY></HTML>
------=_NextPart_000_004D_01C725C1.88C360D0--Congrats, Chris! (and I hate you--I would kill to
do a Guy Clark record--I am such a fan...:) )
"Chris Latham" <latham_c@bellsouth.net> wrote:
>For those of you who will be voting for the 49th annual Grammys, please
give
>a listen to, and consider Guy Clark's "Workbench Songs" for Best
>Contemporary Folk/Americana Album (Catagory 68 in Field 14 - Folk). Edited
>and mixed entirely in PARIS.
>
>THANK YOU,
>Chris
>
>I have to say that is the best Guy Clark CD I've ever heard. I'm no longer
a member of NARAS, but I've been telling everyone who will listen to go and
buy the CD. Great work.
David
>
> For those of you who will be voting for the 49th annual Grammys, please
> give
> a listen to, and consider Guy Clark's "Workbench Songs" for Best
> Contemporary Folk/Americana Album (Catagory 68 in Field 14 - Folk).
> Edited
> and mixed entirely in PARIS.
>
> THANK YOU,
> ChrisThad,
I think your points are valid. However I think the reason most Recording
engineers don't like to talk Audio science, is because of the "unexplainable"
anomolies that occurr with sound. Matters not if it's digital or analog,
but rather how does it sound..
We need factions like AES who discuss such theorectical and new ideas and
advancements in audio reproduction. However, once the science down, then
comes the art of it all.. Music is still the reason for whatwe are discussing.
Music is emotional, yet is a science as well.
For your camp to continue to de-value the human side (use your ears) of the
equation is not right as well.
I think both sides are right, but the science campers cannot speak to a guy
who's main tool is his "ears" and not a scope.
"TCB" <nobody@ishere.com> wrote:
>
>Actually, I wasn't referring specifically to you, I hear similar things
all
>the time all over the place. The first time I went through this on this
forum
>a couple of years ago was the Great CD Burning Speed Debate. In that one
>Derek and I came up with about a gazillion ways to show that you could rip-burn-rip-burn
>over and over again at all kinds of different speeds and wind up with exactly
>the same data or audio CD. And I mean the same as in I slurped the whole
>audio file *as a string* into perl and checked the samples. Having done
that,
>and thereby proven beyond a shadow of a doubt, I was told, roughly, that
>clearly I couldn't hear well enough for these esoteric discussions. 'Use
>your ears, dude.'
>
>So, it is possible to write a DAW with a filter on the master bus? Yes,
of
>course it is. Why anyone would want to do such a thing is beyond me since
>there are conventions that are pretty much constant throughout the digital
>audio world about how signals should be mixed together. So if DAW X is a
>little more present in the second to the top octave (I think you mentioned
>one being so) I would call that either a bug or mistaken perception. If
I
>could do the export file, flip polarity trick and the files didn't null
I'd
>say, 'Interesting, let's be sure my test is good. Is there an EQ on a track
>in one mix and not the other? Is there a group track that is doubling the
>guitars in one mix and not the other?' If, on the other hand, the tracks
>did null I'd say, 'Hmmmmmm, maybe I'm hearing a difference where there isn't
>one.'
>
>Lastly, and this is just a quirk for me, I find it odd that musicians and
>audio engineers are so disinterested in taking seriously expert opinion.
>This is rampant in the audiophile world where off the record the engineers
>themselves will tell you they're not sure the $3k speaker cables they used
>to hook up their new speaker line makes any difference. But with Neil, for
>example, his mixes are 20 times better than mine for that kind of music.
>If he gave me advice and opinion I would take it very seriously. But for
>some reason when people like me and Dedric, who have developed extensive
>knowledge into how computers work, are very often brushed off very quickly.
>Dedric isn't even a jerk about, while I'm a jerk about it only sometimes,
>so I find that reaction to be, well, odd. But like I said, whatever gets
>ya through the day, I'm not looking for converts and nobody is paying me
>to post here.
>
>TCB
>
>"LaMont" <jjdpro@ameritech.net> wrote:
>>
>>Thad, I assume that you ar ereferring to me (using your ears).
>>
>>Look, I think we are talking about two differnt things here:
>>
>>1) Digital data
>>
>>2) Software (DAWS) coding
>>
>>You and Dedric have been concentrating on the laws of Digital audio. That's
>>fine. But, I'm talking about the Software that we use to decode our digital
>>audio.
>>
>>Like my previous post states, are we saying that DAW software can't be
written
>>for certain sonic results?
>>
>>
>>"TCB" <nobody@ishere.com> wrote:
>>>
>>>Neil,
>>>
>>>You're using an analog waveform that is leading you to think incorrectly
>>>about sampling. This is (very roughly) how it would look if you're working
>>>with 16 bit samples.
>>>
>>>0101010101010101
>>>0101010111011101
>>>0101110111010101
>>>0101111111010100
>>>0101110101111101
>>>0111011101111101
>>>0111110101110101
>>>0100010111000100
>>>0100011101010101
>>>0001011100010101
>>>0000010111111100
>>>0001000001010111
>>>0100000111110101
>>>0111011101010000
>>>0101011101000000
>>>0101011111000101
>>>0101010101010101
>>>
>>>The easiest way to think of how the sampler works is that it looks at
the
>>>incoming voltage to the converter and asks 'Is this in the top or bottom
>>>half of the possible amplitudes I can measure.' If it's in the top half
>>it
>>>writes a 1, if it's in the bottom half, it writes a zero. The next bit
>asks,
>>>'Now that I know which half of my measurable voltage I'm looking at, is
>>the
>>>voltage in the top half of that half or the bottom half?' That's bit number
>>>two. Then it's on to, 'Now that I know what quarter it's it, is it in
the
>>>top or bottom half of that quarter?' And so on sixteen time giving it
a
>>resolution
>>>of 2 to the sixteenth power.
>>>
>>>In other words, asking if the bits under the sample would sound is like
>>asking
>>>how the road would drive if it were 30 feet underground.
>>>
>>>Now then, to get back to the original argument, people like me (and I
think
>>>Dedric but I'll let him speak for himself) get a little hacked off when
>>someone
>>>says, 'you have to use your ears' when it's possible using various computer
>>>tools to check exactly how many of those samples match in two given files.
>>>The nulling trick is just a very easy way to get a quick read on one aspect,
>>>which is to answer the question 'do these two files match?' But there
are
>>>others and I've used them. And the sameness between properly written (by
>>>which I mean lacking in serious bugs) audio applications is startling
and
>>>their differences so minor that other errors (analog cables, dust on the
>>>speaker cone, humidity and temperature in the room) are far more likely
>>to
>>>cause a difference.
>>>
>>>Personally I think this all stems from romanticism about music and the
>purity
>>>of art. I have yet to hear someone tell me they need financial calculations
>>>down to 25 decimal points. They need them done to (at most) five decimal
>>>points because the smallest commonly used financial divisor is the basis
>>>point, or one one hundredth of a penny. So internally you calculate to
>five
>>>decimal places and round up or down from there and get on with your life.
>>>As geeky as finance guys can get, nobody ever says, 'You know, Thad, that
>>>last basis point just isn't really punchy enough for this deal. LBO guys
>>>need really punchy returns, so can you run that calculation out a few
more
>>>bits to get a punchier basis point?' Scientists are also extremely careful
>>>to keep 'false precision' out of their calculations, so if one instrument
>>>will measure to four decimal points and the others will measure to 12
they
>>>understand that everything the higher resolution instruments measure beyond
>>>four accurate decimal points is worthless. They usually won't even record
>>>the data to be sure they don't claim greater precision than they have,
>because
>>>that's considered a horribly embarrassing junior high school mistake.
But
>>>musicians and audio engineers think that just because the data is sound
>>data
>>>somehow it enters a nebulous zone where that last one hundredth of a penny
>>>can be punchier. Hey, if it gets you through the day, that's fine by me,
>>>but there are things about digital audio that can be proven true or false
>>>using the data. For things that can't be proven true or false with the
>data
>>>itself there is ABY testing, which is a controlled way to use the most
>precise
>>>audio measuring instruments available (our ears, at least until bats will
>>>wear headphones) to see if things sound different. When it's not in the
>>data,
>>>and it's not in the ABY, I say it doesn't exist.
>>>
>>>TCB
>>>
>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>
>>>>Dedric - first of all, great explanation - esp. your 2nd
>>>>paragraph. Next, let's take a look at something in the form of
>>>>the best "graph" I can do in this NG's format... let's assume
>>>>that each dot in the simple graph below is a sample point on a
>>>>segment of a waveform, and let's futher assume that each "I"
>>>>below represents four bits (I don't want to make it too
>>>>vertically large, for ease of reading) - so we're dealing with
>>>>a 16-bit wav file, with the 5th "dot" from the start point on
>>>>the left being a full-amplitude, zero-db-line 16 bit sample.
>>>>
>>>>Now.... really, all I have to do to get a "null" is to have the
>>>>amplitude match at each "dot" on the waveform, yes? This, of
>>>>course, is a very simplistic graphic example, so bear with
>>>>me... but if I have each "dot" matching in amplitude &
>>>>therefore can get a null, what about the bits & content thereof
>>>>in between the extremes between the maxes & zero-line
>>>>crossings? Are you saying that there can be no variables in
>>>>sound between those sections that would still result in a null?
>>>>What about all the "I"'s that represent bits in between the
>>>>maxes & the minimums?
>>>>
>>>> .
>>>> . I .
>>>> . I I I . What about the stuff in here?
>>>> . I I I I I . .....or in here????
>>>>. I I I I I I I .
>>>>-------------------------------------
>>>> . I I I I I I I .
>>>> . I I I I I . Again, what about this region?
>>>> . I I I . ... or this region?
>>>> . I .
>>>> .
>>>>
>>>>Neil
>>>
>>
>"TCB" <nobody@ishere.com> wrote:
>Lastly, and this is just a quirk for me, I find it odd that musicians and
>audio engineers are so disinterested in taking seriously expert opinion.
>This is rampant in the audiophile world where off the record the engineers
>themselves will tell you they're not sure the $3k speaker cables they used
>to hook up their new speaker line makes any difference. But with Neil, for
>example, his mixes are 20 times better than mine for that kind of music.
>If he gave me advice and opinion I would take it very seriously. But for
>some reason when people like me and Dedric, who have developed extensive
>knowledge into how computers work, are very often brushed off very quickly.
>Dedric isn't even a jerk about, while I'm a jerk about it only sometimes,
>so I find that reaction to be, well, odd. But like I said, whatever gets
>ya through the day, I'm not looking for converts and nobody is paying me
>to post here.
First of all, thanks for the compliment, and secondly, I hope I
haven't come across as one of those who brushes off the facts
stated by those more knowledgeable than I with regard to the
technical aspects of all this stuff - I guess I'm part of the
"Use your ears, but also pay attention to the data" crowd.
I'm the first one to acknowledge that I don't understand some of
of the more technical elements of the digiworld, and it appears
that my interpretations of how certain things work in the
digital realm are/were flawed... so, despite the fact that
you're not paid to post here, I thank you & Dedric & others for
getting into the detail you have... some of it's over my head,
admittedly, but you've explained it clearly enough to where
I "get it" a little bit better.
NeilDedric good post..
However, I have PT-M-Powered/M-audio 410 interface for my laptop and it has
that same sound (no eq, zero fader) that HD does. I know their use the same
48 bit fix mixer. I load up the same file in Nuendo (no eq, zero fader)..results.
different sonic character.
PT having a top end touch..Nuendo, nice smooth(flat) sound. And I'm just
taking about a stereo wav file nulled with no eq..nothing ..zilch..nada..
Now, there are devices (keyboards, dum machines) on the market today that
have a Master Buss Compressor and EQ set to on with the top end notched up.
Why? because it gives their product an competitive advantageover the competition..
Ex: Yahama's Motif ES, Akai's MPC 1000, 2500, Roland's Fantom.
So, why would'nt a DAW manufactuer code in an extra (ooommf) to make their
DAW sound better. Especially, given the "I hate Digtal Summing" crowd? And,
If I'm a DAW manufactuer, what would give my product a sonic edge over the
competition?
We live in the "louder is better" audio world these days, so a DAW that can
catch my attention 'sonically" will probaly will get the sell. That's what
happend to me back in 1997 when I heard Paris. I was floored!!! Still to
this day, nothing has floored me like that "Road House Blues Demo" I heard
on Paris.
Was it the hardware ? was it the software. I remember talking with Edmund
at the 2000 winter Namm, and told me that he & Steve set out to reproduce
the sonics of big buck analog board (eq's) and all.. And, summing was a big
big issue for them because they (ID) thought that nobody has gotten it(summing)
right. And by right, they meant, behaved like a console with a wide lane
for all of those tracks..
"Dedric Terry" <dedric@echomg.com> wrote:
>"LaMont" <jjdpro@ameritech.net> wrote in message news:458be8d5$1@linux...
>>
>> Okay...
>> I guess what I'm saying is this:
>>
>> -Is it possible that diferent DAW manufactuers "code" their app
>> differently
>> for sound results.
>
>Of course it is *possible* to do this, but only if the DAW has a specific
>sound shaping purpose
>beyond normal summing/mixing. Users talk about wanting developers to add
a
>"Neve sound" or "API sound" option to summing engines,
>but that's really impractical given the amount of dsp required to make a
>decent emulation (with convolution, dynamic EQ functions,
>etc). For sake of not eating up all cpu processing, that could likely only
>surface as is a built in EQ, which
>no one wants universally in summing, and anyone can add at will already.
>
>So it hasn't happened yet and isn't likely to as it detours from the basic
>tenant of audio recording - recreate what comes in as
>accurately as possible.
>
>What Digi did in recoding their summing engine was try to recover some
>of the damage done by the 24-bit buss in Mix systems. Motorola 56k dsps
are
>24-bit fixed point chips and I think
>the new generation (321?) still is, but they use double words now for
>48-bits). And though plugins could process at 48-bit by
>doubling up and using upper and lower 24-bit words for 48-bit outputs, the
>buss
>between chips was 24-bits, so they had to dither to 24-bits after every
>plugin. The mixer (if I recall correctly) also
>had a 24-bit buss, so what Digi did is to add a dither stage to the mixer
to
>prevent this
>constant truncation of data. 24-bits isn't enough to cover summing for
more
>than a few tracks without
>losing information in the 16-bit world, and in the 24-bit world some
>information will be lost, at least at the lowest levels.
>
>Adding a dither stage (though I think they did more than that - perhaps
>implement a 48-bit double word stage as well),
>simply smoothed over the truncation that was happening, but it didn't solve
>the problem, so with HD
>they went to a double-word path - throughout I believe, including the path
>between chips. I believe the chips
>are still 24-bit, but by doubling up the processing (yes at a cost of twice
>the overhead), they get a 48-bit engine.
>This not only provided better headroom, but greater resolution. Higher
bit
>depths subdivide the amplitude with greater resolution, and that's
>really where we get the definition of dynamic range - by lowering the signal
>to quantization noise ratio.
>
>With DAWs that use 32-bit floating point math all the way through, the only
>reason for altering the summing
>is by error, and that's an error that would actually be hard to make and
get
>past a very basic alpha stage of testing.
>There is a small difference in fixed point math and floating point math,
or
>at least a theoretical difference in how it affects audio
>in certain cases, but not necessarily in the result for calculating gain
in
>either for the same audio file. Where any differences might show up is
>complicated, and I believe only appear at levels below 24-bit (or in
>headroom with tracks pushed beyond 0dBFS), or when/if
>there areany differences in where each amplitude level is quantized.
>
>Obviously there can be differences if the DAW has to use varying bit depths
>throughout a single summing path to accomodate hardware
>as well as software summing, since there may be truncation or rounding along
>the way, but that impacts the lowest bit
>level, and hence - spacial reproduction, reverb tails perhaps, and "depth",
>not the levels most music so the differences are most
>often more subtle than not. But most modern DAWs have eliminated those
>"rough edges" in the math by increasing the bit depth to accomodate normal
>summing required for mixing audio.
>
>So with Lynn's unity gain summing test (A files on the CD I believe), DAWs
>were never asked to sum beyond 24-bits,
>at least not on the upper end of the dynamic range, so everything that could
>represent 24-bits accurately would cancel. The only ones
>that didn't were ones that had a different bit depth and/or gain structure
>whether hybrid or native
>(e.g. Paris' subtracting 20dB from tracks and adding it to the buss). In
>this case, PTHD cancelled (when I tested it) with
>Nuendo, Samplitude, Logic, etc because the impact of the 48-bit fixed vs.
>32-bit float wasn't a factor.
>
>When trying other tests, even when adding and subtracting gain, Nuendo,
>Sequoia and Sonar cancel - both audibly and
>visually at inaudible levels, which only proves that one isn't making an
>error when calculating basic gain. Since a dB is well defined,
>and the math to add gain is simple, they shouldn't. The fact that they
all
>use 32-bit float all the way through eliminates a difference
>in data structure as well, and this just verifies that. There was a time
>that supposedly Logic (v3, v4?) was partly 24-bit, or so the rumor went,
>but it's 32-bit float all the way through now just as Sonar, Nuendo/Cubase,
>Samplitude/Sequoia, DP, Audition (I presume at least).
>I don't know what Acid or Live use. Saw promotes a fixed point engine,
but
>I don't know if it is still 24-bit, or now 48 bit.
>That was an intentional choice by the developer, but he's the only one I
>know of that stuck with 24-bit for summing
>intentionally, esp. after the Digi Mix system mixer incident.
>
>Long answer, but to sum up, it is certainly physically *possible* for a
>developer to code something differently intentionally, but not
>in reality likely since it would be breaking some basic fixed point or
>floating point math rules. Where the differences really
>showed up in the past is with PT Mix systems where the limitation was really
>significant - e.g. 24 bit with truncation at several stages.
>
>That really isn't such an issue anymore. Given the differences in workflow,
>missing something in workflow or layout differences
>is easy enough to do (e.g. Sonar doesn't have group and busses the way
>Nuendo does, as it's outputs are actually driver outputs,
>not software busses, so in Sonar, busses are actually outputs, and sub
>busses are actually busses in Nuendo. There are no,
>or at least I haven't found the equivalent of a Nuendo group in Sonar -
that
>affects the results of some tests (though not basic
>summing) if not taken into account, but when taken into account, they work
>exactly the same way).
>
>So at least when talking about apps with 32-bit float all the way through,
>it's safe to say (since it has been proven) that summing isn't different
>unless
>there is an error somewhere, or variation in how the user duplicates the
>same mix in two different apps.
>
>Imho, that's actually a very good thing - approaching a more consistent
>basis for recording and mixing from which users can make all
>of the decisions as to how the final product will sound and not be required
>to decide when purchasing a pricey console, and have to
>focus their business on clients who want "that sound". I believe we are
>actually closer to the pure definition of recording now than
>we once were.
>
>Regards,
>Dedric
>
>
>>
>> I the answer is yes, then,the real task is to discover or rather un-cover
>> what's say: Motu's vision of summing, versus Digidesign, versus Steinberg
>> and so on..
>>
>> What's under the hood. To me and others,when Digi re-coded their summing
>> engine, it was obvious that Pro Tools has an obvious top end (8k-10k)
>> bump.
>> Where as Steinberg's summing is very neutral.
>>
>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>Hi Neil,
>>>
>>>Jamie is right. And you aren't wacked out - you are thinking this through
>>
>>>in a reasonable manner, but coming to the wrong
>>>conclusion - easy to do given how confusing digital audio can be. Each
>> word
>>>represents an amplitude
>>>point on a single curve that is changing over time, and can vary with
a
>>
>>>speed up to the Nyquist frequency (as Jamie described).
>>>The complex harmonic content we hear is actually the frequency modulation
>> of
>>>a single waveform,
>>>that over a small amount of time creates the sound we translate - we don't
>>
>>>really hear a single sample at a time,
>>>but thousands of samples at a time (1 sample alone could at most represent
>> a
>>>single positive or negative peak
>>>of a 22,050Hz waveform).
>>>
>>>If one bit doesn't cancel, esp. if it's a higher order bit than number
24,
>>
>>>you may hear, and will see that easily,
>>>and the higher the bit in the dynamic range (higher order) the more
>>>audible
>>
>>>the difference.
>>>Since each bit is 6dB of dynamic range, you can extrapolate how "loud"
>>>that
>>
>>>bit's impact will be
>>>if there is a variation.
>>>
>>>Now, obviously if we are talking about 1 sample in a 44.1k rate song,
then
>>
>>>it simply be a
>>>click (only audible if it's a high enough order bit) instead of an obvious
>>
>>>musical difference, but that should never
>>>happen in a phase cancellation test between identical files higher than
>> bit
>>>24, unless there are clock sync problems,
>>>driver issues, or the DAW is an early alpha version. :-)
>>>
>>>By definition of what DAWs do during playback and record, every audio
>>>stream
>>
>>>has the same point in time (judged by the timeline)
>>>played back sample accurately, one word at a time, at whatever sample
>>>rate
>>
>>>we are using. A phase cancellation test uses that
>>>fact to compare two audio files word for word (and hence bit for bit since
>>
>>>each bit of a 24-bit word would
>>>be at the same bit slot in each 24-bit word). Assuming they are aligned
>> to
>>>the same start point, sample
>>>accurately, and both are the same set of sample words at each sample
>>>point,
>>
>>>bit for bit, and one is phase inverted,
>>>they will cancel through all 24 bits. For two files to cancel completely
>>
>>>for the duration of the file, each and every bit in each word
>>>must be the exact opposite of that same bit position in a word at the
same
>>
>>>sample point. This is why zooming in on an FFT
>>>of the full difference file is valuable as it can show any differences
in
>>
>>>the lower order bits that wouldn't be audible. So even if
>>>there is no audible difference, the visual followup will show if the two
>>
>>>files truly cancel even a levels below hearing, or
>>>outside of a frequency change that we will perceive.
>>>
>>>When they don't cancel, usually there will be way more than 1 bit
>>>difference - it's usually one or more bits in the words for
>>>thousands of samples. From a musical standpoint this is usually in a
>>>frequency range (low freq, or high freq most often) - that will
>>>show up as the difference between them, and that usually happens due to
>> some
>>>form of processing difference between the files,
>>>such as EQ, compression, frequency dependant gain changes, etc. That is
>> what
>>>I believe you are thinking through, but when
>>>talking about straight summing with no gain change (or known equal gain
>>
>>>changes), we are only looking at linear, one for one
>>>comparisons between the two files' frequency representations.
>>>
>>>Regards,
>>>Dedric
>>>
>>>> Neil wrote:
>>>>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>>>> The tests I did were completely blank down to -200 dB (far below the
>>
>>>>>> last
>>>>>
>>>>>> bit). It's safe to say there is no difference, even in
>>>>>> quantization noise, which by technical rights, is considered below
the
>>
>>>>>> level
>>>>>
>>>>>> of "cancellation" in such tests.
>>>>>
>>>>> I'm not necessarily talking about just the first bit or the
>>>>> last bit, but also everything in between... what happens on bit
>>>>> #12, for example? Everything on bit #12 should be audible, but
>>>>> in an a/b test what if thre are differences in what bits #8
>>>>> through #12 sound like, but the amplutide is stll the same on
>>>>> both files at that point, you'll get a null, right? Extrapolate
>>>>> that out somewhat & let's say there are differences in bits #8
>>>>> through #12 on sample points 3, 17, 1,000, 4,523, 7,560, etc,
>>>>> etc through 43,972... Now this is breaking things down well
>>>>> beyond what I think can be measured, if I'm not mistaken (I
>>>>> dn't know of any way we could extract JUST that information
>>>>> from each file & play it back for an a/b test; but would not
>>>>> that be enough to have to "null-able" files that do actually
>>>>> sound somewhat different?
>>>>>
>>>>> I guess what I'm saying is that since each sample in a musical
>>>>> track or full song file doesn't represent a pure, simple set of
>>>>> content like a sample of a sine wave would - there's a whole
>>>>> world of harmonic structure in each sample of a song file, and
>>>>> I think (although I'll admit - I can't "prove") that there is
>>>>> plenty of room for some variables between the first bit & the
>>>>> last bit while still allowing for a null test to be successful.
>>>>>
>>>>> No? Am I wacked out of my mind?
>>>>>
>>>>> Neil
>>>>>
>>>
>>
>
>.....It seems you all have forgot to talk about the difference with listening
from different monitors, with different "room problems", that is summing the
results to our ears as a last instance.....
Erling
"LaMont" <jjdpro@ameritech.net> skrev i melding news:458c0fc6$1@linux...
>
> Thad,
> I think your points are valid. However I think the reason most Recording
> engineers don't like to talk Audio science, is because of the
> "unexplainable"
> anomolies that occurr with sound. Matters not if it's digital or analog,
> but rather how does it sound..
>
> We need factions like AES who discuss such theorectical and new ideas and
> advancements in audio reproduction. However, once the science down, then
> comes the art of it all.. Music is still the reason for whatwe are
> discussing.
> Music is emotional, yet is a science as well.
>
> For your camp to continue to de-value the human side (use your ears) of
> the
> equation is not right as well.
>
> I think both sides are right, but the science campers cannot speak to a
> guy
> who's main tool is his "ears" and not a scope.
>
>
> "TCB" <nobody@ishere.com> wrote:
>>
>>Actually, I wasn't referring specifically to you, I hear similar things
> all
>>the time all over the place. The first time I went through this on this
> forum
>>a couple of years ago was the Great CD Burning Speed Debate. In that one
>>Derek and I came up with about a gazillion ways to show that you could
>>rip-burn-rip-burn
>>over and over again at all kinds of different speeds and wind up with
>>exactly
>>the same data or audio CD. And I mean the same as in I slurped the whole
>>audio file *as a string* into perl and checked the samples. Having done
> that,
>>and thereby proven beyond a shadow of a doubt, I was told, roughly, that
>>clearly I couldn't hear well enough for these esoteric discussions. 'Use
>>your ears, dude.'
>>
>>So, it is possible to write a DAW with a filter on the master bus? Yes,
> of
>>course it is. Why anyone would want to do such a thing is beyond me since
>>there are conventions that are pretty much constant throughout the digital
>>audio world about how signals should be mixed together. So if DAW X is a
>>little more present in the second to the top octave (I think you mentioned
>>one being so) I would call that either a bug or mistaken perception. If
> I
>>could do the export file, flip polarity trick and the files didn't null
> I'd
>>say, 'Interesting, let's be sure my test is good. Is there an EQ on a
>>track
>>in one mix and not the other? Is there a group track that is doubling the
>>guitars in one mix and not the other?' If, on the other hand, the tracks
>>did null I'd say, 'Hmmmmmm, maybe I'm hearing a difference where there
>>isn't
>>one.'
>>
>>Lastly, and this is just a quirk for me, I find it odd that musicians and
>>audio engineers are so disinterested in taking seriously expert opinion.
>>This is rampant in the audiophile world where off the record the engineers
>>themselves will tell you they're not sure the $3k speaker cables they used
>>to hook up their new speaker line makes any difference. But with Neil, for
>>example, his mixes are 20 times better than mine for that kind of music.
>>If he gave me advice and opinion I would take it very seriously. But for
>>some reason when people like me and Dedric, who have developed extensive
>>knowledge into how computers work, are very often brushed off very
>>quickly.
>>Dedric isn't even a jerk about, while I'm a jerk about it only sometimes,
>>so I find that reaction to be, well, odd. But like I said, whatever gets
>>ya through the day, I'm not looking for converts and nobody is paying me
>>to post here.
>>
>>TCB
>>
>>"LaMont" <jjdpro@ameritech.net> wrote:
>>>
>>>Thad, I assume that you ar ereferring to me (using your ears).
>>>
>>>Look, I think we are talking about two differnt things here:
>>>
>>>1) Digital data
>>>
>>>2) Software (DAWS) coding
>>>
>>>You and Dedric have been concentrating on the laws of Digital audio.
>>>That's
>>>fine. But, I'm talking about the Software that we use to decode our
>>>digital
>>>audio.
>>>
>>>Like my previous post states, are we saying that DAW software can't be
> written
>>>for certain sonic results?
>>>
>>>
>>>"TCB" <nobody@ishere.com> wrote:
>>>>
>>>>Neil,
>>>>
>>>>You're using an analog waveform that is leading you to think incorrectly
>>>>about sampling. This is (very roughly) how it would look if you're
>>>>working
>>>>with 16 bit samples.
>>>>
>>>>0101010101010101
>>>>0101010111011101
>>>>0101110111010101
>>>>0101111111010100
>>>>0101110101111101
>>>>0111011101111101
>>>>0111110101110101
>>>>0100010111000100
>>>>0100011101010101
>>>>0001011100010101
>>>>0000010111111100
>>>>0001000001010111
>>>>0100000111110101
>>>>0111011101010000
>>>>0101011101000000
>>>>0101011111000101
>>>>0101010101010101
>>>>
>>>>The easiest way to think of how the sampler works is that it looks at
> the
>>>>incoming voltage to the converter and asks 'Is this in the top or bottom
>>>>half of the possible amplitudes I can measure.' If it's in the top half
>>>it
>>>>writes a 1, if it's in the bottom half, it writes a zero. The next bit
>>asks,
>>>>'Now that I know which half of my measurable voltage I'm looking at, is
>>>the
>>>>voltage in the top half of that half or the bottom half?' That's bit
>>>>number
>>>>two. Then it's on to, 'Now that I know what quarter it's it, is it in
> the
>>>>top or bottom half of that quarter?' And so on sixteen time giving it
> a
>>>resolution
>>>>of 2 to the sixteenth power.
>>>>
>>>>In other words, asking if the bits under the sample would sound is like
>>>asking
>>>>how the road would drive if it were 30 feet underground.
>>>>
>>>>Now then, to get back to the original argument, people like me (and I
> think
>>>>Dedric but I'll let him speak for himself) get a little hacked off when
>>>someone
>>>>says, 'you have to use your ears' when it's possible using various
>>>>computer
>>>>tools to check exactly how many of those samples match in two given
>>>>files.
>>>>The nulling trick is just a very easy wa
|
|
|
|
| Re: Paris Skins / examples 3 [message #76037 is a reply to message #76036] |
Wed, 15 November 2006 22:18 |
Yanoska
Messages: 32
Registered: January 2007
|
Member |
|
|
y to get a quick read on one
>>>>aspect,
>>>>which is to answer the question 'do these two files match?' But there
> are
>>>>others and I've used them. And the sameness between properly written (by
>>>>which I mean lacking in serious bugs) audio applications is startling
> and
>>>>their differences so minor that other errors (analog cables, dust on the
>>>>speaker cone, humidity and temperature in the room) are far more likely
>>>to
>>>>cause a difference.
>>>>
>>>>Personally I think this all stems from romanticism about music and the
>>purity
>>>>of art. I have yet to hear someone tell me they need financial
>>>>calculations
>>>>down to 25 decimal points. They need them done to (at most) five decimal
>>>>points because the smallest commonly used financial divisor is the basis
>>>>point, or one one hundredth of a penny. So internally you calculate to
>>five
>>>>decimal places and round up or down from there and get on with your
>>>>life.
>>>>As geeky as finance guys can get, nobody ever says, 'You know, Thad,
>>>>that
>>>>last basis point just isn't really punchy enough for this deal. LBO guys
>>>>need really punchy returns, so can you run that calculation out a few
> more
>>>>bits to get a punchier basis point?' Scientists are also extremely
>>>>careful
>>>>to keep 'false precision' out of their calculations, so if one
>>>>instrument
>>>>will measure to four decimal points and the others will measure to 12
> they
>>>>understand that everything the higher resolution instruments measure
>>>>beyond
>>>>four accurate decimal points is worthless. They usually won't even
>>>>record
>>>>the data to be sure they don't claim greater precision than they have,
>>because
>>>>that's considered a horribly embarrassing junior high school mistake.
> But
>>>>musicians and audio engineers think that just because the data is sound
>>>data
>>>>somehow it enters a nebulous zone where that last one hundredth of a
>>>>penny
>>>>can be punchier. Hey, if it gets you through the day, that's fine by me,
>>>>but there are things about digital audio that can be proven true or
>>>>false
>>>>using the data. For things that can't be proven true or false with the
>>data
>>>>itself there is ABY testing, which is a controlled way to use the most
>>precise
>>>>audio measuring instruments available (our ears, at least until bats
>>>>will
>>>>wear headphones) to see if things sound different. When it's not in the
>>>data,
>>>>and it's not in the ABY, I say it doesn't exist.
>>>>
>>>>TCB
>>>>
>>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>>
>>>>>Dedric - first of all, great explanation - esp. your 2nd
>>>>>paragraph. Next, let's take a look at something in the form of
>>>>>the best "graph" I can do in this NG's format... let's assume
>>>>>that each dot in the simple graph below is a sample point on a
>>>>>segment of a waveform, and let's futher assume that each "I"
>>>>>below represents four bits (I don't want to make it too
>>>>>vertically large, for ease of reading) - so we're dealing with
>>>>>a 16-bit wav file, with the 5th "dot" from the start point on
>>>>>the left being a full-amplitude, zero-db-line 16 bit sample.
>>>>>
>>>>>Now.... really, all I have to do to get a "null" is to have the
>>>>>amplitude match at each "dot" on the waveform, yes? This, of
>>>>>course, is a very simplistic graphic example, so bear with
>>>>>me... but if I have each "dot" matching in amplitude &
>>>>>therefore can get a null, what about the bits & content thereof
>>>>>in between the extremes between the maxes & zero-line
>>>>>crossings? Are you saying that there can be no variables in
>>>>>sound between those sections that would still result in a null?
>>>>>What about all the "I"'s that represent bits in between the
>>>>>maxes & the minimums?
>>>>>
>>>>> .
>>>>> . I .
>>>>> . I I I . What about the stuff in here?
>>>>> . I I I I I . .....or in here????
>>>>>. I I I I I I I .
>>>>>-------------------------------------
>>>>> . I I I I I I I .
>>>>> . I I I I I . Again, what about this region?
>>>>> . I I I . ... or this region?
>>>>> . I .
>>>>> .
>>>>>
>>>>>Neil
>>>>
>>>
>>
>True, every part of the signal chain is important. But I don't think
anyone here is assuming otherwise.
For testing purposes, the monitors and room can be eliminated as a
variable by using the same monitors and room when auditioning the output
of several DAWs. And clamp your head into the exact same position. :^)
The idea is to design any test so that all variables except the DAWs are
eliminated. Then, if there is a difference, it can only be because the
DAWs handle audio files differently. And if there is no difference, they
handle audio files identically.
Cheers,
-Jamie
www.JamieKrutz.com
erlilo wrote:
> ....It seems you all have forgot to talk about the difference with listening
> from different monitors, with different "room problems", that is summing the
> results to our ears as a last instance.....
>
> Erling
>
>
> "LaMont" <jjdpro@ameritech.net> skrev i melding news:458c0fc6$1@linux...
>> Thad,
>> I think your points are valid. However I think the reason most Recording
>> engineers don't like to talk Audio science, is because of the
>> "unexplainable"
>> anomolies that occurr with sound. Matters not if it's digital or analog,
>> but rather how does it sound..
>>
>> We need factions like AES who discuss such theorectical and new ideas and
>> advancements in audio reproduction. However, once the science down, then
>> comes the art of it all.. Music is still the reason for whatwe are
>> discussing.
>> Music is emotional, yet is a science as well.
>>
>> For your camp to continue to de-value the human side (use your ears) of
>> the
>> equation is not right as well.
>>
>> I think both sides are right, but the science campers cannot speak to a
>> guy
>> who's main tool is his "ears" and not a scope.
>>
>>
>> "TCB" <nobody@ishere.com> wrote:
>>> Actually, I wasn't referring specifically to you, I hear similar things
>> all
>>> the time all over the place. The first time I went through this on this
>> forum
>>> a couple of years ago was the Great CD Burning Speed Debate. In that one
>>> Derek and I came up with about a gazillion ways to show that you could
>>> rip-burn-rip-burn
>>> over and over again at all kinds of different speeds and wind up with
>>> exactly
>>> the same data or audio CD. And I mean the same as in I slurped the whole
>>> audio file *as a string* into perl and checked the samples. Having done
>> that,
>>> and thereby proven beyond a shadow of a doubt, I was told, roughly, that
>>> clearly I couldn't hear well enough for these esoteric discussions. 'Use
>>> your ears, dude.'
>>>
>>> So, it is possible to write a DAW with a filter on the master bus? Yes,
>> of
>>> course it is. Why anyone would want to do such a thing is beyond me since
>>> there are conventions that are pretty much constant throughout the digital
>>> audio world about how signals should be mixed together. So if DAW X is a
>>> little more present in the second to the top octave (I think you mentioned
>>> one being so) I would call that either a bug or mistaken perception. If
>> I
>>> could do the export file, flip polarity trick and the files didn't null
>> I'd
>>> say, 'Interesting, let's be sure my test is good. Is there an EQ on a
>>> track
>>> in one mix and not the other? Is there a group track that is doubling the
>>> guitars in one mix and not the other?' If, on the other hand, the tracks
>>> did null I'd say, 'Hmmmmmm, maybe I'm hearing a difference where there
>>> isn't
>>> one.'
>>>
>>> Lastly, and this is just a quirk for me, I find it odd that musicians and
>>> audio engineers are so disinterested in taking seriously expert opinion.
>>> This is rampant in the audiophile world where off the record the engineers
>>> themselves will tell you they're not sure the $3k speaker cables they used
>>> to hook up their new speaker line makes any difference. But with Neil, for
>>> example, his mixes are 20 times better than mine for that kind of music.
>>> If he gave me advice and opinion I would take it very seriously. But for
>>> some reason when people like me and Dedric, who have developed extensive
>>> knowledge into how computers work, are very often brushed off very
>>> quickly.
>>> Dedric isn't even a jerk about, while I'm a jerk about it only sometimes,
>>> so I find that reaction to be, well, odd. But like I said, whatever gets
>>> ya through the day, I'm not looking for converts and nobody is paying me
>>> to post here.
>>>
>>> TCB
>>>
>>> "LaMont" <jjdpro@ameritech.net> wrote:
>>>> Thad, I assume that you ar ereferring to me (using your ears).
>>>>
>>>> Look, I think we are talking about two differnt things here:
>>>>
>>>> 1) Digital data
>>>>
>>>> 2) Software (DAWS) coding
>>>>
>>>> You and Dedric have been concentrating on the laws of Digital audio.
>>>> That's
>>>> fine. But, I'm talking about the Software that we use to decode our
>>>> digital
>>>> audio.
>>>>
>>>> Like my previous post states, are we saying that DAW software can't be
>> written
>>>> for certain sonic results?
>>>>
>>>>
>>>> "TCB" <nobody@ishere.com> wrote:
>>>>> Neil,
>>>>>
>>>>> You're using an analog waveform that is leading you to think incorrectly
>>>>> about sampling. This is (very roughly) how it would look if you're
>>>>> working
>>>>> with 16 bit samples.
>>>>>
>>>>> 0101010101010101
>>>>> 0101010111011101
>>>>> 0101110111010101
>>>>> 0101111111010100
>>>>> 0101110101111101
>>>>> 0111011101111101
>>>>> 0111110101110101
>>>>> 0100010111000100
>>>>> 0100011101010101
>>>>> 0001011100010101
>>>>> 0000010111111100
>>>>> 0001000001010111
>>>>> 0100000111110101
>>>>> 0111011101010000
>>>>> 0101011101000000
>>>>> 0101011111000101
>>>>> 0101010101010101
>>>>>
>>>>> The easiest way to think of how the sampler works is that it looks at
>> the
>>>>> incoming voltage to the converter and asks 'Is this in the top or bottom
>>>>> half of the possible amplitudes I can measure.' If it's in the top half
>>>> it
>>>>> writes a 1, if it's in the bottom half, it writes a zero. The next bit
>>> asks,
>>>>> 'Now that I know which half of my measurable voltage I'm looking at, is
>>>> the
>>>>> voltage in the top half of that half or the bottom half?' That's bit
>>>>> number
>>>>> two. Then it's on to, 'Now that I know what quarter it's it, is it in
>> the
>>>>> top or bottom half of that quarter?' And so on sixteen time giving it
>> a
>>>> resolution
>>>>> of 2 to the sixteenth power.
>>>>>
>>>>> In other words, asking if the bits under the sample would sound is like
>>>> asking
>>>>> how the road would drive if it were 30 feet underground.
>>>>>
>>>>> Now then, to get back to the original argument, people like me (and I
>> think
>>>>> Dedric but I'll let him speak for himself) get a little hacked off when
>>>> someone
>>>>> says, 'you have to use your ears' when it's possible using various
>>>>> computer
>>>>> tools to check exactly how many of those samples match in two given
>>>>> files.
>>>>> The nulling trick is just a very easy way to get a quick read on one
>>>>> aspect,
>>>>> which is to answer the question 'do these two files match?' But there
>> are
>>>>> others and I've used them. And the sameness between properly written (by
>>>>> which I mean lacking in serious bugs) audio applications is startling
>> and
>>>>> their differences so minor that other errors (analog cables, dust on the
>>>>> speaker cone, humidity and temperature in the room) are far more likely
>>>> to
>>>>> cause a difference.
>>>>>
>>>>> Personally I think this all stems from romanticism about music and the
>>> purity
>>>>> of art. I have yet to hear someone tell me they need financial
>>>>> calculations
>>>>> down to 25 decimal points. They need them done to (at most) five decimal
>>>>> points because the smallest commonly used financial divisor is the basis
>>>>> point, or one one hundredth of a penny. So internally you calculate to
>>> five
>>>>> decimal places and round up or down from there and get on with your
>>>>> life.
>>>>> As geeky as finance guys can get, nobody ever says, 'You know, Thad,
>>>>> that
>>>>> last basis point just isn't really punchy enough for this deal. LBO guys
>>>>> need really punchy returns, so can you run that calculation out a few
>> more
>>>>> bits to get a punchier basis point?' Scientists are also extremely
>>>>> careful
>>>>> to keep 'false precision' out of their calculations, so if one
>>>>> instrument
>>>>> will measure to four decimal points and the others will measure to 12
>> they
>>>>> understand that everything the higher resolution instruments measure
>>>>> beyond
>>>>> four accurate decimal points is worthless. They usually won't even
>>>>> record
>>>>> the data to be sure they don't claim greater precision than they have,
>>> because
>>>>> that's considered a horribly embarrassing junior high school mistake.
>> But
>>>>> musicians and audio engineers think that just because the data is sound
>>>> data
>>>>> somehow it enters a nebulous zone where that last one hundredth of a
>>>>> penny
>>>>> can be punchier. Hey, if it gets you through the day, that's fine by me,
>>>>> but there are things about digital audio that can be proven true or
>>>>> false
>>>>> using the data. For things that can't be proven true or false with the
>>> data
>>>>> itself there is ABY testing, which is a controlled way to use the most
>>> precise
>>>>> audio measuring instruments available (our ears, at least until bats
>>>>> will
>>>>> wear headphones) to see if things sound different. When it's not in the
>>>> data,
>>>>> and it's not in the ABY, I say it doesn't exist.
>>>>>
>>>>> TCB
>>>>>
>>>>> "Neil" <IUOIU@OIU.com> wrote:
>>>>>> Dedric - first of all, great explanation - esp. your 2nd
>>>>>> paragraph. Next, let's take a look at something in the form of
>>>>>> the best "graph" I can do in this NG's format... let's assume
>>>>>> that each dot in the simple graph below is a sample point on a
>>>>>> segment of a waveform, and let's futher assume that each "I"
>>>>>> below represents four bits (I don't want to make it too
>>>>>> vertically large, for ease of reading) - so we're dealing with
>>>>>> a 16-bit wav file, with the 5th "dot" from the start point on
>>>>>> the left being a full-amplitude, zero-db-line 16 bit sample.
>>>>>>
>>>>>> Now.... really, all I have to do to get a "null" is to have the
>>>>>> amplitude match at each "dot" on the waveform, yes? This, of
>>>>>> course, is a very simplistic graphic example, so bear with
>>>>>> me... but if I have each "dot" matching in amplitude &
>>>>>> therefore can get a null, what about the bits & content thereof
>>>>>> in between the extremes between the maxes & zero-line
>>>>>> crossings? Are you saying that there can be no variables in
>>>>>> sound between those sections that would still result in a null?
>>>>>> What about all the "I"'s that represent bits in between the
>>>>>> maxes & the minimums?
>>>>>>
>>>>>> .
>>>>>> . I .
>>>>>> . I I I . What about the stuff in here?
>>>>>> . I I I I I . .....or in here????
>>>>>> . I I I I I I I .
>>>>>> -------------------------------------
>>>>>> . I I I I I I I .
>>>>>> . I I I I I . Again, what about this region?
>>>>>> . I I I . ... or this region?
>>>>>> . I .
>>>>>> .
>>>>>>
>>>>>> Neil
>
>First of all, I don't represent a 'camp.' Second, music is emotional for us,
not the computer. The computer doesn't care if it's calculating reverb tails
or running SQL queries. So there _are_ some things about digital audio that
we can say are true or false absolutely, or to a certain predictable degree
of error. To argue against those things with the 'use your ears' argument
is as useful as arguing about gravity, and whether you believe in gravity
or not you still ain't gonna fall up when you jump off the park bench. For
the rest, I've never said here or anywhere that ears shouldn't be used, but
only that claims should be backed up by repeatable, statistically significant
results in careful tests. So 'everybody who can hear will hear this' doesn't
cut it for me. 'We did this test under these conditions and these were the
results' does.
TCB
"LaMont" <jjdpro@ameritech.net> wrote:
>
>Thad,
>I think your points are valid. However I think the reason most Recording
>engineers don't like to talk Audio science, is because of the "unexplainable"
>anomolies that occurr with sound. Matters not if it's digital or analog,
>but rather how does it sound..
>
>We need factions like AES who discuss such theorectical and new ideas and
>advancements in audio reproduction. However, once the science down, then
>comes the art of it all.. Music is still the reason for whatwe are discussing.
>Music is emotional, yet is a science as well.
>
>For your camp to continue to de-value the human side (use your ears) of
the
>equation is not right as well.
>
>I think both sides are right, but the science campers cannot speak to a
guy
>who's main tool is his "ears" and not a scope.
>
>
>"TCB" <nobody@ishere.com> wrote:
>>
>>Actually, I wasn't referring specifically to you, I hear similar things
>all
>>the time all over the place. The first time I went through this on this
>forum
>>a couple of years ago was the Great CD Burning Speed Debate. In that one
>>Derek and I came up with about a gazillion ways to show that you could
rip-burn-rip-burn
>>over and over again at all kinds of different speeds and wind up with exactly
>>the same data or audio CD. And I mean the same as in I slurped the whole
>>audio file *as a string* into perl and checked the samples. Having done
>that,
>>and thereby proven beyond a shadow of a doubt, I was told, roughly, that
>>clearly I couldn't hear well enough for these esoteric discussions. 'Use
>>your ears, dude.'
>>
>>So, it is possible to write a DAW with a filter on the master bus? Yes,
>of
>>course it is. Why anyone would want to do such a thing is beyond me since
>>there are conventions that are pretty much constant throughout the digital
>>audio world about how signals should be mixed together. So if DAW X is
a
>>little more present in the second to the top octave (I think you mentioned
>>one being so) I would call that either a bug or mistaken perception. If
>I
>>could do the export file, flip polarity trick and the files didn't null
>I'd
>>say, 'Interesting, let's be sure my test is good. Is there an EQ on a track
>>in one mix and not the other? Is there a group track that is doubling the
>>guitars in one mix and not the other?' If, on the other hand, the tracks
>>did null I'd say, 'Hmmmmmm, maybe I'm hearing a difference where there
isn't
>>one.'
>>
>>Lastly, and this is just a quirk for me, I find it odd that musicians and
>>audio engineers are so disinterested in taking seriously expert opinion.
>>This is rampant in the audiophile world where off the record the engineers
>>themselves will tell you they're not sure the $3k speaker cables they used
>>to hook up their new speaker line makes any difference. But with Neil,
for
>>example, his mixes are 20 times better than mine for that kind of music.
>>If he gave me advice and opinion I would take it very seriously. But for
>>some reason when people like me and Dedric, who have developed extensive
>>knowledge into how computers work, are very often brushed off very quickly.
>>Dedric isn't even a jerk about, while I'm a jerk about it only sometimes,
>>so I find that reaction to be, well, odd. But like I said, whatever gets
>>ya through the day, I'm not looking for converts and nobody is paying me
>>to post here.
>>
>>TCB
>>
>>"LaMont" <jjdpro@ameritech.net> wrote:
>>>
>>>Thad, I assume that you ar ereferring to me (using your ears).
>>>
>>>Look, I think we are talking about two differnt things here:
>>>
>>>1) Digital data
>>>
>>>2) Software (DAWS) coding
>>>
>>>You and Dedric have been concentrating on the laws of Digital audio.
That's
>>>fine. But, I'm talking about the Software that we use to decode our digital
>>>audio.
>>>
>>>Like my previous post states, are we saying that DAW software can't be
>written
>>>for certain sonic results?
>>>
>>>
>>>"TCB" <nobody@ishere.com> wrote:
>>>>
>>>>Neil,
>>>>
>>>>You're using an analog waveform that is leading you to think incorrectly
>>>>about sampling. This is (very roughly) how it would look if you're working
>>>>with 16 bit samples.
>>>>
>>>>0101010101010101
>>>>0101010111011101
>>>>0101110111010101
>>>>0101111111010100
>>>>0101110101111101
>>>>0111011101111101
>>>>0111110101110101
>>>>0100010111000100
>>>>0100011101010101
>>>>0001011100010101
>>>>0000010111111100
>>>>0001000001010111
>>>>0100000111110101
>>>>0111011101010000
>>>>0101011101000000
>>>>0101011111000101
>>>>0101010101010101
>>>>
>>>>The easiest way to think of how the sampler works is that it looks at
>the
>>>>incoming voltage to the converter and asks 'Is this in the top or bottom
>>>>half of the possible amplitudes I can measure.' If it's in the top half
>>>it
>>>>writes a 1, if it's in the bottom half, it writes a zero. The next bit
>>asks,
>>>>'Now that I know which half of my measurable voltage I'm looking at,
is
>>>the
>>>>voltage in the top half of that half or the bottom half?' That's bit
number
>>>>two. Then it's on to, 'Now that I know what quarter it's it, is it in
>the
>>>>top or bottom half of that quarter?' And so on sixteen time giving it
>a
>>>resolution
>>>>of 2 to the sixteenth power.
>>>>
>>>>In other words, asking if the bits under the sample would sound is like
>>>asking
>>>>how the road would drive if it were 30 feet underground.
>>>>
>>>>Now then, to get back to the original argument, people like me (and I
>think
>>>>Dedric but I'll let him speak for himself) get a little hacked off when
>>>someone
>>>>says, 'you have to use your ears' when it's possible using various computer
>>>>tools to check exactly how many of those samples match in two given files.
>>>>The nulling trick is just a very easy way to get a quick read on one
aspect,
>>>>which is to answer the question 'do these two files match?' But there
>are
>>>>others and I've used them. And the sameness between properly written
(by
>>>>which I mean lacking in serious bugs) audio applications is startling
>and
>>>>their differences so minor that other errors (analog cables, dust on
the
>>>>speaker cone, humidity and temperature in the room) are far more likely
>>>to
>>>>cause a difference.
>>>>
>>>>Personally I think this all stems from romanticism about music and the
>>purity
>>>>of art. I have yet to hear someone tell me they need financial calculations
>>>>down to 25 decimal points. They need them done to (at most) five decimal
>>>>points because the smallest commonly used financial divisor is the basis
>>>>point, or one one hundredth of a penny. So internally you calculate to
>>five
>>>>decimal places and round up or down from there and get on with your life.
>>>>As geeky as finance guys can get, nobody ever says, 'You know, Thad,
that
>>>>last basis point just isn't really punchy enough for this deal. LBO guys
>>>>need really punchy returns, so can you run that calculation out a few
>more
>>>>bits to get a punchier basis point?' Scientists are also extremely careful
>>>>to keep 'false precision' out of their calculations, so if one instrument
>>>>will measure to four decimal points and the others will measure to 12
>they
>>>>understand that everything the higher resolution instruments measure
beyond
>>>>four accurate decimal points is worthless. They usually won't even record
>>>>the data to be sure they don't claim greater precision than they have,
>>because
>>>>that's considered a horribly embarrassing junior high school mistake.
>But
>>>>musicians and audio engineers think that just because the data is sound
>>>data
>>>>somehow it enters a nebulous zone where that last one hundredth of a
penny
>>>>can be punchier. Hey, if it gets you through the day, that's fine by
me,
>>>>but there are things about digital audio that can be proven true or false
>>>>using the data. For things that can't be proven true or false with the
>>data
>>>>itself there is ABY testing, which is a controlled way to use the most
>>precise
>>>>audio measuring instruments available (our ears, at least until bats
will
>>>>wear headphones) to see if things sound different. When it's not in the
>>>data,
>>>>and it's not in the ABY, I say it doesn't exist.
>>>>
>>>>TCB
>>>>
>>>>"Neil" <IUOIU@OIU.com> wrote:
>>>>>
>>>>>Dedric - first of all, great explanation - esp. your 2nd
>>>>>paragraph. Next, let's take a look at something in the form of
>>>>>the best "graph" I can do in this NG's format... let's assume
>>>>>that each dot in the simple graph below is a sample point on a
>>>>>segment of a waveform, and let's futher assume that each "I"
>>>>>below represents four bits (I don't want to make it too
>>>>>vertically large, for ease of reading) - so we're dealing with
>>>>>a 16-bit wav file, with the 5th "dot" from the start point on
>>>>>the left being a full-amplitude, zero-db-line 16 bit sample.
>>>>>
>>>>>Now.... really, all I have to do to get a "null" is to have the
>>>>>amplitude match at each "dot" on the waveform, yes? This, of
>>>>>course, is a very simplistic graphic example, so bear with
>>>>>me... but if I have each "dot" matching in amplitude &
>>>>>therefore can get a null, what about the bits & content thereof
>>>>>in between the extremes between the maxes & zero-line
>>>>>crossings? Are you saying that there can be no variables in
>>>>>sound between those sections that would still result in a null?
>>>>>What about all the "I"'s that represent bits in between the
>>>>>maxes & the minimums?
>>>>>
>>>>> .
>>>>> . I .
>>>>> . I I I . What about the stuff in here?
>>>>> . I I I I I . .....or in here????
>>>>>. I I I I I I I .
>>>>>-------------------------------------
>>>>> . I I I I I I I .
>>>>> . I I I I I . Again, what about this region?
>>>>> . I I I . ... or this region?
>>>>> . I .
>>>>> .
>>>>>
>>>>>Neil
>>>>
>>>
>>
>"TCB" <nobody@ishere.com> wrote:
"As geeky as finance guys can get, nobody ever says, 'You know, Thad, that
last basis point just isn't really punchy enough for this deal. LBO guys
need really punchy returns, so can you run that calculation out a few more
bits to get a punchier basis point?'"
Don't be so sure. A long, long time ago I wrote some "summing" code for
a general ledger. One of the outputs of the system was an income statement.
A long discussion ensued, amongst some very bright people, about where to
do the rounding. There were many camps.
Were we to round the individual transactions within a GL number, and sum
these?
Were we to sum the individual transactions unrounded, then round the total
of the GL number?
Were we to sum the unrounded GL numbers associated with a specific income
statement line, then round that total?
People are funny.
Chuck"j to the c r o n" <fu@hydrorecords.com> wrote:
>wouldn't be so bad if it was actually funny
>
>PS. Why do people link to stuff they find offensive? :P
I was curious what other people think. It doesn't matter if it's funny to
an adult, my concern is the kids that saw this, it was on regular television.
Where do we draw the line? Where are our standards?
There are many different subjects that could offend many different kinds
of people. It shouldn't matter if it's funny or not, it should matter if
it is appropriate. The schmucks that run the entertainment business seem
to want to twist and pervert everything. I know that this wasn't pointed
directly at Christmas, but, can't they just leave Christmas alone?
Nobody wants a moderator, or somebody to stomp on their freedom of speech,
but what about decency standards? With out them a lot of people could be
offended, including the Schmucks that run the entertainment business.
Just my thoughts. Maybe not so appropriate for this NG.
James
>
>"James McCloskey" <excelsm@hotmail.com> wrote in message
>news:458aefa2$1@linux...
>>
>> http://www.nbc.com/Saturday_Night_Live/uncensored.shtml
>
>Tom Bruhl GOOOODDDD.....KVM BBBBBAAAAADDDDDD
Tom Bruhl wrote:
> Okay Aaron I hope I get this right.
>
> The first diagram is what I think might not work due to the position of
> the splitter. I'm afraid it won't amplify the signal if
> placed at the end of the line. If I'm wrong then that's great! You'll
> want to widen your view for this one.
>
> _Example 1:_
>
>
> 21"
> VGA monitor & 22" LCD with VGA input
> 2
> x 5' extensions
> Video
> Splitter (Aarons active type)
> 20'
> extension
> KVM #1 out
> KVM #1 In
> VGA #1 out
> Paris Comp - G550
> VGA #2 out
> KVM #2 In
> KVM #2 out
> 20'
> extension
> Video
> Splitter (Aaron's active type)
> 2
> x 5' extensions
>
> 21" VGA monitor & 22" LCD with VGA input
>
>
>
> KVM #1 out
> (Shares with above from this point on.)
> KVM
> #3 in
> VGA #1 out
> Cubase Comp - 7300GT
> DVI out to VGA connector out
> KVM #4 in
> KVM #2 out
> (Shares with above from this point on.)
>
>
> OR:
> _Example 2:_
>
> 21"
> VGA monitor & 22" LCD with VGA input
>
> 2 x 5' extensions
>
> Passive Video Splitter (Y cable)
>
> 20' extension
>
> KVM #1 out
> KVM
> #1 In
> Video amplifier (Like
> splitter in example 1 with only one output)
> VGA #1 out
> Paris Comp - G550
> VGA #2 out
> Video amplifier (Like
> splitter in example 1 with only one output)
>
> KVM #2 In
>
> KVM #2 out
>
> 20' extension
>
> Passive Video Splitter (Y cable)
> 2
> x 5' extensions
>
> 21" VGA
> monitor & 22" LCD with VGA input
>
>
>
>
> KVM #1 out (Shares with above from this point on.)
> KVM
> #3 in
> Video amplifier (Like
> splitter in example 1 with only one output)
> VGA #1 out
> Cubase Comp - 7300GT
> DVI out to VGA connector out
> Video amplifier (Like
> splitter in example 1 with only one output)
> KVM #4 in
>
> KVM #2 out (Shares with above from this point on.)
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
>
> "Aaron Allen" <know-spam@not_here.dude
> <mailto:know-spam@not_here.dude>> wrote in message
> news:458c050f@linux...
> I'm not getting the visual you have in mind.. diagram it?
>
> AA
>
> "Tom Bruhl" <arpegio@comcast.net <mailto:arpegio@comcast.net>>
> wrote in message news:458b7f6f@linux...
> Thanks Aaron,
> Will this work on the backside of a KVM? I can't put it in front.
> I guess if it needed to be in front of the KVM I could just get a
> single output jobbie and split it at the other side. Do those
> exist?
>
>
> "Aaron Allen" <know-spam@not_here.dude
> <mailto:know-spam@not_here.dude>> wrote in message
> news:458b23a2@linux...
> I hope you're not doing a passive Y split. Look for a VGA
> splitter on google, something like
>
> http://www.pcmall.com/pcmall/shop/detail~dpno~767745.asp
>
> They're cheap and OH so worth it
>
> AA
>
> "Tom Bruhl" <arpegio@comcast.net
> <mailto:arpegio@comcast.net>> wrote in message
> news:4589ba0a$1@linux...
> Thanks Chris.
> This graphics stuff is confusing to me.
>
> I have to convert the DVI to VGA because of the 20' length
> as I understand. I was going to split it with a Y cable
> too.
> They'll split to a 21"VGA and a 22"LCD. Will that work on
> the far side of a KVM? I know the resolutions won't match
> but I'll only need the VGAs when mixing so I'll adjust the
> 7300 differently for that application. Sooner or later
> I'll get
> 4 matching 22" LCDs and quad cards for both comps.
>
> Will this work?
> Tom
>
>
>
>
>
>
>
> "Chris Ludwig" <chrisl@adkproaudio.com
> <mailto:chrisl@adkproaudio.com>> wrote in message
> news:45894ef0$1@linux...
> HI Tom,
> These monitors will be fine with your card. The
> 7300GT supports that
> resolution on both the VGA and DVI connections.I
> would run both with VGA
> connections. Having one hooked up DVI and one VGA
> may make the 2 look
> very different in brightness and such.
> Chris
>
>
> Tom Bruhl wrote:
>
> > My card has one VGA, one DVI and one S Video
> connector on it.
> >
> >
> > I choose Polesoft Lockspam to fight spam, and you?
> > http://www.polesoft.com/refer.html
>
>
> --
> Chris Ludwig
> ADK
> chrisl@adkproaudio.com
> <mailto:chrisl@adkproaudio.com>
> <mailto:chrisl@adkproaudio.com>
> www.adkproaudio.com <http://www.adkproaudio.com>
> <http://www.adkproaudio.com/>
> (859) 635-5762
--
Chris Ludwig
ADK Pro Audio
(859) 635-5762
www.adkproaudio.com
chrisl@adkproaudio.comBut were the totals warm and punchy? If so, then you did the summing right.
Seriously, though, it can matter when this kind of rounding takes place.
A lot of times it's getting statements from custodial banks to match what's
internal, and they might round differently than the logic of the internal
database. And if you're running, say, $19 billion plus those rounding errors
can add up to more than the cost of a smoothie or two. Still, in contrast
to audio people the finance types are interested in managing inevitable imprecision
instead of finding precision where there really is none. At least usually
they are . . .
TCB
"chuck duffy" <c@c.com> wrote:
>
>"TCB" <nobody@ishere.com> wrote:
>
>"As geeky as finance guys can get, nobody ever says, 'You know, Thad, that
>last basis point just isn't really punchy enough for this deal. LBO guys
>need really punchy returns, so can you run that calculation out a few more
>bits to get a punchier basis point?'"
>
>Don't be so sure. A long, long time ago I wrote some "summing" code for
>a general ledger. One of the outputs of the system was an income statement.
>
>
>A long discussion ensued, amongst some very bright people, about where to
>do the rounding. There were many camps.
>
>Were we to round the individual transactions within a GL number, and sum
>these?
>
>Were we to sum the individual transactions unrounded, then round the total
>of the GL number?
>
>Were we to sum the unrounded GL numbers associated with a specific income
>statement line, then round that total?
>
>People are funny.
>
>Chuck
>
>This is a multi-part message in MIME format.
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boundary="----=_NextPart_001_000A_01C725E2.462A6A90"
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charset="iso-8859-1"
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Hi Chris,
My dual KVM works great as is. Is the dual monitor idea so bad? How =
else will I be able to sit=20
in my sweet spot and bounce between 2 and maybe even three comps with =
one mouse/keyboard?
They make quad KVMs. I might end up with that in the end. 3 comps each =
with 4 head cards,=20
four monitors, one keyboard/mouse.=20
Why is the KVM soooooo Baaaad? How else can I achieve this?
This other scenario would be even more complex than described in the =
diagram below.
The ultimate setup would have two QWERTY keyboards/mice with KVM sharing =
with Paris/Cubase=20
and Cubase independently.
The room as it relates to the photo:
My MIDI setup mimics what is shown to left of the
mixing position here except there's a desk spot for legs on the right =
along
with a controller. I'd like a 2 x 22" LCD widescreens, QWERTY & mouse=20
there along with the KVM QWERTY/mouse in the center as shown for Paris
and mixdown work.
What would you suggest?
Getting more confused by the minute...
Tom
"Chris Ludwig" <chrisl@adkproaudio.com> wrote in message =
news:458c3c62$1@linux...
Tom Bruhl GOOOODDDD.....KVM BBBBBAAAAADDDDDD
Tom Bruhl wrote:
> Okay Aaron I hope I get this right.
> =20
> The first diagram is what I think might not work due to the position =
of=20
> the splitter. I'm afraid it won't amplify the signal if
> placed at the end of the line. If I'm wrong then that's great! =
You'll=20
> want to widen your view for this one.
> =20
> _Example 1:_
> =20
> =20
> =
21" =
> VGA monitor & 22" LCD with VGA input
> =
2=20
> x 5' extensions
> =
Video=20
> Splitter (Aarons active type)
> =
20'=20
> extension
> KVM #1 =
out
> KVM #1 In=20
> VGA #1 out
> Paris Comp - G550=20
> VGA #2 out =20
> KVM #2 In=20
> KVM #2 =
out
> =
20'=20
> extension
> =
Video=20
> Splitter (Aaron's active type)
> =
2=20
> x 5' extensions
> =
=20
> 21" VGA monitor & 22" LCD with VGA input
> =20
> =20
> =20
> KVM #1 =
out=20
> (Shares with above from this point on.)
> KVM=20
> #3 in =20
> VGA #1 out
> Cubase Comp - 7300GT
> DVI out to VGA connector out
> KVM #4 in
> KVM #2 =
out=20
> (Shares with above from this point on.)
> =20
> =20
> OR:
> _Example 2:_
> =20
> =
=
21"=20
> VGA monitor & 22" LCD with VGA input
> =
=
=20
> 2 x 5' extensions
> =
=20
> Passive Video Splitter (Y cable)
> =
=20
> 20' extension
> =
=20
> KVM #1 out
> =
KVM=20
> #1 In=20
> Video amplifier (Like =
> splitter in example 1 with only one output)
> VGA #1 out
> Paris Comp - G550=20
> VGA #2 out
> Video amplifier (Like =
> splitter in example 1 with only one output)
> =20
> KVM #2 In=20
> =
=20
> KVM #2 out
> =
=20
> 20' extension
> =
=20
> Passive Video Splitter (Y cable)
> =
=
2=20
> x 5' extensions
> =
=20
> 21" VGA =
> monitor & 22" LCD with VGA input
> =20
> =20
> =20
> =
=20
> KVM #1 out (Shares with above from this point on.)
> =
KVM=20
> #3 in =20
> Video amplifier (Like=20
> splitter in example 1 with only one output)
> VGA #1 out
> Cubase Comp - 7300GT
> DVI out to VGA connector out
> Video amplifier (Like=20
> splitter in example 1 with only one output)
> =
KVM #4 in
> =
=20
> KVM #2 out (Shares with above from this point on.)
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
> =20
>=20
> "Aaron Allen" <know-spam@not_here.dude
> <mailto:know-spam@not_here.dude>> wrote in message
> news:458c050f@linux...
> I'm not getting the visual you have in mind.. diagram it?
> =20
> AA
>=20
> "Tom Bruhl" <arpegio@comcast.net =
<mailto:arpegio@comcast.net>>
> wrote in message news:458b7f6f@linux...
> Thanks Aaron,
> Will this work on the backside of a KVM? I can't put it in =
front.
> I guess if it needed to be in front of the KVM I could just =
get a
> single output jobbie and split it at the other side. Do =
those
> exist?
> =20
>=20
> "Aaron Allen" <know-spam@not_here.dude
> <mailto:know-spam@not_here.dude>> wrote in message
> news:458b23a2@linux...
> I hope you're not doing a passive Y split. Look for a =
VGA
> splitter on google, something like
> =20
> http://www.pcmall.com/pcmall/shop/detail~dpno~767745.asp
> =20
> They're cheap and OH so worth it
> =20
> AA
>=20
> "Tom Bruhl" <arpegio@comcast.net
> <mailto:arpegio@comcast.net>> wrote in message
> news:4589ba0a$1@linux...
> Thanks Chris.=20
> This graphics stuff is confusing to me.
> =20
> I have to convert the DVI to VGA because of the 20' =
length
> as I understand. I was going to split it with a Y =
cable
> too.
> They'll split to a 21"VGA and a 22"LCD. Will that =
work on
> the far side of a KVM? I know the resolutions won't =
match
> but I'll only need the VGAs when mixing so I'll =
adjust the
> 7300 differently for that application. Sooner or =
later
> I'll get
> 4 matching 22" LCDs and quad cards for both comps.
> =20
> Will this work?
> Tom
> =20
> =20
> =20
> =20
> =20
> =20
>=20
> "Chris Ludwig" <chrisl@adkproaudio.com
> <mailto:chrisl@adkproaudio.com>> wrote in =
message
> news:45894ef0$1@linux...
> HI Tom,
> These monitors will be fine with your card. The
> 7300GT supports that
> resolution on both the VGA and DVI connections.I
> would run both with VGA
> connections. Having one hooked up DVI and one =
VGA
> may make the 2 look
> very different in brightness and such.
> Chris
>=20
>=20
> Tom Bruhl wrote:
>=20
> > My card has one VGA, one DVI and one S Video
> connector on it.
> >
> >
> > I choose Polesoft Lockspam to fight spam, and =
you?
> > http://www.polesoft.com/refer.html =20
>=20
>=20
> --=20
> Chris Ludwig
> ADK
> chrisl@adkproaudio.com
> <mailto:chrisl@adkproaudio.com>
> <mailto:chrisl@adkproaudio.com>
> www.adkproaudio.com <http://www.adkproaudio.com>
> <http://www.adkproaudio.com/>
> (859) 635-5762
--=20
Chris Ludwig
ADK Pro Audio
(859) 635-5762
www.adkproaudio.com
chrisl@adkproaudio.com
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<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
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<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>Hi Chris,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>My dual KVM works great as is. Is =
the dual=20
monitor idea so bad? How else will I be able to sit =
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>in my sweet spot </FONT><FONT =
face=3DArial size=3D2>and=20
bounce between 2 and maybe even three comps with one=20
mouse/keyboard?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>They make quad KVMs. I might end =
up with that=20
in the end. 3 comps each with 4 head cards, </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>four monitors, one </FONT><FONT =
face=3DArial=20
size=3D2>keyboard/mouse.</FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Why is the KVM =
soooooo Baaaad? How else=20
can I achieve this?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>This other scenario would be even more =
complex than=20
described in the diagram below.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>The ultimate setup would have two =
QWERTY=20
keyboards/mice with KVM sharing with Paris/Cubase </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>and Cubase independently.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>The room as it relates to the =
photo:</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>My MIDI setup mimics what is shown =
to left of=20
the</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>mixing position here except there's a =
desk spot for=20
legs on the right along</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>with a controller. I'd like a 2 x =
22" LCD=20
widescreens, </FONT><FONT face=3DArial size=3D2>QWERTY & mouse =
</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>there along with the =
KVM QWERTY/mouse in the=20
center as shown for Paris</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>and mixdown work.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>What would you suggest?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Getting more confused by the =
minute...</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Tom</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<BLOCKQUOTE=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Chris Ludwig" <<A=20
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A>> =
wrote in=20
message <A =
href=3D"news:458c3c62$1@linux">news:458c3c62$1@linux</A>...</DIV>Tom=20
Bruhl GOOOODDDD.....KVM BBBBBAAAAADDDDDD<BR><BR><BR><BR><BR>Tom Bruhl=20
wrote:<BR>> Okay Aaron I hope I get this right.<BR>> =
<BR>> The=20
first diagram is what I think might not work due to the position of =
<BR>>=20
the splitter. I'm afraid it won't amplify the signal if<BR>> =
placed=20
at the end of the line. If I'm wrong then that's great! =
You'll=20
<BR>> want to widen your view for this one.<BR>> <BR>> =
_Example=20
1:_<BR>> <BR>> =20
=
<BR> > &a mp;nbsp;  =
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; =20
21" <BR>> VGA monitor & 22" LCD with VGA=20
=
input<BR> > &a mp;nbsp; =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp;=20
2 <BR>> x 5'=20
=
extensions<BR> > &a mp;nbsp; &=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; =20
Video <BR>> Splitter (Aarons active=20
=
type)<BR> > &a mp;nbsp; =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
; =20
20' <BR>>=20
=
extension<BR> > &a mp;nbsp; &n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
=20
KVM #1=20
=
out<BR> > &a mp;nbsp; &n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;=20
KVM #1 In=20
=
<BR> > &a mp;nbsp;  =
;   ;   ;=
=20
VGA #1 out<BR>> Paris Comp - G550=20
=
<BR> > &a mp;nbsp;  =
;   ;   ;=
=20
VGA #2 out =20
=
<BR> > &a mp;nbsp;  =
;   ;   ;=
&=
nbsp; & nbsp; =20
KVM #2 In=20
=
<BR> > &a mp;nbsp;  =
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; =20
KVM #2=20
=
out<BR> > &a mp;nbsp; &n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; =20
20' <BR>>=20
=
extension<BR> > &a mp;nbsp; &n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; =20
Video <BR>> Splitter (Aaron's active=20
=
type)<BR> > &a mp;nbsp; =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp;=20
2 <BR>> x 5'=20
=
extensions<BR> > &a mp;nbsp; &=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ; =20
<BR>> 21" VGA monitor & 22" LCD with VGA input<BR>> =20
<BR>> <BR>> =20
=
<BR> > &a mp;nbsp;  =
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; =20
KVM #1 out <BR>> (Shares with above from this point=20
=
on.)<BR> > &a mp;nbsp; &=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p;=20
KVM <BR>> #3=20
=
in &nbs p; &nbs p=
;   ;   ;=
=20
=
<BR> > &a mp;nbsp;  =
;   ;   ;=
=20
VGA #1 out<BR>> Cubase Comp -=20
=
7300GT<BR> > &a mp;nbsp;  =
; =20
DVI out to VGA connector=20
=
out<BR> > &a mp;nbsp; &n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; =20
KVM #4=20
=
in<BR> > &a mp;nbsp; &nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; =20
KVM #2 out <BR>> (Shares with above from this point =
on.)<BR>> =20
=
<BR> > &a mp;nbsp;  =
;   ;   ;=
&=
nbsp; =20
=
<BR> > &a mp;nbsp;  =
;   ;   ;=
=20
OR:<BR>> _Example 2:_<BR>> =20
=
<BR> > &a mp;nbsp;  =
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
21" <BR>> VGA =
monitor=20
& 22" LCD with VGA=20
=
input<BR> > &a mp;nbsp; =
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ; =20
<BR>> 2 x 5'=20
=
extensions<BR> > &a mp;nbsp; &=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p;=20
<BR>> Passive Video Splitter (Y=20
=
cable)<BR> > &a mp;nbsp;  =
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
=20
<BR>> 20'=20
=
extension<BR> > &a mp;nbsp; &n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; =20
=
<BR> > &a mp;nbsp;  =
; =20
KVM #1=20
=
out<BR> > &a mp;nbsp; &n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
=20
KVM <BR>> #1 In=20
=
<BR> > &a mp;nbsp;  =
;   ;   ;=
&=
nbsp; & nbsp; =20
Video amplifier (Like <BR>> splitter in example 1 with only one=20
=
output)<BR> > &a mp;nbsp; &nbs=
p; &nbs p; &nbs p=
;   ; =20
VGA #1 out<BR>> Paris Comp - G550=20
=
<BR> > &a mp;nbsp;  =
;   ;   ;=
=20
VGA #2=20
=
out<BR> > &a mp;nbsp; &n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;=20
Video amplifier (Like <BR>> splitter in example 1 with only one=20
=
output)<BR> > &a mp;nbsp; &nbs=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; =20
<BR>> KVM #2 In=20
=
<BR> > &a mp;nbsp;  =
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; =20
<BR>> KVM #2=20
=
out<BR> > &a mp;nbsp; &n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; =20
<BR>> 20'=20
=
extension<BR> > &a mp;nbsp; &n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; =20
=
<BR> > &a mp;nbsp;  =
;   ;   ;=
=20
Passive Video Splitter (Y=20
=
cable)<BR> > &a mp;nbsp;  =
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; =20
2 <BR>> x 5'=20
=
extensions<BR> > &a mp;nbsp; &=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp;=20
=
<BR> > &a mp;nbsp;  =
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; =20
21" VGA <BR>> monitor & 22" LCD with VGA input<BR>> =20
<BR>> <BR>> =20
=
<BR> > &a mp;nbsp;  =
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; =20
<BR>> KVM #1 out (Shares with above from this point=20
=
on.)<BR> > &a mp;nbsp; &=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ; =20
KVM <BR>> #3=20
=
in &nbs p; &nbs p=
;   ;   ;=
=20
=
<BR> > &a mp;nbsp;  =
;   ;   ;=
&=
nbsp; & nbsp; =20
Video amplifier (Like <BR>> splitter in example 1 with only one=20
=
output)<BR> > &a mp;nbsp; &nbs=
p; &nbs p; &nbs p=
;   ; =20
VGA #1 out<BR>> Cubase Comp -=20
=
7300GT<BR> > &a mp;nbsp;  =
; =20
DVI out to VGA connector=20
=
out<BR> > &a mp;nbsp; &n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; =20
Video amplifier (Like <BR>> splitter in example 1 with only one=20
=
output)<BR> > &a mp;nbsp; &nbs=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; =20
KVM #4=20
=
in<BR> > &a mp;nbsp; &nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; =20
<BR>> KVM #2 out (Shares with above from this point =
on.)<BR>> =20
<BR>> <BR>> <BR>> <BR>> =
<BR>> =20
<BR>> <BR>> <BR>> <BR>> =
<BR>> =20
<BR>> <BR>> <BR>> <BR>> =
<BR>> =20
<BR>> <BR>> <BR>> <BR>> =
<BR>> =20
<BR>> <BR>> <BR>> <BR>> =
<BR>> =20
<BR>> <BR>> <BR>> <BR>> =
<BR>> =20
<BR>> <BR>> <BR>> <BR>> =
<BR>> =20
<BR>> <BR>> <BR>> =
<BR>> "Aaron=20
Allen" <<A=20
=
href=3D"mailto:know-spam@not_here.dude">know-spam@not_here.dude</A><BR>&g=
t; =20
<<A=20
=
href=3D"mailto:know-spam@not_here.dude">mailto:know-spam@not_here.dude</A=
>>>=20
wrote in message<BR>> <A=20
=
href=3D"news:458c050f@linux">news:458c050f@linux</A>...<BR>> &nbs=
p; =20
I'm not getting the visual you have in mind.. diagram=20
it?<BR>> =
<BR>> =20
AA<BR>> <BR> > &a mp;nbsp; =
"Tom=20
Bruhl" <<A =
href=3D"mailto:arpegio@comcast.net">arpegio@comcast.net</A> <<A=20
=
href=3D"mailto:arpegio@comcast.net">mailto:arpegio@comcast.net</A>>>=
;<BR> > &a mp;nbsp; =20
wrote in message <A=20
=
href=3D"news:458b7f6f@linux">news:458b7f6f@linux</A>...<BR>> &nbs=
p; &nbs p;=20
Thanks Aaron,<BR> > &a mp;nbsp; =
Will=20
this work on the backside of a KVM? I can't put it in=20
front.<BR> > &a mp;nbsp; I guess =
if it=20
needed to be in front of the KVM I could just get=20
a<BR> > &a mp;nbsp; single =
output jobbie=20
and split it at the other side. Do=20
those<BR> > &a mp;nbsp; =20
exist?<BR> > &a mp;nbsp; =
<BR>>=20
=
<BR> > &a mp;nbsp;  =
; =20
"Aaron Allen" <<A=20
=
href=3D"mailto:know-spam@not_here.dude">know-spam@not_here.dude</A><BR>&g=
t; &nbs p; &nbs p=
;=20
<<A=20
=
href=3D"mailto:know-spam@not_here.dude">mailto:know-spam@not_here.dude</A=
>>>=20
wrote in=20
=
message<BR> > &a mp;nbsp; &nbs=
p; =20
<A=20
=
href=3D"news:458b23a2@linux">news:458b23a2@linux</A>...<BR>> &nbs=
p; &nbs p; =20
I hope you're not doing a passive Y split. Look for a=20
=
VGA<BR> > &a mp;nbsp; &n=
bsp; =20
splitter on google, something=20
=
like<BR> > &a mp;nbsp; &=
nbsp; =20
=
<BR> > &a mp;nbsp;  =
; =20
<A=20
=
href=3D"http://www.pcmall.com/pcmall/shop/detail~dpno~767745.asp">http://=
www.pcmall.com/pcmall/shop/detail~dpno~767745.asp</A><BR>> =
=20
=
<BR> > &a mp;nbsp;  =
; =20
They're cheap and OH so worth=20
=
it<BR> > &a mp;nbsp; &nb=
sp; =20
=
<BR> > &a mp;nbsp;  =
; =20
AA<BR>>=20
=
<BR> > &a mp;nbsp;  =
; =20
"Tom Bruhl" <<A=20
=
href=3D"mailto:arpegio@comcast.net">arpegio@comcast.net</A><BR>> =
&=
nbsp; =20
<<A=20
=
href=3D"mailto:arpegio@comcast.net">mailto:arpegio@comcast.net</A>>>=
; wrote=20
in=20
=
message<BR> > &a mp;nbsp; &nbs=
p; &nbs p;=20
<A=20
=
href=3D"news:4589ba0a$1@linux">news:4589ba0a$1@linux</A>...<BR>> =
&=
nbsp; =20
Thanks Chris.=20
=
<BR> > &a mp;nbsp;  =
; =20
This graphics stuff is confusing to=20
=
me.<BR> > &a mp;nbsp; &n=
bsp; &n bsp;=20
=
<BR> > &a mp;nbsp;  =
; =20
I have to convert the DVI to VGA because of the 20'=20
=
length<BR> > &a mp;nbsp;  =
;   ;=20
as I understand. I was going to split it with a Y=20
=
cable<BR> > &a mp;nbsp; =
=20
=
too.<BR> > &a mp;nbsp; &=
nbsp; =20
They'll split to a 21"VGA and a 22"LCD. Will that work=20
=
on<BR> > &a mp;nbsp; &nb=
sp; =20
the far side of a KVM? I know the resolutions won't=20
=
match<BR> > &a mp;nbsp; =
=20
but I'll only need the VGAs when mixing so I'll adjust=20
=
the<BR> > &a mp;nbsp; &n=
bsp; =20
7300 differently for that application. Sooner or=20
=
later<BR> > &a mp;nbsp; =
=20
I'll=20
=
get<BR> > &a mp;nbsp; &n=
bsp; =20
4 matching 22" LCDs and quad cards for both=20
=
comps.<BR> > &a mp;nbsp;  =
;   ; =20
=
<BR> > &a mp;nbsp;  =
; =20
Will this=20
=
work?<BR> > &a mp;nbsp; =
=20
=
Tom<BR> > &a mp;nbsp; &n=
bsp; &n bsp;=20
=
<BR> > &a mp;nbsp;  =
;   ;=20
=
<BR> > &a mp;nbsp;  =
;   ;=20
=
<BR> > &a mp;nbsp;  =
;   ;=20
=
<BR> > &a mp;nbsp;  =
;   ;=20
=
<BR> > &a mp;nbsp;  =
;   ;=20
<BR>>=20
=
<BR> > &a mp;nbsp;  =
;   ; =20
"Chris Ludwig" <<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A><BR>>=
&=
nbsp; & nbsp; =20
<<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">mailto:chrisl@adkproaudio.com</A>&=
gt;>=20
wrote in=20
=
message<BR> > &a mp;nbsp; &nbs=
p; &nbs p; =20
<A=20
=
href=3D"news:45894ef0$1@linux">news:45894ef0$1@linux</A>...<BR>> =
&=
nbsp; & nbsp;=20
HI=20
=
Tom,<BR> > &a mp;nbsp; &=
nbsp; & nbsp; =20
These monitors will be fine with your card.=20
=
The<BR> > &a mp;nbsp; &n=
bsp; &n bsp; =20
7300GT supports=20
=
that<BR> > &a mp;nbsp; &=
nbsp; & nbsp; =20
resolution on both the VGA and DVI=20
=
connections.I<BR> > &a mp;nbsp; &nbs=
p; &nbs p; =20
would run both with=20
=
VGA<BR> > &a mp;nbsp; &n=
bsp; &n bsp; =20
connections. Having one hooked up DVI and one=20
=
VGA<BR> > &a mp;nbsp; &n=
bsp; &n bsp; =20
may make the 2=20
=
look<BR> > &a mp;nbsp; &=
nbsp; & nbsp; =20
very different in brightness and=20
=
such.<BR> > &a mp;nbsp; =
=20
Chris<BR>> <BR>>=20
=
<BR> > &a mp;nbsp;  =
;   ; =20
Tom Bruhl wrote:<BR>>=20
=
<BR> > &a mp;nbsp;  =
;   ; =20
> My card has one VGA, one DVI and one S=20
=
Video<BR> > &a mp;nbsp; =
=20
connector on=20
=
it.<BR> > &a mp;nbsp; &n=
bsp; &n bsp; =20
=
><BR> > &a mp;nbsp; &=
nbsp; & nbsp; =20
=
><BR> > &a mp;nbsp; &=
nbsp; & nbsp; =20
> I choose Polesoft Lockspam to fight spam, and=20
=
you?<BR> > &a mp;nbsp; &=
nbsp; & nbsp; =20
> <A=20
=
href=3D"http://www.polesoft.com/refer.html">http://www.polesoft.com/refer=
..html</A> =20
<BR>> <BR>>=20
=
<BR> > &a mp;nbsp;  =
;   ; =20
--=20
=
<BR> > &a mp;nbsp;  =
;   ; =20
Chris=20
=
Ludwig<BR> > &a mp;nbsp;  =
;   ; =20
=
ADK<BR> > &a mp;nbsp; &n=
bsp; &n bsp; =20
<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A><BR>>=
&=
nbsp; & nbsp; =20
<<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">mailto:chrisl@adkproaudio.com</A>&=
gt;<BR> > &a mp;nbsp; &n=
bsp; &n bsp; =20
<<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">mailto:chrisl@adkproaudio.com</A>&=
gt;<BR> > &a mp;nbsp; &n=
bsp; &n bsp; =20
<A href=3D"http://www.adkproaudio.com">www.adkproaudio.com</A> <<A=20
=
href=3D"http://www.adkproaudio.com">http://www.adkproaudio.com</A>><BR=
> > &a mp;nbsp; &a mp;n=
bsp; &n bsp; =20
<<A=20
=
href=3D"http://www.adkproaudio.com/">http://www.adkproaudio.com/</A>><=
BR> > &a mp;nbsp; =
=20
(859) 635-5762<BR><BR>-- <BR>Chris Ludwig<BR><BR>ADK Pro =
Audio<BR>(859)=20
635-5762<BR><A =
href=3D"http://www.adkproaudio.com">www.adkproaudio.com</A><BR><A=20
=
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A></BLOCKQ=
UOTE></BODY></HTML>
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------=_NextPart_000_0009_01C725E2.462A6A90--Lamont - what is the output chain you are using for each app when comparing
the file in Nuendo
vs ProTools? On the same PC, I presume (and is this PT HD or M-Powered?)?
Since these can't use the same output driver, you would have to depend on
the D/A being
the same, but clocking will be different unless you have a master clock, and
both interfaces
are locking with the same accuracy. This was one of the issues that came up
at Lynn Fuston's
D/A converter shootout - when do you lock to external clock and incur the
resulting jitter,
and when do you trust the internal clock - and if you do lock externally,
how good is the PLL
in the slave device? These issues can cause audible changes in the top end
that have nothing to do
with the software itself. If you say that PTHD through the same converter
output as Nuendo (via? RME?
Lynx?) using the same master clock, sounds different playing a single audio
file, then I take your word
for it. I can't tell you why that is happening - only that an audible
difference really shouldn't happen due
to the software alone - not with a single audio file, esp. since I've heard
and seen PTHD audio cancel with
native DAWs. Just passing a single 16 or 24 bit track down the buss to the
output driver should
be, and usually is, completely transparent, bit for bit.
The same audio file played through the same converters should only sound
different if something in
the chain is different - be it clocking, gain or some degree of unintended,
errant dsp processing. Every DAW should
pass a single audio file without altering a single bit. That's a basic level
of accuracy we should always
expect of any DAW. If that accuracy isn't there, you can be sure a heavy
mix will be altered in ways you
didn't intend, even though you would end up mixing with that factor in place
(e.g. you still mix for what
you want to hear regardless of what the platform does to each audio track or
channel).
In fact you should be able to send a stereo audio track out SPDIF or
lightpipe to another DAW, record it
bring the recorded file back in, line them up to the first bit, and have
them cancel on and inverted phase
test. I did this with Nuendo and Cubase 4 on separate machines just to be
sure my master clocking and
slave sync was accurate - it worked perfectly.
Also be sure there isn't a variation in the gain even by 0.1 dB between the
two. There shouldn't
and I wouldn't expect there to be one. Also could PT be set for a different
pan law? Shouldn't make a
difference even if comparing two mono panned files to their stereo
interleaved equivalent, but for sake
of completeness it's worth checking as well. A variation in the output
chain, be it drivers, audio card
card, or converters would be the most likely culprit here.
The reason DAW manufacturers wouldn't add any sonic "character"
intentionally is that the
ultimate goal from day one with recording has been to accurately reproduce
what we hear.
We developed a musical penchant for sonic character because the hardware
just wasn't accurate,
and what it did often sent us down new creative paths - even if by force -
and we decided it was
preferred that way.
Your point about what goes into the feature presets to sell synths is right
for sure, but synths are about
character and getting that "perfect piano" or crystal clear bell pad, or fat
punchy bass without spending
a mint on development, adding 50G onboard sample libraries, or costing $15k,
so what they
lack in actual synthesis capabilities, they make up with EQ and effects on
the output. That's been the case
for years, at least since we had effects on synths at least. But even with
modern synths such as the Fantom,
Tritons, etc, which are great synths all around, of course the coolest,
widest and biggest patches
will make the biggest impression - so in come the EQs, limiters, comps,
reverbs, chorus, etc. The best
way to find out if a synth is really good is to bypass all effects and see
what happens. Most are pretty
good these days, but about half the time, there are presets that fall
completely flat in fx bypass.
DAWs aren't designed to put a sonic fingerprint on a sound the way synths
are - they are designed
to *not* add anything - to pass through what we create as users, with no
alteration (or as little as possible)
beyond what we add with intentional processing (EQ, comps, etc). Developers
would find no pride
in hearing that their DAW sounds anything different than whatever is being
played back in it,
and the concept is contrary to what AES and IEEE proceedings on the issue
propose in general
digital audio discussions, white papers, etc.
What ID ended up doing with Paris (at least from what I gather per Chuck's
findings - so correct me if I'm missing part of the equation Chuck),
is drop the track gain by 20dB or so, then added it back at the master buss
to create the effect of headroom (probably
because the master buss is really summing on the card, and they have more
headroom there than on the tracks
where native plugins might be used). I don't know if Paris passed 32-bit
float files to the EDS card, but sort of
doubt it. I think Chuck has clarified this at one point, but don't recall
the answer.
Also what Paris did is use a greater bit depth on the hardware than ProTools
did - at the time PT was just
bring Mix+ systems to market, or they had been out for a year or two (if I
have my timeline right) - they
were 24-bit fixed all the way through. Logic and Cubase were native DAWs,
but native was still too slow
to compete with hardware hybrids. Paris trumped them all by running 32-bit
float natively (not new really, but
better than sticking to 24-bit) and 56 or so bits in hardware instead of
going to Motorola DSPs at 24.
The onboard effects were also a step up from anything out there, so the demo
did sound good.
I don't recall which, but one of the demos, imho, wasn't so good (some
sloppy production and
vocals in spots, IIRC), so I only listened to it once. ;-)
Coupled with the gain drop and buss makeup, this all gave it a "headroom" no
one else had. With very nice
onboard effects, Paris jumped ahead of anything else out there easily, and
still respectably holds its' own today
in that department.
Most demos I hear (when I listen to them) vary in quality, usually not so
great in some area. But if a demo does
sound great, then it at least says that the product is capable of at least
that level of performance, and it can
only help improve a prospective buyer's impression of it.
Regards,
Dedric
"LaMont " <jjdpro@ameritech.net> wrote in message news:458c14c0$1@linux...
>
> Dedric good post..
>
> However, I have PT-M-Powered/M-audio 410 interface for my laptop and it
> has
> that same sound (no eq, zero fader) that HD does. I know their use the
> same
> 48 bit fix mixer. I load up the same file in Nuendo (no eq, zero
> fader)..results.
> different sonic character.
>
> PT having a top end touch..Nuendo, nice smooth(flat) sound. And I'm just
> taking about a stereo wav file nulled with no eq..nothing ..zilch..nada..
>
> Now, there are devices (keyboards, dum machines) on the market today that
> have a Master Buss Compressor and EQ set to on with the top end notched
> up.
> Why? because it gives their product an competitive advantageover the
> competition..
> Ex: Yahama's Motif ES, Akai's MPC 1000, 2500, Roland's Fantom.
>
> So, why would'nt a DAW manufactuer code in an extra (ooommf) to make their
> DAW sound better. Especially, given the "I hate Digtal Summing" crowd?
> And,
> If I'm a DAW manufactuer, what would give my product a sonic edge over the
> competition?
>
> We live in the "louder is better" audio world these days, so a DAW that
> can
> catch my attention 'sonically" will probaly will get the sell. That's what
> happend to me back in 1997 when I heard Paris. I was floored!!! Still to
> this day, nothing has floored me like that "Road House Blues Demo" I heard
> on Paris.
>
> Was it the hardware ? was it the software. I remember talking with Edmund
> at the 2000 winter Namm, and told me that he & Steve set out to reproduce
> the sonics of big buck analog board (eq's) and all.. And, summing was a
> big
> big issue for them because they (ID) thought that nobody has gotten
> it(summing)
> right. And by right, they meant, behaved like a console with a wide lane
> for all of those tracks..
>
>
>
>
> "Dedric Terry" <dedric@echomg.com> wrote:
>>"LaMont" <jjdpro@ameritech.net> wrote in message news:458be8d5$1@linux...
>>>
>>> Okay...
>>> I guess what I'm saying is this:
>>>
>>> -Is it possible that diferent DAW manufactuers "code" their app
>>> differently
>>> for sound results.
>>
>>Of course it is *possible* to do this, but only if the DAW has a specific
>
>>sound shaping purpose
>>beyond normal summing/mixing. Users talk about wanting developers to add
> a
>>"Neve sound" or "API sound" option to summing engines,
>>but that's really impractical given the amount of dsp required to make a
>
>>decent emulation (with convolution, dynamic EQ functions,
>>etc). For sake of not eating up all cpu processing, that could likely
>>only
>
>>surface as is a built in EQ, which
>>no one wants universally in summing, and anyone can add at will already.
>>
>>So it hasn't happened yet and isn't likely to as it detours from the basic
>
>>tenant of audio recording - recreate what comes in as
>>accurately as possible.
>>
>>What Digi did in recoding their summing engine was try to recover some
>>of the damage done by the 24-bit buss in Mix systems. Motorola 56k dsps
> are
>>24-bit fixed point chips and I think
>>the new generation (321?) still is, but they use double words now for
>>48-bits). And though plugins could process at 48-bit by
>>doubling up and using upper and lower 24-bit words for 48-bit outputs, the
>
>>buss
>>between chips was 24-bits, so they had to dither to 24-bits after every
>
>>plugin. The mixer (if I recall correctly) also
>>had a 24-bit buss, so what Digi did is to add a dither stage to the mixer
> to
>>prevent this
>>constant tru
|
|
|
|
| Re: Paris Skins / examples 4 [message #76038 is a reply to message #76037] |
Wed, 15 November 2006 22:19 |
Yanoska
Messages: 32
Registered: January 2007
|
Member |
|
|
ncation of data. 24-bits isn't enough to cover summing for
> more
>>than a few tracks without
>>losing information in the 16-bit world, and in the 24-bit world some
>>information will be lost, at least at the lowest levels.
>>
>>Adding a dither stage (though I think they did more than that - perhaps
>
>>implement a 48-bit double word stage as well),
>>simply smoothed over the truncation that was happening, but it didn't
>>solve
>
>>the problem, so with HD
>>they went to a double-word path - throughout I believe, including the path
>
>>between chips. I believe the chips
>>are still 24-bit, but by doubling up the processing (yes at a cost of
>>twice
>
>>the overhead), they get a 48-bit engine.
>>This not only provided better headroom, but greater resolution. Higher
> bit
>>depths subdivide the amplitude with greater resolution, and that's
>>really where we get the definition of dynamic range - by lowering the
>>signal
>
>>to quantization noise ratio.
>>
>>With DAWs that use 32-bit floating point math all the way through, the
>>only
>
>>reason for altering the summing
>>is by error, and that's an error that would actually be hard to make and
> get
>>past a very basic alpha stage of testing.
>>There is a small difference in fixed point math and floating point math,
> or
>>at least a theoretical difference in how it affects audio
>>in certain cases, but not necessarily in the result for calculating gain
> in
>>either for the same audio file. Where any differences might show up is
>
>>complicated, and I believe only appear at levels below 24-bit (or in
>>headroom with tracks pushed beyond 0dBFS), or when/if
>>there areany differences in where each amplitude level is quantized.
>>
>>Obviously there can be differences if the DAW has to use varying bit
>>depths
>
>>throughout a single summing path to accomodate hardware
>>as well as software summing, since there may be truncation or rounding
>>along
>
>>the way, but that impacts the lowest bit
>>level, and hence - spacial reproduction, reverb tails perhaps, and
>>"depth",
>
>>not the levels most music so the differences are most
>>often more subtle than not. But most modern DAWs have eliminated those
>
>>"rough edges" in the math by increasing the bit depth to accomodate normal
>
>>summing required for mixing audio.
>>
>>So with Lynn's unity gain summing test (A files on the CD I believe), DAWs
>
>>were never asked to sum beyond 24-bits,
>>at least not on the upper end of the dynamic range, so everything that
>>could
>
>>represent 24-bits accurately would cancel. The only ones
>>that didn't were ones that had a different bit depth and/or gain structure
>
>>whether hybrid or native
>>(e.g. Paris' subtracting 20dB from tracks and adding it to the buss). In
>
>>this case, PTHD cancelled (when I tested it) with
>>Nuendo, Samplitude, Logic, etc because the impact of the 48-bit fixed vs.
>
>>32-bit float wasn't a factor.
>>
>>When trying other tests, even when adding and subtracting gain, Nuendo,
>
>>Sequoia and Sonar cancel - both audibly and
>>visually at inaudible levels, which only proves that one isn't making an
>
>>error when calculating basic gain. Since a dB is well defined,
>>and the math to add gain is simple, they shouldn't. The fact that they
> all
>>use 32-bit float all the way through eliminates a difference
>>in data structure as well, and this just verifies that. There was a time
>
>>that supposedly Logic (v3, v4?) was partly 24-bit, or so the rumor went,
>>but it's 32-bit float all the way through now just as Sonar,
>>Nuendo/Cubase,
>
>>Samplitude/Sequoia, DP, Audition (I presume at least).
>>I don't know what Acid or Live use. Saw promotes a fixed point engine,
> but
>>I don't know if it is still 24-bit, or now 48 bit.
>>That was an intentional choice by the developer, but he's the only one I
>
>>know of that stuck with 24-bit for summing
>>intentionally, esp. after the Digi Mix system mixer incident.
>>
>>Long answer, but to sum up, it is certainly physically *possible* for a
>
>>developer to code something differently intentionally, but not
>>in reality likely since it would be breaking some basic fixed point or
>>floating point math rules. Where the differences really
>>showed up in the past is with PT Mix systems where the limitation was
>>really
>
>>significant - e.g. 24 bit with truncation at several stages.
>>
>>That really isn't such an issue anymore. Given the differences in
>>workflow,
>
>>missing something in workflow or layout differences
>>is easy enough to do (e.g. Sonar doesn't have group and busses the way
>>Nuendo does, as it's outputs are actually driver outputs,
>>not software busses, so in Sonar, busses are actually outputs, and sub
>>busses are actually busses in Nuendo. There are no,
>>or at least I haven't found the equivalent of a Nuendo group in Sonar -
> that
>>affects the results of some tests (though not basic
>>summing) if not taken into account, but when taken into account, they work
>
>>exactly the same way).
>>
>>So at least when talking about apps with 32-bit float all the way through,
>
>>it's safe to say (since it has been proven) that summing isn't different
>
>>unless
>>there is an error somewhere, or variation in how the user duplicates the
>
>>same mix in two different apps.
>>
>>Imho, that's actually a very good thing - approaching a more consistent
>
>>basis for recording and mixing from which users can make all
>>of the decisions as to how the final product will sound and not be
>>required
>
>>to decide when purchasing a pricey console, and have to
>>focus their business on clients who want "that sound". I believe we are
>
>>actually closer to the pure definition of recording now than
>>we once were.
>>
>>Regards,
>>Dedric
>>
>>
>>>
>>> I the answer is yes, then,the real task is to discover or rather
>>> un-cover
>>> what's say: Motu's vision of summing, versus Digidesign, versus
>>> Steinberg
>>> and so on..
>>>
>>> What's under the hood. To me and others,when Digi re-coded their summing
>>> engine, it was obvious that Pro Tools has an obvious top end (8k-10k)
>
>>> bump.
>>> Where as Steinberg's summing is very neutral.
>>>
>>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>>Hi Neil,
>>>>
>>>>Jamie is right. And you aren't wacked out - you are thinking this
>>>>through
>>>
>>>>in a reasonable manner, but coming to the wrong
>>>>conclusion - easy to do given how confusing digital audio can be. Each
>>> word
>>>>represents an amplitude
>>>>point on a single curve that is changing over time, and can vary with
> a
>>>
>>>>speed up to the Nyquist frequency (as Jamie described).
>>>>The complex harmonic content we hear is actually the frequency
>>>>modulation
>>> of
>>>>a single waveform,
>>>>that over a small amount of time creates the sound we translate - we
>>>>don't
>>>
>>>>really hear a single sample at a time,
>>>>but thousands of samples at a time (1 sample alone could at most
>>>>represent
>>> a
>>>>single positive or negative peak
>>>>of a 22,050Hz waveform).
>>>>
>>>>If one bit doesn't cancel, esp. if it's a higher order bit than number
> 24,
>>>
>>>>you may hear, and will see that easily,
>>>>and the higher the bit in the dynamic range (higher order) the more
>>>>audible
>>>
>>>>the difference.
>>>>Since each bit is 6dB of dynamic range, you can extrapolate how "loud"
>
>>>>that
>>>
>>>>bit's impact will be
>>>>if there is a variation.
>>>>
>>>>Now, obviously if we are talking about 1 sample in a 44.1k rate song,
> then
>>>
>>>>it simply be a
>>>>click (only audible if it's a high enough order bit) instead of an
>>>>obvious
>>>
>>>>musical difference, but that should never
>>>>happen in a phase cancellation test between identical files higher than
>>> bit
>>>>24, unless there are clock sync problems,
>>>>driver issues, or the DAW is an early alpha version. :-)
>>>>
>>>>By definition of what DAWs do during playback and record, every audio
>
>>>>stream
>>>
>>>>has the same point in time (judged by the timeline)
>>>>played back sample accurately, one word at a time, at whatever sample
>
>>>>rate
>>>
>>>>we are using. A phase cancellation test uses that
>>>>fact to compare two audio files word for word (and hence bit for bit
>>>>since
>>>
>>>>each bit of a 24-bit word would
>>>>be at the same bit slot in each 24-bit word). Assuming they are aligned
>>> to
>>>>the same start point, sample
>>>>accurately, and both are the same set of sample words at each sample
>>>>point,
>>>
>>>>bit for bit, and one is phase inverted,
>>>>they will cancel through all 24 bits. For two files to cancel
>>>>completely
>>>
>>>>for the duration of the file, each and every bit in each word
>>>>must be the exact opposite of that same bit position in a word at the
> same
>>>
>>>>sample point. This is why zooming in on an FFT
>>>>of the full difference file is valuable as it can show any differences
> in
>>>
>>>>the lower order bits that wouldn't be audible. So even if
>>>>there is no audible difference, the visual followup will show if the two
>>>
>>>>files truly cancel even a levels below hearing, or
>>>>outside of a frequency change that we will perceive.
>>>>
>>>>When they don't cancel, usually there will be way more than 1 bit
>>>>difference - it's usually one or more bits in the words for
>>>>thousands of samples. From a musical standpoint this is usually in a
>>>>frequency range (low freq, or high freq most often) - that will
>>>>show up as the difference between them, and that usually happens due to
>>> some
>>>>form of processing difference between the files,
>>>>such as EQ, compression, frequency dependant gain changes, etc. That is
>>> what
>>>>I believe you are thinking through, but when
>>>>talking about straight summing with no gain change (or known equal gain
>>>
>>>>changes), we are only looking at linear, one for one
>>>>comparisons between the two files' frequency representations.
>>>>
>>>>Regards,
>>>>Dedric
>>>>
>>>>> Neil wrote:
>>>>>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>>>>> The tests I did were completely blank down to -200 dB (far below the
>>>
>>>>>>> last
>>>>>>
>>>>>>> bit). It's safe to say there is no difference, even in
>>>>>>> quantization noise, which by technical rights, is considered below
> the
>>>
>>>>>>> level
>>>>>>
>>>>>>> of "cancellation" in such tests.
>>>>>>
>>>>>> I'm not necessarily talking about just the first bit or the
>>>>>> last bit, but also everything in between... what happens on bit
>>>>>> #12, for example? Everything on bit #12 should be audible, but
>>>>>> in an a/b test what if thre are differences in what bits #8
>>>>>> through #12 sound like, but the amplutide is stll the same on
>>>>>> both files at that point, you'll get a null, right? Extrapolate
>>>>>> that out somewhat & let's say there are differences in bits #8
>>>>>> through #12 on sample points 3, 17, 1,000, 4,523, 7,560, etc,
>>>>>> etc through 43,972... Now this is breaking things down well
>>>>>> beyond what I think can be measured, if I'm not mistaken (I
>>>>>> dn't know of any way we could extract JUST that information
>>>>>> from each file & play it back for an a/b test; but would not
>>>>>> that be enough to have to "null-able" files that do actually
>>>>>> sound somewhat different?
>>>>>>
>>>>>> I guess what I'm saying is that since each sample in a musical
>>>>>> track or full song file doesn't represent a pure, simple set of
>>>>>> content like a sample of a sine wave would - there's a whole
>>>>>> world of harmonic structure in each sample of a song file, and
>>>>>> I think (although I'll admit - I can't "prove") that there is
>>>>>> plenty of room for some variables between the first bit & the
>>>>>> last bit while still allowing for a null test to be successful.
>>>>>>
>>>>>> No? Am I wacked out of my mind?
>>>>>>
>>>>>> Neil
>>>>>>
>>>>
>>>
>>
>>
>Awesome Chris. Congratulatons on the nomination. Wish I was a big enough
cat to vote.
MR
"Chris Latham" <latham_c@bellsouth.net> wrote in message
news:458b70fc@linux...
> For those of you who will be voting for the 49th annual Grammys, please
give
> a listen to, and consider Guy Clark's "Workbench Songs" for Best
> Contemporary Folk/Americana Album (Catagory 68 in Field 14 - Folk).
Edited
> and mixed entirely in PARIS.
>
> THANK YOU,
> Chris
>
>This is a multi-part message in MIME format.
------=_NextPart_000_0014_01C725FA.9A559460
Content-Type: text/plain;
charset="iso-8859-1"
Content-Transfer-Encoding: 7bit
I've got a Mac G4 (OSX 10.4) and a PC running Windows 98SE and Paris. I
want to be able to move .wav files made on the G4 over to Paris, and from
Paris back over to the Mac --if I want to apply FX there that I cannot in
Paris. As I understand it, the G4 has an ethernet card built in. I assume
that I have to get a NIC card for the PC. Do I need to get some sort of
software for each machine so that the PC will see the Mac and vice versa?
This is totally new territory. I wish it was as simple as plugging in a
cable between two boxes, but I fear it is not...
Thanks much,
MR
------=_NextPart_000_0014_01C725FA.9A559460
Content-Type: text/html;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1555" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>I've got a Mac G4 (OSX 10.4) and a PC =
running=20
Windows 98SE and Paris. I want to be able to move .wav files made =
on the=20
G4 over to Paris, and from Paris back over to the Mac --if I want to =
apply FX=20
there that I cannot in Paris. As I understand it, the G4 has an =
ethernet=20
card built in. I assume that I have to get a NIC card for the PC. =
Do I=20
need to get some sort of software for each machine so that the PC will =
see the=20
Mac and vice versa? This is totally new territory. I wish it =
was as=20
simple as plugging in a cable between two boxes, but I fear it is=20
not...</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Thanks much,</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>MR</FONT></DIV></BODY></HTML>
------=_NextPart_000_0014_01C725FA.9A559460--This is a multi-part message in MIME format.
------=_NextPart_000_0023_01C7268C.A5A724E0
Content-Type: text/plain;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
I think it is....
Most PC's these days have built in Ethernet cards, and windows has a =
network setup wizard that will run you through the process.
easy....really.
--=20
Martin Harrington
www.lendanear-sound.com
"Mike R." <emarenot@yahoo.com> wrote in message =
news:458c6d36$1@linux...
I've got a Mac G4 (OSX 10.4) and a PC running Windows 98SE and Paris. =
I want to be able to move .wav files made on the G4 over to Paris, and =
from Paris back over to the Mac --if I want to apply FX there that I =
cannot in Paris. As I understand it, the G4 has an ethernet card built =
in. I assume that I have to get a NIC card for the PC. Do I need to get =
some sort of software for each machine so that the PC will see the Mac =
and vice versa? This is totally new territory. I wish it was as simple =
as plugging in a cable between two boxes, but I fear it is not...
Thanks much,
MR
------=_NextPart_000_0023_01C7268C.A5A724E0
Content-Type: text/html;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.5730.11" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT size=3D2>I think it is....</FONT></DIV>
<DIV><FONT size=3D2>Most PC's these days have built in Ethernet cards, =
and windows=20
has a network setup wizard that will run you through the =
process.</FONT></DIV>
<DIV><FONT size=3D2>easy....really.</FONT><BR>-- <BR>Martin =
Harrington<BR><A=20
href=3D"http://www.lendanear-sound.com">www.lendanear-sound.com</A><BR></=
DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Mike R." <<A=20
href=3D"mailto:emarenot@yahoo.com">emarenot@yahoo.com</A>> wrote in =
message=20
<A href=3D"news:458c6d36$1@linux">news:458c6d36$1@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>I've got a Mac G4 (OSX 10.4) and a PC =
running=20
Windows 98SE and Paris. I want to be able to move .wav files =
made on the=20
G4 over to Paris, and from Paris back over to the Mac --if I want to =
apply FX=20
there that I cannot in Paris. As I understand it, the G4 has an =
ethernet=20
card built in. I assume that I have to get a NIC card for the =
PC. Do I=20
need to get some sort of software for each machine so that the PC will =
see the=20
Mac and vice versa? This is totally new territory. I wish =
it was=20
as simple as plugging in a cable between two boxes, but I fear it is=20
not...</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Thanks much,</FONT></DIV>
<DIV><FONT face=3DArial =
size=3D2>MR</FONT></DIV></BLOCKQUOTE></BODY></HTML>
------=_NextPart_000_0023_01C7268C.A5A724E0--Dedric, my simple test is simple..
Using the same audio interface, with the same stereo file..null-ed to zero..No
eq, for fx. Master fader on zero..
Nuendo, Pro-Tools -Mpowered(native)... yields a sonic difference that I have
referenced before.. The sound coming from PT-M has a nice top end , where
as Neundo has a nice flatter sound quality.
Same audio interface. M-audio 410..Using Mackies & Blue-Sky pro monitors..
Same test at the big room..PT-HD & Neundo Logic Audio(macG5-Dual) Using the
192 interface.
Same results..But adding Logic audio's sound ..(Broad, thick)
Somethings going on.
Chucks post about how paris handles audio is a theory..Only Edmund can truly
give us the goods on what's really what..
I disagree that manufactuers don;t set out o put a sonic print on their products.
I think they do.
I have been fortunate to work on some digital mixers and I can tell you that
each one has their own sound. The Sony Dmx-100 was modeled after SSL 4000g
(like it's Big Brother).And you what? That board (Dmx-100) sound very warm
and it's eq tries to behave and sound just like an SSL.. Unlike he Yamaha
Dm2000(version 1.x) which has a very Clean, neutral sound..However, some
complained that it was tooo Vanila and thus, Yamaha add a version 2.0 which
added Vintage type Eq's, modeled analog input gain saturation fx too give
the user a choice Btw Clean and Neutral vs sonic Character.
So, if digital conoles can be given a sonic character, why not a software
mixer?
The truth is, there are some folks who want a neutral mixer and then there
are others who want a sonic footprint imparted. and these can be coded in
the digital realm.
The apllies with the manufactuers. They too have their vision on what They
think and want their product to sound.
I love reading on gearslutz the posts from Plugin developers and their interpretations
and opinions about what makes their Neve 1073 Eq better and what goes into
making their version sound like it does.. Each Developer has a different
vision as to what the Neve 1073 should sound like. And yet they all sound
good , but slightly different.
You stated that you use Vegas. Well as you know, Vegas has a very generic
sound..Just plain and simple. But, i bet you can tell the difference on
your system when you play that same file in Neundo (No, fx, eq, null-edzerro)..
???
"Dedric Terry" <dedric@echomg.com> wrote:
>Lamont - what is the output chain you are using for each app when comparing
>the file in Nuendo
>vs ProTools? On the same PC, I presume (and is this PT HD or M-Powered?)?
>Since these can't use the same output driver, you would have to depend on
>the D/A being
>the same, but clocking will be different unless you have a master clock,
and
>both interfaces
>are locking with the same accuracy. This was one of the issues that came
up
>at Lynn Fuston's
>D/A converter shootout - when do you lock to external clock and incur the
>resulting jitter,
>and when do you trust the internal clock - and if you do lock externally,
>how good is the PLL
>in the slave device? These issues can cause audible changes in the top
end
>that have nothing to do
>with the software itself. If you say that PTHD through the same converter
>output as Nuendo (via? RME?
>Lynx?) using the same master clock, sounds different playing a single audio
>file, then I take your word
>for it. I can't tell you why that is happening - only that an audible
>difference really shouldn't happen due
>to the software alone - not with a single audio file, esp. since I've heard
>and seen PTHD audio cancel with
>native DAWs. Just passing a single 16 or 24 bit track down the buss to
the
>output driver should
>be, and usually is, completely transparent, bit for bit.
>
>The same audio file played through the same converters should only sound
>different if something in
>the chain is different - be it clocking, gain or some degree of unintended,
>errant dsp processing. Every DAW should
>pass a single audio file without altering a single bit. That's a basic level
>of accuracy we should always
>expect of any DAW. If that accuracy isn't there, you can be sure a heavy
>mix will be altered in ways you
>didn't intend, even though you would end up mixing with that factor in place
>(e.g. you still mix for what
>you want to hear regardless of what the platform does to each audio track
or
>channel).
>
>In fact you should be able to send a stereo audio track out SPDIF or
>lightpipe to another DAW, record it
>bring the recorded file back in, line them up to the first bit, and have
>them cancel on and inverted phase
>test. I did this with Nuendo and Cubase 4 on separate machines just to
be
>sure my master clocking and
>slave sync was accurate - it worked perfectly.
>
>Also be sure there isn't a variation in the gain even by 0.1 dB between
the
>two. There shouldn't
>and I wouldn't expect there to be one. Also could PT be set for a different
>pan law? Shouldn't make a
>difference even if comparing two mono panned files to their stereo
>interleaved equivalent, but for sake
>of completeness it's worth checking as well. A variation in the output
>chain, be it drivers, audio card
>card, or converters would be the most likely culprit here.
>
>The reason DAW manufacturers wouldn't add any sonic "character"
>intentionally is that the
>ultimate goal from day one with recording has been to accurately reproduce
>what we hear.
>We developed a musical penchant for sonic character because the hardware
>just wasn't accurate,
>and what it did often sent us down new creative paths - even if by force
-
>and we decided it was
>preferred that way.
>
>Your point about what goes into the feature presets to sell synths is right
>for sure, but synths are about
>character and getting that "perfect piano" or crystal clear bell pad, or
fat
>punchy bass without spending
>a mint on development, adding 50G onboard sample libraries, or costing $15k,
>so what they
>lack in actual synthesis capabilities, they make up with EQ and effects
on
>the output. That's been the case
>for years, at least since we had effects on synths at least. But even with
>modern synths such as the Fantom,
>Tritons, etc, which are great synths all around, of course the coolest,
>widest and biggest patches
>will make the biggest impression - so in come the EQs, limiters, comps,
>reverbs, chorus, etc. The best
>way to find out if a synth is really good is to bypass all effects and see
>what happens. Most are pretty
>good these days, but about half the time, there are presets that fall
>completely flat in fx bypass.
>
>DAWs aren't designed to put a sonic fingerprint on a sound the way synths
>are - they are designed
>to *not* add anything - to pass through what we create as users, with no
>alteration (or as little as possible)
>beyond what we add with intentional processing (EQ, comps, etc). Developers
>would find no pride
>in hearing that their DAW sounds anything different than whatever is being
>played back in it,
>and the concept is contrary to what AES and IEEE proceedings on the issue
>propose in general
>digital audio discussions, white papers, etc.
>
>What ID ended up doing with Paris (at least from what I gather per Chuck's
>findings - so correct me if I'm missing part of the equation Chuck),
>is drop the track gain by 20dB or so, then added it back at the master buss
>to create the effect of headroom (probably
>because the master buss is really summing on the card, and they have more
>headroom there than on the tracks
>where native plugins might be used). I don't know if Paris passed 32-bit
>float files to the EDS card, but sort of
>doubt it. I think Chuck has clarified this at one point, but don't recall
>the answer.
>
>Also what Paris did is use a greater bit depth on the hardware than ProTools
>did - at the time PT was just
>bring Mix+ systems to market, or they had been out for a year or two (if
I
>have my timeline right) - they
>were 24-bit fixed all the way through. Logic and Cubase were native DAWs,
>but native was still too slow
>to compete with hardware hybrids. Paris trumped them all by running 32-bit
>float natively (not new really, but
>better than sticking to 24-bit) and 56 or so bits in hardware instead of
>going to Motorola DSPs at 24.
>The onboard effects were also a step up from anything out there, so the
demo
>did sound good.
>I don't recall which, but one of the demos, imho, wasn't so good (some
>sloppy production and
>vocals in spots, IIRC), so I only listened to it once. ;-)
>
>Coupled with the gain drop and buss makeup, this all gave it a "headroom"
no
>one else had. With very nice
>onboard effects, Paris jumped ahead of anything else out there easily, and
>still respectably holds its' own today
>in that department.
>
>Most demos I hear (when I listen to them) vary in quality, usually not so
>great in some area. But if a demo does
>sound great, then it at least says that the product is capable of at least
>that level of performance, and it can
>only help improve a prospective buyer's impression of it.
>
>Regards,
>Dedric
>
>"LaMont " <jjdpro@ameritech.net> wrote in message news:458c14c0$1@linux...
>>
>> Dedric good post..
>>
>> However, I have PT-M-Powered/M-audio 410 interface for my laptop and it
>> has
>> that same sound (no eq, zero fader) that HD does. I know their use the
>> same
>> 48 bit fix mixer. I load up the same file in Nuendo (no eq, zero
>> fader)..results.
>> different sonic character.
>>
>> PT having a top end touch..Nuendo, nice smooth(flat) sound. And I'm just
>> taking about a stereo wav file nulled with no eq..nothing ..zilch..nada..
>>
>> Now, there are devices (keyboards, dum machines) on the market today that
>> have a Master Buss Compressor and EQ set to on with the top end notched
>> up.
>> Why? because it gives their product an competitive advantageover the
>> competition..
>> Ex: Yahama's Motif ES, Akai's MPC 1000, 2500, Roland's Fantom.
>>
>> So, why would'nt a DAW manufactuer code in an extra (ooommf) to make their
>> DAW sound better. Especially, given the "I hate Digtal Summing" crowd?
>> And,
>> If I'm a DAW manufactuer, what would give my product a sonic edge over
the
>> competition?
>>
>> We live in the "louder is better" audio world these days, so a DAW that
>> can
>> catch my attention 'sonically" will probaly will get the sell. That's
what
>> happend to me back in 1997 when I heard Paris. I was floored!!! Still
to
>> this day, nothing has floored me like that "Road House Blues Demo" I heard
>> on Paris.
>>
>> Was it the hardware ? was it the software. I remember talking with Edmund
>> at the 2000 winter Namm, and told me that he & Steve set out to reproduce
>> the sonics of big buck analog board (eq's) and all.. And, summing was
a
>> big
>> big issue for them because they (ID) thought that nobody has gotten
>> it(summing)
>> right. And by right, they meant, behaved like a console with a wide lane
>> for all of those tracks..
>>
>>
>>
>>
>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>"LaMont" <jjdpro@ameritech.net> wrote in message news:458be8d5$1@linux...
>>>>
>>>> Okay...
>>>> I guess what I'm saying is this:
>>>>
>>>> -Is it possible that diferent DAW manufactuers "code" their app
>>>> differently
>>>> for sound results.
>>>
>>>Of course it is *possible* to do this, but only if the DAW has a specific
>>
>>>sound shaping purpose
>>>beyond normal summing/mixing. Users talk about wanting developers to
add
>> a
>>>"Neve sound" or "API sound" option to summing engines,
>>>but that's really impractical given the amount of dsp required to make
a
>>
>>>decent emulation (with convolution, dynamic EQ functions,
>>>etc). For sake of not eating up all cpu processing, that could likely
>>>only
>>
>>>surface as is a built in EQ, which
>>>no one wants universally in summing, and anyone can add at will already.
>>>
>>>So it hasn't happened yet and isn't likely to as it detours from the basic
>>
>>>tenant of audio recording - recreate what comes in as
>>>accurately as possible.
>>>
>>>What Digi did in recoding their summing engine was try to recover some
>>>of the damage done by the 24-bit buss in Mix systems. Motorola 56k dsps
>> are
>>>24-bit fixed point chips and I think
>>>the new generation (321?) still is, but they use double words now for
>>>48-bits). And though plugins could process at 48-bit by
>>>doubling up and using upper and lower 24-bit words for 48-bit outputs,
the
>>
>>>buss
>>>between chips was 24-bits, so they had to dither to 24-bits after every
>>
>>>plugin. The mixer (if I recall correctly) also
>>>had a 24-bit buss, so what Digi did is to add a dither stage to the mixer
>> to
>>>prevent this
>>>constant truncation of data. 24-bits isn't enough to cover summing for
>> more
>>>than a few tracks without
>>>losing information in the 16-bit world, and in the 24-bit world some
>>>information will be lost, at least at the lowest levels.
>>>
>>>Adding a dither stage (though I think they did more than that - perhaps
>>
>>>implement a 48-bit double word stage as well),
>>>simply smoothed over the truncation that was happening, but it didn't
>>>solve
>>
>>>the problem, so with HD
>>>they went to a double-word path - throughout I believe, including the
path
>>
>>>between chips. I believe the chips
>>>are still 24-bit, but by doubling up the processing (yes at a cost of
>>>twice
>>
>>>the overhead), they get a 48-bit engine.
>>>This not only provided better headroom, but greater resolution. Higher
>> bit
>>>depths subdivide the amplitude with greater resolution, and that's
>>>really where we get the definition of dynamic range - by lowering the
>>>signal
>>
>>>to quantization noise ratio.
>>>
>>>With DAWs that use 32-bit floating point math all the way through, the
>>>only
>>
>>>reason for altering the summing
>>>is by error, and that's an error that would actually be hard to make and
>> get
>>>past a very basic alpha stage of testing.
>>>There is a small difference in fixed point math and floating point math,
>> or
>>>at least a theoretical difference in how it affects audio
>>>in certain cases, but not necessarily in the result for calculating gain
>> in
>>>either for the same audio file. Where any differences might show up is
>>
>>>complicated, and I believe only appear at levels below 24-bit (or in
>>>headroom with tracks pushed beyond 0dBFS), or when/if
>>>there areany differences in where each amplitude level is quantized.
>>>
>>>Obviously there can be differences if the DAW has to use varying bit
>>>depths
>>
>>>throughout a single summing path to accomodate hardware
>>>as well as software summing, since there may be truncation or rounding
>>>along
>>
>>>the way, but that impacts the lowest bit
>>>level, and hence - spacial reproduction, reverb tails perhaps, and
>>>"depth",
>>
>>>not the levels most music so the differences are most
>>>often more subtle than not. But most modern DAWs have eliminated those
>>
>>>"rough edges" in the math by increasing the bit depth to accomodate normal
>>
>>>summing required for mixing audio.
>>>
>>>So with Lynn's unity gain summing test (A files on the CD I believe),
DAWs
>>
>>>were never asked to sum beyond 24-bits,
>>>at least not on the upper end of the dynamic range, so everything that
>>>could
>>
>>>represent 24-bits accurately would cancel. The only ones
>>>that didn't were ones that had a different bit depth and/or gain structure
>>
>>>whether hybrid or native
>>>(e.g. Paris' subtracting 20dB from tracks and adding it to the buss).
In
>>
>>>this case, PTHD cancelled (when I tested it) with
>>>Nuendo, Samplitude, Logic, etc because the impact of the 48-bit fixed
vs.
>>
>>>32-bit float wasn't a factor.
>>>
>>>When trying other tests, even when adding and subtracting gain, Nuendo,
>>
>>>Sequoia and Sonar cancel - both audibly and
>>>visually at inaudible levels, which only proves that one isn't making
an
>>
>>>error when calculating basic gain. Since a dB is well defined,
>>>and the math to add gain is simple, they shouldn't. The fact that they
>> all
>>>use 32-bit float all the way through eliminates a difference
>>>in data structure as well, and this just verifies that. There was a time
>>
>>>that supposedly Logic (v3, v4?) was partly 24-bit, or so the rumor went,
>>>but it's 32-bit float all the way through now just as Sonar,
>>>Nuendo/Cubase,
>>
>>>Samplitude/Sequoia, DP, Audition (I presume at least).
>>>I don't know what Acid or Live use. Saw promotes a fixed point engine,
>> but
>>>I don't know if it is still 24-bit, or now 48 bit.
>>>That was an intentional choice by the developer, but he's the only one
I
>>
>>>know of that stuck with 24-bit for summing
>>>intentionally, esp. after the Digi Mix system mixer incident.
>>>
>>>Long answer, but to sum up, it is certainly physically *possible* for
a
>>
>>>developer to code something differently intentionally, but not
>>>in reality likely since it would be breaking some basic fixed point or
>>>floating point math rules. Where the differences really
>>>showed up in the past is with PT Mix systems where the limitation was
>>>really
>>
>>>significant - e.g. 24 bit with truncation at several stages.
>>>
>>>That really isn't such an issue anymore. Given the differences in
>>>workflow,
>>
>>>missing something in workflow or layout differences
>>>is easy enough to do (e.g. Sonar doesn't have group and busses the way
>>>Nuendo does, as it's outputs are actually driver outputs,
>>>not software busses, so in Sonar, busses are actually outputs, and sub
>>>busses are actually busses in Nuendo. There are no,
>>>or at least I haven't found the equivalent of a Nuendo group in Sonar
-
>> that
>>>affects the results of some tests (though not basic
>>>summing) if not taken into account, but when taken into account, they
work
>>
>>>exactly the same way).
>>>
>>>So at least when talking about apps with 32-bit float all the way through,
>>
>>>it's safe to say (since it has been proven) that summing isn't different
>>
>>>unless
>>>there is an error somewhere, or variation in how the user duplicates the
>>
>>>same mix in two different apps.
>>>
>>>Imho, that's actually a very good thing - approaching a more consistent
>>
>>>basis for recording and mixing from which users can make all
>>>of the decisions as to how the final product will sound and not be
>>>required
>>
>>>to decide when purchasing a pricey console, and have to
>>>focus their business on clients who want "that sound". I believe we are
>>
>>>actually closer to the pure definition of recording now than
>>>we once were.
>>>
>>>Regards,
>>>Dedric
>>>
>>>
>>>>
>>>> I the answer is yes, then,the real task is to discover or rather
>>>> un-cover
>>>> what's say: Motu's vision of summing, versus Digidesign, versus
>>>> Steinberg
>>>> and so on..
>>>>
>>>> What's under the hood. To me and others,when Digi re-coded their summing
>>>> engine, it was obvious that Pro Tools has an obvious top end (8k-10k)
>>
>>>> bump.
>>>> Where as Steinberg's summing is very neutral.
>>>>
>>>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>>>Hi Neil,
>>>>>
>>>>>Jamie is right. And you aren't wacked out - you are thinking this
>>>>>through
>>>>
>>>>>in a reasonable manner, but coming to the wrong
>>>>>conclusion - easy to do given how confusing digital audio can be. Each
>>>> word
>>>>>represents an amplitude
>>>>>point on a single curve that is changing over time, and can vary with
>> a
>>>>
>>>>>speed up to the Nyquist frequency (as Jamie described).
>>>>>The complex harmonic content we hear is actually the frequency
>>>>>modulation
>>>> of
>>>>>a single waveform,
>>>>>that over a small amount of time creates the sound we translate - we
>>>>>don't
>>>>
>>>>>really hear a single sample at a time,
>>>>>but thousands of samples at a time (1 sample alone could at most
>>>>>represent
>>>> a
>>>>>single positive or negative peak
>>>>>of a 22,050Hz waveform).
>>>>>
>>>>>If one bit doesn't cancel, esp. if it's a higher order bit than number
>> 24,
>>>>
>>>>>you may hear, and will see that easily,
>>>>>and the higher the bit in the dynamic range (higher order) the more
>>>>>audible
>>>>
>>>>>the difference.
>>>>>Since each bit is 6dB of dynamic range, you can extrapolate how "loud"
>>
>>>>>that
>>>>
>>>>>bit's impact will be
>>>>>if there is a variation.
>>>>>
>>>>>Now, obviously if we are talking about 1 sample in a 44.1k rate song,
>> then
>>>>
>>>>>it simply be a
>>>>>click (only audible if it's a high enough order bit) instead of an
>>>>>obvious
>>>>
>>>>>musical difference, but that should never
>>>>>happen in a phase cancellation test between identical files higher than
>>>> bit
>>>>>24, unless there are clock sync problems,
>>>>>driver issues, or the DAW is an early alpha version. :-)
>>>>>
>>>>>By definition of what DAWs do during playback and record, every audio
>>
>>>>>stream
>>>>
>>>>>has the same point in time (judged by the timeline)
>>>>>played back sample accurately, one word at a time, at whatever sample
>>
>>>>>rate
>>>>
>>>>>we are using. A phase cancellation test uses that
>>>>>fact to compare two audio files word for word (and hence bit for bit
>>>>>since
>>>>
>>>>>each bit of a 24-bit word would
>>>>>be at the same bit slot in each 24-bit word). Assuming they are aligned
>>>> to
>>>>>the same start point, sample
>>>>>accurately, and both are the same set of sample words at each sample
>>>>>point,
>>>>
>>>>>bit for bit, and one is phase inverted,
>>>>>they will cancel through all 24 bits. For two files to cancel
>>>>>completely
>>>>
>>>>>for the duration of the file, each and every bit in each word
>>>>>must be the exact opposite of that same bit position in a word at the
>> same
>>>>
>>>>>sample point. This is why zooming in on an FFT
>>>>>of the full difference file is valuable as it can show any differences
>> in
>>>>
>>>>>the lower order bits that wouldn't be audible. So even if
>>>>>there is no audible difference, the visual followup will show if the
two
>>>>
>>>>>files truly cancel even a levels below hearing, or
>>>>>outside of a frequency change that we will perceive.
>>>>>
>>>>>When they don't cancel, usually there will be way more than 1 bit
>>>>>difference - it's usually one or more bits in the words for
>>>>>thousands of samples. From a musical standpoint this is usually in
a
>>>>>frequency range (low freq, or high freq most often) - that will
>>>>>show up as the difference between them, and that usually happens due
to
>>>> some
>>>>>form of processing difference between the files,
>>>>>such as EQ, compression, frequency dependant gain changes, etc. That
is
>>>> what
>>>>>I believe you are thinking through, but when
>>>>>talking about straight summing with no gain change (or known equal gain
>>>>
>>>>>changes), we are only looking at linear, one for one
>>>>>comparisons between the two files' frequency representations.
>>>>>
>>>>>Regards,
>>>>>Dedric
>>>>>
>>>>>> Neil wrote:
>>>>>>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>>>>>> The tests I did were completely blank down to -200 dB (far below
the
>>>>
>>>>>>>> last
>>>>>>>
>>>>>>>> bit). It's safe to say there is no difference, even in
>>>>>>>> quantization noise, which by technical rights, is considered below
>> the
>>>>
>>>>>>>> level
>>>>>>>
>>>>>>>> of "cancellation" in such tests.
>>>>>>>
>>>>>>> I'm not necessarily talking about just the first bit or the
>>>>>>> last bit, but also everything in between... what happens on bit
>>>>>>> #12, for example? Everything on bit #12 should be audible, but
>>>>>>> in an a/b test what if thre are differences in what bits #8
>>>>>>> through #12 sound like, but the amplutide is stll the same on
>>>>>>> both files at that point, you'll get a null, right? Extrapolate
>>>>>>> that out somewhat & let's say there are differences in bits #8
>>>>>>> through #12 on sample points 3, 17, 1,000, 4,523, 7,560, etc,
>>>>>>> etc through 43,972... Now this is breaking things down well
>>>>>>> beyond what I think can be measured, if I'm not mistaken (I
>>>>>>> dn't know of any way we could extract JUST that information
>>>>>>> from each file & play it back for an a/b test; but would not
>>>>>>> that be enough to have to "null-able" files that do actually
>>>>>>> sound somewhat different?
>>>>>>>
>>>>>>> I guess what I'm saying is that since each sample in a musical
>>>>>>> track or full song file doesn't represent a pure, simple set of
>>>>>>> content like a sample of a sine wave would - there's a whole
>>>>>>> world of harmonic structure in each sample of a song file, and
>>>>>>> I think (although I'll admit - I can't "prove") that there is
>>>>>>> plenty of room for some variables between the first bit & the
>>>>>>> last bit while still allowing for a null test to be successful.
>>>>>>>
>>>>>>> No? Am I wacked out of my mind?
>>>>>>>
>>>>>>> Neil
>>>>>>>
>>>>>
>>>>
>>>
>>>
>>
>
>Best of luck to you guys. Guy Clark has been getting lots of airplay on our
local NPR affiliate here. I've been so busy that it's been like my life's
background music lately and I don't know if it's your new project, but I've
been enjoying it.
Merry Christmas,
Deej
"Chris Latham" <latham_c@bellsouth.net> wrote in message
news:458b70fc@linux...
> For those of you who will be voting for the 49th annual Grammys, please
> give a listen to, and consider Guy Clark's "Workbench Songs" for Best
> Contemporary Folk/Americana Album (Catagory 68 in Field 14 - Folk).
> Edited and mixed entirely in PARIS.
>
> THANK YOU,
> Chris
>Hi Mike,
This is relatively easy to do. You will need an Ethernet network card in
the PC and a Category 5 Ethernet "crossover" cable.
Install the card in the PC, install the drivers as per the instructions that
came with the card, and Windows will automatically install file and printer
sharing services. Connect the crossover cable to the modular (RJ45) jack
on each machine.
On the Mac, you must use the Network system preferences control to enable
Windows file sharing.
Assign the Mac an IP address of 192.168.0.1 and the PC an address of 192.168.0.2.
Set the network (subnet) mask value to 255.255.0.0 on each machine.
Apple has instructions on how to do this on their web site and it can be
easily located.
Also, have you considered using an external storage device, such as a USB
flash drive or external hard drive, to move the files back and forth? This
may be much simpler and quicker if you're not into doing the network thing.
Hope this helps. Peace & happy holidays!
Mark
"Mike R." <emarenot@yahoo.com> wrote:
>
>
>I've got a Mac G4 (OSX 10.4) and a PC running Windows 98SE and Paris. I
>want to be able to move .wav files made on the G4 over to Paris, and from
>Paris back over to the Mac --if I want to apply FX there that I cannot in
>Paris. As I understand it, the G4 has an ethernet card built in. I assume
>that I have to get a NIC card for the PC. Do I need to get some sort of
>software for each machine so that the PC will see the Mac and vice versa?
>This is totally new territory. I wish it was as simple as plugging in a
>cable between two boxes, but I fear it is not...
>Thanks much,
>MR
>
>
><!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
><HTML><HEAD>
><META http-equiv=3DContent-Type content=3D"text/html; =
>charset=3Diso-8859-1">
><META content=3D"MSHTML 6.00.2800.1555" name=3DGENERATOR>
><STYLE></STYLE>
></HEAD>
><BODY bgColor=3D#ffffff>
><DIV><FONT face=3DArial size=3D2>I've got a Mac G4 (OSX 10.4) and a PC =
>running=20
>Windows 98SE and Paris. I want to be able to move .wav files made =
>on the=20
>G4 over to Paris, and from Paris back over to the Mac --if I want to =
>apply FX=20
>there that I cannot in Paris. As I understand it, the G4 has an =
>ethernet=20
>card built in. I assume that I have to get a NIC card for the PC. =
>Do I=20
>need to get some sort of software for each machine so that the PC will =
>see the=20
>Mac and vice versa? This is totally new territory. I wish it =
>was as=20
>simple as plugging in a cable between two boxes, but I fear it is=20
>not...</FONT></DIV>
><DIV><FONT face=3DArial size=3D2>Thanks much,</FONT></DIV>
><DIV><FONT face=3DArial size=3D2>MR</FONT></DIV></BODY></HTML>
>
>I can't tell you why you hear ProTools differently than Nuendo using a
single file.
There isn't any voodoo in the software, or hidden character enhancing dsp.
I'll see if
I can round up an M-Powered system to compare with next month.
For reference, everytime I open Sequoia I think I might hear a broader,
clean,
and almost flat (spectrum, not depth) sound, but I don't - it's the same as
Nuendo, fwiw.
Also I don't think what I was referring to was a theory from Chuck - I
believe that was what he
discovered in the code.
Digital mixers all have different preamps and converters. Unless you are
bypassing every
EQ and converter and going digital in and out to the same converter when
comparing, it would be hard
to say the mix engine itself sounds different than another mixer, but taken
as a whole, then
certainly they may very well sound different. In addition, hardware digital
mixers may use a variety of different paths between the I/O, channel
processing, and summing,
though most are pretty much software mixers on a single chip or set of dsps
similar to ProTools,
with I/O and a hardware surface attached.
I know it may be hard to separate the mix engine as software in either a
native DAW
or a digital mixer, from the hardware that translates the audio to something
we hear,
but that's what is required when comparing summing. The hardware can
significantly change
what we hear, so comparing digital mixers really isn't of as much interest
as comparing native
DAWs in that respect - unless you are looking to buy one of course.
Even though I know you think manufacturers are trying to add something to
give them an edge, I am 100%
sure that isn't the case - rather they are trying to add or change as little
as possible in order to give
them the edge. Their end of digital audio isn't about recreating the past,
but improving upon it.
As we've discussed and agreed before, the obsession with recreating
"vintage" technology is as much
fad as it is a valuable creative asset. There is no reason we shouldn't
have far superior hardware and software EQs and comps
than 20, 30 or 40 years ago. No reason at all, other than market demand,
but the majority of software, and new
hardware gear on the market has a vintage marketing tagline with it.
Companies will sell any bill of
goods if customers will buy it.
There's nothing unique about the summing in Nuendo, Cubase, Sequoia/Samp,
or Sonar, and it's pretty safe to include Logic and DP in that list as well.
One of the reasons I test
these things is to be sure my DAW isn't doing something wrong, or something
I don't know about.
Vegas - I use it for video conversions and have never done any critical
listening tests with it. What I have heard
briefly didn't sound any different. It certainly looks plain vanilla
though. What you are describing is exactly
what I would say about the GUIs of each of those apps, not that it means
anything. Just interesting.
That's one reason I listen eyes closed and double check with phase
cancellation tests and FFTs - I am
influenced creatively by the GUI to some degree. I actually like Cubase 4's
GUI better than Nuendo 3.2,
though there are only slight visual differences (some workflow differences
are a definite improvement for me though).
ProTools' GUI always made me want to write one dimensional soundtracks in
mono for public utilities, accounting offices
or the IRS while reading my discreet systems analysis textbook - it was also
grey. ;-)
Regards,
Dedric
"LaMont" <jjdpro@ameritech.net> wrote in message news:458c82fd$1@linux...
>
> Dedric, my simple test is simple..
> Using the same audio interface, with the same stereo file..null-ed to
> zero..No
> eq, for fx. Master fader on zero..
>
> Nuendo, Pro-Tools -Mpowered(native)... yields a sonic difference that I
> have
> referenced before.. The sound coming from PT-M has a nice top end , where
> as Neundo has a nice flatter sound quality.
> Same audio interface. M-audio 410..Using Mackies & Blue-Sky pro monitors..
>
> Same test at the big room..PT-HD & Neundo Logic Audio(macG5-Dual) Using
> the
> 192 interface.
> Same results..But adding Logic audio's sound ..(Broad, thick)
>
> Somethings going on.
>
> Chucks post about how paris handles audio is a theory..Only Edmund can
> truly
> give us the goods on what's really what..
>
> I disagree that manufactuers don;t set out o put a sonic print on their
> products.
> I think they do.
>
> I have been fortunate to work on some digital mixers and I can tell you
> that
> each one has their own sound. The Sony Dmx-100 was modeled after SSL 4000g
> (like it's Big Brother).And you what? That board (Dmx-100) sound very warm
> and it's eq tries to behave and sound just like an SSL.. Unlike he Yamaha
> Dm2000(version 1.x) which has a very Clean, neutral sound..However, some
> complained that it was tooo Vanila and thus, Yamaha add a version 2.0
> which
> added Vintage type Eq's, modeled analog input gain saturation fx too give
> the user a choice Btw Clean and Neutral vs sonic Character.
>
> So, if digital conoles can be given a sonic character, why not a software
> mixer?
> The truth is, there are some folks who want a neutral mixer and then there
> are others who want a sonic footprint imparted. and these can be coded in
> the digital realm.
> The apllies with the manufactuers. They too have their vision on what They
> think and want their product to sound.
>
> I love reading on gearslutz the posts from Plugin developers and their
> interpretations
> and opinions about what makes their Neve 1073 Eq better and what goes into
> making their version sound like it does.. Each Developer has a different
> vision as to what the Neve 1073 should sound like. And yet they all sound
> good , but slightly different.
>
> You stated that you use Vegas. Well as you know, Vegas has a very generic
> sound..Just plain and simple. But, i bet you can tell the difference on
> your system when you play that same file in Neundo (No, fx, eq,
> null-edzerro)..
> ???
>
>
> "Dedric Terry" <dedric@echomg.com> wrote:
>>Lamont - what is the output chain you are using for each app when
>>comparing
>
>>the file in Nuendo
>>vs ProTools? On the same PC, I presume (and is this PT HD or M-Powered?)?
>>Since these can't use the same output driver, you would have to depend on
>
>>the D/A being
>>the same, but clocking will be different unless you have a master clock,
> and
>>both interfaces
>>are locking with the same accuracy. This was one of the issues that came
> up
>>at Lynn Fuston's
>>D/A converter shootout - when do you lock to external clock and incur the
>
>>resulting jitter,
>>and when do you trust the internal clock - and if you do lock externally,
>
>>how good is the PLL
>>in the slave device? These issues can cause audible changes in the top
> end
>>that have nothing to do
>>with the software itself. If you say that PTHD through the same converter
>
>>output as Nuendo (via? RME?
>>Lynx?) using the same master clock, sounds different playing a single
>>audio
>
>>file, then I take your word
>>for it. I can't tell you why that is happening - only that an audible
>>difference really shouldn't happen due
>>to the software alone - not with a single audio file, esp. since I've
>>heard
>
>>and seen PTHD audio cancel with
>>native DAWs. Just passing a single 16 or 24 bit track down the buss to
> the
>>output driver should
>>be, and usually is, completely transparent, bit for bit.
>>
>>The same audio file played through the same converters should only sound
>
>>different if something in
>>the chain is different - be it clocking, gain or some degree of
>>unintended,
>
>>errant dsp processing. Every DAW should
>>pass a single audio file without altering a single bit. That's a basic
>>level
>
>>of accuracy we should always
>>expect of any DAW. If that accuracy isn't there, you can be sure a heavy
>
>>mix will be altered in ways you
>>didn't intend, even though you would end up mixing with that factor in
>>place
>
>>(e.g. you still mix for what
>>you want to hear regardless of what the platform does to each audio track
> or
>>channel).
>>
>>In fact you should be able to send a stereo audio track out SPDIF or
>>lightpipe to another DAW, record it
>>bring the recorded file back in, line them up to the first bit, and have
>
>>them cancel on and inverted phase
>>test. I did this with Nuendo and Cubase 4 on separate machines just to
> be
>>sure my master clocking and
>>slave sync was accurate - it worked perfectly.
>>
>>Also be sure there isn't a variation in the gain even by 0.1 dB between
> the
>>two. There shouldn't
>>and I wouldn't expect there to be one. Also could PT be set for a
>>different
>
>>pan law? Shouldn't make a
>>difference even if comparing two mono panned files to their stereo
>>interleaved equivalent, but for sake
>>of completeness it's worth checking as well. A variation in the output
>
>>chain, be it drivers, audio card
>>card, or converters would be the most likely culprit here.
>>
>>The reason DAW manufacturers wouldn't add any sonic "character"
>>intentionally is that the
>>ultimate goal from day one with recording has been to accurately reproduce
>
>>what we hear.
>>We developed a musical penchant for sonic character because the hardware
>
>>just wasn't accurate,
>>and what it did often sent us down new creative paths - even if by force
> -
>>and we decided it was
>>preferred that way.
>>
>>Your point about what goes into the feature presets to sell synths is
>>right
>
>>for sure, but synths are about
>>character and getting that "perfect piano" or crystal clear bell pad, or
> fat
>>punchy bass without spending
>>a mint on development, adding 50G onboard sample libraries, or costing
>>$15k,
>
>>so what they
>>lack in actual synthesis capabilities, they make up with EQ and effects
> on
>>the output. That's been the case
>>for years, at least since we had effects on synths at least. But even
>>with
>
>>modern synths such as the Fantom,
>>Tritons, etc, which are great synths all around, of course the coolest,
>
>>widest and biggest patches
>>will make the biggest impression - so in come the EQs, limiters, comps,
>
>>reverbs, chorus, etc. The best
>>way to find out if a synth is really good is to bypass all effects and see
>
>>what happens. Most are pretty
>>good these days, but about half the time, there are presets that fall
>>completely flat in fx bypass.
>>
>>DAWs aren't designed to put a sonic fingerprint on a sound the way synths
>
>>are - they are designed
>>to *not* add anything - to pass through what we create as users, with no
>
>>alteration (or as little as possible)
>>beyond what we add with intentional processing (EQ, comps, etc).
>>Developers
>
>>would find no pride
>>in hearing that their DAW sounds anything different than whatever is being
>
>>played back in it,
>>and the concept is contrary to what AES and IEEE proceedings on the issue
>
>>propose in general
>>digital audio discussions, white papers, etc.
>>
>>What ID ended up doing with Paris (at least from what I gather per Chuck's
>
>>findings - so correct me if I'm missing part of the equation Chuck),
>>is drop the track gain by 20dB or so, then added it back at the master
>>buss
>
>>to create the effect of headroom (probably
>>because the master buss is really summing on the card, and they have more
>
>>headroom there than on the tracks
>>where native plugins might be used). I don't know if Paris passed 32-bit
>
>>float files to the EDS card, but sort of
>>doubt it. I think Chuck has clarified this at one point, but don't recall
>
>>the answer.
>>
>>Also what Paris did is use a greater bit depth on the hardware than
>>ProTools
>
>>did - at the time PT was just
>>bring Mix+ systems to market, or they had been out for a year or two (if
> I
>>have my timeline right) - they
>>were 24-bit fixed all the way through. Logic and Cubase were native DAWs,
>
>>but native was still too slow
>>to compete with hardware hybrids. Paris trumped them all by running
>>32-bit
>
>>float natively (not new really, but
>>better than sticking to 24-bit) and 56 or so bits in hardware instead of
>
>>going to Motorola DSPs at 24.
>>The onboard effects were also a step up from anything out there, so the
> demo
>>did sound good.
>>I don't recall which, but one of the demos, imho, wasn't so good (some
>>sloppy production and
>>vocals in spots, IIRC), so I only listened to it once. ;-)
>>
>>Coupled with the gain drop and buss makeup, this all gave it a "headroom"
> no
>>one else had. With very nice
>>onboard effects, Paris jumped ahead of anything else out there easily, and
>
>>still respectably holds its' own today
>>in that department.
>>
>>Most demos I hear (when I listen to them) vary in quality, usually not so
>
>>great in some area. But if a demo does
>>sound great, then it at least says that the product is capable of at
>>least
>
>>that level of performance, and it can
>>only help improve a prospective buyer's impression of it.
>>
>>Regards,
>>Dedric
>>
>>"LaMont " <jjdpro@ameritech.net> wrote in message news:458c14c0$1@linux...
>>>
>>> Dedric good post..
>>>
>>> However, I have PT-M-Powered/M-audio 410 interface for my laptop and it
>
>>> has
>>> that same sound (no eq, zero fader) that HD does. I know their use the
>
>>> same
>>> 48 bit fix mixer. I load up the same file in Nuendo (no eq, zero
>>> fader)..results.
>>> different sonic character.
>>>
>>> PT having a top end touch..Nuendo, nice smooth(flat) sound. And I'm just
>>> taking about a stereo wav file nulled with no eq..nothing
>>> ..zilch..nada..
>>>
>>> Now, there are devices (keyboards, dum machines) on the market today
>>> that
>>> have a Master Buss Compressor and EQ set to on with the top end notched
>
>>> up.
>>> Why? because it gives their product an competitive advantageover the
>>> competition..
>>> Ex: Yahama's Motif ES, Akai's MPC 1000, 2500, Roland's Fantom.
>>>
>>> So, why would'nt a DAW manufactuer code in an extra (ooommf) to make
>>> their
>>> DAW sound better. Especially, given the "I hate Digtal Summing" crowd?
>
>>> And,
>>> If I'm a DAW manufactuer, what would give my product a sonic edge over
> the
>>> competition?
>>>
>>> We live in the "louder is better" audio world these days, so a DAW that
>
>>> can
>>> catch my attention 'sonically" will probaly will get the sell. That's
> what
>>> happend to me back in 1997 when I heard Paris. I was floored!!! Still
> to
>>> this day, nothing has floored me like that "Road House Blues Demo" I
>>> heard
>>> on Paris.
>>>
>>> Was it the hardware ? was it the software. I remember talking with
>>> Edmund
>>> at the 2000 winter Namm, and told me that he & Steve set out to
>>> reproduce
>>> the sonics of big buck analog board (eq's) and all.. And, summing was
> a
>>> big
>>> big issue for them because they (ID) thought that nobody has gotten
>>> it(summing)
>>> right. And by right, they meant, behaved like a console with a wide lane
>>> for all of those tracks..
>>>
>>>
>>>
>>>
>>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>>"LaMont" <jjdpro@ameritech.net> wrote in message
>>>>news:458be8d5$1@linux...
>>>>>
>>>>> Okay...
>>>>> I guess what I'm saying is this:
>>>>>
>>>>> -Is it possible that diferent DAW manufactuers "code" their app
>>>>> differently
>>>>> for sound results.
>>>>
>>>>Of course it is *possible* to do this, but only if the DAW has a
>>>>specific
>>>
>>>>sound shaping purpose
>>>>beyond normal summing/mixing. Users talk about wanting developers to
> add
>>> a
>>>>"Neve sound" or "API sound" option to summing engines,
>>>>but that's really impractical given the amount of dsp required to make
> a
>>>
>>>>decent emulation (with convolution, dynamic EQ functions,
>>>>etc). For sake of not eating up all cpu processing, that could likely
>
>>>>only
>>>
>>>>surface as is a built in EQ, which
>>>>no one wants universally in summing, and anyone can add at will already.
>>>>
>>>>So it hasn't happened yet and isn't likely to as it detours from the
>>>>basic
>>>
>>>>tenant of audio recording - recreate what comes in as
>>>>accurately as possible.
>>>>
>>>>What Digi did in recoding their summing engine was try to recover some
>>>>of the damage done by the 24-bit buss in Mix systems. Motorola 56k dsps
>>> are
>>>>24-bit fixed point chips and I think
>>>>the new generation (321?) still is, but they use double words now for
>>>>48-bits). And though plugins could process at 48-bit by
>>>>doubling up and using upper and lower 24-bit words for 48-bit outputs,
> the
>>>
>>>>buss
>>>>between chips was 24-bits, so they had to dither to 24-bits after every
>>>
>>>>plugin. The mixer (if I recall correctly) also
>>>>had a 24-bit buss, so what Digi did is to add a dither stage to the
>>>>mixer
>>> to
>>>>prevent this
>>>>constant truncation of data. 24-bits isn't enough to cover summing for
>>> more
>>>>than a few tracks without
>>>>losing information in the 16-bit world, and in the 24-bit world some
>>>>information will be lost, at least at the lowest levels.
>>>>
>>>>Adding a dither stage (though I think they did more than that - perhaps
>>>
>>>>implement a 48-bit double word stage as well),
>>>>simply smoothed over the truncation that was happening, but it didn't
>
>>>>solve
>>>
>>>>the problem, so with HD
>>>>they went to a double-word path - throughout I believe, including the
> path
>>>
>>>>between chips. I believe the chips
>>>>are still 24-bit, but by doubling up the processing (yes at a cost of
>
>>>>twice
>>>
>>>>the overhead), they get a 48-bit engine.
>>>>This not only provided better headroom, but greater resolution. Higher
>>> bit
>>>>depths subdivide the amplitude with greater resolution, and that's
>>>>really where we get the definition of dynamic range - by lowering the
>
>>>>signal
>>>
>>>>to quantization noise ratio.
>>>>
>>>>With DAWs that use 32-bit floating point math all the way through, the
>
>>>>only
>>>
>>>>reason for altering the summing
>>>>is by error, and that's an error that would actually be hard to make and
>>> get
>>>>past a very basic alpha stage of testing.
>>>>There is a small difference in fixed point math and floating point math,
>>> or
>>>>at least a theoretical difference in how it affects audio
>>>>in certain cases, but not necessarily in the result for calculating gain
>>> in
>>>>either for the same audio file. Where any differences might show up is
>>>
>>>>complicated, and I believe only appear at levels below 24-bit (or in
>>>>headroom with tracks pushed beyond 0dBFS), or when/if
>>>>there areany differences in where each amplitude level is quantized.
>>>>
>>>>Obviously there can be differences if the DAW has to use varying bit
>>>>depths
>>>
>>>>throughout a single summing path to accomodate hardware
>>>>as well as software summing, since there may be truncation or rounding
>
>>>>along
>>>
>>>>the way, but that impacts the lowest bit
>>>>level, and hence - spacial reproduction, reverb tails perhaps, and
>>>>"depth",
>>>
>>>>not the levels most music so the differences are most
>>>>often more subtle than not. But most modern DAWs have eliminated those
>>>
>>>>"rough edges" in the math by increasing the bit depth to accomodate
>>>>normal
>>>
>>>>summing required for mixing audio.
>>>>
>>>>So with Lynn's unity gain summing test (A files on the CD I believe),
> DAWs
>>>
>>>>were never asked to sum beyond 24-bits,
>>>>at least not on the upper end of the dynamic range, so everything that
>
>>>>could
>>>
>>>>represent 24-bits accurately would cancel. The only ones
>>>>that didn't were ones that had a different bit depth and/or gain
>>>>structure
>>>
>>>>whether hybrid or native
>>>>(e.g. Paris' subtracting 20dB from tracks and adding it to the buss).
> In
>>>
>>>>this case, PTHD cancelled (when I tested it) with
>>>>Nuendo, Samplitude, Logic, etc because the impact of the 48-bit fixed
> vs.
>>>
>>>>32-bit float wasn't a factor.
>>>>
>>>>When trying other tests, even when adding and subtracting gain, Nuendo,
>>>
>>>>Sequoia and Sonar cancel - both audibly and
>>>>visually at inaudible levels, which only proves that one isn't making
> an
>>>
>>>>error when calculating basic gain. Since a dB is well defined,
>>>>and the math to add gain is simple, they shouldn't. The fact that they
>>> all
>>>>use 32-bit float all the way through eliminates a difference
>>>>in data structure as well, and this just verifies that. There was a
>>>>time
>>>
>>>>that supposedly Logic (v3, v4?) was partly 24-bit, or so the rumor went,
>>>>but it's 32-bit float all the way through now just as Sonar,
>>>>Nuendo/Cubase,
>>>
>>>>Samplitude/Sequoia, DP, Audition (I presume at least).
>>>>I don't know what Acid or Live use. Saw promotes a fixed point engine,
>>> but
>>>>I don't know if it is still 24-bit, or now 48 bit.
>>>>That was an intentional choice by the developer, but he's the only one
> I
>>>
>>>>know of that stuck with 24-bit for summing
>>>>intentionally, esp. after the Digi Mix system mixer incident.
>>>>
>>>>Long answer, but to sum up, it is certainly physically *possible* for
> a
>>>
>>>>developer to code something differently intentionally, but not
>>>>in reality likely since it would be breaking some basic fixed point or
>>>>floating point math rules. Where the differences really
>>>>showed up in the past is with PT Mix systems where the limitation was
>
>>>>really
>>>
>>>>significant - e.g. 24 bit with truncation at several stages.
>>>>
>>>>That really isn't such an issue anymore. Given the differences in
>>>>workflow,
>>>
>>>>missing something in workflow or layout differences
>>>>is easy enough to do (e.g. Sonar doesn't have group and busses the way
>>>>Nuendo does, as it's outputs are actually driver outputs,
>>>>not software busses, so in Sonar, busses are actually outputs, and sub
>>>>busses are actually busses in Nuendo. There are no,
>>>>or at least I haven't found the equivalent of a Nuendo group in Sonar
> -
>>> that
>>>>affects the results of some tests (though not basic
>>>>summing) if not taken into account, but when taken into account, they
> work
>>>
>>>>exactly the same way).
>>>>
>>>>So at least when talking about apps with 32-bit float all the way
>>>>through,
>>>
>>>>it's safe to say (since it has been proven) that summing isn't different
>>>
>>>>unless
>>>>there is an error somewhere, or variation in how the user duplicates the
>>>
>>>>same mix in two different apps.
>>>>
>>>>Imho, that's actually a very good thing - approaching a more consistent
>>>
>>>>basis for recording and mixing from which users can make all
>>>>of the decisions as to how the final product will sound and not be
>>>>required
>>>
>>>>to decide when purchasing a pricey console, and have to
>>>>focus their business on clients who want "that sound". I believe we are
>>>
>>>>actually closer to the pure definition of recording now than
>>>>we once were.
>>>>
>>>>Regards,
>>>>Dedric
>>>>
>>>>
>>>>>
>>>>> I the answer is yes, then,the real task is to discover or rather
>>>>> un-cover
>>>>> what's say: Motu's vision of summing, versus Digidesign, versus
>>>>> Steinberg
>>>>> and so on..
>>>>>
>>>>> What's under the hood. To me and others,when Digi re-coded their
>>>>> summing
>>>>> engine, it was obvious that Pro Tools has an obvious top end (8k-10k)
>>>
>>>>> bump.
>>>>> Where as Steinberg's summing is very neutral.
>>>>>
>>>>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>>>>Hi Neil,
>>>>>>
>>>>>>Jamie is right. And you aren't wacked out - you are thinking this
>>>>>>through
>>>>>
>>>>>>in a reasonable manner, but coming to the wrong
>>>>>>conclusion - easy to do given how confusing digital audio can be.
>>>>>>Each
>>>>> word
>>>>>>represents an amplitude
>>>>>>point on a single curve that is changing over time, and can vary with
>>> a
>>>>>
>>>>>>speed up to the Nyquist frequency (as Jamie described).
>>>>>>The complex harmonic content we hear is actually the frequency
>>>>>>modulation
>>>>> of
>>>>>>a single waveform,
>>>>>>that over a small amount of time creates the sound we translate - we
>
>>>>>>don't
>>>>>
>>>>>>really hear a single sample at a time,
>>>>>>but thousands of samples at a time (1 sample alone could at most
>>>>>>represent
>>>>> a
>>>>>>single positive or negative peak
>>>>>>of a 22,050Hz waveform).
>>>>>>
>>>>>>If one bit doesn't cancel, esp. if it's a higher order bit than number
>>> 24,
>>>>>
>>>>>>you may hear, and will see that easily,
>>>>>>and the higher the bit in the dynamic range (higher order) the more
>>>>>>audible
>>>>>
>>>>>>the difference.
>>>>>>Since each bit is 6dB of dynamic range, you can extrapolate how "loud"
>>>
>>>>>>that
>>>>>
>>>>>>bit's impact will be
>>>>>>if there is a variation.
>>>>>>
>>>>>>Now, obviously if we are talking about 1 sample in a 44.1k rate song,
>>> then
>>>>>
>>>>>>it simply be a
>>>>>>click (only audible if it's a high enough order bit) instead of an
>>>>>>obvious
>>>>>
>>>>>>musical difference, but that should never
>>>>>>happen in a phase cancellation test between identical files higher
>>>>>>than
>>>>> bit
>>>>>>24, unless there are clock sync problems,
>>>>>>driver issues, or the DAW is an early alpha version. :-)
>>>>>>
>>>>>>By definition of what DAWs do during playback and record, every audio
>>>
>>>>>>stream
>>>>>
>>>>>>has the same point in time (judged by the timeline)
>>>>>>played back sample accurately, one word at a time, at whatever sample
>>>
>>>>>>rate
>>>>>
>>>>>>we are using. A phase cancellation test uses that
>>>>>>fact to compare two audio files word for word (and hence bit for bit
>
>>>>>>since
>>>>>
>>>>>>each bit of a 24-bit word would
>>>>>>be at the same bit slot in each 24-bit word). Assuming they are
>>>>>>aligned
>>>>> to
>>>>>>the same start point, sample
>>>>>>accurately, and both are the same set of sample words at each sample
>>>>>>point,
>>>>>
>>>>>>bit for bit, and one is phase inverted,
>>>>>>they will cancel through all 24 bits. For two files to cancel
>>>>>>completely
>>>>>
>>>>>>for the duration of the file, each and every bit in each word
>>>>>>must be the exact opposite of that same bit position in a word at the
>>> same
>>>>>
>>>>>>sample point. This is why zooming in on an FFT
>>>>>>of the full difference file is valuable as it can show any differences
>>> in
>>>>>
>>>>>>the lower order bits that wouldn't be audible. So even if
>>>>>>there is no audible difference, the visual followup will show if the
> two
>>>>>
>>>>>>files truly cancel even a levels below hearing, or
>>>>>>outside of a frequency change that we will perceive.
>>>>>>
>>>>>>When they don't cancel, usually there will be way more than 1 bit
>>>>>>difference - it's usually one or more bits in the words for
>>>>>>thousands of samples. From a musical standpoint this is usually in
> a
>>>>>>frequency range (low freq, or high freq most often) - that will
>>>>>>show up as the difference between them, and that usually happens due
> to
>>>>> some
>>>>>>form of processing difference between the files,
>>>>>>such as EQ, compression, frequency dependant gain changes, etc. That
> is
>>>>> what
>>>>>>I believe you are thinking through, but when
>>>>>>talking about straight summing with no gain change (or known equal
>>>>>>gain
>>>>>
>>>>>>changes), we are only looking at linear, one for one
>>>>>>comparisons between the two files' frequency representations.
>>>>>>
>>>>>>Regards,
>>>>>>Dedric
>>>>>>
>>>>>>> Neil wrote:
>>>>>>>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>>>>>>> The tests I did were completely blank down to -200 dB (far below
> the
>>>>>
>>>>>>>>> last
>>>>>>>>
>>>>>>>>> bit). It's safe to say there is no difference, even in
>>>>>>>>> quantization noise, which by technical rights, is considered below
>>> the
>>>>>
>>>>>>>>> level
>>>>>>>>
>>>>>>>>> of "cancellation" in such tests.
>>>>>>>>
>>>>>>>> I'm not necessarily talking about just the first bit or the
>>>>>>>> last bit, but also everything in between... what happens on bit
>>>>>>>> #12, for example? Everything on bit #12 should be audible, but
>>>>>>>> in an a/b test what if thre are differences in what bits #8
>>>>>>>> through #12 sound like, but the amplutide is stll the same on
>>>>>>>> both files at that point, you'll get a null, right? Extrapolate
>>>>>>>> that out somewhat & let's say there are differences in bits #8
>>>>>>>> through #12 on sample points 3, 17, 1,000, 4,523, 7,560, etc,
>>>>>>>> etc through 43,972... Now this is breaking things down well
>>>>>>>> beyond what I think can be measured, if I'm not mistaken (I
>>>>>>>> dn't know of any way we could extract JUST that information
>>>>>>>> from each file & play it back for an a/b test; but would not
>>>>>>>> that be enough to have to "null-able" files that do actually
>>>>>>>> sound somewhat different?
>>>>>>>>
>>>>>>>> I guess what I'm saying is that since each sample in a musical
>>>>>>>> track or full song file doesn't represent a pure, simple set of
>>>>>>>> content like a sample of a sine wave would - there's a whole
>>>>>>>> world of harmonic structure in each sample of a song file, and
>>>>>>>> I think (although I'll admit - I can't "prove") that there is
>>>>>>>> plenty of room for some variables between the first bit & the
>>>>>>>> last bit while still allowing for a null test to be successful.
>>>>>>>>
>>>>>>>> No? Am I wacked out of my mind?
>>>>>>>>
>>>>>>>> Neil
>>>>>>>>
>>>>>>
>>>>>
>>>>
>>>>
>>>
>>
>>
>"James McCloskey" <excelsm@hotmail.com> wrote in message
news:458c31f7$1@linux...
>
> "j to the c r o n" <fu@hydrorecords.com> wrote:
>>wouldn't be so bad if it was actually funny
>>
>>PS. Why do people link to stuff they find offensive? :P
>
> I was curious what other people think. It doesn't matter if it's funny to
> an adult, my concern is the kids that saw this, it was on regular
> television.
> Where do we draw the line? Where are our standards?
>
> There are many different subjects that could offend many different kinds
> of people. It shouldn't matter if it's funny or not, it should matter if
> it is appropriate. The schmucks that run the entertainment business seem
> to want to twist and pervert everything. I know that this wasn't pointed
> directly at Christmas, but, can't they just leave Christmas alone?
>
> Nobody wants a moderator, or somebody to stomp on their freedom of speech,
> but what about decency standards? With out them a lot of people could be
> offended, including the Schmucks that run the entertainment business.
>
> Just my thoughts. Maybe not so appropriate for this NG.
Fair enough. Luckily SNL is on pretty late and they've been pushing the
limits for years. I remember maybe 15 years ago I checked it out and there
was some chick with her legs in the sky and some dude was simulating boning
her. Honestly I think if something crosses the line and its funny its more
appropriate than being offensive and not funny.
Have you noticed that mainstream movies have started to include explicit
sex? I'm afraid thats the direction things are headed. Even viagra
commercials are offensive... "4 hour erection" Just what I want to hear
when I'm watching football with my family on Thanksgiving.
I hate how everything is perverted but thats the way the pop culture works.
I've actually researched the subject... check out this book called Media
Virus "Hidden Agendas in Popular Culture" by Douglas Rushkoff (the same dude
that did the Frontline "The Persuaders" on PBS)Hi Tom,
Sorry you took the post too serious. I was just amused by the
complicated setup. I think you've been drinking from DJ bottle..:)
I think it would be cooler if you had separate monitors for everything.
It'd be like a spaceship.
Chris
Tom Bruhl wrote:
> Hi Chris,
> My dual KVM works great as is. Is the dual monitor idea so bad? How
> else will I be able to sit
> in my sweet spot and bounce between 2 and maybe even three comps with
> one mouse/keyboard?
>
> They make quad KVMs. I might end up with that in the end. 3 comps
> each with 4 head cards,
> four monitors, one keyboard/mouse.
> Why is the KVM soooooo Baaaad? How else can I achieve this?
>
> This other scenario would be even more complex than described in the
> diagram below.
> The ultimate setup would have two QWERTY keyboards/mice with KVM
> sharing with Paris/Cubase
> and Cubase independently.
>
> The room as it relates to the photo:
> My MIDI setup mimics what is shown to left of the
> mixing position here except there's a desk spot for legs on the right
> along
> with a controller. I'd like a 2 x 22" LCD widescreens, QWERTY & mouse
> there along with the KVM QWERTY/mouse in the center as shown for Paris
> and mixdown work.
>
> What would you suggest?
>
> Getting more confused by the minute...
> Tom
>
>
>
>
>
>
>
> "Chris Ludwig" <chrisl@adkproaudio.com
> <mailto:chrisl@adkproaudio.com>> wrote in message
> news:458c3c62$1@linux...
> Tom Bruhl GOOOODDDD.....KVM BBBBBAAAAADDDDDD
>
>
>
>
> Tom Bruhl wrote:
> > Okay Aaron I hope I get this right.
> >
> > The first diagram is what I think might not work due to the
> position of
> > the splitter. I'm afraid it won't amplify the signal if
> > placed at the end of the line. If I'm wrong then that's great!
> You'll
> > want to widen your view for this one.
> >
> > _Example 1:_
> >
> >
> >
> 21"
> > VGA monitor & 22" LCD with VGA input
> >
> 2
> > x 5' extensions
> >
> Video
> > Splitter (Aarons active type)
> >
> 20'
> > extension
> >
> KVM #1 out
> > KVM #1 In
> > VGA #1 out
> > Paris Comp - G550
> > VGA #2 out
> > KVM #2 In
> >
> KVM #2 out
> >
> 20'
> > extension
> >
> Video
> > Splitter (Aaron's active type)
> >
> 2
> > x 5' extensions
> >
>
> > 21" VGA monitor & 22" LCD with VGA input
> >
> >
> >
> > KVM
> #1 out
> > (Shares with above from this point on.)
> > KVM
> > #3 in
> > VGA #1 out
> > Cubase Comp - 7300GT
> > DVI out to VGA connector out
> > KVM #4 in
> > KVM
> #2 out
> > (Shares with above from this point on.)
> >
> >
> > OR:
> > _Example 2:_
> >
> >
> 21"
> > VGA monitor & 22" LCD with VGA input
> >
>
> > 2 x 5' extensions
> >
>
> > Passive Video Splitter (Y cable)
> >
>
> > 20' extension
> >
>
> > KVM #1 out
> >
> KVM
> > #1 In
> > Video amplifier
> (Like
> > splitter in example 1 with only one output)
> > VGA #1 out
> > Paris Comp - G550
> > VGA #2 out
> > Video amplifier
> (Like
> > splitter in example 1 with only one output)
> >
> > KVM #2 In
> >
>
> > KVM #2 out
> >
>
> > 20' extension
> >
>
> > Passive Video Splitter (Y cable)
> >
> 2
> > x 5' extensions
> >
>
> > 21"
> VGA
> > monitor & 22" LCD with VGA input
> >
> >
> >
> >
>
> > KVM #1 out (Shares with above from this point on.)
> >
> KVM
> > #3 in
> > Video amplifier (Like
> > splitter in example 1 with only one output)
> > VGA #1 out
> > Cubase Comp - 7300GT
> > DVI out to VGA connector out
> > Video amplifier (Like
> > splitter in example 1 with only one output)
> >
> KVM #4 in
> >
>
> > KVM #2 out (Shares with above from this point on.)
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> >
> > "Aaron Allen" <know-spam@not_here.dude
> <mailto:know-spam@not_here.dude>
> > <mailto:know-spam@not_here.dude>> wrote in message
> > news:458c050f@linux...
> > I'm not getting the visual you have in mind.. diagram it?
> >
> > AA
> >
> > "Tom Bruhl" <arpegio@comcast.net
> <mailto:arpegio@comcast.net> <mailto:arpegio@comcast.net>>
> > wrote in message news:458b7f6f@linux...
> > Thanks Aaron,
> > Will this work on the backside of a KVM? I can't put it
> in front.
> > I guess if it needed to be in front of the KVM I could
> just get a
> > single output jobbie and split it at the other side. Do
> those
> > exist?
> >
> >
> > "Aaron Allen" <know-spam@not_here.dude
> <mailto:know-spam@not_here.dude>
> > <mailto:know-spam@not_here.dude>> wrote in message
> > news:458b23a2@linux...
> > I hope you're not doing a passive Y split. Look for
> a VGA
> > splitter on google, something like
> >
> >
> http://www.pcmall.com/pcmall/shop/detail~dpno~767745.asp
> < http://www.pcmall.com/pcmall/shop/detail%7Edpno%7E767745.asp>
> >
> > They're cheap and OH so worth it
> >
> > AA
> >
> > "Tom Bruhl" <arpegio@comcast.net
> <mailto:arpegio@comcast.net>
> > <mailto:arpegio@comcast.net>> wrote in message
> > news:4589ba0a$1@linux...
> > Thanks Chris.
> > This graphics stuff is confusing to me.
> >
> > I have to convert the DVI to VGA because of the
> 20' length
> > as I understand. I was going to split it with a
> Y cable
> > too.
> > They'll split to a 21"VGA and a 22"LCD. Will
> that work on
> > the far side of a KVM? I know the resolutions
> won't match
> > but I'll only need the VGAs when mixing so I'll
> adjust the
> > 7300 differently for that application. Sooner
> or later
> > I'll get
> > 4 matching 22" LCDs and quad cards for both comps.
> >
> > Will this work?
> > Tom
> >
> >
> >
> >
> >
> >
> >
> > "Chris Ludwig" <chrisl@adkproaudio.com
> <mailto:chrisl@adkproaudio.com>
> > <mailto:chrisl@adkproaudio.com>> wrote in
> message
> > news:45894ef0$1@linux...
> > HI Tom,
> > These monitors will be fine with your card. The
> > 7300GT supports that
> > resolution on both the VGA and DVI connections.I
> > would run both with VGA
> > connections. Having one hooked up DVI and
> one VGA
> > may make the 2 look
> > very different in brightness and such.
> > Chris
> >
> >
> > Tom Bruhl wrote:
> >
> > > My card has one VGA, one DVI and one S Video
> > connector on it.
> > >
> > >
> > > I choose Polesoft Lockspam to fight spam,
> and you?
> > > http://www.polesoft.com/refer.html
> >
> >
> > --
> > Chris Ludwig
> > ADK
> > chrisl@adkproaudio.com
> <mailto:chrisl@adkproaudio.com>
> > <mailto:chrisl@adkproaudio.com>
> > <mailto:chrisl@adkproaudio.com>
> > www.adkproaudio.com
> <http://www.adkproaudio.com> <http://www.adkproaudio.com>
> > <http://www.adkproaudio.com/>
> > (859) 635-5762
>
> --
> Chris Ludwig
>
> ADK Pro Audio
> (859) 635-5762
> www.adkproaudio.com <http://www.adkproaudio.com>
> chrisl@adkproaudio.com <mailto:chrisl@adkproaudio.com>
>
>
> ------------------------------------------------------------ ------------
>
--
Chris Ludwig
ADK
chrisl@adkproaudio.com <mailto:chrisl@adkproaudio.com>
www.adkproaudio.com <http://www.adkproaudio.com/>
(859) 635-5762Dedric, check out this post from our dear friend Fredo: Neundo Moderator:
Explaining how Steingberg's audio engine works. Note the trade-offs..Meaning,
Steinberg's way of coding an audio-engine 32bit float is different than say
Magix Samplitude:
Fredo
Administrative Moderator
Joined: 29 Dec 2004
Posts: 4213
Location: Belgium
Posted: Fri Dec 08, 2006 2:33 pm Post subject:
------------------------------------------------------------ --------------------
I think I see where the problem is.
In my scenario's I don't have any track that goes over 0dBfs, but I have
always lowered one channel to compensate with another.
So, I never whent over the 0dB fs limit.
Here's the explanation:
As soon as you go over 0dB, technically you are entering the domain of distortion.
In a 32bit FP mixer, that is not the case since there is unlimited headroom.
Now follow me step by step please - read this slow and make sure you understand
-
At the end of each "stage", there is an adder (a big calculator) which adds
all the numbers from the individual tracks that are routed to this "adder".
The numbers are kept in the 80-bit registers and then brought back to 32bit
float.
This process of bringing back the numbers from 80-bit (and more) to 32bit
is kept to an absolute minimum.
This adding/bringing back to 32bit is done at 3 places: After a plugin slot
(VST-specs for all plugin manufacturers) - Group Tracks and Master Tracks.
Now, as soon as you boost the volume above 0dB, you get more than 32bits.
Stay below 0dB and you will stay below 32 bits.
When the adders dump their results, the numbers are brought back from any
number of bits (say 60bit) to 32 bit float.
These numbers are simply truncated which results in distortion; that's the
noise/residue you find way down low.
There is an algortithm that protects us from additive errors - so these errors
can never come into the audible range.
So, as soon as you go over 0dB, you will see these kind of artifacts.
It is debatable if this needs to be dithered or not. The problem -still is-
that it is very difficult to dither in a Floating Point environment.
Fact remains that the error shouldn't be bigger than 2 to 3 LSB's.
Is this a problem?
In real world applictations: NO.
In scientific -unrealistic- tests (forcing the erro ): YES.
The alternative is having a Fixed point mixer, where you already would be
in trouble as soon as you boost one channel over 0dBfs. (or merge two files
that are @ 0dB)
Also, this problem will be pretty much gone as soon as we switch to the 64
bit engine.
For the record, the test where Jake hears "music" as residue must be flawed.
You should hear noise/distortion from square waves.
HTH
Fredo
"Dedric Terry" <dedric@echomg.com> wrote:
>I can't tell you why you hear ProTools differently than Nuendo using a
>single file.
>There isn't any voodoo in the software, or hidden character enhancing dsp.
>I'll see if
>I can round up an M-Powered system to compare with next month.
>
>For reference, everytime I open Sequoia I think I might hear a broader,
>clean,
>and almost flat (spectrum, not depth) sound, but I don't - it's the same
as
>Nuendo, fwiw.
>Also I don't think what I was referring to was a theory from Chuck - I
>believe that was what he
>discovered in the code.
>
>Digital mixers all have different preamps and converters. Unless you are
>bypassing every
>EQ and converter and going digital in and out to the same converter when
>comparing, it would be hard
>to say the mix engine itself sounds different than another mixer, but taken
>as a whole, then
>certainly they may very well sound different. In addition, hardware digital
>mixers may use a variety of different paths between the I/O, channel
>processing, and summing,
>though most are pretty much software mixers on a single chip or set of dsps
>similar to ProTools,
>with I/O and a hardware surface attached.
>
>I know it may be hard to separate the mix engine as software in either a
>native DAW
>or a digital mixer, from the hardware that translates the audio to something
>we hear,
>but that's what is required when comparing summing. The hardware can
>significantly change
>what we hear, so comparing digital mixers really isn't of as much interest
>as comparing native
>DAWs in that respect - unless you are looking to buy one of course.
>
>Even though I know you think manufacturers are trying to add something to
>give them an edge, I am 100%
>sure that isn't the case - rather they are trying to add or change as little
>as possible in order to give
>them the edge. Their end of digital audio isn't about recreating the past,
>but improving upon it.
>As we've discussed and agreed before, the obsession with recreating
>"vintage" technology is as much
>fad as it is a valuable creative asset. There is no reason we shouldn't
>have far superior hardware and software EQs and comps
>than 20, 30 or 40 years ago. No reason at all, other than market demand,
>but the majority of software, and new
>hardware gear on the market has a vintage marketing tagline with it.
>Companies will sell any bill of
>goods if customers will buy it.
>
>There's nothing unique about the summing in Nuendo, Cubase, Sequoia/Samp,
>or Sonar, and it's pretty safe to include Logic and DP in that list as well.
>One of the reasons I test
>these things is to be sure my DAW isn't doing something wrong, or something
>I don't know about.
>
>Vegas - I use it for video conversions and have never done any critical
>listening tests with it. What I have heard
>briefly didn't sound any different. It certainly looks plain vanilla
>though. What you are describing is exactly
>what I would say about the GUIs of each of those apps, not that it means
>anything. Just interesting.
>
>That's one reason I listen eyes closed and double check with phase
>cancellation tests and FFTs - I am
>influenced creatively by the GUI to some degree. I actually like Cubase
4's
>GUI better than Nuendo 3.2,
>though there are only slight visual differences (some workflow differences
>are a definite improvement for me though).
>
>ProTools' GUI always made me want to write one dimensional soundtracks in
>mono for public utilities, accounting offices
>or the IRS while reading my discreet systems analysis textbook - it was
also
>grey. ;-)
>
>Regards,
>Dedric
>
>"LaMont" <jjdpro@ameritech.net> wrote in message news:458c82fd$1@linux...
>>
>> Dedric, my simple test is simple..
>> Using the same audio interface, with the same stereo file..null-ed to
>> zero..No
>> eq, for fx. Master fader on zero..
>>
>> Nuendo, Pro-Tools -Mpowered(native)... yields a sonic difference that
I
>> have
>> referenced before.. The sound coming from PT-M has a nice top end , where
>> as Neundo has a nice flatter sound quality.
>> Same audio interface. M-audio 410..Using Mackies & Blue-Sky pro monitors..
>>
>> Same test at the big room..PT-HD & Neundo Logic Audio(macG5-Dual) Using
>> the
>> 192 interface.
>> Same results..But adding Logic audio's sound ..(Broad, thick)
>>
>> Somethings going on.
>>
>> Chucks post about how paris handles audio is a theory..Only Edmund can
>> truly
>> give us the goods on what's really what..
>>
>> I disagree that manufactuers don;t set out o put a sonic print on their
>> products.
>> I think they do.
>>
>> I have been fortunate to work on some digital mixers and I can tell you
>> that
>> each one has their own sound. The Sony Dmx-100 was modeled after SSL 4000g
>> (like it's Big Brother).And you what? That board (Dmx-100) sound very
warm
>> and it's eq tries to behave and sound just like an SSL.. Unlike he Yamaha
>> Dm2000(version 1.x) which has a very Clean, neutral sound..However, some
>> complained that it was tooo Vanila and thus, Yamaha add a version 2.0
>> which
>> added Vintage type Eq's, modeled analog input gain saturation fx too give
>> the user a choice Btw Clean and Neutral vs sonic Character.
>>
>> So, if digital conoles can be given a sonic character, why not a software
>> mixer?
>> The truth is, there are some folks who want a neutral mixer and then there
>> are others who want a sonic footprint imparted. and these can be coded
in
>> the digital realm.
>> The apllies with the manufactuers. They too have their vision on what
They
>> think and want their product to sound.
>>
>> I love reading on gearslutz the posts from Plugin developers and their
>> interpretations
>> and opinions about what makes their Neve 1073 Eq better and what goes
into
>> making their version sound like it does.. Each Developer has a different
>> vision as to what the Neve 1073 should sound like. And yet they all sound
>> good , but slightly different.
>>
>> You stated that you use Vegas. Well as you know, Vegas has a very generic
>> sound..Just plain and simple. But, i bet you can tell the difference
on
>> your system when you play that same file in Neundo (No, fx, eq,
>> null-edzerro)..
>> ???
>>
>>
>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>Lamont - what is the output chain you are using for each app when
>>>comparing
>>
>>>the file in Nuendo
>>>vs ProTools? On the same PC, I presume (and is this PT HD or M-Powered?)?
>>>Since these can't use the same output driver, you would have to depend
on
>>
>>>the D/A being
>>>the same, but clocking will be different unless you have a master clock,
>> and
>>>both interfaces
>>>are locking with the same accuracy. This was one of the issues that came
>> up
>>>at Lynn Fuston's
>>>D/A converter shootout - when do you lock to external clock and incur
the
>>
>>>resulting jitter,
>>>and when do you trust the internal clock - and if you do lock externally,
>>
>>>how good is the PLL
>>>in the slave device? These issues can cause audible changes in the top
>> end
>>>that have nothing to do
>>>with the software itself. If you say that PTHD through the same converter
>>
>>>output as Nuendo (via? RME?
>>>Lynx?) using the same master clock, sounds different playing a single
>>>audio
>>
>>>file, then I take your word
>>>for it. I can't tell you why that is happening - only that an audible
>>>difference really shouldn't happen due
>>>to the software alone - not with a single audio file, esp. since I've
>>>heard
>>
>>>and seen PTHD audio cancel with
>>>native DAWs. Just passing a single 16 or 24 bit track down the buss
to
>> the
>>>output driver should
>>>be, and usually is, completely transparent, bit for bit.
>>>
>>>The same audio file played through the same converters should only sound
>>
>>>different if something in
>>>the chain is different - be it clocking, gain or some degree of
>>>unintended,
>>
>>>errant dsp processing. Every DAW should
>>>pass a single audio file without altering a single bit. That's a basic
>>>level
>>
>>>of accuracy we should always
>>>expect of any DAW. If that accuracy isn't there, you can be sure a heavy
>>
>>>mix will be altered in ways you
>>>didn't intend, even though you would end up mixing with that factor in
>>>place
>>
>>>(e.g. you still mix for what
>>>you want to hear regardless of what the platform does to each audio track
>> or
>>>channel).
>>>
>>>In fact you should be able to send a stereo audio track out SPDIF or
>>>lightpipe to another DAW, record it
>>>bring the recorded file back in, line them up to the first bit, and have
>>
>>>them cancel on and inverted phase
>>>test. I did this with Nuendo and Cubase 4 on separate machines just to
>> be
>>>sure my master clocking and
>>>slave sync was accurate - it worked perfectly.
>>>
>>>Also be sure there isn't a variation in the gain even by 0.1 dB between
>> the
>>>two. There shouldn't
>>>and I wouldn't expect there to be one. Also could PT be set for a
>>>different
>>
>>>pan law? Shouldn't make a
>>>difference even if comparing two mono panned files to their stereo
>>>interleaved equivalent, but for sake
>>>of completeness it's worth checking as well. A variation in the output
>>
>>>chain, be it drivers, audio card
>>>card, or converters would be the most likely culprit here.
>>>
>>>The reason DAW manufacturers wouldn't add any sonic "character"
>>>intentionally is that the
>>>ultimate goal from day one with recording has been to accurately reproduce
>>
>>>what we hear.
>>>We developed a musical penchant for sonic character because the hardware
>>
>>>just wasn't accurate,
>>>and what it did often sent us down new creative paths - even if by force
>> -
>>>and we decided it was
>>>preferred that way.
>>>
>>>Your point about what goes into the feature presets to sell synths is
>>>right
>>
>>>for sure, but synths are about
>>>character and getting that "perfect piano" or crystal clear bell pad,
or
>> fat
>>>punchy bass without spending
>>>a mint on development, adding 50G onboard sample libraries, or costing
>>>$15k,
>>
>>>so what they
>>>lack in actual synthesis capabilities, they make up with EQ and effects
>> on
>>>the output. That's been the case
>>>for years, at least since we had effects on synths at least. But even
>>>with
>>
>>>modern synths such as the Fantom,
>>>Tritons, etc, which are great synths all around, of course the coolest,
>>
>>>widest and biggest patches
>>>will make the biggest impression - so in come the EQs, limiters, comps,
>>
>>>reverbs, chorus, etc. The best
>>>way to find out if a synth is really good is to bypass all effects and
see
>>
>>>what happens. Most are pretty
>>>good these days, but about half the time, there are presets that fall
>>>completely flat in fx bypass.
>>>
>>>DAWs aren't designed to put a sonic fingerprint on a sound the way synths
>>
>>>are - they are designed
>>>to *not* add anything - to pass through what we create as users, with
no
>>
>>>alteration (or as little as possible)
>>>beyond what we add with intentional processing (EQ, comps, etc).
>>>Developers
>>
>>>would find no pride
>>>in hearing that their DAW sounds anything different than whatever is being
>>
>>>played back in it,
>>>and the concept is contrary to what AES and IEEE proceedings on the issue
>>
>>>propose in general
>>>digital audio discussions, white papers, etc.
>>>
>>>What ID ended up doing with Paris (at least from what I gather per Chuck's
>>
>>>findings - so correct me if I'm missing part of the equation Chuck),
>>>is drop the track gain by 20dB or so, then added it back at the master
>>>buss
>>
>>>to create the effect of headroom (probably
>>>because the master buss is really summing on the card, and they have more
>>
>>>headroom there than on the tracks
>>>where native plugins might be used). I don't know if Paris passed 32-bit
>>
>>>float files to the EDS card, but sort of
>>>doubt it. I think Chuck has clarified this at one point, but don't recall
>>
>>>the answer.
>>>
>>>Also what Paris did is use a greater bit depth on the hardware than
>>>ProTools
>>
>>>did - at the time PT was just
>>>bring Mix+ systems to market, or they had been out for a year or two (if
>> I
>>>have my timeline right) - they
>>>were 24-bit fixed all the way through. Logic and Cubase were native DAWs,
>>
>>>but native was still too slow
>>>to compete with hardware hybrids. Paris trumped them all by running
>>>32-bit
>>
>>>float natively (not new really, but
>>>better than sticking to 24-bit) and 56 or so bits in hardware instead
of
>>
>>>going to Motorola DSPs at 24.
>>>The onboard effects were also a step up from anything out there, so the
>> demo
>>>did sound good.
>>>I don't recall which, but one of the demos, imho, wasn't so good (some
>>>sloppy production and
>>>vocals in spots, IIRC), so I only listened to it once. ;-)
>>>
>>>Coupled with the gain drop and buss makeup, this all gave it a "headroom"
>> no
>>>one else had. With very nice
>>>onboard effects, Paris jumped ahead of anything else out there easily,
and
>>
>>>still respectably holds its' own today
>>>in that department.
>>>
>>>Most demos I hear (when I listen to them) vary in quality, usually not
so
>>
>>>great in some area. But if a demo does
>>>sound great, then it at least says that the product is capable of at
>>>least
>>
>>>that level of performance, and it can
>>>only help improve a prospective buyer's impression of it.
>>>
>>>Regards,
>>>Dedric
>>>
>>>"LaMont " <jjdpro@ameritech.net> wrote in message news:458c14c0$1@linux...
>>>>
>>>> Dedric good post..
>>>>
>>>> However, I have PT-M-Powered/M-audio 410 interface for my laptop and
it
>>
>>>> has
>>>> that same sound (no eq, zero fader) that HD does. I know their use the
>>
>>>> same
>>>> 48 bit fix mixer. I load up the same file in Nuendo (no eq, zero
>>>> fader)..results.
>>>> different sonic character.
>>>>
>>>> PT having a top end touch..Nuendo, nice smooth(flat) sound. And I'm
just
>>>> taking about a stereo wav file nulled with no eq..nothing
>>>> ..zilch..nada..
>>>>
>>>> Now, there are devices (keyboards, dum machines) on the market today
>>>> that
>>>> have a Master Buss Compressor and EQ set to on with the top end notched
>>
>>>> up.
>>>> Why? because it gives their product an competitive advantageover the
>>>> competition..
>>>> Ex: Yahama's Motif ES, Akai's MPC 1000, 2500, Roland's Fantom.
>>>>
>>>> So, why would'nt a DAW manufactuer code in an extra (ooommf) to make
>>>> their
>>>> DAW sound better. Especially, given the "I hate Digtal Summing" crowd?
>>
>>>> And,
>>>> If I'm a DAW manufactuer, what would give my product a sonic edge over
>> the
>>>> competition?
>>>>
>>>> We live in the "louder is better" audio world these days, so a DAW that
>>
>>>> can
>>>> catch my attention 'sonically" will probaly will get the sell. That's
>> what
>>>> happend to me back in 1997 when I heard Paris. I was floored!!! Still
>> to
>>>> this day, nothing has floored me like that "Road House Blues Demo" I
>>>> heard
>>>> on Paris.
>>>>
>>>> Was it the hardware ? was it the software. I remember talking with
>>>> Edmund
>>>> at the 2000 winter Namm, and told me that he & Steve set out to
>>>> reproduce
>>>> the sonics of big buck analog board (eq's) and all.. And, summing was
>> a
>>>> big
>>>> big issue for them because they (ID) thought that nobody has gotten
>>>> it(summing)
>>>> right. And by right, they meant, behaved like a console with a wide
lane
>>>> for all of those tracks..
>>>>
>>>>
>>>>
>>>>
>>>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>>>"LaMont" <jjdpro@ameritech.net> wrote in message
>>>>>news:458be8d5$1@linux...
>>>>>>
>>>>>> Okay...
>>>>>> I guess what I'm saying is this:
>>>>>>
>>>>>> -Is it possible that diferent DAW manufactuers "code" their app
>>>>>> differently
>>>>>> for sound results.
>>>>>
>>>>>Of course it is *possible* to do this, but only if the DAW has a
>>>>>specific
>>>>
>>>>>sound shaping purpose
>>>>>beyond normal summing/mixing. Users talk about wanting developers to
>> add
>>>> a
>>>>>"Neve sound" or "API sound" option to summing engines,
>>>>>but that's really impractical given the amount of dsp required to make
>> a
>>>>
>>>>>decent emulation (with convolution, dynamic EQ functions,
>>>>>etc). For sake of not eating up all cpu processing, that could likely
>>
>>>>>only
>>>>
>>>>>surface as is a built in EQ, which
>>>>>no one wants universally in summing, and anyone can add at will already.
>>>>>
>>>>>So it hasn't happened yet and isn't likely to as it detours from the
>>>>>basic
>>>>
>>>>>tenant of audio recording - recreate what comes in as
>>>>>accurately as possible.
>>>>>
>>>>>What Digi did in recoding their summing engine was try to recover some
>>>>>of the damage done by the 24-bit buss in Mix systems. Motorola 56k dsps
>>>> are
>>>>>24-bit fixed point chips and I think
>>>>>the new generation (321?) still is, but they use double words now for
>>>>>48-bits). And though plugins could process at 48-bit by
>>>>>doubling up and using upper and lower 24-bit words for 48-bit outputs,
>> the
>>>>
>>>>>buss
>>>>>between chips was 24-bits, so they had to dither to 24-bits after every
>>>>
>>>>>plugin. The mixer (if I recall correctly) also
>>>>>had a 24-bit buss, so what Digi did is to add a dither stage to the
>>>>>mixer
>>>> to
>>>>>prevent this
>>>>>constant truncation of data. 24-bits isn't enough to cover summing
for
>>>> more
>>>>>than a few tracks without
>>>>>losing information in the 16-bit world, and in the 24-bit world some
>>>>>information will be lost, at least at the lowest levels.
>>>>>
>>>>>Adding a dither stage (though I think they did more than that - perhaps
>>>>
>>>>>implement a 48-bit double word stage as well),
>>>>>simply smoothed over the truncation that was happening, but it didn't
>>
>>>>>solve
>>>>
>>>>>the problem, so with HD
>>>>>they went to a double-word path - throughout I believe, including the
>> path
>>>>
>>>>>between chips. I believe the chips
>>>>>are still 24-bit, but by doubling up the processing (yes at a cost of
>>
>>>>>twice
>>>>
>>>>>the overhead), they get a 48-bit engine.
>>>>>This not only provided better headroom, but greater resolution. Higher
>>>> bit
>>>>>depths subdivide the amplitude with greater resolution, and that's
>>>>>really where we get the definition of dynamic range - by lowering the
>>
>>>>>signal
>>>>
>>>>>to quantization noise ratio.
>>>>>
>>>>>With DAWs that use 32-bit floating point math all the way through, the
>>
>>>>>only
>>>>
>>>>>reason for altering the summing
>>>>>is by error, and that's an error that would actually be hard to make
and
>>>> get
>>>>>past a very basic alpha stage of testing.
>>>>>There is a small difference in fixed point math and floating point math,
>>>> or
>>>>>at least a theoretical difference in how it affects audio
>>>>>in certain cases, but not necessarily in the result for calculating
gain
>>>> in
>>>>>either for the same audio file. Where any differences might show up
is
>>>>
>>>>>complicated, and I believe only appear at levels below 24-bit (or in
>>>>>headroom with tracks pushed beyond 0dBFS), or when/if
>>>>>there areany differences in where each amplitude level is quantized.
>>>>>
>>>>>Obviously there can be differences if the DAW has to use varying bit
>>>>>depths
>>>>
>>>>>throughout a single summing path to accomodate hardware
>>>>>as well as software summing, since there may be truncation or rounding
>>
>>>>>along
>>>>
>>>>>the way, but that impacts the lowest bit
>>>>>level, and hence - spacial reproduction, reverb tails perhaps, and
>>>>>"depth",
>>>>
>>>>>not the levels most music so the differences are most
>>>>>often more subtle than not. But most modern DAWs have eliminated those
>>>>
>>>>>"rough edges" in the math by increasing the bit depth to accomodate
>>>>>normal
>>>>
>>>>>summing required for mixing audio.
>>>>>
>>>>>So with Lynn's unity gain summing test (A files on the CD I believe),
>> DAWs
>>>>
>>>>>were never asked to sum beyond 24-bits,
>>>>>at least not on the upper end of the dynamic range, so everything that
>>
>>>>>could
>>>>
>>>>>represent 24-bits accurately would cancel. The only ones
>>>>>that didn't were ones that had a different bit depth and/or gain
>>>>>structure
>>>>
>>>>>whether hybrid or native
>>>>>(e.g. Paris' subtracting 20dB from tracks and adding it to the buss).
>> In
>>>>
>>>>>this case, PTHD cancelled (when I tested it) with
>>>>>Nuendo, Samplitude, Logic, etc because the impact of the 48-bit fixed
>> vs.
>>>>
>>>>>32-bit float wasn't a factor.
>>>>>
>>>>>When trying other tests, even when adding and subtracting gain, Nuendo,
>>>>
>>>>>Sequoia and Sonar cancel - both audibly and
>>>>>visually at inaudible levels, which only proves that one isn't making
>> an
>>>>
>>>>>error when calculating basic gain. Since a dB is well defined,
>>>>>and the math to add gain is simple, they shouldn't. The fact that they
>>>> all
>>>>>use 32-bit float all the way through eliminates a difference
>>>>>in data structure as well, and this just verifies that. There was a
>>>>>time
>>>>
>>>>>that supposedly Logic (v3, v4?) was partly 24-bit, or so the rumor went,
>>>>>but it's 32-bit float all the way through now just as Sonar,
>>>>>Nuendo/Cubase,
>>>>
>>>>>Samplitude/Sequoia, DP, Audition (I presume at least).
>>>>>I don't know what Acid or Live use. Saw promotes a fixed point engine,
>>>> but
>>>>>I don't know if it is still 24-bit, or now 48 bit.
>>>>>That was an intentional choice by the developer, but he's the only one
>> I
>>>>
>>>>>know of that stuck with 24-bit for summing
>>>>>intentionally, esp. after the Digi Mix system mixer incident.
>>>>>
>>>>>Long answer, but to sum up, it is certainly physically *possible* for
>> a
>>>>
>>>>>developer to code something differently intentionally, but not
>>>>>in reality likely since it would be breaking some basic fixed point
or
>>>>>floating point math rules. Where the differences really
>>>>>showed up in the past is with PT Mix systems where the limitation was
>>
>>>>>really
>>>>
>>>>>significant - e.g. 24 bit with truncation at several stages.
>>>>>
>>>>>That really isn't such an issue anymore. Given the differences in
>>>>>workflow,
>>>>
>>>>>missing something in workflow or layout differences
>>>>>is easy enough to do (e.g. Sonar doesn't have group and busses the way
>>>>>Nuendo does, as it's outputs are actually driver outputs,
>>>>>not software busses, so in Sonar, busses are actually outputs, and sub
>>>>>busses are actually busses in Nuendo. There are no,
>>>>>or at least I haven't found the equivalent of a Nuendo group in Sonar
>> -
>>>> that
>>>>>affects the results of some tests (though not basic
>>>>>summing) if not taken into account, but when taken into account, they
>> work
>>>>
>>>>>exactly the same way).
>>>>>
>>>>>So at least when talking about apps with 32-bit float all the way
>>>>>through,
>>>>
>>>>>it's safe to say (since it has been proven) that summing isn't different
>>>>
>>>>>unless
>>>>>there is an error somewhere, or variation in how the user duplicates
the
>>>>
>>>>>same mix in two different apps.
>>>>>
>>>>>Imho, that's actually a very good thing - approaching a more consistent
>>>>
>>>>>basis for recording and mixing from which users can make all
>>>>>of the decisions as to how the final product will sound and not be
>>>>>required
>>>>
>>>>>to decide when purchasing a pricey console, and have to
>>>>>focus their business on clients who want "that sound". I believe we
are
>>>>
>>>>>actually closer to the pure definition of recording now than
>>>>>we once were.
>>>>>
>>>>>Regards,
>>>>>Dedric
>>>>>
>>>>>
>>>>>>
>>>>>> I the answer is yes, then,the real task is to discover or rather
>>>>>> un-cover
>>>>>> what's say: Motu's vision of summing, versus Digidesign, versus
>>>>>> Steinberg
>>>>>> and so on..
>>>>>>
>>>>>> What's under the hood. To me and others,when Digi re-coded their
>>>>>> summing
>>>>>> engine, it was obvious that Pro Tools has an obvious top end (8k-10k)
>>>>
>>>>>> bump.
>>>>>> Where as Steinberg's summing is very neutral.
>>>>>>
>>>>>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>>>>>Hi Neil,
>>>>>>>
>>>>>>>Jamie is right. And you aren't wacked out - you are thinking this
>>>>>>>through
>>>>>>
>>>>>>>in a reasonable manner, but coming to the wrong
>>>>>>>conclusion - easy to do given how confusing digital audio can be.
>>>>>>>Each
>>>>>> word
>>>>>>>represents an amplitude
>>>>>>>point on a single curve that is changing over time, and can vary with
>>>> a
>>>>>>
>>>>>>>speed up to the Nyquist frequency (as Jamie described).
>>>>>>>The complex harmonic content we hear is actually the frequency
>>>>>>>modulation
>>>>>> of
>>>>>>>a single waveform,
>>>>>>>that over a small amount of time creates the sound we translate -
we
>>
>>>>>>>don't
>>>>>>
>>>>>>>really hear a single sample at a time,
>>>>>>>but thousands of samples at a time (1 sample alone could at most
>>>>>>>represent
>>>>>> a
>>>>>>>single positive or negative peak
>>>>>>>of a 22,050Hz waveform).
>>>>>>>
>>>>>>>If one bit doesn't cancel, esp. if it's a higher order bit than number
>>>> 24,
>>>>>>
>>>>>>>you may hear, and will see that easily,
>>>>>>>and the higher the bit in the dynamic range (higher order) the more
>>>>>>>audible
>>>>>>
>>>>>>>the difference.
>>>>>>>Since each bit is 6dB of dynamic range, you can extrapolate how "loud"
>>>>
>>>>>>>that
>>>>>>
>>>>>>>bit's impact will be
>>>>>>>if there is a variation.
>>>>>>>
>>>>>>>Now, obviously if we are talking about 1 sample in a 44.1k rate song,
>>>> then
>>>>>>
>>>>>>>it simply be a
>>>>>>>click (only audible if it's a high enough order bit) instead of an
>>>>>>>obvious
>>>>>>
>>>>>>>musical difference, but that should never
>>>>>>>happen in a phase cancellation test between identical files higher
>>>>>>>than
>>>>>> bit
>>>>>>>24, unless there are clock sync problems,
>>>>>>>driver issues, or the DAW is an early alpha version. :-)
>>>>>>>
>>>>>>>By definition of what DAWs do during playback and record, every audio
>>>>
>>>>>>>stream
>>>>>>
>>>>>>>has the same point in time (judged by the timeline)
>>>>>>>played back sample accurately, one word at a time, at whatever sample
>>>>
>>>>>>>rate
>>>>>>
>>>>>>>we are using. A phase cancellation test uses that
>>>>>>>fact to compare two audio files word for word (and hence bit for bit
>>
>>>>>>>since
>>>>>>
>>>>>>>each bit of a 24-bit word would
>>>>>>>be at the same bit slot in each 24-bit word). Assuming they are
>>>>>>>aligned
>>>>>> to
>>>>>>>the same start point, sample
>>>>>>>accurately, and both are the same set of sample words at each sample
>>>>>>>point,
>>>>>>
>>>>>>>bit for bit, and one is phase inverted,
>>>>>>>they will cancel through all 24 bits. For two files to cancel
>>>>>>>completely
>>>>>>
>>>>>>>for the duration of the file, each and every bit in each word
>>>>>>>must be the exact opposite of that same bit position in a word at
the
>>>> same
>>>>>>
>>>>>>>sample point. This is why zooming in on an FFT
>>>>>>>of the full difference file is valuable as it can show any differences
>>>> in
>>>>>>
>>>>>>>the lower order bits that wouldn't be audible. So even if
>>>>>>>there is no audible difference, the visual followup will show if the
>> two
>>>>>>
>>>>>>>files truly cancel even a levels below hearing, or
>>>>>>>outside of a frequency change that we will perceive.
>>>>>>>
>>>>>>>When they don't cancel, usually there will be way more than 1 bit
>>>>>>>difference - it's usually one or more bits in the words for
>>>>>>>thousands of samples. From a musical standpoint this is usually in
>> a
>>>>>>>frequency range (low freq, or high freq most often) - that will
>>>>>>>show up as the difference between them, and that usually happens due
>> to
>>>>>> some
>>>>>>>form of processing difference between the files,
>>>>>>>such as EQ, compression, frequency dependant gain changes, etc. That
>> is
>>>>>> what
>>>>>>>I believe you are thinking through, but when
>>>>>>>talking about straight summing with no gain change (or known equal
>>>>>>>gain
>>>>>>
>>>>>>>changes), we are only looking at linear, one for one
>>>>>>>comparisons between the two files' frequency representations.
>>>>>>>
>>>>>>>Regards,
>>>>>>>Dedric
>>>>>>>
>>>>>>>> Neil wrote:
>>>>>>>>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>>>>>>>> The tests I did were completely blank down to -200 dB (far below
>> the
>>>>>>
>>>>>>>>>> last
>>>>>>>>>
>>>>>>>>>> bit). It's safe to say there is no difference, even in
>>>>>>>>>> quantization noise, which by technical rights, is considered below
>>>> the
>>>>>>
>>>>>>>>>> level
>>>>>>>>>
>>>>>>>>>> of "cancellation" in such tests.
>>>>>>>>>
>>>>>>>>> I'm not necessarily talking about just the first bit or the
>>>>>>>>> last bit, but also everything in between... what happens on bit
>>>>>>>>> #12, for example? Everything on bit #12 should be audible, but
>>>>>>>>> in an a/b test what if thre are differences in what bits #8
>>>>>>>>> through #12 sound like, but the amplutide is stll the same on
>>>>>>>>> both files at that point, you'll get a null, right? Extrapolate
>>>>>>>>> that out somewhat & let's say there are differences in bits #8
>>>>>>>>> through #12 on sample points 3, 17, 1,000, 4,523, 7,560, etc,
>>>>>>>>> etc through 43,972... Now this is breaking things down well
>>>>>>>>> beyond what I think can be measured, if I'm not mistaken (I
>>>>>>>>> dn't know of any way we could extract JUST that information
>>>>>>>>> from each file & play it back for an a/b test; but would not
>>>>>>>>> that be enough to have to "null-able" files that do actually
>>>>>>>>> sound somewhat different?
>>>>>>>>>
>>>>>>>>> I guess what I'm saying is that since each sample in a musical
>>>>>>>>> track or full song file doesn't represent a pure, simple set of
>>>>>>>>> content like a sample of a sine wave would - there's a whole
>>>>>>>>> world of harmonic structure in each sample of a song file, and
>>>>>>>>> I think (although I'll admit - I can't "prove") that there is
>>>>>>>>> plenty of room for some variables between the first bit & the
>>>>>>>>> last bit while still allowing for a null test to be successful.
>>>>>>>>>
>>>>>>>>> No? Am I wacked out of my mind?
>>>>>>>>>
>>>>>>>>> Neil
>>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>
>>>
>>>
>>
>
>This is a multi-part message in MIME format.
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charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
So I gave Chris a little Christmas chuckle did I?
I'm glad I could be of service!
"Chris Ludwig" <chrisl@adkproaudio.com> wrote in message =
news:458cc07c$1@linux...
Hi Tom,
Sorry you took the post too serious. I was just amused by the=20
complicated setup. I think you've been drinking from DJ bottle..:)
I think it would be cooler if you had separate monitors for =
everything.=20
It'd be like a spaceship.
Chris
Tom Bruhl wrote:
> Hi Chris,
> My dual KVM works great as is. Is the dual monitor idea so bad? =
How=20
> else will I be able to sit
> in my sweet spot and bounce between 2 and maybe even three comps =
with=20
> one mouse/keyboard?
> =20
> They make quad KVMs. I might end up with that in the end. 3 comps=20
> each with 4 head cards,
> four monitors, one keyboard/mouse.=20
> Why is the KVM soooooo Baaaad? How else can I achieve this?
> =20
> This other scenario would be even more complex than described in the =
> diagram below.
> The ultimate setup would have two QWERTY keyboards/mice with KVM=20
> sharing with Paris/Cubase
> and Cubase independently.
> =20
> The room as it relates to the photo:
> My MIDI setup mimics what is shown to left of the
> mixing position here except there's a desk spot for legs on the =
right=20
> along
> with a controller. I'd like a 2 x 22" LCD widescreens, QWERTY & =
mouse
> there along with the KVM QWERTY/mouse in the center as shown for =
Paris
> and mixdown work.
> =20
> What would you suggest?
> =20
> Getting more confused by the minute...
> Tom
> =20
> =20
> =20
> =20
> =20
> =20
>
> "Chris Ludwig" <chrisl@adkproaudio.com
> <mailto:chrisl@adkproaudio.com>> wrote in message
> news:458c3c62$1@linux...
> Tom Bruhl GOOOODDDD.....KVM BBBBBAAAAADDDDDD
>
>
>
>
> Tom Bruhl wrote:
> > Okay Aaron I hope I get this right.
> >=20
> > The first diagram is what I think might not work due to the
> position of
> > the splitter. I'm afraid it won't amplify the signal if
> > placed at the end of the line. If I'm wrong then that's =
great!=20
> You'll
> > want to widen your view for this one.
> >=20
> > _Example 1:_
> >=20
> >=20
> > =
=
=20
> 21"
> > VGA monitor & 22" LCD with VGA input
> > =
=20
> 2
> > x 5' extensions
> > =
=20
> Video
> > Splitter (Aarons active type)
> > =
=20
> 20'
> > extension
> > =20
> KVM #1 out
> > KVM #1 In
> > VGA #1 out
> > Paris Comp - G550
> > VGA #2 out=20
> > KVM #2 In
> > =20
> KVM #2 out
> > =
=20
> 20'
> > extension
> > =
=20
> Video
> > Splitter (Aaron's active type)
> > =
=20
> 2
> > x 5' extensions
> > =
=
=20
>
> > 21" VGA monitor & 22" LCD with VGA input
> >=20
> >=20
> >=20
> > =
KVM
> #1 out
> > (Shares with above from this point on.)
> > KVM
> > #3 in =20
> > VGA #1 out
> > Cubase Comp - 7300GT
> > DVI out to VGA connector out
> > KVM #4 in
> > =
KVM
> #2 out
> > (Shares with above from this point on.)
> >=20
> > =20
> > OR:
> > _Example 2:_
> >=20
> > =
=
=20
> 21"
> > VGA monitor & 22" LCD with VGA input
> > =
=
=20
>
> > 2 x 5' extensions
> > =
=20
>
> > Passive Video Splitter (Y cable)
> > =
=20
>
> > 20' extension
> > =
=20
>
> > KVM #1 out
> > =
=20
> KVM
> > #1 In
> > Video amplifier
> (Like
> > splitter in example 1 with only one output)
> > VGA #1 out
> > Paris Comp - G550
> > VGA #2 out
> > Video amplifier
> (Like
> > splitter in example 1 with only one output)
> > =
=20
> > KVM #2 In
> > =
=20
>
> > KVM #2 out
> > =
=20
>
> > 20' extension
> > =
=20
>
> > Passive Video Splitter (Y cable)
> > =
=
=20
> 2
> > x 5' extensions
> > =
=20
>
> > =
21"
> VGA
> > monitor & 22" LCD with VGA input
> >=20
> >=20
> >=20
> > =
=20
>
> > KVM #1 out (Shares with above from this point on.)
> > =
=20
> KVM
> > #3 in =20
> > Video amplifier =
(Like
> > splitter in example 1 with only one output)
> > VGA #1 out
> > Cubase Comp - 7300GT
> > DVI out to VGA connector out
> > Video amplifier =
(Like
> > splitter in example 1 with only one output)
> > =
=20
> KVM #4 in
> > =
=20
>
> > KVM #2 out (Shares with above from this point on.)
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >=20
> >
> > "Aaron Allen" <know-spam@not_here.dude
> <mailto:know-spam@not_here.dude>
> > <mailto:know-spam@not_here.dude>> wrote in message
> > news:458c050f@linux...
> > I'm not getting the visual you have in mind.. diagram it?
> > =20
> > AA
> >
> > "Tom Bruhl" <arpegio@comcast.net
> <mailto:arpegio@comcast.net> <mailto:arpegio@comcast.net>>
> > wrote in message news:458b7f6f@linux...
> > Thanks Aaron,
> > Will this work on the backside of a KVM? I can't put =
it
> in front.
> > I guess if it needed to be in front of the KVM I could
> just get a
> > single output jobbie and split it at the other side. =
Do
> those
> > exist?
> > =20
> >
> > "Aaron Allen" <know-spam@not_here.dude
> <mailto:know-spam@not_here.dude>
> > <mailto:know-spam@not_here.dude>> wrote in message
> > news:458b23a2@linux...
> > I hope you're not doing a passive Y split. Look =
for
> a VGA
> > splitter on google, something like
> > =20
> > =20
> http://www.pcmall.com/pcmall/shop/detail~dpno~767745.asp
> < http://www.pcmall.com/pcmall/shop/detail%7Edpno%7E767745.asp>
> > =20
> > They're cheap and OH so worth it
> > =20
> > AA
> >
> > "Tom Bruhl" <arpegio@comcast.net
> <mailto:arpegio@comcast.net>
> > <mailto:arpegio@comcast.net>> wrote in message
> > news:4589ba0a$1@linux...
> > Thanks Chris.
> > This graphics stuff is confusing to me.
> > =20
> > I have to convert the DVI to VGA because of =
the
> 20' length
> > as I understand. I was going to split it with =
a
> Y cable
> > too.
> > They'll split to a 21"VGA and a 22"LCD. Will
> that work on
> > the far side of a KVM? I know the resolutions
> won't match
> > but I'll only need the VGAs when mixing so =
I'll
> adjust the
> > 7300 differently for that application. Sooner
> or later
> > I'll get
> > 4 matching 22" LCDs and quad cards for both =
comps.
> > =20
> > Will this work?
> > Tom
> > =20
> > =20
> > =20
> > =20
> > =20
> > =20
> >
> > "Chris Ludwig" <chrisl@adkproaudio.com
> <mailto:chrisl@adkproaudio.com>
> > <mailto:chrisl@adkproaudio.com>> wrote in
> message
> > news:45894ef0$1@linux...
> > HI Tom,
> > These monitors will be fine with your =
card. The
> > 7300GT supports that
> > resolution on both the VGA and DVI =
connections.I
> > would run both with VGA
> > connections. Having one hooked up DVI and
> one VGA
> > may make the 2 look
> > very different in brightness and such.
> > Chris
> >
> >
> > Tom Bruhl wrote:
> >
> > > My card has one VGA, one DVI and one S =
Video
> > connector on it.
> > >
> > >
> > > I choose Polesoft Lockspam to fight =
spam,
> and you?
> > > http://www.polesoft.com/refer.html=20
> >
> >
> > --
> > Chris Ludwig
> > ADK
> > chrisl@adkproaudio.com
> <mailto:chrisl@adkproaudio.com>
> > <mailto:chrisl@adkproaudio.com>
> > <mailto:chrisl@adkproaudio.com>
> > www.adkproaudio.com
> <http://www.adkproaudio.com> <http://www.adkproaudio.com>
> > <http://www.adkproaudio.com/>
> > (859) 635-5762
>
> --=20
> Chris Ludwig
>
> ADK Pro Audio
> (859) 635-5762
> www.adkproaudio.com <http://www.adkproaudio.com>
> chrisl@adkproaudio.com <mailto:chrisl@adkproaudio.com>
>
>
> =
------------------------------------------------------------ ------------
>
--=20
Chris Ludwig
ADK
chrisl@adkproaudio.com <mailto:chrisl@adkproaudio.com>
www.adkproaudio.com <http://www.adkproaudio.com/>
(859) 635-5762
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<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
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<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>So I gave Chris a little Christmas =
chuckle did=20
I?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>I'm glad I could be of =
service!</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<BLOCKQUOTE=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Chris Ludwig" <<A=20
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A>> =
wrote in=20
message <A =
href=3D"news:458cc07c$1@linux">news:458cc07c$1@linux</A>...</DIV>Hi=20
Tom,<BR>Sorry you took the post too serious. I was just amused by the=20
<BR>complicated setup. I think you've been drinking from DJ =
bottle..:)<BR>I=20
think it would be cooler if you had separate monitors for everything. =
<BR>It'd=20
be like a spaceship.<BR><BR><BR>Chris<BR><BR><BR>Tom Bruhl =
wrote:<BR><BR>>=20
Hi Chris,<BR>> My dual KVM works great as is. Is the dual =
monitor=20
idea so bad? How <BR>> else will I be able to sit<BR>> in =
my sweet=20
spot and bounce between 2 and maybe even three comps with <BR>> one =
mouse/keyboard?<BR>> <BR>> They make quad KVMs. I =
might end=20
up with that in the end. 3 comps <BR>> each with 4 head=20
cards,<BR>> four monitors, one keyboard/mouse. <BR>> Why is the =
KVM=20
soooooo Baaaad? How else can I achieve this?<BR>> =
<BR>> This=20
other scenario would be even more complex than described in the =
<BR>>=20
diagram below.<BR>> The ultimate setup would have two QWERTY =
keyboards/mice=20
with KVM <BR>> sharing with Paris/Cubase<BR>> and Cubase=20
independently.<BR>> <BR>> The room as it relates to the=20
photo:<BR>> My MIDI setup mimics what is shown to left of =
the<BR>>=20
mixing position here except there's a desk spot for legs on the right =
<BR>>=20
along<BR>> with a controller. I'd like a 2 x 22" LCD =
widescreens,=20
QWERTY & mouse<BR>> there along with the KVM QWERTY/mouse in =
the center=20
as shown for Paris<BR>> and mixdown work.<BR>> <BR>> =
What would=20
you suggest?<BR>> <BR>> Getting more confused by the=20
minute...<BR>> Tom<BR>> <BR>> <BR>> =20
<BR>> <BR>> <BR>> =20
<BR>><BR>> "Chris Ludwig" <<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A><BR>>=
=20
<<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">mailto:chrisl@adkproaudio.com</A>&=
gt;>=20
wrote in message<BR>> <A=20
=
href=3D"news:458c3c62$1@linux">news:458c3c62$1@linux</A>...<BR>> =
=20
Tom Bruhl GOOOODDDD.....KVM=20
=
BBBBBAAAAADDDDDD<BR>><BR>><BR>><BR >><BR>>  =
; =20
Tom Bruhl wrote:<BR>> > Okay Aaron I =
hope I get=20
this right.<BR>> >=20
<BR>> > The first diagram is what I =
think might=20
not work due to the<BR>> position=20
of<BR>> > the splitter. I'm afraid =
it=20
won't amplify the signal if<BR>> > =
placed at the=20
end of the line. If I'm wrong then that's great!=20
<BR>> =
You'll<BR>> >=20
want to widen your view for this one.<BR>> =
>=20
<BR>> > _Example=20
1:_<BR>> > =
<BR>> >=20
<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; =20
<BR>> 21"<BR>> =
> VGA=20
monitor & 22" LCD with VGA input<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
; =20
<BR>> 2<BR>> =
> x 5'=20
extensions<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; =20
<BR>> Video<BR>> =
>=20
Splitter (Aarons active type)<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; =20
<BR>> 20'<BR>> =
>=20
extension<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp;=20
<BR>> KVM #1 =
out<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ; =20
KVM #1 In<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p;=20
VGA #1 out<BR>> > Paris Comp -=20
G550<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p;=20
VGA #2 out <BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ; =20
KVM #2 In<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp;=20
<BR>> KVM #2 =
out<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; =20
<BR>> 20'<BR>> =
>=20
extension<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; =20
<BR>> Video<BR>> =
>=20
Splitter (Aaron's active type)<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
; =20
<BR>> 2<BR>> =
> x 5'=20
extensions<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; =20
<BR>><BR>> > 21" VGA monitor & =
22" LCD=20
with VGA input<BR>> >=20
<BR>> > <BR>> =
>=20
<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp;=20
KVM<BR>> #1 =
out<BR>> =20
> (Shares with above from this point =
on.)<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ; =20
KVM<BR>> > #3=20
=
in &nbs p; &nbs p=
;   ; =20
<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; =20
VGA #1 out<BR>> > Cubase Comp -=20
7300GT<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
DVI out to VGA connector out<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ; =20
KVM #4 in<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp;=20
KVM<BR>> #2 =
out<BR>> =20
> (Shares with above from this point =
on.)<BR>> =20
> <BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;=20
<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p;=20
OR:<BR>> > _Example=20
2:_<BR>> > =
<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; <BR>> =20
21"<BR>> > VGA monitor & 22" LCD =
with VGA=20
input<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; =20
<BR>><BR>> > 2 x 5'=20
extensions<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ; =20
<BR>><BR>> > Passive Video Splitter =
(Y=20
cable)<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; =20
<BR>><BR>> > 20'=20
extension<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; =20
<BR>><BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp;=20
KVM #1 out<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; =20
<BR>> KVM<BR>> =
> #1=20
In<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ; =20
Video amplifier<BR>> =20
(Like<BR>> > splitter in example 1 with =
only one=20
output)<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p;=20
VGA #1 out<BR>> > Paris Comp -=20
G550<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p;=20
VGA #2 out<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ; =20
Video amplifier<BR>> =20
(Like<BR>> > splitter in example 1 with =
only one=20
output)<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; =20
<BR>> > KVM #2=20
In<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp;=20
<BR>><BR>> > KVM #2=20
out<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; =20
<BR>><BR>> > =
20'=20
extension<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; =20
<BR>><BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
Passive Video Splitter (Y cable)<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; =20
<BR>> 2<BR>> =
> x 5'=20
extensions<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; &n b=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
; =20
<BR>><BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp;=20
21"<BR>> =
VGA<BR>> >=20
monitor & 22" LCD with VGA input<BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; =20
<BR>><BR>> > KVM #1 out (Shares with =
above=20
from this point on.)<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; =20
<BR>> KVM<BR>> =
> #3=20
=
in &nbs p; &nbs p=
;   ; =20
<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ; =20
Video amplifier (Like<BR>> > splitter in =
example=20
1 with only one output)<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; =20
VGA #1 out<BR>> > Cubase Comp -=20
7300GT<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
DVI out to VGA connector out<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ; =20
Video amplifier (Like<BR>> > splitter in =
example=20
1 with only one output)<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; =20
<BR>> KVM #4 =
in<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; &nb s=
p; &nbs p; &nbs p=
;   ;   ;=
&=
nbsp; & nbsp; & n=
bsp; &n bsp; =20
<BR>><BR>> > KVM #2 out (Shares with =
above=20
from this point on.)<BR>> >=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
>=20
<BR>> > <BR>> =
><BR>> > =
"Aaron=20
Allen" <<A=20
=
href=3D"mailto:know-spam@not_here.dude">know-spam@not_here.dude</A><BR>&g=
t; =20
<<A=20
=
href=3D"mailto:know-spam@not_here.dude">mailto:know-spam@not_here.dude</A=
>><BR>> =20
> <<A=20
=
href=3D"mailto:know-spam@not_here.dude">mailto:know-spam@not_here.dude</A=
>>>=20
wrote in message<BR>> =
> =20
<A=20
=
href=3D"news:458c050f@linux">news:458c050f@linux</A>...<BR>> &nbs=
p; =20
> I'm not getting the visual you have in =
mind..=20
diagram it?<BR>> =
> =20
<BR>> > =20
AA<BR>> =
><BR>> =20
> &a mp;nbsp; "Tom Bruhl" =
<<A=20
=
href=3D"mailto:arpegio@comcast.net">arpegio@comcast.net</A><BR>> =
=20
<<A =
href=3D"mailto:arpegio@comcast.net">mailto:arpegio@comcast.net</A>>=20
<<A=20
=
href=3D"mailto:arpegio@comcast.net">mailto:arpegio@comcast.net</A>>>=
;<BR>> =20
> &a mp;nbsp; wrote in message =
<A=20
=
href=3D"news:458b7f6f@linux">news:458b7f6f@linux</A>...<BR>> &nbs=
p; =20
> &a mp;nbsp; Thanks=20
Aaron,<BR>> =20
> &a mp;nbsp; Will this work on =
the=20
backside of a KVM? I can't put =
it<BR>> in=20
front.<BR>> =20
> &a mp;nbsp; I guess if it =
needed to=20
be in front of the KVM I could<BR>> just =
get=20
a<BR>> =20
> &a mp;nbsp; single output =
jobbie and=20
split it at the other side. Do<BR>> =20
those<BR>> =20
> &a mp;nbsp; =20
exist?<BR>> =20
> &a mp;nbsp; =20
<BR>> ><BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp;=20
"Aaron Allen" <<A=20
=
href=3D"mailto:know-spam@not_here.dude">know-spam@not_here.dude</A><BR>&g=
t; =20
<<A=20
=
href=3D"mailto:know-spam@not_here.dude">mailto:know-spam@not_here.dude</A=
>><BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp;=20
<<A=20
=
href=3D"mailto:know-spam@not_here.dude">mailto:know-spam@not_here.dude</A=
>>>=20
wrote in message<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp;=20
<A=20
=
href=3D"news:458b23a2@linux">news:458b23a2@linux</A>...<BR>> &nbs=
p; =20
=
> &a mp;nbsp; &a mp;nb=
sp; I=20
hope you're not doing a passive Y split. Look=20
for<BR>> a =
VGA<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp;=20
splitter on google, something like<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp;=20
<BR>> =20
> &a mp;nbsp; =
<BR>> <A=20
=
href=3D"http://www.pcmall.com/pcmall/shop/detail~dpno~767745.asp">http://=
www.pcmall.com/pcmall/shop/detail~dpno~767745.asp</A><BR>> =
=20
<<A=20
=
href=3D" http://www.pcmall.com/pcmall/shop/detail%7Edpno%7E767745.asp">htt=
p://www.pcmall.com/pcmall/shop/detail%7Edpno%7E767745.asp</A>><BR>>=
=20
=
> &a mp;nbsp; &a mp;nb=
sp;=20
<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp;=20
They're cheap and OH so worth it<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp;=20
<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp;=20
AA<BR>> =
><BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
"Tom Bruhl" <<A=20
=
href=3D"mailto:arpegio@comcast.net">arpegio@comcast.net</A><BR>> =
=20
<<A=20
=
href=3D"mailto:arpegio@comcast.net">mailto:arpegio@comcast.net</A>><BR=
>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
<<A=20
=
href=3D"mailto:arpegio@comcast.net">mailto:arpegio@comcast.net</A>>>=
; wrote=20
in message<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
<A=20
=
href=3D"news:4589ba0a$1@linux">news:4589ba0a$1@linux</A>...<BR>> =
=20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
Thanks Chris.<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
This graphics stuff is confusing to =
me.<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
I have to convert the DVI to VGA because of=20
the<BR>> 20' =
length<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
as I understand. I was going to split it with=20
a<BR>> Y =
cable<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
too.<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
They'll split to a 21"VGA and a 22"LCD. =20
Will<BR>> that work=20
on<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
the far side of a KVM? I know the=20
resolutions<BR>> won't=20
match<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
but I'll only need the VGAs when mixing so=20
I'll<BR>> adjust=20
the<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
7300 differently for that application. =20
Sooner<BR>> or=20
later<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
I'll get<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
4 matching 22" LCDs and quad cards for both=20
comps.<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
Will this work?<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
Tom<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; =20
<BR>> ><BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
"Chris Ludwig" <<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A><BR>>=
=20
<<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">mailto:chrisl@adkproaudio.com</A>&=
gt;<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
<<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">mailto:chrisl@adkproaudio.com</A>&=
gt;>=20
wrote in<BR>> =20
message<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
<A=20
=
href=3D"news:45894ef0$1@linux">news:45894ef0$1@linux</A>...<BR>> =
=20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
HI Tom,<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
These monitors will be fine with your card.=20
The<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
7300GT supports that<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
resolution on both the VGA and DVI=20
connections.I<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
would run both with VGA<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
connections. Having one hooked up DVI =
and<BR>> one=20
VGA<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
may make the 2 look<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
very different in brightness and such.<BR>> =
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
Chris<BR>> =
><BR>> =20
><BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
Tom Bruhl wrote:<BR>> =20
><BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
> My card has one VGA, one DVI and one S=20
Video<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
connector on it.<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
><BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
><BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
> I choose Polesoft Lockspam to fight =
spam,<BR>> =20
and you?<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
> <A=20
=
href=3D"http://www.polesoft.com/refer.html">http://www.polesoft.com/refer=
..html</A>=20
<BR>> ><BR>> =20
><BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
--<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
Chris Ludwig<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
ADK<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A><BR>>=
=20
<<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">mailto:chrisl@adkproaudio.com</A>&=
gt;<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
<<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">mailto:chrisl@adkproaudio.com</A>&=
gt;<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
<<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">mailto:chrisl@adkproaudio.com</A>&=
gt;<BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
<A=20
=
href=3D"http://www.adkproaudio.com">www.adkproaudio.com</A><BR>> =
=20
<<A =
href=3D"http://www.adkproaudio.com">http://www.adkproaudio.com</A>>=20
<<A=20
=
href=3D"http://www.adkproaudio.com">http://www.adkproaudio.com</A>><BR=
>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
<<A=20
=
href=3D"http://www.adkproaudio.com/">http://www.adkproaudio.com/</A>><=
BR>> =20
=
> &a mp;nbsp; &a mp;nb=
sp; &nb sp; =20
(859) 635-5762<BR>><BR>> --=20
<BR>> Chris=20
Ludwig<BR>><BR>> ADK Pro=20
Audio<BR>> (859)=20
635-5762<BR>> <A=20
href=3D"http://www.adkproaudio.com">www.adkproaudio.com</A> <<A=20
=
href=3D"http://www.adkproaudio.com">http://www.adkproaudio.com</A>><BR=
>> =20
<A href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A> =
<<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">mailto:chrisl@adkproaudio.com</A>&=
gt;<BR>><BR>><BR>>=20
=
------------------------------------------------------------ ------------ <=
BR>><BR><BR>--=20
<BR>Chris Ludwig<BR>ADK<BR><A=20
href=3D"mailto:chrisl@adkproaudio.com">chrisl@adkproaudio.com</A> =
<<A=20
=
href=3D"mailto:chrisl@adkproaudio.com">mailto:chrisl@adkproaudio.com</A>&=
gt;<BR><A=20
href=3D"http://www.adkproaudio.com">www.adkproaudio.com</A> <<A=20
=
href=3D"http://www.adkproaudio.com/">http://www.adkproaudio.com/</A>><=
BR>(859)=20
635-5762</BLOCKQUOTE></BODY></HTML>
------=_NextPart_000_009A_01C72636.981EBFF0--This is a multi-part message in MIME format.
------=_NextPart_000_00A5_01C72637.1A3D4BF0
Content-Type: text/plain;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
I got things really cooking here with my new rig except the
Creamware folks are on vacation from today until 1/2/07.=20
I picked a great day to install the Pro card . . .
Hurry up and wait.
Now, Cubase 4 ? ? ? Should I wait?
Naw, well maybe . . .
NOT!
I choose Polesoft Lockspam to fight spam, and you?
http://www.polesoft.com/refer.html
------=_NextPart_000_00A5_01C72637.1A3D4BF0
Content-Type: text/html;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2800.1400" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>I got things really cooking here with =
my new rig=20
except the</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Creamware folks are on vacation =
from today=20
until 1/2/07. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I picked a great day to install the Pro =
card . .=20
..</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Hurry up and wait.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Now, Cubase 4 ? ? ? Should I=20
wait?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2> =20
Naw, well maybe . . .</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>NOT!</FONT></DIV>
<DIV><FONT size=3D2><BR><BR>I choose Polesoft Lockspam to fight spam, =
and=20
you?<BR><A=20
href=3D"http://www.polesoft.com/refer.html">http://www.polesoft.com/refer=
..html</A> </FONT></DIV></BODY ></HTML>
------=_NextPart_000_00A5_01C72637.1A3D4BF0--Thanks guys, I'll try this when I get back from vacation for the holiday.
Peace and Joy,
AAHey Mike! On the Mac, go to help on the menu bar at the top and type in "networking"
or "networking a PC and a Mac". It should all be there, just scroll down
for the answers.
James
"Mike R." <emarenot@yahoo.com> wrote:
>
>
>I've got a Mac G4 (OSX 10.4) and a PC running Windows 98SE and Paris. I
>want to be able to move .wav files made on the G4 over to Paris, and from
>Paris back over to the Mac --if I want to apply FX there that I cannot in
>Paris. As I understand it, the G4 has an ethernet card built in. I assume
>that I have to get a NIC card for the PC. Do I need to get some sort of
>software for each machine so that the PC will see the Mac and vice versa?
>This is totally new territory. I wish it was as simple as plugging in a
>cable between two boxes, but I fear it is not...
>Thanks much,
>MR
>
>
><!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
><HTML><HEAD>
><META http-equiv=3DContent-Type content=3D"text/html; =
>charset=3Diso-8859-1">
><META content=3D"MSHTML 6.00.2800.1555" name=3DGENERATOR>
><STYLE></STYLE>
></HEAD>
><BODY bgColor=3D#ffffff>
><DIV><FONT face=3DArial size=3D2>I've got a Mac G4 (OSX 10.4) and a PC =
>running=20
>Windows 98SE and Paris. I want to be able to move .wav files made =
>on the=20
>G4 over to Paris, and from Paris back over to the Mac --if I want to =
>apply FX=20
>there that I cannot in Paris. As I understand it, the G4 has an =
>ethernet=20
>card built in. I assume that I have to get a NIC card for the PC. =
>Do I=20
>need to get some sort of software for each machine so that the PC will =
>see the=20
>Mac and vice versa? This is totally new territory. I wish it =
>was as=20
>simple as plugging in a cable between two boxes, but I fear it is=20
>not...</FONT></DIV>
><DIV><FONT face=3DArial size=3D2>Thanks much,</FONT></DIV>
><DIV><FONT face=3DArial size=3D2>MR</FONT></DIV></BODY></HTML>
>
>Here's a new AMP that's coming out. They have video demos of different styles.
I think they're in the ball park with that Pantera, Zakk Wylde triple rectum
fryer sound, if your in to that sort of thing. There's some good sounds,
blues and clean etc., but then again, I'm a drummer, what the heck do I know!!!
; )
http://www.the-valve.com/videos.html
http://www.the-valve.com/
James"TCB" <nobody@ishere.com> wrote:
>
>That's too bad. I think people have an instinctive thing against the sound
>of Live as well, just because it also loops like ACID does. Live sounds
like
>a properly written native DAW when working with non time stretched tracks.
>The sound quality on the stretched audio is amazing, all things considered,
>but the non-stretched sound is indistinguishable from SX. Too bad the only
>really truly awful sounding app has to bring down a perfectly nice sounding
>one.
>
>TCB
I'm a recent convert to Live 6 after testing its audio engine. I had also
been under the impression it would affect the quality of my mixdowns but
am happy to see thats not the case. Great program although I still use SX3
for some midi editing recording.This is a multi-part message in MIME format.
------=_NextPart_000_039F_01C72653.C1A67080
Content-Type: text/plain;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
So, do you have it there and you are having problems with it?=20
Deej
"Tom Bruhl" <arpegio@comcast.net> wrote in message =
news:458cd310@linux...
I got things really cooking here with my new rig except the
Creamware folks are on vacation from today until 1/2/07.=20
I picked a great day to install the Pro card . . .
Hurry up and wait.
Now, Cubase 4 ? ? ? Should I wait?
Naw, well maybe . . .
NOT!
I choose Polesoft Lockspam to fight spam, and you?
http://www.polesoft.com/refer.html
------=_NextPart_000_039F_01C72653.C1A67080
Content-Type: text/html;
charset="iso-8859-1"
Content-Transfer-Encoding: quoted-printable
<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
<HTML><HEAD>
<META http-equiv=3DContent-Type content=3D"text/html; =
charset=3Diso-8859-1">
<META content=3D"MSHTML 6.00.2900.2180" name=3DGENERATOR>
<STYLE></STYLE>
</HEAD>
<BODY bgColor=3D#ffffff>
<DIV><FONT face=3DArial size=3D2>So, do you have it there and you are =
having=20
problems with it? </FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Deej</FONT></DIV>
<BLOCKQUOTE dir=3Dltr=20
style=3D"PADDING-RIGHT: 0px; PADDING-LEFT: 5px; MARGIN-LEFT: 5px; =
BORDER-LEFT: #000000 2px solid; MARGIN-RIGHT: 0px">
<DIV>"Tom Bruhl" <<A=20
href=3D"mailto:arpegio@comcast.net">arpegio@comcast.net</A>> wrote =
in message=20
<A href=3D"news:458cd310@linux">news:458cd310@linux</A>...</DIV>
<DIV><FONT face=3DArial size=3D2>I got things really cooking here with =
my new rig=20
except the</FONT></DIV>
<DIV><FONT face=3DArial size=3D2>Creamware folks are on vacation =
from today=20
until 1/2/07. </FONT></DIV>
<DIV><FONT face=3DArial size=3D2>I picked a great day to install the =
Pro card . .=20
.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Hurry up and wait.</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>Now, Cubase 4 ? ? ? Should I=20
wait?</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2> =
Naw, well maybe . . .</FONT></DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2></FONT> </DIV>
<DIV><FONT face=3DArial size=3D2>NOT!</FONT></DIV>
<DIV><FONT size=3D2><BR><BR>I choose Polesoft Lockspam to fight spam, =
and=20
you?<BR><A=20
=
href=3D"http://www.polesoft.com/refer.html">http://www.polesoft.com/refer=
..html</A> </FONT></DIV></BLOCKQUOTE ></BODY></HTML>
------=_NextPart_000_039F_01C72653.C1A67080--Thank you very very much all!!!
Merry Christmas.
MR
"James McCloskey" <excelsm@hotmail.com> wrote in message
news:458ce4ef$1@linux...
>
> Hey Mike! On the Mac, go to help on the menu bar at the top and type in
"networking"
> or "networking a PC and a Mac". It should all be there, just scroll down
> for the answers.
>
> James
>
> "Mike R." <emarenot@yahoo.com> wrote:
> >
> >
> >I've got a Mac G4 (OSX 10.4) and a PC running Windows 98SE and Paris. I
> >want to be able to move .wav files made on the G4 over to Paris, and from
> >Paris back over to the Mac --if I want to apply FX there that I cannot in
> >Paris. As I understand it, the G4 has an ethernet card built in. I
assume
> >that I have to get a NIC card for the PC. Do I need to get some sort of
> >software for each machine so that the PC will see the Mac and vice versa?
> >This is totally new territory. I wish it was as simple as plugging in a
> >cable between two boxes, but I fear it is not...
> >Thanks much,
> >MR
> >
> >
> ><!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
> ><HTML><HEAD>
> ><META http-equiv=3DContent-Type content=3D"text/html; =
> >charset=3Diso-8859-1">
> ><META content=3D"MSHTML 6.00.2800.1555" name=3DGENERATOR>
> ><STYLE></STYLE>
> ></HEAD>
> ><BODY bgColor=3D#ffffff>
> ><DIV><FONT face=3DArial size=3D2>I've got a Mac G4 (OSX 10.4) and a PC =
> >running=20
> >Windows 98SE and Paris. I want to be able to move .wav files made =
> >on the=20
> >G4 over to Paris, and from Paris back over to the Mac --if I want to =
> >apply FX=20
> >there that I cannot in Paris. As I understand it, the G4 has an =
> >ethernet=20
> >card built in. I assume that I have to get a NIC card for the PC. =
> >Do I=20
> >need to get some sort of software for each machine so that the PC will =
> >see the=20
> >Mac and vice versa? This is totally new territory. I wish it =
> >was as=20
> >simple as plugging in a cable between two boxes, but I fear it is=20
> >not...</FONT></DIV>
> ><DIV><FONT face=3DArial size=3D2>Thanks much,</FONT></DIV>
> ><DIV><FONT face=3DArial size=3D2>MR</FONT></DIV></BODY></HTML>
> >
> >
>Hi Lamont,
I've posted this several times in the past, but here's the scoop. Edmund
did not write the summing code. It's deep within the DSP code running on
the ESP2 chips. It was written by some very talented guys at Ensoniq. I
really dig everything that Edmund and Stephen did, but the summing just isn't
part of it.
The stuff I posted is not really a theory. The PARIS mix engine source code
is freely available for download. Anyone with a little time, patience and
the ESP2 patent can clearly see what is going on. It's only a couple hundred
lines of code.
Chuck
"Dedric Terry" <dedric@echomg.com> wrote:
>I can't tell you why you hear ProTools differently than Nuendo using a
>single file.
>There isn't any voodoo in the software, or hidden character enhancing dsp.
>I'll see if
>I can round up an M-Powered system to compare with next month.
>
>For reference, everytime I open Sequoia I think I might hear a broader,
>clean,
>and almost flat (spectrum, not depth) sound, but I don't - it's the same
as
>Nuendo, fwiw.
>Also I don't think what I was referring to was a theory from Chuck - I
>believe that was what he
>discovered in the code.
>
>Digital mixers all have different preamps and converters. Unless you are
>bypassing every
>EQ and converter and going digital in and out to the same converter when
>comparing, it would be hard
>to say the mix engine itself sounds different than another mixer, but taken
>as a whole, then
>certainly they may very well sound different. In addition, hardware digital
>mixers may use a variety of different paths between the I/O, channel
>processing, and summing,
>though most are pretty much software mixers on a single chip or set of dsps
>similar to ProTools,
>with I/O and a hardware surface attached.
>
>I know it may be hard to separate the mix engine as software in either a
>native DAW
>or a digital mixer, from the hardware that translates the audio to something
>we hear,
>but that's what is required when comparing summing. The hardware can
>significantly change
>what we hear, so comparing digital mixers really isn't of as much interest
>as comparing native
>DAWs in that respect - unless you are looking to buy one of course.
>
>Even though I know you think manufacturers are trying to add something to
>give them an edge, I am 100%
>sure that isn't the case - rather they are trying to add or change as little
>as possible in order to give
>them the edge. Their end of digital audio isn't about recreating the past,
>but improving upon it.
>As we've discussed and agreed before, the obsession with recreating
>"vintage" technology is as much
>fad as it is a valuable creative asset. There is no reason we shouldn't
>have far superior hardware and software EQs and comps
>than 20, 30 or 40 years ago. No reason at all, other than market demand,
>but the majority of software, and new
>hardware gear on the market has a vintage marketing tagline with it.
>Companies will sell any bill of
>goods if customers will buy it.
>
>There's nothing unique about the summing in Nuendo, Cubase, Sequoia/Samp,
>or Sonar, and it's pretty safe to include Logic and DP in that list as well.
>One of the reasons I test
>these things is to be sure my DAW isn't doing something wrong, or something
>I don't know about.
>
>Vegas - I use it for video conversions and have never done any critical
>listening tests with it. What I have heard
>briefly didn't sound any different. It certainly looks plain vanilla
>though. What you are describing is exactly
>what I would say about the GUIs of each of those apps, not that it means
>anything. Just interesting.
>
>That's one reason I listen eyes closed and double check with phase
>cancellation tests and FFTs - I am
>influenced creatively by the GUI to some degree. I actually like Cubase
4's
>GUI better than Nuendo 3.2,
>though there are only slight visual differences (some workflow differences
>are a definite improvement for me though).
>
>ProTools' GUI always made me want to write one dimensional soundtracks in
>mono for public utilities, accounting offices
>or the IRS while reading my discreet systems analysis textbook - it was
also
>grey. ;-)
>
>Regards,
>Dedric
>
>"LaMont" <jjdpro@ameritech.net> wrote in message news:458c82fd$1@linux...
>>
>> Dedric, my simple test is simple..
>> Using the same audio interface, with the same stereo file..null-ed to
>> zero..No
>> eq, for fx. Master fader on zero..
>>
>> Nuendo, Pro-Tools -Mpowered(native)... yields a sonic difference that
I
>> have
>> referenced before.. The sound coming from PT-M has a nice top end , where
>> as Neundo has a nice flatter sound quality.
>> Same audio interface. M-audio 410..Using Mackies & Blue-Sky pro monitors..
>>
>> Same test at the big room..PT-HD & Neundo Logic Audio(macG5-Dual) Using
>> the
>> 192 interface.
>> Same results..But adding Logic audio's sound ..(Broad, thick)
>>
>> Somethings going on.
>>
>> Chucks post about how paris handles audio is a theory..Only Edmund can
>> truly
>> give us the goods on what's really what..
>>
>> I disagree that manufactuers don;t set out o put a sonic print on their
>> products.
>> I think they do.
>>
>> I have been fortunate to work on some digital mixers and I can tell you
>> that
>> each one has their own sound. The Sony Dmx-100 was modeled after SSL 4000g
>> (like it's Big Brother).And you what? That board (Dmx-100) sound very
warm
>> and it's eq tries to behave and sound just like an SSL.. Unlike he Yamaha
>> Dm2000(version 1.x) which has a very Clean, neutral sound..However, some
>> complained that it was tooo Vanila and thus, Yamaha add a version 2.0
>> which
>> added Vintage type Eq's, modeled analog input gain saturation fx too give
>> the user a choice Btw Clean and Neutral vs sonic Character.
>>
>> So, if digital conoles can be given a sonic character, why not a software
>> mixer?
>> The truth is, there are some folks who want a neutral mixer and then there
>> are others who want a sonic footprint imparted. and these can be coded
in
>> the digital realm.
>> The apllies with the manufactuers. They too have their vision on what
They
>> think and want their product to sound.
>>
>> I love reading on gearslutz the posts from Plugin developers and their
>> interpretations
>> and opinions about what makes their Neve 1073 Eq better and what goes
into
>> making their version sound like it does.. Each Developer has a different
>> vision as to what the Neve 1073 should sound like. And yet they all sound
>> good , but slightly different.
>>
>> You stated that you use Vegas. Well as you know, Vegas has a very generic
>> sound..Just plain and simple. But, i bet you can tell the difference
on
>> your system when you play that same file in Neundo (No, fx, eq,
>> null-edzerro)..
>> ???
>>
>>
>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>Lamont - what is the output chain you are using for each app when
>>>comparing
>>
>>>the file in Nuendo
>>>vs ProTools? On the same PC, I presume (and is this PT HD or M-Powered?)?
>>>Since these can't use the same output driver, you would have to depend
on
>>
>>>the D/A being
>>>the same, but clocking will be different unless you have a master clock,
>> and
>>>both interfaces
>>>are locking with the same accuracy. This was one of the issues that came
>> up
>>>at Lynn Fuston's
>>>D/A converter shootout - when do you lock to external clock and incur
the
>>
>>>resulting jitter,
>>>and when do you trust the internal clock - and if you do lock externally,
>>
>>>how good is the PLL
>>>in the slave device? These issues can cause audible changes in the top
>> end
>>>that have nothing to do
>>>with the software itself. If you say that PTHD through the same converter
>>
>>>output as Nuendo (via? RME?
>>>Lynx?) using the same master clock, sounds different playing a single
>>>audio
>>
>>>file, then I take your word
>>>for it. I can't tell you why that is happening - only that an audible
>>>difference really shouldn't happen due
>>>to the software alone - not with a single audio file, esp. since I've
>>>heard
>>
>>>and seen PTHD audio cancel with
>>>native DAWs. Just passing a single 16 or 24 bit track down the buss
to
>> the
>>>output driver should
>>>be, and usually is, completely transparent, bit for bit.
>>>
>>>The same audio file played through the same converters should only sound
>>
>>>different if something in
>>>the chain is different - be it clocking, gain or some degree of
>>>unintended,
>>
>>>errant dsp processing. Every DAW should
>>>pass a single audio file without altering a single bit. That's a basic
>>>level
>>
>>>of accuracy we should always
>>>expect of any DAW. If that accuracy isn't there, you can be sure a heavy
>>
>>>mix will be altered in ways you
>>>didn't intend, even though you would end up mixing with that factor in
>>>place
>>
>>>(e.g. you still mix for what
>>>you want to hear regardless of what the platform does to each audio track
>> or
>>>channel).
>>>
>>>In fact you should be able to send a stereo audio track out SPDIF or
>>>lightpipe to another DAW, record it
>>>bring the recorded file back in, line them up to the first bit, and have
>>
>>>them cancel on and inverted phase
>>>test. I did this with Nuendo and Cubase 4 on separate machines just to
>> be
>>>sure my master clocking and
>>>slave sync was accurate - it worked perfectly.
>>>
>>>Also be sure there isn't a variation in the gain even by 0.1 dB between
>> the
>>>two. There shouldn't
>>>and I wouldn't expect there to be one. Also could PT be set for a
>>>different
>>
>>>pan law? Shouldn't make a
>>>difference even if comparing two mono panned files to their stereo
>>>interleaved equivalent, but for sake
>>>of completeness it's worth checking as well. A variation in the output
>>
>>>chain, be it drivers, audio card
>>>card, or converters would be the most likely culprit here.
>>>
>>>The reason DAW manufacturers wouldn't add any sonic "character"
>>>intentionally is that the
>>>ultimate goal from day one with recording has been to accurately reproduce
>>
>>>what we hear.
>>>We developed a musical penchant for sonic character because the hardware
>>
>>>just wasn't accurate,
>>>and what it did often sent us down new creative paths - even if by force
>> -
>>>and we decided it was
>>>preferred that way.
>>>
>>>Your point about what goes into the feature presets to sell synths is
>>>right
>>
>>>for sure, but synths are about
>>>character and getting that "perfect piano" or crystal clear bell pad,
or
>> fat
>>>punchy bass without spending
>>>a mint on development, adding 50G onboard sample libraries, or costing
>>>$15k,
>>
>>>so what they
>>>lack in actual synthesis capabilities, they make up with EQ and effects
>> on
>>>the output. That's been the case
>>>for years, at least since we had effects on synths at least. But even
>>>with
>>
>>>modern synths such as the Fantom,
>>>Tritons, etc, which are great synths all around, of course the coolest,
>>
>>>widest and biggest patches
>>>will make the biggest impression - so in come the EQs, limiters, comps,
>>
>>>reverbs, chorus, etc. The best
>>>way to find out if a synth is really good is to bypass all effects and
see
>>
>>>what happens. Most are pretty
>>>good these days, but about half the time, there are presets that fall
>>>completely flat in fx bypass.
>>>
>>>DAWs aren't designed to put a sonic fingerprint on a sound the way synths
>>
>>>are - they are designed
>>>to *not* add anything - to pass through what we create as users, with
no
>>
>>>alteration (or as little as possible)
>>>beyond what we add with intentional processing (EQ, comps, etc).
>>>Developers
>>
>>>would find no pride
>>>in hearing that their DAW sounds anything different than whatever is being
>>
>>>played back in it,
>>>and the concept is contrary to what AES and IEEE proceedings on the issue
>>
>>>propose in general
>>>digital audio discussions, white papers, etc.
>>>
>>>What ID ended up doing with Paris (at least from what I gather per Chuck's
>>
>>>findings - so correct me if I'm missing part of the equation Chuck),
>>>is drop the track gain by 20dB or so, then added it back at the master
>>>buss
>>
>>>to create the effect of headroom (probably
>>>because the master buss is really summing on the card, and they have more
>>
>>>headroom there than on the tracks
>>>where native plugins might be used). I don't know if Paris passed 32-bit
>>
>>>float files to the EDS card, but sort of
>>>doubt it. I think Chuck has clarified this at one point, but don't recall
>>
>>>the answer.
>>>
>>>Also what Paris did is use a greater bit depth on the hardware than
>>>ProTools
>>
>>>did - at the time PT was just
>>>bring Mix+ systems to market, or they had been out for a year or two (if
>> I
>>>have my timeline right) - they
>>>were 24-bit fixed all the way through. Logic and Cubase were native DAWs,
>>
>>>but native was still too slow
>>>to compete with hardware hybrids. Paris trumped them all by running
>>>32-bit
>>
>>>float natively (not new really, but
>>>better than sticking to 24-bit) and 56 or so bits in hardware instead
of
>>
>>>going to Motorola DSPs at 24.
>>>The onboard effects were also a step up from anything out there, so the
>> demo
>>>did sound good.
>>>I don't recall which, but one of the demos, imho, wasn't so good (some
>>>sloppy production and
>>>vocals in spots, IIRC), so I only listened to it once. ;-)
>>>
>>>Coupled with the gain drop and buss makeup, this all gave it a "headroom"
>> no
>>>one else had. With very nice
>>>onboard effects, Paris jumped ahead of anything else out there easily,
and
>>
>>>still respectably holds its' own today
>>>in that department.
>>>
>>>Most demos I hear (when I listen to them) vary in quality, usually not
so
>>
>>>great in some area. But if a demo does
>>>sound great, then it at least says that the product is capable of at
>>>least
>>
>>>that level of performance, and it can
>>>only help improve a prospective buyer's impression of it.
>>>
>>>Regards,
>>>Dedric
>>>
>>>"LaMont " <jjdpro@ameritech.net> wrote in message news:458c14c0$1@linux...
>>>>
>>>> Dedric good post..
>>>>
>>>> However, I have PT-M-Powered/M-audio 410 interface for my laptop and
it
>>
>>>> has
>>>> that same sound (no eq, zero fader) that HD does. I know their use the
>>
>>>> same
>>>> 48 bit fix mixer. I load up the same file in Nuendo (no eq, zero
>>>> fader)..results.
>>>> different sonic character.
>>>>
>>>> PT having a top end touch..Nuendo, nice smooth(flat) sound. And I'm
just
>>>> taking about a stereo wav file nulled with no eq..nothing
>>>> ..zilch..nada..
>>>>
>>>> Now, there are devices (keyboards, dum machines) on the market today
>>>> that
>>>> have a Master Buss Compressor and EQ set to on with the top end notched
>>
>>>> up.
>>>> Why? because it gives their product an competitive advantageover the
>>>> competition..
>>>> Ex: Yahama's Motif ES, Akai's MPC 1000, 2500, Roland's Fantom.
>>>>
>>>> So, why would'nt a DAW manufactuer code in an extra (ooommf) to make
>>>> their
>>>> DAW sound better. Especially, given the "I hate Digtal Summing" crowd?
>>
>>>> And,
>>>> If I'm a DAW manufactuer, what would give my product a sonic edge over
>> the
>>>> competition?
>>>>
>>>> We live in the "louder is better" audio world these days, so a DAW that
>>
>>>> can
>>>> catch my attention 'sonically" will probaly will get the sell. That's
>> what
>>>> happend to me back in 1997 when I heard Paris. I was floored!!! Still
>> to
>>>> this day, nothing has floored me like that "Road House Blues Demo" I
>>>> heard
>>>> on Paris.
>>>>
>>>> Was it the hardware ? was it the software. I remember talking with
>>>> Edmund
>>>> at the 2000 winter Namm, and told me that he & Steve set out to
>>>> reproduce
>>>> the sonics of big buck analog board (eq's) and all.. And, summing was
>> a
>>>> big
>>>> big issue for them because they (ID) thought that nobody has gotten
>>>> it(summing)
>>>> right. And by right, they meant, behaved like a console with a wide
lane
>>>> for all of those tracks..
>>>>
>>>>
>>>>
>>>>
>>>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>>>"LaMont" <jjdpro@ameritech.net> wrote in message
>>>>>news:458be8d5$1@linux...
>>>>>>
>>>>>> Okay...
>>>>>> I guess what I'm saying is this:
>>>>>>
>>>>>> -Is it possible that diferent DAW manufactuers "code" their app
>>>>>> differently
>>>>>> for sound results.
>>>>>
>>>>>Of course it is *possible* to do this, but only if the DAW has a
>>>>>specific
>>>>
>>>>>sound shaping purpose
>>>>>beyond normal summing/mixing. Users talk about wanting developers to
>> add
>>>> a
>>>>>"Neve sound" or "API sound" option to summing engines,
>>>>>but that's really impractical given the amount of dsp required to make
>> a
>>>>
>>>>>decent emulation (with convolution, dynamic EQ functions,
>>>>>etc). For sake of not eating up all cpu processing, that could likely
>>
>>>>>only
>>>>
>>>>>surface as is a built in EQ, which
>>>>>no one wants universally in summing, and anyone can add at will already.
>>>>>
>>>>>So it hasn't happened yet and isn't likely to as it detours from the
>>>>>basic
>>>>
>>>>>tenant of audio recording - recreate what comes in as
>>>>>accurately as possible.
>>>>>
>>>>>What Digi did in recoding their summing engine was try to recover some
>>>>>of the damage done by the 24-bit buss in Mix systems. Motorola 56k dsps
>>>> are
>>>>>24-bit fixed point chips and I think
>>>>>the new generation (321?) still is, but they use double words now for
>>>>>48-bits). And though plugins could process at 48-bit by
>>>>>doubling up and using upper and lower 24-bit words for 48-bit outputs,
>> the
>>>>
>>>>>buss
>>>>>between chips was 24-bits, so they had to dither to 24-bits after every
>>>>
>>>>>plugin. The mixer (if I recall correctly) also
>>>>>had a 24-bit buss, so what Digi did is to add a dither stage to the
>>>>>mixer
>>>> to
>>>>>prevent this
>>>>>constant truncation of data. 24-bits isn't enough to cover summing
for
>>>> more
>>>>>than a few tracks without
>>>>>losing information in the 16-bit world, and in the 24-bit world some
>>>>>information will be lost, at least at the lowest levels.
>>>>>
>>>>>Adding a dither stage (though I think they did more than that - perhaps
>>>>
>>>>>implement a 48-bit double word stage as well),
>>>>>simply smoothed over the truncation that was happening, but it didn't
>>
>>>>>solve
>>>>
>>>>>the problem, so with HD
>>>>>they went to a double-word path - throughout I believe, including the
>> path
>>>>
>>>>>between chips. I believe the chips
>>>>>are still 24-bit, but by doubling up the processing (yes at a cost of
>>
>>>>>twice
>>>>
>>>>>the overhead), they get a 48-bit engine.
>>>>>This not only provided better headroom, but greater resolution. Higher
>>>> bit
>>>>>depths subdivide the amplitude with greater resolution, and that's
>>>>>really where we get the definition of dynamic range - by lowering the
>>
>>>>>signal
>>>>
>>>>>to quantization noise ratio.
>>>>>
>>>>>With DAWs that use 32-bit floating point math all the way through, the
>>
>>>>>only
>>>>
>>>>>reason for altering the summing
>>>>>is by error, and that's an error that would actually be hard to make
and
>>>> get
>>>>>past a very basic alpha stage of testing.
>>>>>There is a small difference in fixed point math and floating point math,
>>>> or
>>>>>at least a theoretical difference in how it affects audio
>>>>>in certain cases, but not necessarily in the result for calculating
gain
>>>> in
>>>>>either for the same audio file. Where any differences might show up
is
>>>>
>>>>>complicated, and I believe only appear at levels below 24-bit (or in
>>>>>headroom with tracks pushed beyond 0dBFS), or when/if
>>>>>there areany differences in where each amplitude level is quantized.
>>>>>
>>>>>Obviously there can be differences if the DAW has to use varying bit
>>>>>depths
>>>>
>>>>>throughout a single summing path to accomodate hardware
>>>>>as well as software summing, since there may be truncation or rounding
>>
>>>>>along
>>>>
>>>>>the way, but that impacts the lowest bit
>>>>>level, and hence - spacial reproduction, reverb tails perhaps, and
>>>>>"depth",
>>>>
>>>>>not the levels most music so the differences are most
>>>>>often more subtle than not. But most modern DAWs have eliminated those
>>>>
>>>>>"rough edges" in the math by increasing the bit depth to accomodate
>>>>>normal
>>>>
>>>>>summing required for mixing audio.
>>>>>
>>>>>So with Lynn's unity gain summing test (A files on the CD I believe),
>> DAWs
>>>>
>>>>>were never asked to sum beyond 24-bits,
>>>>>at least not on the upper end of the dynamic range, so everything that
>>
>>>>>could
>>>>
>>>>>represent 24-bits accurately would cancel. The only ones
>>>>>that didn't were ones that had a different bit depth and/or gain
>>>>>structure
>>>>
>>>>>whether hybrid or native
>>>>>(e.g. Paris' subtracting 20dB from tracks and adding it to the buss).
>> In
>>>>
>>>>>this case, PTHD cancelled (when I tested it) with
>>>>>Nuendo, Samplitude, Logic, etc because the impact of the 48-bit fixed
>> vs.
>>>>
>>>>>32-bit float wasn't a factor.
>>>>>
>>>>>When trying other tests, even when adding and subtracting gain, Nuendo,
>>>>
>>>>>Sequoia and Sonar cancel - both audibly and
>>>>>visually at inaudible levels, which only proves that one isn't making
>> an
>>>>
>>>>>error when calculating basic gain. Since a dB is well defined,
>>>>>and the math to add gain is simple, they shouldn't. The fact that they
>>>> all
>>>>>use 32-bit float all the way through eliminates a difference
>>>>>in data structure as well, and this just verifies that. There was a
>>>>>time
>>>>
>>>>>that supposedly Logic (v3, v4?) was partly 24-bit, or so the rumor went,
>>>>>but it's 32-bit float all the way through now just as Sonar,
>>>>>Nuendo/Cubase,
>>>>
>>>>>Samplitude/Sequoia, DP, Audition (I presume at least).
>>>>>I don't know what Acid or Live use. Saw promotes a fixed point engine,
>>>> but
>>>>>I don't know if it is still 24-bit, or now 48 bit.
>>>>>That was an intentional choice by the developer, but he's the only one
>> I
>>>>
>>>>>know of that stuck with 24-bit for summing
>>>>>intentionally, esp. after the Digi Mix system mixer incident.
>>>>>
>>>>>Long answer, but to sum up, it is certainly physically *possible* for
>> a
>>>>
>>>>>developer to code something differently intentionally, but not
>>>>>in reality likely since it would be breaking some basic fixed point
or
>>>>>floating point math rules. Where the differences really
>>>>>showed up in the past is with PT Mix systems where the limitation was
>>
>>>>>really
>>>>
>>>>>significant - e.g. 24 bit with truncation at several stages.
>>>>>
>>>>>That really isn't such an issue anymore. Given the differences in
>>>>>workflow,
>>>>
>>>>>missing something in workflow or layout differences
>>>>>is easy enough to do (e.g. Sonar doesn't have group and busses the way
>>>>>Nuendo does, as it's outputs are actually driver outputs,
>>>>>not software busses, so in Sonar, busses are actually outputs, and sub
>>>>>busses are actually busses in Nuendo. There are no,
>>>>>or at least I haven't found the equivalent of a Nuendo group in Sonar
>> -
>>>> that
>>>>>affects the results of some tests (though not basic
>>>>>summing) if not taken into account, but when taken into account, they
>> work
>>>>
>>>>>exactly the same way).
>>>>>
>>>>>So at least when talking about apps with 32-bit float all the way
>>>>>through,
>>>>
>>>>>it's safe to say (since it has been proven) that summing isn't different
>>>>
>>>>>unless
>>>>>there is an error somewhere, or variation in how the user duplicates
the
>>>>
>>>>>same mix in two different apps.
>>>>>
>>>>>Imho, that's actually a very good thing - approaching a more consistent
>>>>
>>>>>basis for recording and mixing from which users can make all
>>>>>of the decisions as to how the final product will sound and not be
>>>>>required
>>>>
>>>>>to decide when purchasing a pricey console, and have to
>>>>>focus their business on clients who want "that sound". I believe we
are
>>>>
>>>>>actually closer to the pure definition of recording now than
>>>>>we once were.
>>>>>
>>>>>Regards,
>>>>>Dedric
>>>>>
>>>>>
>>>>>>
>>>>>> I the answer is yes, then,the real task is to discover or rather
>>>>>> un-cover
>>>>>> what's say: Motu's vision of summing, versus Digidesign, versus
>>>>>> Steinberg
>>>>>> and so on..
>>>>>>
>>>>>> What's under the hood. To me and others,when Digi re-coded their
>>>>>> summing
>>>>>> engine, it was obvious that Pro Tools has an obvious top end (8k-10k)
>>>>
>>>>>> bump.
>>>>>> Where as Steinberg's summing is very neutral.
>>>>>>
>>>>>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>>>>>Hi Neil,
>>>>>>>
>>>>>>>Jamie is right. And you aren't wacked out - you are thinking this
>>>>>>>through
>>>>>>
>>>>>>>in a reasonable manner, but coming to the wrong
>>>>>>>conclusion - easy to do given how confusing digital audio can be.
>>>>>>>Each
>>>>>> word
>>>>>>>represents an amplitude
>>>>>>>point on a single curve that is changing over time, and can vary with
>>>> a
>>>>>>
>>>>>>>speed up to the Nyquist frequency (as Jamie described).
>>>>>>>The complex harmonic content we hear is actually the frequency
>>>>>>>modulation
>>>>>> of
>>>>>>>a single waveform,
>>>>>>>that over a small amount of time creates the sound we translate -
we
>>
>>>>>>>don't
>>>>>>
>>>>>>>really hear a single sample at a time,
>>>>>>>but thousands of samples at a time (1 sample alone could at most
>>>>>>>represent
>>>>>> a
>>>>>>>single positive or negative peak
>>>>>>>of a 22,050Hz waveform).
>>>>>>>
>>>>>>>If one bit doesn't cancel, esp. if it's a higher order bit than number
>>>> 24,
>>>>>>
>>>>>>>you may hear, and will see that easily,
>>>>>>>and the higher the bit in the dynamic range (higher order) the more
>>>>>>>audible
>>>>>>
>>>>>>>the difference.
>>>>>>>Since each bit is 6dB of dynamic range, you can extrapolate how "loud"
>>>>
>>>>>>>that
>>>>>>
>>>>>>>bit's impact will be
>>>>>>>if there is a variation.
>>>>>>>
>>>>>>>Now, obviously if we are talking about 1 sample in a 44.1k rate song,
>>>> then
>>>>>>
>>>>>>>it simply be a
>>>>>>>click (only audible if it's a high enough order bit) instead of an
>>>>>>>obvious
>>>>>>
>>>>>>>musical difference, but that should never
>>>>>>>happen in a phase cancellation test between identical files higher
>>>>>>>than
>>>>>> bit
>>>>>>>24, unless there are clock sync problems,
>>>>>>>driver issues, or the DAW is an early alpha version. :-)
>>>>>>>
>>>>>>>By definition of what DAWs do during playback and record, every audio
>>>>
>>>>>>>stream
>>>>>>
>>>>>>>has the same point in time (judged by the timeline)
>>>>>>>played back sample accurately, one word at a time, at whatever sample
>>>>
>>>>>>>rate
>>>>>>
>>>>>>>we are using. A phase cancellation test uses that
>>>>>>>fact to compare two audio files word for word (and hence bit for bit
>>
>>>>>>>since
>>>>>>
>>>>>>>each bit of a 24-bit word would
>>>>>>>be at the same bit slot in each 24-bit word). Assuming they are
>>>>>>>aligned
>>>>>> to
>>>>>>>the same start point, sample
>>>>>>>accurately, and both are the same set of sample words at each sample
>>>>>>>point,
>>>>>>
>>>>>>>bit for bit, and one is phase inverted,
>>>>>>>they will cancel through all 24 bits. For two files to cancel
>>>>>>>completely
>>>>>>
>>>>>>>for the duration of the file, each and every bit in each word
>>>>>>>must be the exact opposite of that same bit position in a word at
the
>>>> same
>>>>>>
>>>>>>>sample point. This is why zooming in on an FFT
>>>>>>>of the full difference file is valuable as it can show any differences
>>>> in
>>>>>>
>>>>>>>the lower order bits that wouldn't be audible. So even if
>>>>>>>there is no audible difference, the visual followup will show if the
>> two
>>>>>>
>>>>>>>files truly cancel even a levels below hearing, or
>>>>>>>outside of a frequency change that we will perceive.
>>>>>>>
>>>>>>>When they don't cancel, usually there will be way more than 1 bit
>>>>>>>difference - it's usually one or more bits in the words for
>>>>>>>thousands of samples. From a musical standpoint this is usually in
>> a
>>>>>>>frequency range (low freq, or high freq most often) - that will
>>>>>>>show up as the difference between them, and that usually happens due
>> to
>>>>>> some
>>>>>>>form of processing difference between the files,
>>>>>>>such as EQ, compression, frequency dependant gain changes, etc. That
>> is
>>>>>> what
>>>>>>>I believe you are thinking through, but when
>>>>>>>talking about straight summing with no gain change (or known equal
>>>>>>>gain
>>>>>>
>>>>>>>changes), we are only looking at linear, one for one
>>>>>>>comparisons between the two files' frequency representations.
>>>>>>>
>>>>>>>Regards,
>>>>>>>Dedric
>>>>>>>
>>>>>>>> Neil wrote:
>>>>>>>>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>>>>>>>> The tests I did were completely blank down to -200 dB (far below
>> the
>>>>>>
>>>>>>>>>> last
>>>>>>>>>
>>>>>>>>>> bit). It's safe to say there is no difference, even in
>>>>>>>>>> quantization noise, which by technical rights, is considered below
>>>> the
>>>>>>
>>>>>>>>>> level
>>>>>>>>>
>>>>>>>>>> of "cancellation" in such tests.
>>>>>>>>>
>>>>>>>>> I'm not necessarily talking about just the first bit or the
>>>>>>>>> last bit, but also everything in between... what happens on bit
>>>>>>>>> #12, for example? Everything on bit #12 should be audible, but
>>>>>>>>> in an a/b test what if thre are differences in what bits #8
>>>>>>>>> through #12 sound like, but the amplutide is stll the same on
>>>>>>>>> both files at that point, you'll get a null, right? Extrapolate
>>>>>>>>> that out somewhat & let's say there are differences in bits #8
>>>>>>>>> through #12 on sample points 3, 17, 1,000, 4,523, 7,560, etc,
>>>>>>>>> etc through 43,972... Now this is breaking things down well
>>>>>>>>> beyond what I think can be measured, if I'm not mistaken (I
>>>>>>>>> dn't know of any way we could extract JUST that information
>>>>>>>>> from each file & play it back for an a/b test; but would not
>>>>>>>>> that be enough to have to "null-able" files that do actually
>>>>>>>>> sound somewhat different?
>>>>>>>>>
>>>>>>>>> I guess what I'm saying is that since each sample in a musical
>>>>>>>>> track or full song file doesn't represent a pure, simple set of
>>>>>>>>> content like a sample of a sine wave would - there's a whole
>>>>>>>>> world of harmonic structure in each sample of a song file, and
>>>>>>>>> I think (although I'll admit - I can't "prove") that there is
>>>>>>>>> plenty of room for some variables between the first bit & the
>>>>>>>>> last bit while still allowing for a null test to be successful.
>>>>>>>>>
>>>>>>>>> No? Am I wacked out of my mind?
>>>>>>>>>
>>>>>>>>> Neil
>>>>>>>>>
>>>>>>>
>>>>>>
>>>>>
>>>>>
>>>>
>>>
>>>
>>
>
>Fredos post was really good. At some point, for performance reasons developers
may choose to implement accumulators in CPU registers . This is most likely
what fredo calls an adder, I think.
It's possible that developers of other products don't do this, but the tradeoff
would be increased CPU and FPU utilization.
Fredo said there are three points where truncation occurs. In paris, at
least to my understanding there are many more than that.
Your not supposed to be able to hear this truncation distortion, but how
the hell knows, and that's not math :-)
Chuck
"LaMOnt" <jjdpro@ameritech.net> wrote:
>
>Dedric, check out this post from our dear friend Fredo: Neundo Moderator:
>Explaining how Steingberg's audio engine works. Note the trade-offs..Meaning,
>Steinberg's way of coding an audio-engine 32bit float is different than
say
>Magix Samplitude:
>
>Fredo
>Administrative Moderator
>
>
>Joined: 29 Dec 2004
>Posts: 4213
>Location: Belgium
> Posted: Fri Dec 08, 2006 2:33 pm Post subject:
>
> ------------------------------------------------------------ --------------------
>
>I think I see where the problem is.
>In my scenario's I don't have any track that goes over 0dBfs, but I have
>always lowered one channel to compensate with another.
>So, I never whent over the 0dB fs limit.
>
>Here's the explanation:
>
>As soon as you go over 0dB, technically you are entering the domain of distortion.
>
>In a 32bit FP mixer, that is not the case since there is unlimited headroom.
>
>
>Now follow me step by step please - read this slow and make sure you understand
>-
>
>At the end of each "stage", there is an adder (a big calculator) which adds
>all the numbers from the individual tracks that are routed to this "adder".
>
>The numbers are kept in the 80-bit registers and then brought back to 32bit
>float.
>This process of bringing back the numbers from 80-bit (and more) to 32bit
>is kept to an absolute minimum.
>This adding/bringing back to 32bit is done at 3 places: After a plugin slot
>(VST-specs for all plugin manufacturers) - Group Tracks and Master Tracks.
>
>
>Now, as soon as you boost the volume above 0dB, you get more than 32bits.
>Stay below 0dB and you will stay below 32 bits.
>When the adders dump their results, the numbers are brought back from any
>number of bits (say 60bit) to 32 bit float.
>These numbers are simply truncated which results in distortion; that's the
>noise/residue you find way down low.
>There is an algortithm that protects us from additive errors - so these
errors
>can never come into the audible range.
>So, as soon as you go over 0dB, you will see these kind of artifacts.
>
>It is debatable if this needs to be dithered or not. The problem -still
is-
>that it is very difficult to dither in a Floating Point environment.
>Fact remains that the error shouldn't be bigger than 2 to 3 LSB's.
>
>Is this a problem?
>In real world applictations: NO.
>In scientific -unrealistic- tests (forcing the erro ): YES.
>
>The alternative is having a Fixed point mixer, where you already would be
>in trouble as soon as you boost one channel over 0dBfs. (or merge two files
>that are @ 0dB)
>Also, this problem will be pretty much gone as soon as we switch to the
64
>bit engine.
>
>
>For the record, the test where Jake hears "music" as residue must be flawed.
>You should hear noise/distortion from square waves.
>
>HTH
>
>Fredo
>
>
>
>
>
>"Dedric Terry" <dedric@echomg.com> wrote:
>>I can't tell you why you hear ProTools differently than Nuendo using a
>>single file.
>>There isn't any voodoo in the software, or hidden character enhancing dsp.
>
>>I'll see if
>>I can round up an M-Powered system to compare with next month.
>>
>>For reference, everytime I open Sequoia I think I might hear a broader,
>
>>clean,
>>and almost flat (spectrum, not depth) sound, but I don't - it's the same
>as
>>Nuendo, fwiw.
>>Also I don't think what I was referring to was a theory from Chuck - I
>
>>believe that was what he
>>discovered in the code.
>>
>>Digital mixers all have different preamps and converters. Unless you are
>
>>bypassing every
>>EQ and converter and going digital in and out to the same converter when
>
>>comparing, it would be hard
>>to say the mix engine itself sounds different than another mixer, but taken
>
>>as a whole, then
>>certainly they may very well sound different. In addition, hardware digital
>>mixers may use a variety of different paths between the I/O, channel
>>processing, and summing,
>>though most are pretty much software mixers on a single chip or set of
dsps
>
>>similar to ProTools,
>>with I/O and a hardware surface attached.
>>
>>I know it may be hard to separate the mix engine as software in either
a
>
>>native DAW
>>or a digital mixer, from the hardware that translates the audio to something
>
>>we hear,
>>but that's what is required when comparing summing. The hardware can
>>significantly change
>>what we hear, so comparing digital mixers really isn't of as much interest
>
>>as comparing native
>>DAWs in that respect - unless you are looking to buy one of course.
>>
>>Even though I know you think manufacturers are trying to add something
to
>
>>give them an edge, I am 100%
>>sure that isn't the case - rather they are trying to add or change as little
>
>>as possible in order to give
>>them the edge. Their end of digital audio isn't about recreating the past,
>
>>but improving upon it.
>>As we've discussed and agreed before, the obsession with recreating
>>"vintage" technology is as much
>>fad as it is a valuable creative asset. There is no reason we shouldn't
>
>>have far superior hardware and software EQs and comps
>>than 20, 30 or 40 years ago. No reason at all, other than market demand,
>
>>but the majority of software, and new
>>hardware gear on the market has a vintage marketing tagline with it.
>>Companies will sell any bill of
>>goods if customers will buy it.
>>
>>There's nothing unique about the summing in Nuendo, Cubase, Sequoia/Samp,
>>or Sonar, and it's pretty safe to include Logic and DP in that list as
well.
>
>>One of the reasons I test
>>these things is to be sure my DAW isn't doing something wrong, or something
>
>>I don't know about.
>>
>>Vegas - I use it for video conversions and have never done any critical
>
>>listening tests with it. What I have heard
>>briefly didn't sound any different. It certainly looks plain vanilla
>>though. What you are describing is exactly
>>what I would say about the GUIs of each of those apps, not that it means
>
>>anything. Just interesting.
>>
>>That's one reason I listen eyes closed and double check with phase
>>cancellation tests and FFTs - I am
>>influenced creatively by the GUI to some degree. I actually like Cubase
>4's
>>GUI better than Nuendo 3.2,
>>though there are only slight visual differences (some workflow differences
>
>>are a definite improvement for me though).
>>
>>ProTools' GUI always made me want to write one dimensional soundtracks
in
>
>>mono for public utilities, accounting offices
>>or the IRS while reading my discreet systems analysis textbook - it was
>also
>>grey. ;-)
>>
>>Regards,
>>Dedric
>>
>>"LaMont" <jjdpro@ameritech.net> wrote in message news:458c82fd$1@linux...
>>>
>>> Dedric, my simple test is simple..
>>> Using the same audio interface, with the same stereo file..null-ed to
>
>>> zero..No
>>> eq, for fx. Master fader on zero..
>>>
>>> Nuendo, Pro-Tools -Mpowered(native)... yields a sonic difference that
>I
>>> have
>>> referenced before.. The sound coming from PT-M has a nice top end , where
>>> as Neundo has a nice flatter sound quality.
>>> Same audio interface. M-audio 410..Using Mackies & Blue-Sky pro monitors..
>>>
>>> Same test at the big room..PT-HD & Neundo Logic Audio(macG5-Dual) Using
>
>>> the
>>> 192 interface.
>>> Same results..But adding Logic audio's sound ..(Broad, thick)
>>>
>>> Somethings going on.
>>>
>>> Chucks post about how paris handles audio is a theory..Only Edmund can
>
>>> truly
>>> give us the goods on what's really what..
>>>
>>> I disagree that manufactuers don;t set out o put a sonic print on their
>
>>> products.
>>> I think they do.
>>>
>>> I have been fortunate to work on some digital mixers and I can tell you
>
>>> that
>>> each one has their own sound. The Sony Dmx-100 was modeled after SSL
4000g
>>> (like it's Big Brother).And you what? That board (Dmx-100) sound very
>warm
>>> and it's eq tries to behave and sound just like an SSL.. Unlike he Yamaha
>>> Dm2000(version 1.x) which has a very Clean, neutral sound..However, some
>>> complained that it was tooo Vanila and thus, Yamaha add a version 2.0
>
>>> which
>>> added Vintage type Eq's, modeled analog input gain saturation fx too
give
>>> the user a choice Btw Clean and Neutral vs sonic Character.
>>>
>>> So, if digital conoles can be given a sonic character, why not a software
>>> mixer?
>>> The truth is, there are some folks who want a neutral mixer and then
there
>>> are others who want a sonic footprint imparted. and these can be coded
>in
>>> the digital realm.
>>> The apllies with the manufactuers. They too have their vision on what
>They
>>> think and want their product to sound.
>>>
>>> I love reading on gearslutz the posts from Plugin developers and their
>
>>> interpretations
>>> and opinions about what makes their Neve 1073 Eq better and what goes
>into
>>> making their version sound like it does.. Each Developer has a different
>>> vision as to what the Neve 1073 should sound like. And yet they all sound
>>> good , but slightly different.
>>>
>>> You stated that you use Vegas. Well as you know, Vegas has a very generic
>>> sound..Just plain and simple. But, i bet you can tell the difference
>on
>>> your system when you play that same file in Neundo (No, fx, eq,
>>> null-edzerro)..
>>> ???
>>>
>>>
>>> "Dedric Terry" <dedric@echomg.com> wrote:
>>>>Lamont - what is the output chain you are using for each app when
>>>>comparing
>>>
>>>>the file in Nuendo
>>>>vs ProTools? On the same PC, I presume (and is this PT HD or M-Powered?)?
>>>>Since these can't use the same output driver, you would have to depend
>on
>>>
>>>>the D/A being
>>>>the same, but clocking will be different unless you have a master clock,
>>> and
>>>>both interfaces
>>>>are locking with the same accuracy. This was one of the issues that
came
>>> up
>>>>at Lynn Fuston's
>>>>D/A converter shootout - when do you lock to external clock and incur
>the
>>>
>>>>resulting jitter,
>>>>and when do you trust the internal clock - and if you do lock externally,
>>>
>>>>how good is the PLL
>>>>in the slave device? These issues can cause audible changes in the top
>>> end
>>>>that have nothing to do
>>>>with the software itself. If you say that PTHD through the same converter
>>>
>>>>output as Nuendo (via? RME?
>>>>Lynx?) using the same master clock, sounds different playing a single
>
>>>>audio
>>>
>>>>file, then I take your word
>>>>for it. I can't tell you why that is happening - only that an audible
>>>>difference really shouldn't happen due
>>>>to the software alone - not with a single audio file, esp. since I've
>
>>>>heard
>>>
>>>>and seen PTHD audio cancel with
>>>>native DAWs. Just passing a single 16 or 24 bit track down the buss
>to
>>> the
>>>>output driver should
>>>>be, and usually is, completely transparent, bit for bit.
>>>>
>>>>The same audio file played through the same converters should only sound
>>>
>>>>different if something in
>>>>the chain is different - be it clocking, gain or some degree of
>>>>unintended,
>>>
>>>>errant dsp processing. Every DAW should
>>>>pass a single audio file without altering a single bit. That's a basic
>
>>>>level
>>>
>>>>of accuracy we should always
>>>>expect of any DAW. If that accuracy isn't there, you can be sure a heavy
>>>
>>>>mix will be altered in ways you
>>>>didn't intend, even though you would end up mixing with that factor in
>
>>>>place
>>>
>>>>(e.g. you still mix for what
>>>>you want to hear regardless of what the platform does to each audio track
>>> or
>>>>channel).
>>>>
>>>>In fact you should be able to send a stereo audio track out SPDIF or
>>>>lightpipe to another DAW, record it
>>>>bring the recorded file back in, line them up to the first bit, and have
>>>
>>>>them cancel on and inverted phase
>>>>test. I did this with Nuendo and Cubase 4 on separate machines just
to
>>> be
>>>>sure my master clocking and
>>>>slave sync was accurate - it worked perfectly.
>>>>
>>>>Also be sure there isn't a variation in the gain even by 0.1 dB between
>>> the
>>>>two. There shouldn't
>>>>and I wouldn't expect there to be one. Also could PT be set for a
>>>>different
>>>
>>>>pan law? Shouldn't make a
>>>>difference even if comparing two mono panned files to their stereo
>>>>interleaved equivalent, but for sake
>>>>of completeness it's worth checking as well. A variation in the output
>>>
>>>>chain, be it drivers, audio card
>>>>card, or converters would be the most likely culprit here.
>>>>
>>>>The reason DAW manufacturers wouldn't add any sonic "character"
>>>>intentionally is that the
>>>>ultimate goal from day one with recording has been to accurately reproduce
>>>
>>>>what we hear.
>>>>We developed a musical penchant for sonic character because the hardware
>>>
>>>>just wasn't accurate,
>>>>and what it did often sent us down new creative paths - even if by force
>>> -
>>>>and we decided it was
>>>>preferred that way.
>>>>
>>>>Your point about what goes into the feature presets to sell synths is
>
>>>>right
>>>
>>>>for sure, but synths are about
>>>>character and getting that "perfect piano" or crystal clear bell pad,
>or
>>> fat
>>>>punchy bass without spending
>>>>a mint on development, adding 50G onboard sample libraries, or costing
>
>>>>$15k,
>>>
>>>>so what they
>>>>lack in actual synthesis capabilities, they make up with EQ and effects
>>> on
>>>>the output. That's been the case
>>>>for years, at least since we had effects on synths at least. But even
>
>>>>with
>>>
>>>>modern synths such as the Fantom,
>>>>Tritons, etc, which are great synths all around, of course the coolest,
>>>
>>>>widest and biggest patches
>>>>will make the biggest impression - so in come the EQs, limiters, comps,
>>>
>>>>reverbs, chorus, etc. The best
>>>>way to find out if a synth is really good is to bypass all effects and
>see
>>>
>>>>what happens. Most are pretty
>>>>good these days, but about half the time, there are presets that fall
>>>>completely flat in fx bypass.
>>>>
>>>>DAWs aren't designed to put a sonic fingerprint on a sound the way synths
>>>
>>>>are - they are designed
>>>>to *not* add anything - to pass through what we create as users, with
>no
>>>
>>>>alteration (or as little as possible)
>>>>beyond what we add with intentional processing (EQ, comps, etc).
>>>>Developers
>>>
>>>>would find no pride
>>>>in hearing that their DAW sounds anything different than whatever is
being
>>>
>>>>played back in it,
>>>>and the concept is contrary to what AES and IEEE proceedings on the issue
>>>
>>>>propose in general
>>>>digital audio discussions, white papers, etc.
>>>>
>>>>What ID ended up doing with Paris (at least from what I gather per Chuck's
>>>
>>>>findings - so correct me if I'm missing part of the equation Chuck),
>>>>is drop the track gain by 20dB or so, then added it back at the master
>
>>>>buss
>>>
>>>>to create the effect of headroom (probably
>>>>because the master buss is really summing on the card, and they have
more
>>>
>>>>headroom there than on the tracks
>>>>where native plugins might be used). I don't know if Paris passed 32-bit
>>>
>>>>float files to the EDS card, but sort of
>>>>doubt it. I think Chuck has clarified this at one point, but don't recall
>>>
>>>>the answer.
>>>>
>>>>Also what Paris did is use a greater bit depth on the hardware than
>>>>ProTools
>>>
>>>>did - at the time PT was just
>>>>bring Mix+ systems to market, or they had been out for a year or two
(if
>>> I
>>>>have my timeline right) - they
>>>>were 24-bit fixed all the way through. Logic and Cubase were native
DAWs,
>>>
>>>>but native was still too slow
>>>>to compete with hardware hybrids. Paris trumped them all by running
>>>>32-bit
>>>
>>>>float natively (not new really, but
>>>>better than sticking to 24-bit) and 56 or so bits in hardware instead
>of
>>>
>>>>going to Motorola DSPs at 24.
>>>>The onboard effects were also a step up from anything out there, so the
>>> demo
>>>>did sound good.
>>>>I don't recall which, but one of the demos, imho, wasn't so good (some
>>>>sloppy production and
>>>>vocals in spots, IIRC), so I only listened to it once. ;-)
>>>>
>>>>Coupled with the gain drop and buss makeup, this all gave it a "headroom"
>>> no
>>>>one else had. With very nice
>>>>onboard effects, Paris jumped ahead of anything else out there easily,
>and
>>>
>>>>still respectably holds its' own today
>>>>in that department.
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| Re: Paris Skins / examples 4 [message #76040 is a reply to message #76038] |
Wed, 15 November 2006 23:28 |
highmtn
Messages: 13
Registered: November 2006
|
Junior Member |
|
|
r />
>>>>
>>>>"LaMont " <jjdpro@ameritech.net> wrote in message news:458c14c0$1@linux...
>>>>>
>>>>> Dedric good post..
>>>>>
>>>>> However, I have PT-M-Powered/M-audio 410 interface for my laptop and
>it
>>>
>>>>> has
>>>>> that same sound (no eq, zero fader) that HD does. I know their use
the
>>>
>>>>> same
>>>>> 48 bit fix mixer. I load up the same file in Nuendo (no eq, zero
>>>>> fader)..results.
>>>>> different sonic character.
>>>>>
>>>>> PT having a top end touch..Nuendo, nice smooth(flat) sound. And I'm
>just
>>>>> taking about a stereo wav file nulled with no eq..nothing
>>>>> ..zilch..nada..
>>>>>
>>>>> Now, there are devices (keyboards, dum machines) on the market today
>
>>>>> that
>>>>> have a Master Buss Compressor and EQ set to on with the top end notched
>>>
>>>>> up.
>>>>> Why? because it gives their product an competitive advantageover the
>>>>> competition..
>>>>> Ex: Yahama's Motif ES, Akai's MPC 1000, 2500, Roland's Fantom.
>>>>>
>>>>> So, why would'nt a DAW manufactuer code in an extra (ooommf) to make
>
>>>>> their
>>>>> DAW sound better. Especially, given the "I hate Digtal Summing" crowd?
>>>
>>>>> And,
>>>>> If I'm a DAW manufactuer, what would give my product a sonic edge over
>>> the
>>>>> competition?
>>>>>
>>>>> We live in the "louder is better" audio world these days, so a DAW
that
>>>
>>>>> can
>>>>> catch my attention 'sonically" will probaly will get the sell. That's
>>> what
>>>>> happend to me back in 1997 when I heard Paris. I was floored!!! Still
>>> to
>>>>> this day, nothing has floored me like that "Road House Blues Demo"
I
>
>>>>> heard
>>>
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| Re: Paris Skins / examples 4 [message #76042 is a reply to message #76040] |
Thu, 16 November 2006 02:05 |
Dale
Messages: 77
Registered: September 2005
|
Member |
|
|
t;>"LaMont" <jjdpro@ameritech.net> wrote in message
>>>>>>news:458be8d5$1@linux...
>>>>>>>
>>>>>>> Okay...
>>>>>>> I guess what I'm saying is this:
>>>>>>>
>>>>>>> -Is it possible that diferent DAW manufactuers "code" their app
>>>>>>> differently
>>>>>>> for sound results.
>>>>>>
>>>>>>Of course it is *possible* to do this, but only if the DAW has a
>>>>>>specific
>>>>>
>>>>>>sound shaping purpose
>>>>>>beyond normal summing/mixing. Users talk about wanting developers
to
>>> add
>>>>> a
>>>>>>"Neve sound" or "API sound" option to summing engines,
>>>>>>but that's really impractical given the amount of dsp required to make
>>> a
>>>>>
>>>>>>decent emulation (with convolution, dynamic EQ functions,
>>>>>>etc). For sake of not eating up all cpu processing, that could likely
>>>
>>>>>>only
>>>>>
>>>>>>surface as is a built in EQ, which
>>>>>>no one wants universally in summing, and anyone can add at will already.
>>>>>>
>>>>>>So it hasn't happened yet and isn't likely to as it detours from the
>
>>>>>>basic
>>>>>
>>>>>>tenant of audio recording - recreate what comes in as
>>>>>>accurately as possible.
>>>>>>
>>>>>>What Digi did in recoding their summing engine was try to recover some
>>>>>>of the damage done by the 24-bit buss in Mix systems. Motorola 56k
dsps
>>>>> are
>>>>>>24-bit fixed point chips and I think
>>>>>>the new generation (321?) still is, but they use double words now for
>>>>>>48-bits). And though plugins could process at 48-bit by
>>>>>>doubling up and using upper and lower 24-bit words for 48-bit outputs,
>>> the
>>>>>
>>>>>>buss
>>>>>>between chips was 24-bits, so they had to dither to 24-bits after every
>>>>>
>>>>>>plugin. The mixer (if I recall correctly) also
>>>>>>had a 24-bit buss, so what Digi did is to add a dither stage to the
>
>>>>>>mixer
>>>>> to
>>>>>>prevent this
>>>>>>constant truncation o
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