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Hi Luggers,
Need a question answered: Which RME package will give me 8 analogue
I/O and at least 16 lightpipe I/O and maybe spdif at the same time?
George
________________________________________________________
www.utopiaparkwaymusic.com
Mac AND PC: The only way to fly 8-}
--- In logic-users@yahoogroups.com, "Maurits van de Kamp"
<maurits@b...> wrote:
> > --- In logic-users@yahoogroups.com, Nick Batzdorf
<recording@e...> wrote:
> opinions. We
> > agree that there IS without a doubt a sonic difference between
these
> programs.
>
> That wasn't actually what this discussion was about. (It was about the
> influence of a digital bus on the sound, not different DAWs.) But if
you
> have to bring it up again, here's the simple answer that has to be
> concluded from all the previous times the summing discussion came up:
>
I went back an re-traced the entire thread - and realized the error of my
post - sorry to
drive you nuts with that - However also in the mean time I made so calls to
people who I
know and have worked with in the past that are very, very knowledgeable
about this -
including being on the Board of AES to set standards and guidelines for all
this type of
stuff.
It is true, summing is summing. End of story - I need to apologise for
'starting this up
again' especially after heading the post down the wrong path. What I found
is that the
reallity is how ANY digital console/DAW or whatever compensates for
distortion(as bit
resolution declines, like in a fade situation) and the emense amount of data
and how it is
truncated or dithered. The reality is that a 24 bit file "plus"
another 24 bit file does not
yeild another 24 bit file - so where and how does the manufacturer start to
dither? It is
clear that early PT rigs merely truncated it(this was from them - not me).
Now that is no
longer an 'acceptable' means. It is also how the computer/program
compensates for
timing things in every aspect - things as simple as processing left and
right channels, how
it filters off the frequencies above and below limit of sampling frequency.
I mean geeze, I
just got three totally different opinions on how to handle the phaze issues
of different
digital consoles. I got "absolute phaze should be the determining
factor" then I got "no -
we must have a way to adjust/align sub-samples to make adjustments for
distortions(which are of course random)" I will leave that alone, it
made my head hurt and
I couldn't write it down fast enough.
I was also referenced to an article in this months (actually I am not sure
if that is March or
April) Mix magazine - it is by Steve St. Croix and it is actually on HD
video - I quote one of
the guys I spoke with "read that article, it is the same idea in audio,
it is all how you get
there (to the end)...." "Six guys all in the woods all needing to
get to out. Several different
paths all lead to the same end result, getting out."
As far as the FTP transfer and the loss of data - I wasn't comparing it to
the original
source. I just needed a common reference for these DAW's.
I would love, and I am not trying to be a jerk, to see/hear the results of
where " Logic and
PT have been mathematically proven to provide 100% identical summing results
when all
plugins andmix settings are the same." I will not be able to do this in
my current set up
for a while, and I am sure that a test like this would have to have the
results documented
and stored somewhere.
Thanks again, sorry for the anquish. Any further comment I will take to
AHOT.
Aaron Swihart
I am trying to connect my good old Unitor 8 ( not the MKII ! ) to a G5 via a
USB to
serial adapter... No luck so far... Does the Unitor 8 work under MacOs X ???
Thank you...
JY
> To summarize: I'm not saying I've witnessed 1+1 = 3, nor am I saying
> that the sum of two reciprocals doesn't equal zero. What I witnessed
> was a difference that five people described the same way but can't
> explain. I also witnessed pretty convincing proof that *audible* (not
> mathematical) cancellation of two files doesn't mean that they're
> identical.
this is a critical point. this comes up too often to be totally poo
poo'd by phase inversion and pedantic math rants. while we all
obviously already know about the subjectivity of perception, i remember
hearing the same math related arguments when digital was younger and
had problems that we were later able to find, test, and prove. i simply
think this shows that there are things going on in digital audio that
we don't yet have the right test and controls for. and i used to side
with the math folks, saying all those luddites who claimed vinyl was
better were nostalgic old fools. that is until i found out that their
complaints about digital audio were later proven to be correct.
i for one am glad it's improved, with better clocking and conversion
and more accurate math digital is growing up, but i am in the business
of trusting my ears. and anything else is somewhat irrelevant. even
though i am using a daw, i'm using my ears more. the theory is
interesting, but the long posts about math aren't totally compelling
anymore as we've been through these arguments earlier on with digital
audio and they didn't pan out at the time because they didn't take into
account the fact that the wrong tests were being used and that people
with great ears tended to say the same thing.
i've heard arguments about sonic differences from too many people i do
trust to ignore. and while i am not interested in perpetuating
untruthful information on digital audio, i say the jury is still out on
what is causing us to hear differences. and i don't think that it's all
some psychological hoo haa. i think we should be curious as to if there
are differences there somehow that we haven't been able to account for
yet.
cheers,
paul
pounce international
http://marsh.prosoundweb.com/ - keyboard forum moderator
Hi group,
> Maurits van de Kamp <maurits@...> replied & wrote :
> Subject: Re: Totalmix vs CueMix
>
>> . . . files
>> canceled each other out audibly . . .
> In that case the sound is positively absolutely indisputably identical.
>
>> Independently, we all wrote down the same descriptions of what we
>> heard. It was very subtle but real . . . . this proves to me that .
>> . .
>> something else is going on to make them sound different
> The only thing (phase cancellation) doesn't reveal is what's going on
> in the mind of the
> listener.
>> some fool out there is going to say that this
>> was all psychological
The solution may be far simpler. I was trying to replicate a mix a few
years ago -
had all the fader levels, tone settings etc. all set up the same. I
played the original
mix from DAT, and the re-setup mix from the desk. I A-B-ed them with
both running
from the same start-point. Switching from one to the other you got a
continuous song,
but they sounded different. (Ignore the DAT being ever-so-slightly
different on playback,
this was a LOT different it seemed to me.)
The solution was not to A-B two synchronous tracks.
The original set up would look like this :
MIX 1 A B C D E F G H
MIX 2 A B C D E F G H
so you listened to the CAPS part of MIX 1, and the CAPS part of mix 2:
MIX 1 A b C d E f G h
MIX 2 a B c D e F g H
I then time slipped one mix so I heard part "A" of mix 1; and then
part "A" of mix 2 immediately. Thus:
MIX 1 A b C d E f G h
MIX 2 A b C d E f G h
The mixes now sounded identical. It's especially noticeable when
switching
in the middle of a smooth piece of audio - a chord perhaps - no change,
assuming your switch is silent. The time gap was less than 4 seconds.
Anything much longer and my brain / ears could not compare with any
accuracy at all. Maybe this is what's happening?
I concur with Maurits - identical digital == same sound, exactly, every
time.
Regards,
Mike.
from Mike Levon
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www.holyground.co.uk
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From: "George Leger III" <george3@...>
> Hi Luggers,
> Need a question answered: Which RME package will give me 8 analogue
> I/O and at least 16 lightpipe I/O and maybe spdif at the same time?
>
> George
RME Fireface.
Regards:
Darth
"f-erenc szabo" <zerobeat@...> wrote
> Michael Levine <mlmusic@...> previously wrote:
>> ivan's recent post about reverse playback made me think about a
>> ProTools function I really miss in Logic
> This technique isn't as hard as you suggest, although I certainly
> wish Logic had varispeed... a wish I've had since 1993.
Is it possible to sync Logic to a SMPTE-like time code which can be
manipulated? Even better, can a simple MTP box be built which
would allow variable speed output? I'd buy one straight away!
Regards,
Mike.
from Mike Levon
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H O L Y G R O U N D R E C O R D S
EnglandÕs 1st independent record label
www.holyground.co.uk
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>
> Hi Luggers,
> Need a question answered: Which RME package will give me 8 analogue
> I/O and at least 16 lightpipe I/O and maybe spdif at the same time?
>
> George
George -
I'm using a fireface with a similar configuration BUT without the high
lightpipe requirements. BTW - you'll need connectors/convertors from them if
you do this. The pre-amps are pretty good and the system configures like a
dream. I ended up going to an AES-16 by Lynx for the other I/O and
connecting the Fireface through an AES connector along with some Liquid
Channel Pres. You might consider something similar where you route the
lightpipes into an outboard box and then use the AES for a matrix/router.
Brian
In a message dated 3/31/05 6:17:06 AM, rory_horan@... writes:
> Hi all, I'm new to computers, newer to Mac's and even newer to Logic 6
> so please forgive the very basic Q's. I started 30hrs ago. I'm running
> Logic 6 on a G5 with Roxio Toast. I've mixed live band recordings to
> two tracks but don't know how to burn it. Do i need to save or export
> it in a particular format? Or am i way off the mark?!
> Reply's are much appreciated.
>
You need to bounce it to your hard drive as a Stereo File (typically SDII)
using Logics' Stereo outputs 1/2. The resulting file is the one you drag
into
Toast and burn as an Audio file.
Back on the subject of the TotalMix in-the-hardware summing sounding
better (or at least different) to Logic's summing, here's my take on why:
Logic:
32-bit floating point math for summing, ie 24 bits of precision at any
sample value.
TotalMix:
RME CoreAudio driver is supplied 32-bit floating point data by Logic,
and the driver is responsible for converting this to the format
required by the hardware - RME uses 40-bit FIXED POINT math for
summing, ie precision depends on actual sample value. The precision is
greater than 32-bit float down to -96dBFS (low 24 bits used, high 16
bits zeroed), less than 32-bit float below that.
In other words, there's mathematical room for the two summing
processes to produce different results.
John Pitcairn
-----------------------------------------------------------------
LC Xview software LC display, LC Xmu Logic Control emulation,
Logic environments & stuff: http://www.opuslocus.com
-----------------------------------------------------------------
Op Thursday 31 March 2005 20:07, schreef paul kavicky:
> and i don't think that it's all
> some psychological hoo haa.
As long as you close yourself off to parts of the scope like this, don't
expect a sensible discussion or a usable result.
The chain of events is this:
Digital domain -> D/A conversion -> Analog domain -> ears ->
brain
The digital domain is perfectly calculatable and repeatable, bar
transmission
errors etc. (Note that all examples so far of digital
"imperfection" where
not in this stage, but the next one!!)
D/A conversion: Pretty critical, a lot known but it is a never ending
science.
Analog domain: Influenced by anything that makes noise and anything that
induces signal to the path.
Ears: Idem
Brain: Almost totally unknown. And this is where we do our measurements.
Now anyone who searches for mysteries in the first domain, and discards any
psychological influence as "hoo haa" has absolutely no clue what
he's talking
about.
Maurits.
Op Friday 01 April 2005 00:06, schreef John Pitcairn:
> In other words, there's mathematical room for the two summing
> processes to produce different results.
Agreed, in this case there is a clear mathematical difference. (Not that
anyone would really hear that) ;o)
Maurits.
From: Mike Levon <mike@...>
>The solution was not to A-B two synchronous tracks.
>The original set up would look like this :
>
>MIX 1 A B C D E F G H
>MIX 2 A B C D E F G H
>
>so you listened to the CAPS part of MIX 1, and the CAPS part of mix 2:
>
>MIX 1 A b C d E f G h
>MIX 2 a B c D e F g H
>
>I then time slipped one mix so I heard part "A" of mix 1; and
then
>part "A" of mix 2 immediately. Thus:
>
>MIX 1 A b C d E f G h
>MIX 2 A b C d E f G h
>
>The mixes now sounded identical. It's especially noticeable when
>switching
>in the middle of a smooth piece of audio - a chord perhaps - no change,
>assuming your switch is silent. The time gap was less than 4 seconds.
>Anything much longer and my brain / ears could not compare with any
>accuracy at all. Maybe this is what's happening?
Nope. We were looping the same section.
>I concur with Maurits - identical digital == same sound, exactly, every
>time.
That's what they call *stating the bleeding obvious* in the art world. :)
The actual equation to describe this heavy scientific experiment is:
*Should be* identical digital == different sound every time.
Therefore *should be* ‚ *is*.
--
Nick Batzdorf - editor
Virtual Instruments magazine
(the world of softsynths and samplers)
www.Virtualinstrumentsmag.com
1-877 VImagzn (846-2496), 818/905-9101, cell 590-9101
--- In logic-users@yahoogroups.com, Maurits van de Kamp <maurits@b...>
wrote:
> Agreed, in this case there is a clear mathematical difference. (Not
> that anyone would really hear that) ;o)
Well, I don't know about that. The difference I hear is that summing
in TotalMix sounds somewhat more open & transparent, with a
better-defined stereo image. It's not a massive difference, ie nothing
like the difference in going from 16- to 24-bit, or even from
Multiface to Fireface converter quality, but it's enough that I'd
think it worth setting up and using for the final master on some material.
John Pitcairn
-----------------------------------------------------------------
LC Xview software LC display, LC Xmu Logic Control emulation,
Logic environments & stuff: http://www.opuslocus.com
-----------------------------------------------------------------
Hi All,
After having done a couple of long sessions I have found two annoying
bugs; that I am wondering if anyone can corroborate.
The first time you set the record path, everything is fine. You hit
record and get the little progress bar counting down the remaining
time, as expected. As soon as you switch screen sets though, two
strange things happen. First, the recording progress bar no longer
appears. There's no way to get it back without rebooting Logic. And
second, when you are in the set record path dialogue box, hitting
return or enter no longer works to OK a dialogue box. It does work at
the beginning, but disappears when the progress bar disappears.
The second thing; L7 doesn't seem to be happy when I have to open many
songs during a session. Sure enough after about the seventh or eighth
song, Logic crashes upon trying to open subsequent songs. When I
restart logic, the count restarts again, but it just won't open beyond
a certain number of songs. And it doesn't matter wether they are opened
via double clicking in the Finder, or by navigating through the
"open"
hierarchy.
Anyone else have this???
--------
Eli Krantzberg
http://www.nightshiftorchestra.com
Almat Productions
On Mar 31, 2005, at 1:24 PM, Darth wrote:
>
>
> From: "George Leger III" <george3@...>
>> Hi Luggers,
>> Need a question answered: Which RME package will give me 8 analogue
>> I/O and at least 16 lightpipe I/O and maybe spdif at the same time?
>>
>> George
>
> RME Fireface.
> Regards:
> Darth
>
>
RME Fireface or RME Multi face (No mic Pre's though)
Ciao
Gio
> Hi Luggers,
> Need a question answered: Which RME package will give me 8 analogue
> I/O and at least 16 lightpipe I/O and maybe spdif at the same time?
>
Hi Georggers
The FireFace800 will give you this out of the box.
For a cheaper version, you could go for the Hammerfall DSP (HDSP) 9652 -
this has 3 sets of ADAT I/O and an SPDIF I/O. You can attached an Alesis
AI-3 or Behringer ADA8000 to one of the ADAT I/Os to provide the analogue
side.
I can strongly recommend the RME interfaces - extremely good performance!
Best wishes,
--
The best of times to you all...
Pants
On 31/03/2005, at 6:16 AM, Brian Mikiten wrote:
> I'm getting an odd effect where the entire mix shifts to the left
> channel
> using PSP and then back again over time. Do you see this as well? I'm
> only
> using it on the final mix channels.
It's called phasing and it's caused by latency.
:::::::::::::::::::::::::::::::::::::::::::::::::::
Paul Najar
Jaminajar Music Production
www.jaminajar.com
On 31/03/2005, at 5:05 AM, Peter Duemmler wrote:
>
> Logic´s native mixer only has pre-pan sends... (Pro, eh?).
> Send the tracks to the busses with the Vintage Warmers through their
> outputs, not through sends.
> Or use the I/O Helper plug-in for the panning in the tracks (THAT
> affects the aux sends), but be careful, then there´s NO
pan-law...
You will also run into phase issues caused by latency doing it the way
you describe because there is not plugin delay compensation on busses
yet so if you're mixing dry tracks with bus processing at the outputs
the signals will arrive at different times.
A more phase coherent way would be to not have any dry signal and
control the amount of the Vintage warmer you want in the plugin itself.
:::::::::::::::::::::::::::::::::::::::::::::::::::
Paul Najar
Jaminajar Music Production
www.jaminajar.com
On 01/04/2005, at 12:50 AM, Joe Frey wrote:
>
>
> All, Is there an easy way to do a pr-pr-pr-pre-echo on vocals? The ones
> that are sort of a ramp up to the beginning of the vocal line? TIA.
> JoeF
Record your vocal through a normal reverb to a track. Reverse it in the
sample editor and move the resultant audio earlier in the track till
you get what you want.
:::::::::::::::::::::::::::::::::::::::::::::::::::
Paul Najar
Jaminajar Music Production
www.jaminajar.com
Hey can anybody help out here. Im having troubles when Im playing midi
tracks.
My setup:
Logic Platinium 5.5.0
Lots of plugins
Soundcard: Audiophile 2496
I tried to set my buffer size up, but it didnt solve the problem.
Anybody
Thanks
Mikkel
On Mar 31, 2005, at 1:29 PM, logic-users@yahoogroups.com wrote:
> Re: Saving audio from Logic to burn with Roxio Toast
>
Stereo File (typically SDII)
Not so much anymore. I am weaning myself from this file type as it is
by far the least compatable of the bunch. I am standardizing on AIFF
due to it's interoperability. AES standards call for .WAVs on FAT32
volumes which are not compatable with Macintosh in my experience.
Hi to all.
I got Logic pro 6.4.1 (OS 9.2, G4 466, Digi 001 board, MT4 MIDI interface,
120 G
Firewire HD and 20 G SCSI HD).
With logic plat 4.3.7.I have done all the time sysex bulkdump from my sound
modules to the sequencer without any problem, but with logic pro I cant do
it
because the data its wrong recorded ( I can see it on the event edit window
were the
sysex data from my roland jv 2080 its mixed with data from other
manufactures
sound modules ) when i try to send this data to the jv2080 it give me an
error on the
sound module display . When I do it on a empty song (just the deafult song
not my
autoload song) the sysex its recorded correctly, but all get wrong when the
song its
filled with sequences, and plugings. It seems that the problem with the
sysex
recording began when the song have a lot of data. (Plug-ins, midi data, an
audio)
Any idea about that? Can you please reproduce it on your systems?
>-----Original Message-----
>From: Joe Frey [mailto:joe.frey@...]
>Sent: Thursday, March 31, 2005 6:51 AM
>To: logic-users@yahoogroups.com
>Subject: [LUG] Easyway to do a pre echo on a vocal?
>
>
>
>All, Is there an easy way to do a pr-pr-pr-pre-echo on vocals?
>The ones that are sort of a ramp up to the beginning of the
>vocal line? TIA. JoeF
Joe,
Reverse the track you want the effect on, set up your reverb or delay
(any time based effect works), Bounce the output of the effect to
another track, reverse original and the bounced effect(forward again),
mix to taste. Mmmmmmm!
You can bounce the original and effect pre-mixed, but having them
separate will allow you more freedom to dial the effect in.
b o y d p o s t </> s o u n d d e s i g n </> c o m p o s i t i
o n
</> s u r r e a l s o f t w a r e
IS there a way of bouncing QT instruments (as they dont go to an
output) or do you have to do it in external software to Logic?
cheers
Tom
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