PlanetZ forums - scopeusers.com

Hi,
I am not really sure about 'summing with scope' really means,......is that like grouping all the tracks in the sequencer app and route each individual group channel to stereo tracks in the scope mixer?

hc.

the grouping isn't essential, it's about 'summing' together the different tracks (yes, and then also groups) to the main out of the mixer.
Scope is said to do that much better than native mixers, as Scope is realtime calculating in the DSPs as opposed to native summing where buffering is involved.
Correct me if I don't describe it correctly guys

I always wondered if there's a way to prove that. I've done a couple of test before, but I didn't hear substantial differences. I must admit that the acoustic of my studio is crappy. So I belive that the sum is better, but I almost take it as a principle.

Has anyone ever done a "scientific" test about this?

yes it's true.
measurable data would be hard to come by. for some people it wouldn't matter. i know people who think that their $150 cassette 4 track machines sound pro.

you can't do much of a scientific test about it, since there is no formal/absolute description of what a perfet mix should sound like, or be done. the best you can do is compare the two mixes (native vs scope.)

you'll want to use a mix with everything (master, channels) set at unity gain to make sure the gain stages doesn't affect/colour the sound, and then check and make sure that the two mixes have the same loudness to make sure you don't perceive a mix as sounding better just because it's louder.

i guess you could also run some analysis on the bitdepth of the mixing/gain stages of the mixing engines, but that won't garantee that higher bit-depth will sound better since it also depends on how it's programmed.

garyb wrote:... i know people who think that their $150 cassette 4 track machines sound pro.

I'd taken a Tascam 246 for that amount yesterday, if the inputs had been symmetric...

I found a background info in a market survey about analog summing highly instructive - I haven't heared the (unlabeled) audio examples yet, but it made me reconsider my own opinion.

a digital stream in stereo at CD rate has 88.2k individual values per second to describe the audio (2x44.1k).
Looks (and reads) much, yet the (say) 16 source tracks contain 1.4 million single data events, which are mangled in some (hopefully smart) mathematical way to fit into the available 88.2 output events...

I admit I never saw it that way - beyond it's original topic (as a review intro) it made clear why differences in software mix engines (have to) exist.
It's not so much about the limits of absolute values or rounding the 22nd digit somewhere - it's simply a matter of 'density'.

A perfect analog summing device wouldn't loose any of this information due to it's infinite resolution.
The difference is clearly audible - if it's perceived better is an entirely different question, which they passed as a poll to the readers.

that's why I was interested in the afformentioned multitracker, it has a pretty good anlog performance and $150 would have been worth trying.

cheers, Tom

see?
actually, the 246 was a good machine, not professional, but good. it was much more than $150 originally...more like $250, 1980's $250......

without taking this too far from the topic...
yes, considering it's specs for a 'full' production the 246 is indeed not 'professional'
on the other hand it's a pretty versatile switchboard for monitoring and afaik it was quite expensive in monetary value it's days, yet very affordable regarding it's functionality (in that context it has something in common with Scope...)

I consider the specs from the manual 'honest' values opposed to what's published today, where often simply the figures of individual parts are reprinted from the circuit manufacturer's charts.
Can't help it, but I'd rather trust in this 20-pounds veteran than any gear the same amount could buy me today

I would have fed it 3 stereo busses via the A16 - we could as well use any quality analog mixer instead (for the hypothesis), and record the stereo sum back in Scope.
Would it benefit from the analog processing or not ?

Scope is known for it's excellent summing - would there be a difference in one big mix versus 3 submixes summed up analog ?
Even with a provocantly cheap piece of gear ? (it deals with a 'solid' level already, so the lowest SNR figures wouldn't matter much)

cheers, Tom

To me, these "scientific" summing tests are quite useless and might even yield the same resuls everywhere.
Scope summing superiority comes into play when you actually do something.
If Scope as an environment only was superior because its tools would help you mix better, then I say it would be totally worth it.

It's good to actually hear what's going on.

I admit I never saw it that way - beyond it's original topic (as a review intro) it made clear why differences in software mix engines (have to) exist.
It's not so much about the limits of absolute values or rounding the 22nd digit somewhere - it's simply a matter of 'density'.

For software based summing this explanation attempt is highly questionable. 44.1 kHz sampling rate is 44.1 kHz and stays 44.1 kHz with any software (jitter taken aside), regardless how much sources you add. There is no "fit inbetween" as well as there is none in the analog world. The sources are summed and there's no one sitting to decide which value is taken or not. The example sounds like saying 1+1 is 4 because there are two 1 at the same time.
Another debate of course is about levels, where multiplication & division takes place ...

The difference between native & scope mixing is, that most (if not all) popular sequencers use 32 bit float, while scope uses 32 bit integer for calculation.
The advantage of float calc is that you can boost the signal above zero dB without having clip distortion so you have a virtually endless headroom. Basically 32 bit float is 24 bit int with an 8 bit mantissa. It is very easy to create really hot mixes, but with some experience you can distinct between float and integer mixes because all heavy driven float mixes have a similar sound.
32 bit int on the other hand uses the whole bitdepth resulting in clip distortion above zero dB, but also in a better-to-judge sound as the mantissa thingie changes the sound (i.e. transient response and depth) of the signal.

As voidar said 32bit integer through the whole signal path comes into the game as soon as you actually do something ..

Thanks to everyone, really interesting!!!

Wolf, about the bit depth, it's the same for ASIO? I mean, could I say the same for the asio interface and use these reasons to chose one instead another?

well, for discussion's sake I'd like to stick with the (unscientific) 'density' argument for a while and contradict Wolf's theory.
There's probably few doubt that an 'instrument' in an individual track is much more present than in a mix (of say 16)
It's also unquestioned that our (human) ability to share our attention is somewhat limited, so the 'full' perceiption of every detail in an imaginary, loss-less mix may be restricted to a very small group of extraordinarily talented individuals.
Let alone this would be an entirely statistic driven process and not a creative one...

Yet there seems to be more in the algorithms than just adding numbers.
I've been through the afforementioned sound examples blindly - and there was a clear winner in punch and transparancy, which I found a bit over the top, so I picked a less impressive one.

Now comes the funny part: as the article stated it was about 10 analog summers versus a Protools HD mix (to represent the digital domain), I was convinced the most outstanding must be ProTools, as it was so much different from all the others.
Bloody wrong - that was the AMS Neve 8816, and my personal favourite turned out to be the Dangerous Minds 2 Bus.

Protools was among the 'a bit muddy, nothing special' group.
Too bad there wasn't a Scope mix for reference...

addition: not sure if all files have been prepared appropriately - after looking at the waveforms the Neve seems a bit too impressive - intentionally cheating ?

cheers, Tom

garyb wrote:yes it's true.
measurable data would be hard to come by. for some people it wouldn't matter. i know people who think that their $150 cassette 4 track machines sound pro.

I have heard some of Neil's 88.2kHZ mixes ITB vs 4 x stereo stems mixed in Cubase and lightpiped to his Scope rig. There was an audible difference. Improved detail/depth and better stereo imaging to my ears with the Scope summing. Others heard it too. This was done on the Paris forum and I assure you that the engineers there are total summing fanatics. They can hear the summing of multiple hummingbird farts at 1000 meters and tell you whether the hummingbirds sucked organic nectar or refined sugar water. This platform is starting to develop a following over at my old home on the Paris forum. This is quite a compliment to Scope, IMHO, as Paris has the best digital summing I've ever heard.

Hi,
imho, floatingpoint summing is nice as long as singal scales are almost identical.
(For 32 bit floatingpint signals, you have a audiosignal -24 bit and a scale - 8bit)
If you are summing floating-signals with different scales (loud and quiet signals), fixed point seems to be better.


Cahrly Steinberg wrotes:

Words by Charlie Steinberg (about mix engine technology and sound)
-------------------------------------
The 'Logic sounds better' thing is not new and always comes down to pan
law
beeing 0 dB in Logic, as opposed to the defaults of -3dB in Nuendo and
-6 dB
in Cubase. this also explains why 'Nuendo sounds better than Cubase',
because
higher panning law makes centered signals (such as bass and bass drum)
sound
louder (= 'better') and 'more punchy'. one can argue either methods
endlessly,
i think the -3 dB default of Nuendo is a good compromise because most
desks
appear to use this. however punchy your mix sounds is entirely up to
you,
regardless of the panning law you choose; and if your mix sounds better
when
you switch to a different panning law, then you should think twice
about how
you mix in general at least, in Nuendo and Cubase, you can adjust
the
panning law so to fit your 'taste'.
the idea behind -6 dB panning law is that when you would put your 2
stereo
speakers into 1 box, wherever you set the pan, there wouldn't be any
difference
in volume (loudness), and this becomes obvious when you listen to the
mix in
Mono, because the sum of left and right when panning a channel is
always equal.
this is not true for any other panning law than 6dB.
using 0 dB on the other hand, when you are all to the left or right,
the sum
will be 0dB, but when you pan to the middle it will become +6dB.
personally this
feels plain wrong to me (despite the math, just listening to it) and
makes me
insecure as to how much panning to apply, but it is really a matter of
taste
(and what you are used to; if you come from Logic, you'd probably be
much better
off choosing 0 dB). as said above, -3dB appears to be a good
compromise, but
whichever law you choose, your ears are the one and only final resort
to jugde
how 'punchy' you want your mix to be.
Goobers perfect summing bus. (simplified example)
Input from channel 1
Input from channel 2 =3
Input from channel 4 =1
Add inputs together...
Output= 14
Now, how could any data massaging algorhythm improve on this? All
integer summing
busses should sound exactly the same until ditherd.
The only problem is that the VST engine uses a 32bit float, meaning
that mixing large and
small numbers could cause smaller signals to be lost due to mantissa
priority confusion!
sorry but this needs explanation...
the nominal range of VST floating point is -1.0 to + 1.0.
any signal going in or out the VST system is converted to this range.
as 'external' signals are always integer currently, they are limited to
maximum numbers (both positive or negative). so whatever
the integer format (16 or 24 bit), the maximum and minimum values that
those bits can represent are + 1.0 and -1.0 expressed in floating point
terms.
in a 32 bit floating point value, this will cover the mantissa
of 24 bit, so it is always *at least* as precise as a 24 bit integer
value (so it is at least 'as good'). the exponent is always 0.
considering a signal of 0dB peak, this means the largest values will
be + 1.0 and - 1.0 resp. smaller signals will go down to -144 dB with
100% precision, because the mantissa is 24 bits large (also there
is a sign bit). going below -144 dB is also possible, where the
farther you go, the less precise will the numbers become; not that
this could ever have any impact, because a) there are currently and
commonly no converters to deal with more than 24 bits nor b) is there
any analog equipment that i know which can deal with dynamic ranges
of over 144 dB anyways.
only when you drive signals above 0dB will the exponent get
changed. in this case the unbeleiveable headroom of the IEEE floating
point mechanism becomes active. it works like a VCA: when
you exceed 0 dB, the control voltage will rise, where below 0dB it
will be constant (actually it changes to a negative 'voltage' when
you get below -144 dB). as with the lower numbers, at a certain point,
there is a loss of precision, but this is very small unless you go
crazy with your signals.
this gets us to

This is where a small number dissapears when added to a larger one as
the integer has too
few bits to hold both values.
I'm assuming they use a 24bit number with 7 bit mantissa and sign.
It would have to a fairly extreme situation for this to happen, and I'm
sure it could not be
audible, but who knows? Using a floating point number may give better
results for sperate
channels in theory, but could cause problems when many are added.
'extreme situation', you can say that. for a signal of -140 dB
to be masked, you would probably need some +140 dB on the other end.
not only will this never happen in anybodys' studio (unless a plug
goes mad , but even if it happened would you not really be able
to hear that your -140 dB signal is gone, because your speaker mebrane
has arrived at the opposite wall meanwhile.
the bottom line of all of this is:
- first off we are dealing with numbers. we don't want any
colorizing so we 'only' need to be precise. maths can not be
cheated. if any digital system sounds 'better' or 'worse' than
another when playing back an unaltered signal and using the same
hardware, either there is some bug, or there is colorization
applied by intention.
- second, the nominal range of 32 bit floats is always 100% precise
within 144 dB (or 24 bits) (i think it's even 25 bits because we
have an extra sign bit but it's late and i'm too lazy to look it
up).
- third, there is an incredible headroom with floats as opposed to
integer values. there is loss of precision however when you exceed
the range of the mantissa (those 24 bits), but that only happens when
you exceed that range big time and even then, the change of the
numbers that result is in the range of 1 bit which is again inaudible.
and when you bring it all down again to the range that your converter
can handle, the 'imprecise bits' usually fall outside that range
anyways.
- fourth and finally, there may be problems when *converting* those
floating point numbers from and to integer numbers. we are currently
double-checking this so to make sure that when you render to file
(export audio), the outcome is exact. but even if we dumb programmers
create errors in this area would you not be able to tell the difference
(at least with 24 bit files), because again the error would be in the
range of 1 bit, which is measureable but no equipment can reproduce
it anyways. but yes, we certainly need to be 100% correct for your
convenience and we will.
that gets us to 'bus summing'. as somebody pointed out
earlier, this is in fact nothing but adding values; and this adding
operation is just as precise as your pocket (or scientific) calculator.
would you trust it?
BTW, Ksmith, did you hear about the test when the same tracks were
summed in a PT Rig
and the Oxford? The results could be put out of phase giving zero.
[ http://www.virtualstudio.org/VS_AUTO_DI ... -02-15/117 ]http://
www.virtualstudio.org/VS_AUTO_DIGEST/00-02-15/117
I think both PT and the sony use integer busses though.
Either they both use the same 'summing bus' or we are all caught up in
some invented
problem.
possibly
in theory, you should not overdrive the channels more than +6dB and of
course
keep your master at 0 dB. this puts you on the 'mathematically prooven
exact
side of the fence', so to speak. in practice, even if you drive many
channels
much into the red, it doesn't make for more than an inaudible (albeit
possibly
measureable) difference.
actually, floats behave a bit like analog equipment: if you overdrive
it,
it does something on its own. but for what a float does then to become
perceivable, you would have to write a plug which drives your signal up
a few hundred dB's....certainly more than your analog tape recorder
at +6dB, which is clearly audible.
charlie

Lima wrote:Thanks to everyone, really interesting!!!

Wolf, about the bit depth, it's the same for ASIO? I mean, could I say the same for the asio interface and use these reasons to chose one instead another?

Read my posts on http://www.planetz.com/phpBB2/viewtopic.php?t=20251

For full 32-bit int integrity you should really use ASIO1 modules, be it FLT or 32 depending on what your ASIO-host wants to see.

For representing a single audio-stream, 32-bit float will retain all the information from a 32-bit int word. When summing two of these I would rather use 32-bit int or 64-bit float.

AndreD wrote:Hi,
imho, floatingpoint summing is nice as long as singal scales are almost identical.
(For 32 bit floatingpint signals, you have a audiosignal -24 bit and a scale - 8bit)
If you are summing floating-signals with different scales (loud and quiet signals), fixed point seems to be better.


Cahrly Steinberg wrotes:

Words by Charlie Steinberg (about mix engine technology and sound)
-------------------------------------
The 'Logic sounds better' thing is not new and always comes down to pan
law
beeing 0 dB in Logic, as opposed to the defaults of -3dB in Nuendo and
-6 dB
in Cubase. this also explains why 'Nuendo sounds better than Cubase',
because
higher panning law makes centered signals (such as bass and bass drum)
sound
louder (= 'better') and 'more punchy'. one can argue either methods
endlessly,
i think the -3 dB default of Nuendo is a good compromise because most
desks
appear to use this. however punchy your mix sounds is entirely up to
you,
regardless of the panning law you choose; and if your mix sounds better
when
you switch to a different panning law, then you should think twice
about how
you mix in general at least, in Nuendo and Cubase, you can adjust
the
panning law so to fit your 'taste'.
the idea behind -6 dB panning law is that when you would put your 2
stereo
speakers into 1 box, wherever you set the pan, there wouldn't be any
difference
in volume (loudness), and this becomes obvious when you listen to the
mix in
Mono, because the sum of left and right when panning a channel is
always equal.
this is not true for any other panning law than 6dB.
using 0 dB on the other hand, when you are all to the left or right,
the sum
will be 0dB, but when you pan to the middle it will become +6dB.
personally this
feels plain wrong to me (despite the math, just listening to it) and
makes me
insecure as to how much panning to apply, but it is really a matter of
taste
(and what you are used to; if you come from Logic, you'd probably be
much better
off choosing 0 dB). as said above, -3dB appears to be a good
compromise, but
whichever law you choose, your ears are the one and only final resort
to jugde
how 'punchy' you want your mix to be.
Goobers perfect summing bus. (simplified example)
Input from channel 1
Input from channel 2 =3
Input from channel 4 =1
Add inputs together...
Output= 14
Now, how could any data massaging algorhythm improve on this? All
integer summing
busses should sound exactly the same until ditherd.
The only problem is that the VST engine uses a 32bit float, meaning
that mixing large and
small numbers could cause smaller signals to be lost due to mantissa
priority confusion!
sorry but this needs explanation...
the nominal range of VST floating point is -1.0 to + 1.0.
any signal going in or out the VST system is converted to this range.
as 'external' signals are always integer currently, they are limited to
maximum numbers (both positive or negative). so whatever
the integer format (16 or 24 bit), the maximum and minimum values that
those bits can represent are + 1.0 and -1.0 expressed in floating point
terms.
in a 32 bit floating point value, this will cover the mantissa
of 24 bit, so it is always *at least* as precise as a 24 bit integer
value (so it is at least 'as good'). the exponent is always 0.
considering a signal of 0dB peak, this means the largest values will
be + 1.0 and - 1.0 resp. smaller signals will go down to -144 dB with
100% precision, because the mantissa is 24 bits large (also there
is a sign bit). going below -144 dB is also possible, where the
farther you go, the less precise will the numbers become; not that
this could ever have any impact, because a) there are currently and
commonly no converters to deal with more than 24 bits nor b) is there
any analog equipment that i know which can deal with dynamic ranges
of over 144 dB anyways.
only when you drive signals above 0dB will the exponent get
changed. in this case the unbeleiveable headroom of the IEEE floating
point mechanism becomes active. it works like a VCA: when
you exceed 0 dB, the control voltage will rise, where below 0dB it
will be constant (actually it changes to a negative 'voltage' when
you get below -144 dB). as with the lower numbers, at a certain point,
there is a loss of precision, but this is very small unless you go
crazy with your signals.
this gets us to

This is where a small number dissapears when added to a larger one as
the integer has too
few bits to hold both values.
I'm assuming they use a 24bit number with 7 bit mantissa and sign.
It would have to a fairly extreme situation for this to happen, and I'm
sure it could not be
audible, but who knows? Using a floating point number may give better
results for sperate
channels in theory, but could cause problems when many are added.
'extreme situation', you can say that. for a signal of -140 dB
to be masked, you would probably need some +140 dB on the other end.
not only will this never happen in anybodys' studio (unless a plug
goes mad , but even if it happened would you not really be able
to hear that your -140 dB signal is gone, because your speaker mebrane
has arrived at the opposite wall meanwhile.
the bottom line of all of this is:
- first off we are dealing with numbers. we don't want any
colorizing so we 'only' need to be precise. maths can not be
cheated. if any digital system sounds 'better' or 'worse' than
another when playing back an unaltered signal and using the same
hardware, either there is some bug, or there is colorization
applied by intention.
- second, the nominal range of 32 bit floats is always 100% precise
within 144 dB (or 24 bits) (i think it's even 25 bits because we
have an extra sign bit but it's late and i'm too lazy to look it
up).
- third, there is an incredible headroom with floats as opposed to
integer values. there is loss of precision however when you exceed
the range of the mantissa (those 24 bits), but that only happens when
you exceed that range big time and even then, the change of the
numbers that result is in the range of 1 bit which is again inaudible.
and when you bring it all down again to the range that your converter
can handle, the 'imprecise bits' usually fall outside that range
anyways.
- fourth and finally, there may be problems when *converting* those
floating point numbers from and to integer numbers. we are currently
double-checking this so to make sure that when you render to file
(export audio), the outcome is exact. but even if we dumb programmers
create errors in this area would you not be able to tell the difference
(at least with 24 bit files), because again the error would be in the
range of 1 bit, which is measureable but no equipment can reproduce
it anyways. but yes, we certainly need to be 100% correct for your
convenience and we will.
that gets us to 'bus summing'. as somebody pointed out
earlier, this is in fact nothing but adding values; and this adding
operation is just as precise as your pocket (or scientific) calculator.
would you trust it?
BTW, Ksmith, did you hear about the test when the same tracks were
summed in a PT Rig
and the Oxford? The results could be put out of phase giving zero.
[ http://www.virtualstudio.org/VS_AUTO_DI ... -02-15/117 ]http://
www.virtualstudio.org/VS_AUTO_DIGEST/00-02-15/117
I think both PT and the sony use integer busses though.
Either they both use the same 'summing bus' or we are all caught up in
some invented
problem.
possibly
in theory, you should not overdrive the channels more than +6dB and of
course
keep your master at 0 dB. this puts you on the 'mathematically prooven
exact
side of the fence', so to speak. in practice, even if you drive many
channels
much into the red, it doesn't make for more than an inaudible (albeit
possibly
measureable) difference.
actually, floats behave a bit like analog equipment: if you overdrive
it,
it does something on its own. but for what a float does then to become
perceivable, you would have to write a plug which drives your signal up
a few hundred dB's....certainly more than your analog tape recorder
at +6dB, which is clearly audible.
charlie

One reason I prefer to sum in Paris is due to it's 52 bit fixed point summing architecture. It is possible to process tracks at 32 bit ploat and then lightpipe them to Paris and this is what (apparently) happens from there on:

The following was posted by another other engineer who purchased the hardware patent from EMU after they deropped the system.

“ 1. esp2s (the chips on the EDS card that are the basis of paris-DSP's in the way the the SHARC's are the CW DSP's) work internally at 24bits of resolution regardless of the bit depth selected for the project. The *only* thing this setting controls is whether the bounce to disk is at 16 or 24 bits. When you mix bounce at 16 bits the least significant bits are truncated before being written. It should be noted that (IMHO) the vast majority of the resolution of the least significant bits is going to be converter self noise, system noise or other inaudible junk. I doubt that anything past 20bits is significant in a bounce in paris. I'm sure this will cause some contention with some, but it's only my opinion.

2. There is saturation occuring in the ESP2, in hardware, at the instruction level. I'm not an expert in this by any means, and I may be a bit off in my explanation - but here's what I have figured out after careful reading of the ESP2 patent. I'm trying to put some of it in laymens terms below with a bit of background.

In general There a couple basic instructions / math operations that go on in a mix. I'll stick to two:

1. Mixing streams of audio together is ADDITION. The addition operators in the ESP2 are *saturating*

2. Changing gain is MULTIPLICATION. The multiplication operators in the ESP2 are *saturating* . What does this mean? Well it involves the oft quoted "52bit" accumulator line. Now for a bit about what this means

1. An accumulator is used to store the result of a series of operations such as additions or multiplications.
2. The actual accumulator in paris is a 48 bits word with 4 control bits for a total of 52.
3. The 4 extra bits provide 4 guard bits for use in *detecting overlow/underflow* in the result of any calculation.

There's the background - now here's what happens:

When a series of additions or multiplications occurs on the esp2 the "intermediate result" is stored in the accumulator. The accumulator is 48 bits for a reason. Two 24bit values, or a series of 24bit values added together may produce a value that is greater than can be held in 24 bits. If in the course of this "accumulation" a result greater than 24bits is produced an "overflow" occurs. These overflows are tracked by the 4 control bits.
When an overflow occurs the "final result" or output of the operation or series of operations is *saturated*. This is done by setting the output value to saturated to h7FFFFF,FFFFFF (the largest positive number that can be represented by a 48-bit word)

When an "underflow" occurs the output of the operation or series of operations is set to h800000,000000 (the largest negative number)

The final step to producing the real output value is to take the most significant 24bits of the accumulator (7FFFFF) and send them on their way.

Other keys to the paris sound lie in the way it handles panning, pre-eq gain (this is really interesting) and reserving enough headroom to sum all those submixes together I can write more about those later as I get some kind of clue.

Anyway - here's a really concrete way for you to see how saturation at the instruction level affects the sound.

Open a project with 16 tracks or less. Drive the hell out of the mix, push it way into the red and make sure the submix clip lights come on occasionally. Don't use any paris eqs, directx or eds effects on this mix. Keep it dry and confined to faders and panning only. Make it SLAMMING. Now add another submix to the project. Take the submix with your slamming hot mix and switch it to a NATIVE submix. I'm sure you will be able to hear the difference in seconds. All kinds of gnarly nasty **** going on. “

Basically, what all this accomplishes when mixing in DSP is that the signal that is being received is being attenuated by 24dB, with no audible degradation. then, like an analog console, the Paris mixer has multiple gain stages available using the channel, global and submix faders as well as EQ makeup gain trim pots and very flexible compressor settings. These can be pushed and pulled in a manner similar to mixing on an analog board while achieving almost unheard of RMS levels when mixing in the digital realm…..

Additionally, Paris may have the lowest latency of any DSP based system for tracking audio. This helpful in delivering a comfortable real time headphone mix to performers while being able to deliver simultaneous DSP effects directly to the cue mix……..similar to the way it works with an analog mixer.

Paris latencies:
Round trip via insert (EDS "External") through SPDIF I/O = 1 Sample
Round trip via insert (EDS "External") through MEC 24 bit
analog Out to analog In = 60 Samples = 1.36ms/44.1KHz or 1.25ms/48K
Total record and monitor path (Channel In/Stereo Master Out) MEC 24 bit
analog In to analog Out = 66 Samples = 1.5ms/44.1Khz or 1.375ms/48KHz
EDS compressor with no lookahead = 2 Samples
EDS EQ = 0 Samples

I'm finding lots of sonic similarities between Paris and CW. The mixes sound very organic and punchy rather than bright and shiny like Pro Tools mixes, but that's good, IMHO. It's easy enough to dial in an overly bright top end with a filter if I want that.

DJ, you love always quoting all, eh?
Just wanne say a thread doesn't get more readable that way

(I just do that if I wanne be sure to hold what someone else might want to delete...and I don't)

and you huub, you always have to comment these things doesn't make the thread more readable, eighter

friendly greez
Roman

hubird wrote:DJ, you love always quoting all, eh?
Just wanne say a thread doesn't get more readable that way

(I just do that if I wanne be sure to hold what someone else might want to delete...and I don't)

Sorry. I will never do this again. I deserve to die.

I hold back for months already, and DJ knows what quoting is..., but this one again was so extremely redundent/annoying, I had to grab my chance
cheers.