Another thread about summing in scope

[Mike Goodwin]

Shocking news is that here are some utterly fanboys waiting to be busted. Just kidding. But gary...this is way beyond your level. I will prepare audio examples in line with files and screenshots so once for all we will kill this summing myth. I am really interested to read your comment after that. But today is the day and i will have some party just as you guys. Then i have my birthday soon etc.etc.. So see ya soon in a week or so.

I am glad that here are people that actually know technical side. Thank you Schroomz...

Please do!

The rest of this post if a mix of info and questions that are rattling around in my head...

Here is an interesting little bit of info from the Live manual. It is from the section of the manual that goes into depth about Lives audio quality. ...

"Please note that, while 64-bit summing is applied to each single mix point, Live’s internal
processing is still done at 32-bit. Thus, signals that are mixed across multiple summing
points may still result in an extremely small amount of signal degradation. This combination
of 64-bit summing within a 32-bit architecture strikes an ideal balance between audio quality
and CPU/memory consumption."

I am trying to think of when I would be summing across multiple points. Would creating summing busses and summing them into the master qualify as summing across multiple points? If so then I would be getting this extremely small about of degradation in just about every single one of my mixes. This leads one to ask if the Scope mixers suffer from this same problem. Anyone know if this is the case?

Getting back to the delay issues when using inserts and so on in the scope mixer......
Is there not an automatic delay compensation built into the scope mixers? I am going to get into the manual again today if I can to see if I can answer my own question but thought I would throw this one out there just in case there is more to it that what the manual covers.

Another thing that came up in this thread that has been running around in my head is the talk of entering and exiting the Scope environment via ASIO drivers. Is there room for degradation in signal there? I get the idea that if you are running at different bit depths and sample rates there is dithering going on but if you where to work with 24bits files and set the host program to run at 24 bits would it not just be a direct digital transfer? I did not even consider this in the past but I suppose it is a possibility. Again I bet even if there is only tiny bits of degradation it is the concern that when you add them all up in one signal chain you start to get errors that are large enough to create a level of degradation work thinking about. For example in my case I would be...

From my host program (having to make sure that all sample rates and bit depths match) via ASIO into the scope environment.
Then straight into the 2448 and from single channels into mix busses and then summing into the master channel.
Once I am happy with the mix and want to print it I have to go back via ASIO into Live or another recording program.
Then last but not least dither it down to 16bits from 24 bits..
It seems to me that there are many places there that problems can creap in and that fails to mention any insert FX and AUX summing.

So it is not just one step, it is when all of a sudden you have dithered two or three times as well as gone through two sample conversions and through two ASIO conversions, if indeed they are non-neutral operations.

Sorry for the ADD thread but I figured I may as keep all this in one thread/post.

P.S. Happy New Year to you and your families!

[garyb]

no, Scope doesn't have latency compensation, but then neither do real mixers. when you insert a digital delay or compressor into a hardware mixer or use a patchbay, there is a small amount of latency added. normally, this is not a problem. a vst interface MUST have latency compensation because the plugin is using the same cpu for processing as the mixer and playback mechanism is. much more serious latencies are the result. there is a single sample delay on the stm2448 and 4896, but you'd need to use it on the OTHER tracks, not the effected one, to compensate for a latency that was unlistenable.

there's a lot of debate about non-issues. going in and out of Scope via asio over and over will certainly create errors. that was never suggested, although if you get a good sound, who cares?

Mike, your mention of decreased bit depth across multiple points is exactly the problem. there are cpu resource issues in a complex mix, as you pointed out, which are not an issue in the dsp world. mpodrug, whatever incredible work you do, i hope it accounts for this in a real world, not a lab environment. i think we ALL agree that in theory, digital is digital, and that if the math is applied correctly, it doesn't matter what the platform is.

mpodrug, YOU are convinced about what you know. great! i also don't mind you bringiong forth info! that's good! i don't mind being wrong, although you'll have a hard time convincing me that i get equal or better mixes natively, my experience tells me otherwise, and NO experiment that you can show me will take into account all the factors involved. indeed, an experiment that WOULD take all the factors involved would be so complex as to be bad science, but hey, it's more fun to bust "fanboys" than it is to make good music, huh?

and lastly, THIS IS NOT AN ALL OR NOTHING DEBATE! natuarlly, the best solution is a mix of native and dsp. i say that the Scope mixers sound better that the native ones. that can't be disproved as it's a subjective opinion. all another can do is agree or disagree.

[Mike Goodwin]

no, Scope doesn't have latency compensation

And how is that considered to be OK?! That would create problems in many situations that "native" mixers would not have. This gives the Scope mixers a MASSIVE disadvantage, in my opinion.

there's a lot of debate about non-issues. going in and out of Scope via asio over and over will certainly create errors. that was never suggested, although if you get a good sound, who cares?

I will have to do some searching around on the forum for more info about going in and out via ASIO. In the end I have to agree with you though. If my mixes sound better in the end when I mix with Scope then ultimately who cares. I am all about work flow and feel. They are undeniably very important to a good "product".

Mike, your mention of decreased bit depth across multiple points is exactly the problem. there are cpu resource issues in a complex mix, as you pointed out, which are not an issue in the dsp world.

That is what I was hoping. Is there any reading material on this? There I did not see any audio fact sheet about the scope mixers in the manual for them. It would be good to see some type of solid info from Creamwre/Sonic Core about this.

mpodrug, YOU are convinced about what you know. great! i also don't mind you bringiong forth info! that's good! i don't mind being wrong, although you'll have a hard time convincing me that i get equal or better mixes natively, my experience tells me otherwise, and NO experiment that you can show me will take into account all the factors involved. indeed, an experiment that WOULD take all the factors involved would be so complex as to be bad science, but hey, it's more fun to bust "fanboys" than it is to make good music, huh?

and lastly, THIS IS NOT AN ALL OR NOTHING DEBATE! natuarlly, the best solution is a mix of native and dsp. i say that the Scope mixers sound better that the native ones. that can't be disproved as it's a subjective opinion. all another can do is agree or disagree.

I think this is where you are getting confused. The only thing this thread is trying to prove are things that can be proven or dissproven. The is not an argument about work flow, ease of use, flexibility, weather or not you like it better and so on. It is about math and digital accuracy. It is about dispelling the myth that summing straight digital sounds different in different programs, hosts, digital mixers. It is not dealing with summing across multiple points or other complex routing either (admittedly making the piratical usefulness of this thread rather debatable).

[garyb]

sure, math is math. purely summing a couple of tracks should be identical. in fact, i would expect it to be MOST accurate INSIDE the sequencer(well, that's what i would expect).

in a mix, there's no need to go back and forth to the sequencer(unless that's where you record the final mix, which would make this sentence irrelevant). latency issues are no more than hardware, so compensation is no more a problem in the Scope world than in hardware. for music that such things really mattered on, there is a plugin made called "phasefix"(i believe) and the sample delay built into the mixer, but this must be done manually, and it's a non-issue for most music as dsp processors are dedicated to a single task and so are, for the most part, truly "realtime". as i said, a patchbay and a digital reverb in the hardware realm is just as latency inducing.

Scope IS oversampled in most of the environment, and that oversampling is not load dependant as it is in the sequencer. for the exact official figures, you'd have to ask S/C. as i remember the stm2448 is 48bit, but i could be mis-remembering( )...

i am aware of the point of the thread. my point is "digital accuracy" is not as valuable as "i like the sound of it better". the best "hifi" and professional audio hardware is not always the most clinical, in fact, it is rarely the most clinical(which is not to say that Scope is less accurate or that being less accurate would make it better).

i would think that cubase could sum a few digital tracks more accurately than any analog hardware could dream of.

[Warp69]

no, Scope doesn't have latency compensation

And how is that considered to be OK?! That would create problems in many situations that "native" mixers would not have. This gives the Scope mixers a MASSIVE disadvantage, in my opinion.

And the same is true for any mixing console (analogue or digital) which use outboards. And for Protools.

[Mike Goodwin]

no, Scope doesn't have latency compensation

And how is that considered to be OK?! That would create problems in many situations that "native" mixers would not have. This gives the Scope mixers a MASSIVE disadvantage, in my opinion.

And the same is true for any mixing console (analogue or digital) which use outboards. And for Protools.

So true. I guess the difference is that in the analog world all compressors and EQ's do not introduce problematic amounts of delay. Where in the software world this is not the case. I guess the beauty of the situation is that anything that I need to sum in sample accuracy I can do in my host program before patching into the scope mixer. I do see that it could get confusing if I am using fx sends in my host as well as in my scope mixer.

I think my big curiosity now is the ASIO degradation factor. I did a quick search here but did not seem to find any threads on it. I could have easily missed it though. Is there anyone that can give a link to some info?

[Mike Goodwin]

latency compensation compensates what ?

The processing delay of the internal effects in the internal (if so) sequencer.

Protools can have this for DSP and native
Cubase/Nuendo can have this for native

latency compensation for any outboard like microphone, synth, effect does work only during playback.

The latency of the input path and in the native case the ADC of the OS (like ASIO and AU) cannot be compensated during recording and monitoring.

Yes, I am referring to playback compensation. Of course there is no way to compensate for delays when recording in real time.

[ChampionSound]

[...]

I think my big curiosity now is the ASIO degradation factor. I did a quick search here but did not seem to find any threads on it. I could have easily missed it though. Is there anyone that can give a link to some info?

This is something I really want to know too. Untill now, I always recorded my audio through the ASIO 24bit modules into Cubase. I've read several times that VDAT is superior to other audio recording apps (but unfortunately I don't own VDAT).
I found a somewhat related thread was which heading pretty offtopic here.

I really like to know if sampling audio in the STS samplers will result in similar recording quality as recording into VDAT, as the audiopath stays in the scope environment, in contrast to using ASIO. Of course the sampling time will depend on the amount of RAM available, but it may be a good option for people without VDAT (like me) but like to preserve as much as possible audio quality.
I think this is a bit offtopic for here and maybe needs its own thread?

[garyb]

latency compensation compensates what ?

The processing delay of the internal effects in the internal (if so) sequencer.

Protools can have this for DSP and native
Cubase/Nuendo can have this for native

latency compensation for any outboard like microphone, synth, effect does work only during playback.

The latency of the input path and in the native case the ADC of the OS (like ASIO and AU) cannot be compensated during recording and monitoring.

Yes, I am referring to playback compensation. Of course there is no way to compensate for delays when recording in real time.

Mike, hardware has similar latencies to Scope when using, say, a compressor. Scope won't have the kind of latencies that vst native will, because vst native uses the cpu like the rest of the computer. there's NO way in hell that the process can be anything near "realtime". compensation is a must in that case. Scope uses a dedicated processor for devices, they are realtime, just like the dsps in a hardware compressor or digital reverb. no compensation is required for MOST processes.

[Mike Goodwin]

there's NO way in hell that the process can be anything near "realtime". compensation is a must in that case. Scope uses a dedicated processor for devices, they are realtime, just like the dsps in a hardware compressor or digital reverb. no compensation is required for MOST processes.

If I understand you right you are wrong about that. For example Air EQ from eiosis runs at 0 sample latency. A quote from there site...

AirEQ works in all mixing situations because of its zero delay technology. It's the only equalizer on the market with such quality, no processing delay, and an incredibly low CPU usage. You can also use it for tracking or in a live situation; AirEQ will always bring you the best results imaginable from an equalizer.

Also with the Sonalksis SV-315 comperssor and SV-719 gate. Here is a quoat from there site
- Zero latency throughput (just like a real analogue compressor) - enables the compressor to be used in the monitor path for tracking situations

If my memory serves me right the Waves SSL bundle of plugins all run at 0 or 1 sample latency. I am quite sure that they have several others that do as well. These are just the ones that come to mind right away. Also these plugins are not very CPU heavy. In face the Air EQ is the lightest EQ I have and it sounds fantastic. It goes to show that it is not always about bruit force and more about brilliant designs.

It seems to me that many folks feel/think just because it is in a computer running on an Intel chip it will, sound worse, work slower, be less accurate and so on. In reading that quote from the guy at Sonalksis it seems that programming for shark DSP's may have a distinct sound due to the fact that they are introducing aliasing, most often thought of as a very unwanted byproduct of digital. And in the end less acurate than programming for a native CPU. That is if his feelings are indeed correct. Pretty wild.

[garyb]

i don't believe it. is it magic or necromancy? is it a time machine? 0 sample latency is not possible, even with analog hardware. the processor must take some time, i assume there's some "look ahead" function there.

should you use those eqs and comps(like air)? of course, why not?

[astroman]

one should be tolerant with the marketing geeks, but the quotes above (about the AirEQ and the Sonalksis) are way over the top.
Plain bs - how many different types of analog compressors do you know ?
btw every capacitor needs a tiny bit of time to load... and unload... and load... that's it's very nature

audio processing on Sharc DSPs is supposed to be superior because the chip was designed with audio processing as a main target.
Both the Pentium and the Motorola 56k DSP (for example) are general purpose devices.
Of course this includes audio applications as well, but the AD engineers provided optimized tools making it more likely to deliver a high-quality result than in the native world.

That type of math isn't simply 'math' as quoted a couple of times in this thread.
It's damn tricky and you have almost always > 6 options how to implement a certain processing.
Numeric precision is a pretty low concern - according to the designer of the Access Virus ...every idiot can copy a filter algorithm from a math book, but you need a talent to identify those (algorithms) that will sound in a pleasing way...

I have the Scope Hexachorus Effect, and the very same thing (probably it's predecessor) as pure Intel code (from Triple Dat)...
indeed I might prefer the Intel version

sorry for repeating myself, but since I have a preamp device from the late 60s, I don't give any sh*t on spec figures any more. This thing just sounds gorgeous with 0.4% of THD, while a 0.002% 'modern' device is plain flat and boring.
I have no idea if it's in the parts, the design, or whatever... but the amount of detail, precision and dynamic is amazing.
It's relatively important for me, as it's (in) my practice amp setup for guitar and bass, providing better tone control and (significantly!) less fatigue.

But I'm not sure if you could tell the difference in a mix at all

my 2nd rule of thumb (if I may add...)
use more than one different soundsource.
a complex arrangement all from only Scope synths and fx is very likely to sound rather boring.
Add a crappy device like the DX7, an analog Drumbox or some native Reverb somewhere.
Don't use all samples - let a real guitar player or bassist do some parts, etc etc.
Even if the latter recording aren't world class (because one lacks the gear), the track will gain in lifelyness and it's ability to adress the listener - imho.

it may be useful to know that a Scope mixer channel can slip 1-3 samples under obscure conditions...
while stacking hi-hats and kick-attacks it may be worth checking the alignment, but anytime else it's pretty pointless.
As is my (earlier) example of the difference between VDAT and .wav via Asio.
Assuming it wasn't a malfunction of software or the card itself, the difference wouldn't shriek your listener away
it's just good to know about the fact and take it into account
sorry, if any (unnecessary) confusion was caused by mentioning it...

cheers, Tom
ps: happy new year

[garyb]

there's a post with good sense.

[Fede]

AirEQ works in all mixing situations because of its zero delay technology. It's the only equalizer on the market with such quality, no processing delay, and an incredibly low CPU usage. You can also use it for tracking or in a live situation; AirEQ will always bring you the best results imaginable from an equalizer.

Also with the Sonalksis SV-315 comperssor and SV-719 gate. Here is a quoat from there site
- Zero latency throughput (just like a real analogue compressor) - enables the compressor to be used in the monitor path for tracking situations

If my memory serves me right the Waves SSL bundle of plugins all run at 0 or 1 sample latency.

Hi, I've to say that when I bought my first Pulsar card v1.2 it was marketed exactly as "zero latency".
I still have the packaging with those capitals letters, so I can confirm this, I was very impressed.

As in analogue world a circuit takes a certain amount of time to load as astro described, in dsp processing memory cells of filters (substantially capacitors again) have to load too, without taking into account complex algorithms in which the elaboration involves more "stages"...
no, real-time in hardware is really only marketing.
It is not realistic too, bacause there aren't things in nature that behave in a similar way, everything in nature has a small delay which is part of what we call the "impulse response". The meaning of that word should be re-thinked in a realistic way:
- one thing is extra-long buffering in software drivers, inefficient scheduling of intel cpus and architectures.
- one thing is dsp processing, where the processing work starts at clock "0" and the output begins to be non zero after n clock cycles. This is inevitable and dramatically similar to everything. We can call this real-time, exactly as analog, if we think that n clocks delay (together with resulting phase/frequencies cancellations) is part of the impulse response of this particular system. This is called a "causal system". In nature there can be only such systems. "non causal" systems are impossible in our space-time dimension. (There are some examples of "non causal" filters which we can approximate and use at the cost of compensating the time scale, *but* the output must be delayed furthermore in order to achieve them).
Maybe one day with terahertz chips it will be possible to confinate all processing delays in non hearable frequencies zones...

Don't think of sequencers, scope cards are designed to take an analog input and return an analog output in the exact way an effect unit does.

Fede

[Mike Goodwin]

how many different types of analog compressors do you know ?

Is there a productive point to this question that I am missing?


As for marketing bull, you can choose to believe what you like but it is easy to see that this could be done on modern computers. Sonalksis are one of the most reputable companies around. With a background from Neve I tend to feel that they have a bit of a reputation at stake. If they can manage to do there computations in less time than it takes for one sample at the given sample rate to pass then they can. After all there is life between samples

Do you truly feel that nothing can happen in the time of one sample? If so that is just silly. I can run very CPU heavy tasks (rendering video-converting audio files) in the background with a session running in my DAW with these plugins in use with out any problems. Get the picture? There is a lot of room to move we just cant sence it.

Good on them for coming up with efficient code. Five years ago people would have said that it was impossible for someone to code an eq that sounds as good and uses as little CPU as Air EQ as well. But it is there and it is almost unbelievable. Just like all new technology, at first is seems like magic but then it becomes clear that it is advancement.

As for mixing multiple sound sources I have to agree. Even mixing things from scope and VST manufacturers creates a deeper mix. Throw in some hardware for good measure if you can and even better.

[Mike Goodwin]

As in analogue world a circuit takes a certain amount of time to load as astro described, in dsp processing memory cells of filters (substantially capacitors again) have to load too, without taking into account complex algorithms in which the elaboration involves more "stages"...
no, real-time in hardware is really only marketing.
It is not realistic too, bacause there aren't things in nature that behave in a similar way, everything in nature has a small delay which is part of what we call the "impulse response". The meaning of that word should be re-thinked in a realistic way:
- one thing is extra-long buffering in software drivers, inefficient scheduling of intel cpus and architectures.
- one thing is dsp processing, where the processing work starts at clock "0" and the output begins to be non zero after n clock cycles. This is inevitable and dramatically similar to everything. We can call this real-time, exactly as analog, if we think that n clocks delay (together with resulting phase/frequencies cancellations) is part of the impulse response of this particular system. This is called a "causal system". In nature there can be only such systems. "non causal" systems are impossible in our space-time dimension. (There are some examples of "non causal" filters which we can approximate and use at the cost of compensating the time scale, *but* the output must be delayed furthermore in order to achieve them).
Maybe one day with terahertz chips it will be possible to confinate all processing delays in non hearable frequencies zones...

Don't think of sequencers, scope cards are designed to take an analog input and return an analog output in the exact way an effect unit does.

Fede

Wow Dude Ok there is zero latency and then there is zero sample latency. The audio interface will introduce latency sure. I am not talking about that. All I am trying to say is that there is software that can do what it needs to do in less than 1/44100th - 1/96000th of a second.

[garyb]

cool, and if you like those tools, use them. they go before the final gain stage(the fader) anyway. it's quite a natural signal flow to insert an eq or compressor on the sequencer track and then feed out it's own asio output into a Scope mixer. or not iif you prefer.

i find the Scope mixer superior, and for the most part, Scope compressors to be hardware quality. i think that they're a bargain in price, but then i actually own and know how to use hardware. i find the Scope signal flow to be natural to my real gear brain. i find the Scope mixer to rival my Soundcraft 1600, which was making REAL records for MONEY before some here had stopped soiling their diapers. everyone else doesn't need to feel the same, but it's good to know that Scope is good enough to be nominated for Grammys and Oscars for it's sound. this tells me that my gear isn't the issue. i don't care what $2000 vst eq package someone wants to tout, good sounding gear always is good sounding gear, that's why a real 1176 is still worth $1500 and so is an old beat up minimoog. that's why i just got $500 for an old tape echo and that's why i'm not impressed with marketing garbage. everyone knowsthat 99% of the computer gear can't hold a candle to real hardware, that's why they're always bragging about how much more "analog" the newest product sounds or bragging about having a tube. almost no one can afford the real stuff, so they sell kids who don't know the difference a counterfit and tell them it's the real thing but better. meanwhile, anyone who REALLY matters in the business is hording all the real, top quality gear, that's hardly made anymore, or at least hardly made except at exhorbitant prices. if a Scope card doesn't sound good enough, then why did a $250,000 Fairlight board come witrh a Scope card with stock effects?

does native work and sound great? OF COURSE IT CAN AND OFTEN DOES! a Scope card isn't meant to replace native, it's a massive augmentation. it really confuses me how little use some people get out of such a prime collection of tools, but that's just me. i've never understood why people who claim to be into music and sound and who want their own studio don't learn more about the actual art of music production and instead look for cheats and walkthroughs, like the whole things a video game or VH-1 original drama, but again, that's just me and my problems...

is Sonalksis great? no doubt, and well worth using, although Waves is a buch of hyped crap for the money that they charge. i haven't tried AIR, but i really use very little eq, unless i just want to screw something up. normally, an engineer likes to use the right mic and mic pre, and the thing is perfectly recorded and eq'd from the get-go, which always sounds better. mostly, i use things like lowcut filters or maybe a shelving filter, so the stock peq4 is normally pretty effective. i do eq the whole mix, usually with the BX, but first, i try to get a balanced mix, so that once again, a low cut or so is all that's normally needed. having massive amounts of eq needed is the sign of a bad sounding room, poor mic choice, crappy recording technique or bad samples. it indicates a second rate engineer, excepting of course, when it's for special effects.

there you go, i'm ranting away again. thanks for the opportunity.

if i said anything good, i hope it helps, if not, please ignore it.

[maky325]

no, Scope doesn't have latency compensation, but then neither do real mixers. when you insert a digital delay or compressor into a hardware mixer or use a patchbay, there is a small amount of latency added. normally, this is not a problem. a vst interface MUST have latency compensation because the plugin is using the same cpu for processing as the mixer and playback mechanism is. much more serious latencies are the result. there is a single sample delay on the stm2448 and 4896, but you'd need to use it on the OTHER tracks, not the effected one, to compensate for a latency that was unlistenable.

there's a lot of debate about non-issues. going in and out of Scope via asio over and over will certainly create errors. that was never suggested, although if you get a good sound, who cares?

Mike, your mention of decreased bit depth across multiple points is exactly the problem. there are cpu resource issues in a complex mix, as you pointed out, which are not an issue in the dsp world. mpodrug, whatever incredible work you do, i hope it accounts for this in a real world, not a lab environment. i think we ALL agree that in theory, digital is digital, and that if the math is applied correctly, it doesn't matter what the platform is.

mpodrug, YOU are convinced about what you know. great! i also don't mind you bringiong forth info! that's good! i don't mind being wrong, although you'll have a hard time convincing me that i get equal or better mixes natively, my experience tells me otherwise, and NO experiment that you can show me will take into account all the factors involved. indeed, an experiment that WOULD take all the factors involved would be so complex as to be bad science, but hey, it's more fun to bust "fanboys" than it is to make good music, huh?

and lastly, THIS IS NOT AN ALL OR NOTHING DEBATE! natuarlly, the best solution is a mix of native and dsp. i say that the Scope mixers sound better that the native ones. that can't be disproved as it's a subjective opinion. all another can do is agree or disagree.

Garry dont try to turn your head around now. My english is pretty bad but all people can see that i cleared out i am surprised that you (IMO very experienced and super nice user) is talking that scope mixer is better sounding . Your sentence "summing is summing yet cubase sounds nothing like an ssl or euphonix console even with no processing. cubase alone sounds different than run through the Scope mixer. this is ALSO undeniable" clearly show this. Then again i cleared that i am surprised that you very experienced user is talking like that in THREAD about summing. From what you are saying people can easily think and believe in ongoing myth that somehow if you mix/sum in scope you will have better or different results.

I also cleared that if difference exist it does not came from summing alone, it is from using scope internal effects or working modes or on hand experience whic is not better/worse, just different. I said and cleared all that before all of what i have just quoted from you. That said i was and i am still surprised by your comment.

I also want to clear out that i am not making any user any campaign or ant gary campaing LOL. For what i can say you are (and you will be) still legend here and you did very helpful things for many of us. Your status is deserved!! That was the reason why i was so confused reading your words.

So i am preparing files with downloadable examples. I hate to tell you but all what you are saying(given that you are saying this in summing thread where we talk about summing not about mixing exclusively in scope) in this thread about summing and better sounding mixers/mixes is simply not true. Whatever monitor pair you have or whatever room treatment you have it is beyond my mind how can you with all your experience hear undeniable substantial difference with your multiple confirmations ??

that said i can saying all this from my ass but that is the reason why i am making files. However today is my birthday and i have some things to do I will post my results in short time (few days).

I believe it is not too late but happy new year to all at planetZ. Everything best!!

[garyb]

however YOU get YOUR best results is fine by me.

i have my preferences and i think this is a silly discussion. i also think it's silly for you to post examples. i'm sure you can get the results you are happiest with, with the gear you like best. you're right. Cubase and Euphonix are identical.

look, it doesn't matter. no mix is simply a, b and c tracks simply combined, but if that's all you are doing, then i'm sure that it hardly matters how you combine them, within reason, of course.

am i being foolish? ok, fine. music, the internet and computers are also rather foolish, but i'm still playing with all of them. i'm not surprised to be foolish again.

btw- stop trying to save my feelings while you assisinate my character. i'm already insulted, but it doesn't matter since we've already established my foolishness.

[maky325]

however YOU get YOUR best results is fine by me.

i have my preferences and i think this is a silly discussion. i also think it's silly for you to post examples. i'm sure you can get the results you are happiest with, with the gear you like best. you're right. Cubase and Euphonix are identical.

look, it doesn't matter. no mix is simply a, b and c tracks simply combined, but if that's all you are doing, then i'm sure that it hardly matters how you combine them, within reason, of course.

am i being foolish? ok, fine. music, the internet and computers are also rather foolish, but i'm still playing with all of them. i'm not surprised to be foolish again.

btw- stop trying to save my feelings while you assisinate my character. i'm already insulted, but it doesn't matter since we've already established my foolishness.

I am not trying to assisinate your character. Guys that was not my intention. My english is bad i having big problems in ordering what i want to say. Please stop it. Also i am not pointing on your music or your talent whatsoever. Or your preferences. I really dont care for that. I am intrigued on all this by ongoing myth about summing and by some comments of experienced users in past (you included).

So i dont have some crazy "however i get my result" technique. I will just show my results based on proven facts. You can now try to discredit this by putting all in "look, it doesn't matter" bag or "i also think it's silly for you to post examples" but i will post results anyway.

So with hope some future users wont fall in marketing myth some people like to make (i am not directing this to you).

Also i think it is so sad you are trying to mislead my words in public. I NEVER, EVER said Cubase or Euphonix are identical. NEVER!! I said they are using very same summing, just as i said that there are much more things going inside which gives them different sound. Gary this is so sad. You are trying to discredit someone on his bad language skills. Your bad. But people can read this thread from start so you are shooting yourself here claiming (again) i am killing your character. Fine just fine.

Keep in mind i wont make any points to direct users. I will make thread about summing overall not about summing with results and your personal results compared with mine. I am really not interested in personal fight with you and this really would not be smart. Really Gary i am not having anything against you! But each time you try to discredit me with absurd things i will react you can be sure about that.

If you find yourself insulted after results than you can blame yourself. I am not responsible if you said some naive things in past. But if you dont mind then this is fine also. I really have to go offline now so with no disrespect intended see ya soon. Hopefully you wont be angry.


edit: For some more fun, i still remember my first two days on using scope mixer.. It was something like"wow this is sounding better then anything i heard for a long time"....But later i found it was not mixer it was fantastic Scope synths which are way ahead of most of what you can find in VST world. (and some different FX, even free ones)...eehhh..i am still discovering wonderful scope world..