niclaus, post: 430844, member: 33719 wrote: Yes, i know, that is not what i meant.

Yes, I know what you meant. You are asking and wondering the same thing as me.

I just tried something.
Opened a 96k session, export some files @44.1, reimported them in the same session @96k (so two SRCs in the box), flip the polarity, and i get a null... Not what i expected...

TheJackAttack, post: 430843, member: 34377 wrote: I think the gist is that if you have very high quality ADC and DAC as separate units a dual computer setup provides superior bit rate and sample rate conversion (uncoupled clocks). If you have mediocre converters your mileage will vary versus your ITB skill and the program writer's skill. I'd like to think that even middle road converters will achieve smoother results than ITB but I have no data to back this up.

Spot on.

I've done extensive test in all this, built an incredible mixing system so I am pretty confident I have a good handle on what is worth doing or not. Capturing to something like a DSD recorder is cool but nothing like mixing into the capture DAW. Having a mastering DAW on the capture side be as truthful as possible is the ultimate mind blow.
NOTE: "mastering" does not mean mastering, it means better summing ;) .
The other critical component is the monitoring system must be connected to the capture DAW DA. If not, you are summing and mixing rather blind.

Chris, I don't want to derail this great discussion but I've been itching to know if you do the majority of your mixing on DAW 1 and then just touch up with DAW 2? My guess is that it depends upon each mix and the options are endless.
Sorry for butting in.

vibrations1951, post: 430850, member: 34341 wrote: Chris, I don't want to derail this great discussion but I've been itching to know if you do the majority of your mixing on DAW 1 and then just touch up with DAW 2? My guess is that it depends upon each mix and the options are endless.
Sorry for butting in.

You are most definitely NOT derailing the thread, its only getting better! Because you are entering this approach, you are already asking advanced question that I look forward to answering more and more.

I do ALL my mixing on DAW 1. DAW 2 is where I monitor DAW1 from. DAW 2 is the sum, where I analyze mixes, create space, use spacial effects, M/S process, export and more. DAW 2 is everything involved in what your mastering section is about.

DAW 1 master section is disabled.

audiokid, post: 430851, member: 1 wrote: You are most definitely NOT derailing the thread, its only getting better! Because you are entering this approach, you are already asking advanced question that I look forward to answering more and more.

I do ALL my mixing on DAW 1. DAW 2 is where I monitor DAW1 from. DAW 2 is the sum, where I analyze mixes, create space, use spacial effects, M/S process, export and more. DAW 2 is everything involved in what your mastering section is about.

DAW 1 master section is disabled.

OK great! This is a big help. That is how I'm approaching things right now. Monitoring off DAW2 while I mix on DAW 1. From what little I've had a chance to try mixing so far it just seems easier this way.??? Maybe easier is a poor choice of words but it's the bottom line for me. Easier to hear, easier to make changes, easier to hear those changes. I haven't done any finishing with DAW 2 yet but I will. Baby steps for me right now...havin' a ball!
Namaste

TheJackAttack, post: 430843, member: 34377 wrote: I think the gist is that if you have very high quality ADC and DAC as separate units a dual computer setup provides superior bit rate and sample rate conversion (uncoupled clocks). If you have mediocre converters your mileage will vary versus your ITB skill and the program writer's skill. I'd like to think that even middle road converters will achieve smoother results than ITB but I have no data to back this up.

I've heard the results of uncoupled daws on stuff I'd mixed down differently. It's hard to accrue data on 'sounds better to me'. My line of thinking is I know for sure (as anything can be) it isn't going to be worse provided my system is specd nicely. So I'm incorporating the daw 2 as a necessary part of my new digs.

niclaus, post: 430848, member: 33719 wrote: I just tried something.
Opened a 96k session, export some files @44.1, reimported them in the same session @96k (so two SRCs in the box), flip the polarity, and i get a null... Not what i expected...

I could be way off but I think null testing would be more revealing if we were talking linear recording formats? Just thinking out loud, silently thru text.

kmetal, post: 430853, member: 37533 wrote: I could be way off but I think null testing would be more revealing if we were talking linear recording formats? Just thinking out loud, silently thru text.

As an example:
I would use null testing to compare 1 DAW mix to a 2 DAW. Check converters and transparencies.
To hear what an analog product did in a mix.

In some cases, more for just testing and learning about stuff... , you of course have to be taking the analog pass into consideration. This would be similar to saying, I'm used to my bed, what does this sheet feel like now.

Doesn't the analog pass inherently make the waveforms slightly different? Or I guess what I'm asking is shouldn't it?

Yes! Which can be a good change. But, you can still compare in ways that are beside the change.

Which is what i meant by saying
"I know what my bed feels like"
Bed=the core of the pass (hybrid summing system) before added gear or colour is added.

Which is where transparent summing excels. You can always add colour.

Being said, Im now well versed on what "I want" as "my sound" to a point I am going for the Folcrom and Millennia m-2b as my colour (pass of choice).

Until my new rig arrives, this is still speculation.
I look forward in sharing what I'm talking about.

What a thrill seeing all the interest in this now. Its so awesome.

So does the null show anything qualitative? Are you able to 'see' which passes 'sound' better? To my thinking, a sample based system just has to hit the right 'dots'. so as long as the extremes were met, the points in between are averaged or thrown out? Similar to a median line on a scatter plot?

I'm curious to hear from Boulder and mr ease, and people who are fans of the null test, if they can hear a difference they can't see, or see a difference they can't here. In they're repspective listing areas.

Good questions.

Visual and auditory combined help each other. Example, I may use a visual editor of a plugin graph on the Capture DAW to pin out all the freq I see and hear in something until I hear no more can be added or subtracted to get a null.

Analog is not constant and you can hear this when you try and null something. No two captures are the same. Uncoupling is even less., which is a good or bad thing. Which is why you ultimately want good converters, a good summing console and good monitoring DA. You will at some point want to learn more about what you are doing, not be mixing through a bunch of swinging products screwing with you all once. This is why people say, once ITB, stay ITB.
So if you are the daring individual who goes OTB, this is why I start with transparent gear and add colour after. At least thats the idea anyway. Some of us already know what we want and some of us never get the chance to even know what this is all about.
In my case, I have now moved into a more personal way of mixing. A Folcrom and m-2b are my next journey. If this doesn't work, then I will just stick with 2 DAW's and no gear in between at all. Racks of gear is behind me now but a very specific tone and another Bricasti might be exactly what I want.

When we use analog in a chain, (API 2500, MEQ-5, API 5500 etc) the analog can fool you into thinking its actually creating space when in fact its fooling you because of some inconsistent phase swinging, thus sounding like its wider.
To find these or demystify something you are wondering about, or emulate something you really like, using a graphic EQ to pin the remainders of something you think is an attribute, soon becomes apparent if you can find out why its doing what it is, in a pass. More so than not... the analog pass is actually shifting something and when you start stacking analog gear, you also are stacking randon swinging which imho, takes a good thing and makes it worse. (less is more) :love: makes a tighter bigger mix to my ears.
All this may seem irrelevant to most people but anyone who is looking at analog gear, should know more about the gear than just taking a bunch of gearslut opinions on something.
A 2 DAW system opens up new ways of listening. Our business is all about listening, not guessing. When you actually discover how a lot of expensive gear, or cheap for that matter plays in your chain, it gives you a whole new respect for less is more and where to use a tool (plug-in or hardware) in a chain. Its a big topic but in a nut shell, I love being able to hear and sometimes see what gear does rather than just saying I like it.

To reach a bit more into hardware... The BAX LPF is pretty damn important as the last piece in a "busy" chain or mix. Why? Its really easy to push high freq that effect the AD (round trip or DAW2 capture). When you strap on a LPF and stop those ultra freq from passing through to the capture DAW, the mids sound straighter and warmer, more upfront. This is actually why Chris Muth put those filters in there. I get it now. Which I now hold off on adding high freqs in the mix. And even better, why I look for phase issue way ahead so I'm not trying to export them at the finish line. ;)
When you help your AD like this, the mix seems to export better as well, which takes us back to the OP.. I would never have discovered this with such conviction had I not done some isolation tests between one mix to another, in a 2 DAW system.

My list goes on.
During this process I have learned how powerful my DAW really is, and how important it is to hear those changes I was missing before. It took $100,000 worth of mixing and mastering gear to learn I can do it ITB for thousands less. 2 DAW's makes it way more fun and interesting. Plus you can track HD and never have to bounce again. That is, if you don't want to.

audiokid, post: 430854, member: 1 wrote: As an example:
I would use null testing to compare 1 DAW mix to a 2 DAW. Check converters and transparencies.
To hear what an analog product did in a mix.

Yes, right, but at least the null testing is telling me the SRC in PT does not degrade the audio quality in a horrid manner.

I will test the 2 DAWs conversion tomorrow and i will let you know what i hear.

audiokid, post: 430862, member: 1 wrote: The BAX LPF is pretty damn important as the last piece in a "busy" chain or mix. Why? Its really easy to push high freq that effect the AD (round trip or DAW2 capture). When you strap on a LPF and stop those ultra freq from passing through to the capture DAW, the mids sound straighter and warmer, more upfront.

What is the corner frequency you usually use for this?

DonnyThompson, post: 430864, member: 46114 wrote: What is the corner frequency you usually use for this?

24 hz low , 18 khz top but what ever sounds right on the low.

niclaus, post: 430863, member: 33719 wrote: I will test the 2 DAWs conversion tomorrow and i will let you know what i hear.

You may be attempting to be trying something Dave is doing or what I do. Dave and I are doing it different. I'm wondering which?

what I do.

procedure

• track something on DAW1 at 96k SR then bounce down one version to 44.1. Capture it also on an uncoupled DAW2 pass at 44.1 and compare the two 44.1 versions. If they null you are not doing this successfully.

What do you hear?

audiokid, post: 430862, member: 1 wrote: In my case, I have now moved into a more personal way of mixing. A Folcrom and m-2b are my next journey.

Are you saying you want to put the Millennia in the transfer chain from Daw1 to Daw2 ?

pcrecord, post: 430882, member: 46460 wrote: Are you saying you want to put the Millennia in the transfer chain from Daw1 to Daw2 ?

Yes.

audiokid, post: 430879, member: 1 wrote: You may be attempting to be trying something Dave is doing or what I do. Dave and I are doing it different. I'm wondering which?

what I do.

procedure

• track something on DAW1 at 96k SR then bounce down one version to 44.1. Capture it also on an uncoupled DAW2 pass at 44.1 and compare the two 44.1 versions. If they null you are not doing this successfully.

What do you hear?

Yes, that is what i am going to try tomorrow when i am at the studio.
I Will let you know for sure.
Thanks.

audiokid, post: 430878, member: 1 wrote: 24 hz low , 18 khz top but what ever sounds right on the low.

I usual HPF a bit higher than that, generally around 35hz, but occasionally higher... Most of the time I'm working with a bandwidth of 35hz - 18 k.

Do you ever lo-pass any higher than 18k? I'm just curious, I know there are some engineers who like to occasionally tap into those ultra-high 20k + bands a bit, (although I would assume that that this would depend on the SR used, along with the quality of the converters in the gain-chain).

I've read where some of these modern pro-level converters are claiming to being able to reach up to 60k, and I can't help but wonder if their audience isn't maybe sleeping hanging upside-down in a cave, or peeing on fire hydrants. LOL

DonnyThompson, post: 430890, member: 46114 wrote: I usual HPF a bit higher than that, generally around 35hz, but occasionally higher... Most of the time I'm working with a bandwidth of 35hz - 18 k.

Do you ever lo-pass any higher than 18k? I'm just curious, I know there are some engineers who like to occasionally tap into those ultra-high 20k + bands a bit, (although I would assume that that this would depend on the SR used, along with the quality of the converters in the gain-chain).

I've read where some of these modern pro-level converters are claiming to being able to reach up to 60k, and I can't help but wonder if their audience isn't maybe sleeping hanging upside-down in a cave, or peeing on fire hydrants. LOL

there is plenty of evidence that even if someone can't hear the dog frequencies (above 20k, which a lot of people can) that interaction in the harmonics of lower freqs can be affected creating audible effect. with 24 /48 i could see benefit in low passing to avoid the effects of filtering in the conversion. all that stuff is going to be filtered out by the converters anyhow so why not do it in a way that you control?

one very well known tale is when Neve was installing the first ISA at AIR London, Geoff Emrik heard a 40k oscillation on one channel strip. Neve didn't believe him until he discovered a fault which when corrected made the offending oscillations go away. so i think it's fair to assume that higher freqs do matter. imo, if you are going to hp at 40hZ and lp at 18k, why bother with digital at all? you can get that kind of performance from an old TEAC/TASCAM RTR.

My settings are not set in stone. Those are starting points and very subjective to each mix I get.
Being said, now that i have removed all the analog bloat, i no longer need the BAX or outboard filters.
The straighest wire from DAW 1 to DAW 2 is what I hear fullest and most smooth.
I only suggest the BAX to people who are using outboard gear to be aware of the importance of using filters. That is, if you are sizzling.
The whole hybrid trip is a Big money pit to me. Its missleading us into thinking software can't do what gear does.
The whole Hybrid game can quickly become a waste of money if you are missing key parts to the puzzle. Example: the da on the capture side, dont bounce etc. But, this is of course the Dangerous approach which I tend to trust alot.

I chuckle reading rave shoutouts on product used in a chain /example: mastering level EQ etc then sizzle the hell out of the round trip through example, lower end conversion.
I can only imagine how much trouble the conversion is having trying to sum all the information through a round trip mix then bouncing it to a lower sr.

Sorry iphone typing here.

In all my comments, make note that i am no longer into hybrid mixing but i am still really engaged in a two DAW system.
Once I set up the Folcrom I'm interested in how a mix will sound with just the m-2b.

Maybe some members can send me a few mixes and we can have fun discussing it.

Kurt Foster, post: 430893, member: 7836 wrote: there is plenty of evidence that even if someone can't hear the dog frequencies (above 20k, which a lot of people can) that interaction in the harmonics of lower freqs can be affected creating audible effect. with 24 /48 i could see benefit in low passing to avoid the effects of filtering in the conversion. all that stuff is going to be filtered out by the converters anyhow so why not do it in a way that you control?

exactly. Which is what the BAX shine on.

Killer explanation chris. Well said.

niclaus, post: 430863, member: 33719 wrote: Yes, right, but at least the null testing is telling me the SRC in PT does not degrade the audio quality in a horrid manner.

I will test the 2 DAWs conversion tomorrow and i will let you know what i hear.

A null test will tell you it's the same data, not necessarily if it's degrading more subjective things like soundstage. 6db at 10k is that much no matter what, but that says nothing about the timbre of the eq, or whatever it may be your doing,

kmetal, post: 430904, member: 37533 wrote: Killer explanation chris. Well said.

Thanks Kyle.

Generally speaking.

I used a null test to isolate reverb bleed on a left channel of a mix that the recordist swore he had turned off the effect. I heard a weird phasing and upper freq tail in the mix and after I did this test using the adding and subtracting of freqs, I found the reverb, that for some reason never turned off in his Pro Tools mix.
This not only made me look good but it also opened up the mystery of why didn't that plug-in completely turn off in Pro tools. Yet to be answered.

Some might wonder how I did that.
It was in a shootout of converters. The shootout was corrupted. The guy providing various ADDA conversions, added reverb on a track. My guess, he was curious to hear how a session sounded when he applied reverb to one of the converters. Listening for that smooth tail... (he should have uses a Bricasti lol)
A group of ME apparently selling some BS sophisticated nulling process missed it. I found it using a 2 DAW system and the process I shared a few post back.

I asked for the same session passed through the other converters and low and behold, even though the converters were different brands they were close enough that I was able to isolate reverb on that channel creating a phase issue. Had I not found that, The brand would have been falsely shunned by that group.
The recordist providing those sessions finally admitted he tried a verb and thought he had turned it off.

My point is, nulling can help you discover other things in sound even if they aren't the exact take..

Interesting, I'm gonna add nulling to my bag of tricks. That's a great case were the repeatability factor of the digital audio, or the averaging I'm speaking of, is amazingly useful.

niclaus, post: 430824, member: 33719 wrote: Exactly, that is What i mean...
So the ME just import the files to the final SR (since you said he didn't have two DAWs).
So protools is doing the SRC?

Yes. Thru his converter. His converter is an Omni that has been upgraded by Black Lion. He opens my files as a PT session, records the tracks back into his session at the destination SR...thus his converters do the conversion of unlinked unclocked files.

As for me "doing something pretty odd" this is in fact a method we used in mixing on SSL desks....at least a close proximity of it....I mixed a lot of stuff on desks that you grouped everything in similar instrumentals and sent them out thru the 2-bus from a separate bus. This is the only way to control summing from multiple sources and not blow up the mix at the 2-bus.....standard stuff....

The fact that PT HD has this function available is a very good thing for me. My comment about not knowing what I was missing was more in reference to learning something about my program I didn't know existed and am happier about it. I made plenty of recordings NOT knowing this and they are sonically good....so its more a learning experience rather than needing to FIX something thats wrong.

Davedog, post: 430915, member: 4495 wrote: Yes. Thru his converter. His converter is an Omni that has been upgraded by Black Lion. He opens my files as a PT session, records the tracks back into his session at the destination SR...thus his converters do the conversion of unlinked unclocked files

I'm still confused here which I'm beginning to think its because I don't understand the Mastering process and quite possibly... can Pro Tools play different SR at the same time.
How can he load your example: 48k session then re-track (uncoupled) back to his DAW in real time at a different SR?
Is he doing a round trip as well?

Davedog, post: 430915, member: 4495 wrote: The fact that PT HD has this function available is a very good thing for me. My comment about not knowing what I was missing was more in reference to learning something about my program I didn't know existed and am happier about it

Indeed! Kudo's

I'm still trying to understand this function, which I'm beginning to think I won't because I can't duplicate this without wire. :)

audiokid, post: 430917, member: 1 wrote: I'm still confused here which I'm beginning to think its because I don't understand the Mastering process and quite possibly... can Pro Tools play different SR at the same time.
How can he load your example: 48k session then re-track (uncoupled) back to his DAW in real time at a different SR?
Is he doing a round trip as well?

I was going to ask the same thing And that is what i am trying to understand from the beginning.
Protools cannot record and play files at a different SR in the same session, at least from what i know.

Anyways, i did my tests this morning, here it is :

I took two 96k files from a same song, one was the mix and the other was the mastered version.
I loaded them in a protools HD9
Took the analog output from that one (192 I/O) and plugged it in the analog input of Protools HD9 number 2 (192 I/O as well)
Recorded those two files in PT number 2 @44.1
Then I imported and converted the 2 original files (96k) in protools number 2.

I tried to null them and of course they would not null.
Here is what i hear (and this is the same thing for the two files)

- i really don't hear a huge difference between the converted and the recorded versions. I'd say the recorded versions tend to tame the high end, to be a little bit smoother, but honestly i am pretty sure I would not be able to pick it up in a A/B testing. (And i listened in our main studio, which i know by heart)

- i tried a matching EQ software between the two versions, and it gives me a perfectly flat answer.

- when i try to null them, i only get stuff higher than, say, 4 KHz.

What do you guys think? Does that make any sense to you?
Chris, what should i be listening for? What should be the biggest difference here?

Maybe things would be different through better converters but i wanted to keep it as simple as possible.
Maybe i'll try to go through our neve converter.

niclaus, post: 430936, member: 33719 wrote: when i try to null them, i only get stuff higher than, say, 4 KHz.

Hmmm.... curious to find out if this might perhaps be a result of the added conversion step (?)

Excellent!

niclaus, post: 430936, member: 33719 wrote: Chris, what should i be listening for? What should be the biggest difference here?

Your findings are accurate to what I expected. This subtle change, plus many other aspects of why I do this add up to many things.

niclaus, post: 430936, member: 33719 wrote: Maybe things would be different through better converters but i wanted to keep it as simple as possible.
Maybe i'll try to go through our neve converter.

Indeed. Now you know what your Avid AD sounds like. It would be interesting to hear the Neve, yes!
Was the audio a tone or a mix? How did they mono? Phase is a huge factor here.

DonnyThompson, post: 430938, member: 46114 wrote: Hmmm.... curious to find out if this might perhaps be a result of the added conversion step (?)

Both are being converter, one is the bounce and the other is the uncoupled pass to 44.1, right?
If the change was a lot, this would indeed tell us that Avid AD and/or code has a problem. What it tells us is its different and this is how little accumulates.

niclaus

niclaus, post: 430936, member: 33719 wrote: when i try to null them, i only get stuff higher than, say, 4 KHz.

On which version?

It's very easy to get confused between the various factors involved when using separate source and destination systems (two-box method).

For example, on the use of nulling to gauge results: once you have an analog step in the process, you will not get absolute nulling, but it may appear to be better at low frequencies than high (assuming amplitudes are matched). This alone could explain what Niclaus is seeing. A lack of perfect nulling does not mean the process is faulty or that you would hear a difference between the results even if the visual waveform shows one. There's a well-known demonstration where the harmonics of a clarinet waveform are phase-randomised with respect to the fundamental so the waveform looks completely different, yet it sounds identical because the ear is not phase-sensitive for steady-state sounds.

"[="http://recording.org/threads/hybrid-setup-summing-mixer-questions.53286/"]Hybrid setup, summing mixer questions[/]="http://recording.or…"]Hybrid setup, summing mixer questions[/]" is a 2012 thread worth reading.

This is an extract from the 2014 thread "[[url=http://="http://recording.or…"]Analogue transfers between uncoupled DAW's[/]="http://recording.or…"]Analogue transfers between uncoupled DAW's[/]":

...the tests that I did with what I called the "two-box" method all show that my best improvement came from (a) using a high sampling rate for the source tracks, (b) not using a digital SRC, (c) performing an analog mixdown from box 1 to box 2, and (d) not locking the sampling clocks of the two boxes (if they happen to be at the same nominal rate).

All these conditions interact in different ways, so it's hard to say whether you would get a better result with your proposed trial, but don't let me stop you trying it out!

Incidentally, I even got some improvement by using 48KHz source rates rather than going all the way to 96KHz for box 1 when box 2 was at the CD standard of 44.1KHz. This seemed to show that there is something going on right at the top of the audio band which when multiplied by the number of source tracks causes unpleasantness in the mix if digitized at the same rate as the source.

Boswell, post: 430948, member: 29034 wrote: For example, on the use of nulling to gauge results: once you have an analog step in the process, you will not get absolute nulling, but it may appear to be better at low frequencies than high (assuming amplitudes are matched).

Excellent Bos, this is a very important step I missed mentioning when I was talking about my nulling process, via your two box or my two DAW ;) process!
When null testing, I forensically adjust levels and in some additional tests include +- freqs to isoloate suspects until there is absolutely nothing possible remaining to fool me.

Generally speaking, I'm assuming its hard for the general public to comprehend some aspects to this until you've actually used two systems like this. Its all very powerful to me.

Yes the signal need to be at equal level and timmed perfectly !

pcrecord, post: 430953, member: 46460 wrote: and timmed perfectly !

And timed properly too!

I chuckle of that group of ME's who designed some loop back null test thingy to test conversions. Manual labour goes a long ways when it comes to null testing.
Thanks for mentioning that too Marco!

I understand the confusion my posts have caused and I will ask my ME his method. He may very well have another capture device in his chain. I don't know for sure. I do know he doesn't like any sharing through a disc of any sort. You guys have fun with your 'nulling'

Also....he sends files for duplication through a DDP and prefers everything be zipped for protection against corruption.

audiokid, post: 430943, member: 1 wrote:
Was the audio a tone or a mix? How did they mono? Phase is a huge factor here.

Both are being converter, one is the bounce and the other is the uncoupled pass to 44.1, right?
If the change was a lot, this would indeed tell us that Avid AD and/or code has a problem. What it tells us is its different and this is how little accumulates.

niclaus

On which version?

They were two mixes (same mix actually but one unmastered and a mastered version).
Well they "monoed" pretty much the same, no notable differences...

(about nulling, yes, i know things should time aligned and leveled)

I totally understand that the analog process make it impossible to get a null, but at least it tells us what is different and what is not (even though i understand that difference would be different every time you go through that analog trip).
And what i find interesting is that there were almost nothing up to 4k in the difference (when trying to null).

Again, the difference (when switching from A to B) between the two versions (bounced (actually, converted by protools) and recorded) was really REALLY thin, and even though i am not doubting that you guys see a difference in the workflow as you use that method everyday, i am trying to understand what could be the plus side of doing something when i cannot tell the difference.
I am not saying i think you waste your time doing that (i respect you guys and you work too much for that), i am just trying to understand why you feel it is so important to proceed this way everyday when you could be lazy and just launch an offline conversion.
To make it clear, i'd really like to listen to those two files, and really see a difference that would make me say "oh shit, why didn't i do that sooner"...

Or maybe the kind of files i used do not let me see the difference. I used heavy rock, and maybe those distorted guitars and heavily compressed drums did not let me hear what i would have on some other materials.

If any of you have files that could be more interesting to try, i'd be happy to do the test and share the results here.
(Sorry but i cannot share the files of my tests since i am not sure the client would like me to)

Again, i am not doubting anything, i am just waiting to be convinced.

(oh, and for the record, i most always mix as Dave do (or my version of it), so i never go through the master fader thing myself either)

Davedog
Yes, that would be great if you could get some clarification from your ME. Thank you for that.

niclaus, post: 430956, member: 33719 wrote: (oh, and for the record, i most always mix as Dave do (or my version of it), so i never go through the master fader thing myself either)

THIS was my point of my post in the first place. A lot of people using ProTools are still trying to stuff everything in their mix into the 2-bus. Some people would say it's because ProTools is a flawed DAW program but I'll say that ANY mix to a 2-bus without sub-buses attached will collapse whether its analog or digital. I do not think that this method is available to PT LE users though I could be wrong on that. Using HD has opened a lot of things for me. And not because it's the almighty ProTools.....its just a tool that works for me.

Davedog, post: 430957, member: 4495 wrote: THIS was my point of my post in the first place. A lot of people using ProTools are still trying to stuff everything in their mix into the 2-bus. Some people would say it's because ProTools is a flawed DAW program but I'll say that ANY mix to a 2-bus without sub-buses attached will collapse whether its analog or digital. I do not think that this method is available to PT LE users though I could be wrong on that. Using HD has opened a lot of things for me. And not because it's the almighty ProTools.....its just a tool that works for me.

Trident 24 consoles have underpowered stock power that can cause serious lack of depth and headroom in their busing, perhaps because it's a split console?

I've been using mix buses for a bit now with good results. Although like anything it's probably best 'as needed' I've heard some awful phase problems particularly on snare drums by sending the signal via auxes and a drum bus. This was my mistake in DP, where in the older version you had to add pluggin (trim for instance) for the delay compensation to kick in.

I think of busing/grouping more in the light of gain staging.

Also. If one was a master and one was a mix, they wouldn't ever null out because of the mastering processing, even at equal gain, because compression and limiting and EQ are changing the file even in a small way. Or did I miss something?