Mixing ITB .. Which outboard??

[TLMUSIC]

Hi, im looking to buy some outboard equipment. So far i mix exclusively in the box. The only outboard i have currently is an Avalon 737.

What i am wondering is .. what could i get that will most benefit me .. im thinking perhaps a 2bus comp, as at least everything will be running through it, as opposed to just having one analog eq on just one track etc ...

I've been looking at the SSL G series comp .. I mix primarily hip-hop / pop ..

Please share any thoughts..

Thanks!

[peeder]

If your ADC + DAC costs less than $2000 stay in the box. Even if you have one great ADC + DAC, you'll need another DA to monitor with. See:

DAW -> DAC -> OTB comps etc. -> ADC -> DAW -> DAC #2 -> Monitors.

If you monitor without going back through the ADC/DAW/DAC loop you will not be monitoring what you will be delivering as you mix, and you will be unsatisfied with the result. If your converters are poor you will just lose imaging, depth and detail.

I use the SSL bus compressor in the Duende which is actually quite an inexpensive solution overall and I just monitor out a Lavry DA10.

I will run individual parts through outboard, print them back into the DAW, nudge them back to where they were for phase/timing accuracy, and mix from there ITB.

[manthe]

I have another take on this, a little different from above. While what he said is technically accurate, it is a balancing act. If you're using a *really* low-end audio IF, stay ITB. In my opinion, if your converters are mediocre or better, OTB is viable. You will lose a little fidelity on the round trip, but you have to judge the following:

- is it *actually* audible/noticeable. Be honest here. If you have a pretty decent monitoring setup and you still can not hear the difference, the VAST majority of anyone who hears your mixes will never be able to tell.

- If you do notice 'something', is it enough to offset the benefits of OTB mixing with good outboard gear.

I mix using a combo. I have mediocre converters and a decent monitoring setup. I notice a *slight* reduction in fidelity after an DAAD round-trip, but not nearly enough to stop me. I have some good outboard compressors, EQs, reverbs/FX, etc. Sometimes I just REALLY want the sound from my real 1176LN, instead of my UAD-1 copy.

One thing that I ALWAYS get with OTB that is extremely difficult and usually imposible ITB is depth. The difference is amazingly drastic, IMO. The SSL Duende actually does a really good job of imparting an OTB sound to the tracks it touches. It, alone has kept me ITB a lot of times that I would have gone out. If I buy a second one, who knows??

Anyhow, I have a lot of UAD and SSL plugs that I love and use. They are valuable to my mixes. Some time in the not-too-distant future, I'll invest an ass-load of money in high-end converters. At that point, OTB will not be a sacrifice. But, with proper gain staging and PROPERLY RECORDED SOURCE TRACKS (I follow the '-12 to -8 for digital' philosophy), I really do not suffer much loss at all. Also, if I know I will be doing a lot of OTB on a song, I will record at 96k. It just makes sense to me to have the extra information available in the files if I'm going to be doing a lot of destructive editing.

At any rate, to answer your question: I would go with a really good 2bus comp. Especially if this comp is versatile enough to use on other tracks...that is to say, something you could run a few things through individually before you strapped it to the Master Bus. A good 'glue' compressor can make a HUGE difference on a mix and is, IMO a great place to start when going OTB.

You might also think about some good tube amps with line drivers to run the master bus signal through before/after it hits the compressor. I do this all of the time to add character (I WONT say 'warmth' ) to my mix...just watch your gain staging carefully!

Good Luck!

[kats]

I'm pretty much with Menthe here. I'd stick ITB except for perhaps a good 2bus compressor. The only other outboard I would consider is whatever you need for tracking - so a good vox compressor would be usefull.

I hate to sound like a broken record, but I think you should choose your workflow and embrace it. I mix otb 100%, but if I were to switch I wouldn't kill my workflow and convenience of ITB just to be quasi OTB - it's the worst of both worlds IMO.

Go one way or the other and don't lose any sleep over it.

[intolerance]

What DAW are you using? If you're using PTLE, or anything else without PDC (plugin delay compensation) forget it, you're better off ITB. If however, you're using Cubase or the like with automatic PDC, and have decent DA/AD, then I'd ask yourself "What would be the biggest improvement via hardware, that plugs currently suffer?". I would vote against a 2 buss compressor personally, and lean towards a Distressor and decent outboard reverb. To me, this is where outboard is noticeably better than the like plugs. Eq's plugs are getting pretty darn close these days, specifically Waves V-Series, SSL, and API emulations. (the later being my favorite recently.) My 2 anyhow.

[lukejs]

HI...I've been debating this for awhile, and I think I'm finally gonna buy an external compressor and EQ.. I have an Apogee AD16-x, and Mini DAC, and plan on buying a Drawmer 1968 m.e., and an API 5500 for 2 bus duties, and "quasi" mastering... how's that combo grab ya ? think it would be a good combo ?

[peeder]

Originally Posted by intolerance

What DAW are you using? If you're using PTLE, or anything else without PDC (plugin delay compensation) forget it, you're better off ITB.

Note that on the 2 bus delay compensation is irrelevant, as nothing will go out of phase (as long as you're using the same pair of converters left and right!).

Delay Compensation isn't impossible otherwise, you need to use pings, but as I wrote I just print back individual parts and manually realign them.

DAWs that offer plugin delay compensation may still require manual delay compensation when going out to hardware.

[Andy M.]

If you have a digital I/O on your soundcard, then look for a processor with Digital I/O (can't think of any compressors offhand though). That way it doesn't matter how good or bad your converters are because they won't be used.

I once saw a creative au-dodgy card connected to a Kurzweil FX box via the SPDIF - sounded great...

[raal]

Originally Posted by TLMUSIC

Please share any thoughts..

mr. kats summed it up nicely. there are guys doing great things ITB. if you're already there just keep going is my advice, but i'm no one to talk as i do the exact opposite.

have a nice D.Control and i've been using and digging ITB EQs and recently some ITB comps have impressed, but i keep buying outboard.... it's like a sickness... so if you don't want to follow my advice, and you want to get something that'll be useful for tracking as well as mixing, maybe get yourself a nice all arounder like a GML 8900 or the original Avalon L44 w/built in parallel processing, or maybe a Fatso or a wonderful box like the Chandler Zener w/built in side chain filter for a little color. G comp is a great piece but maybe not quite as versatile. if you're flirting with the dark side, any of these might be a good place to start but if you do, don't say i didn't warn you... and welcome to the hotel california.

Originally Posted by kats

worst of both worlds

think i just found a name for the studio. my partner's is 'shabby road'.

[Animus]

Originally Posted by peeder

If your ADC + DAC costs less than $2000 stay in the box. Even if you have one great ADC + DAC, you'll need another DA to monitor with. See:

DAW -> DAC -> OTB comps etc. -> ADC -> DAW -> DAC #2 -> Monitors.

If you monitor without going back through the ADC/DAW/DAC loop you will not be monitoring what you will be delivering as you mix, and you will be unsatisfied with the result. If your converters are poor you will just lose imaging, depth and detail.

That's funny. I only have the RME ADI Pro range and 2 Multifaces and I routinely send out to my hardware all the time, including a summing mixer and a Massive Passive on the 2 buss, and the mix always sounds way better coming back in then it did going out. Paying 2k for 2 channels of ad/da would sound even better I am sure but not everyone has that kind of money for the amount of i/o they would need. I am quite happy with my studio now with the integration of real hardware.

[AndyUF2]

Originally Posted by Andy M.

If you have a digital I/O on your soundcard, then look for a processor with Digital I/O (can't think of any compressors offhand though). That way it doesn't matter how good or bad your converters are because they won't be used.

I once saw a creative au-dodgy card connected to a Kurzweil FX box via the SPDIF - sounded great...

TC makes a comp with SPIDIF in/out...haven't heard it, though, so I can't vouch for the sound. But it sounds tempting. Combine that with their new digital patchbay system, and you can get OTB without conversion.

[Electronique]

Originally Posted by Andy M.

If you have a digital I/O on your soundcard, then look for a processor with Digital I/O (can't think of any compressors offhand though). That way it doesn't matter how good or bad your converters are because they won't be used.

But If there were a compressor with digital in/out, it would itself have ad/da within.. And the onboard converters would probably be worse than those on your interface..??

Much better to go analog in to a compressor??

[Animus]

Originally Posted by AndyUF2

TC makes a comp with SPIDIF in/out...haven't heard it, though, so I can't vouch for the sound. But it sounds tempting. Combine that with their new digital patchbay system, and you can get OTB without conversion.

Well, there's still no difference. You are then using the converters on that compressor that has digital i/o. It still has to convert it to audio then back to digital, and what is the "quality" of the onboard converters. unless I am missing something here.

[severe]

Originally Posted by Animus

It still has to convert it to audio then back to digital

It does? What would it be converting running S/PDIF?


TLMUSIC, if you decide to run more outboard, why not just run your ADC channels to a mixer, use it's i/o, then run 2 channels of DAC back ITB. This way you've made one round-trip of conversion.

[TEMAS]

Originally Posted by Animus

Well, there's still no difference. You are then using the converters on that compressor that has digital i/o. It still has to convert it to audio then back to digital, and what is the "quality" of the onboard converters. unless I am missing something here.

Hmm, I agree. Unless the TC compressor mentioned is the Triple C, which is itself a digital compressor I believe. So there would be no conversion at all.

But I really think the original poster was searching for a quality piece of analog outboard, regardless of what converters he has.

[Animus]

Originally Posted by TEMAS

Hmm, I agree. Unless the TC compressor mentioned is the Triple C, which is itself a digital compressor I believe. So there would be no conversion at all.

But I really think the original poster was searching for a quality piece of analog outboard, regardless of what converters he has.

In that case, I would just stick to ITB as there are plenty of quality digital compressor plugins.

[peeder]

I got a question about latency compensation in PTLE via PM from this thread and I thought I'd share my response here for the general public good...


When I do an outboard pass on an individual part I bring it back in printed on a separate audio track and then I move it back in time to align with the original signal by eye, zooming in on the waveform. Then I inactivate the old track and mix from there with the new one.

When I'm trying to do real-time mixing with hardware inserts (or latency-including plugins) I load up ping regions (that I hand-drew with the pencil tool) into every audio track out past the end of the song, aligned using grid mode. Then I record those pings back in to my mixdown track (I record to disk rather than bounce to disk) and I see whether they are aligned or not (I set reverbs/delays to 100% dry for this). I time delay using the time adjuster plugin everything that comes earlier than the slowest of the pings, sometimes on the individual tracks, sometimes on busses. I sometimes need a couple layers of delays in order to align sends/returns when I have latency on the busses.

This is one reason why I generally just print back hardware passes and manually realign (the other reason is recall). And when using UAD or Duende I will try to make sure there's an equal number of those plugs everywhere so that they all have the same latencies.

And I'll reiterate that most DAWs have this problem even if they claim automatic latency compensation, and this is the general solution. This could be automated and a few DAWs do automate via pings. I predict all will by the end of the decade but you're going to have to really bear down on the programmers for this it appears. No idea why this wasn't solved generally 10 years ago, but probably most of the DAW coders are doing electronica or something where phase is less of a problem.

[manthe]

SONAR 7's new hardware insert feature is flippin' amazing for this!!

It is basically a 'plugin' that allows you to load an out->input path and does continuous 'pings' to keep things time aligned, in real time. It does a fantastic job, even on multiple tracks. Last night, I had to acoustic guitar tracks running to 2, separate EQs, then to 2 separate compressors. I had a vocal track from the same song going to an external reverb. All 3 were perfectly aligned with everything else.

S7 has 'revolutionized' my OTB mixing. It really makes it viable and allows me to do tweaks in real time as well as to hear what combinations of multiple tracks, going OTB sound like, in real time...without the headache of having to print destructively, time align, test, then move on to the next track, print destructively, time align.....etc.

SONAR is really for real now, IMO. I wish they had it for Mac. I use PCs and Macs. I use PTLE on my Mac (as well as MOTU DP 4.5). I would LOVE to be able to use SONAR! That would make life so much easier for me.

Sorry...OT

Originally Posted by peeder

I got a question about latency compensation in PTLE via PM from this thread and I thought I'd share my response here for the general public good...


When I do an outboard pass on an individual part I bring it back in printed on a separate audio track and then I move it back in time to align with the original signal by eye, zooming in on the waveform. Then I inactivate the old track and mix from there with the new one.

When I'm trying to do real-time mixing with hardware inserts (or latency-including plugins) I load up ping regions (that I hand-drew with the pencil tool) into every audio track out past the end of the song, aligned using grid mode. Then I record those pings back in to my mixdown track (I record to disk rather than bounce to disk) and I see whether they are aligned or not (I set reverbs/delays to 100% dry for this). I time delay using the time adjuster plugin everything that comes earlier than the slowest of the pings, sometimes on the individual tracks, sometimes on busses. I sometimes need a couple layers of delays in order to align sends/returns when I have latency on the busses.

This is one reason why I generally just print back hardware passes and manually realign (the other reason is recall). And when using UAD or Duende I will try to make sure there's an equal number of those plugs everywhere so that they all have the same latencies.

And I'll reiterate that most DAWs have this problem even if they claim automatic latency compensation, and this is the general solution. This could be automated and a few DAWs do automate via pings. I predict all will by the end of the decade but you're going to have to really bear down on the programmers for this it appears. No idea why this wasn't solved generally 10 years ago, but probably most of the DAW coders are doing electronica or something where phase is less of a problem.

[severe]

Originally Posted by peeder

I got a question about latency compensation in PTLE via PM from this thread and I thought I'd share my response here for the general public good...


When I do an outboard pass on an individual part I bring it back in printed on a separate audio track and then I move it back in time to align with the original signal by eye, zooming in on the waveform. Then I inactivate the old track and mix from there with the new one.

When I'm trying to do real-time mixing with hardware inserts (or latency-including plugins) I load up ping regions (that I hand-drew with the pencil tool) into every audio track out past the end of the song, aligned using grid mode. Then I record those pings back in to my mixdown track (I record to disk rather than bounce to disk) and I see whether they are aligned or not (I set reverbs/delays to 100% dry for this). I time delay using the time adjuster plugin everything that comes earlier than the slowest of the pings, sometimes on the individual tracks, sometimes on busses. I sometimes need a couple layers of delays in order to align sends/returns when I have latency on the busses.

This is one reason why I generally just print back hardware passes and manually realign (the other reason is recall). And when using UAD or Duende I will try to make sure there's an equal number of those plugs everywhere so that they all have the same latencies...

Holy sh*t, what a pain in the a*s!

One could tell me, "it ain't all that bad" till they're blue in the face, but that's fookin ridiculous at this point in the game.

Is this all run on a computer that you have to pedal?

[Brad McGowan]

My advice to the original poster would be to find one piece of gear that possesses a sonic trait, “color”, “mojo”, etc. that you dream about in your sleep and run your whole ITB mix through that. It could be a mic preamp, a compressor, an EQ, a mixer, a cassette 4-track, a vintage hi-fi preamp, a tube DI, a Casio SK1…anything. Just find something that makes the music you are creating sound like MUSIC and run your mixes through it. I think doing so can really bring an otherwise flat and 2D ITB mix to a whole new level of sonic joy. You’re going to have to experiment and listen and it will take much time…perhaps years. But you’ll know when you’ve found something that you connect with. And don’t get caught up in the popularity trends on this forum. There have been certain items over the years that people rave about. Some of them I was seduced into buying, but later decided they were not my cup of tea. It’s perfectly fine if you think the Gearslutz flavor of the week is boring sounding, or too dirty. Other’s sense of aesthetics is really irrevelant to the work you are doing in your studio. For instance, there seem to be a lot of folks here that work on what I would call “modern rock”. A lot of those guys are having great success using the Steven Slate samples in their mixes. However, despite all the rave reviews, that product would be totally useless to me since I don’t work on music that calls for those sounds. Gear is a very personal choice, and often it takes many years to figure it all out. Trial and error is really the best way to arrive at the best answer. Make Visa and Mastercard your new best friends.

Good luck!

Brad

[peeder]

Originally Posted by severe

Holy sh*t, what a pain in the a*s!

One could tell me, "it ain't all that bad" till they're blue in the face, but that's fookin ridiculous at this point in the game.

Is this all run on a computer that you have to pedal?

Yes it's a pain in the ass but a mix can take 8 hours and this takes 5 minutes once you're in the swing of it. It's not enough to get me to bail on my tools.

But it is EMBARASSING for the programmers involved. I would go to my manager and tell them that I was ashamed of working on a product that had this problem. It also affects HD even though their latency compensation is claimed to be automatic. Many of the plugins sadly lie about how much time they are taking to process. So a ping strategy is really the best way of doing things although the delay-type plugins must be set dry to not show confusing results.

[bass man]

OK , i am little bit confused here !

Are you saying that if i have recorded like lets say 8 tracks

5 for the drums , 1 for the sax ,one for the double bass and one for the guitar and u use lets say Waves r comp on the bass and tl eq on the guitar that the bass and the guitar are going to have latency in the mix or monitoring and come "later" than the OHs or sax ????
Is that what you are saying , that the plug ins are having latency ?
If you have two channels one with plug ins and one with out they wont be in the same place ? When you bounce the mix treated and untreated tracks are not going to be at the same place ?
Please answer ,this sounds scary and i never noticed that in my mixes ...

[thethrillfactor]

Originally Posted by peeder

Yes it's a pain in the ass but a mix can take 8 hours and this takes 5 minutes once you're in the swing of it. It's not enough to get me to bail on my tools.

But it is EMBARASSING for the programmers involved. I would go to my manager and tell them that I was ashamed of working on a product that had this problem. It also affects HD even though their latency compensation is claimed to be automatic. Many of the plugins sadly lie about how much time they are taking to process. So a ping strategy is really the best way of doing things although the delay-type plugins must be set dry to not show confusing results.

I think lie is the wrong choice of word here.

What generally happens in the plug in world is that new plugs are released after the software has been sold so there is no way to update the compensation until a later version is sold with the correct one. Also keep in mind that HD compensation allows compensation up to a certain number of samples and once you pass this point it can't compensate.

And lastly tracking hardware inserts and shifting them back is not the most ideal either. The amount of latency going in and out of a converter is not a whole number but a whole number broken down to sub samples. Unfortunately no DAW that's out there(including PTools) shows a subsample grid display(and we are talking about 4 decimal points). The lowest they show is a 1 sample grid and while this is cool for plug ins its just not as accurate for analog inserts or digital processors.

[robot gigante]

Originally Posted by TLMUSIC

Hi, im looking to buy some outboard equipment. So far i mix exclusively in the box. The only outboard i have currently is an Avalon 737.

What i am wondering is .. what could i get that will most benefit me .. im thinking perhaps a 2bus comp, as at least everything will be running through it, as opposed to just having one analog eq on just one track etc ...

I've been looking at the SSL G series comp .. I mix primarily hip-hop / pop ..

Please share any thoughts..

Thanks!

What aren't you specifically getting out of what you have now that you would like to have? Break it down.

ITB Reverbs that you have working OK for you? Compressors? EQ's? Are you wanting a 2-bus comp because you like to compress the 2-bus ITB but the plugins aren't working for you?

[elcubo]

one of the best threads...keep posting please...

[peeder]

Originally Posted by bass man

OK , i am little bit confused here !

Are you saying that if i have recorded like lets say 8 tracks

5 for the drums , 1 for the sax ,one for the double bass and one for the guitar and u use lets say Waves r comp on the bass and tl eq on the guitar that the bass and the guitar are going to have latency in the mix or monitoring and come "later" than the OHs or sax ????
Is that what you are saying , that the plug ins are having latency ?
If you have two channels one with plug ins and one with out they wont be in the same place ? When you bounce the mix treated and untreated tracks are not going to be at the same place ?
Please answer ,this sounds scary and i never noticed that in my mixes ...

Yep this is a problem that is solved to varying extents by varying DAWs. Pro Tools LE is uniquely lame in this category as it doesn't make any attempt to delay compensate at all. Many of the plugins that ship with it don't have any latency. But your Ren Comp iirc has a fixed latency of 64 samples...so tracks run through it in PTLE will play back 64 samples later than everything else. This will subtly affect the groove of your tracks in a way that might escape you but you'll still feel it, which is the worst kind of bug...your mix sucks and you don't know why!

Worse is when you try to parallel compress or do something else where phase variance shows up as nasty comb filtering. Then you really hear it...maybe that isn't worse since you're at least alerted to the problem and have to fix it or give up that strategy. You can also get slap echoes you didn't want.

DAWs other than PTLE try to solve this but break down when going out on hardware inserts. Some DAWs may have solved that problem too (Sonar 7 is mentioned above?) but I've never tried them to see. Some people think you just set it for your hardware and you can then forget it but that is not my experience. Also it appears that firewire has variable latency (about 20 samples on my machine) from startup to startup, meaning that you will have to recompensate every time you boot your machine!

Programmers would generally like to do something sexy like a new virtual instrument or whatever rather than solve a nagging pain in the ass like this. It's dirty laundry, made more complex by the need for 3rd party cooperation, and it's unseemly to market a bug's absence as a feature.

I don't tolerate any comb filtering (though I haven't gone to the extent of sub-samples! Mr. thrillfactor, do you realize that the comb filters on subsample phase variations will only affect frequencies above Nyquist? Or have I got that wrong somehow?) or unwanted slap delays but I will tolerate latency variations of a few samples here and there which don't cause comb filtering because the tracks are isolated. I check every mix though to be certain.

[thethrillfactor]

Originally Posted by peeder

I don't tolerate any comb filtering (though I haven't gone to the extent of sub-samples! Mr. thrillfactor, do you realize that the comb filters on subsample phase variations will only affect frequencies above Nyquist? Or have I got that wrong somehow?) but I will tolerate latency variations of a few samples here and there which don't cause comb filtering. I check every mix though to be certain.

The combfiltering becomes an issue when doing any parallel processing or if you use plug ins or hardware inserts on some live drums and not others. You will get phase cancellations in the higher freqs depending on the closeness of the compensation.

The latency becomes less noticeable at the higher sampling rates.

Lastly what i meant was that the round trip through the converters is not a whole number. Like i think at 44.1k on a Digi 192/Aurora 16 the manual reads 98 samples?(i forget).
But in actuality its 98.something,something,something,something. Those something,something,somethings are the subsamples divisions.

As of right now there is no way to fully compensate unfortunately.

Even using the Eventide Precision Time delay/align(which breaks the samples down to subsamples) i wasn't able to get a perfect null(because the divisions are not finite enough even at the higher rates which give you more choices). This is still one place where analog is still king.

Now can you hear it is the question.

Before i started using the Eventide plug in i wasn't sure how much of a difference it would make, but i knew from doing parallel processing so long on a console their is a slight timbral difference when doing it ITB even with the latency correction. When i tried it immediately you can hear the sounds are punchier and clearer even though at 44.1K the plug in rolls off the top somewhat. I noticed this the most when doing a parallel drum sub which along with the kicks and snares was my biggest concern(bass is not as noticeable because there are hardly any phase shifts there). To my ears though it still sounds tighter and punchier on a console but at least its close.