MIXING Pro Tools 10/11 ANALOG SUMMING & DELAY COMPENSATION HELP!
Micah421
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#1
3rd February 2014
Old 3rd February 2014
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MIXING Pro Tools 10/11 ANALOG SUMMING & DELAY COMPENSATION HELP!

Hey everyone,

Ive been doing some test and I have found some "maddening" results.

My home rig consist of:

Mac 3.7 Quad i7 24 gigs Ram
Apollo 8/8
Lynx Aurora 8 ADAT
Dangerous D-Box
Pro-Tools 10/11 HD and Native
Lots Of Outboard Gear used as "Hardware Inserts"

In my mix template I route all my tracks(they all usually have different plugins creating different algo's of latency) to the appropriate analog summing inputs...8 total, 6 stereo 2 mono(drums/instruments/bass/vocals/fx returns). From those "summed" inputs coming from my D box I route the stereo returns to a Stereo Aux Track that I usually have a few plugins AND Hardware Inserts on to process through, such as a Limiter, EQ ect. From that Stereo Aux track I then route it to a Stereo Audio Track to print my final mix and Monitor from.

For this Test:

Hardware Inserts Apollo 1-8
Summing is handled by Apollo--->ADAT Lynx 8/8 analog

Keys------> Lynx out--->Dangerous 1-2 input
Drums----> Lynx out--->Dangerous 3-4 input
Guitars---> Lynx out--->Dangerous 5-6 input
Bass------> Lynx out--->Dangerous 7 input
Vocals----> Lynx out--->Dangerous 8 input

Dangerous 1-2 Outs-----Lynx 1-2 inputs AuxTrack (Master Aux)

Master Aux(with plugins/hardware inserts)------->Stereo Audio Print Track

I noticed however that each "stem" (drums, keys, guitars, bass, vocals ect) gets printed at different latency "sample timings" . I have attached a screen shot to show the results. I used tab-to-transient to measure the amount of latency on each "bus" print.

Here are the results in this particular test. Please keep in mind obviously each real world "mix"session the latency times will vari depending on the plugins used.

Delay compensation set to "long"
Low Latency set to off
I tried control+apple on the "print" track with same results

Keys were off by : 5907 samples
Drums were off by : 5903 samples
Guitars were off my 5903 samples
Bass was off by 5899 samples
Vocals were off by 5911 samples


Now, why of course does this matter? Some of you may be asking.

Well this means every "stem" (drums, guitars, keys, vocals, bass ect) running through the summing is not phase aligned, impacting the quality of the final mix., If I'm thinking of this correctly.

So my question is, does everyone who uses hardware inserts and analog summing just live with this? Possibly unknowing? Are there any work-arounds?

I went a step further and tried this bypassing any hardware and routing everything through AUX buses inside pf pro-tools with similar results(not as many samples but still all consistently different.

Hope I've explained this well enough to understand, its difficult to think of the proper way to without being here to visually see results but hopefully screen shot helps.
Attached Thumbnails
MIXING Pro Tools 10/11 ANALOG SUMMING & DELAY COMPENSATION HELP!-screen-shot-2014-02-02-10.52.05-pm.jpg  
#2
3rd February 2014
Old 3rd February 2014
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Well there you have it.
Protools "vanilla" non HD version (so without the cards) will never ever work great with ADC on the "hardware insert" plugin. Try to paralel comp on a snare. It will not line up. Why? The DEA/AAE is crippled in a sort of way.
Even you put the in the options the offset of your AD/DA the compensatie will always drift in samples. You hit play. it works, you stop, replay... phase. You change your buffer... it stop working. It's a nightmare with 3rd party interfaces to get it right.
Scott Greiner has some great post here on the forum why and how to find a solution.

What you can do is that you stem out your instruments from the converter in to you summingbox. Between these 2 you putt your hardware processing (fx, compression or eq or what ever). But don't use these as a insert with the "hardware insert" plugin. The extra AD/DA trip and the funky ADC will never aline how it should be... Only if you go the Avid HD(x)/Native PCI/Thunderbolt interface card way with the AVID converters... And even then I counter props with it...
#3
3rd February 2014
Old 3rd February 2014
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Quote:
Originally Posted by Micah421 View Post
Hey everyone,

Ive been doing some test and I have found some "maddening" results.

My home rig consist of:

Mac 3.7 Quad i7 24 gigs Ram
Apollo 8/8
Lynx Aurora 8 ADAT
Dangerous D-Box
Pro-Tools 10/11 HD and Native
Lots Of Outboard Gear used as "Hardware Inserts"

In my mix template I route all my tracks(they all usually have different plugins creating different algo's of latency) to the appropriate analog summing inputs...8 total, 6 stereo 2 mono(drums/instruments/bass/vocals/fx returns). From those "summed" inputs coming from my D box I route the stereo returns to a Stereo Aux Track that I usually have a few plugins AND Hardware Inserts on to process through, such as a Limiter, EQ ect. From that Stereo Aux track I then route it to a Stereo Audio Track to print my final mix and Monitor from.

For this Test:

Hardware Inserts Apollo 1-8
Summing is handled by Apollo--->ADAT Lynx 8/8 analog

Keys------> Lynx out--->Dangerous 1-2 input
Drums----> Lynx out--->Dangerous 3-4 input
Guitars---> Lynx out--->Dangerous 5-6 input
Bass------> Lynx out--->Dangerous 7 input
Vocals----> Lynx out--->Dangerous 8 input

Dangerous 1-2 Outs-----Lynx 1-2 inputs AuxTrack (Master Aux)

Master Aux(with plugins/hardware inserts)------->Stereo Audio Print Track

I noticed however that each "stem" (drums, keys, guitars, bass, vocals ect) gets printed at different latency "sample timings" . I have attached a screen shot to show the results. I used tab-to-transient to measure the amount of latency on each "bus" print.

Here are the results in this particular test. Please keep in mind obviously each real world "mix"session the latency times will vari depending on the plugins used.

Delay compensation set to "long"
Low Latency set to off
I tried control+apple on the "print" track with same results

Keys were off by : 5907 samples
Drums were off by : 5903 samples
Guitars were off my 5903 samples
Bass was off by 5899 samples
Vocals were off by 5911 samples


Now, why of course does this matter? Some of you may be asking.

Well this means every "stem" (drums, guitars, keys, vocals, bass ect) running through the summing is not phase aligned, impacting the quality of the final mix., If I'm thinking of this correctly.

So my question is, does everyone who uses hardware inserts and analog summing just live with this? Possibly unknowing? Are there any work-arounds?

I went a step further and tried this bypassing any hardware and routing everything through AUX buses inside pf pro-tools with similar results(not as many samples but still all consistently different.

Hope I've explained this well enough to understand, its difficult to think of the proper way to without being here to visually see results but hopefully screen shot helps.

You aren't using protools properly from first glance.

Protools ADC calculates the delay adjustments so the audio aligns at the master faders. You don't have any master faders. Create a master fader for each stem and run the test again and see how the results look.
#4
3rd February 2014
Old 3rd February 2014
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Quote:
Originally Posted by knownby View Post
You aren't using protools properly from first glance.

Protools ADC calculates the delay adjustments so the audio aligns at the master faders. You don't have any master faders. Create a master fader for each stem and run the test again and see how the results look.
Does this apply only to third party interfaces or must it be done also with Avid interfaces? I am using PT HD 11 Native with an AVID HD I/O and an external summing unit.
#5
3rd February 2014
Old 3rd February 2014
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Hmm...not quite. The master faders are always "there" whether you've got them in the session or not.

As to the OP - part of it is possibly down to the interface driver. With a 100% avid system, PT knows what it's looking at converter-wise, and compensates accordingly. With a 3rd party system, it's reliant on that driver. As soon as you connect converters via digital audio, it has no way of knowing what's connected, and manual compensation might be necessary.

A few samples between totally unrelated stems isn't going to cause problems. A few hundred samples will make a difference to the timing, but again not phase. The only time you'll notice a phase discrepancy is if you're splitting a multi-miked source across different conversion - non/coherent sources (eg bass and drums) aren't going to phase-cancel any more than they will without a few sample variation.

Quote:
Originally Posted by knownby View Post
You aren't using protools properly from first glance.

Protools ADC calculates the delay adjustments so the audio aligns at the master faders. You don't have any master faders. Create a master fader for each stem and run the test again and see how the results look.
#6
3rd February 2014
Old 3rd February 2014
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Quote:
Originally Posted by Blues Bird View Post
Does this apply only to third party interfaces or must it be done also with Avid interfaces? I am using PT HD 11 Native with an AVID HD I/O and an external summing unit.
For HD10 with TDM cards it does. I can't comment on the rest as I haven't used them.

Try it and see? Just duplicate a bass track - route them out two different interface outputs without using master faders and place a high latency plugin (noise reduction for example) on one and see if they are still in sync.

You could also try it with a hardware insert on one.

If they are out of sync, try creating master faders for the two outputs and see if it fixes it.
#7
3rd February 2014
Old 3rd February 2014
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Quote:
Originally Posted by psycho_monkey View Post
Hmm...not quite. The master faders are always "there" whether you've got them in the session or not.
That is true, but I have witnessed issues with ADC that only get resolved when you physically create the master faders.
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3rd February 2014
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This is yet another reason I prefer Cubase over PT on a Native system : ASIO. I use a lot of hardware inserts and never have a problem with samples being out of phase(or time I should say). Cubase "pings" the interface and calculates the latency itself and thus compensates for it automatically. Very intuitive. I have never had a good result trying to use delay compensation in PT however. The "having to set it yourself" for each different interface is maddening and seems to never work.

In my estimation, if one insists on using PT and mixing with hardware, HD is the only way to go.
Micah421
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#9
3rd February 2014
Old 3rd February 2014
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Quote:
Originally Posted by cowboycoalminer View Post
This is yet another reason I prefer Cubase over PT on a Native system : ASIO. I use a lot of hardware inserts and never have a problem with samples being out of phase. Cubase "pings" the interface and calculates the latency itself and thus compensates for it automatically. Very intuitive. I have never had a good result trying to use delay compensation in PT however. The "having to set it yourself" for each different interface is maddening and seems to never work.

In my estimation, if one insists on using PT and mixing with hardware, HD is the only way to go.
Hey Cowboy, can you send me a link where it said that Cubase "pings" your hardware interface to calculate your latency to compensate automatically? Man that's the first I've heard of this.

Why in the world would Pro Tools not do this??? I have to read that Cubase does this and try it for myself as a test.
Micah421
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#10
3rd February 2014
Old 3rd February 2014
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Quote:
Originally Posted by knownby View Post
That is true, but I have witnessed issues with ADC that only get resolved when you physically create the master faders.
I'm possibly not understanding you correctly. The whole purpose of Analog summing is to route through separate buses. You cant route I/0 with a master track, it always controls your stereo bus. Are you sure you understand, or am I confused as to what your saying?
Micah421
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3rd February 2014
Old 3rd February 2014
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Quote:
Originally Posted by LeMauce View Post
Well there you have it.
Protools "vanilla" non HD version (so without the cards) will never ever work great with ADC on the "hardware insert" plugin. Try to paralel comp on a snare. It will not line up. Why? The DEA/AAE is crippled in a sort of way.
Even you put the in the options the offset of your AD/DA the compensatie will always drift in samples. You hit play. it works, you stop, replay... phase. You change your buffer... it stop working. It's a nightmare with 3rd party interfaces to get it right.
Scott Greiner has some great post here on the forum why and how to find a solution.

What you can do is that you stem out your instruments from the converter in to you summingbox. Between these 2 you putt your hardware processing (fx, compression or eq or what ever). But don't use these as a insert with the "hardware insert" plugin. The extra AD/DA trip and the funky ADC will never aline how it should be... Only if you go the Avid HD(x)/Native PCI/Thunderbolt interface card way with the AVID converters... And even then I counter props with it...
Thanks for taking time to reply! I'll search for Scott Greiner

Its quite maddening, especially when I've spent all this money on gear, when I print from my Analog Summing I need everything to be in time obviously.

I've got a buddy that just bought 3 HDX cards for his post room. He does a lot of 5.1 mixes that involve complex aux routing and he swears everytime he uses Native plugs his print masters are off and he has to line them up after printing back to the original 2-pop and consolidate the file again. He just spent 14k....does not seem acceptable...am I right?
Micah421
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#12
3rd February 2014
Old 3rd February 2014
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Quote:
Originally Posted by psycho_monkey View Post
Hmm...not quite. The master faders are always "there" whether you've got them in the session or not.

As to the OP - part of it is possibly down to the interface driver. With a 100% avid system, PT knows what it's looking at converter-wise, and compensates accordingly. With a 3rd party system, it's reliant on that driver. As soon as you connect converters via digital audio, it has no way of knowing what's connected, and manual compensation might be necessary.

A few samples between totally unrelated stems isn't going to cause problems. A few hundred samples will make a difference to the timing, but again not phase. The only time you'll notice a phase discrepancy is if you're splitting a multi-miked source across different conversion - non/coherent sources (eg bass and drums) aren't going to phase-cancel any more than they will without a few sample variation.
Thanks for taking the time to reply!

Your right it involves the timing not phase....good call....man timing is everything in music though.....ESPECIALLY when doing parallel compression with hardware inserts! Man it is a nightmare

should I not worry about my results?
#13
3rd February 2014
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Quote:
Originally Posted by Micah421 View Post
I'm possibly not understanding you correctly. The whole purpose of Analog summing is to route through separate buses. You cant route I/0 with a master track, it always controls your stereo bus. Are you sure you understand, or am I confused as to what your saying?
In PT you can create a so called master fader for every physical output, not only for your main out. ( this can be especially be useful to make sure that you hit your outboard or your summing mixer not to hard (usually you want the output level to be +4 dBu.)
#14
3rd February 2014
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Quote:
Originally Posted by Micah421 View Post
Hey Cowboy, can you send me a link where it said that Cubase "pings" your hardware interface to calculate your latency to compensate automatically? Man that's the first I've heard of this.

Why in the world would Pro Tools not do this??? I have to read that Cubase does this and try it for myself as a test.
This is from the C 4 manual. I'm sure the language may have changed a bit in V7;

• To us eexternal effects,you need audio hardware with multiple inputs and outputs. To use external instruments, a MIDI interface must be connected to your computer.
An external effect will require at least one input and one output (or input/ output pairs for stereo effects) – in addition to the input/output ports you use for recording and monitoring.
?20
VST Connections: Setting up input and output busses
• As always,audio hardware with low-latency drivers is a good thing to have.
Cubase will compensate for the input/output latency and ensure that the audio processed through external effects isn’t shifted in time.

Now from real world use, I can tell you when you pull up the Hardware insert window from in DAW after strapping something on, there is a button on the GUI so that you can ping the round trip manually. I have never had to do this. It always comes up to zero samples from the start because like I said, Cubase does the work for you when the insert is created.

If you want to look into this further, simply go to Steinbergs webpage and go to downloads. From there you can download the entire manual and read for yourself.

You've seen my posts about Cubase before. This is yet another reason I like Cubase. It's a rock solid DAW for a Native system on Mac or Windows. I'm told it runs even better on Windows. Hard to imagine cause it never hiccups on my Mac.
#15
3rd February 2014
Old 3rd February 2014
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Quote:
Originally Posted by Micah421 View Post
Hey Cowboy, can you send me a link where it said that Cubase "pings" your hardware interface to calculate your latency to compensate automatically? Man that's the first I've heard of this.

Why in the world would Pro Tools not do this??? I have to read that Cubase does this and try it for myself as a test.
Because pt is designed as a closed system on the HD side (and that goes for HD Tdm and HD native), there's no need - the latency is "known".

On the vanilla pt (which is effectively what you're running) the "non-standard" interface thing is still fairly new...and yes, it should be a feature. Again, if you were running an Avid interface here, it would "know" latencies.
#16
3rd February 2014
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Quote:
Originally Posted by knownby View Post
That is true, but I have witnessed issues with ADC that only get resolved when you physically create the master faders.
This sounds very odd - and I will check it before commenting further.
#17
3rd February 2014
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Quote:
Originally Posted by psycho_monkey View Post
Again, if you were running an Avid interface here, it would "know" latencies.
Right. So I'm wondering if using say, Apogee Symphony converters on an HD rig would pose the same problems as are encountered with Native systems. Since the Apogee is not a known converter.

Is this why the Symphony has PT HD mode?
#18
3rd February 2014
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Quote:
Originally Posted by Micah421 View Post
I'm possibly not understanding you correctly. The whole purpose of Analog summing is to route through separate buses. You cant route I/0 with a master track, it always controls your stereo bus. Are you sure you understand, or am I confused as to what your saying?

Master faders can be assigned to every physical output of your audio interface.
Just above the fader on master tracks is a routing drop down menu that let's you select the routing.

If you use 16 outputs on your interface, just create 16 mono master tracks at once and they will automatically be routed to outputs 1 through 16.
#19
4th February 2014
Old 4th February 2014
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Quote:
Originally Posted by cowboycoalminer View Post
Right. So I'm wondering if using say, Apogee Symphony converters on an HD rig would pose the same problems as are encountered with Native systems. Since the Apogee is not a known converter.

Is this why the Symphony has PT HD mode?
HD "sees" the Apogees and Lynxes etc as 192s or HDIOs - they emulate the Avid interfaces, so as far as PT sees, it's just dealing with it's own interfaces.

Historically, Lynx made a big thing that their boxes had the same converter latency (for this is what's making the difference) as the Digi interfaces, so therefore it should be a non-issue.

This wasn't the case for the AD/DA-16Xes, but I read in another thread that Don from Apogee said that the Symphony (at least when connected in PT HD mode) also has the same latency as the Avid HD IO PT sees it as.

This is different to connecting a converter digitally - obviously here there's no data stream, only audio, so PT has no way of seeing what is connected. Not a problem when mixing, or recording providing you're not splitting multimiked sources across converters, because the differences are only a few samples. More of an issue for hardware inserts.

In theory - connecting the symphony via eg USB and then using PT vanilla with the Apogee USB driver SHOULD produce a known result - the driver knows what's connected, and should compensate for it's own converter. But whether or not PT adds a "default" correction or not I don't know. I did a test on my old system (002 running with a Lynx via ADAT). The 002 was I think 1 sample out; the Lynx 16 samples EARLY (when connected via ADAT ie PT was over-compensating, presumably the 002 has a longer converter latency than the Lynx). This is a slightly different situation to having a 3rd party driver though.
#20
4th February 2014
Old 4th February 2014
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Quote:
Originally Posted by knownby View Post
You aren't using protools properly from first glance.

Protools ADC calculates the delay adjustments so the audio aligns at the master faders. You don't have any master faders. Create a master fader for each stem and run the test again and see how the results look.
Quote:
Originally Posted by knownby View Post
That is true, but I have witnessed issues with ADC that only get resolved when you physically create the master faders.
Right. I've just tested this on my system (HD3 with Avid IO/192s) and as I suspected, no problems.

I've got a mult of a track - going to 2 outputs. I put an RX3 denoiser (turned all the way off, but not bypassed) on one of the tracks. No phase slip, delay comp happy, no problems.

What you suggest is really technically unfeasible (as I said - master faders have NO effect on delay comp - in fact, any plugins you put on them are NOT delay compensated themselves!) and are always in the path, even if not visible.

I don't know what the OP's problem is, or what behaviour you saw, but this is unlikely to be the issue.
#21
4th February 2014
Old 4th February 2014
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Quote:
Originally Posted by psycho_monkey View Post
I don't know what the OP's problem is, or what behaviour you saw, but this is unlikely to be the issue.

So tell him not to try it. It doesn't bother me.

Everyone can just sit here and ponder why things that are 'technically unfeasible' are happening. Its meant to work... right?
#22
4th February 2014
Old 4th February 2014
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Quote:
Originally Posted by knownby View Post
So tell him not to try it. It doesn't bother me.

Everyone can just sit here and ponder why things that are 'technically unfeasible' are happening. Its meant to work... right?
Erm...you just suggested something I thought was technically incorrect...I tested it to make sure....it WAS technically incorrect.

Of course things that are "unfeasible" can happen. In this case - if what you described WAS happening, there's something wrong with the setup, because in a correctly set up rig (ie mine in this case) that doesn't happen.

Aren't you concerned that your rig appears to behave in a way that is not intended?

I'm not able to test this with a non-HD rig at the moment...but I'd be very surprised if the PT architecture varies between the versions. Delay comp simply doesn't depend on master faders, and I tested that to make sure!
#23
4th February 2014
Old 4th February 2014
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Quote:
Originally Posted by psycho_monkey View Post
Right. I've just tested this on my system (HD3 with Avid IO/192s) and as I suspected, no problems.

I've got a mult of a track - going to 2 outputs. I put an RX3 denoiser (turned all the way off, but not bypassed) on one of the tracks. No phase slip, delay comp happy, no problems.

What you suggest is really technically unfeasible (as I said - master faders have NO effect on delay comp - in fact, any plugins you put on them are NOT delay compensated themselves!) and are always in the path, even if not visible.

I don't know what the OP's problem is, or what behaviour you saw, but this is unlikely to be the issue.
Could please describe how exactly you've measured whether there is a phase slip or not. I want to try it with my system.
Thank you.
#24
6th February 2014
Old 6th February 2014
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Quote:
Originally Posted by Micah421 View Post
Thanks for taking time to reply! I'll search for Scott Greiner

Its quite maddening, especially when I've spent all this money on gear, when I print from my Analog Summing I need everything to be in time obviously.

I've got a buddy that just bought 3 HDX cards for his post room. He does a lot of 5.1 mixes that involve complex aux routing and he swears everytime he uses Native plugs his print masters are off and he has to line them up after printing back to the original 2-pop and consolidate the file again. He just spent 14k....does not seem acceptable...am I right?

Just asking... you use a UAD apollo on a Imac, right? Isn't the uad console panel (you know that mixerboard, dsp controler programme of your soundcard) setup funky with some strange routing/plugins on some channelouts and some not?
And second... Using 2 diff kinds of soundcards with diff DA converters/wordclock is a bit asking for troubles in "the perfect in sync department".

Put try to do this... protools. AUX tracks for your stems. Aux -> DA -> (Hardware, comp/eq/what ever) -> summing -> DA back to protools. Still problems?
Just for the love of god, don't use "hardware insert" plugin in protools without AVID Native/HDX card+converters... it will never lineup/work.
#25
1 Week Ago
Old 1 Week Ago
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Hello,

I would like to ask what do you mean by: Create a master fader for each stem and run the test again and see how the results look.?
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