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ADAT will add latency? or not?
Old 11th April 2016
  #1
Gear Head
 

ADAT will add latency? or not?

Hi all! I am finalizing my upgrades and wanted your opinions or help if you have the time. I am thinking of running a Motu 16a, through thunderbolt on a powerful MacBook Pro, Logic pro X. I will have those 16 outs sent to a 2-Bus+. I have an Apollo Duo, I would like to set it up as an independent ADAT converter and use all 8 ADAT I/O channels. I'll record at 24 96.
My questions.
1. Will Going over ADAT be the same latency as recording straight into the MOTU? Recording in low latency mode, for example, guitar through the Apollo inputs and Logic plugins) What can I expect? Also, Will playback through the ADAT/APOLLO be in sync with the outs of the MOTU?
2. As far as I know, It doesn't matter which of the converters is first in the chain, I can clock both to whichever's clock sounds better. Is that right? Or is there something more I should know here?
Thank you!
Old 11th April 2016
  #2
Lives for gear
Light moves pretty fast. If the interfaces are properly synced, there won't be issues. Word clock works better than adat syncing for my setup, but I haven't used your particular interfaces.
If it's not working you're likely to get drop-outs (pops and clicks), not a noticeable latency issue.
Old 11th April 2016
  #3
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there will be a difference between the analog inputs and the ADAT inputs .. since you are using 2 different A/D converters (motu, apollo). there will be processing difference between the 2 converters .. not sure how to find out exactly how much, except for recording the same source with one microphone split between the motu and apollo converters and then measuring it in your DAW to see if they line up or not ..

my guess is they will NOT line up ..
Old 11th April 2016
  #4
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Maybe a tinnnnnny bit (sub 1ms I'm guessing) due to processing differences in the two converters, as mentioned above.
The much bigger problem is syncing everything, get a good word clock and you should be fine.

Last edited by EMMST; 11th April 2016 at 06:41 PM.. Reason: Fix
Old 11th April 2016
  #5
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You may not be able to run two different audio interfaces at the same time from one computer. Typically when adding 8 channels of ADAT via an extra ADAT port on the main interface it's a stand alone converter with ADAT I/O or stand alone mic pre / converter with ADAT output. Hit your product manuals. Additionally per ADAT format when you double the sample rate, you halve the number of channels so for that one ADAT port you get 8 channels at 44.1k / 48k, or you can get 4 channels at 88.2k / 96k, under SMUX with ADAT.
Old 12th April 2016
  #6
Gear Head
 

Thanks for the replies! I don't think I will actually use Audio divers with the apollo, Just Adat I/O and UAD plugs. Im thinking during multi-channel recording Ill have to go ALL into the Motu, I don't think samples will line up perfectly either. but for 1 or 2 tracks at a time I can choose converters. Ill try clocking each to each other and see what my sync Is like. And yes, Motu 16a and Apollo duo both have 2 sets of Adat so no problem. Ok thanks guys!
Old 12th April 2016
  #7
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I use an ADAT box into my interface and the ADAT tracks are indeed slightly time delayed (about half a millisecond).
Old 13th April 2016
  #8
Tui
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I did some tests, years ago, and found that ADAT delays by a few samples. However, the results were inconsistent, i.e., the values kept changing every time I turned the units on.
Old 13th April 2016
  #9
TNM
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Quote:
Originally Posted by steven mc. View Post
Hi all! I am finalizing my upgrades and wanted your opinions or help if you have the time. I am thinking of running a Motu 16a, through thunderbolt on a powerful MacBook Pro, Logic pro X. I will have those 16 outs sent to a 2-Bus+. I have an Apollo Duo, I would like to set it up as an independent ADAT converter and use all 8 ADAT I/O channels. I'll record at 24 96.
My questions.
1. Will Going over ADAT be the same latency as recording straight into the MOTU? Recording in low latency mode, for example, guitar through the Apollo inputs and Logic plugins) What can I expect? Also, Will playback through the ADAT/APOLLO be in sync with the outs of the MOTU?
2. As far as I know, It doesn't matter which of the converters is first in the chain, I can clock both to whichever's clock sounds better. Is that right? Or is there something more I should know here?
Thank you!
i'm using focusrite 8 pre with the apollo 8 thunderbolt , running synths into all 16 inputs (apollo's 8, focusrite's 8) and honestly i haven't experienced any timing problems when i record them as 16 tracks into the DAW. I'm sure there is something as the focusrite is using different converters, but it's so microscopic its' not noticeable. The only problem might be if you are splitting drums across different interfaces, so just pick one for them and do it.
Old 13th April 2016
  #10
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Quote:
Originally Posted by steven mc. View Post
Hi all! I am finalizing my upgrades and wanted your opinions or help if you have the time. I am thinking of running a Motu 16a, through thunderbolt on a powerful MacBook Pro, Logic pro X. I will have those 16 outs sent to a 2-Bus+. I have an Apollo Duo, I would like to set it up as an independent ADAT converter and use all 8 ADAT I/O channels. I'll record at 24 96.
My questions.
1. Will Going over ADAT be the same latency as recording straight into the MOTU? Recording in low latency mode, for example, guitar through the Apollo inputs and Logic plugins) What can I expect? Also, Will playback through the ADAT/APOLLO be in sync with the outs of the MOTU?
2. As far as I know, It doesn't matter which of the converters is first in the chain, I can clock both to whichever's clock sounds better. Is that right? Or is there something more I should know here?
Thank you!
Just make sure the Apollo can run as a standalone converter, meaning its analog ins 1-8 will output to adat out 1-8. The adat out 1-8 could be outputs 9-16, the analog outs being out 1-8. I have a Twin so all this doesn't apply to baby Apollo.

KA
Old 13th April 2016
  #11
I say just try it. I work with multiple computers with their own audio interfaces, I've sent digital audio via coaxial s/pdif and 8 channel optical adat. The differences are so minute I can't hear any difference BEFORE I apply any delay compensation, or later physically realign the audio.

There's only ONE way I have even perceived anything at all, and that's by sending the SAME digital audio recording of say a track, from one PC to the other, while the 1st audio had say been EXPORT/IMPORTED and then compare the two together in the receiving PC. Only then can I hear a very slight phase difference. But then they also have to share the same left, right or center field with each other to hear this phase. Meaning that if I did this test with an exact same stereo audio file, export/import one, and digital audio SEND the other, and THEN I took each file, having just used the LEFT only part of the file, and other one using just the far right part of the file, I CANNOT hear any phase or perceive anything different...it's sounds exactly the same to me if I was just listening to the original stereo audio file.

So to sum up what I'm saying, if the parts, tracks or whatever are different (not the same audio recording) I doubt most anyone could hear the difference from any delay, especially if and when in the same computer. My example with multiple computers would probably result in a higher amount of delay, which again, is not perceivable under normal operations. I've only 'corrected' delays before, just because I knew it existed. Other times I haven't bothered.

Now jitter may be something different, and there may be some drifting in and out, especially if you compare an original file, with a digitally audio send file. But again it's minute in my experience in comparing them. I also agree with using Word Clock, as it's said to be more 'pure'. I'm about to do a total reset up, using pure Word Clock on everything...just because! Although, clocking via the same digital audio cable, along with audio hasn't been a problem for me...with stereo audio. I haven't really worked with my adat, sending all 8 channels very often, and have not done extensive testing like I did with stereo s/pdif over coaxial cable.
Old 23rd December 2017
  #12
Quote:
Originally Posted by Tui View Post
I did some tests, years ago, and found that ADAT delays by a few samples. However, the results were inconsistent, i.e., the values kept changing every time I turned the units on.
As crazy as this sounds I've been noticing the same thing for a while. Was recently just syncing up my DP88 to my Apollo 8 and the latency reported was a bit larger than usual. I started using clocking via adat since I moved my interface and WC cable wasn't long enough. Think I'll order a larger WC cable...
Old 23rd December 2017
  #13
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Quote:
Originally Posted by szyam View Post
As crazy as this sounds I've been noticing the same thing for a while. Was recently just syncing up my DP88 to my Apollo 8 and the latency reported was a bit larger than usual. I started using clocking via adat since I moved my interface and WC cable wasn't long enough. Think I'll order a larger WC cable...
I'm afraid, it doesn't have anything with that..
Reported latency of audio interface to DAW isn't affected by connection of external converters. Those numbers (input and output latency in samples) are calculated at audio interface driver for particular sample rate. Usually it reflects roundtrip latency, when using its own analog I/Os (naturally it can't know latency of some 3rd party converter).
Maybe with Apollo, this reported figure changes also according to the settings of Console, where you can have inserted plugins with own processing latency, however I can't verify that now.

External converter will be always bit off to reported latency values.. not just because ADAT I/O handling, but because there can be different converter chips with different inherent delay, than those at main interface.
If you really want to have everything aligned (usually it's not really needed, unless you want some parallel processing with analog outboard, but in such case, it's better to have also analog blend outside of box).. You need to test that and measure exact roundtrip offset (in samples - it can be either positive or negative) using some sharp spike signal passing through analog loop, which you can then compensate in DAW. By compensation I mean either manual nudge of recorded items position, some sample delay at affected tracks or using some I/O insert plugin with variable offset (that depends on used DAW and your workflow).

Anyway, that offset value should be always fixed for particular setup (working sample rate, DAW software, the same effect chain in Console, analog I/O combination). It definitely shouldn't drift. Offset can be sub-sample (that depends on converter), but still fixed.
It could theoretically drift, when sync would be dropping. But it's not really common, because you'll hear that as clicks or pops.. or it should be somewhat visually indicated via some LED or so.
It doesn't matter, whether you're using Word Clock or ADAT as clock source, a device either reliably locks or not.. and then you have also quite other troubles than latency.

Michal
Old 23rd December 2017
  #14
Quote:
Originally Posted by msmucr View Post
I'm afraid, it doesn't have anything with that..
Reported latency of audio interface to DAW isn't affected by connection of external converters. Those numbers (input and output latency in samples) are calculated at audio interface driver for particular sample rate. Usually it reflects roundtrip latency, when using its own analog I/Os (naturally it can't know latency of some 3rd party converter).
Maybe with Apollo, this reported figure changes also according to the settings of Console, where you can have inserted plugins with own processing latency, however I can't verify that now.

External converter will be always bit off to reported latency values.. not just because ADAT I/O handling, but because there can be different converter chips with different inherent delay, than those at main interface.
If you really want to have everything aligned (usually it's not really needed, unless you want some parallel processing with analog outboard, but in such case, it's better to have also analog blend outside of box).. You need to test that and measure exact roundtrip offset (in samples - it can be either positive or negative) using some sharp spike signal passing through analog loop, which you can then compensate in DAW. By compensation I mean either manual nudge of recorded items position, some sample delay at affected tracks or using some I/O insert plugin with variable offset (that depends on used DAW and your workflow).

Anyway, that offset value should be always fixed for particular setup (working sample rate, DAW software, the same effect chain in Console, analog I/O combination). It definitely shouldn't drift. Offset can be sub-sample (that depends on converter), but still fixed.
It could theoretically drift, when sync would be dropping. But it's not really common, because you'll hear that as clicks or pops.. or it should be somewhat visually indicated via some LED or so.
It doesn't matter, whether you're using Word Clock or ADAT as clock source, a device either reliably locks or not.. and then you have also quite other troubles than latency.

Michal
Ahh OK, that makes sense. I'll try tweaking some other stuff and continue clocking with ADAT. I wasn't using any plugins in console but RTL seemed a bit more than before. However, in my daw, Studio One, I do use their pipeling plugin for sending a ping and adjusting. So I guess it doesn't matter too much, I didn't notice any phasing or weirdness, but last time I had to adjust a few samples, and this last time was more like 60 samples. Something weird is happening, will just have to investigate a bit more. Cheers
Old 23rd December 2017
  #15
Tui
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Tui's Avatar
Quote:
Originally Posted by msmucr View Post
Anyway, that offset value should be always fixed for particular setup (working sample rate, DAW software, the same effect chain in Console, analog I/O combination). It definitely shouldn't drift. Offset can be sub-sample (that depends on converter), but still fixed.
It could theoretically drift, when sync would be dropping. But it's not really common, because you'll hear that as clicks or pops.. or it should be somewhat visually indicated via some LED or so.
It doesn't matter, whether you're using Word Clock or ADAT as clock source, a device either reliably locks or not.. and then you have also quite other troubles than latency.

Michal
Yes, this is all clear. However, when I performed my tests, I measured roundtrip latencies and they weren't consistent. They would unpredictably change by a few samples between reboots. I don't know what the culprit was, the external FX units (Eventide and TC), my RME interface, or the computer/Logic.
Old 23rd December 2017
  #16
Lives for gear
Here are a couple of data points: The Audient ASP800 has 39 samples A/D conversion delay in its ADAT output, and the ART TubeOpto 8 has a 32 sample delay for its A/D output. Calculate the fraction of a millisecond based on your sample rate.
Old 23rd December 2017
  #17
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Quote:
Originally Posted by Tui View Post
Yes, this is all clear. However, when I performed my tests, I measured roundtrip latencies and they weren't consistent. They would unpredictably change by a few samples between reboots. I don't know what the culprit was, the external FX units (Eventide and TC), my RME interface, or the computer/Logic.
I'd guess, there could be also some other factors involved, because that's definitely not normal situation IME (and no technical reason for drifting)... Currently I'm using two RME interfaces at different rigs and offsets for respective sample rates are fixed regardless of clocking source (AES from A/D converter, WC input or internal).
Similarly with SSL MADI card and Orion 32 and Ferrofish converters, previously also Lynx Auroras and AES cards.
However automatic ping detection of leading edge at measurement signal transient at some DAWs can be sometimes inaccurate (especially when signal is bit deteriorated by passing through some outboard effects). I usually manually measure distance in samples at DAW using some range selection tool across source and recorded track.. and test it also with straight loopback from analogue out to in before connecting other gear.
Also when signal is passing through outboard digital effects, offset could be affected by selected patch/program there (for example at our TC6K it depends on lookahead, oversampling etc. which can be used in particular engine algorithm).

Michal
Old 24th December 2017
  #18
Tui
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Quote:
Originally Posted by msmucr View Post
I'd guess, there could be also some other factors involved, because that's definitely not normal situation IME (and no technical reason for drifting)... Currently I'm using two RME interfaces at different rigs and offsets for respective sample rates are fixed regardless of clocking source (AES from A/D converter, WC input or internal).
Similarly with SSL MADI card and Orion 32 and Ferrofish converters, previously also Lynx Auroras and AES cards.
However automatic ping detection of leading edge at measurement signal transient at some DAWs can be sometimes inaccurate (especially when signal is bit deteriorated by passing through some outboard effects). I usually manually measure distance in samples at DAW using some range selection tool across source and recorded track.. and test it also with straight loopback from analogue out to in before connecting other gear.
Also when signal is passing through outboard digital effects, offset could be affected by selected patch/program there (for example at our TC6K it depends on lookahead, oversampling etc. which can be used in particular engine algorithm).

Michal
Yes, understood. I measured latency by passing a signal through the FX boxes in full bypass mode and then aligning the audio in Logic. You'd expect latency to be consistent, but it wasn't.
Old 1st January 2018
  #19
So I got my Apollo 8 Duo mkii set up with a fairly low RTL, about 3-4ms depending on how i set the buffers. When in S1, using the pipeline plugin, the ping's are pretty much lined up from the get go. I have to adjust about 1 or 2 samples. However on the DP88, I need to move up about 50 or so samples to get it lined up. Quite a bit of difference. I'm also now using WC but like mentioned above, no real gains from that. Is there anything I can do to get the ADAT interface RTL lower? Are there any units out there that have an inherently lower RTL or latency due to drivers? ATM, I'm mainly ising these to connect to external stereo hardware for master bus so not too bad, as everything is running through them. But would be nice if bother interfaces were a little closer to latency values... Cheers
Old 1st January 2018
  #20
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Quote:
Originally Posted by szyam View Post
....
Is there anything I can do to get the ADAT interface RTL lower? Are there any units out there that have an inherently lower RTL or latency due to drivers?
... But would be nice if bother interfaces were a little closer to latency values... Cheers
Alas, the ADAT latency is not affected by drivers at all. It's strictly a hardware conversion function. If you dig into the data sheet for the A/D and D/A converter chips used in the ADAT/preamp box, you'll see a 'Group Delay' figure expressed in samples as xx/Fs.

That's the number causing the concern, and it cannot be overcome or adjusted. In addition there's usually another control/support chip along the way that will add more samples of delay. As I mentioned in my previous post, the ART TubeOpto 8 documents 32-samples delay for A/D conversion and (not mentioned before) it has 22-samples delay for D/A conversion, totaling 54 samples round-trip. As you can see, that number is comparable to what you saw with the DP88 round-trip results.

A little more info: It seems that the Behringer ADA8200 uses the CS4270 AD/DA chip. Its data sheet shows that in 'single speed mode' (44.1K and 48K) it has 10/Fs Group Delay for D/A, and 12/Fs in A/D. So you already know that its round trip latency cannot be less than 22 samples.

I don't generally dig down to this level of info for electronic stuff, but it does always bug me when people only read a single figure (usually just the dynamic range number) from the gigantic multi-page data sheets generated by the dedicated engineers that design this stuff. I salute you!


HTH

Last edited by MediaGary; 1st January 2018 at 02:12 AM.. Reason: Added mini-rant.
Old 1st January 2018
  #21
Tui
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Quote:
Originally Posted by MediaGary View Post
Alas, the ADAT latency is not affected by drivers at all. It's strictly a hardware conversion function. If you dig into the data sheet for the A/D and D/A converter chips used in the ADAT/preamp box, you'll see a 'Group Delay' figure expressed in samples as xx/Fs.

That's the number causing the concern, and it cannot be overcome or adjusted. In addition there's usually another control/support chip along the way that will add more samples of delay. As I mentioned in my previous post, the ART TubeOpto 8 documents 32-samples delay for A/D conversion and (not mentioned before) it has 22-samples delay for D/A conversion, totaling 54 samples round-trip. As you can see, that number is comparable to what you saw with the DP88 round-trip results.

A little more info: It seems that the Behringer ADA8200 uses the CS4270 AD/DA chip. Its data sheet shows that in 'single speed mode' (44.1K and 48K) it has 10/Fs Group Delay for D/A, and 12/Fs in A/D. So you already know that its round trip latency cannot be less than 22 samples.
This is really interesting, thanks for sharing.

Would you say that the same figures apply to optical SPDIF and ADAT, or are there differences?
Old 1st January 2018
  #22
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Quote:
Originally Posted by Tui View Post
Would you say that the same figures apply to optical SPDIF and ADAT, or are there differences?
MediaGary was talking mainly about different inherent latency of used converter chips in some external converter versus chips at main interface itself (and reported by interface driver), this is usually main factor in observed offset.

However it can be also other way around - eg. outboard converter can have lower latency. This the case at RME AIO and more recent ADI8 external converters, where built-in AD and DA converters at AIO has higher latency than at those external converters (incl. handling of digital transfers).

Handling of ADAT or SPDIF optical transfers itself is usually very quick (couple of samples) and typically there's no practical difference between those.

Also those datasheet values can't be really used as kind of definitive reference, when you know model and vendor of used chip in the particular converter and you don't find latency values for whole box at its specs.
At modern chips there are few switchable types of reconstruction and decimation filters with different group delay and you don't know, which was used by designers. The shortest roundtrip latency isn't necessarily absolute priority there (eg. those filters has also other different parameters, which are considered there).
Also some converters has built-in asynchronous sample rate conversion, which is used to attenuate incoming jitter and optimize conversion (which works then at fixed sampling frequency regardless of input rate - say 210kHz). Such process naturally adds some latency.
So if converter vendor doesn't specify any figure, it needs to be tested for each working sampling frequency, you can't reliably guess the figure.

Michal
Old 1st January 2018
  #23
MediaGary and Michal, thanks much for the info! This was news to me. I just assumed for some reason that the adat extension would be synched up more closely. And, it's obv not bad, I still need to experiment and see if it's going to create any issues, or work out OK.

I guess this makes the case for "stackable" interfaces to me. Like TB daisy-chaining Apollo's, the new Presonus Quantum's for example. Then I'm assuming everything would be on the same playing field.

So, does this latency create any issues while tracking, or using external hardware while mixing? Or is it easily compensated for by modern DAW's. HNY all, cheers.
Old 1st January 2018
  #24
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Quote:
Originally Posted by szyam View Post
...
So, does this latency create any issues while tracking, or using external hardware while mixing? Or is it easily compensated for by modern DAW's. HNY all, cheers.
A little strategy goes a long way: If you have all your drums coming in via the ADAT portion of your interface, their timing relative to one another isn't an issue, and their timing relative to the rest of the instruments is as if the drum set is another 6 inches away from the band.

I guess you can overtly create a condition wherein you're close-miking a few things spread across the two paths of your interface (native/ADAT) and are able to induce some kind of comb filtering as a result, but that's easily avoided with choosing your inputs.

If you're using external analog inserts while mixing, then consolidating them for consistent round-trip latency is certainly beneficial, with the same easily-accomplished precautions that I mentioned for tracking.

And yes, if you have to compensate, there's a free plugin from Voxengo that allows you to delay tracks or buses by specific amounts of time or samples. It's called Voxengo Sound Delay. I'm sure there are others that do this too.

HTH
Old 18th October 2018
  #25
OK, gonna ask a question that might be a little dumb, and I think I know the answer here, but... I totally get that there is a bit of latency while tracking with ADAT. I was just doing some experiments and though it's just a ms or 2, I can see that it affects the phase relationships of various bits of audio, and I'm not too into it.

However, say I want to just use the ADAT interface (DP88 in my case) in the DAW (S1) to use HW as plugins, the DAW compensates for the latency correct? So, I'm assuming the DAW would just shift the whole production around based on issues like this? So in essence, the ADAT latency here would be noticed or a problem at all...

I'm assuming its the same for plugins too, as they inherently bring their own latencies to the party as well...

Thanks in advance!
Old 29th January 2019
  #26
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Quote:
Originally Posted by szyam View Post
OK, gonna ask a question that might be a little dumb, and I think I know the answer here, but... I totally get that there is a bit of latency while tracking with ADAT. I was just doing some experiments and though it's just a ms or 2, I can see that it affects the phase relationships of various bits of audio, and I'm not too into it.

However, say I want to just use the ADAT interface (DP88 in my case) in the DAW (S1) to use HW as plugins, the DAW compensates for the latency correct? So, I'm assuming the DAW would just shift the whole production around based on issues like this? So in essence, the ADAT latency here would be noticed or a problem at all...

I'm assuming its the same for plugins too, as they inherently bring their own latencies to the party as well...

Thanks in advance!
I’m going to restart this thread because I see a real need for it. I predict that quite a few people like myself and szyam who both have Apollo units and a Presonus DP88 connected via adat in order to add additional hardware inserts to the Apollos can benefit.

Personally I could not afford the extra thousand dollars to move up to the unit with more inns and outs so I chose to pick up a DP88 because from what I understand they sound great and are fairly inexpensive (mine was $360 to my door from a guy on eBay).

I will measure my round-trip latency in samples and post that information here sometime this week. I know that that could vary from system to system but it should get most people who have similar equipment in the ballpark at least to test and see if they are on the right track.

Thanks szyam for putting this important thread on my radar.
Old 30th January 2019
  #27
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I ran through the latency exercises and determined that for my PC the Apollo x8 does automatic compensation and there is no issue there.

Once the DP88 was connected it was phase city until I did the math once I determined that the conversion added 29 samples. So....29 divided by my sample rate of 44,100 x 1000 = 0.66

I plugged this figure into the hardware insert compensation tab in ProTools and everything worked great.

This figure will vary for every sampling rate but the math stays the same. Samples, divided by sampling rate times 1000.

Hope that helps some of you guys. This is a cheap way to add more hardware inserts for sure.
Old 30th January 2019
  #28
Quote:
Originally Posted by Secret80sMan View Post
I ran through the latency exercises and determined that for my PC the Apollo x8 does automatic compensation and there is no issue there.

Once the DP88 was connected it was phase city until I did the math once I determined that the conversion added 29 samples. So....29 divided by my sample rate of 44,100 x 1000 = 0.66

I plugged this figure into the hardware insert compensation tab in ProTools and everything worked great.

This figure will vary for every sampling rate but the math stays the same. Samples, divided by sampling rate times 1000.

Hope that helps some of you guys. This is a cheap way to add more hardware inserts for sure.
Awesome. Also, not sure if you know, but you can reduce the Apollo latency a little more by turning off the "Input Delay Compensation" and making sure the "Cue Mix" is at the lowest number, 2 i believe. Both in Console settings...
Old 1st February 2019
  #29
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Quote:
Originally Posted by szyam View Post
Awesome. Also, not sure if you know, but you can reduce the Apollo latency a little more by turning off the "Input Delay Compensation" and making sure the "Cue Mix" is at the lowest number, 2 i believe. Both in Console settings...
Thanks bud! I knew about the latency setting and have it off but not about cue mix so will check on that.

Am kinda surprised so few discussions have centered on adding hardware inserts via adat. It’s a good cheap way to do it if you can’t afford the BIG unit of something from the get go.
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