Old 03-19-2019, 08:19 AM   #1
MixR
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Default Pro Tools to REAPER: Using Reainsert

Question:

Hello all,
Why is there are massive amount of PDC added as soon as I add Reainsert to a channel?

My block size is 128, SR 44.1kHz, PDC is 4864 samples (38x block size).
When I switch to 1024 block size, PDC is 7168 samples (7x block size).

Obviously these inserts will be compensated with ADC. But using them will yank the system delay up massively. The reported latency inside the plug-in is 330 samples at 128 (2122 at 1024) which is expected (2x block size + RME hardware + Core Audio Safety Buffer, digital I/O so no converter delay).

So where and how does the extra PDC come into play?


Context:

I am still struggling to make my transition from Pro Tools HD to REAPER, something that is going on in the background whilst I am continuing to work in PT on a day to day basis.

My setup involves a lot of outboard hardware and I am trying to get an idea how to successfully port that over to the REAPER world. Currently I am using around 60 hardware inserts with outboard gear in Pro Tools plus another 48 using REAPER as a realtime FX rig. The plan is to switch it so that REAPER becomes the main DAW and Pro Tools the FX rig.

In my brief trial with 16 inserts Reainsert seems to work as expected. CPU usage is about 4% at block size 128 so if that scales correctly I can expect around 25% CPU usage for 100 I/O. I would be very grateful to know figures from other users to help gauge system performance for when I switch to my native REAPER rig more permanently.
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Old 03-20-2019, 07:29 AM   #2
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You have "Anticipative FX multiprocessing" on from preferences. That causes the extra DPC latency in this instance since it tries to "render ahead" the reainsert plugin by how many milliseconds you have on the setting (200 by default).

You can disable/enable anticipative FX multiprocessing on track by track basis by right clicking the track and selecting "Track performance options/allow anticipative FX"

Remember, the reported interface latency isn't always correct (depends on your drivers) so the setting on Reainsert "Automatic interface latency adjustment" doesn't always correspond to the real loopback latency you have.

Only way to measure the real delay is to make loop back testing and measure it.
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Old 03-20-2019, 08:44 AM   #3
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Thank you, Icchan - that has fixed it!

The reported interface latency seems in the ball park but with the help of the Ping function to adjust for extra latency I can get the signal to null when polarity is inverted.

The next test will be to see how well it works with an analogue insert.

One small problem will be the fact that my converters have a Roundtrip Delay of 69.9 samples which is unfortunate in that it is not a number down to the nearest sample (the smallest increment Reainsert understands).
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Old 03-20-2019, 12:18 PM   #4
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Quote:
Originally Posted by MixR View Post
Thank you, Icchan - that has fixed it!

The reported interface latency seems in the ball park but with the help of the Ping function to adjust for extra latency I can get the signal to null when polarity is inverted.

The next test will be to see how well it works with an analogue insert.

One small problem will be the fact that my converters have a Roundtrip Delay of 69.9 samples which is unfortunate in that it is not a number down to the nearest sample (the smallest increment Reainsert understands).
I have a hard time to believe that it can be franctions of samples. Samples are the atoms of digital audio really...
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Old 03-20-2019, 02:29 PM   #5
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Quote:
Originally Posted by mlprod View Post
I have a hard time to believe that it can be franctions of samples. Samples are the atoms of digital audio really...
Well... it's a bit annoying and also inconvenient - but it's still a fact:

A/D: 41.7 samples
D/A: 28.2 samples

And then there is this:
https://en.wikipedia.org/wiki/Subatomic_particle

;-)
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Old 03-20-2019, 02:36 PM   #6
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Quote:
Originally Posted by MixR View Post
Well... it's a bit annoying and also inconvenient - but it's still a fact:

A/D: 41.7 samples
D/A: 28.2 samples

And then there is this:
https://en.wikipedia.org/wiki/Subatomic_particle

;-)
Lets say samples are quarks then=)
Are those figures from the AD/DA specs?
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Old 03-21-2019, 04:51 AM   #7
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Originally Posted by mlprod View Post
Lets say samples are quarks then=)
Are those figures from the AD/DA specs?
Yes, they are!

It would be great if Reainsert could use fractions of samples (or milliseconds) as units - st the moment everything is rounded to the nearest sample.
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Old 03-21-2019, 06:41 AM   #8
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Originally Posted by MixR View Post
Yes, they are!

It would be great if Reainsert could use fractions of samples (or milliseconds) as units - st the moment everything is rounded to the nearest sample.
I'm electronics designer by trade... I would like to know what AD converter we're talking about exactly? I mean I'd liek to read that datasheet as well.

I do know that thee are several reasons that AD converters can have sampling times that are fractions of the samples real time length because of oversampling and different converter topologies, but I'd like to know anyway

And frankly, those sub sample conversion times are only relevant in very specific circumstances. Most of the time Nyquist sampling theorem holds and it doesn't really matter where in the signal you sample, as long as you sample often enough and your sampling clock is steady (minimal amount of jitter).

So in my opinion you can just round that delay to the nearest whole sample and forget about the sub sample timing issue. Yes, your waveform would be corrected in DAW just a hair (0.1 samples) early of what it really is, but unless you're trying to sync up multiple devices with different converters and different converter latencies it's not a real issue as long as you don't try to put exactly the same signal through that different converter at the same time and then try to do summing or other stuff to it with another copy of the signal etc.... They wouldn't necessarily null out and you would get distortion from that timing issue.

But as long as you don't split them up like that (let's say left stereo channel goes to the converter 1 and right goes to converter 2 with different conversion time) there's no issue
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Old 03-21-2019, 07:33 AM   #9
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Quote:
Originally Posted by Icchan View Post
I'm electronics designer by trade... I would like to know what AD converter we're talking about exactly? I mean I'd liek to read that datasheet as well.

I do know that thee are several reasons that AD converters can have sampling times that are fractions of the samples real time length because of oversampling and different converter topologies, but I'd like to know anyway

And frankly, those sub sample conversion times are only relevant in very specific circumstances. Most of the time Nyquist sampling theorem holds and it doesn't really matter where in the signal you sample, as long as you sample often enough and your sampling clock is steady (minimal amount of jitter).

So in my opinion you can just round that delay to the nearest whole sample and forget about the sub sample timing issue. Yes, your waveform would be corrected in DAW just a hair (0.1 samples) early of what it really is, but unless you're trying to sync up multiple devices with different converters and different converter latencies it's not a real issue as long as you don't try to put exactly the same signal through that different converter at the same time and then try to do summing or other stuff to it with another copy of the signal etc.... They wouldn't necessarily null out and you would get distortion from that timing issue.

But as long as you don't split them up like that (let's say left stereo channel goes to the converter 1 and right goes to converter 2 with different conversion time) there's no issue
Euphonix AM713 and MA713. Installation manual attached. As I have stated in my original post I am currently using around 60 analogue I/O for hardware inserts. A fair few of those are used for parallel processing. In those instances even the sub-sample delay can be audible.

However, it is not a show stopper but with many channels of I/O in use the tighter your delay compensation works the better for your overall sonics.

Using Precision Time Align from Eventide helps getting the parallel-processed return tracks aligned even better. If Reainsert's Additional Delay Compensation setting would support milliseconds I could type in i.e. 1.5850ms into the Box instead - problem solved!
Attached Files
File Type: zip INSTALLATION GUIDE AM713 MA703 DM714 MD704.pdf.zip (878.6 KB, 111 views)
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