PC amp sims that sounds as good as axe fx and kemper?

[pipelineaudio]

file:///D:/downloads/AES_Latency.pdf

[zeekat]

Originally Posted by pipelineaudio

https://community.bose.com/ziiiw76728/attachments/ziiiw76728/Hearphones/594/1/AES_Latency.pdf

Fix'd the link. Interesting - quick glance shows there's sometimes such thing as latency being too short . Lots of data to understand and digest.

"Consider also a study by Chris Chafe and Michael
Gurevich on Network Time Delay and Ensemble
Accuracy [5]. The study suggests that the sweet spot
for musicians to play with each other in tempo is
11.5ms. This, of course, corresponds to a distance
between the musicians in free air. Statistically, a delay
of less than 11.5ms caused 74% of musicians to speed
up, and a delay of more than 11.5ms caused 85% of the
musicians to slow down. Additionally, they determined
that the longest propagation delay in free air for a
typical ensemble is around 20ms. "

I wonder if it can be applied to playing along to your recorded stuff, or only to ensembles adjusting the timing to each other.

[Stews]

If there was too low a latency for guitar then wouldn't it be impossible to record acoustic guitar?

It might be confusing things reading about orchestra players hearing other instruments - latency is about hearing your own instrument.

When it comes to the 12ms latency, I doubt it would have much effect on my timing, it's more about feel. When you're playing with a band, players aren't within 12ms of being metronomic to the grid are they?

[Goldreap]

Originally Posted by Stews

So would the 64 samples be in addition to the 64 samples set in the ASIO control panel?
Would that mean the latency would be doubled?

Yes, my tests show that.

Originally Posted by brainwreck

I think it works such that, for example, the 7/64 you see in the fx window means that the focused plugin has 7 samples latency, where the entire effects chain has 64 samples latency (of the 64 samples asio buffer), not an additional latency added to the asio buffer....

Yes the 64 is the latency of the entire chain, but I believe that it is added to the ASIO buffer

Originally Posted by brainwreck

.....And last that I remember seeing, if you go over by even 1 sample of the asio buffer, an additional buffer of the same size is added to the asio buffer. So say that you have a bunch of plugins that total up to 65 samples latency and your buffer size is 64 samples, an additional 64 samples buffer is added to the asio buffer for a total of 128 samples latency.....

This is correct, but...

Originally Posted by brainwreck

......In other words, your buffer size is doubled when exceeding the base buffer size..........

The added buffer size (equal to the ASIO buffer size) happens anyway.. you don't have to exceed the ASIO buffer size,

Originally Posted by Stews

I had a hopeful idea that might be it.
At first I was surprised to see the amp sim plugin added latency but if what you say is correct, it seems they'd be perfectly fine having a small number of latency as it wouldn't have any effect unless the person's buffer was set ridiculously low........

Not the case I believe. I tested it like this:
Set my ASIO buffer to 2048 (obvious latency).
Loaded Thermionic Serpent amp sim which has 32 samples latency.
I could hear that the added latency was a lot more than 32.
But to be sure, I kept loading more instances to the chain....with 8 instances the chain buffer shows 16384 (easy to hear that). Each instance is adding 2048 not 32.

But anyway, with normal ASIO say 128, the extra 128 added with Thermionic doesn't bother me .

BTW, I'm no expert, so happy to be corrected on any of the above.

[Stews]

How did you manage to test it?

Any time I tried a loopback test, reaper seems to always compensate for the extra latency and it's only ever the same as the asio buffer

[drumphil]

Glenn with the Spectre Sound Studios youtube channel has a bunch of reviews and comparisons of amp sims and modeling amps and cabinet sims and such.

[brainwreck]

Goldreap, you are right on plugin latency being additional to the asio buffer size. I wasn't in front of my recording pc when I said the above. Taking a look today, I see that when adding multiple instances of a plugin that adds any latency, a whole buffer size is added for each new instance. So if a plugin has 7 samples latency and the buffer size is 64, an additional latency of 64 samples is added per instance of the plugin.

Stews, you can right-click a track's recording button to toggle plugin delay compensation (pdc). So then, setup a dry track (no plugins) with monitoring on. Setup another track with the latent plugin, with monitoring on, and check that pdc is on. Arm both, record something percussive, and check the gap between peaks. You should notice that adding an additional instance of the plugin will cause an additional 64 samples latency (or whatever the buffer size is).

Apologies for the confusion.

[brainwreck]

Originally Posted by drumphil

Glenn with the Spectre Sound Studios youtube channel has a bunch of reviews and comparisons of amp sims and modeling amps and cabinet sims and such.

He has a good channel and info for metalheads, not so much for anyone else.

[karbomusic]

Originally Posted by brainwreck

He has a good channel and info for metalheads, not so much for anyone else.

I agree.

[karbomusic]

Since we are on the subject, you might check this out where 4ms minimum can be obviously heard and takes all the PDC variables out of the equation (I know the tests above are about how it feels to play with it but this is what it sounds like when it's off by 4ms). I'd bet I could take that latency down another ms or two and it still be noticeable. My point has always been, latency in this regard is in "addition" to "all" other real-world expected latency, not in lieu of.

FYI that Alanfoz's correction is correct so consider 4ms instead of my initial oversight of 2ms. To me, the latency sounds like a band that isn't tight.

https://forum.cockos.com/showthread....hlight=latency

[Stews]

Originally Posted by brainwreck

Goldreap, you are right on plugin latency being additional to the asio buffer size. I wasn't in front of my recording pc when I said the above. Taking a look today, I see that when adding multiple instances of a plugin that adds any latency, a whole buffer size is added for each new instance. So if a plugin has 7 samples latency and the buffer size is 64, an additional latency of 64 samples is added per instance of the plugin.

Stews, you can right-click a track's recording button to toggle plugin delay compensation (pdc). So then, setup a dry track (no plugins) with monitoring on. Setup another track with the latent plugin, with monitoring on, and check that pdc is on. Arm both, record something percussive, and check the gap between peaks. You should notice that adding an additional instance of the plugin will cause an additional 64 samples latency (or whatever the buffer size is).

Apologies for the confusion.

I was trying that but the option didn't seem to do anything. If that definitely works for you I'll try again tonight or tomorrow, whenever I get time.

Does it mean that the latency I was hearing each time was actually double?

I'm surprised the amp sim plugins have latency if it doubles the overall latency. There are complex plugins with zero latency, I'd imagine that would be a priority for real time.

[brainwreck]

Originally Posted by Stews

I'm surprised the amp sim plugins have latency if it doubles the overall latency. There are complex plugins with zero latency, I'd imagine that would be a priority for real time.

I remember doing some latency / pdc testing some years back involving recording the input of a track and the output of a track to see how everythng lines up. What I found is that it's all kind of messy. I would think that preserving performance timing (and not influencing it) should be a top priority in digital recording, probably more important than converter accuracy, etc. But it doesn't seem to be so. Latency is an issue. Latency compensation is an issue. Probably the best that we can do is to treat the computer like a tape machine, monitoring our playing outside of the digital chain, assuming that we have consistant interface latency compensation so that everything always lines up exactly as we heard it while recording.

[Goldreap]

Originally Posted by Stews

How did you manage to test it?

Any time I tried a loopback test, reaper seems to always compensate for the extra latency and it's only ever the same as the asio buffer

It wasn't a stringent test, just basically hearing that at ASIO 2048, a plugin with latency 32 adds another 2048.
But Brainwreck's suggestion (post #287) demonstrates it.

But don't be confused about the 'Preserve PDC delayed monitoring in recorded items' which I believe Brainwreck is talking about. This setting won't change what you hear when monitoring...it just moves the recorded material later in the timeline to compensate for players who when hearing latency actually play early to get what they're hearing back in time...that's another story, but for our purposes toggling this setting does reveal (after recording) the plugin induced delay we hear when monitoring.

Another way to test this is to set ASIO to 2048 then simultaneously record the outputs of two tracks, both being fed by your one guitar, one track with Reaverb set to Max FFT 32, the other with Reaverb set to Max FFT 1024. When armed for recording you'll hear the same latency on both tracks and after recording you'll see that both tracks line up exactly....REAPER sees both the 32 and 1024 as 2048.

[SmajjL]

What a guitar dude would put after a tb-303 is what I would like to know.

[brainwreck]

Stews, I think it works such that with latency compensation on and pdc on, what you were hearing while you were recording is what you should hear during playback. In other words, audio interface roundtrip latency + additional buffer introduced by latent plugin. So say that you have 264 samples roundtrip latency when using no plugins, introducing a plugin with a latency of 7 samples will be adding 64 samples additional latency to that: 264 + 64 = 328.

Interestingly, on my current interface I get something a little higher than that.

When all is taken into consideration (interface, daw, plugins), is there really such a thing as sample accurate recording and playback?

[karbomusic]

Originally Posted by brainwreck

When all is taken into consideration (interface, daw, plugins), is there really such a thing as sample accurate recording and playback?

It the resulting recording, I think there is. I'm not seeing where you adjusted the record/playback offset - meaning, there is always 'some' unexplained latency somewhere IME and that is the setting to account for it - secondly as mentioned before, what the player uses as their timing queue becomes very important right about now because it will undo record latency compensation (or not) depending on which queue they use.

[brainwreck]

Originally Posted by karbomusic

It the resulting recording, I think there is. I'm not seeing where you adjusted the record/playback offset

What I did was insert a click source on a track and record the output back to a track (so that I get a sine to look at, rather than a perfect square). Then I sent that out and back in to check for alignment. Alignment was within a sample, without any additional latency compensation other than the reported latency from the interface = 264 samples. I then added amplitube to that track with pdc turned on and with the amp and cab bypassed (so as to get a clean signal back). I sent that out and back in, resulting in 17 samples higher than 264 + 64.

[brainwreck]

Repeating the above test lined up, but this time I used a tone generated by js tone generator as the source. The offset that I saw before may be to do with my interface.

[Reason]

Originally Posted by SmajjL

What a guitar dude would put after a tb-303 is what I would like to know.

I've read (unsubstantiated) that Josh Wink used a ProCo RAT on Higher State, and I've also heard that Hardfloor use the same.

I've tried a couple RAT sims on Venom VB-303 and preferred the Distorque 104+.

Of course, with sims you can put it through a Marshall stack.

[SmajjL]

3o0o3hh yeah! thank you! and Distorque emulates MXR Distortion (also) and is a VST?, thank you twice!! glad I asked

Also, for a reference of what I am looking for, this is close, but not the best example link though, but here you go, at 5:00.
https://www.youtube.com/watch?v=s2aWug-unKs


Wait, free? how about I drop what I was supposed to do and try this right now.

& edit (because tiny off topic borrowing/pushing luck)

https://www.youtube.com/watch?v=1IdlDMIgCoE

[politicalEconomist]

Just to add to the original topic of this thread. I just ran across this free Amp Sim and was very impressed.

This is now my "go to" recommendation for free/inexpensive Amp Sim plugs.

https://tonelib.net/gfx-overview/