Before you buy new SB hardware please consider if you want to support a company with such low environmental ethics that lets you throw away technically working hardware because they are too lazy to program new drivers while there are companies like RME…
he uses Apple computers, Apple are the kings of throwaway with their constant decision to use different connectors to the whole world so they can sell you expensive chargers/cables. The EU tried to standardise chargers to stop this years ago and all the main manufactures come on board except Apple.
My point? if you use Apple computers you’re NOT ecologically sound, which is fine but renders your argument pointless in this context.
I love RME gear too by the way i still have a 9652hdsp. All my steinberg hardware still works on windows 10 even my Houston controller. I chose to upgrade because of other decisions not because they stopped working. UR824,MR816, Midex 8, CC121, Houston controller,All still working under windows 10 x64.
that’s only half the story of course :
strictly speaking the CC121 isn’t fully supported on cubase 10.5 - apparently that’s coming. It’s supposed to be a fully supported product of course
midex 8 didn’t work with 64 bit OS for a long long long time - until SB relented and released beta drivers…too late for me as I’d long since disposed of mine.
MR816 isn’t supported under the current MAC OS
well I’ve not had any issues with CC121 using it everyday.
Midex 8, not sure how long but not long enough for me to get rid and I’ve had it now since new.
as to your last point it’s one of the reasons I use windows. Apple tried to kill off firewire like they did with PCI , they’ve done a U turn with PCI though recently.
Anyway this is getting off topic and not helping the OP.
I was just trying to help him with with my answers to his questions and my experience of 30 years using steinberg hardware and software.
So, sorry TNM for drifting OT.
wow, we are going to go to taking jabs at me for using Apple?
I tried to have a hackintosh built, the supposed best builder in Melbourne didn’t even answer my email. I can’t do it myself anymore as it’s too much for my lower back to be hunched over a computer case… I have sever back problems. And OSX is my OS of choice for good reason. I have bootcamp installed, and Win 10 runs fine, but it’s high DPI support for the 5K imac screen is horrendous… even the windows support guy at GS agrees with this and explained why it’s so bad and so hard for everything to support it. Even apollo console and pro tools are not high dpi on windows.
Environmentally friendly? Are you joking? I had my previous macbook for 5 years, and this one I will be trading in for the 16" replacement, through their buyback program. They melt them and reuse the metal, they told me. Um… what’s the issue?
my previous imac was used till it was dead. Windows users are upgrading parts all the time… so… what happens to those parts? resale? what’s the difference? I can resell a computer. People that know what they are doing still strip apples for parts. Apple are using USB c for charging, for years now, so again, what are you on about? They are using standard, current USB protocol for charging… Hardly ANY windows laptop does that except for a few.
As far as everything else, 2.2ms output at 32 buffer is very good for USB… But you didn’t say the sample rate.
Everything else remains unanswered.
I doubt it can be as good as HDX, with HDX I can choose 100 different effects, and the cards are so powerful as far as “audio card dsp” is concerned, they can each run hundreds of plugins. I just checked the manual and I can only use 16 sweet morph strips with the AXR4, which means I can’t even put an EQ and comp on every channel… that’s telling me it’s incredibly underpowered.
With one card in HDX i can put a heat tape, EQ and comp on every channel of 64 input channels, and a couple different verbs for monitoring.
This was not the point though… HDX destroys everything as far as integrated monitoring. It has full latency compensation for inputs so they are in time with DAW playback tracks, no matter what effects you use. It can handle 192 inputs with three cards, which is way more than the 84 of three AXR4. (and that’s only at 48K). The cards are each around the price of an AXR4T, however no analog i/o of course, so my idea is to get something like an ORion HD3 with 64 ins that has a HDX connector, and then still have the second HDX card free for when I want to add another 64 ins. When the time comes, just get another Orion, where street price is not far off the AXR4T. So yeah, overall, HDX is more expensive, but 3 AXR4T here are 10 grand at best in AUD, plus 6 banks of adat 8 A/D i’d need to max them out…
But if the DSP is this weak and I have no way of using a guitar amp, it’s 100% pointless to pursue, hence my questions. Maybe i understood the manual and the instances wrong, I don’t know. But I do know I am correct re guitar amps which is why I am asking if i can use a native guitar amp whilst still using the DSP reverb.
Don’t worry about latency, I know what I and my singers feel comfortable with… I just need to know the figures… Basically, it has to be under 3ms total roundtrip through the DSP effects… that’s the absolute max… and still I can’t find this info anywhere. HDX is 1.7ms at 44K, 0.7 at 96K. It is around 2.5ms total RTL with eq, comp, tape inserts, and a nice reverb send bus. At 44K. that’s LOW. UA Apollo Console (and the coming luna) is much higher than that when you pile on the good effects.
TNM I wasn’t having a jab at you at all so I apologise if you took it that way, I was just saying to the other poster that the ecological argument was a little spurious at best in this case.
Anyway. When I said like HDX I meant as regard latency, NOT processing power so again, I’m sorry if I gave that impression.
I don’t need to have EQ, comp, guitar sims on all monitor channels so it’s not a scenario I’ve come across. I record musicians , do film scores, mix and produce fairly traditional music so the Steinberg hardware has suited me for my needs in regards to stability/sound and low latency monitoring.
I cannot give you the figures your asking for sorry.
I was only trying to help.
p.s. the hackintosh thing isn’t that hard .I’ve just finished the machine in my signature and it’s running a Vanilla Mojave very nicely along side windows 10. I’m using a Dell 30 inch Cinema display so I don’t have the hires issues I know exist, it’s one of the reasons I have a dual boot machine in case I go 4/5k in future and find that this situation doesn’t get worked out. Cubase and most other things I’ve found seem to be a lot snappier under windows 10 though so that’s still my chosen platform.
p.p.s here’s what I was referring to with Apple:
Ok I am sorry if I sounded angry… I can’t build my own desktop anymore though that’s out of the question… You have no way of understanding how serious my back issues are, but put it this way, I have to have a special chair with features just to be able to sit in the studio room and compose.
I absolutely do not need a guitar input on every channel LOL… I would need just one. But my point is that AXR4 removed the dsp guitar amps from the previous generation of steinberg dsp interfaces. In fact you are in a unique position to help me, as you could let me know whether a native guitar amp, just one, could be used in the monitor path of the AXR4 whilst also using DSP effects. This is what I meant by “hybrid monitoring”. Logic can do it with Apogee, and Avid can do it with HDX if you put the native aax plugin before the DSP one. I am hoping steinberg has a similar system in place for the AXR4, since it is missing modulation plugins and guitar amp plugins completely from the dsp lineup.
Look, I don’t really need to have an eq and comp per channel, not always, often I can make do with the outboard effects going into the device… but…the fact that it can’t really surprised me, unless I understood the manual wrong.
The way I understand it is 1 global reverb and 16 sweet morph strips, over the 28 inputs (12 analog built in, and 16 adat analog in). Even less if the better dedicated vintage EQ and comp plugins are used… Am I incorrect?
no worries, the internet is a tricky place for communication sometimes…well…most times
I don’t think that it will work hybrid due to the fact the AXR installs extensions into cubase so when you set up an input from the AXR on your channel it adds a ‘hardware AXR’ menu . This menu is where you send to you reverb, add inserts. The only inserts you can add are the DSP plugins.
You can add regular inserts on the track and with 32 sample latency you’d have no problem latency wise but you’d have to turn off direct monitoring I think to be able to monitor through the software which possibly could mean loosing the zero latency thing.
I’ll give it a go tomorrow when I’m in the studio.
Ok… thanks a lot.
I wonder then, maybe not combining dsp and native, but if I could have one input only native and the others on DSP. I doubt it but who knows. In any case, that’s a deal breaker unless steinberg expands their line of effects. If i have to monitor all 80 inputs at 32 buffer using native CPU just for one or two tracks I need native effects on, it’s not even worth considering. I don’t have a strong enough cpu for that… I designed my system around dsp monitoring and external instruments which is why i only got the 8 core imac pro. It will have to be apollo for now and aax dsp end of year then with the new thunderbolt daisy chainable ones avid are releasing, pre packaged ready to go. Just got to save a bit. Well, a lot hehe.
Even if I had a 16 core new mac pro I would not want to be monitoring 80 inputs at 32 buffer… The whole point of DSP is 100% reliable performance with zero cpu load… but one or two tracks native? Yeah, that’s fine.
can OSX do ASIO direct monitoring yet? , that was the main reason I went with windows back in the early days. With direct monitoring you could ‘direct monitor’ all you hardware and it would be zero latency and then use the limited DSP on the tracks you needed it as direct monitoring is independent of buffering and separate from the DSP interface monitoring. The interface DSP monitoring system was mainly brought into play for Mac users who had no way at the time to not monitor through the software maybe that’s changed in recent years. If you’ve got a bootcamp partition you could try it and see how it works on windows. Then again if the GUI is a PIA that’s not going to float your boat. Just going through options here.
HDX is the best scenario if you need large monitor mixes with FX if you don’t have a desk and hardware. The steinberg units are fantastic for smaller , simpler things and have suited me well over the years, I’ve also got a lot of hardware so if I need comp/eq on the way in I use hardware. I’ve also got a studio full of boutique tube amps and a Kemper so that covers everything from that point of view too.
. I looked at the Apollo and the RME thunderbolt as they’re a similar price but the total integration of the AXR4 plus the sound quality won me over. Having used UR and MR units for years now I can’t go back to using a separate audio interface monitor mixer
Good luck with your system, at the end of the day we’re spoilt now. I l happened to hear an album I did 20 years ago the other day , which still sounds great and pointed out to someone that the computer used to record it was less powerful than my current phone!!
OSX doesn’t have asio at all… but you have confused me now… I thought asio direct monitoring was for a simple a/d to d/a passthrough to eliminate latency, but that you also couldn’t use effects. I have never heard of people using effects with asio DM before your post today.
Yes that’s true, you can still use the on board DSP FX though of course, must not have explained myself well
Hi Norbury, so you are saying that companies added their own standalone console app mixers for mac os, which doesn’t have asio?
Well fair enough. Mac OS uses core audio… and it makes you wonder, why doesn’t UA offer a DM asio mode on windows with their DSP effects? A bit of a missed opportunity if you ask me.
I hate to play devil’s advocate, but core audio is one reason I choose mac over windows… It absolutely crushes asio… Obviously there’s a way to do it as Apogee and logic have dsp integration with core audio… regardless though, if we take asio DM out of the equation, yes, that’s a good feature… however with core audio, I can have multiple apps open using the very same outputs, at different buffer sizes, and even sample rates!
Right now I have PT open at 96K, 128 buffer and seamlessly switch to logic at 44K, 32 buffer just using the inbuilt audio on the macbook. That’s core audio. Secondly, you can make aggregate devices, which is basically combining multiple devices to be seen as one. So… no comparison for me…
Steinberg are the only developer who won’t do native core audio and still have an asio wrapper for Cubendo…
It’s really made me think… Cubase 10.5’s performance has finally outpaced Logic in around 80% of real world use cases… imagine if Steinberg used native core audio instead of having a wrapping layer to asio… But I suppose maybe that’s too difficult to keep windows and OSX versions parallel.
Unless I am mistaken, I think last time I checked wavelab that used the asio wrapper as well, but I have no idea about V10.
Anyway, for arming tracks and monitoring them at 32 buffer with effects, Cubase 10.5 is now the best performing DAW on OSX, with Logic 2nd, PT/Reaper third. It’s come a long way.
But at 128 buffer, PT is king still. So this is the way I see it… I use DSP for monitoring and 128 to play VI’s, best of all worlds kind of thing. Even if I switched to Nuendo full time, I’d still want the same sort of setup… keep the main buffer at 128 to get an overall balance of good performance and still low enough latency to play VI’s. This is why I asked about the output latency of the AXR (the output is the one that determines realtime VI playing latency with a midi keyboard).
If it’s no more than 4MS for the thunderbolt one at 128 buffer, that will be fine. Anything up to 5MS really, max, is OK for VI’s. Maybe not for drums though but you get used to it.
External input monitoring, well, that’s another story, that’s for sure! 128 way too high for 99% of vocalists, and the external synths I use already HAVE d/a latency of their own as well as DSP latency… so I like keeping it as low as possible. Believe it or not, the apollo’s DSP mixer is also way too high when you use the MK2 UA effects, which they don’t mention anywhere pre buying. You have to use the MK1 effects which have been outdone by native stuff these days.
Luna, the revelation UA keep sprouting about, will be subject to the same input latency as all apollos, and VI’s are at a fixed buffer of 128 that can never be changed. I know that’s way OT but i thought someone reading might find it interesting, and not get bedazzled by UA’s marketing spin like I did.
This is also why I am trying to find out everything FIRST this time instead of buying something without all the facts. If I go out and buy an AXR4T right nw and it can’t do what I need, then it’s entirely my fault… I have to be patient and get the info I need.
Ed from Steinberg support (really nice fellow) asked me for my questions via PM but he hasn’t replied yet. I keep hoping he will get back to me with the answers!
core audio is actually less efficient needing 3 kernel calls to ASIO’s 2, so ASIO technically will always outshine Core audio in performance. I agree ASIO doesn’t do aggregate devices but in 35 years I’ve never needed to use that feature so I’ll take better performance Windows ASIO devices can switch between apps no problem if you have multi client drivers, all my steinberg units have had this so again it’s not a limitation I’ve come across. ASIO4 all will give low latency drivers for on board sound chips on a laptop for example if you want to use the on board sound. I’ve done gigs using this playing a VSTI rack on my HP elitebook and on board sound from the output jack
Latency and monitoring is indeed a thorny old problem and…sometimes the numbers don’t match the experience either.
Cubase is one DAW that has a single button ‘constrain plugin delay compensation’ which is very useful not sure how other DAW’s get round this.
Even if you have 32 buffer setting, if you’re monitoring through software and have say a look ahead limiter or your master bus…that will give you huge latency with monitoring. Hit the constrain button and bang you’ve bypassed any latent plugins, this is another thing people forget when working
anyway, hope you get your answers and end up with a system you’re happy with.
May I ask why Nuendo and not cubase? I used to have both up to V6 then decided Nuendo became too post orientated so just carried on with Cubase. I’ve had no problem doing my movie scores in Cubase.
cause i want to get into some video and post work.
I don’t mean to be rude, but i am not a noob re latency… I have been campaigning for other DAWs to offer a version of CDC from Cubase for, well, a decade, to no avail… I know exactly how it works.
I am so sensitive to latency, that I discovered all the problems with UA apollos, and uncovered all their lies (that they still sell their interfaces on today), entirely by ear… and everything I questioned their support about is 100% confirmed with no workaround other than using the older 0 additional latency plugins (the ones from 2005) to monitor with. I have no problem calling them liars, as right now they have an ad for their fender plugin saying you can track through it with NO latency. Even after everything I have discussed with them they still continue the BS…They use the word “NO”. Which to me means 0. The fender plugin has 131 samples latency on it’s own (from memory, or 121, something like that), on top of the base 2.3 RTL at 44K of apollo. So that’s almost 5MS RTL, without ANY compressors, EQ’s or possible desired tape plugins. Native is much faster.
This is why I am trying to find out if AXR is actually like 2ms through the effects or much higher like apollo is.
Logic goes one better btw re low latency recording - they call it low latency mode and you click a button, same thing as Cubase, but you set the threshold. For example I can allow all plugins with latency, but below 1MS “total” chain latency, to function. This means I can still monitor with softube plugins which have 4 samples latency, which Cubase disables with CDC. I only use a zero latency safety limiter on the master bus (blue cat protector 2) and before that, the glue in 0 latency mode to mix into (all oversampling disabled so it’s true zero latency).
If I was using Logic still though, or even Cubase, my options would be broader as I could keep latent plugins on the master and use their low latency modes at record time… I do miss this a LOT in pro tools. PT has a low latency mode but it disables ALL plugins… not the same thing.
I have never, ever seen a multi client asio driver use the same outputs at different buffer and sample rates… I have seen them use different outputs on the same interface… so… who knows… regardless, I use aggregate so that’s moot.
Well, how am I supposed to get the answers if no one with an interface is willing to help? A simple loopback test using the FREE Ceentrance utility for example, without cubase loaded, just the AXR through it’s own DSP mixer app going through the vintage EQ and comp?
Or what about someone trying just one track with native plugins instead of DSP?
And if that can’t be done, can it be switched seamlessly to native mode, just so someone may use a native guitar amp, then once the guitar part recorded, switched back to DSP for all the other inputs?
No point wishing answers come my way if no one is willing to spend a few minutes trying things out. I guess it’s a NO from me for now, as even steinberg is not answering me… so… i’ll be on my merry way and maybe even buy one HDX card to get started…
Ultimately I have to be honest… I doubt I could live without the way PT does clip gain anyway… so maybe I am deluding myself. I do love Cubendo’s midi, chord track, drum editor and midi inserts… it’s all midi stuff I miss when in PT… but the way it handles audio is great…
The other thing was the matter of cost… PT ultimate with all the post features is very expensive to maintain… like 400 USD a YEAR… and around 2 grand to buy… vs 400 for Cubase crossgrade to nuendo… so there’s that. Currently I am on PT Vanilla… the upgrade price is like 1800 USD. LOL. And the DSP cards are about 2.5K each real world pricing. Then there’s I/O!
But if Steinberg themselves aren’t willing to give me the answers to what I need, I am at a loss as to what more I can do from my end… maybe they just don’t know, even about their own interface!?!
I didn’t mean core audio was more efficient than asio head to head… Steinberg are using a wrapper… How can THAT be more efficient in this case? That was my point…They are converting core audio TO asio… isn’t that adding extra layers? I know asio performs great on windows… and I myself tell everyone about asio 4 all which I use on my bootcamp partitions
I didn’t get to the studio today and I’ve been installing a new machine over the last week so not had time for anything.
I’ll do the test tomorrow for you if it’s quick and easy
do I need to physically connect outputs to inputs?
oh… thanks so much!
Yeah, just connect output 1 to input 1… that’s all you need to do… Put effect on input 1, run the utility… bob’s your uncle! That will answer one of the major questions for me.
I would then ask you, how easy is it to disable DSP mode to temporarily have the device in native mode, to as I said, use a native monitoring plugin I might need, then once that track is recorded to audio, flip it back to all inputs being through the DSP mixer?
If there’s no hybrid monitoring, this could be a workaround.
TNM, I can’t find that RTL program anywhere. The web site doesn’t have it on there anymore and I can’t find any external downloads or any other tools to do it after an hour of searching.
Spoke to Vin @daw bench and he recommended a utility.
The RTL of AD/DA going from an output to an input (loopback) @32 samples is 4.6ms. adding the DSP FX makes NO DIFFERENCE to this delay at all.
If you look at the screen grab you can see there’s an EQ and Comp on the tested channel.
To bypass DSP zero latency monitoring you just un-check ‘Direct monitoring’ in the studio/Audio device tab.