Do Pro Users Record the Axe-Fx through a mic pre?

[Synchronicity]

Thanks for the kind words!

Because you are in real time with your axe and are not using it to echo/input monitor through your DAW, you should not get any latency.

If you were to hear any audible latency, it would most likely be due to Studio One needing an asio offset.

For example, we have an offset option in SONAR that literally fixes drifting audio. This comes from the interface to DAW communication.

I'll dig out one of my old videos to show you what I mean when I get a chance. But sometimes, recorded audio can be early or late. We can test for this by running a noise file on a track and then taking the output of that noise file and feeding it into an input of our interface

We hit record and hope that when the audio records that it is perfect with the test file we had. Chances are it will be late. So we would measure in samples, how off it is...input the offset into our DAW (if applicable) and rerecord the test file to see that it is in perfect sync.

If you are getting latency ( which I know you aren't ) there are loads of things that can cause it. In your particular situation, you'd only pick up latency if you were using your interface to record other things.

I've not tried this using the axe fx interface, but if I loaded a piano VSTi and tried to play it in real time, I believe I would vet latency in the axe with its default buffer size of 2048.

I'd have to set it to at least 128 to be able to play without it being late. That's the only time you run into latency with the setup you and I have. My console works the same way.

Everything is live and input monitoring is not needed, so I have zero latency. I hope these were the answers you were looking for.

Thanks a lot man yeah the only time I've noticed any latency is when using the axe fx as an interface while recording MIDI instruments, and in that case I either turn the buffer way down to as low as possible without audio artifacts, or just switch to ASIO4ALL and bypass the axe fx and go straight from my PC speakers or headphones for recording any MIDI stuff.

But yeah the main thing I was thinking of is what you've mentioned as the drifting audio that isn't in sync, but I'm 99% sure Studio One is automatically compensating for that like the option you have in SONAR, perhaps it could still be offset by a couple of ms without manual tweaking but I'll have a proper scrutinising look tomorrow

Thanks again, hope you have a nice weekend!

 

[pauly]

Gday hochha,
I'm with you there - as with most of the neve products (excluding some digital copycats ) the rndi imparts something special into my bass signal. I have been recording a dry track via the rndi, and a stereo track from the axe fx, and have found that in the mix, the dry track with plugins ends up the winner most times.

Thanks
Pauly

The "pros' should use a Neve RNDI as the instrument interface then run to the AxeFx then direct to an audio interface. The RNDIs are incredible I have several and will be getting more!

 

[Danny Danzi]

Thanks a lot man yeah the only time I've noticed any latency is when using the axe fx as an interface while recording MIDI instruments, and in that case I either turn the buffer way down to as low as possible without audio artifacts, or just switch to ASIO4ALL and bypass the axe fx and go straight from my PC speakers or headphones for recording any MIDI stuff.

But yeah the main thing I was thinking of is what you've mentioned as the drifting audio that isn't in sync, but I'm 99% sure Studio One is automatically compensating for that like the option you have in SONAR, perhaps it could still be offset by a couple of ms without manual tweaking but I'll have a proper scrutinising look tomorrow

Thanks again, hope you have a nice weekend!

You're very welcome. I have studio One 2 too and will check this out also because I'm curious.

I'll do the test there actually and we'll see how off it is. I still have the test file around here somewhere. I'll be in touch.

You have a nice weekend also, thanks!

 

[Admin M@]

I think the issue with USB M@, is most of the USB units suffer with latency issues.

I think USB in this case would specifically refer only to using the Axe-Fx as the audio interface, right? That's what I took the poll to mean...

 

[Synchronicity]

Gday hochha,
I'm with you there - as with most of the neve products (excluding some digital copycats ) the rndi imparts something special into my bass signal. I have been recording a dry track via the rndi, and a stereo track from the axe fx, and have found that in the mix, the dry track with plugins ends up the winner most times.

Thanks
Pauly

Just a little tip, but it's always best to record bass in mono I hear, for it to fit better in the mix/audio spectrum

 

[pauly]

Always

 

[hochha]

I try not to should on anyone.
I am sure that most Pro Audio studios have more high end Neve or Neve level quality mic pres, di's and interfaces than an $250 RNDI
There are also a myriad of ways to skin a cat and yours is just one.
If I was to use an RNDI with an Axe I would record the guitar direct with it and use the thru output to go to the Axe for a sound that is inspiring for tracking and then later re amp and tweak from the di recording.

Did you tell Matt and Chris about using a $250 interface to do the AxeFx tone matching process?

 

[Stratoblaster]

In my extensive experience most pros record the unit from the analog outs direct into the console. Second most popular would be AES. Third would be analog into a preamp. I've never seen SPDIF or USB used in a pro studio... but then, I do know that a number of "guest solos" performed by Fractal artists I work with have been recorded via USB into a laptop on the bus or in the hotel.

Interesting, thanks for this...was always curious as to how pros/studios generally record with the AFX.

 

[USMC_Trev]

I try not to should ..

Not. A. Verb.

 

[THX]

To the OP: I'm not a "Pro" by any stretch of the imagination, but have played for 30+ years. I'm currently using S/PDIF, but only because my current DAW interface (MOTU 1248) doesn't have an AES/EBU interface. S/PDIF and AES/EBU are virtually identical, with S/PDIF being the "consumer" version of the "pro" AES/EBU format. There is a extra "flag" in the data bitstream for AES/EBU making them incompatible with each other, but other than that, they're essentially the same thing with different connectors (75 Ohm RCA for S/PDIF and 110 Ohm XLR for AES/EBU).

I'm some what of a "purist" in that I try to absolutely limit the number of A/D and D/A conversions, so once I get the signal into the AXE-FX, I only want to deal with it in the digital domain. One nice thing that the MOTU interface will let me do with S/PDIF is to double the 48 kHz rate of the AXE-FX to 96 kHz, so I do all of my recording and live sound at 96 kHz. It's not 100% perfect, but it does allow me to run everything in my DAW at 96 kHz instead of 48 kHz. I also play a lot of analog synth (70's & 80's stuff) and it sounds better at the higher rates.

I would run everything at 192 kHz, 24 bit if the AXE-FX and my interface FULLY supported it (the MOTU 1248 does support 192 kHz, but you lose their great built-in reverb, limiter, compressor and EQ processing). I've done a lot of audiophile monitoring and currently use a Sennheiser HDVD-800 with HD-800 headphones through balanced headphone connectors and the difference between GOOD, WELL MASTERED, 48 kHz and 192 kHz (ALL other factors being equal), is simply astounding! It's like a veil being lifted. I cannot wait for the next iteration of the AXE-FX to have "hopefully" have higher sampling rates.

 

[Danny Danzi]

Thanks a lot man yeah the only time I've noticed any latency is when using the axe fx as an interface while recording MIDI instruments, and in that case I either turn the buffer way down to as low as possible without audio artifacts, or just switch to ASIO4ALL and bypass the axe fx and go straight from my PC speakers or headphones for recording any MIDI stuff.

But yeah the main thing I was thinking of is what you've mentioned as the drifting audio that isn't in sync, but I'm 99% sure Studio One is automatically compensating for that like the option you have in SONAR, perhaps it could still be offset by a couple of ms without manual tweaking but I'll have a proper scrutinising look tomorrow

Thanks again, hope you have a nice weekend!

Ok the test results are in for studio one. I have an older version, but that doesn't usually matter as I am getting the same settings using all my interfaces...which is a good thing as consistency is important. I'm going to create a small video for you so you can see how to do this yourself. This is not using the Fractal as an interface though because I couldn't get behind my console to hook that up. It should be easy enough to do. The reason for this, is to show you how off Studio One 2 can be even though they claim compensation. SONAR claims the same and literally has a setting showing what it is compensating for after it profiles your interface. I still had to use a manual offset. So give me like 30 minutes or so, and I'll shoot the video off along with the test file if you or anyone else decides to mess with this.

 

[Danny Danzi]

Hey Danny, love your videos btw, great stuff!

Just wondering if you'd know the answer to this;

If you're using the Axe FX as an interface and recording directly via USB, but monitoring out of the back output into a couple of studio monitors.

There is no latency (to me) of the direct guitar audio (unless monitoring in software at the same time), but perhaps there is some latency on the recording as it travels through the USB cable and into my DAW? I'm using Studio One 2 which I think may even automatically correct for latency on recordings...

If that is the case, is there any issue of where the latency can become apparent or create issues?

Thanks alot!

Ok, check out this vid. It shows you how Studio One 2 did and needed to be manually set.



Here's the test file: www.dannydanzi.com/Fun%20Stuff/48kTestFile.wav

 

[Synchronicity]

Ok, check out this vid. It shows you how Studio One 2 did and needed to be manually set.



Here's the test file: www.dannydanzi.com/Fun%20Stuff/48kTestFile.wav

Thanks so much Danny! You're an amazing part of the community, so grateful for your extensive help and expertise, will get onto syncing up Studio One 2 when I get up tomorrow.

Cheers man, take care!

 

[J.C]

I may have polled incorrectly, but I use the usb of the Axe FX for guitar tone, then switch to my steinberg for everything else.

 

[AlienSexGod]

To the OP: I'm not a "Pro" by any stretch of the imagination, but have played for 30+ years. I'm currently using S/PDIF, but only because my current DAW interface (MOTU 1248) doesn't have an AES/EBU interface. S/PDIF and AES/EBU are virtually identical, with S/PDIF being the "consumer" version of the "pro" AES/EBU format. There is a extra "flag" in the data bitstream for AES/EBU making them incompatible with each other, but other than that, they're essentially the same thing with different connectors (75 Ohm RCA for S/PDIF and 110 Ohm XLR for AES/EBU).

I'm some what of a "purist" in that I try to absolutely limit the number of A/D and D/A conversions, so once I get the signal into the AXE-FX, I only want to deal with it in the digital domain. One nice thing that the MOTU interface will let me do with S/PDIF is to double the 48 kHz rate of the AXE-FX to 96 kHz, so I do all of my recording and live sound at 96 kHz. It's not 100% perfect, but it does allow me to run everything in my DAW at 96 kHz instead of 48 kHz. I also play a lot of analog synth (70's & 80's stuff) and it sounds better at the higher rates.

I would run everything at 192 kHz, 24 bit if the AXE-FX and my interface FULLY supported it (the MOTU 1248 does support 192 kHz, but you lose their great built-in reverb, limiter, compressor and EQ processing). I've done a lot of audiophile monitoring and currently use a Sennheiser HDVD-800 with HD-800 headphones through balanced headphone connectors and the difference between GOOD, WELL MASTERED, 48 kHz and 192 kHz (ALL other factors being equal), is simply astounding! It's like a veil being lifted. I cannot wait for the next iteration of the AXE-FX to have "hopefully" have higher sampling rates.

I agree once you are digital you stay there unless in rare circumstances you need some tone shaping from a super high end tube eq or something. You are also correct that the digital audio data in spdif is exactly the same... the only reason to use AES over spdif is very long cable runs.

Fractal haven't confirmed my earlier question if USB audio is the same or is compressed somewhat as I suspect?

As for recording digital out of the the Axe @ 96khz I am pretty sure the Axe is only 48khz @ 24 bit and you are converting that to 96khz and locking yourself into huge, slow to edit and copy audio files. for ZERO audio quality benefit.

There is an argument for recording an analog source through great convertors @ 96khz or 192khz but in blind tests the difference even on high end convertors & monitors is very marginal. The law of diminishing returns comes in at the steepest end of an exponential curve here. No one can hear anything above 24khz which 48kHz recording covers very well. The additional samples per millisecond may make a difference but can you hear it in a blind test with more than 50% success rate? I doubt it. There was a audible benefit going 16>20>24 bit as that actually gave more detail. I think you would gain more consistently discernable audible quality in many other areas than 96khz recording. It may make you feel good doing it but largely I believe like $150 guitar cables it's minimal difference (compared to say a mogami/canare cable with switchcraft or neutral connectors at 20% of the price) and basically therefore snake oil.

 

[KSPeace]

I will pitch in here just to throw the proverbial wrench.... as a musician, engineer and producer I am looking at exactly what kind of sound is the artist wanting to convey, as to how I am going to set it up to record. Now, I don't know about all of you, but, I find it really hard to get a crappy guitar sound from the axe to the DAW ... so in that instance I am running lines out to an old crappy cab that we can hear the paper rattling on the cone, mic'd up with a turd of a 57. As a PRO, I am PROducing what sound the artist (be that me or someone else) is looking for.

Do you have one more guitar? More than one mic to put on your cab? I mean, the word PRO keeps getting thrown around and to me the PRO way to do it is totally different avenues for different musicians. My approach is different than yours. Yours is different from Cliffs. Cliffs differs from Steve Vai. Via differs from Satch. Are any of us more right than wrong? more wrong than right? Can you utilize the SPIDIF? Do you want to record a dry track along with a polished track? How many sets of cans do you need different mixes for? Too many variables, and zero of them is any more right or wrong than the next.

 

[pauly]

I strongly disagree.

If we were talking a recording of a single guitar, maybe it's not such a difference.

When recording music however, we may have (say) 40 tracks and every combining action (mix bus), plugin, conversion, change etc needs to make a change to the recorded data. My strong opinion is that the higher the resolution (of recording), the better the overall result.

As an example, lets say one track is a stereo set of overheads. That may have 5 plugins on it, each imparting thier own changes. That track is then bussed into a 'Drumkit' stereo bus (digitally), and then that stereo bus has 3 plugins on it. That bus is subsequently combined with all the other busses, effects, and tracks and printed to a stereo mix track (oh the mixing also effects the quality of course). Then the track is mastered which may have an untold amount of processing, but lets say it's multiband compression, & EQ, & dithering only. - Then the master is converted to the desired output formats such as 16 bit 44.1 for CD, 24 bit 48 for listening etc etc.

During that process - the more bits, the better - And there will be an audible difference. It will just sound better.

Thanks
Pauly

SNIP

There is an argument for recording an analog source through great convertors @ 96khz or 192khz but in blind tests the difference even on high end convertors & monitors is very marginal. The law of diminishing returns comes in at the steepest end of an exponential curve here. No one can hear anything above 24khz which 48kHz recording covers very well. The additional samples per millisecond may make a difference but can you hear it in a blind test with more than 50% success rate? I doubt it. There was a audible benefit going 16>20>24 bit as that actually gave more detail. I think you would gain more consistently discernable audible quality in many other areas than 96khz recording. It may make you feel good doing it but largely I believe like $150 guitar cables it's minimal difference (compared to say a mogami/canare cable with switchcraft or neutral connectors at 20% of the price) and basically therefore snake oil.

 

[∞Fractals]

For music in general, mo' bits = mo' betta

I remember going from superb analog via very high-end FET transducers on vinyl (class A amplification e.g. Threshold Amps to Koss electrostatic speakers) to digital CDs w/ 44.1k @ 16 bits; sucked big time (still does IMO).

Same old arguments about what you can and cannot technically hear BS. (Let's not even talk about mpeg.) I still waiting for digital to match analog; there is hope with 24 bits and higher sampling.

IMO 96k @ 24 bits is a step up and can easily be heard. I have no issue with more signal/data ... just because it slows down your workflow, well I remember the very first digital processors and buying my first 1 gig HD (over $1000) to master a CD with, and yes, I would lay on the floor of the studio waiting for the processing.

Now a 1 gig flash drive is free and I bet my iPhone 6s w/ a A9 chipset has more power than my old Atari ST on a Motorola 030. (I even wonder about the 040 on my prior Apple Quadra although it had a DSP as well.)

Point being, don't let processing power determine fidelity (unless you are FAS and need real-time processing HP (Shark DSPs), but that's a slightly different animal.)

There are trade offs, and while fidelity is one perspective, so is ease of use and things (random access, modeling 250 amps, etc. ) we can do in the digital domain that would have been impossible in the analog world. YMMV.

 

[vangrieg]

IMO 96k @ 24 bits is a step up and can easily be heard. I have no issue with more signal/data ... just because it slows down your workflow, well I remember the very first digital processors and buying my first 1 gig HD (over $1000) to master a CD with, and yes, I would lay on the floor of the studio waiting for the processing.

It's not just that. First, if we're still talking Axe FX recording, it's worth remembering that no matter what happens inside it, all that goes out is 48 kHz 24 bit if you're using digital, less than 24 bit in reality if you're using analog. Nothing better can come out of the unit, period. So if it's the highest possible fidelity you're after, go with digital at 48 kHz 24 bit while recording axe FX. That's simple as a nail, IMO.

What happens to the signal afterwards is getting moot maybe. For processing it, oversampling will be beneficial a lot of times. However, plugins do oversample whether you want it or not. They will upsample, do what they need to do, and then downsample, and most of the time you cannot turn it off. And the signal will go through this every time it enters a new plugin. Then you'll downsample it all to 44.1 kHz 16 bit. And every time this resampling will introduce errors. Does it help if all this craziness happens at an already crazy rate of 96 kHz? I'm not so sure. And if it does, why stop there? Why not run your projects at 192?

 

[∞Fractals]

First, if we're still talking Axe FX recording, it's worth remembering that no matter what happens inside it, all that goes out is 48 kHz 24 bit if you're using digital, less than 24 bit in reality if you're using analog. Nothing better can come out of the unit, period. So if it's the highest possible fidelity you're after, go with digital at 48 kHz 24 bit while recording axe FX. That's simple as a nail, IMO.

RE: AXE FX ... No question ... in full agreement.

I was speaking generally about being able to hear the differences in sample rates and bit depth in music.