audio interface outputs to inputs
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- KVRist
- Topic Starter
- 102 posts since 4 Feb, 2012 from residant in ksa/jeddah
hello, i have an audio interface 'roland ua-s10' & using a daw 'cubase' and monitoring through the headphones and i would like to get a step closer to real analogue finishing for cheap. i'm thinking of connecting the analog outputs to the analog inputs to get benifit of the pre-amps .. i wonder if anybody tried this before - still not have the cables so didn't tried it myself yet .. any body knows if it works or it will damage the unit?
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- KVRian
- 626 posts since 15 Jun, 2015
If you're looking to add character to your signal, you need to add a preamp to your signal chain that will give you some character. If you want to go cheap, look for a used ART Tube MP studio premap. They can usually be found on eBay or Craig's list for about $20.
Connecting your audio interface's outputs to inputs will just give you feedback unless you've come some sort of special routing software. But even if you could do it safely, audio interfaces are generally built to have transparent preamps. You're not going to get much (or any) character from them.
Connecting your audio interface's outputs to inputs will just give you feedback unless you've come some sort of special routing software. But even if you could do it safely, audio interfaces are generally built to have transparent preamps. You're not going to get much (or any) character from them.
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- KVRAF
- 6419 posts since 22 Jan, 2005 from Sweden
Level will be strange, since preamps probably are bypassed with line inputs from line outputs.
I assume very little character will be added with just going through - no amplification.
Some inputs are adjustable from line level up to microphone levels in sensitivity - so unless you amplify anything, I wonder how much character is added that is of benefit.
I've done this connection for pure recording of roundtrip latency tests to see if daw compensate correctly. Let metronome click go out and record that again and see if lines up.
I think hardware emulating plugins might be a better choice. Like Waves NLS or something that has three different analog consoles. It really brings something of value, I think - even just putting the bus component of stereobus out. Then you can continue to selected tracks or all tracks within the mix as well.
But just try your idea, awful lot of work to rerecord everything you want analog touch on. But your taste is what matters.
I assume very little character will be added with just going through - no amplification.
Some inputs are adjustable from line level up to microphone levels in sensitivity - so unless you amplify anything, I wonder how much character is added that is of benefit.
I've done this connection for pure recording of roundtrip latency tests to see if daw compensate correctly. Let metronome click go out and record that again and see if lines up.
I think hardware emulating plugins might be a better choice. Like Waves NLS or something that has three different analog consoles. It really brings something of value, I think - even just putting the bus component of stereobus out. Then you can continue to selected tracks or all tracks within the mix as well.
But just try your idea, awful lot of work to rerecord everything you want analog touch on. But your taste is what matters.
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- KVRist
- Topic Starter
- 102 posts since 4 Feb, 2012 from residant in ksa/jeddah
thanks for help, i though it warms up the sound and make to the sound something closer to analog summing, like instrument separation and so...
just have another question, what happens if connected a synth line-out to the mic xlr pre-ams inputs? just would like to know
just have another question, what happens if connected a synth line-out to the mic xlr pre-ams inputs? just would like to know
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- KVRAF
- 15253 posts since 8 Mar, 2005 from Utrecht, Holland
Worst case: you blow up a OPAMP chip, and replacing that can cost as much as a whole new interface.walidantar wrote:what happens if connected a synth line-out to the mic xlr pre-ams inputs?
XLR inputs are usually Mic level, that is a signal in the order of some milliVolts.
Synths usually output at Line level, which is a signal in the order of one Volt.
So the signal is a magnitude 100 or so too hot.
You can compensate with a proper DI box (one with -20 / -40 dB pad switches) or by setting the synth's volume very low. The latter will give significant extra noise.
We are the KVR collective. Resistance is futile. You will be assimilated. 
My MusicCalc is served over https!!
My MusicCalc is served over https!!
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- KVRAF
- 3080 posts since 17 Apr, 2005 from S.E. TN
I get the impression that many mastering guys like to send audio thru expensive outboard chains and then re-record the results. They call it "pitching and catching". Sometimes using different audio interfaces or programs, or even different computers for the pitching and catching. Sometimes send to the analog chain from one program/interface combination, then re-record the analog chain output with another program/interface.
I don't know the benefits of this practice. So many mastering engineers practice the rigamarole, maybe it makes sense or they would do it some other way. Or possibly at least some of it is expensive superstition. Dunno. If it makes profit from happy clients then its all good. Don't mess with success. Don't look the horse which lays the golden eggs in the mouth.
There is also the practice of re-amping-- Record guitar or bass parts fairly dry and flat with a good direct box, then later on feed the clean track to a guitar amp, adjust to taste, mic the amp and re-record the amp into a new DAW track. The advantage being that you can take your time experimenting with it, try different amps and speakers and mics and settings to find which works best with the track. Maybe better than getting stuck with a guitar+amp tone that turns out not fitting the song, and either having to put up with the "imperfect sound" or bite the bullet and re-record the track hoping to get it right next time.
Sometimes folk will run vocal or whatever out of the computer to a favorite hardware compressor or EQ and re-record the result.
Even if the above instances might sometimes make sense, I doubt that just re-recording the interface-out to the interface-in would accomplish much except to add at least some small amount of extra noise, hum and distortion to the track. In the above cases, there is also the probability of at least slightly degrading the signal, but the user is hoping the analog mangling will make it sound sufficiently better that the degradation is a tolerable side-effect.
The fellas serious about it, just take as much care possible to minimize degradation from the extra D-A and A-D conversions. Solid wiring and real good interfaces can make the degradation so small that it might be difficult to measure and perhaps impossible to hear. But a lesser interface and/or sloppy wiring might have problems.
I don't know the benefits of this practice. So many mastering engineers practice the rigamarole, maybe it makes sense or they would do it some other way. Or possibly at least some of it is expensive superstition. Dunno. If it makes profit from happy clients then its all good. Don't mess with success. Don't look the horse which lays the golden eggs in the mouth.
There is also the practice of re-amping-- Record guitar or bass parts fairly dry and flat with a good direct box, then later on feed the clean track to a guitar amp, adjust to taste, mic the amp and re-record the amp into a new DAW track. The advantage being that you can take your time experimenting with it, try different amps and speakers and mics and settings to find which works best with the track. Maybe better than getting stuck with a guitar+amp tone that turns out not fitting the song, and either having to put up with the "imperfect sound" or bite the bullet and re-record the track hoping to get it right next time.
Sometimes folk will run vocal or whatever out of the computer to a favorite hardware compressor or EQ and re-record the result.
Even if the above instances might sometimes make sense, I doubt that just re-recording the interface-out to the interface-in would accomplish much except to add at least some small amount of extra noise, hum and distortion to the track. In the above cases, there is also the probability of at least slightly degrading the signal, but the user is hoping the analog mangling will make it sound sufficiently better that the degradation is a tolerable side-effect.
The fellas serious about it, just take as much care possible to minimize degradation from the extra D-A and A-D conversions. Solid wiring and real good interfaces can make the degradation so small that it might be difficult to measure and perhaps impossible to hear. But a lesser interface and/or sloppy wiring might have problems.
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- KVRAF
- 6419 posts since 22 Jan, 2005 from Sweden
Following episodes on Pensados Place http://www.pensadosplace.tv you learn many things.
What seems to be a trend is the external summing boxes that create something that old analog consoles would.
Things I read about have been recording on an old real-to-real of good quality and then back into computer, to get natural tape saturation.
I had a real-to-real in the 80's, doing some sound-on-sound before I got sequencers and stuff. And doing bias current adjustments to tape you record a sine on 1 khz and 10 khz to adjust current to produce a linear curve. Recording this sine, and looking the oscilloscope you see the even harmonics added - how the sine has really strong first even harmonics added. So doing this on music, makes it all sound better to ears than original.
Harmonics are multiples of basenote pitch frequency, even sound pleasent to ears, odd ones 3,5 where, 1 is basenote and not harmonic, sound harsch to our ears.
Experiment doing a C3 note on a keyboard, then double up with also sounding C3+C4 and how much richer it sounds. That's even harmonics - very simplified.
If you have a Hammond organ, plugins or hardware - this is additive synthesis. Creme colored bars are even harmonics, and brown ones odd harmonics. So you make your sound from combining these even+odd into what you want. To make it complicated, every bar is not a pure sine, but in principle this is how it is. Every timbre of an instrument is a combination of even+odd harmonics - that is what makes a violin apart from a guitar etc.
So even harmonics are pleasing to ears, and odd ones brings character.
Other units that bring something extra are high quality analog stuff, like Neve, which use solid state in combination with transformers. Units that favour even harmonics in material making it sound just a bit fuller.
Then there is tube based stuff, which also favours even harmonics.
Just doing a re-amping thing with standard solid state preamps, just bring the same amount of even+odd harmonics - so you will probably don't feel it's worth the trouble - being analog and all. What you are after is distortion on second harmonics.
When you do specs for hifi equipment like amps, they measure THD, harmonic distorition on third harmonic, which is the first odd harmonic - base note being first pitch. Measuring on third is to do with that this is unpleasent to ears.
So my tape example above - seing a very strong distortion, but on first even harmonic - it makes it sound more pleasent to ears than original.
But to do it all ITB, I think Waves NLS is a really good start. I think it was Spike console that was an SSL analog console brought into digital realm. Just putting the bus component on stereo bus out made a huge difference.
But it's all down to taste, at least if you have experienced enough ears to hear the difference if deteriorating it more than brings something better. If is guaranteed it sounds different, but is it better?
And even if listening to mixes a lot, doing too long sessions really is dangerous, next time you open a project the next day you think - what was I thinking doing this? Your ears fatigue.
I listened to a Pensado episode, where they made a visit to another studio. And folks there talked about how much time they spent experimenting with the new preamps they bought, and various mikes etc. What you learn is that nothing creates itself - you have to experiment to learn what you've got to make best use of it.
And if you think you spent enough time on a mix, doing one week - maybe consider what demon producer Max Martin said:
- be ready to spend a month on a mix to really get it right
What seems to be a trend is the external summing boxes that create something that old analog consoles would.
Things I read about have been recording on an old real-to-real of good quality and then back into computer, to get natural tape saturation.
I had a real-to-real in the 80's, doing some sound-on-sound before I got sequencers and stuff. And doing bias current adjustments to tape you record a sine on 1 khz and 10 khz to adjust current to produce a linear curve. Recording this sine, and looking the oscilloscope you see the even harmonics added - how the sine has really strong first even harmonics added. So doing this on music, makes it all sound better to ears than original.
Harmonics are multiples of basenote pitch frequency, even sound pleasent to ears, odd ones 3,5 where, 1 is basenote and not harmonic, sound harsch to our ears.
Experiment doing a C3 note on a keyboard, then double up with also sounding C3+C4 and how much richer it sounds. That's even harmonics - very simplified.
If you have a Hammond organ, plugins or hardware - this is additive synthesis. Creme colored bars are even harmonics, and brown ones odd harmonics. So you make your sound from combining these even+odd into what you want. To make it complicated, every bar is not a pure sine, but in principle this is how it is. Every timbre of an instrument is a combination of even+odd harmonics - that is what makes a violin apart from a guitar etc.
So even harmonics are pleasing to ears, and odd ones brings character.
Other units that bring something extra are high quality analog stuff, like Neve, which use solid state in combination with transformers. Units that favour even harmonics in material making it sound just a bit fuller.
Then there is tube based stuff, which also favours even harmonics.
Just doing a re-amping thing with standard solid state preamps, just bring the same amount of even+odd harmonics - so you will probably don't feel it's worth the trouble - being analog and all. What you are after is distortion on second harmonics.
When you do specs for hifi equipment like amps, they measure THD, harmonic distorition on third harmonic, which is the first odd harmonic - base note being first pitch. Measuring on third is to do with that this is unpleasent to ears.
So my tape example above - seing a very strong distortion, but on first even harmonic - it makes it sound more pleasent to ears than original.
But to do it all ITB, I think Waves NLS is a really good start. I think it was Spike console that was an SSL analog console brought into digital realm. Just putting the bus component on stereo bus out made a huge difference.
But it's all down to taste, at least if you have experienced enough ears to hear the difference if deteriorating it more than brings something better. If is guaranteed it sounds different, but is it better?
And even if listening to mixes a lot, doing too long sessions really is dangerous, next time you open a project the next day you think - what was I thinking doing this? Your ears fatigue.
I listened to a Pensado episode, where they made a visit to another studio. And folks there talked about how much time they spent experimenting with the new preamps they bought, and various mikes etc. What you learn is that nothing creates itself - you have to experiment to learn what you've got to make best use of it.
And if you think you spent enough time on a mix, doing one week - maybe consider what demon producer Max Martin said:
- be ready to spend a month on a mix to really get it right
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- KVRAF
- 3080 posts since 17 Apr, 2005 from S.E. TN
Yes its a matter of taste more than science. The rich and famous probably have skills which help explain achievements, but nobody gets rich and famous enough to become immune to superstition and placebo effect.
Maybe the tendency to treat giant heaps of expensive precision gear as mere fancy fuzz stompboxes is why I can't hardly listen to pop music anymore. All that time money and tubes, transformers, etc seems to output typical music too distorted to tolerate. Musical taste is one thing but "no dynamics wall of distortion" is something else.
Fancy vintage gear might have nonlinearities when pushed, but was generally designed and intended to be as clean possible, not expensive fuzz boxes.
Sure sometimes distortion sounds great. And sometimes it don't. Sure sometimes fat sounds great, and sometimes thin sounds great. Wall-to-wall fat is not long-term gratifying.
Listen to a pianist who octave doubles near everything and it gets boring fast. Same for a guitarist who wes-montgomery-style octave doubles all his leads all night. If "fatness" is so ideal why aren't 12 string guitars more popular? Because of the octave doubling and extra richness they ought to be lots fatter and therefore near irresistably better to play or listen to, compared to mere 6 stringers.
Even harmonics are not necessarily mellow and odd harmonics are not necessarily harsh. Well-played clarinet (especially bass clarinet and all clarinets in the lower registers) are among the mellowest of instruments and the sound is predominantly odd harmonics. Same deal for well-done bassoon or oboe. The high school bassoonist at his best might sound like a woeful sick cow, but a pro makes beautiful mellow tones out of those odd harmonics.
Hammond registrations of all white drawbars (the even harmonics) don't sound especially fat. More dull and churchy-sounding. Or transistor organs with the flute filters and all 16, 8, 4, 2 foot tabs pressed. Might sound fat for a few minutes before becoming oppressively dull.
Mellow octave doubling 2nd harmonic distortion can be real purty on electric bass. It can even sound pleasing as a "long term standard tone" ala old ampeg B15 and B18 fliptop bass amps pushed a bit. But typical speakers have so much natural second harmonic bass distortion that if you excessively intentionally "mellow distort" the bass and it sounds better live or on "low distortion" studio monitors-- When a consumer's speakers octave double what was already octave-doubled maybe it sounds gawdawful on the customer's system.
Maybe the tendency to treat giant heaps of expensive precision gear as mere fancy fuzz stompboxes is why I can't hardly listen to pop music anymore. All that time money and tubes, transformers, etc seems to output typical music too distorted to tolerate. Musical taste is one thing but "no dynamics wall of distortion" is something else.
Fancy vintage gear might have nonlinearities when pushed, but was generally designed and intended to be as clean possible, not expensive fuzz boxes.
Sure sometimes distortion sounds great. And sometimes it don't. Sure sometimes fat sounds great, and sometimes thin sounds great. Wall-to-wall fat is not long-term gratifying.
Listen to a pianist who octave doubles near everything and it gets boring fast. Same for a guitarist who wes-montgomery-style octave doubles all his leads all night. If "fatness" is so ideal why aren't 12 string guitars more popular? Because of the octave doubling and extra richness they ought to be lots fatter and therefore near irresistably better to play or listen to, compared to mere 6 stringers.
Even harmonics are not necessarily mellow and odd harmonics are not necessarily harsh. Well-played clarinet (especially bass clarinet and all clarinets in the lower registers) are among the mellowest of instruments and the sound is predominantly odd harmonics. Same deal for well-done bassoon or oboe. The high school bassoonist at his best might sound like a woeful sick cow, but a pro makes beautiful mellow tones out of those odd harmonics.
Hammond registrations of all white drawbars (the even harmonics) don't sound especially fat. More dull and churchy-sounding. Or transistor organs with the flute filters and all 16, 8, 4, 2 foot tabs pressed. Might sound fat for a few minutes before becoming oppressively dull.
Mellow octave doubling 2nd harmonic distortion can be real purty on electric bass. It can even sound pleasing as a "long term standard tone" ala old ampeg B15 and B18 fliptop bass amps pushed a bit. But typical speakers have so much natural second harmonic bass distortion that if you excessively intentionally "mellow distort" the bass and it sounds better live or on "low distortion" studio monitors-- When a consumer's speakers octave double what was already octave-doubled maybe it sounds gawdawful on the customer's system.
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- KVRer
- 8 posts since 22 Jun, 2017
I doubt you would do any harm, opamps are typically quite robust. Remember that you will be losing SNR with that configuration. You are basically adding DAC noise + preamp noise + ADC noise to your signal... is the 'analog finish' you are looking for worth it?
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- KVRist
- Topic Starter
- 102 posts since 4 Feb, 2012 from residant in ksa/jeddah
i've tried it and applied the internal compressor, i preferred not to do it because of the latency issues - roland audio interfaces when comes to latency are not perfect so it sounded a bit like delay/chorus applied .. and also as RichieWitch said that the DACs in audio interfaces are transparent .. it's not worth it
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- KVRAF
- 6419 posts since 22 Jan, 2005 from Sweden
Latency part is only while realtime - that will brings some chorus phasing to it.walidantar wrote:i've tried it and applied the internal compressor, i preferred not to do it because of the latency issues - roland audio interfaces when comes to latency are not perfect so it sounded a bit like delay/chorus applied .. and also as RichieWitch said that the DACs in audio interfaces are transparent .. it's not worth it
Daw should adjust so it all align properly to the sample when recorded clip.
Do roundtrip latency check and pdc test by recording metronome with this loopback and see if it align with grid. If it doesn't daw is crap or interface does not report correct latency, and look in daw for setting for general settings adjustment as milliseconds or samples that make it line up.
But unless really high end preamps or tape machine etc - you will not get the "analog" thingy.
But experimenting is fun so no harm in trying things.
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- KVRist
- Topic Starter
- 102 posts since 4 Feb, 2012 from residant in ksa/jeddah
thanks dear, actually i often play in real-time 'which's the cause i experienced as you said chorus phasing'Latency part is only while realtime - that will brings some chorus phasing to it.
Daw should adjust so it all align properly to the sample when recorded clip.
i often play in real-time for entertainment as i'm very interested in such things but not taking it mush in a serious way; just for fun
but not a big matter, maybe some time in the future i get a real thing, a better audio interface, analog summing mixer & analogue mastering parametric equalizer
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- KVRAF
- 15253 posts since 8 Mar, 2005 from Utrecht, Holland
Believe me: you can do without those last two analog boxes. Digital delivers better quality. There's no added pixie dust by analog equipment.walidantar wrote:maybe some time in the future i get a real thing, a better audio interface, analog summing mixer & analogue mastering parametric equalizer
If you feel your current mixes are missing a certain je ne sais quoi it's more likely your mixing skills are just not up to par than inserting whatever tool will fix it.
We are the KVR collective. Resistance is futile. You will be assimilated.
My MusicCalc is served over https!!
My MusicCalc is served over https!!
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- KVRAF
- 6419 posts since 22 Jan, 2005 from Sweden
The good analog summing boxes do add something like analog consoles do according to an episode on http://pensadosplace.tv .BertKoor wrote:Believe me: you can do without those last two analog boxes. Digital delivers better quality. There's no added pixie dust by analog equipment.
One of ten latest, I think, but don't remember the guest.
But think the Waves NLS(non-linear summer) is rather nice. Even just using the stereo bus component do nice things.
So until my ears are fully educated, I'll use NLS.
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- KVRAF
- 9576 posts since 16 Dec, 2002
If youre wanting to add some analog character try one of Shattered Glass free plugs
SGA1566 and Code Red Free sound lovely
http://www.shatteredglassaudio.com/gear ... id=digital
Also check out the extensive range from AirWindows that do all sorts of magicness
SGA1566 and Code Red Free sound lovely
http://www.shatteredglassaudio.com/gear ... id=digital
Also check out the extensive range from AirWindows that do all sorts of magicness
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