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How to Record Hardware Synths w/ No Latency in Ableton Desktop Synthesizers
Old 13th December 2013
  #1
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Thread Starter
How to Record Hardware Synths w/ No Latency in Ableton

Hi guys,

I am trying to record my hardware synths in Ableton, but I can't get the timing dead on. I have used the walk throughs and some googling but I still can't get it perfect.

Is it even possible to have my soundcard settings and track delays set so perfectly that I can record a take, and the recording will be 100% on time. IE: Recording a kick drum and having the recording w/ the kick perfectly aligned with 1.1 of the bar (and not having to adjust)

If that is possible, what's the best way to go about determining my settings?

Basically I seem to get close, but I am always a little off, which causes problems when recording long pieces (in sync w/ existing audio) especially with sounds that don't have a clear transient (to re-align midway through the recording).

This is really causing me problems with my poly synths, drum machines, and external delay effects b/c the timing is integral to the groove, and it's a pain to align after the fact sometimes.

Any help or comments are greatly appreciated.
Old 13th December 2013
  #2
Old 13th December 2013
  #3
Can you give an description of your MIDI signal flow?
Old 13th December 2013
  #4
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Stevism's Avatar
yeah let's talk more about your midi out flow and your audio in flow.

if you're OCD about timing there really isn't anyway to get it 100% perfect. for this sort of surgical precision i think the best way is to record digestible bits, then go and zoom in ultra close and align the actual waveforms. sometimes i play it as slow as my DAW's playback speed will allow, and work on stuff in that timeframe.

otherwise, a lot of DAWs are good at identifying the overall latency and telling you in ms. i nudge my midi info back by this amount, and then turn off the option to "correct for latency" if it's on. that way i can jam relatively close in time, and record without worrying how it will get moved around once i'm done recording.
Old 13th December 2013
  #5
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Thread Starter
Hi guys,

The jitter sucks, I have accepted defeat there. What I am experiencing is a bit of delay before the audio is recorded. For example a 4 on floor kick pattern in a drum machine in midi on 1.1, 1.2, 1.3, 1.4 when recorded, the kicks are more like 1.17, 1.27, 1.37, 1.47.

How can I zero in and adjust the latency so it's dead on (so I don't have to adjust the start point of the audio recording). I have tried to adjust the delay compensation in "external instrument" but I can't seem to get it perfect.

My setup:

1. Ableton Live, using the External Input or External Effect devices
2. Motu 828 Mk3 Hybrid Interface (sending out the midi)
3. Kawai Midi Patchbay In, sending Midi out to each synth on it's own channel (IE: Moog on Channel 1, Juno on 2, etc)
4. Audio In on Motu 828 Mk3 Hybrid Interface (again w/ each synth on it's own channel).
Old 13th December 2013
  #6
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Quote:

How can I zero in and adjust the latency so it's dead on (so I don't have to adjust the start point of the audio recording). I have tried to adjust the delay compensation in "external instrument" but I can't seem to get it perfect.
this is gonna be so obvious to you in a second, but give yourself a bar of silence so the midi clock can lock. Probably a 1/4 note would be enough, but as a rule I use a full bar.
Old 13th December 2013
  #7
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Thread Starter
Quote:
Originally Posted by doulos30 View Post
this is gonna be so obvious to you in a second, but give yourself a bar of silence so the midi clock can lock. Probably a 1/4 note would be enough, but as a rule I use a full bar.
Thanks for this - follow up question:
Should I be doing what you suggested in addition to adjusting latency in the "external instrument" device?
Old 13th December 2013
  #8
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Stevism's Avatar
when you're talking milliseconds, it's really hard to get it perfect because the latency only has to vary every so slightly during playback/recording for it to get thrown off a little bit. and from my brief understanding, there are a number of factors that lead to this slightly changing latency shift...such as CPU usage, strain on the interfaces, etc.

even then it won't be 100% perfect because it's not analog. the conversion processes alone will keep it a bit delayed. when you think about it milliseconds accuracy is incredible for everything that's being accomplished.

now what would be interested would be a plug-in on midi and audio tracks that monitored in real time what the current latency was, and slightly shifted the upcoming midi notes accordingly. that might lead to a lot of inaccurate problems but with time could be kind of cool. maybe i will start working on something like that.






or maybe i'm just a latency noob that doesn't know how to accurately solve this problem lol.
Old 14th December 2013
  #9
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You are about to enter a world of pain my friend. Really boils down to 3 things : use another DAW, Expert Sleepers, Innerlock systems. And IM not joking when I say thats it. That is it. Ive pretty much given up on external sequencing until I get Expert Sleepers - I just record to audio and arrange that way but its not the way I want to work its the way Im forced to work. Im so bloody used to the GUI I can find my way around quick and easy
Old 14th December 2013
  #10
Quote:
Originally Posted by Stevism View Post
if you're OCD about timing there really isn't anyway to get it 100% perfect. for this sort of surgical precision i think the best way is to record digestible bits, then go and zoom in ultra close and align the actual waveforms.
+1

I use an 828mk3 for my main interface as well, although I don't use the MIDI output.


Is every recorded track off by the same amount, and does it drift over time? I use my old Layla20 (and two cheaply acquired sister units) with an old desktop to multitrack record from the 828 and my other hardware, and I definitely notice it get off as time goes on when examining the individually recorded tracks on the timeline. But luckily they are all off the same amount, so a global 'slice and pull back' tends to work.
Old 14th December 2013
  #11
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Mate you should check out expert sleepers as you have a dc coupled interface you can use a couple of floating ring ts cables (diode and capacitor) and you can use silent way - that's for cv control - I'm not sure about midi over this connection but check - you'll get sample accurate audio and midi. Otherwise you simply can't do it without drift - sometimes ill just keep a loop going in giest and keep a line running from a shruthi (or any external synth) and you can hear it drifting! It's a joke.
Old 14th December 2013
  #12
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Quote:
Originally Posted by doulos30 View Post
this is gonna be so obvious to you in a second, but give yourself a bar of silence so the midi clock can lock. Probably a 1/4 note would be enough, but as a rule I use a full bar.
Which daw do you use?
Old 14th December 2013
  #13
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intuitionnyc's Avatar
I have spent a great deal of time (and sanity) in Ableton Live when trying to get my hardware timing right with my software. And I say "timing right" because there are several problems when timing your DAW, MIDI and Audio hardware together.

1) Getting your MIDI to be on time when you are inputing MIDI messages into your DAW. (for example, you playing on a keyboard and your notes entering in the MIDI event in the DAW exactly how you played them). For this, I have found that Logic tracks your MIDI events much better than Ableton, but Ableton is pretty good. For Ableton, I was blown away that when my MIDI messages were triggering drum sounds which were quantized, the audio was off by different increments. Logic was astoundingly better.

2) The latency that occurs when you are tracking (playing and recording at the same time) is another timing issue. For example, you press a key on the keyboard/controller and there is a delay between when you press the key and when you hear the sound of the note that you pressed through your monitors. Much of this is determined by the buffer size (usually found in your DAW's preferences). The lower the buffer setting the more "instantaneous" you will hear the note you played. Lowering your buffer setting will also be taxing on your CPU power (especially if you have plug-ins and lots of tracks playing). Some people track and mix at different buffer settings. While this is effective this can be a pain in the ass. There will always be latency because of the fact that you are pressing a key, the message gets converted to a MIDI message and out your computer to you MIDI interface, to your piece of hardware and then the audio goes BACK to your computer (not to mention if it has to go through plug-ins). You can get it down close enough so that it is negligible. Now, some audio interfaces and DAWs allow for zero latency monitoring, but you usually can't put too many (or any) plug-ins on while playing. This is because the audio must go through a series of calculations when going through plug-ins and it takes time. With Zero Latency Monitoring, you're pretty much hearing the audio before it enters your DAW for computing and recording. This is also an effective way to track.

3) The latency that occurs when you want to record audio from your hardware after you've recorded your MIDI events is also troublesome sometimes (this ties into #2). Again, it comes down to buffer size. BUT, most DAW's have a time compensation error adjustment for such. Especially after you've recorded it. Here's sorta how to do it:

Set up a sound in your hardware that has instantaneous sound if you press a key. Like a sine wave or a sawtooth wave on a synth, or a kick drum or something with a VERY low attack. Not a pad or something with a long attack. Now have a note on the first beat trigger the hardware. Record this sound. Look at your waveform after recording. You can usually highlight the space between the 1st beat and where your waveform starts. If there is a negligible amount of time, then you should be recording on time. If there is space (let's say 43 milliseconds), then adjust the timing error on your track by -43 milliseconds and it should be pretty close to on time. This time can vary from one piece of external hardware to another.

Many people will record their events and just adjust the timing error compensation by ear until it "sits right" in the mix.

If you use ableton, they have a cool way to setup your external hardware on the Ableton website. Look on their or youtube for the video "setting up your external hardware with Push." It's a cool video because it allows you to set up multiple external hardware setup "plug-ins." It's much easier than routing a midi channel to an audio channel. Also, you can save them and their latency settings. This way, each piece of external hardware (say a access virus, jupiter 6, tb-303, etc.) all have different latency settings.

Please let me know if what i've said is inaccurate or if you have any questions. I don't have all the answers, just stuff I've experienced and what helps me. ALSO, check the video on reducing latency in Ableton Live on youtube.
Old 14th December 2013
  #14
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Kissed's Avatar
Quote:
Originally Posted by intuitionnyc View Post
I have spent a great deal of time (and sanity) in Ableton Live when trying to get my hardware to get timing right. And I say "timing right" because there are several problems when timing your DAW, MIDI and Audio hardware together.

1) Getting your MIDI to be on time when you are inputing MIDI messages into your DAW. (for example, you playing on a keyboard and your notes entering in the MIDI event in the DAW exactly how you played them). For this, I have found that Logic tracks your MIDI events much better than Ableton, but Ableton is pretty good. For Ableton, I was blown away that when my MIDI messages were triggering drum sounds which were quantized, the audio was off by different increments. Logic was astoundingly better.

2) The latency that occurs when you are tracking (playing and recording at the same time) is another timing issue. For example, you press a key on the keyboard/controller and there is a delay between when you press the key and when you hear the sound of the note that you pressed through your monitors. Much of this is determined by the buffer size (usually found in your DAW's preferences). The lower the buffer setting the more "instantaneous" you will hear the note you played. Lowering your buffer setting will also be taxing on your CPU power (especially if you have plug-ins and lots of tracks playing). Some people track and mix at different buffer settings. While this is effective this can be a pain in the ass. There will always be latency because of the fact that you are pressing a key, the message gets converted to a MIDI message and out your computer to you MIDI interface, to your piece of hardware and then the audio goes BACK to your computer (not to mention if it has to go through plug-ins). You can get it down close enough so that it is negligible. Now, some audio interfaces and DAWs allow for zero latency monitoring, but you usually can't put too many (or any) plug-ins on while playing. This is because the audio must go through a series of calculations when going through plug-ins and it takes time. With Zero Latency Monitoring, you're pretty much hearing the audio before it enters your DAW for computing and recording. This is also an effective way to track.

3) The latency that occurs when you want to record audio from your hardware after you've recorded your MIDI events is also troublesome sometimes (this ties into #2). Again, it comes down to buffer size. BUT, most DAW's have a time compensation error adjustment for such. Especially after you've recorded it. Here's sorta how to do it:

Set up a sound in your hardware that has instantaneous sound if you press a key. Like a sine wave or a sawtooth wave on a synth, or a kick drum or something with a VERY low attack. Not a pad or something with a long attack. Now have a note on the first beat trigger the hardware. Record this sound. Look at your waveform after recording. You can usually highlight the space between the 1st beat and where your waveform starts. If there is a negligible amount of time, then you should be recording on time. If there is space (let's say 43 milliseconds), then adjust the timing error on your track by -43 milliseconds and it should be pretty close to on time. This time can vary from one piece of external hardware to another.

Many people will record their events and just adjust the timing error compensation by ear until it "sits right" in the mix.

If you use ableton, they have a cool way to setup your external hardware on the Ableton website. Look on their or youtube for the video "setting up your external hardware with Push." It's a cool video because it allows you to set up multiple external hardware setup "plug-ins." It's much easier than routing a midi channel to an audio channel. Also, you can save them and their latency settings. This way, each piece of external hardware (say a access virus, jupiter 6, tb-303, etc.) all have different latency settings.

Please let me know if what i've said is inaccurate or if you have any questions. I don't have all the answers, just stuff I've experienced and what helps me. ALSO, check the video on reducing latency in Ableton Live on youtube.
Well written, thanks for this.
Old 14th December 2013
  #15
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Thank you and well written. Ive basically tried everything and I basically need to get either expert sleepers, innerlock, or I just found this one - its a hardware sequencer, looper and more and it claims 2ms timing with ableton live - 250eur. FYRD Instruments - Has maps for shruthi, volca, mopho, tetra. Im going to get one I think. Perfect with the Silent way suite.
Old 15th December 2013
  #16
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intuitionnyc's Avatar
Quote:
Originally Posted by dogma View Post
Thank you and well written. Ive basically tried everything and I basically need to get either expert sleepers, innerlock, or I just found this one - its a hardware sequencer, looper and more and it claims 2ms timing with ableton live - 250eur. FYRD Instruments - Has maps for shruthi, volca, mopho, tetra. Im going to get one I think. Perfect with the Silent way suite.
The Expert Sleepers Silent Way stuff is VERY cool. Although, I do not have first hand knowledge with it. I have a modular rig and I use MOTU Volta to "interact" with my modular. I am not sure about Innerlock. I have to research that.

But, I am not sure if Expert Sleepers is going to eliminate latency. What exactly are you purchasing, Expert Sleepers hardware or Expert Sleepers Silent Way? Or BOTH? I am considering both in the future.

It seems that Expert Sleepers Hardware is just going to communicate with your audio interface. Think of it as another way to get audio into your computer. SO, there is still going to be latency similar to what you are already experiencing. Even using a hardware sequencer is still going to have latency if you are going to route it into your computer.

PLEASE tell me if I'm missing something here. I, just like everyone on here, am still learning all the time. I am by no means an expert. If there is a way to reduce latency further using the interlock, please let me know.

BTW, it keeps auto-correcting interlock.
Old 15th December 2013
  #17
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Re expert sleepers - it's simple yet complicated if that makes sense. Any expert sleepers hardware (in your case you have a dc coupled interface so you don't need any additional harware, just some cables) so the hardware is just to access the silent way software which is really a creative suite but to use it it must be sample accurate so the byproduct of this is sample accurate modulation, timing even on midi (that requires a seperate but of harware but exactly the same principles). For anyone with a modular or hePs of cv it's amazing - have a look at the sonic state review but remember that the harware has improved (made simpler) and the software is at v 2.5. There's things like lfo step sequencer (it's kind of like max for cv and midi)
The fryde looks super cool though as an add on to silent way as it's more midi orientated and performance based - except for how it handles drum sequencing (algorithmic ally based)
Innerclock just does midi and it's expensive as hell but a complete midi solution for timing. It does nothing else though. It's like $600 . Silent way hardware and software (the software is $60 or there abouts) and say an es3 and midi extender is about $400
Old 15th December 2013
  #18
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enossified's Avatar
It's not rocket science...it takes a small amount of time for MIDI data to reach the hardware and a small amount of time for the hardware to trigger. The actual audio latency is already compensated so it's not an issue.

Suck it up....if 100% timing precision is what you want, stay ITB and you'll have it

PS in case you didn't know this, before digital recording allowed us to see it noone gave two s**ts about small timing deviations
Old 15th December 2013
  #19
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Quote:
Originally Posted by enossified View Post
It's not rocket science...it takes a small amount of time for MIDI data to reach the hardware and a small amount of time for the hardware to trigger. The actual audio latency is already compensated so it's not an issue.

Suck it up....if 100% timing precision is what you want, stay ITB and you'll have it

PS in case you didn't know this, before digital recording allowed us to see it noone gave two s**ts about small timing deviations
Yeah that's one of the weaker arguments I've heard against what everyone wants - correct timing. Not a matter f sucking it up and no the audio is not compensated for in all daws either at all or in varying degrees of success
As a drummer I can tell the slightest deviations in timing and if you use external gear syncing it is just reality in 2013 my friend - wish it wasn't but it is. So you can accept it or do something about it your choice - I choose to do something about it. Plus notice how you never see these things second hand? Wonder why that is?
Old 15th December 2013
  #20
Gear Head
 

Hey everyone,

I don't intend to troll this topic in any way (as the creator of the MTRX-8) but here are some of my thoughts on timing with DAW that can help you clarify what you want/need.

- latency is simply the amount of time between an action and its consequence. When you hit a drum pad for instance, you ideally want the sample to be triggered instantly (i.e. with no latency). As enossified said, there is always a more or less small amount of time needed for the drum pad to capture the hit, for the chip inside the drum pad to convert it into a MIDI message, for the MIDI message to be sent by the device and received by the DAW, and for the DAW to actually trigger the sample. You need to remember one thing though: the latency is not always a fixed value. In the case of the MTRX-8, you can experience a "high" latency on the beggining/start of the sequencer, but after that start, the latency (when you change a parameter) is almost zero. In this situation, tweaking the sync delay parameter in Live almost solve the problem of latency. So my point is: latency is bad but some latency can be fixed, as latency is a constant deviation from ideal (due to hardware/software construction).

- jitter is something different from latency. Jitter is a variable deviation from ideal. There can be many things causing jittering (like noisy environments...) but one cause of jittering that is really annoying when you record MIDI data is the loss of MIDI clock messages. As the MIDI clock system is not absolute (it doesn't send the actual time of the system for instance) but rather relative (it sends a signal which frequency refers to the BPM) a loss of just 1 MIDI clock message will "shift" all the future MIDI data 1/100th note off-beat. The problem is this kind of jitter is additive: you loss 2 more MIDI clock messages and the shift will then be 1/32th note (which is pretty huge).

For the MTRX-8, I decided to create a new kind of MIDI sync system that adds a bit of "absoluteness" in the system: each clock message (within 2 bars) are different. This way, if the MTRX-8 lose a MIDI sync message (or your DAW failed to send it), they will automatically synced again when the next message will be received (as the "type" of the message will inform the MTRX-8 on where the DAW is in a window of 2 bars).

What's great in this kind of system is that the jitter still exists but loss its "additive" nature and thus jitter never turns into drift.

Hope it helps!
Old 15th December 2013
  #21
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Thanks very much for the explanation. Now the sync in live - is the best method for using multiple synths simaltaniously using the external instrument or midi to audio or.... And how many pieces can be used with one unit and still maintain sync?
Old 16th December 2013
  #22
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enossified's Avatar
Quote:
Originally Posted by dogma View Post
Yeah that's one of the weaker arguments I've heard against what everyone wants - correct timing.As a drummer I can tell the slightest deviations in timing and if you use external gear syncing it is just reality in 2013 my friend - wish it wasn't but it is.
I'm not a drummer but I can hear correct timing, too. All of the complaints I read about timing have people looking at waveforms and making measurements, etc. That's not "correct" timing but perfect timing. That's something that music played by human musicians never achieves, why is it that with electronic music people get so obsessed with it.

MIDI outside of a computer cannot achieve perfect timing, it's physically impossible. MIDI is a serial protocol running at a low data rate, it takes about 250 microseconds to send a single note message, so two notes programmed in a sequencer to start at exactly the same time can never actually start closer than 250 microseconds from each other. It will never be perfect.

Audio latency with DAWs can be compensated for if you know how to do it....all you need is to play a recording of an impulse, loop it out of the computer through your interface and back in, measure the delay between the two impulses and tell the DAW what it is.

DAWs let you drag audio and MIDI back and forth in time to line it up, sure it's tedious but if you need everything perfect there you have it...perfection.
Old 16th December 2013
  #23
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crufty's Avatar
I have no problems adjusting two kicks

What doesn't sync ?

Fun fact--Human brain will automatically error correct up to 80ms of lag
Old 16th December 2013
  #24
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Quote:
Originally Posted by enossified View Post
I'm not a drummer but I can hear correct timing, too. All of the complaints I read about timing have people looking at waveforms and making measurements, etc. That's not "correct" timing but perfect timing. That's something that music played by human musicians never achieves, why is it that with electronic music people get so obsessed with it.

MIDI outside of a computer cannot achieve perfect timing, it's physically impossible. MIDI is a serial protocol running at a low data rate, it takes about 250 microseconds to send a single note message, so two notes programmed in a sequencer to start at exactly the same time can never actually start closer than 250 microseconds from each other. It will never be perfect.

Audio latency with DAWs can be compensated for if you know how to do it....all you need is to play a recording of an impulse, loop it out of the computer through your interface and back in, measure the delay between the two impulses and tell the DAW what it is.

DAWs let you drag audio and MIDI back and forth in time to line it up, sure it's tedious but if you need everything perfect there you have it...perfection.
Thanks but I realize all that and Im not looking for perfect timing Im talking about something livable. I dont measure anything but for instance yesterday I simply pulled up a external instrument, set everything correctly - drew some midi notes and played my shruthi. I also just had a couple of instances of geist playing drums and kontakt a system 100 line and you could literally hear it drifting out of sync. Im just reiterating my real world experience. Certainly not any controlled tests or anything like that at all
Old 16th December 2013
  #25
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If with that knowledge you can tell me how to get it playing in sync id be eternally grateful. And so would a lot of Ableton users. This is not the case in Logic or Reaper.
Old 16th December 2013
  #26
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crufty's Avatar
What os and midi interface ?
Old 16th December 2013
  #27
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OSX 10.6.8 and ML 10.8.4 - Tried various midi interfaces and it follows the same pattern of behavior with each interface. Daw - LIve9 Logic X - fine in logic.
Old 16th December 2013
  #28
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Even with expert sleeper / cv driven stuff, there's still an audio latency. I usually have to do about 30-50ms of delay compensation to get things lined up.
Old 16th December 2013
  #29
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owensands's Avatar
I record hardware all the time with ableton. Here are some tips. Its a pain the ass.

1) Get VERY good sound card with AMAZING drivers. RME or something with ASIO drivers that are just as good. End of discussion there.
2) Get a beast of a computer and overclock the CPU(this will help get super low latency results with your interface) I use a 3rd gen I5 OC'd to 4.7 and its amazing.
3) Dial the latency down to below 128 buffer size on your SC. I use 32 and 64, mostly 64.
4) use an external instrument plugin with a pre-programmed midi clip on it and record to an audio track that is getting its audio from NOT the inputs of where the source is coming in from your SC but the channel that has the external instrument plugin on it. Arm this audio track and leave the other unarmed. record in 8, 16 32 bar clips
5) this will yield almost dead on to the grid even not using any delay compensation etc. If it isn't(it should be only very slightly off and when I say slightly I mean OCD slightly) then create a warp point somewhere towards the middle of the audio clip. Delete the warp marker at the beginning of the audio clip. Go back to the one you created in the middle and hold down shift on PC and click drag on the warp marker. This will slide the audio clip whatever direction you need to go to snap to grid or your groove and it will sound great.

I use this method with the following midi synths with perfect results.
Juno 106
waldorf pulse
boomstar 4075
DSI mophox4
MI Ambika

I use Expert sleepers for Modular(CV via ADAT) and record to an audio track that is getting audio from the channels on my interface. This sometimes needs a sight adjustment just like above because I use ableton to monitor. If you use direct monitoring from your SC it should be almost dead on. And buffer size will affect this greatly when monitoring from ableton so again lower the buffer size the better.
Old 16th December 2013
  #30
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Mate that was excellent. I will try some of the methods you describe and see how I go. Number 4 though - I use a Metric Halo 2882 and it uses firewire returns which is how its drivers work - so there are 18 available "channels" but I have only 8 physical ins and 8 outs - so you can choose 1-18 mono or in stereo. So in the MIO mixer I allocate a mix strip an i/o plug which routes the audio to the daw (theres heaps more to it - yu can pull up an AU and take a system snapshot so all settings are saved on the actual MIO itself - sorry Im unneccessarily complicating it but point is the method you describe is how it is or can be set up.
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