Hello i open this topic to discuss the ways of syncing MD in the ableton.
I find out that the latency produced when i was monitoring from the ableton
then i turn the ableton’s monitoring of and i start monitoring from my audio interface (motu gives monitoring zero latency). As a result now i can use the daw only as a recorder.
Any better way? To integrade the machinedrum to ableton with zero latency of the sequencer and without loosing the monitoring from inside the daw and as a result loose all the plug-in possibilities.
When you want to monitor the plugins inside ableton, you will incur latency.
There’s really not a way around that…the ‘zero latency monitoring’ where you monitor the direct input signal is just one way around it.
If you had a fancy card with built in DSP (like a unversal audio), sometimes they provide fx that you can use before your daw, and so you can monitor those with little to no latency, but not really what yer asking.
Are you talking about an audible delay between what you hear and when you mute stuff on the machinedrum?
I’m using an Ultralite mk3 via USB and I have it stable at right around 4.5 ms latency overall. Go into the ableton settings and lower the sample buffer in the hardware settings. More taxing on your computer, but should improve audible latency considerably.
There is also the delay compensation settings, but I don’t mess with that
afaik…there will never be zero latency in ableton.
how i do it:
i have high latency on purpose, when i press play on MD the 10ms late recordings will always be in sync no matter how much more latency causing things will occur in ableton …
in other words: everything i do in ableton that is not comming from hardware i.e. singing will be recorded with an initial 10 ms delay.
i wont hear the delay afterwards since everything is delayed and nsyc (yeah baaaby)
Play with the Midi delay settings and the audio offset settings in either the track or the external instrument plug in (depending how you’re bringing the audio in).
I have a click track on both the MD and Live (Metronome) and twiddle until they sync up. From then, the MD will play in time with the audio from Live.
When it comes to rendering, you will likely have to manually tweak the latency offset back by nudging the 1st Warp Marker a touch.
Works well (once you get your head around it)
For me, I have the following set up…
Prefs:
Midi Sync > Midi Ports > Output > Sync Delay = -27ms
Audio > Driver compensation = 0ms (I’ve heard this makes very little difference and just adds latency elsewhere)
On the tracks
I have an External Instrument and it is set to 0ms Latency.
It will be the same way.
If you process with software, there will be latency.
You can get it low, but you can’t eliminate it - that’s how modern audio processing works.
My read on the OP was that the MD was not in sync with Ableton - the delay in processing the audio from the MD didn’t sit in time with Ableton’s sounds.
Eg: two kick drums playing at the same time would be out of sync.
If this is the case, then use latency compensation as per the posts above.
In Ableton the track delay under every track in the session view is the monitoring delay if you adjust it you and make it fits you, you will notice that you here the monitoring as you adjust but the recording stays the same.
So this is not what we are looking for.
In the Preferences > Audio at the Latency section the Driver Error Compensation is the latency that comes from native plugins like the ableton compressor , or the gate, or the limiter e.t.c. You can test it. If you open 10compressors in a channel you will start hearing slaty the latency if you open 20 compressors the latency will be more obvious to your ears.
So the Driver Error Compensation is fixing this.
The actual way to adjust the monitoring/recording latency is in the
Preferences > Midi Sync. You go to the output port that feeds your machinedrum or monomachine click on the arrow to open the list
And you adjust the MIDI Clock Sync Delay.
For my setup the Machinedrum needs -30.5ms and for Monomachine -29.5ms. to be syncing in ableton but not to be synced together there will be always a little drifting in the sync.
I find the Machinedrum easier to sync than the Monomachine with this way.
Now this works perfectly in a setup with only one Drummachine and a couple of Synthesizers. If you want to sync 5 or 6 clocks together it will be harder.
Monomachine and Machinedrum with that way aren’t completely synced together like the way a motu or rme audio interface gives the zero onboard latency.
But! Only the Monomachine or the Machinedrum can be monitoring sync to the Ableton very easy.
I also notice that synthesizers that has internal sequencers like the DSI Evolver or Tetra. Needs others values to be put in the Midi Clock Sync to delay to be correct sync when are playing via midi notes from the pianoroll of the daw or a midi keyboard or when they running their internal sequencer
Perfectly understand what you say for a hobby so as other people said don’t SYNC Record like a multi pistes and then RE ADJUST to the Grid, you only have the beginning to Adjust if Tempo is the same as you DAW no Problem .
OR JAM and the MODULATION is the important thing, OR SOUND DESIGN is the important thing so in this case just SAMPLE and do ALL in your DAW
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Nope that’s computer the problem as far as all people told me.
SYNC Elektron between each other is nice (maybe Turbo midi help a bit in this territory for sync between Elktron machines)
Introduce a computer, a soundcard, VST and you will encounter some issue.
That’s why people ON STAGE use PRO MIDI Sync solution. Ok that’s not cheap but it’s really worth the price. So Roland release Roland SBX-1 Sync Box… But the TWO one i TRUST is [innerclocksystems SYNCGEN 2 PRO and [url=“http://www.s-n-d.com/acme4e_a.html”]s-n-d ACME4](http://www.innerclocksystems.com/new ics sync-gen ii pro.html)
Sync GEN 2 Pro is the most affordable. Computer is not the master, Clock is Master and all is Slaved to an Audio Pulse (very short noise to get a tight sync)
Ableton slaved to external MIDI clock is usually much worse than the other way around.
The problem the OP describes is sound card driver latency.
The buffer setting (64 / 128 / 256 / 512) determines the time required for audio from the sound card inputs to reach the host.
Lower settings are faster but require more CPU.
Some sound card drivers are better than others, it’s one of the reasons people love RME sound cards - the drivers are the best performing for low latency recording.
