UMC 1820 audio in latency

So, I write techno that should be tight af.
Got a live last year and bought myself a behri umc1820 and ada8200 to give me in the box mixing.
Love the quality and the ability to add compression or whatever you want on my hw synths. But timing was skew. Researched a fair bit and tried adjusting UMC buffer in cubase but MPC didn't use it. Have eventually run MPC sounds via program into submix then added a simple delay of 18ms. All lined up.

BUT does anyone know how to delay the metronome? The 18ms delay is fine but metronome is now confusing. Recently joined the forum read more posts than I can remember and started on mpc1000 years ago.

Cheers

Sapro wrote:So, I write techno that should be tight af.
Got a live last year and bought myself a behri umc1820 and ada8200 to give me in the box mixing.
Love the quality and the ability to add compression or whatever you want on my hw synths. But timing was skew. Researched a fair bit and tried adjusting UMC buffer in cubase but MPC didn't use it. Have eventually run MPC sounds via program into submix then added a simple delay of 18ms. All lined up.

BUT does anyone know how to delay the metronome? The 18ms delay is fine but metronome is now confusing. Recently joined the forum read more posts than I can remember and started on mpc1000 years ago.

Cheers

It's impossible to delay the metronome, just create a custom click track with a simple delay you will unmute when you need it. End of Story.

The umc1820 should do 0.5ms latency or lower are you using asio4all?

blocksize 64 to 128, periods 2 or 3, rate and depth can be whatever. if you can set that up you'll acheive a lower latency.

I think he's got the problem that he's using the 1820 as his audio device on the MPC, but cannot modify the buffer size/sample rate/etc... to reduce the latency on the MPC through the 1820. At least that's how I read OP's post.

AFAIK, MPC and Force use fixed buffer size of 128 samples in standalone, and this is non-negotiable. Try testing with an in-out passthrough via the 1820 and MPC. If total latency is significantly less than the 18 ms, then the 18 ms isn't caused by the MPC's audio buffer length but something else.

NearTao wrote:I think he's got the problem that he's using the 1820 as his audio device on the MPC, but cannot modify the buffer size/sample rate/etc... to reduce the latency on the MPC through the 1820. At least that's how I read OP's post.

That's the one, although later when I am in my studio I am going to try the Behringer application to change buffer size and see if that sticks.

Also thankyou to replier who mentioned making my own click track. That was my backup plan

Ignore me im tired..

Is there evidence that the audio interface has any say in what buffer size the MPC decides to use? The MPC is the "computer" here, and I've always thought that the computer doesn't ask the audio interface, it just uses some buffer size. It might ask the USER but not the interface. That's what I thought.

HouseWithoutMouse wrote:Is there evidence that the audio interface has any say in what buffer size the MPC decides to use? The MPC is the "computer" here, and I've always thought that the computer doesn't ask the audio interface, it just uses some buffer size. It might ask the USER but not the interface. That's what I thought.

Nope, the audio interface is just a dumb device that does what the host machine tells it to.

Those buffer settings only apply to how the DAW in Windows or MacOS handles the chunks of audio coming in and out. It won't do anything for the MPC, which would need its own method of setting its own internal buffer, which Akai decided we only need 128 samples..

Monotremata wrote:

HouseWithoutMouse wrote:Is there evidence that the audio interface has any say in what buffer size the MPC decides to use? The MPC is the "computer" here, and I've always thought that the computer doesn't ask the audio interface, it just uses some buffer size. It might ask the USER but not the interface. That's what I thought.

Nope, the audio interface is just a dumb device that does what the host machine tells it to.

Those buffer settings only apply to how the DAW in Windows or MacOS handles the chunks of audio coming in and out. It won't do anything for the MPC, which would need its own method of setting its own internal buffer, which Akai decided we only need 128 samples..

Is that on two inputs or the full 32? (Though why they did that is beyond me as it will only handle 8 stereo channels). I wonder if it is larger with more channels and yes changing in UMC control panel on pc made no difference.

Test the latency yourself. Route a monitoring-enabled audio input straight to an output in the MPC, and record a stereo signal where one side goes through the MPC (and audio interface) and the other side as a direct analog connection without entering the MPC system at all. Record this pair to another recording device such as a computer. The difference between the left and right sides is the total pass-through latency of the MPC+interface, including AD/DA conversions and everything.

Do the same test with and without 32 ins/outs mode. Or with the UMC 1820 vs MPC internal audio.

Long story short it is now about 3 seconds which fits with the above post. Dunno what has changed it has been disconnected for standalone but hey I'm happy. J can cope with that nice small delay to set in my send. Thanks all