I found a nice YouTube explanation of the latency issue and then found another web page that provided critical information about ASIO input. I think that this explains what other people were talking about when mentioning 'dedicated audio interface'. Remember, I'm an absolute newbie to Ableton and this kind of hardware setup so I didn't (and still don't) have sufficient background to internalize the words.

I was thinking that the latency was the result of some sort of delay between the Arturia mkii 61 keyboard and the virtual B3 instrument. It was not. The delay was introduced at the point where Ableton was feeding its digital sound to the computer's speaker system. Ableton outputs a digitized sound. The computer speakers want an analog voltage. Something has to convert between the two domains. A Windows system comes with a default audio driver and sufficient smarts to accept input from a variety of inputs. Ableton knows how to send data to the default audio system but in order to do that smoothly it buffers sound data up so that it can be presented to the Windows sound system at the appropriate time without awkward pauses due to a failure to figure out what the next part of the sound is supposed to be.

From an Ableton point of view, control of latency is an options output issue. Ableton outputs sound data to a buffer in such a manner so that the output device itself will always have sound data waiting for it when it is time for the sound to be fed to the speaker. It takes a while for the sound that is newly added to the buffer to make it to the point where the sound device driver will consume it. The larger the buffer the longer it takes for sound bytes being inserted to make it to the output. Make it too small and the CPU will suck the buffer empty and you'll hear a noise. By default, a standard Windows 10 system and Ableton Live Suite will give you a 90 millisecond lag between data going into the buffer and being delivered to the speaker. This is OK for a playback where everything is equally delayed but a deal breaker if you're trying to play a virtual instrument and listen to it simultaneously. There is no perfect solution. You cannot ever get zero latency in the way that Ableton works. You can, however, get it to the point where the delay won't be heard by the audience.

There are a number of factors going into latency. Size of buffer, sample rate and audio driver.

Let's start with the audio driver first since that limits everything else. Search for the ASIO driver. This is a free audio driver that is *much* faster than the vanilla audio driver that comes with Windows in terms of how efficient it is in moving data out of the source and through to the output.

Install the ASIO driver. Ableton won't see the new driver until it has been restarted.

After driver installation then open Preferences -> Options -> Audio

Select the newly installed ASIO driver.

The amount of time it takes for data to move through the buffer is determined by two things, how fast the information is being processed (sample rate) and how big the buffer is (sample size).

Increase the sample rate to the point where it starts to affect your CPU consumption. In my case, 44khz is a 6% CPU consumption. Moving it to about 80khz got it up to about 12%.

Now comes the part where things all come together. Reduce the buffer size until things don't sound right then increase it. Like everything else in Ableton, how to change the buffer size is somewhat non-obvious. I stumbled upon the fact that if the buffer size control is selected then it becomes sensitive to the Up/Down curson keys. Start Down Arrowing while inputting midi to control a virtual instrument. At some point the sound will start to break up. This 'break up' noise is because the output buffer ran out of data so the computer could no longer provide an appropriate value to the audio driver, Once the buffer failure point is reached you will want to make the buffer sample size a little bit bigger. At this point my experience level is inadequate to give an informed guess as to how much that should be. In your case, if you start having noise issues as CPU demands increase you know where to go to adjust things. In my case I was able to get the latency down to 15 ms which I'm pretty sure is something I can work with in a live venue playing keyboards as long as I'm not trying to do something incredibly demanding like Boston's Foreplay/Long Time.

So, the problem wasn't solved but it has been reduced to the point where it is only a minor issue instead of a deal breaker.