When I change from one song to another, the latency change and I have to click on TOOLS - OPTIONS - AUDIO ENGINE and click on OK the get the good latency back. (I use the buffert size 129). The buffert size remains the same, but I can clearly feel that the latency is getting worse when I go from one song to another.
I need to add that I am not a very knowledgeable user.
Karl
Hi,
this sounds like the kind of thing we’ve been experiencing with certain songs.
Brad has been helping and there are workarounds:
When I use an external audio interface (scarlet 6i6 2nd) I don’t have this problem, just when I connect the piano directly to the computer with an usb cable. So let’s use the external audio interface…
One thing that’s annoying is your buffer size of 129. Are you sure it is not 128? Maybe this could be a reason for the problem.
Regards, humphrey
Hi Humphrey,
I can set a buffert size in Cantabile and then I can set a buffert size on the Focusrite ASIO control panel. Is it important that they are the same? How is it working, ho has priority!?!
Today I tried with buffert size 88 on both and I think the latency remained the same even when I changed songs.
Karl
Hi Karl,
I’m not an expert concering latency settings. My feeling is: it would be good to have settings matched if there are 2 places to adjust them.
The other aspect is that most soundcards I know have organizesd their buffersizes in multiples of 16. So typical values to chose are 32 (sometimes 48), 64, 128,… I don’t think this is mandatory but also have no clue what happens if other settings are used.
Only my 2 cent, regards, humphrey
If you would use Brad’s book “Glitch Free”, it says, “Latency can be calculated by dividing the buffer size by the sample rate and using some
simple algebra we can work out the required buffer size:
Latency = BufferSize / SampleRate
BufferSize = Latency * SampleRate
BufferSize = 0.01 * 44100 = 441
At a sample rate of 44.1KHz, 10ms is 441 samples. Since some sound cards only support
buffer sizes that are powers of 2 this is often rounded up to 512 samples (about 12ms) or
down to 256 samples (6ms) – depending on what your computer is capable of.”
Your Focusrite is going to handle any Latency just fine. According to what soundcard is in your computer, and the computer actual processing abilities, you may have to deal with some latency. I can run off my meager laptop sound card without issues, but running through my Focusrite is very reliable, because it is like a “super charged” sound card that can process so many things with a very low latency near 0. More information as what you are running, and trying to achieve would be helpful to anyone analyzing your problem. Screen shots would also help tremendously.
Regards
Corky
EDIT:
More from “Glitch Free”:
“I recommend starting with your audio software configuration – a sample
rate 44,100 KHz and a buffer size of 256 samples.
If you’re not processing input audio (i.e. all your sounds are coming from within your audio
software) set the buffer size of 256. If you are processing input audio the total latency will be
affected by the length of the input buffer and the output buffer. Setting the buffer size to
128 samples gives a total end-to-end latency of 256 samples.
You certainly don’t want to go below 44,100 KHz or audio quality will noticeable degrade
and in general you don’t need to go higher than this.
Remember the higher you go with the sample rate the more data needs to be processed and
the less time it has to do it. Say for example you doubled the sample rate to 88,100 KHz –
the number of samples per second has doubled and if you leave the buffer size at 256 samples
the latency has dropped to about 3 milliseconds. If you do increase the sample rate, make
sure you also increase the buffer size to compensate.
So when would you increase the sample rate? The main reason for doing this if you have a
long chain of plugins or effects that result in the audio signal being processed many times.
Increasing the sample rate can reduce the chances of audio artefacts creeping in.
The reason I recommend 256 samples is that it’s well below the 10ms human hearing limit and
provides enough head room for any additional latency that the sound card might introduce -
there will be some. Also, it’s a big enough buffer that it can be processed efficiently.
For nearly all applications these default settings will work just fine - and more importantly
they’ll work reliably. There are however exceptions:
• If you’re running more audio effects and/or plugins than your computer can handle
you may need to increase the buffer size to compensate.
• If you’re routing audio between multiple programs or devices where each introduces its
own latency you might need to optimize the latency down as far as possible so that the
total latency of the entire chain is acceptable. In general keep the buffer as big as you
can while maintaining acceptable latency”