Thanks, that helps. You rock!
Tony
There are a couple of good videos here to get the latency down as low as possible in Reaper.
Either one works. 14/14 is not to shabby I was running with mid 30s until I change my AI recently. And as Lbro described there was a noticeable delay but I followed the first video and got it down to 7/7 or there abouts with a sample rate at 128. I might go to 64 or lower but I was getting some crackling on vox and thought that may be buffer related. Bumped it back to 128 and it was clean again. It was just before the Open Mic so did not want to mess any further.
On the Ai front have to agree with LB if you are going 2 mics 1 Gtr you need at least three separate inputs, which most like means having to go for 4 in 4 out, off course when you come to upgrade.
Cheers
Toby
Thanks Tony, glad you enjoyed it.
Visuals are all clips downloaded from Pixabay.com and transition effects added in Corel Ultimate Studio.
âPro audioâ typically targets latency of 5ms or less. (Note that you can never completely eliminate latency, but bringing it down to those levels makes it unnoticeable to most people.) With that said, you can still get by with higher latency, it just makes some things more difficult. If you get into double digit latency values youâre probably in the âobvious and annoyingâ range.
The problem is that to achieve lower levels of latency you need to balance the sample rate and buffer size (among other factors) with the capabilities of your hardware. For a given sample rate, the smaller your buffer size, the less latency youâll get. However, the smaller your buffer size the more demand you put on the system hardware (and audio drivers, for that matter). Some hardware may not be capable of achieving âpro audioâ levels of latency without xruns (which create unacceptable pops, crackles, etc.).
Your screenshot shows 14ms of latency, which is higher than Iâd want. You can try decreasing the buffer size (i.e., number of samples). Right now youâre at a buffer size of 512 samples. Try lowering it to 256, then to 128, and so on. At each stage, test it out and see if your hardware is âkeeping upâ at the lower latency rate, or if it is starting to fail (xruns like audio stretching, pops/cracks, et cetera). This would be a good âfirst stepâ to finding the limits of your hardware/software and the lowest latency setting that you can reasonably achieve. Keep in mind that the load on your hardware can vary depending on your project, too. If you have many tracks, or many plugins/effects running in the DAW it will increase the load, which can also affect what buffer size, etc. you want to choose.
Bottom line: youâll need to experiment a bit to see what configuration and buffer size works best for your circumstances.
Thereâs some decent information in Focusriteâs article on latency and interfaces. Thatâs kind of the tip of the iceberg. The Ardour Manual also has an introduction to latency and a slightly more in-depth discussion of the subject.
For what itâs worth, hereâs what Iâm currently using/getting:
The image doesnât show it, but the configuration is using 3 periods per buffer (which seemed to work better for me than 2 periods per buffer â I think I read something about 3 being a good thing to try for USB-connected interfaces).
Thatâs on an older Intel i5-2550K system running Linux/Jack and using a Focusrite Scarlett gen 2 interface. (I recently bought a new computer, but I havenât had the chance to migrate everything over to it, yet.)
Thanks all for the detail on latency, plenty of education for me there, have a super day
Been reading a bit and watching videos about latency. Checking if this simple explanation is accurate within itâs narrow scopeâŠ
When recording vocals, to a pre recorded guitar track, excessive latency could cause the vocals to appear later than you expect?
Yes, and it can be very distracting for the performer if theyâre monitoring and hearing a âdelayâ as theyâre singing and playing even as theyâre singing and playing it. (Many interface have a âdirect monitoringâ feature which can avoid the issue, although you wonât hear any computer-side effects youâve added in the DAW.)
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âPeriods/bufferâ is the number of âchunksâ the buffer is split into. The default for most audio software seems to be 2 periods per buffer, which is âdouble bufferingâ: two buffers combined into a single block of memory.
On my system 3 periods works better than 2 when using a USB interface. However, looking into it a bit, this may be something that is specific to Linux. If youâre on Windows or Mac (and not using JACK) you can probably just ignore it and use the default. (In fact, if youâre using ASIO or Core Audio Iâm not sure you can change it.)
In short: I think itâs Linux-specific, and you donât need to worry about it. I didnât think about it being a Linux thing when I wrote that post. I probably shouldnât have mentioned it. (Donât want to add complication when itâs not necessary.)
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Thanks for clarifying, Jason. Iâm sure there are other Linux users in the Community, so worth a mention with the Linux context clarified.
Certainly no similar option for my UMC1820 in Windows. But at least I have around 5/6ms total in Reaper at 64 samples. Need to check this again on vox, as the crackles I was getting was before I reset the Latency Offset. If they are still there Iâll go back to 128.
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Iâve currently got Garage Band and Reaper installed on my mac. Itâs been a great exercise reading about latency because it gives me something to work towards (lower latency). Iâve done a small amount of recording to Garage Band and will be in a new locale next week with time hopefully to dedicate to a bit more recording.
Oh but thereâs a great open mic to rehearse for at the new locale. Isnât it wonderful with music that we have so many creative outlets.
Another option Iâll look at after comparing Garage Band and Reaper is evaluating Appleâs Logic Pro. They now offer a 90 day trial so plenty of time to compare and then decide on which I want to stick with. Fun fun fun.
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