Re: audio and BIAB
D F Tweedie
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Hi Jim ...
You don't mention whether you have an interface or are using the computer's onboard audio devices. This can make a big difference for recording latency, i.e., the delay between playing or singing into your mic and then hearing the audio back after it is processed through any program.
One big advantage of an interface in addition to dedicated drivers permitting lower buffer sizes is that many have a feature that lets you hear your audio back after your preamp section but before it is processed by the computer/ program, thereby reducing latency to insignificant amounts.
Coming back to your question, ASIO is almost always the preferred device with any program that can utilize it.
The buffer size is tied to the audio driver. When you change buffer size "in a program," you are not really changing it in the program. More precisely, you are changing it 'through the program' accessing the driver's software applet. So yes, if it is changed it is changed for all applications until the next time you change it regardless of which programs you load in the meantime.
On the other hand changing a device's buffer size, as opposed to changing devices, is trivial. Even professional studios do it regularly when the alternate between recording (lowest buffer size available) and mixing (buffer size as large as you need to lower CPU demand on hungry plugins). The glitching sounds you hear are almost always the result of the buffer emptying faster than the CPU can process the audio.
And yes, the best practice is to change your buffer size depending upon what you are doing.
You probably notice the lag in BIAB when 'rendering' new RealTracks. However, once rendered, playback is relatively seamless and should proceed at the lowest buffer setting you can manage for recording purposes.
If BIAB isn't up to the task you can always export the RealTrack audio and import it into Logic and record in that program. For most purposes I can think of that would be superior to recording in BIAB.
In general it is plugins and especially virtual instruments (VSTIs) that place large demand on CPU and require large buffer size. Playing back pre-recorded audio and recording one input at a time should work fine under the lowest buffer setting your audio device permits.
From: "Jim MIngs jmings2003@... [Band-in-a-Box]"
Sent: Thursday, May 24, 2018 8:50 AM
Subject: [Band-in-a-Box] audio and BIAB
I have 3 options in audio settings: ASIO, MME, and WAS. I seem to experience chatter in each to varying degrees when recording audio into and out of BIAB. Also, audio is out of sync a good percentage of the time unless I freeze all tracks and that still can be out of sync. I am also experimenting with buffer size. A smaller size for recording and bigger for playback. This is a hassle. Is there a good compromise? When I change buffer sizes in BIAB does it affect buffer size in other programs? Is there a magic number? I realize that there are many better ways to record audio than BIAB, but it would sure be nice to get decent recordings while one is writing without having to drag files from program to program. Decent recordings happen sometimes. ;-) The stars have to be aligned... Any help would be greatly appreciated.