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Best Practices For Sampling in BM3 (and in general)
Since getting BM3 I've been (re)sampling a lot more. Mostly taking my fav drumkits from other apps and resampling them into BM3. Each time I think I did it right I end up finding something online that makes me want to throw everything out and start over resampling again (which I've done... 3 times). The latest was finally finding the "normalise" button in the editor screen after completing 12 meticulously chopped, ordered and named 16pad kits (internal screams... and f#ck it I'm not redoing them anymore).
Anyway, what're most important best practices for a hobbyist going forward?
24-16bit / 44.1khz / -3 to -6db?
Comments
For starters, use the browser instead of the database. Save your stuff in folders, copy your samples when you save banks. Whenever working on building banks use the zip backup option when you finish, because it is such a time consuming process...
24 bit sampling does sound a lot better than 16 bit so go with that. Definitely normalise your samples.
With resampling you can make one long sample of spaced hits, percussive or melodic. If you space if out over a bar to allow the tail to fade out completely you can use the split slice and get a near perfect slice almost every time. That will get even easier when transient detection gets released.
Honestly probably better to just resample what you want to use for projects you're working on. For example I'm just enjoying Thor at the moment. When you start getting into things like looping and setting polyphony and envelopes and modulation and added effects and playing with chords the end result is so much unexpectedly better than the original.
Take your time and don't delete your works.
I'm still not sure which I prefer:
IMO it doesn't matter. Others may passionately argue that point.
Agree on this as things tend to add up and overload the master. BM3 now has the ability to set the default cut, which before was fixed at -6db.
I also tend to avoid the "threshold" setting in BM3's sampler. I would use that if I was sampling a live input such as a guitar, but for sampling an app, I set the threshold to it's minimum. I feel like somehow I might miss some of the initial attack curve of the sound. I'd rather get it all then trim the sample than miss anything.
I put in a feature request for a "trim side silence" button in BM3, which would trim all silence from the beginning and end of a sample. That's a feature I really like about the Edison editor from FL Studio on the desktop.
But, the amazingly handy buttons in the BM3 editor that take you to the beginning or end of a sample and zoom in all the way do make this chore fairly easy.
Definitely.
BM3 forced me to get/keep my sh#t in order so after zipping I send em straight to the cloud. Perfect.
I'm not a fan of normalising when sampling instruments personally. If I want to keep my dynamics similar to how the instrument is in reality, then my feeling is that automatic normalising will delete anything that I 'play in'. Usually I record into Auria Pro, as I can normalise everything together in proportion there.
Normalising is a quick way to assure a certain level isn't exceeded, but it doesn't tell anything about the perceived loudness.
The latter is basically represented by the dense section of the waveform, not the peaks.
The higher the dense part (extreme: sausage shape without peaks) the louder it feels.
But at an expense: it usually sounds like shit, so you really want to keep some peaks to have the sound still 'breathing'.
The dataformat depends on what you record, how many tracks the mix will contain and how much storage space you have available.
If you sample synth output or prerecorded drums, 16bit at -3dB is ok (some safety)
At 24bits you can leave 6-10 dB headroom.
After all it doesn't matter much for the DAW will convert everything into the format of it's mix engine anyway, which is always 24bit and above.
With microphone recordings of acoustic sources like guitar and vocals 24bit is the preferred format to have clean tails in the sound.
(in 16 bit silent parts may start to get grainy)
With an electric guitar/bass cabinet the source is too loud and noisy for 24bit tracking.
There's just no advantage of the higher resolution, it can even make a track worse with so much 'noise bits' into there.
It's common practice, though - but still remains pointless.
Will it? All I've heard so far is an app that acts as an interface for files you have stored in iCloud, and can like open them in other apps. Still sandboxed. And useless if you don't want to store your samples in icloud.
Yepp, and direct access to iOS FileProviders such as AudioShare
(BM3 will most definitely support this, so managing BM3 content with Files.app will be a breeze).
And I have a good feeling Gadget & Cubasis will support this as well...
Yeah, but I have nothing in itunes, and avoid it like the plague. I'm obviously the odd case. Hope you can find it some use, I'll just wait and see, but so far I haven't seen any personal benefit from audioshare.
Sometimes it's good to go with mono samples as well, for example chopping some loops will sound more seamless when mono than stereo. Also, BM3 developers ought to get on this bit but when slicing you can adjust the fade out. I always set it up to 2 seconds, for whatever reasons it's the magic number to prevent any clipping. You can also drop the sustain a bit, which will lower the volume after the initial attack. My default polyphony setting is 2, but I'll often bring it up to 4 or 10 depending on what kind of sound it is.
Eventually I'll put up some videos showing my workflow and what I'm getting out of it.
Let me help clear the confusion.
What we mean here is the 'Documents' folder that every app that generates files saves to by default.
That 'Documents' folder is often called 'iTunes Shared Documents' or simply 'iTunes'.
It's got nothing do with 'iTunes on the Computer' other than that iTunes is just one way to access the files.
The way AudioShare shares it's documents for now is thru iOS FIle Provider extension.
You can not access those files directly from the computer when you hook it up with a USB cable which some user use for backing up the content in AudioShare.(You can access them using the Wifi-Drive feature though).
So Files.app in iOS11 will be a huge productivity boost for all areas of usage
@Samu thanks for the explanation, I was uninformed
Ah yeah, I always sample in mono when building instruments. Maybe for loop slicing I could use stereo files, but for 'notes', mono.
Following!
I also try to remember to remove reverb and delay FX when sampling. I prefer to add them in through the DAW for better control. That is unless the app I'm sampling has a delay or reverb of a specific character I can't get another way or can't be bothered to try to recreate.
@Max23 not sure about the pr0n as you say but Sector is made by Kymatica who has also given us AudioShare. I love being able to feed it MP4 video and trim bits without having to convert the whole thing to wav.
Is there a best practice for hard-coding sample loop points? I'll often use the same samples in different apps, and having to find zero-crossing, non-clicky loop points each and every time makes me think I must have missed a trick.