Using diff sample rates in same project... not a good idea?

[nms]

I know that supposedly this is not supposed to be a good idea.. but practically speaking.. is it really something to avoid doing? Normally any samples I use are 44.1 but right now in a project I'm working on I have a 48khz bass tone mapped out on drum rack and transposed to 6 higher notes. Because of the transpose I thought it best to keep the sample in its original higher quality 48khz form.

should I just leave it as is and don't worry about it? or should I bounce the finished bassline to 44.1k?
or.. should I just transpose the notes individually in sound forge and save them at 44.1khz? (*that seems like the smartest thing to do)

looking for a suggestion here from someone who knows the science behind this! thx

[leedsquietman]

you can't have different sample rates in the same project, as the speed/pitch will be affected and you won't get things in sync and some things will sound like alvin and the chipmunks version.

you need to get everything to one universal sample rate.

[chapelier fou]

I am absolutely not sure of that. Ableton must do a kind of conversion. But I am sure it's better to work everytime with the same sample rate and bit depth.

[IP]

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Like leedsquietman said all your sounds (and your project settings) should have the same sample rate.

I mainly do all my recordings and work @ 88200.
If i need to use some samples recorded @ 44100 - 48000 etc
i use Audiomove to convert them @ 88200 and after that i drop them into live
(else live will use a kind of onthefly conversion which is not the best think to work with. except the quality it uses more resources too ... correct me if im wrong )

Generally be careful ...
dont skip converting your samples to match your project's sample rate
and also dont use ANY software for that ..

[SubFunk]

I am absolutely not sure of that. Ableton must do a kind of conversion. But I am sure it's better to work everytime with the same sample rate and bit depth.

after my knowledge this is what happens in some DAWs and hardware... a sort of realtime sample conversion if you use different sample rates in one project.

not a good idea at all... it is just a possible cause for trouble... we had that once with hardware running a yamaha DM2000 a d&b D12 amp and a Tascam CDRW all wired via ASE/EBU only the D12 only runs on 48khz and the Tascam CDRW (i can't recall the exact model number) only on 44.1khz only. the yamaha had a hard time dealing with that, in fact it was in this case only possible with a workaround not running over the AS/EBU slot card, but over the built in... because the desk needs to realtime convert all the time... i would leave my hands of it. it's a headache!

[nms]

dont skip converting your samples to match your project's sample rate
and also dont use ANY software for that ..

if I don't use any software to convert the samples (transpose & sample rate) they will remain as they are now.. what do you mean by that? I've been using soundforge for a decade and trust that with audio editing.. should I just use that and transpose the notes in it then convert & save the files at 44.1?

[Tone Deft]

you can't have different sample rates in the same project, as the speed/pitch will be affected and you won't get things in sync and some things will sound like alvin and the chipmunks version.

you need to get everything to one universal sample rate.

absolutely not true. I change sample rates in the same project all the time. try it.

make sure the Hi Quality SRC mode box is checked in the preferences and you're fine. ethios4 did a test once and it was shown that without Hi Quality mode on there are some artifacts, with it on those artifacts are mostly gone.

Live sample rates the incoming data to the same rate (I don't know what).

[SubFunk]

^^^ true, but i would avoid it, if possible.

[Tone Deft]

^^^ true, but i would avoid it, if possible.

why?

the only think I might avoid is using 44.1k because it's the most problematic rate to convert to and from. even then the artifacts are below -60dB which is very very very low in the mix, basically inaudible, quieter than the fan on your computer from the other side of the room.

[SubFunk]

^^^ true, but i would avoid it, if possible.

why?

the only think I might avoid is using 44.1k because it's the most problematic rate to convert to and from. even then the artifacts are below -60dB which is very very very low in the mix, basically inaudible, quieter than the fan on your computer from the other side of the room.

first, as i mentioned here already i had problems (atefacts and general problems) ok it was a hardware issue (ASE/EBU)...

but i guess as the sample rate is changed in realtime while you go, it most probably sucks a hell lot of CPU crunching... and that with already a not CPU friendly application... that is why i said: it is possible, but i would avoid it. (admit that i did not tried the different CPU behaviour in a project yet)

it's a bit like with normalizing... any programmer, and probably R.H. will tell you that in theory it is just a calculation thing, not changing or having any effect on your audio except the desired one whatsoever... but the reality often shows different. at least that is my experience...

so i would avoid it, if not 100% necessary.

[Tone Deft]

^^^ true, but i would avoid it, if possible.

why?

the only think I might avoid is using 44.1k because it's the most problematic rate to convert to and from. even then the artifacts are below -60dB which is very very very low in the mix, basically inaudible, quieter than the fan on your computer from the other side of the room.

first, as i mentioned here already i had problems (atefacts and general problems) ok it was a hardware issue (ASE/EBU)...

but i guess as the sample rate is changed in realtime while you go, it most probably sucks a hell lot of CPU crunching... and that with already a not CPU friendly application... that is why i said: it is possible, but i would avoid it. (admit that i did not tried the different CPU behaviour in a project yet)

seriously? whatever man... you had some piece of gear that was poorly designed.

as for the CPU thing, ALL incoming data is SRC'ed.

buncha bullsh1t.

[Tone Deft]

on the contrary, all the improvements to your workflow far outweigh your complaints.

for example I record live instruments (bass, guitar etc) at 192kHz for low latency. then as the CPU load goes up I move to 96kHz. when I'm done recording stuff I move to 48kHz when I do stuff where latency doesn't matter.

all the stuff you're talking about it a moot point compared to the ease of workflow when changing sample rates. use the right rate for the job you need to get done.

[SubFunk]

^^^


DM 2000, d&b D12 poorly designed??? LOL you not have a clue do you?

as for the CPU thing, i know in theory your are right, but hey everyone to his own... use as many different sample rates as you want.
and normalise the shit out of your audio...

i don't, because it prooved not to be a good outcome. neither with Peak, sonic HD, Live, Logic, PT or any audio prog i ever used.

and poorly designed hardware... LOL again.

[leedsquietman]

There are times in my experience where artifacts are noticeable (and certainly not at -60, I'm talking warped audio here (especially 48Khz > 44.1Khz), ranging from blurry sounds to clicks and pops), this is often because SRCs are not equal and running it as a real time process is not really what this type of task was designed for.

I'm not critiquing Live's SRC ability here, *in hi quality mode* because that opens up that can of worms about the infinite wave tests, which I think are flawed as they don't test a realistic scenario.

Even if you did run different sample rates in the same project successfully, on the fly converting is taking CPU power away from other tasks.

It just makes sense to get everything on a level playing field. Good file management and good audio practices make it easier to get the best results quicker IMO.

I have to say that my experience is closer to SubFunk's experience here, but you know Tone that I respect you and your opinion. You probably have a good hardware combination that works well for this, whereas other hardware combinations produces a less successful result.

[nms]

I change sample rates in the same project all the time. try it.
make sure the Hi Quality SRC mode box is checked in the preferences and you're fine. ethios4 did a test once and it was shown that without Hi Quality mode on there are some artifacts, with it on those artifacts are mostly gone.
Live sample rates the incoming data to the same rate (I don't know what).

I've avoided using the high qual mode since I saw that it boosted 0db peaking audio to clip past 0db and seemed to add more high end so that alteration sketched me out a bit.

the samples I'm using are bass tones though for my bassline.. so I have them laid out in a drum rack via simpler... so there is no high quality option. I opted for this method rather than using an instrument rack because I've got multiple bass sounds and it just works well for laying out the whole thing.

also, on the topic of using samples that are all the same rate in live...
there is no setting in live to change your project's working sample rate. so is there one? what sample rate can we use that won't cause any extra on-the-fly conversion?