What sample rate in Maschine / computer?

What's good?

I'm undecided what sample rate to choose in Maschine. Default was 44.1k Hz. I upped it to 48k Hz.
Does it even matter for playback or is it only important for actual sampling into Maschine?

Another thing adding to my confusion: The default sample rate setting for my Mac internal speakers is 44.1k Hz.
(You can check yours by going to "Audio MIDI Setup" > Built In Output)
Do I have to change that setting to 48k Hz as well or is most stuff on the web (YouTube etc.) in 44.1k Hz and it's better to keep the default setting?

What sample rate do you use? What sample rate is your Built In Output set to?

Thanks.

 

Personally for music stuff I would just work at 44.1 and avoid any unneccesary sample rate conversion later on, unless you have a very specific need for a higher sample rate. For instance, when working to video 48k is much more common. Or if you have world class outboard gear and are recording acoustic instruments, sometimes working at 96k has some slight benefits. These days though, most software instruments are good about not creating aliasing like in the past, so there's not much to gain working at 48 if you're just going to have to downsample later on.

 

I use 88.2 KHz.
Not because of sound qualitiy but latency is 5.4 ms instead of 8.3 ms.

And not a big impact on CPU.







I export 44.1 though mostly.
48 is for video (whyever it is) and 96 is double of that.
so i use 88.2 which is double for 44.1
And again, its for latency not for sound quality.

 

That's a good point about the latency, something I tend to forget since I don't usually have issues there.

One thing to point out is that it doesn't matter if you're running at exactly double your target SR. The way modern SRC works there's so much oversampling happening anyway that the exact ratio has no effect on the final results. It's not as easy as just cutting it in half, so to speak.

 

I do 48/24.
Audio only gets bounced to 44.1/16

Everybody does YouTube and TV nowadays so I can't ignore it.

 

when you work with sounds that are in specific frequency range, and and not an entire one, like a mix down, then 44 is fine. remember, this is the playback and record, the waves are still processed at 64 bit.

 

Yeah that's an important point. Comparisons done with loopback tests and cable are important.

Interfaces are fully optimized on a specific setting. In the case of my Focusrite Saffire Pro it is at 48Khz

 

Dope. Thanks for all the replies!
Now, that I know that I'm good with both, 48 and 44.1 kHz, I'm still wondering what I should set my ouput sample rate to.
The default sample rate of my built in speakers is 44.1 kHz (which I think the majority of music on the internet has).
But that would mean all my stuff that is 48 kHz would get resampled for playback. If I set it my output sample rate to 48 everything at 44.1 gets resampled.

What's the optimum here for output sample rate? It surely ain't adjusting it every time!?

 

Just record at 48/24 and export to requirement once you've mixed/mastered.
CD for instance is 44.1/16.
There's plenty of articles out there to Google if you want.

 

Maybe you didn't understand my question or I don't understand your answer.
I'm not asking what sample rate I should record / produce at.
I'm asking what output sample rate I should use (not of the track export, the hardware output sample rate). The output sample rate for my mac speakers is atm 44.1 kHz.
It's confusing af to me that my output sample rate differs from the audio's sample rate.
Is there some middle ground setting for the output sample rate?

How do you go about this? It's really bugging me that I can't find a concise answer for this question.
Literally spent two full days researching this topic which is why I turned to this forum for answers since some very knowledgeable people seem to be active here.

 

Where can that info be found? I have a cheap 2i2 and would like to check, it's probably the same as yours though

 

Your speakers are analogue devices so digital to analogue conversion will take place prior to digital audio leaving your audio card.

Any digital audio data that is sent from your DAW to your audio card will include a header as well as the raw PCM data.

The header specifies the file type, sample rate, sample size, bit size etc.

The simple answer to your question is that it doesn't really matter because the DAC will perform the conversion based on the information in the header (i.e. it knows the sample rate of the data being sent to it).

 

If you're going to work at 44.1, then I would set your soundcard to also work at 44.1. There's no benefit to mismatching samplerates, if anything it will likely lead to more CPU overhead.

 

Thanks the reply, man. Thing is I read somewhere else that everything gets resampled if it's not matching the output sample rate. Now, reading your statement here I got two conflicting arguments. I really want yours to be true, it would make this thing so much easier.

True, you are 100% right but the thing that's bothering me is when I work / play stuff that is not matching my output sample rate.
For example I got a song produced at 48 kHz and play it with my 44.1 kHz set speakers --> Will quality suffer? Accurate playback?
Do I really need to go into my settings and change my output sample rate for accurate output? Maybe not during playback but let's say I pull a 48k beat into a DAW for mixing --> Change settings or doesn't matter?

Tbh, I'm surprised there's so little information on this on the web.

 

mm, not to be hard on CakeAlexS soundcard, but I only saw this in "old" and/or entry level sound cards that don't have fixed sample rates at every common values, but down/up sample. Never saw this in higher levels audio interfaces. I think the Emu sound cards for example did that, but that's a while ago.

 

When you are using a DAW the sample rate is set within the DAW and you should find that you can no longer alter the sample rate from within your audio interfaces settings page.

If you set the sample rate in the DAW to 48000Hz, all incoming audio will be recorded at 48000Hz and all outgoing PCM sample data will be at 48000Hz. The DAW will make the audio interface aware of this sample rate via the ASIO driver.

If some of the sample data you are outputting has been previously recorded at a different sample rate, the sample rate of this data will be converted(up/down sampled) in real time to 48000Hz.

 

Strange, my experience has generally been that doubling the sample rate gives me a higher CPU hit than halving the buffer size of the lower sample rate.

 

Try a few. If you can't hear a difference, don't worry about it. If you can hear a difference, use the one that sounds better. You don't need to waste 2+ days worrying about this.

 

There is a guy who maintains a list of interfaces and the optimal sample rate settings for latency. I just done 10 mins of googling and I can't find it sorry.

Regardless best way is to do your own loopback tests (cable from output to input, record a tone and measure the delay, Google for full details) with different samples rates and compare. You can find out for yourself.

 

24 bit gives better headroom than 16 bit when recording.

Higher sample rates always sound better and if they don't you will convince itself it will