# FL Studio 12 – “Mix in buffer switch” issue



## Kubler (Jan 12, 2018)

Hi everyone,

I noted lately that enabling the "Mix in buffer switch" option would save nearly half my CPU load. So obviously I'd like to keep it this way, the problem is that then, the whole playback of my project starts playing at low speed and clicking furiously. It sounded like some kind of sample rate issue at first so I checked my audio settings, but everything is at the same rate, the project, my external audio card and even my headset.













I assume then that the issue is related to the processing, but I'm still a newbie in that kind of DAW adjustment and I can't tell what I should change in my settings… Any help would be greatly appreciated !

Thanks in advance


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## Ecliptiq Audio (Jan 12, 2018)

Hi,

If you enable "Mix in buffer switch", it means that the output audio is mixed in ASIO's buffer. For some interfaces, it allows to reach lower latencies. However in some cases if the driver can't process fast enough, it can cause clicking because the underrun counter is bypassed and buffer underruns are more audible.

You can try adjusting the buffer length, although it's already long enough as I see. Unfortunately I don't have any other solution.

Sidenote: In my opinion, unless you recording vocals or instruments, you don't need 96kHz sample rate, 48K is enough. But I don't know if everybody agrees with that.

Regards,
Peter


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## Kubler (Jan 12, 2018)

So considering that "Asus U7 Asio" is my external card's driver, it means that it is not powerful enough to handle my mix's processing ?

Maybe I should check if I can still increase my buffer's length then — I don't care much about latency because I don't use MIDI keyboards or such. If it's not possible, I guess I'll have to look for a new audio card.

For the sidenote : I have reminiscences of an engineer explaining to me that because of a physical effect of which I forgot the name, mixing in x kHz would result in only x/2 kHz being actually emitted… So I thought that mixing in 96 kHz would be "true 48 kHz" or something like this.

That being said, mixing in 48 kHz would probably save me some CPU, so I guess I'll have to chose between having good audio performances and being a maniac


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## Ecliptiq Audio (Jan 12, 2018)

I think he explained the Nyquist-Shannon theorem. The theorem states that when the sampling frequency is greater than twice the maximum frequency of the signal being sampled, the original signal can be faithfully reconstructed. For us humans, we need to capture the 20kHz maximum because that's what we can hear, so even 48kHz is enough for that.


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## Kubler (Jan 12, 2018)

Yep that has to be the Nyquist-Shannon theorem, and apparently I had misunderstood what I had been told.

Thanks for clarifying this to me and thank you for your help !


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