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garyjam

New user with latency/pops/clicks issues, please help.

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I thought before I begin with Cakewalk I would read up on how to improve latency.  So i read the online documentation which simply says:

"If you are using ASIO drivers, click on the ASIO Panel button. This will launch your audio interface’s proprietary control panel. Adjust the buffer settings accordingly."

Uhhhh.  what does "accordingly" mean??    so much for helpful instruction manuals?!

I'm using the most popular audio interface the Focusrite 2i2.  when i get to the control panel, I have choices for sample rate and buffer size.    But I don't know how to adjust them "accordingly."

I have a brand new computer, and I got a really good processor and extra memory, just so I would be able to record music without pops and clicks and latency.

I don't know if my settings are good or bad but I did notice some pops and clicks when started my first recording.

I was hoping that I would be able to go into the documentation and it would clearly tell me how to improve this... but alas it just tells me "accordingly" which I guess it means to move the slider to the "safer" side but at what point is it too far?  the latency increases as I move it over there.

any advice is appreciated. 

thanks!

gary in vermont

 

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5 minutes ago, garyjam said:

 

"If you are using ASIO drivers, click on the ASIO Panel button. This will launch your audio interface’s proprietary control panel. Adjust the buffer settings accordingly."

Uhhhh.  what does "accordingly" mean??    so much for helpful instruction manuals?!

I'm using the most popular audio interface the Focusrite 2i2.  when i get to the control panel, I have choices for sample rate and buffer size.    But I don't know how to adjust them "accordingly."

 

I am using a PC as per my signature and I do not get “latency/pops/clicks”

To me, “Adjust the buffer settings accordingly” simply means adjust so that the latency is tolerable (or barely noticeable) but not getting pops/clicks.

There is a document doing the rounds that can help reduce pops and clicks.

See:

https://download.cantabilesoftware.com/GlitchFree.pdf

One other thing I would suggest is to use the Beta Focusrite drivers.

The release notes for this specifically mentions fixes for Cakewalk by Bandlab.

Using a non Firefox browser, head over to here

http://beta.focusrite.com

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27 minutes ago, Promidi said:

I am using a PC as per my signature and I do not get “latency/pops/clicks”

To me, “Adjust the buffer settings accordingly” simply means adjust so that the latency is tolerable (or barely noticeable) but not getting pops/clicks.

There is a document doing the rounds that can help reduce pops and clicks.

See:

https://download.cantabilesoftware.com/GlitchFree.pdf

One other thing I would suggest is to use the Beta Focusrite drivers.

The release notes for this specifically mentions fixes for Cakewalk by Bandlab.

Using a non Firefox browser, head over to here

http://beta.focusrite.com

thanks that is great info!  I will definitely try the beta version.      so there is one general question here for me..., it seems that the idea is to find a "sweet spot" between latency and pops/clicks.   the issue i have here is, this is subjective.  some people can hear the latency better than others and some can hear the pops/clicks better than others.   I guess there isn't an objective tool that can take care of this?  I'm also just a little surprised that with the speed of computers today that we are facing a choice between the two, or even that we are on the edge of it being tolerable.   aren't todays computers fast enough to make multi track audio recordings without signifcant latency and zero pops/clicks?   I just worry sometimes if there is a pop or a click, what is that exactly?  Is it in the recording? or is it in the playback? will it be in the exported wav file?   and during that milliseconds of a pop/click, what is happening to the recording?  is it on top of the recording or is it actually pushing the recording back in time for a few milliseconds every time ?        

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When I started using CbB, recording was flawless.  It was only with playback that I experienced the very rare pop and click.

That was usually when notes requiring CPU heavy synths (MSoundfactory comes to mind) kicks in.

In my case, the setting that fixed that, was to disable C-States and Intel Speedstep.  That's basically telling the CPU to run at full speed at all times rather than stepping down the clock speed when a higher speed in not needed.  As you can imagine, this can play havoc for DAW use that needs a batphone to a full speed CPU.  What you can also do is set CPU min and Max to 100%

These days, it's not just the shear power of PCs that makes for flawless DAW operation.  It's more how they are optimised.  Remember, on my lowly, 2014 i5 4670 CPU, I do not get clicks and pops during playback within CbB. 

For me, 256 samples seems to be my sweet spot.  I do play VSTis inside CbB in real-time and I do not notice the very slight latency.  Remember, if you are 2 metres from your monitors, the delay there is 5.8 milliseconds.

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Hey, been there. Was there in May, actually.

This thread (linked below) goes off at tangents, but it’s about exactly your problem. I’d take a look especially at Bitflipper’s long post on DPC and my summing up (and its addendum) towards the end of the thread.

 

 

Edited by Mark Bastable

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"The most popular" rumor about audio interfaces and latency:

  1. Latency importance is oversized (each 30cm from an audio source adds 1ms latency in any case). In general, latency limits are:
    • vocal monitoring throw software - <3ms. Not used in practice, since partial direct monitoring (with  zero latency) solve the problem. You normally want just reverb added, and it can have 50ms+ latency.
    • e-guitar soft sim monitoring. Preferably <5ms.
    • e-drums with soft synth monitoring. Preferably <7ms.
    • MIDI keyboard with soft synth monitoring. 10-15ms is tolerable is most situations. Under 20-25ms is playable.
    • for anything else latency is not important
  2. CPU power has little to do with lowest possible latency. The difference is like a truck vs sport-car, you can drive with 10t but that does not mean you can drive fast. Sure, in case you need that 10t (in audio case many heavy soft-synths and effects) you need a car which can do that. CPU characteristics are declared as "power", not as "speed", even so CPU frequency is naturally perceived like a speed. The fact is, any 10MHz DSP easily beats in latency (many times) most powerful 5HGz desktops. The key to success with latency is strict audio optimization in BIOS and OS. And there can be brick walls (in hardware and drivers).
  3. "The most popular audio interface" is Realtek. 2i2 is a good entry level music audio interface with pre-amps, it is not in top league in any category (latency, drivers, sound quality) but it is reasonable for many use cases. Stable usable latency is around 8ms, and so it can be inconvenient with e-guitar soft sims only. For comparison, under the same relaxed settings on the same system top (in latency) interfaces have under 5ms.
  4. The lowest allowed by driver latency is rarely usable in practice even on optimized top system. When looking into "latency charts", buffer sizes under 32 are not meaningful.  Check what particular interface does with buffers 64/128 on 48kHz.

"Accordingly" in the context means the highest setting of latency you are still convenient or the lowest your system and the project allows without pops and clicks, in case you get problems in your convenient range. That is human, project, system and tasks dependent.

 

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