Hi, guys. I've been sitting on this one for a while, and I couldn't find the answer myself. I'm wondering does anyone have the same problem as me. Or maybe it's normal, I don't know.

So, When switching ASIO latency settings in the DAW I get a very noticeable difference in sound. I tested it with Live and Studio One. And they both do this. But you'll get a different sound for different latency settings and S1 even glitches out on the bigger ones. At 500ms it's unusable glitchy no audio in S1 (plus DAW's interface is lagging too), but in Ableton, it gets a better sound. And by a better sound I mean it gets better transients response the clarity is better. And it is quite random. In S1 on 100ms, it sounds flatter but stops lagging, on 80ms it sounds better than on 100ms. On 300ms It sounds good too, and I think it sounds a bit different, but it lags sometimes. On 40 ms it sounds flatter again. In Ableton the pattern is different.

For a long time I worked in Ableton only and I kinda thought that it was normal and the bigger values are better for sound. But I didn't extensively test it back then. Now I'm incorporating s1 for mixing and blah blah blah it got me worried.

The question is: Does anybody know wtf is going on or experiencing the same stuff?

It should be said that I've got quite an old and shitty interface. And no I'm DEFINITELY not fooling myself that it sounds different.

 

Last edited: Mar 11, 2017

what does dpc latency cheker say on your sys & latency mon

to answer your other question
besides differences in headroom (between cubase and live) never had/heard such severe diferences in daws

 

I have no answer for you, but have experienced and realized this myself.

 

What is your soundcard ? Drivers + soundcard vs pc components makes thousands of combinations. What is the sound when you switch to a WASAPI or Windows driver or generic ASIO mode ?

 

CHeck if Audio resolution are set correctly everywhere ( OS, Audio driver if that applies, Ableton , and any other software )
Example : If you set your soundcard to 44mhz in OS, remember that ableton will overide this and play at 96mhz ( or whatever it is set to.)
If you set them all at the same rate, then you really can compare the audio output

 

i mean one explanation for ableton could be the preference for stable
audiostream for live use vs accuracy and you should check global warp vs no warp
and vst vs samples. and sometimes it could be one bad/heavy plugin
but this is just hypothetical

list the stuff you used in the projects
and i try to recreate with one crappy notebook + interface combo and a workstation
i could test live, sam11, and i think sonar (demo me think)

.ps
sometimes live acted strange if 24bit waves with long silence (from a freeze)
run in a 16bit preject

 

this was linked several times in this Forum,it gives a clue, what the problem might be:

 

OS version ?
Soundcard model ?
Project settings ? (44.1,88.2,96 Khz)

Try dpc latency (like @dondada said).

 

You mean khz? The problem is that different latency settings cause different sound inside each DAW. But yeah I work at 96.

It's around 1000 most of the time. "Your system is capable of streaming live video or audio without dropouts"

I can't actually switch to WASAPI inside the DAW's and check different latency settings but inside the media player it's different to ASIO inside the DAW s, is it supposed to be? Generic (second driver option i have) ASIO driver only lets me go 48khz, does the same thing - slightly different sound for a different latency setting. The sound card, you'll laugh, is Creative Audigy 4.

Thanks for the vid, seems to be very informative.

Maybe it is, I will test with just audio.

So, I did a test in Ableton where I exported a track mixdown with different latencies on ASIO driver at 48khz with 4ms 80 ms 150ms and 300 ms one after another. Loaded it back into the session at 150ms, and not only they don't completely null, one synth part seems to be almost not affected at all but still sounds different with different latencies when I put them against each other with reversed phase on one. It's like some phase shift happening on it. I though it was just a real-time playback problem. This is weird.

 

geez that is bad! 500 is barely legal i would say. for reference mine is around 63. get yourself latency mon and check what could be the (i think) many culprits. lan, wifi, internal soundcard (gfx sound drivers), 3984 or what is it firewire, legacy drivers unchecked system resources, hdd or simply old system?

sorry to say, but work ahead my friend ;)

what about warp, was it off? somethimes daws ad little time/silence before and end of clip. but then again once rendered it should match. anyways get the system as fast and stable as it gets.
and check the dpc while audio is playing in the daw, too.

 

Last edited: Mar 12, 2017

There is another point to check : CPU throttle

Check this :
http://avid.force.com/pkb/articles/en_US/How_To/en378495

 

If I were you,I will buy a real soundcard first.All your problems come from this one.This is not a good soundcard to make music.

A good computer,a good soundcard and good speakers are essential to make music.

500ms of latency is just...wow, I work at 24ms with my RME Fireface UCX,and I've never any glitch or problem,with Live or Studio One or Cubase or Protools.And I can't choose a latency like 500ms,maybe one tenth max!!!

Before my RME I had a M-Audio Delta 192 and had always problems like you,with glitches,artefacts,and my cpu load was raising and falling all the time,it was never stable. All my problems have disappeared with my RME.And the sound quality is another planet. I love it!!!

I know a good soundcard is expensive,but try to make some savings and change your Audigy,I'm sure you won't regret that.Maybe a used Fireface 400 ou USB.Or something else!

And the link of Pipotron3000 give precious advices,do it if you haven't do it yet.

a very low latency is usefull for any live situation,besides that not.That's not usefull when you're mixing your project for exemple.
Personally I work at 1024 samples because my pc is old now .With very low latency it doesn't have enough power.

 

Last edited: Mar 12, 2017

Best Setting should range from 64ms or 128 and even 256 at most I don't think anyone uses 512ms latency its way to much and always messes up
also go into your Bios and check make sure you have C2 and C4 Enabled this will throttle down your pc apps when working strictly with Audio it really works Good Luck

also if your on a tight budget switch to a M-Audio 410 ? pci card there only around 40 bucks on ebay
and will blow away your sound blaster card for audio recordings

 

The numbers you are talking about are not milliseconds but samples.

I think the M-Audio 410 is not a good idea,it's an old product and you take the risk that M-Audio don't make drivers anymore in the close future.And it's a Firewire card,for Mac or you need a firewire card to use it on pc.

 

Last edited: Mar 12, 2017

did you try asio4all by any chance?
since it was the sounblaster who started it all. maybe the audigy will benifit there too

 

I'd get a good soundcard, until then I'd forget asio and just use it to listen back to finished mix to check. I use a tablet sometimes to write and use decent headphones to monitor along with windows sounds drivers and it's usually not far off, beats going back to my early pc days and spending half my free time messing about with crappy asio problems. Good luck getting your problems sorted out.

 

Your Creative driver may not be bit-perfect, and passing though the kernal-mixer. I don't know, but I thought these issues were hashed out when Windows 7 drivers first arrived on scene. For me, this was a problem on XP with my GINA24 card until they introduced 'pure-wave', which bypassed the OS bulls*** and established a connection from hardware to DAW. I currently have 0 problems with RME.

Hopefully, at least, your bounced/rendered files are unaffected. I'm not sure of a solution unless you buy a modern audio interface. Maybe try ASIO4ALL - could outperform your Creative drivers? Ultimately, it's time to ditch the Creative Labs. You'll be much happier with a 64 sample buffer too.

*edit*

s***, i'm looking at a Scarlett 2i2 for $118.00 on eBay. That'll way outperform the Creative Labs driver. I see 2nd hand 2i2s for like 90 bucks on there too.

http://www.ebay.com/itm/Focusrite-S...971855?hash=item43fef0540f:g:ALAAAOSwr~lYtEC2

 

Last edited: Mar 12, 2017

Did you update your soundcard's drivers ?
Humm ... also latency has nothing to do with sound quality. it works or not.
(when it's not working, the sound is chopped + crackles 'n pops )
did you export to wav to compare or you are playing live ?
( in case you exported to WAV, you need to set " No Dither " in the export window. )

 

Last edited: Mar 14, 2017

Thanks for the help and suggestions, everybody! I've downloaded ASIO4all and it does perform massively better. I think my issue is pretty much fixed. Who would know that a driver could fuck things up. I thought it either works or it doesn't. Yeah, I'm moving to another interface as soon as I can.
EDIT: Eh. It doesn't perform better. It's another different sound again. And it does the same thing - different sound for a different buffer setting. Even on a single source. I suppose i'll just get a decent interface and start from there.

 

Last edited: Mar 16, 2017