[10068]

Background and setup; I am using the following:

* Stock (aka unmodified) Auzentech X-Fi Prelude with the latest drivers,
* WinXP Pro SP3,
* Newest version of Foobar2000, (v0.9.6.6 at the time of this writing)
* ASIO output obviously

* Auzentech drivers' config set to Audio Creation Mode @ 44.1kHz,
* Auzentech drivers' config set to Headphones output type, (I found over time that I actually prefer this to 2.1 Speakers output! I was surprised...)
* Auzentech drivers' config set to Bit-Matched Playback of course.
* No resampling / oversampling used in Foobar2000,
* Only DSPs used in Foobar's output are Advanced Limiter and EQ, (see below)
* Grado SR-80s,
* No headphone amp in output path.

So: I was messing around with output settings because the treble was bothering me... if you've seen my other thread(s), I have been dealing with Tinnitus lately. I've had it in some form or another since 2004-2005 but it got particularly bothersome recently. This combined with my life-long sensitivity to the very high frequencies made for listening on headphones impossible at some times

Really had me bummed; as everyone here should feel as well, music is life.

I set up and began using the EQ in Foobar (to curve off steeply @ and after 7kHz; reaching -20dB at 20kHz) in a way that to my ears impacted SQ the least as I was able to get it to -- which took some fiddling with configurations for sure -- and that helped take the edge off; unfortunately, treble was still an issue.
------


Lo and behold!~ I had had ASIO Buffer Latency set to 10ms in the ASIO Settings for "Auzentech X-FI [ASIO]" within the ASIO Output Settings sections, since as long as I can remember. I vaguely recall that at the time, it sounded most "snappy" to me if that makes any sense, without interrupting the flow -- at any less than 10ms, the sound signal gets cut off periodically if you approach 100% CPU usage due to other apps running simultaneously; comparable to if you have the output Buffer in the main Foobar Output Settings set too low. I changed this to 300ms on whim to see if it helped (heck, I’d tried everything else...) and it made a night-and-day difference! Awesome.

Today has been about a week (-ish) since this change and bass has better impact, volume and realism, and treble is nice and smooth. Fantastic.

Hopefully this is useful information to my fellow 'Fi-ers

And no, I don't think this is placebo. Trust me, my ears are complete sissies right now; I’d pay for it if the treble was as harsh as before. (Not that it was harsh relatively to begin with, but you know what I mean).


Happy listening!


EDIT: And try it out yourself; let me know what you think!

 

[endless402]

i'm using the m-audio firewire and my latency is the minimum setting? doesnt take up any cpu usage

 

[12Bass]

ASIO latency adjustments should not change the sound unless the computer is having problems keeping up with the buffered audio stream and causing dropouts (clicks/pops). Small buffers create a much greater strain on the CPU and subsystems. While low latency is desirable for recording, I'm not sure it is that useful for playback. Setting processor scheduling for background services can help achieve stable low latency operation.

 

[10068]

Quote:

Originally Posted by 12Bass /img/forum/go_quote.gif ASIO latency adjustments should not change the sound unless [snip]

Buzzkill!

 

[daglesj]

I too was always under the impression that ASIO latency was more a technical/performance issue for the CPU/audio gear rather than a 'sound quality' feature. For playback it shouldnt really matter whether you set it to 5ms or 500ms.

But hey if you think it works then good for you. Nice one.

 

[10068]

Seriously no one has tried this, played around with this setting or noticed anything changing the sound in regards? I'm kind of surprised...

 

[Bojamijams]

latency is only for recording.. it has NO effect on playback

 

[10068]

Quote:

Originally Posted by Bojamijams /img/forum/go_quote.gif latency is only for recording.. it has NO effect on playback

If that's correct, then why would foobar bother to even have the option to change it?

 

[Slogra]

For playback you don't need a small buffer, you can set the buffer as big as your want. It will lower the chance on pops and skips. A delay in the sound is not a problem at all.

However if you are making some live music on your pc, then you want a smallest delay as possible, so you need a small buffer. For example when you press a key your (musical) keyboard you don't want the sound from your pc after a second, you want it immidiately.

 

[12Bass]

During playback, large buffers can create a minor annoyance by introducing a delay on EQ (or other) adjustments. Other than that, there should be no effect on sound quality.

 

[Bojamijams]

Quote:

Originally Posted by Sduibek /img/forum/go_quote.gif If that's correct, then why would foobar bother to even have the option to change it?

Its not foobar that lets you change it, its the ASIO plugin that lets you change it, as that is the main point of ASIO.

 

[10068]

Quote:

Originally Posted by Bojamijams /img/forum/go_quote.gif Its not foobar that lets you change it, its the ASIO plugin that lets you change it, as that is the main point of ASIO.

FULL SIZE: [1280x960] = http://img40.imageshack.us/img40/284...foobar2000.jpg

 

[Shizdan]

What about asio4all?

 

[Slogra]

Some cards don't have ASIO support, or just plain bad ASIO support. So in that case you can use asio4all, which works really well imho (good quality and low latency). Kernel Streaming output is also a good alternative.
Personally i just use waveout because it is bitperfect on my soundcard already

 

[olblueyez]

WASAPI works well and eliminates all of these questions.