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Author Topic: Output to a DAC using Foobar+ASIO  (Read 4006 times)
nosh
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« on: March 22, 2011, 12:07:15 PM »

I recently upgraded my sound system and got a DAC to go along with the new equipment.

The DAC has been set as the default output device in Windows and it shows up as "USB Audio DAC" in Foobar. Foobar also supports an ASIO plugin. Does one of these methods have an advantage over the other?

Someone on an audiophile forum mentioned that ASIO would give far superior sound quality, I'm wondering if he misunderstood and assumed I was comparing it to DirectSound or some other method which would involve Windows processing the audio. If it's all the same I'd much rather not use ASIO and save myself the hassle of configuring and updating the ASIO component.
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nosh
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« Reply #1 on: March 22, 2011, 03:46:16 PM »

Solved: The "USB Audio DAC" option does route the sound through kmixer, so it was a false assumption on my part that it did not. Should have got a hint by the fact that I could hear other Windows sounds while the music was playing.
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40hz
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« Reply #2 on: March 22, 2011, 03:59:45 PM »

All depends on what you plan on doing with your DAC box.

ASIO is a low latency driver which can give you better audio quality mainly by virtue of the large number of professional 3rd-party plugins you can get that use it.

It also allows for programming direct low-level access to audio hardware components without needing to go through intermediate processing steps. This pays off big time when you're doing something like playing a software synth through your DAC and you don't want to deal with occasional time lags that may crop up between between pressing a key and hearing the sound.

So if you're doing synthesis or sampling (especially using keyboards in real time) or you plan on doing live recording or studio type post production work on audio tracks, ASIO is probably the better choice. Especially if you want to start exploring the world of pro-audio plugins.

For simple playback, home theater, games, and much everything else, DirectSound should work just fine.

That's my 2ΒΆ anyway. smiley

« Last Edit: March 22, 2011, 04:01:29 PM by 40hz » Logged

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nosh
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« Reply #3 on: March 22, 2011, 04:10:56 PM »

I would have to agree with you. I can hardly not tell the difference between the two.  smiley
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f0dder
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« Reply #4 on: March 22, 2011, 04:11:55 PM »

I didn't think ASIO vs. regular drivers had anything to do with sound quality, only latency?
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nosh
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« Reply #5 on: March 22, 2011, 04:15:48 PM »

kmixer apparently plays spoilsport. Shouldn't worry those who've moved past XP.
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f0dder
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« Reply #6 on: March 22, 2011, 04:34:20 PM »

From that site:
Quote
The downside to this? If there are errors on the CD, you will hear them as there is not any correction being made to overcome the error.
Like, HUH?! - this seems to imply that it's the sound drivers, rather than the cd firmware, doing the error correction? That sounds pretty wrong to me.

OTOH, the post might be right about the default sound mixer (do realize the post date, though, algorithm might have been improved since 2004... somebody should look into this), but even if that's true, wouldn't that only happen when multiple sound streams are played back? Or is it enough to have adjusted the volume control?
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nosh
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« Reply #7 on: March 22, 2011, 09:49:08 PM »

Hopefully, this article is more credible.

http://code.google.com/p/...iadrivers/wiki/Bitperfect

Quote
Contrary to popular belief, the kmixer of Windows 2000 and XP doesn't modify the sound and is thus bitperfect if these four conditions are met:

   1. The PCM/wave volume slider of the mixer (sndvol32.exe) must be set at its maximum. Some start-up applications modify the volume slider (e.g. hardware monitoring tools from Asus).
   2. The player must be compiled for the same architecture that the OS was compiled for - e.g. 32 bit player on a 32 bit OS, 64 bit player on a 64 bit OS. This is the case for the vast majority of installations because the 64 bit version of Windows XP isn't very commonly used.
   3. Applications other than the player mustn't play sounds, otherwise the two output streams will be potentially sample rate converted and mixed.
   4. Applications which are using the soundcard for recording have to use the same sample rate as concurrently running applications which are playing sounds - the soundcard has only one clock generator and hence this limitation arises.
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f0dder
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« Reply #8 on: March 24, 2011, 12:11:56 PM »

Goodie, nosh - that makes a helluvalotta sense smiley
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nosh
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« Reply #9 on: March 24, 2011, 01:46:03 PM »

Yeah and they're easy to implement rules, so I'm skipping ASIO altogether and going with DirectSound. Played a Test CD yesterday, couldn't tell any difference between ASIO and DS.
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