ASIO4All Explanation - Page 4

Do I need to change it though? isn't 24 bit better or same thing anyway?

edit: another question., about volume control. Do I need to keep it maxed? Meaning both in foobar and windows volume control knob or it doesn't matter? I can control my volume from my speaker pre amp so, was wondering if keeping preset windows volume affects anything, and if it does, at what do I need to keep it.

Yeah, I mean USB-ASIO.

I did download the demo, and during installation it required to plug the DAC and the software recognized the DAC. Everything worked well.

I will test it now, but #"%¤%#¤ the 30 sec beep of this demo is annoying...

Dunno is it better or worse, but USB-ASIO seems to widen the soundstage a tad and brings out more details. The bass of my DT880s have more body.

But definately a good thing to pass through windows mixer. The price is however really high, compared to the "improvements" and free ASIO4ALL...

EDIT:

Listened to Matt Mays & El Torpedo - "Cocaine Cowgirl". Result: OMG the highs! The price came a bit closer to acceptable...

Suzka, you don't need anything special to get bit-perfect playback on the X-fi. Go into "Audio Creation Mode", select "output" (I think that's it, it's on the lower left of the Audio Creation Mode control panel) then select "bit matched playback" at whatever the bitrate is of your source (44.1 khz for CD, FLAC, mp3 from CD sources).

Be aware all eq functions and software volume control will go away...but that's what you want...you want to control volume and eq in the analog domain, to avoid having to resample and losing information in the process. Not only are you avoiding the horrible re-sampling done in k-mixer, but you're avoiding resampling altogether (hopefully you're using the analog volume on an external amplifier).

You can use ASIO to achieve bit-perfect playback, but ASIO != bit perfect. Often, ASIO is simply used to handle resampling outside of Windows K-mixer, or to provide low latency (for applications where latency is a factor) to recording devices and the like. To have bit perfect, you have to have a soundcard or device that doesn't internally re-sample to 48 khz as a hardware function. Most Creative Labs soundcards prior to X-Fi, and the vast majority of USB soundcards are worthless when it comes to trying to achieve bit-perfect playback, because they resample to 48khz irregardless of how you feed them (If you own an Audigy 2 NX, Extigy, or Turtle Beach Roadie, I don't think there's anything you can do to achieve bit-perfect sound from 44.1 sources...you could, however, perform software resampling of your files outside of k-mixer, IIRC). If you want a cheap means to get bit-perfect playback from a PCI soundcard and stream it digitally to an external decoder/reciever via optical, then check out the Chaintech AV-710. It's $25.

I believe it's not an issue in Vista because the new version of k-mixer simply uses floating point math to determine the optimum numbers to send out in the resulting digital stream, when you're doing something like asking windows to control volume or eq. Yes, there can be numeric differences in the result, but these differences are well, well below the level of human hearing, and this process is nothing like the butchery exhibited by k-mixer. For handling audio, Vista is a fine craftsman where XP is a hamfisted dolt.

I'm lost in one regard. Why does resampling to a higher bit rate negatively affect sound quality? Rounding errors? Phase errors? Nyquist errors and jitter? How?

I hope we are not confusing sampling rate (number of samples per second) with bit depth (16-bit vs 24-bit vs 32-bit) - using a larger "word" size should not resample anything, should it, but would merely fit the top 16 bits up front into the 24-bit word, right? The extra bit depth is very useful for performing DSP calculations with greater accuracy, so I certainly hope it is considered a good thing, especially since every single professional grade plug-in performs a bit-depth conversion upwards to 24-bit or 32-bit before applying its DSP algorithms (some automatically converting back down to 16-bit as an option afterwards). The bit depth permits finer variation between the sampled levels, since it permits many more numerical values, being a higher bit-depth word.

Terry

PS - I do see the value of avoiding resampling down to 15-bits by windows, and the benefits of ASIO there and elsewhere, just missing why the Creative Labs cards of yore that always resampled up to 48 were considered troublesome.

Thanks and I appreciate your time on this topic, seeing this thread got me to finally try ASIO with foobar.

I plan to compare the ASIO to the "stock" SQ.

You can always use software resampler like this one http://www.mega-nerd.com/SRC/fb2k.html to avoid all resampling woes, hardware or kmixer based.

Sure, EnOYiN, I understood the benefit of avoiding letting Windows get its mits on the signal. As you felt might be coming across, there was a general impression that all Creative cards that upsampled were "bad" (they did this to match the audio to their soundfont standards, as one reason, and because for a time there was a popular movement in favor of 48 over 44.1 for any card that did recording, and for direct support of SPDIF.) If Windows gives even the old AWE-32 a good signal, it can do a fine job with it - as fine as one usually would need (though its digital out for some reason had worse distortion specs than the analog out! But we're talking inaudible differences there.)

Anyway, this is a great article (and is evolving nicely!) and I hope it will get everyone to using this little piece of software!

(BTW, DTS stands for "Digital Theater Systems" sound.) http://www.google.com/search?client=...=Google+Search

Terry

SB Live! 5.1 (SB0100), latest drivers from Creative (5.12.2.252), Foobar2000 v0942, ASIO4All v2 (ASIO Virtual Devices),

SB Live! Audio [9000] 1 - 32 bit - Left
SB Live! Audio [9000] 2 - 32 bit - Right
SB Live! Audio [9000] 3 - 32 bit - <None>
SB Live! Audio [9000] 4 - 32 bit - <None>
SB Live! Audio [9000] 5 - 32 bit - <None>
SB Live! Audio [9000] 6 - 32 bit - <None>

+ and - in Foobar works volume fine, Windoze WAV/MP3 Volume does not work ... does that mean I got it working correctly? And the Offline settings are what you said to do.