I bought an Audigy2 ZS which does 24/92 out. Since CD music is recorded in 16/44 would making the output 24/92 help out a lot? What would I do to foobar to make ALL my music output at 24/92?

Thanks,
Joe

Let me guess... you asked this even without opening the Preferences window.
That's not how you're going to learn about FB2K.
Go read through every single dialog box and instruction; supplement what you don't know using the FB2K wiki, which is linked from the front page of HA.org.

Furthermore, this question has been asked many times, hence I recommend you use the Search function.

Only after doing these 2 things, and you're still clueless, then it is all in your right to ask.

Anyway, it's 24/96.

CD-music is indeed sampled at 16/44.1, but if you have ripped your CDs to a lossy format like mp3, ogg vorbis or musepack, well, these files do not have bit-depth per se (I'm not familiar with this subject though).

To change your bit-depth, go to the Playback dialog in Preferences. To resample your files, you have to get the standard DSP array from here. Then go to the DSP dialog in Preferences, activate the Resampler DSP, then go to the Resampler dialog to modify the samplerate. Do not activate slow mode.
I cannot advise you as to the sample rate you must set, because I don't know what to do on the A2 ZS.

Make sure to disable EAX and all other effects, and disable calibration, or set basic calibration only (there may be more to do, but again I'm also clueless). Otherwise, the A2 ZS will not play at 24/96.

I don't have an audigy so I can only tell what I've read here (and elsewhere):

• With this card there's resampling to 48kHz on playback with probably not-so-decent quality. This can be avoided using fb2k resampler to 48kHz; if you set output to 24bit (or 24bit padded to 32 if you use kernel streaming) you avoid dither noise added to the signal or truncation distortion caused by changing the resolution back to 16bit after resampling.
• It might be possible to disable internal resampling so fb2k resampling isn't necessary. There's a recent thread about this - use the search function to find it.
• There have been claims (especially at head-fi forums) that upsampling CD audio to high resolution like 96kHz improves sound, but I haven't seen any evidence for this (= double blind tests). If upsampling to 24/96 'helps a lot', there's something seriously wrong with this card, I'd say. I don't know how your card handles 96kHz input - hopefully it isn't resampled back to 48kHz internally... You'll probably find information about this using the search function too.
• If your card doesn't downsample 96kHz input, resampling to 96kHz with fb2k most likely won't do any harm - besides useless waste of processing power in worst case.

To set fb2k output to 24/96 enable resampler DSP (fast mode is enough) to 96kHz and set output resolution to 24bit fixed point in playback settings. A good test to find some problems is playing back udial.wav sample from "Test your soundcard for clipping" thread (-> search ...).

Audigy2 is not 24bit/96 Khz... Well I know that's what it says on the box (I bought one ^^) but it's not. All it does is resample 44Khz -> 48Khz because it can't output 44Khz....
So I think you should set output to 16Bit, You can resample to 48Khz (this way your audigy won't have to do it, and the audigy resampler is said to "color" the sound), but resampling to 96Khz is useless.


err... Unless the "ZS" is different from the original Audigy2, I don't know actually.

SBLive or Audigy1 based cards should be used with 16bit 48kHz samples, and Audigy2 based cards should be used with 24bit 48kHz samples for optimal quality.

are you sure about that 24 bit thing ? I remember reading a while ago that Audigy2 was actually 16bit...
well, can't tell the difference anyway...

Audigy2 can perfectly output 24/96, except for DSP effects (EAX or speaker calibration; when applying these, the signal will be downsampled to 16/48).

It's the Audigy1. I know because I have one.

Here's a good post about the Audigy2 (non zs) doing 24/96

http://www.3dss-forums.com/cgi-bin/wwwthre...=2&fpart=1&vc=1

Basically, I turn off a load of features on the A2 and then it can do 24/96 but that's doing it via winamp and I like foobar better mainly because I love smaller programs

Anyhow, in the playback menu i selected 24bit bit depth fixed, then how do I crank the resampeling rate to 96?

Foobar's prefs has me pretty confused, i dont know too much about noise shaping or a lot of this audio lingo, and honestly with work and 4 classes in college, I wish I had more time to learn this stuff.

So I got 24 by bringing the bit depth up under playback, what about the 96khz sampeling rate? I just go to resampler and crank it to 96? If the Audigy2 is resampeling up from 44.1 to 48, shoudnt i just have my output 24/48? Can I do that?

Just enable resampler in preferences - playback - dsp manager and set it to 96. But it's not recommended - it will only eat a lot of your CPU and you will not have any benefits and quality gains. Set it to 48khz.

I set it to resample to 48khz and the bottom info bar on fb2k it says 44100...

It's as should be. It's because your files are 44,1 khz. It's showing real sample rate of your files.

Well I assume that you've already enabled Resampler correctly in the DSP Manager.

The info bar displays the info of the original file, which should be at 44100 kHz, especially if you ripped it from a CD or downloaded (TSK!) it from the net.

when im encoding my wav files with lame, if I resample the files to 48000 during the encoding process is that better than having the A2 resample the file on the fly? (im not talking about cpu usage, but quality)

And i dont understand how taking a 16/44.1 file and upsampling can help quality. Is it like taking a 600x600 image and stretching it to 800x800? So you see the pixels and all the detail, or is that a bad analogy?

Quality isn't better in any case! Audigy resamples everything to 48khz internally. But foobar's resampler is higher quality than audigy's internal resampling.

so how would i get maximum quality with my files? why do people even play their mp3's at 24bit depth? Would using 24bit padded to 32bit give me the best results with kernel streaming? how come with waveout when i select 24bit output foobar crashes? lots of questions

If you want higher possible quality with your hardware:
1.Set output depth to 24bit fixed point
2.enable resampler in DSP menu. set it to 48000hz. Place resampler at the top of dsp chain before other dsp plugins
3.encode your mp3 files at 44100hz

I hope this will help you.

i set it to 24 bit depth and 96khz, why instead of 48? because I can dammit!

Why do you want to resample to 96 kHz? Your Audigy will resample the output 48 kHz anyway. You are just wasting CPU time for useless job. Altho there shouldn't be any difference in audio quality.

LAME is optimised for the 44.1 kHz samplerate.

The crux of the problem here is that Creative cards follow the AC97 specification, among which one of the criteria is to resample ALL audio to 48 kHz.

So, as a fellow poster stated earlier, one uses FB2K's resampler only because it exhibits a slight edge over the A2's resampler in quality. The difference is most likely to be inaudible with A2's latest drivers though.

ALL audio eh? Then why would they advertise the audigy2 as 24bit/92khz if they cant hit it? Sounds like a false-advertisement law-suit from what your telling me.

It's just an advertisement. nVidia is advertising in GF4MX documents that it is DX8.1 card... Yes GF4MX works when DX8.1 is installed and yes Audigy2 works with 24bit/92kHz input...

Sue them about the Audigy1 too, while you're at it.

Audigy2 can do 24/96 with using p16v chip, but that chip cannot process any effects and is used only for that.