Fb2k settings for RME Digi96/8 Pad
Feb 28, 2004 at 4:09 AM Thread Starter Post #1 of 27

kelesh

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Upsampling, resampling, output, etc.

I've searched and found 24 padded to 32bit to be the recommended setting. What resampling setting do I want. Also, what waveout and asio settings?

For Output(and in the RME's settings), do I want to choose "Digi96/8 PAD Out(1+2)(1)" or "Digi96/8 Pad Out(7+8)(1)" or any of the ones inbetween?

Which RME Mode do I want(DIGI Settings)? 2048 or 1024 Samples?

Changing these in my rme digi's settings seems to have no change. Right now I have them on 7+8 *shrug*

Regarding volume settings: setting windows master and wave volume to max, settings RME DIgi settings to -15.0. This leaves me with my META42's volume control between .7 and .85 rotation, with 1 being the max volume. Is this a good idea?

edit: Asio does not seem to work. i can't get any foobar settings to work with the asio output. any ideas?
 
Feb 28, 2004 at 5:24 AM Post #2 of 27
to use the 24bit padded to 32bit setting with foobar you must make sure that the ASIO part of the Digisettings is at a 32bit resolution (1024 or 256 samples). 1024 samples is fine for general music listening, you'd really only use a smaller value if you knew what it did
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You might also want to add the resampler (SSRC) plugin to the DSP manager and then change the settings for the resampler to 96,000Hz, this dramatically improves the music quality, though you'll get the most improvement if you use lossless files or source cd's, not MP3 or vorbis.

ASIO is generally better as the output plugin, the settings you want for the config page of the plugin are:
Device: ASIO Digi96 series
Thread Priority: Highest (time critical is overkill)
Buffer Size: 63
Shift output channels: 0
and the Use direct input monitor unchecked.

If it's a but stuttery then go back to the playback tab of the options and move back the Playback Thread Priority a notch to 2/3 the way along.

I can't answer about the output channel part or the right setting for the volume though.

Hope that helps
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Feb 29, 2004 at 2:34 AM Post #3 of 27
awesome!! thanks!
 
Feb 29, 2004 at 6:03 AM Post #4 of 27
Resampling doesnt make it sound much better, even with lossless audio. It makes it sound slightly different, but not always better. And the difference between an mp3 and a flac isn't always noticeable imo...but then again I don't have an amp yet, so that may be why. ANd you don't need the buffer size at the max.. That will make fastforwarding and the like a real pain. I keep mine at 7 and almost never have skips from other normal cpu processing (p4 2.4), and can browse a track with little delay and without the stuttering that occurs with realtime.
 
Feb 29, 2004 at 6:33 AM Post #5 of 27
i am having static pops in my music with a 63 buffer.. digi Mode set to 1024 sample.

Seems to be less with a buffer of 7, but I still hear them rampant in APE files with foobar.


Using Asio(rime critical, buffer size 7), resampler to 96,000, continuator(crossfader), crossfeed, volume control, and advanced limiter.

Any ideas?



Athlon XP 1700+, 1gb pc2100 crucial ram.
 
Feb 29, 2004 at 7:32 AM Post #6 of 27
also, do i want to use 1-2 or 7-8 for the RME's output? is there any difference?
 
Feb 29, 2004 at 7:40 AM Post #8 of 27
Removing resampler removes the pops. I think I will just use waveout with resampler, continuator, crossfeed, and advanced limiter.

Waveout and asio sound the same to me. I assume they should be the same, right? If they were different it would have to mean that RME's drivers have problems. Is this correct?


I can't really tell the difference between Lame extreme mp3s and FLAC's, so I think I will just stick with mp3s for now, anyway.

Using asio with resampler causes pops in some mp3s and in AAC files as well. Pretty much goes away without resampler, but isn't that a bad deal?

Asio + no resampler VS. Waveout + resampler?
 
Feb 29, 2004 at 10:49 AM Post #9 of 27
why does 88.2 khz resampling seem to work on Asio? it plays everything fine except for flac's, they still pop like crazy.
 
Feb 29, 2004 at 11:07 AM Post #10 of 27
if 44.1 to 48khz is okay, then 44.1 to 96khz should also be okay :/

i didn't try 48khz though, but i assume it would have worked fine.
 
Feb 29, 2004 at 11:58 PM Post #12 of 27
Quote:

Originally posted by Distroyed
Sounds like your cpu cant handle the higher resampling rates. Nothing you can do about that but upgrade. I know my pc cant handly 96k with the equalizer at the same time.


Uh, I highly doubt an Athlon XP1700+ is too slow for mp3 playback(I will raise foobar 2k's priority just to check). My cpu usage is pretty low. is it possible its spiking and not showing me?

Also, resampling to 88200 khz is fine, but 96000 is not, so the issue is probably not cpu speed.
 
Mar 1, 2004 at 12:06 AM Post #13 of 27
I agret that it maybe CPU power issue also. resampling takes up quite a bit of CPU and memory power.
 
Mar 1, 2004 at 3:07 AM Post #14 of 27
if I use my audigy 2, resampling to 96000 using waveout works fine for APE's and MP3's, so I think my cpu can handle it.

Its got to be an issue with my ASIO output and my RME settings. Any ideas?

Also, is 88200 better/worse than 96000?

I noticed everybody suggests 96000 for the rme and asio. Why is this?

Would 48000 be better than 88200?
 
Mar 1, 2004 at 4:07 AM Post #15 of 27
Here is what I do:

FLAC & alt-preset standard MP3->foobar2k v.80->24-bit padded to 32 bit->ASIO priority set to high (not time critical)->96000 resampling (without slow mode - try turning it off too, I noticed no difference and less erratic behavior)->advanced limiter

I think the two key things for you to try if you haven't done so already are:

1. set the ASIO to high and NOT time critical.

2. DO NOT check the "slow mode" in the resampling options - its performance enhancements are negligable above 44100Hz anyway (read the blurb on the bottom of the resampling menu). I noticed no audible difference at all but an increase in performance when the slow mode was turned off.

After you try these settings THEN start to worry.
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