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convert 24bit flacs sampled at 88.2khz to 96k or 48k? - Page 2

post #16 of 20
Thread Starter 

Well for me, I did hear an improvement in slightly less background hiss and slightly better soundstage when I used asio/wasapi for onboard realtek sound chip, ha.  It probably didn't make much of a difference for my Xonar DG(Not that I can tell), but I do it anyways. 

 

The other factor is that I like undisturbed music, and asio/wasapi will ensure that, even if I bump into some fancy flash advertisements on web

 

it's not that hard to do actually, for foobar, download either the asio4all plugin or wasapi output support plugin, install them, and select them in the playback/output section of foobar

post #17 of 20

Thanks for the help everyone. I think I've settled on the solution that works best for me. I use my E17 to process all sounds on my computer.

 

Since most of my files are 44.1 / 16 bit, that's what I'm setting in the Windows control panel. I'm also using the normal SoX plugin and setting that to 44.1 in foobar. That should give me no extra processing or resampling most of the time.

 

For my 96 / 24 files, I'll have to change the Windows control panel to 96 / 24 and the plugin to 96000. That should also cause no resampling.

 

For my 88.2 / 24 files, I'll change the Windows control panel to 96 / 24 since there's no option for 88.2 / 24, but change the plugin target to 88200. I think that's the only time any resampling will occur...

 

Sound about right?

post #18 of 20
Quote:

Originally Posted by StratocasterMan View Post

 

For my 88.2 / 24 files, I'll change the Windows control panel to 96 / 24 since there's no option for 88.2 / 24, but change the plugin target to 88200. I think that's the only time any resampling will occur...

 

Sound about right?

 

In that case set the plugin also to 96 kHz so that you will get high quality resampling from sox instead of the Windows audio engine. :-) That's the whole point of the plugin.

post #19 of 20

Okay, thanks, xnor! biggrin.gif

post #20 of 20
Quote:
Originally Posted by xnor View Post

Here's a link to the download. The normal version resamples everything to the target sample rate, unless the track already is at the target sample rate. In that case it does nothing (same as the Windows audio engine if the configured format matches the track's format).

 

If you want to minimize processing then set the target to 44.1 kHz since most of your tracks are already at that sample rate (=> no resampling) and also set the the format in the control panel to 44.1 kHz. I don't know why the E17 doesn't display 44.1 kHz but it seems to support 24 bit input so you should choose that in the Windows control panel.

 

A completely different route would be to give the WASAPI or ASIO plugins a try, which can be found on the foobar2000.org/components page and resampling only those tracks (using mod2) that have a sampling rate that is not supported by your DAC.

 

edit: the correct way to install fb2k plugins is using File - Preferences - Components - Install...

 

Thank you so much - this was really helpful ... I was getting frustrated with foobar and 88.2 kHz sample rates on the E17 - the SoX mod2 is perfect for this and I missed the mod - I can down (or up) sample for just that sample rate (or any unsupported rates) and use native E17 functionality for the supported rates. (And I can't install other software like jriver media center on my work computer - so I am stuck (not a bad thing) with foobar).

 

Also if the E17 is set for 44.1 (either 16 or 24 bit) it always displays 16 .. this is a known display limitation only - it absolutely does support 24/44.1 (and 24/48) even if it can't display this) 

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