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

Maybe dbPoweramp can do that.

In dbpoweramp, near DSP effects click Add, Add DSP effects > Resample. Unless your version is very different to mine.

Personally I'd probably downsample to 44.1 (exactly half 88.2 which is mathematically good), but that's because the last time I DBT'd 96 vs 44.1 I had a hard time. Might be worth having another go.

Why do you want to resample and convert the files? I have an E17. I just set it for 96 / 24 in my control panel, it shows 96 / 24 on the E17 display, and then it plays 88.2 / 24 files just fine. (I believe it just automatically resamples them on the fly. I don't know, but it definitely plays them with no problem.)

I'd use foobar2k with the sox mod plugin and configure it to only resample (that is the correct term here) 88.2 kHz to either 48 or 96 kHz. Which one doesn't really matter unless you're concerned with inaudible frequencies above 21+ kHz in which case you should go for 96 kHz.

@StratocasterMan: Windows? Yes with DirectSound or WASAPI in shared mode this will result in on-the-fly resampling by the Windows audio engine.

The plugin mod version will only resample the sample rates you specify, so for all other non-88 kHz albums there's no processing at all.

There are two mod versions:

Quote:

Originally Posted by xnor 

I'd use foobar2k with the sox mod plugin and configure it to only resample (that is the correct term here) 88.2 kHz to either 48 or 96 kHz. Which one doesn't really matter unless you're concerned with inaudible frequencies above 21+ kHz in which case you should go for 96 kHz.

 

@StratocasterMan: Windows? Yes with DirectSound or WASAPI in shared mode this will result in on-the-fly resampling by the Windows audio engine.

Yes, Windows Vista and foobar 2000. I haven't customized foobar 2000 much at all since my initial install, so I'm pretty sure that means I'm using DirectSound. I haven't done anything to switch to WASAPI or ASIO. I guess I am using shared mode since all of my normal computer sounds such as web videos go through the E17. I'm also using Spotify and MOG.

If I'm understanding things correctly, since I haven't gone to WASAPI or ASIO, that means everything is not bit-perfect. While it seems that there is a theoretical advantage to having everything bit-perfect, I haven't tried to do that because I was afraid it would cause me more trouble than it's worth. I had doubts about whether I would be able to hear the difference and I was concerned I'd end up having some files or regular computer sounds that wouldn't play.

So, I guess the question is, should I attempt to go to WASAPI and the sox mod plugin as the original poster is doing? It seems like it might be a little difficult to configure correctly and I don't want to lose functionality with other computer sounds like streaming music or video. Most of my files are 44.1 / 16, but I do have some 96 / 24 and 88.2 / 24 and 320 kbps MP3.

Do you think it'd be wise to change my configuration?

Nothing wrong with DirectSound but I'd use the sox plugin (normal version) to resample to the sample rate that you configured in the Windows control panel. That way you get high-quality resampling (last time I checked, the Windows resampler didn't produce very good results).

Quote:

Originally Posted by xnor 

Nothing wrong with DirectSound but I'd use the sox plugin (normal version) to resample to the sample rate that you configured in the Windows control panel. That way you get high-quality resampling (last time I checked, the Windows resampler didn't produce very good results).

Thanks for the help, but now I've really confused myself!  I put the file foo_dsp_resampler.dll in my foobar components folder. When I go into my foobar configuration, I get the same options that alflying described in post #9. I don't see anything about *_mod or *_mod2.

Edit: The file is the normal version. I do see where I could download the *_mod or the *_mod 2 if I needed to install them instead of the normal version.

I can choose the 'Target samplerate'

I'm still confused about which options I should select in the foobar configuration and also in the Windows control panel. The FiiO E17 never displays 44.1, it only displays 96k or 48k. My files are usually 44.1 / 16 so it would make sense for me to choose whatever options would be best for those. I've always been confused about the best thing to do for those 44.1 files since the FiiO E17 displays 48k (or 96), but not 44.1.

I'd like to set it up so that it's best for 44.1 / 16 files on an ordinary basis, but I'd also like to know what changes to make in the foobar configuration and the Windows control panel when I do want to play the 96 / 24 or 88.2 / 24 files I have.

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...