I'm wondering which sampling rate is better for storing audio files in the best quality, 44.1Khz WAV or 48Khz WAV?
I've searched countless times but everytime it is either one or the other, no certainty.
So does anyone here have any insight on the pros and cons of each one?

DVDs and some games use 48khz, CDs use 44.1khz. It depends on what audio you're storing. If the standard quality of the audio you're storing is 44.1khz, then there's no need to store it at 48khz, as there is no quality gained with the upgrade, you're just forcing it to be something that's not necessary. Wav doesn't lose sound quality. Same goes for games that use 32khz or lower, storing it in its original quality is always best.

basically what jessie said...

check what the source file is and keep it the same. i wouldnt recommend storing in wav though because of filesize. lots of smaller lossless codecs for that...

What they said. Keep it at the original sample rate.

If you actually have the choice (such as when playing emulated game music: GBS, GSF, USF, SPC and all that), i say go for 44.1, since it's the only sample rate you can burn as standard audio CDs.

basically what jessie said...

check what the source file is and keep it the same. i wouldnt recommend storing in wav though because of filesize. lots of smaller lossless codecs for that...

u can convert wav to mp3 or whatever format u perfer later.

You can do the same with FLAC, but it's a lot smaller than WAV for storing purposes.

i vote for keeping the original sample rate. for archiving.

if you need it for a CD, make sure it's 44.1.
if you make it for DVD-Audio, make sure it'ss 48.

ssrc gives the best sample rate conversion qualities.
it's open source, iirc.

not worth it so much to buy the most expensive audio editing tool based on its sample rate conversion.

check SSRC against other sample rate converters here.
SSRC Comparisons (http://anonym.to/?http://src.infinitewave.ca/)

same goes for bit depth.
if it's 24bit depth, keep it 24 bit depth.
if it's 16 bit dpeth, keep it 16 bit depth.

if you want to down convert, do some research on "dithering" and see if it applies to you.
different dithering algorithms give different results, most often based on different music styles, genres, and structure.

converting to a lossless codec is ideal. save space, keep quality.

FLAC is the best. Despite it not being supported on stand-alone players.
The compression is good, and the seek-time is excellent.

Many will say APE gives the best compression ratio and is supported on devices like iPods and other Mac stuff.
However, it's seek-time is extremely horrible. There's about a 2 second delay before it jumps to the position you want it to.
This could cause some problems for some players.
It's just down-right awful of a lossless codec.

FLAC is the way. Although it still many bugs and hasn't been updated since 20007, it's still the all-around universal lossless codec.

Though i can't name them all from memory, there are dozens of players that support FLAC, even without including those that Rockbox (http://www.rockbox.org/) has been ported to. All the Cowon and Sandisk players support it natively, for starters.

Also, what "many bugs" are you referring to? I've never run into a single one.

I've run across a number of AIFF files that couldn't convert to FLAC directly. It was the weirdest thing.
I had to convert to a different lossless format then to FLAC.
Several others I know had the same issue with AIF/F.

Cowon...I should look into this.
Right now I have Coby.

I'd bet your AIFF files were 32bit floating point files, which flac unfortunately doesn't support. Do you think it might have changed when converting to the other format?

I couldn't find a site that says what exactly FLAC's limitations are, it would be pretty useful to have a list of what bit depths and sample rates it can handle.

I don't think it's been spelled out clearly in this thread, but it's important to remember that changing the sample rate is NOT strictly speaking a lossless process.

Upsample 44.1 to 48 and back again - and you do NOT have the same audio. You have quantisation errors all over the place - you introduced them during the upsampling and then you introduced more in the downsampling.

Yet another reason why it is insane (not to mention plain destructive) to archive at any sample rate other than native.

Off the back of this, people who rip CDs at 48khz (ostensibly because it's "better" - but higher numbers are *always* better, right?) need to get repeatedly kicked in the balls.

I'd bet your AIFF files were 32bit floating point files, which flac unfortunately doesn't support. Do you think it might have changed when converting to the other format?

I couldn't find a site that says what exactly FLAC's limitations are, it would be pretty useful to have a list of what bit depths and sample rates it can handle.

Strangely, no they were actually 24bit. Several of us couldn't figure out what was happening.
:itsamystery:


I don't think it's been spelled out clearly in this thread, but it's important to remember that changing the sample rate is NOT strictly speaking a lossless process.

Upsample 44.1 to 48 and back again - and you do NOT have the same audio. You have quantisation errors all over the place - you introduced them during the upsampling and then you introduced more in the downsampling.

Yet another reason why it is insane (not to mention plain destructive) to archive at any sample rate other than native.

Off the back of this, people who rip CDs at 48khz (ostensibly because it's "better" - but higher numbers are *always* better, right?) need to get repeatedly kicked in the balls.