44100hz vs 48000hz - Upsampling vs Downsampling (1 Viewer)

[Jeditrav]

Hi,

not really a 'problem' as such, just looking for people's thoughts on the topic.

In the past I've been using basic Realtek AC97 motherboard audio and everything was resampled to 48000hz; didn't have a choice so just lived with it. I've recently upgraded to a discrete soundcard (Asus Xonar HDAV1.3 Deluxe) which allows the user to choose the sample rate at which audio is output - 44100, 48000, 96000, 192000hz. Unfortunately, it doesn't auto-select a sample rate based on the media playing, it will just resample anything that doesn't match the chosen sample rate.

My music library consists primarily of CD's ripped to FLAC format - these are 16bit @ 44100hz; ideally these would be output from the soundcard at 44100hz.

Many of my .avi's have audio encoded at 48000hz, and the digital tv channels here in New Zealand that don't have an AC3 soundtrack broadcast in AAC encoded at 48000hz; ideally these would be output at 48000hz.

So, in order to keep from getting up and changing the output sample rate every time I change media, I'm wondering what would be the preferred output sample rate to set and forget about, in people's opinion.

If I set it to 44100hz then I get the correct sample rate for my music, but video and tv gets downsampled - if I set it to 48000hz, then all my music gets upsampled. Ordinarily, I would say that upsampling would be preferable to downsampling, and as such I should set it to 48000hz. But, my music is 'lossless CD quality' whereas the audio on my .avi's and the AAC soundtrack on the TV broadcast is all in a 'lossy compressed' format, so do I want to 'damage' a CD track by upsampling from 44100hz to 48000hz, or further 'damage' an already compressed audio signal by downsampling from 48000hz to 44100hz?

How much 'damage' is done to a CD track by upsampling to 48000hz?

Would it be better to upsample a CD track from 44100hz to 48000hz or 96000hz?

Are there any other solutions? I output any AC3 soundtracks via bitstreaming to my receiver which then does the decoding - the AAC tv broadcast is handled by the Monogram AAC decoder, and my .avi's audio is dealt with by ffdshow (I believe, if I remember correctly); is it possible to 'capture' the audio from my video files/TV broadcast and encode into AC3 to be output to my receiver? This way, I could set the soundcard output at 44100hz, my music would be played 'perfect' and all video/TV audio would be output as an AC3 data stream to be decoded by the receiver (which can auto switch between sample rates depending on the source material; why can't soundcards do this??!!).

Anyways, thanks for listening - any thoughts on the topic would be greatly appreciated

 

[pilehave]

I would choose the downsampling...upsampling doesn't magically create more data than available already, so I would prefer listening to my music in the correct samplerate, and then let tv/video get downsampled, if the movie is good enough you won't notice

The best solution would be to dump your soundcard and buy one that outputs whatever it is fed...but I guess you don't wan't to spend even more money

 

[tourettes]

Isn't it pretty sad that the only(?) audio card that is able to bitstream the Blu-ray high end formats directly to the receiver is not able to chose the output frequency on it's own?

I might go with the upsampling to 192k choice (true, upsampling wont create any better sounding output than the original) as 44.1k to 192k upsampiling is most likely not creating as bad artefacts as the much smaller scale downsampling from 48 to 44.1. Its always much harder to upsample / downsample (resize with video) when the source and destination are close to each other.

Best way would be to ask Asus why their drivers wont allow automatic selection for the output frequency. The send best would be to test all the different choices and pick the one that sounds best to your ears.

 

[Owlsroost]

The real answer is - whatever sounds best to your ears....

That said, generally I'd agree with Tourettes - upsample to 96k or 192k - how good or bad this sounds is going to depend on the quality of the resampling software/hardware (which is where your ears become useful as a test tool ).

Keep in mind (unless you are using ASIO or Kernel Streaming to send data to the sound card) that the Windows sound mixer will resample all audio that doesn't match the output sampling rate anyway...

Tony

(I use FFDShow to upsample MP TV/Video/DVD to 96k/24bit, and have the sound card set to 96k/24bit).

 

[bazzz]

It's highly unlikely you'd be able to tell the difference, so don't worry about it. Why not get someone to change the sample rate for you (and not tell you) and see if you can notice anything.

 

[Jeditrav]

Cheers for your input guys!

I'd have to say that keeping the quality of my lossless music is more important to me than the quality loss incurred from downsampling an already compromised, compressed, tv/video soundtrack. While I don't *think* I can hear a difference in the music quality when resampling from 44100hz to 192000hz, there is that little 'niggle' in the back of my mind that's telling me that 44100 doesn't go evenly into 192000 and as such, what I'm hearing isn't 'perfect', and that does 'hinder' my enjoyment! (stoopid brain!). I've maintained the AC3 soundtrack for all my important movies, etc, so that'll just get bitstreamed and isn't affected.

I've raised the issue with Asus, it'll be interesting to hear what they have to say. For now I guess I'll just stick with 44100hz output.

Here's a query - I've heard of this ASIO plugin (which I haven't tried out yet) which provides 'bit-perfect' playback; what is it about the BASS engine, or the internal player, that isn't 'bit-perfect'? What's it doing to mess with my music?

Cheers!

 

[edterbak]

Thats just windows itself messing around with the sound. Volume controle etc. on a bit-perfect signal is not possible. So you have to change the volume by receiver only. I use WMPlayer (from ASIO plugin section) to play DTS/AC3 files @44.1+48khz bitperfect . It does this wonderfully. Thanks to symphy