[penaloza]

I was looking into the M-audio revolution and I hit a couple of topics which touched on upsampling. The idea of a 24/96 sound card was initially unappealing to me as I don't use my pc for recording purposes, but I read a few posts and looked at Foobar's site, and I'm intrigued but still confused. From what I understand: it's possible to upsample audio (both mp3 and redbook) on the pc (bit rate as well as sampling rate) and then if you have good enough sound card (that supports ASIO), output to a digital or analog output on your sound card? Am I missing anything here?

 

[badmonkey]

Bump ^^^^^

I'm confused here too. I've being playing with Foobar and the MAD plug-in for Quintessential. Well I can't get the MAD plug-in to work without crashing, so screw that, but Foobar runs fine either with kernel streaming or with the ASIO plugin.

Now apparently it's a good idea to use upsampling, i.e. convert 44.1KHz to 96KHz. I don't hear any difference (I don't have very good headphones but anyway), but I was wondering why this should make any difference. My logic tells me that any resampling will be harmful, and 96 is not a multiple of 44.1 so wouldn't 88.2KHz be a better figure to go to if there is some advantage? Am I confused.

It's also a good idea to go from 16 bits to 24 bits, so I'm told. This makes a little more sense to me as it's just increasing the data bandwidth and doesn't involve any resampling, but I'd still appreciate a concise explanation.

I have an M-Audio Revolution btw.

 

[xtreme4099]

aye you need better phones to hear the difference ...but yeah .. the highs are more pronounced and sharper ... i even tried the newest asio plugin for winamp2 upto 192000khz ...yeah there is a difference ...but runnin it is incredibly cpu intensive ... this on athlon xp 2100

 

[badmonkey]

Thx but I know there is a difference, I'm just trying to understand it, cos it doesn't make any theoretical sense to me.

 

[itim100]

The only explanation I can offer:

Somewhere i read that there is frequency overhead of some type; most cards that only have an actual 44.1 rate only can use about 35khz in effect. This makes no sense to me, but perhaps the extra 96 as opposed to 88.2 takes care of this overhead.

Yeah, I don't really understand the 44.1->96 thing, but the overhead is something to think about.

Where can you get these plugins for foobar? I am fine with the program as is, but it would be nice to have a better interface, perhaps with the ability to move within the song playing.

 

[xtreme4099]

they should be in the dsp manager -- listed as a resampler ..

 

[badmonkey]

Quote:

Originally posted by itim100 The only explanation I can offer: Somewhere i read that there is frequency overhead of some type; most cards that only have an actual 44.1 rate only can use about 35khz in effect. This makes no sense to me, but perhaps the extra 96 as opposed to 88.2 takes care of this overhead. Yeah, I don't really understand the 44.1->96 thing, but the overhead is something to think about. Where can you get these plugins for foobar? I am fine with the program as is, but it would be nice to have a better interface, perhaps with the ability to move within the song playing.

Thx dude I don't know about general plugins, and I don't think it supports skins to change the interface, so until I get better headphones so I can really hear the difference I'm gonna stick with Sonique, cos Foobar's interface is just crap.

 

[lan]

If you use any DSPs (attenuation, crossfeed, etc.) or replaygain, it's good to have 24bit soundcard as this editing can be outputted at higher resolution. If you have an MP3, MPC, or certain other files, they can be output as 24bit which will preserve more of the sound during decoding. Eventhough the original WAV was 16bit 44.1khz, it's taken out of the time domain.

44.1khz -> 96khz would just take more processing power compared to 88.2khz.

I hear that the analogue filter has an easier time at 88.2 and 96khz because it doesn't cut into the audible frequencies.

I think upsampling is good on a certain level of equipment. There's a point where things at 44.1 are so good it's better than upsampling.

 

[aos]

Quote:

I hear that the analogue filter has an easier time at 88.2 and 96khz because it doesn't cut into the audible frequencies.

That would be the main benefit that I can see with upsampling on a PC. However you are using one and the same filter on the sound card so the only likely difference is in the oversampling process and the digital filter that is in the DAC chip (oversampling as in what most DAC chips do; they might oversample 44kHz at 8 times, 96kHz at 4 times and 192kHz at 2 times). Other than that, you don't get the jitter reduction effect you get with hardware asynchronous upsampling as the clock you do D/A conversion with is the same at 44 and 96 and 192 kHz.