INTEL HD Audio (Realtek ALC889A) Analog line output, any sense in upsampling ?
Nov 12, 2008 at 8:54 PM Thread Starter Post #1 of 12

timmiethecat

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Hi there !

Quite new here, so please forgive anything.

I've been running this setup now for a while:

Motherboard: Gigabyte P35-DS3R with onboard sound "Intel HD Audio by Realtek ALC889A"

I am using the analog line output, which goes directly into a Pioneer stereo amp firing two of these little monsters as nearfield monitors:
Pico Lino by a germany diy loudspeaker publication called "hobby hifi"(flat response from 50Hz to nearly 20khz on-axis, no resonances)

For playback I am using foobar 2000 (0.9.5.3) which I preferebly feed with flac and other non-mp3 sources. I have routed the ouput to ASIO4ALL and it sounds just fabulous compared to the way it was before (going through the windows mixer)

Using the analog line output of the soundcard, does it make any difference if I upsample via PPHS or SSRC or SRC before output ?
I can do it, no prob, all the way up to 192000 khz. But does it make any difference to the analog signal or is it just a waste of CPU-time ?

BTW, does anybody know the internal sampling rate of that chip ? I guess all intel HD-Audio are more or less the same (7.1 channels etc....)

many thanks
 
Nov 12, 2008 at 11:28 PM Post #2 of 12
I would not consider upsampling. Stick with the original source. That means 16bits/44kHz for audio cds. You will not get better SQ in my opinion by upsampling, in fact it will not sound as good. Make sure you set bit/frequency settings in Foobar and Audio settings to match the original source of your material. If you want to go to 24/96khz, I suggest getting DVD-A music and you can get that on the net.
 
Nov 12, 2008 at 11:53 PM Post #3 of 12
I tried upsampling with the same audio chip, and couldn't hear any difference at all. You may aswell leave the bitrate/sampling rate the same as your content for better performance (although mp3s do sound better when using 24-bit as opposed to 16-bit... unless you dither, of course).
 
Nov 13, 2008 at 12:06 AM Post #4 of 12
thanks guys for your replies !!

Okay, I set everything back to no plugins at all, no pphs or src or ssrc. Don't notice any difference, probably all placebo.
Output is still running through newest Asio4all @384 samples.

One last question remains: which is the internal sampling rate of this chip and does any internal resampling happen in the chip ?

If so, I believe upsampling the signal via some plugin could be better than the upsampling within the onboard soundcard.

Is that right or am I getting too confused with it all ?
 
Nov 13, 2008 at 12:15 AM Post #5 of 12
Quote:

Originally Posted by timmiethecat /img/forum/go_quote.gif
thanks guys for your replies !!

Okay, I set everything back to no plugins at all, no pphs or src or ssrc. Don't notice any difference, probably all placebo.
Output is still running through newest Asio4all @384 samples.

One last question remains: which is the internal sampling rate of this chip and does any internal resampling happen in the chip ?

If so, I believe upsampling the signal via some plugin could be better than the upsampling within the onboard soundcard.

Is that right or am I getting too confused with it all ?



Actually, the Intel HD audio codecs do all resampling at driver level. (This means that the resampling of the audio is software-based.)
 
Nov 13, 2008 at 1:09 AM Post #6 of 12
Quote:

Originally Posted by punk_guy182 /img/forum/go_quote.gif
I would not consider upsampling. Stick with the original source. That means 16bits/44kHz for audio cds. You will not get better SQ in my opinion by upsampling, in fact it will not sound as good. Make sure you set bit/frequency settings in Foobar and Audio settings to match the original source of your material. If you want to go to 24/96khz, I suggest getting DVD-A music and you can get that on the net.


You don't have a choice. You can resample it in the player, or let it be resampled later. I'd take the route of known control within the player, myself. I find SRC's sinc medium to be universally transparent.

However, I was less than impressed by the output quality of the board (my board is GA-P35-DS3R v2.0), so I don't know how obvious it would be.

Quote:

Originally Posted by timmiethecat /img/forum/go_quote.gif
One last question remains: which is the internal sampling rate of this chip and does any internal resampling happen in the chip ?

If so, I believe upsampling the signal via some plugin could be better than the upsampling within the onboard soundcard.

Is that right or am I getting too confused with it all ?



I think it's right, mostly, but I also think Eagle_Driver is right. I can't play 44.1 directly.
 
Nov 13, 2008 at 1:36 PM Post #8 of 12
Quote:

Originally Posted by skamp /img/forum/go_quote.gif
The ALC889a supports 44.1kHz just fine, as well as 48, 88.2, 96, 176.4 and 192kHz. It doesn't resample anything.


While the above may be true for the ALC889a, there are a lot of codec chips used for many implementations of the Intel HD audio which natively supports only 48, 96 and 192 kHz. And since all of those codec chips lack hardware DSP capability, the resampling of 44.1, 88.2 and 176.4 kHz audio takes place in the drivers.
 
Nov 13, 2008 at 8:24 PM Post #11 of 12
I'll accept the possibility of it being a driver issue; but, no, I can't.
 
Nov 13, 2008 at 9:16 PM Post #12 of 12
Quote:

Originally Posted by skamp /img/forum/go_quote.gif
Also I'd be curious to know what HDA chips don't support 44.1kHz PCM.


Eh, I'm answering my own question: the Realtek ALC861 chip, for one, only supports 48/96kHz PCM. I also looked at Intel's specs and I'm stumped to learn that 44.1kHz support is not a requirement. That makes no sense to me at all, given that the large majority of music releases is sampled at that rate.
 

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