I have used a Terratec PCI sound card until now. I have opened another thread because I require help with setting it up in Windows 7. It seems that I won't get a satisfactory solution, therefore I'm looking for alternatives (besides buying a brand new card).

My Gigabyte GA-P35C DS3R has onboard sound. I connected a coax cable from the SPDIF out of the motherboard to my surround receiver. Does this mean I achieve perfect sound quality because data is transmitted digitally? Or can there be influencing factors degrading the sound? If so, what should I look for when setting up the onboard sound? Or should I buy something new?

br
foo_on_air

I had to change the sampling frequency for my Realtek onboard sound. Default was 48khz, resampling everything.

Here's an image showing the option.

@foo_on_air

I don't think so. AFAIK, this just means that you can use an external DAC , instead of the one built-in in your sound card. Off course , if your surround receiver have a better DAC, then you'll get better sound.

When your onboard codec doesn't employ forced resampling you usually get very high quality DA conversion with external receivers. At least not worse than very good dedicated sound cards.

The DA converters in surround receivers are often of excellent quality, since the respective ICs have become quite inexpensive and the required infrastructure (e.g. stable power rails) is present in receivers, anyway.

The transmission is digitally perfect. And even if there's resampling, the audio should be darn-near perfect. Usually, you either get digital perfection or terrible sound... When something goes wrong in the digital domain, it usually goes very wrong! ...Think about your TV... In the old days, you could get a little snow, maybe a little noise in the audio, or maybe some rolling if the sync got off. Now, when something goes wrong with the digital transmission the whole picture freezes-up or gets very "blocky".

But, there is still some potential for noise getting-into the analog signal either before the ADC or after the DAC. For example, I have a DVD recorder and if I pause a DVD during playback and turn the volume way up, I can hear broadcast audio leaking through from the tuner built-into the DVD recorder!!!! I didn' texpect that since everything should be digital at that point, and there should be no analog signal path when playing a DVD... maybe a bad design.... It seems like the ADC could be switched-off.... I'm also getting some other (very low level) noise. (See this post if you're interested.)

http://www.hifisentralen.no/cgi/yabb3/YaBB...;num=1103185900

A nice little test (sorry about Norwegian language). A small group of dedicated audiophiles received a CD of the same song subject to various degradations. One of them was using the built-in soundcard of a cheap motherboard (analog), and using the cheapest signal cable for recording the output of a CD-player.

Even given generous time, and the opportunity to listen in their own setting at their own pace, the test was not able to prove perceived degradation.

-k

But it will still resample anything not natively in the SR you select, right? E.g., if you set SR to 44.1 (e.g., for CD-derived audio) won't it resample a 96kHz file (e.g. from DVD-A derived audio) down to 44.1?

Exactly.

Thats a problem with many soundcards. I've set it to 44khz and my player resamples the few tracks with different samplingrates.

That quite did the job! Two things allowed me to set the SR to a reasonable setting (although I still don't have all options):
1. Installed beta driver for Vista
2. Now I can choose between 24 bit and either 44,1 / 48 / 96 kHz (in the device properties of the 6fire). The bit depth will always stay on 24 bit. What does this mean in terms of sound quality when a 16 bit source is resampled to 24? At least it sounds much more like music than before

I have this surround set: Boston Acoustics Digital BA7500. I really like the sound, still after so many years because it produces a nice clear tonal range, just the mids could be more present. Should I prefer digital out via onboard ALC889A and let the DAC in the surround set do the work or go for the analogue outs of the Terratec DMX 6fire LT 24/96?

"Resampling" from 16 bits to 24 is lossless - there will be no change in quality.

That is indeed a nice little test; translating Norwegian is a pain in the butt tho..

Is there any Norwegian-speaking person who will be generous enough to translate the entire test in English? I'm sick of Google-translated gibberish texts LOL :X

WASAPI exclusive should take care of that. It bypasses the mixer/resampler. Have you tried that?

For translating Siegfried Linkwitz on his web-page recommends Systran.

http://www.systransoft.com/

When my sound card is set to 24 bit 96 kHz and I am playing back DVD multi-channel audio the surround sources are mostly 48 kHz (Dolby, DTS) so if I stick to my higher setting nothing should affect quality right? I don't know how it is with LPCM because this is a very HQ format but I don't have such a disc anyway (and there are hardly any?!).



How can I do that?

br

You should set it to exactly what the source is. Doesn't matter if it's higher or lower. It's an additional processing step that degrades quality (though not distinguishable for the most time).
The mixer bypassing is even less likely to be distinguishable (if at all). Don't know about win7 but in XP it was enough to set volume to 100% to be fine. But if it makes you feel warm and fuzzy there's nothing wrong in using WASAPI. You probably only need an appropriate output plugin for your player.

I tried WASAPI and when I set this API as output in foobar's playback options something strange happens when I chose different sampling rates (kHz) for the internal sound card clock in the Terratec driver panel. There is only one that is able to play, all other settings give the error message of being not supported by the device. Doesn't happen when I set the API to DS (direct sound?).

Seems like the driver will pass only correct sampling rate with wasapi, ie. if source is other than selected => no resampling => differing sampling rates between expected and source => error.

With DS => windows resamples the audio automagically in order to match the cards expected sampling rate.

Solution: Use DS or resample with the provided resampler to selected sampling rate