Soundcard digital pathways?, If you skip the DAC are they 'perfect'? Options

[puntloos]

I was wondering,

If you use a soundcard's digital out and tell it not to equalize/eax/otherprocessing, is the digital quality 'perfect'?

Case in point I have a professional DSP hooked up to my SB audigy's digital out. If I were to upgrade the audigy (not that I have the $$$ but for argument's sake ), would I get any improvement in the sound quality of my mpc audio playback?

Basically I hear some 'stories' from people that say 'yes' to the last question but I find it hard to believe that the digital ".wav" data from winamp would get distorted by the sound card that's hooked up to it. provided of course I don't use the onboard DAC's or some hardware DSP feature like resampling or spatialisation crap, but Im telling my computer to have all EQ's off and just play back the file.

Anyone know the answer here?

[MartinO2]

I just wonder, what kind of resampling method is used in SB cards. It is a system like DAC -> ADC, so the digital audio signal is analogized and then sampled wiht desired sampling frequency or it is system based on interpolation of input signal? I wasnt able to find this information anywhere and I really need it for my project..

Thank for any help or reference...

[eevan]

I just wonder, what kind of resampling method is used in SB cards. It is a system like DAC -> ADC, so the digital audio signal is analogized and then sampled wiht desired sampling frequency or it is system based on interpolation of input signal? I wasnt able to find this information anywhere and I really need it for my project..

Thank for any help or reference...

I think it's an interpolation method. Here's the part of kX driver's Techical Guide to Playback and Recording

DSP Resampling

It is widely known that 10k1 and 10k2-based audio cards perform audio resampling even when the incoming audio signal is 16/48. This happens due to not-so-perfect implementation of the SRC algorithms in hardware. For Audigy and Audigy2 cards (and, probably, for 10k1-based cards with chip revision >= 7 as well) the 'modified' 16/48 audio stream can be restored by using 'b2b' or 'FXBusX' plugins in the kX DSP.

The nature of the SRC bug causes all the audio data to contain partially-wrong 16th bit, thus giving you '15.5-bit playback' (and not '16bit'). The known solutions (the DSP plugins mentioned above) restore the 16th bit of the audio data, but only for the original 16/48 content. That is, if the incoming signal is not 16/48, but is, say, performed at 44100, 22050 or any other frequency, the 'b2b' and 'FXBusX' plugins, obviously, won't "correct" it, but will change the signal a little bit. That's why this correction is not turned on by the kX Audio Driver by default.

What is the difference between using b2b and FXBusX?(advanced topic)
The main difference is that FXBusX not only restores the 16th bit, but also performs sound truncation to 24 bits (optionally obtaining the least 8 bits from FXBus2 sources, if available), while the 'b2b' plugin restores the 16th bit only. From the user point of view this causes the following problem: if the truncated audio data generated by the FXBusX plugin is passed thru the Routing and Epilog plugins, it is automatically restored to '15.5bit' state (due to mathematical conversion), while the 24-bit audio data is not. So, when using FXBusX for bit-to-bit 16/48 playback, it is preferred to avoid adding any volume controls and route the output directly to the epiloglt (lite version!). The 'B2B' doesn't truncate the audio data and can be easily inserted between, say, FXBus 0,1 and the Routing.

The output signal (for instance, of the SPDIF outputs) might get truncated / rounded by hardware. This option is card-dependent (certain cards perform that, while the others don't). That's why it is recommended to check bit-to-bit playback and the particular B2B/FXBusX chain before using it (for instance, by trying the 'Direct SPDIF Recording' method and a SPDIF loopback cable).

Hope this helps.