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[Johnny Stork]

Running the latest release on Windows 7, Asus Xonar Essence STX and the Asus ASIO drivers. But will often studder requiring a restart of MC. Odd time will also lockup. Suggestion for options to stabilize or possibly try different ASIO drivers like the ASIO4ALL, or that "universal" one (forgot name) I think I saw mentioned somewhere?

[kstuart]

ASIO4ALL is a universal one.  I've had good success using it with MC18 (although I don't have a Xonar DAC).

It simply uses the ASIO standard to connect to the player software and then uses the low level kernel streaming interface Windows to bypass the normal Windows audio software and connect to the DAC (it's the same kernel streaming interface that WASAPI uses to connect to hardware).

If you do try it, look for the latest ASIO4ALL - V2.11 Beta 1 (which may take some work as the author has apparently lost control of the web site).

Of course, you can also try the two WASAPI drivers included in MC18.

[Johnny Stork]

Thanks kindly kstuart! I actually found the unified driver so have that running now and will see if it's any better and if not, will go back to the ASIO4ALL which I also used in the past.

Cheers and thanks again for your help/suggestion.

[mwillems]

Thanks kindly kstuart! I actually found the unified driver so have that running now and will see if it's any better and if not, will go back to the ASIO4ALL which I also used in the past.

Cheers and thanks again for your help/suggestion.

I have an Asus Essence ST (similar but not the same as the STX), and the stock drivers were unusably bad.  I use the Unified Drivers and have had no trouble since installing them. 

Some others have had trouble when they installed the unified driver package that included the Asus Control Panel.  If you have any trouble with them, try uninstalling and reinstalling the low latency option that bypasses the Asus Xonar Control Panel.  That version is much more stable and performs much better (for me and the other folks I've talked to about it). 

[Mikkel]

@Johnny Stork: If you have to abandon ASIO (like me :-/) then there is not much point in running ASIO4ALL. I assume you would like the card to switch sample rate and/or bit-rate depending on the source. In that case ASIO4ALL won't help you since it cannot bypass the ASUS mixer (at least it does not on my Essence ST card). So either use ASIO or just switch to Wasapi Event Style.

If you want to use ASIO4ALL anyway remember to untick the "always resample to 48khz" (or whatever the name).

As a last resort you could do as me: set the ASUS control panel to some desired sample rate (I set it to 192), use Wasapi Event Style, and ask JRiver to output to whatever bit-depth you prefer. No harm will be done to the signal reaching your ears :-).


Best regards,
Mikkel

[kstuart]

mikkel wrote: 'In that case ASIO4ALL won't help you since it cannot bypass the ASUS mixer (at least it does not on my Essence ST card)."

Unless I am missing something, ASIO4ALL does not go through any ASUS software.  You might have to uninstall the ASUS software to get it all out of the signal path.

MC18->ASIO output driver->ASIO4ALL->Windows kernel streaming intefeace->WDM drivers->DAC

[mwillems]

mikkel wrote: 'In that case ASIO4ALL won't help you since it cannot bypass the ASUS mixer (at least it does not on my Essence ST card)."

Unless I am missing something, ASIO4ALL does not go through any ASUS software.  You might have to uninstall the ASUS software to get it all out of the signal path.

MC18->ASIO output driver->ASIO4ALL->Windows kernel streaming intefeace->WDM drivers->DAC

The Asus ST automatically resamples all incoming sound to the sample rate set in its control panel (or in the windows playback settings) in all output formats but native ASIO.  With ASIO, a software package (like JRiver) can automatically change the sample rate of the card dynamically on a track by track basis, but the card will not allow automatic changes to its sample rate in any other output format (like WASAPI or kernel streaming). There's an entry about it on the JRiver wiki: http://wiki.jriver.com/index.php/Audio_Output_Modes "The only way to prevent a Creative Labs X-Fi based card or Asus Xonar card from resampling all incoming audio is to use ASIO and the driver that came with the card. With these cards, WASAPI exclusive will not change the master clock of the card."  

ASIO4All does not allow for this automatic sample rate switching with any of the Asus Xonar cards I've used (Mikkel noted this above), which makes sense because it's not actually an ASIO interface on the hardware side.  This can be confirmed in a number of ways, but the easiest way is that many Asus Xonars make an audible relay click when changing sample rates.  ASIO4All does not produce the clicks, native ASIO does.  

So you're stuck resampling everything if you can't use the native ASIO drivers.  Several of the Creative DAC cards work similarly, as does at least one M-Audio card I've personally tested, and I've read reports on these forums that some USB DACs work the same way.  If you want to automatically change sample rates with many DACs, you need to use their native ASIO drivers. Otherwise those cards tend to lock in a sample rate and resample everything

You can set the sample rate high, but you're still resampling everything before playback (either in JRiver or at the card itself).

[Johnny Stork]

Thank you to everyone who has responded, you have all been VERY helpful! So let me see if I have understood everything correctly. Of course my ultimate goal is the highest sound quality possible. I believe I do know the difference, have more than one "serious" audio system and have the empty bank accounts to prove it!

The computer running the Asus ST CAN use a Musical Fidelity up-sampling DAC, but I have decided not to use it since it only supports 24/96 in/out and so it is relegated to my headphone system. All audio and DAC will be handled by the computer/ST.

I have a decent collection of high-bit-rate FLAC's, right up to 24/192 so getting bit-perfect processing with minimal re-sampling is desirable, if that is the best route to go.

Note: I am currently using the Xonar Unified Drivers, 1815 v 1.64 beta, in low latency mode.

"The Asus ST automatically re-samples all incoming sound to the sample rate set in its control panel"

(1) Is this in reference to the Windows setting shown in the screenshot? If so, which would you say does a better job of re/up-sampling? The Asus ST card itself, or MC, if I want to re/up-sample everything NOT 24/192 to 24/192?

(2) Although I have not done any serious A/B testing myself, would you say that re/up-sampling anything NOT 24/192 to 24/192 (either in the ST or MC) improves, or degrades sound quality?

(3) If I have set upsampling in the Windows settings (screenshot) to be done by the Asus ST, does that mean I should DISABLE output formating in MC? It looks like I am re/up-sampling twice based on the screenshots? Once by the ST and then again by MC?

(4) I suppose my basic/main questions are about the "best" or suggested way to get the best sound quality possible, with no external DAC, using the Asus ST with lossless FLAC's ranging from 16/44 to 24/192?

[mwillems]

Thank you to everyone who has responded, you have all been VERY helpful! So let me see if I have understood everything correctly. Of course my ultimate goal is the highest sound quality possible. I believe I do know the difference, have more than one "serious" audio system and have the empty bank accounts to prove it!

The computer running the Asus ST CAN use a Musical Fidelity up-sampling DAC, but I have decided not to use it since it only supports 24/96 in/out and so it is relegated to my headphone system. All audio and DAC will be handled by the computer/ST.

I have a decent collection of high-bit-rate FLAC's, right up to 24/192 so getting bit-perfect processing with minimal re-sampling is desirable, if that is the best route to go.

Note: I am currently using the Xonar Unified Drivers, 1815 v 1.64 beta, in low latency mode.

"The Asus ST automatically re-samples all incoming sound to the sample rate set in its control panel"

(1) Is this in reference to the Windows setting shown in the screenshot? If so, which would you say does a better job of re/up-sampling? The Asus ST card itself, or MC, if I want to re/up-sample everything NOT 24/192 to 24/192?

(2) Although I have not done any serious A/B testing myself, would you say that re/up-sampling anything NOT 24/192 to 24/192 (either in the ST or MC) improves, or degrades sound quality?

(3) If I have set upsampling in the Windows settings (screenshot) to be done by the Asus ST, does that mean I should DISABLE output formating in MC? It looks like I am re/up-sampling twice based on the screenshots? Once by the ST and then again by MC?

(4) I suppose my basic/main questions are about the "best" or suggested way to get the best sound quality possible, with no external DAC, using the Asus ST with lossless FLAC's ranging from 16/44 to 24/192?

If you're using the Unified Asus Drivers *and* are using the ASIO output mode in JRiver (under audio settings), the Asus ST will automatically change it's clock to match the sample rate of your recordings within the limitations of the card.  That's the advantage of using the ASIO driver.  So if you use the ASIO output mode, the card will not be resampling anything (unless the card doesn't support the sample rate in question).

You appear to have JRiver set to upsample everything to 192 KHz, which I would not recommend if you are using the ASIO output mode setting.  I would recommend setting your output format to "no change" for all listed sample rates.  Then JRiver will output a sound file at the source sample rate, and the ASIO output mode will automatically change the card's sample rate on the fly (ignoring the windows setting).

If you are not using the ASIO output mode, then the situation is different.  Then the card is resampling everything to the rate set in the windows control panel.  In that case, my understanding is that JRiver's upsampling is superior to the card's upsampling, so you should set output format as you currently have it.

Does that make sense?

[Mikkel]

Thank you to everyone who has responded, you have all been VERY helpful! So let me see if I have understood everything correctly. Of course my ultimate goal is the highest sound quality possible. I believe I do know the difference, have more than one "serious" audio system and have the empty bank accounts to prove it!

The computer running the Asus ST CAN use a Musical Fidelity up-sampling DAC, but I have decided not to use it since it only supports 24/96 in/out and so it is relegated to my headphone system. All audio and DAC will be handled by the computer/ST.

I have a decent collection of high-bit-rate FLAC's, right up to 24/192 so getting bit-perfect processing with minimal re-sampling is desirable, if that is the best route to go.

Note: I am currently using the Xonar Unified Drivers, 1815 v 1.64 beta, in low latency mode.

"The Asus ST automatically re-samples all incoming sound to the sample rate set in its control panel"

(1) Is this in reference to the Windows setting shown in the screenshot? If so, which would you say does a better job of re/up-sampling? The Asus ST card itself, or MC, if I want to re/up-sample everything NOT 24/192 to 24/192?

(2) Although I have not done any serious A/B testing myself, would you say that re/up-sampling anything NOT 24/192 to 24/192 (either in the ST or MC) improves, or degrades sound quality?

(3) If I have set upsampling in the Windows settings (screenshot) to be done by the Asus ST, does that mean I should DISABLE output formating in MC? It looks like I am re/up-sampling twice based on the screenshots? Once by the ST and then again by MC?

(4) I suppose my basic/main questions are about the "best" or suggested way to get the best sound quality possible, with no external DAC, using the Asus ST with lossless FLAC's ranging from 16/44 to 24/192?

1. If you use Wasapi Event Style (exclusive), kernel streaming, ASIO4ALL or ASIO it will bypass the windows mixer. I don't know the windows resampler but the ASUS hardware resampler works fine (as do JRiver). One of a few reasons to use ASIO is that you don't have to worry if you *think* you *probably* *just may* have heard a difference :-). As mentioned I set JRiver to output 24 bit and the ASUS panel to 192khz. From listening tests I may just as well have set it to 16/44 but well, it doesn't harm the original source.

2. Converting to 24bit raises the noise floor of the hardware. It doesn't harm anything, it does not do anything perceptibly good either. Resampling won't harm the original audio either. No content above what was originally recorded is added. If in doubt do an off-line upsampling and compare the files in Audacity (or some other audio software).

3. Leave JRiver to output the original sample rate. Otherwise JRiver will do software upsampling which there is no need for (it takes CPU resources). It won't do any harm, though, since it will just feed the audio card with a 192khz sample rate which the audio card then doesn't need to resample.

4. There may be exceptions but any solution will suit you as long as the resampler is of good quality (JRiver is and the ASUS is). I have been worried about this myself. Since then I have investigated the matter quite thoroughly and the only reason to use ASIO (if you ask me) is that you don't have to worry whether or not you hear a difference. That can be quite enough reason to use ASIO, of course :-).


Best regards,
Mikkel

[Johnny Stork]

Thanks again for all your wonderful detailed and clear suggestions, and explanations fellas. very helpful and I have made the suggested changes.