[earnmyturns]

 
Jitter shouldn't be an issue. USB, however, doesn't send a continuous signal, but is far more complex. Supposedly all the negotiation between components causes the USB chips to generate noise, which can leak into the DAC and cause distortion, hence why audiophile players that hog the USB connection are supposed to improve the sound quality.

While electrical and protocol protocol details are very different between asynchronous audio USB and S/PDIF, both can introduce jitter. In the S/PDIF case, the ideal square wave signal put out by the sender gets munged in transmission, affecting leading edge detection and thus timing. For accurate D>A, you want samples to arrive to the DAC at exactly the right time. In my ideal world, no matter whether the digital input is USB, S/PDIF, AES/SBU, or whatever, it would be buffered and reissued to the D>A proper synchronously with a very accurate clock. I've read a lot of gear descriptions for DACs and for USB>S/PDIF or USB>/AES/SBU at many price points, but they typically do not provide precise details on the different data paths (for instance, on how clock is handled in USB vs S/PDIF for DACs that accept both).

 

[earnmyturns]

  I was reading this review, it goes into some details about the different input impressions he had, within the realm of comparison to an MSB analog. 
 
He said USB was the closest he got to what the MSB was doing, while bnc/rca softened the signal (paraphrasing).
With the regeneration system the Ygg has built in, all of this input talk is the highest grade of "take it with a grain of salt" imo. 

Reading a bit in between the lines all that has been written here about the Ygg, its USB receiver stage seems to do a lot more to recreate an accurate clock than its S/PDIF stages, which seem to rely on the source clock and would thus be subject to leading edge detection jitter at the Ygg end (I'm happy to be corrected, and with decent cables that might be immaterial anyway).

 

[steved21]

Earnmyturns
 
Question then,, based on the info I understand from your last two posts, do you think its better to feed my Yaggi to a Wyred via USB to the usb  input of my Yaggi  OR USB converter to  the AES / EBU impute on my Yaggi?   OR  ( yes third option)  Ethernet to a SimAudio Moon Mind 180  then out via AES /EBU to my Yaggi input, which takes USB out of the equation altogether.  The Mind180 would then make me use the Moon interface which is quite  good but it takes me away from Audirvana, I'm 6 of one half dozen of the other on the interface.
Thanks, Steve

 

[earnmyturns]

 
Earnmyturns
 
Question then,, based on the info I understand from your last two posts, do you think its better to feed my Yaggi to a Wyred via USB to the usb  input of my Yaggi  OR USB converter to  the AES / EBU impute on my Yaggi?   OR  ( yes third option)  Ethernet to a SimAudio Moon Mind 180  then out via AES /EBU to my Yaggi input, which takes USB out of the equation altogether.  The Mind180 would then make me use the Moon interface which is quite  good but it takes me away from Audirvana, I'm 6 of one half dozen of the other on the interface.
Thanks, Steve

We won't really know until there's more precise info on how various digital inputs are handled by the DAC. From what Schiit has published, the Ygg's USB input has specific provisions for re-clocking that seem to be more sophisticated than what their other multibit DACs have. It's not so clear what the S/PDIF or AES/EBU's inputs do to correct any timing issues from the source, or whether those matter with reasonable sources and cables. Unfortunately, there are so many variables here, and so many unknowns, that close listening is the only option. Which, unfortunately, is very difficult to achieve because very few if any of us have access to a complete inventory of devices and cables to do the needed side-by-side comparisons. Given all this, my positive experiences so far with Schiit gear and advice would make me choose, other things being equal, whatever input method they'd recommend for each of their DACs. I don't have a Ygg (yet?) because it's too bulky for my headphone setup and I don't have (yet?) a speaker amp with balanced analog input, but if I had one, I'd start with USB and then compare with alternative inputs. For my Bifrost Multibit I use a Bel Canto mLink USB>S/PDIF converter with a good coax cable because 1) it sounds a bit better to me; 2) I had that gear around anyway for a comparison; and 3) my USB sources are relatively cheap Linux-based UPnP/DLNA streamers with so-so power supplies. 

 

[Xcalibur255]

   
You shouldn't have to switch sample rates with any sound card as long as your playback application can take exclusive control of your playback device.  Even on-board S/PDIF will follow the source file's sampling rate in applications like foobar2000, JRiver, etc in exclusive mode.  The "default format" bit depth/sample rate setting in the control panel is for shared mode sounds only.

 
This isn't how my sound card works (which is a rather old HT Omega Claro+ for the record).  For PCM output you have to specify a sampling rate in the driver control panel, and the driver enforces that sampling rate globally.  So if I want to play back something at 96khz I have to go into that control panel and switch it, and vise versa when going back to 44.1 files.
 
I have spent hours googling information on this subject and what I've come across led me to believe this was the norm, but it's hard to find any actual clear discussion on the subject to begin with.  Unless I'm just doing a poor job of looking.
 
When Yggy comes in a few days things will get complicated more so because using USB means I have to change the default audio output device in Windows whenever I want to use my other audio devices that aren't going to be connected to the DAC.  Windows really wasn't made to support a multitude of audio gear all at once I'm finding out.
 
If anybody could point me to a specific make/model of soundcard currently in production that has a driver setting which simply passes the native sample rate of the material being played I'd greatly appreciate it.

 

[SeanS]

   
This isn't how my sound card works (which is a rather old HT Omega Claro+ for the record).  For PCM output you have to specify a sampling rate in the driver control panel, and the driver enforces that sampling rate globally.  So if I want to play back something at 96khz I have to go into that control panel and switch it, and vise versa when going back to 44.1 files.

 
The output rate you choose at the driver control panel is for Windows mixer.  You can avoid having Windows mixer send data to your DAC by choosing another audio device, such as your internal/motherboard audio, as the default in the control panel.  Of course, you loose all Windows sounds through the DAC.
 
At this point you run a playback app that lets you choose the audio device.  I run Foobar2000 in Kernel streaming mode to get bit-perfect output to the USB DAC.

 

[Tbone289]

   
The output rate you choose at the driver control panel is for Windows mixer.  You can avoid having Windows mixer send data to your DAC by choosing another audio device, such as your internal/motherboard audio, as the default in the control panel.  Of course, you loose all Windows sounds through the DAC.
 
At this point you run a playback app that lets you choose the audio device.  I run Foobar2000 in Kernel streaming mode to get bit-perfect output to the USB DAC.

 
Sean is correct, but I feel this needs some clarifying...
 
The depth and sample rate in the playback device properties is for "shared mode".  You should be running your playback application in "exclusive mode" by checking "allow applications to take exclusive control of this device" on the advanced tab of the output device.  "Give exclusive mode applications priority" also should be checked.  Choose your windows default device as the one you want windows sounds to output to.  The output device selected in your player application (foobar, JRiver, etc.) is independent of the default device, and will not be affected by the depth & sample rate you choose for "shared mode".
 
If you set this up correctly, you should not have to change bit depth and sample rate in the control panel, regardless of how old your sound card is.
 
I personally have my DAC as the default device for windows but, since I have "allow applications to take exclusive control of this device" checked, windows will not output to the DAC while foobar is playing, because foobar has exclusive control of the DAC.  Foobar will change bit rate and depth to the DAC as needed, with no intervention.

 

[Xcalibur255]

   
The output rate you choose at the driver control panel is for Windows mixer.  You can avoid having Windows mixer send data to your DAC by choosing another audio device, such as your internal/motherboard audio, as the default in the control panel.  Of course, you loose all Windows sounds through the DAC.
 
At this point you run a playback app that lets you choose the audio device.  I run Foobar2000 in Kernel streaming mode to get bit-perfect output to the USB DAC.

 
I'm using ASIO output in JRiver.  I understand what you're saying but in my setup the C-media control panel seems to throw a blanket over everything else.  Maybe because the ASIO solution is also provided by C-media it still slaves it to the driver settings.
 
This soundcard is actually my only audio device, there is no on-board audio on my particular motherboard (rare, I know). 
 
I'll be able to test a few things once my Yggdrasil arrives.  Using USB will place a second audio device in the system after all.

 

[Xcalibur255]

   
Sean is correct, but I feel this needs some clarifying...
 
The depth and sample rate in the playback device properties is for "shared mode".  You should be running your playback application in "exclusive mode" by checking "allow applications to take exclusive control of this device" on the advanced tab of the output device.  "Give exclusive mode applications priority" also should be checked.  Choose your windows default device as the one you want windows sounds to output to.  The output device selected in your player application (foobar, JRiver, etc.) is independent of the default device, and will not be affected by the depth & sample rate you choose for "shared mode".
 
If you set this up correctly, you should not have to change bit depth and sample rate in the control panel, regardless of how old your sound card is.
 
I personally have my DAC as the default device for windows but, since I have "allow applications to take exclusive control of this device" checked, windows will not output to the DAC while foobar is playing, because foobar has exclusive control of the DAC.  Foobar will change bit rate and depth to the DAC as needed, with no intervention.

 
Yep my understanding of all of this is solid, and this is how everything is configured.  These C-media 8788 based cards are installed as DMA devices if I'm not mistaken.  Their driver settings seem to control things above the level of the windows mixer.  One problem I've always had is not knowing what the DAC is actually receiving (it has no sample rate indicators), but my ears hear a clear difference when I play back a 96k file after switching the PCM output rate to 96k in the soundcard driver.  If I leave the setting at 44.1 it actually sounds a little worse what I expect a regular 44.1 file to sound like because of the downsampling the card is doing to the audio.

 

[Tbone289]

Well, I don't blame you for shopping for a new sound card then.  There is certainly no way I would put up with that one.

 

[StefanJK]

  Well, I don't blame you for shopping for a new sound card then.  There is certainly no way I would put up with that one.

Yep, that sounds crazy. nothing you should put up with.  I was going to weight in before, but I didn't.  I had a HT Omega Claro+ soundcard a while back, but go rid of it, but not for this reason.  My memory is getting too vague...got replaced by Gungnir.  

 

[Xcalibur255]

  Well, I don't blame you for shopping for a new sound card then.  There is certainly no way I would put up with that one.

 
It's been a pain ever since I started accumulating hi-res tracks in my library, but I figured with my luck I'd put a new card in and still have the same problem so I've never done anything with it.  I would have thought that in 2016 somebody would make a product that explicity states it can pass audio bit perfect via a pass through setting, but I haven't been able to find one.  The target market seems to be gamers and a heap of simulated surround sound modes for headsets.
 
So, crossing my fingers that USB on Yggy still sounds good for me despite everybody claiming it's the worst input to use because any other solution seems more messy right now.

 

[claud W]

Two weeks, my ass. "unexpected production delays"

 

[Astral Abyss]

That's fine... we need some metal issues to crop up that the only fix for is black powder coating...  

 

[gefski]

All this talk about which input is best can drive a person crazy.

I went hunting for a PC sound card that isn't crazy expensive (read:  a professional product) which is capable of passing the native sampling rate of the source being played as opposed to jumping into the control panel and changing the output sampling setting manually every time the need arises.  Apparently this is just a completely alien concept because I can't find one that can do it.  I was hoping to get on board with using the coax input this way, but having to manually change sampling rates each time you change between a high-res and redbook song is a bit of a usability snag.

So I guess USB it is.

I've abandoned PCs because of USB driver issues.

But for those that would like to go soundcard, I am recalling that in 2013(?) Absolute Sound really liked the ESI juli@ soundcard. Just a thought.