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S/PDIF Setting when used with external DAC?

post #1 of 18
Thread Starter 

I am trying to figure out the best way or setting connecting my HTPC (Foobar) to an external DAC (w/ ASRC upsampling). Currently, the DAC is connected to onboard S/PDIF via toslink cable. Under Windows 7's audio control panel, the output can be set from 16bit / 44.1hz to 24bit / 192hz. My music collection contains audio file in various format ranging from 16/44.1 to 24/192 but mostly in 16/44.1. Now should I have set the audio output in Windows to the maximum allowed 24/192 or 16/44.1?

 

Also, will I benefit from a dedicate sound card such as Xonar STX instead of onboard S/PDIF. Both will be connected to an external DAC via Toslink cable. Does HTPC process the audio signal at all when external DAC is used? For some reason, I felt the SQ from DX100 DAP->DAC->Amp->headphonesHTPC->DAC->Amp->headphones is not as good as Dx100 DAP->DAC->Amp->headphones for some reason.

post #2 of 18
Quote:
Originally Posted by psun786 View Post

I am trying to figure out the best way or setting connecting my HTPC (Foobar) to an external DAC (w/ ASRC upsampling). Currently, the DAC is connected to onboard S/PDIF via toslink cable. Under Windows 7's audio control panel, the output can be set from 16bit / 44.1hz to 24bit / 192hz. My music collection contains audio file in various format ranging from 16/44.1 to 24/192 but mostly in 16/44.1. Now should I have set the audio output in Windows to the maximum allowed 24/192 or 16/44.1?

 

Also, will I benefit from a dedicate sound card such as Xonar STX instead of onboard S/PDIF. Both will be connected to an external DAC via Toslink cable. Does HTPC process the audio signal at all when external DAC is used? For some reason, I felt the SQ from DX100 DAP->DAC->Amp->headphonesHTPC->DAC->Amp->headphones is not as good as Dx100 DAP->DAC->Amp->headphones for some reason.

Which external DAC and headphone amplifier are you using?

 

Not worth buying the STX sound card for a separate S/PDIF output.

I've heard that external USB-S/PDIF converters help with digital audio quality.

 

If you wanted to add Dolby Headphone surround sound for movies and gaming, then the Xonar DX or D1 sound card (used $55) would add that feature.

Just plug the external DAC into the Xonar's optical output.

post #3 of 18

there is no reason to limit your outputs.  In your output audio device settings just enable all possible outputs and each specific setting can be used when needed.  Though if you're using wasabi or KS or something like that I think all the windows settings get bypassed anyway. And no, I wouldn't waste money on another sound card you're not going to use just to get another spdif output.
 

post #4 of 18
Thread Starter 

The DAC is a PCM1792 based Hlly SMK 4, the ASRC chip inside is SRC4192. There aren't any review available for this DAC as it is a fairly new product. But it perform very close to Sony's popular CDP-XA50ES player. Sony has a better bass transparency while SMK 4's mid and treble are more pleasing. It doesn't have USB input so I am limited to either Toslink or Coaxial cable. The amp is SPL Auditor, driving k702 or q701.

 

My question is... is there any difference in term of sound quality when outputting via Toslink cable between onboard S/PDIF out and dedicated sound card S/PDIF out? I am assuming the Toslink out is unprocessed raw signal so the hardware doesn't matter, just like bitstream via HDMI, right?

 

Regarding the output format under Win 7's audio control panel, does selecting 24/192khz mean everything gets upsampled to 24/192khz before output or simply meaning the S/PDIF support audio file encoded up to 24/192khz (play music at its encoded rate)? I certainly do not want my HTPC does any kind of software upsample before output, all upsampling should be done by DAC.

 

@ PurpleAngel

I don't need the surround sound from the sound card.The HTPC is also connected to a receiver via HDMI, I use PowerDVD to bitstream PCM signal to receiver for surround sound.

 

Thank you

post #5 of 18
Thread Starter 
Quote:
Originally Posted by tme110 View Post

there is no reason to limit your outputs.  In your output audio device settings just enable all possible outputs and each specific setting can be used when needed.  Though if you're using wasabi or KS or something like that I think all the windows settings get bypassed anyway. And no, I wouldn't waste money on another sound card you're not going to use just to get another spdif output.
 

 

Thank you.

 

I noticed when I select 24/192 in audio output device setting, my DAC "auto detect" would display signal as 192k even when 16/44.1 is playing. Why would the DAC detect 192k if the audio device setting simply means all possible outputs can be used when needed?

 

Is there any setting in Foobar that allows program to bypass Win7's sample rate like Wasabi and KS?

 

Update: nvm, I found the bypass plugins


Edited by psun786 - 7/11/12 at 6:08pm
post #6 of 18

Not sure on that one though I know the reason most dacs don't display sample rates is because it doesn't work very well (I've heard the display itself impacts the DAC) and some DACs don't display the rate it's decoding but what it pulls from metadata.
 

post #7 of 18
Quote:
Originally Posted by psun786 View Post

Thank you.

I noticed when I select 24/192 in audio output device setting, my DAC "auto detect" would display signal as 192k even when 16/44.1 is playing. Why would the DAC detect 192k if the audio device setting simply means all possible outputs can be used when needed?

Is there any setting in Foobar that allows program to bypass Win7's sample rate like Wasabi and KS?

Update: nvm, I found the bypass plugins

When you set 24/192 in Windows, it will upsample to 24/192 and make everything conform to that output. If you want it to change, it has to re-lock depending on the source material (it has to change clocking, either from clock synth or crystals). Overall I'd leave it at 16/44.1 or 16/48, and not worry about it (there's a lot bigger fish to fry), but for compatability reasons you may want to switch to 24/192 or at least 24/44.1 or 24/48 to play back the higher bitrate files. It is possible to do the native playback, but again it will require the DtoA to relock whenever it changes fs or depth (which may result in audible clicks and momentary signal dropouts).

As far as quality between onboard and a soundcard, probably nothing to write home about, but the soundcard might offer additional input, decoding, or processing features that are unavailable to your onboard solution. It's more of a preference thing at that point. Do you need or want those features?
post #8 of 18
Quote:
Originally Posted by psun786 View Post

The DAC is a PCM1792 based Hlly SMK 4, the ASRC chip inside is SRC4192. There aren't any review available for this DAC as it is a fairly new product. But it perform very close to Sony's popular CDP-XA50ES player. Sony has a better bass transparency while SMK 4's mid and treble are more pleasing. It doesn't have USB input so I am limited to either Toslink or Coaxial cable. The amp is SPL Auditor, driving k702 or q701.

 

My question is... is there any difference in term of sound quality when outputting via Toslink cable between onboard S/PDIF out and dedicated sound card S/PDIF out? I am assuming the Toslink out is unprocessed raw signal so the hardware doesn't matter, just like bitstream via HDMI, right?

 

Regarding the output format under Win 7's audio control panel, does selecting 24/192khz mean everything gets upsampled to 24/192khz before output or simply meaning the S/PDIF support audio file encoded up to 24/192khz (play music at its encoded rate)? I certainly do not want my HTPC does any kind of software upsample before output, all upsampling should be done by DAC.

 

@ PurpleAngel

I don't need the surround sound from the sound card.The HTPC is also connected to a receiver via HDMI, I use PowerDVD to bitstream PCM signal to receiver for surround sound.

 

Thank you

I believe there is two types of signals that go over optical 2-channel PCM or DDL Dolby Digital Live, which I guess is up to 6 channels compressed audio.

I would think there would be no difference between outputing 2-channel PCM, over the motherboard S/PDIF Toslink optical output verses a sound cards S/PDIF Toslink optical output, in general.

Maybe if the sound card had a shield to protect itself from the electrical noise from the inside of the computer?

I believe Schiit recommends coaxial>optical>USB.

 

Also there seems to be USB-S/PDIF external converters which I guess improve the digital signal.

 

I leave my sample rate at 96khz.

post #9 of 18
Thread Starter 
Quote:
Originally Posted by obobskivich View Post

When you set 24/192 in Windows, it will upsample to 24/192 and make everything conform to that output. If you want it to change, it has to re-lock depending on the source material (it has to change clocking, either from clock synth or crystals). Overall I'd leave it at 16/44.1 or 16/48, and not worry about it (there's a lot bigger fish to fry), but for compatability reasons you may want to switch to 24/192 or at least 24/44.1 or 24/48 to play back the higher bitrate files. It is possible to do the native playback, but again it will require the DtoA to relock whenever it changes fs or depth (which may result in audible clicks and momentary signal dropouts).
As far as quality between onboard and a soundcard, probably nothing to write home about, but the soundcard might offer additional input, decoding, or processing features that are unavailable to your onboard solution. It's more of a preference thing at that point. Do you need or want those features?

Thank you very much, you answered all my questions.

Now, how can I get foobar to play files in its native rate? I tried wasapi plugin but all audio files sounded a little distorted (especially noticeable during vocal) when the plugin is active.

If it is not possible with foobar, any paid program out there that can deliver what I need? My collection includes single flac, cued ape, wav, and sacd iso.
post #10 of 18
Quote:
Originally Posted by psun786 View Post

Thank you very much, you answered all my questions.
Now, how can I get foobar to play files in its native rate? I tried wasapi plugin but all audio files sounded a little distorted (especially noticeable during vocal) when the plugin is active.
If it is not possible with foobar, any paid program out there that can deliver what I need? My collection includes single flac, cued ape, wav, and sacd iso.

Really not sure on this one - I'm not that much of a stickler about everything native all the time ("bitperfect"), and I only rip from CD to 16/44.1 anyways. redface.gif I really have no issue leaving my setup at 16/44.1 or 24/48 constantly. Sounds good to me.

I know that with X-Fi soundcards, they do hardware SRC that's lower noise than any DtoA you can buy. I know that software SRC can be equally competent, but I'm pretty sure this isn't universal (and real-time is usually where you have to trade-off performance, that was one of the big X-Fi selling points - I haven't read anything about SoundCore (Recon)'s SRC engine but I don't see any reason for Creative to go backwards since they already have the tech). I know nothing about the 8770 or 8788 or other C-Media chips and what they do in HW or not.

If you're getting distortion that's a problem, so I'd avoid whatever configuration nets that. If you just leave Windows at 16/48 (which is default),does your DAC just lock-up there and life is good?
post #11 of 18
Quote:
Originally Posted by obobskivich View Post


When you set 24/192 in Windows, it will upsample to 24/192 and make everything conform to that output. If you want it to change, it has to re-lock depending on the source material (it has to change clocking, either from clock synth or crystals). Overall I'd leave it at 16/44.1 or 16/48, and not worry about it (there's a lot bigger fish to fry), but for compatability reasons you may want to switch to 24/192 or at least 24/44.1 or 24/48 to play back the higher bitrate files. It is possible to do the native playback, but again it will require the DtoA to relock whenever it changes fs or depth (which may result in audible clicks and momentary signal dropouts).
As far as quality between onboard and a soundcard, probably nothing to write home about, but the soundcard might offer additional input, decoding, or processing features that are unavailable to your onboard solution. It's more of a preference thing at that point. Do you need or want those features?

I can say with 100% certainty that this is not happening on my computer or system.  I have all output rates  enabled and the music is leaving my computer at the level it was mastered at - both thru windows enabled music and with foobar etc.

post #12 of 18
Quote:
Originally Posted by tme110 View Post

I can say with 100% certainty that this is not happening on my computer or system.  I have all output rates  enabled and the music is leaving my computer at the level it was mastered at - both thru windows enabled music and with foobar etc.

And you can prove this how? You just said that displays on DtoA converters lie and degrade the signal, so how do you verify this?

Really, a person can go mad trying to achieve "bitperfect" - and for what?
post #13 of 18

because I have different DACs that play differet rates and react differently  with different inputs and my optical output does not work at 24/192 and is very staticy at 24/174 so I have different ways to determine exactly what bitrate and depth are playing.  And you have to know how your DACs react because my lower end DACs play anything you give them but they downsample to whatever it can play and you have no clue what that is while my higher end DACs don't do anything if it can't play the signal.
 


Edited by tme110 - 7/11/12 at 9:19pm
post #14 of 18
Quote:
Originally Posted by tme110 View Post

because I have different DACs that play differet rates and react differently  with different inputs and my optical output does not work at 24/192 and is very staticy at 24/174 so I have different ways to determine exactly what bitrate and depth are playing.  And you have to know how your DACs react because my lower end DACs play anything you give them but they downsample to whatever it can play and you have no clue what that is while my higher end DACs don't do anything if it can't play the signal.

Should I interpret this as: "because I can hear the difference" and/or "I am able to sense the difference" ? If not, you've lost me. confused_face_2.gif


If they're really re-locking with every signal (which is not impossible), you'll have an audio cut-out as it relocks to the new target fs. This is normal. On my equipment (all of which have displays) you'll usually get a "NO DATA" or "SIG UNLOCK" notice when the source changes fs during playback, and then it will update to show whatever the new input is (and if it can't handle the input you get a similar warning as above - I've never seen any of these components "tricked" by a source (either in terms of displaying the wrong fs or bit-depth, or displaying the wrong channel map, or the wrong file format, etc)). I'm not aware of any DtoA that will re-lock instantly, with no drop, and that's why SRC is helpful. Your converters then run at a single rate, and the data is just padded up or dithered down to satisfy them. If it's done right, you don't hear it, and there should be no timing errors. Like I said, it's absolutely maddening to try and make things "perfect."
post #15 of 18
Thread Starter 
Quote:
Originally Posted by tme110 View Post

I can say with 100% certainty that this is not happening on my computer or system.  I have all output rates  enabled and the music is leaving my computer at the level it was mastered at - both thru windows enabled music and with foobar etc.

I think both u and obobskivich are correct regarding the subject, but are referring different settings. There are two settings under device audio control. One is "supported format", which should display all bitrates supported by the hardware. Another one is called shared output bitrate (under advance tab).

If you have a program that bypass Win 7 mixer and can play native bitrate, your DAC will play what ever the bitrate the audio was mastered. However, foobar does not bypass Win 7 mixer by default, which requires wasapi plugin as a workaround. For some reason, I am experiencing minor distortion when wasapi is active.

I am not sure if bit-perfect sounds better in RL but I do like to have my DAC apply ASRC instead of HTPC. There should be a clear different in term of SQ between software upsampling and a decent ASRC chip. Unfortuntely, I dont have a viable way to bypass Win7 mixer to verify this.

Again, can anyone recommend me a program that play audio file in its native bitrate bypassing Win7 mixer?

Thx
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