SPDIF Sound Card Choice

[ChrisR1983]

Guys

I was looking at purchasing an Asus Xonar Essence SX. Initially I'd be using the onboard DAC whilst I finish building my own DAC but then I thought: if I'm to eventually end up just using the digital out on this card, is it overkill?

Money is not much of a problem so I'm not trying to save any money but I'm just thinking, is there a better option? I've taken a look at the ESI Juli@ which looks ok. What else is really good, has the ability to play decent sound alone but will be awesome connected to my WM8471 based DAC?

Sorry for the "which is better" style post but who's used what, and what's decent?

Cheers

Chris

[2KW]

Hey Chris,

Did you mean 8741 ??

Anyway, I have an Asus Zonar D2/PM and a Juli@ and I prefer the Juli@ using the SPDIF output into a modded DCX2496. Using the Juli@ you could also directly connect the I2S of the Juli@ to your DAC to bypass the SPDIF conversion. The only downside to the Juli@ for me has been that is it doesn't have a headphone out.

Cheers,
Paul.

[5th element]

Originally Posted by 2KW

Using the Juli@ you could also directly connect the I2S of the Juli@ to your DAC to bypass the SPDIF conversion.

This is precisely what I've done with my Essence ST (I assume this is the one the Chris is talking about. There's the ST and the STX, but not the SX. The ST has the reclocking low jitter chip and the expansion header, the STX does not.)

Tapping I2S from the ST is very easy, three of the pins on the expansion header carry the master, bit and LR clock. You only need to solder one wire directly to the board (for the data stream) and that can be done on the reverse side.

Previously I had an Maudio revolution 7.1, this uses the same Envy chip from VIA as the Julia@. In my setup the ST sounds significantly better then the revolution did. Whether or not this is down to the CMedia CMI8788 PCI interface chip that the Asus uses, or the fact that the ST uses some low jitter reclocking IC before the CMI8788, I don't know.

The Xonar card has better measured performance, something that is less important to some people, but the advantage of this is that the Xonar ST makes for a very useful tool for measuring the performance of the equipment you build.

I would definitely recommend that you consider placing the DAC chip/IV/line driver in the PC case close to the sound card and tap the I2S lines. Of course use a separate transformer externally, with the regulators close to the DAC chip.

You will probably be surprised at how good this will sound, I'm using a dual mono balanced output implementation with two PCM1794s. As these are hard wire controlled, it makes using them very easy.

The advantage of using a balanced output is that you can lift the ground connection between the PC and whatever it is you are going to drive.

I also assume that you are referring to the WM8741 DAC chip made by Wolfson, the WM8471 doesn't seem to exist. This chip looks like it would work well too, offering hard wire control and working with the full range of sampling frequencies, without requiring any circuit changes.

I also recommend you use the THS4031/32 for I/V conversion. If you do you will need to place a resistor in series with the feedback capacitor to decouple the output from the capacitive load, 22R will be fine. If you don't do this the 4031 isn't happy and as a result runs a lot hotter and has significantly higher distortion. We're talking -70dB, which seriously compromises the system linearity.

If you want more information with regards to doing this don't hesitate to ask.

[ChrisR1983]

Thanks for the replies.

5th Element,

Thanks for this - looks like we're on the same wavelength. As I wrote the original post at work I was taking care that my boss didn't see me and wrote it complete wrong. Yes, I meant the ST and the WM8741 :-)

I'm going to order the ST next week (after driving round Manchester to see if I can pick it up anywhere there and then) and have a play around. Might have to give you a shout when it comes to locating the I2S if that's ok?

I'm using the WM8741 evaluation board that I finally managed to get fired up last night. Not tested it yet with anything but I'm looking forward to seeing how it sounds

Cheers

Chris

[5th element]

Originally Posted by ChrisR1983

Might have to give you a shout when it comes to locating the I2S if that's ok?

Sure, no problem.

[phofman]

The Envy-based cards (Maya, Juli, Prodigy, etc.) feature two crystals for 44.1kHz and 48kHz families and generate their clock by simple frequency division, unlike the Essence cards which utilize PLL from a single crystal. The "jitter reduction chip" is just a precision PLL synthesis chip, producing the appropriate clock for the required fs.

I personally would go for pure frequency division and two crystals. Another question is PCI vs. PCIe, there are not many (any?) PCIe cards with two crystals.

[5th element]

Originally Posted by phofman

The "jitter reduction chip" is just a precision PLL synthesis chip, producing the appropriate clock for the required fs.

I don't think the jitter reduction chip is told to create different frequencies based on the sampling frequency. I do of course know that it can do this, I was browsing the data sheet a few days ago. However the rest of the cards using the Cmedia chips don't feature anything other then a simple crystal at one fixed frequency.

I was under the impression that the jitter reduction chip simply resynthesised this single frequency as a low jitter clock and that was fed into the Cmedia chip.

The rest of the clock synthesis the Cmedia chip handles itself.

I could be wrong on this and in the not too distant future I am going to take the ST out of the computer again to figure out exactly what's going on.

As far as I know this is the chip used - http://www.cirrus.com/en/pubs/proDat...2000-CP_F1.pdf.

If I can I will try and get the scope on the output to see what's going on.

[phofman]

5th element,

Fortunately we do not have to take a guess about functions of CS2000 in Essence ST since we have linux alsa drivers source code freely available.

A short analysis of git.alsa-project.org Git - alsa-kmirror.git/blob - pci/oxygen/xonar_pcm179x.c shows CS2000 registers are modified only when switching from userspace between PCM1792A oversampling of x64 and x128. Clearly that control is independent of the actual sample rate. You are right, CS2000 only provides stable single frequency clock for CMI8788 which employs its internal PLL for 44.1kHz Fs family, just as in the STX model.

Nevertheless, there is still PLL used for CD-audio formats in both models which in my eyes makes the cards less suitable for "hard core" audiophiles than the cards with two separate crystals.

[1audio]

In checking the specs for the cmedia CM8788 stuff I see the following limitations:
1) 24.576 crystal only, needs PLL or DDS technology to make the 44.1 frequency chain. A single crystal is a selling point per cmedia.
2) The drivers don't support 176.4K sample rates at all.

Its not clear from the specs but I would not be surprised to find an internal sample rate converter.

The Via chipset uses two crystals so no PLL or similar, however they left out support for 176.4. The Juli@ uses an external clock chain to create the 176.4K sample rate. It does deliver "bit perfect" for all of the standard sample rates in every test I have tried.

I would have really liked for the cmedia stuff to meet these requirements however they don't see them as a real issue at any level.

The Juli@ still seems to be the most flexible and useable mid priced card (and the only route I know with a no pll solution, even the high end cards all have PLL's).

[ChrisR1983]

Hi Guys

Sorry for hi-jacking my thread back

Well I went for the Asus card in the end. I've gotten everything set up and to be completely honest I'm very impressed. I haven't connected it to my DAC yet and everything is crammed into my computer room. I'm going to move it into my home theatre over the weekend and get a media server set up (well its a hi-fi but the only way I can get my Mrs to agree is by calling it a theatre room)

Anyway - I'm using JRiver media centre 15 as it gets quite good reviews. I've ripped a few audio CD's using MC15 to flac files. I'm outputting using ASIO. There's a nifty setting that display's ASIO settings before the audio starts playing but it says 44.1Khz 2 channel. All the settings I've changed to 24bit 192KHz. Is this normal?

As for sounds - the first track I played was take 5. It sounded great. Bass is much more prominent and the sound stage is excellent. The treble seems a bit screechy but that's the way my scan-speak 2-ways are set up.

Any ideas how I set this up or should I give up and use foobar?

Cheers

Chris