Xonar Essence ST hardcore mods
Jan 3, 2010 at 9:50 AM Thread Starter Post #1 of 8

phwatts

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Dear all,

Since the Essence ST's performance is finally good enough to compete with serious component hi-fi, I'm taking the plunge to replace my Meridian setup.

However, I think there are still a few areas where the ST can be improved, but despite hundreds of postings & threads it seems most are only going in circles with trying opamp permutations. Without treading on toes, simply spending lots of money on exotic opamps while there is so much room for improvement elsewhere is IMO similar to using sandpaper on a rough tree log. At least there aren't long discussions on how one brand of boutique solder made a quantum leap difference :wink: Don't get me wrong, I also do it, but I would prefer to leave component tweaking by ear for last.

I was wondering who might be interested in being part of it, either just by means of comments & input or actively participating.

I'm an EE specializing in digital design and multilayer PCB layout, mostly digital media interfaces for audio&video so I'm familiar with the environment - I do not consider the proposed project as over-ambitious, but it will definitely take considerable time.

As my intention is both stereo & HT, it will require the HDAV module too. However, simply swapping the PCM1796 with PCM1792 DAC's is not good enough.

I have identified four problem areas:

1) Despite the on-board regulation and "HyperGrounding", the power & grounding environment in a PC is hardly clean and a far step from component hi-fi.
2) The quality and complexity of the I-V and summing stages are low-quality and similar to sub-$500 CD players. Although the DAC's and opamps are rather good, filter & decoupling capacitors are ceramic and MKT instead of MKP.
3) Unless the expansion board connects with TDM, the digital audio appears to be daisy-chained I2S with single-ended lines; hardly a good idea given the high clock frequencies. Given the high clock accuracy, going for ASRC will (probably) not improve things though.
4) It appears that volume is adjusted either in the processor or in the DAC's by means of the SPI interface. Either way, digital attenuation is an undesirable method since it reduces dynamic range.

I therefore intend on doing the following. The exact details and extent will depend on the actual implementation as I haven't bought my ST yet and would first like to get some answers from current owners:

1) Build a proper external supply for the ST card with a 50VA toroidal transformer and regulation. This will reside within the PC chassis.
2) Disable/remove all unnecessary circuitry to reduce power consumption and ground pollution. This includes the DAC, should the headphone out not be used.
3) Add a small daughterboard that plugs into the expansion header, with iCouplers and differential line drivers (LVDS) for the digital audio signals. This board has its own high-speed breakout connector. The SDATA signal from the front channels will be tapped from the board and added here, so that all 8 channels are presented on the header. This method assumes that I2S is used for the audio, but it could be TDM. This connector now provides properly tailored audio, completely isolated, for all eight channels. The LVDS electrical format also allows a fairly long cable without loss, compared to the limitation of 3.3V single-ended TTL of the stock connection. There are neat 110ohm differential cables to choose from; CAT6 bundles could be used if really necessary.
4) Build a high-end 8-channel external DAC; residing outside the PC chassis. This will be similar to the HDAV board but with a few notable differences:

- Dedicated power supplies with transformers. This will include a properly regulated 3.3VDD, 5VCC and +-15V.
- Much better quality filter components. Not going overboard with $1300 Audio Note capacitors though! Wima MKP/FKP has worked very well in past projects.
- Volume control in the analog domain by means of TI's PGA2320. I've used it in numerous projects and it works excellent. Admittedly it's an extra opamp in the signal path, but it's a much better trade-off compared to digital attenuation.
- Proper line drivers for long cables and improved bass. Choices I've tried in the past are BUF634 or THS6012; the latter is very similar to TPA6120A2 but intended for line drivers.
- PCM1794A DAC's. The reason to use it above the PCM1792 is simple, all the settings that is currently software controlled is unnecessary and can be hard-configured, and the circuit path is much cleaner without having more noisy digital lines. The only caveat is volume control, which depends on how the ST implements it. Hopefully it's done in the DAC's via the SPI bus. In that case, a small MCU can be used to hijack the SPI bus, identify the volume control signalling and re-format it for the PGA chips. However, if attenuation is performed in the processor, a different strategy will be required. By far the easiest would be to leave all the software at max volume and build a custom volume control on the DAC itself. This is easy by means of a cheap pot acting as voltage reference with an MCU with internal ADC, which translates it into SPI for the PGA chips. I have done all of this before.

There's a choice to opt for a single large board, or rather 4x identical stacked stereo boards, each with its pros&cons. The implementation of the volume and TDM vs. I2S, and generally how much duplication will be possible will play a role in this. The increased real estate and lower production cost will allow adding more separation such as individual regulation, and dual-mono for increased dynamic range by summing the currents and using one DAC per channel. This I have also done before.

Anyways, I'm looking forward to comments. Of particular help would be current ST owners (preferably those with HDAV boards) to do some reverse-engineering in places. Simply checking connectivity with a continuity meter will already help a stack in order to determine some of the unknowns. I hope that this will allow me to avoid buying an HDAV board. The exact clocking will also determine whether the excessive signal tailoring is necessary - even though benign, buffers & line drivers add jitter so it might be that cons outweigh the pros. All the same can be done with straight TTL though.

Naturally this will not be cheap, but rather benign compared to the price of the component hi-fi it will contend with.

Regards,
Pierre
 
Jan 3, 2010 at 5:59 PM Post #2 of 8
I have no issues with software type attenuation as long as it is done in 24 bit mode or better & no truncation of 16 bit code is done before the signal reaches -48db below max setting. At this level you would never be able to hear any truncation as the softest sounds available is below the noise floor of the opamps by 24db & below the human hearing threshold by at least that much.

At that level it is better than achievable by attenuation chips which mostly cannot be direct coupled & therefore have signal losses due to the caps unless very high quality caps are used & potentiometers have miss tracking issues with low volume settings which software attenuation does not have. High quality coupling caps that have very low loss are very expensive & way too large for use on a sound card. Plus thier large size exposes the to picking up excess noise from inside the computer, not a good thing. Most step type potentiometer that use many high quality resistors instead of a single variable resistor per channel have excellent traking but too few steps to find the perfect volume.They tend to be too soft at one volume & too loud at the next.

As for power supply mods those could help but to me the noise level on this card is already exceptionally low for being in a computer. Below the threshold of hearing. They may still have a positive effect sonically though as I do find the the better the quality power supply to the DAC's the better.
 
Jan 3, 2010 at 7:28 PM Post #3 of 8
@PHwatts, I think you find many opamps tests in the main thread as those are easiest for people new to electronics. There are many areas that can be improved on virtually any audio device and they are usually in power quality, filtering and just straight component quality. A few have already tested external supplies and other have modded the cards in different ways.
Your right though, there is more sound quality to be had in this design, but it will require modification outside the realm of the average user. As for DAC communication, IIRC, I2S is used.
 
Jan 3, 2010 at 8:47 PM Post #4 of 8
@ Germanium & ROBSCIX, thanks for the comments. Indeed you are correct regarding software attenuation, but that is assuming all audio to be 16-bit. Although that is true with the majority of audio, higher bitrates are increasingly being used, and then software attenuation becomes a problem.

The PGA chips use 128 different feedback resistors on a noninverting opamp, configurable via SPI. This gives 0.5dB steps, which is more than fine enough. Especially for balanced applications this is a very good solution, and also used by Levinson IIRC as 4-gang pots simply will not work. Passive stepped attenuators like D.A.C.T. are simply too expensive.

Regarding the environment, the whole idea is for the DAC board(s) to be placed outside the PC chassis. That is why the LVDS transmission is proposed, in order to allow a longer cable. The iCouplers will isolate the grounds as well, although it's probably not necessary.

About power supplies, technically this whole project is overkill as the ST already performs superbly and few will hear the difference. However, we know that in the upper echelons of audio everything counts, and the intention of this project is simply to see how far the ST technology can be exploited without going overboard (refer again to the $1300 Audio Note capacitors...) On the other hand, most of the circuitry in my Meridian & Krell gear will also be considered overkill when analyzed, so I'm just comparing apples to apples.
 
Jan 4, 2010 at 12:20 AM Post #6 of 8
Even with 24 bit files software attenuation is fine unless you are doing mixing & mastering as then you are cutting & then boosting the signal again but even there software attenuation is best as long as it is done with high floating point precision such as 64 bit floating point. That way no truncation occures & the reboosted signal is not truncated as it would be with 24 bit integer attenuation. The losses with 24 bit files & 24bit integer software attenuation the truncation would occure below the amps ability to resolve it so it would not be an issue for normal listening. Again noise from the opamps would swamp any distortion caused by truncation & bear in mind even the opamps noise is well below our ability to hear it unless boosted to absurd levels.

I do agree there are situation in high end equipment that they were able to use hardware based attenuation with very high grade resistors & solenoid relay operation for switching those resistors that have excellent tracking & very small steps in volume but bear in mind these preamps are extremely expensive & do not offer anything better than what software attenuation offers for normal listening. Software attenuation can offer even smaller increments of attenuation than .5db. I'm talking even well below .1db in practical terms.

I will say though that the volume control in windows is not as fine as I would like it & foobars software based volume control is excellent.
 
Oct 15, 2010 at 7:41 AM Post #7 of 8
Given "..I'm an EE specializing in digital design and multilayer PCB layout, mostly digital media interfaces for audio&video so I'm familiar with the environment"  Its sounds like it would be not much more work and much more satisfying to design and build your own board from scratch. 
 
 
Oct 15, 2010 at 1:01 PM Post #8 of 8
 
It sounds like it would be not much more work and much more satisfying to design and build your own board from scratch. 
 


Exactly my thoughts...polishing a turd is a waste of time. The ST is plagued by crummy software drivers, way too many active components in its signal path and a 500ps jitter spec'ed DSP that'll ruin the SQ anyway
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