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Thanks a lot for the new uni package. I have a question for anybody who has D2X soundcard...are you able to play 24/96 files with bass redirection? I have 5.1 setup but for music I am using only front stereo and sub. Don't like any upmixing so I am looking for somebody who can confirm that he is able to play these hi-res files on 5.1 setup like stereo + sub (2.1). I'm using foobar and KS output but still no luck in getting bass redirection with hi-res files. Everything working nice with redbook files (16/44.1) .....once hit play play on 96khz file spectrum visualization disappears in xonar d2x audio center. BTW using windows xp sp2 and as output in xonar driver is set as 5.1 Is it some kind of bug or incompatibility issue of foobar with xonar driver? Any help is welcome and appreciated. Hope for finding someone with similar problem and possible solution.
Because kernel streaming, WASAPI, ASIO etc. are low lever APIs, it will not support any effects (ie. bass redirection, equalizer etc. are not supported using this APIs)..
Only using DirectSound those effect are supported because this driver is WDM, not UAA.
UAA drivers (like Creative drivers for PCI-E Titanium cards) can use APO effects for WASAPI output.
Hi, i have installed the Unified Drivers 1.41, however when ASIO was not detected i looked in the FAQ and when they told me to replace the .inis in the WIN7 folder... I found that i have no WIN7 folder in the install. Is it meant to be like that?
[merged comments]
Disregard last post, i was installing the ASIO 1.0 patch which was buggering everything up
@Virus That is not the cause of this problem with 16/44.1 everything works even bass redirection and all through KS output or ASIO. Switching to DS out does't help still the same....
So why you not surround mixer or matrix mixer foobar dsp plugin? It produce nicer upmixing effect than default driver mode.
In foobar I also used PPHS resampler to 96khz on ultra mode ;p
@tom
thx for mentioning it, now aimp´s asio works also (haven´t tested it for some time ^^)
In AIMP 3 you can use WASAPI exclusive mode for bit exac playback.
Bug:
with Uni 1.41 32Bit playback via MusicBee dosen't work anymore; ASIO plyback too
I don't want any upmix or lame resampler just redirect bass that is all. This kind of plugins is only destroying soud. The only one I use is gapless crossfader.....
got the "cannot verify the digital signature" error on my Xonar D2X, Win7 64bit, using the 1.41 drivers 🙁
can't verify digital singature too
Windows7 pro 64bits
ASUS xonar DG
hi everyone.
I wanted to ask if is it possible to enable Dolby headphone when using the Rca outputs in the Xonar STX with this drivers.
If not is there something i can modify so i can use this as i need it badly.
Thx
Sorry , can't be done.
I modded this card with Burson + clock, now I have one of the best audio card on the planet (no kidding) !
Fernando, re: "modded Burson + clock", please provide some details as I am considering same mods to Xonar Essense STX. Part #'s & where you sourced...
Thanks for sharing
What card did you mod?
Like Ron said, please provide some details or links.
Modding this Asus sound card is the right/best thing to do to have the one of the best sound out of a computer. The same really HiFI sound as from very expensive dedicated DAC devices. The hardware used on this card is very capable to do it! (I personally use Asus STX only on stereo and for music listening)
Specially the best and easiest way to mod it with the best results, is to mod the Essence STX, or ST card. The hardware on this Asus sound card is quite good (maybe the best) in this class. It is very strange that Asus it self did not used correctly the hardware they choose it for build up this card. They are very big mistakes made by Asus designers in designing this card. At they presented afterwards their product and promoted STX/ST cards as the best HiFi card on the marked is only a big lie! They want to prove the "good" results by showing snapshots from oscilloscopes, spectrum analysers and so on. Asus only forget at ordinary consumer do not hear the sound with oscilloscopes, but with they own ears... But anyway...
The most important one have to change/mod it on this card is the clock (oscillator), and the power supply for the respectively integrated circuits. It is essential that the clock have the lowest jitter for at this card to sound exceptional. The same essential is that the right and quiet as possible power supply feed the electronics on card.
Unfortunately, to mod this Asus card in the right way is not easy at all. One have to have high enough skills in electronics, and the right tools too. The easiest way to do it is to change the clock/oscillator on the card with a better one. The way to do this is already describe it with pictures on the net. Just Google it. Note that the best and right way is to solder the oscillator output directly to the clock input pin of the CMI8788 processor. Desolder carefully and lift up this in clock pin of the CMI8788 processor. Then solder it directly to the oscillator output. The oscillator have to be glued on the card in one way previously. The both pins have to be positioned in right position, and then solder the pins together. Do not touch these components after, and be carefully to not change the oscillator position after soldering of the pins are made.
More I will try here to suggest the follow:
- remove all the "best" yellow electrolytic capacitors on this card and replace it with SMD ceramics with corresponding values. Note that near to PCM 1792 DAC on the card are 2 x 47µF capacitors (near the coper shield in the middle of the card). These have to be replaced with a equivalent value of 300µF/6v ceramic SMD. The best way to remove the yellow capacitors is not de soldering it, but broke theirs legs by moving it back and forth many times until their be separated from the legs. This take al little time, and one have to have patience to do this job.... The remaining legs on the card will help to soldering the SMD capacitors.
- they are many passive components witch Asus use at the output of the analog signal on this card. These have to be removed. They are quite small and it have to be desolder from their places right on the output (before RCA connectors). An only 10-20 ohm resistor have to be between the op amp output leg and the RCA out connector. No capacitor to GND!
- the analog part of the card (op amps) have to be feed it with +/-15v from a separate, dedicated hi performance power supply. The actually Asus design use +12v from ordinary computer power supply. Asus regulate and filter this +12v from the switching power supply of the computer, with an +12 regulator... There is now regulation at all in this way!
Then they create a kind of negative power supply on the card using an local oscillator. The result is a -11v witch feeds directly the op amps and analog processing... This is the worst design one ever have seen in the audio (HiFi) field!
- the another important change is to replace the op amps with another and better ones. Important here is the slew rate of this op amps. The highest, the better dynamics. The best way is to remove carefully all about these standard opp amps, and solder the new ones directly on the card (the same place...). The best way to remove the present component from the card is to cut it out. DO NOT DESOLDER IT! Desoldering on the card destroy the traces.
It is important that one not touch the components of the +5v line witch came from the computer power supply. This power line and its components is used internally by the processor to control the right distribution of the power supply on the card, the relays, and so on. Else, one get error messages.
- I will also suggest to take the output signal right from the headphone output. Just solder on this output a non inductive (SMD) 100-300 ohm resistor between the headphone connector pins on the card. Then make an adaptor for have 2 x RCA out from the headphone jack. Using in this way to output the signal will improve much the sound. The headphone circuit on the card has better performance than the opp amps from the standard analog output. Do not forget to set up in the Xonar panel the output from headphone (on a 300 ohm headphone is better). Sett your amplifier volume at the minimum one previously testing this way...
Finally, I have to say that I`ve used many month of work and studies to make all the necessary mods on this STX card i have. The result was at the end that It sounds fantastic! So fantastic, that I think now to change the way and use a dedicated performance DAC trough USB connection... This have to be the best of the best at the moment... But anyway, I think that one have to try to get most of this card, because the results are unbelievable.
I just wander why Asus did not all these in their designing process...
does your mod make the card now independent from shitty asus drivers? if not, then its maybe not the best card on the planet :p
The hardware mods of the Asus cards do not make the card independent from using the drivers (original or Uni ones). The software drivers make possible this card to work. The drivers is need to the CMI8788 processor and in the digital to analog conversion process to make it work accordingly.
<blockquote cite="#commentbody-2541">
semtexxl :
Thanks a lot for the new uni package. I have a question for anybody who has D2X soundcard…are you able to play 24/96 files with bass redirection? I have 5.1 setup but for music I am using only front stereo and sub. Don’t like any upmixing so I am looking for somebody who can confirm that he is able to play these hi-res files on 5.1 setup like stereo + sub (2.1). I’m using foobar and KS output but still no luck in getting bass redirection with hi-res files. Everything working nice with redbook files (16/44.1) …..once hit play play on 96khz file spectrum visualization disappears in xonar d2x audio center. BTW using windows xp sp2 and as output in xonar driver is set as 5.1 Is it some kind of bug or incompatibility issue of foobar with xonar driver? Any help is welcome and appreciated. Hope for finding someone with similar problem and possible solution.
I have not been able to make it work with anything better than 24/48…..I think it has something to do with the source you want to up-mix from.i.e. it can’t give you better than you already have from the source file. As for redirect ;try lowering it from the default 120Hz to ~80Hz
My Asus essence st can suddenly make high pitch EEEEEEEEEE noise through my headphone, and I worry that this could damage my expensive headphone someday. I just set to 44.1KHZ and HIFI mode on and this could happen anytime especially when I open a folder. Why can this annoying thing happen?
The discussion: http://vip.asus.com/forum/view.aspx?id=2009022208...
I just think maybe I can find answers here. Thanks.
Sorry to say but there isn't a solution for this problem.
You can read about some suggestions I gave to another guy how reported this, here.
Asus should be sued for this, it could damage your ear drums.
Fernando and Con, Thank you for sharing all of this information with everyone.
I will now photograph my STX and identify all component locations and space (x-y-z volume) of new modded components & assemblies, such as low jitter clock daughter board...
I have been looking at Burson's main website for ideas.
I have electronics engineering background, good soldering skills and access to several Hong Kong shops specializing in smt component repairs & upgrades - also several Burson retailers here... I now must carefully consider & plan...
Also and v. important, Many Thanks to "CarvedInside" for the ongoing Uni Driver development program, your hard work means a great deal to many Asus Users everywhere!
I could publish some picture of the mods, but I don`t know how on this forum/site. I personally did not use an Burson’s clock (witch can be quite expensive), but an oscillator I found it on Ebay (after an research on possible mods for STX on the internet). There is an so called Vanguard (gold) crystal oscillator. I has printed on it 0,3ppm. I did not measure this ppm. This is not easy at all without very special measurements tools. Anyway I just believe that what is printed on it, is very near reality, because it was just an amazing upgrade in sound quality after connected this clock. I`ve tried first with an 1ppm oscillator, so I could hear the difference....
Few more details. I`ve used dedicated shunt regulators as follow on the cards circuits: for the oscillator (+5,00v), for the analog part of PCM1792 (+5,1v), digital part of PCM 1792, CMI8788, and the rest of the other chips (3,33v), and +/- 14,5v for I/V filter and analog out (op amps). I`ve removed the standard 7805 regulator on the card.
There is an (quite big) ferrite bead on the back side (in the middle of the card), witch connect the digital part of the card to the 3,3v power supply from the computer (via motherboard PCI or PCI-e connector). This ferrite bead have to be removed to separate the card completely from the computer power supply. The +5v witch came in to the card from the computer is not used in the audio process, but it give the information to the main processor that the card is connected correctly to the computer. Filtering better the +12v from the computer, and with shunt regulators in place, this power can be used as well. I`ve used, as I said, dedicated power supply and trafo connected to the wall (controlled by a power on relay), to power the whole sound card.
The relays on the card DO NOT HAVE TO BE REMOVED. Few of these are not need it any more after the necessary mods, but they have to stay in place. Or if removed, one have to simulate with resistors their presence. The main processor control in one way at they works as programmed. Else come error messages.
Do not exced +/-15v for the analog part (out) of the card.
CarvedIns, do you know something about the asus xonar not being able to do proper automatic sample rate conversion with wasapi?
someone at the jriver forum posted a few months ago that wasapi doesnt automatically change the clock but resamples instead.
refers to wasapi exclusive
nope, don't know anything about this.
If you want to check this out, try monitoring with DPC latency checker, it should be higher on the audio files with higher sample rate. Let us now of your findings.
[corrected some mistakes here, delete the other post please]
heres what I found:
the latency is indeed different for files with different sample rates:
44.1kHz ~75-115
96kHz ~130-210
192kHz ~220-320
I tested with winamp and the adionSOFT wasapi plugin, asus drivers 6.12.8.1756 for Vista 64 (all other versions crack when browsing through explorer etc.)
when you do now switch between these kind of tracks from one to the other, then you dont get any sound for those tracks whose sample rates deviate from that track you played frst. meaning If I play 44kHz first, then you wont get sound for 96 and 192 tracks. but the DPC latency value stays the same for all 3 track, its that of the 44kHz track.
same when you begin with another track. 96 first, and you wont get sound for 44 and 192 and the latency will be as for that of the 96 track for all other tracks.
however, you can still get all tracks to play without adapting the sample rate set in the asus control center if you close and restart winamp. so you might not get the tracks with different sample rates from that track you started first to play when switching directly from one to the other. but those tracks will still play fine when you close & reopen winamp and start playback with them. latency will also correspond correctly to that track's sample rate, no matter how it is set up in the asus center.
but if you then switch back to tracks with other sample rates (on the fly, without restarting winamp), it will be again as described above: no sound and latency will be as that of the track's sample rate you played first.
one thing to mention: that plugin still seems to have some problems e.g. when you stop playback (not pausing though!) and then want to start again, winamp will crash.
still, even if that should be responsible for the no sound thing, theres still the latency value which doesnt change when changing to tracks with different sample rates on the fly.
so my guess is that, unfortunately, wasapi or wasapi in combination with the asus xonar (essence (stx)) really cannot change the sample rate on the fly while playing.
The auto sample rate was introduced in 1788 drivers I suppose, at least then it was for ASIO.
So my advice is to try the latest drivers and see if anything changes. For the cracks/pops while browsing through explorer you can :
1) disable windows sounds(if they aren't disabled).
2) increase the buffer length if you have that option somewhere
Another thing you should is to try WASAPI in foobar and see it the issues are still present, at least this way you will know if the problems come from the wasapi plug-in or the driver.
I tested with both versions now, uni 1794 and vista 6.12.8.1756 drivers. Im not sure whether there is a difference in latency between both version. at the beginning, when I switched to uni 1794 I thought yes, but after switching back to the vista drivers it looked about the same to me. if there is a difference then it might be too small for me to recognize without more intensive testing.
I used foobar and jr16 this time. the results were about the same for both progs, although overall latency values varied a bit between both. but I guess thats normal. all in all, compared to winamp, the latency gap between all 3 sample rate settings was not as high for those two progs. there was much more overlapping from one step to the other and sometimes even from 44.1 to 192.
when I switched between those different tracks while playing (44, 96, 192), without changing the rate in asus center, I had sound in all cases. so that no sound thing was just due to the buggy winamp plugin.
regarding the latency at sample rate changed, Im not 100% sure. but it seems that the overall amplitude of the latency values between all 3 tracks found when testing is higher at 44kHz than for when I compare all 3 tracks at 192kHz.
when playing the 44kHz track at 44kHz set in asus center, then minimum values of 70something can be reached. playing the same track at 192kHz set in asus center, latency wont drop below ~110. so there is a change of latency here, its higher than when playing that track at its correct sample rate.
when I play that 44kHz track at 192kHz in asus center, it also has ~approx (maybe minimal lower) the same latency as when playing the native 192kHz track at this sample rate.
when playing the 192kHz track at 44kHz instead, then the sample rate is about as high as when playing it with native 192, actually Im tempted to say it might be even a little higher than. the difference is not as high as when playing the 44kHz at wrong 192kHz though.
so what to conclude? if latency is a true indicator for the acutal sample rate of playback, it *might* really be that wasapi (with the xonar) cannot do automatic sample rate change during playback. at least the quite higher sample rate when playing 44kHz at 192kHz compared to when playing it at its native sample rate might prove this. maybe also the tiny bit higher latency of the 192kHz track playing at 44kHz compared to when playing it at its native 192kHz (so maybe downsampling uses less latency as upsampling?).