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Old Mar 26, 2005, 01:55 PM   #1
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exclamation kX 3538h workaround for SB0400, Audigy2 Value

Hi!

Just installed a Audigy2 value and was a bit curious...

So, there is a way to use the kX-driver, before Eugene has added full functionality for the SB0400. This test was made with the actual 3538h driver update.

1. Set SPDIF output frequency to 96000Hz (in kX-Mixer / Master page) (Thanks popej )

2. Delete DSP completely and make this setup (for example):
www.electricstart.de/pic/sb0400.gif

... the output of the MX6 is routed to all three outputs only for testing - they are independent. Saved setups can be reloded as usual.

3. Note, that the analog output signal appears on the correspondig analog plug, but only with corresponding Digital Out selected in the k2lt plug!

4. You can add all plugins you want between the src and k2lt.
Recording is possible via MME and ASIO, for HQ-device thereīs no solution yet.

Finally: A2Value runs with kX!!!

To EUGENE: Iīm ready for doing some tests related to the A2Value and new software stuff.

Greetings!
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Last edited by TravelRec.; Mar 27, 2005 at 06:10 PM.
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Old Mar 26, 2005, 02:12 PM   #2
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Travelrec... You are a stud.
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Old Mar 26, 2005, 02:25 PM Threadstarter Thread Starter   #3
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Not that easy as you think, Nappy. Iīve ordered an A2ZS, but got the SB0400 itīs quite cheaper. And before I would send it back, I rather tried it with kX, so there may be a light for all the unhappy users with A2 Value. I plan some expandation of the thing, after full support is added . Unfortunately the AD_EXT is missed, but functionality seems to be the same as like with the big version. Time for testing ...
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Old Mar 26, 2005, 02:50 PM   #4
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the image link is down... www.1asphost.com (free; 100 mb storage, 1 gb transfer/month) just a suggestion
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Old Mar 26, 2005, 02:55 PM   #5
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It works for me; here's a mirror:

http://www.naptastic.com/sb0400.gif
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Old Mar 26, 2005, 03:24 PM   #6
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so basically the spdif outs work but the analogs do not?
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Old Mar 26, 2005, 06:14 PM   #7
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Quote:
Originally Posted by TravelRec.
1. Avoid the automatic launching kX-Mixer at startup, by deleting the kX-Mixer startup value in the registry-key:
HKEY_LOCAL_MACHINE\SOFTWARE\Microsoft\Windows\Curr entVersion\Run
I think this step is not necessary, my Audigy2 Value runs without that. But you should set SPDIF Out to 96000 in kx mixer master window.
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Old Mar 26, 2005, 07:05 PM Threadstarter Thread Starter   #8
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Hmm, not tested yet, but thanks for that hint. One thing I also discovered: if I save the whole system config with "save settings" and reload it, the sound stops. If I only save the DSP-config from the DSP-Window and reload that, then all is okay. I will check tomorrow a bit around.

To Chester01: the analog outs work, but you have to set the digital outputs as the physical outputs to enable the analog outs (bit confusing, but true). Think this is related to the difference of the custom chips...

BTW: my card came without the digital module, but it has an I/O plug for 1xSPDIF in and 1x SPDIF out. I think this will be my next project ...
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Old Mar 26, 2005, 09:04 PM   #9
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Quote:
Originally Posted by TravelRec.
One thing I also discovered: if I save the whole system config with "save settings" and reload it, the sound stops. If I only save the DSP-config from the DSP-Window and reload that, then all is okay.
In my setup global save and restore works OK.
Maybe that's the reason that I dont have to start kx mixer manually? I wonder what's the difference between automatic mixer start, manual start and settings reload?
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Old Mar 27, 2005, 01:03 PM Threadstarter Thread Starter   #10
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Yeah, okay - manually start of the mixer is not necessary if you set the output SPDIF frqeuency to 96000 Hz. I can confirm that. Now the restoring of saved *.kx files works okay also for me.

News: The SB0400 [color=red]has[/color] the UDA capability - even if there is no Philips UDA-Chip soldered on the card . This means, that you can record 2 analog stereo tracks seperately at the same time (via ASIO). This feature is available with new proFX AC97 plugin, set UDA enable and set as source "Line", set as normal AC97 source "Aux" (has to be connected via molex plug adapter) and voila: 2 separate stereo tracks in cubase available .

Greetings!
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Old Mar 27, 2005, 07:18 PM   #11
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does the audigy 2 zs have uda?
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Old Mar 27, 2005, 07:22 PM   #12
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Quote:
Originally Posted by TravelRec.
News: The SB0400 [color=red]has[/color] the UDA capability - even if there is no Philips UDA-Chip soldered on the card
Actually there is UDA1361T on SB0400 :-)
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Old Mar 28, 2005, 12:22 AM   #13
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System Specs

Quote:
Originally Posted by Chester01
does the audigy 2 zs have uda?
yes, thought i haven't scoured the card to find the chip, i can activate it within kX
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Old Mar 28, 2005, 12:57 AM   #14
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Travelrec: What kind of input SNR are you getting on that card with the 4-channel imputs?
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Old Mar 28, 2005, 01:55 AM   #15
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Travelrec: great. I will release a new version for a2 tests soon

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Old Mar 29, 2005, 01:07 PM   #16
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Quote:
Originally Posted by Nappylady
Travelrec: What kind of input SNR are you getting on that card with the 4-channel imputs?
I have made some measurements, you can see results here:
http://www.anpo.republika.pl/sb0400/index.html
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Old Mar 29, 2005, 03:58 PM   #17
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*jaw drops*

You mean that card gives you 4 analog inputs with 90+dB signal to noise ratio!?! Where can I get one?!
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Old Mar 29, 2005, 05:17 PM   #18
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Quote:
Originally Posted by Nappylady
You mean that card gives you 4 analog inputs with 90+dB signal to noise ratio!?! Where can I get one?!
I think most of Audigy2 cards can do that due to flexibility of kx driver and profx plugins. You can record 2 different mono inputs (or 1 stereo) through AC97 and stereo mix of 4 inputs through UDA. Unfortunately my Audigy2 Value has only 3 input sockets - mic, line_in and aux_in :-)
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Old Mar 29, 2005, 05:32 PM   #19
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Quote:
Originally Posted by popej
I think most of Audigy2 cards can do that due to flexibility of kx driver and profx plugins. You can record 2 different mono inputs (or 1 stereo) through AC97 and stereo mix of 4 inputs through UDA. Unfortunately my Audigy2 Value has only 3 input sockets - mic, line_in and aux_in :-)
There is a forth but it is hiding.

The digital I/O is a 4-pole jack and it it yields digital I/O. If you look at the UK creative site you will find there is a digital I/O dongle for it that yields both optical and wire SPDIF both input and output.

What the connections are I don't know "mono" to RCA yields SPDIF out, that can be seen from the Audigy 2 Docs, the other two must yield both SPDIF-in and power for the wire to optical conversion.

The dongle is not released in North America and the above is only true of the Audigy Value and one other card that is mentioned on the UK site.
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Old Mar 29, 2005, 05:40 PM   #20
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Quote:
Originally Posted by popej
I have made some measurements, you can see results here:
http://www.anpo.republika.pl/sb0400/index.html
Did you see if there was any difference at 96000. The DSP as far as I know only runs at 48000 but if the ADC bypasses the DSP function as it should (using the I2S protocol??) then it may be able to input to Windows at 96000 and to the 8 channel output DAC as well.
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Old Mar 29, 2005, 07:26 PM   #21
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Wave HQ device is supposed to handle 96/24, but I'm not able to make it work. Foobar does not paly with wave HQ and RMAA behaves strangly, when set to HQ. The best I can get is playing 24/48 through FBusX and recording 16/48 from k2lt. Sending 96/24 to DSP results in resampling and playback quality is worst then with 24/48.

A2 Value, A2 Notebook and Audigy4 have different DSP than previous Audigy versions, probably something is changed in p16 part. I wonder if anyone succeeded in making wave HQ work on new chips.
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Old Mar 30, 2005, 10:31 AM Threadstarter Thread Starter   #22
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Hi there!

Made also some testing these days and I must say, that it makes no real difference in noise input whether you use AC97 or UDA input, both of them provide around -76db peak and -90 db RMS. On my Audigy 1 the UDA has a much better S/N ratio of around -83 peak and -92 RMS. For me this is audible, but for the most purposes this will not be a problem. The sound of the SB0400 UDA input is slightly better/clearer. After a long searching on the card I īve not found the UDA-chip, so I suppose, it is integrated in the custom chip. I only found the sigmatel AC97 chip and the control chip that exists on real A2 cards too and some other small logic gates. Iīve ordered some components to replicate the digital I/O, when I finished the thing, I post a schematic and an "How To Build" file.

CU!
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Old Mar 30, 2005, 12:34 PM   #23
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Quote:
Originally Posted by TravelRec.
Made also some testing these days and I must say, that it makes no real difference in noise input whether you use AC97 or UDA input, both of them provide around -76db peak and -90 db RMS. On my Audigy 1 the UDA has a much better S/N ratio of around -83 peak and -92 RMS. For me this is audible, but for the most purposes this will not be a problem. The sound of the SB0400 UDA input is slightly better/clearer.
I think UDA is limited by analog mixer on AC97. It is Sigmatel STAC9750T, which has SNR about 90dB on analog part. And its ADC gives up to 10 times more distortion then UDA. Probably this is a low cost part even for Sigmatel :-)

Quote:
Originally Posted by TravelRec.
After a long searching on the card I īve not found the UDA-chip, so I suppose, it is integrated in the custom chip.
See this picture http://myhard.com/imagesnew/hardware/0107/VALUE/6.jpg .
I think UDA is this small chip on the left.
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Old Mar 30, 2005, 01:56 PM   #24
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Quote:
Originally Posted by popej
I think UDA is limited by analog mixer on AC97. It is Sigmatel STAC9750T, which has SNR about 90dB on analog part. And its ADC gives up to 10 times more distortion then UDA. Probably this is a low cost part even for Sigmatel :-)



See this picture http://myhard.com/imagesnew/hardware/0107/VALUE/6.jpg .
I think UDA is this small chip on the left.
The UDA is chip U9 it is a Philips 1361T and on Mine at any rate is well labelled. The SIgmatel STAC9750T is chip U7.

I do not know why you suggest that the AC97 mixer would be involved. I would suppose from what has been documented in the KX Project that Philips I2S protocol is used by the 1361T to communicate to the CAT0108 that is U19.

One point that I am curious about is whether one can get on and off U19 without dropping from 96000 to the 48000 at which the DSP part of the chip runs.

If that can be done then 96000 can be run through the card from the UDA input to Windows, to U8 (the 8 channel DAC) and to the SPDIF-out as well, the SPDIF-in may well participate.

If that is so doing mixing either with the AC97 or in the DSP both have more limitations than this part of the hardware.
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Old Mar 30, 2005, 04:56 PM   #25
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Quote:
Originally Posted by John Lodge
I do not know why you suggest that the AC97 mixer would be involved. I would suppose from what has been documented in the KX Project that Philips I2S protocol is used by the 1361T to communicate to the CAT0108 that is U19.
I'm talking about analog part. Line-in goes directly to AC97 and through its analog mixer to analog input of UDA. There is no way to bypass AC97 without hardware mod (which seems to be quit simple :-)
Quote:
Originally Posted by John Lodge
One point that I am curious about is whether one can get on and off U19 without dropping from 96000 to the 48000 at which the DSP part of the chip runs.
I hope this is possible, but not with current kx drivers.
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Old Mar 30, 2005, 06:09 PM   #26
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Quote:
Originally Posted by popej
I'm talking about analog part. Line-in goes directly to AC97 and through its analog mixer to analog input of UDA. There is no way to bypass AC97 without hardware mod (which seems to be quit simple .
Gotya! I can see the noise floor difference between the AC97 ADC and the UDA ADC but now I look as the STAC9750 specs on the analog side the noise floor is undoubtedly being limited by the AC97 mixer or it would be a little lower.

Tisk!

This means that it is important to line up the analog gains by driving the input to near its 1Vrms limit (or 2Vrms if the AC97 has attenuation at its input).

...John
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Old Mar 30, 2005, 07:14 PM   #27
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I made RMAA measurements with Creative drivers. Definitely UDA and output DAC can work at 96kHz without any resampling. Results are here:
http://www.anpo.republika.pl/sb0400/index.html
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Old Mar 31, 2005, 07:42 AM Threadstarter Thread Starter   #28
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Hmm, very interesting, needs to be confirmed anyhow - I canīt believe that there is no resampling, the signal has to run through the DSP and this part runs at 48k (or not?).
I made some more comparison tests between UDA (thanks for showing me the chip , on my Audigy1 it has 24 pins so I was on the wrong track) and AC97. I found on AC97 some strange noise (audible as digital floating) when higer gains in AC97Mixer (ProFx) are applied, this is not the case with the UDA, it is clean and at +12 db gain even very quiet. Of course the better choice for analog recording.
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Old Mar 31, 2005, 09:36 AM   #29
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Quote:
Originally Posted by TravelRec.
Hmm, very interesting, needs to be confirmed anyhow - I canīt believe that there is no resampling, the signal has to run through the DSP and this part runs at 48k (or not?)
I think signal goes through p16v (I mean hardware not plugin), there is no excessive IM distortion and bandwith is greater then 40kHz. Wave HQ should do the same in kx driver, but on my A2 Value I cannot make it work.
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Old Mar 31, 2005, 02:18 PM   #30
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popej, thanks for the extra measurements. The spectral responses also seem to indicate that downsampling is not happening.

Also, I thought that the p16v (chip) was integrated on the SB0400, so that part of the U19 (the Creative DSP etc.,) can run at 96000 while the DSP part specifically can only run at 48000.

This means in regard to 96000 data streams, that: we can see 'em but we can't touch 'em 'cause if we do it means taking signal into the DSP where it is downsampled to 48000.

So this limits the mixing etc., that can be done.
In analog: the analog mixer in the AC97 will be OK.
In digital: the digital volume control in U8, the 8 channel cirrus DAC is OK.

...What else, what happens to SPDIF???

I wonder if the ProFX controls are designed with this in mind.

I would have thought MX6 would use DSP resources, but in Adobe Audition I am seeing wider spectral responses akin to those Popej illustrated with his RightMark.


Could be that the conclusions are wrong and resampling is occuring. This could probably be seen with something like a 30Khz signal. If the stream is being downsampled and then upsampled most that frequency will vanish but with 96000 straight through it will be preserved.
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