pomFritz

Hello, if I 'AC3 decode' a 24/96 signal, is it downsampled to 48 KHz somewhere in the path during decoding or routing?

Max M.

Mm, well, assuming the AC3 format itself supports only upto 48kHz sample rates - the answer would be irrelevant. In other words, you can't "ac3 decode 24/96" since 24/96 cannot be encoded as AC3 without prior downsampling (i.e. you would get your downsampling right in an encoder before the signal actually enters your soundcard).

I took a look at your previous posts and it seems you often use "AC3" as a synonym for SPDIF and any digital intefaces/formats at all (and vice-versa). So maybe the answer could be more helpful if you provide more details on the device/media you actually get your digital input from (perhaps that 24/96 signal you're recieving has nothing to do with AC3 at all...)

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pomFritz

OK, let be more precise. First, I'm a little bit confused with all the surround mess terms AC3/DOLBY I or II/DTS or HD/Surround/and others. I say AC3 because in Creative drivers, the decoder says Dolby/DTS and I thought that it corresponds to AC3 decode in KX.

I want to build an "amp" PC under Win 7. The surround arrives encoded in the SPDIF input of card A. Then it's decoded by card A and ideally sent to KX ASIO. But, because of the 48 KHz limitation, I plan to add 3 more cards B (front), C (rear) and D (center+sub).

4x Audigy 2 ZS cards, 3x externals AD converters (front/rear/center+sub), KX and ASIO4ALL :
---
external surround or stero 24/96 -> (A) SPDIF in, decode
--- front ---
-> (A) SPDIF1 out
-> (B) P16V SPDIF in -> ASIO software -> (B) SPDIF out
-> external AD converter 1
--- rear ---
-> (A) SPDIF2 out
-> (C) P16V SPDIF in -> ASIO software-> (C) SPDIF out
-> external AD converter 2
--- center + sub ---
-> (A) SPDIF3 out
-> (D) P16V SPDIF in -> ASIO software-> (D) SPDIF out
-> external AD converter 3

The audio signal from player to external AD converter must be digital and not downsampled somewhere in the path.

Any simplier solution is welcome
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Max M.

Well, I'd said it's too complicated and in fact it won't worth it, let's see:

* When the input format is that "External Surround" (which is transferred as AC3 - yes) it *is* already at 48 kHz (or below) hence there's no point to overcome it via p16v and your current setup (as in Surround to ASIO channels problem) is just enough.

* If the input is "Stereo 24/96" (DVD Audio media for example) then it is not the AC3 format and no AC3 decoding is involved. Those you need only one (stereo) ASIO channel and don't need those additional cards again.
As far as I understand you're planning to overcome the 48kHz limitation of kX ASIO by using Asio4All set to use kX Wave HQ ports, right? That would probably work (in theory) but still seems to be too complicated, I would consider a much simpler alternative:

"If it's really DVD Audio (or just whatever 24/96 file) why don't just play it with any media player that has built-in VST plugin support (for example I use foobar2k)? - so you could apply all the effects you want directly in place without any resampling and without all that crazy software/kx/asio4all/spdif/multicard rerouting (just set the player to output to kX Wave HQ)."
(P.S. Actually, can't just that "ASIO Software" you use for effects play those 24/96 media on its own?)

---

Notice that by putting more cards into your setup you just introduce another point of signal degradation: the Audigy cannot lock its clocks to an incoming spdif stream so it's always has to be some kind of resampling at its SPDIF receiver itself (In particular, the p16v spdif input does resample anything to its own 96kHz (including 96kHz input streams - just because those are running at different clock (i.e. those are "different 96kHz")). So by using Audigy's SPDIF input you automatically accept some kind of resampling (which is not necessary a *down*sampling but still not cool) one way or another.

The problem of different clocks also somewhat applies even to "Direct SPDIF recording" feature (just in case you read about it and begining to think if it can help someway): Yes, you can record at 96kHz (let's forget it's limited to 16 bits) and *then* play back the recorded file via "Wave HQ" and no resampling takes place - But there's no way to monitor the incoming stream in *real-time* w/o resampling because the recording and playback have to be performed at different clocks (recording should be running at the spdif input stream clock while playback should go at the card clocks) - i.e. the monitoring w/o syncing would just result in clicks.

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pomFritz

Thank you much for you very instructive answer.

You are right, my error was that I was speaking about decoding HD versions of Dolby and DTS, which seems can't be done neither by Creative, nor by KX.

Yes, this is my actual setup.

I want to separate the player (Blu-Ray, audio, games, hardware, etc...) from the mixing console related to my room setup (sub EQ, etc...) and to access to the mixing console remotely (tablet PC).

I like my externals DA converters (Musical Fidelity V-DAC). I don't want to search and spend money in an external amp and in good speakers cables.

Those cards are the only I know with 4 digital outputs. What you tell me about the SPDIF sync doesn't sounds good to me . Back to brainstorming...

pomFritz

Hehe, thanks to Kx driver and some others softs, it works

My goal was to decode DTS HD and send 24/96 decoded channels (fronts + rears + center + sub), through Audigy 2 ZS, SPDIF outputs to externals stereo DA converters (one per speaker) without downsampling (48 KHz, etc...) somewhere in the path. After days and hours ... the light was

Windows 7 64:
1. Install Creative driver SBAX_PCDRV_LB_2_18_0017 -> 5.1 decoded though SPDIFs in 24/96 doesn't work
2. Install Kx 3551 -> 5.1 decoded though SPDIFs in 24/96 (Wave HQ) doesn't work
3. Clear Kx DSP (not sure if needed, but I did it, because all that goes to DSP is downsampled to 48 KHz)
4. Disable front/rear swap in Kx Mixer (from what I understood, it's related to Audigy internal DA converters)
5. Enable 96000 Hz in Kx Mixer
6. Connect speakers to Audigy 2 SPDIF outs: fronts to SPDIF0, rears to SPDIF3, center (left) + sub (right) to SPDIF1 (I think SPDIF2 is for 7.1 laterals)
7. Set Wave HQ as output
=> Enjoy

After that you can do any soft config you want. I use:
- ffdshow audio to set separately channels volumes (Win MCE remote works only for master volume), channels routing and stereo to surround conversion (cool with Media Portal for TV)
- DTS HD decoder filter found in Arcsoft totalmediatheatre_2.1.6.105_2.1.6.125_update_all.e xe
- MPC-HC as player
You can install a Dolby TrueHD decoder filter if needed.

I have checked every Audigy 2 SPDIF channels output with an EMU 1616 and it detects the 96 KHz clock

Now I can enjoy Dark Side of the Moon and Wish you Where Here Immersion Editions in 5.1 24/96 (it's not DTS HD, just LPCM).
And I hope I can play my guitar in 5.1 . I will try to install ASIO4ALL and my RME Babyface (because ASIO4ALL allows wou to mix Ins (RME) and Outs (Audigy 2) from different sound cards).

Of course, it would have been be better to use a second PC as DTS HD decoder, but at the moment, it doesn't seem possible to decode a DTS HD stream through SPDIF input.

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