DTS over SPDIF on aureon space

[vimax]

My problem is that I cannot play dts over spdif on a aureon space.

I read the dts faq, the whole audio faq in fact, and it said that the only way to play it is through vlc or in linux. I also used the search with a fair amount of patience, but with no useful results.

The problem is vlc works fine for decoding a lot of dts files, but I it still doesn't work if I choose A/52 over spdif in the menu, or even DTS->SPDIF encapsulation mixer in audio properties.

I accidentaly found a way to play dts files in a tutorial on converting DTS with graphedit using IV audio decoder. I noticed that if I render the output and in the filter properties choose spdif, the signal is sent correctly over spdif.

So far this is the only way I managed to send a dts file obtained from a wav file with hypercube dtsparser. This works fine with me, but I would really like to find a way to save that setting in the filter itself, because otherwise the only way to play it graphedit which is not a very friendly player.

Probably this should have been posted in the programming section but if there is a user-friendly solution...
What I am trying to get is a iviaudio.ax with spdif as a default option.
If I save the option in graphedit, save the .grf, load it as a IGraphBuilder in VC++, is there any way to save that object in a .ax so that I have a filter with spdif as a default option?

I have aureon space as a soundcard, with the latest drivers and onkyo TX-SR502E as the reciever. I don't think it's a problem of upsampling.The reason I say it's not a problem of upsampling is that if I play a wav ripped from a dtsaudiocd in winamp or wmp the reciever recognises the signal as dts 5.0 but I don't hear anything. every 5-10 seconds there is a tick of a sound in which the signal is reported as 5.1dts.

Now, the reason I say it's not the upsampling problem is that if I play the file with wmpclassic I can hear the hissing on the digital output which the reciever recognises as 44.1khz pcm. I noticed the soundcard has the same chipset as the m-audio audiophile so I guess it should work.

thank you for your patience with a newbie in the dts problem.

[vimax]

I recently tried to record the digital output on 96khz, 48khz, and 44.1khz, all in 16bit stereo and my results were:

on 96khz (I know the dts I had was not played in that sampling rate but I thought the extra resolution would help) the sound was just the undecoded dts, hissing

on 48khz the sound stopped, I guess the reciever was trying to decode something

on 44.1khz there were very short fragments (well under a second) of decoded (I thought I recognise it) sound.

Again I confirmed that there were no upsampling problems because in the 44.1khz if I lowered the volume, the silence became a loud hissing, and the reciever recognized the signal as 44.1Khz PCM
Should I try to record the stream in 24bit? or maybe mono?

I was sure that if I recorded correctly the output directly digital I would get wavdts files that I could play with everything but it seems I was wrong.
Anyone knows why it didn't work?

[Rockaria]

If you can play dvds(mostly 48k dts or dd) spdif passthrough without problems using windvd, it is most likely the upsampling problem. (http://forum.doom9.org/showthread.php?t=95113)

[vimax]

thank you for the answer, and sorry I forgot to finish the sentence. As I said twice I don't think it's the upsampling problem. I read that thread before I started this one because I think it's something else.

To sum up:

I would like either to get a wave I can play with anything and the reciever recognize it as dts

or a iviaudio.ax with spdif as default so I can play raw dts in wmp to be decoded by the reciever. (like in graphedit, only that in windows media player, to my knoledge you cannot change the filter settings)

[Rockaria]

Well, when they make the dtscd from original 48k dts 5.1ch(dvd audio or anything), they downsample the sample rate to 44.1k dts5.1 and wrap the format to dtswav which is 44.1k 2ch pcm externally to be written to the cd audio format.

When you play the dtswav file or cd with non-dts(wav)-aware player, you will hear the 2ch pcm hissing signals.

When you play them with vlc or foobar(renamed to *.dts) and set to spdif passthrough, if the sound card remarks(not decode & upsample) the signal as 48k dts for the rereceiver, I guess there will be some disconnections(or loose playing) while decoding the signal.
Or maybe the receiver is simply mis-recognizing the the 44.1k dts as 48k dts(even if the sound card does not remark to 48k).

I admit it may not be exactly the resampling problem.

[vimax]

it worked!! with foobar
at first I was dissapointed because I couldn't find any spdif options in foobar, but I read on another forum that it works if I use kernel streaming.

I do get BSOD quite often, but I think it's because I chose 24bit, now it's on 16bit and it hasn't crashed yet.

I'll see what I can find for a library manager for foobar, maybe with a nice skin.

Thank you very much!