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I have serveral concert bluray's containing 2 channel LPCM 24 bit audiotracks, and was wondering what exactly is beeing outputted via optical out when playing these in my setup.(my DAC has no display)

My physical setup:

Xtreamer Ultra2 ((Version:2.99.2 (3.0 RC 2)) > optical out > Benchmark DAC (24/96 capable) > balanced out > Primare i30 (stereo amp).

Audio Output: Optical/Spdif (Also tried "Analog" since that's supposed to leave the digital out untouched)
Speaker Config: 2.0
Boost Volume level on downmix: disabled
Output stereo to all speakers: disabled
-Dolby Digital capable receiver: disabled
-DTS capable receiver: disabled
-AAC capable receiver: disabled
-Multichannel LPCM capable receiver: disabled
-DTSHD capable receiver: disabled
Audio Output Device: ALC892 Digital S/PDIF
Passthrough Device: ALC892 Digital S/PDIF
GUI Sounds: Disabled

My setup is strictly for hifi stereo, and basicly I'm wondering if Frodo/AudioEngine is capable to output the untouched 2 channel LPCM track via my Xtreamers spdif/optical out?
This worked nicely in my "Dune HD Base 3", but it had specific settings for it.

I've read the AudioEngine wikis etc, but it really isn't mentioning much regarding untouched 2 channel 24bit LPCM sound. The focus here beeing two channel hd audio not multichannel.

Please enlighten me

Also dont just answer that if my hardware is capable of it it is beeing outputted because there is no proof in that when going through xbmc/frodo settings or in any osd debug/decode info that I can tell of at least

enable debug log and see

Thanks, I just tried that but I cant make out anything relevant from the debug log.
There is no mentioning of 16bit or 24 bit anywhere.

Pastebin

Please, next time - don't cut logfiles ...

Alright I won't, didn't think the rest was important.
Here it is in all its complete "glory": PASTEBIN
I changed the audiotrack while playing the clip from "DolbyDigital" to Stereo (lpcm 24bit) after a few seconds.

Anyway so "AE_FMT_S16NE" means it's 16bit then?

Thanks for replying, appreciated.
Now I just have to figure out if any settings could help me achieve 24bit lpcm output, I'll try the openelec forum I guess.

I thought optical will not support such high res files 24/96?

It does, it actually supports up to 24bit/192Khz if you have a multistrand fibre cable (not the cheapest plastic ones).

But that doesnt matter, I can't even get the right signal to get to the spdif/toslink output.
Guess I'll have to buy an expensive hdmi audio breakout box to get the lpcm signal.

You cut the logfile again :-) Especially the startup part is of great relevance.

Darn! well third time's a charm
pastebin

Observe that I changed audio tracks three times to get to the lpcm 24bit audiotrack from the bluray!

Thanks for looking into it!

Check the m_dataFormats of your sinks and compare, it does not look good. It is transcoding all the time. If you can upload a sample I can check, if it takes wrong codepaths.

Hm, but what about "AE_FMT_S32NE"?
That's also the one beeing used when playing back the lpcm stream.
EDIT: NWM now I noticed it changes to FMT_FLOAT instead later.

I'll try to make a clip of it for upload, thanks for the help so far

Damn to bad..

EDIT2:
Changed my Audio Output from "Analog" (which acording to the AE wiki should disable processing on the digital out) to Optical/Spdif insted, doubt it did any difference but here's a new debug log

LINK

I guess I'm screewed anyway to bad I only just bought this hardware.

Check your speaker setup. At the moment you listen 5.1 Tracks on a 2.0 speaker setup, it must be downmixed. And as the Optical only really has 2 channels, it must be done for those.

My speaker setup is always 2.0 both physically and in the settings.

Perhaps you happened to notice the audio track change I did a few times while the debug log vwas active, but I always select the 2ch lpcm track in the end.
I just pressed the audio button a few times to see the osd that indicates that the 2ch lpcm is selected so it' cycled through both the ac3 soundtrack as well as the dts-hd master track before getting to the stere lpcm track.

Also do you know of any nifty software that can create a short clip out of the m2ts file for you to have a look at?

Really appreciate the help, thanks.

Herse my audio setup in xbmc btw:
Audio Output: Optical/Spdif (Also tried "Analog" since that's supposed to leave the digital out untouched)
Speaker Config: 2.0
Boost Volume level on downmix: disabled
Output stereo to all speakers: disabled
-Dolby Digital capable receiver: disabled
-DTS capable receiver: disabled
-AAC capable receiver: disabled
-Multichannel LPCM capable receiver: disabled
-DTSHD capable receiver: disabled
Audio Output Device: ALC892 Digital S/PDIF
Passthrough Device: ALC892 Digital S/PDIF
GUI Sounds: Disabled

Hi there,

I'm trying to obtain a bitperfect audio streaming too with a similar configuration Openelec on ATV 1 XBMC 12.0-RC3

Audio Output: Optical/Spdif (Also tried "Analog")
Speaker Config: 2.0
Boost Volume level on downmix: disabled
Output stereo to all speakers: disabled
-Dolby Digital capable receiver: disabled
-DTS capable receiver: disabled
-AAC capable receiver: disabled
-Multichannel LPCM capable receiver: disabled
-DTSHD capable receiver: disabled
Audio Output Device: HDA Intel ALC889A Digital S/PDIF

ATV is connected with tosslink cable to an M-DAC Audiolab that can run bitperfect test in addition to show actual bit/frequency values.
Bitperfect test wav files I have are only 16/44.1kHz and 24/96kHz soI cannot run also the 24/48kHz test, but:
- the setup is bitperfect for 24/96kHz (test passed with no errors playing the 5 min wav file)
- With 16/44.1 the DAC is seeing 24/44.1, resulting in test failure

I have also tried to set "audiophile" option in advancedsettings.xml but with no results
I've also tried output USB with exactly the same results.



Now I am kind of loss because I've been fighting oversampling to 48kHz moving back from ATV 2 to ATV1, now the frequency is ok but not the bit depth.


any hints


thanks

> aldonz

You're already having better luck than me

But anyway try this in your advancedsettings.xml file:
<advancedsettings>
<gui>
<algorithmdirtyregions>1</algorithmdirtyregions>
<nofliptimeout>1500</nofliptimeout>
</gui>
<audio>
<audiophile>1</audiophile>
<allowtranscode44100>1</allowtranscode44100>
</audio>
</advancedsettings>

Specifically the " <allowtranscode44100>1</allowtranscode44100>" option might help, probably not but it's worth a shot