[f1r3fly]

Hi all,

I'm using XBMC 12 Frodo from a fresh install of XBMCbuntu, connected with a SPDIF optical link to a DAC (Cambridge Audio upsampling DAC actually) with a high-end old fashioned 2.0 audio system
No pulseaudio or anything, direct way from XBMC to SPDIF only using ALSA

So when I play a standard FLAC file, obviously I get a 44.1 kHz sampling.
When I play a 192k FLAC file, it doesn't work as my DAC only accepts 96 kHz (and it's also a S/PDIF limitation as far as I know)
When I play a DVDrip file, ok, 44.1 kHz, but when I play a HD Movie (even with DTS Master Audio), it looks like the AudioEngine resamples at 48 kHz.

Now here is my advancedsettings file
<audio>
<resample>0</resample>
<forceDirectSound>0</forceDirectSound>
<audiophile>1</audiophile>
<audiosinkbufferdurationmsec>50</audiosinkbufferdurationmsec>
<allowtranscode44100>0</allowtranscode44100>
<streamsilence>0</streamsilence>
</audio>

I have a few questions about the new AudioEngine :

I don't want to use the <resample> tag, because the DAC can handle 44.1 files, it would be a shame to make the AE upsample it to 96k as my DAC does that better imho (upsampling to 192 kHz with filters and fancy stuff huh).
But I would like to downsample anything that is higher than 96k to 96k.
Is there any tweak who could do the trick ?

In fact, the tag I would need is <allowtranscode96000> so it would downsample anything higher.

Bonus question : Would any solution apply to movies audiostream ? Like, playing a DTS Master Audio (192/24) would be transcoded and downsampled to PCM 96/24 ?

Thanks for your help, guys

[DDDamian]

Here's how it works: when you play a decoded file (FLAC, mp3, wav) it looks at the sample rate of the file, and then at the closest match the device reports. In your case, playing a 192khz FLAC would downsample to 96khz as the DAC driver should report that as the closest match. A Debug Log would tell what your DAC reports.

The <allowtranscode44100> is only applicable if you are transcoding back to AC3 (Dolby). Thus the "transcode" part. Only applies to multichannel really.

Your DTS-MA question is a doozy. Basically there is no open-source decoder for DTS-MA - we either bitstream it or we don't play it - we can only play the DTS core. This DTS core can be decoded to PCM, unlike the lossless DTS-MA. DTS-MA is always at a sample rate (actually link speed) of 192khz. Your DAC *will not* pass this signal, not only because it requires 192khz but it can only be passed via HDMI as it requires a protected pathway called HDCP, which is only available on HDMI.

[f1r3fly]

Hey thanks for your really quick answer.

I'll check again, but it looks like DTS movies are played correctly.
Actually to make things easier, I didn't talk about HDMI also. I use the "both" pseudo-device in XBMC 12* so the stream is output to direct-SPDIF and also HDMI, and the TV sends it to the DAC via SPDIF (but synchronized with the screenplay, the motionflow induces about 100 ms latency)

I'll give a try with DTS-MA directly to the DAC, and also try to provide a debug log with 192k FLAC to the DAC also !

*btw thanks for this feature, beloved developpers, it was a pain in the ass to create it in ALSA conf since Camelot

[f1r3fly]

Here it is :
http://pastebin.com/bKPCGDKq

It gets interesting from 03:02:10, trying a standard FLAC file (44.1) then a 192k (but I noticed the issue may be that the DAC doesn't support 32 bits float ?)
About the DTS-HDMA, never mind, it looks that the file has a 48k sampling.

So with my current config, I'll use the most of my DAC, playing 44.1k files at 44.1k and 192/24 files at 96/24 ? No issue then

Anyway, thanks for the work achieved in XBMC guys, it seriously kicks ass