AudioEngine branch - DO NOT REQUEST BINARY BUILDS

@fincheresque - Yes, this is an issue I am looking into, I need to find a way to enumerate the device capabilities without opening it. You can find out what might be using the device by running "lsof -n | grep asound" (betting its PulseAudio)

Off topic, butMemphiz sounds correct, looks like your firewall is broken and should be clamping your MSS to your PMTU:
iptables -A FORWARD -p tcp --tcp-flags SYN,RST SYN -j TCPMSS --clamp-mss-to-pmtu

Reducing your MTU has the same effect, but the above is the more correct way to reduce it for an entire network

(2012-04-24 11:03)Memphiz Wrote:  guess activating debugging in XBMC system settings would do the trick aswell

I thought that too. But it did not work unfortunately.
I tried to remove the debug logging entry in advancedsettings in order to see if it interferes somehow, but it didn't help. So I went over to append the debug parameter to the bin.

Hello to all,
I've been trying to pass 2 channel audio (flac, wav) through my optical output to
my preamp. It seems to be always resampled (the xbmc gain is functioning which
is not the case when output is a multi channel stream. Is that possible or am I just wasting
my (your) time. Unfortunately I do not have an HDMI receiver to test on..........


P.S. hope I am not off topic........

@panic_cpu, SPDIF doesnt support sample rates other then 44100hz, and 48000hz, and many AVRs dont support 44100hz, so we lock it to 48000hz.

(2012-04-26 20:43)gnif Wrote:  @panic_cpu, SPDIF doesnt support sample rates other then 44100hz, and 48000hz, and many AVRs dont support 44100hz, so we lock it to 48000hz.

This means all CD red book and DTS gets re-sampled, or am I missing something?

@HeresJohnny - yes your correct, there is nothing we can do about that, SPDIF has been deprecated by the industry for a reason, if you care for audio quality move to HDMI.

Those that have been having issues with their AVR locking onto DTS-HD MA streams, please give the latest version a go... we believe it is fixed now.

@gnif
I understand what you're trying to do and I think it's basically a good thing to make software more compatible. However, I feel it goes a bit against the idea to provide a high quality audio engine and then introduce forced resampling. I'd suggest adding two different output devices, one called "LOCKED SPDIF" where resampling takes places and another simply "SPDIF" where all resolutions possible with this standard are allowed. If you add a checkbox for this, you'd use up space in the GUI which would only be needed for one of several possible output devices. Just my humble opinion, of course

The following is the output of /proc/asound/card0/pcm1p/sub0/hw_params while playing a wav through spdif:
access: RW_INTERLEAVED
format: S32_LE
subformat: STD
channels: 2
rate: 44100 (44100/1)
Does that mean that after been resampled to 48000 by the AudioEngine something is resampling it back to 44100!!!!!!
Corresponding tests with DTS and Dolby Digital show an output of 48000.


Sorry

@HeresJohnny & panic_cpu - gnif and I spoke about this, and we'll add the option to not resample 44.1->48khz for spdif, but it will be off by default and have to be enabled via advancedsettings as it can cause issues with some receivers. It'll be in a future commit.

For some of us 2 channel playback is of far greater importance than multi channel and unfortunately
HDMI preamps are in the 4 digit value. Thanks and congrats for this amazing software without which
XBMC, as good as it is, cannot be considered complete.