Linux - What needed packages are missing from Ubuntu Minimal 10.10?
Hi there, I asked a question a few weeks ago that never got answered, basically. I'm using XBMC 11.0 final installed on a minimal ubuntu 10.10 setup.

The problem is that any audio files that are in a non standard sample rate, e.g 22khz\32khz are giving silence when played in XBMC. The vizualisations indicate that sound is being output and my receiver is acting as if there is a loose connection, on the LED of the receiver the word "HDMI" flickers on and off. Wheras it should stay solid - and does for all other media.

From research I have gathered that XBMC isn't converting the 22khz\32khz audio for output at a a standard sample rate like 44.1khz\48khz, etc.

Further reading showed that this could be down to missing dependencies in my Ubuntu minimal install. I have tried different versions of ubuntu with exactly the same results.

The strange thing is that these audio files are say 30 mins long but XBMC is saying they are about 2 hours long in some cases.

I have verified it is not a problem in the audio streams by playing them on a computer and even my sqeezebox setup plays them fine, both a radio and a squeezebox receiver. The audio files in question are from different sources too and analysis shows they all have non standard sample rates but that is their only similarity.

So, is this a bug in XBMC? a dependency issue? I would really like to solve this.

I have already installed the libmad0 dependency that is always needed to get any mp3 to play correctly at all.

if you are using passthrough, there is no sample rate conversion afaik, its up to the receiver to handle it.
Thanks for the reply,

To get audio to work, I had to set the passthrough and audio output device to both "custom" and "plughw:0,9"

I've tried various combinations and all sorts of settings but don't seem to be getting anywhere.

Do you know how I could set a sample rate conversion? I've tried messing around with the XML file that contains adanced settings that mentions sample rate conversion but again, to no avail. Nothing seems to make any difference and changing plughw:0,9 breaks playback completely.
you could try alsa dmixer, that should do the sample rate conversion, but don't ask me how, apart from creating an asoundrc...
Thanks for the tip Smile

It's a bit over my head unfortunately...I'm still surprised this is even nessacary, though I don't question you are right in that it would work, surely there's an easier way and it shouldn't really be happening in the first place.

Is anyone who uses their card as a passthrough device (most?) having no joy playing non standard sample rate audio, surely not? (Well I don't know lol). Thanks for your posts though it's very much appreciated.
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What needed packages are missing from Ubuntu Minimal 10.10?00