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Old 3rd October 2013, 10:58 AM   #101
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Great phofman - many thanks.

I'll read your posts and then dive into the source code.
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Old 3rd October 2013, 11:01 AM   #102
phofman is offline phofman  Czech Republic
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Well, the hint to kernel source code was more of a joke :-) But it will definitely be a good learning experience
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Old 3rd October 2013, 11:10 AM   #103
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I'm good with C source code, I used to write production code for telcos a few years ago and have recently got back into it, writing a Win32 API sampler for my music side projects.

The bigger hassle for me will be to actually find the time to work through the source code!
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Old 3rd October 2013, 01:44 PM   #104
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FWIW to those interested in using MPD on the RPi, my system is averaging 1.6% CPU usage and 5.5% memory usage when playing 44.1 wave files.

I'm not sure if these figures are typical, nevertheless it's nice to see the system resources so lightly loaded for playing audio.
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Old 3rd October 2013, 01:55 PM   #105
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This seems like an interesting project. I'm tempted to give it a try and see how well it works.

Right now I've found the only way I can get digital sound from this is to use an HDMI splitter, or use the TOSLINK out from a TV supporting audio over HDMI.

Now exactly ideal.

Would I be able to use something like a Behringer UCA202 in order to get a digital out?

Alternativly there is I2C which I totally unfamiliar with.

Would it be possible to hook the I2C from the RasbPi to a DCX2496? If so how easy is it to do?

Thanks
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Old 3rd October 2013, 02:09 PM   #106
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Quote:
Originally Posted by MikeHunt79 View Post

Right now I've found the only way I can get digital sound from this is to use an HDMI splitter, or use the TOSLINK out from a TV supporting audio over HDMI.

I have a temporary solution in place currently whilst I plot my next move, it might be of interest to you: I'm using a cheap USB external soundcard, namely the Creative Soundblaster Live card.

It's completely plug and play with my Raspian distro - all you need is a USB cable, and then of course your choice of cable into the DAC.

Actually, you don't even need cables - I've just about been able to couple my RPi, soundcard, and DAC together with adapters. (My theory of course is that taking the cables out of the equation is a good thing, which logic tells me is true ... but I'm sure somebody will be along soon to correct me )

Quote:
Originally Posted by MikeHunt79 View Post
Alternativly there is I2C which I totally unfamiliar with.
Thanks
Good luck there. I have some simple projects running using the GPIO, but I imagine it would be very challenging to get I2C/S working efficiently.

I think there is a thread on this elsewhere, IIRC. Anyway, I'm sure the more experienced members of this forum will guide you better than I can in this domain.
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Old 3rd October 2013, 02:42 PM   #107
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A picture of the soundcard, with RPi plugged in and optical cable to the DAC.



(I'll be replacing the card with a WaveIO board shortly.)

Click the image to open in full size.
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Old 3rd October 2013, 04:38 PM   #108
phofman is offline phofman  Czech Republic
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Quote:
Originally Posted by quinnpad View Post
FWIW to those interested in using MPD on the RPi, my system is averaging 1.6% CPU usage and 5.5% memory usage when playing 44.1 wave files.

I'm not sure if these figures are typical, nevertheless it's nice to see the system resources so lightly loaded for playing audio.
There is no reason a pure playback without any resampling should take more than that. Plus MPD uses quite large alsa buffers which makes it reduce CPU load too.

In complete opposite direction is the nrpacks=1 setup which forces the USB controller throw IRQ every 1ms (at least on UHCI-compatible machines) which must be services by the CPU.

Recent alsa patches remove this setup alltogether, calculating appropriate nrpacks value based on requested buffer/period sizes Re: [alsa-devel] Buffer size for ALSA USB PCM audio -- Linux USB
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Old 3rd October 2013, 06:07 PM   #109
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Quote:
Originally Posted by phofman View Post
There is no reason a pure playback without any resampling should take more than that. Plus MPD uses quite large alsa buffers which makes it reduce CPU load too.

In complete opposite direction is the nrpacks=1 setup which forces the USB controller throw IRQ every 1ms (at least on UHCI-compatible machines) which must be services by the CPU.

Recent alsa patches remove this setup alltogether, calculating appropriate nrpacks value based on requested buffer/period sizes Re: [alsa-devel] Buffer size for ALSA USB PCM audio -- Linux USB
That's really interesting.

A few weeks ago I had added
options snd-usb-audio nrpacks=1
to my alsa-base.conf file to cure a logging problem in the mpd code, and so I am running in this mode ... but we are not seeing the high CPU load.

The recent patch which has removed the aforementioned static int from headers and source files can't be on my system, since I haven't performed any package updates in a long time.

What gives? Is something not working correctly?

FWIW, I only ever play 44.1 PCM (.wav) files, so I'm not sure I understand what need there might be for resampling in my context. (Although this area is admittedly beyond my knowledge for the moment.)
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Old 3rd October 2013, 06:44 PM   #110
phofman is offline phofman  Czech Republic
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Here is a nice function to list parameters of running loaded modules. https://wiki.archlinux.org/index.php...ule_parameters

Just copy the code into terminal window and run aa_mod_parameters . That way you can check your loaded snd-usb-audio module parameters.
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