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Date:      Sat, 26 Jan 2013 23:53:18 +0100
From:      "Ralf Mardorf" <ralf.mardorf@alice-dsl.net>
To:        "FreeBSD multi" <freebsd-multimedia@freebsd.org>, "Ruslan Bukin" <br@bsdpad.com>
Cc:        Jack devel <jack-devel@lists.jackaudio.org>, FreeBSD quest <freebsd-questions@freebsd.org>, John Rigg <j@jrigg.co.uk>
Subject:   Good news for snd_hdspe ADAT sync slave
Message-ID:  <op.wrji64upqhadp0@freebsd>

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Hi,

special thanks to John Rigg :), he mentioned sync. Thank you very much!

In general the driver should use master for sync as default, since the  
default is slave, it's good for my needs, because my ADA8000 is an  
elCheapo ADAT device, the RME card should be able to be the better sync  
slave. The short explanation can be found in the Jack devel mailing list  
archive, it was also mentioned by John.

So I switched the ADA8000 from slave to 48.0 KHz master (44.1 KHz is  
available too).

I read  
http://www.freebsd.org/cgi/man.cgi?query=mixer&manpath=FreeBSD+8-current  
and run

$ mixer
Mixer vol      is currently set to  75:75
Mixer pcm      is currently set to  75:75
Mixer rec      is currently set to  75:75

So there are no sync settings available.
I only tested outputs.

The latency is very bad, -p16 or any other value is ignored.
Good and bad news, ADAT out does work without glitches for

jackd -r -doss -r48000 -p16 -n2 -w16 -C/dev/dsp4.0 -P/dev/dsp4.0
= Channel 1 and 5, all other channels are dead.

jackd -r -doss -r48000 -p16 -n2 -w16 -C/dev/dsp5.0 -P/dev/dsp5.0
= Channel 3 and 7, all other channels are dead.

jackd -r -doss -r48000 -p16 -n2 -w16 -C/dev/dsp6.0 -P/dev/dsp6.0
= Channel 2 and 4, all other channels are dead.

jackd -r -doss -r48000 -p16 -n2 -w16 -C/dev/dsp7.0 -P/dev/dsp7.0
= Channel 6 and 8, allother channels are dead.

Comparison to Linux

On Linux latency isn't perfect either, but good enough to do serious audio  
work. I don't get completely rid of xruns on Linux using this card, but  
theoretically it anyway could be used for professional work, resp. not for  
professional usage, but to get lower professional audio quality. Regarding  
to xruns I don't have experiences with FreeBSD, for the test everything  
was ok, but this doesn't say that much.

On Linux the two analog IOs, but only ADAT channel 1 and 2 can be used,  
but all channels at the same time, I never tested SPDIF and AES/EBU on  
Linux. On FreeBSD the analog outputs and all 8 ADAT outputs are available,  
but not at the same time.

FreeBSD is missing TotalMix, ALSA and MIDI don't work on my machine. On  
Linux the HDSP mixer does run, but HDSP config doesn't and I can't store  
alsa mixer settings. On Linux ALSA and MIDI are perfect, especially when  
using Jack 2 with the alsarawmidi switch, there's quasi zero MIDI hardware  
jitter.

I forgot to test the excellent phones output on FreeBSD, but it anyway  
only makes sense in combination with TotalMix.

Regards,
Ralf

PS: The link to the coaxial SPDIF test is  
http://lists.freebsd.org/pipermail/freebsd-multimedia/2013-January/013827.html  
, I suspect for the optical SPDIF test I need to do some settings, so I  
didn't even try it, but the gear is at hand. I don't have AES/EBU at hand  
and I won't test AES as second SPDIF. I still need an app to test the  
inputs on FreeBSD.






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