[linux-audio-user] Recording from SPDIF with Audiophile

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Jan Depner wrote:
>
>> Moreover I found out that xmms and audiacity play
>> it a bit faster and higher than the original data, even though I always
>> use the DAT's master clock! :-(
>
>The problem there is that your DAT is probably at 44100 and your card
>defaults to 48000.  Check in envy24control.

You are right - sorry! When playing around I changed from a low level 48
kHz to a full level 44,1 kHz tape...

Frank Barknecht wrote:
>
> Please upgrade...
>
> ... if alone because dmix'ing as I described in c't will not work with
> this ALSA version and the Audiophile. ;)

Hm, I am a bit averse from upgrading because I am afraid that this may
result in conflicts with all that YaST stuff (and actually I do not need
dmix for recording). Do I also have to build a new kernel to upgrade
ALSA?

>> Well, -f S16_LE -r 48000 -c 2 is identical to -f dat and actually also
>> results in a mute file.

> Your problem probably is, that you are *not* recording from your
> digital input, but from the device called "default", which arecord
> uses by default. "default" corresponds to "plughw:0W unless you
> changed something in asoundrc (but you didn't do this).

Yes, I think this is the reason for my problem. As I said there are no
asoundrc files on my box. So I am going to learn about asoundrc and then
create one.

> Although I also have the Audiophile, I don't have any digital audio
> gear, so I never tried to record from that and thus I don't know the
> name of the digital ALSA device off-hand, but maybe someone else here
> does?

I am sure I can find this information somewhere on the ALSA webite.

> BTW: OSS emulation on the Audiophile can be a bit tricky sometimes
> because of the chipset, so you should try to use ALSA wherever
> possible with this card.

Sounds like a good idea to me. Anyhow I think that arecord is the
perfect tool for recording from DAT - if it works... ;-)

davidrclark@xxxxxxxxxxxxx wrote:
> Regarding low levels with some 24/96 cards: The inputs are lowered to 8.3%
> to account for 12 (or so) channels so that clipping won't occur, I presume.
> So if you have a 2-channel 24/96 card, your inputs are way too low
> when using ICE1712, for example.  (This is true for arecord, not qarecord.)
> If you do arecord with verbose output (-v), you will see exactly what the
> reduction is.  I should mention that this is with analog --- I would expect
> the same with SPDIF.

Is this also true if you do not use the mixer? What a nonsense! :-(

> Using qarecord, this problem doesn't exist.  I looked at the code, but
> again couldn't find where the input levels were maintained versus arecord
> where they are lowered.

At present I cannot access my Linux box to find out if qarecord is
installed. I am going to check for this as soon as possible.

Thank you all for your valuable help so far! I am confident of getting
it running now. :-)

Ciao,
 HippiE

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