New subject: [linux-audio-user] Recording from SPDIF with Audiophile

2 Jan 2004

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(a)earthlink.net 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

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