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
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.
Transformation table:
0 <- 0*0.0833333 + 1*0.0833333 + 2*0.0833333 + 3*0.0833333 + 4*0.0833333 +
5*0.0833333 + 6*0.0833333 + 7*0.0833333 + 8*0.0833333 + 9*0.0833333 +
10*0.0833333 + 11*0.0833333
I would be interested in how to alter the routing myself, if anyone has
the information. I looked at some of the configuration files but could
not immediately see how to do this. I would like NO reduction on inputs.
It appears to have something to do with ttable routing and gain factors.
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.
Thanks to anyone for information on how to do this. I'm sure there are
a number of folks out there who have cranked up their volumes, only to
clip on the card, then lower them again, only to throw away perfectly good
signal. But many of them may not realize that this is totally unnecessary.
Jan Depner wrote:
>
> What version of Audacity are you using?
1.1.1. Note that there is no problem in *playing* files with audacity.
Harald Milz wrote:
>
>>In the meantime I tried to use arecord (which unfortunately does not
>>provide man pages)
>
> It does - I have one on my machine!
Ah yes, there is one on my one, too. Seems that I typed something
wrong?! But man does not really provide more information than --help (at
least in version 0.9.0rc7). :-(
>>arecord -f S16_LE -r 48000 test.wav
>>
>>*but* this results in a mono file with too low signal level.
>
> You may have to set -c 2 as well to get stereo, this is not the default.
Well, -f S16_LE -r 48000 -c 2 is identical to -f dat and actually also
results in a mute file.
> Alsio the hardware device maybe.
In the meantime I checked modules.conf by the great c't article of Frank
Barknecht. Everything seems to be okay there. But I could not find
/etc/asoundrc or ~/.asoundrc. May be this is the reason for my problem -
I am going to find out next.
> As for low level - are you sure the DAT is correctly leveled? After all,
> these tools do nothing with the level except if explicitly (and kindly)
> asked.
Yes, arecord seems to be exactly what I need for a 1:1 DAT copy.
Nevertheless I still can only record mono files with definitely much
lower level than the input signal (which takes the full range of the
envy24control meters). 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! :-(
By the way: Does anybody know, what HardwareSettings -> VolumeChange
means in envy24control?
Ciao,
HippiE
So far, I have gotten to install on my Debian system, the MPU401 and USB-Audio
drivers. Here is my cat /proc/asound/cards:
1 [UM1 ]: USB-Audio - UM-1
EDIROL UM-1 at usb-00:07.2-2
2 [UART ]: MPU-401 UART - MPU-401 UART
MPU-401 UART at 0x300, polled
Both are listed as RWxE (or similar) but I cannot remember how to get that at
present.
amidi -l cannot read the cards.
(I still cannot get snd-cs46xx to install for my Dman2044 because the pci
auto-config system thinks it has a Maestro AGOPO ESS chip. About to give up
on this and install my old MediaVision just to have something on which to
listen.)
The MPU-401 is feeding a Yamaha sw60xg card. Yamaha did it this way for
windows2000 so I tried it here.
Windows programs running under WINE can play to the MPU401 beautifully. Trying
to get MIDI in from the UM1 will hang the polling program. Nothing native to
the Linux can record or play from either one of them.
Are there any options that I need to feed the usb-audio to get that working?
How do I use these devices with the Linux?
Mark Constable wrote:
>
> http://alsa.opensrc.org/index.php?action=find&find=envy24control
>
> Probably more exciting info somewhere but the above may help.
Thank you. There is some information about installing and OSS emulation,
but unfortunately nothing helpful about how to use envy24control.
Anahata wrote:
>
> Does this have something to do with Audacity only working with OSS sound
> drivers?
>
> I think it can be made to work, either by later versions of Audacity
> having ALSA support or by using ALSA's OSS emulation, but as you can
> probably tell, I haven't tried this. I am using Audacity with OSS,
> which works fine for my purposes, including recording via S/PDIF input.
Harald Milz wrote:
>
> man arecord maybe.
In the meantime I tried to use arecord (which unfortunately does not
provide man pages) to avoid OSS emulation problems:
arecord -f dat test.wav
produces mute files again. However it works with
arecord -f S16_LE -r 48000 test.wav
*but* this results in a mono file with too low signal level.
I also did not succeed in playing around with the
recording/muting/volume settings of alsamixer. :-(
HippiE
I'm trying to use the Gate plugins (from the swh plugin collection) with
Ardour. Seems like that isn't quite the right place for them - there
seems to be no place for the non-audio input / output?
So what other app could I use them in?
thanks
John
Jan Depner wrote:
>
> Try changing /dev/dsp to /dev/dsp0. Usually /dev/dsp is a link to
> /dev/dsp0 but not always.
This link is okay.
> Also, can you see the signal in envy24control?
Yes, I can see it an route it - directly or via the mixer - to the
analog outputs.
> I was under the impression that there might be a problem with
> the SPDIF inputs on the envy24 chipset cards with ALSA but I could be
> wrong.
I do not think so because I can mix, DA and output the signal.
HippiE
tim hall wrote:
>
> Why are you using such an old version of alsa?
> Perhaps this is the way SuSE does it (?:-)
Right, this is part of SuSE 8.2.
>>But how can I record the data?
>
> What do you mean by 'data'? If you just want the data, surely there is some
> kind of way of 'mounting' the tape drive directly - As if it was a Tape
> ARchive and then using standard tools to copy it where you want it - 'dd' ?
Mounting the DAT recorder?! Sounds interesting... ;-)
> You probably haven't read enough, and my experience is that it's easy to
> overcomplicate matters. There may be a simpler solution. Oh yeah, and you
> need to ask clearer questions.
What I want to have is a 1:1 copy of the audio data of a DAT deck. For
this I use its SPDIF-out and plug it to the SPDIF-in of the Audiophile.
Using envy24control I can route the signal to the cards internal mixer
or to its outputs. But how can I record it to an audio file?
Recording with Audacity results in a mute file. In its preferences I
found the sole option /dev/dsp as recording device.
I am determined to read more about it if you can tell me *what* to
read...
Ciao,
HippiE
Hi there,
I still try to record a DAT signal. In the meantime I installed a
M-Audio Audiophile 2496 as card 0. I can configure it using
envy24control (alsa 0.9.0) and route the signal to the mixer or directly
to the outputs.
But how can I record the data? Programs like Audacity or krecord do not
"hear" it. What do I do wrong?
Ciao,
HippiE
> -----Original Message-----
> From: Dave Phillips [mailto:dlphilp@bright.net]
>
> PlanetCCRMA is definitely has the more current apps versions, but
> AGNULA Demudi is really quite nice too. I have both systems installed
> here, and I must say that I'm impressed by the AGNULA core system. Yes,
> it's not pretty (yet), but it's well designed where it counts, and the
> 1.1 release promises to bring the system into a more up to date
> condition.
I will 3rd (or 4th?) the Agnula recommendation. I had originally installed
ccrma (which is fantastic for those who want to just get up and go!) but I
wanted to do a lot of stuff with PD that wasn't yet supported under ccrma,
but was supported under debian out of the PD CVS. I installed a base woody
system, then set all of my apt stuff to testing, and used the Agnula disks
to get the sound stuff set up. After some initial X and mouse problems with
agnula (now being tracked in their bug databse) I was good to go.
> I'm not sure when AGNULA 1.1 will be ready for release but
> I'm sure it's Real Soon Now. ;)
January 15th according to some recent posts of the Agnula list.
> Next month's column for the Linux Journal On-line will briefly
> describe my experiences with PlanetC and AGNULA.
Cool. By the way Dave GREAT article in CMJ. I just got back from vacation,
checked the mail, and saw the cover! Having just jumped into the Linux world
I was happy to see the article. (A month or 2 to late for me, but I'm sure
it will help a lot for others!)
m.