20 Jun
2006
20 Jun
'06
3:51 a.m.
So, another dumb question from a hardheaded linux audio user
(is there any other kind? ;-)
I've got an RME hammerfall lite (9636/52)
I've got a Yamaha AW4416 with an MY8AT module (8 channels ADAT i/o).
I've got fibre cables connecting them together.
The Yamaha seems happy, reporting 8 channels locked at 48kHz.
What I don't have is any idea how to get anything good to
happen on the linux side. I've been googling my brains out
for the last two or three months, and the very best information
I have been able to find anywhere is this 3 year old thread:
http://www.music.columbia.edu/pipermail/linux-audio-dev/2003-March/003328.h…
Which in turn recommands alsa-lib/doc/asoundrc.txt
which, looking at that, doesn't really help much at all, I'm afraid,
you pretty much have to know how it works already in
order to make head or tail of it.
Is there any better information out there? I just want
to be able to transfer tracks from the PC to the AW4416, and
back, preferably as many as 8 at a time, or as few as 1 at
a time, but, judging from my googling, it seems next to nobody
is doing anything like this. I would estimate less than ten
people on the entire planet (and that is my inflated guess)
have an RME hammerfall AND are using the ADAT interfaces.
Is that about right? Am I that far out in the weeds?
I would really like to get this working... otherwise, I've
just blown $700 (RME + yamaha ADAT interface + cables)
for nothing , and I'm back to burning CDs to transfer
tracks between AW and PC.
Any hints appreciated... thanks.
-- steve
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20 Jun
20 Jun
9:39 a.m.
New subject: [linux-audio-user] Recording ADAT inputs on RME hammerfall 9636/52
On Mon, 2006-06-19 at 22:51 -0700, Stephen Cameron wrote:
So, another dumb question from a hardheaded linux
audio user
(is there any other kind? ;-)
I've got an RME hammerfall lite (9636/52)
I've got a Yamaha AW4416 with an MY8AT module (8 channels ADAT i/o).
I've got fibre cables connecting them together.
The Yamaha seems happy, reporting 8 channels locked at 48kHz.
What I don't have is any idea how to get anything good to
happen on the linux side. I've been googling my brains out
for the last two or three months, and the very best information
I have been able to find anywhere is this 3 year old thread:
http://www.music.columbia.edu/pipermail/linux-audio-dev/2003-March/003328.h…
Which in turn recommands alsa-lib/doc/asoundrc.txt
which, looking at that, doesn't really help much at all, I'm afraid,
you pretty much have to know how it works already in
order to make head or tail of it.
a) make sure you are on a distro that makes hdspconf and hdspmixer
available. several braindead distributions have removed the alsa package
that includes these tools from their repositories.
b) run hdspmixer to initialize the HDSP mixer to a sane default
c) use hdspconf to set the clock master/slave relationship between the
the AW4416 and the HDSP. i would recommend *not* use ADAT autosync for
anything critical - i do not find it 100% reliable. you're better off
getting a word clock cable. personally, i would make the HDSP the master
and the AW4416 the slave, but you may prefer it the other way around.
d) run JACK-enabled software.
e) profit!
--p
10:53 a.m.
On 6/20/06, Paul Davis <paul(a)linuxaudiosystems.com> wrote:
On Mon, 2006-06-19 at 22:51 -0700, Stephen Cameron
wrote:
<SNIP>
> I've got an RME hammerfall lite (9636/52)
<SNIP>
a) make sure you are on a distro that makes hdspconf
and hdspmixer
available. several braindead distributions have removed the alsa package
that includes these tools from their repositories.
b) run hdspmixer to initialize the HDSP mixer to a sane default
Paul,
Hammerfall, not HDSP. Does any of this work in his case? It didn't
when I was running my Hammerfall under Linux.
Steve,
I have both an HDSP9652 and a Hammerfall Lite. My Hammerfall is now
in a Windows box so I cannot step you though an exact solution but I
have used it under Linux and it works well. I'm pretty sure some group
of us can answer your questions and get you there pretty quickly.
The Hammerfall will work correctly with Alsa drivers and the Jack
server. I think your most simple answer is as Paul outlined, although
I think that you cannot use hdspconf to set the clock rate of the
Hammerfall unless hdspconf has been updated in some way I'm not aware
of.
To set the card's clock rate will likely require use of some amixer
commands. man amixer and then play around with reading out the
controls in the driver. The correct solution here is to configure
either your Hammerfall or the AW4416 as a clock slave. I don't have
any problems using the ADAT lines to sync clocks so you could try that
to begin with. It will certainly work well enough to test the
connection. You can go to a spdif or Word Clock sync later if you need
it. I'm not sure if you can do Word Clock sync on a DIGI9636 anyway.
That connection may be on the DIGI9652 expander card. I cannot
remember.
Anyway, it will work. Possibly you can post back your currect
settings from /proc/asound/cards and the output of amixer's listing of
controls and er can get you lined up pretty quickly.
Cheers,
Mark
11:45 a.m.
--- Mark Knecht <markknecht(a)gmail.com> wrote:
On 6/20/06, Paul Davis
<paul(a)linuxaudiosystems.com> wrote:
Ah, thank you, that was going to be my next question.
(and this morning, I tried to build hdspmixer from alsa-tools-1.0.11
and it wanted fltk lib... tried the
latest fltk lib v. 1.1xxx and v. 2.xxx something, and though
configure stopped complaining, hdspmixer
would compile with neither version of the fltk libs. Maybe it
needs some specific earlier version? (or newer CVS version?)
It complained about no such member something->fl_widget. (I
can reproduce the exact message when I get home tonight
if anybody here wants me to.)
But it sounds like I might not be able to use hdspmixer
anyhow, even if I could build it.
On Mon, 2006-06-19 at 22:51 -0700, Stephen
Cameron wrote:
<SNIP>
> I've got an RME hammerfall lite
(9636/52)
<SNIP>
a) make sure you are on a distro that makes
hdspconf and hdspmixer
available. several braindead distributions have removed the alsa package
that includes these tools from their repositories.
b) run hdspmixer to initialize the HDSP mixer to a sane default
Paul,
Hammerfall, not HDSP. Does any of this work in his case? It didn't
when I was running my Hammerfall under Linux.
Steve,
I have both an HDSP9652 and a Hammerfall Lite. My Hammerfall is now
in a Windows box so I cannot step you though an exact solution but I
have used it under Linux and it works well. I'm pretty sure some group
of us can answer your questions and get you there pretty quickly.
The Hammerfall will work correctly with Alsa drivers and the Jack
server. I think your most simple answer is as Paul outlined, although
I think that you cannot use hdspconf to set the clock rate of the
Hammerfall unless hdspconf has been updated in some way I'm not aware
of.
To set the card's clock rate will likely require use of some amixer
commands. man amixer and then play around with reading out the
controls in the driver. The correct solution here is to configure
either your Hammerfall or the AW4416 as a clock slave. I don't have
any problems using the ADAT lines to sync clocks so you could try that
to begin with. It will certainly work well enough to test the
connection. You can go to a spdif or Word Clock sync later if you need
it. I'm not sure if you can do Word Clock sync on a DIGI9636 anyway.
That connection may be on the DIGI9652 expander card. I cannot
remember.
Ok, this gives me some hints to play around with, and encouragement.
Thanks a bunch! I imagine when I run, say qjackconnect,
I should expect to see a bunch of inputs and outputs to
the hammerfall? But maybe my config files (asound... ) have
to be correct first?
Anyway, it will work. Possibly you can post back
your currect
settings from /proc/asound/cards and the output of amixer's listing of
controls and er can get you lined up pretty quickly.
Ok, I will do that tonight.
Thanks again,
-- steve
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3:01 p.m.
On 6/20/06, Stephen Cameron <smcameron(a)yahoo.com> wrote:
Should not be required or useful (as far as I know) for the DIGI9636.
hdspconf and hdspmixer only work with the HDSP cards, not the older
Hammerfall cards.
This is a common mistake. I make it about 40% of the time people write here...
You should not need any config files. I don't use them. There is one
file which is created by the system called /etc/asound.state which
sometimes is outdated by Alsa updates. If that happens you'll get
messages when Alsa starts (probably boot time for you - what distro?)
and the messages will say things like a 'control' has incorrect
values, etc. If that happens just erase asound.state and let the
system create a new one.
Will look in and see if I can help.
Cheers,
Mark
--- Mark Knecht <markknecht(a)gmail.com> wrote:
<SNIP>
Hammerfall,
not HDSP. Does any of this work in his case? It didn't
when I was running my Hammerfall under Linux.
Ah, thank you, that was going to be my next question.
(and this morning, I tried to build hdspmixer from alsa-tools-1.0.11
and it wanted fltk lib... tried the
latest fltk lib v. 1.1xxx and v. 2.xxx something, and though
configure stopped complaining, hdspmixer
would compile with neither version of the fltk libs. Maybe it
needs some specific earlier version? (or newer CVS version?)
It complained about no such member something->fl_widget. (I
can reproduce the exact message when I get home tonight
if anybody here wants me to.)
But it sounds like I might not be able to use hdspmixer
anyhow, even if I could build it.
Nope...
Steve,
I have both an HDSP9652 and a Hammerfall Lite. My Hammerfall is now
in a Windows box so I cannot step you though an exact solution but I
have used it under Linux and it works well. I'm pretty sure some group
of us can answer your questions and get you there pretty quickly.
The Hammerfall will work correctly with Alsa drivers and the Jack
server. I think your most simple answer is as Paul outlined, although
I think that you cannot use hdspconf to set the clock rate of the
Hammerfall unless hdspconf has been updated in some way I'm not aware
of.
To set the card's clock rate will likely require use of some amixer
commands. man amixer and then play around with reading out the
controls in the driver. The correct solution here is to configure
either your Hammerfall or the AW4416 as a clock slave. I don't have
any problems using the ADAT lines to sync clocks so you could try that
to begin with. It will certainly work well enough to test the
connection. You can go to a spdif or Word Clock sync later if you need
it. I'm not sure if you can do Word Clock sync on a DIGI9636 anyway.
That connection may be on the DIGI9652 expander card. I cannot
remember.
Ok, this gives me some hints to play around with, and encouragement.
Thanks a bunch! I imagine when I run, say qjackconnect,
I should expect to see a bunch of inputs and outputs to
the hammerfall? But maybe my config files (asound... ) have
to be correct first? Anyway, it will work. Possibly you can post
back your currect
settings from /proc/asound/cards and the output of amixer's listing of
controls and er can get you lined up pretty quickly.
Ok, I will do that tonight.
10:57 p.m.
--- Mark Knecht <markknecht(a)gmail.com> wrote:
You should not need any config files. I don't use them. There is one
file which is created by the system called /etc/asound.state which
sometimes is outdated by Alsa updates. If that happens you'll get
messages when Alsa starts (probably boot time for you - what distro?)
and the messages will say things like a 'control' has incorrect
values, etc. If that happens just erase asound.state and let the
system create a new one.
Will look in and see if I can help.
Cheers,
Mark
Well, here's what I've tried so far tonight:
Ran jack as root.
Ran qjackctl as root.
--------------
On the setup/Settings page:
Checked: Realime/hw monitor/hw meter.
Left other messages alone.
Priority: 0
Prames/period: 1024 (a guess)
sample rate:22050 (default...)
Periods/Buffer: 2
Port Max: 128
Timeout: 500ms
Interface: hw:1
Dither: none
Audio:duplex
Input device:(default)
output device:(default)
input channels: 0
output channels: 0
input latency: 0
output latency: 0
Latency: 92.9 msec
-----------
The "input channels: 0" and "output channels: 0"
seem suspicious. If I try to change them Jack
won't start and says: "ALSA: cannot set channel
count to 1 for capture" If I leave them zero, jack
will start (but I don't know if that's right.)
On the AW4416:
On the Setup/Patch In page, it looks like
SL2-1 through SL2-8 are mapped to mixer input channels
17-24. For the mixer 1-24/rtn-1-2, I pushed the faders
up. On the "Home" screen, 1-24/rtn, I can monitor the
inputs.
I ran ecasound like so:
ecasound -c -b:1024 -i:planet-zoing-zoing-banjomix.wav -o:jack
Then "start"
Then in qjackctl, I can see ecasound in the "capture" column,
and I routed it to every single output (all 18 of them --
presumably 2 s/pdif outs + 16 ADAT outs.) So I am seeing
the hammerfall having 18 ports, which is encouraging
I got nothing on the AW4416 where I would expect the sound
from the wav file to show up on all 8 ADAT inputs.
That's where things stand right now.
-- steve
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To set the card's clock rate will likely
require use of some amixer
commands. man amixer and then play around with reading out the
controls in the driver. The correct solution here is to configure
either your Hammerfall or the AW4416 as a clock slave. I don't have
any problems using the ADAT lines to sync clocks so you could try that
to begin with. It will certainly work well enough to test the
connection. You can go to a spdif or Word Clock sync later if you need
it. I'm not sure if you can do Word Clock sync on a DIGI9636 anyway.
That connection may be on the DIGI9652 expander card. I cannot
remember.
Ok, this gives me some hints to play around with, and encouragement.
Thanks a bunch! I imagine when I run, say qjackconnect,
I should expect to see a bunch of inputs and outputs to
the hammerfall? But maybe my config files (asound... ) have
to be correct first? Anyway, it will work. Possibly you can post
back your currect
settings from /proc/asound/cards and the output of amixer's listing of
controls and er can get you lined up pretty quickly.
Ok, I will do that tonight.
21 Jun
21 Jun
1:07 a.m.
On 6/20/06, Stephen Cameron <smcameron(a)yahoo.com> wrote:
<SNIP>
Well, here's what I've tried so far tonight:
Ran jack as root.
Ran qjackctl as root.
Let's try running Jack from within QJC. That way you can set up a Jack
profile with settings that make sense and then you can save them.
I suspect, for no good reason, that the mixer and the Hammerfall
wouldn't be happy if the Hammerfall is really running at 22K.
In QJC make sure that you are choosing the right soundcard. Is it
possible that you have a built in sound card that is set up for card 0
and the Hammerfall is actually card 1?
mark@lightning ~ $ cat /proc/asound/cards
0 [CK804 ]: NFORCE - NVidia CK804
NVidia CK804 with ALC850 at 0xda103000, irq 225
1 [DSP ]: H-DSP - Hammerfall DSP
RME Hammerfall HDSP 9652 at 0xda000000, irq 66
mark@lightning ~ $
If yours looks anything like this then the default Jack setting is
running Jack with the built in card. However since you see 18 IO's it
sounds like the Hammerfall and not a built in unless they are virtual
ports...
From within QJC I start Jack here with the command
shown, and again
this is my HDSP9652, not a Hammerfall, but it should be very close
if
not exactly the same, and this is what I see in QJC's message window:
(you do NOT need to run 64/2 for testing...)
20:01:52.019 jackd -R -P80 -p512 -dalsa -dhw:1 -r44100 -p64 -n2
20:01:52.035 JACK was started with PID=4148 (0x1034).
jackd 0.102.14
Copyright 2001-2005 Paul Davis and others.
jackd comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details
JACK compiled with System V SHM support.
loading driver ..
apparent rate = 44100
creating alsa driver ... hw:1|hw:1|64|2|44100|0|0|nomon|swmeter|-|32bit
control device hw:1
configuring for 44100Hz, period = 64 frames, buffer = 2 periods
ALSA: final selected sample format for capture: 32bit little-endian
ALSA: use 2 periods for capture
ALSA: final selected sample format for playback: 32bit little-endian
ALSA: use 2 periods for playback
20:01:54.084 Server configuration saved to "/home/mark/.jackdrc".
Hope this helps somehow,
Mark
1:38 a.m.
Mark Knecht wrote;
Ok, I will have to try that one.
On 6/20/06, Stephen Cameron
<smcameron(a)yahoo.com> wrote:
<SNIP>
Actually I did. After I typed "Run jack as root", I meant to
replace that line with "Run qjackctl as root", not keep both
lines.
Well, here's what I've tried so far tonight:
Ran jack as root.
Ran qjackctl as root.
Let's try running Jack from within QJC. That way you can set up a Jack
profile with settings that make sense and then you can save them.
I suspect, for no good reason, that the mixer and the Hammerfall
wouldn't be happy if the Hammerfall is really running at 22K.
Ok. I don't know what to make of that.
In QJC make sure that you are choosing the right soundcard. Is it
possible that you have a built in sound card that is set up for card 0
and the Hammerfall is actually card 1?
I set in the Setup window of qjackctl -> interface: "hw:1"
but actually, this is a good point, I actually
have _three_ soundcards, I think. The first is
the completely useless one on the motherboard,
the 2nd is an "audigy2 value" and the third is
the RME. So, now that you mention it, I would
think it should be hw:2, but somehow I still
think it's "hw:1" Maybe it's because the motherboard
soundcard is so utterly useless it doesn't even
show up for work? Er, come to think of it I may have
disabled it in the ROM setup of my computer because
it was only causing problems... I can't recall if I
did that or not. I am sure I thought about trying to do
it. In any case, I think "hw:1" is correct, because of this:
[scameron@zuul ~]$ arecord -l
**** List of CAPTURE Hardware Devices ****
card 0: Audigy2 [Audigy 2 Value [SB0400]], device 0: emu10k1 [ADC Capture/Standard PCM
Playback]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: Audigy2 [Audigy 2 Value [SB0400]], device 1: emu10k1 mic [Mic Capture]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: Audigy2 [Audigy 2 Value [SB0400]], device 2: emu10k1 efx [Multichannel Capture/PT
Playback]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: R15 [RME Digi9636 (Rev 1.5)], device 0: RME Digi9636 (Rev 1.5) [RME Digi9636 (Rev
1.5)]
Subdevices: 1/1
Subdevice #0: subdevice #0
[scameron@zuul ~]$
[scameron@zuul ~]$ cat /proc/asound/cards
0 [Audigy2 ]: Audigy2 - Audigy 2 Value [SB0400]
Audigy 2 Value [SB0400] (rev.0, serial:0x10011102) at 0x9000, irq
11
1 [R15 ]: RME9636 - RME Digi9636 (Rev 1.5)
RME Digi9636 (Rev 1.5) at 0xea000000, irq 10
[scameron@zuul ~]$
mark@lightning ~ $ cat /proc/asound/cards
0 [CK804 ]: NFORCE - NVidia CK804
NVidia CK804 with ALC850 at 0xda103000, irq 225
1 [DSP ]: H-DSP - Hammerfall DSP
RME Hammerfall HDSP 9652 at 0xda000000, irq 66
mark@lightning ~ $
If yours looks anything like this then the default Jack setting is
running Jack with the built in card. However since you see 18 IO's it
sounds like the Hammerfall and not a built in unless they are virtual
ports...
Yeah, I set it to "hw:1" so I think it's correct.
From within QJC I start Jack here with the command
shown, and again
this is my HDSP9652, not a Hammerfall, but it should be very close
if
not exactly the same, and this is what I see in QJC's message window:
(you do NOT need to run 64/2 for testing...)
20:01:52.019 jackd -R -P80 -p512 -dalsa -dhw:1 -r44100 -p64 -n2 20:01:52.035 JACK was started with PID=4148 (0x1034).
jackd 0.102.14
Copyright 2001-2005 Paul Davis and others.
jackd comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details
JACK compiled with System V SHM support.
loading driver ..
apparent rate = 44100
creating alsa driver ... hw:1|hw:1|64|2|44100|0|0|nomon|swmeter|-|32bit
control device hw:1
configuring for 44100Hz, period = 64 frames, buffer = 2 periods
ALSA: final selected sample format for capture: 32bit little-endian
ALSA: use 2 periods for capture
ALSA: final selected sample format for playback: 32bit little-endian
ALSA: use 2 periods for playback
20:01:54.084 Server configuration saved to "/home/mark/.jackdrc".
Hope this helps somehow,
Mark
__________________________________________________
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Tired of spam? Yahoo! Mail has the best spam protection around
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3:19 p.m.
On 6/20/06, Stephen Cameron <smcameron(a)yahoo.com> wrote:
Mark Knecht wrote;
Ok, I will have to try that one.
On 6/20/06, Stephen Cameron
<smcameron(a)yahoo.com> wrote:
<SNIP>
Actually I did. After I typed "Run jack as root", I meant to
replace that line with "Run qjackctl as root", not keep both
lines.
Well, here's what I've tried so far tonight:
Ran jack as root.
Ran qjackctl as root.
Let's try running Jack from within QJC. That way you can set up a Jack
profile with settings that make sense and then you can save them.
I suspect, for no good reason, that the mixer and the Hammerfall
wouldn't be happy if the Hammerfall is really running at 22K.
Ok. I don't know what to make of that.
In QJC make sure that you are choosing the right soundcard. Is it
possible that you have a built in sound card that is set up for card 0
and the Hammerfall is actually card 1?
I set in the Setup window of qjackctl -> interface: "hw:1"
but actually, this is a good point, I actually
have _three_ soundcards, I think. The first is
the completely useless one on the motherboard,
the 2nd is an "audigy2 value" and the third is
the RME. So, now that you mention it, I would
think it should be hw:2, but somehow I still
think it's "hw:1" Maybe it's because the motherboard
soundcard is so utterly useless it doesn't even
show up for work? Er, come to think of it I may have
disabled it in the ROM setup of my computer because
it was only causing problems... I can't recall if I
did that or not. I am sure I thought about trying to do
it. In any case, I think "hw:1" is correct, because of this:
[scameron@zuul ~]$ arecord -l
**** List of CAPTURE Hardware Devices ****
card 0: Audigy2 [Audigy 2 Value [SB0400]], device 0: emu10k1 [ADC Capture/Standard PCM
Playback]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: Audigy2 [Audigy 2 Value [SB0400]], device 1: emu10k1 mic [Mic Capture]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 0: Audigy2 [Audigy 2 Value [SB0400]], device 2: emu10k1 efx [Multichannel
Capture/PT
Playback]
Subdevices: 1/1
Subdevice #0: subdevice #0
card 1: R15 [RME Digi9636 (Rev 1.5)], device 0: RME Digi9636 (Rev 1.5) [RME Digi9636 (Rev
1.5)]
Subdevices: 1/1
Subdevice #0: subdevice #0
[scameron@zuul ~]$
[scameron@zuul ~]$ cat /proc/asound/cards
0 [Audigy2 ]: Audigy2 - Audigy 2 Value [SB0400]
Audigy 2 Value [SB0400] (rev.0, serial:0x10011102) at 0x9000, irq
11
1 [R15 ]: RME9636 - RME Digi9636 (Rev 1.5)
RME Digi9636 (Rev 1.5) at 0xea000000, irq 10
[scameron@zuul ~]$
mark@lightning ~ $ cat /proc/asound/cards
0 [CK804 ]: NFORCE - NVidia CK804
NVidia CK804 with ALC850 at 0xda103000, irq 225
1 [DSP ]: H-DSP - Hammerfall DSP
RME Hammerfall HDSP 9652 at 0xda000000, irq 66
mark@lightning ~ $
If yours looks anything like this then the default Jack setting is
running Jack with the built in card. However since you see 18 IO's it
sounds like the Hammerfall and not a built in unless they are virtual
ports...
Yeah, I set it to "hw:1" so I think it's correct.
From within QJC I start Jack here with the command
shown, and again
this is my HDSP9652, not a Hammerfall, but it should be very close
if
not exactly the same, and this is what I see in QJC's message window:
(you do NOT need to run 64/2 for testing...)
20:01:52.019 jackd -R -P80 -p512 -dalsa -dhw:1 -r44100 -p64 -n2 20:01:52.035 JACK was started with PID=4148
(0x1034).
jackd 0.102.14
Copyright 2001-2005 Paul Davis and others.
jackd comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details
JACK compiled with System V SHM support.
loading driver ..
apparent rate = 44100
creating alsa driver ... hw:1|hw:1|64|2|44100|0|0|nomon|swmeter|-|32bit
control device hw:1
configuring for 44100Hz, period = 64 frames, buffer = 2 periods
ALSA: final selected sample format for capture: 32bit little-endian
ALSA: use 2 periods for capture
ALSA: final selected sample format for playback: 32bit little-endian
ALSA: use 2 periods for playback
20:01:54.084 Server configuration saved to "/home/mark/.jackdrc".
Hope this helps somehow,
Mark
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
http://mail.yahoo.com
3:37 p.m.
On 6/20/06, Stephen Cameron <smcameron(a)yahoo.com> wrote:
<SNIP>
Nor I. We need to figure this out. If the card is truly not running at
44/48K then I suspect your mixer will give you no sound, as it seems
to be.
<SNIP>
I suspect, for
no good reason, that the mixer and the Hammerfall
wouldn't be happy if the Hammerfall is really running at 22K.
Ok. I don't know what to make of that. The first is
the completely useless one on the motherboard,
the 2nd is an "audigy2 value" and the third is
the RME. So, now that you mention it, I would
think it should be hw:2, but somehow I still
think it's "hw:1" Maybe it's because the motherboard
soundcard is so utterly useless it doesn't even
show up for work?
No, Linux/Alsa is not like Windows. It does not automatically load
drivers for you. If you do not load a driver for the onboard chip then
it won't work and won't show up in the list, which looking below it
did not. That is NOT the problem.
<SNIP>
[scameron@zuul ~]$ cat /proc/asound/cards
0 [Audigy2 ]: Audigy2 - Audigy 2 Value [SB0400]
Audigy 2 Value [SB0400] (rev.0, serial:0x10011102) at 0x9000, irq
11
1 [R15 ]: RME9636 - RME Digi9636 (Rev 1.5)
RME Digi9636 (Rev 1.5) at 0xea000000, irq 10
[scameron@zuul ~]$
You have two cards recognized by Alsa and the Hammerfall is hw:1
One thing that has caused problems with some RME cards in the past is
the revision of firmward on the card. You seem to have Ver. 1.5 which
appears to be the same one I had at the time:
http://article.gmane.org/gmane.linux.alsa.devel/36449
<COPY>
mark <at> gigastudio ~ $ cat /proc/asound/cards
0 [M2496 ]: ICE1712 - M Audio Audiophile 24/96
M Audio Audiophile 24/96 at 0xa400, irq 10
1 [R15 ]: RME9636 - RME Digi9636 (Rev 1.5)
RME Digi9636 (Rev 1.5) at 0xe2000000, irq 9
2 [CMI8738MC6 ]: CMI8738-MC6 - C-Media PCI CMI8738-MC6
C-Media PCI CMI8738-MC6 (model 55) at 0xd800, irq 11
mark <at> gigastudio ~ $
<COPY>
At this point I might suggest getting either meterbridge or
jackbitscope (hope that's the name) and making sure that you are
actually sending data on the Jack ports. You might also try a simpler
application like Aqualung or alsaplayer as a verification of what you
are doing with ecasound.
I'm a bit lost as to where you might look next. On my HDSP 9652 I Can
explore the /proc/asound/card1 directory a bit and see Alsa reporting
things about the card. Note that unless the card is operating much of
this was not reporting anything. However, after starting Jack I start
getting info, most of which isn't helpful but the last one was. It
shows sample rate:
mark@lightning ~ $ cat /proc/asound/card1/pcm0c/sub0/status
state: RUNNING
trigger_time: 1150911228.996154244
tstamp : 1150911244.400714450
delay : 80
avail : 80
avail_max : 96
-----
hw_ptr : 679376
appl_ptr : 679296
mark@lightning ~ $
mark@lightning ~ $ cat /proc/asound/card1/pcm0c/sub0/sw_params
tstamp_mode: NONE
period_step: 1
sleep_min: 0
avail_min: 64
xfer_align: 64
start_threshold: 0
stop_threshold: 128
silence_threshold: 0
silence_size: 0
boundary: 4611686018427387904
mark@lightning ~ $
mark@lightning ~ $ cat /proc/asound/card1/pcm0c/sub0/hw_params
access: MMAP_NONINTERLEAVED
format: S32_LE
subformat: STD
channels: 26
rate: 44100 (44100/1)
period_size: 64
buffer_size: 128
tick_time: 1000
mark@lightning ~ $
Maybe you can do something similar and see what you discover?
Cheers,
Mark
3:55 p.m.
On 6/21/06, Mark Knecht <markknecht(a)gmail.com> wrote:
On 6/20/06, Stephen Cameron
<smcameron(a)yahoo.com> wrote:
<SNIP>
Nor I. We need to figure this out. If the card is truly not running at
44/48K then I suspect your mixer will give you no sound, as it seems
to be.
<SNIP>
I
suspect, for no good reason, that the mixer and the Hammerfall
wouldn't be happy if the Hammerfall is really running at 22K.
Ok. I don't know what to make of that. Maybe you can do something similar and see what you
discover?
Cheers,
Mark
BTW - You can always take the ADAT cable and connected it from the
output back to an input, send audio out and then see if you can record
audio coming back in. It's a simple test to make sure the
Alsa/Hammerfall set up is actually working. If that fails the problem
is in your Linux setup, or the ADAT cables. If it works then it's
either in the mixer or an interface problem where the mixer and the
Hammerfall are not somehow compatible. (Or how you've configured your
mixer I suppose....
4:09 p.m.
Mark Krecht wrote:
[snip]
BTW - You can always take the ADAT cable and connected
it from the
output back to an input, send audio out and then see if you can record
audio coming back in. It's a simple test to make sure the
Alsa/Hammerfall set up is actually working. If that fails the problem
is in your Linux setup, or the ADAT cables. If it works then it's
either in the mixer or an interface problem where the mixer and the
Hammerfall are not somehow compatible. (Or how you've configured your
mixer I suppose....
Oh, now that's a good idea. I can probably do the same
on the AW as well. Thanks.
I found jackbitscope here, btw:
http://www.ecs.soton.ac.uk/~njl98r/code/ladspa/bitscope/
Thanks for that...
I'll post what I find out tonight.
-- steve
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22 Jun
22 Jun
5:20 a.m.
Wednesday 21 June 2006 20:08, Stephen Cameron:
I found jackbitscope here, btw:
http://www.ecs.soton.ac.uk/~njl98r/code/ladspa/bitscope/
For the record: The author moved the page to
http://www.ecs.soton.ac.uk/~njl98r/code/audio/
and renamed jackbitscope to "bitmeter". Great tool.
2:16 p.m.
--- Stephen Cameron <smcameron(a)yahoo.com> wrote:
Mark Krecht wrote:
[snip]
Well, something came up and I wasn't able to try out as much
as I wanted to last night, but I did try the loopback test
with the AW4416 talking to itself over 50 foot ADAT cable,
and that worked. At least I know it's possible to use the
cables under some circumstance.
-- steve
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BTW - You can always take the ADAT cable and
connected it from the
output back to an input, send audio out and then see if you can record
audio coming back in. It's a simple test to make sure the
Alsa/Hammerfall set up is actually working. If that fails the problem
is in your Linux setup, or the ADAT cables. If it works then it's
either in the mixer or an interface problem where the mixer and the
Hammerfall are not somehow compatible. (Or how you've configured your
mixer I suppose....
Oh, now that's a good idea. I can probably do the same
on the AW as well. Thanks.
I found jackbitscope here, btw:
http://www.ecs.soton.ac.uk/~njl98r/code/ladspa/bitscope/
Thanks for that...
I'll post what I find out tonight.
2:37 p.m.
On 6/22/06, Stephen Cameron <smcameron(a)yahoo.com> wrote:
--- Stephen Cameron <smcameron(a)yahoo.com>
wrote:
Well, that's a good test as it says the ADAT interface for the mixer
is working.
Now you do the same tests with the Hammerfall until you have that
functioning. At that point (hopefully) you'll reattach the Hammerfall
to the mixer and get better results.
good luck,
Mark
Mark Krecht wrote:
[snip]
Well, something came up and I wasn't able to try out as much
as I wanted to last night, but I did try the loopback test
with the AW4416 talking to itself over 50 foot ADAT cable,
and that worked. At least I know it's possible to use the
cables under some circumstance.
-- steve BTW - You can always take the ADAT cable and
connected it from the
output back to an input, send audio out and then see if you can record
audio coming back in. It's a simple test to make sure the
Alsa/Hammerfall set up is actually working. If that fails the problem
is in your Linux setup, or the ADAT cables. If it works then it's
either in the mixer or an interface problem where the mixer and the
Hammerfall are not somehow compatible. (Or how you've configured your
mixer I suppose....
Oh, now that's a good idea. I can probably do the same
on the AW as well. Thanks.
I found jackbitscope here, btw:
http://www.ecs.soton.ac.uk/~njl98r/code/ladspa/bitscope/
Thanks for that...
I'll post what I find out tonight. 
4:06 p.m.
--- Stephen Cameron <smcameron(a)yahoo.com> wrote:
I've been running my studio on a 50 foot snake; ADAT,
keyboard, mouse, spdif, power, etc, for several years.
No problems under any circumstances. Are you
experiencing something weird?
Ron
--- Stephen Cameron <smcameron(a)yahoo.com>
wrote:
same
http://www.ecs.soton.ac.uk/~njl98r/code/ladspa/bitscope/
Mark Krecht wrote:
[snip]
> BTW - You can always take the ADAT cable and
connected it from the
> output back to an input, send audio out and
then
see if you can record
> audio coming back in. It's a simple test
to make
sure the
> Alsa/Hammerfall set up is actually working.
If
that fails the problem
> is in your Linux setup, or the ADAT cables.
If
it works then it's
> either in the mixer or an interface problem
where the mixer and the
> Hammerfall are not somehow compatible. (Or
how
you've configured your
mixer I
suppose....
Oh, now that's a good idea. I can probably do the on the AW as well. Thanks.
I found jackbitscope here, btw:
Thanks for
that...
I'll post what I find out tonight.
Well, something came up and I wasn't able to try out
as much
as I wanted to last night, but I did try the
loopback test
with the AW4416 talking to itself over 50 foot ADAT
cable,
and that worked. At least I know it's possible to
use the
cables under some circumstance. -- steve
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5:12 p.m.
I've been running my studio on a 50 foot snake;
ADAT,
keyboard, mouse, spdif, power, etc, for several years.
No problems under any circumstances. Are you
experiencing something weird?
No, not really. Just seems that most places sell cables
up to 10m, and 10m seems to be the max distance recommended,
and until you came along, I had not encountered anyone using
50 foot ADAT cables so I was a little bit worried that 50
feet might prove to be a problem, so I was glad to be able
to find out it doesn't seem to be a problem.
-- steve
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7:48 p.m.
--- Stephen Cameron <smcameron(a)yahoo.com> wrote:
The only place I see problems on the 50ft snake is
with dual monitors. When the driver asks the monitors
what they are the second one doesn't always return an
answer. My assumption at the time was that the problem
was due to cable length. I wrote the display types
into the xorg.conf file which solves that. I'm no
longer convinced the problem was cable length. It's
probably a driver issue.
Ron
I've been
running my studio on a 50 foot snake;
ADAT,
keyboard, mouse, spdif, power, etc, for several
years.
No problems under any circumstances. Are you
experiencing something weird?
No, not really. Just seems that most places sell
cables
up to 10m, and 10m seems to be the max distance
recommended,
and until you came along, I had not encountered
anyone using
50 foot ADAT cables so I was a little bit worried
that 50
feet might prove to be a problem, so I was glad to
be able
to find out it doesn't seem to be a problem. -- steve
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21 Jun
21 Jun
11:30 a.m.
New subject: [linux-audio-user] Recording ADAT inputs on RME hammerfall 9636/52
Ran jack as root.
Ran qjackctl as root.
--------------
On the setup/Settings page:
Checked: Realime/hw monitor/hw meter.
Left other messages alone.
Priority: 0
Prames/period: 1024 (a guess)
sample rate:22050 (default...)
Periods/Buffer: 2
Port Max: 128
Timeout: 500ms
Interface: hw:1
Dither: none
Audio:duplex
Input device:(default)
output device:(default)
input channels: 0
output channels: 0
input latency: 0
output latency: 0
Latency: 92.9 msec
-----------
The "input channels: 0" and "output channels: 0"
seem suspicious. If I try to change them Jack
won't start and says: "ALSA: cannot set channel
count to 1 for capture" If I leave them zero, jack
will start (but I don't know if that's right.)
its correct. qjackctl could do a better job of presenting the idea that
"zero = use however many channels the hardware offers".
Then in qjackctl, I can see ecasound in the
"capture" column,
and I routed it to every single output (all 18 of them --
presumably 2 s/pdif outs + 16 ADAT outs.) So I am seeing
the hammerfall having 18 ports, which is encouraging
please jack_lsp -c and post the results here, that way we can see
precisely what connections you have set up.
I got nothing on the AW4416 where I would expect the
sound
from the wav file to show up on all 8 ADAT inputs.
and where do you expect them to show up?
1:50 p.m.
--- Paul Davis <paul(a)linuxaudiosystems.com> wrote:
[...]
Ah, I see. Thanks for the explanation.
Ok, I'll do that when I get home tonight. I noticed ecasound
doesn't show up in qjackctl's connection window until I
start it playing... then, after it's playing, I can set up
connections. Wonder if there's a way to set up those connections
before I start the playing?
On the pre-fader mixer input channel meters in the AW4416.
Likely you don't really care too much about the esoteric
workings of the AW4416, but in case you do, there is
a pdf here which is not too bad.
http://www.yamaha-europe.com/yamaha_europe/download/products/20_proaudio/mu…
ugh, big url... http://tinyurl.com/px3ot
(well, the block diagram at the end cuts off the left hand side,
which is unfortunate, as this is where the input meters are
shown.) Anyway, on page 3 of that pdf, where it says
"If you select F1 [1-24/RTN] you can see the meter array for
all 24 possible inputs..." That is the screen I'm looking at.
I have the 8 inputs of the Yamaha ADAT interface assigned
to mixer channels 17-24, so expect to see something on the
meters for those channels when I get all this working.
-- steve
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The
"input channels: 0" and "output channels: 0"
seem suspicious. If I try to change them Jack
won't start and says: "ALSA: cannot set channel
count to 1 for capture" If I leave them zero, jack
will start (but I don't know if that's right.)
its correct. qjackctl could do a better job of presenting the idea that
"zero = use however many channels the hardware offers". Then in qjackctl, I can see ecasound in the
"capture" column,
and I routed it to every single output (all 18 of them --
presumably 2 s/pdif outs + 16 ADAT outs.) So I am seeing
the hammerfall having 18 ports, which is encouraging
please jack_lsp -c and post the results here, that way we can see
precisely what connections you have set up. I got nothing on the AW4416 where I would expect
the sound
from the wav file to show up on all 8 ADAT inputs.
and where do you expect them to show up?
20 Jun
20 Jun
10:05 a.m.
Hi Stephen,
So, another dumb question from a hardheaded linux
audio user
(is there any other kind? ;-)
There's the highly sexy, me. I estimate one good
looker in the group and couldn't careless if there are
others.
I've got an RME hammerfall lite (9636/52)
I've got a Yamaha AW4416 with an MY8AT module (8
channels ADAT i/o).
I've got fibre cables connecting them together.
Do you mean ADAT cables?
The Yamaha seems happy, reporting 8 channels locked
at 48kHz.
What I don't have is any idea how to get anything
good to
happen on the linux side. I've been googling my
brains out
for the last two or three months, and the very best
information
I have been able to find anywhere is this 3 year old
thread:
http://www.music.columbia.edu/pipermail/linux-audio-dev/2003-March/003328.h…
Which in turn recommands alsa-lib/doc/asoundrc.txt
which, looking at that, doesn't really help much at
all, I'm afraid,
you pretty much have to know how it works already in
order to make head or tail of it.
Is there any better information out there? I just
want
to be able to transfer tracks from the PC to the
AW4416, and
back, preferably as many as 8 at a time, or as few
as 1 at
a time, but, judging from my googling, it seems next
to nobody
is doing anything like this.
If your talking about transfering a file rather than
doing a realtime record pass then you are correct;
assuming your using ADAT cables. I'm not aware of an
alternative cable for the 4416 and the RME cards are
ADAT. You can't use those interfaces for file
transfer.
I would estimate less
than ten
people on the entire planet (and that is my inflated
guess)
have an RME hammerfall AND are using the ADAT
interfaces.
Is that about right? Am I that far out in the
weeds?
The AW4416 has a SCSI interface, it runs Linux
(irrelevant for this issue but see Tux during the
boot) so you could figure out the AW filesystem
format, build and mount a drive for both systems and
use that to transfer files. I suppose. Try fat32 or
whatever is used in Windows. There's a Windows
application for extracting files from AW archives. I'd
try running it in Wine.
Otherwise, I imagine Ardour to be the most intuitive
solution for realtime recording. The AW will slave to
Ardour MTC which works well as master for multiple
passes of eight track groups. My last tests of Ardour
as the slave weren't successful...it's been awhile.
I would really like to get this working...
otherwise, I've
just blown $700 (RME + yamaha ADAT interface +
cables)
for nothing , and I'm back to burning CDs to
transfer
tracks between AW and PC.
Are you using an HDR because of mobility?
Ron
Any hints appreciated... thanks.
-- steve
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2:15 p.m.
--- R Parker <rtp405(a)yahoo.com> wrote:
Yes.
That (computer is not in the music room)
and because I've had it for ages, and
it's basically got the majority of an O2R
digital mixer in it, which is not a bad
little piece of gear. Even if I didn't use
the HD in the AW, the mixer section and
preamps/compressors/eq's and so on that are in it
would still be useful.
It's really a very nice, convenient machine for
tracking. Not so good for editing, and there's no nice
way to get things into or out of it. Mostly
what little music I do is internet collaborations,
so I end up with a process like this:
download/decode tracks (mp3 ->wav), burn cds,
import tracks to AW, record my parts, burn cds,
import tracks to pc, encode/upload. Rather tedious.
Would be nice to transform that process to:
download/decode tracks, transfer to AW over ADAT
cables, record my parts, transfer to PC over ADAT
cables, encode/upload tracks.
That's where I'm trying to get to.
-- steve
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Hi Stephen,
[...]
I've got
fibre cables connecting them together.
Do you mean ADAT cables? If your talking about transfering a file rather than
doing a realtime record pass then you are correct;
assuming your using ADAT cables. I'm not aware of an
alternative cable for the 4416 and the RME cards are
ADAT. You can't use those interfaces for file
transfer.
No, I'm not talking about transferring files, I
mean realtime recording. (I *wish* there were a
nice way to transfer tracks as wav files to/from the
AW4416, but there just isn't. I think the newer
(but smaller) AW2816 or 16G might have a usb port
that allows something nice, but I'm not really sure.
I would estimate less
The SCSI interface is only used for backups.
However, the hard drives in the AW are easily
pluggable... might be possible to rig a usb/ide
interface for easy access to the hard drive
from a linux host... BUT... I think this has
already been tried by smarter people
than me. It's a proprietary filesystem called
SFS, iirc.
http://ampfea.org/pipermail/a-list/2001-May/004596.html
Someone has had some success making a program to read
the backup CDs, but it's not open source, and windows-only
(AWextract) and has other limitations (doesn't really
understand the complete backup format.)
than ten
people on the entire planet (and that is my inflated
guess)
have an RME hammerfall AND are using the ADAT
interfaces.
Is that about right? Am I that far out in the
weeds?
The AW4416 has a SCSI interface, it runs Linux
(irrelevant for this issue but see Tux during the
boot) so you could figure out the AW filesystem
format, build and mount a drive for both systems and
use that to transfer files. I suppose. Try fat32 or
whatever is used in Windows. There's a Windows
application for extracting files from AW archives. I'd
try running it in Wine. Otherwise, I imagine Ardour to be the most intuitive
solution for realtime recording. The AW will slave to
Ardour MTC which works well as master for multiple
passes of eight track groups. My last tests of Ardour
as the slave weren't successful...it's been awhile.
This is my plan, eventually.
I would really like to get this working...
otherwise, I've
just blown $700 (RME + yamaha ADAT interface +
cables)
for nothing , and I'm back to burning CDs to
transfer
tracks between AW and PC.
Are you using an HDR because of mobility?
2:38 p.m.
--- Stephen Cameron <smcameron(a)yahoo.com> wrote:
--- R Parker <rtp405(a)yahoo.com> wrote:
It's really a very nice, convenient machine for
tracking.
Until you want to record and monitor 8+ tracks. And
then it's infuriating.
Not so good for editing, and there's no
nice
way to get things into or out of it. Mostly
what little music I do is internet collaborations,
so I end up with a process like this:
download/decode tracks (mp3 ->wav), burn cds,
import tracks to AW, record my parts, burn cds,
import tracks to pc, encode/upload. Rather tedious.
I think you can redesign for a much better studio. Is
your PC hardware capable of running jackd without
xruns?
Would be nice to transform that process to:
download/decode tracks, transfer to AW over ADAT
cables, record my parts, transfer to PC over ADAT
cables, encode/upload tracks.
Don't use the HDR but use the mixer and monitoring
interfaces with i/o routed to the RME card all the
time. Record to the AW HDR at a remote locations;
practice space, bar, etc.
I'm looking at your setup while occupied with
something else. Am I overlooking your reasons for
recording to the AW?
Ron
That's where I'm trying to get to.
-- steve
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3:21 p.m.
--- R Parker <rtp405(a)yahoo.com> wrote:
I think so, eventually. That is my hope. I forget
the processor speed, but it's not ancient, nor the
fastest thing, an Athlon 1700+, I think. 1Gb RAM.
Hard disk seems kind of slow, but that's just gut
feeling I haven't tried measuring, and maybe I'm
comparing against what I'm used to at work (loads of
hardware RAID systems.) I still have work
to do to get a suitable kernel running.
[...]
--- Stephen Cameron <smcameron(a)yahoo.com>
wrote:
I suppose so. That's not been an issue for me as of yet.
The AW is probably overkill for what I do (record myself
along with a few other tracks.)
It's really a very nice, convenient machine
for
tracking.
Until you want to record and monitor 8+ tracks. And
then it's infuriating. Not so good
for editing, and there's no
nice
way to get things into or out of it. Mostly
what little music I do is internet collaborations,
so I end up with a process like this:
download/decode tracks (mp3 ->wav), burn cds,
import tracks to AW, record my parts, burn cds,
import tracks to pc, encode/upload. Rather tedious.
I think you can redesign for a much better studio. Is
your PC hardware capable of running jackd without
xruns? Don't use the HDR but use the mixer and
monitoring
interfaces with i/o routed to the RME card all the
time. Record to the AW HDR at a remote locations;
practice space, bar, etc.
I'm looking at your setup while occupied with
something else. Am I overlooking your reasons for
recording to the AW?
Probably 2 reasons. 1) it's proven to work
for me (albeit inconvenently) The linux
approach is as yet not working (for me, obviously
it's working for others.) So I'm taking
smaller steps first, rather than trying to jump
to the imagined optimal solution immediately.
2) This is the main reason.
The PC isn't in the room with all the
music making gear, and there's not enough
room to move either into the other's room.
Hence my (yet unproven) 50 foot adat cables.
I think they're spec'ed to 10 meters, and
50 feet exceeds that... So essentially *all*
my recording is remote... my living room
is remote, lol.
Well, this may all end up with me buying a
house that is compatible with shorter ADAT cables
and having the PC in the same room as the AW, heh.
Also, I've noticed it's much harder for me to get
a good sounding recording with the computer, guitar
pickups tend to get a lot of RF noise/whine from monitors,
fan motors, etc, esp. with high gain amp settings, and
mics pick up fan noise, like when the CPU fan changes
gears suddenly, and none of those have been as
much of a problem with my AW. But with the AW as a front
end to the RME, maybe that would be alleviated somewhat,
since previously my computer recording experiments have
been through line-in of a cheap sound card (audigy2 value).
-- steve
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