On Mon, 2004-01-05 at 08:57, Takashi Iwai wrote:
> At Wed, 24 Dec 2003 22:20:57 +0100,
> Eckhard Jokisch wrote:
> >
> > Hello guys,
> > I just noticed that the controls ( or their names) of the digital inputs of
> > Live!-drive are changed. If I want ot get sound out of the coaxial inputs I
> >
On Fri, 2004-01-02 at 06:10, Martin Langer wrote:
> On Sat, Dec 06, 2003 at 06:15:19PM -0800, Mark Knecht wrote:
> >
> > Note ADAT3 which do not exist on
> > this card.
> >
> >
> > [EMAIL PROTECTED] mark]$ cat /proc/asound/card0/rme9652
> [...]
> &
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Justin
> Cormack
> Sent: Monday, December 29, 2003 10:50 AM
> To: Mark Knecht
> Cc: alsa list
> Subject: RE: [Alsa-devel] Why does Alsa sometimes not find the HDSP
> 9652? -SOL
t value I can make the boot time and still get the
card to be recognized.
I hope this helps someone in the future not have this problem. Thanks for
your help.
Cheers,
Mark
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Mark Knecht
> Sen
eceived anything on this
from Jaroslav.
Can you offer any insight into the root cause of this problem? Why
should lspci be able to see the card, but Alsa cannot start the driver?
(going back tot he original post.)
Thanks,
Mark
---
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On Mon, 2003-12-22 at 04:43, Jaroslav Kysela wrote:
> > how did this code imply a duplicate control? removing it is
> > *wrong*, utterly wrong. it is an important control for some purposes.
>
> It seems that this control is available only for 9652 not for 9632
> version. The same control was used
On Mon, 22 Dec 2003 02:27 pm, Glenn Maynard wrote:
> ...
> I don't think OpenAL has a future. Do you really want to invest your time
> in it?
From the linked text...
OpenAL is useful mostly for games and multimedia where you want
3D positional audio to be rendered in a realistic fashion. Some
der version and causing problems?
Let me know,
Mark
On Sun, 2003-12-21 at 09:11, Justin Cormack wrote:
> On Sun, 2003-12-21 at 16:03, Mark Knecht wrote:
> > Justin,
> >Hi. Thanks. syslog from that series of boots is attached.
> >
> > Note the first boot, line 304 looks
mes I have to
reboot to get them loaded. There is no pattern. This morning's cold boot
worked jsut fine. I got both the HDSP and the UPS.
I'm not clear about how to mess with the memory allocator, but I'm not
sure that's appropriate after viewing the attached file.
Thanks for your h
Wizard root # /etc/init.d/alsasound start
* Loading ALSA drivers...
* Loading: snd-seq-oss
* Loading: snd-pcm-oss
* Loading: snd-mixer-oss
* Loading: snd-via82xx
* Loading: snd-hdsp
/lib/modules/2.4.20-gentoo-r9/kernel/sound/pci/rme9652/snd-hdsp.o:
init_module:
No such device
Hint: insmod er
On Thu, 2003-12-18 at 19:02, Mark Knecht wrote:
>Has someone intentionally renumbered where audio should be?
>
> Mark
>
Is anyone addressing this?
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ggle.
For some reason it appears that the numbering of the audio ports has
changed. I am very sure that the order used to be ADAT1 on inputs 1-8 in
Jack, ADAT2 on inputs 9-16, and I suppose spdif was on 17-18 although
I've never used it.
I haven't used
On Tue, 2003-12-09 at 06:31, Jaroslav Kysela wrote:
> On Tue, 9 Dec 2003, Mark Knecht wrote:
> > Then how do we explain that in dxs_support=2 my sound is so bad? I'm
> > using 44.1KHz material and supposedly everything is running at 44.1KHz.
>
> To be honest, we do
On Tue, 2003-12-09 at 05:35, Jaroslav Kysela wrote:
> On Tue, 9 Dec 2003, Mark Knecht wrote:
>
> > I am disappointed that there is no option to set the PCM ADC to 44100 in
>
> ADC is for capture not for playback.
Well, of course you are right about this. Is it 5:30AM where
On Tue, 2003-12-09 at 02:21, Sergey Vlasov wrote:
> On Mon, Dec 08, 2003 at 08:08:37PM -0800, Mark Knecht wrote:
> > On Mon, 2003-12-08 at 13:04, Sergey Vlasov wrote:
> >
> > > Most likely you need the dxs_support option instead of ac97_clock.
> > >
> > >
Good
So, I am currently using Mode 4 and getting pretty good results,
although I think Mode 1 would be even better for me if the PCM ADC was
set to the right value.
Cheers,
Mark
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D
hat for other sound chips in this family?
I'll try them all out later today.
Thanks,
Mark
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er and xmms both sound equally bad though OSS. alsaplayer
says it's playing at 44100. alsaplayer through Jack and my HDSP 9652
sounds good.
Any other ideas about what to try?
- Mark
Wizard root # !cat
cat /proc/asound/card0/codec97#0/ac97#0-0
0-0/0: Realtek ALC650 rev 2
Capabilities :
DAC
On Mon, 2003-12-08 at 07:49, Clemens Ladisch wrote:
> Mark Knecht wrote:
> >BTW - how can I get a list of all possible module options for a given
> > card or device? Is there any way to find out about this without reading
> > the code?
>
> modinfo snd-via82xx
ul. I'm having similar problems, so I'll try this out
later today.
BTW - how can I get a list of all possible module options for a given
card or device? Is there any way to find out about this without reading
the code?
Thanks,
Mark
--
RV_CARDS - 1)] = 48000};
> I actuially wrote a mail earlier today on the same subject (called
> viasomehting8235 clock problems or something). It seems to work fine
> after I make this change, btw.
>
> 784 - Michael C. Piantedosi - [EMAIL PROTECTED]
Intere
r could this be a similar problem?
The sound is so bad as to be unlistenable.
Current .asoundrc and modules.conf below...
Thanks,
Mark
pcm.via82xx {
type hw
card 0
}
ctl.via82xx {
type hw
card 0
}
pcm.hdsp {
typ
g like
that.
Cheers,
Mark
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st here that should be looked into.
Thanks,
Mark
pcm.hdsp {
type hw
card 0
}
ctl.hdsp {
type hw
card 0
}
pcm_slave.hdsp {
pcm "hw:0"
channels 26
}
pcm.playback_5_6 {
type dshare
the problem with the OSS emulation
interface and that needs to be looked at?
I also tried building the alsa-xmms plugin, but it fails with
Alsa-1.0.0rc2, so I'll file bug reports on that elsewhere.
Thanks,
Mark
Postscript:
Without support from a developer somewhere who is int
On Sat, 2003-12-06 at 17:28, Mark Knecht wrote:
> Hi,
>The Hammerfall Light driver has a couple of small bugs in the way it
> identifies port status in /proc/asound where it lists the spdif port as
> an ADAT port. I'm not using spdif on this machine so I cannot check to
>
this.
Cheers,
Mark
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is isn't completely right yet.
And, since I'm remote I don't know what this sounds like, but I presume
it's OK. I'll let you know in 5-6 hours.
Anyway, rc2 is much better. aoss is happy even if I'm not. (YET!!!)
Thanks,
Mark
e I will either make the effort to modify the Gentoo
ebuild myself, or wait for Gentoo to catch up. They were only 2 days behind
this time on rc1 making me think an Alsa developer must be right in the
ebuild loop these days.
Thanks for all your help. Hopefully we can get this wor
>
> I am using 1.0.0.rc1 on Gentoo with a 2.4.20-r9 kernel. I got the same
> behavior on 0.9.8 and reported it but never got a response.
>
I didn't see an announcement, but the Alsa page now shows 1.0.0rc2. Are you
suggesting that this might actually have been fixed in rc2? If so, what was
the cau
gt; (You can update this single package independantly of the other ALSA
> packages.)
>
I am using 1.0.0.rc1 on Gentoo with a 2.4.20-r9 kernel. I got the same
behavior on 0.9.8 and reported it but never got a response.
Thanks,
Mark
---
This
On Thu, 2003-12-04 at 05:15, Erik Inge Bolsø wrote:
> On Thu, 4 Dec 2003, Mark Knecht wrote:
> >pcm.playback_5_6 {
> > type dshare
> > slave hdsp
> Uhmm... for _playback_, shouldn't that be "type dmix"? Or am I confused
> again?
>
TW - if you get a chance, take a look at /proc/asound/card0/hdsp (modify
the card number if required and I'm guessing on the hdsp part) and see if
it's listing spdif under the status section? Mine doesn't, or didn't, the
last time I looked. It indicated 3 ADAT ports which this card d
>
> Whenever I try to activate double speed (88.2/96kHz) mode with my RME
> Hammerfall Lite (DIGI 9636), snd_pcm_hw_params fails with a 'Device or
> resource busy' message. This is with the number of channels set to 10,
> since the number of ADAT channels is halved. Is this a known
> defect of
0 now.
>
>
> Takashi
Thanks Takashi. I'm running Codeweavers Crossover Office, so I'll pass this
on to them first. Possibly I'll also try Transgaming's version of Wine which
is likely newer and more up to date.
- Mark
-
els of my HDSP 9652. Or are
you saying I need rc2?
Also, why does aoss change the sample rate from 44.1K to 48K?
Thanks,
Mark
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On Thu, 2003-12-04 at 05:06, Mark Knecht wrote:
> bash-2.05b$ aoss xmms
>
> (At this point the xmms gui opens. I then press play)
>
> xmms: alsa-oss.c:222: oss_dsp_hw_params: Assertion `err >= 0' failed.
> /usr/bin/aoss: line 9: 2841 Aborted
>
bash-2.05b$ aoss xmms
(At this point the xmms gui opens. I then press play)
xmms: alsa-oss.c:222: oss_dsp_hw_params: Assertion `err >= 0' failed.
/usr/bin/aoss: line 9: 2841 Aborted
LD_PRELOAD=${exec_prefix}/lib/libaoss.so $*
bash-2.05b$
Contents of my .asoundrc file:
pcm.hd
** Thu Dec 4 04:41:38 2003
Starting '/opt/cxoffice/bin/wineloader' '--'
'winepath.exe' '--long' '--'
'/home/mark/.cxoffice/dotwine/fake_windows/SIERRA/CAESAR3DEMO/c3.exe'
Argument conversion:
[/home/mark/.cxoffice/dotwine/fake_windows
ps OSS
> automagically does this cleanup? Mark, have you tried using the real
> OSS? Do the pops happen with OSS?
No, I haven't tried real OSS. I'm fairly new to Linux audio and have
never used anything other than Alsa in all it's glory. I don't know how
OSS would be instal
se
on the HDSP 9652 when using OSS applications. The results are the same
as I reported over the last few weeks.
I have not run significant Alsa audio yet to talk about how that's
working. The OSS fix was the one I was most hoping for, and I haven't
had many Alsa problems anyway.
- Thank
dio starts playing. This last group is ADAT-1 out
going to the AI-3 and then into a Pro Tools input.
I hope this helps explain the problem.
Cheers,
Mark
HDSPnoise.xpm.gz
Description: GNU Zip compressed data
> > Then you can compare (maybe via some editor) if the whole
> > .wav file is in the recorded stream (plus some zero samples at
> begging and
> > end of this stream). Or, you can send me your source and recorded files
> > and I'll compare them for you
editor) if the whole
> .wav file is in the recorded stream (plus some zero samples at begging and
> end of this stream). Or, you can send me your source and recorded files
> and I'll compare them for you (only mark which file is master).
>
>
nsynced
there is far more digital noise on ADAT-2, channels 7 & 8 than on the other
6 channels. I can see no reason for this, but it happens.
If you get a chance to respond today then I'll get you what I can
quickly. I'll be traveling after
dphone amps
ADAT-3 <==> Hammerfall Light/WinME/GigaStudio/Reaktor-or-Linux soft
synths
Maybe this is an ADAT-2 problem specifically and wouldn't happen if the
Alesis was on ADAT-1?
Thanks for your help.
Mark
>
> Ok, it's a thing that we would like to eliminate,
pages. It does not happen when starting a native Alsa
application like alsaplayer.
Does this sound at all like what you're experiencing?
- Mark
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Steve
> deRosier
> Sent: Wedn
ashi Iwai:
> > > At Mon, 17 Nov 2003 01:37:02 +,
> > >
> > > Mark Hubbard wrote:
> > > > On Friday 14 Nov 2003 15:31, Takashi Iwai wrote:
> > > > > At Fri, 14 Nov 2003 15:11:22 +, Mark Hubbard wrote:
> > > > > > On Fr
On Friday 14 Nov 2003 15:31, Takashi Iwai wrote:
> At Fri, 14 Nov 2003 15:11:22 +0000, Mark Hubbard wrote:
> > On Friday 14 Nov 2003 13:20, Takashi Iwai wrote:
> > > as a future plan, we'll define dmix as default for el-cheapo
> > > soundcards, but e.g. not fo
On Friday 14 Nov 2003 13:20, Takashi Iwai wrote:
> as a future plan, we'll define dmix as default for el-cheapo
> soundcards, but e.g. not for sb live, which supports such a function
> on hardware.
And what is your definition of an "el-cheapo" soundcard? :)
I object to this future plan as even us
On Tue, 2003-11-04 at 01:41, Frank Barknecht wrote:
> Hallo,
> Mark Knecht hat gesagt: // Mark Knecht wrote:
>
> > # OSS/Free portion - card #1 (HDSP9652)
> > alias sound-service-0-0 snd-mixer-oss
> > alias sound-service-0-1 snd-seq-oss
> > alias sound-servic
r script. Why wouldn't this be better for Alsa to be in CVS?
Thanks!
Cheers,
Mark
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o box you could try copying the hacks
> I made to the Planet CCRMA alsasound script so that it loads anything that has
> not been loaded so far. That should fix the problem.
>
I will take a close look at that script on my Planet box
anks in advance for any ideas.
Cheers,
Mark
### This file is automatically generated by modules-update
#
# Please do not edit this file directly. If you want to change or add
# anything please take a look at the files in /etc/modules.d and read
# the manpage for modules-update.
#
### module
;
> Thomas
>
Hi Thomas,
Welcome back, and thanks for all the efforts. Your work has been so
helpful in getting my system running. It wouldn't be running without
you.
I really look forward to getting all of this into CVS and no longer
having to do patches!
Thanks again.
Che
>The left front channel is always mute when the M-Audio Revolution
>(ice1724) driver is started/restarted with all mixer settings saved
>on exit. The other channels are fine.
>Moving the DAC volume slider in alsamixer unmutes this channel.
I should add:
1) This bug does not exist in version 0.9.
it. What
firmware revision are you using on your card? I'm on firmware rev 65.
(Or 101 in decimal.)
Wizard root # lspci
00:0e.0 Multimedia audio controller: Xilinx Corporation RME Hammerfall
DSP (rev 65)
I hope this helps you with your debug.
Good luck!
Cheers,
Mark
snap.xpm.gz
Description: GNU Zip compressed data
The left front channel is always mute when the M-Audio Revolution
(ice1724) driver is started/restarted with all mixer settings saved
on exit. The other channels are fine.
Moving the DAC volume slider in alsamixer unmutes this channel.
---
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gt; Datasheets for all those.
>
> Cheers
> James
>
Thanks James. I'll keep that in mind.
I guess you are answering my question in the negative? Alsa does not
today support the Realtek chips on this motherboard?
Thanks,
Mark
EN find
out if there's Alsa support?
I don't think so!
I'm looking at motherboards and deciding which one to get. Is ther some
issue here that you know about?
Thanks,
Mark
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with
it after I set it up.
>From the Asus page:
Audio RealtekR ALC650 6CH w/built in HP amplifier
Integrated APU(Audio Processor Unit) SoundStorm?/ DolbyR Digital (AC-3)
Encoder
Cheers,
Mark
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On Wednesday 15 Oct 2003 14:00, Takashi Iwai wrote:
> At Wed, 15 Oct 2003 13:41:40 +0100,
>
> Liam Girdwood wrote:
> > Hi,
> >
> > Unfortunately WM8770 driver development is not in my immediate to do
> > list. However, I do remember someone had done some work on the 8770
> > (possibly on the Aureon
e card.
BTW - what firmware does your HDSP 9652 have? I suppose that a
different revision of firmware could certainly account for differences.
Cheers,
Mark
snap.xpm.gz
Description: GNU Zip compressed data
be of some help.
Cheers,
Mark
On Fri, 2003-10-10 at 01:55, Nick Arnold wrote:
> Okay, as usual I'm being ambiguous. :( Sorry. We attached a signal generator
> to the ADAT input channels and used our software to record the first 12
> capture channels of the HDSP. (Also, I'm tal
On Wed, 2003-10-08 at 20:12, Nick Arnold wrote:
> The symptoms:
>
>
> We connect a sine wave to the first 12 channels and record them, and see
> three channels -- 0, 4, and 8 -- are "corrupted". The remaining channels
> appear to be well-formed and look as we expect them to.
When yo
On Wed, 2003-10-08 at 20:04, Nick Arnold wrote:
> Not sure what the status of things are, so:
>
> - has the patch been applied to the ALSA tree?
Unfortunately, no.
> - alternatively, is there a new version of the patch that will apply
> cleanly to the 0.9.7a release?
Unfortunatel
> >
> >I agree. It does sound that way, but read the thread please. In my
> >machine's case I am dual boot and in my description I said this does not
> >happen under Windows.
>
> that would suggest that the driver is not setting the clock source
> correctly. sync/lock/nolock is done by the h/w, and
On Tue, 2003-10-07 at 06:01, Paul Davis wrote:
> > I agree that they may be related. Possibly my noise only happens
> >continuously when trying to sync to an external 48K source, and possibly
> >this is just a sign of it never syncing. When I set the Pref. Sync. Ref.
> >to ADAT1 and use AutoSync
ts to sync which I presume it should.
So, Alsa developers, what's this about?
Thanks,
Mark
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this and how I might fix it. If you want more technical data or
have some test you think I should try please let me know.
Thanks in advance,
Mark
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Surround sound with the M-Audio Revolution still doesn't work
using latest cvs compile despite the recent "fix" to ak4xxx-adda.c on
11th September:
- fixed the reset of AK4355 codec.
the surround sounds on m-audio revo 7.1 should work now.
- write to only the register image instead of i/o writin
Surround sound with the M-Audio Revolution still doesn't work
using latest cvs compile despite the recent "fix" to ak4xxx-adda.c
on 11th September:
- fixed the reset of AK4355 codec.
the surround sounds on m-audio revo 7.1 should work now.
- write to only the register image instead of i/o writin
boot time and the apcupsd daemon
did not start.
I'll look at all of this in the upcoming week or two and see if I can get
2.6 actually working.
Thanks for your time and for sharing your experience.
- Mark
---
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warnings on alsa-driver tree, due to redefinition of symbols.
> but you can ignore them in general.
Way beyond me, I think, but thanks for the info. Maybe it will sink in
later.
Cheers,
Mark
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> At Wed, 24 Sep 2003 14:24:29 -0700,
> Mark Knecht wrote:
> >
> > I see that in the Gentoo kernel tree (and maybe all 2.6 kernels)
> > linux/drivers/sound has moved to linux/sound.
>
> the OSS drivers are located now on linux/sound/oss.
> the other sound dire
I see that in the Gentoo kernel tree (and maybe all 2.6 kernels)
linux/drivers/sound has moved to linux/sound.
Sorry for wasting bandwidth.
- Mark
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Mark Knecht
> Sent: Wednesday, September 24
or is there
some other way to go about this?
I'm not likely to do this, but there were some questions on the Gentoo
lists, so I got interested.
Thanks,
Mark
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Natasha is having email server problems and asked me to forward this to
the list.
- Mark
On Sun, 2003-09-14 at 09:39, Natasha Barrett wrote:
> For some time I have been trying to get the HDSP 9652 sound card
> working under LINUX (with Pd).
> Everything was fine in an old, slower mac
nowledge Thomas didn't do any
editing on the MIDI side of things, but we didn't talk about that level of
detail much.
I just didn't want anyone to be surprised later if it turns out it has
something to do with the audio side of the hdsp 9652 driver.
- Mark
---
current input data has been read. I
> don't know why it does this, but you may try to remove/disable lines
> 3181, 3182, 3188, and 3189 in hdsp.c.
Thanks Clemens. I'll take a look at this over the weekend probably, alth
to
try.
Thanks,
Mark
> -Original Message-
> From: Paul Davis [mailto:[EMAIL PROTECTED]
> Sent: Thursday, September 11, 2003 6:06 AM
> To: Clemens Ladisch
> Cc: Mark Knecht; Alsa-Devel
> Subject: Re: [Alsa-devel] MIDI getting killed by Jack?
>
>
> >When an int
1 80:0
aconnect 80:1 80:1
aconnect 80:1 130:0
I had no immediate problems driving zynaddsubfx on 130:0. (I think it was
130:0)
Cheers,
Mark
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Mark Knecht
> Sent: Wednesday, September 10, 2003 9:
ked to the MidiSport. I will likely rewire what is hooked to what
this evening and see if I can say that it is only the HDSP input, or
something else.
Thanks,
Mark
64:X - HDSP 9652
80:X - MidiSport 2x2
Wizard root # lspci
00:00.0 Host bridge: VIA Technologies, Inc. VT8366/A/7 [Apollo
KT266/A/33
On Tue, 2003-09-02 at 06:48, Steve Harris wrote:
> On Tue, Sep 02, 2003 at 05:42:01 -0700, Mark Knecht wrote:
> > On Tue, 2003-09-02 at 01:54, Steve Harris wrote:
> > > http://www.m-audio.com/products/m-audio/fw410.php
> > >
> > > Maybe the first supported
any like M-Audio under NDA on the MLAN part of it,
and then possibly release a driver.
If the device was purely a 61883 device it might be a bit easier. There
are some 'early' Linux 61883 drivers out there.
Complicated...
- Mark
---
On Thu, 2003-08-28 at 20:21, Erik de Castro Lopo wrote:
> > also, you can try to swap the card order,
> >
> > options snd-ens1371 index=1,0
>
> Tired this, ran "update-modules" and restarted alsa.
>
> This changes the way that the two cards are listed in the /proc/asound/cards
> output but o
I looked at outputs.
Anyway, maybe some of this will give you some ideas. I hope you find the
solution. I'm interested in what you're trying as I'd like to do it
myself one of these days.
Best of luck,
Mark
---
This sf.n
Clemens,
Thanks very much for writing all this up! I'll study it and maybe come
back with a few questions.
With best regards,
Mark
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED] Behalf Of Clemens Ladisch
> Sent: Thursday, August 28, 2003 8
> >> Mark Knecht is currently using this code with success, but the card
> >> behaved strangely with rev. 104 (0x68) firmware. He had to
> downgrade it to
> >> rev. 101 (0x65) for the card to work properly.
> >>
> >> [and Mark concured he had to dow
e see Paul and Thomas's responses.
spdif does run at 96Khz (apparently) so the two channels are both available.
Sorry for any confusion.
- Mark
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On Thu, 2003-08-28 at 03:26, Nick Arnold wrote:
> > Mark Knecht is currently using this code with success, but the card
> > behaved strangely with rev. 104 (0x68) firmware. He had to downgrade it to
> > rev. 101 (0x65) for the card to work properly.
> >
> > [and Mar
On Wed, 2003-08-27 at 18:19, Mark Knecht wrote:
> OK, so modprobe snd-seq fixed this. I suppose this should be in
> modules.conf?
>
> Thanks,
> Mark
>
OK, by placing snd-seq at the end of my modules.autoload file this is
loaded at boot time and Rosegarden runs fine so
; the moment, please correct me if I'm wrong.
>
> Thomas
In fact, I am testing some MIDI stuff this evening on the HDSP 9652 and
it's working very nicely so far. I'm not far along, but no problems yet.
Cheers,
Mark
-
What causes this? What should have installed or created /dev/snd/seq?
(RME HDSP 9652) MIDI does work using /dev/midi00
Thanks,
Mark
bash-2.05b$ rosegardensequencer
kbuildsycoca running...
rosegardensequencer: Registering with DCOP server
rosegardensequencer: created plugin manager
OK, so modprobe snd-seq fixed this. I suppose this should be in
modules.conf?
Thanks,
Mark
On Wed, 2003-08-27 at 18:13, Mark Knecht wrote:
> What causes this? What should have installed or created /dev/snd/seq?
>
> (RME HDSP 9652) MIDI does work using /dev/midi00
>
> Thanks,
&
Please keep any answers extremely simple. I'm just looking for a little
really basic education.
Thanks in advance,
Mark
---
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receive of audio
2) spdif clocking (lock/sync) indications
spdif transmit is fine at 44.1KHz.
I look forward to hearing how you progress once you get set up to use
this card at 96KHz.
Cheers,
Mark
> -Original Message-
> From: [EMAIL PROTECTED]
> [mailto:[EMAIL PROTECTED]
to the
second card. I'd try this and not worry about the clocking issues, or
give the second card a clock based on the first card's outputs if you
have the right hardware. (spdif clocks, etc.)
Good luck,
Mark
>
> Erik
---
This SF
#x27;ll be around to help
when you do get back to it, and Thomas has made lots of headway with his
tools recently (hdspconf, hdspmixer) so maybe they'll help you get to the
bottom of this.
Best of luck,
Mark
> -Original Message-
> From: Dan Nyborg [mailto:[EMAIL PROTECTED]
>
at RME's site for quite a while
so there could be new stuff too.
Cheers,
Mark
---
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