hi and thanks for the help so far!,
the hdsp module is loaded /sbin/lsmod:
snd-hdsp 46732 0 (autoclean)
cat /proc/asound/cards
0 [default ]: H-DSP - Hammerfall DSP
RME Hammerfall DSP + Multiface at 0xfeae0000, irq 5
the motherboards intel sound chip has now disappeared but thats ok...
i ran alsaconf again and now and everything seems a lot healthier
the hdsploader works (the red Host light is now off) as does hdspconf and hdspmixer
so everything looks fine but when i start pd it hangs for about 20 seconds and then outputs the following errors, then quits.
input channels = 2, output channels = 2
device name hw:0; channels in 2, out 2
audio buffer set to 50
opening sound input...
using non-interleaved audio input
Sample width set to 4 bytes
ALSA: set input channels to 14
ALSA lib pcm_hw.c:324:(snd_pcm_hw_hw_params) SNDRV_PCM_IOCTL_HW_PARAMS failed: Device or resource busy
snd_pcm_hw_params (input): Device or resource busy
pd: pcm.c:4887: snd_pcm_sw_params_current: Assertion `pcm->setup' failed.
Pd: signal 6
socket receive error: Connection reset by peer (104)
Aborted
sorry about this!
rob
Greetings:
I've already written to the usual lists about this problem, but as yet
I've read no solution.
During playback of audio in JACK apps (Snd, Ardour, SC3) the sound wil
often start out fine but after a few minutes it begins to distort in a
"zipping" fashion, like it's out of sync with itself. After a few
minutes it returns to stability. It's starting to be a real problem for
me, so I thought I'd ask here again if anyone else has experienced this
problem and what did you do to fix it.
Here's the system:
Planet CCRMA RH 9 w. kernel 2.4.24-ll
ALSA 1.0.1
JACK 0.94
SBLive Value
gForce2 w. nVidia driver
XFree86 4.3.0
The machine is no screamer, an 800 MHz Duron, but I haven't had this
problem with it before. I have suspicions about the nVidia card and
driver, and I'm also wondering about moving the SBLive to another slot.
Any and all advice on this problem will be vastly appreciated.
Btw, I don't have the problem during playback from XMMS or xine.
Best regards,
dp
On Fri, 6 Feb 2004 linux-audio-user-request(a)music.columbia.edu wrote:
| From: Dave Phillips <dlphilp(a)bright.net>
| Subject: Re: [linux-audio-user] Should I Bother Learning Csound?
| Considering the low cost of each of these systems, why not try them
| all ? You might also want to consider Common Lisp Music, a Lisp-based
| system. Btw, both CLM and SC3 are very much object-oriented languages,
| while Csound and RTcmix are modeled after more procedural languages.
Well, that would be ideal, but I'm trying to get the most power for the
smallest amount of time spent learning it. Given all the comments, I
think I will give Csound a shot. I use java at work, so I'm used to
using a crappy language just because it has good libraries. When writing
similar code over and over, I have resorted to writing my own
template/preprocessor before. If I really end up liking what csound can
do, but hate the language constructs, I might go down that road. I think
I might have seen something like that done already Python, but I can't
recall where right now.
| Finally I would urge the beginner to make a real study of some other
| language, i.e., C/C++, Java, whatever, along with learning a sound &
| music programming language. That assumes the time for such study, but
| consider it time well spent, you'll learn a lot by the inevitable
| comparisons.
|
agreed.
Hello all,
On the GNU/Linux Centre stand at Sounds Expo 2004 (London, 10-12 Feb)
I'm planning to take along a laptop with some DJ software, and I was
going to take my typical DJ set music with me. But - I thought - why
not take some 'Made with Linux' music instead!
Apart from our own stuff, I'd like to take along as much music made by
the community as possible. So please send me links for your oggs and
mp3s, offlist if you like - any genre is fine.
Cheers
Daniel
Hi folks,
I have been using Linux for about 2 years now. Started with MDK 8.0 and
have mostly stuck with them. Using 9.2 now.I'm not a command line expert
at any measure. In fact, if not for the vast improvements in the GUI I
would not have the time to get anywhere with linux.
I started awhile ago hunting around for audio apps spured by Austin
Acton's "Mandrake 9.1 Audio workstation how to" advertised in the
Mandrake emails. I figured out how to setup URPMI and found apps like
Jack, Ardour (Since, like 8 releases ago) Rosegarden, Fluidsynth and a
vast surprising variety of other cool audio stuff. I thank "Thac" for
his efforts on the "Thac's RPMS" pages as he keeps the Mandrake packages
for the audio stuff up to current release more than any other repository
I've found.(I never seem to have much success at building from source).
Anyway, my biggest headache is getting my laptop fine tuned to work at
it's best with Linux Audio under Mandrake. I have used Win XP and
Cakewalk Sonar since its release and I get fairly solid performance out
of it. I use it live for it's midi playback capabilities. (Save midis to
Sonar format maintaining mix, edit data easily, etc.) I would one day
like to use Linux instead but "Live" is where the rubber meets the road!
Cant afford to have crashes and glitches in the middle of a show. I also
use Sonar for recording. As promising as Ardour is, I just dont have the
time to figure out everything. Sorry if that seems non contributing but
it's just the way it is. I can promise that when it is sorted, I WILL
pay/contribute financially to have it, if it meets my needs. Sure would
like to see the inbuilt midi capabilities in it. But I know the focus is
on the general recording and editing side and thats cool. Must be a lot
of work!
I want to try using a spare laptop for live fx processing (Over vocals
on a PA)at this point. Some of the LADSPA plugs seem very cool (Gverb
for one) and I can get Jack-realtime working but dont know if it's
working as good as it could. I have been messing with Ecamegapedal with
Jack.
So heres my stuff:
Dell CPxJ 650mhz, 256megs Ram, 20 gigHD (5400RPM I think)
I installed Andrew Mortens(??) Multimedia kernel RPM and it boots
fine...although I dont know where the scheduling is sitting or if it is
even making a difference.
The Dell Latitude has a Maestro3 PCI audio accelerator chip. (Not real
high end I know but works sufficiently well with Sonar XL)
If I set Jack parameters in Qjackctl any lower than 1024 I get Xrun
mayhem. (at 44100) I have also tried this running from consoles as root
and it does not seem to make much difference. Am I expecting too much?
Anyway, sorry for such a long post but I was told in the Ardour users
list that this was where I should ask all this stuff...
I do luv linux (it's kinda addictive) and would like to completely sever
the umbilical from Bill someday. :)
thanks for any help in advance.
rob canning wrote :
> hi fernando, thomas, list,
>
> thanks very much for your help, i feel like i may be getting close now but am still having problems...
>
> i did an apt get install on the new kernel set it asd default with GRUB
>
> when i do alsaconf i no longer get the insmod errrors but i do still get:
>
> [root@localhost robcanning]# /usr/sbin/alsaconf
> /usr/sbin/alsaconf: line 120: modinfo: command not found
> /usr/sbin/alsaconf: line 127: modinfo: command not found
> /usr/sbin/alsaconf: line 135: modinfo: command not found
>
> when i start alsasound
> [robcanning@localhost robcanning]$ /etc/rc.d/init.d/alsasound start
> ALSA driver already running
> Sound driver snd-hdsp is already loaded
>
>
> when i start alsamixer
> Card: Intel ICH5 ││ Chip: Analog Devices AD1985 ││ Item: Master
>
> and with the -c 1 argument
> [robcanning@localhost robcanning]$ /usr/bin/alsamixer -c 1
> No mixer elems found
>
>
> my modules.conf file looks like this (i have commented out the oss stuff):
>
> # --- BEGIN: Generated by ALSACONF, do not edit. ---
> # --- ALSACONF verion 1.0.1 ---
> alias char-major-116 snd
> alias char-major-14 soundcore
> #alias sound-service-0-0 snd-mixer-oss
> #alias sound-service-0-1 snd-seq-oss
> #alias sound-service-0-3 snd-pcm-oss
> #alias sound-service-0-8 snd-seq-oss
> #alias sound-service-0-12 snd-pcm-oss
> alias snd-card-0ntitlntitled 1ntitled 1ed 1 snd-hdsp
> alias sound-slot-0 snd-intel8x0
> # --- END: Generated by ALSACONF, do not edit. ---
> post-install sound-slot-0 /bin/aumix-minimal -f /etc/.aumixrc -L >/dev/null 2>&1 || :
> pre-remove sound-slot-0 /bin/aumix-minimal -f /etc/.aumixrc -S >/dev/null 2>&1 || :
> alias sound-slot-1 snd-hdsp
> post-install sound-slot-1 /bin/aumix-minimal -f /etc/.aumixrc -L >/dev/null 2>&1 || :
> pre-remove sound-slot-1 /bin/aumix-minimal -f /etc/.aumixrc -S >/dev/null 2>&1 || :
> # -- Keep modules from being autocleaned
> add options -k snd-card-0
>
> when i run the audio software pd the Hamerfall is listed in its audio devices but when i change to it from the default (intel) it crashes with:
>
> input channels = 2, output channels = 2
> alsa: changing output device to agree with input device
> device name hw:1; channels in 2, out 2
> audio buffer set to 50
> ALSA lib pcm_hw.c:1055:(snd_pcm_hw_open) open /dev/snd/pcmC1D0c failed: No such
> device
> snd_pcm_open (input): No such device
> ALSA lib pcm_hw.c:1055:(snd_pcm_hw_open) open /dev/snd/pcmC1D0p failed: No such
> device
> snd_pcm_open (output): No such device
> ALSA lib pcm_hw.c:494:(snd_pcm_hw_start) SNDRV_PCM_IOCTL_START failed: Bad file
> descriptor
> audio I/O stuck... closing audio
>
> ALSA lib pcm_hw.c:751:(snd_pcm_hw_close) close failed
> : Bad file descriptor
> snd_pcm_close (input): Bad file descriptor
> Segmentation fault
> [robcanning@localhost robcanning]$ pd_gui: pd process exited
>
>
> when i do a dmesg i see the lines
> ALSA ../../alsa-kernel/pci/rme9652/hdsp.c:5054: card initialization pending : waiting for firmware
> PCI: Setting latency timer of device 00:1f.5 to 64
> intel8x0: clocking to 48000
>
> the red Host light is permanantly on on the front of the card - it was never like this with a pcmcia card on windows - should that light turn off on linux with a pci card?
>
Hi Rob,
You have to initialize the iobox running the hdsploader command.
After that you should run hdspmixer (a totalmix clone) that will set the
default levels. You'll also want to check hdspconf (windows/mac control
panel clone). All this tools are part of the alsa-tools package.
Hdsploader also need the alsa-firmware package to be installed. I
believe tha Planet installs all this automatically.
Note that you can't use alsamixer with the hdsp, but you can use amixer.
(googling for hdsp+linux should lead you to more information on this)
Note also that the card can only play 24/32 bits non-interleaved
streams. If you want to play 16 bits and/or interleaved files you have
to use the alsa-lib's plug layer.
Here are a few examples :
aplay -Dhw:X,0 32or24bits.wav
aplay -Dplughw:X,0 16bits.wav
where X is the hdsp alsa card number.
Thomas
hi fernando, thomas, list,
thanks very much for your help, i feel like i may be getting close now but am still having problems...
i did an apt get install on the new kernel set it asd default with GRUB
when i do alsaconf i no longer get the insmod errrors but i do still get:
[root@localhost robcanning]# /usr/sbin/alsaconf
/usr/sbin/alsaconf: line 120: modinfo: command not found
/usr/sbin/alsaconf: line 127: modinfo: command not found
/usr/sbin/alsaconf: line 135: modinfo: command not found
when i start alsasound
[robcanning@localhost robcanning]$ /etc/rc.d/init.d/alsasound start
ALSA driver already running
Sound driver snd-hdsp is already loaded
when i start alsamixer
Card: Intel ICH5 ││ Chip: Analog Devices AD1985 ││ Item: Master
and with the -c 1 argument
[robcanning@localhost robcanning]$ /usr/bin/alsamixer -c 1
No mixer elems found
my modules.conf file looks like this (i have commented out the oss stuff):
# --- BEGIN: Generated by ALSACONF, do not edit. ---
# --- ALSACONF verion 1.0.1 ---
alias char-major-116 snd
alias char-major-14 soundcore
#alias sound-service-0-0 snd-mixer-oss
#alias sound-service-0-1 snd-seq-oss
#alias sound-service-0-3 snd-pcm-oss
#alias sound-service-0-8 snd-seq-oss
#alias sound-service-0-12 snd-pcm-oss
alias snd-card-0ntitlntitled 1ntitled 1ed 1 snd-hdsp
alias sound-slot-0 snd-intel8x0
# --- END: Generated by ALSACONF, do not edit. ---
post-install sound-slot-0 /bin/aumix-minimal -f /etc/.aumixrc -L >/dev/null 2>&1 || :
pre-remove sound-slot-0 /bin/aumix-minimal -f /etc/.aumixrc -S >/dev/null 2>&1 || :
alias sound-slot-1 snd-hdsp
post-install sound-slot-1 /bin/aumix-minimal -f /etc/.aumixrc -L >/dev/null 2>&1 || :
pre-remove sound-slot-1 /bin/aumix-minimal -f /etc/.aumixrc -S >/dev/null 2>&1 || :
# -- Keep modules from being autocleaned
add options -k snd-card-0
when i run the audio software pd the Hamerfall is listed in its audio devices but when i change to it from the default (intel) it crashes with:
input channels = 2, output channels = 2
alsa: changing output device to agree with input device
device name hw:1; channels in 2, out 2
audio buffer set to 50
ALSA lib pcm_hw.c:1055:(snd_pcm_hw_open) open /dev/snd/pcmC1D0c failed: No such
device
snd_pcm_open (input): No such device
ALSA lib pcm_hw.c:1055:(snd_pcm_hw_open) open /dev/snd/pcmC1D0p failed: No such
device
snd_pcm_open (output): No such device
ALSA lib pcm_hw.c:494:(snd_pcm_hw_start) SNDRV_PCM_IOCTL_START failed: Bad file
descriptor
audio I/O stuck... closing audio
ALSA lib pcm_hw.c:751:(snd_pcm_hw_close) close failed
: Bad file descriptor
snd_pcm_close (input): Bad file descriptor
Segmentation fault
[robcanning@localhost robcanning]$ pd_gui: pd process exited
when i do a dmesg i see the lines
ALSA ../../alsa-kernel/pci/rme9652/hdsp.c:5054: card initialization pending : waiting for firmware
PCI: Setting latency timer of device 00:1f.5 to 64
intel8x0: clocking to 48000
the red Host light is permanantly on on the front of the card - it was never like this with a pcmcia card on windows - should that light turn off on linux with a pci card?
thanks again,
rob
www.robcanning.utvinternet.com
Hi,
We have a couple of machines that we are having problems with. Has
anyone been able to sucessfully use a Digigram VX222 with the ALSA
drivers?
What kernel should I use? Patches?
We are getting hard lockups of our system. We have turned off all power
management. It seems to be a problem with activity on either the sound
card driver and/or the NIC driver (e100). The board is an Intel
S845WD1-E.
At this point if we do not come up with a solution by next week, the
higher ups are going to make us use Win2k. Ugh.
--
Torleiv Ringer
Broadcast Systems Analyst
Minnesota Public Radio
http://www.mpr.org