[Alsa-devel] Compiling but not installing Alsa
Hi, I did my first Alsa CVS download this morning just to take a look at the code. I am a PlanetCCRMA user, but wanted to know more about Alsa. I have not yet found a README or INSTALL file on how to do a build. Are there any written instructions in the CVS tree? I looked at the Makefile, which I can't read well, and I get the feeling that the default operation would be to build AND install the code. As a RPM based user I would like to build the code, but not install it until the Planet updates the appropriate RPM. Is this possible with the current CVS code? Thanks, Mark --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP
Hi,
Where are the definitions for PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP and
PCI_VENDOR_ID_XILINX kept in the alsa code?
Thanks,
Mark
static struct pci_device_id snd_hdsp_ids[] __devinitdata = {
{
.vendor= PCI_VENDOR_ID_XILINX,
.device= PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP,
.subvendor = PCI_ANY_ID,
.subdevice = PCI_ANY_ID,
}, /* RME Hammerfall-DSP */
{ 0, },
};
---
This sf.net email is sponsored by:ThinkGeek
Welcome to geek heaven.
http://thinkgeek.com/sf
___
Alsa-devel mailing list
[EMAIL PROTECTED]
https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP
Thanks Paul.
I'm still struggling with getting this new card going. Fernanado and I
are working through the issues one at a time by doing a Planet RPM for
the newest version alsa and have applied the one line patch that you
pointed me at the other day. However, I'm still not running.
1) Fernando applied the one line patch applied:
--- hdsp.c-orig 2002-12-03 19:28:40.0 +
+++ hdsp.c 2002-12-03 19:28:06.0 +
@@ -2966,6 +2966,7 @@
switch (rev 0xff) {
case 0xa:
+ case 0x64:
/* hdsp_initialize_firmware() will reset this */
hdsp-card_name = RME Hammerfall DSP;
break;
2) Alsaconf works, sort of. modules.conf gets built, but my machine will
not reboot after running it. Don't know why.
3) When I boot I see the following message in /var/log/messages
Dec 9 12:39:40 Godzilla kernel: Hammerfall memory allocator: buffers
allocated for 1 cards
Dec 9 12:39:40 Godzilla kernel: RME Hammerfall-DSP: no cards found
Dec 9 12:39:40 Godzilla insmod:
/lib/modules/2.4.19-1.ll/kernel/drivers/sound/pci/rme9652/snd-hdsp.o:
init_module: No such device
4) lspci -v shows the card is there:
00:0f.0 Multimedia audio controller: Xilinx, Inc.: Unknown device 3fc5
(rev 64)
Flags: bus master, medium devsel, latency 32, IRQ 10
Memory at f600 (32-bit, non-prefetchable) [size=64K]
I'm completely puzzled. What are we doing wrong?
Thanks,
Mark
On Mon, 2002-12-09 at 20:11, Paul Davis wrote:
Where are the definitions for PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP and
PCI_VENDOR_ID_XILINX kept in the alsa code?
one of two places. either in the kernel source (if you have a much,
much newer kernel (2.5)) or at the top of either rme9652.c or hdsp.c
(there are conditional #define's there to check if they are already
defined in the kernel's PCI ID header.
--p
---
This sf.net email is sponsored by:ThinkGeek
Welcome to geek heaven.
http://thinkgeek.com/sf
___
Alsa-devel mailing list
[EMAIL PROTECTED]
https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] PCI_DEVICE_ID_XILINX_HAMMERFALL_DSP
We will recheck. I have immense faith in Fernando, but everyone makes mistakes. On Mon, 2002-12-09 at 21:12, Paul Davis wrote: 3) When I boot I see the following message in /var/log/messages Dec 9 12:39:40 Godzilla kernel: Hammerfall memory allocator: buffers allocated for 1 cards Dec 9 12:39:40 Godzilla kernel: RME Hammerfall-DSP: no cards found Dec 9 12:39:40 Godzilla insmod: /lib/modules/2.4.19-1.ll/kernel/drivers/sound/pci/rme9652/snd-hdsp.o: init_module: No such device are you sure you have the new module installed? i know of at least 2 people using the patch you have used that have got their new 9652's working. --p --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] HDSP 9652 - a couple of small (?) issues
Hi, I'm finding that I seem to have two problems with this new card: 1) I am unable to turn down the volume with alsamixer. All the way up or down, the volume is always very loud. Has anyone else seen this? Is there some other tool which will actually control the volume? 2) If I use alsamixer and set volume bars and then use 'alsactl store' to store the levels, if I use 'alsactl restore' the restore process always sets channel 0 back to 0. Other channels are restored at their saved levels. Is alsactl the right tool to use to accomplish this? I'm not at all clear who to report this to. Who works on alsactl and alsamixer? Thanks, Mark --- This sf.net email is sponsored by: With Great Power, Comes Great Responsibility Learn to use your power at OSDN's High Performance Computing Channel http://hpc.devchannel.org/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] possible problems with rc6 aplay
Patrick,
I'm not an Alsa expert so take all of this with a grain of salt. The
difference between 48K and 44.1K is indeed about a whole step, so that's
consistent with your results. You have 48000 samples that are supposed
to take one second to play, but you are taking more than one second to
play then. The result is the output tuning is low.
Since CDs are ALWAYS 44.1K, this would make sense when you burn a CD.
You could get around this by resampling the 48K input down to 44.1K.
There is some open source software for doing that.
This will change the quality of the sound a bit, and is the main
reason I always work at 44.1K.
I don't remember how to do it, but there is an option I've seen in
some .asoundrc files that allows you to set the frequency of the
Hammerfall. I have two Hammerfalls, so I suppose I had better learn to
do that one of these days.
Good luck,
Mark
On Sun, 2002-12-15 at 13:59, patrick reardon wrote:
hi everyone:
i'm running on a PIII with kernel 2.4.18 and Alsa 0.9.0rc6 and a Hammerfall 9636
card.
Alsa has been working fine for the last year, or so it seems. recently a scsi CD
burner
was installed. i have some recordings of live performances made with arecord,
version
0.9.0beta8a. they play back just fine, but when i tried to burn them to CD, they
were low
by about 2 to 3 half steps.
Joerg Shilling suggested that Alsa was writing the wrong headers. so i upgraded to
rc6
and on the first try on each of the old WAV files, aplay also played them too
slowly.
however, on subsequent runs, everything was fine again. i don't understand this
behaviour
at all.
someone on LAU suggested that since it was too low by about 2-3 half steps, data was
being
recorded at 48000 but Alsa thought it was at 44100.
info in /proc/asound/hammerfall/rme9652:
snip-
.
.
Latency: 4096 samples (2 periods of 16384 bytes)
Hardware pointer (frames): 0
Passthru: no
Clock mode: autosync
Pref. sync source: ADAT1
IEC958 input: Coaxial
IEC958 output: Coaxial only
IEC958 quality: Consumer
IEC958 emphasis: off
IEC958 Dolby: off
IEC958 sample rate: error flag set
ADAT Sample rate: 44100Hz
.
.
-snip-
for months up until about an hour ago the ADAT sample rate read 48000. in that hour
i
changed my .asoundrc from
---snip---
pcm.hammerfall { #hammerfall is the alias for snd-rme9652 in
/etc/modules.conf
type hw
card0
}
ctl.hammerfall {
type hw
card0
}
---snip---
to the following
---snip---
pcm.rme9652 { #changed from hammerfall to rme9652 on 12.15.2002
type hw
card 0
}
ctl.rme9652 { #same as above comment
type hw
card 0
}
-snip-
after the .asoundrc change i recorded a fresh WAV and burned it to CD but with the
same
problem -- too slow. also, with the new .asoundrc, version rc6 plays WAV's recorded
with
the old .asoundrc and version rc6 a little too fast. i'm at a loss for new ideas to
debug
this.
can anyone enlighten me about this, or does anyone know where i can download some
reference WAV files (for example, a middle C tone) to check whether the burning
problem
might involve Alsa or whether it's something else in my setup?
any pointers would be greatly appreciated.
tia,
patrick
---
This sf.net email is sponsored by:
With Great Power, Comes Great Responsibility
Learn to use your power at OSDN's High Performance Computing Channel
http://hpc.devchannel.org/
___
Alsa-devel mailing list
[EMAIL PROTECTED]
https://lists.sourceforge.net/lists/listinfo/alsa-devel
---
This sf.net email is sponsored by:
With Great Power, Comes Great Responsibility
Learn to use your power at OSDN's High Performance Computing Channel
http://hpc.devchannel.org/
___
Alsa-devel mailing list
[EMAIL PROTECTED]
https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] possible problems with rc6 aplay
Paul, I'm using two Hammerfalls in separate boxes. Please try to come up with a solution, either automatically or by asking questions in some configuration process, that allows two Linux boxes to choose which to make the master. It is important in my case. Thanks, Mark On Sun, 2002-12-15 at 19:13, Paul Davis wrote: Latency: 4096 samples (2 periods of 16384 bytes) Hardware pointer (frames): 0 Passthru: no Clock mode: autosync Pref. sync source: ADAT1 IEC958 input: Coaxial IEC958 output: Coaxial only IEC958 quality: Consumer IEC958 emphasis: off IEC958 Dolby: off IEC958 sample rate: error flag set ADAT Sample rate: 44100Hz if you're hammerfall is configured as shown above (and no, the name change makes no difference), then the SR that it uses will be determined by your external converter connected to the first ADAT port. nothing that ALSA does (or any program using ALSA does) will alter the SR. thats because you are synced to ADAT1, not the card's internal clock, thus the SR is determined by the clock signal arriving at ADAT1, which presumably comes from a converter somewhere back up the ADAT chain. its been on my to-do list for some time to make master the default clock mode on the hammerfall, which avoids any ambiguity about the sample rate used by the card. i've held back because its really not the right option for most studio-ish users, who have external converters that probably have rate switches on them and they expect the hammerfall to follow the switch setting. does any of this make it any clearer? its really a bit of problem that the rate setting code doesn't do a full 100% check on all this stuff. an app can set the rate to 44100, and appear to have succeeded, but it will have no difference on the actual rate if the sync source is not the clock's internal clock. this is true, btw, for most digital cards. if you tried to record at 44100, but your external converters were running at 48kHz (as you suggest they have been), then the recordings will be at 48kHz with the sync source set as shown above. --p --- This sf.net email is sponsored by: With Great Power, Comes Great Responsibility Learn to use your power at OSDN's High Performance Computing Channel http://hpc.devchannel.org/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel --- This sf.net email is sponsored by: With Great Power, Comes Great Responsibility Learn to use your power at OSDN's High Performance Computing Channel http://hpc.devchannel.org/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] possible problems with rc6 aplay
Martin, That might certainly be an answer. How would this amixer switch get st in the first place? I wouldn't mind doing it by hand once as long as it was then loaded after that. I'm having an interesting problem with this setup with now that's probably based in this area. If I bring up these two systems with the main DAW in Linux, and the slave system in Windows everything is fine. The DAW controls the frequency via my running jack, but even at first boot the two sides lock together just fine. If I then boot the DAW into Windows, the two sides start making noise through the speakers, and if I look at the RME app in Windows on the slave machine, the frequency is bouncing around and so is the mode saying it's master or slave. The worst part is I get ugly noise out of my speakers unless I tell one f the two Windows machines what mode to be in. Is sort of makes sense... Going back into Linux solves the noise problem. Mark On Mon, 2002-12-16 at 02:17, Martin Langer wrote: On Sun, Dec 15, 2002 at 08:38:54PM -0800, Mark Knecht wrote: Paul, I'm using two Hammerfalls in separate boxes. Please try to come up with a solution, either automatically or by asking questions in some configuration process, that allows two Linux boxes to choose which to make the master. It is important in my case. What about the amixer switch? You can use it for switching between master, world, ... modes, but I have only small personal experiences with external hardware using rme32. Another problem I see is the frequency of your master mode. In my opinion you can't set your card to master mode without defining it's frequency before. On rme32 I have three master three modes (32/44.1/48 kHz). If you have a freshly loaded driver and switch to master mode at first it's output frquency is totally undefined. But if you play at first some audio stuff with your rme32 it's no problem and the card uses this last frequency. But using master clock mode without defining a frequency before isn't plausible for me and defining one master mode for each frequency was only a quick solution by me. Any comments or better solutions? martin Thanks, Mark On Sun, 2002-12-15 at 19:13, Paul Davis wrote: Latency: 4096 samples (2 periods of 16384 bytes) Hardware pointer (frames): 0 Passthru: no Clock mode: autosync Pref. sync source: ADAT1 IEC958 input: Coaxial IEC958 output: Coaxial only IEC958 quality: Consumer IEC958 emphasis: off IEC958 Dolby: off IEC958 sample rate: error flag set ADAT Sample rate: 44100Hz if you're hammerfall is configured as shown above (and no, the name change makes no difference), then the SR that it uses will be determined by your external converter connected to the first ADAT port. nothing that ALSA does (or any program using ALSA does) will alter the SR. thats because you are synced to ADAT1, not the card's internal clock, thus the SR is determined by the clock signal arriving at ADAT1, which presumably comes from a converter somewhere back up the ADAT chain. its been on my to-do list for some time to make master the default clock mode on the hammerfall, which avoids any ambiguity about the sample rate used by the card. i've held back because its really not the right option for most studio-ish users, who have external converters that probably have rate switches on them and they expect the hammerfall to follow the switch setting. does any of this make it any clearer? its really a bit of problem that the rate setting code doesn't do a full 100% check on all this stuff. an app can set the rate to 44100, and appear to have succeeded, but it will have no difference on the actual rate if the sync source is not the clock's internal clock. this is true, btw, for most digital cards. if you tried to record at 44100, but your external converters were running at 48kHz (as you suggest they have been), then the recordings will be at 48kHz with the sync source set as shown above. --p --- This sf.net email is sponsored by: With Great Power, Comes Great Responsibility Learn to use your power at OSDN's High Performance Computing Channel http://hpc.devchannel.org/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel --- This sf.net email is sponsored by: With Great Power, Comes Great Responsibility Learn to use your power at OSDN's High Performance Computing Channel http://hpc.devchannel.org/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel -- 2b|!2b
Re: [Alsa-devel] possible problems with rc6 aplay
Patrick, I believe the AI-3 operates at 48K if it is not receiving a clock via it's ADAT input. If the ADAT input is applied and provides 44.1K, then it is my understanding that the AI-3 operates at 44.1K. Mark On Mon, 2002-12-16 at 14:53, patrick reardon wrote: yes, thnx, it's much clearer now. my converter is external with no rate switches and from the manual i just discovered it always operates at 48000 (Alesis AI-3). i'm still uncertain how to change the card's configuration though. alsactl doesn't seem to provide an obvious way to do this (looking at the man page). but my asound.state file has --- This sf.net email is sponsored by: With Great Power, Comes Great Responsibility Learn to use your power at OSDN's High Performance Computing Channel http://hpc.devchannel.org/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: alsasound init script (Re: [Alsa-devel] possible problems withrc6 aplay )
On Mon, 2002-12-16 at 18:51, Paul Davis wrote: i think it would something like this: options snd-hdsp snd_index=0 options snd-usb-foo snd_index=1 i'm sure that takashi or jaroslav will correct me if i got this wrong. --p Paul, This makes perfect sense, and it isn't what I did. (!!) The PlanetCCRMA has a Nano-HOWTO on how to install the MidiMan 2X2 by hand. It's a little USB-based MIDI interface (not a sound card) that is not recognized by alsaconf, so we do a bit of editing by hand. alsaconf sets up modules.conf for the HDSP # --- BEGIN: Generated by ALSACONF, do not edit. --- # --- ALSACONF verion 0.9.0 --- alias char-major-116 snd alias snd-card-0 snd-hdsp alias char-major-14 soundcore alias sound-slot-0 snd-card-0 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 options snd major=116 cards_limit=1 device_mode=0666 options snd-hdsp index=0 # --- END: Generated by ALSACONF, do not edit. --- We then modify one line in the file to look like this: options snd major=116 cards_limit=2 device_mode=0666 and we also do the following: SNIP from the Planet add usb-midi and audio to the /etc/hotplug/blacklist file So that the OSS audio and usb-midi modules are not automatically loaded when the device reconnects after the firmware download. Add ``usb-midi'' and ``audio'' in separate lines at the end of the list of blacklisted modules in that file. End SNIP I think, according to your info, that the problem is caused by not having some sort of options snd-midiman index=1 line. That makes sense to me, except that I don't know what to put there since there actually isn't a driver. The goal is to get the system to make some devices in /dev/snd. This works fine on a cold boot, but fails sometimes on a warm boot. (At least I think it does, since sometimes I get pcmC1D0 when I have no pcmC0D0 Thanks, Mark --- This sf.net email is sponsored by: With Great Power, Comes Great Responsibility Learn to use your power at OSDN's High Performance Computing Channel http://hpc.devchannel.org/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] alsaplayer not working with alsa?
Hi, When running alsaplayer version 0.99.73 without jack, I'm seeing this failure: [mark@Godzilla mark]$ alsaplayer -v alsaplayer 0.99.73 [mark@Godzilla mark]$ alsaplayer alsaplayer: pcm.c:6293: snd_pcm_unlink_ptr: Assertion `0' failed. AlsaPlayer interrupted by signal 6 [mark@Godzilla mark]$ alsaplayer works very well when jack is running. Anyone else seeing this? Thanks, Mark --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [linux-audio-user] Re: [Alsa-devel] alsaplayer not working withalsa?
Thanks Steve. Merry Xmas! On Mon, 2002-12-23 at 05:34, Steve Harris wrote: On Sun, Dec 22, 2002 at 01:19:05 -0800, Mark Knecht wrote: [mark@Godzilla mark]$ alsaplayer alsaplayer: pcm.c:6293: snd_pcm_unlink_ptr: Assertion `0' failed. AlsaPlayer interrupted by signal 6 [mark@Godzilla mark]$ alsaplayer works very well when jack is running. Anyone else seeing this? Yes. I suspect that Fernando needs to update alsaplayer. - Steve --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [linux-audio-user] Re: [Alsa-devel] alsaplayer not workingwith alsa?
No, just standard Intel hardware... On Mon, 2002-12-23 at 06:05, Takashi Iwai wrote: Hi, if you're using a ppc snd-powermac driver, then it's likely a long-standing bug of alsa-lib (somehwere deep inside). it appears only on ppc with alsaplayer. the maintainer is me, but, sorry, atm i haven't had a hardware for debugging this... have a merry christmas! Takashi At 23 Dec 2002 05:46:45 -0800, Mark Knecht wrote: Thanks Steve. Merry Xmas! On Mon, 2002-12-23 at 05:34, Steve Harris wrote: On Sun, Dec 22, 2002 at 01:19:05 -0800, Mark Knecht wrote: [mark@Godzilla mark]$ alsaplayer alsaplayer: pcm.c:6293: snd_pcm_unlink_ptr: Assertion `0' failed. AlsaPlayer interrupted by signal 6 [mark@Godzilla mark]$ alsaplayer works very well when jack is running. Anyone else seeing this? Yes. I suspect that Fernando needs to update alsaplayer. - Steve --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [linux-audio-user] Re: [Alsa-devel] alsaplayer notworking with alsa?
Takashi-san,
I am attaching my current asound.state file. The installed card is an
RME HDSP 9652, their new card. My current modules.conf looks like:
alias parport_lowlevel parport_pc
alias eth0 eepro100
alias usb-controller usb-uhci
# --- BEGIN: Generated by ALSACONF, do not edit. ---
# --- ALSACONF verion 0.9.0 ---
alias char-major-116 snd
alias snd-card-0 snd-hdsp
alias char-major-14 soundcore
alias sound-slot-0 snd-card-0
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
options snd major=116 cards_limit=3 device_mode=0666
options snd-hdsp index=0
options snd-usb-audio index=1
# --- END: Generated by ALSACONF, do not edit. ---
# -- Keep modules from being autocleaned
add options -k snd-card-0
add options -k snd-card-1
The alsaplayer failure when jack is not running looks like:
[mark@Godzilla mark]$ alsaplayer
alsaplayer: pcm.c:6293: snd_pcm_unlink_ptr: Assertion `0' failed.
AlsaPlayer interrupted by signal 6
[mark@Godzilla mark]$
Please let me know if thee is more information you'd like to look at.
Thanks,
Mark
On Tue, 2003-01-07 at 07:58, Takashi Iwai wrote:
At 27 Dec 2002 14:56:22 -0800,
Mark Knecht wrote:
No, just standard Intel hardware...
well, then you have an exotic one :)
could you show /etc/asound.state?
perhaps the card lacks of some mixer controls which are required.
thanks,
Takashi
state.DSP {
control.1 {
comment.access 'read write'
comment.type IEC958
iface PCM
name 'IEC958 Playback Default'
value
''
}
control.2 {
comment.access 'read write inactive'
comment.type IEC958
iface PCM
name 'IEC958 Playback PCM Stream'
value
''
}
control.3 {
comment.access read
comment.type IEC958
iface MIXER
name 'IEC958 Playback Con Mask'
value
'3b00'
}
control.4 {
comment.access read
comment.type IEC958
iface MIXER
name 'IEC958 Playback Pro Mask'
value
'1f00'
}
control.5 {
comment.access 'read write'
comment.type INTEGER
comment.range '0 - 65536 (step 1)'
iface PCM
name Mixer
value.0 0
value.1 0
value.2 0
}
control.6 {
comment.access 'read write'
comment.type ENUMERATED
comment.item.0 ADAT1
comment.item.1 Coaxial
comment.item.2 Internal
iface PCM
name 'IEC958 Input Connector'
value Internal
}
control.7 {
comment.access 'read write'
comment.type BOOLEAN
iface PCM
name 'IEC958 Output also on ADAT1'
value false
}
control.8 {
comment.access 'read write'
comment.type ENUMERATED
comment.item.0 Internal
comment.item.1 Word
comment.item.2 'ADAT Sync'
comment.item.3 IEC958
comment.item.4 ADAT1
comment.item.5 ADAT2
comment.item.6 ADAT3
iface PCM
Re: [Alsa-devel] Re: HDSP 9652 MIDI - A timing disaster?
On Mon, 2003-01-13 at 09:04, Clemens Ladisch wrote: Mark Knecht wrote: I recently purchased an RME HDSP 9652 card. The card is working fine for audio, but the MIDI interface is a timing disaster. The interface works, but won't keep time. A 2 minute song is Rosegarden takes abut 2:45 to play every time. You can hear how the HDSP isn't delivering closely spaced MIDI events together, but is sort of smearing them out. The hdsp driver doesn't send more than one MIDI byte per timer tick. IMHO it should be modified to send in a loop until the FIFO is full (however, I don't know if the HDSP has a FIFO at all). And it should start sending in output_trigger() instead of delaying it to the next timer tick. Clemens, Thanks for the response. One comment I forgot to make in the first post. This MIDI interface works fine under Windows, so whatever causes the problem is purely a Alsa MIDI issue. If we can figure it out, then we can fix it. I agree that it sounds like this sort of one note per timer tick. When the interface is supposed to send a chord, it sends what sounds like an arpegiated chord. It's all smeared out. Is there some example code I could look at to understand implementing a FIFO? However, if there is a FIFO Full indication, doesn't we need to know _how_ it's indicated? I would assume it's different for all cards? (Bus possibly similar for cards from the same manufacturer? Also, this is the HDSP 9652, which is a single PCI card. Is this problem showing up for the DigiFace/MultiFace type cards? Thanks, Mark --- This SF.NET email is sponsored by: FREE SSL Guide from Thawte are you planning your Web Server Security? Click here to get a FREE Thawte SSL guide and find the answers to all your SSL security issues. http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0026en ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] patch #1 for hdsp MIDI
Paul,
Thanks for looking onto this. We'll try to get it into the Alsa RPM
and tested soon.
Cheers,
Mark
On Mon, 2003-01-13 at 13:34, Paul Davis wrote:
Index: hdsp.c
===
RCS file: /cvsroot/alsa/alsa-kernel/pci/rme9652/hdsp.c,v
retrieving revision 1.16
diff -u -u -r1.16 hdsp.c
--- hdsp.c 7 Jan 2003 10:36:32 - 1.16
+++ hdsp.c 13 Jan 2003 13:32:32 -
@@ -817,10 +817,18 @@
static inline int snd_hdsp_midi_output_possible (hdsp_t *hdsp, int id)
{
+ int fifo_bytes_used;
+
if (id) {
- return (hdsp_read(hdsp, HDSP_midiStatusOut1) 0xff) 128;
+ fifo_bytes_used = hdsp_read(hdsp, HDSP_midiStatusOut1) 0xff;
} else {
- return (hdsp_read(hdsp, HDSP_midiStatusOut0) 0xff) 128;
+ fifo_bytes_used = hdsp_read(hdsp, HDSP_midiStatusOut0) 0xff;
+ }
+
+ if (fifo_bytes_used 128) {
+ return 128 - fifo_bytes_used;
+ } else {
+ return 0;
}
}
---
This SF.NET email is sponsored by: FREE SSL Guide from Thawte
are you planning your Web Server Security? Click here to get a FREE
Thawte SSL guide and find the answers to all your SSL security issues.
http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0026en
___
Alsa-devel mailing list
[EMAIL PROTECTED]
https://lists.sourceforge.net/lists/listinfo/alsa-devel
---
This SF.NET email is sponsored by: FREE SSL Guide from Thawte
are you planning your Web Server Security? Click here to get a FREE
Thawte SSL guide and find the answers to all your SSL security issues.
http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0026en
___
Alsa-devel mailing list
[EMAIL PROTECTED]
https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] hdsp multiface pci.
On Sun, 2003-01-19 at 16:07, Thomas Charbonnel wrote: So the numid=5 26,26,16384 line says: connect software output 1 (called playback in the above table) to line out left, as the syntax of the call is input_source,output_source,value. Thomas, I have possibly more than a passing interest in this subject as I use the HDSP 9652 card that Patrick mentioned and find this whole amixer language completely unfathomable. In your quote above, could you outline what each parameter stands for? numid=5 26,26,16384 numid=5 - Is this a physical connection? A mixer input or output? An abstract number just used to keep track of things? 26,26 - clearly Patrick and I were thinking that these two numbers related to specific hardware inputs and outputs, but apparently not. 16384 - a mixer volume? Thanks in advance for helping. Cheers, Mark --- This SF.NET email is sponsored by: FREE SSL Guide from Thawte are you planning your Web Server Security? Click here to get a FREE Thawte SSL guide and find the answers to all your SSL security issues. http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0026en ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] amixer question - numid=2
Hi, On my HDSP 9652 system executing the command amixer controls yields a list that appears to be 166 items long, starting with numid=1 and ending with numid=166. However, closer study shows numid=2 seems to be missing. Is this an issue with amixer or my card? What function might normally be associated with numid=2? Thanks, Mark --- This SF.NET email is sponsored by: FREE SSL Guide from Thawte are you planning your Web Server Security? Click here to get a FREE Thawte SSL guide and find the answers to all your SSL security issues. http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0026en ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] hdsp multiface pci.
On Sun, 2003-01-19 at 17:00, Roger Williams wrote: Patrick Shirkey [EMAIL PROTECTED] writes: Can we route multiple software outputs to the same hardware output? Yes. For instance, when recording 8 tracks, I'll generate a monitor mix (for JACK's outputs) to outputs 1 2 like this: amixer cset numid=5 26,26,16384 amixer cset numid=5 28,26,16384 amixer cset numid=5 30,26,16384 amixer cset numid=5 32,26,16384 amixer cset numid=5 27,27,16384 amixer cset numid=5 29,27,16384 amixer cset numid=5 31,27,16384 amixer cset numid=5 33,27,16384 -- Roger, OK, I just don't get it yet. This is from the Alsa page on the HDSP cards: * Since the Multiface only have 18 i/o channels, the channel mapping in the matrix mixer is different from the Digiface when operating at 48kHz or lower. [Ed. This is a routing table] input_source: 0-7 (analog), 16-23 (adat), 24-25 (spdif), 26-51 (playback) output_source: 0-7 (analog), 16-23 (adat) 24-25 (spdif), 26-27 (line out) * In your example above, is the use of 26 and 27 on the output part of the command telling the hardware to route whatever it is that you're mixing to the line out connectors on your box? Second, where do the inputs numbered 26-33 come from. I can't figure this part out. Are they outputs from the HDSP hardware mixer? Or do they mean from physical inputs? (somehow renumbered) This is the part I don't get right now. I presume that your use of 16384 is to reduce volume so that after summing 4 channels you do not create output clipping if all the inputs hit maximum volume at the same time? QUESTION - If you wanted to route input 0 directly to an output, could you use a command like: amixer cset numid=5 0,26,16384 amixer cset numid=5 1,27,16384 I think my HDSP 9652 will have different numbering since I have a different number of I/O's on the card, but I want to understand your example since it seems like a good one. Thanks, Mark --- This SF.NET email is sponsored by: FREE SSL Guide from Thawte are you planning your Web Server Security? Click here to get a FREE Thawte SSL guide and find the answers to all your SSL security issues. http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0026en ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] hdsp multiface pci.
Roger, This is very, very helpful. Thanks for sharing this info. I am trying to do similar things with the HDSP 9652, which is similar but a bit different, I'd like to address a couple more things. (Darn...Tennessee just scored!) :-((( 7-7 I just want to clearly understand the physical I/O numbering right now. Once I've got that I'll ask you a couple of questions about the mixer if I may. OK, it appears that the MultiFace has a slightly different numbering system for your analog channels, starting with 0-7 instead of what I think I have which starts with 1-8. Is this correct? If I look at the HDSP amixer info, I seem to start with an index of 1, not 0. Am I looking at the right stuff? (I've done some sorting to make this more readable to me. I have this initial section with 9 controls, and then 26 sections that are identical to the second area I show below, indexing from 1-26. I presume that your first one would start with index=0? [mark@Godzilla mark]$ amixer controls numid=1,iface=PCM,name='IEC958 Playback Default' numid=3,iface=MIXER,name='IEC958 Playback Con Mask' numid=4,iface=MIXER,name='IEC958 Playback Pro Mask' numid=5,iface=PCM,name='Mixer' numid=6,iface=PCM,name='IEC958 Input Connector' numid=7,iface=PCM,name='IEC958 Output also on ADAT1' numid=8,iface=PCM,name='Preferred Sync Source' numid=9,iface=PCM,name='Passthru' numid=10,iface=PCM,name='Line Out' numid=11,iface=MIXER,name='Chn',index=1 numid=12,iface=PCM,name='Input Peak',index=1 numid=13,iface=PCM,name='Output Peak',index=1 numid=14,iface=PCM,name='Playback Peak',index=1 numid=15,iface=PCM,name='Playback RMS',index=1 numid=16,iface=PCM,name='Input RMS',index=1 On Sun, 2003-01-19 at 22:24, Roger Williams wrote: Mark Knecht [EMAIL PROTECTED] writes: Second, where do the inputs numbered 26-33 come from. As far as the Multiface's analogue I/O goes, channels appear to be numbered like this: When you say 'appear to be numbered like this', how did you determine this? Was it documented somewhere? Or did you have to test yourself? Multiface inputs 1-8 = amixer source channels 0-7 Multiface outputs 1-8 = amixer destination channels 0-7 Multiface line (headphone) outputs = amixer destination channels 26-27 alsa_pcm:playback_1-8 = amixer source channels 26-33 OK, here's where you throw me. Let me list out what I thought I knew about the MultiFace, and then correct me where I'm wrong please. The MultiFace has 18 inputs - 8 analog, 8 ADAT, 2 s/pdif, and 20 outputs (including the Headphone outs if they are really separate from everything else. They may not be...) so I would have expected you to list: (Yea!!! Raiders score! 14-7) MultiFace analog inputs 1-8 = amixer source channels 0-7 MultiFace ADAT inputs 9-16 = amixer source channels W-X (8-16?) MultiFace s/pdif inputs 17-18 = amixer source channels Y-Z (17-18?) Basically, to use the 18 inputs, they all must have unique numbers. Correct? MultiFace analog outputs 1-8= amixer dest. channels 0-7 MultiFace ADAT outputs 9-16 = amixer dest. channels W-X (8-16?) MultiFace s/pdif outputs 17-18 = amixer dest. channels Y-Z (17-18?) MultiFace Line outputs 1-2 = amixer dest. channels 26-27 Maybe all of the features of the MultiFace not actually supported in the current driver, so you didn't list them, or maybe you just didn't use them, so you didn't write them down? In my case I am guessing that my physical I/O numbering will be as follows: HDSP 9652 ADAT-1 inputs 1-8 = amixer source channels 1-8 HDSP 9652 ADAT-2 inputs 9-16= amixer source channels 8-16 HDSP 9652 ADAT-3 inputs 17-24 = amixer source channels 17-24 HDSP 9652 s/pdif inputs 25-26 = amixer source channels 25-26 and similar numbering for the outputs. Do you think this is right? Any other thoughts or comments? Thanks, Mark --- This SF.NET email is sponsored by: FREE SSL Guide from Thawte are you planning your Web Server Security? Click here to get a FREE Thawte SSL guide and find the answers to all your SSL security issues. http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0026en ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] hdsp multiface pci.
Roger, I think the light is just starting to turn on, or at least I can only hope... ;-) See if you can help me improve the following description. I can think of my HDSP mixer as a device with 52 inputs and 26 outputs. The inputs look like: 1) 26 mixer inputs come from the HDSP's physical inputs - numbered 0-25 in the mixer - called Alsa_pcm:capture_1 - 26 in Jack 2) 26 mixer inputs come from the Alsa:playback group - numbered 26-51 in the mixer - called Alsa_pcm:playback_1 - 26 in Jack The outputs look like: 1) 26 mixer outputs come from the mixer - numbered 0-25 out of the mixer By default, the HDSP driver then takes the Alsa_pcm:playback group and hooks them to (more or less) matching output destinations: Alsa_pcm:playback_1 - HDSP Output 0 Alsa_pcm:playback_2 - HDSP Output 1 ... Alsa_pcm:playback_26 - HDSP Output 25 so that if I connect capture_1 to playback_1 then whatever comes in on the ADAT-1, channel 0 input will get sent back out on ADAT-1, channel 1 output. To test this idea, and using your headphone mix example, I should be able to create a hardware monitoring setup by routing physical inputs directory to physical outputs using something like this: amixer cset numid=5 0,0,32768 amixer cset numid=5 1,1,32768 ... amixer cset numid=5 25,25,32768 which would route each physical input to each physical output, 1 for 1, at a volume of unity gain. (As per Marcus's notes again.) What is not at all clear yet is whether a single input can go to multiple outs with different gains. For instance, if I execute: amixer cset numid=5 0,0,1 amixer cset numid=5 0,1,3 I am attempting to take physical input 0 and sending it to both the left and right outputs, but at different volumes. Is this legal? I'm not sure. In my case, I use Alsa_pcm:playback_1/2 for my main speakers, and playback_3/4 for my headphones. The problem I've been having is that I didn't know how to set the volume on my speakers with no mixer. Using this information, I could execute: amixer cset numid=5 26,0,3000 amixer cset numid=5 27,1,3000 and now my playback_1/2 should come out of my speakers, but at considerably reduced volumes. I'm going to stop and get feedback, as this is almost making sense. In the meantime, I might as well try the commands above and see if my speaker volume is under control. Thanks very much for your help! Cheers, Mark On Mon, 2003-01-20 at 02:14, Roger Williams wrote: Mark Knecht [EMAIL PROTECTED] writes: This is very, very helpful. What will be very, very helpful will be Thomas's TotalMix clone! :) ... it appears that the MultiFace has a slightly different numbering system for your analog channels, starting with 0-7 instead of what I think I have which starts with 1-8. My amixer controls dump looks the same as yours. I don't know what index=1 means, but it doesn't appear to mean that our cset indexing begins at 1. I've got to assume that Paul or Thomas already know everything there is to know about the HDSP mixer, and I hate to explore known territory, but when you don't have a map... When you say 'appear to be numbered like this', how did you determine this? Was it documented somewhere? Or did you have to test yourself? Well, it matches Marcus Andersson's notes on the ALSA HDSP page: input_source: 0-7 (analog), 16-23 (adat), 24-25 (spdif), 26-51 (playback) output_source: 0-7 (analog), 16-23 (adat) 24-25 (spdif), 26-27 (line out) I found output_source confusing, but it makes sense if I rename it destination. The MultiFace has 18 inputs ... so I would have expected you to list: Some of the numbering is discontinuous (because the Multiface doesn't have the Digiface's ADAT2 range of channels), so the full Multiface I/O list is: Inputs to HDSP Mixer Multiface analogue inputs 1-8 = amixer source channels 0-7 Multiface ADAT inputs 1-8 = amixer source channels 16-23 Multiface SPDIF input = amixer source channels 24-25 alsa_pcm:playback_1-26 = amixer source channels 26-51 Outputs from HDSP Mixer Multiface analogue outputs 1-8 = amixer destination channels 0-7 Multiface ADAT outputs 1-8 = amixer destination channels 16-23 Multiface SPDIF output = amixer destination channels 24-25 Multiface line (headphone) output = amixer destination channels 26-27 Mapping between Multiface inputs and alsa_pcm:capture channels Multiface analogue inputs 1-8 = capture_1-8 Multiface ADAT inputs 1-8 = capture_9-16 Multiface SPDIF input = capture_17-18 In my case I am guessing that my physical I/O numbering will be HDSP 9652 ADAT-1 inputs 1-8 = amixer source channels 1-8 HDSP 9652 ADAT
Re: [Alsa-devel] hdsp multiface pci.
Roger, First, thanks for taking time to follow this through with me. This has been helpful and I do at least feel like I'm starting to understand what the software is tying to do. When I tried some amixer commands I was disappointed to find that they do nothing at all on my machine. No command I tried seems to control volume to my headphones or speakers at all. I guess the HDSP 9652 is not supported at this time. Makes me feel like I bought the wrong product. Anyway, I will continue to study and understand this stuff and hope that one day some of these features actually work. Thanks for your help! Cheers, Mark On Mon, 2003-01-20 at 06:25, Roger Williams wrote: Mark Knecht [EMAIL PROTECTED] writes: I can think of my HDSP mixer as a device with 52 inputs and 26 outputs. In the case of the 9652, you don't have headphone outputs, so you don't have the Digiface's 27th and 28th outputs. 1) 26 mixer inputs come from the HDSP's physical inputs - numbered 0-25 in the mixer - called Alsa_pcm:capture_1 - 26 in Jack Yup. 2) 26 mixer inputs come from the Alsa:playback group - numbered 26-51 in the mixer - called Alsa_pcm:playback_1 - 26 in Jack Yup. 1) 26 mixer outputs come from the mixer - numbered 0-25 out of the mixer Yup. By default, the HDSP driver then takes the Alsa_pcm:playback group and hooks them to (more or less) matching output destinations: Perhaps, but I'm not sure about that. I always explicitly set up the HDSP mixer routes. I just removed all of my ALSA modules, removed the HDSP Cardbus card, power-cycled the Multiface, and reinstalled everything; and I had to issue an amixer cset numid=5 26,0,32768 command to connect alsa_pcm:playback_1 to Multiface output 1. so that if I connect capture_1 to playback_1 then whatever comes in on the ADAT-1, channel 0 input will get sent back out on ADAT-1, channel 1 output. You don't connect capture_1 to playback_1, because playback_1 is a signal coming from ALSA, going into the HDSP mixer. But you _can_ connect the signal arriving on the ADAT1:1 input (which _also_ drives ALSA's capture_1 input) to the ADAT1:1 output with an amixer cset numid=5 0,0,32768 command. (But playback_1 isn't routed to the ADAT1:1 output except by coincidence or a default setting -- playback_1 doesn't have anything to do with your ADAT1:1 input - ADAT1:1 output connection.) amixer cset numid=5 0,0,32768 amixer cset numid=5 1,1,32768 ... amixer cset numid=5 25,25,32768 which would route each physical input to each physical output, 1 for 1, at a volume of unity gain. (As per Marcus's notes again.) Yup. What is not at all clear yet is whether a single input can go to multiple outs with different gains. For instance, if I execute: amixer cset numid=5 0,0,1 amixer cset numid=5 0,1,3 Sure, that works just fine. I'll run a combined setup like this: Monitor submix == amixer cset numid=5 0,26,2 amixer cset numid=5 1,27,2 amixer cset numid=5 2,26,15000 amixer cset numid=5 3,27,15000 amixer cset numid=5 4,26,1 amixer cset numid=5 5,27,1 DAT safety recording == amixer cset numid=5 0,24,16384 amixer cset numid=5 1,25,16384 amixer cset numid=5 2,24,12288 amixer cset numid=5 3,25,12288 amixer cset numid=5 4,24,8192 amixer cset numid=5 5,25,8192 The changes I make in the first group (headphones submix) don't have any effect on the signal being recorded by the DAT, which is set up in the second group. I am attempting to take physical input 0 and sending it to both the left and right outputs, but at different volumes. Is this legal? Sure. Consider RME's description of TotalMix, which is nothing more than a software interface (OK, I'm a hardware guy) to the HDSP mixer: - setting up delay-free submixes (headphone mixes) - unlimited routing of inputs and outputs (free utilisation, patchbay function) - distributing signals to several outputs at a time - simultaneous playback of different programs over only one stereo channel - mixing of the input signal to the playback signal (complete ASIO Direct Monitoring) - integration of external devices (effects etc). in real-time - mixdown of three ADAT inputs to one (realizing two additional inputs) RME calls the HDSP mixer in the Multiface a 720 channel mixer, and the one in the Digiface 1456 channels. (The 9652 would be 1352 channels, because it doesn't have the headphone outputs.) For the Multiface, that's [18 hardware input channels + 18 playback channels] x 20 hardware output channels. (There are only 18 playback channels because playback_8-15 aren't used in the Multiface.) In my case, I use Alsa_pcm:playback_1/2 for my main speakers, and playback_3/4 for my
RE: [Alsa-devel] hdsp multiface pci.
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Roger Williams Sent: Wednesday, January 22, 2003 10:31 AM To: Paul Davis Cc: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: Re: [Alsa-devel] hdsp multiface pci. Paul Davis [EMAIL PROTECTED] writes: This would explain why snd_hdsp works (i.e. you can playback and record through the default I/O), although amixer doesn't do anything: the driver doesn't have to set up any default connections, because those are part of the FPGA reset state... actually, the driver does have to set up default connections, which are zero gain for every possible routing... Yes, I understand that that's what the driver is _supposed_ to do. That's how it works on my HDSP PCI and Cardbus systems. But isn't it true that the current driver doesn't actually set up any of those zero-gain connections for the HDSP 9652? This is the only point that I was trying to make to Mark -- the default unity-gain connections (i.e. playback - H/W output) that he has been using are FPGA configuration defaults, not explicitly set up by the driver. Yes, and I think I understood that this was probably what was happening as our conversation progressed. When I made that statement it was a bit earlier on, if I remember correctly, and I was explaining my 'vision' of what I thought was going on. That should not be confused with the trusth! ;-) It makes perfect sense that the RME card itself has some default connections. Thanks, Mark --- This SF.net email is sponsored by: Scholarships for Techies! Can't afford IT training? All 2003 ictp students receive scholarships. Get hands-on training in Microsoft, Cisco, Sun, Linux/UNIX, and more. www.ictp.com/training/sourceforge.asp ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] HDSP 9652 MIDI Timing - Much improved, but no Port 1...
Paul, Takashi-san and Clemens, Hi. A couple of days ago Fernando got a new RPM built for Alsa which includes the recent HDSP 9652 MIDI timing fix that you worked together on and supplied about 10 days ago. I wanted to report back that the timing is now much improved. I haven't used it a lot yet, but I am playing moderately complicated songs now and the timing seems fine. Thanks. There is one problem that has come up new in this release. The HDSP 9652 has two MIDI ports. With this release I can only get MIDI out on Port 2. I cannot get MIDI out at all on Port 1. Both ports used to work on the previous version, albeit with bad timing, so this fix this has changed this aspect of the driver. I also tried using kaconnect to look at the connections between Rosegarden and the alsa_sequencer. It shows that Rosegarden is hooked to 64:32 External MIDI 0 only. A second group, 64:0 External MIDI 0 shows up, but kaconnect will not connect to it. I do not know what that means, but it seems to be part of it. It strikes me that I have not done any recording with this device, so I should do some of that before we make any more driver changes. However, I wanted to report back a big thanks for a step in the right direction, even if we are not quite all the way there yet and allow you to look at what might be causing this problem. If there is any specific testing you'd like me to do, please let me know and I'll try to get to it as soon as possible. I'd like to get this card fully supported (yes, the mixer too Paul!) ;-) as I get 2-3 emails a week from people asking me if they should buy the card and I'd like to tell them yes. Thanks much, Mark --- This SF.NET email is sponsored by: SourceForge Enterprise Edition + IBM + LinuxWorld = Something 2 See! http://www.vasoftware.com ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] amixer question - numid=2
Thanks! On Fri, 2003-01-24 at 11:15, Takashi Iwai wrote: At 19 Jan 2003 10:59:39 +, Mark Knecht wrote: Hi, On my HDSP 9652 system executing the command amixer controls yields a list that appears to be 166 items long, starting with numid=1 and ending with numid=166. However, closer study shows numid=2 seems to be missing. Is this an issue with amixer or my card? What function might normally be associated with numid=2? don't worry, no bug: it's not a mixer control but a control for a PCM stream. that's why amixer doesn't show this element. ciao, Takashi --- This SF.NET email is sponsored by: SourceForge Enterprise Edition + IBM + LinuxWorld = Something 2 See! http://www.vasoftware.com ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] buffer_size and period_size
Paul, So with -p 64 -n 2 settings, what number of bytes of audio data is transferred across the PCI bus between each interrupt? I guess I had mistakenly thought -p was setting the number of bytes. I no longer think that is true. Also, does the number of bytes transferred change based on how many channels are enabled? Or does my RME always transfer 26 channels of data even if I am not using some channels? I am assuming that a card like the RME is a bus master, moves so many bytes, and then interrupts to tell the system that the bytes are there. Is this basically the case? Thanks, Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Paul Davis Sent: Monday, January 27, 2003 9:32 AM To: Jozef Kosoru Cc: [EMAIL PROTECTED] Subject: Re: [Alsa-devel] buffer_size and period_size I would like to fully understand the exact meaning of buffer_size and period_size and how can I compute the final latency in the full duplex processing from these variables. period_size = frames between interrupts from the hardware buffer size = total frames for the hardware buffer max output latency = buffer_size min output latency = buffer_size - period_size max input latency = period_size + interrupt overhead min input latency = 1 frame + interrupt overhead the latency numbers assume: a) the buffer is generally full b) you process data 1 period at a time c) your s/w keeps up with the h/w d) its an average, computed across a period's worth of data --p --- This SF.NET email is sponsored by: SourceForge Enterprise Edition + IBM + LinuxWorld = Something 2 See! http://www.vasoftware.com ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel --- This SF.NET email is sponsored by: SourceForge Enterprise Edition + IBM + LinuxWorld = Something 2 See! http://www.vasoftware.com ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Automounting my 1394 drives causes Alsa to not load...
Hi, Bit of a problem. This is Redhat 7.3, PlanetCCRMA flow, and the machine has 256MB. Alsa has been working reasonably well for me, but I have two problems that I would really like to fix: 1) Old problem - if my MidiSport 2x2 is plugged in when I cold boot, then Alsa gets loaded when the MidiSport is found. When I get to the part of the boot process where Alsa is supposed to get started, I get a 'Failed' message, telling me Alsa is already running. Even this is OK, but then later when Linux attempt to load the HDSP 9652 drivers, they fail one out of two times saying they cannot allocate memory. 2) When I try to auto-mount my 1394 hard drives by creating an auto-mount entry in /etc/fstab, they may not be turned on, which is legal. However, in this situation Alsa always fails to load. I have to make the drive 'noauto' to get Alsa to start correctly. Both of these problems seem to be solved by warm booting the system. However, that takes time and I certainly shouldn't have to do that. What can I do to solve these problems so that Alsa will come up correctly on my first cold boot. Cheers, Mark --- This SF.NET email is sponsored by: SourceForge Enterprise Edition + IBM + LinuxWorld = Something 2 See! http://www.vasoftware.com ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Automounting my 1394 drives causes Alsa to notload...
On Sat, 2003-02-01 at 12:31, Fernando Pablo Lopez-Lezcano wrote: 1) Old problem - if my MidiSport 2x2 is plugged in when I cold boot, then Alsa gets loaded when the MidiSport is found. When I get to the part of the boot process where Alsa is supposed to get started, I get a 'Failed' message, telling me Alsa is already running. Even this is OK, but then later when Linux attempt to load the HDSP 9652 drivers, they fail one out of two times saying they cannot allocate memory. I think the solution to this one is to blacklist the alsa driver so that hotplug does not load it while the system is starting up. To do that just add a line with snd-usb-audio to the end of /etc/hotplug/blacklist You probably already have audio and usb-midi there (the oss kernel modules that deal with usb audio and midi). Fernando, Thanks! Early indications are that this helps. I'll keep an eye on it and see how it goes. Cheers, Mark --- This SF.NET email is sponsored by: SourceForge Enterprise Edition + IBM + LinuxWorld = Something 2 See! http://www.vasoftware.com ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] Where to report?
Hi, Is Alsa-Dev the right place to report problems with Linux MIDI? (Such as stuck note problems with soft synths.) Thanks, Mark --- This SF.NET email is sponsored by: SourceForge Enterprise Edition + IBM + LinuxWorld = Something 2 See! http://www.vasoftware.com ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: Further info: [Alsa-devel] Bug: ALSA Sequencer or MTPAV - easy to reproduce
Will do. I'll send it along this evening. Thanks, Mark -Original Message- From: Jaroslav Kysela [mailto:[EMAIL PROTECTED]] Sent: Wednesday, February 12, 2003 1:26 PM To: Mark Knecht Cc: [EMAIL PROTECTED] Subject: RE: Further info: [Alsa-devel] Bug: ALSA Sequencer or MTPAV - easy to reproduce On Wed, 12 Feb 2003, Mark Knecht wrote: Jaroslav, Hi. Actually, I had been looking around for where to report this sort of problem. I'm using an HDSP 9652 for MIDI input and getting stuck notes on all soft synths I'm using. (amSynth, ZynAddSubFx and iiwusynth) I'm at a bit of a loss as to how to debug this, but I do see the problem. I have found that it is independent of MIDI applications, as I see the problem if I just use kaconnect to hook MIDI input to the soft synth and qjackconnect to hook analog output to my speakers. Could you try the command 'dd if=/dev/snd/midiC0D0 of=abcd bs=1' and play some notes on connected keyboard? In the file abcd will be the raw context of midi input, so we can determine, if it's driver or sequencer problem. Jaroslav - Jaroslav Kysela [EMAIL PROTECTED] Linux Kernel Sound Maintainer ALSA Project, SuSE Labs --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: Further info: [Alsa-devel] Bug: ALSA Sequencer or MTPAV - easyto reproduce
On Wed, 2003-02-12 at 13:25, Jaroslav Kysela wrote: On Wed, 12 Feb 2003, Mark Knecht wrote: Jaroslav, Hi. Actually, I had been looking around for where to report this sort of problem. I'm using an HDSP 9652 for MIDI input and getting stuck notes on all soft synths I'm using. (amSynth, ZynAddSubFx and iiwusynth) I'm at a bit of a loss as to how to debug this, but I do see the problem. I have found that it is independent of MIDI applications, as I see the problem if I just use kaconnect to hook MIDI input to the soft synth and qjackconnect to hook analog output to my speakers. Could you try the command 'dd if=/dev/snd/midiC0D0 of=abcd bs=1' and play some notes on connected keyboard? In the file abcd will be the raw context of midi input, so we can determine, if it's driver or sequencer problem. Jaroslav Hi, OK, as requested, here's a few chords and some notes. However, I cannot hear the soft synth when doing this, and I normally only get a stuck note once every 5-10 minutes, so there's no guarantee that there's anything interesting in here. Maybe I could record something, using Rosegarden, until I get a stuck note, and give you a MIDI file? Don't know if that would be of much help as this problem is not terribly repeatable yet. Is there any way I can pipe this input to my soft synth so I can hear what I'm doing? Cheers, Mark midi_notes Description: Binary data
RE: [Alsa-devel] playing underruns
Also check out the Planet for more info on this. Fernando has some suggestions for Redhat there. Cheers, Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Paul Davis Sent: Thursday, February 13, 2003 11:11 AM To: Chris Raphael Cc: [EMAIL PROTECTED] Subject: Re: [Alsa-devel] playing underruns /sbin/hdparm /dev/hda2 I get: /dev/hda2: multcount = 16 (on) I/O support = 0 (default 16-bit) unmaskirq = 0 (off) using_dma = 1 (on) keepsettings = 0 (off) nowerr= 0 (off) readonly = 0 (off) readahead = 8 (on) geometry = 4864/255/73, sectors = 36869175, start = 4225095 with similar results for the other hda's. I don't know if this is the question you were asking, though, since this doesn't seem to have much info. yep, this doesn't look too good, though its not a complete disaster. please read this: http://linux.oreillynet.com/pub/a/linux/2000/06/29/hdparm.html keep in mind that some distributions (RH included, i think) have fixed this issue somewhat, though they may not have gone far enough for low latency audio. I don't have the low latency patch. you will probably need it. the standard kernel in RH7 is not up to the task. --p --- This SF.NET email is sponsored by: FREE SSL Guide from Thawte are you planning your Web Server Security? Click here to get a FREE Thawte SSL guide and find the answers to all your SSL security issues. http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0026en ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel --- This SF.NET email is sponsored by: FREE SSL Guide from Thawte are you planning your Web Server Security? Click here to get a FREE Thawte SSL guide and find the answers to all your SSL security issues. http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0026en ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] IEEE 1394
Pavel, You're in the wrong forum. Go to www.linux1394.org and pick up the information you need to get started there. If you want to develop 1394 applications there are some mailing lists there with other like minded people. Good luck, Mark -Original Message-From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of PavelSent: Thursday, February 13, 2003 4:34 AMTo: [EMAIL PROTECTED]; [EMAIL PROTECTED]Subject: [Alsa-devel] IEEE 1394 Hi, I would like to ask about situation in Linux about one problem. Does Linux kernel or Alsa drivers supports IEEE 1394 standart? If yes, could you recomend me some references and advices to be able to use it andprogram itto createapplications with IEEE 1394? If no, could you recomend me some advices or some documentation to be able to create IEEE 1394 driver? Thanks Ing. Pavel Nikitenko
[Alsa-devel] Please explain Alsa Interface MIDI numbering PLEASE!
Hi, I'm having a great deal of confusion about how Alsa is handling my MIDI hardware. This is spilling over into unintended consequences in Rosegarden that I think none of us understand. Couple someone with some background in this please explain? Thanks. I have two 2-port MIDI devices on this system. One is an RME HDSP 9652 with two MIDI ports, and the other is a hot pluggable MidiSport 2x2. I attach two screen shots of kaconnect, one with and one without the 2x2 plugged in. My questions: 1) In the screen shot without_2x2.png I see two read ports and two write ports. Please explain why they are called 64:0 External MIDI 0 64:32 External MIDI 0 Why is my HDSP given the apparent name '64'? Why the :0 and :32? I would have thought :0 and :16 would make more sense from a channel numbering point of view, or :0 and :1 from an interface point of view. What's going on? 2) In the screen shot with_2x2.png I've plugged in the MidiSport 2x2. New devices show up in kaconnect. However, instead of showing 2 read ports and 2 write ports, I am getting 4 read ports and no write ports. Please explain why the MidiSport is given the names 72:0 External MIDI 1 72:1 External MIDI 1 72:2 External MIDI 1 72:3 External MIDI 1 Shouldn't this be just :0 and :1 for both the read and write ports? I get the feeling that BOTH of the drivers for these devices are hosed. What's up with these things? I do not understand why Alsa gives these devices numbers in the first place, nor how the numbers are assigned. How can I change the names that are displayed so that 64:0 External MIDI 0 shows the name HDSP 9652 Port 1 64:32 External MIDI 0 shows the name HDSP 9652 Port 2 72:0 External MIDI 1 shows the name MidiSport 2x2 Port A 72:1 External MIDI 1 shows the name MidiSport 2x2 Port B Thanks very, very much in advance, Mark attachment: with_2x2.pngattachment: without_2x2.png
Re: [Alsa-devel] Please explain Alsa Interface MIDI numberingPLEASE!
Pedro, I run on the PlanetCCRMA flow. My current Alsa appears to be from 1/21/03, or about a month ago. Was rc7 after that? Thanks, Mark On Sun, 2003-02-16 at 11:43, Pedro Lopez-Cabanillas wrote: On Sunday 16 February 2003 19:53, Mark Knecht wrote: 2) In the screen shot with_2x2.png I've plugged in the MidiSport 2x2. New devices show up in kaconnect. However, instead of showing 2 read ports and 2 write ports, I am getting 4 read ports and no write ports. Please explain why the MidiSport is given the names 72:0 External MIDI 1 72:1 External MIDI 1 72:2 External MIDI 1 72:3 External MIDI 1 Shouldn't this be just :0 and :1 for both the read and write ports? Yes. This was a bug in snd-usb-audio for 0.9.0rc7, fixed now in ALSA CVS, see: http://www.mail-archive.com/alsa-devel@lists.sourceforge.net/msg06225.html Regards, Pedro -- ALSA Library Bindings for Pascal http://alsapas.alturl.com --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Please explain Alsa Interface MIDI numberingPLEASE!
Sorry... I see the date on the email about the patch was a few days later in February, so I definitely do not have the patch. Thanks On Sun, 2003-02-16 at 11:46, Mark Knecht wrote: Pedro, I run on the PlanetCCRMA flow. My current Alsa appears to be from 1/21/03, or about a month ago. Was rc7 after that? Thanks, Mark On Sun, 2003-02-16 at 11:43, Pedro Lopez-Cabanillas wrote: On Sunday 16 February 2003 19:53, Mark Knecht wrote: 2) In the screen shot with_2x2.png I've plugged in the MidiSport 2x2. New devices show up in kaconnect. However, instead of showing 2 read ports and 2 write ports, I am getting 4 read ports and no write ports. Please explain why the MidiSport is given the names 72:0 External MIDI 1 72:1 External MIDI 1 72:2 External MIDI 1 72:3 External MIDI 1 Shouldn't this be just :0 and :1 for both the read and write ports? Yes. This was a bug in snd-usb-audio for 0.9.0rc7, fixed now in ALSA CVS, see: http://www.mail-archive.com/alsa-devel@lists.sourceforge.net/msg06225.html Regards, Pedro -- ALSA Library Bindings for Pascal http://alsapas.alturl.com --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] Please explain Alsa Interface MIDI numbering PLEASE!
Pedro, Is there any online information about how to use Midiman's firmware? Thanks, Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED]]On Behalf Of Pedro Lopez-Cabanillas Sent: Sunday, February 16, 2003 12:34 PM To: Mark Knecht Cc: Alsa-Devel; Rosegarden-Devel; Fernando Pablo Lopez-Lezcano Subject: Re: [Alsa-devel] Please explain Alsa Interface MIDI numbering PLEASE! On Sunday 16 February 2003 20:46, Mark Knecht wrote: Pedro, I run on the PlanetCCRMA flow. My current Alsa appears to be from 1/21/03, or about a month ago. Was rc7 after that? Yes, 0.9.0rc7 is dated 2003-01-28 PlanetCCRMA's ALSA drivers came from a CVS snapshot taken at 2003-01-21 The bug was introduced at 2003-01-10 So, you should use current CVS driver, or wait for a PlanetCCRMA update, or use Midiman's firmware. For other USB MIDI fully compliant devices, like Evolution's keyboards, a fixed driver is needed. Regards, Pedro On Sun, 2003-02-16 at 11:43, Pedro Lopez-Cabanillas wrote: On Sunday 16 February 2003 19:53, Mark Knecht wrote: 2) In the screen shot with_2x2.png I've plugged in the MidiSport 2x2. New devices show up in kaconnect. However, instead of showing 2 read ports and 2 write ports, I am getting 4 read ports and no write ports. Please explain why the MidiSport is given the names 72:0 External MIDI 1 72:1 External MIDI 1 72:2 External MIDI 1 72:3 External MIDI 1 Shouldn't this be just :0 and :1 for both the read and write ports? Yes. This was a bug in snd-usb-audio for 0.9.0rc7, fixed now in ALSA CVS, see: http://www.mail-archive.com/alsa-devel@lists.sourceforge.net/msg06225.htm l Regards, Pedro -- ALSA Library Bindings for Pascal http://alsapas.alturl.com --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [linux-audio-user] Re: [Alsa-devel] Please explain AlsaInterface MIDI numbering PLEASE!
Clemens, Thanks. This is helpful, although questions remain. FYI - my machine doesn't have pmidi, so I cannot run that now. [mark@Godzilla mark]$ aconnect -io client 0: 'System' [type=kernel] 0 'Timer ' 1 'Announce' client 64: 'External MIDI 0' [type=kernel] 0 'MIDI 0-0' 32 'MIDI 0-1' client 72: 'External MIDI 1' [type=kernel] 0 'Midisport 2x2 Port 0' 1 'Midisport 2x2 Port 1' 2 'Midisport 2x2 Port 2' 3 'Midisport 2x2 Port 3' [mark@Godzilla mark]$ The above information is certainly a bit more readable, but it seems to still be, at the least, inconsistent. 1) For client 64, which is an HDSP 9652, there are two rawmidi ports. However, the info above says they are labeled '0' and '32'. Should they not be 0 1? If this is an error, then what needs to be fixed? The HDSP 9652 driver? 2) Why does the HDSP 9652 not tell me its name like the MidiSport does? The MidiSport info above is with the Win2K firmware installed as per your extraction program. It actually didn't change from the way Fernando had me install it, so I suppose that his installation had the real Midiman firmware and not the open source firmware. Apparently I'll continue to get the wrong number of ports on that device until I can get Alsa itself upgraded. Thanks, Mark On Mon, 2003-02-17 at 01:21, Clemens Ladisch wrote: Mark Knecht wrote: 1) In the screen shot without_2x2.png I see two read ports and two write ports. Please explain why they are called 64:0 External MIDI 0 64:32 External MIDI 0 Why is my HDSP given the apparent name '64'? This isn't the name, it's the sequencer client number. In theory, it should not be necessary to identify devices by this. 0-63 are reserved for the ALSA core. 64-127 are used by sound cards, with each card getting 8 (64-71, 72-79, etc.). 128-255 are for use by applications. Why the :0 and :32? I would have thought :0 and :16 would make more sense from a channel numbering point of view, or :0 and :1 from an interface point of view. What's going on? These ports are not native sequencer ports implemented directly by the driver but are emulated on top the rawmidi ports. There can be 256 ports per sequencer client, and 8 rawmidi devices per card, so each rawmidi device (which can have an unspecified number of subdevices=ports) is mapped to a group of 32 (256/8) sequencer ports. If the two rawmidi ports would have been subdevices of one device, they would have been mapped to port numbers :0 and :1. How can I change the names that are displayed so that 64:0 External MIDI 0 shows the name HDSP 9652 Port 1 64:32 External MIDI 0 shows the name HDSP 9652 Port 2 72:0 External MIDI 1 shows the name MidiSport 2x2 Port A 72:1 External MIDI 1 shows the name MidiSport 2x2 Port B External MIDI x is the client name, which is the same for all ports of the same client. It seems that kaconnect doesn't show the port name, which would be what you want. Please complain to the author of kaconnect. :-) To show the port names, run aconnect -io or pmidi -l. HTH Clemens --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [linux-audio-user] Re: [Alsa-devel] Please explain Alsa Interface MIDI numbering PLEASE!
0 'Timer ' 1 'Announce' client 64: 'External MIDI 0' [type=kernel] 0 'MIDI 0-0' 32 'MIDI 0-1' client 72: 'External MIDI 1' [type=kernel] 0 'Midisport 2x2 Port 0' 1 'Midisport 2x2 Port 1' 2 'Midisport 2x2 Port 2' 3 'Midisport 2x2 Port 3' [mark@Godzilla mark]$ The above information is certainly a bit more readable, but it seems to still be, at the least, inconsistent. 1) For client 64, which is an HDSP 9652, there are two rawmidi ports. However, the info above says they are labeled '0' and '32'. Should they not be 0 1? If this is an error, then what needs to be fixed? The HDSP 9652 driver? no, the port numbers are 0 and 32, but in the name, its 0 and 1. Well, OK, I guess I don't understand the meaning of 'ports' then. The HDSP only has two sets of in and out connectors. Are these not ports? Or does the Alsa spec think that each 'port' is somehow combination of a MIDI connector and a channel or something? How is it that a single input uses up 32 port number? (HDSP 9652 MIDI 1 seems to go from port 0 to port 31, and I guess #2 goes from 32-63.) 2) Why does the HDSP 9652 not tell me its name like the MidiSport does? its using a copy of some generic ALSA code that just calls the ports MIDI C P where C=card number and P=physical port number. i'll change this when i add the fixes for the mixer and the h/w names. This would be very helpful. Thanks! --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] kaconnect question and enhancement request
Hi, I'm sorry, but I'm not at all sure who developed kaconnect. I like this little app quite a bit, however, it won't allow one thing I'd certainly like - to be able to hook a MIDI input to its corresponding MIDI output. Why doesn't this work? I can hook MidiSport 2x2 In A to Out B, but not to Out A, which I would really like to be able to do sometimes. Is there a technical reason that this cannot be done? As an enhancement request, I sure would like kaconnect to have the ability to filter certain MIDI events, like controllers, or key pressure, etc. I understand that I might be difficult to do this on a path by path basis, although it would be useful at times. Saving a set of connections would be cool also. Anyway, thanks for this little app that I use every day. Cheers, Mark --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] kaconnect question and enhancement request
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Takashi Iwai Sent: Tuesday, February 25, 2003 2:10 AM To: Mark Knecht Cc: Alsa-Devel Subject: Re: [Alsa-devel] kaconnect question and enhancement request no idea.. doesn't aconnect in alsa-utils work? Takashi-san, Thanks. I tried aconnect, and read through the --help stuff, but couldn't figure out the command to hook together two ports. Could you give an example? Everything I tried resulted in error messages. I also wondered about hooking a single input to multiple outputs, so if you could show an example of that, I would appreciate it. Thanks, Mark --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] kaconnect question and enhancement request
On Tue, 2003-02-25 at 07:52, Takashi Iwai wrote: for connecting between the same input and output, just run like % aconnect 64:0 64:0 Takashi-san, Thanks. This works fine. If 64:0 is connected to 64:0 in aconnect, then kaconnect shows it and will allow it to be disconnected. However kaconnect will not make the connection itself, so I suppose this is an oversight in kaconnect. Thanks for your help. Cheers, Mark --- This SF.net email is sponsored by: Scholarships for Techies! Can't afford IT training? All 2003 ictp students receive scholarships. Get hands-on training in Microsoft, Cisco, Sun, Linux/UNIX, and more. www.ictp.com/training/sourceforge.asp ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] HDSP 9652 MIDI IN - stuck notes
Hi, I've had a miserable stuck note problem with Alsa MIDI for a while, so I finally sat down this evening to try and determine where the problem was coming from, and it appears to be the HDSP 9652 MIDI input. My system has two 2-port MIDI interfaces, the HDSP 9652 (Alsa MIDI 64:0 64:32) and a MidiSport 2x2 (Alsa MIDI 72:0 72:1) I am using three external hardware synths on different ports (Alsa 64:0, 72:0 and 72:1) and additionally I ran a copy each of the Linux synths ZynAddSubFx (Alsa 130:0) and amSynth (Alsa 128:0) at the same time. (I.e. - 5 synths in parallel, all being driven by a single MIDI input.) To do the tests, I use kaconnect and aconnect to build routing in the MIDI stack to connect a single input to all outputs, which looks like the attachment when I attach all outputs to the MidiSport Port A input. In this configuration I can play continually, basically overdriving the whole system with lots of MIDI events, including controllers and sustain pedal, and I have no problems. I can do this for at least 15 minutes with no problem on any of the synths. If I then change my keyboard to drive the HDSP 9652 input 1 (Alsa 64:0) and change the connections internally to drive all of outputs, I get stuck notes pretty much immediately. It seems a bit worse with the sustain pedal, but does not seem to be effected at all by controllers. It is heavily effected by the MIDI note density. If I hit only one or two notes, I'm not likely to get it, but using the sustain pedal I can create the problem in under a minute. To be sure it's the input and not the outputs (as much as I can be) I have external synths attached to the Alsa outputs on 64:0, 72:0 and 72:1. When I get a stuck note, I seem to get it on both internal soft synths and all three external hardware synths at the same time. For this reason I deduce that it is the HDSP input that is not clearing out whatever event queue that holds this stuff and somehow the note never shuts off. I'll be happy to try any other tests anyone wants me to look into. I'd be curious to know if this is a problem on any of the other RME products that have MIDI or whether it's specifically a HDSP 9652 issue. Also, have any other HDSP 9652 owners seen this? Please let me know what I might be able to do to help further get this solved. Knowing now that the MidiSport doesn't have a problem, I can just use it and for me that's fine. However, for others that might be depending on the HDSP 9652 as a primary interface this would not be acceptable. Thanks, Mark attachment: snapshot1.png
Re: [Alsa-devel] [FOR] justin carmack (also mark knecht)
As always Paul, thanks for the efforts. On Thu, 2003-02-27 at 03:48, Paul Davis wrote: justin - sorry, i lost your email address. i got much clearer info on the mixer controls for the hdsp-9652 from RME, and have fixed the code. i have to get one more piece of information from them and then i will release a patch. the patch will also contain a consolidation of other work to recognize the various firmware revs and different h/w, plus a few other fixes from the last month. --p --- This SF.NET email is sponsored by: SourceForge Enterprise Edition + IBM + LinuxWorld = Something 2 See! http://www.vasoftware.com ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] HDSP 9652 MIDI IN - stuck notes
On Fri, 2003-02-28 at 04:14, Takashi Iwai wrote: At 26 Feb 2003 20:47:41 -0800, Mark Knecht wrote: If I then change my keyboard to drive the HDSP 9652 input 1 (Alsa 64:0) and change the connections internally to drive all of outputs, I get stuck notes pretty much immediately. It seems a bit worse with the sustain pedal, but does not seem to be effected at all by controllers. It is heavily effected by the MIDI note density. If I hit only one or two notes, I'm not likely to get it, but using the sustain pedal I can create the problem in under a minute. To be sure it's the input and not the outputs (as much as I can be) I have external synths attached to the Alsa outputs on 64:0, 72:0 and 72:1. When I get a stuck note, I seem to get it on both internal soft synths and all three external hardware synths at the same time. For this reason I deduce that it is the HDSP input that is not clearing out whatever event queue that holds this stuff and somehow the note never shuts off. to be sure, the configuration which doesn't work is like below, ok? HDSP MIDI1 input - softsynth and/or HDSP MIDI1 input - HDSP MIDI1 output - external device Both don't work, and when they fail, they both fail at the same time in the same way, with a note stuck on. That's why I titled the thread HDSP MIDI In - stuck notes. (I'm actually using MIDI 0, not MIDI 1) - softsynth | HDSP MIDI 0 -- | - HDSP MIDI 0 output - external synth Using the USB MIDI in does not fail: - softsynth | MidiSport 0 -- | - HDSP MIDI 0 output - external synth at least, we need to check whether the interrupts for MIDI are generated properly. please try the following. 1. connect HDSP MIDI1 input to HDSP MIDI1 output via aconnect. 2. trigger a note from MIDI1 input. check whether the IRQ count in /proc/interrupts (for HDSP) increases. during this test, don't use HDSP audio. [EMAIL PROTECTED] card1]$ more /proc/interrupts CPU0 0: 62488 XT-PIC timer 1:816 XT-PIC keyboard 2: 0 XT-PIC cascade 5:749 XT-PIC usb-uhci, usb-uhci, usb-uhci, eth0 8: 1 XT-PIC rtc 10: 94 XT-PIC hdsp 11: 5 XT-PIC ohci1394 12: 7033 XT-PIC PS/2 Mouse 14: 8556 XT-PIC ide2 15: 9005 XT-PIC ide3 NMI: 0 ERR: 0 [EMAIL PROTECTED] card1]$ --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] HDSP 9652 MIDI IN - stuck notes
On Fri, 2003-02-28 at 04:56, Mark Knecht wrote: at least, we need to check whether the interrupts for MIDI are generated properly. please try the following. 1. connect HDSP MIDI1 input to HDSP MIDI1 output via aconnect. 2. trigger a note from MIDI1 input. check whether the IRQ count in /proc/interrupts (for HDSP) increases. during this test, don't use HDSP audio. [EMAIL PROTECTED] card1]$ more /proc/interrupts CPU0 0: 62488 XT-PIC timer 1:816 XT-PIC keyboard 2: 0 XT-PIC cascade 5:749 XT-PIC usb-uhci, usb-uhci, usb-uhci, eth0 8: 1 XT-PIC rtc 10: 94 XT-PIC hdsp 11: 5 XT-PIC ohci1394 12: 7033 XT-PIC PS/2 Mouse 14: 8556 XT-PIC ide2 15: 9005 XT-PIC ide3 NMI: 0 ERR: 0 [EMAIL PROTECTED] card1]$ BTW - The HDSP interrupts above do not represent a failure. All I did is what you asked me to do. If you asked me to wait for a failure, we'd have 1000's on interrupts at least, I'm sure, and I don't know how we would identify that one did not happen. Also, if I wasn't clear earlier, the failure is ONE stuck note. The MIDI input keeps working, and subsequent notes work properly. (both on and off) It's just that a single note gets stuck every 1-2 minutes. --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] RE: [Alsa-user] Midisport 2x2 spawns before Alsa, taking the device0 slot event though modules.conf puts snd-usb-midi as 3rd device - result potential hardlocks when opening dsp
Ivica, Welcome to _MY_ nightmare!!! I've been dealing with this for months!! Actually, your nightmare is a little different, but in my version, just before the Frankenstein monster shows up, my HDSP 9652 won't initialize, the system saying it doesn't have enough memory. Then, just as the monster raises its arms to grab me, I do a warm boot and the monster goes away. Alsa and Jack work wonderfully and the sky is blue... Oh yes, and sometimes my 1394 drives cannot be used either, until after the same warm boot. Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ivica Bukvic Sent: Tuesday, February 25, 2003 3:26 PM To: [EMAIL PROTECTED]; [EMAIL PROTECTED] Subject: [Alsa-user] Midisport 2x2 spawns before Alsa, taking the device0 slot event though modules.conf puts snd-usb-midi as 3rd device - result potential hardlocks when opening dsp Hi all, I was just wondering how can I force snd-usb-midi to assume device 2 slot, rather than the default 0 when spawning, since at boot time on my machine USB gets initialized before alsa and therefore if I have Midisport 2x2 hooked up, it ends up being my default /dev/dsp device (which actually does not exist) and that makes apps like pd hardlock my machine if trying to access them. So my question is twofold: 1) Is there anything I can configure about my boot process, where Alsa gets initialized before USB (I am assuming not, since Alsa does not start until someone logs in, please correct me if I am wrong). 2) Is there anything I can specify in my modules.conf or elsewhere in order to force the device to be used as snd-card-2 and not snd-card-0 (1 I would like to reserve for hdsp). Obviously the quick and dirty fix is to hook-up the midisport once I've logged in and if the device was already hooked-up simply restart alsa once the midisport was removed (which I am doing), but that seems like a real hacky solution to this issue and does not address the root of the problem. Any help is greatly appreciated! Sincerely, Ivica Ico Bukvic, composer, multimedia sculptor, programmer, webmaster computer consultant http://meowing.ccm.uc.edu/~ico To be or not to be - Shakespeare To be is to do - Socrates To do is to be - Sartre Do be do be do - Sinatra 2b || ! 2b - ? I am - God --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-user mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-user --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [PATCH] HDSP rms and peak registers correct offsets (was: Re:[Alsa-devel] hdsp driver)
Thomas, Will this work cover the HDSP 9652 also, or is that different and has to wait? Thanks, Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Thomas Charbonnel Sent: Sunday, March 02, 2003 3:43 PM To: Paul Davis Cc: alsa-devel Subject: [PATCH] HDSP rms and peak registers correct offsets (was: Re:[Alsa-devel] hdsp driver) On Sun, 2003-03-02 at 21:24, Thomas Charbonnel wrote: On Thu, 2003-02-27 at 19:43, Paul Davis wrote: i haven't had time today to get the patch for the hdsp ready. however, the new source works here (i have some CVS sync issues). i'll get it out to the list on monday (i'm gone for the weekend). Did you sort out the rms register issue ? I'm currently implementing the metering part of HDSPMixer and the rms values returned are obviously bogus. Thomas Ok, I sorted it out myself, the offsets for the rms and peak registers were wrong. Sorry Paul, I could have seen this much earlier if I had done more extensive tests: I kept on testing inputs 1 and 2 while the input rms offset pointed on input 3... :-\ The correct offsets can be found in the attached patch. Thomas --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [PATCH] HDSP rms and peak registers correct offsets (was:Re:[Alsa-devel] hdsp driver)
Thomas, Thanks for the update. Mark -Original Message- From: Thomas Charbonnel [mailto:[EMAIL PROTECTED] Sent: Monday, March 03, 2003 6:40 AM To: Mark Knecht Cc: alsa-devel Subject: RE: [PATCH] HDSP rms and peak registers correct offsets (was:Re:[Alsa-devel] hdsp driver) On Mon, 2003-03-03 at 15:23, Mark Knecht wrote: Thomas, Will this work cover the HDSP 9652 also, or is that different and has to wait? Thanks, Mark No, sorry, I'm afraid you'll have to wait. Thomas --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] patch for h-dsp driver
Paul,
Thanks for the work. As I am Planet flow, I'll have to wait for
Fernando's next Alsa build cycle to test the HDSP 9652 mixer, but I'll be on
it within minutes of his release, I promise. (And I understand it is
untested and may not work at all.)
Or, if someone wants to help me through the struggle of building a
kernel, Alsa and Jack from scratch, I'm off on Friday and would be happy to
try to do that. Best, though, if the Alsa and Jack builds were actually RPMs
so as to no upset the Planet based RPM flow. (That's a lot to ask, I know.)
Cheers,
Mark
-Original Message-
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] Behalf Of Paul Davis
Sent: Tuesday, March 04, 2003 1:08 PM
To: [EMAIL PROTECTED]
Subject: [Alsa-devel] patch for h-dsp driver
Enclosed below is a substantial patch (against current CVS) for the
HDSP driver. It does a number of things:
* read access to mixer
* remove force_firmware option, since it does nothing (the
h/w can't be reloaded without a power cycle)
* fix RMS meter offsets (from Thomas Charbonnel)
* fix number of output peak meter controls (from Thomas Charbonnel)
* handle various newer PCI rev numbers
* create a more informational card_name field
* support for HDSP rev11 (0xb) mixer
* attempts to implement HDSP9652 mixer control (untested)
* add new rev11 config file (sort-of-firmware)
the new firmware will be enclosed in a separate message, compressed, etc.
Still to come: RMS meter controls for the HDSP9652 hardware, firmware
loading from a file, use of udelay() to handle wait loops.
--p
Index: hdsp.c
===
RCS file: /cvsroot/alsa/alsa-kernel/pci/rme9652/hdsp.c,v
retrieving revision 1.26
diff -u -u -r1.26 hdsp.c
--- hdsp.c4 Mar 2003 16:47:35 - 1.26
+++ hdsp.c4 Mar 2003 20:47:31 -
@@ -23,7 +23,6 @@
#include sound/driver.h
#include linux/delay.h
#include linux/interrupt.h
-#include linux/init.h
#include linux/slab.h
#include linux/pci.h
@@ -42,13 +41,14 @@
#include multiface_firmware.dat
#include digiface_firmware.dat
+#include multiface_firmware_rev11.dat
+#include digiface_firmware_rev11.dat
static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX; /* Index 0-MAX */
static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;/* ID for
this card */
static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP; /*
Enable this card */
static int precise_ptr[SNDRV_CARDS] = { [0 ... (SNDRV_CARDS-1)]
= 0 }; /* Enable precise pointer */
static int line_outs_monitor[SNDRV_CARDS] = { [0 ...
(SNDRV_CARDS-1)] = 0}; /* Send all inputs/playback to line outs */
-static int force_firmware[SNDRV_CARDS] = { [0 ...
(SNDRV_CARDS-1)] = 0}; /* Force firmware reload */
MODULE_PARM(index, 1- __MODULE_STRING(SNDRV_CARDS) i);
MODULE_PARM_DESC(index, Index value for RME Hammerfall DSP interface.);
@@ -65,19 +65,17 @@
MODULE_PARM(line_outs_monitor,1- __MODULE_STRING(SNDRV_CARDS) i);
MODULE_PARM_DESC(line_outs_monitor, Send all input and playback
streams to line outs by default.);
MODULE_PARM_SYNTAX(line_outs_monitor, SNDRV_ENABLED ,
SNDRV_BOOLEAN_FALSE_DESC);
-MODULE_PARM(force_firmware,1- __MODULE_STRING(SNDRV_CARDS) i);
-MODULE_PARM_DESC(force_firmware, Force a reload of the I/O box
firmware);
-MODULE_PARM_SYNTAX(force_firmware, SNDRV_ENABLED ,
SNDRV_BOOLEAN_FALSE_DESC);
MODULE_AUTHOR(Paul Davis [EMAIL PROTECTED]);
MODULE_DESCRIPTION(RME Hammerfall DSP);
MODULE_LICENSE(GPL);
MODULE_CLASSES({sound});
-MODULE_DEVICES({{RME,Hammerfall-DSP}});
+MODULE_DEVICES({{RME Hammerfall-DSP},
+ {RME HDSP-9652}});
typedef enum {
Digiface,
- Multiface
-} HDSP_Type;
+ Multiface,
+} HDSP_IO_Type;
#define HDSP_MAX_CHANNELS26
#define DIGIFACE_SS_CHANNELS 26
@@ -123,9 +121,9 @@
#define HDSP_playbackPeakLevel 4096 /* 26 * 32 bit values */
#define HDSP_inputPeakLevel 4224 /* 26 * 32 bit values */
-#define HDSP_outputPeakLevel4100 /* 26 * 32 bit values */
+#define HDSP_outputPeakLevel4352 /* 26 * 32 bit values */
#define HDSP_playbackRmsLevel 4612 /* 26 * 64 bit values */
-#define HDSP_inputRmsLevel 4884 /* 26 * 64 bit values */
+#define HDSP_inputRmsLevel 4868 /* 26 * 64 bit values */
#define HDSP_IO_EXTENT 5192
@@ -287,10 +285,6 @@
#define HDSP_LONG_WAIT4
#define HDSP_SHORT_WAIT 100
-/* Computing addresses for adjusting gains */
-
-#define INPUT_TO_OUTPUT_KEY(in,out) ((64 * (out)) + (in))
-#define PLAYBACK_TO_OUTPUT_KEY(chn,out) ((64 * (out)) + 32 + (chn))
#define UNITY_GAIN 32768
#define MINUS_INFINITY_GAIN 0
@@ -335,42 +329,43 @@
};
struct _hdsp {
- spinlock_t lock;
+ spinlock_t lock;
snd_pcm_substream_t *capture_substream;
snd_pcm_substream_t
Re: [Alsa-devel] HDSP 9652 MIDI IN - stuck notes
Takashi-san, Was this of any use? Are you looking into this, or do you need more data? Thanks, Mark On Fri, 2003-02-28 at 05:09, Mark Knecht wrote: On Fri, 2003-02-28 at 04:56, Mark Knecht wrote: at least, we need to check whether the interrupts for MIDI are generated properly. please try the following. 1. connect HDSP MIDI1 input to HDSP MIDI1 output via aconnect. 2. trigger a note from MIDI1 input. check whether the IRQ count in /proc/interrupts (for HDSP) increases. during this test, don't use HDSP audio. [EMAIL PROTECTED] card1]$ more /proc/interrupts CPU0 0: 62488 XT-PIC timer 1:816 XT-PIC keyboard 2: 0 XT-PIC cascade 5:749 XT-PIC usb-uhci, usb-uhci, usb-uhci, eth0 8: 1 XT-PIC rtc 10: 94 XT-PIC hdsp 11: 5 XT-PIC ohci1394 12: 7033 XT-PIC PS/2 Mouse 14: 8556 XT-PIC ide2 15: 9005 XT-PIC ide3 NMI: 0 ERR: 0 [EMAIL PROTECTED] card1]$ BTW - The HDSP interrupts above do not represent a failure. All I did is what you asked me to do. If you asked me to wait for a failure, we'd have 1000's on interrupts at least, I'm sure, and I don't know how we would identify that one did not happen. Also, if I wasn't clear earlier, the failure is ONE stuck note. The MIDI input keeps working, and subsequent notes work properly. (both on and off) It's just that a single note gets stuck every 1-2 minutes. --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel --- This SF.net email is sponsored by: Etnus, makers of TotalView, The debugger for complex code. Debugging C/C++ programs can leave you feeling lost and disoriented. TotalView can help you find your way. Available on major UNIX and Linux platforms. Try it free. www.etnus.com ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] alsaconf -L - where's Waldo?
Hi, The --help option says that I can get a log file from alsaconf using the -L option. If I look in /tmp while alsaconf is running, I see some files, but when alsaconf has finished, the files are gone... [EMAIL PROTECTED] mark]$ ls /tmp alsaconf.3nMgvm alsaconf.txNTFm ksocket-mark orbit-mark ssh-XX7Jn9me alsaconf.Ef6KCm kde-mark mcop-mark ssh-XX1CS1IB ssh-XXIbrHpS [EMAIL PROTECTED] mark]$ ls /tmp kde-mark ksocket-mark mcop-mark orbit-mark ssh-XX1CS1IB ssh-XX7Jn9me ssh-XXIbrHpS [EMAIL PROTECTED] mark]$ Is this file possibly put somewhere else? I'm trying to figure out why alsaconf is complaining about this card. Loading driver.. Starting sound driver snd-hdsp /lib/modules/2.4.20-1.12.ll.acpi/kernel/drivers/sound/pci/rme9652/snd-hdsp.o: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg init_module: No such device /lib/modules/2.4.20-1.12.ll.acpi/kernel/drivers/sound/pci/rme9652/snd-hdsp.o: insmod /lib/modules/2.4.20-1.12.ll.acpi/kernel/drivers/sound/pci/rme9652/snd-hdsp.o failed /lib/modules/2.4.20-1.12.ll.acpi/kernel/drivers/sound/pci/rme9652/snd-hdsp.o: insmod snd-hdsp failed [FAILED] Setting default volumes.. === Now ALSA is ready to use. For adjustment of volumes, please use alsamixer or gamix. Have a lot of fun! [EMAIL PROTECTED] etc]# --- This sf.net email is sponsored by:ThinkGeek Welcome to geek heaven. http://thinkgeek.com/sf ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] NEW hdsp 9652 problems - {WAS: alsaconf -L - where's Waldo?)
Takashi, I have two sound devices (on-board Via device and an RME HDSP 9652) but I only want to configure the RME. Currently alsaconf finds the RME, attempts to install the drivers, and then the drivers fail saying they cannot find the cards. I had hoped that possibly the log file would give me more information on why. I suppose it will not if it's only for ISA devices. (BTW - alsaconf does find this Via chip and can configure it correctly.) I think that the --help for alsaconf should make this more clear. Currently it says: -L|--log logging on /tmp/alsaconf.log No idea it only covers ISA devcies. OK, so I start with a clean /etc/modules.conf looking like this: alias parport_lowlevel parport_pc alias eth0 bcm4400 alias usb-controller ehci-hcd alias usb-controller1 usb-uhci lspci -v shows: SNIP 00:0e.0 Multimedia audio controller: Xilinx, Inc. RME Hammerfall DSP (rev 68) Flags: bus master, medium devsel, latency 32, IRQ 10 Memory at e980 (32-bit, non-prefetchable) [size=64K] SNIP I then run alsaconf. It shows the hdsp and the via82xx. I choose the hdsp and get the following: Loading driver.. Starting sound driver snd-hdsp /lib/modules/2.4.20-1.12.ll.acpi/kernel/drivers/sound/pci/rme9652/snd-hdsp.o: Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters. You may find more information in syslog or the output from dmesg init_module: No such device /lib/modules/2.4.20-1.12.ll.acpi/kernel/drivers/sound/pci/rme9652/snd-hdsp.o: insmod /lib/modules/2.4.20-1.12.ll.acpi/kernel/drivers/sound/pci/rme9652/snd-hdsp.o failed /lib/modules/2.4.20-1.12.ll.acpi/kernel/drivers/sound/pci/rme9652/snd-hdsp.o: insmod snd-hdsp failed [FAILED] Setting default volumes.. === Now ALSA is ready to use. For adjustment of volumes, please use alsamixer or gamix. Have a lot of fun! [EMAIL PROTECTED] etc]# modules.conf now looks like: alias parport_lowlevel parport_pc alias eth0 bcm4400 alias usb-controller ehci-hcd alias usb-controller1 usb-uhci # --- BEGIN: Generated by ALSACONF, do not edit. --- # --- ALSACONF verion 0.9.0 --- alias char-major-116 snd alias snd-card-0 snd-hdsp alias char-major-14 soundcore alias sound-slot-0 snd-card-0 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 options snd major=116 cards_limit=1 device_mode=0666 options snd-hdsp index=0 # --- END: Generated by ALSACONF, do not edit. The card is not recognized on reboot either. dmesg shows a log string of these messages: Hammerfall memory allocator: buffers allocated for 1 cards RME Hammerfall-DSP: no cards found Hammerfall memory allocator: buffers allocated for 1 cards RME Hammerfall-DSP: no cards found Now, this all worked fine on the previous 2.4.19-1.ll kernel from the PlanetCCRMA site **BEFORE** I updated firmware on the HDSP 9652 for new Win XP driver support. After updating firmware for the card, the alsaconf program I had for the 2.4.19-1.ll kernel no longer recognized the card at all and cannot write anything in modules.conf. I have a new PC in my studio, using a newer kernel from the Planet not released widely yet. (2.4.20-1.12.ll.acpi) The kernel is dated Sunday, Feb. 2nd. The alsa RPM is dated Feb. 18th and appears to be alsa 0.9.0-45, if that makes sense. I understand some of this might be Fernando's numbering system for the Planet. The alsaconf that comes with this kernel recognizes the card, but the driver doesn't load and says there's no card. Chicken egg problem apparently... Thanks for any help you can provide. Cheers, Mark On Fri, 2003-03-21 at 05:59, Takashi Iwai wrote: At 21 Mar 2003 05:50:47 -0800, Mark Knecht wrote: Hi, The --help option says that I can get a log file from alsaconf using the -L option. If I look in /tmp while alsaconf is running, I see some files, but when alsaconf has finished, the files are gone... yep, they are temporary working directories. [EMAIL PROTECTED] mark]$ ls /tmp alsaconf.3nMgvm alsaconf.txNTFm ksocket-mark orbit-mark ssh-XX7Jn9me alsaconf.Ef6KCm kde-mark mcop-mark ssh-XX1CS1IB ssh-XXIbrHpS [EMAIL PROTECTED] mark]$ ls /tmp kde-mark ksocket-mark mcop-mark orbit-mark ssh-XX1CS1IB ssh-XX7Jn9me ssh-XXIbrHpS [EMAIL PROTECTED] mark]$ Is this file possibly put somewhere else? I'm trying to figure out why alsaconf is complaining about this card. /tmp/alsaconf.log is created only when the legacy cards are probed. it's for a deubgging purpose only... Loading driver.. Starting sound driver snd-hdsp /lib/modules/2.4.20-1.12.ll.acpi/kernel/drivers/sound/pci/rme9652/snd-hdsp.o: Hint: insmod errors can
Re: [Alsa-devel] NEW hdsp 9652 problems - {WAS: alsaconf -L -where's Waldo?)
On Fri, 2003-03-21 at 08:39, Justin Cormack wrote: On Fri, 2003-03-21 at 16:17, Mark Knecht wrote: Takashi, I have two sound devices (on-board Via device and an RME HDSP 9652) but I only want to configure the RME. Currently alsaconf finds the RME, attempts to install the drivers, and then the drivers fail saying they cannot find the cards. I had hoped that possibly the log file would give me more information on why. I suppose it will not if it's only for ISA devices. (BTW - alsaconf does find this Via chip and can configure it correctly.) You need a more recent version of the driver, probably the cvs version for this card. The configurator probably checks the PCI id but not the revision nuumber. Justin, Thanks for the reply. It makes sense to me, however, since I use the PlanetCCRMA flow, and alsa is supplied as an RPM, I have no way (that I know of) to just get a new driver, compile it and try it out. I need a new RPM, I think. Thanks for the ideas! Cheers, Mark --- This SF.net email is sponsored by:Crypto Challenge is now open! Get cracking and register here for some mind boggling fun and the chance of winning an Apple iPod: http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0031en ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] NEW hdsp 9652 problems - {WAS: alsaconf -L -where's Waldo?)
On Fri, 2003-03-21 at 09:45, Paul Davis wrote: Now, this all worked fine on the previous 2.4.19-1.ll kernel from the PlanetCCRMA site **BEFORE** I updated firmware on the HDSP 9652 for new Win XP driver support. After updating firmware for the card, the alsaconf program I had for the 2.4.19-1.ll kernel no longer recognized the card at all and cannot write anything in modules.conf. you cannot use the current ALSA driver with the new firmware. i submitted a patch to handle the new firmware, but it has not been put into CVS yet. meanwhile, thomas charbonnel has extended the work i did and added some more stuff, and so we are working on a mega-patch that fixes lots of issues with full H-DSP support. once thomas gets his tasklet problems fixed (for MIDI I/O without affecting PCM latency), we'll submit the patch again. --p Paul, Well, I guess thanks then for saving the rest of my vacation Friday from being a total waste of time. I took the day off to focus on getting this Alsa stuff going. Now I find I cannot. Too bad for me I guess. I guess I will take my vacation day and rewire my studio to get this Linux box out of the audio path completely. In a few hours I'll be back to where I was last September, other than using Linux for email now, it just doesn't seem like a positive 6 months. Not your fault. All mine for even trying to use this Linux stuff seriously I suppose. Such disappointment. Mark --- This SF.net email is sponsored by:Crypto Challenge is now open! Get cracking and register here for some mind boggling fun and the chance of winning an Apple iPod: http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0031en ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] NEW hdsp 9652 problems - {WAS: alsaconf -L -where's Waldo?)
On Fri, 2003-03-21 at 11:05, Paul Davis wrote: Well, I guess thanks then for saving the rest of my vacation Friday from being a total waste of time. I took the day off to focus on getting this Alsa stuff going. Now I find I cannot. Too bad for me I guess. mark - i'm not trying to apportion blame, but i would note that if you had not updated the firmware, this would not have happened. also, i had hoped that takashi or jaroslav would have applied the patches already mailed to the mailing list to CVS, but they chose not for various reasons. Paul, I completely accept responsibility for this series of problems. Between September and December I had an opportunity to choose what hardware to buy. I made my choice, right or wrong. From December until today I have waited for this hardware to really work, but it doesn't work today. Two weeks ago, wanting to compare Linux and Windows in more depth, and making no real progress with Linux, I pressed the button and allowed RME's software to update my card so that I could use the best they have to offer in Windows. It killed Linux. That was my choice, wise or not. On the positive side, I must tell people that Linux kicks but. I can run my box with Linux/Alsa/Jack, doing recording, or playing CDs, at sub 3mS latencies. I've pushed it down to 1.2mS and it's not failed, but I think it might. The same box, running Win XP, cannot do better than 12mS and seems to work better at 23mS. Guess which one I'd rather be using when I use soft synths live? This issue is not your fault, nor your responsility to fix. You are quite busy with two very important projects, Jack and Ardour. You don't need to become a focal point for developing specifc drivers. I don't expect it of you. I have come to understand just how much Alsa support is on a very adhoc basis. It is not possible, I think, for any user like myself to depend on a specific piece of hardware having support, or really knowing when any piece of hardware will _ever_ be supported. I've also come to find out just how much Alsa and Jack are still at the point of changing protocols. This causes things to break. Beyond that, there is a general sense that if developers want to change things and cause stuff to break, then users should just put up with it, or learn to compile kernels, or spend their time reading web sites to find solutions for problems that didn't exist the day before. I don't think that works very well, but I have no power to influence the outcome of you or anyone else in the development community when you press your buttons. The unfortunate part of this adventure into Linuxland is that 6 months has gone by and I have barely written music. This is not what I want, nor the way I'm willing to spend my time. Will this machine ever work in the future? I don't know, but I hope so. In the meantime I have a shiny new Athlon XP 2600+, 512MB of PC2700 DRAM, RME HDSP 9652 and a disk drive that's doing 45-50MB/Sec sitting here doing nothing but fetchmail. Cheers, Mark i use my H-DSP almost every day, and it works without problems for most things (you know most of the issues). the problems you have had are unfortunate, and i regret them, but unfortunately i do not have write access to ALSA CVS (i'm not sure i want it), and trying to support the very confusing situation that RME have created with the H-DSP has been hard to do (2 PCI rev numbers, 2 incompatible ROM-based firmware versions, 2 versions of the driver-loaded firmware, plus the hdsp-9652 which is a lot like the h-dsp but has no driver-loaded firmware and different register access rules sigh). i believe that my work and now thomas's will be available very soon, and will make the hdsp a truly phenomenal card under linux. thomas also has written an almost-finished version of totalmix for linux (with bitmaps from RME), and that is really impressive to see running. next time you go to upgrade the firmware on anything (your BIOS, a SCSI adapter, a sound card), its best to be sure to ask first if it will affect existing driver's ability to interact with the h/w. --p --- This SF.net email is sponsored by:Crypto Challenge is now open! Get cracking and register here for some mind boggling fun and the chance of winning an Apple iPod: http://ads.sourceforge.net/cgi-bin/redirect.pl?thaw0031en ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] HDSP 9652 Status Request
Hi, I'm just checking in to see what progress has been made on getting this patch Paul did to support RME's new firmware for the HDSP series into CVS. I've been literally down for 4-5 weeks now waiting for this. I am completely unable to run Alsa at all without this patch. Can someone please provide an update on where all of this has progressed to? Thanks, Mark * From: Paul Davis * Subject: Re: [Alsa-devel] NEW hdsp 9652 problems * Date: Fri, 21 Mar 2003 10:06:21 -0800 Now, this all worked fine on the previous 2.4.19-1.ll kernel from the PlanetCCRMA site **BEFORE** I updated firmware on the HDSP 9652 for new Win XP driver support. After updating firmware for the card, the alsaconf program I had for the 2.4.19-1.ll kernel no longer recognized the card at all and cannot write anything in modules.conf. you cannot use the current ALSA driver with the new firmware. i submitted a patch to handle the new firmware, but it has not been put into CVS yet. meanwhile, thomas charbonnel has extended the work i did and added some more stuff, and so we are working on a mega-patch that fixes lots of issues with full HDSP support. once thomas gets his tasklet problems fixed (for MIDI I/O without affecting PCM latency), we'll submit the patch again. --p --- This SF.net email is sponsored by: ValueWeb: Dedicated Hosting for just $79/mo with 500 GB of bandwidth! No other company gives more support or power for your dedicated server http://click.atdmt.com/AFF/go/sdnxxaff00300020aff/direct/01/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] HDSP 9652 Status Request
On Thu, 2003-04-03 at 21:46, Thomas Charbonnel wrote: Hi, I'm just checking in to see what progress has been made on getting this patch Paul did to support RME's new firmware for the HDSP series into CVS. I've been literally down for 4-5 weeks now waiting for this. I am completely unable to run Alsa at all without this patch. Can someone please provide an update on where all of this has progressed to? Thanks, Mark Hi Mark, I have been working on this patch for a moment. It is nearly finished now. Unfortunately I couldn't spend much time on it this week. I'll be back on it this evening. Expect a release quite soon. It'll be done by the end of the week-end. Exciting features include : 96 kHz support, midi I/O done in a tasklet, firmware loading from userspace application (for multiface digiface users, comes handy if the power supply of the io box gets disconnected, avoids a reboot), plus nice fltk GUI to control and display card settings (sample rate, preferred sync source, spdif bits, sync state, etc...). You'll have to wait a little more for the totalmix clone, but it's nearly finished too. Thomas Thomas, Thanks for the update. I wait patiently, if not anxiously, to be able to use Linux again. Over the last two weeks I've gone back to Windows, written 6 new pieces using Acid Pro and Pro Tools. I look very forward to getting Linux support operational again so that I can add all of these soft synths into the mix. Thanks for all your hard work. Cheers, Mark --- This SF.net email is sponsored by: ValueWeb: Dedicated Hosting for just $79/mo with 500 GB of bandwidth! No other company gives more support or power for your dedicated server http://click.atdmt.com/AFF/go/sdnxxaff00300020aff/direct/01/ ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] Hammerfall DSP System Problems
Hi Frank, I'm an HDSP 9652 non-user for many of the same reasons. A couple of comments below. Mark Hello, I recently purchased a Hammerfall DSP system to use on Windows and especially LINUX for use on Ardour. I have the multiface system. I must say I am confused on the support of the newest version of this pci card (I know that I have firmware version 11). I downloaded the current CVS on alsa-dev and compiled it fine. However when I run modprobe I get the following error: /sbin/modprobe snd-hdsp /lib/modules/2.4.18-3/kernel/sound/pci/rme9652/snd-hdsp.o: init_module: No such device [MWK] This is sometimes caused by a mismatch between between the firmware revision in your card/box and those firmware revisions that the driver knows about. Run lspci to see what your card has, and read through the driver code to see what the driver knows about. NOTE TO DEVELOPERS - Can't you make this message init_module: No such device more descriptive, such as init_module: Checked for devices with PCI Device ID's XXX, firmware revisions AA, BB, CC, DD EE - No devices found? Many people, including me, don't know how to read source code and can't give you as much direct information. Just a thought. Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters /lib/modules/2.4.18-3/kernel/sound/pci/rme9652/snd-hdsp.o: insmod /lib/modules/2.4.18-3/kernel/sound/pci/rme9652/snd-hdsp.o failed /lib/modules/2.4.18-3/kernel/sound/pci/rme9652/snd-hdsp.o: insmod snd-hdsp failed Then when I just offer the command insmod snd-hdsp I get the following: /sbin/insmod snd-hdsp Using /lib/modules/2.4.18-3/kernel/sound/pci/rme9652/snd-hdsp.o /lib/modules/2.4.18-3/kernel/sound/pci/rme9652/snd-hdsp.o: unresolved symbol snd_hammerfall_get_buffer_R5bab1fdf /lib/modules/2.4.18-3/kernel/sound/pci/rme9652/snd-hdsp.o: unresolved symbol snd_rawmidi_receive_R1e37534c [MWK] I have seen these sorts of errors when: 1) The kernel source code that the driver was compiled against is not the same as the kernel that's running. For instanace, you have a standard distribution kernel and a custom kernel from kernel.org. If you are running the standard kernel, but the Alsa driver was compiled with /usr/src/linux pointing at the custom kernel, then problems ensue. 2) The C compiler used to compile Alsa does not match the compiler used to compile the kernel. SNIP When I run hdsploader it completes but says nothing like this: hdsploader - firmware loader for RME Hammerfall DSP cards Looking for HDSP + Multiface or Digiface cards : [MWK] Again, this strikes me that Alsa does not understand your firmware revision since it isn't even finding a card. Good luck!! --- This SF.net email is sponsored by: Etnus, makers of TotalView, The best thread debugger on the planet. Designed with thread debugging features you've never dreamed of, try TotalView 6 free at www.etnus.com. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] hammerfall_mem.c - obsolete - to be removed
Jaroslav, Is this true with Alsa 0.9.4? Should I remove the loading of this module? Thanks, Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Jaroslav Kysela Sent: Friday, June 06, 2003 5:06 AM To: ALSA development Cc: Paul Davis; Takashi Iwai Subject: [Alsa-devel] hammerfall_mem.c - obsolete - to be removed Hi, I think that hammerfall_mem.c should be removed from the ALSA tree, because we have unified preallocation / buffer cache system in the snd-page-alloc module (alsa-kernel/core/memalloc.c) which completely replaces the original single purpose code. Jaroslav - Jaroslav Kysela [EMAIL PROTECTED] Linux Kernel Sound Maintainer ALSA Project, SuSE Labs --- This SF.net email is sponsored by: Etnus, makers of TotalView, The best thread debugger on the planet. Designed with thread debugging features you've never dreamed of, try TotalView 6 free at www.etnus.com. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel --- This SF.net email is sponsored by: Etnus, makers of TotalView, The best thread debugger on the planet. Designed with thread debugging features you've never dreamed of, try TotalView 6 free at www.etnus.com. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] aplay: pcm_write:1025: write error SNDRV_PCM_IOCTL_PREPAREfailed:device or resource busy
Sorry this is a bit long. Mostly just data. I hope it helps figure this out. Sort of strange that the chip is identified in one place as a VIA8233 and in another place as a VIA8235 On Wed, 2003-05-21 at 04:09, Takashi Iwai wrote: At 18 May 2003 17:05:32 -0700, Mark Knecht wrote: Hi, I brought up Alsa Friday on a Gentoo platform and used it for most of Friday evening and Saturday in one form or another. Along the way I was building some new kernels to fix some SCSI issues and a few other items, which required that Alsa get rebuilt also as per the Gentoo Alsa instructions. This process went on for most of the day. Later Saturday evening, following who knows how many 'unrelated' changes, Alsa stopped working. The error messages look like this: Wizard root # aplay wave/sequen~1.wav Playing WAVE 'wave/sequen~1.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Mono aplay: pcm_write:1025: write error: Input/output error Wizard root # or when trying to play a game like frozen-bubble, that app locks up hard and in my console I see: ALSA Lib pcm_hw.c:467:(snd_pcm_hw_prepare)SNDRV_PCM_IOCTL_PREPARE failed:device or resource busy alsaplayer starts to run, sees the CD (I.e. - shows track length) and then won't play the CD. I'll attach a bit of data, but can anyone point me toward what might be causing this? the chip looks like a VIA8235, so the (first) pcm device supports multiple playbacks but with the same sample rate. please check whether there is any other app running and using the pcm device and if it uses different sample rates (e.g. 48kHz)... Takashi Takashi-san, To the best of my knowledge there are no other apps using sound in any way. The environment is fluxbox, which doesn't use sound. I get the same results in a console after a cold boot before starting fluxbox also. I've tried playing 48KHz and 44.1KHz wave files and get the same results. There is a strange thing I noticed about how the chip is set up though. I've loaded the via82xx driver. In /proc/pci the chip identifies itself as an VIA8233: Wizard root # grep audio /proc/pci Multimedia audio controller: Xilinx, Inc. RME Hammerfall DSP (rev 104). Multimedia audio controller: VIA Technologies, Inc. VT8233 AC97 Audio Controller (rev 80). Wizard root # However, in aplay it identifies itself as a VIA 8235: bash-2.05b$ aplay -l List of PLAYBACK Hardware Devices card 0: 8235 [VIA 8235], device 0: VIA 8235 [VIA 8235] Subdevices: 4/4 Subdevice #0: subdevice #0 Subdevice #1: subdevice #1 Subdevice #2: subdevice #2 Subdevice #3: subdevice #3 card 0: 8235 [VIA 8235], device 1: VIA 8235 [VIA 8235] Subdevices: 1/1 Subdevice #0: subdevice #0 bash-2.05b$ There are no processes running that are obviously usign audio, as far as I can see: bash-2.05b$ ps aux USER PID %CPU %MEM VSZ RSS TTY STAT START TIME COMMAND root 1 0.2 0.0 1352 496 ?S20:54 0:04 init root 2 0.0 0.0 00 ?SW 20:54 0:00 [keventd] root 3 0.0 0.0 00 ?SWN 20:54 0:00 [ksoftirqd_CPU0] root 4 0.0 0.0 00 ?SW 20:54 0:00 [kswapd] root 5 0.0 0.0 00 ?SW 20:54 0:00 [bdflush] root 6 0.0 0.0 00 ?SW 20:54 0:00 [kupdated] root 8 0.0 0.0 00 ?SW 20:54 0:00 [khubd] root12 0.0 0.0 00 ?SW 20:54 0:00 [kjournald] root36 0.0 0.1 1748 916 ?S20:54 0:00 /sbin/devfsd /dev root 843 0.0 0.1 1428 572 ?S20:54 0:00 /usr/sbin/syslogd -m 0 root 846 0.0 0.1 1472 576 ?S20:54 0:00 /usr/sbin/klogd -c 3 -2 root 849 0.0 0.1 1472 568 ?S20:54 0:00 /usr/sbin/crond root 1089 0.0 0.2 2732 1284 ?S20:54 0:00 /usr/sbin/sshd xfs 1242 0.0 0.9 6264 4772 ?S20:54 0:00 [xfs] root 1252 0.0 0.2 2288 1236 ?S20:54 0:00 [login] root 1253 0.0 0.0 1336 424 vc/2 S20:54 0:00 /sbin/agetty 38400 tty2 linux root 1254 0.0 0.0 1336 424 vc/3 S20:54 0:00 /sbin/agetty 38400 tty3 linux mark 1322 0.0 0.2 2292 1372 vc/1 S20:54 0:00 -bash mark 1331 0.0 0.1 2036 976 vc/1 S20:55 0:00 /bin/sh /usr/X11R6/bin/startx mark 1342 0.0 0.1 2324 624 vc/1 S20:55 0:00 xinit /home/mark/.xinitrc -- -deferglyphs 16 root 1343 0.3 2.5 92644 13356 ? R20:55 0:06 [X] mark 1361 0.0 0.5 5212 2648 vc/1 S20:55 0:00 /usr/bin/fluxbox mark 1560 0.0 0.3 3692 1832 ?S21:13 0:00 /usr/bin/gconfd-1 12 mark 1628 0.0 0.8 6928 4332 ?R21:21 0:00 [xterm] mark 1629 0.0 0.2 2312 1396 pts/0S21:21 0:00 bash mark 1638 0.0 0.1 2664 800 pts/0R21:23 0:00 ps aux My
Re: [Alsa-devel] position at sun/paul davis
Orm, I must say that I think this is the biggest bunch of crap I've seen on a Linux list in a long time. This list is no place for this sort of discussion and it's really a low act on your part to publish innuendo and rumor like it was the truth. Even if it was true (AND I'M NOT SAYING IT IS. I HAVE NO INFORMATION AT ALL.) that Paul was not hired for some job at Sun, there are about a 1000 other potential reasons why that might have happened vs. some rumor of his technical capabilities, and I do not see what bearing a Sun-Ray job has on Alsa development anyway. I think the fact that you are publishing nothing but rumor is really a lousy thing to do and that you should apologize to the group here and to Paul personally for such a childish act. Even if you have real data, this is no place to play these games, and this is no way to do it. If this was a moderated reflector I would ask to have you banned. What a piece of work you are. I am not a member of the Jack Development list so my response is not going there. If a member of that list sees any reason to send my note there, please do. With best regards, Mark On Fri, 2003-05-30 at 00:37, Orm Finnendahl wrote: Dear Javier Diaz, I just learned that Sun rejected the application of Paul Davis for its job opening in San Jose concerning the architecting of SunRay(thin client) apparently for not considering him competent and appropriate. I'm working in Europe mainly for electronic music studios in the academic field. It is well known here that Paul Davis is one of the principal developers in the open source linux audio community. It seems ironic that one of the motors of linux audio development isn't even considered for a position in a major company, using software which is partly based on his free work. In addition not being able to judge Davis' competence doesn't enhance our trust in the quality of the products Sun develops. Be assured, the news about this decision is spread here in Europe and will have a significant impact on our choice of hardware purchase in the future. Please notify the responsible members of your staff. Orm Finnendahl -- Date: Tue, 27 May 2003 14:02:53 PDT To: [EMAIL PROTECTED] From: [EMAIL PROTECTED] Subject: [Jackit-devel] could you please post this opening for a 12 month contract in San Jose, CA. DUTIES Work as a member of a team in architecting Audio framework for SunRay(thin client) on Linux.Work involves in-depth understanding of the Audio frameworks in Linux space such as (OSS and ALSA) and providing multi-user pseudo audio device interface for SunRay.Must have experience on implementation of CSS and/or ALSA framework and has contributed tothe Linux Opensrc community. In depth Linux kernal experience is a must.*** SKILLS Strong C and Linux kernel programming/debugging skills. Working knowledge of Audio drivers based on OSS and ALSA. In-depth working knowledge of ALSA kernel interfaces. Demonstrated contributions to ALSA framework is desirable. Working knowledge of Linux distributions including Rehat, Suse. Minimum of two years experience in driver development in Linux. Minimum of five years experience in systems software development. EDUCATION BS in CS/EE with 9+ years of industry experience. Best Regards, Javier Diaz Contract Manager Yoh Company Hardware, Software and Wireless Technologies (408) 654-9192 Ext. 256 (408) 654-9197 Fax Want to know greater talent? Go to http://www.yoh.com --- This SF.net email is sponsored by: eBay Get office equipment for less on eBay! http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel --- This SF.net email is sponsored by: eBay Get office equipment for less on eBay! http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] position at sun/paul davis
Orm, Imagine my surprise when I rejoined Alsa-Dev yesterday for the first time in at least a month and this was the second email I received?! I am not a member of the Jack lists anymore, so I did not see anything that was posted by anyone, including Paul, there. I looked in the archives of this list, found nothing, and simply had to think that it was some weird personal attack against Paul. I now see that it was not and would like to apologize. I consider both your explanation and especially your graciousness about explaining it to be first rate. Thank you. If you were offended by anything I said, please accept my apologies. Paul is a hard working individual, and quite capable of doing most programming tasks I'm sure. Whether he was qualified for the job at Sun that focused on the Linux kernel, or whether there were other issues involved, I do not nor want to know. I still think this wasn't the place for this sort of message, but I suppose that since Paul's original post started it you were simply responding to that. That said, even rereading your post, I found it hard to be sure you were supporting Paul and anti-Sun. I think had that been clearer I might not have written anything at all. Please accept both my apologies as well as my thanks for how you handled this response. Again, in fairness, please forward this response to any list or person that might have received my previous post. With best regards, Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Orm Finnendahl Sent: Friday, May 30, 2003 7:38 AM To: [EMAIL PROTECTED] Subject: Re: [Alsa-devel] position at sun/paul davis Am Freitag, den 30. Mai 2003 um 07:00:02 Uhr (-0700) schrieb Mark Knecht: Orm, I must say that I think this is the biggest bunch of crap I've seen on a Linux list in a long time. This list is no place for this sort of discussion and it's really a low act on your part to publish innuendo and rumor like it was the truth. Hi Mark, my sincere apologies, if I offended you, Paul or anybody else on this list. Paul posted this information on jackit-devel, as was the job offer so I considered my post appropriate. No low act intended on my side. I'll quote his mail below in case you didn't read it. sorry fore the noise. I'll take the blame. Orm --- ---Original Message--- From: Paul Davis [EMAIL PROTECTED] Sent: 05/28/03 03:55 PM To: [EMAIL PROTECTED] Subject: [Jackit-devel] Re: position at sun I'd just like to point out that Sun decided that my resume wasn't adequate for the position that was posted here yesterday. Apparently, Sun's representative said the candidate must have in depth Linux kernel experience. Also experience in implementation of OSS and/or ALSA framework and has contributed to the Linux Opensrc community. I'm taking this as a personal affront for the time being. If you think you or someone else has a better resume to match what Sun advertised, you should probably get in touch with them. Be sure to send Sun my regards. Oh well, Ardour is a more interesting project anyway :) --p --- --- This SF.net email is sponsored by: eBay Get office equipment for less on eBay! http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel --- This SF.net email is sponsored by: eBay Get office equipment for less on eBay! http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] aplay: pcm_write:1025: write error SNDRV_PCM_IOCTL_PREPAREfailed:device or resource busy
Takashi, While I find this interesting, it doesn't address the problem. I sent a bunch of information and data in the last two emails. Have I sent what you need? Are you able to make any assessment about why aplay will not work with this this sound chip on my Asus motherboard? With best regards, Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Takashi Iwai Sent: Monday, June 02, 2003 4:12 AM To: Mark Knecht Cc: Alsa-Devel Subject: Re: [Alsa-devel] aplay: pcm_write:1025: write error SNDRV_PCM_IOCTL_PREPAREfailed:device or resource busy At 29 May 2003 21:43:39 -0700, Mark Knecht wrote: Sorry this is a bit long. Mostly just data. I hope it helps figure this out. Sort of strange that the chip is identified in one place as a VIA8233 and in another place as a VIA8235 in fact, both VIA8233 and 8235 have the same PCI ID. that's why lspci shows the same name (while 8235 has the revision 0x50). Takashi --- This SF.net email is sponsored by: eBay Get office equipment for less on eBay! http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel --- This SF.net email is sponsored by: eBay Get office equipment for less on eBay! http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] aplay: pcm_write:1025: write error SNDRV_PCM_IOCTL_PREPAREfailed:device or resource busy
Takashi, Thanks for the fast response. Can you give me the exact format for adding the module options? I'm not sure which module to add this to, nor have I ever added module options by hand before. Please be clear as I am not a software person. /etc/modules.conf? Which line? Thanks, Mark -Original Message- From: Takashi Iwai [mailto:[EMAIL PROTECTED] Sent: Monday, June 02, 2003 6:56 AM To: Mark Knecht Cc: Alsa-Devel Subject: Re: [Alsa-devel] aplay: pcm_write:1025: write error SNDRV_PCM_IOCTL_PREPAREfailed:device or resource busy At Mon, 2 Jun 2003 06:41:12 -0700, Mark Knecht wrote: Takashi, While I find this interesting, it doesn't address the problem. I sent a bunch of information and data in the last two emails. Have I sent what you need? Are you able to make any assessment about why aplay will not work with this this sound chip on my Asus motherboard? well, i'm still not sure how can it happen. it looks like an unusual case. anyway, in general, BIOS on some ASUS boards have a bug, and the first PCM device won't work on them. could you try to add the module option dxs_support=2 or 3 ? the option 2 will disable the DXS channel support and works as VIA8233A chip. the option 3 suppresses the sample rate 48k only. Takashi With best regards, Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Takashi Iwai Sent: Monday, June 02, 2003 4:12 AM To: Mark Knecht Cc: Alsa-Devel Subject: Re: [Alsa-devel] aplay: pcm_write:1025: write error SNDRV_PCM_IOCTL_PREPAREfailed:device or resource busy At 29 May 2003 21:43:39 -0700, Mark Knecht wrote: Sorry this is a bit long. Mostly just data. I hope it helps figure this out. Sort of strange that the chip is identified in one place as a VIA8233 and in another place as a VIA8235 in fact, both VIA8233 and 8235 have the same PCI ID. that's why lspci shows the same name (while 8235 has the revision 0x50). Takashi --- This SF.net email is sponsored by: eBay Get office equipment for less on eBay! http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel --- This SF.net email is sponsored by: eBay Get office equipment for less on eBay! http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] aplay: pcm_write:1025: write error SNDRV_PCM_IOCTL_PREPAREfailed:device or resource busy
just add the following line to the end of /etc/modules.conf: options snd-via82xx dxs_support=2 (or dxs_support=3) and run /etc/init.d/alsasound restart. note that the option above is available only on the very recent ALSA version. ciao, Takashi Thanks. That's very clear. I'll try it this evening. (At work right now.) As for Alsa versions, this is a Gentoo ebuild of Alsa. It identifies itself as '0.9.3c-r1'. (Is that right? I might have thought 0.9.3-rc1?) Anyway, that's what it says. New enough for these options? Thanks much! Mark --- This SF.net email is sponsored by: eBay Get office equipment for less on eBay! http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] aplay: pcm_write:1025: write error SNDRV_PCM_IOCTL_PREPAREfailed:device or resource busy
anyway, in general, BIOS on some ASUS boards have a bug, and the first PCM device won't work on them. Can I tell aplay to use any other PCM device? The second device? How might I do this? aplay hw:1 sound.wav??? --- This SF.net email is sponsored by: eBay Get office equipment for less on eBay! http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] aplay: pcm_write:1025: write error SNDRV_PCM_IOCTL_PREPAREfailed:device or resource busy
aplay hw:1 sound.wav??? please use -D option, for example, % aplay -D hw:0,1 sound.wav for the second pcm device. the first 0 means the first card and you cannot ommit it to specify the pcm device number. Thanks. This explanation makes sense. I'll try it out this evening and get back to you then. Cheers, Mark --- This SF.net email is sponsored by: eBay Get office equipment for less on eBay! http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] aplay: pcm_write:1025: write error SNDRV_PCM_IOCTL_PREPAREfailed:device or resource busy
On Mon, 2003-06-02 at 07:25, Takashi Iwai wrote:
just add the following line to the end of /etc/modules.conf:
options snd-via82xx dxs_support=2
(or dxs_support=3)
and run /etc/init.d/alsasound restart.
note that the option above is available only on the very recent ALSA
version.
A solution is found! But I'd like to understand why the other failed:
Wizard root # aplay -D hw:0,1 /home/mark/data/wave/seque~10.wav
Playing WAVE '/home/mark/data/wave/seque~10.wav' : Signed 16 bit Little
Endian, Rate 44100 Hz, Mono
And I get sound just fine...
Why is specifying the device required? As root I have no .asoundrc file,
but as a user I do:
pcm.via82xx {
type hw
card 0
}
ctl.via82xx {
type hw
card 0
}
Apparently my version is not new enough for your new option:
Wizard modules.d # /etc/init.d/alsasound restart
* Shutting down ALSA
modules [ ok ]
* Initialising ALSA
* Starting sound driver: snd-via82xx Warning: ignoring dxs_support=2,
no such parameter in this module
Module snd-via82xx loaded, with
warnings [ ok ]
Wizard modules.d #
And pcm2p did not exist...
Wizard root # ls /proc/asound/card0/
ac97#0 ac97#0regs id pcm0c pcm0p pcm1c pcm1p
Wizard root #
Wizard root # aplay -v /home/mark/data/wave/seque~10.wav
Playing WAVE '/home/mark/data/wave/seque~10.wav' : Signed 16 bit Little
Endian, Rate 44100 Hz, Mono
Plug PCM: Hardware PCM card 0 'VIA 8235' device 0 subdevice 0
Its setup is:
stream : PLAYBACK
access : RW_INTERLEAVED
format : S16_LE
subformat: STD
channels : 1
rate : 44100
exact rate : 44100 (44100/1)
msbits : 16
buffer_size : 22050
period_size : 4410
period_time : 10
tick_time: 1
tstamp_mode : NONE
period_step : 1
sleep_min: 0
avail_min: 4410
xfer_align : 4410
start_threshold : 22050
stop_threshold : 22050
silence_threshold: 0
silence_size : 0
boundary : 1445068800
aplay: pcm_write:1025: write error: Input/output error
Wizard root #
---
This SF.net email is sponsored by: eBay
Get office equipment for less on eBay!
http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5
___
Alsa-devel mailing list
[EMAIL PROTECTED]
https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] Quick question
-Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of David Stuart Sent: Tuesday, June 03, 2003 6:53 AM To: David E. Storey Cc: alsa Subject: Re: [Alsa-devel] Quick question Ah! I do exist! :) SNIP 1) It's relatively easy to install (from rpms anyway) 2) One does not have to tinker too much with a default RedHat (our tested distro) installation to use ALSA. 3) It has good full-duplex support, as opposed to the default OSS setup. 4) JACK is the future, but not yet the present (seems to me) SNIP Hi, Since you're on Redhat, you can install Alsa and Jack very easily using the PlanetCCRMA flow in less than an hour. Check it out if you're interested. Cheers, Mark --- This SF.net email is sponsored by: eBay Get office equipment for less on eBay! http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] How do I make -D hw:0,1 default?
Hi,
Yesterday evening I found that I could use aplay -D hw:0,1 and get sound
out of Alsa. This was cool, but I don't normally do anything with aplay.
Other apps that use sound are still locking up hard.
I thought the .asoundrc file was supposed to make this happen
automatically, but apparently the one on the sound card page at the Alsa
site does not do this for me. Can someone tell me how to make this the
default?
.asoundrc from Alsa site:
pcm.via82xx {
type hw
card 0
}
ctl.via82xx {
type hw
card 0
}
Maybe I need to include the ',1' interface somehow? How would I do that?
Thanks,
Mark
---
This SF.net email is sponsored by: eBay
Get office equipment for less on eBay!
http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5
___
Alsa-devel mailing list
[EMAIL PROTECTED]
https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] still losing with hdsp (me too)
SNIP Xilinx Corporation RME Hammerfall DSP (rev 0b) 0b hexadecimal == 11 decimal so, you have revision 11 firmware... This looks like the problem I'm having that didn't go away by downgrading the firmware. Then again, I'm using the cardbus interface. Anyway, rev 10 firmware gets loaded and rev 11 doesn't. So maybe try downgrading? the (me too) part: Sorry to burn the devel-list bandwidth again. I'm still wishing that somewhere I'd get a hint of what's going on with my system. (The snd-hdsp module locks up when I try to use it. Start aplay or pd and wind up with a module that can't be rmmod'ed because of device or resource busy.) Jaakko, Hi. Sorry if this email doesn't help. I'm currently not using my HDSP 9652 at all since the drivers just don't work, but in the meantime I tried using Alsa with my motherboard's sound chip - a Via8235. I had similar problems as you with that chip. Often apps like aplay locked up for at least a minute or two, and some game apps like Frozen Bubble would lock up forever as far as I could tell. I have also seen the same 'device or resource busy' messages. Last evening I found that at least for aplay, if I specified the specific hardware interface, such as aplay -D hw:0,1 wavefile.wav then the system works fine, at least for aplay. However, just starting aplay using aplay wavefile.wav resulted in error messages. The system still hangs on Frozen Bubble and many other games, so I need to know how to make this -D hw:0,1 option the default. My .asoundrc file does not seem to make that happen. Takashi-san said that some motherboards have BIOS problems. Possibly that's the cause in my case, but I'm not so sure. Maybe you can try aplay -l and get some info about the interface that will help. Again, sorry for probably wasting bandwidth. cheers, Mark --- This SF.net email is sponsored by: eBay Get office equipment for less on eBay! http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] How do I make -D hw:0,1 default?
.asoundrc from Alsa site:
pcm.via82xx {
type hw
card 0
}
ctl.via82xx {
type hw
card 0
}
Maybe I need to include the ',1' interface somehow? How would I do that?
Thanks,
Mark
Would this be correct?
(Info taken from
http://alsa.opensrc.org/index.php?page=.asoundrc
)
pcm.via82xx {
type hw
card 0
device 0
subdevice 1
}
ctl.via82xx {
type hw
card 0
device 0
subdevice 1
}
Or do I need to give it a different name and call it specifically for every
application that needs sound? This is the part I cannot seem to find info
on
Thanks,
Mark
---
This SF.net email is sponsored by: eBay
Get office equipment for less on eBay!
http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5
___
Alsa-devel mailing list
[EMAIL PROTECTED]
https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] How do I make -D hw:0,1 default?
.asoundrc from Alsa site:
pcm.via82xx {
type hw
card 0
}
ctl.via82xx {
type hw
card 0
}
Maybe I need to include the ',1' interface somehow? How would I do that?
Add 'device 1' to {} section.
Jaroslav
Thanks Jaroslav!
Will this make device 1 default for all applications that require sound but
do not ask for anything specific in terms of interfaces?
Also, please ignore the email I just sent to alsa-devel 10 seconds before
your answer arrived at my desk.
Cheers,
Mark
---
This SF.net email is sponsored by: eBay
Get office equipment for less on eBay!
http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5
___
Alsa-devel mailing list
[EMAIL PROTECTED]
https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] How do I make -D hw:0,1 default?
On Tue, 2003-06-03 at 11:36, Mark Knecht wrote:
Add 'device 1' to {} section.
Jaroslav
Thanks Jaroslav!
Will this make device 1 default for all applications that require sound but
do not ask for anything specific in terms of interfaces?
Also, please ignore the email I just sent to alsa-devel 10 seconds before
your answer arrived at my desk.
Cheers,
Mark
Jaroslav,
This did not work for me. I made the edits to the .asoundrc file,
tried aplay with and without the -D, then tried restarting alsa and
tried aplay again. It didn't work.
I did read some time ago that someone had trouble getting the
.asoundrc file to link up with the Alsa driver due to the exact name
used in the .asoundrc file. Could something like that be happening here?
Thanks,
Mark
bash-2.05b$ aplay data/wave/seque~10.wav
Playing WAVE 'data/wave/seque~10.wav' : Signed 16 bit Little Endian,
Rate 44100 Hz, Mono
aplay: pcm_write:1025: write error: Input/output error
bash-2.05b$ aplay -D hw:0,1 data/wave/seque~10.wav
Playing WAVE 'data/wave/seque~10.wav' : Signed 16 bit Little Endian,
Rate 44100 Hz, Mono
bash-2.05b$ more .asoundrc
pcm.via82xx {
type hw
card 0
device 1
}
ctl.via82xx {
type hw
card 0
device 1
}
bash-2.05b$
Wizard root # /etc/init.d/alsasound restart
* Shutting down ALSA
modules
[ ok ]
* Initialising
ALSA
[ ok ]
* Starting sound driver: snd-via82xx
Wizard root # lsmod
Module Size Used byNot tainted
snd-via82xx11692 0
snd-ac97-codec 34624 0 [snd-via82xx]
snd-mpu401-uart 3456 0 [snd-via82xx]
snd-rawmidi14432 0 [snd-mpu401-uart]
snd-seq-device 4192 0 [snd-rawmidi]
snd-pcm60928 0 [snd-via82xx]
snd-timer 15240 0 [snd-pcm]
snd30852 0 [snd-via82xx snd-ac97-codec
snd-mpu401-uart snd-rawmidi snd-seq-device snd-pcm snd-timer]
ide-cd 27048 0 (autoclean)
snd-page-alloc 4908 0 [snd-via82xx snd-pcm]
Wizard root #
Wizard root # exit
logout
bash-2.05b$ aplay data/wave/seque~10.wav
Playing WAVE 'data/wave/seque~10.wav' : Signed 16 bit Little Endian,
Rate 44100 Hz, Mono
aplay: pcm_write:1025: write error: Input/output error
bash-2.05b$
---
This SF.net email is sponsored by: eBay
Get office equipment for less on eBay!
http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5
___
Alsa-devel mailing list
[EMAIL PROTECTED]
https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] How do I make -D hw:0,1 default?
On Tue, 2003-06-03 at 11:58, Jaroslav Kysela wrote:
Will this make device 1 default for all applications that require sound but
do not ask for anything specific in terms of interfaces?
The applications should use 'default' device name in that case. You can
override it, of course:
pcm.!default {
... put your configuration here ...
}
You will propably copy pcm.via82xx configuration.
Or you can redirect the default device to some another pcm:
pcm.!default pcm.via82xx
Jaroslav
Our messages are crossing in the ether! ;-)
OK, using pcm.!default worked great for the aplay example, but the game
frozen-bubble still freezes up. Possibly it's a different problem?
Starting it from a terminal yields:
bash-2.05b$ frozen-bubble
[[ Frozen-Bubble-1.0.0 ]]
SNIP
[SDL Init] [Graphics...] [Levels] [Sound Init]
Ready.
SNIP - At this point I try to start playing the game, but the game is
hung and the following message appears...
ALSA lib pcm_hw.c:467:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy
Killed
bash-2.05b$ !aplay
aplay -D hw:0,1 data/wave/seque~10.wav
Playing WAVE 'data/wave/seque~10.wav' : Signed 16 bit Little Endian,
Rate 44100
Hz, Mono
bash-2.05b$
Looking at frozen-bubble's --help listing, I do not see a way to direct
that game's sound to anything special, nor do I think I should have to,
should I?
Thanks,
Mark
---
This SF.net email is sponsored by: eBay
Get office equipment for less on eBay!
http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5
___
Alsa-devel mailing list
[EMAIL PROTECTED]
https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] How do I make -D hw:0,1 default?
On Tue, 2003-06-03 at 16:30, Florian Schmidt wrote:
hi.. maybe i can add something.. here's a snippet from my .asoundrc
pcm.!default {
type hw
card 0
}
ctl.!default {
type hw
card 0
}
this creates a default pcm device which can point to any kind of pcm
device.. here it refers to card 0 which is a hw device. maybe you cann
add a device 1 or subdevice 1 to make it use the right pcm..
Florian,
Yes. Thanks. This is something Jaroslav pointed out later in the
morning. With this strategy I got at least a simple alsa application to
play sounds cleanly, so this was helpful.
Can you shed any light on getting a game like frozen-bubble to use
Alsa? When I try to start this app it lock up with error messages about
resources being busy.
Am I correct in guessing that an app like frozen-bubble is using the
OSS portion of Alsa? Is there some setup that I can do to point OSS
sound requests to this default device?
Thanks much,
Mark
---
This SF.net email is sponsored by: Etnus, makers of TotalView, The best
thread debugger on the planet. Designed with thread debugging features
you've never dreamed of, try TotalView 6 free at www.etnus.com.
___
Alsa-devel mailing list
[EMAIL PROTECTED]
https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] Re: Alsa-devel digest, Vol 1 #1207 - 14 msgs
Have you also symlinked /dev/dsp to /dev/adsp0 as that will allow oss apps to use the hw:0,1 channel? It not required. See to alsa-kernel/Documentation/OSS-Emulation.txt, look for 'dsp_map' and 'adsp_map'. Jaroslav Patrick and Jaroslav, Interesting ideas. I'll goo look at these docs this evening. I think this may specifically be a Via82xx driver problem. If I change only one line in modules.conf (snd-via82xx to snd-hdsp) the problem does away, so I don't think it's anything specific about my setup, although it could have something to do with my specific motherboard I guess. Anyway, thanks for the ideas. Mark --- This SF.net email is sponsored by: Etnus, makers of TotalView, The best thread debugger on the planet. Designed with thread debugging features you've never dreamed of, try TotalView 6 free at www.etnus.com. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] Re: Alsa-devel digest, Vol 1 #1207 - 14 msgs
On Wed, 2003-06-04 at 11:24, Nathaniel Gray wrote: On Wednesday 04 June 2003 06:48 am, [EMAIL PROTECTED] wrote: Florian, Yes. Thanks. This is something Jaroslav pointed out later in the morning. With this strategy I got at least a simple alsa application to play sounds cleanly, so this was helpful. Can you shed any light on getting a game like frozen-bubble to use Alsa? When I try to start this app it lock up with error messages about resources being busy. That means that some other program is using the soundcard (only one program at a time can use it). If you're running KDE then ARTS is a likely suspect and you can try running artsdsp frozenbubble. Otherwise you can try fuser /dev/dsp or fuser /dev/snd/pcmC0D1p to find out the PID of the process that's using it. Someday the dmix plugin might be usable in which case this problem will go away. Nathanial, Hi. Thanks for the ideas. While I know it's not practical for you to have gone back and reread all the the traffic on this over the last week or two, I think this answer, while technically correct, possibly doesn't apply in this case: 1) As I've reported, I'm running fluxbox which has no specific sound support, servers, etc., to the best of my knowledge. There is certainly nothing obvious when looking at processes. I've used Alsa under both Gnome and KDE, so I am familiar with these potential problems. 2) The single application answer would apply to other Alsa apps, like aplay also. However, I can run aplay multiple times before I run frozen-bubble, and I can run it immediately after killing frozen-bubble after the hand. If some other application was using Alsa, and blocking frozen-bubble, then I would expect it to block ALL other Alsa apps, and not just frozen-bubble. Am I correct in guessing that an app like frozen-bubble is using the OSS portion of Alsa? Is there some setup that I can do to point OSS sound requests to this default device? That's pretty much automatic. There's nothing you need to do. Probably true, but I was told that using aplay would be automatic also, and it wasn't. I've had to associate a pcm.!default entity to get it to work, so something is not quite right. I continue to wonder if something similar is effecting the OSS part of this machine. I will have to look into the fuser idea. It sounds interesting. Currently I have a different sound card loaded and this problem does not appear. I'll have to load up the Via driver and try this out. None the less, thanks for your insights. I will continue to look for the answer, but this is software and I am blind. With best regards, Mark --- This SF.net email is sponsored by: Etnus, makers of TotalView, The best thread debugger on the planet. Designed with thread debugging features you've never dreamed of, try TotalView 6 free at www.etnus.com. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] HDSP 9652 - Some first results, but lockups and noactual audio
Hi, Just a bit more info. By rebooting and doing a couple of power cycles on the external A/D/D/A I have been able to get the RME to acknowledge that it's seeing an ADAT sync at 44.1K, so I think at least the clock is being sent. This specific external unit (AI-3 for now) supports a loopback, where the ADAT in can be looped and sent back on its ADAT out. In doing this, I cannot see the audio signal I send it, implying that it is actually not receiving one. Also, I sent the wave file recorded using jackrec to my Windows box. (A copy of a Yellowjackets tune) It plays fine, so it appears that jack and alsaplayer are working OK. Thanks in advance again. Cheers, Mark On Sat, 2003-04-12 at 10:37, Mark Knecht wrote: [EMAIL PROTECTED] card0]# more /proc/asound/card0/hdsp RME HDSP 9652 (Card #1) Buffers: capture dde0 playback ddc0 IRQ: 10 Registers bus: 0xe880 VM: 0xe0967000 Control register: 0x1008098 Status register: 0x6086a48 Status2 register: 0x8061 FIFO status: 0 MIDI1 Output status: 0xff00 MIDI1 Input status: 0xff80 MIDI2 Output status: 0xff00 MIDI2 Input status: 0xff5f Buffer Size (Latency): 1024 samples (2 periods of 4096 bytes) Hardware pointer (frames): 1024 Passthru: no Line out: on Firmware version: 1 Sample Clock Source: Internal 44.1 kHz Preferred Sync Reference: ADAT1 AutoSync Reference: ADAT1 AutoSync Frequency: 48000 System Clock Mode: Master System Clock Frequency: 44100 IEC958 input: Internal IEC958 output: Coaxial only IEC958 quality: Consumer IEC958 emphasis: off IEC958 NonAudio: off IEC958 sample rate: Error flag set ADAT1: Lock ADAT2: No Lock ADAT3: No Lock SPDIF: No Lock Word Clock: No Lock ADAT Sync: No Lock [EMAIL PROTECTED] card0]# [EMAIL PROTECTED] mark]$ cat /proc/asound/card0/hdsp RME HDSP 9652 (Card #1) Buffers: capture de60 playback de40 IRQ: 10 Registers bus: 0xe880 VM: 0xe0967000 Control register: 0x1008090 Status register: 0x2040008 Status2 register: 0x8041 FIFO status: 0 MIDI1 Output status: 0xff00 MIDI1 Input status: 0xff3d MIDI2 Output status: 0xff00 MIDI2 Input status: 0xff39 Buffer Size (Latency): 64 samples (2 periods of 256 bytes) Hardware pointer (frames): 0 Passthru: no Line out: on Firmware version: 1 Sample Clock Source: Internal 44.1 kHz Preferred Sync Reference: ADAT1 AutoSync Reference: ADAT1 AutoSync Frequency: 44100 System Clock Mode: Master System Clock Frequency: 44100 IEC958 input: Internal IEC958 output: Coaxial only IEC958 quality: Consumer IEC958 emphasis: off IEC958 NonAudio: off IEC958 sample rate: Error flag set ADAT1: Sync ADAT2: No Lock ADAT3: No Lock SPDIF: No Lock Word Clock: No Lock ADAT Sync: No Lock [EMAIL PROTECTED] mark]$ --- This SF.net email is sponsored by: Etnus, makers of TotalView, The debugger for complex code. Debugging C/C++ programs can leave you feeling lost and disoriented. TotalView can help you find your way. Available on major UNIX and Linux platforms. Try it free. www.etnus.com ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] HDSP 9652 - Some first results, but lockups and noactual audio
Thomas, Thanks for the response. I'm wondering if the whole problem could be that I've not run amixer at all? Is this required to actually route through the mixer of the HDSP 9652? In the previous driver the mixer was disabled, but if it is now enabled, then maybe I need to learn to use amixer and not alsamixer, which worker before? Mark On Sat, 2003-04-12 at 12:37, Thomas Charbonnel wrote: [ I'm not CC-ing anymore to Planetccrma as I am not a subscriber, and anyway Fernando's on alsa-devel ] Hi, Just a bit more info. By rebooting and doing a couple of power cycles on the external A/D/D/A I have been able to get the RME to acknowledge that it's seeing an ADAT sync at 44.1K, so I think at least the clock is being sent. What do you want to do here ? See the hdsp clock reported on you external D/A device or sync the card with the external device ? From your previous post I see that your external device is sending a 48kHz clock signal to the card ADAT1 input, but the card is ignoring it because the clock mode is set to internal 44.1kHz. If you want the card to slave to the external device, then you should : * set the external device to the proper rate. * switch the card to AutoSync mode : amixer cset numid=11 0 If you want the external device to be slaved : * switch the card to the appropriate rate : amixer cset numid=11 x (amixer cget numid=11 will give you the options for x) * switch the external device to slave mode. If the problem is as I think a mixer problem the clock signal should be received and interpreted correctly by the external device even if no audio signal is sent. This specific external unit (AI-3 for now) supports a loopback, where the ADAT in can be looped and sent back on its ADAT out. In doing this, I cannot see the audio signal I send it, implying that it is actually not receiving one. Also, I sent the wave file recorded using jackrec to my Windows box. (A copy of a Yellowjackets tune) It plays fine, so it appears that jack and alsaplayer are working OK. Thanks in advance again. Cheers, Mark On Sat, 2003-04-12 at 10:37, Mark Knecht wrote: [EMAIL PROTECTED] card0]# more /proc/asound/card0/hdsp RME HDSP 9652 (Card #1) Buffers: capture dde0 playback ddc0 IRQ: 10 Registers bus: 0xe880 VM: 0xe0967000 Control register: 0x1008098 Status register: 0x6086a48 Status2 register: 0x8061 FIFO status: 0 MIDI1 Output status: 0xff00 MIDI1 Input status: 0xff80 MIDI2 Output status: 0xff00 MIDI2 Input status: 0xff5f Buffer Size (Latency): 1024 samples (2 periods of 4096 bytes) Hardware pointer (frames): 1024 Passthru: no Line out: on Firmware version: 1 Sample Clock Source: Internal 44.1 kHz Preferred Sync Reference: ADAT1 AutoSync Reference: ADAT1 AutoSync Frequency: 48000 System Clock Mode: Master System Clock Frequency: 44100 IEC958 input: Internal IEC958 output: Coaxial only IEC958 quality: Consumer IEC958 emphasis: off IEC958 NonAudio: off IEC958 sample rate: Error flag set ADAT1: Lock ADAT2: No Lock ADAT3: No Lock SPDIF: No Lock Word Clock: No Lock ADAT Sync: No Lock [EMAIL PROTECTED] card0]# [EMAIL PROTECTED] mark]$ cat /proc/asound/card0/hdsp RME HDSP 9652 (Card #1) Buffers: capture de60 playback de40 IRQ: 10 Registers bus: 0xe880 VM: 0xe0967000 Control register: 0x1008090 Status register: 0x2040008 Status2 register: 0x8041 FIFO status: 0 MIDI1 Output status: 0xff00 MIDI1 Input status: 0xff3d MIDI2 Output status: 0xff00 MIDI2 Input status: 0xff39 Buffer Size (Latency): 64 samples (2 periods of 256 bytes) Hardware pointer (frames): 0 Passthru: no Line out: on Firmware version: 1 Sample Clock Source: Internal 44.1 kHz Preferred Sync Reference: ADAT1 AutoSync Reference: ADAT1 AutoSync Frequency: 44100 System Clock Mode: Master System Clock Frequency: 44100 IEC958 input: Internal IEC958 output: Coaxial only IEC958 quality: Consumer IEC958 emphasis: off IEC958 NonAudio: off IEC958 sample rate: Error flag set ADAT1: Sync ADAT2: No Lock ADAT3: No Lock SPDIF: No Lock Word Clock: No Lock ADAT Sync: No Lock [EMAIL PROTECTED] mark]$ --- This SF.net email is sponsored by: Etnus, makers of TotalView, The debugger for complex code. Debugging C/C++ programs can leave you feeling lost and disoriented. TotalView can help you find your way. Available on major UNIX and Linux platforms. Try it free. www.etnus.com ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
[Alsa-devel] hdspconf - cool!
Thomas, Thanks very much for creating this new little configuration app for the HDSP devices. Very nice. I have one question. If I set the Sample Clock Source to 'Auto Sync', then I'm seeing the AutoSync Ref. box going between ADAT1/48K and no setting at all. It's just bouncing back and forth. I do not remember seeing this card do anything like that under Windows. I've also just noticed that if I set the clock at 48K, and then move it to AutoSync, I'm seeing the card going back and forth between master and slave. Kind of strange. Great start though. Very nice to have a cool little config utility! Thanks! Cheers, Mark --- This SF.net email is sponsored by: Etnus, makers of TotalView, The debugger for complex code. Debugging C/C++ programs can leave you feeling lost and disoriented. TotalView can help you find your way. Available on major UNIX and Linux platforms. Try it free. www.etnus.com ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: RE: [Alsa-devel] Hammerfall DSP System Problems
/sbin/modprobe snd-rawmidi /sbin/modprobe snd-hammerfall-mem /sbin/modprobe snd-hwdep /sbin/modprobe snd-hdsp After I issue the last command I get the following errors: /lib/modules/2.4.19/kernel/sound/pci/rme9652/snd-hdsp.o: init_module: No such device Hint: insmod errors can be caused by incorrect module parameters, including invalid IO or IRQ parameters I am having the very same problems with my HDSP 9652. What's even stranger is, in my case, that for one evening the card worked, at least as far as loading the driver. (No sound, but the driver loaded.) I was able to look at /proc/asound/card0 (or whatever the directory/device was at the time...) and see sensible stuff. I even took some screen shots and sent email to another individual as I was debugging why I had no sound. Later that evening I powered down. Upon rebooting the next morning it failed, and has failed ever since. That was last Thursday evening. I too am using 0.9.4 as provided in the Gentoo emerge, along with a custom compiled kernel. Like you I think the HDSP driver understands my firmware revision. I have also downgraded and upgraded my firmware without success. This whole Alsa/HDSP thing is amazingly weird... - Mark --- This SF.net email is sponsored by: Etnus, makers of TotalView, The best thread debugger on the planet. Designed with thread debugging features you've never dreamed of, try TotalView 6 free at www.etnus.com. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] Hammerfall DSP System Problems
Later that evening I powered down. Upon rebooting the next morning it failed, and has failed ever since. That was last Thursday evening. the same thing happened to me a week ago. i have, as usual, not had time to investigate further. Well, it is actually quite comforting to know that folks with far more experience than I sometimes see the same strange behaviors! This whole Alsa/HDSP thing is amazingly weird... at the moment, i regret that we made the move to user-space loading of the firmware when we did. i think now that it was a mistake to do it at that time. it has caused a fundamental change in the way the driver sets up, but it came at a time when we weren't confident that setup was working for various cases. I'm sure that it will get worked out. I think Joshua is currently having some good successes with the HDSP 9652, and I'm currently learning how to modify a Gentoo ebuild to accomplish doing a patch. I also have (after more pain than I wish to recount) Windows on this Gentoo box in a dual boot config, and have the RME firmware tools to upgrade and downgrade, so I think I'm almost set to be able to try this stuff out again. I'm sure over the next few weeks and months this will get straightened out. :-) Mark --- This SF.net email is sponsored by: Etnus, makers of TotalView, The best thread debugger on the planet. Designed with thread debugging features you've never dreamed of, try TotalView 6 free at www.etnus.com. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] Hammerfall DSP System Problems
so I think I'm almost set to be able to try this stuff out again. For what it's worth, i had problems with alsa 0.9.4 SMP. i think there are some spinlock changes that aren't very well debugged yet. On the other hand, alsa 0.9rc7 seems fairly stable. i've gotten some lockups, but i've also used jack for 4-5 hours without any problem. Well, I found last night that if I removed Alsa and then rebuilt it, that as long as I didn't power down 0.9.4-rc1 stayed up and I could see my card even after a warm boot, but again, after a cold boot Alsa no longer recognized the card. So, my new strategy is that every morning, as I brew my tea, I now do: emerge -C alsa-driver alsa-utils alsa-tools ACCEPT_KEYWORDS=~x86 emerge alsa-driver alsa-utils alsa-tools and then warm boot to get the driver to find the card. I still don't get sound, but at least my CPU knows the card is there. I guess life is good when you finally know how to cope with it! ;-) Mark --- This SF.net email is sponsored by: Etnus, makers of TotalView, The best thread debugger on the planet. Designed with thread debugging features you've never dreamed of, try TotalView 6 free at www.etnus.com. ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] alsa as rpm or binary
PlanetCCRMA would be my best guess -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Ralf Haller Sent: Thursday, June 12, 2003 11:09 AM To: [EMAIL PROTECTED] Subject: [Alsa-devel] alsa as rpm or binary Hi, I want to install alsa on an computer running RedHat 9 from rpm or binary. There is no development environment installed on the computer since it only has a 1.2 GB harddrive with about 100 MB free space. Where can I get an appropriate rpm of binary? Ralf --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] hdsp patch
Paul, Hi. Thanks for sendign these along. First steps look good. I've managed to patch things and it does install. QUESTION - I am not explicitly loading snd-hammerfall-mem, but it is getting loaded when I start Alsa. Is this correct? I should _not_ explicitly load that file anywhere, but just let the driver load it? Also, I seem to have had some problems with the second patch you sent. It didn't apply. I'm sure it's just me. IT was so simple so I just typed it into the file. Were these sent as attachments, or as text within the email? Hopefully I'll get some sound later today. Currently nothing is hooked up to the card, so that will take a while to switch over as everything is quite buried right now. Thanks, Mark Wizard alsa-driver # lsmod Module Size Used byNot tainted snd-hdsp 32556 0 snd-rawmidi15040 0 [snd-hdsp] snd-seq-device 4352 0 [snd-rawmidi] snd-pcm64928 0 [snd-hdsp] snd-timer 15876 0 [snd-pcm] snd-page-alloc 5404 0 [snd-pcm] snd-hwdep 5216 0 [snd-hdsp] snd32836 0 [snd-hdsp snd-rawmidi snd-seq-device snd-pcm snd-timer snd-hwdep] snd-hammerfall-mem 1920 0 [snd-hdsp] radeon107972 1 agpgart11920 3 (autoclean) ide-cd 27080 0 (autoclean) cdrom 25984 0 (autoclean) [ide-cd] Wizard alsa-driver # Wizard card0 # more hdsp RME HDSP 9652 (Card #1) Buffers: capture dee0 playback df00 IRQ: 17 Registers bus: 0x2880 VM: 0xe08e6000 Control register: 0x100805e Status register: 0x280 Status2 register: 0x8701 FIFO status: 0 MIDI1 Output status: 0xff00 MIDI1 Input status: 0xff3a MIDI2 Output status: 0xff00 MIDI2 Input status: 0xff39 Buffer Size (Latency): 8192 samples (2 periods of 32768 bytes) Hardware pointer (frames): 0 Passthru: no Line out: on Firmware version: 1 Sample Clock Source: Internal 32 kHz Preferred Sync Reference: ADAT1 AutoSync Reference: None AutoSync Frequency: 0 System Clock Mode: Master System Clock Frequency: 32000 IEC958 input: Internal IEC958 output: Coaxial only IEC958 quality: Consumer IEC958 emphasis: off IEC958 NonAudio: off IEC958 sample rate: Error flag set ADAT1: No Lock ADAT2: No Lock ADAT3: No Lock SPDIF: No Lock Word Clock: No Lock ADAT Sync: No Lock Wizard card0 # On Fri, 2003-06-13 at 21:55, Paul Davis wrote: this patch fixes some basic problems with the hdsp driver with respect to the hdsp9652 card. it also cleans up a few minor issues with naming in the driver, and slightly rationalizes initialization to involve the minimum of special-casing for the hdsp9652. --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] hdsp patch
Paul,
I've been trying for the last day or so to get some sound out of the
card. Still no luck. The setup does work fine when I boot into Windows.
I've certainly had a few problems on this end, like getting
/etc/asound.state into a funny configuration that had both the on-board
Via chipset and the HDSP 9652 in it. That's fixed, but still no sound.
I'm running as root. I've tried both Jack and straight Alsa with
aplay and alsaplayer. Everything acts like I should be getting sound,
but I don't. The Alsa drivers appear to be loaded. Restarting Alsa looks
pretty normal.
alsamixer says everything is turned up to 30. 'M' doesn't seem to
mute or unmute and channels for this card.
Can you clarify - do I need to make any 'connections' through the
HDSP 9652 to get the alsa_pcm:playback_1/2 to be enabled and supplying
audio to my amp? If so, what commands are you using?
I'm attaching asound.state, .asoundrc and a little more info. Let me
know what else you want to see.
Thanks for any pointers you can provide.
Cheers,
Mark
Wizard root # lsmod
Module Size Used byNot tainted
snd-hdsp 32556 3
snd-rawmidi15040 0 [snd-hdsp]
snd-seq-device 4352 0 [snd-rawmidi]
snd-pcm64928 2 [snd-hdsp]
snd-timer 15876 0 [snd-pcm]
snd-hwdep 5216 0 [snd-hdsp]
snd32836 1 [snd-hdsp snd-rawmidi snd-seq-device
snd-pcm snd-timer snd-hwdep]
radeon107972 1
agpgart11920 3 (autoclean)
ide-cd 27080 0 (autoclean)
cdrom 25984 0 (autoclean) [ide-cd]
snd-page-alloc 5404 0 [snd-pcm]
snd-hammerfall-mem 1920 0 [snd-hdsp]
Wizard root #
Wizard root # cat /proc/asound/card0/hdsp
RME HDSP 9652 (Card #1)
Buffers: capture df00 playback dee0
IRQ: 17 Registers bus: 0xe880 VM: 0xe08e6000
Control register: 0x10080b3
Status register: 0x2043088
Status2 register: 0x8041
FIFO status: 0
MIDI1 Output status: 0xff00
MIDI1 Input status: 0xff5e
MIDI2 Output status: 0xff00
MIDI2 Input status: 0xff4b
Buffer Size (Latency): 128 samples (2 periods of 512 bytes)
Hardware pointer (frames): 0
Passthru: no
Line out: on
Firmware version: 1
Sample Clock Source: Internal 44.1 kHz
Preferred Sync Reference: ADAT1
AutoSync Reference: ADAT1
AutoSync Frequency: 44100
System Clock Mode: Master
System Clock Frequency: 44100
IEC958 input: Internal
IEC958 output: Coaxial only
IEC958 quality: Consumer
IEC958 emphasis: off
IEC958 NonAudio: off
IEC958 sample rate: Error flag set
ADAT1: Sync
ADAT2: No Lock
ADAT3: No Lock
SPDIF: No Lock
Word Clock: No Lock
ADAT Sync: No Lock
Wizard root #
On Fri, 2003-06-13 at 21:55, Paul Davis wrote:
this patch fixes some basic problems with the hdsp driver with respect
to the hdsp9652 card. it also cleans up a few minor issues with naming
in the driver, and slightly rationalizes initialization to involve
the minimum of special-casing for the hdsp9652.
the basic problem with the hdsp9652 was related to 8 bit versus 32 bit
offsets when addressing the mixer memory. once this was fixed,
everything worked. this driver continues to work fine on my
pci+digiface unit as well.
my apologies for this taking so long - it has taken a long time to ask
RME the right question, and quite a long time to get the
answer. once i got down to it, the fix took 5 minutes!
now we just need to solve the multiface initialization problems :(
--p
state.'' {
control.1 {
comment.access 'read write'
comment.type IEC958
iface PCM
name 'IEC958 Playback Default'
value
''
}
control.2 {
comment.access 'read write inactive'
comment.type IEC958
iface PCM
name 'IEC958 Playback PCM Stream'
value
''
}
control.3 {
comment.access read
comment.type IEC958
iface MIXER
name 'IEC958 Playback Con Mask'
value
Re: [Alsa-devel] hdsp patch
On Sun, 2003-06-15 at 19:42, Paul Davis wrote: RME HDSP 9652 (Card #1) Buffers: capture df00 playback dee0 IRQ: 17 Registers bus: 0xe880 VM: 0xe08e6000 Control register: 0x10080b3 You don't have the correct version of the driver. It would print: RME Hammerfall HDSP 9652 (Card #1) Buffers: capture f700 playback f6e0 IRQ: 11 Registers bus: 0xfebb VM: 0xf88af000 Control register: 0x10080f9 Control2 register: 0x800 notice the extra Control2 register printout. Cool. Something to look for anyway 15 minutes later. Bingo! OK, so the new driver hadn't gotten moved to the right place. It seems to be there now. I'm getting sound, but it's full volume and I don't seem to be able to turn it down. (And I only have a few more minutes before my kid goes to sleep. Or tries to...) ;-) I have alsamixer up and running, and the volumes turned down to 6 and it's still screaming loud. Is this like the old driver where the mixer didn't work at all? Or have I not set the right things? The other thing I notice is I only seem to be able to set 24 values in my little script to set volumes. The driver I just replaced allowed me to set all 26. OK, so a lot of progress, but I need to be able to reduce the volume badly!!! What can I send you to see if it's my problem? Also, please answer - do I need to set routing paths through the HDSP FPGA to get the mixer working? Can you supply a script to do that if it's necessary? Thanks, Mark --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] hdsp patch
On Sun, 2003-06-15 at 20:26, Mark Knecht wrote: Cool. Something to look for anyway 15 minutes later. Bingo! OK, so the new driver hadn't gotten moved to the right place. It seems to be there now. I'm getting sound, but it's full volume and I don't seem to be able to turn it down. (And I only have a few more minutes before my kid goes to sleep. Or tries to...) ;-) I have alsamixer up and running, and the volumes turned down to 6 and it's still screaming loud. Is this like the old driver where the mixer didn't work at all? Or have I not set the right things? The other thing I notice is I only seem to be able to set 24 values in my little script to set volumes. The driver I just replaced allowed me to set all 26. OK, so a lot of progress, but I need to be able to reduce the volume badly!!! What can I send you to see if it's my problem? Also, please answer - do I need to set routing paths through the HDSP FPGA to get the mixer working? Can you supply a script to do that if it's necessary? Thanks, Mark BTW: Wizard rme9652 # cat /proc/asound/card0/hdsp RME Hammerfall HDSP 9652 (Card #1) Buffers: capture de40 playback de20 IRQ: 17 Registers bus: 0xe880 VM: 0xe4cf7000 Control register: 0x10080de Control2 register: 0x800 Status register: 0x2040008 Status2 register: 0x8061 FIFO status: 0 MIDI1 Output status: 0xff00 MIDI1 Input status: 0xff00 MIDI2 Output status: 0xff00 MIDI2 Input status: 0xff00 Buffer Size (Latency): 8192 samples (2 periods of 32768 bytes) Hardware pointer (frames): 0 Passthru: no Line out: on Firmware version: 1 Sample Clock Source: Internal 48 kHz Preferred Sync Reference: ADAT1 AutoSync Reference: ADAT1 AutoSync Frequency: 48000 System Clock Mode: Master System Clock Frequency: 48000 IEC958 input: Internal IEC958 output: Coaxial only IEC958 quality: Consumer IEC958 emphasis: off IEC958 NonAudio: off IEC958 sample rate: Error flag set ADAT1: Sync ADAT2: No Lock ADAT3: No Lock SPDIF: No Lock Word Clock: No Lock ADAT Sync: No Lock Wizard rme9652 # --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] hdsp patch
On Sun, 2003-06-15 at 20:42, Paul Davis wrote:
I have alsamixer up and running, and the volumes turned down to 6 and
it's still screaming loud. Is this like the old driver where the mixer
didn't work at all? Or have I not set the right things?
no, the mixer works, but unfortunately it appears that i didn't test
enough. the code still appears to be not quite right - there's a
tricky detail that you can only write 32 bit values to the mixer, but
each mixer element is 16 bits, so you always have to write the one you
want to modify, plus its neighbour. looks like i don't have that
quite right yet.
OK, well I'll stay tuned for possible patches and testing whatever you
need. Thanks.
If it makes any difference in simplifying your initial testing, I have
my speakers hooked to the HDSP9652 ADAT1 port, using channels 1 2. I'm
not using any other channels in that group, or any other ADAT outputs as
of yet.
If you think some other outputs do work (in terms of modifying the
volumes, let me know and I'll switch to them.
I don't suppose there is any .asoundrc magic that could reduce the
volume automagically in Alsa itself instead of in the driver? (A JAck
volume control?) ;-)
The other thing I notice is I only seem to be able to set 24 values in
my little script to set volumes. The driver I just replaced allowed me
to set all 26.
i'll check on that. probably just a mistake on my part about the
number of channels (i tend to forget the s/pdif outs).
I figured as much. No problem right now.
Also, please answer - do I need to set routing paths through the HDSP
FPGA to get the mixer working?
no. the mixer is (dis|en)abled by flipping a control register bit. if
its on, its on. what needs work is the hdsp_write_gain() function.
--p
OK, I was just remembering Roger Williams telling my some stuff back in
January when I was first trying to get this working (when we didn't know
that the volume controls didn't work) about needing to break
connections. He sent a little script file for the Digiface, but I was
not sure if this was required, or just something he kept around for test
purposes:
#!/bin/bash
ADAT1=0 1 2 3 4 5 6 7
ADAT2=8 9 10 11 12 13 14 15
ADAT3=16 17 18 19 20 21 22 23
SPDIF=24 25
function disconnect () {
for output in $@; do
input=0
while [ $input -le 25 ]; do
echo -n .
amixer cset numid=5 $input,$output,0 /dev/null
input=$((input+1))
done
done
echo
}
echo -n Disconnecting ADAT1 outputs
disconnect $ADAT1
echo -n Disconnecting ADAT2 outputs
disconnect $ADAT2
echo -n Disconnecting ADAT3 outputs
disconnect $ADAT3
echo -n Disconnecting SPDIF output
disconnect $SPDIF
Thanks,
Mark
---
This SF.NET email is sponsored by: eBay
Great deals on office technology -- on eBay now! Click here:
http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5
___
Alsa-devel mailing list
[EMAIL PROTECTED]
https://lists.sourceforge.net/lists/listinfo/alsa-devel
Re: [Alsa-devel] hdsp patch
On Sun, 2003-06-15 at 20:56, Paul Davis wrote: OK, I was just remembering Roger Williams telling my some stuff back in January when I was first trying to get this working (when we didn't know that the volume controls didn't work) about needing to break connections. He sent a little script file for the Digiface, but I was not sure if this was required, or just something he kept around for test purposes: no, this doesn't do anything except write values to the mixer, and its the function inside the driver that does this which is broken. Thanks for the explanation. I'll watch the list for updates. Cheers, Mark --- This SF.NET email is sponsored by: eBay Great deals on office technology -- on eBay now! Click here: http://adfarm.mediaplex.com/ad/ck/711-11697-6916-5 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] hdsp and midi problems
Marcus, The HDSP 9652 MIDI interface has (for me under Linux) always dropped lots of notes. Mostly I find it interesting that it drops note off information and doesn't seem to drop note-on. I cannot be sure how well it handles controller information. do you mean dropping notes in reading MIDI? well, IIRC, you reported also that the behavior of MIDI out is strange... oh sigh. Yes, but I tested Alsa (a long time ago) with both MIDI-IN driving soft synths, and then MIDI-IN/MIDI-OUT driving hardware synths and got the same 'note stuck on' problem. For that reason I say it's a 'MIDI-IN dropped note' problem, but certainly I do not know this for sure. dropping a note is surely a bug of the driver. unless the FIFO overflows, no dropping should happen. btw, could you try the attached patch? (and in-advance-scheduled excuse if it causes a hang-up :) I would be happy to try a bit of patching. However, I am on Redhat/PlanetCCRMA and Gentoo, both of which provide prebuilt Alsa environments. I can try patching the Gentoo one which is pretty recent, but it may take a few days before I can get back to you. --- This SF.Net email is sponsored by: INetU Attention Web Developers Consultants: Become An INetU Hosting Partner. Refer Dedicated Servers. We Manage Them. You Get 10% Monthly Commission! INetU Dedicated Managed Hosting http://www.inetu.net/partner/index.php ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] New mixer api.
Hi, If a user has a 5.1 sound card, but only has 2 speakers connected, it would be nice if the user could tell alsa this fact. One of my SACD players has this feature built in. It's nice in that I can listen to a 5.1 SACD through stereo headphones and hear all 6 of the channels mixed in a more or less normal way. I know this isn't exactly what you are proposing, but I think it's similar. One of these days you are going to have an audio DVD in your PC, and maybe be using a stereo sound card, and you'll run into the same issue. Cheers, Mark --- This SF.Net email is sponsored by: INetU Attention Web Developers Consultants: Become An INetU Hosting Partner. Refer Dedicated Servers. We Manage Them. You Get 10% Monthly Commission! INetU Dedicated Managed Hosting http://www.inetu.net/partner/index.php ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] hdsp driver status
Thomas, Great to have you back! Looking forward to the matrix mixer actually working one of these days. I hope you can work it into your design somewhere. Thanks, Mark -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] Behalf Of Thomas Charbonnel Sent: Thursday, July 03, 2003 4:13 AM To: [EMAIL PROTECTED] Subject: [Alsa-devel] hdsp driver status Hi, I've been off the net and busy for a long time. I apologize to anyone whose mail I didn't answer during this period. I'm slowly catching up with the huge activity that has been going on this list, focusing on the hdsp driver issues. I'd like to summarize the driver status : Fixed : * Marcus, Jaroslav : thanks for having solved the long (ever ? :) lasting hdsp midi bug ! Done: * Takashi : I succesfully tested your memory preallocation patch (only the hdsp part of it, actually), thanks for it. (Thanks too for the addition of hdspconf to cvs, it is indeed the latest version). Remaining: * HDSP9652: the matrix mixer isn't yet handled right. * Rev11 + Multiface: either the firmware file is wrong or the io-box firmware upload scenario has changed for this card revision. Has anyone ever had success with this combination (or rev11 + Digiface) ? The attached patch : * Reenables the hdsp_check_for_iobox function Paul bypassed in his previous patch. This function is needed for mobile setups. It handles the cases where the iobox has been unplugged/replugged or lost power supply. * Fixes the channel limit Mark has reported on the HDSP9652 (24 instead of 26). * Fixes a locking issue reported by Joshua N Pritikin. David, Jesse and Patrick, I'm back at work on the totalmix clone. It should be out for testing really soon. Sorry for having been so unresponsive lately. Thomas --- This SF.Net email sponsored by: Free pre-built ASP.NET sites including Data Reports, E-commerce, Portals, and Forums are available now. Download today and enter to win an XBOX or Visual Studio .NET. http://aspnet.click-url.com/go/psa0016ave/direct;at.asp_061203_01/01 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] hdsp driver status
The documentation is the driver code. Paul is in contact with people from RME. As of today I'm not, but I'll try to contact them. I'd be as happy as you to see the driver working smoothly in all possible configurations. Thomas Thomas, We RME users are very happy to have you on our side! Can you give some insight (if you have enough info to comment!) into how much of the driver for the HDSP/DF/MF is also used for the HDSP 9652? I'm sort of concerned that after 6 months of no mixer on the HDSP 9652, the DF/MF mixer will get fixed, and then I'll find out it has nothing to do the HDSP 9652 and be let down again. Is it one common design, or is the mixer different? Thanks, Mark --- This SF.Net email sponsored by: Free pre-built ASP.NET sites including Data Reports, E-commerce, Portals, and Forums are available now. Download today and enter to win an XBOX or Visual Studio .NET. http://aspnet.click-url.com/go/psa0016ave/direct;at.asp_061203_01/01 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] hdsp driver status
On Thu, 2003-07-03 at 18:20, Thomas Charbonnel wrote: I'm sort of concerned that after 6 months of no mixer on the HDSP 9652, the DF/MF mixer will get fixed, and then I'll find out it has nothing to do the HDSP 9652 and be let down again. The MF/DF mixer works, but firmware upload to the iobox fails with some card revision. On the HDSP 9652 no firmware has to be uploaded (hdsploader is unecessary), but the mixer is broken. Is it one common design, or is the mixer different? It is one common design BUT the mixer is different... Thomas Thanks for the description. I've written the next paragraph 5 times, but just don't want to do this all over again. Anyone wanna buy an HDSP 9652 cheap? Bye, Mark --- This SF.Net email sponsored by: Free pre-built ASP.NET sites including Data Reports, E-commerce, Portals, and Forums are available now. Download today and enter to win an XBOX or Visual Studio .NET. http://aspnet.click-url.com/go/psa0016ave/direct;at.asp_061203_01/01 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] Trivial hdsp.c patch to support rev 50 cards
This adds rev 50 to the list of supported cards. i will have a slightly longer patch that includes this, later today. also included will be a no_mixer option for the hdsp9652. --p Paul, Is this envisioned an option in modules.conf, or a compile time option? Also, while I have your attention, can you clarify whether the RME Hammerfall 9652 and 9636 have mixers? I made one of my 9636 machines into dual boot last evening and then found alsamixer didn't work. I assume that means the 9636 has no mixer, or is this a bug I should be reporting somewhere? Thanks, Mark --- This SF.Net email sponsored by: Free pre-built ASP.NET sites including Data Reports, E-commerce, Portals, and Forums are available now. Download today and enter to win an XBOX or Visual Studio .NET. http://aspnet.click-url.com/go/psa0013ave/direct;at.aspnet_072303_01/01 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] Trivial hdsp.c patch to support rev 50 cards
Is this envisioned an option in modules.conf, or a compile time option? modules.conf. Also, while I have your attention, can you clarify whether the RME Hammerfall 9652 and 9636 have mixers? I made one of my 9636 machines into dual boot last evening and then found alsamixer didn't work. I assume that means the 9636 has no mixer, or is this a bug I should be reporting somewhere? the digi9652 and digi9636 have no mixer. --p Thanks for the info. So then the only way to set volume levels on a DIGI9652 or DIGI9636 when using Alsa is either in the application, or using some form of a script like: #!/bin/bash for i in $(seq 1 26);do amixer -c 0 sset 'Chn',$i 85% done Thanks, Mark --- This SF.Net email sponsored by: Free pre-built ASP.NET sites including Data Reports, E-commerce, Portals, and Forums are available now. Download today and enter to win an XBOX or Visual Studio .NET. http://aspnet.click-url.com/go/psa0013ave/direct;at.aspnet_072303_01/01 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
RE: [Alsa-devel] Trivial hdsp.c patch to support rev 50 cards
haven't tried recording anything to see if the same problem exists there. it's tough to test these things as i'm guaranteed a lock-up, ie: my drives have to be fsck'd each time. does anyone know of a way to safely debug this? can i catch the playback thread before it pees on the floor, NO! Geez hdsp, stream goes out the urinal! ...hold that thing, damnit!!! ? dan. The only safe way to debug this is to let someone else do it. ;-) Use ext3/reiserfs and at least the time delays won't be as bad. Do not have any other applications (email, etc.) open at the same time to protect those databases. Have a big, cold beer handy. (Sorry the mono thing didn't work. I can play Penguin-Command with lots of sound and my HDSP 9652 driver works fine. Attempt alsaplayer for 1 second and I lose all sound. Go figure...) ;-) --- This SF.Net email sponsored by: Free pre-built ASP.NET sites including Data Reports, E-commerce, Portals, and Forums are available now. Download today and enter to win an XBOX or Visual Studio .NET. http://aspnet.click-url.com/go/psa0013ave/direct;at.aspnet_072303_01/01 ___ Alsa-devel mailing list [EMAIL PROTECTED] https://lists.sourceforge.net/lists/listinfo/alsa-devel
