Re: [Alsa-devel] USB audio kernel panic

2004-05-12 Thread Patrick Shirkey
Clemens Ladisch wrote:
Patrick Shirkey wrote:

Using the latest usb-audio code from cvs I get a kernel panic when
loading the drivers.
I'm running kernel-2.2.21 with gcc-2.95.4


Works fine with my 2.2.19(?).

ALSA 1.0.4 fixed some horrible bugs in the USB compatibility code for
2.2.x kernels.
Where exactly does this crash happen?

I just updated to the latest sf.net cvs. hdspmixer is working again but 
still get a kernel error with usb-audio.

This is the error message and the corresponding section of /proc/ksyms. 
(The actual symbol doesn't appear)

---

Unable to handle kernel paging request at virtual address 28302e63
 printing eip:
e08eafc0
*pde = 
Oops: 
CPU:0
EIP:0010:[e08eafc0]Tainted: PF


.
.
.
e08e7000 
__insmod_snd-usb-audio_O/lib/modules/2.4.21/kernel/sound/usb/snd-usb-au
dio.o_M40A2C2B5_V132117 [snd-usb-audio]
e08e7060 __insmod_snd-usb-audio_S.text_L29104   [snd-usb-audio]
e08eb02c snd_hack_usb_set_interface [snd-usb-audio]
e08ed8a8 snd_usbmidi_disconnect [snd-usb-audio]
e08edff4 snd_usb_create_midi_interface  [snd-usb-audio]
e08ee220 __insmod_snd-usb-audio_S.rodata_L7690  [snd-usb-audio]
e08f01e0 __insmod_snd-usb-audio_S.data_L4548[snd-usb-audio]
e08f13c0 __insmod_snd-usb-audio_S.bss_L64   [snd-usb-audio]
(END)


--
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Apparently upon the beginning of the barrage, the donkey broke 
discipline and panicked, toppling the cart. At that point, the rockets 
disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
the arrival of U.S. troops.

United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM


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[Alsa-devel] USB audio kernel panic

2004-05-08 Thread Patrick Shirkey
Using the latest usb-audio code from cvs I get a kernel panic when 
loading the drivers. After that I cannot unload the modules and have to 
reboot if I want to try again.

This doesn't happen with the hdsp driver or cmipci driver which I also use.

I'm running kernel-2.2.21 with gcc-2.95.4

--
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Apparently upon the beginning of the barrage, the donkey broke 
discipline and panicked, toppling the cart. At that point, the rockets 
disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
the arrival of U.S. troops.

United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM


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Re: [Alsa-devel] Frontend Development

2004-04-18 Thread Patrick Shirkey
Chris wrote:
ehh.. that had nothing to do with my mail... its for a project of my own...

You asked for code and urls to help you develop a new gui for alsaplayer 
so I sent you a link that has a lot of useful information about alsaplayer.

I wonder if you meant to ask for info on developing a frontend for 
aplay/arecord?

If not check the alsaplayer cvs to get some ideas and tips.

On Sunday 18 April 2004 12:43 am, Patrick Shirkey wrote:

Chris wrote:

by front-end i mean GUI in XFree. Im open to whatever toolkits, i just
need a starting point. :)
There are a few guis for alsaplayer already. Check the cvs
http://www.alsaplayer.org



--
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Apparently upon the beginning of the barrage, the donkey broke 
discipline and panicked, toppling the cart. At that point, the rockets 
disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
the arrival of U.S. troops.

United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM


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Re: [Alsa-devel] Frontend Development

2004-04-17 Thread Patrick Shirkey
Chris wrote:
by front-end i mean GUI in XFree. Im open to whatever toolkits, i just need a 
starting point. :)

There are a few guis for alsaplayer already. Check the cvs 
http://www.alsaplayer.org

--
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Apparently upon the beginning of the barrage, the donkey broke 
discipline and panicked, toppling the cart. At that point, the rockets 
disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
the arrival of U.S. troops.

United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM


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Re: [Alsa-devel] USB audio devices

2004-03-10 Thread Patrick Shirkey
Karsten Wiese wrote:
We can also vary the exact USB frame time.
With UHCI 1.1 USB Hosts there is the SOF Register. 
It is setable from 0 to 255, the default being 127.
Using this SOF-Register, we can set the actual USB Frame Rate from 
((12000 - 127) / 12000)ms to ((12000 + 128) / 12000)ms. That is about -/+ 1%:
More than enough to adjust the USB-Frame rate to let us get 44(44100) or 48
(48000) Sample Frames everyy USB Frame.
It really works here already with the us428: The trick is:
We first make the USB-Frame longer until we capture 1 Sample Frame more  45 
(for 44100). then the USB-Frame is shortened until we capture only 43 Sample 
Frames for one USB Frame. and so on. 
If we use a period size of multiples of 44 (for 44100) then :
- Latency is almost at its best (only 1/44 less than optimal.
- User Prog Scheduling Jitter is minimized.
- Alsas keeps constant Period size.

Hallaluhjah

Please please please make this standard for the usb audio driver.



--
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Http://www.djcj.org/LAU/guide/ - The Linux Audio Users guide
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Apparently upon the beginning of the barrage, the donkey broke 
discipline and panicked, toppling the cart. At that point, the rockets 
disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
the arrival of U.S. troops.

United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM


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Re: [Alsa-devel] Soundcard matrix: removing some M-Audio USB devices

2004-03-01 Thread Patrick Shirkey
James Courtier-Dutton wrote:
Frank Barknecht wrote:

Hallo,
James Courtier-Dutton hat gesagt: // James Courtier-Dutton wrote:

You will be lucky to find any sound card complying with the USB Audio 
spec, as the spec is written so badly.


This is the USB spec I'm referring, not the USB AUDIO spec. Those
M-Audio devices aren't even recognized as being USB devices by the
kernel.
ciao


So lsusb gives no output for the device?
I have seen some usb devices that connect, and then for some reason the 
kernel does not like them, and disconnects them before you have a chance 
to see them in lsusb.

Which part of the USB spec do they break ?

They have inappropriate altsettings. There is a patch which Sampo 
Savolainen just released but it probably won't be accepted to the Kernel.

I'll make a mention of it.

--
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Apparently upon the beginning of the barrage, the donkey broke 
discipline and panicked, toppling the cart. At that point, the rockets 
disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
the arrival of U.S. troops.

United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM


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[Alsa-devel] Fwd: [ALSA: doc notes additions]

2004-02-15 Thread Patrick Shirkey
The following has been submitted as a note for the documentation of 
intel8x0.

acharnley [EMAIL PROTECTED]/abr
Sunday, 15 February 2004/td
Support for nForce 2 chipset (Shuttle, Slackware 9.1) appears broke
with xine 0.96 after ALSA 0.9.6. XMMS et all still working fine.
Any changes to .asound as mentioned above cause any audio app to lock up.
I have had to remove .asound and use defaults.
I am _desperately_ trying to convert standard 2 channel audio to
Dolby Digital stream in XMMS and pass through instead of passing
2 channel through IEC958 and having to use ESP effect on my
external AMP.
I see ALSAMIXER has duplication options and send analog to IEC958,
which would be perfect, but none of those controls work. :(






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Re: [Alsa-devel] Soundcard Matrix guidelines ?

2004-01-26 Thread Patrick Shirkey
Giuliano Pochini wrote:
Who is the maintainer of the soundcard matrix ? What is the preferred
format for sending him updates ?
I am supposed to be that person :)

For now you can make additions to the relevant page and send me the 
revision.

Thanks.



--
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Apparently upon the beginning of the barrage, the donkey broke 
discipline and panicked, toppling the cart. At that point, the rockets 
disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
the arrival of U.S. troops.

United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM


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Re: [Alsa-devel] Documentation Bugs

2004-01-25 Thread Patrick Shirkey
Torrey Hoffman wrote:
I've been working through the .asoundrc documentation and have found
several errors.
Thanks for the detailed report.

I have fixed the obvious errors but I'm not sure what the problem is 
with the rate_convert example. I have left it as is for now.



--
3. The rate conversion plug documentation (or code?) is broken.
I'm not sure if this is a documentation bug or a driver/alsa-lib bug. 

My .asoundrc at this point closely follows the documentation (with
corrections as noted in bug#2) and looks like:
pcm.ac97 {
type hw
card 0
device 0
}
pcm_slave.sl2 {
pcm ac97
rate 44100
}
pcm.rate_convert {
type rate
slave sl2
}
The documentation says:  
Now you can call this newly created virtual device by: aplay -D
rate_convert test.wav

But when I try that, I get this error:

Playing WAVE 'test.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, Stereo
aplay: set_params:805: Broken configuration for this PCM: no configurations available
Note that aplay -D ac97 test.wav works, but plays too fast (at 48000),
and gives the warning please, try the plug plugin (-Dplug:ac97). If I
do that, it works perfectly.
So I think the documentation is wrong, or at least incomplete, or rate
conversion does not work.


--
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Http://www.djcj.org - The Linux Audio Users guide

 Apparently upon the beginning of the barrage, the donkey broke 
discipline and panicked, toppling the cart. At that point, the rockets 
disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
the arrival of U.S. troops.

United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM


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Re: [Alsa-devel] The LAD History

2004-01-16 Thread Patrick Shirkey
holborn wrote:
Sorry i'm new :-) from 2003  today is my first visit to linuxaudio.org 
home page, i never know about their existence. Then  can someone explain 
here in a few words a little history about when/how LAD and linuxaudio was 
born and their dependencies?

Sorry this kind of discussion is to mutch for my poor English, i need hours to 
read it, but i try ...  :-)

There is much information available on the linux audio developers website.

www.linuxdj.org

the info for linuxaudio.org can be found on the lad archives and the 
linuxadudio.org website.

--
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 Apparently upon the beginning of the barrage, the donkey broke 
discipline and panicked, toppling the cart. At that point, the rockets 
disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
the arrival of U.S. troops.

United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM


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Re: [Alsa-devel] hdsp vs 16bit vs dmix

2004-01-13 Thread Patrick Shirkey
Justin Cormack wrote:


I think it may be something else in the hardware setup, as it works fine
for me. Are the disks running dma? Is there some interrupt oddness? What
kernel is it? What kind of machine? What filesystem? The fact that it
works with JACK suggests that it is a small delay that aplay doesnt cope
with well.
Thanks. After this and Takashi's message I have ben convinced to look 
into the kernel and latency stuffs again.



--
Patrick Shirkey - Boost Hardware Ltd.
Http://www.boosthardware.com
Http://www.djcj.org - The Linux Audio Users guide

 Apparently upon the beginning of the barrage, the donkey broke 
discipline and panicked, toppling the cart. At that point, the rockets 
disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
the arrival of U.S. troops.

United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM


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Re: [Alsa-devel] hdsp vs 16bit vs dmix

2004-01-08 Thread Patrick Shirkey
Takashi Iwai wrote:
At Thu, 08 Jan 2004 02:29:13 +0900,
Patrick Shirkey wrote:
I am seeing strange things happen while using the hdsp and dmix plugin 
and aplay as a test app.

JACK works perfectly.

I can get sound from aplay if I use this commandline:

aplay -f cd -D plughw audio.wav
Playing WAVE 'audio.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, 
Stereo
underrun!!! (at least 40.147 ms long)
underrun!!! (at least 86.223 ms long)
underrun!!! (at least 145.116 ms long)
underrun!!! (at least 79.109 ms long)
underrun!!! (at least 248.555 ms long)
underrun!!! (at least 180.442 ms long)
underrun!!! (at least 96.117 ms long)
Aborted by signal Interrupt...	

But I get an xrun every second or so which makes the card useless for 
native alsa apps .


i'm not sure which plugin plays wrong yet.

run aplay with -v option and see which plugins and what configurations
are used.  for example, if the native hardware requires 32bit samples,
prepare 32bit samples WAV instead of 16bit, and reduce the plugin.
i don't think the rate conversion is used, so the plugins are linear,
route, and copy plugins.
my rough guess is the copy plugin.  try aplay -M option, so that mmap
mode is used.
IMO the test case should be a 16bit file because this is the CD standard.

---
 aplay -v -M -f cd -D plughw audio.wav
Playing WAVE 'audio.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, 
Stereo
Plug PCM: Route conversion PCM (sformat=S32_LE)
Transformation table:
0 - 0
1 - 1
2 - none
3 - none
4 - none
5 - none
6 - none
7 - none
8 - none
9 - none
10 - none
11 - none
12 - none
13 - none
14 - none
15 - none
16 - none
17 - none
Its setup is:
stream   : PLAYBACK
access   : MMAP_INTERLEAVED
format   : S16_LE
subformat: STD
channels : 2
rate : 44100
exact rate   : 44100 (44100/1)
msbits   : 16
buffer_size  : 8192
period_size  : 4096
period_time  : 92879
tick_time: 1
tstamp_mode  : NONE
period_step  : 1
sleep_min: 0
avail_min: 4096
xfer_align   : 4096
start_threshold  : 8192
stop_threshold   : 8192
silence_threshold: 0
silence_size : 0
boundary : 1073741824
Slave: Hardware PCM card 0 'Hammerfall DSP' device 0 subdevice 0

Its setup is:
stream   : PLAYBACK
access   : MMAP_NONINTERLEAVED
format   : S32_LE
subformat: STD
channels : 18
rate : 44100
exact rate   : 44100 (44100/1)
msbits   : 24
buffer_size  : 8192
period_size  : 4096
period_time  : 92879
tick_time: 1
tstamp_mode  : NONE
period_step  : 1
sleep_min: 0
avail_min: 4096
xfer_align   : 4096
start_threshold  : 8192
stop_threshold   : 8192
silence_threshold: 0
silence_size : 0
boundary : 1073741824
underrun!!! (at least 683.032 ms long)
Status:
state   : XRUN
trigger_time: 1073628220.377968000
tstamp  : 1073628221.60968000
delay   : 0
avail   : 8192
avail_max   : 8192
underrun!!! (at least 280.077 ms long)
Status:
state   : XRUN
trigger_time: 1073628222.81239
tstamp  : 1073628223.92462000
delay   : 0
avail   : 8192
avail_max   : 8192
underrun!!! (at least 179.903 ms long)
Status:
state   : XRUN
trigger_time: 1073628224.413069000
tstamp  : 1073628224.592968000
delay   : 0
avail   : 8192
avail_max   : 8192
underrun!!! (at least 143.721 ms long)
Status:
state   : XRUN
trigger_time: 1073628225.18556
tstamp  : 1073628225.329276000
delay   : 0
avail   : 8192
avail_max   : 8192
underrun!!! (at least 9.202 ms long)
Status:
state   : XRUN
trigger_time: 1073628225.729596000
tstamp  : 1073628225.738793000
delay   : 0
avail   : 8192
avail_max   : 8192
underrun!!! (at least 126.904 ms long)
Status:
state   : XRUN
trigger_time: 1073628227.274901000
tstamp  : 1073628227.401799000
delay   : 0
avail   : 8192
avail_max   : 8192
underrun!!! (at least 225.002 ms long)
Status:
state   : XRUN
trigger_time: 1073628227.898006000
tstamp  : 1073628228.123003000
delay   : 0
avail   : 8192
avail_max   : 8192
underrun!!! (at least 163.350 ms long)
Status:
state   : XRUN
trigger_time: 1073628229.57122000
.
.
.
--
 aplay -f cd -D default audio.wav
ALSA lib pcm.c:1900:(snd_pcm_open_conf) Cannot open shared library 
/usr/lib/alsa-lib/libasound_module_pcm_plughw.so
aplay: main:502: audio open error: No such file or directory


this one is clear.  there is no plughw type.
it's a defined pcm name.
Hmm. This would appear to be a problem. While I'm not ecouraging people 
to use the hdsp as a standard consumer playback device... My customer 
(who runs windows mostly) is expecting to be able to view videos etc 
with this device simply because he can in his windows env. Should I be 
encouraging him to purchase a second card or can this be fixed using ALSA?



--
Patrick Shirkey - Boost Hardware Ltd.
Http://www.boosthardware.com
Http://www.djcj.org - The Linux Audio Users guide

 Apparently upon the beginning of the barrage, the donkey broke 
discipline

[Alsa-devel] hdsp vs 16bit vs dmix

2004-01-07 Thread Patrick Shirkey
I am seeing strange things happen while using the hdsp and dmix plugin 
and aplay as a test app.

JACK works perfectly.

I can get sound from aplay if I use this commandline:

aplay -f cd -D plughw audio.wav
Playing WAVE 'audio.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, 
Stereo
underrun!!! (at least 40.147 ms long)
underrun!!! (at least 86.223 ms long)
underrun!!! (at least 145.116 ms long)
underrun!!! (at least 79.109 ms long)
underrun!!! (at least 248.555 ms long)
underrun!!! (at least 180.442 ms long)
underrun!!! (at least 96.117 ms long)
Aborted by signal Interrupt...	

But I get an xrun every second or so which makes the card useless for 
native alsa apps .

aplay -f cd -D hw:0 audio.wav
Playing WAVE 'audio.wav' : Signed 16 bit Little Endian, Rate 44100 Hz, 
Stereo
aplay: set_params:822: Access type not available

If I try the dmix plugin (which also works with the pci card):

 pcm.!default {
   type plug
   slave.pcm dmixer
}
pcm.dmixer  {
type dmix
ipc_key 1234
slave {
pcm hw:0
period_time 0
period_size 1024
buffer_size 4096
#periods 128
rate 44100
format s16_le
}
bindings {
0 0
1 1
}
}
ctl.dmixer {
type hw
card 0
}
--
 aplay -f cd -D default audio.wav
ALSA lib pcm_hw.c:324:(snd_pcm_hw_hw_params) SNDRV_PCM_IOCTL_HW_PARAMS 
failed: Device or resource busy
ALSA lib pcm_direct.c:712:(snd_pcm_direct_initialize_slave) unable to 
install hw params
ALSA lib pcm_dmix.c:827:(snd_pcm_dmix_open) unable to initialize slave
aplay: main:502: audio open error: Device or resource busy
--

For fun I tried this syntax:

 pcm.!default {
   type plughw
   slave.pcm dmixer
}
--
 aplay -f cd -D default audio.wav
ALSA lib pcm.c:1900:(snd_pcm_open_conf) Cannot open shared library 
/usr/lib/alsa-lib/libasound_module_pcm_plughw.so
aplay: main:502: audio open error: No such file or directory
--

--
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 Apparently upon the beginning of the barrage, the donkey broke 
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Tethered to the now toppled cart, the donkey was unable to escape before 
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United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM


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[Alsa-devel] HDSP as normal user.

2004-01-05 Thread Patrick Shirkey
Using latest cvs I am unable to run the hdsp as a normal user. It works 
as root user however.

Any ideas for fixing this?

--
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 Apparently upon the beginning of the barrage, the donkey broke 
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disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
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Re: [Alsa-devel] HDSP as normal user.

2004-01-05 Thread Patrick Shirkey
Takashi Iwai wrote:
At Mon, 05 Jan 2004 19:18:44 +0900,
Patrick Shirkey wrote:
Using latest cvs I am unable to run the hdsp as a normal user. It works 
as root user however.

Any ideas for fixing this?


this sounds like a permission problem.
check the permission of /dev/snd/* files.
on some systems, the permission of these files is changed dynamically
at each login over xdm via PAM.
Hmm, seems I had to completely logout for my changes to the perms to 
have effect. Thanks for the tip.

I'll add something to the online docs about this.



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 Apparently upon the beginning of the barrage, the donkey broke 
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Tethered to the now toppled cart, the donkey was unable to escape before 
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Re: [Alsa-devel] DMIX and capture stream

2004-01-04 Thread Patrick Shirkey
David Lloyd wrote:
On Sun, 4 Jan 2004, Florian Schmidt wrote:


On Sun, 04 Jan 2004 00:07:26 -0500
Paul Davis [EMAIL PROTECTED] wrote:

I'm working on changing DMIX to allow clients to open the capture
stream.  
i don't get it. dmix is for playback, not capture. what would be the
semantics of this?
Hmm, i don't know the OP's answer to this, but i like to ask a very
related question:
It is possible to create a playback pcm device with the help of the dmix
plugin. This way several apps can playback at once - nice. But now what
about using two apps that do full duplex at the same time?


This is exacly my point.  The fact that dmix and dsnoop are half-duplex is
where I have a problem.  There seems to be no way to allow more than one
OSS-compatibility client to share audio in a satisfactory way.  Which is
to say, you can have two doing playback, or two doing capture, or maybe
one each (haven't tried yet), but you can't do two full-duplex, or one
full-duplex and one capture or playback.  This is a bug in my opinion.  
It seems to me that this issue is not addressed by anyone, and that's why 
I started this code change, to allow capture to pass through so that one 
client can do full-duplex with multiple other clients still able to 
playback.  This is not too much to ask IMHO.

Recent discussion of this has made it clear that in the near future 
Jaroslav and Takashi are planning on implementing better 
interoperability between dmix and dsnoop.

It's not high on the list of priorities because there are already other 
software mixing apps/deamons that work. It is high on the list of 
desired features though :)

As you have found out the changes required are not superficial but 
require advanced understanding of ALSA internals.

dmix and dsnoop are less than a year old and there are still apps that 
don't provide correct support for them. If you can't wait, you could try 
 to use JACK, artsd, gstreamer

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 Apparently upon the beginning of the barrage, the donkey broke 
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disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
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Re: [Alsa-devel] New hardware - New developer / tester

2003-12-20 Thread Patrick Shirkey
Doug wrote:
Hi,

My goal is to become ridiculously involved in linux audio, mainly (at 
least for now) device driver development.  I have an EZBus audio 
device/control surface/mixer, and an EZ8 ADAT I/O card (www.event1.com). 
 My goal was to write an ALSA driver for the EZ8 card and EZbus for use 
with Ardour/JACK to do 8 track live recording and be a 100% open source 
band (we are currently using XP / Cubase).  The snd-usb-audio works with 
the EZBus already, and how suprised I was to get sound out of the EZ8 
card using the ice1712 driver.  There is some noise there, and neither 
with alsamixer nor envy24control could I get any type of level control. 
 Theres lots that need to be adapted to the EZ8 I think (channel 
assignments/levels, clock source, sample rate, etc), so thats what I 
would like to do as I learn device driver programming and linux audio. 
Right now I'm deep into the Oreilly Linux Device Drivers book, and 
browsing the ALSA developers docs.

So, here are my questions:

Is there a process for registering devices in the matrix and determining 
how 'functional' is enough to be considered 'supported'?  Do I just 
shout loudly, 'EZBus works!' ? :)

If you could send an html addition to the appropriate manufacturer that 
would be helpful. Otherwise just these:

manufacturer, card name, chip, module name, needs midi firmware

Do any of the existing ice1712 developers have chip documentation I 
could have?  Is it allowed / polite / acceptable for me to tear into the 
existing ice1712 code and learn from it, then either use code from it to 
create this driver or add complete EZ8 functionality to the existing 
driver?

Probably best to start with that and ask some questions as you go.

I assume this is the list to ask driver specific questions, which I'm 
sure I will have alot of as I dive into kernel land?

yes.

I read the home site asking for testers, I have alot of PC hardware at 
my disposal, do developers just post requests for people to test code as 
the need arises?

There are times when people need testing done. Just monitor this list.

Sorry about my generally 'clueless' disposition, but I guess I have to 
start somewhere :)

Welcome aboard.

--
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 Apparently upon the beginning of the barrage, the donkey broke 
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Tethered to the now toppled cart, the donkey was unable to escape before 
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United Press International
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Re: [Alsa-devel] usb midi devices.

2003-12-17 Thread Patrick Shirkey
Clemens Ladisch wrote:
Martin Langer wrote:

And there are some other cards (Sonica, ...) which need a firmware download,
but there's no linux loader available.


MobilePre, Sonica, Transit, Ozone (all from M-Audio).

Thanks.

BTW:  Patrick, what would be the preferred format for submitting
changed to the soundcard list?  (There's no file to send a patch for.)
For now just make additions to each manufacturer. I am working on a more 
simple backend but haven't finished it yet.

ATM I have just split each manufacturer into their own file and parse it 
when the user asks to see it.

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 Apparently upon the beginning of the barrage, the donkey broke 
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[Alsa-devel] usb midi devices.

2003-12-14 Thread Patrick Shirkey
Could someone send me a list of all the usbmidi devices that require 
firmware to be inserted. I am redesigning the docs for these cards.

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 Apparently upon the beginning of the barrage, the donkey broke 
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Re: [Alsa-devel] Newbie driver prgrmmr

2003-12-14 Thread Patrick Shirkey
Coen Leermakers wrote:
Hi all,

Does anyone here know where to find some good manuals and howto's on
driver programming? I myself have some programming experience, and want
to take this some further. Since I have a sound card which is not (yet)
compatible with ALSA, I decided the ALSA-Dev mailing list was to be my
staging ground :)
There is a detailed howto available via the documentation page.



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 Apparently upon the beginning of the barrage, the donkey broke 
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[Alsa-devel] listing devices.

2003-12-14 Thread Patrick Shirkey
I'm looking but I can't find a way to get the list of all available 
devices including virtual devices defined in the .asoundrc

Is there any function that provides this?

If not has there been any thought put into how it can be accomplished?

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 Apparently upon the beginning of the barrage, the donkey broke 
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Tethered to the now toppled cart, the donkey was unable to escape before 
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Re: [Alsa-devel] listing devices.

2003-12-14 Thread Patrick Shirkey
Patrick Shirkey wrote:
I'm looking but I can't find a way to get the list of all available 
devices including virtual devices defined in the .asoundrc

Is there any function that provides this?

If not has there been any thought put into how it can be accomplished?

Sorry missed the thread on query devices in a non blcking way from a 
few days ago. I'll pick it up there.
	



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 Apparently upon the beginning of the barrage, the donkey broke 
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Tethered to the now toppled cart, the donkey was unable to escape before 
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Re: [Alsa-devel] Query devices in a non-blocking fashion

2003-12-14 Thread Patrick Shirkey
If we have a DB of info how would we define the abilities of each device?

I assume this info is available in the driver layer because there is a 
point where ALSA will return false eg. if a card is not able to run at 
48000Hz

My opinion is that a simple function could be included in alsactl which 
scans for available devices, makes a list of their abilities. Everyone 
uses post-insert alsactl restore in the modules.conf file so it would 
be essentially a non issue from a user perspective.

Couldn't this be saved in the config settings for alsactl?



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 Apparently upon the beginning of the barrage, the donkey broke 
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[Alsa-devel] Fwd: Re: [linux-audio-user] Sampling rate problems on a VXPocket v2

2003-12-11 Thread Patrick Shirkey
Fwd'd to alsa-devel.

Geoffrey Martin wrote:


Hi,

I'm using the following:
  IBM T40 laptop
  Digigram VXPocket v2
  RedHat 9
  Kernel 2.4.22-ac4
  ALSA drivers 0.9.8
  vxloader 0.9.7-i686
Measurement gear
  Audio Precision System Two Cascade
  PrismSound DSA-1
Some of my other gear is having difficulties locking to the S/PDIF output
of the card at 44.1 kHz and I have a suspicion as to why this is
happening...
I have control over the sampling rate of the S/PDIF output of the card as I
would expect, however...
In the S/PDIF Channel Status Block information there is a portion of Byte 3
that is used to send the intended sampling rate (selectable between three
values, 32 kHz, 44.1 kHz and 48 kHz, depending on the arrangement of bits
0-3 in Byte 3. For specific information on this, please see Application
Note AN22REV2 from www.crystal.com). It is also possible, using bit 0 of
Byte 0, to indicate whether the signal is Professional or Consumer format.
Using my present configuration, I do not appear to have control over either
of these two variables. For example, even when I am sending a 44.1 kHz
signal at a rate of 44.1 kHz (or close enough... it's actually 44.0993, -17
ppm) Byte 3 of the Channel Status Block is saying that the signal is a 48
kHz transmission.
Are the VXPocket drivers in ALSA capable of changing these values in the
card? The official drivers for Mac OSX allow me to change the Pro/Consumer
flag using an application, and the sampling rate indicator changes
automatically with the actual sampling rate of the card's output.
Cheers
-geoff



Geoff Martin Ph.D.
Tonmeister, Bang  Olufsen a/s
email: [EMAIL PROTECTED]
web: www.bang-olufsen.com
web: www.tonmeister.ca
phone: +45 96 84 49 54





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 Apparently upon the beginning of the barrage, the donkey broke 
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Re: [Alsa-devel] alsa-oss devel-files and alsa-firmware license

2003-12-11 Thread Patrick Shirkey
Thorsten Leemhuis wrote:
- What the purpose of
/usr/lib/libaoss.a
/usr/lib/libaoss.so
after compiling alsa-oss? Nobody should be compiling with them,
or missed I something? Maybe see bug
aoss is a clever little app which hijacks oss applications output before 
it gets to the alsa emulation layer and translates it to native ALSA.



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 Apparently upon the beginning of the barrage, the donkey broke 
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Re: [Alsa-devel] dsnoop

2003-12-10 Thread Patrick Shirkey
Jaroslav Kysela wrote:


'periods 0' and 'period_time 0' is missing here. You need to overwrite the 
default values.


Thanks that works. I will update the .asoundrc page for the online docs.

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 Apparently upon the beginning of the barrage, the donkey broke 
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Tethered to the now toppled cart, the donkey was unable to escape before 
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Re: [Alsa-devel] Which API for MIDI?

2003-12-10 Thread Patrick Shirkey
Tim Goetze wrote:


i found the sequencer API to be well-documented. 
Internally right?

it's somewhat lacking
in code examples, but there are quite a few applications out there
that use it (pmidi for a start maybe, and then we have rosegarden,
muse etc).
I just spotted this older doc

http://www.alsa-project.org/~frank/alsa-sequencer/

Never seen or heard of it before.

Unsurprisingly it is quite comprehensive. How much has the api changed 
in the past 4 years? If someone who is well versed in the current 
implementation wants to vett this doc and provide an update I'll happily 
add it to the site docs.

You'll get your name in the big bright lights and everyone will 
associate you with Guru status.

This has to be one of the last core programming docs needed.

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 Apparently upon the beginning of the barrage, the donkey broke 
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Re: [Alsa-devel] Alsa sequencer port type flags

2003-12-10 Thread Patrick Shirkey
David Garcia Garzon wrote:
(Context: Keep doing with the sequencer chapter of the Alsa user conceptual 
model, an spanish documentation to be translated soon, at:
http://www.salleurl.edu/~is04069/Modders/Docs/alsadoc-sequencer.html)


Nice work. I have added a link to the docs page. Let us know if you need 
it hosted permanently.

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 Apparently upon the beginning of the barrage, the donkey broke 
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Tethered to the now toppled cart, the donkey was unable to escape before 
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[Alsa-devel] dsnoop

2003-12-09 Thread Patrick Shirkey
I'm using older alsa drivers so maybe this is fixed already.

Advanced Linux Sound Architecture Driver Version 0.9.3c.
Compiled on May 25 2003 for kernel 2.4.19 with versioned symbols.
I can sucessfully run two instances of arecord with the dsnoop plugin 
but I get an error with my intel8x0 that doesn't occur while using my 
cmipci:

arecord: set_params:847: Can't use period equal to buffer size (7526 == 
7526)

The number above is directly related to the buffer_size option below. 
7526 corresponds to buffer size of 8192. If I use a lower size like 4096 
it is:

arecord: set_params:847: Can't use period equal to buffer size (3763 == 
3763)

Here's the definition I use in my .asoundrc

pcm.mixin {
type dsnoop
ipc_key 5978293 # must be unique for all dmix plugins
ipc_key_add_uid yes
slave {
pcm hw:0,0
channels 2
period_size 1024
buffer_size 4096
rate 44100
}
bindings {
0 0
0 1
}
}


--
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Http://www.boosthardware.com
Http://www.djcj.org - The Linux Audio Users guide

 Apparently upon the beginning of the barrage, the donkey broke 
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United Press International
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By P. MITCHELL PROTHERO
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Re: [Alsa-devel] Re: doc-page

2003-12-07 Thread Patrick Shirkey
Martin Langer wrote:
Thank you very much, Patrick for the work you've done until now.

But one point I still miss is some kind of browseable Changelog. And a good
complete changelog history is also good for detecting bugs.
Well, only the users in the mailling lists can know if driver xy has changed
(or maybe improved) since the last version or not. So normal users have to
try out each version. And that's far away from perfect IMHO.
Nice idea. I think I know how this can be accomplished too.

I'll look into it.

--
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Http://www.djcj.org - The Linux Audio Users guide

 Apparently upon the beginning of the barrage, the donkey broke 
discipline and panicked, toppling the cart. At that point, the rockets 
disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
the arrival of U.S. troops.

United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM


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Re: [Alsa-devel] Alsa-OSS - Pop at start of song update (rc2)

2003-12-07 Thread Patrick Shirkey
Mark Knecht wrote:
Hi,

   The only practical idea I can offer is that someone in the Alsa camp
should take up the task of developing a set of instructions for
configuring Mozilla. et. all, in such away as to make OSS not be used. I
think Alsa should support, but not depend on, OSS at all. Today it
depends on OSS by not providing clear info on how to use Alsa inside of
browsers.
This is probably useful but the main problem I see is you are using a 
few apps that could be running alsa native but you are using the oss 
drivers instead.

mplayer will use whatever driver you specify in the config file. I 
assume xine is the same. Thats pretty much the best fix AFAICT. Then you 
only have to find a way to make sure you're java plugins don't make 
noise and stop using xmms until they fix the alsa driver.

Anything that xmms can stream mplayer can do too. Anything that doesn't 
require streaming alsaplayer can do.

Have you had major problems with java plugins? In this case you should 
look into ways to get around that. (Some people have been known to 
invest in a second $10 card just for browsing).





--
Patrick Shirkey - Boost Hardware Ltd.
Http://www.boosthardware.com
Http://www.djcj.org - The Linux Audio Users guide

 Apparently upon the beginning of the barrage, the donkey broke 
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disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
the arrival of U.S. troops.

United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM


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[Alsa-devel] Re: doc-page

2003-12-04 Thread Patrick Shirkey
Florian Schmidt wrote:
Hi Patrick. I saw your changes to the doc page and have one or two
remarks:
1.] The layout is a litle bit messed up here. The black line goes
through Supported Soundcards and Userspace Tools. If i use a smaller
font, it's ok. so i don't know, if it's a mozilla thing [Mozilla Debian
Package 1.5-3]..
Yeah I removed it (was just playing around with layout).

2.] There's quite a lot of info in the wiki allready and i wonder if we
could just create [initially empty] entries in the wiki for the missing
links [like dmix/dsnoop, 5.1 sound...]. There's already a dmix page in
the wiki which we could link to. As it grows more mature it can  someday
maybe be integrated into the normal documentation..
Using the wiki for drafts for these pages also has the advantage of that
it is very easy to set up initial pages which then can be improved.
Let's say: it becomes much easier for the user to contribute
documentation this way.. Also users who don't know what a wiki is will
soon learn if links from the doc pages go to it :)
I would prefer to have actual data before we make a link. It drives me 
nuts when I go to find info on the official site for a project and the 
links just lead to dead ends. The extra clicks for nothing make me feel 
annoyed.

3.] Contrary to my first draft of the TOC i would suggest moving the
soundcard matrix to the system maintainers section. Also the name Sound
Card Matrix  User documentation is misleading. I suppose this means
the user supplied info regarding each driver. I would rather have User
documentation as a single Topic for documentation that is really aimed
at the end user [This contains the userspace tools and the .asoundrc,
though this also overlaps with system maintainers stuff]
People have given a lot of suggestions for the matrix which I'm working 
on adding. I'm trying to find a way to combine all these things into a 
neat little package. Until then we will just have to make do with links 
to all the various sources as our main stay.

P.S.: If you want help with it, i could just do the changes to a local
copy and then send you a diff.
Sure. Any contributions are welcome.

--
Patrick Shirkey - Boost Hardware Ltd.
Http://www.boosthardware.com
Http://www.djcj.org - The Linux Audio Users guide

 Apparently upon the beginning of the barrage, the donkey broke 
discipline and panicked, toppling the cart. At that point, the rockets 
disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
the arrival of U.S. troops.

United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM


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Re: [Alsa-devel] ALSA 1.0.0rc1 released

2003-12-03 Thread Patrick Shirkey
Mark Knecht wrote:
I wish I didn't have to use OSS at all. Has anyone accomplished getting
all types of web-based multi-media to work just using Alsa? Any
mime.type files to lead the way?
I have recently been running all sorts with mplayer. at least all the 
sorenson codec I could find plus mpegs and .wmvs. all through the alsa 
driver.

If you want help setting it up let me know. It has taken me a while to 
get it right.





--
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Http://www.djcj.org - The Linux Audio Users guide

 Apparently upon the beginning of the barrage, the donkey broke 
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disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
the arrival of U.S. troops.

United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM


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Re: [Alsa-devel] Soundcard Matrix patch.

2003-12-01 Thread Patrick Shirkey
Manuel Jander wrote:
Hi,

Here is patch to the current Soundcard matrix, to reflect the current
state of the Aureal driver.
committed. Thanks for the patch.

--
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 Apparently upon the beginning of the barrage, the donkey broke 
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United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
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Re: [Alsa-devel] soundcard matrix (more)(

2003-12-01 Thread Patrick Shirkey
Peter Enderborg wrote:
The yamaha Motif products have mlan interfaces. I guess that they are 
not supported by alsa. Should be in the lists too.

Peter Enderborg wrote:

Could someone please add firewire 410 from m-audio and put a big NO 
sign on it. (there should be blacklist part
of the matrix of avoid devices) Since its m-Lan we wont get any driver 
if we can not put pressure on yamaha.

These are good points. It will become a faq unless we add a note. What 
is the official ALSA viewpoint on mlan devices?



--
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 Apparently upon the beginning of the barrage, the donkey broke 
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disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
the arrival of U.S. troops.

United Press International
Rockets on donkeys hit major Baghdad sites
By P. MITCHELL PROTHERO
Published 11/21/2003 11:13 AM


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[Alsa-devel] Fwd: [was Re: [linux-audio-user] Multiple (three) RME Hammerfalls... any experience]

2003-11-29 Thread Patrick Shirkey
I have forwarded this to alsa-devel as there is probably more luck 
finding the answer here and it would be a valuable thread to have as a 
resource.

R Parker wrote:
Hi,

I can't help you much, however, unless JACK has
recently been worked on I think there's a 32 track
limitation. It's known to be technically trivial.
Otherwise I don't believe alsa would be a problem,
assuming you've got wordclock i/o on the hammerfall
cards. 

Midi Time Code syncronization in Ardour could also be
problematic, it doesn't work with my Tascam DM-24.
Even though transport location is accurate, the
automation system can't interpret Ardour mtc. Paul has
announced his intention to implement SMPTE but I'm not
sure if that's been done yet.
I wonder what the status of 96kHz adat is with the
hammerfall drivers.
Anyway, that's not much help. Sorry.

ron

--- Ronald van Engelen [EMAIL PROTECTED] wrote:

Hi,

After rereading my message, I saw too many
unclarities and typos, so
I'll try again.
The problem is that we have to transfer a few hours
of 72-channel music
from three synced Mackie SDR24/96's to a harddisk. I
don't really know 
what format the music is recorded in.

As my colleague is unable to use three RME
Hammerfall DSP card's in his
dual G4 powermac with OSX (don't know exact
version), I wondered if I
could use Linux to capture the music to harddisk.
If anybody has any experience, please let me know.
Otherwise we will
proceed with one card and thus 24 channels at a
time, which does work.
Thanks,
Ronald van Engelen
Op vr 28-11-2003, om 20:06 schreef Ronald van
Engelen:
Hi there,

A colleague of mine is having major problems using
multiple (three) RME

Hammerfall cards using Mac OS 9 and X and
consulted me for help. 

The problem is that he has to transfer a few hours
of 72-channel music

(recorded with Mackie HDR's) to his harddisk for
further mixing and

processing.

I only have experience with (single and dual)
stereo cards using alsa,

jack, ardour and ladspa. I wonder if anybody have
experience with

multiple RME cards using alsa, and particullary
experience doing such a

task with mentioned tools? 

I haven't had the chance to examine his
(Mac-related) problem but it

sounds like he might suffering from problems
related to his

power-supply?

Regards,
Ronald van Engelen



--
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Http://www.boosthardware.com
Http://www.djcj.org - The Linux Audio Users guide

 Apparently upon the beginning of the barrage, the donkey broke 
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disconnected from the timer, leaving them strewn around the street. 
Tethered to the now toppled cart, the donkey was unable to escape before 
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United Press International
Rockets on donkeys hit major Baghdad sites
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Published 11/21/2003 11:13 AM


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Re: [Alsa-devel] USB Audio problems

2003-11-21 Thread Patrick Shirkey
Takashi Iwai wrote:


no, it has nothing to do with the latency but oopsen / lock-up.
the problem exists only at stopping the streams.  if the everything is
ok, it's not a big issue.  but, when the device runs with small
latency, which may result in xrun often, has the higher probability to
hit this bug, because ALSA tries to stop the stream if xrun happens.
Thanks. There should be a disclaimer somewhere for this problem.

--
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Being on stage with the band in front of crowds shouting, Get off! No! 
We want normal music!, I think that was more like acting than anything 
I've ever done.

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Re: [Alsa-devel] USB Audio problems

2003-11-20 Thread Patrick Shirkey
Takashi Iwai wrote:
the current code is surely buggy, because it issues sync unlink
inside the spinlocked context.  it's problematic on 2.6 kernel or SMP
system, and may result in kernel oops.  i added async_unlink module
option to change the behavior in the new version.
but it's still disabled as default, because unlinking multiple urbs
asynchronouly on 2.4 kernel causes kernel panic.  sigh.
unfortunately, there is no perfect solution to satisfy all versions
yet.  perhaps we need to redesign the linked-pcm streams to make it
possible to call prepare callback without spinlocks.

Ahhh, Finally an explanaiton for why my root /var/ partition is always 
being filled up with usb errors.

Would this also have a detrimental effect on latency? If so then I can 
let other usb-audio user know why.

--
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[Alsa-devel] [Fwd: nforce 2 support in ALSA?]

2003-10-20 Thread Patrick Shirkey
Anyone know the ETA for this driver?

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Being on stage with the band in front of crowds shouting, Get off! No! 
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---BeginMessage---
Hi, I'm working with an nforce 2 motherboard, and trying to set up 4.1 
surround. i see that nforce is supported, but is support for nforce2 on the 
way?

Thanks,

-Paul





---End Message---


Re: [Alsa-devel] Soundcard matrix maintainer - ASI

2003-09-15 Thread Patrick Shirkey
Eliot Blennerhassett wrote:
Hi Matrix maintainer,

Is there any chance that the entries for all AudioScience cards except AS4215
can be updated.
At least mention that a beta driver can be downloaded from 

http://www.audioscience.com/internet/download/beta/

I'm not sure of the criteria for changing an entry from green to white.

thanks and regards

Sorry for the delay in this. I am suffering from multiple component 
meltdowns.

I was almost there two weeks ago and then I lost my videocard.

Maybe another two weeks and everything will be back in order.

--

Patrick Shirkey - Boost Hardware Ltd.
Http://www.boosthardware.com
Http://www.djcj.org - The Linux Audio Users guide

Being on stage with the band in front of crowds shouting, Get off! No!
We want normal music!, I think that was more like acting than anything
I've ever done.
Goldie, 8 Nov, 2002
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Re: [Alsa-devel] [ANN] hdspmixer-1.1

2003-07-30 Thread Patrick Shirkey
Using a multiface with original firmware rev.

I was using the app last night and it seemed to work well except that 
today we have only a very long blank track.

Testing today with ardour reveals that the mute buttons have no effect 
on the recorded signal.

How do the mute buttons work? I fond it difficult to distinguish between 
on and off. Why is the default color red? Does that mean they are on by 
default?

IMO the default should be no color and when they are on they should be red.

I also missed a control for the headphone output.

I also found that aplay could only work at 16 bit. I will update to 
current cvs soon to check that though.



--

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Http://www.djcj.org - The Linux Audio Users guide

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Re: [Alsa-devel] some more anon cvs funkiness...

2003-07-07 Thread Patrick Shirkey
Jaroslav Kysela wrote:


I am slowly thinking to switch back our CVS repository to our dedicated 
server. SF seems overloaded these days.

		Jaroslav

It will be this way until August at the earliest. They are switching 
systems on the site to cope with massive demand and have fsked the anon 
cvs until it is ready.



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Re: [Alsa-devel] hdsp driver status

2003-07-03 Thread Patrick Shirkey
Thomas Charbonnel wrote:
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) ?
I am using multiface which hasn't been flashed. I have upgraded to a 
newer kernel (2.4.21) (0.9.4 tarball) and now I don't get the error 
about finding hwdep when I run hdsploader but there is still no sound 
coming from the device.

My friend tells me that in windows it will always have the WC button on 
when it is working correctly.

In this case after running hdsploader the box flashes and then all the 
lights turn off.

This device has worked with older versions of the alsa drivers.



--
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Http://www.djcj.org - The Linux Audio Users guide

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Re: [Alsa-devel] hdsp driver status

2003-07-03 Thread Patrick Shirkey
Patrick Shirkey wrote:
I'll also try with amixer just incase.
Yep. Things changed again. This time Jeds rich and I'm happy.

The line outs are still kaput but the outputs are working.

I got a nice clear clean signal from output 1 and 2 after running:

amixer -c 1 cset numid=23 0,0,4000;
amixer -c 1 cset numid=24 0,0,4000;
amixer -c 1 cset numid=25 0,0,4000;
amixer -c 1 cset numid=26 0,0,4000;
amixer -c 1 cset numid=27 0,0,4000;
amixer -c 1 cset numid=28 0,0,4000;
amixer -c 1 cset numid=29 0,0,4000;
amixer -c 1 cset numid=30 0,0,4000;
amixer -c 1 cset numid=31 0,0,4000;
amixer -c 1 cset numid=32 0,0,4000;
amixer -c 1 cset numid=33 0,0,4000;
amixer -c 1 cset numid=34 0,0,4000;
amixer -c 1 cset numid=34 0,0,4000;
amixer -c 1 cset numid=35 0,0,4000;
amixer -c 1 cset numid=36 0,0,4000;
amixer -c 1 cset numid=37 0,0,4000;
amixer -c 1 cset numid=38 0,0,4000;
amixer -c 1 cset numid=39 0,0,4000;
amixer -c 1 cset numid=40 0,0,4000;
Someone might want to make a script out of this...

Now I can start recording :)

--
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Http://www.djcj.org - The Linux Audio Users guide

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Re: [Alsa-devel] hdsp driver status

2003-07-03 Thread Patrick Shirkey
Patrick Shirkey wrote:
Patrick Shirkey wrote:

I'll also try with amixer just incase.


Yep. Things changed again. This time Jeds rich and I'm happy.

The line outs are still kaput but the outputs are working.

I got a nice clear clean signal from output 1 and 2 after running:

One more addition. After doing this I have to adjust the level in 
alsamixer before I can hear the sound.

Kind of odd. amixer will do it's thing happily but alsamixer thinks 
everything is still mute.

--
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We want normal music!, I think that was more like acting than anything 
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[Alsa-devel] hdsp multiface.

2003-07-02 Thread Patrick Shirkey
I have this device and with the latest incarnation of the driver I 
cannot get audio out. I had it working about 6 months ago but things 
have changed a lot since then.

It will initialise and I provide an entry in /proc/asound/.

-
./hdsploader
hdsploader - firmware loader for RME Hammerfall DSP cards
Looking for HDSP + Multiface or Digiface cards :
Card 0 : C-Media PCI CMI8738 (model 37) at 0x9800, irq 10
Card 1 : RME Hammerfall DSP (Multiface) at 0xe900, irq 11
Upload firmware for card hw:1
Error opening hwdep device on card hw:1.
-
0 [first  ]: CMI8738 - C-Media PCI CMI8738
 C-Media PCI CMI8738 (model 37) at 0x9800, irq 10
1 [hdsp   ]: H-DSP - Hammerfall DSP
 RME Hammerfall DSP (Multiface) at 0xe900, irq 11
-
I don't have a win partition so cannot check if it is the firmware 
loading problem others have witnessed.

This device has not been flashed.

//I'm not sure how to find out the firmware revision.

This is from lspci.

00:0d.0 Multimedia audio controller: Xilinx, Inc.: Unknown device 3fc5 
(rev 0a)

I use this command to set the mixer levels for the line out:
--
amixer -D hw:1 cset numid=5 0,0,16384
numid=5,iface=PCM,name='Mixer'
  ; type=INTEGER,access=rw---,values=3,min=0,max=65536,step=1
  : values=16384,0,0
--
This is what aplay returns but jack will connect:

--
aplay -f cd -D hw:1,0 file.wav
Playing WAVE '/art/1REC/020403-studio.wav' : Signed 16 bit Little 
Endian, Rate 44100 Hz, Stereo
aplay: set_params:800: Access type not available
--

With jack I can connect alsaplayer but I get not sound from the outputs.

 jack_lsp
alsa_pcm:capture_1
alsa_pcm:capture_2
alsa_pcm:capture_3
alsa_pcm:capture_4
alsa_pcm:capture_5
alsa_pcm:capture_6
alsa_pcm:capture_7
alsa_pcm:capture_8
alsa_pcm:capture_9
alsa_pcm:capture_10
alsa_pcm:capture_11
alsa_pcm:capture_12
alsa_pcm:capture_13
alsa_pcm:capture_14
alsa_pcm:capture_15
alsa_pcm:capture_16
alsa_pcm:capture_17
alsa_pcm:capture_18
alsa_pcm:playback_1
alsa_pcm:playback_2
alsa_pcm:playback_3
alsa_pcm:playback_4
alsa_pcm:playback_5
alsa_pcm:playback_6
alsa_pcm:playback_7
alsa_pcm:playback_8
alsa_pcm:playback_9
alsa_pcm:playback_10
alsa_pcm:playback_11
alsa_pcm:playback_12
alsa_pcm:playback_13
alsa_pcm:playback_14
alsa_pcm:playback_15
alsa_pcm:playback_16
alsa_pcm:playback_17
alsa_pcm:playback_18


--
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Http://www.boosthardware.com
Http://www.djcj.org - The Linux Audio Users guide

Being on stage with the band in front of crowds shouting, Get off! No! 
We want normal music!, I think that was more like acting than anything 
I've ever done.

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Re: [Alsa-devel] NV Soundstorm support? / Recording broke!

2003-06-23 Thread Patrick Shirkey
Daniel MacDonald wrote:
I've been thinking of starting a petition as I'm sick
of waiting for this. 

Good idea.

The main reason for my building impatience is as Mr.
PBD said that he's have that long, long, LONG awaited
Ardour rc out by now- what happened Paul? We expect
release candidates to have bugs- as long as it
compiles under the major distros that'll be enough.
Bring forth that glorious tarball pleeze!
The beta is being tested now and when that is ready the rc will probably 
be released.

Anyway, I have installed ALSA 0.9.4 on Redhat 9 XFS
and although sound output works fine, recording (both
line in and mic) doesn't want to work at all (under
Audacity CVS, but I tried it with the GNOME sound
recorder too). I did get recording to work on this
same machine but when I was running Mandrake 9.1 and
ALSA 0.9.3. Has anybody else reported/ solved this
problem?
Never seen it but you should try using alsamixer and arecord/aplay. 
There is a howto via the quicktoots project at the LAU guide (see below).

I tried downloading the latest alsa-driver CVS but for
some reason it refused to download past the patches
bit and it was complaining about some ALSAKERNEL stuff
so I just gave up with that.
The tarballs are working AFAIK. Did you use the ones from alsa-project.org?

--
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Http://www.djcj.org - The Linux Audio Users guide

Being on stage with the band in front of crowds shouting, Get off! No! 
We want normal music!, I think that was more like acting than anything 
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[Alsa-devel] ./snddevices

2003-06-17 Thread Patrick Shirkey
I have a report that the manually running snddevices script is needed to 
get 0.9.4 working.

Last time I asked I got the impression that it was being automatically 
handled and even saw the output messages after running make install.

However I and others using the quattro have a bizarre problem where we 
cannot access the alsa layer as a normal user.

This is the message I get before/after running ./snddevices manually:
-
snd_pcm_open: Permission denied (quattro1)
Failed to initialize plugin!
Failed to register plugin: /usr/local/lib/alsaplayer/output/libalsa_out.so
Failed to load output plugin alsa. Trying defaults.
snd_pcm_open: Permission denied (quattro1)
Failed to initialize plugin!
-
Can someone let me know what is wrong or has support for the old dev 
system been dropped now?

--
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Being on stage with the band in front of crowds shouting, Get off! No! 
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Re: [Alsa-devel] Please help!

2003-06-08 Thread Patrick Shirkey
Jacek Blizinski wrote:
I'm using a Mandrake 9.1 box with an Intel (intel8x0) card and KDE as my 
desktop environment. Thanks for any help you can provide!

Problems:
1. Doesnt do alsactl restore on startup
2. On Flash, games and other apps, the sound lags about 2-3 seconds in any WM.

Here are my KDE sound server options:
__ARTS TAB__
Start aRts server on KDE startup
Run soundserver with realtime priority
Autosuspend if idle for 60 secs
__SOUND I/O TAB__
Enable full duplex operation
Sound quality: 16 bits (high)
Audio buffer size: 232 milliseconds (10 fragments with 4096 bytes)


Change the latency setting (4096). Try 1024 or 512.

--
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Being on stage with the band in front of crowds shouting, Get off! No! 
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Re: [Alsa-devel] ALSA Soundcard Matrix

2003-06-08 Thread Patrick Shirkey
Levin Fritz wrote:
Hi,
what's the policy for adding new soundcards to the ALSA Soundcard Matrix?
I recently bought a Terratec Aureon 5.1 Fun soundcard and it works fine
with ALSA. So if you guys add soundcards based on user's emails, here's
one ;-)
Here's the relevant data:
Vendor:  Terratec
Product: Aureon 5.1 Fun
Chipset: CMI8738
Notes:   [Tio]
I have only tested the line out, headphone and toslink (optical) out so
far, though.

Thanks for the report. Now added.

--
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Re: [Alsa-devel] Quick question

2003-06-04 Thread Patrick Shirkey
David Stuart wrote:
Apparently the ALSA community is too busy to answer my little questions,
this is the sadly the 3rd or 4th time I has posted to this list with no
answer. :(
I don't think my question is too difficult. All I'm asking for is a
little help! Either a small snippet or a pointer to some other
information.. Maybe my question is too newbie-ish, but you can't develop
new experts without said experts being newbies at one time.
On Mon, 2003-06-02 at 16:01, David Stuart wrote:

Does anyone have a short example of capturing from one device and doing
a playback (of the captured data) on another?


Check out ecasound. It can do this quite easily apparently.



--
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Http://www.djcj.org - The Linux Audio Users guide

Being on stage with the band in front of crowds shouting, Get off! No! 
We want normal music!, I think that was more like acting than anything 
I've ever done.

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[Alsa-devel] USB quattro testing

2003-05-27 Thread Patrick Shirkey
I have been testing with an intel8x0 running into the quattro while
running all the four inputs. Input 3 and 4 started off working but after
killing off jack and rm /tmp/jack a few times it has become impossible
to get sound out of it with either alsa or jack even though the intel8x0 
was still functional.

This is the jack comandline as root with lowlatency on.

jackd -v -R -a -d alsa -d q4b -r 44100 -p 4096 -n3

Testing with jack only through i/o 3,4 I get a crystal clear sound.

The usb-audio driver has lost the ability to read the first two channels 
completely, replacing them with white noise and the multi or route 
plugins don't want to play nicely with jack even though this device is 
sample synced.

I tested recording with jackrec and got a lot of audio dropouts while
playing. I then used alsaplayer running with the input plugin and that 
was interesting. It gave the audio stream more of a pumping sound which 
was also adjustable by the pitch control so I could get a nice warped 
effect going. It wouldn't stand up to live recording but it could be fun 
to play with.

After rebooting and mounting /tmp instead of /tmp/jack (as well as 
spending two or three hours trying to get it back before hand). I am 
currently listening with ams in a direct loop and it sounds the same as 
alsaplayer input. This is a good recording of what I get using a sine 
wave through inputs 3,4 recorded with the ams wave output while using 
2in,4out.

	http://www.djcj.org/audio/test/052703-amsrec-4i2o.wav

This is with arecord -f cd for inputs 1,2:

	http://www.djcj.org/audio/test/052703-arec-2a.wav

=
pcm.quattro {
type multi;
slaves.a.pcm hw:2,0;
slaves.a.channels 2;
slaves.b.pcm hw:2,1;
slaves.b.channels 2;

 bindings.0.slave a;
 bindings.0.channel 0; # left
 bindings.1.slave a;
 bindings.1.channel 1; # right
 bindings.2.slave b;
 bindings.2.channel 0; # left
 bindings.3.slave b;
 bindings.3.channel 1; # right

}
ctl.quattro {
type hw;
card 2;
}
pcm.q4b {
type route;
slave.pcm quattro;
ttable.0.0 1;
ttable.1.1 1;
ttable.2.2 1;
ttable.3.3 1;
}

ctl.q4b {
 type hw;
card 2;
}
===




--
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Http://www.djcj.org - The Linux Audio Users guide

Being on stage with the band in front of crowds shouting, Get off! No!
We want normal music!, I think that was more like acting than anything
I've ever done.
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[Alsa-devel] User notes online

2003-04-05 Thread Patrick Shirkey
I have whipped up a page to which displays all the user notes contributions.

http://www.alsa-project.org/alsa-doc/doc-php/notes-full.php3

It can also be accessed quickly through the main matrix page.

I have been feeling slightly guilty that I'm probably the only person 
who has a full view of what people are contributing to the notes and 
managed to figure out how to generate a page so y'all can find out these 
things too.

Hopefully this will make things easier for people when they are looking 
at purchasing new cards/devices too.

It should also give developers an idea of which cards are most used. 
Currently the people who use the via82xx module are the most prolific 
contributers.

Hopefully it doesn't make any problems for the server. I'm not sure how 
much it affects the machine/s running things.

--
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[Alsa-devel] quattro bug

2003-04-02 Thread Patrick Shirkey
I found out last night that input 3 is not working for the latest cvs.

All the other i/o ports are functioning correctly though.

--
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[Alsa-devel] alsa-utils make error

2003-04-01 Thread Patrick Shirkey
I get an error on line 221 in alsactl.c

undefined call to snd_ctl_elem_info_is_first

with latest cvs

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[Alsa-devel] Fwd: via82xx problem

2003-03-28 Thread Patrick Shirkey
This was submitted to the notes for via82xx.

Felix Braun
Saturday, 29 March 2003
I have a VIA VT8233 (on a ASUS A7V8X motherboard). The last version that
produces sound output is alsa 0.9rc_6. All later version remain silent.
I've not been able to get sound recording to work at all.
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Re: [Alsa-devel] dsnoop, dshare and dmix plugins

2003-03-20 Thread Patrick Shirkey
Fernando Pablo Lopez-Lezcano wrote:
It would be nice that the user not need to type magic numbers into a
configuration file, unique or not. I don't know what it involves, but it
would make using the configuration file much easier. 

Even nicer is if this could be embedded like the dmix was eventually so 
that it is unnecessary to use the .asoundrc :)

--
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[Alsa-devel] dmix vs alsaplayer [was jackd, zynaddsubfx and iiwusynth]

2003-03-03 Thread Patrick Shirkey
Levi Burton wrote:

Has anyone tried the dmix plugin thats part of alsa rc8?  This looks 
interesting, but im scared if I install rc8, I will be bitten by that 
alsa-lib bug again.

I was fscking around with it on Saturday. So far I have managed to use 
multiple instances of aplay but other apps like alsaplayer bork at it. I 
have yet to figure out why but I think I should probably submit a bug 
report somewhere.

I had to remove /dev/dsp otherwise alsaplayer will use the oss drivers.

Both of these commands give the same error.

$ alsaplayer -o alsa -d plug:'dmix:RATE=44100' track_8.wav
$ alsaplayer -o alsa -d plug:\'dmix:SLAVE=\hw:2,1\,RATE=44100\' 
track_8.wav


error on set_period_size (512)
Unavailable hw params:
ACCESS:  RW_INTERLEAVED
FORMAT:  S16_LE
SUBFORMAT:  STD
SAMPLE_BITS: 16
FRAME_BITS: 32
CHANNELS: 2
RATE: 44100
PERIOD_TIME: (125011 125012)
PERIOD_SIZE: 5513
PERIOD_BYTES: 22052
PERIODS: (2 3)
BUFFER_TIME: (371519 371520)
BUFFER_SIZE: 16384
BUFFER_BYTES: 65536
TICK_TIME: ALL
error on set_period_size (1024)
Unavailable hw params:
ACCESS:  RW_INTERLEAVED
FORMAT:  S16_LE
SUBFORMAT:  STD
SAMPLE_BITS: 16
FRAME_BITS: 32
CHANNELS: 2
RATE: 44100
PERIOD_TIME: (125011 125012)
PERIOD_SIZE: 5513
PERIOD_BYTES: 22052
PERIODS: (2 3)
BUFFER_TIME: (371519 371520)
BUFFER_SIZE: 16384
BUFFER_BYTES: 65536
TICK_TIME: ALL
failed to configure output device...trying OSS
error opening /dev/dsp
I could not find a suitable output module on your
system. Make sure they're in /usr/local/lib/alsaplayer/output/.
Use the -o parameter to select one.
failed to load output plugin (alsa). exitting...

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Being on stage with the band in front of crowds shouting, Get off! No! 
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Re: [Alsa-devel] aserver, smix

2003-03-01 Thread Patrick Shirkey
Jaroslav Kysela wrote:

Are you using latest CVS drivers? Previous drivers are not compatible.

		Jaroslav

Yes, I did a cvs update today.

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[Alsa-devel] ALSA - software mixing docs.

2003-03-01 Thread Patrick Shirkey
I have added some info to the asoundrc docs on using the new dmix plugin

http://www.alsa-project.org/alsa-doc/doc-php/asoundrc.php3?company=Genericcard=Genericchip=Genericmodule=Generic#softmix

It allows software mixing of multiple streams at one time in a similar 
way to ESD,ARTS,JACK.

Currently it is only available with latest cvs so needs testing with 
various apps to make sure it works as desired.

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Re: [Alsa-devel] Interesting problem with hotplug and Midisport 2x2on Mdk 9.0

2003-02-28 Thread Patrick Shirkey
Takashi Iwai wrote:

I see it too with the quattro and another user has reported it with the 
audigy. It would be nicer if we didn't have to remember to unload the 
usb-audio driver before hotplugging. Is it possible to make that happen.


then there is still a bug in this regard.
are you using MIDI on quattro, too?
It is automatically used. Clemens would have to explain as I cannot 
remember the exact details.

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Re: [Alsa-devel] aserver, smix

2003-02-28 Thread Patrick Shirkey
Jaroslav Kysela wrote:

Note that we have actually the experimental dmix plugin (direct mixing) 
which works for i386 architecture in CVS. The usage is simple:

aplay -Dplug:dmix filename

You may pass 'plug:dmix' to any other application.

When I try this I get this error:

~$ aplay -D plug:dmix -f cd track_8.wav
ALSA lib pcm_hw.c:371:(snd_pcm_hw_sw_params) SNDRV_PCM_IOCTL_SW_PARAMS 
failed: Invalid argument
ALSA lib pcm_dmix.c:1378:(snd_pcm_dmix_initialize_slave) unable to 
install sw params
ALSA lib pcm_dmix.c:1534:(snd_pcm_dmix_open) unable to initialize slave

aplay: main:479: audio open error: Invalid argument

Using the example for the .asoundrc file in the original dmix thread is 
the same.

~$ aplay -D dmix -f cd track_8.wav
ALSA lib pcm_hw.c:371:(snd_pcm_hw_sw_params) SNDRV_PCM_IOCTL_SW_PARAMS 
failed: Invalid argument
ALSA lib pcm_dmix.c:1378:(snd_pcm_dmix_initialize_slave) unable to 
install sw params
ALSA lib pcm_dmix.c:1534:(snd_pcm_dmix_open) unable to initialize slave

aplay: main:479: audio open error: Invalid argument

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Re: [Alsa-devel] Interesting problem with hotplug and Midisport 2x2on Mdk 9.0

2003-02-28 Thread Patrick Shirkey
Patrick Shirkey wrote:
Takashi Iwai wrote:

I see it too with the quattro and another user has reported it with 
the audigy. It would be nicer if we didn't have to remember to unload 
the usb-audio driver before hotplugging. Is it possible to make that 
happen.


then there is still a bug in this regard.
are you using MIDI on quattro, too?
It is automatically used. Clemens would have to explain as I cannot 
remember the exact details.

I another bug I was just reminded of is that if I remove the alsa 
modules while the usb device is still switched on I get a complete hang 
where I have to restart with alt+sysreq+b

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Re: [Alsa-devel] Interesting problem with hotplug and Midisport 2x2on Mdk 9.0

2003-02-27 Thread Patrick Shirkey
Takashi Iwai wrote:

hmm, the hot unplugging of ALSA usb devices was improved in the recent
version - basically i didn't see problems with the devices i have.
but i've not tested with midi devices, so this might be a problem.
just make sure that you unload snd-usb-audio (snd-usb-midi doesn't
exist any longer) before unplugging the device.



I see it too with the quattro and another user has reported it with the 
audigy. It would be nicer if we didn't have to remember to unload the 
usb-audio driver before hotplugging. Is it possible to make that happen.

--
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[Alsa-devel] Parsing asoundrc for devices.

2003-02-26 Thread Patrick Shirkey
I need a way to find out the details of the devices I create with the 
asoundrc file.

It would be extremely useful to have it as part of the alsa-tools or 
maybe just part of alsactl/amixer.

Is this on anyone short term TODO list or will I need to look into it 
myself?

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Re: [Alsa-devel] Parsing asoundrc for devices.

2003-02-26 Thread Patrick Shirkey
Paul Davis wrote:
I need a way to find out the details of the devices I create with the 
asoundrc file.

It would be extremely useful to have it as part of the alsa-tools or 
maybe just part of alsactl/amixer.

Is this on anyone short term TODO list or will I need to look into it 
myself?


more than snd_card_next () ?

I don't quite follow this. Is that a fictional call which an app could 
use to get these details or is it an already functioning piece of code 
that I haven't seen in action?

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[Alsa-devel] cs46xx: alsa-docs note

2003-02-26 Thread Patrick Shirkey
Sorry I must have missed sending this here.

The following has been submitted as a note for the documentation of cs46xx.

sender: [EMAIL PROTECTED]
date:   Friday, 03 January 2003
Terratec DMX XFire 1024:

The DMX XFire 1024 doesn't get initialized correctly the first time
the alsa modules are loaded. Sound is heavily distorted. Usually
reloading the modules solves this problem. One can do that
autmatically by loading, unloading and reloading the modules in
the bootscripts.
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[Alsa-devel] via82xx: alsa-docs note

2003-02-26 Thread Patrick Shirkey
I'm wondering if this is a bug that can be fixed or not.

The following has been submitted as a note for the documentation of via82xx.

sender: [EMAIL PROTECTED]
date:   Saturday, 18 January 2003
I don't use sonypi with via82xx on my VAIO Laptop
because sonypi and via82xx try to use the same region
and I obtained this error message :
 unable to grab ports 0x1000-0x10ff

My chipset is :
00:07.5 Multimedia audio controller: VIA Technologies, Inc. VT82C686 
AC97 Audio Controller (rev 50)
Subsystem: Sony Corporation: Unknown device 80f6
Flags: medium devsel, IRQ 5
I/O ports at 1000 [size=256]
I/O ports at 1c54 [size=4]
I/O ports at 1c50 [size=4]
Capabilities: [c0] Power Management version 2



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Re: [Alsa-devel] Parsing asoundrc for devices.

2003-02-26 Thread Patrick Shirkey
Paul Davis wrote:
I don't quite follow this. Is that a fictional call which an app could 
use to get these details or is it an already functioning piece of code 
that I haven't seen in action?


the latter.

Sorry to be lame but...

Does anyone know if it is used in any apps that I might have in my 
system or have I not seen it in action because no one uses it yet?



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Re: [Alsa-devel] aserver, smix

2003-02-26 Thread Patrick Shirkey
Jozef Kosoru wrote:
Hi!

A couple of questions:

I would like to ask what is the main purpose of the aserver. It can run
on the same machine as the client only (AFAIK) and it cannot mix multiple
client streams into the one device. So I've missed the point of that
application -- is there something for which it could be good for?
ALSA plugin documentation mentions the 'smix' plugin. But I cannot find
any further information nor the source code in the library for that. Is
it something which will be introduced it the future ALSA releases?
According the doc it is exactly something I've been looking for a long
time; pcm device sharing.
This has been discussed many times.

Currently the only ways to get pcm device sharing, I assume you mean 
using multiple devices as one virtual device are to write it up in your 
.asoundrc or use the patch which has been submitted but not applied to 
jack for this purpose.

The former is a little tricky to grok but it has been shown to work. The 
later requires a bit more of a deeper understanding of jack and coding.

There is info in the online docs. Just click on the link in the asoundrc 
 section for your device and the file is explained in depth.

If however you mean hardware mixing (multiple streams at the same time) 
then it is dependant on the device and the manufacturers supplied tech 
details.

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[Alsa-devel] [Fwd: usbaudio panic]

2003-02-24 Thread Patrick Shirkey
Sending on behalf of John
---BeginMessage---
Hi --

I'm sorry to bother you, but would you be so kind
as to forward this to the alsa-devel list?
I've tried to mail it there myself without
success;  for details about that see
http://sourceforge.net/tracker/?group_id=1atid=21func=detailaid=683706

=

Using linux 2.4.16 - alsa rc6 - usbaudio - extigy ...
While some audio is playing,
power down the extigy using its front-panel switch.
Result:  instant system crash.

I had no trouble reproducing this.
-- Got a kernel panic at interrupt level once
 (something about module 'hotplug').
-- Got no message at all once, possibly because
I was running Xwindows on the console at the time..
If you have trouble reproducing it, let me
know and I'll see if I can provide more information.

---End Message---


[Alsa-devel] FYI: hdsp note

2003-02-24 Thread Patrick Shirkey
The following has been submitted as a note for the documentation of hdsp.

johannes.taelman(a)rug.ac.be
In order to make alsa-hdsp recognize the hdsp I had to downgrade
the firmware of the card(bus) to revision 10. There is a flash update
tool available at 
http://www.rme-audio.de/english/download/drivers_archive.htm
You'll have to do that in windows...



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[Alsa-devel] Website maintainer away.

2003-02-06 Thread Patrick Shirkey
Hi.

I will be away until the 24 so any changes to the website will have to 
go through Jaroslav until then.

Rgds.

--
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[Alsa-devel] hdsp multiface pci.

2003-01-19 Thread Patrick Shirkey
 #5 'ADAT2'
  : values=0


--


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Re: [Alsa-devel] hdsp multiface pci.

2003-01-19 Thread Patrick Shirkey
Thomas Charbonnel wrote:

On Sun, 2003-01-19 at 13:08, Patrick Shirkey wrote:


Using latest cvs, I have been trying to get this device working for my 
friend. We have had some success but it is unreliable.

We could get sound out of the lineout port after using this command

 amixer -D hw:0 cset numid=5 0,0,16384



This is supposed to route the signal received on analog input 1 to
analog output 1.



numid=5,iface=PCM,name='Mixer'
  ; type=INTEGER,access=rw---,values=3,min=0,max=65536,step=1
  : values=16384,0,0

Using

 	numid=5 26,26,16384
 	numid=5 27,27,16384



The above should indeed be the correct way to route software playback
channels 1-2 to the line outs (jack's alsa_pcm:playback_1 and _2, for
example).



Sorry I'm not fully grasping this. Why are 26 and 27 directly related to 
alsa_pcm:playback_1 and _2?

I thought from reading the notes on the hdsp page that pcm1 and pcm2 
would be numbers 0 and 1 not 26 and 27.

eg.
==
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.

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)
==

See how it says 0-7 (analog)?	

I haven't found an explanation for the following in the docs describing 
how to use the amixer app so could you please clarify this for me?

What is this line saying?

  	numid=5 26,26,16384

something like:

connect 26 to 5 with a volume of 16384




or
 	numid=5 10,10,16384 --- 10 appears to be line out id




What makes you say so ?




These are the id's in question:

numid=5,iface=PCM,name='Mixer'
  ; type=INTEGER,access=rw---,values=3,min=0,max=65536,step=1
  : values=16384,0,0

numid=10,iface=PCM,name='Line Out'
  ; type=BOOLEAN,access=rw---,values=1
  : values=on

numid=26,iface=PCM,name='Playback Peak',index=3
  ; type=INTEGER,access=r,values=2,min=0,max=0,step=0
  : values=0,0

numid=27,iface=PCM,name='Playback RMS',index=3
  ; type=INTEGER64,access=r,values=1,min=0,max=0,step=0
  : values=0



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Re: [Alsa-devel] hdsp multiface pci.

2003-01-19 Thread Patrick Shirkey
Another question.

What are each of these?


IEC958 Playback Con Mask
IEC958 Playback Pro Mask
Chn
IEC958 Input Connector
IEC958 Output also on ADAT1
IEC958 Playback Default
Input Peak
Input RMS
Line Out
Mixer
Output Peak
Passthru
Playback Peak
Playback RMS
Preferred Sync Source
=

Some are self explanatory but I will add it to the notes if someone will 
go through and explain each one.

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Re: [Alsa-devel] hdsp multiface pci.

2003-01-19 Thread Patrick Shirkey
Thomas Charbonnel wrote:



You have to understand that software outputs are not directly related to
physical outputs. On simpler hardware, software and physical outputs can
be considered the same. On the hdsp system, thanks to the internal
matrix mixer, software outputs are totally abstracted from physical
ones. On the multiface you have 18 software outputs, each of wich can be
independently routed to any of the 18 physical outputs. For convenience,
the hdsp linux driver's default behaviour is to have a 1:1 routing
policy: each software output is by default routed to to corresponding
hardware output.
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.



Great explanation. I have added it as an editors note on the hdsp page.

so it's default is a fifo?

i.e 	software output 26 - analog output 1
	software output 27 - analog output 2

Therefore can I do this?

numid=5 26,0,16384			
numid=5 27,1,16384
numid=5 28,2,16384
numid=5 29,3,16384
numid=5 30,4,16384
numid=5 31,5,16384
numid=5 32,6,16384
numid=5 33,7,16384

Or is that the default setting (except with no volume).



Here it looks like you're mixing up the amixer numid parameter number
with the hdsp internal channel numbers.
Numid 5 is the actual alsa matrix mixer control. Numid 10 is the global
line out switch. Numid 26 and 27 are read only, and correspond to the
internal peak and rms calculation the card does to provide a 0% CPU
metering solution (to be used in the forthcoming HDSPMixer totalmix
clone !...)

I hope this makes things clearer.



Getting there. So we have to tell the control called numid=5 to route 
the output from software 1 to line out before we can hear it through 
line out?

Can we route multiple software outputs to the same hardware output?

eg.	numid=5 26,26,16384			
	numid=5 27,27,16384
	numid=5 28,26,16384
	numid=5 29,27,16384
	numid=5 30,26,16384
	numid=5 31,27,16384
	numid=5 32,26,16384
	numid=5 33,27,16384

Or is that what the .asoundrc is for?



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Re: [Alsa-devel] hdsp multiface pci.

2003-01-19 Thread Patrick Shirkey
Patrick Shirkey wrote:

Thomas Charbonnel wrote:




You have to understand that software outputs are not directly related to
physical outputs. On simpler hardware, software and physical outputs can
be considered the same. On the hdsp system, thanks to the internal
matrix mixer, software outputs are totally abstracted from physical
ones. On the multiface you have 18 software outputs, each of wich can be
independently routed to any of the 18 physical outputs. For convenience,
the hdsp linux driver's default behaviour is to have a 1:1 routing
policy: each software output is by default routed to to corresponding
hardware output.


In essence the hdsp can run 18 software streams at the same time as 
processing upto 18 hardware input streams?

Do I need to set the volume for the analog inputs? Or will I be able to 
capture from them without setting anything in amixer?





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[Alsa-devel] Re: [linux-audio-user] quattro distortion under mandrake 9.0 - moredetails

2003-01-18 Thread Patrick Shirkey
iriXx wrote:


any suggestions would be very gratefully received!



Sorry. I forgot you weren't using the latest cvs. If you would like some 
help installing it again (after your last round) I could ssh in and do 
it for you.

Let me know off list if you think that is a good idea.

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[Alsa-devel] Re: Some ALSA documents on line

2003-01-09 Thread Patrick Shirkey
Takashi Iwai wrote:

Hi Patrick,



- ALSA Driver API Reference

	http://www.alsa-project.org/~iwai/alsa-driver-api/index.html
	http://www.alsa-project.org/~iwai/alsa-driver-api.pdf
	http://www.alsa-project.org/~iwai/alsa-driver-api.sgml



Is this generated daily in cvs or are you doing it manually?


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[Alsa-devel] cs46xx - Terratec DMX XFire 1024

2003-01-03 Thread Patrick Shirkey
[EMAIL PROTECTED] wrote:

The following has been submitted as a note for the documentation of cs46xx. 


   	   	Richard Stevens   
   	   	Friday, 03 January 2003/td


Terratec DMX XFire 1024:

The DMX XFire 1024 doesn't get initialized correctly the first time 
the alsa modules are loaded. Sound is heavily distorted. Usually 
reloading the modules solves this problem. One can do that 
autmatically by loading, unloading and reloading the modules in 
the bootscripts.
		




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Re: [Alsa-devel] Converted Writing an ALSA Driver to DocBook.

2002-12-23 Thread Patrick Shirkey
Kevin Conder wrote:

In case any one was curious, I converted Takashi's Writing an ALSA
Driver document to DocBook/SGML. It's available for download here:

http://www.kevindumpscore.com/download/writing-an-alsa-driver.sgml




I have downloaded it to add to the site. I will render it to html so we 
can provide a nice way to browse it online.




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Re: alsasound init script (Re: [Alsa-devel] possible problems withrc6 aplay )

2002-12-23 Thread Patrick Shirkey
Takashi Iwai wrote:

At Fri, 20 Dec 2002 23:53:13 +0100,
Pedro Lopez-Cabanillas wrote:


Clemens, I remember that /proc/asound/cards returned this bus/device  
information for USB MIDI devices some time ago (rc3?), like it does for my SB 
128 PCI.


it would be nice to provide a new generic proc file which shows the
low-level hardware information as the card number, the module name and
the PCI/ISA-PnP/USB/PCMCIA IDs like:

0 snd-ens1371 0x0123 0xABCD
1 snd-usb-audio 0x 0x



Could you add to that the irq number?



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Re: [Alsa-devel] Converted Writing an ALSA Driver to DocBook.

2002-12-23 Thread Patrick Shirkey
Takashi Iwai wrote:




In case any one was curious, I converted Takashi's Writing an ALSA
Driver document to DocBook/SGML. It's available for download here:

http://www.kevindumpscore.com/download/writing-an-alsa-driver.sgml




I have downloaded it to add to the site. I will render it to html so we 
can provide a nice way to browse it online.


nice.  if you finish the set up, please let me know the url.
i'll add it as the reference to the new version.



After playing around with the docbook install for a few hours it's 
probably more appropriate if Kevin can put the html versions on line for 
me to download. I will be here for a while doing this otherwise.


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Re: [Alsa-devel] usbaudio won't do 24-bit or 32-bit i/o...

2002-12-23 Thread Patrick Shirkey
Takashi Iwai wrote:

At Sat, 21 Dec 2002 07:00:45 +0900,
Patrick Shirkey wrote:


Takashi Iwai wrote:


hmmm, it seems that the problem is on usb-uhci driver.
mysteriously, the midi part of quattro causes the kernel panic.  the
audio part seems ok.
you can avoid this either by using uhci driver or by commenting out
the entry for quattro in alsa-kernel/usb/usbquirks.h.



I am trying it out on my ohci machine too and the latest driver has 
fixed the problem with not being able to capture from all the input.

Thanks.

I have commented out the midi quirk though in alsa-kernel/usb/usbquirk.h

I will be saving this cvs for sure.

The 24_3le support in arecord still captures the left channel maxed and 
the right at normal volume on pcm0 and pcm1.


did you update the alsa-lib, too?
there was a bug regarding 24bit conversion both on alsa-driver and
alsa-lib.



I updated all the cvs files and recompiled alsa-driver, alsa-lib and 
alsa-utils.

I will try again soon.




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Re: [Alsa-devel] usbaudio won't do 24-bit or 32-bit i/o...

2002-12-16 Thread Patrick Shirkey
Takashi Iwai wrote:


 
hmm, it's weird.  it would be nice if we can know at which point the
driver became broken...


It was working nicely before the drivers were merged but wouldn't paly 
with jack. Now it likes jack.


the attached patch will (hopefully) do the same thing as qinit in the
kernel at the initialization (applied to the latest cvs).  please let
me know whether it works.


Ouch. The latest cvs results in a hard lock when loading the usb-audio 
driver.





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Re: [Alsa-devel] usbaudio won't do 24-bit or 32-bit i/o...

2002-12-14 Thread Patrick Shirkey
Takashi Iwai wrote:



could you tell me the rcs version numbers of the files on
alsa-kernel/usb you are using (18 Nov.) ?  i've checked the files via
cvs but i couldn't see any differences around the date.



I have managed to test a more upto date version since then and it is the 
same. I have no idea when this happened though as I wasn't actively 
testing the 24 bit support until recently. I have a vague memory of 
testing it much earlier in the year but I think that was only for playback.

I currently cannot record from input 1 and 2 either.

Working are:

output 1,2,3,4
input 3,4

I also have to initialise both pcms with the small utility you made.




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Re: [Alsa-devel] soundcard matrix: broken links etc.

2002-12-10 Thread Patrick Shirkey
John S. Denker wrote:

Hi --

The soundcard matrix needs attention.
  http://www.alsa-project.org/~goemon/



This version is now deprecated. How did you get access to it? I noticed 
the other day that google search for ALSA lists this page as second.

We should make the index for geomon point to it too. Jaroslav or Dan can 
you do that?

If it was through a link on the site I will squash it.




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Re: [Alsa-devel] soundcard matrix: broken links etc.

2002-12-10 Thread Patrick Shirkey
John S. Denker wrote:


However, the apparently non-deprecated version
  http://www.alsa-project.org/alsa-doc/
contains almost as many broken links etc.
Here's the summary:

 Code Occurrences  What to do
 200 1 There are broken fragments which must be fixed.
 301 1 You should update the link.
 302 28Usually nothing.
 401 4 The link is not public. You'd better specify it.
 403 1 The link is forbidden! This needs fixing.
 404 57The link is broken. Fix it NOW!
 405 1 The server does not allow HEAD requests.
 500 37Either the hostname is incorrect or ...

Interested parties may grab the details from
  http://www.av8n.net/~jsd/alsa/checked-links2.html
  http://www.av8n.net/~jsd/alsa/checked-links2.html.gz



Thanks for that. If you have the inclination to send in a patch that 
would be very appreciated. Otherwise it will have to wait for me to get 
some more time.


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Re: [Alsa-devel] usbaudio won't do 24-bit or 32-bit i/o, and won'tdo 96000 frames per second

2002-12-07 Thread Patrick Shirkey
Jaroslav Kysela wrote:

 
 Another big joke from the Creative PR staff ;-((
 Patrick, could you add this info to USB device web pages?
 

Done.

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Re: [Alsa-devel] usbaudio won't do 24-bit or 32-bit i/o, and won'tdo 96000 frames per second

2002-12-05 Thread Patrick Shirkey
Clemens Ladisch wrote:

John S. Denker wrote:


Request #1: USB driver supporting 24-bit i/o.
Request #2: USB driver supporting 96000 frames per second.



What you want is already supported by the snd-usb-audio driver. IIRC
Patrick Shirkey reported that 24bit@96kHz works with the M-Audio Quattro.



Actually I have just yesterday noticed a specific problem with the 24 
bit support.

Also you should know that the quattro only supports 24_3le which is 24 
bits, three bytes. This is contrary to possibly all other pro devices 
which support 24bits, four bytes or something like that. Meaning that 
you have to use a special bit depth just for the quattro. This could be 
the problem you are seeing.

Currently I am using the cvs from the 18 November. I will update and 
check again. I have initialised both pcms and I cannot record a signal 
through the first pcm hw:1,0 although arecord doesn't complain.

Using 24_3le,2,44100 I get a normal signal on the left (input 3) but 
maxed signal on the right (input 4). If I press in the input level 
button on the device input 3 is boosted (again normal operation) but 
input 4 is still maxed.

I have verified it is the same for 88200 and 96000hz too.

arecord will not work for 88200 or 96000 at 16 bit eg.

arecord -D hw:1,1 -c 2 -f s16_le -r 88200 -d 10 
/art/1music/rec/test-10sec-88-16.wav
Recording WAVE '/art/1music/rec/test-10sec-88-16.wav' : Signed 16 bit 
Little Endian, Rate 88200 Hz, Stereo
ALSA lib pcm_hw.c:428:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE 
failed: Invalid argument
arecord: set_params:814: Unable to install hw params:
ACCESS:  RW_INTERLEAVED
FORMAT:  S16_LE
SUBFORMAT:  STD
SAMPLE_BITS: 16
FRAME_BITS: 32
CHANNELS: 2
RATE: 88200
PERIOD_TIME: (46439 46440)
PERIOD_SIZE: 4096
PERIOD_BYTES: 16384
PERIODS: 4
BUFFER_TIME: (185759 185760)
BUFFER_SIZE: 16384
BUFFER_BYTES: 65536
TICK_TIME: 1




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Re: [Alsa-devel] usb quattro not recording

2002-12-03 Thread Patrick Shirkey
Patrick Shirkey wrote:

Much to my surprise the quattro has stopped being able to record with 
cvs from Nov 27 2002 and also a fresh checkout today.


It's definitely broken as going back to the 18 November works. Damn, 
looks like I'll have to take the computer to the next gig too ;)


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Re: [Alsa-devel] [Alsa-user] chipset CX5530

2002-12-02 Thread Patrick Shirkey
That's very informative Takashi.

I have stored it for inclusion in the alsa-docs at a later stage when 
this information is more fully explained.


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[Alsa-devel] usb quattro not recording

2002-12-02 Thread Patrick Shirkey
Much to my surprise the quattro has stopped being able to record with 
cvs from Nov 27 2002 and also a fresh checkout today.

Neither pcm devices are able to capture.

Playback is fine however.

It might be telling that jack hangs here in capture only mode:


jackd -d alsa -d quattro1 -C
jackd 0.40.0
Copyright 2001-2002 Paul Davis and others.
jackd comes with ABSOLUTELY NO WARRANTY
This is free software, and you are welcome to redistribute it
under certain conditions; see the file COPYING for details

loading driver ..
creating alsa driver ... quattro1|1024|2|48000|swmon
starting engine



$ cat /proc/asound/card2/pcm0c/
info  oss   sub0
$ cat /proc/asound/card2/pcm0c/info
card: 2
device: 0
subdevice: 0
stream: CAPTURE
id: USB Audio
name: USB Audio
subname: subdevice #0
class: 0
subclass: 0
subdevices_count: 1
subdevices_avail: 1
$ cat /proc/asound/card2/pcm0c/oss
$ cat /proc/asound/card2/pcm0c/sub0/
cat: /proc/asound/card2/pcm0c/sub0/: Is a directory
$ cat /proc/asound/card2/pcm0c/sub0/
hw_params  info   prealloc   status sw_params
$ cat /proc/asound/card2/pcm0c/sub0/hw_params
closed
$ cat /proc/asound/card2/pcm0c/sub0/info
card: 2
device: 0
subdevice: 0
stream: CAPTURE
id: USB Audio
name: USB Audio
subname: subdevice #0
class: 0
subclass: 0
subdevices_count: 1
subdevices_avail: 1
$ cat /proc/asound/card2/pcm0c/sub0/prealloc
64
$ cat /proc/asound/card2/pcm0c/sub0/status
closed
$ cat /proc/asound/card2/pcm0c/sub0/sw_params
closed



How embarrasing. I thought I was recording on Friday night and told my 
friend we wouldn't need his MD. Now I have 4 hours of pure silence :(

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[Alsa-devel] unresolved symbol snd.o

2002-11-29 Thread Patrick Shirkey
I tried to update the cvs here last night and was getting an error 
installing the alsa-driver.

Something about unresolved symbols in snd.o

Going back to cvs from the 18/11 the problem is not there.

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Re: [Alsa-devel] USB devices should not oops after disconnect now

2002-11-29 Thread Patrick Shirkey
Jaroslav Kysela wrote:

Hello all,

	the subject says everything. CVS contains first code to avoid 
oops / systems hangs when a running hotplug device is removed from the 
system. Please, test and report problems.


I have updated and now when I disconnect the device it will not register 
on reconnect. No connection found messages are written to dmesg an it 
doesn't appear in /proc/asound/cards. However it doesn't hang or cause 
an oops either :) This is with the maudio quattro.



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Re: [Alsa-devel] USB devices should not oops after disconnect now

2002-11-29 Thread Patrick Shirkey
Jaroslav Kysela wrote:


Note that you'll have to terminate all applications using that soundcard
before you'll try to reconnect the device again. In next code, hopefully
the applications will be somehow terminated to make the cleanup process
automatic.




Ok. That's the probable cause as I am having problems with jack not 
releasing the device.

It seems that hotpluging out is a tricky issue because we have to decide 
whether the application/s that are using the device should also be 
terminated.

Is it possible to transparently swap snd-usb-audio with snd-dummy if it 
has been installed? That could be a secondary default if people would 
prefer their apps do not automatically segv.




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[Alsa-devel] kernel oops - snd-usb-audio

2002-11-26 Thread Patrick Shirkey
Advanced Linux Sound Architecture Driver Version 0.9.0rc6.
Compiled on Nov 15 2002 for kernel 2.4.19 with versioned symbols.

I got an oops while playing with jackd. This probably happened while 
trying to do duplex with all four channels at 44100hz. I'm trying to 
figure out how to get capture working for four channels instead of two. 
I was able to get jack to do playback only with them although I haven't 
tested to see if the streams are independant or copied. Last time I 
tried they were copied. ie. channel 12=34 respectively.

Here's the oops:


dmesg:

ALSA ../alsa-kernel/usb/usbaudio.c:1037: 4:8:4: cannot set freq 44100 to 
ep 0x86
ALSA ../alsa-kernel/usb/usbaudio.c:1037: 4:8:4: cannot set freq 44100 to 
ep 0x86
.
.
.
.
ALSA ../alsa-kernel/usb/usbaudio.c:1037: 4:8:4: cannot set freq 44100 to 
ep 0x86
ALSA ../alsa-kernel/usb/usbaudio.c:1037: 4:8:4: cannot set freq 44100 to 
ep 0x86
Unable to handle kernel NULL pointer dereference at virtual address 0098
 printing eip:
d8a27614
*pde = 
Oops: 
CPU:0
EIP:0010:[d8a27614]Tainted: P
EFLAGS: 00013246
eax:    ebx:    ecx: c5427480   edx: c5427480
esi:    edi: fffa   ebp: c5427680   esp: c94e7f30
ds: 0018   es: 0018   ss: 0018
Process jackd (pid: 2743, stackpage=c94e7000)
Stack: d671715c d7290c00 c5427480  d8a20ec9 c5427480  

   c5427680 4140 bf3ff8e4 0009 d8a22d3b c5427680 4140 
bf3ff8e4
   002f c01a9d0d d746c000 c94e6000 ca7ca100 c0115ad3 c94e7fbc 
c9fe9b80
Call Trace:[d8a20ec9] [d8a22d3b] [c01a9d0d] [c0115ad3] 
[c0147553]
  [c0109007]

Code: 8b 93 98 00 00 00 8b 02 83 f8 03 74 56 8b 02 83 f8 05 74 48
 6usb.c: USB disconnect on device 4
remove_proc_entry: dev/pcmC2D0p busy, count=1


=
cat /proc/ksyms|sort|less:


d8a25e00 snd_pcm_hw_constraint_ratdens_Red85c352[snd-pcm]
d8a25ec0 snd_pcm_hw_constraint_msbits_Rc5abfad7 [snd-pcm]
d8a25f50 snd_pcm_hw_constraint_step_R88603cc4   [snd-pcm]
d8a25fd0 snd_pcm_hw_constraint_pow2_R61dbf58b   [snd-pcm]
d8a260e0 _snd_pcm_hw_params_any_R6667da40   [snd-pcm]
d8a262a0 snd_pcm_hw_param_value_min_R132c8ec7   [snd-pcm]
d8a26330 snd_pcm_hw_param_value_max_R6d375137   [snd-pcm]
d8a263d0 _snd_pcm_hw_param_setempty_Rb661a635   [snd-pcm]
d8a26450 _snd_pcm_hw_param_setinteger_Rfe98f3f6 [snd-pcm]
d8a26650 snd_pcm_hw_param_first_R07c16536   [snd-pcm]
d8a26810 snd_pcm_hw_param_last_R466fc8c9[snd-pcm]
d8a268a0 _snd_pcm_hw_param_min_Ra3be0ddb[snd-pcm]
d8a26cb0 _snd_pcm_hw_param_set_R43ab6d9f[snd-pcm]
d8a26ff0 snd_pcm_hw_param_mask_R10ab0f83[snd-pcm]
d8a271e0 snd_pcm_hw_param_near_Rce7de4dd[snd-pcm]
d8a27720 snd_pcm_lib_ioctl_R73f9a607[snd-pcm]
d8a27770 snd_pcm_playback_ready_R46350c5e   [snd-pcm]
d8a277d0 snd_pcm_capture_ready_R3d59b3c6[snd-pcm]
d8a27810 snd_pcm_playback_data_R2f8436e5[snd-pcm]
d8a278d0 snd_pcm_capture_empty_R21832c9f[snd-pcm]
d8a27bf0 snd_pcm_period_elapsed_R0bcb3f9f   [snd-pcm]
d8a28170 snd_pcm_lib_write_Rd40be517[snd-pcm]
d8a28370 snd_pcm_lib_writev_Rc0e2f4bb   [snd-pcm]
d8a28870 snd_pcm_lib_read_R93425d33 [snd-pcm]
d8a28a40 snd_pcm_lib_readv_R5b4bc6ed[snd-pcm]
d8a28b80 snd_pcm_lib_buffer_bytes_R6065543e [snd-pcm]
d8a28ba0 snd_pcm_lib_period_bytes_Rca6f41e2 [snd-pcm]
d8a28f10 snd_pcm_format_signed_Re2c5f868[snd-pcm]
d8a28f40 snd_pcm_format_unsigned_R95a60aa5  [snd-pcm]
d8a28f60 snd_pcm_format_linear_R8733f967[snd-pcm]
d8a28f80 snd_pcm_format_little_endian_R8bbdc311 [snd-pcm]
d8a28fb0 snd_pcm_format_big_endian_R93878ece[snd-pcm]



At the moment I cannot unload the modules or access the usb device 
(maudio quattro).

I can still use other cards on my system but the quattro is completely 
unresponsive, even using the direct monitor switch.


--
Patrick Shirkey - Boost Hardware Ltd.
For the discerning hardware connoisseur
Http://www.boosthardware.com
Http://www.djcj.org - The Linux Audio Users guide


Being on stage with the band in front of crowds shouting, Get off! No! 
We want normal music!, I think that was more like acting than anything 
I've ever done.

Goldie, 8 Nov, 2002
The Scotsman



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[Alsa-devel] snd-azt3328 bug.

2002-11-24 Thread Patrick Shirkey
I have forwarded this on to hte alsa-devel list as this is a bug. Please 
sign up to the alsa-devel mailing list to help get this driver working 
correctly.


i do it by 
http://alsa-project.org/alsa-doc/doc-php/template.php3?company=AzTechamp;card=PCI+168amp;chip=AZF3328amp;module=azt3328
why i use  alsa-0.9.0rc6   driver   and
  when run to

[root@localhost root]# modprobe snd-azt3328
/lib/modules/2.4.18-14/kernel/sound/pci/snd-azt3328.o: unresolved symbol 
snd_mpu401_uart_interrupt_R4ea4dc60
/lib/modules/2.4.18-14/kernel/sound/pci/snd-azt3328.o: unresolved symbol 
snd_mpu401_uart_new_R0588a8ba
/lib/modules/2.4.18-14/kernel/sound/pci/snd-azt3328.o: insmod 
/lib/modules/2.4.18-14/kernel/sound/pci/snd-azt3328.o failed
/lib/modules/2.4.18-14/kernel/sound/pci/snd-azt3328.o: insmod 
snd-azt3328 failed


--
Patrick Shirkey - Boost Hardware Ltd.
For the discerning hardware connoisseur
Http://www.boosthardware.com
Http://www.djcj.org - The Linux Audio Users guide


Being on stage with the band in front of crowds shouting, Get off! No! 
We want normal music!, I think that was more like acting than anything 
I've ever done.

Goldie, 8 Nov, 2002
The Scotsman
---BeginMessage---
The following has been submitted as a note for the documentation of azt3328. Save this 
as /alsa-doc/additions/azt3328.php3. If there is not a file already you may need to 
use the template. Don't forget to add any new pages to cvs.

 cvs add www/alsa-doc/additions/azt3328.php3


/td/tr
trtd align=center

!-- Use the above if there are multiple entries in a file --
table border=0 cellpadding=4 cellspacing=0 width=620
tr valign=top
td bgcolor=#bed1be colspan=2
table border=0 cellpadding=2 cellspacing=0 width=620
tr valign=toptd
lzshun   nbsp nbsp a href=mailto: 
[EMAIL PROTECTED][EMAIL PROTECTED]/abr
Monday, 25 November 2002/td
td align=right
br/td
/tr
tr bgcolor=#f0f0f0td colspan=2
pre
  Patrick Shirkey

i do it by 
http://alsa-project.org/alsa-doc/doc-php/template.php3?company=AzTechamp;card=PCI+168amp;chip=AZF3328amp;module=azt3328
why i use  alsa-0.9.0rc6   driver   and
  when run to
[root@localhost root]# modprobe snd-azt3328
/lib/modules/2.4.18-14/kernel/sound/pci/snd-azt3328.o: unresolved symbol 
snd_mpu401_uart_interrupt_R4ea4dc60
/lib/modules/2.4.18-14/kernel/sound/pci/snd-azt3328.o: unresolved symbol 
snd_mpu401_uart_new_R0588a8ba
/lib/modules/2.4.18-14/kernel/sound/pci/snd-azt3328.o: insmod 
/lib/modules/2.4.18-14/kernel/sound/pci/snd-azt3328.o failed
/lib/modules/2.4.18-14/kernel/sound/pci/snd-azt3328.o: insmod snd-azt3328 failed

amp;#12288;amp;#12288;amp;#12288;amp;#12288;amp;#12288;amp;#12288;amp;#12288;amp;#12288;[EMAIL PROTECTED]
amp;#12288;amp;#12288;amp;#12288;amp;#12288;amp;#12288;amp;#12288;amp;#12288;amp;#12288;amp;#12288;amp;#12288;2002-11-23

  please tell me how can i use my soundcard work  under linux 
 THANKS

amp;#27492;
/pre
/td/tr
/table
/td
/tr
/table




---End Message---


Re: [Alsa-devel] HDSP Line Out Volume Control?

2002-11-21 Thread Patrick Shirkey
Thomas Charbonnel wrote:


I asked Paul once about the matrix mixer.
His answer was :

quote
first value: input source
second value: output source
third value: gain

input source: 0-25  (physical channels), 26-51 (audio output streams)
output source: 0-27 (physical channels)

output channels 26 and 27 are the two line outs on the I/O box.
/quote

As Michael is stating, full volume (100% in alsamixer, or full 65536
value in the matrix mixer) causes distortion. 50%, or 32786, seems to be
the highest distortion-free value. I already mentioned this on the list
but got no answer.

I succesfully tested the matrix mixer, using both amixer and the alsa
mixer api. Here's an amixer example with the values you picked above :
amixer cset numid=5 26,26,32786
amixer cset numid=5 27,27,32786


This is the kind of information that should be submitted as a note to 
the alsa-docs. If you submit it then people will associate your name 
with the info.

Currently not one person has provided anything for the advanced multi 
channel devices :(


--
Patrick Shirkey - Boost Hardware Ltd.
For the discerning hardware connoisseur
Http://www.boosthardware.com
Http://www.djcj.org - The Linux Audio Users guide


Being on stage with the band in front of crowds shouting, Get off! No! 
We want normal music!, I think that was more like acting than anything 
I've ever done.

Goldie, 8 Nov, 2002
The Scotsman



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