Re: [Alsa-devel] direct midi event dispatching

2004-05-24 Thread Clemens Ladisch
Gilles Dégottex wrote:
 Why SND_SEQ_EVENT_NOTE event doesnt accept direct dispatching ?
 technicaly, there is surely a reason, but conceptualy, I dont see. Why do a
 difference ?

SND_SEQ_EVENT_NOTE isn't a 'real' event but gets split into two
events, SND_SEQ_EVENT_NOTE_ON and _OFF, and the latter must be sent at
a later time.  Scheduling events to be sent later requires a timer
which isn't avilable without a queue.

The SND_SEQ_EVENT_NOTE event is the only event that uses the
event.data.note.duration field, but you cannot know what the value in
this field means without having a timer which specifies how long one
timer tick lasts.


HTH
Clemens





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[Alsa-devel] [PATCH] em10k1x fixes

2004-05-24 Thread Martin Langer

Hi,

here is a patch with some emu10k1x fixes:

- it adds emu10k1x to the Makefile
- include sound/info.h added
- move snd_iprintf to the right position; it compiles now
- fix typo in snd_iprintf

bye,
martin

-- 
 Living on earth may be expensive,
   but it includes an annual free trip around the sun. 
Index: alsa-driver/pci/Makefile
===
RCS file: /cvsroot/alsa/alsa-driver/pci/Makefile,v
retrieving revision 1.24
diff -u -r1.24 Makefile
--- alsa-driver/pci/Makefile8 Apr 2004 16:38:15 -   1.24
+++ alsa-driver/pci/Makefile24 May 2004 15:20:02 -
@@ -11,7 +11,7 @@
 
 clean-files := intel8x0.c
 
-obj-$(CONFIG_SND) += pdplus/
+obj-$(CONFIG_SND) += pdplus/ emu10k1/
 
 snd-hdspm-objs := hdspm.o
 
Index: alsa-driver/pci/emu10k1/emu10k1x.c
===
RCS file: /cvsroot/alsa/alsa-driver/pci/emu10k1/emu10k1x.c,v
retrieving revision 1.1
diff -u -r1.1 emu10k1x.c
--- alsa-driver/pci/emu10k1/emu10k1x.c  14 May 2004 13:42:49 -  1.1
+++ alsa-driver/pci/emu10k1/emu10k1x.c  24 May 2004 15:20:26 -
@@ -33,6 +33,7 @@
 #include sound/initval.h
 #include sound/pcm.h
 #include sound/ac97_codec.h
+#include sound/info.h
 
 MODULE_AUTHOR(Francisco Moraes [EMAIL PROTECTED]);
 MODULE_DESCRIPTION(EMU10K1X);
@@ -744,10 +745,11 @@
   snd_info_buffer_t * buffer)
 {
emu10k1x_t *emu = snd_magic_cast(emu10k1x_t, entry-private_data, return);
-   snd_iprintf(buffer, Registers:\n\n);
unsigned long value,value1,value2;
unsigned long flags;
int i;
+
+   snd_iprintf(buffer, Registers:\n\n);
for(i = 0; i  0x40; i+=4) {
spin_lock_irqsave(emu-emu_lock, flags);
value = inl(emu-port + i);
@@ -759,7 +761,7 @@
value = snd_emu10k1x_ptr_read(emu, i, 0);
value1 = snd_emu10k1x_ptr_read(emu, i, 1);
value2 = snd_emu10k1x_ptr_read(emu, i, 2);
-   snd_iprintf(buffer, %02X: %08lX %08lX %08lX\n, i, value, value, 
value2);
+   snd_iprintf(buffer, %02X: %08lX %08lX %08lX\n, i, value, value1, 
value2);
}
 }
 


Re: [Alsa-devel] [PATCH] em10k1x fixes

2004-05-24 Thread Takashi Iwai
At Mon, 24 May 2004 19:00:54 +0200,
Martin Langer wrote:
 
 Hi,
 
 here is a patch with some emu10k1x fixes:
 
 - it adds emu10k1x to the Makefile

it's not needed.  emu10k1 is already in alsa-kernel/pci/Makefile.

 - include sound/info.h added
 - move snd_iprintf to the right position; it compiles now
 - fix typo in snd_iprintf

applied to cvs.


thanks,

Takashi


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Re:[Alsa-devel] line in not supported Audigy and Audigy2 ?

2004-05-24 Thread Peter Zubaj
Hi,

If you think Line In on back side of card then:

try to unmute and set volume for
Line In
Analog Mix (best 100 %)
Front (best 100 %)
Master (best 100 %)

I am sure that it works on audigy 1.

Peter Zubaj




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[Alsa-devel] 1.0.5rc1 release

2004-05-24 Thread Jaroslav Kysela
Hi all,

ALSA 1.0.5rc1 is out. Please, report (especially 
compilation) problems.

Thanks,
Jaroslav

-
Jaroslav Kysela [EMAIL PROTECTED]
Linux Kernel Sound Maintainer
ALSA Project, SuSE Labs


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[Alsa-devel] ATI IXP SPDIF

2004-05-24 Thread Alex Song
hi,

i am trying to get SPDIF output working on ATI IXP/Realtek ALC655 and i am
using atiixp.c from cvs (version 1.9) and linux kernel 2.6.5. i tried
playing back wav files through aplay and ac3 files with ac3play and i
couldn't get any output from either SPDIF over AC97 or SPDIF Direct. analog
output is fine. from the cvs logs i gather that SPDIF on ATI IXP should work
for some cases at least, has anyone got it to work? if so with what
files/hardware/settings etc?

i am not very familiar with sound programming or sound card driver
operation, but i am willing to help if needed.

cheers,

alex

=
Alex Song
Software Engineer
XVD Corporation
44061 Nobel Drive
Fremont, CA 94538, USA
Work: (510) 770-5700 x266
Fax: (510) 770-5769
Email: [EMAIL PROTECTED]
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[Alsa-devel] Advanced 5.1 + DMIX configuration problems...

2004-05-24 Thread Bradee-oh!
I've posted the following message on the alsa-user list a few times and 
gotten no response.  Especially since it's a fairly complicated setup 
and probably a rare one that could reveal certain bugs that wouldn't be 
found otherwise, I thought I'd give the development list a shot so the 
people who know the nitty-gritty under the hood stuff can comment

-
I first started experimenting with the stereo - 5.1 configuration and
got it to work great!
Then in the interests of everyone sharing the card, I fiddled around
with dmixing the new 5.1 device and succeeded - now stereo apps are
upmixed to 5.1, share with each other, AND share with real 5.1 apps.
Then I tried to throw OSS into the mix using AOSS.  Here's where my
problem is...
A stereo OSS app DOES work in the configuration - for a very short
amount of time.  After a variable amount of time (1 to 5 minutes,
roughly) the OSS app will be cruising along and then the next time it
tries to play a sound, it will freeze.  Not because the sound card is
blocked, because it happens when nothing else is accessing the card.
But, in the process of freezing, it does block the sound card as nothing
else (ALSA or OSS) can use it until I kill the OSS app.
After it happens once, it seems to happen very quickly every subsequent
time until I reboot.  By very quickly, I mean the OSS app may get one or
two sounds off before it freezes 10 seconds into execution.
Here is the asound.conf I am using -
pcm.!default{
  type plug
  slave.pcm dsp0
}
pcm.dsp0{
  type plug
  slave {
pcm mixed51
channels 6
rate 48000
  }
  ttable.0.0 1
  ttable.1.1 1
  ttable.0.2 1
  ttable.1.3 1
  ttable.0.4 0.2
  ttable.1.4 0.2
  ttable.0.5 0.5
  ttable.1.5 0.5
}
pcm.mixed51 {
  type dmix
  ipc_key 1099234
  ipc_key_add_uid TRUE
  slave {
pcm hw:0,1
rate 48000
channels 6
period_time 0
period_size 1024
buffer_time 0
buffer_size 4096
  }
}
ctl.mixer0 {
  type hw
  card 0
}
My soundcard is a CMIPCI (8738) and I am running ALSA 1.04_rc2, as
included in kernel 2.6.5.  Notice the hardware device is hw:0,1 as that
is the 5.1 device for this card.
The way I use this setup is as follows -
-OSS apps using AOSS obviously route directly to dsp0.  That upmixes the
stereo to 5.1 then uses mixed51 to dmix it.
-Stereo ALSA apps using the default PCM get rerouted to dsp0, or
alternately I tell them specifically to use dsp0
-5.1 ALSA apps I tell to use mixed51 directly as they don't need the
stereo-5.1 hack but DO need to share the card via dmix
Any explanations or solutions are MUCH appreciated - even if the 
official developer explanation is don't do that, it doesn't work   tia

-Brady
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[Alsa-devel] emu10k1x patch

2004-05-24 Thread [EMAIL PROTECTED]
A few minor fixes to the emu10k1x driver. Also, it would be nice to update the 
soundcards page to list it as supported (mention that only PCM playback is currently 
supported).

I will try to test a few more things before 1.0.5 gets final, I hope.

Francisco
Index: pci/emu10k1/emu10k1x.c
===
RCS file: /cvsroot/alsa/alsa-driver/pci/emu10k1/emu10k1x.c,v
retrieving revision 1.2
diff -r1.2 emu10k1x.c
185c185
   .channels_min = 1,
---
   .channels_min = 2,
188,189c188,189
   .period_bytes_min = 64,
   .period_bytes_max = (32*1024),
---
   .period_bytes_min = (16*1024),
   .period_bytes_max = (16*1024),
732,733c732,733
   //  snd_emu10k1x_intr_enable(chip, (INTE_CH_0_LOOP1);
   //  snd_emu10k1x_intr_enable(chip, INTE_CH_0_LOOP2);
---
   snd_emu10k1x_intr_enable(chip, (INTE_CH_0_LOOP|INTE_CH_0_HALF_LOOP)1);   snd_emu10k1x_intr_enable(chip, (INTE_CH_0_LOOP|INTE_CH_0_HALF_LOOP)2);