[Alsa-devel] how to cross compile the alsa util?

2004-05-26 Thread Roc Wu
Hello alls:

I cross compiled the alsa-lib-1.04 to arm platform.
The 2.6.6 kernel including alsa driver is ok on our
ARM board, so I want to test the driver. how to do it?

I think maybe the first step is to crosscompile the
alsa utils to play a wave audio file.

CC=arm-linux-gcc ./configure --host=i686-linux
--target=arm-linux
make
make install

then compiled and installed the libasound.*/ asound.h
to /usr/lib and /usr/include/sys.
Then I try to cross compile the alsa-utils:

CC=arm-linux-gcc ./configure --host=arm-linux
--target=arm-linux

checking for libasound headers version >= 0.9.0... not
present.
configure: error: Sufficiently new version of
libasound not found.

So how to make it and is there any simple method to
test the alsa?

Thanks a lot
Best Regards

Bryan Wu


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RE: [Alsa-devel] ATI IXP SPDIF

2004-05-26 Thread Alex Song
hi,

i happened to have that ASUS mobo (P4R800-VM) which has an ATIIXP/AD1888 and
both PCM and ac3 pass through works with SPDIF.
but the board i am trying to get working has an ATIIXP/ALC655 and i had a
look at the realtek site and they had some drivers
(http://www.realtek.com.tw/downloads/dlac97-2.aspx?lineid=5&famid=12&series=
8&Software=True) which looks like they were based off alsa-1.0.4. just doing
a quick diff i see that they added some SPDIF stuff amongst other things.
after testing and some code diffing this is what i figured:

alsa-1.0.4  broken atiixp-spdif broken alc655-spdif
realtek-alsa-1.0.4  broken atiixp-spdif working alc655-spdif
alsa-cvsworking atiixp-spdifbroken alc655-spdif

i am going to try and hack together some combination of realtek-alsa-1.0.4
and alsa-cvs and see if what i figured is right or not. it would be great if
one of the alsa developers can look into realtek-alsa-1.0.4 and merge those
changes into alsa.

cheers,

alex

> -Original Message-
> From: Takashi Iwai [mailto:[EMAIL PROTECTED]
> Sent: Tuesday, May 25, 2004 2:45 AM
> To: Alex Song
> Cc: '[EMAIL PROTECTED]'
> Subject: Re: [Alsa-devel] ATI IXP SPDIF
> 
> 
> At Mon, 24 May 2004 16:16:42 -0700,
> Alex Song wrote:
> > 
> > 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?
> 
> it works fine on my test machine.  it's an ASUS mobo (forgot the
> model).
> 
>   % ac3dec -C some-48k.ac3
> 
> or
> 
>   % aplay -Dplug:spdif foo.wav
> 
> but the direct SPDIF mode doesn't work...
> 
> --
> Takashi Iwai <[EMAIL PROTECTED]>  ALSA Developer - 
> www.alsa-project.org
> 


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Re: [Alsa-devel] RME 9632 Mixer...

2004-05-26 Thread Thomas Charbonnel
Ooops :
> for out_left in $(seq 0 2 16);
should be :
for out_left in $(seq 0 2 14);

Thomas







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Re: [Alsa-devel] RME 9632 Mixer...

2004-05-26 Thread Thomas Charbonnel
> Ed Wildgoose wrote:
>> Can anyone suggest how to script the controls to default to some known
>> values?
>
> "alsactl store" saves the current values of all mixer control in
> /etc/asound.state, "alsactl restore" restores them.
>
>
> HTH
> Clemens
>
>

 Hi Clemens, hi Ed,

This is not a standard mixer control, alsactl is of no use here.
Ed, here's a script for 1:1 routing (hdspmixer preset 1)

#! /bin/bash

for out_left in $(seq 0 2 16);
do
let out_right=$out_left+1
let in_left=$out_left+16
let in_right=$out_right+16
amixer cset numid=5 $in_left,$out_left,32768
amixer cset numid=5 $in_right,$out_right,32768
done;

This script assumes the hdsp is the default card, if it's not, just add
'-c X' (where X is your hdsp's alsa card number) between 'amixer' and
'cset'.

Thomas





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Re: [Alsa-devel] RME 9632 Mixer...

2004-05-26 Thread Clemens Ladisch
Ed Wildgoose wrote:
> Can anyone suggest how to script the controls to default to some known
> values?

"alsactl store" saves the current values of all mixer control in
/etc/asound.state, "alsactl restore" restores them.


HTH
Clemens




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Re: [Alsa-devel] RME 9632 Mixer...

2004-05-26 Thread Patrice Tisserand
On Tue, 2004-05-25 at 21:54, Ed Wildgoose wrote:
> ...is driving me nuts.  Everytime I turn on the PC I need to find a PC 
> with SSH to adjust the mixer controls so that I can hear some sound (no 
> mouse and a tiny TV resolution on the box itelf)
> 
> Can anyone suggest how to script the controls to default to some known 
> values?  I'm using alsa from around kernel 2.6.5 ish (which I believe 
> Thomas said has had the alsactl options removed, certainly I can't 
> figure out how to make them work).  Pointers to relevant parts of the 
> driver appreciated even so that I can write something to set the mixers 
> to a nice default.
> 
Don't know if it will work with 2.6.5, but you can try to add a script
which call amixer after hdsploader.
You will find some information on hdsp routing at the following url:
http://pd.klingt.org/files/hdsp-howto.html#controlling_output_routing

Bye,
Patrice.
-- 
Knowledge belongs to humanity.



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Re: [Alsa-devel] weird xrun problem

2004-05-26 Thread Jaroslav Kysela
On Tue, 25 May 2004, Giuliano Pochini wrote:

> On Sun, 23 May 2004 11:21:41 +0200 (CEST)
> Jaroslav Kysela <[EMAIL PROTECTED]> wrote:
> 
> > On Sat, 22 May 2004, Giuliano Pochini wrote:
> >
> > > Period 0 is 6624 frames and Period 1 is 6592 frames long (32 frames
> > > shorter). I put some debug code and this is the output for a the playback
> > > test above:
> >
> > ALSA does not know about this. All period sizes must be equal.
> 
> I thought about this again. Are you sure all periods must be aqual ?  
> When I record or play something using unequal periods, sound is perfect.
> It means ALSA reads and writes the last (sometimes a lot) smaller period
> as well.

The driver must not acknowledge the settled period with the midlevel and 
application before the specified time. Of course, if you're running into
a rounding problem, you can acknowledge more periods at one time.
In other cases, it's not very practical to program hardware to use 
different period sizes, because you need more interrupts for processing.

Jaroslav

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


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[Alsa-devel] Re: [Alsa-dev]detecing if a device is already used by an other app

2004-05-26 Thread Måns Rullgård
Cournapeau David <[EMAIL PROTECTED]> writes:

> Hi there,
>
> I am currently using two computers which have a crappy intl8x0
> audio chipset, and these chipset cannot be used by two different
> apps in the same time. For example, if an audio app is using the
> soundcard, trying to launch xmms ( with alsa output plugin) gives
> an error message which says something like "check that no other
> app is blocking the device". But if I launch alsaplayer (alsa
> output plugin) instead of xmms, the app doesn't complain: it
> 'just' blocks on the snd_pcm_open call.
> Basically, after having looked at the source, is seems like this
> difference is coming from the flag in snd_pcm_open: if
> SND_PCM_NONBLOCK  is used, opening an already opened device
> failed, if SND_PCM_ASYNC is used, the app just blocks, until the
> other app releases the device.

This is correct.

> Is there a (simple) way to use the blocking call and detecting if
> the device is already opened ?

You can open it in non-blocking mode and set it to blocking later, if
you want that behavior when writing data.

-- 
Måns Rullgård
[EMAIL PROTECTED]



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[Alsa-devel] [Alsa-dev]detecing if a device is already used by an other app

2004-05-26 Thread Cournapeau David
Hi there,
   I am currently using two computers which have a crappy intl8x0 audio 
chipset, and these chipset cannot be used by two different apps in the 
same time. For example, if an audio app is using the soundcard, trying 
to launch xmms ( with alsa output plugin) gives an error message which 
says something like "check that no other app is blocking the device". 
But if I launch alsaplayer (alsa output plugin) instead of xmms, the app 
doesn't complain: it 'just' blocks on the snd_pcm_open call.
   Basically, after having looked at the source, is seems like this 
difference is coming from the flag in snd_pcm_open: if SND_PCM_NONBLOCK  
is used, opening an already opened device failed, if SND_PCM_ASYNC is 
used, the app just blocks, until the other app releases the device.

   Is there a (simple) way to use the blocking call and detecting if 
the device is already opened ?

   Thank you,
   David
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[Alsa-devel] [PATCH] Gateway M675 tweak

2004-05-26 Thread Kevin Mack
Nobody complained, but this will direct mixer output
to speaker, headphone and line-out instead of just
the front(DAC-A) signal.


--- ../alsa-driver-1.0.5rc1/alsa-kernel/pci/ac97/ac97_patch.c   2004-05-24 
07:24:42.0 -0600
+++ alsa-kernel/pci/ac97/ac97_patch.c   2004-05-26 00:12:39.0 -0600
@@ -648,7 +648,7 @@
/* VARIOUS */ 0x0040
};
static unsigned short m675_regs[4] = {
-   /* OUTSEL */ 0x9040, /* CL:FR, SR:FR, LO:DS, LI:FR, MI:DS */
+   /* OUTSEL */ 0xfc70, /* CL:MX, SR:MX, LO:DS, LI:MX, MI:DS */
/* IOMISC */ 0x2102, /* HP amp on */
/* INSEL */ 0x0203, /* LI:LI, MI:FR */
/* VARIOUS */ 0x0041 /* stereo mic */


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