Advanced Linux Sound Architecture

>....
>in addition to IEC958 Playback Switch, there is a control which
>configures the IEC958 status bits.  these bits give some important
>information such as sample rate, transfer mode, etc.
>
>the problem is that they are represented in byte arrays and pretty
>cryptic for end-users.  so, i wrote a small utility to dump and modify
>these bits:
>         http://www.alsa-project.org/~iwai/iecdump-0.1.tar.bz2
>or
>         ftp://ftp.suse.com/pub/people/tiwai/misc/iecdump-0.1.tar.bz2
>
>(the latter will appear later due to sync)
>
>you can get the current status via iecdump and modify with iecset.
>you'll need to set up at least the correct sample rate to play back a
>file.
>
>(also don't forget to turn on "IEC958 Playback Switch" in addition.)

Thank you very much. I will try them immediately.


Dennis

Your problem is eerily similar to my experience with recent versions of 
ALSA with the FlightGear and TORCS simulations. The problem does appear 
related to interrupt handling in ALSA.  With my hardware configuration, 
ALSA 0.9.5 still has this problem.

hi,

Firstly, please excuse my ignorance of the alsa sound system and now:

I'm trying to route an ac3 stream from my RME's optical in through ac3dec=
 and=20
then out the analogue ports of the same RME but am having little success.
i know the optical in works, the ac3 stream coming in is heard as noise f=
rom=20
my analogue outs (using amixer cset to set the input), what i need is to=20
essentially pipe the incoming stream first into ac3dec and not my outs.

any help in this regard (pointers to other resources, mixer setups, confi=
g=20
files asoundrc, asound.state) would be of great help.

thanks,
Ashendra Singh

Hello! 

Surely, I did unmute all the items I found in alsamixer.
This problem never happened on other types of sound chipset, 
and everything works fine even on other SiS7012 devices (by pci id) 
with another sound chipsets.

That's that confusing me a lot :(

В ?? 09.07.2003, в 19:18, Weber, Geoffrey M. пишет:
> Hello Michael:
> 
> May be something way too simple, but did you go into the mixer and un-mute
> the relevant items?  Just thought I'd check...
> 
>     Geoff Weber
>     gweber@...
> 
> 
> -----Original Message-----
> From: Mikhail Kouznetsov [mailto:mkuzn@...]
> Sent: Wednesday, 09 July, 2003 09:59
> To: alsa-users mail list
> Subject: [Alsa-user] Problem with alsa-0.9.2 (kernel-2.4.20) on ADI
> AD1980 sound chipset
> 
> 
> 
> Hello there!
> 
> I've experienced troubles with at least two ASUS motherboards: P4S533-X and
> P4S533-MX.
> Both built on SiS chipsets (651/645DX), and both have SiS7012 audio
> controller (as detected by pci)
> on ADI AD1980 sound chipset.
> The problem is in full sound absence, while the alsa drivers (snd-intel8x0,
> snd-*-oss, etc) loads
> perfectly and all mixers (alsamixer, aumix) works fine. The most strange
> thing is that sound playing 
> programs works like everything is ok, i.e. their seems to be _really_ play
> sound files with no any sound 
> available in the headphones at all.. :(
> 
> Could someone hint me on the possible solution?..
> Is there any probability this problem is solved in alsa 0.9.4? 
> Btw, I didn't find detailed changelog between 0.9.2 and 0.9.4 on the web. Is
> it available?
> 
> Thanks in advance! 
-- 
Michael Kouznetsov
ASPLinux Developers Team
http://www.asplinux.ru
mkuzn@...

Dear ALSA developers,

Ok, I fixed the previous problem; now running the
latest kernel + latest modules (and alsa 0.9.4).

The REAL problem I have is the following:

Recording has always been extremely hard to impossible
with linux - one of the applications that I REALLY
cannot get working is the speech recognition tool of
IBM: ViaVoice.

I downloaded the commercial sound drivers from Front4
and with that it works perfectly.  In the past I've also
got this to work with the OSS/Free stuff.

However... with ALSA, it has never has worked; and I've tried
a lot of things :(.

I can't see this anymore being anything else but a bug
in ALSA.  The problem however is to find WHY it doesn't
work (and then fix it).

I'd like to get some guidance with that.

My setup is as follows:

1) It works (recording speech with microphone) when using
   the Front4 commercial sound drivers.
2) With ALSA, playback works and I hear myself back over
   the speakers when I talk into the microphone.  I even
   can let it 'sing around' by capturing Mic (of course) and
   turning on max. "AC97" and "Capture == Igain".  Also
   "AC97 Capture" was set to the maximum.  In fact, I turned
   EVERYTHING on to the maximum with alsamixer, unmuted
   everything and captured Mic [1]... No recording.

I am willing to get to the bottom of this, but I am out
of ideas right now :/

Please help.

-- 
Carlo Wood <carlo@...>

Hello,

	I'm new to Alsa development. I'm porting a library that works fine
with MacOS CoreAudio and DirectSound, and that need to play severals =
sounds
at the same time, with severals format, number of channels, and sample =
rate.
	I've successfully been able to play one sound using the
snd_pcm_open("plughw:0,0"...) and snd_pcm_writei, but if I want to play
another sound, snd_pcm_open hangs and I'm unable to paly another sound. =
I
though Alsa was responsible for mixing and hardware channel allocation ?

	I have a via_82xx Alsa driver, but it does not seems to handle
severals pcm playback at the same time : I cannot open device with name
"plughw:1,0","plughw:0,1",etc...

	And, latest problem : Under gnome, when opening a fileselector
and/or clicking in a terminal, playback stops without any reason.

	Any help would be really appreciated :-)

	Regards,

David

Hello there!

I've experienced troubles with at least two ASUS motherboards: P4S533-X and P4S533-MX.
Both built on SiS chipsets (651/645DX), and both have SiS7012 audio controller (as detected by pci)
on ADI AD1980 sound chipset.
The problem is in full sound absence, while the alsa drivers (snd-intel8x0, snd-*-oss, etc) loads
perfectly and all mixers (alsamixer, aumix) works fine. The most strange thing is that sound playing 
programs works like everything is ok, i.e. their seems to be _really_ play sound files with no any sound 
available in the headphones at all.. :(

Could someone hint me on the possible solution?..
Is there any probability this problem is solved in alsa 0.9.4? 
Btw, I didn't find detailed changelog between 0.9.2 and 0.9.4 on the web. Is it available?

Thanks in advance! 

-- 
Michael Kouznetsov
ASPLinux Developers Team
http://www.asplinux.ru
mkuzn@...

I downloaded alsa-driver-0.9.5.tar.bz2

Then I did:

tar -xjf alsa-driver*.tar.bz2
cd alsa-driver-*
./configure --with-isapnp=no --with-cards=emu10k1 --with-sequencer=yes
make
sudo make install
sudo chmod 666 /dev/dsp* /dev/mixer* /dev/sequencer* /dev/midi*
cd ..

which was succesful.


Finally, I try to load the modules but get:

/usr/src/sound/alsa>modprobe snd-emu10k1
WARNING: Error inserting snd (/lib/modules/2.5.74-debug/kernel/sound/acore/snd.o): Invalid module format
WARNING: Error inserting snd_hwdep (/lib/modules/2.5.74-debug/kernel/sound/acore/snd-hwdep.o): Invalid module format
WARNING: Error inserting snd_util_mem (/lib/modules/2.5.74-debug/kernel/sound/synth/snd-util-mem.o): Invalid module format
WARNING: Error inserting snd_page_alloc (/lib/modules/2.5.74-debug/kernel/sound/acore/snd-page-alloc.o): Invalid module format
WARNING: Error inserting snd_ac97_codec (/lib/modules/2.5.74-debug/kernel/sound/pci/ac97/snd-ac97-codec.o): Invalid module format
WARNING: Error inserting snd_seq_device (/lib/modules/2.5.74-debug/kernel/sound/acore/seq/snd-seq-device.o): Invalid module format
WARNING: Error inserting snd_timer (/lib/modules/2.5.74-debug/kernel/sound/acore/snd-timer.o): Invalid module format
WARNING: Error inserting snd_pcm (/lib/modules/2.5.74-debug/kernel/sound/acore/snd-pcm.o): Invalid module format
WARNING: Error inserting snd_rawmidi (/lib/modules/2.5.74-debug/kernel/sound/acore/snd-rawmidi.o): Invalid module format
FATAL: Error inserting snd_emu10k1 (/lib/modules/2.5.74-debug/kernel/sound/pci/emu10k1/snd-emu10k1.o): Invalid module format


Does 0.9.5 not work with 2.5 kernels?

Of course I could use the ALSA kernel modules of
the 2.5 kernel - but that gives me another problem:
Then /dev/controlC0 was being accessed, but didn't
exist.  I assumed that was because the alsa tar balls
that I downloaded (0.9.4) were newer then the kernel
modules in 2.5.74.

Can anyone give me a hint?

Or is it just that kernel 2.5.74 compiles and installs
the modules but didn't create the (new) device nodes
that are needed?

-- 
Carlo Wood <carlo@...>

Ryan Underwood wrote:
> What is the lowest speed CPU that anyone has had success with the dmix
> plugin on?  It seems that even when only playing one stream, if the cpu
> is too slow, the sound cuts out.

My Pentium 200 can do five streams or so without any problems. I never
tested with more, so I don't know what the upper limit is.

What CPU do you have?

> Also, is there any way to prevent the OSS device from being blocked if
> the ALSA is being used?  The dmix plugin prevents alsa applications from
> blocking, but OSS applications still block if an ALSA application is in
> use.  [...]
> Could the same be done for ALSA, e.g. using a wrapper program against
> ALSA API instead of the OSS compatibility dsp device?

Yes. The aoss script from the alsa-oss package will redirect accesses
to /dep/dspX to an ALSA PCM device named dspX (which must be defined
in your .asoundrc).


HTH
Clemens

Well,

I figured this one out on my own.  The problem was
mentioned in the Xine FAQ and was caused by audio /
video syncing problems.  When I played a DVD, I got
continuous messages about corrections to audio / video
sync.  This caused dropouts and hence the warbling I
was hearing.

The solution turned out to be a sound card
/etc/asound.state setting.  I had both control 47
(Multi Track Rate Locking) and control 48 (Multi Track
Rate Reset) set to true.  Setting them to false solved
the synching and dropout problems.

I guess what was happening is that Xine was trying to
set a particular rate and since I had everything
locked it was doing resampling or something and
basically fighting with the sound card driver.

All appears well now,

- Tony -

__________________________________
Do you Yahoo!?
SBC Yahoo! DSL - Now only $29.95 per month!
http://sbc.yahoo.com

Greetings--
I've been trying for days to use an M-Audio (Midiman) Audiophile 2496
soundcard using the ALSA ice1712 driver. I've followed the instructions
for the driver on the ALSA web site, but something is still off, because
I can't hear any sound from the card. (And I have turned up all the
levels for the card using Alsamixer.) 

kernel: 2.4.20-18.9 (RedHat 9)
alsa version: 0.9.4
system: AMD Athlon 1.3Ghz (VIA KT133A chipset)

Here's the sound-related part of /etc/modules.conf:
#soundcard configuration
        # ALSA portion
        alias char-major-116 snd
        alias snd-card-0 snd-ice1712
	# module options should go here

        # OSS/Free portion
        alias char-major-14 soundcore
	
	# card #1
	alias sound-service-0-0 snd-mixer-oss
	alias sound-service-0-1 snd-seq-oss
	alias sound-service-0-3 snd-pcm-oss
	alias sound-service-0-8 snd-seq-oss
	alias sound-service-0-12 snd-pcm-oss

alias sound-slot-0 snd-ice1712
post-install sound-slot-0 /bin/aumix-minimal -f /etc/.aumixrc -L
>/dev/null 2>&1 || :
pre-remove sound-slot-0 /bin/aumix-minimal -f /etc/.aumixrc -S
>/dev/null 2>&1 || :

lsmod shows this:
Module                  Size  Used by    Not tainted
snd-ice1712            49576   0
snd-ak4xxx-adda         5428   0  [snd-ice1712]
snd-cs8427              7304   0  [snd-ice1712]
snd-ac97-codec         47720   0  [snd-ice1712]
snd-i2c                 4612   0  [snd-ice1712 snd-cs8427]
snd-mpu401-uart         5136   0  [snd-ice1712]
snd-pcm-oss            44836   0  (autoclean)
snd-mixer-oss          16184   0  (autoclean) [snd-pcm-oss]
snd-pcm                84480   0  [snd-ice1712 snd-pcm-oss]
snd-page-alloc          8388   0  [snd-pcm]
snd-seq-midi            5248   0  (unused)
snd-seq-midi-event      5672   0  [snd-seq-midi]
snd-seq                46864   0  [snd-seq-midi snd-seq-midi-event]
snd-timer              19716   0  [snd-pcm snd-seq]
snd-rawmidi            18496   0  [snd-mpu401-uart snd-seq-midi]
snd-seq-device          6204   0  [snd-seq-midi snd-seq snd-rawmidi]
snd                    43716   0  [snd-ice1712 snd-ak4xxx-adda
snd-cs8427 snd-ac97-codec snd-i2c snd-mpu401-uart snd-pcm-oss
snd-mixer-oss snd-pcm snd-seq-midi snd-seq-midi-event snd-seq snd-timer
snd-rawmidi snd-seq-device]
soundcore               6468   5  [snd]

modinfo snd-ice1712 shows this:
filename:   
/lib/modules/2.4.20-18.9/kernel/sound/pci/ice1712/snd-ice1712.o
description: "ICEnsemble ICE1712 (Envy24)"
author:      "Jaroslav Kysela <perex@...>"
license:     "GPL"
parm:        index int array (min = 1, max = 8), description "Index
value for ICE1712 soundcard."
parm:        id string array (min = 1, max = 8), description "ID string
for ICE1712 soundcard."
parm:        enable int array (min = 1, max = 8), description "Enable
ICE1712 soundcard."
parm:        omni int array (min = 1, max = 8), description "Enable
Midiman M-Audio Delta Omni I/O support."

modinfo soundcore shows this:
filename:    /lib/modules/2.4.20-18.9/kernel/drivers/sound/soundcore.o
description: "Core sound module"
author:      "Alan Cox"
license:     "GPL"

Soundcard detection from the System Settings menu lists the card--
Model: ICE1712
Module: snd-ice1712
But when I click "Play sample sound" I get nothing.

The card works in Windoh's, but no dice in Linux.

I've studied a lot of Linux sound lists and web sites, but still haven't
figured out what I'm doing wrong. Any help will be appreciated.

Thanks,
Tom

Hello. I am assembling a Digital Audio Workstation with a computer I
built recently. It's a 1.8Ghz Athlon on a VIA mobo running Gentoo Linux
1.4. My audio card is the M-Audio Delta 1010-LT, which I bought
specifically because of it's ALSA support.

The version I have compiled is 0.9.2 from the gentoo ports system. And my
first impressions are terrible. Performance is aweful, with even the
slightest GUI action causing the sound to drop out when playing a .ogg
file from XMMS and ogg123.

I have my kernel compiled for low latency and preemptive threads, which I
assumed would improve performance. I followed the kernel building
instructions on this webpage:

http://myweb.cableone.net/eviltwin69/ALSA_JACK_ARDOUR.html

I know that others are having good success with their ALSA drivers. So I'm
not trying to flame this list. I really want them to work! I do have some
experience with programming, so I'd help this project in any way I can.

-l[e^2]
------------------------------
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people experimenting with music, art and technology