problems with rme digi96/8

problems with rme digi96/8

[Gorm David Lai]

Hi, I really hope somebody can help me.

We have recently bought a RME Digi96/8 card. This connects via adat to a
mixing console.
I have implemented the code in the ALSA 0.9.0 HOWTO for playback. This
works fine on a sound blaster card in stereo.
However with the digi96/8 card the mixing console does not get any
input. The mixing console syncs fine on the adat connection, it just
does not seem to get any sound input.

Any help, would be greatly appreciated as I am getting desperate.

Regards

Gorm - Programmer at Center for Advanced Visualization and Interaction



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Re: problems with rme digi96/8

[Martin Langer]

On Thu, Jun 05, 2003 at 12:17:32PM +0200, Gorm David Lai wrote:
> Hi, I really hope somebody can help me.
> 
> We have recently bought a RME Digi96/8 card. This connects via adat to a
> mixing console.
> I have implemented the code in the ALSA 0.9.0 HOWTO for playback. This
> works fine on a sound blaster card in stereo.
> However with the digi96/8 card the mixing console does not get any
> input. The mixing console syncs fine on the adat connection, it just
> does not seem to get any sound input.
> 
> Any help, would be greatly appreciated as I am getting desperate.
> 

ADAT isn't well tested, so everything is possible. But a look to wiki page
on http://opensrc.org/alsa/index.php?page=rme96 looks not so bad.

Are you using plughw:0,1 or hw:0,1? What's your alsa version? 


martin


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Re: problems with rme digi96/8

[Gorm David Lai]

On Thu, 2003-06-05 at 21:03, Martin Langer wrote:
> On Thu, Jun 05, 2003 at 12:17:32PM +0200, Gorm David Lai wrote:
> > Hi, I really hope somebody can help me.
> > 
> > We have recently bought a RME Digi96/8 card. This connects via adat to a
> > mixing console.
> > I have implemented the code in the ALSA 0.9.0 HOWTO for playback. This
> > works fine on a sound blaster card in stereo.
> > However with the digi96/8 card the mixing console does not get any
> > input. The mixing console syncs fine on the adat connection, it just
> > does not seem to get any sound input.
> > 
> > Any help, would be greatly appreciated as I am getting desperate.
> > 
> 
> ADAT isn't well tested, so everything is possible. But a look to wiki page
> on http://opensrc.org/alsa/index.php?page=rme96 looks not so bad.
> 
> Are you using plughw:0,1 or hw:0,1? What's your alsa version? 
> 
> 
> martin
> 
> 
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> 
Well, I test and set only legal settings on the hardware. So every
setting I set returns success.

I normally use hw:0,1, but just tried with plughw:0,1. Sadly with no
luck.
My legal settings do not follow those on the page. The period size seems
to match, but my only legal buffer size is 4096. This seems to be
contrary to the info on the page. I also use a sample  size of 16.

I am not the one who installed alsa, but I think it is 0.9.1. The aplay
version is 0.9.3.

/Gorm



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Re: problems with rme digi96/8

[Gorm David Lai]

I just thought of something new. It seems as if the hardware buffer on
the card gets filled up, but is never emptied anywhere.

The reason is that in the beginning everything seems to go fine, but
then I suddenly get the following error:

ALSA lib pcm_hw.c:466:(snd_pcm_hw_prepare) SNDRV_PCM_IOCTL_PREPARE
failed: Device or resource busy

This also happens on the soundblaster card, but there I can see that I
am still sometimes allowed to pass samples to the card, which makes the
sound play fine. On the rme digi96/8 card however, after this error
first happens, no more samples are allowed to be written.

Any idea on how to get the card to play the samples in the buffer? (if
this is indeed the error).

/Gorm


On Thu, 2003-06-05 at 21:03, Martin Langer wrote:
> On Thu, Jun 05, 2003 at 12:17:32PM +0200, Gorm David Lai wrote:
> > Hi, I really hope somebody can help me.
> > 
> > We have recently bought a RME Digi96/8 card. This connects via adat to a
> > mixing console.
> > I have implemented the code in the ALSA 0.9.0 HOWTO for playback. This
> > works fine on a sound blaster card in stereo.
> > However with the digi96/8 card the mixing console does not get any
> > input. The mixing console syncs fine on the adat connection, it just
> > does not seem to get any sound input.
> > 
> > Any help, would be greatly appreciated as I am getting desperate.
> > 
> 
> ADAT isn't well tested, so everything is possible. But a look to wiki page
> on http://opensrc.org/alsa/index.php?page=rme96 looks not so bad.
> 
> Are you using plughw:0,1 or hw:0,1? What's your alsa version? 
> 
> 
> martin
> 
> 
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> Alsa-devel@lists.sourceforge.net
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> 




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Re: problems with rme digi96/8

[Martin Langer]

On Fri, Jun 06, 2003 at 12:06:33PM +0200, Gorm David Lai wrote:
> On Thu, 2003-06-05 at 21:03, Martin Langer wrote:
> > On Thu, Jun 05, 2003 at 12:17:32PM +0200, Gorm David Lai wrote:
> > > 
> > > We have recently bought a RME Digi96/8 card. This connects via adat to a
> > > mixing console.
> > 
> > ADAT isn't well tested, so everything is possible. But a look to wiki page
> > on http://opensrc.org/alsa/index.php?page=rme96 looks not so bad.
> > 
> > Are you using plughw:0,1 or hw:0,1? What's your alsa version? 
> > 
> Well, I test and set only legal settings on the hardware. So every
> setting I set returns success.
> 
> I normally use hw:0,1, but just tried with plughw:0,1. Sadly with no
> luck.
> My legal settings do not follow those on the page. The period size seems
> to match, but my only legal buffer size is 4096. This seems to be
> contrary to the info on the page. I also use a sample  size of 16.
 
I read on this page:

For using ADAT, in case of 16 bit:
- 512 frames and 8 periods ( 512*8=4096 ) 
- 128 frames and 32 periods ( 128*32=4096 )

Why does it not match your 16bit case? Have you choosen spdif values instead?

But the best would be a view into the hardware description, which was under
ftp://ftp.alsa-project.org/documentation/rme/ IIRC. 


martin




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Re: problems with rme digi96/8

[Gorm David Lai]

On Mon, 2003-06-09 at 19:23, Martin Langer wrote:
> On Fri, Jun 06, 2003 at 12:06:33PM +0200, Gorm David Lai wrote:
> > On Thu, 2003-06-05 at 21:03, Martin Langer wrote:
> > > On Thu, Jun 05, 2003 at 12:17:32PM +0200, Gorm David Lai wrote:
> > > > 
> > > > We have recently bought a RME Digi96/8 card. This connects via adat to a
> > > > mixing console.
> > > 
> > > ADAT isn't well tested, so everything is possible. But a look to wiki page
> > > on http://opensrc.org/alsa/index.php?page=rme96 looks not so bad.
> > > 
> > > Are you using plughw:0,1 or hw:0,1? What's your alsa version? 
> > > 
> > Well, I test and set only legal settings on the hardware. So every
> > setting I set returns success.
> > 
> > I normally use hw:0,1, but just tried with plughw:0,1. Sadly with no
> > luck.
> > My legal settings do not follow those on the page. The period size seems
> > to match, but my only legal buffer size is 4096. This seems to be
> > contrary to the info on the page. I also use a sample  size of 16.
>  
> I read on this page:
> 
> For using ADAT, in case of 16 bit:
> - 512 frames and 8 periods ( 512*8=4096 ) 
> - 128 frames and 32 periods ( 128*32=4096 )
> 
> Why does it not match your 16bit case? Have you choosen spdif values instead?
> 
> But the best would be a view into the hardware description, which was under
> ftp://ftp.alsa-project.org/documentation/rme/ IIRC. 
> 
> 
> martin
> 
> 
I misunderstood the buffer size, I thought it was in kb not i frames.
Not I understand. Also from the documentation I read that the card has a
hardware buffer size of 64KB, which matches 4096 frames in 8 channels.
I have tried the hw:0,0 for spdif btw, but that doesn't work either.

Now I have also tried setting some of the sw_params. I seems the value
set by snd_pcm_sw_params_set_start_threshold affects things greatly.
What happens is that i can fill the buffer with samples, though at some
point I suddenly get an underrun, as if the entire buffer is emptied at
once. The value set by snd_pcm_sw_params_set_start_threshold
approximately determines how often this happens.

As said earlier these problems only occur with the rme card. 

I looked in the cards documentation, but I am not sure what I can use
most of this for, unless I want to bypass ALSA entirely.

/Gorm





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Re: problems with rme digi96/8

[Gorm David Lai]

I also have problems with aplay. Maybe this is related?

The following command:

aplay -D hw:0,1 -c 8 ~/cavi/sound/01/01\ Track\ \ 1.wav

Which gives the following output:

Playing WAVE '/users/lai/cavi/sound/01/01 Track  1.wav' : Signed 16 bit
Little Endian, Rate 44100 Hz, Stereo
aplay: set_params:810: Channels count non available


/Gorm



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Re: problems with rme digi96/8

[Gorm David Lai]

Ok. Sorry for spamming the list, but I got a hint that may point to
somewhere where someone can help me.

If I use aplay with plughw:0,1 as opposed to hw:0,1 aplay acts as if it
is playing, but no sound reaches the digital mixing console, though it
still seems to sync to the signal. This seems to be the same problem I
have in my code.

So maybe it is a configuration problem? 

/Gorm




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Re: problems with rme digi96/8

[Anders Torger]

Test this:

aplay -D hw:0,1 -r 44100 -f S16_LE -c 8 /dev/zero

it should work. It works on my card.

The problem in your 

aplay -D hw:0,1 -c 8 ~/cavi/sound/01/01\ Track\ \ 1.wav

is that aplay seems to ignore your -c 8, and reads the wav header 
instead, and sets it to stereo, as you can see in the aplay output, 
where it cleary says that it plays in "Stereo".

/Anders Torger

On Wednesday 11 June 2003 15.17, Gorm David Lai wrote:
> I also have problems with aplay. Maybe this is related?
>
> The following command:
>
> aplay -D hw:0,1 -c 8 ~/cavi/sound/01/01\ Track\ \ 1.wav
>
> Which gives the following output:
>
> Playing WAVE '/users/lai/cavi/sound/01/01 Track  1.wav' : Signed 16
> bit Little Endian, Rate 44100 Hz, Stereo
> aplay: set_params:810: Channels count non available
>
>
> /Gorm
>
>
>
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> Alsa-devel@lists.sourceforge.net
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Re: problems with rme digi96/8

[Gorm David Lai]

Thanks for the advice, though it still doesn't work. 
It seems to play, and sets the right sync on the mixing console. The
console just doesn't get any input.
Do I have to unmute the card anywhere?

/Gorm


On Wed, 2003-06-11 at 15:48, Anders Torger wrote:
> 
> Test this:
> 
> aplay -D hw:0,1 -r 44100 -f S16_LE -c 8 /dev/zero
> 
> it should work. It works on my card.
> 
> The problem in your 
> 
> aplay -D hw:0,1 -c 8 ~/cavi/sound/01/01\ Track\ \ 1.wav
> 
> is that aplay seems to ignore your -c 8, and reads the wav header 
> instead, and sets it to stereo, as you can see in the aplay output, 
> where it cleary says that it plays in "Stereo".
> 
> /Anders Torger
> 
> On Wednesday 11 June 2003 15.17, Gorm David Lai wrote:
> > I also have problems with aplay. Maybe this is related?
> >
> > The following command:
> >
> > aplay -D hw:0,1 -c 8 ~/cavi/sound/01/01\ Track\ \ 1.wav
> >
> > Which gives the following output:
> >
> > Playing WAVE '/users/lai/cavi/sound/01/01 Track  1.wav' : Signed 16
> > bit Little Endian, Rate 44100 Hz, Stereo
> > aplay: set_params:810: Channels count non available
> >
> >
> > /Gorm
> >
> >
> >
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> > best thread debugger on the planet. Designed with thread debugging
> > features you've never dreamed of, try TotalView 6 free at
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> > Alsa-devel@lists.sourceforge.net
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Re: problems with rme digi96/8

[Anders Torger]

On Wednesday 11 June 2003 16.03, Gorm David Lai wrote:
> Thanks for the advice, though it still doesn't work.
> It seems to play, and sets the right sync on the mixing console. The
> console just doesn't get any input.
> Do I have to unmute the card anywhere?

Note that my example uses /dev/zero (zero input) so it should be silent. 
Put something with sound in it instead of /dev/zero.

No, you shouldn't need to unmute the card (if you are not using analog 
output), the digital output is just digital, no volume control for it 
exists. A classical mistake is of course to connect ADAT input to ADAT 
input, but I suppose you have checked that.

/Anders



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Re: problems with rme digi96/8

[Gorm David Lai]

On Wed, 2003-06-11 at 16:15, Anders Torger wrote:
> 
> Note that my example uses /dev/zero (zero input) so it should be silent. 
> Put something with sound in it instead of /dev/zero.
> 
> No, you shouldn't need to unmute the card (if you are not using analog 
> output), the digital output is just digital, no volume control for it 
> exists. A classical mistake is of course to connect ADAT input to ADAT 
> input, but I suppose you have checked that.
> 
> /Anders
> 
I thought so about the digital output, but wanted to make sure.
I know the ADAT is set up right, because everything works in Windows XP.

I tried your example. Aplay is rather stupid, so it is seems it needs a
soundfile with a equal amount of channels that can be allocated on the
card; in this case 8.
As far as i know wave files only support stereo. So i tried feeding it
noise, like

aplay -D hw:0,1 -r 44100 -f S16_LE -c 8 ~/plg-second-printing-update.ps

aplay starts to play with the output:

Playing raw data '/users/lai/plg-second-printing-update.ps' : Signed 16
bit Little Endian, Rate 44100 Hz, Channels 8

But no sound. I am nearly giving up now...

/Gorm





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Re: problems with rme digi96/8

[Jaroslav Kysela]

On 12 Jun 2003, Gorm David Lai wrote:

> I tried your example. Aplay is rather stupid, so it is seems it needs a

Not too stupid, but you should use right device (plughw vs. hw) and 
probably right alsa-lib configuration (~/.asoundrc).

						Jaroslav

-----
Jaroslav Kysela <perex@suse.cz>
Linux Kernel Sound Maintainer
ALSA Project, SuSE Labs



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Re: problems with rme digi96/8

[Giuliano Pochini]

On 12-Jun-2003 Jaroslav Kysela wrote:
> On 12 Jun 2003, Gorm David Lai wrote:
>
>> I tried your example. Aplay is rather stupid, so it is seems it needs a
>
> Not too stupid, but you should use right device (plughw vs. hw) and
> probably right alsa-lib configuration (~/.asoundrc).

The format written in the sound file has priority over command
line options. It's a bit annoying and should be changed IMHO.


Bye.



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Re: problems with rme digi96/8

[Jaroslav Kysela]

On Thu, 12 Jun 2003, Giuliano Pochini wrote:

> 
> On 12-Jun-2003 Jaroslav Kysela wrote:
> > On 12 Jun 2003, Gorm David Lai wrote:
> >
> >> I tried your example. Aplay is rather stupid, so it is seems it needs a
> >
> > Not too stupid, but you should use right device (plughw vs. hw) and
> > probably right alsa-lib configuration (~/.asoundrc).
> 
> The format written in the sound file has priority over command
> line options. It's a bit annoying and should be changed IMHO.

No, you can explicitly change the format (if something is wrong) using
command line parameters. But my point was that aplay does not do any
conversion. If someone use 'hw' device then hardware must match the stream
parameters, otherwise 'plughw' should be used to instruct alsa-lib to do
necessary conversions.

						Jaroslav

-----
Jaroslav Kysela <perex@suse.cz>
Linux Kernel Sound Maintainer
ALSA Project, SuSE Labs



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Re: problems with rme digi96/8

[Gorm Lai]

Hi All!

Thanks for all the help you offered. I got everything, but unfortunately I
cannot explain why. The machine staff had first installed it in my
workstation on which I do not have root access. However in the end, I got
them to take it out, and then I put it in the demo machine in which it
supposed to work in the end anyhow. I have root access on this machine and
is responsible for configuring everything myself. Here everything worked
nearly out of the box; installation of drivers, libs, and utils went fine.

Thanks again. - I learned a lot in the process.

Gorm



----- Original Message -----
From: "Jaroslav Kysela" <perex@suse.cz>
To: "Giuliano Pochini" <pochini@shiny.it>
Cc: <alsa-devel@lists.sourceforge.net>
Sent: Thursday, June 12, 2003 3:18 PM
Subject: Re: [Alsa-devel] problems with rme digi96/8


> On Thu, 12 Jun 2003, Giuliano Pochini wrote:
>
> >
> > On 12-Jun-2003 Jaroslav Kysela wrote:
> > > On 12 Jun 2003, Gorm David Lai wrote:
> > >
> > >> I tried your example. Aplay is rather stupid, so it is seems it needs
a
> > >
> > > Not too stupid, but you should use right device (plughw vs. hw) and
> > > probably right alsa-lib configuration (~/.asoundrc).
> >
> > The format written in the sound file has priority over command
> > line options. It's a bit annoying and should be changed IMHO.
>
> No, you can explicitly change the format (if something is wrong) using
> command line parameters. But my point was that aplay does not do any
> conversion. If someone use 'hw' device then hardware must match the stream
> parameters, otherwise 'plughw' should be used to instruct alsa-lib to do
> necessary conversions.
>
> Jaroslav
>
> -----
> Jaroslav Kysela <perex@suse.cz>
> Linux Kernel Sound Maintainer
> ALSA Project, SuSE Labs
>
>
>
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