Re: Alsa-devel digest, Vol 1 #810 - 3 msgs

Re: Alsa-devel digest, Vol 1 #810 - 3 msgs

[Bob McElrath]

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alsa-devel-request@lists.sourceforge.net [alsa-devel-request@lists.sourceforge.net] wrote:
> Date: Tue, 29 Oct 2002 08:55:06 +0100
> From: Benny Sjostrand <gorm@cucumelo.org>
> To: alsa-devel@lists.sourceforge.net
> Subject: [Alsa-devel] [PATCH] cs46xx, independant rear PCM channel, IEC958 PCM channel
> 
> This is a multi-part message in MIME format.
> --------------090804060704000507060108
> Content-Type: text/plain; charset=us-ascii; format=flowed
> Content-Transfer-Encoding: 7bit
> 
> Hi!
> 
> First, this stuff is still very experimental, it should work very well 
> with a
> Game Theater XP course that's card I've got and the only cs46xx sound 
> card I can
> test with.
> 
> I've got some reports that stuffs dont seems to work very well with
> the Santa Cruz card, course for some unknown reason the driver dont
> manage to raise GPIO 7 and 9 on the secondary codec that
> enables the rear output, you probably will see the message
> "cs46xx_setup_eapd_slot() Failure to write the GPIO pins for
> slot 12. " and that will probably freeze you computer
> for a while too.
> 
> In this patch, for the Santa Cruz the amplifier and the GPIO's are
> setup on init, then not touched anymore, like it's done for the
> Hercules XP card with hope and a little of luck maybe that
> works better.
> 
> Finally, this patch contains the following changes:
> - Rear PCM channel
> - IEC958 PCM channel, in hope that I'm close to get raw AC3 working soon
> - Turn on Amplifier and setup EGPIO for Santa Cruz card on init
> - Cleanup's in voyectra_setup_eapd_slot(...) (hope it's not broken ...)
> 
> PS. If you have time to help testing this patch, dont forget to turn on
> all debuging stuff, that always helps to find bugs ...

With a GTXP on an alpha. ;)

I managed to stall the codec by playing with the IEC958.  I turned the
IEC958 input all the way down, unmuted it, then unmuted the IEC958
output.  I was playing an mp3 at the time, and the mp3 stopped and the
codec just puts out a constant ~8khz tone.  This is repeatable.

The 3D Depth seems to work now, sounds great.  I don't think it worked
before... ;)

Playing 2 pcm inputs at the same time seems to work great.

Recording still seems to not work.  I still get about a 1Hz ticking in
the right channel when recording (no matter what the mixer settings).

I don't see a debug setting on the modules snd-cs46xx or snd -- is this
a compile time setting?  Anyway, let me know if you want me to send you
some debug output.

What is the easy way to test playing to the rear pcm channel?

Cheers,
-- Bob

Bob McElrath
Univ. of Wisconsin at Madison, Department of Physics

    "The surest way to corrupt a youth is to instruct him to hold in higher
    esteem those who think alike than those who think differently." 
        -- Nietzsche

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Re: Re: Alsa-devel digest, Vol 1 #810 - 3 msgs

[Peter Heatwole]

On 2002.10.29 08:36 Bob McElrath wrote:
> I don't see a debug setting on the modules snd-cs46xx or snd -- is
> this a compile time setting?

    Yes, you set this when you configure the package.  Use the configure
option "--with-debug=detect" for the most verbose debugging output.


> What is the easy way to test playing to the rear pcm channel?

The easy way:  aplay -Dhw:0,1 test.wav

    Just don't forget to unmute and raise the volume for the rear-output.
Also, if you wish to output directly to SPDIF, use:  "aplay -Dhw:0,2".
    
Thanks for testing. ;-)

HTH,
-- Peter Heatwole
"Murphy was just a well known pessimist."



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Re: Re: Alsa-devel digest, Vol 1 #810 - 3 msgs

[Bob McElrath]

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Peter Heatwole [peter599@gotnet.net] wrote:
> On 2002.10.29 08:36 Bob McElrath wrote:
> >I don't see a debug setting on the modules snd-cs46xx or snd -- is
> >this a compile time setting?
> 
>    Yes, you set this when you configure the package.  Use the configure
> option "--with-debug=detect" for the most verbose debugging output.
> 
> 
> >What is the easy way to test playing to the rear pcm channel?
> 
> The easy way:  aplay -Dhw:0,1 test.wav

Seems to work well.  The relevant mixer settings are PCM1 and Surround1
(both of which must be turned up and unmuted.  However it doesn't seem
to deal well with odd rates:

(0)<mcelrath@draal:/home/mcelrath> aplay -Dhw:0,1 /usr/share/sounds/zipper.wav 
Playing WAVE '/usr/share/sounds/zipper.wav' : Unsigned 8 bit, Rate 11025 Hz, Mono
aplay: pcm_write:968: write error: Input/output error

>    Just don't forget to unmute and raise the volume for the rear-output.
> Also, if you wish to output directly to SPDIF, use:  "aplay -Dhw:0,2".

Unfortunately I don't have any hardware which can deal with SPDIF
(except my Marvel TV card which only has an output).

> Thanks for testing. ;-)

No prob.

> HTH,

Have The Ham?

Cheers,
-- Bob

Bob McElrath (rsmcelrath@students.wisc.edu) 
Univ. of Wisconsin at Madison, Department of Physics

    "The surest way to corrupt a youth is to instruct him to hold in higher
    esteem those who think alike than those who think differently." 
        -- Nietzsche

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Re: Re: Alsa-devel digest, Vol 1 #810 - 3 msgs

[Benny Sjostrand]

>
>
>Seems to work well.  The relevant mixer settings are PCM1 and Surround1
>(both of which must be turned up and unmuted.  However it doesn't seem
>to deal well with odd rates:
>
>(0)<mcelrath@draal:/home/mcelrath> aplay -Dhw:0,1 /usr/share/sounds/zipper.wav 
>Playing WAVE '/usr/share/sounds/zipper.wav' : Unsigned 8 bit, Rate 11025 Hz, Mono
>aplay: pcm_write:968: write error: Input/output error
>  
>

Hm.,, just weird, does it work OK with the fron channel (hw:0,0) ???

What do get on dmesg output ???

/Benny




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Re: Re: Alsa-devel digest, Vol 1 #810 - 3 msgs

[Bob McElrath]

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Benny Sjostrand [gorm@cucumelo.org] wrote:
> >
> >
> >Seems to work well.  The relevant mixer settings are PCM1 and Surround1
> >(both of which must be turned up and unmuted.  However it doesn't seem
> >to deal well with odd rates:
> >
> >(0)<mcelrath@draal:/home/mcelrath> aplay -Dhw:0,1 
> >/usr/share/sounds/zipper.wav Playing WAVE '/usr/share/sounds/zipper.wav' : 
> >Unsigned 8 bit, Rate 11025 Hz, Mono
> >aplay: pcm_write:968: write error: Input/output error
> > 
> >
> 
> Hm.,, just weird, does it work OK with the fron channel (hw:0,0) ???

Exact same results for hw:0,0.  (i.e. it doesn't work)

> What do get on dmesg output ???

I still need to recompile with debug.  I will send debug messages later.

Benny Sjostrand [gorm@cucumelo.org] wrote:
> >
> >With a GTXP on an alpha. ;)
> >
> Cool! stupid question from me, which endianess does the Alpha 
> architecture have ??
> well, I believe the cs46xx is never tested on Big-Endian architecture

Alpha is little endian.  (The processor itself can be either, but
linux/alpha uses little endian)

> >I managed to stall the codec by playing with the IEC958.  I turned the
> >IEC958 input all the way down, unmuted it, then unmuted the IEC958
> >output.  I was playing an mp3 at the time, and the mp3 stopped and the
> >codec just puts out a constant ~8khz tone.  This is repeatable.
> > 
> A old bug that I supposed was fixed, but apparentily it's not, probably 
> course
> I've never tested to play with SPDIF in and SPDIF out control at the same
> time. Hopefully I'll find a way to get around it ....
> 
> Just to confirm; if you only play with SPDIF input mixer control, leaving
> the SPDIF output unmuted it dont happen right ?

Yes.

BTW I recommend renaming the IEC958 to "SPDIF" or "Digital In/Out" since
nobody knows what an IEC958 is...

> >Recording still seems to not work.  I still get about a 1Hz ticking in
> >the right channel when recording (no matter what the mixer settings).
> >
> Recording should work fine, did you have the DAC capture SPIDF input 
> "on" ?? then
> it could be SPDIF input related ...

I cannot record anything at all, from any input (I haven't tried from
SPDIF -- I don't have a reliable SPDIF source anyway)  I get the 1Hz
ticking.

Cheers,
-- Bob

Bob McElrath (rsmcelrath@students.wisc.edu) 
Univ. of Wisconsin at Madison, Department of Physics

    "The surest way to corrupt a youth is to instruct him to hold in higher
    esteem those who think alike than those who think differently." 
        -- Nietzsche

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recording problems with snd-cs46xx (was: Re: Re: Alsa-devel digest, Vol 1 #810 - 3 msgs)

[Kai Vehmanen]

Hi,

this issue is over a year old, but in case someone stumbles on
to the same problem. The solution is already documented at

   http://alsa.opensrc.org/index.php?page=cs46xx

... but it took me a while to find it. :)

On Tue, 29 Oct 2002, Bob McElrath wrote:

>>>Recording still seems to not work.  I still get about a 1Hz ticking in
> I cannot record anything at all, from any input (I haven't tried from
> SPDIF -- I don't have a reliable SPDIF source anyway)  I get the 1Hz

I had this very same problem with alsa-driver-0.9.8. The problem turned 
out to be that I had not selected 'ADC' as a capture source in alsamixer. 
So you need to select all three: 'ADC', 'Mic' and 'Capture' for the 
recording to work properly.

PS Tested with IBM Thinkpad models T20 and T21.

-- 
 http://www.eca.cx
 Audio software for Linux!



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Re: Re: Alsa-devel digest, Vol 1 #810 - 3 msgs

[Bob McElrath]

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Benny Sjostrand [gorm@cucumelo.org] wrote:
> >
> >
> >Seems to work well.  The relevant mixer settings are PCM1 and Surround1
> >(both of which must be turned up and unmuted.  However it doesn't seem
> >to deal well with odd rates:
> >
> >(0)<mcelrath@draal:/home/mcelrath> aplay -Dhw:0,1 
> >/usr/share/sounds/zipper.wav Playing WAVE '/usr/share/sounds/zipper.wav' : 
> >Unsigned 8 bit, Rate 11025 Hz, Mono
> >aplay: pcm_write:968: write error: Input/output error
> > 
> 
> Hm.,, just weird, does it work OK with the fron channel (hw:0,0) ???
> 
> What do get on dmesg output ???

The relevant stuff is:
Oct 29 23:52:11 draal kernel: ALSA dsp_spos_scb_lib.c:1188: dsp_spos: creating SRC "SrcTask_SCB0"
Oct 29 23:52:11 draal kernel: ALSA dsp_spos_scb_lib.c:295: dsp_spos: creating SCB <SrcTask_SCB0>
Oct 29 23:52:11 draal kernel: ALSA dsp_spos_scb_lib.c:1355: dsp_spos: setting SRC rate to 44100
Oct 29 23:52:11 draal kernel: ALSA dsp_spos_scb_lib.c:1211: dsp_spos: creating PCM "PCMReader_SCB0" (1)
Oct 29 23:52:11 draal kernel: ALSA dsp_spos_scb_lib.c:295: dsp_spos: creating SCB <PCMReader_SCB0>
Oct 29 23:52:11 draal kernel: ALSA dsp_spos_scb_lib.c:1355: dsp_spos: setting SRC rate to 11025
Oct 29 23:52:21 draal kernel: ALSA dsp_spos_scb_lib.c:231: cs46xx_dsp_proc_free_scb_desc: freeing PCMReader_SCB0
Oct 29 23:52:21 draal kernel: ALSA dsp_spos_scb_lib.c:231: cs46xx_dsp_proc_free_scb_desc: freeing SrcTask_SCB0

Why does it set 44.1kHz first and then set it to 11kHz?

I hear the first piece (frag?) of the sample played repeatedly, as if it
were not servicing interrupts...

Messages for the 2nd pcm channel are identical if you s/(1)/(2)/ above.

As to the recording problem, I have tried all the options I can think
of, all the mixer settings and possible inputs, and I still get ticking.
NO messages are written to the logs during the record.
    arecord -d 2 -f cd blah.wav
I have placed this file here: http://mcelrath.org/blah.wav.bz2

Cheers,
-- Bob

Bob McElrath
Univ. of Wisconsin at Madison, Department of Physics

    "The surest way to corrupt a youth is to instruct him to hold in higher
    esteem those who think alike than those who think differently." 
        -- Nietzsche

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request

[shiva shankar]

Can anyone send me the alsa design document.  I am interested in developing
a ALSA driver.

Thank you,
Shiva




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Re: Alsa-devel digest, Vol 1 #810 - 3 msgs

[Benny Sjostrand]

>
>
>With a GTXP on an alpha. ;)
>  
>
Cool! stupid question from me, which endianess does the Alpha 
architecture have ??
well, I believe the cs46xx is never tested on Big-Endian architecture

>I managed to stall the codec by playing with the IEC958.  I turned the
>IEC958 input all the way down, unmuted it, then unmuted the IEC958
>output.  I was playing an mp3 at the time, and the mp3 stopped and the
>codec just puts out a constant ~8khz tone.  This is repeatable.
>  
>
A old bug that I supposed was fixed, but apparentily it's not, probably 
course
I've never tested to play with SPDIF in and SPDIF out control at the same
time. Hopefully I'll find a way to get around it ....

Just to confirm; if you only play with SPDIF input mixer control, leaving
the SPDIF output unmuted it dont happen right ?

>Recording still seems to not work.  I still get about a 1Hz ticking in
>the right channel when recording (no matter what the mixer settings).
>  
>
Recording should work fine, did you have the DAC capture SPIDF input 
"on" ?? then
it could be SPDIF input related ...

/Benny



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