[Alsa-user] BCD2000

2007-09-04 Thread DJ A. J. Slye
Can someone please write a driver for the BCD2000 and BCD3000 please.. I am a 
DJ and would like to use this device in the linux version of my DJ Software 
Thanks..-
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Re: [Alsa-user] please test the beta CMI8788 driver

2007-09-04 Thread Clemens Ladisch
ATT TynTyn wrote:
 Under Fedora 7 with the newest kernel, I get ~ 0.25 samples every ~
 0.75 sec. on the main output.

That is, 0.25 s of sound at normal pitch, then 0.75 s of silence?

Is it different with 2/4/6 channels (the -c parameter)?


Regards,
Clemens

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Re: [Alsa-user] Mixer problem with cmipci

2007-09-04 Thread Clemens Ladisch
Jan-Benedict Glaw wrote:
 On Thu, 2007-08-30 23:43:26 +0200, Jan-Benedict Glaw [EMAIL PROTECTED] 
 wrote:
 Just out of interest, I ordered another, similar card (Ultron
 Octosound 7.1, lspci: C-Media Electronics Inc CM8738 (rev 10),
 /proc/asound/cards: C-Media PCI CMI8738-MC8 (model 68))

I have the same chip here.

 [...]
 After the FLAC ended, I fired up speaker-test again (and of course
 switched off Four Channel Mode.)  This time, it's different than
 before: front center/LFE, front left/right and side left/right are
 okay, but rear right/left are highly distorted.

I can reproduce this.

 Register 0x04
 ~
 Before:   0x07  CM_ASFC_SHIFT  |  0x00  CM_DSFC_SHIFT  |  CM_BREQ
 After:0x07  CM_ASFC_SHIFT  |  0x07  CM_DSFC_SHIFT  |  CM_BREQ

 (DAC Sample Frequency changed)

0 is the power-on default of 5512 Hz, 7 is 48 kHz.
DAC and ADC aren't quite the correct names, both DMA channels can
be used for either playback and capture, and the driver uses one for
stereo and the other for multichannel playback.

Does the FLAC file have a sample rate of 44.1 kHz or 48 kHz?

 Register 0x08
 ~
 Before:   CM_CHB3D5C  |  CM_SPDIF_SELECT1  |  0x03  CM_CH1FMT_MASK  |  
 0x00  CM_CH0FMT_SHIFT
 After:CM_CHB3D5C  |  CM_SPDIF_SELECT1  |  0x03  CM_CH1FMT_MASK  |  
 0x03  CM_CH0FMT_SHIFT

 (Ch0 Format changed)

0 is 8-bit mono, 3 is 16-bit stereo (or more than two channels).

 Register 0x18
 ~
 Before:   CM_TXVX  |  CM_FM_EN  |  CM_AC3EN2  | CM_VIDWPPRT  |
   CM_ENCENTER  |  CM_FLINKOFF
 After:CM_TXVX  |  CM_FM_EN  |  CM_AC3EN2  | CM_VIDWPPRT  | 
 CM_SPDF_AC97  |  CM_ENCENTER  |  CM_FLINKOFF

 (SPDIF/Out changed from 44.1kHz to 48kHz)

 My guess is that the change to SPDIF/Out frequency and DAC Sample
 Frequency are annoying at best, but irrelevant here. Am I probably
 right that the changed ch0 format is the cause of the distortion?

I think it's the sample rate.

It seems I can 'cure' the distortion by playing a 48 kHz file before
running speaker-test.  Can you confirm this?

The chip has two sample rate registers (ASFC and DSFC), and these are
internally mapped to three destinations (front+side+center DACs, rear
DAC, capture ADC) depending on the state of a bunch of other bits.  I
guess the driver doesn't handle these bits correctly so that the rear
sample rate is still the old value (although the two xSFC registers
appear to have the correct value).


Regards,
Clemens

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Re: [Alsa-user] XLR3 interfaces?

2007-09-04 Thread Clemens Ladisch
James Shatto wrote:
 Have a look at the Edirol UA-25.  It is definitely supported.
 
 Thanks, that looks like Exactly what I'm looking for.  Out of
 curiousity, since I've never used anything like these(yet), what else
 does it do?  Regular sound card stuff, or just a recording interface.

It does recording and playback.  Like many other USB devices, it
probably has few or no software-controllable mixer controls.

What do you mean with regular sound cars stuff?

 Onboard midi, or just an interface?

It has two MIDI jacks, input and output.  What is the difference
between onboard midi and interface?

 Multiple channel output with hardware mixing?

USB devices don't do hardware mixing; sending all those streams over
the bus would take more resources than mixing them in software.


Regards,
Clemens

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Re: [Alsa-user] problems with dmix, a52 and upmix

2007-09-04 Thread Clemens Ladisch
Michael KAufmann wrote:
 Is there any way to make dmix work with a52 as slave?

No.  As it's currently designed, dmix requires a hw device as slave
because it needs access to the sound card's DMA buffer and interrupt for
timing purposes.


Regards,
Clemens

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Re: [Alsa-user] Mixer problem with cmipci

2007-09-04 Thread Jan-Benedict Glaw
On Tue, 2007-09-04 08:56:22 +0200, Clemens Ladisch [EMAIL PROTECTED] wrote:
 Jan-Benedict Glaw wrote:
  On Thu, 2007-08-30 23:43:26 +0200, Jan-Benedict Glaw [EMAIL PROTECTED] 
  wrote:
   [...]
   After the FLAC ended, I fired up speaker-test again (and of course
   switched off Four Channel Mode.)  This time, it's different than
   before: front center/LFE, front left/right and side left/right are
   okay, but rear right/left are highly distorted.
 
 I can reproduce this.
 
  Register 0x04
  ~
  Before: 0x07  CM_ASFC_SHIFT  |  0x00  CM_DSFC_SHIFT  |  CM_BREQ
  After:  0x07  CM_ASFC_SHIFT  |  0x07  CM_DSFC_SHIFT  |  CM_BREQ
 
  (DAC Sample Frequency changed)
 
 0 is the power-on default of 5512 Hz, 7 is 48 kHz.
 DAC and ADC aren't quite the correct names, both DMA channels can
 be used for either playback and capture, and the driver uses one for
 stereo and the other for multichannel playback.
 
 Does the FLAC file have a sample rate of 44.1 kHz or 48 kHz?

44.1kHz:

$ sndfile-info 04_my_immortal.flac 

Version : libsndfile-1.0.17


File : 04_my_immortal.flac
Length : 28262646


Sample Rate : 44100
Frames  : 11651808
Channels: 2
Format  : 0x00170002
Sections: 1
Seekable: TRUE
Duration: 00:04:24.213
Signal Max  : 31624 (-0.31 dB)

  Register 0x18
  ~
  Before: CM_TXVX  |  CM_FM_EN  |  CM_AC3EN2  | CM_VIDWPPRT  |
CM_ENCENTER  |  CM_FLINKOFF
  After:  CM_TXVX  |  CM_FM_EN  |  CM_AC3EN2  | CM_VIDWPPRT  | 
  CM_SPDF_AC97  |  CM_ENCENTER  |  CM_FLINKOFF
 
  (SPDIF/Out changed from 44.1kHz to 48kHz)
 
  My guess is that the change to SPDIF/Out frequency and DAC Sample
  Frequency are annoying at best, but irrelevant here. Am I probably
  right that the changed ch0 format is the cause of the distortion?
 
 I think it's the sample rate.
 
 It seems I can 'cure' the distortion by playing a 48 kHz file before
 running speaker-test.  Can you confirm this?

After alsaplayer'ing one of the 48kHz files that speaker-test uses
with `-t wav', speaker-test also works as expected.  Confirmed.

MfG, JBG

-- 
  Jan-Benedict Glaw  [EMAIL PROTECTED]  +49-172-7608481
Signature of:  Alles sollte so einfach wie möglich gemacht sein.
the second  :  Aber nicht einfacher.  (Einstein)


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[Alsa-user] Problems with 32 bit apps on 64 bit OS using dmix and snd-ioctl32

2007-09-04 Thread James Pearson
I have a problem using 32 bit (i386) playback applications on a 64 bit 
(x86_64) CentOS4/RHEL4 based kernel (based on kernel 2.6.9).

I've installed alsa-driver, alsa-lib and alsa-utils 1.0.13 (newer 
versions of alsa-driver don't build with this kernel)

If I try to run a 32 bit copy of aplay on a 64 bit machine that doesn't 
support hardware mixing (e.g. HDA NVidia) - i.e. uses dmix by default, I 
get:

aplay: pcm_write:1268: write error: Invalid argument

and /var/log/messages reports:

Sep  4 16:45:37 wstemp-b kernel: ioctl32(aplay:28133): Unknown cmd fd(3) 
cmd(40045402){00} arg(ae88) on /dev/snd/timer
Sep  4 16:45:37 wstemp-b kernel: ioctl32(aplay:28133): Unknown cmd fd(5) 
cmd(c008551a){00} arg(080678a0) on /dev/snd/controlC0
Sep  4 16:45:37 wstemp-b kernel: ioctl32(aplay:28133): Unknown cmd fd(5) 
cmd(c008551b){00} arg(080678a0) on /dev/snd/controlC0
Sep  4 16:45:37 wstemp-b kernel: ioctl32(aplay:28133): Unknown cmd fd(3) 
cmd(54a0){00} arg(0320) on /dev/snd/timer

I've done some searching to find a solution - and found out that in more 
recent kernels, snd-ioctl32 has been obsoleted - and if I do use a more 
modern kernel, then using a 32 bit version of aplay works OK on a newer 
64 bit kernel.

However, for other (not sound related) reasons, I need to run the 
RHEL4/CentOS4 2.6.9 kernel ...

I had a look at the alsa-driver code, and acore/ioctl32/timer32_new.c 
contains:

#if 0
 /* ** FIXME **
  * The following four entries are disabled because they conflict
  * with the TCOC* definitions.
  * Unfortunately, the current ioctl32 wrapper uses a single
  * hash table for all devices.  Once when the wrapper is fixed
  * with the table based on devices, they'll be back again.
  */
 MAP_COMPAT(SNDRV_TIMER_IOCTL_START),
 MAP_COMPAT(SNDRV_TIMER_IOCTL_STOP),
 MAP_COMPAT(SNDRV_TIMER_IOCTL_CONTINUE),
 MAP_COMPAT(SNDRV_TIMER_IOCTL_PAUSE),
#endif

If I change the '#if 0' to '#if 1', recompile and reload with this new 
snd-ioctl32 module, then the 32 bit aplay works without an error ... 
although /var/log/messages still reports:

Sep  4 17:05:26 wstemp-b kernel: ioctl32(aplay:28926): Unknown cmd fd(3) 
cmd(40045402){00} arg(adb8) on /dev/snd/timer
Sep  4 17:05:26 wstemp-b kernel: ioctl32(aplay:28926): Unknown cmd fd(5) 
cmd(c008551a){00} arg(080678a0) on /dev/snd/controlC0
Sep  4 17:05:26 wstemp-b kernel: ioctl32(aplay:28926): Unknown cmd fd(5) 
cmd(c008551b){00} arg(080678a0) on /dev/snd/controlC0

... but at least I get sound out!

So, what is the danger of using these 'disabled' ioctls?

I can't find any reference to 'TCOC* definitions' - what are these ???

Thanks

James Pearson

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[Alsa-user] Serial Midi via snd-serial-u16550 not working

2007-09-04 Thread Denny Schierz
hi,

my friend has some problem to get snd-serial working, so i posting his
message to this list:

===
First of all hi everybody. This is my very first post in this forum.

I'm totally new in producing music under linux but after I have
configured programs like Ardour, Rosegarden (in sync via Jack audio
connection kit) and some other programs I want to chance from Windows
(Nuendo/Cubase/Reason) to Linux because the programs were running
excellent and have a big potential because of the function to rewire all
the devices by using JACK.
But befor telling you long stories now to my real problem: Since a few
days I try to get my Technics keyboard SX-KN5000 working by using midi
via the serial port. I want to do this because first I have no midi port
slot for my onboard soundcard of my A8N-SLI Premium motherboard and
second the serial port has a much faster speed and so latencies were
near zero even in big midi-projects. A I've read MANY tutorials the last
days and searched in many different forums to get this problem solved.

After I had first used the 'UART16550 serial MIDI driver' as built-in
kernel module but the midi device did not appear in the list of my devices
Code:
# aconnect -o
client 20: 'MPU-401 UART' [type=kernel]
0 'MPU-401 UART MIDI'

and in many how-tos I've read about using this driver as a loadable
module I've changed this. OK, now I'm at the point where I don't know
how to go on. I have read tutorials like this ones

http://alsa.opensrc.org/index.php/Serial#Simple_serial_converter
http://www.geocities.jp/midi_organ_net/alsa/
http://forums.gentoo.org/viewtopic-t-27919.html?sid=7abc10e4e733063cdc749811f114382c

but no one doesn't really help me. My serial port is com1 (/dev/ttyS0).
Almost all discussions about configuring this driver include that you
first have to make sure, that the serial port isn't in use. So I do a

Code:
# setserial /dev/ttyS0 uart none


setserial without the other parameters says
Code:
# setserial /dev/ttyS0
/dev/ttyS0, UART: unknown, Port: 0x03f8, IRQ: 4


So I know that my port is 0x03f8 and the irq is 4. Now I do a modprobe
to load the driver but I only get an error:
Code:
# modprobe -v snd-serial-u16550 port=0x3F8 irq=4
install { /bin/setserial /dev/ttyS0 uart none; } ; /sbin/modprobe
--first-time --ignore-install snd-serial-u16550
insmod
/lib/modules/2.6.22-gentoo-r2/kernel/sound/drivers/snd-serial-u16550.ko
snd_port=0x3f8 snd_irq=4 snd_speed=38400
FATAL: Error inserting snd_serial_u16550
(/lib/modules/2.6.22-gentoo-r2/kernel/sound/drivers/snd-serial-u16550.ko):
Unknown symbol in module, or unknown parameter (see dmesg)
FATAL: Error running install command for snd_serial_u16550


In my /var/log/messages it says
Code:
snd_serial_u16550: Unknown parameter `snd_port'


I don't understand this. I haven't specified any snd_port in the
modprobe command. I first thought, that I have to, but running modprobe
with snd_port (and e.g. snd_irq) doesn't help. Even if I call it without
any parameters the same error message appears. Maybe someone has the
same error as I or knows any help. Maybe it's just a config error
somewhere. At the beginning I experimented with configuring some
configfiles which you see below. But I don't know if they will be ever
used by my sound system. (Should I mention that all other sound drivers
are all built-in kernel modules)

/etc/modules.d/alsa
Code:
# Alsa kernel modules' configuration file.

# ALSA portion
alias char-major-116 snd
# OSS/Free portion
alias char-major-14 soundcore

##
## IMPORTANT:
## You need to customise this section for your specific sound card(s)
## and then run `update-modules' command.
## Read alsa-driver's INSTALL file in /usr/share/doc for more info.
##
##  ALSA portion
## alias snd-card-0 snd-interwave
## alias snd-card-1 snd-ens1371
##  OSS/Free portion
## alias sound-slot-0 snd-card-0
## alias sound-slot-1 snd-card-1
##
pre-install snd-serial-u16550 /bin/setserial /dev/ttyS0 uart none
alias snd-card-1 snd-serial-u16550
options snd-serial-u16550 port=0x3f8 irq=4 speed=38400

# OSS/Free portion - 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
##  OSS/Free portion - card #2 serial midi
alias sound-service-1-1 snd-seq-oss
alias sound-service-1-8 snd-seq-oss

alias /dev/mixer snd-mixer-oss
alias /dev/dsp snd-pcm-oss
alias /dev/midi snd-seq-oss

# Set this to the correct number of cards.
options snd cards_limit=2


/etc/modprobe.d/sound
Code:
options snd-serial-u16550 port=0x3f8 irq=4


The rest of my soundsystem works fine. I would be thankful for any help
I could get. Is something missing? Please tell me. Here the last thing
which could be helpful:

Code:
# uname -a
Linux marcus 2.6.22-gentoo-r2 #1 SMP Thu Aug 30 18:37:44 CEST 2007 i686
AMD Athlon(tm) 64 X2 Dual Core Processor 4400+ AuthenticAMD GNU/Linux


Thanks
Marcus

===


Re: [Alsa-user] Front Output Jack problems with AD1986A/M5461

2007-09-04 Thread Kevin Hill
Have you checked all of your mixer settings?  The front one might be
PCM2 or Headphone.
And might only be accessible in alsamixer.

Nope, the only things I have listed in alsamixer under playback are:
Master, PCM, Line, CD, Mic, Mic Boos, IEC958, and Aux.

All are unmuted and set at full.

Any other ideas?

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