[Alsa-user] Address family not supported by protocol

2009-02-05 Thread Scott Karlin 
I'm trying to get sound working on my Fedora 10 box but am running
into trouble.  I'm following the steps here:
http://alsa.opensrc.org/index.php/TroubleShooting

Other than no sound, here's an obvious problem:


[root]# alsamixer
E: socket-client.c: socket(): Address family not supported by protocol
ALSA lib pulse.c:272:(pulse_connect) PulseAudio: Unable to connect: Connection 
refused
alsamixer: function snd_ctl_open failed for default: Connection refused


Here's some system information:


[root]# cat /proc/asound/version
Advanced Linux Sound Architecture Driver Version 1.0.17.

[root]# grep VERSION_STR /usr/include/alsa/version.h
#define SND_LIB_VERSION_STR 1.0.19

[root]# cat /proc/asound/cards
 0 [CA0106 ]: CA0106 - CA0106
  Audigy SE [SB0570] at 0xdc00 irq 17

[root]# lspci | egrep -i audio
01:02.0 Multimedia audio controller: Creative Labs CA0106 Soundblaster

[root]# uname -srvpio
Linux 2.6.27.12-170.2.5.fc10.i686 #1 SMP Wed Jan 21 02:09:37 EST 2009 athlon 
i386 GNU/Linux

[root]# aplay -l
 List of PLAYBACK Hardware Devices 
card 0: CA0106 [CA0106], device 0: ca0106 [CA0106]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: CA0106 [CA0106], device 1: ca0106 [CA0106]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: CA0106 [CA0106], device 2: ca0106 [CA0106]
  Subdevices: 1/1
  Subdevice #0: subdevice #0
card 0: CA0106 [CA0106], device 3: ca0106 [CA0106]
  Subdevices: 1/1
  Subdevice #0: subdevice #0

[root]# ls -als /dev/snd
total 0
0 drwxr-xr-x  2 root root 280 Feb  5 14:07 ./
0 drwxr-xr-x 14 root root4500 Feb  5 14:09 ../
0 crw-rw  1 root root 116, 13 Feb  5 14:07 controlC0
0 crw-rw  1 root root 116,  4 Feb  5 14:07 midiC0D0
0 crw-rw  1 root root 116, 12 Feb  5 14:10 pcmC0D0c
0 crw-rw  1 root root 116, 11 Feb  5 14:10 pcmC0D0p
0 crw-rw  1 root root 116, 10 Feb  5 14:07 pcmC0D1c
0 crw-rw  1 root root 116,  9 Feb  5 14:07 pcmC0D1p
0 crw-rw  1 root root 116,  8 Feb  5 14:07 pcmC0D2c
0 crw-rw  1 root root 116,  7 Feb  5 14:07 pcmC0D2p
0 crw-rw  1 root root 116,  6 Feb  5 14:07 pcmC0D3c
0 crw-rw  1 root root 116,  5 Feb  5 14:07 pcmC0D3p
0 crw-rw  1 root root 116,  3 Feb  5 14:07 seq
0 crw-rw  1 root root 116,  2 Feb  5 14:07 timer


[root]# rpm -qa | grep -i alsa\|pulseaudio | sort
alsa-lib-1.0.19-1.fc10.i386
alsa-lib-devel-1.0.19-1.fc10.i386
alsa-plugins-pulseaudio-1.0.18-2.fc10.i386
alsa-utils-1.0.19-1.fc10.i386
bluez-alsa-4.22-2.fc10.i386
kde-settings-pulseaudio-4.1-4.20081031svn.fc10.noarch
pulseaudio-0.9.14-1.fc10.i386
pulseaudio-core-libs-0.9.14-1.fc10.i386
pulseaudio-libs-0.9.14-1.fc10.i386
pulseaudio-libs-glib2-0.9.14-1.fc10.i386
pulseaudio-module-x11-0.9.14-1.fc10.i386
pulseaudio-utils-0.9.14-1.fc10.i386
xine-lib-pulseaudio-1.1.16.1-1.fc10.i386


Any clues on how to track this down and get sound working?

Thanks,
Scott

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Re: [Alsa-user] hda_intel with external DAC

2009-02-05 Thread Lee Revell 
On Wed, Feb 4, 2009 at 1:15 PM, Dirk Mast condo...@gmail.com wrote:
 Until now I used the normal line-out, which was working fine and mixing was
 no problem.


HDA intel does not support hardware mixing, so your distro must be
using PulseAudio or dmix on the default device.

 Using the following .asoundrc

 pcm.hda-intel {
  type hw
  card 0
 }

 ctl.hda-intel {
type hw
card 0
 }


 I can listen to multiple streams (flash, amarok, xmms, whatever).
 But I'm wondering, if this alsa setup is resampling 44.1Khz to 48Khz.


This .asoundrc doesn't do anything, as apps will use the default
device, not hda-intel.

Software mixing *requires* resampling all streams to a common sample
rate, 48Khz by default because not all sound cards support 44.1Khz.
You could configure pulseaudio/dmix/whatever to resample everything to
44.1Khz, but then 48Khz streams will be downsampled.


 When I use the following .asoundrc
 pcm.!default iec958:Intel

 The audio devices get's locked by a single application, which is not nice.


You need to configure your sound server to output to the iec958
device, as the hardware only supports one stereo stream.

 - no resampling in terms of 44.1 - 48
 BUT
 - decoding of those AC3/DTS streams


iec958 only supports 2 channels.  Either you set up xine to decode
those and downmix them to stereo, or you send the raw AC3/DTS stream
to the device and have your receiver decode it.

 How are my .asoundrc / xine / player configs expected to lookalike?

Depends on how your distro sets up the default ALSA device, sorry I
can't help more.  Does Debian testing use pulseaudio?

Lee

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