Intel8x0

From the ALSA wiki

1 Troubleshooting
2 Installation/Configuration
3 Device configuration (.asound)

Troubleshooting

SDL Sound Latency

When using libSDL under gentoo linux with an SiS chip and the intel8x0 driver, make sure you compile with the oss use flag on. Without this flag, oss support is not compiled into libsdl and some sound applications lag by up to half a second.

Multichannel Mixing

I run into this problem with my IBM T40. For example, you want to play music in a flash movie during a mozilla session while you also listen to your mp3s with mplayer or xmms. MPlayer and XMMS provides particular plugins you can load to use ALSA, but this wouldn't solve the problem at all. In fact, you have to use the dmix plugin to get the multiple streams mixed before they access the soundcard. Alsa can do this for you, but you need to tell ALSA to forward the streams first through the dmix plugin. Setup an .asoundrc in your home and follow the first two steps at Dmix Plugin. You should be able to play your mp3s with the dmix plugin. If this isn't working try to unload the ALSA OSS emulation modules, to make sure, that the application really uses alsa. If this works, add this example to your .asoundrc:

 # this is IBM T40 specific, which 
 # uses the snd_intel8x0, and may
 # not work on other laptops

 pcm.amix {
   type dmix
   ipc_key 50557
   slave {
       pcm "hw:0,0"
       period_time 0
       period_size 1024
       buffer_size 8192
   }
   bindings {
       0 0
       1 1
   }
 }

 # route ALSA software through pcm.amix
 pcm.!default {
   type plug
   slave.pcm "amix"
 }

Now try to use eg. mplayer with the alsa plugin to play your mp3s. You should be able to play several mp3s in several mplayer instances. If not try to start your mplayer by defining which device alsa should use:

 mplayer -ao alsa:device=amix foobar.mp3

Remember, we setup the amix device in the .asoundrc first, which uses dmix (pcm.amix). If this device is called pcm.foobar, you then use this instead...

 mplayer -ao alsa:device=foobar foobar.mp3

Teamspeak, Quake3

See DmixPlugin to get it working.

IEC958 controls in the mixer have no effect on sound output on SPDIF connection (Shuttle XPC SS51G)

Thanks to a tip from Ole Andre Schistad (oschista@chello.no) on the alsa-project.org web site under documentation for the Intel i820 card options. The solution is to compile the ALSA drivers, libs and utils with "cvscompile". According to Ole, "The S/PDIF support in this chipset currently requires a CVS version of ALSA...the required device under /dev/alsa is not created with the "release" drivers..." I'm using ALSA version 0.9.6.

Digital output not working (Realtek ALC850)

This driver also works for the Realtek ALC850 chip (as used in nForce4 boards). To use the digital output there you just have to connect a optical cable and set the output volume of the IEC958 Playback AC97-SPSA to zero. This is a bit counter-intuitive, but it works because with a digital output you cannot control the volume from within ALSA, but rather set the output mode. Zero means here output, whereas other values mean no output. You may have to go to the very far right in alsamixer or some graphical mixer to find the slider.

Sound not working on Fujitsu-Siemens Desktops with ICH845

There might be problems with sound not working on workstations by Fujitsu-Siemens with ICH845-based AC97 soundcard onboard. This problem confirmed to occur on Fedora Core 6. Symptoms are sound isn't working, and entries like this appear in dmesg/syslog:

 PCI: Enabling device 0000:00:1f.5 (0000 -> 0003)
 ACPI: PCI Interrupt 0000:00:1f.5[B] -> GSI 17 (level, low) -> IRQ 201
 PCI: Setting latency timer of device 0000:00:1f.5 to 64
 AC'97 0 access is not valid [0xffffffff], removing mixer.
 Unable to initialize codec #0
 ACPI: PCI interrupt for device 0000:00:1f.5 disabled
 Intel ICH: probe of 0000:00:1f.5 failed with error -5

The solution found to overcome a problem is to use the following options for intel8x0 module:

 # modprobe -r snd-intel8x0
 # modprobe snd-index8x0 ac97_quirk=1 xbox=1

This parameters were found in source code for intel8x0 module in alsa-drivers package. Looks like there's no information about them in official docs as for now (Jule 2007).

Installation/Configuration

Debian with KDE

Quick start instructions for Debian with KDE. (verified with Si S 7012 + Realtek codec and nForce2 MCP + Realtek codec)

Install alsa-base
Select intel8x0 in configuration screen
Enable OSS/Free emulation
In KDE, go to control panel-> sound system
Select 'toss' (Threaded Open Sound System). If you select anything else a lot of the output will be scratchy
Restart KDE
Rock'n'Roll

Dell Laptops

The new batches of DELL computers all have this chipset on the motherboard that does everything. The OSS driver is *very* sub-par. Try to watch a DVD or a DIVX movie with it and you'll see what I mean: the audio skips, loses sync, etc. However the ALSA driver solves all problems. [Not for some laptops -- see below.] I compiled it with a brand new Red Hat 7.3 distribution. Things to look for:

you need the kernel source as per documentation, the SRPM is what you want, as Red Hat makes significant changes to the stock kernel. No matter what, you'll need the source of the kernel you are running.
if using the SRPM from Red Hat you'll need to edit the top-level makefile of the kernel distribution, and you'll need to change the EXTRAVERSION variable (4th line) to the version you are running. By default it is something like:

 EXTRAVERSION=-4custom

type uname -a and check the name of your kernel, it should be something like

 Linux caravan 2.4.18-4 #1 Thu May 2 18:47:38 EDT 2002 i686 unknown

if you have an SMP kernel you'll get 2.4.18-4smp. Change the EXTRAVERSION to match that, i.e: -4 or -4smp

Compile and install as per instructions, load the kernel modules. Check /proc/asound that the chipset was correctly detected.
Note that you can now use an OSS mixer, but that no matter what you do no sound comes out of the speakers. Don't despair, this is normal!
you'll NEED to compile the ALSA library and the ALSA tools, and to install them
once this is done, use amixer to unmute the Headphone, ie:

 amixer -c 0 sset Headphone,0 55 unmute

you should get:

 Simple mixer control 'Headphone',0
 Capabilities: pvolume pswitch pswitch-joined
 Playback channels: Front Left - Front Right
 Limits: Playback 0 - 63
 Front Left: Playback 55 [87%] [on]
 Front Right: Playback 55 [87%] [on]

In order to unmute the default speaker jacks do:

 amixer set PCM 100 unmute

be aware that you need the snd-mixer-oss module to be loaded if you wish to adjust the volume with an OSS legacy mixer. This also means programs like xmms!
from now on you should be able to play sounds with any applications, OSS or ALSA, and OSS mixers should be able to handle your card. remember you must install an oss mixer to get oss applications to work correctly - i recommend aumix.
for the best sound quality (at least on my dell smartstep) make sure master and pcm are about two bars below their maximum, and correct volume with the wheel on the side of your computer. otherwise you will notice some frequency range problems.

Dell Inspiron 8600 (and probably others)

To get sound out of the regular laptop speakers, you must MUTE the "External Amplifier Power Down" slider by pressing M. (On mine, it is the farthest slider to the right.)

Mute = sound
Unmute = no sound.

Dell Inspiron Laptops (4150, 8200?, others?)

2003-05-07

NEWSFLASH (Jun 3, 2003): The newly-released BIOS version A06 for the Inspiron 4150 solves this problem completely! I expect that new BIOSes for the other Dell laptops will solve the problem too. I'm leaving the rest of this here just in case anybody else has this problem and wants to understand it, but if you upgrade to a new bios and try the "while /bin/true; ..." test listed below and don't have any problems then you can disregard this issue entirely!

This is one case where the OSS intel8x0 driver works much better than the ALSA snd-intel8x0 driver (at least as of ALSA version 0.9.2). If all you care about is listening to music/CDs/sound-effects in games then just stick with the OSS driver -- it works well for those purposes. If you care about the more advanced features of the ALSA drivers (because you want to use Jack, for example) then you're in for a bumpy ride. Symptoms of the problem:

You find that your sound output often gets polluted with noise.
Start playing a sound or music file with, say, xmms (using the ALSA output plugin) and then run "while /bin/true; do cat /proc/apm; done". The sound will very quickly become heavily polluted with static and noise, which continues even after you stop accessing /proc/apm. Stopping the sound file and restarting it may or may not make the noise go away.

Details about the problem:

It seems to be caused by a bad APM implementation in the BIOS. If you're running apmd, klaptopdaemon, or anything else that reads from /proc/apm periodically it will trigger this problem quite often.
You might have also noticed that your clock runs slow. In fact, running the "while /bin/true..." script causes my clock to run at about 1/2 speed. Here's a link that discusses this aspect of the problem: http://math.stanford.edu/\~carlton/i8200/time.html

Remedies:

BIOS A06 solves this issue for the Inspiron 4150.
You can switch from APM to ACPI for power management. I haven't done much experimentation with this yet because some preliminary investigation suggests that suspend to RAM won't work for 2.4 kernels, which is a showstopper for me. If you're interested in going this route, be sure to see this link: http://acpi.sourceforge.net/wiki/index.php/FixedDsdts
Compile APM with CONFIG_APM_ALLOW_INTS=y. This allows interrupts during APM calls.
Compile APM as a module instead of compiling it into the kernel. This way you can unload the apm module while you're doing music and reload it when you're done. Unfortunately I haven't been able to get the use-count of the apm module down to 0 without exiting X (I stopped apmd and killed all battery monitor applets). If I remove the apm module before starting X and modprobe it after starting X then I can subsequently get the use count back down to 0, but suspend/resume doesn't work properly. So AFAICT you have to restart X to go between apm-enabled and apm-disabled states. It's better than rebooting, but still annoying.

Using the two APM options above (allow interrupts and compile as module) helps somewhat, even when the apm module is loaded. I can now play songs in XMMS with occasional skips but I almost never get the noise pollution that used to show up about once per song. I can even almost use Jack successfully.

Shuttle XPC SS51G

To record with the microphone jack on the front, you need to select "Mic2" in the amixer "Mic Select" field. The default is to use "Mic1", which is the input on the back.

Device configuration (.asound)

Generic configuration

All examples in /etc/asound.conf or \~/.asoundrc

 pcm.nforce-hw {
   type hw
   card 0
 }
 pcm.!default {
   type plug
   slave.pcm "nforce"
 }
 pcm.nforce {
   type dmix
   ipc_key 1234
   slave {
       pcm "hw:0,0"
       period_time 0
       period_size 1024
       buffer_size 4096
       rate 44100
   }
 }
 ctl.nforce-hw {
   type hw
   card 0
 }

Intel 845 (Dell Inspiron 1100 laptop)

 pcm.nforce-hw {
   type hw
   card 0
 }
 pcm.!default {
   type plug
   slave.pcm "nforce"
 }
 pcm.nforce {
   type dmix
   ipc_key 1234
   slave {
       pcm "hw:0,0"
       period_time 0
       period_size 1024
       buffer_size 8192
       rate 44100
   }
   bindings {
       0 0
       1 1
   }
 }
 ctl.nforce-hw {
   type hw
   card 0
 }

NForce 1 MSI K7N420

 pcm.nforce-hw {
   type hw
   card 0
 }
 pcm.!default {
   type plug
   slave.pcm "nforce"
 }
 pcm.nforce {
   type dmix
   ipc_key 1234
   slave {
       pcm "hw:0,0"
       period_time 0
       period_size 512
       buffer_size 4096
       rate 44100
   }
 }
 ctl.nforce-hw {
   type hw
   card 0
 }

ASUS A7N8X Deluxe

 pcm.nforce-hw {
   type hw
   card 0
 }
 pcm.!default {
   type plug
   slave.pcm "nforce"
 }
 pcm.nforce {
   type dmix
   ipc_key 1234
   slave {
       pcm "hw:0,0"
       period_time 0
       period_size 1024
       buffer_size 32768
       rate 48000
   }
 }
 ctl.nforce-hw {
   type hw
   card 0
 }

Shuttle SN41G2

 pcm.nforce-hw {
   type hw
   card 0
 }
 pcm.!default {
   type plug
   slave.pcm "nforce"
 }
 pcm.nforce {
   type dmix
   ipc_key 1234
   slave {
       pcm "hw:0,1"
       period_time 0
       period_size 1024
       buffer_size 4096
       rate 44100
   }
 }
 ctl.nforce-hw {
   type hw
   card 0
 }

nForce 4 digital output

Mainboard is an Asus K8N Deluxe with nForce 4.

cat /proc/asound/pcm 
00-02: Intel ICH - IEC958 : NVidia CK804 - IEC958 : playback 1
00-01: Intel ICH - MIC ADC : NVidia CK804 - MIC ADC : capture 1
00-00: Intel ICH : NVidia CK804 : playback 1 : capture 1

Mixer settings:

amixer set IEC958 unmute
amixer set 'IEC958 Playback AC97-SPSA' 0
amixer set 'IEC958 Playback Source' PCM

AC3/DTS passthrough on S/P-DIF should now work with mplayer:

mplayer -ao alsa:device=hw=0.0 -ac hwdts,hwac3, FILE

Fujitsu Siemens Lifebook E8020 intel8x0 + snd-usb-audio

I have worked on asoundrc that provides duplex functionality i.e. I can play more than one sound sources. I need this to watch TV over mplayer and listen music over xmms, but other progs work as well

# Set default sound card
# Useful so that all settings can be changed to a different card here.
pcm.snd_card0 {
     type hw
     card 0
     device 0
}

pcm.snd_card1 {
     type hw
     card 1
     device 0
}

# Allow mixing of multiple output streams to this device
pcm.output {
     type dmix
     ipc_key 1024
     ipc_perm 0660 # Sound for everybody in your group!
     slave.pcm "snd_card0"

     slave {
          # This stuff provides some fixes for latency issues.
          # buffer_size should be set for your audio chipset.
          period_time 0
          period_size 1024
          buffer_size 8192
          # buffer_size 4096
          # buffer_size 2048
     }

     bindings {
          0 0
          1 1
     }
}

pcm.input {
     type dsnoop
     ipc_key 2048
     slave.pcm "snd_card0"

## Possible artsd full duplex fix:
#     slave {
#          period_time 0
#          period_size 1024
#          buffer_size 8192
#     }

     bindings {
          0 0
          1 1
     }
}
# Allow reading from the default device.
# Also known as record or capture.
pcm.input1 {
     type dsnoop
     ipc_key 2049
     slave.pcm "snd_card1"

## Possible artsd full duplex fix:
     slave {
          period_time 0
          period_size 1024
          buffer_size 8192
          # buffer_size 4096
          # buffer_size 2048
     }

     bindings {
          0 0
          1 1
     }
}
# This is what we want as our default device
# a fully duplex (read/write) audio device.
pcm.duplex {
     type asym
     playback.pcm "output"
     capture.pcm "input"
#     capture.pcm "input1"
}

###################
# CONVERSION PLUG #
###################
# Setting the default pcm device allows the conversion
# rate to be selected on the fly.
# duplex mode allows any alsa enabled app to read/write
# to the dmix plug (Fixes a problem with wine).
pcm.!default {
     type plug
     slave.pcm "duplex"
}
########
# AOSS #
########
# OSS dsp0 device (OSS needs only output support, duplex will break some stuff)
pcm.dsp0 {
     type plug
     slave.pcm "output"
}

#
pcm.dsp1 {
     type plug
     slave.pcm "output1"
}

Below I define some input options as described in the mplayers man page

Unfortunately the output is not working so I use sox to copy the sound and resample it and v4lctl to unmute the tv input

v4lctl volume mute off

This is how I run the mplayer in tv mode

TV_DRV="driver=v4l2:outfmt=yuy2:width=640:height=480"
TV_DEV="device=/dev/video0:input=1"
TV_NORM="norm=PAL:normid=0:chanlist=europe-west"
TV_AUD="alsa:audiorate=48000:forceaudio:amode=1:adevice=input1:volume=75:immediatemode=0"
AUD_O="driver=alsa:noblock:forceaudio:forcechan=2:audiorate=48000:device=output"
VID_O="gl2,pp=lb,denoise3d"

mplayer -ontop -framedrop -stop-xscreensaver  \
            -vf scale \
            -input conf=$HOME/.mplayer/input.conf \
            -tv $TV_DRV:$TV_DEV:$TV_AUD:$TV_NORM\
            -vo $VID_O \
            -ao $AUD_O \
            tv:// 2>&1>/dev/null

And this is how I copy and resample the sound from the tv(usb)card to the intel8x0 card

sox -q -r 48000 -w -s -c 2 -t ossdsp /dev/dsp1 -t alsa -w -s -c 2 duplex

There is a problem to copy/capture the sound from the tv card and (internelly in mplayer) to output it to the other card. If you have any suggestions please write to deloptes at yahoo dot com.

Retrieved from "http://alsa.opensrc.org/Intel8x0"

Category: ALSA modules

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