pandory500/oleavr-rgl-a500-mini-linux-3.10
2
0
Code Issues Pull requests Projects Releases Packages Wiki Activity Actions

Initial commit

Browse source
This commit is contained in:
Ole André Vadla Ravnås 2022-05-07 01:01:45 +02:00
commit 169c65d57e
51358 changed files with 23120455 additions and 0 deletions
Download patch file Download diff file Expand all files Collapse all files

66
Documentation/sound/oss/ALS Normal file
View file

@ -0,0 +1,66 @@
ALS-007/ALS-100/ALS-200 based sound cards
=========================================
Support for sound cards based around the Avance Logic
ALS-007/ALS-100/ALS-200 chip is included. These chips are a single
chip PnP sound solution which is mostly hardware compatible with the
Sound Blaster 16 card, with most differences occurring in the use of
the mixer registers. For this reason the ALS code is integrated
as part of the Sound Blaster 16 driver (adding only 800 bytes to the
SB16 driver).
To use an ALS sound card under Linux, enable the following options as
modules in the sound configuration section of the kernel config:
- 100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support
- FM synthesizer (YM3812/OPL-3) support
- standalone MPU401 support may be required for some cards; for the
ALS-007, when using isapnptools, it is required
Since the ALS-007/100/200 are PnP cards, ISAPnP support should probably be
compiled in. If kernel level PnP support is not included, isapnptools will
be required to configure the card before the sound modules are loaded.
When using kernel level ISAPnP, the kernel should correctly identify and
configure all resources required by the card when the "sb" module is
inserted. Note that the ALS-007 does not have a 16 bit DMA channel and that
the MPU401 interface on this card uses a different interrupt to the audio
section. This should all be correctly configured by the kernel; if problems
with the MPU401 interface surface, try using the standalone MPU401 module,
passing "0" as the "sb" module's "mpu_io" module parameter to prevent the
soundblaster driver attempting to register the MPU401 itself. The onboard
synth device can be accessed using the "opl3" module.
If isapnptools is used to wake up the sound card (as in 2.2.x), the settings
of the card's resources should be passed to the kernel modules ("sb", "opl3"
and "mpu401") using the module parameters. When configuring an ALS-007, be
sure to specify different IRQs for the audio and MPU401 sections - this card
requires they be different. For "sb", "io", "irq" and "dma" should be set
to the same values used to configure the audio section of the card with
isapnp. "dma16" should be explicitly set to "-1" for an ALS-007 since this
card does not have a 16 bit dma channel; if not specified the kernel will
default to using channel 5 anyway which will cause audio not to work.
"mpu_io" should be set to 0. The "io" parameter of the "opl3" module should
also agree with the setting used by isapnp. To get the MPU401 interface
working on an ALS-007 card, the "mpu401" module will be required since this
card uses separate IRQs for the audio and MPU401 sections and there is no
parameter available to pass a different IRQ to the "sb" driver (whose
inbuilt MPU401 driver would otherwise be fine). Insert the mpu401 module
passing appropriate values using the "io" and "irq" parameters.
The resulting sound driver will provide the following capabilities:
- 8 and 16 bit audio playback
- 8 and 16 bit audio recording
- Software selection of record source (line in, CD, FM, mic, master)
- Record and playback of midi data via the external MPU-401
- Playback of midi data using inbuilt FM synthesizer
- Control of the ALS-007 mixer via any OSS-compatible mixer programs.
Controls available are Master (L&R), Line in (L&R), CD (L&R),
DSP/PCM/audio out (L&R), FM (L&R) and Mic in (mono).
Jonathan Woithe
jwoithe@just42.net
30 March 1998
Modified 2000-02-26 by Dave Forrest, drf5n@virginia.edu to add ALS100/ALS200
Modified 2000-04-10 by Paul Laufer, pelaufer@csupomona.edu to add ISAPnP info.
Modified 2000-11-19 by Jonathan Woithe, jwoithe@just42.net
- updated information for kernel 2.4.x.

101
Documentation/sound/oss/AudioExcelDSP16 Normal file
View file

@ -0,0 +1,101 @@
Driver
------
Information about Audio Excel DSP 16 driver can be found in the source
file aedsp16.c
Please, read the head of the source before using it. It contain useful
information.
Configuration
-------------
The Audio Excel configuration, is now done with the standard Linux setup.
You have to configure the sound card (Sound Blaster or Microsoft Sound System)
and, if you want it, the Roland MPU-401 (do not use the Sound Blaster MPU-401,
SB-MPU401) in the main driver menu. Activate the lowlevel drivers then select
the Audio Excel hardware that you want to initialize. Check the IRQ/DMA/MIRQ
of the Audio Excel initialization: it must be the same as the SBPRO (or MSS)
setup. If the parameters are different, correct it.
I you own a Gallant's audio card based on SC-6600, activate the SC-6600 support.
If you want to change the configuration of the sound board, be sure to
check off all the configuration items before re-configure it.
Module parameters
-----------------
To use this driver as a module, you must configure some module parameters, to
set up I/O addresses, IRQ lines and DMA channels. Some parameters are
mandatory while some others are optional. Here a list of parameters you can
use with this module:
Name Description
==== ===========
MANDATORY
io I/O base address (0x220 or 0x240)
irq irq line (5, 7, 9, 10 or 11)
dma dma channel (0, 1 or 3)
OPTIONAL
mss_base I/O base address for activate MSS mode (default SBPRO)
(0x530 or 0xE80)
mpu_base I/O base address for activate MPU-401 mode
(0x300, 0x310, 0x320 or 0x330)
mpu_irq MPU-401 irq line (5, 7, 9, 10 or 0)
A configuration file in /etc/modprobe.d/ directory will have lines like this:
options opl3 io=0x388
options ad1848 io=0x530 irq=11 dma=3
options aedsp16 io=0x220 irq=11 dma=3 mss_base=0x530
Where the aedsp16 options are the options for this driver while opl3 and
ad1848 are the corresponding options for the MSS and OPL3 modules.
Loading MSS and OPL3 needs to pre load the aedsp16 module to set up correctly
the sound card. Installation dependencies must be written in configuration
files under /etc/modprobe.d/ directory:
softdep ad1848 pre: aedsp16
softdep opl3 pre: aedsp16
Then you must load the sound modules stack in this order:
sound -> aedsp16 -> [ ad1848, opl3 ]
With the above configuration, loading ad1848 or opl3 modules, will
automatically load all the sound stack.
Sound cards supported
---------------------
This driver supports the SC-6000 and SC-6600 based Gallant's sound card.
It don't support the Audio Excel DSP 16 III (try the SC-6600 code).
I'm working on the III version of the card: if someone have useful
information about it, please let me know.
For all the non-supported audio cards, you have to boot MS-DOS (or WIN95)
activating the audio card with the MS-DOS device driver, then you have to
<ctrl>-<alt>-<del> and boot Linux.
Follow these steps:
1) Compile Linux kernel with standard sound driver, using the emulation
you want, with the parameters of your audio card,
e.g. Microsoft Sound System irq10 dma3
2) Install your new kernel as the default boot kernel.
3) Boot MS-DOS and configure the audio card with the boot time device
driver, for MSS irq10 dma3 in our example.
4) <ctrl>-<alt>-<del> and boot Linux. This will maintain the DOS configuration
and will boot the new kernel with sound driver. The sound driver will find
the audio card and will recognize and attach it.
Reports on User successes
-------------------------
> Date: Mon, 29 Jul 1996 08:35:40 +0100
> From: Mr S J Greenaway <sjg95@unixfe.rl.ac.uk>
> To: riccardo@cdc8g5.cdc.polimi.it (Riccardo Facchetti)
> Subject: Re: Audio Excel DSP 16 initialization code
>
> Just to let you know got my Audio Excel (emulating a MSS) working
> with my original SB16, thanks for the driver!
Last revised: 20 August 1998
Riccardo Facchetti
fizban@tin.it

152
Documentation/sound/oss/CMI8330 Normal file
View file

@ -0,0 +1,152 @@
Documentation for CMI 8330 (SoundPRO)
-------------------------------------
Alessandro Zummo <azummo@ita.flashnet.it>
( Be sure to read Documentation/sound/oss/SoundPro too )
This adapter is now directly supported by the sb driver.
The only thing you have to do is to compile the kernel sound
support as a module and to enable kernel ISAPnP support,
as shown below.
CONFIG_SOUND=m
CONFIG_SOUND_SB=m
CONFIG_PNP=y
CONFIG_ISAPNP=y
and optionally:
CONFIG_SOUND_MPU401=m
for MPU401 support.
(I suggest you to use "make menuconfig" or "make xconfig"
for a more comfortable configuration editing)
Then you can do
modprobe sb
and everything will be (hopefully) configured.
You should get something similar in syslog:
sb: CMI8330 detected.
sb: CMI8330 sb base located at 0x220
sb: CMI8330 mpu base located at 0x330
sb: CMI8330 mail reports to Alessandro Zummo <azummo@ita.flashnet.it>
sb: ISAPnP reports CMI 8330 SoundPRO at i/o 0x220, irq 7, dma 1,5
The old documentation file follows for reference
purposes.
How to enable CMI 8330 (SOUNDPRO) soundchip on Linux
------------------------------------------
Stefan Laudat <Stefan.Laudat@asit.ro>
[Note: The CMI 8338 is unrelated and is supported by cmpci.o]
In order to use CMI8330 under Linux you just have to use a proper isapnp.conf, a good isapnp and a little bit of patience. I use isapnp 1.17, but
you may get a better one I guess at http://www.roestock.demon.co.uk/isapnptools/.
Of course you will have to compile kernel sound support as module, as shown below:
CONFIG_SOUND=m
CONFIG_SOUND_OSS=m
CONFIG_SOUND_SB=m
CONFIG_SOUND_ADLIB=m
CONFIG_SOUND_MPU401=m
# Mikro$chaft sound system (kinda useful here ;))
CONFIG_SOUND_MSS=m
The /etc/isapnp.conf file will be:
<snip below>
(READPORT 0x0203)
(ISOLATE PRESERVE)
(IDENTIFY *)
(VERBOSITY 2)
(CONFLICT (IO FATAL)(IRQ FATAL)(DMA FATAL)(MEM FATAL)) # or WARNING
(VERIFYLD N)
# WSS
(CONFIGURE CMI0001/16777472 (LD 0
(IO 0 (SIZE 8) (BASE 0x0530))
(IO 1 (SIZE 8) (BASE 0x0388))
(INT 0 (IRQ 7 (MODE +E)))
(DMA 0 (CHANNEL 0))
(NAME "CMI0001/16777472[0]{CMI8330/C3D Audio Adapter}")
(ACT Y)
))
# MPU
(CONFIGURE CMI0001/16777472 (LD 1
(IO 0 (SIZE 2) (BASE 0x0330))
(INT 0 (IRQ 11 (MODE +E)))
(NAME "CMI0001/16777472[1]{CMI8330/C3D Audio Adapter}")
(ACT Y)
))
# Joystick
(CONFIGURE CMI0001/16777472 (LD 2
(IO 0 (SIZE 8) (BASE 0x0200))
(NAME "CMI0001/16777472[2]{CMI8330/C3D Audio Adapter}")
(ACT Y)
))
# SoundBlaster
(CONFIGURE CMI0001/16777472 (LD 3
(IO 0 (SIZE 16) (BASE 0x0220))
(INT 0 (IRQ 5 (MODE +E)))
(DMA 0 (CHANNEL 1))
(DMA 1 (CHANNEL 5))
(NAME "CMI0001/16777472[3]{CMI8330/C3D Audio Adapter}")
(ACT Y)
))
(WAITFORKEY)
<end of snip>
The module sequence is trivial:
/sbin/insmod soundcore
/sbin/insmod sound
/sbin/insmod uart401
# insert this first
/sbin/insmod ad1848 io=0x530 irq=7 dma=0 soundpro=1
# The sb module is an alternative to the ad1848 (Microsoft Sound System)
# Anyhow, this is full duplex and has MIDI
/sbin/insmod sb io=0x220 dma=1 dma16=5 irq=5 mpu_io=0x330
Alma Chao <elysian@ethereal.torsion.org> suggests the following in
a /etc/modprobe.d/*conf file:
alias sound ad1848
alias synth0 opl3
options ad1848 io=0x530 irq=7 dma=0 soundpro=1
options opl3 io=0x388

34
Documentation/sound/oss/ESS Normal file
View file

@ -0,0 +1,34 @@
Documentation for the ESS AudioDrive chips
In 2.4 kernels the SoundBlaster driver not only tries to detect an ESS chip, it
tries to detect the type of ESS chip too. The correct detection of the chip
doesn't always succeed however, so unless you use the kernel isapnp facilities
(and you chip is pnp capable) the default behaviour is 2.0 behaviour which
means: only detect ES688 and ES1688.
All ESS chips now have a recording level setting. This is a need-to-have for
people who want to use their ESS for recording sound.
Every chip that's detected as a later-than-es1688 chip has a 6 bits logarithmic
master volume control.
Every chip that's detected as a ES1887 now has Full Duplex support. Made a
little testprogram that shows that is works, haven't seen a real program that
needs this however.
For ESS chips an additional parameter "esstype" can be specified. This controls
the (auto) detection of the ESS chips. It can have 3 kinds of values:
-1 Act like 2.0 kernels: only detect ES688 or ES1688.
0 Try to auto-detect the chip (may fail for ES1688)
688 The chip will be treated as ES688
1688 ,, ,, ,, ,, ,, ,, ES1688
1868 ,, ,, ,, ,, ,, ,, ES1868
1869 ,, ,, ,, ,, ,, ,, ES1869
1788 ,, ,, ,, ,, ,, ,, ES1788
1887 ,, ,, ,, ,, ,, ,, ES1887
1888 ,, ,, ,, ,, ,, ,, ES1888
Because Full Duplex is supported for ES1887 you can specify a second DMA
channel by specifying module parameter dma16. It can be one of: 0, 1, 3 or 5.

55
Documentation/sound/oss/ESS1868 Normal file
View file

@ -0,0 +1,55 @@
Documentation for the ESS1868F AudioDrive PnP sound card
The ESS1868 sound card is a PnP ESS1688-compatible 16-bit sound card.
It should be automatically detected by the Linux Kernel isapnp support when you
load the sb.o module. Otherwise you should take care of:
* The ESS1868 does not allow use of a 16-bit DMA, thus DMA 0, 1, 2, and 3
may only be used.
* isapnptools version 1.14 does work with ESS1868. Earlier versions might
not.
* Sound support MUST be compiled as MODULES, not statically linked
into the kernel.
NOTE: this is only needed when not using the kernel isapnp support!
For configuring the sound card's I/O addresses, IRQ and DMA, here is a
sample copy of the isapnp.conf directives regarding the ESS1868:
(CONFIGURE ESS1868/-1 (LD 1
(IO 0 (BASE 0x0220))
(IO 1 (BASE 0x0388))
(IO 2 (BASE 0x0330))
(DMA 0 (CHANNEL 1))
(INT 0 (IRQ 5 (MODE +E)))
(ACT Y)
))
(for a full working isapnp.conf file, remember the
(ISOLATE)
(IDENTIFY *)
at the beginning and the
(WAITFORKEY)
at the end.)
In this setup, the main card I/O is 0x0220, FM synthesizer is 0x0388, and
the MPU-401 MIDI port is located at 0x0330. IRQ is IRQ 5, DMA is channel 1.
After configuring the sound card via isapnp, to use the card you must load
the sound modules with the proper I/O information. Here is my setup:
# ESS1868F AudioDrive initialization
/sbin/modprobe sound
/sbin/insmod uart401
/sbin/insmod sb io=0x220 irq=5 dma=1 dma16=-1
/sbin/insmod mpu401 io=0x330
/sbin/insmod opl3 io=0x388
/sbin/insmod v_midi
opl3 is the FM synthesizer
/sbin/insmod opl3 io=0x388

459
Documentation/sound/oss/Introduction Normal file
View file

@ -0,0 +1,459 @@
Introduction Notes on Modular Sound Drivers and Soundcore
Wade Hampton
2/14/2001
Purpose:
========
This document provides some general notes on the modular
sound drivers and their configuration, along with the
support modules sound.o and soundcore.o.
Note, some of this probably should be added to the Sound-HOWTO!
Note, soundlow.o was present with 2.2 kernels but is not
required for 2.4.x kernels. References have been removed
to this.
Copying:
========
none
History:
========
0.1.0 11/20/1998 First version, draft
1.0.0 11/1998 Alan Cox changes, incorporation in 2.2.0
as Documentation/sound/oss/Introduction
1.1.0 6/30/1999 Second version, added notes on making the drivers,
added info on multiple sound cards of similar types,]
added more diagnostics info, added info about esd.
added info on OSS and ALSA.
1.1.1 19991031 Added notes on sound-slot- and sound-service.
(Alan Cox)
1.1.2 20000920 Modified for Kernel 2.4 (Christoph Hellwig)
1.1.3 20010214 Minor notes and corrections (Wade Hampton)
Added examples of sound-slot-0, etc.
Modular Sound Drivers:
======================
Thanks to the GREAT work by Alan Cox (alan@lxorguk.ukuu.org.uk),
[And Oleg Drokin, Thomas Sailer, Andrew Veliath and more than a few
others - not to mention Hannu's original code being designed well
enough to cope with that kind of chopping up](Alan)
the standard Linux kernels support a modular sound driver. From
Alan's comments in linux/drivers/sound/README.FIRST:
The modular sound driver patches were funded by Red Hat Software
(www.redhat.com). The sound driver here is thus a modified version of
Hannu's code. Please bear that in mind when considering the appropriate
forums for bug reporting.
The modular sound drivers may be loaded via insmod or modprobe.
To support all the various sound modules, there are two general
support modules that must be loaded first:
soundcore.o: Top level handler for the sound system, provides
a set of functions for registration of devices
by type.
sound.o: Common sound functions required by all modules.
For the specific sound modules (e.g., sb.o for the Soundblaster),
read the documentation on that module to determine what options
are available, for example IRQ, address, DMA.
Warning, the options for different cards sometime use different names
for the same or a similar feature (dma1= versus dma16=). As a last
resort, inspect the code (search for module_param).
Notes:
1. There is a new OpenSource sound driver called ALSA which is
currently under development: http://www.alsa-project.org/
The ALSA drivers support some newer hardware that may not
be supported by this sound driver and also provide some
additional features.
2. The commercial OSS driver may be obtained from the site:
http://www.opensound.com. This may be used for cards that
are unsupported by the kernel driver, or may be used
by other operating systems.
3. The enlightenment sound daemon may be used for playing
multiple sounds at the same time via a single card, eliminating
some of the requirements for multiple sound card systems. For
more information, see: http://www.tux.org/~ricdude/EsounD.html
The "esd" program may be used with the real-player and mpeg
players like mpg123 and x11amp. The newer real-player
and some games even include built-in support for ESD!
Building the Modules:
=====================
This document does not provide full details on building the
kernel, etc. The notes below apply only to making the kernel
sound modules. If this conflicts with the kernel's README,
the README takes precedence.
1. To make the kernel sound modules, cd to your /usr/src/linux
directory (typically) and type make config, make menuconfig,
or make xconfig (to start the command line, dialog, or x-based
configuration tool).
2. Select the Sound option and a dialog will be displayed.
3. Select M (module) for "Sound card support".
4. Select your sound driver(s) as a module. For ProAudio, Sound
Blaster, etc., select M (module) for OSS sound modules.
[thanks to Marvin Stodolsky <stodolsk@erols.com>]A
5. Make the kernel (e.g., make bzImage), and install the kernel.
6. Make the modules and install them (make modules; make modules_install).
Note, for 2.5.x kernels, make sure you have the newer module-init-tools
installed or modules will not be loaded properly. 2.5.x requires an
updated module-init-tools.
Plug and Play (PnP:
===================
If the sound card is an ISA PnP card, isapnp may be used
to configure the card. See the file isapnp.txt in the
directory one level up (e.g., /usr/src/linux/Documentation).
Also the 2.4.x kernels provide PnP capabilities, see the
file NEWS in this directory.
PCI sound cards are highly recommended, as they are far
easier to configure and from what I have read, they use
less resources and are more CPU efficient.
INSMOD:
=======
If loading via insmod, the common modules must be loaded in the
order below BEFORE loading the other sound modules. The card-specific
modules may then be loaded (most require parameters). For example,
I use the following via a shell script to load my SoundBlaster:
SB_BASE=0x240
SB_IRQ=9
SB_DMA=3
SB_DMA2=5
SB_MPU=0x300
#
echo Starting sound
/sbin/insmod soundcore
/sbin/insmod sound
#
echo Starting sound blaster....
/sbin/insmod uart401
/sbin/insmod sb io=$SB_BASE irq=$SB_IRQ dma=$SB_DMA dma16=$SB_DMA2 mpu_io=$SB_MP
When using sound as a module, I typically put these commands
in a file such as /root/soundon.sh.
MODPROBE:
=========
If loading via modprobe, these common files are automatically loaded when
requested by modprobe. For example, my /etc/modprobe.d/oss.conf contains:
alias sound sb
options sb io=0x240 irq=9 dma=3 dma16=5 mpu_io=0x300
All you need to do to load the module is:
/sbin/modprobe sb
Sound Status:
=============
The status of sound may be read/checked by:
cat (anyfile).au >/dev/audio
[WWH: This may not work properly for SoundBlaster PCI 128 cards
such as the es1370/1 (see the es1370/1 files in this directory)
as they do not automatically support uLaw on /dev/audio.]
The status of the modules and which modules depend on
which other modules may be checked by:
/sbin/lsmod
/sbin/lsmod should show something like the following:
sb 26280 0
uart401 5640 0 [sb]
sound 57112 0 [sb uart401]
soundcore 1968 8 [sb sound]
Removing Sound:
===============
Sound may be removed by using /sbin/rmmod in the reverse order
in which you load the modules. Note, if a program has a sound device
open (e.g., xmixer), that module (and the modules on which it
depends) may not be unloaded.
For example, I use the following to remove my Soundblaster (rmmod
in the reverse order in which I loaded the modules):
/sbin/rmmod sb
/sbin/rmmod uart401
/sbin/rmmod sound
/sbin/rmmod soundcore
When using sound as a module, I typically put these commands
in a script such as /root/soundoff.sh.
Removing Sound for use with OSS:
================================
If you get really stuck or have a card that the kernel modules
will not support, you can get a commercial sound driver from
http://www.opensound.com. Before loading the commercial sound
driver, you should do the following:
1. remove sound modules (detailed above)
2. remove the sound modules from /etc/modprobe.d/*.conf
3. move the sound modules from /lib/modules/<kernel>/misc
(for example, I make a /lib/modules/<kernel>/misc/tmp
directory and copy the sound module files to that
directory).
Multiple Sound Cards:
=====================
The sound drivers will support multiple sound cards and there
are some great applications like multitrack that support them.
Typically, you need two sound cards of different types. Note, this
uses more precious interrupts and DMA channels and sometimes
can be a configuration nightmare. I have heard reports of 3-4
sound cards (typically I only use 2). You can sometimes use
multiple PCI sound cards of the same type.
On my machine I have two sound cards (cs4232 and Soundblaster Vibra
16). By loading sound as modules, I can control which is the first
sound device (/dev/dsp, /dev/audio, /dev/mixer) and which is
the second. Normally, the cs4232 (Dell sound on the motherboard)
would be the first sound device, but I prefer the Soundblaster.
All you have to do is to load the one you want as /dev/dsp
first (in my case "sb") and then load the other one
(in my case "cs4232").
If you have two cards of the same type that are jumpered
cards or different PnP revisions, you may load the same
module twice. For example, I have a SoundBlaster vibra 16
and an older SoundBlaster 16 (jumpers). To load the module
twice, you need to do the following:
1. Copy the sound modules to a new name. For example
sb.o could be copied (or symlinked) to sb1.o for the
second SoundBlaster.
2. Make a second entry in /etc/modprobe.d/*conf, for example,
sound1 or sb1. This second entry should refer to the
new module names for example sb1, and should include
the I/O, etc. for the second sound card.
3. Update your soundon.sh script, etc.
Warning: I have never been able to get two PnP sound cards of the
same type to load at the same time. I have tried this several times
with the Soundblaster Vibra 16 cards. OSS has indicated that this
is a PnP problem.... If anyone has any luck doing this, please
send me an E-MAIL. PCI sound cards should not have this problem.a
Since this was originally release, I have received a couple of
mails from people who have accomplished this!
NOTE: In Linux 2.4 the Sound Blaster driver (and only this one yet)
supports multiple cards with one module by default.
Read the file 'Soundblaster' in this directory for details.
Sound Problems:
===============
First RTFM (including the troubleshooting section
in the Sound-HOWTO).
1) If you are having problems loading the modules (for
example, if you get device conflict errors) try the
following:
A) If you have Win95 or NT on the same computer,
write down what addresses, IRQ, and DMA channels
those were using for the same hardware. You probably
can use these addresses, IRQs, and DMA channels.
You should really do this BEFORE attempting to get
sound working!
B) Check (cat) /proc/interrupts, /proc/ioports,
and /proc/dma. Are you trying to use an address,
IRQ or DMA port that another device is using?
C) Check (cat) /proc/isapnp
D) Inspect your /var/log/messages file. Often that will
indicate what IRQ or IO port could not be obtained.
E) Try another port or IRQ. Note this may involve
using the PnP tools to move the sound card to
another location. Sometimes this is the only way
and it is more or less trial and error.
2) If you get motor-boating (the same sound or part of a
sound clip repeated), you probably have either an IRQ
or DMA conflict. Move the card to another IRQ or DMA
port. This has happened to me when playing long files
when I had an IRQ conflict.
3. If you get dropouts or pauses when playing high sample
rate files such as using mpg123 or x11amp/xmms, you may
have too slow of a CPU and may have to use the options to
play the files at 1/2 speed. For example, you may use
the -2 or -4 option on mpg123. You may also get this
when trying to play mpeg files stored on a CD-ROM
(my Toshiba T8000 PII/366 sometimes has this problem).
4. If you get "cannot access device" errors, your /dev/dsp
files, etc. may be set to owner root, mode 600. You
may have to use the command:
chmod 666 /dev/dsp /dev/mixer /dev/audio
5. If you get "device busy" errors, another program has the
sound device open. For example, if using the Enlightenment
sound daemon "esd", the "esd" program has the sound device.
If using "esd", please RTFM the docs on ESD. For example,
esddsp <program> may be used to play files via a non-esd
aware program.
6) Ask for help on the sound list or send E-MAIL to the
sound driver author/maintainer.
7) Turn on debug in drivers/sound/sound_config.h (DEB, DDB, MDB).
8) If the system reports insufficient DMA memory then you may want to
load sound with the "dmabufs=1" option. Or in /etc/conf.modules add
preinstall sound dmabufs=1
This makes the sound system allocate its buffers and hang onto them.
You may also set persistent DMA when building a 2.4.x kernel.
Configuring Sound:
==================
There are several ways of configuring your sound:
1) On the kernel command line (when using the sound driver(s)
compiled in the kernel). Check the driver source and
documentation for details.
2) On the command line when using insmod or in a bash script
using command line calls to load sound.
3) In /etc/modprobe.d/*conf when using modprobe.
4) Via Red Hat's GPL'd /usr/sbin/sndconfig program (text based).
5) Via the OSS soundconf program (with the commercial version
of the OSS driver.
6) By just loading the module and let isapnp do everything relevant
for you. This works only with a few drivers yet and - of course -
only with isapnp hardware.
And I am sure, several other ways.
Anyone want to write a linuxconf module for configuring sound?
Module Loading:
===============
When a sound card is first referenced and sound is modular, the sound system
will ask for the sound devices to be loaded. Initially it requests that
the driver for the sound system is loaded. It then will ask for
sound-slot-0, where 0 is the first sound card. (sound-slot-1 the second and
so on). Thus you can do
alias sound-slot-0 sb
To load a soundblaster at this point. If the slot loading does not provide
the desired device - for example a soundblaster does not directly provide
a midi synth in all cases then it will request "sound-service-0-n" where n
is
0 Mixer
2 MIDI
3, 4 DSP audio
For example, I use the following to load my Soundblaster PCI 128
(ES 1371) card first, followed by my SoundBlaster Vibra 16 card,
then by my TV card:
# Load the Soundblaster PCI 128 as /dev/dsp, /dev/dsp1, /dev/mixer
alias sound-slot-0 es1371
# Load the Soundblaster Vibra 16 as /dev/dsp2, /dev/mixer1
alias sound-slot-1 sb
options sb io=0x240 irq=5 dma=1 dma16=5 mpu_io=0x330
# Load the BTTV (TV card) as /dev/mixer2
alias sound-slot-2 bttv
alias sound-service-2-0 tvmixer
pre-install bttv modprobe tuner ; modprobe tvmixer
pre-install tvmixer modprobe msp3400; modprobe tvaudio
options tuner debug=0 type=8
options bttv card=0 radio=0 pll=0
For More Information (RTFM):
============================
1) Information on kernel modules: manual pages for insmod and modprobe.
2) Information on PnP, RTFM manual pages for isapnp.
3) Sound-HOWTO and Sound-Playing-HOWTO.
4) OSS's WWW site at http://www.opensound.com.
5) All the files in Documentation/sound.
6) The comments and code in linux/drivers/sound.
7) The sndconfig and rhsound documentation from Red Hat.
8) The Linux-sound mailing list: sound-list@redhat.com.
9) Enlightenment documentation (for info on esd)
http://www.tux.org/~ricdude/EsounD.html.
10) ALSA home page: http://www.alsa-project.org/
Contact Information:
====================
Wade Hampton: (whampton@staffnet.com)

1137
Documentation/sound/oss/MultiSound Normal file
View file

File diff suppressed because it is too large Load diff

6
Documentation/sound/oss/OPL3 Normal file
View file

@ -0,0 +1,6 @@
A pure OPL3 card is nice and easy to configure. Simply do
insmod opl3 io=0x388
Change the I/O address in the very unlikely case this card is differently
configured

218
Documentation/sound/oss/Opti Normal file
View file

@ -0,0 +1,218 @@
Support for the OPTi 82C931 chip
--------------------------------
Note: parts of this README file apply also to other
cards that use the mad16 driver.
Some items in this README file are based on features
added to the sound driver after Linux-2.1.91 was out.
By the time of writing this I do not know which official
kernel release will include these features.
Please do not report inconsistencies on older Linux
kernels.
The OPTi 82C931 is supported in its non-PnP mode.
Usually you do not need to set jumpers, etc. The sound driver
will check the card status and if it is required it will
force the card into a mode in which it can be programmed.
If you have another OS installed on your computer it is recommended
that Linux and the other OS use the same resources.
Also, it is recommended that resources specified in /etc/modprobe.d/*.conf
and resources specified in /etc/isapnp.conf agree.
Compiling the sound driver
--------------------------
I highly recommend that you build a modularized sound driver.
This document does not cover a sound-driver which is built in
the kernel.
Sound card support should be enabled as a module (chose m).
Answer 'm' for these items:
Generic OPL2/OPL3 FM synthesizer support (CONFIG_SOUND_ADLIB)
Microsoft Sound System support (CONFIG_SOUND_MSS)
Support for OPTi MAD16 and/or Mozart based cards (CONFIG_SOUND_MAD16)
FM synthesizer (YM3812/OPL-3) support (CONFIG_SOUND_YM3812)
The configuration menu may ask for addresses, IRQ lines or DMA
channels. If the card is used as a module the module loading
options will override these values.
For the OPTi 931 you can answer 'n' to:
Support MIDI in older MAD16 based cards (requires SB) (CONFIG_SOUND_MAD16_OLDCARD)
If you do need MIDI support in a Mozart or C928 based card you
need to answer 'm' to the above question. In that case you will
also need to answer 'm' to:
'100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support' (CONFIG_SOUND_SB)
Go on and compile your kernel and modules. Install the modules. Run depmod -a.
Using isapnptools
-----------------
In most systems with a PnP BIOS you do not need to use isapnp. The
initialization provided by the BIOS is sufficient for the driver
to pick up the card and continue initialization.
If that fails, or if you have other PnP cards, you need to use isapnp
to initialize the card.
This was tested with isapnptools-1.11 but I recommend that you use
isapnptools-1.13 (or newer). Run pnpdump to dump the information
about your PnP cards. Then edit the resulting file and select
the options of your choice. This file is normally installed as
/etc/isapnp.conf.
The driver has one limitation with respect to I/O port resources:
IO3 base must be 0x0E0C. Although isapnp allows other ports, this
address is hard-coded into the driver.
Using kmod and autoloading the sound driver
-------------------------------------------
Config files in '/etc/modprobe.d/' are used as below:
alias mixer0 mad16
alias audio0 mad16
alias midi0 mad16
alias synth0 opl3
options sb mad16=1
options mad16 irq=10 dma=0 dma16=1 io=0x530 joystick=1 cdtype=0
options opl3 io=0x388
install mad16 /sbin/modprobe -i mad16 && /sbin/ad1848_mixer_reroute 14 8 15 3 16 6
If you have an MPU daughtercard or onboard MPU you will want to add to the
"options mad16" line - eg
options mad16 irq=5 dma=0 dma16=3 io=0x530 mpu_io=0x330 mpu_irq=9
To set the I/O and IRQ of the MPU.
Explain:
alias mixer0 mad16
alias audio0 mad16
alias midi0 mad16
alias synth0 opl3
When any sound device is opened the kernel requests auto-loading
of char-major-14. There is a built-in alias that translates this
request to loading the main sound module.
The sound module in its turn will request loading of a sub-driver
for mixer, audio, midi or synthesizer device. The first 3 are
supported by the mad16 driver. The synth device is supported
by the opl3 driver.
There is currently no way to autoload the sound device driver
if more than one card is installed.
options sb mad16=1
This is left for historical reasons. If you enable the
config option 'Support MIDI in older MAD16 based cards (requires SB)'
or if you use an older mad16 driver it will force loading of the
SoundBlaster driver. This option tells the SB driver not to look
for a SB card but to wait for the mad16 driver.
options mad16 irq=10 dma=0 dma16=1 io=0x530 joystick=1 cdtype=0
options opl3 io=0x388
post-install mad16 /sbin/ad1848_mixer_reroute 14 8 15 3 16 6
This sets resources and options for the mad16 and opl3 drivers.
I use two DMA channels (only one is required) to enable full duplex.
joystick=1 enables the joystick port. cdtype=0 disables the cd port.
You can also set mpu_io and mpu_irq in the mad16 options for the
uart401 driver.
This tells modprobe to run /sbin/ad1848_mixer_reroute after
mad16 is successfully loaded and initialized. The source
for ad1848_mixer_reroute is appended to the end of this readme
file. It is impossible for the sound driver to know the actual
connections to the mixer. The 3 inputs intended for cd, synth
and line-in are mapped to the generic inputs line1, line2 and
line3. This program reroutes these mixer channels to their
right names (note the right mapping depends on the actual sound
card that you use).
The numeric parameters mean:
14=line1 8=cd - reroute line1 to the CD input.
15=line2 3=synth - reroute line2 to the synthesizer input.
16=line3 6=line - reroute line3 to the line input.
For reference on other input names look at the file
/usr/include/linux/soundcard.h.
Using a joystick
-----------------
You must enable a joystick in the mad16 options. (also
in /etc/isapnp.conf if you use it).
Tested with regular analog joysticks.
A CDROM drive connected to the sound card
-----------------------------------------
The 82C931 chip has support only for secondary ATAPI cdrom.
(cdtype=8). Loading the mad16 driver resets the C931 chip
and if a cdrom was already mounted it may cause a complete
system hang. Do not use the sound card if you have an alternative.
If you do use the sound card it is important that you load
the mad16 driver (use "modprobe mad16" to prevent auto-unloading)
before the cdrom is accessed the first time.
Using the sound driver built-in to the kernel may help here, but...
Most new systems have a PnP BIOS and also two IDE controllers.
The IDE controller on the sound card may be needed only on older
systems (which have only one IDE controller) but these systems
also do not have a PnP BIOS - requiring isapnptools and a modularized
driver.
Known problems
--------------
1. See the section on "A CDROM drive connected to the sound card".
2. On my system the codec cannot capture companded sound samples.
(eg., recording from /dev/audio). When any companded capture is
requested I get stereo-16 bit samples instead. Playback of
companded samples works well. Apparently this problem is not common
to all C931 based cards. I do not know how to identify cards that
have this problem.
Source for ad1848_mixer_reroute.c
---------------------------------
#include <stdio.h>
#include <fcntl.h>
#include <linux/soundcard.h>
static char *mixer_names[SOUND_MIXER_NRDEVICES] =
SOUND_DEVICE_LABELS;
int
main(int argc, char **argv) {
int val, from, to;
int i, fd;
fd = open("/dev/mixer", O_RDWR);
if(fd < 0) {
perror("/dev/mixer");
return 1;
}
for(i = 2; i < argc; i += 2) {
from = atoi(argv[i-1]);
to = atoi(argv[i]);
if(to == SOUND_MIXER_NONE)
fprintf(stderr, "%s: turning off mixer %s\n",
argv[0], mixer_names[to]);
else
fprintf(stderr, "%s: rerouting mixer %s to %s\n",
argv[0], mixer_names[from], mixer_names[to]);
val = from << 8 | to;
if(ioctl(fd, SOUND_MIXER_PRIVATE2, &val)) {
perror("AD1848 mixer reroute");
return 1;
}
}
return 0;
}

162
Documentation/sound/oss/PAS16 Normal file
View file

@ -0,0 +1,162 @@
Pro Audio Spectrum 16 for 2.3.99 and later
=========================================
by Thomas Molina (tmolina@home.com)
last modified 3 Mar 2001
Acknowledgement to Axel Boldt (boldt@math.ucsb.edu) for stuff taken
from Configure.help, Riccardo Facchetti for stuff from README.OSS,
and others whose names I could not find.
This documentation is relevant for the PAS16 driver (pas2_card.c and
friends) under kernel version 2.3.99 and later. If you are
unfamiliar with configuring sound under Linux, please read the
Sound-HOWTO, Documentation/sound/oss/Introduction and other
relevant docs first.
The following information is relevant information from README.OSS
and legacy docs for the Pro Audio Spectrum 16 (PAS16):
==================================================================
The pas2_card.c driver supports the following cards --
Pro Audio Spectrum 16 (PAS16) and compatibles:
Pro Audio Spectrum 16
Pro Audio Studio 16
Logitech Sound Man 16
NOTE! The original Pro Audio Spectrum as well as the PAS+ are not
and will not be supported by the driver.
The sound driver configuration dialog
-------------------------------------
Sound configuration starts by making some yes/no questions. Be careful
when answering to these questions since answering y to a question may
prevent some later ones from being asked. For example don't answer y to
the question about (PAS16) if you don't really have a PAS16. Sound
configuration may also be made modular by answering m to configuration
options presented.
Note also that all questions may not be asked. The configuration program
may disable some questions depending on the earlier choices. It may also
select some options automatically as well.
"ProAudioSpectrum 16 support",
- Answer 'y'_ONLY_ if you have a Pro Audio Spectrum _16_,
Pro Audio Studio 16 or Logitech SoundMan 16 (be sure that
you read the above list correctly). Don't answer 'y' if you
have some other card made by Media Vision or Logitech since they
are not PAS16 compatible.
NOTE! Since 3.5-beta10 you need to enable SB support (next question)
if you want to use the SB emulation of PAS16. It's also possible to
the emulation if you want to use a true SB card together with PAS16
(there is another question about this that is asked later).
"Generic OPL2/OPL3 FM synthesizer support",
- Answer 'y' if your card has a FM chip made by Yamaha (OPL2/OPL3/OPL4).
The PAS16 has an OPL3-compatible FM chip.
With PAS16 you can use two audio device files at the same time. /dev/dsp (and
/dev/audio) is connected to the 8/16 bit native codec and the /dev/dsp1 (and
/dev/audio1) is connected to the SB emulation (8 bit mono only).
The new stuff for 2.3.99 and later
============================================================================
The following configuration options are relevant to configuring the PAS16:
Sound card support
CONFIG_SOUND
If you have a sound card in your computer, i.e. if it can say more
than an occasional beep, say Y. Be sure to have all the information
about your sound card and its configuration down (I/O port,
interrupt and DMA channel), because you will be asked for it.
You want to read the Sound-HOWTO, available from
http://www.tldp.org/docs.html#howto . General information
about the modular sound system is contained in the files
Documentation/sound/oss/Introduction. The file
Documentation/sound/oss/README.OSS contains some slightly outdated but
still useful information as well.
OSS sound modules
CONFIG_SOUND_OSS
OSS is the Open Sound System suite of sound card drivers. They make
sound programming easier since they provide a common API. Say Y or M
here (the module will be called sound.o) if you haven't found a
driver for your sound card above, then pick your driver from the
list below.
Persistent DMA buffers
CONFIG_SOUND_DMAP
Linux can often have problems allocating DMA buffers for ISA sound
cards on machines with more than 16MB of RAM. This is because ISA
DMA buffers must exist below the 16MB boundary and it is quite
possible that a large enough free block in this region cannot be
found after the machine has been running for a while. If you say Y
here the DMA buffers (64Kb) will be allocated at boot time and kept
until the shutdown. This option is only useful if you said Y to
"OSS sound modules", above. If you said M to "OSS sound modules"
then you can get the persistent DMA buffer functionality by passing
the command-line argument "dmabuf=1" to the sound.o module.
Say y here for PAS16.
ProAudioSpectrum 16 support
CONFIG_SOUND_PAS
Answer Y only if you have a Pro Audio Spectrum 16, ProAudio Studio
16 or Logitech SoundMan 16 sound card. Don't answer Y if you have
some other card made by Media Vision or Logitech since they are not
PAS16 compatible. It is not necessary to enable the separate
Sound Blaster support; it is included in the PAS driver.
If you compile the driver into the kernel, you have to add
"pas2=<io>,<irq>,<dma>,<dma2>,<sbio>,<sbirq>,<sbdma>,<sbdma2>
to the kernel command line.
FM Synthesizer (YM3812/OPL-3) support
CONFIG_SOUND_YM3812
Answer Y if your card has a FM chip made by Yamaha (OPL2/OPL3/OPL4).
Answering Y is usually a safe and recommended choice, however some
cards may have software (TSR) FM emulation. Enabling FM support with
these cards may cause trouble (I don't currently know of any such
cards, however).
Please read the file Documentation/sound/oss/OPL3 if your card has an
OPL3 chip.
If you compile the driver into the kernel, you have to add
"opl3=<io>" to the kernel command line.
If you compile your drivers into the kernel, you MUST configure
OPL3 support as a module for PAS16 support to work properly.
You can then get OPL3 functionality by issuing the command:
insmod opl3
In addition, you must either add the following line to
/etc/modprobe.d/*.conf:
options opl3 io=0x388
or else add the following line to /etc/lilo.conf:
opl3=0x388
EXAMPLES
===================================================================
To use the PAS16 in my computer I have enabled the following sound
configuration options:
CONFIG_SOUND=y
CONFIG_SOUND_OSS=y
CONFIG_SOUND_TRACEINIT=y
CONFIG_SOUND_DMAP=y
CONFIG_SOUND_PAS=y
CONFIG_SOUND_SB=n
CONFIG_SOUND_YM3812=m
I have also included the following append line in /etc/lilo.conf:
append="pas2=0x388,10,3,-1,0x220,5,1,-1 sb=0x220,5,1,-1 opl3=0x388"
The io address of 0x388 is default configuration on the PAS16. The
irq of 10 and dma of 3 may not match your installation. The above
configuration enables PAS16, 8-bit Soundblaster and OPL3
functionality. If Soundblaster functionality is not desired, the
following line would be appropriate:
append="pas2=0x388,10,3,-1,0,-1,-1,-1 opl3=0x388"
If sound is built totally modular, the above options may be
specified in /etc/modprobe.d/*.conf for pas2, sb and opl3
respectively.

41
Documentation/sound/oss/PSS Normal file
View file

@ -0,0 +1,41 @@
The PSS cards and other ECHO based cards provide an onboard DSP with
downloadable programs and also has an AD1848 "Microsoft Sound System"
device. The PSS driver enables MSS and MPU401 modes of the card. SB
is not enabled since it doesn't work concurrently with MSS.
If you build this driver as a module then the driver takes the following
parameters
pss_io. The I/O base the PSS card is configured at (normally 0x220
or 0x240)
mss_io The base address of the Microsoft Sound System interface.
This is normally 0x530, but may be 0x604 or other addresses.
mss_irq The interrupt assigned to the Microsoft Sound System
emulation. IRQ's 3,5,7,9,10,11 and 12 are available. If you
get IRQ errors be sure to check the interrupt is set to
"ISA/Legacy" in the BIOS on modern machines.
mss_dma The DMA channel used by the Microsoft Sound System.
This can be 0, 1, or 3. DMA 0 is not available on older
machines and will cause a crash on them.
mpu_io The MPU emulation base address. This sets the base of the
synthesizer. It is typically 0x330 but can be altered.
mpu_irq The interrupt to use for the synthesizer. It must differ
from the IRQ used by the Microsoft Sound System port.
The mpu_io/mpu_irq fields are optional. If they are not specified the
synthesizer parts are not configured.
When the module is loaded it looks for a file called
/etc/sound/pss_synth. This is the firmware file from the DOS install disks.
This fil holds a general MIDI emulation. The file expected is called
genmidi.ld on newer DOS driver install disks and synth.ld on older ones.
You can also load alternative DSP algorithms into the card if you wish. One
alternative driver can be found at http://www.mpg123.de/

88
Documentation/sound/oss/PSS-updates Normal file
View file

@ -0,0 +1,88 @@
This file contains notes for users of PSS sound cards who wish to use the
newly added features of the newest version of this driver.
The major enhancements present in this new revision of this driver is the
addition of two new module parameters that allow you to take full advantage of
all the features present on your PSS sound card. These features include the
ability to enable both the builtin CDROM and joystick ports.
pss_enable_joystick
This parameter is basically a flag. A 0 will leave the joystick port
disabled, while a non-zero value would enable the joystick port. The default
setting is pss_enable_joystick=0 as this keeps this driver fully compatible
with systems that were using previous versions of this driver. If you wish to
enable the joystick port you will have to add pss_enable_joystick=1 as an
argument to the driver. To actually use the joystick port you will then have
to load the joystick driver itself. Just remember to load the joystick driver
AFTER the pss sound driver.
pss_cdrom_port
This parameter takes a port address as its parameter. Any available port
address can be specified to enable the CDROM port, except for 0x0 and -1 as
these values would leave the port disabled. Like the joystick port, the cdrom
port will require that an appropriate CDROM driver be loaded before you can make
use of the newly enabled CDROM port. Like the joystick port option above,
remember to load the CDROM driver AFTER the pss sound driver. While it may
differ on some PSS sound cards, all the PSS sound cards that I have seen have a
builtin Wearnes CDROM port. If this is the case with your PSS sound card you
should load aztcd with the appropriate port option that matches the port you
assigned to the CDROM port when you loaded your pss sound driver. (ex.
modprobe pss pss_cdrom_port=0x340 && modprobe aztcd aztcd=0x340) The default
setting of this parameter leaves the CDROM port disabled to maintain full
compatibility with systems using previous versions of this driver.
Other options have also been added for the added convenience and utility
of the user. These options are only available if this driver is loaded as a
module.
pss_no_sound
This module parameter is a flag that can be used to tell the driver to
just configure non-sound components. 0 configures all components, a non-0
value will only attept to configure the CDROM and joystick ports. This
parameter can be used by a user who only wished to use the builtin joystick
and/or CDROM port(s) of his PSS sound card. If this driver is loaded with this
parameter and with the parameter below set to true then a user can safely unload
this driver with the following command "rmmod pss && rmmod ad1848 && rmmod
mpu401 && rmmod sound && rmmod soundcore" and retain the full functionality of
his CDROM and/or joystick port(s) while gaining back the memory previously used
by the sound drivers. This default setting of this parameter is 0 to retain
full behavioral compatibility with previous versions of this driver.
pss_keep_settings
This parameter can be used to specify whether you want the driver to reset
all emulations whenever its unloaded. This can be useful for those who are
sharing resources (io ports, IRQ's, DMA's) between different ISA cards. This
flag can also be useful in that future versions of this driver may reset all
emulations by default on the driver's unloading (as it probably should), so
specifying it now will ensure that all future versions of this driver will
continue to work as expected. The default value of this parameter is 1 to
retain full behavioral compatibility with previous versions of this driver.
pss_firmware
This parameter can be used to specify the file containing the firmware
code so that a user could tell the driver where that file is located instead
of having to put it in a predefined location with a predefined name. The
default setting of this parameter is "/etc/sound/pss_synth" as this was the
path and filename the hardcoded value in the previous versions of this driver.
Examples:
# Normal PSS sound card system, loading of drivers.
# Should be specified in an rc file (ex. Slackware uses /etc/rc.d/rc.modules).
/sbin/modprobe pss pss_io=0x220 mpu_io=0x338 mpu_irq=9 mss_io=0x530 mss_irq=10 mss_dma=1 pss_cdrom_port=0x340 pss_enable_joystick=1
/sbin/modprobe aztcd aztcd=0x340
/sbin/modprobe joystick
# System using the PSS sound card just for its CDROM and joystick ports.
# Should be specified in an rc file (ex. Slackware uses /etc/rc.d/rc.modules).
/sbin/modprobe pss pss_io=0x220 pss_cdrom_port=0x340 pss_enable_joystick=1 pss_no_sound=1
/sbin/rmmod pss && /sbin/rmmod ad1848 && /sbin/rmmod mpu401 && /sbin/rmmod sound && /sbin/rmmod soundcore # This line not needed, but saves memory.
/sbin/modprobe aztcd aztcd=0x340
/sbin/modprobe joystick

1455
Documentation/sound/oss/README.OSS Normal file
View file

File diff suppressed because it is too large Load diff

106
Documentation/sound/oss/README.modules Normal file
View file

@ -0,0 +1,106 @@
Building a modular sound driver
================================
The following information is current as of linux-2.1.85. Check the other
readme files, especially README.OSS, for information not specific to
making sound modular.
First, configure your kernel. This is an idea of what you should be
setting in the sound section:
<M> Sound card support
<M> 100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support
I have SoundBlaster. Select your card from the list.
<M> Generic OPL2/OPL3 FM synthesizer support
<M> FM synthesizer (YM3812/OPL-3) support
If you don't set these, you will probably find you can play .wav files
but not .midi. As the help for them says, set them unless you know your
card does not use one of these chips for FM support.
Once you are configured, make zlilo, modules, modules_install; reboot.
Note that it is no longer necessary or possible to configure sound in the
drivers/sound dir. Now one simply configures and makes one's kernel and
modules in the usual way.
Then, add to your /etc/modprobe.d/oss.conf something like:
alias char-major-14-* sb
install sb /sbin/modprobe -i sb && /sbin/modprobe adlib_card
options sb io=0x220 irq=7 dma=1 dma16=5 mpu_io=0x330
options adlib_card io=0x388 # FM synthesizer
Alternatively, if you have compiled in kernel level ISAPnP support:
alias char-major-14 sb
softdep sb post: adlib_card
options adlib_card io=0x388
The effect of this is that the sound driver and all necessary bits and
pieces autoload on demand, assuming you use kerneld (a sound choice) and
autoclean when not in use. Also, options for the device drivers are
set. They will not work without them. Change as appropriate for your card.
If you are not yet using the very cool kerneld, you will have to "modprobe
-k sb" yourself to get things going. Eventually things may be fixed so
that this kludgery is not necessary; for the time being, it seems to work
well.
Replace 'sb' with the driver for your card, and give it the right
options. To find the filename of the driver, look in
/lib/modules/<kernel-version>/misc. Mine looks like:
adlib_card.o # This is the generic OPLx driver
opl3.o # The OPL3 driver
sb.o # <<The SoundBlaster driver. Yours may differ.>>
sound.o # The sound driver
uart401.o # Used by sb, maybe other cards
Whichever card you have, try feeding it the options that would be the
default if you were making the driver wired, not as modules. You can
look at function referred to by module_init() for the card to see what
args are expected.
Note that at present there is no way to configure the io, irq and other
parameters for the modular drivers as one does for the wired drivers.. One
needs to pass the modules the necessary parameters as arguments, either
with /etc/modprobe.d/*.conf or with command-line args to modprobe, e.g.
modprobe sb io=0x220 irq=7 dma=1 dma16=5 mpu_io=0x330
modprobe adlib_card io=0x388
recommend using /etc/modprobe.d/*.conf.
Persistent DMA Buffers:
The sound modules normally allocate DMA buffers during open() and
deallocate them during close(). Linux can often have problems allocating
DMA buffers for ISA cards on machines with more than 16MB RAM. This is
because ISA DMA buffers must exist below the 16MB boundary and it is quite
possible that we can't find a large enough free block in this region after
the machine has been running for any amount of time. The way to avoid this
problem is to allocate the DMA buffers during module load and deallocate
them when the module is unloaded. For this to be effective we need to load
the sound modules right after the kernel boots, either manually or by an
init script, and keep them around until we shut down. This is a little
wasteful of RAM, but it guarantees that sound always works.
To make the sound driver use persistent DMA buffers we need to pass the
sound.o module a "dmabuf=1" command-line argument. This is normally done
in /etc/modprobe.d/*.conf files like so:
options sound dmabuf=1
If you have 16MB or less RAM or a PCI sound card, this is wasteful and
unnecessary. It is possible that machine with 16MB or less RAM will find
this option useful, but if your machine is so memory-starved that it
cannot find a 64K block free, you will be wasting even more RAM by keeping
the sound modules loaded and the DMA buffers allocated when they are not
needed. The proper solution is to upgrade your RAM. But you do also have
this improper solution as well. Use it wisely.
I'm afraid I know nothing about anything but my setup, being more of a
text-mode guy anyway. If you have options for other cards or other helpful
hints, send them to me, Jim Bray, jb@as220.org, http://as220.org/jb.

107
Documentation/sound/oss/README.ymfsb Normal file
View file

@ -0,0 +1,107 @@
Legacy audio driver for YMF7xx PCI cards.
FIRST OF ALL
============
This code references YAMAHA's sample codes and data sheets.
I respect and thank for all people they made open the information
about YMF7xx cards.
And this codes heavily based on Jeff Garzik <jgarzik@pobox.com>'s
old VIA 82Cxxx driver (via82cxxx.c). I also respect him.
DISCLIMER
=========
This driver is currently at early ALPHA stage. It may cause serious
damage to your computer when used.
PLEASE USE IT AT YOUR OWN RISK.
ABOUT THIS DRIVER
=================
This code enables you to use your YMF724[A-F], YMF740[A-C], YMF744, YMF754
cards. When enabled, your card acts as "SoundBlaster Pro" compatible card.
It can only play 22.05kHz / 8bit / Stereo samples, control external MIDI
port.
If you want to use your card as recent "16-bit" card, you should use
Alsa or OSS/Linux driver. Of course you can write native PCI driver for
your cards :)
USAGE
=====
# modprobe ymfsb (options)
OPTIONS FOR MODULE
==================
io : SB base address (0x220, 0x240, 0x260, 0x280)
synth_io : OPL3 base address (0x388, 0x398, 0x3a0, 0x3a8)
dma : DMA number (0,1,3)
master_volume: AC'97 PCM out Vol (0-100)
spdif_out : SPDIF-out flag (0:disable 1:enable)
These options will change in future...
FREQUENCY
=========
When playing sounds via this driver, you will hear its pitch is slightly
lower than original sounds. Since this driver recognizes your card acts
with 21.739kHz sample rates rather than 22.050kHz (I think it must be
hardware restriction). So many players become tone deafness.
To prevent this, you should express some options to your sound player
that specify correct sample frequency. For example, to play your MP3 file
correctly with mpg123, specify the frequency like following:
% mpg123 -r 21739 foo.mp3
SPDIF OUT
=========
With installing modules with option 'spdif_out=1', you can enjoy your
sounds from SPDIF-out of your card (if it had).
Its Fs is fixed to 48kHz (It never means the sample frequency become
up to 48kHz. All sounds via SPDIF-out also 22kHz samples). So your
digital-in capable components has to be able to handle 48kHz Fs.
COPYING
=======
This program is free software; you can redistribute it and/or modify
it under the terms of the GNU General Public License as published by
the Free Software Foundation; either version 2, or (at your option)
any later version.
This program is distributed in the hope that it will be useful, but
WITHOUT ANY WARRANTY; without even the implied warranty of
MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
General Public License for more details.
You should have received a copy of the GNU General Public License
along with this program; if not, write to the Free Software
Foundation, Inc., 675 Mass Ave, Cambridge, MA 02139, USA.
TODO
====
* support for multiple cards
(set the different SB_IO,MPU_IO,OPL_IO for each cards)
* support for OPL (dmfm) : There will be no requirements... :-<
AUTHOR
======
Daisuke Nagano <breeze.nagano@nifty.ne.jp>

105
Documentation/sound/oss/SoundPro Normal file
View file

@ -0,0 +1,105 @@
Documentation for the SoundPro CMI8330 extensions in the WSS driver (ad1848.o)
------------------------------------------------------------------------------
( Be sure to read Documentation/sound/oss/CMI8330 too )
Ion Badulescu, ionut@cs.columbia.edu
February 24, 1999
(derived from the OPL3-SA2 documentation by Scott Murray)
The SoundPro CMI8330 (ISA) is a chip usually found on some Taiwanese
motherboards. The official name in the documentation is CMI8330, SoundPro
is the nickname and the big inscription on the chip itself.
The chip emulates a WSS as well as a SB16, but it has certain differences
in the mixer section which require separate support. It also emulates an
MPU401 and an OPL3 synthesizer, so you probably want to enable support
for these, too.
The chip identifies itself as an AD1848, but its mixer is significantly
more advanced than the original AD1848 one. If your system works with
either WSS or SB16 and you are having problems with some mixer controls
(no CD audio, no line-in, etc), you might want to give this driver a try.
Detection should work, but it hasn't been widely tested, so it might still
mis-identify the chip. You can still force soundpro=1 in the modprobe
parameters for ad1848. Please let me know if it happens to you, so I can
adjust the detection routine.
The chip is capable of doing full-duplex, but since the driver sees it as an
AD1848, it cannot take advantage of this. Moreover, the full-duplex mode is
not achievable through the WSS interface, b/c it needs a dma16 line which is
assigned only to the SB16 subdevice (with isapnp). Windows documentation
says the user must use WSS Playback and SB16 Recording for full-duplex, so
it might be possible to do the same thing under Linux. You can try loading
up both ad1848 and sb then use one for playback and the other for
recording. I don't know if this works, b/c I haven't tested it. Anyway, if
you try it, be very careful: the SB16 mixer *mostly* works, but certain
settings can have unexpected effects. Use the WSS mixer for best results.
There is also a PCI SoundPro chip. I have not seen this chip, so I have
no idea if the driver will work with it. I suspect it won't.
As with PnP cards, some configuration is required. There are two ways
of doing this. The most common is to use the isapnptools package to
initialize the card, and use the kernel module form of the sound
subsystem and sound drivers. Alternatively, some BIOS's allow manual
configuration of installed PnP devices in a BIOS menu, which should
allow using the non-modular sound drivers, i.e. built into the kernel.
Since in this latter case you cannot use module parameters, you will
have to enable support for the SoundPro at compile time.
The IRQ and DMA values can be any that are considered acceptable for a
WSS. Assuming you've got isapnp all happy, then you should be able to
do something like the following (which *must* match the isapnp/BIOS
configuration):
modprobe ad1848 io=0x530 irq=11 dma=0 soundpro=1
-and maybe-
modprobe sb io=0x220 irq=5 dma=1 dma16=5
-then-
modprobe mpu401 io=0x330 irq=9
modprobe opl3 io=0x388
If all goes well and you see no error messages, you should be able to
start using the sound capabilities of your system. If you get an
error message while trying to insert the module(s), then make
sure that the values of the various arguments match what you specified
in your isapnp configuration file, and that there is no conflict with
another device for an I/O port or interrupt. Checking the contents of
/proc/ioports and /proc/interrupts can be useful to see if you're
butting heads with another device.
If you do not see the chipset version message, and none of the other
messages present in the system log are helpful, try adding 'debug=1'
to the ad1848 parameters, email me the syslog results and I'll do
my best to help.
Lastly, if you're using modules and want to set up automatic module
loading with kmod, the kernel module loader, here is the section I
currently use in my conf.modules file:
# Sound
post-install sound modprobe -k ad1848; modprobe -k mpu401; modprobe -k opl3
options ad1848 io=0x530 irq=11 dma=0
options sb io=0x220 irq=5 dma=1 dma16=5
options mpu401 io=0x330 irq=9
options opl3 io=0x388
The above ensures that ad1848 will be loaded whenever the sound system
is being used.
Good luck.
Ion
NOT REALLY TESTED:
- recording
- recording device selection
- full-duplex
TODO:
- implement mixer support for surround, loud, digital CD switches.
- come up with a scheme which allows recording volumes for each subdevice.
This is a major OSS API change.

53
Documentation/sound/oss/Soundblaster Normal file
View file

@ -0,0 +1,53 @@
modprobe sound
insmod uart401
insmod sb ...
This loads the driver for the Sound Blaster and assorted clones. Cards that
are covered by other drivers should not be using this driver.
The Sound Blaster module takes the following arguments
io I/O address of the Sound Blaster chip (0x220,0x240,0x260,0x280)
irq IRQ of the Sound Blaster chip (5,7,9,10)
dma 8-bit DMA channel for the Sound Blaster (0,1,3)
dma16 16-bit DMA channel for SB16 and equivalent cards (5,6,7)
mpu_io I/O for MPU chip if present (0x300,0x330)
sm_games=1 Set if you have a Logitech soundman games
acer=1 Set this to detect cards in some ACER notebooks
mwave_bug=1 Set if you are trying to use this driver with mwave (see on)
type Use this to specify a specific card type
The following arguments are taken if ISAPnP support is compiled in
isapnp=0 Set this to disable ISAPnP detection (use io=0xXXX etc. above)
multiple=0 Set to disable detection of multiple Soundblaster cards.
Consider it a bug if this option is needed, and send in a
report.
pnplegacy=1 Set this to be able to use a PnP card(s) along with a single
non-PnP (legacy) card. Above options for io, irq, etc. are
needed, and will apply only to the legacy card.
reverse=1 Reverses the order of the search in the PnP table.
uart401=1 Set to enable detection of mpu devices on some clones.
isapnpjump=n Jumps to slot n in the driver's PnP table. Use the source,
Luke.
You may well want to load the opl3 driver for synth music on most SB and
clone SB devices
insmod opl3 io=0x388
Using Mwave
To make this driver work with Mwave you must set mwave_bug. You also need
to warm boot from DOS/Windows with the required firmware loaded under this
OS. IBM are being difficult about documenting how to load this firmware.
Avance Logic ALS007
This card is supported; see the separate file ALS007 for full details.
Avance Logic ALS100
This card is supported; setup should be as for a standard Sound Blaster 16.
The driver will identify the audio device as a "Sound Blaster 16 (ALS-100)".

26
Documentation/sound/oss/Tropez+ Normal file
View file

@ -0,0 +1,26 @@
From: Paul Barton-Davis <pbd@op.net>
Here is the configuration I use with a Tropez+ and my modular
driver:
alias char-major-14 wavefront
alias synth0 wavefront
alias mixer0 cs4232
alias audio0 cs4232
pre-install wavefront modprobe "-k" "cs4232"
post-install wavefront modprobe "-k" "opl3"
options wavefront io=0x200 irq=9
options cs4232 synthirq=9 synthio=0x200 io=0x530 irq=5 dma=1 dma2=0
options opl3 io=0x388
Things to note:
the wavefront options "io" and "irq" ***MUST*** match the "synthio"
and "synthirq" cs4232 options.
you can do without the opl3 module if you don't
want to use the OPL/[34] synth on the soundcard
the opl3 io parameter is conventionally not adjustable.
Please see drivers/sound/README.wavefront for more details.

80
Documentation/sound/oss/VIBRA16 Normal file
View file

@ -0,0 +1,80 @@
Sound Blaster 16X Vibra addendum
--------------------------------
by Marius Ilioaea <mariusi@protv.ro>
Stefan Laudat <stefan@asit.ro>
Sat Mar 6 23:55:27 EET 1999
Hello again,
Playing with a SB Vibra 16x soundcard we found it very difficult
to setup because the kernel reported a lot of DMA errors and wouldn't
simply play any sound.
A good starting point is that the vibra16x chip full-duplex facility
is neither still exploited by the sb driver found in the linux kernel
(tried it with a 2.2.2-ac7), nor in the commercial OSS package (it reports
it as half-duplex soundcard). Oh, I almost forgot, the RedHat sndconfig
failed detecting it ;)
So, the big problem still remains, because the sb module wants a
8-bit and a 16-bit dma, which we could not allocate for vibra... it supports
only two 8-bit dma channels, the second one will be passed to the module
as a 16 bit channel, the kernel will yield about that but everything will
be okay, trust us.
The only inconvenient you may find is that you will have
some sound playing jitters if you have HDD dma support enabled - but this
will happen with almost all soundcards...
A fully working isapnp.conf is just here:
<snip here>
(READPORT 0x0203)
(ISOLATE PRESERVE)
(IDENTIFY *)
(VERBOSITY 2)
(CONFLICT (IO FATAL)(IRQ FATAL)(DMA FATAL)(MEM FATAL)) # or WARNING
# SB 16 and OPL3 devices
(CONFIGURE CTL00f0/-1 (LD 0
(INT 0 (IRQ 5 (MODE +E)))
(DMA 0 (CHANNEL 1))
(DMA 1 (CHANNEL 3))
(IO 0 (SIZE 16) (BASE 0x0220))
(IO 2 (SIZE 4) (BASE 0x0388))
(NAME "CTL00f0/-1[0]{Audio }")
(ACT Y)
))
# Joystick device - only if you need it :-/
(CONFIGURE CTL00f0/-1 (LD 1
(IO 0 (SIZE 1) (BASE 0x0200))
(NAME "CTL00f0/-1[1]{Game }")
(ACT Y)
))
(WAITFORKEY)
<end of snipping>
So, after a good kernel modules compilation and a 'depmod -a kernel_ver'
you may want to:
modprobe sb io=0x220 irq=5 dma=1 dma16=3
Or, take the hard way:
modprobe soundcore
modprobe sound
modprobe uart401
modprobe sb io=0x220 irq=5 dma=1 dma16=3
# do you need MIDI?
modprobe opl3=0x388
Just in case, the kernel sound support should be:
CONFIG_SOUND=m
CONFIG_SOUND_OSS=m
CONFIG_SOUND_SB=m
Enjoy your new noisy Linux box! ;)

170
Documentation/sound/oss/WaveArtist Normal file
View file

@ -0,0 +1,170 @@
(the following is from the armlinux CVS)
WaveArtist mixer and volume levels can be accessed via these commands:
nn30 read registers nn, where nn = 00 - 09 for mixer settings
0a - 13 for channel volumes
mm31 write the volume setting in pairs, where mm = (nn - 10) / 2
rr32 write the mixer settings in pairs, where rr = nn/2
xx33 reset all settings to default
0y34 select mono source, y=0 = left, y=1 = right
bits
nn 15 14 13 12 11 10 9 8 7 6 5 4 3 2 1 0
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
00 | 0 | 0 0 1 1 | left line mixer gain | left aux1 mixer gain |lmute|
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
01 | 0 | 0 1 0 1 | left aux2 mixer gain | right 2 left mic gain |mmute|
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
02 | 0 | 0 1 1 1 | left mic mixer gain | left mic | left mixer gain |dith |
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
03 | 0 | 1 0 0 1 | left mixer input select |lrfg | left ADC gain |
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
04 | 0 | 1 0 1 1 | right line mixer gain | right aux1 mixer gain |rmute|
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
05 | 0 | 1 1 0 1 | right aux2 mixer gain | left 2 right mic gain |test |
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
06 | 0 | 1 1 1 1 | right mic mixer gain | right mic |right mixer gain |rbyps|
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
07 | 1 | 0 0 0 1 | right mixer select |rrfg | right ADC gain |
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
08 | 1 | 0 0 1 1 | mono mixer gain |right ADC mux sel|left ADC mux sel |
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
09 | 1 | 0 1 0 1 |loopb|left linout|loop|ADCch|TxFch|OffCD|test |loopb|loopb|osamp|
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
0a | 0 | left PCM channel volume |
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
0b | 0 | right PCM channel volume |
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
0c | 0 | left FM channel volume |
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
0d | 0 | right FM channel volume |
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
0e | 0 | left wavetable channel volume |
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
0f | 0 | right wavetable channel volume |
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
10 | 0 | left PCM expansion channel volume |
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
11 | 0 | right PCM expansion channel volume |
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
12 | 0 | left FM expansion channel volume |
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
13 | 0 | right FM expansion channel volume |
----+---+------------+-----+-----+-----+----+-----+-----+-----+-----+-----+-----+-----+
lmute: left mute
mmute: mono mute
dith: dithds
lrfg:
rmute: right mute
rbyps: right bypass
rrfg:
ADCch:
TxFch:
OffCD:
osamp:
And the following diagram is derived from the description in the CVS archive:
MIC L (mouthpiece)
+------+
-->PreAmp>-\
+--^---+ |
| |
r2b4-5 | +--------+
/----*-------------------------------->5 |
| | |
| /----------------------------------->4 |
| | | |
| | /--------------------------------->3 1of5 | +---+
| | | | mux >-->AMP>--> ADC L
| | | /------------------------------->2 | +-^-+
| | | | | | |
Line | | | | +----+ +------+ +---+ /---->1 | r3b3-0
------------*->mute>--> Gain >--> | | | |
L | | | +----+ +------+ | | | *->0 |
| | | | | | +---^----+
Aux2 | | | +----+ +------+ | | | |
----------*--->mute>--> Gain >--> M | | r8b0-2
L | | +----+ +------+ | | |
| | | | \------\
Aux1 | | +----+ +------+ | | |
--------*----->mute>--> Gain >--> I | |
L | +----+ +------+ | | |
| | | |
| +----+ +------+ | | +---+ |
*------->mute>--> Gain >--> X >-->AMP>--*
| +----+ +------+ | | +-^-+ |
| | | | |
| +----+ +------+ | | r2b1-3 |
| /----->mute>--> Gain >--> E | |
| | +----+ +------+ | | |
| | | | |
| | +----+ +------+ | | |
| | /--->mute>--> Gain >--> R | |
| | | +----+ +------+ | | |
| | | | | | r9b8-9
| | | +----+ +------+ | | | |
| | | /->mute>--> Gain >--> | | +---v---+
| | | | +----+ +------+ +---+ /-*->0 |
DAC | | | | | | |
------------*----------------------------------->? | +----+
L | | | | | Mux >-->mute>--> L output
| | | | /->? | +--^-+
| | | | | | | |
| | | /--------->? | r0b0
| | | | | | +-------+
| | | | | |
Mono | | | | | | +-------+
----------* | \---> | +----+
| | | | | | Mix >-->mute>--> Mono output
| | | | *-> | +--^-+
| | | | | +-------+ |
| | | | | r1b0
DAC | | | | | +-------+
------------*-------------------------*--------->1 | +----+
R | | | | | | Mux >-->mute>--> R output
| | | | +----+ +------+ +---+ *->0 | +--^-+
| | | \->mute>--> Gain >--> | | +---^---+ |
| | | +----+ +------+ | | | | r5b0
| | | | | | r6b0
| | | +----+ +------+ | | |
| | \--->mute>--> Gain >--> M | |
| | +----+ +------+ | | |
| | | | |
| | +----+ +------+ | | |
| *----->mute>--> Gain >--> I | |
| | +----+ +------+ | | |
| | | | |
| | +----+ +------+ | | +---+ |
\------->mute>--> Gain >--> X >-->AMP>--*
| +----+ +------+ | | +-^-+ |
/--/ | | | |
Aux1 | +----+ +------+ | | r6b1-3 |
-------*------>mute>--> Gain >--> E | |
R | | +----+ +------+ | | |
| | | | |
Aux2 | | +----+ +------+ | | /------/
---------*---->mute>--> Gain >--> R | |
R | | | +----+ +------+ | | |
| | | | | | +--------+
Line | | | +----+ +------+ | | | *->0 |
-----------*-->mute>--> Gain >--> | | | |
R | | | | +----+ +------+ +---+ \---->1 |
| | | | | |
| | | \-------------------------------->2 | +---+
| | | | Mux >-->AMP>--> ADC R
| | \---------------------------------->3 | +-^-+
| | | | |
| \------------------------------------>4 | r7b3-0
| | |
\-----*-------------------------------->5 |
| +---^----+
r6b4-5 | |
| | r8b3-5
+--v---+ |
-->PreAmp>-/
+------+
MIC R (electret mic)

92
Documentation/sound/oss/btaudio Normal file
View file

@ -0,0 +1,92 @@
Intro
=====
people start bugging me about this with questions, looks like I
should write up some documentation for this beast. That way I
don't have to answer that much mails I hope. Yes, I'm lazy...
You might have noticed that the bt878 grabber cards have actually
_two_ PCI functions:
$ lspci
[ ... ]
00:0a.0 Multimedia video controller: Brooktree Corporation Bt878 (rev 02)
00:0a.1 Multimedia controller: Brooktree Corporation Bt878 (rev 02)
[ ... ]
The first does video, it is backward compatible to the bt848. The second
does audio. btaudio is a driver for the second function. It's a sound
driver which can be used for recording sound (and _only_ recording, no
playback). As most TV cards come with a short cable which can be plugged
into your sound card's line-in you probably don't need this driver if all
you want to do is just watching TV...
Driver Status
=============
Still somewhat experimental. The driver should work stable, i.e. it
should'nt crash your box. It might not work as expected, have bugs,
not being fully OSS API compilant, ...
Latest versions are available from http://bytesex.org/bttv/, the
driver is in the bttv tarball. Kernel patches might be available too,
have a look at http://bytesex.org/bttv/listing.html.
The chip knows two different modes. btaudio registers two dsp
devices, one for each mode. They can not be used at the same time.
Digital audio mode
==================
The chip gives you 16 bit stereo sound. The sample rate depends on
the external source which feeds the bt878 with digital sound via I2S
interface. There is a insmod option (rate) to tell the driver which
sample rate the hardware uses (32000 is the default).
One possible source for digital sound is the msp34xx audio processor
chip which provides digital sound via I2S with 32 kHz sample rate. My
Hauppauge board works this way.
The Osprey-200 reportly gives you digital sound with 44100 Hz sample
rate. It is also possible that you get no sound at all.
analog mode (A/D)
=================
You can tell the driver to use this mode with the insmod option "analog=1".
The chip has three analog inputs. Consequently you'll get a mixer device
to control these.
The analog mode supports mono only. Both 8 + 16 bit. Both are _signed_
int, which is uncommon for the 8 bit case. Sample rate range is 119 kHz
to 448 kHz. Yes, the number of digits is correct. The driver supports
downsampling by powers of two, so you can ask for more usual sample rates
like 44 kHz too.
With my Hauppauge I get noisy sound on the second input (mapped to line2
by the mixer device). Others get a useable signal on line1.
some examples
=============
* read audio data from btaudio (dsp2), send to es1730 (dsp,dsp1):
$ sox -w -r 32000 -t ossdsp /dev/dsp2 -t ossdsp /dev/dsp
* read audio data from btaudio, send to esound daemon (which might be
running on another host):
$ sox -c 2 -w -r 32000 -t ossdsp /dev/dsp2 -t sw - | esdcat -r 32000
$ sox -c 1 -w -r 32000 -t ossdsp /dev/dsp2 -t sw - | esdcat -m -r 32000
Have fun,
Gerd
--
Gerd Knorr <kraxel@bytesex.org>

185
Documentation/sound/oss/mwave Normal file
View file

@ -0,0 +1,185 @@
How to try to survive an IBM Mwave under Linux SB drivers
+ IBM have now released documentation of sorts and Torsten is busy
trying to make the Mwave work. This is not however a trivial task.
----------------------------------------------------------------------------
OK, first thing - the IRQ problem IS a problem, whether the test is bypassed or
not. It is NOT a Linux problem, but an MWAVE problem that is fixed with the
latest MWAVE patches. So, in other words, don't bypass the test for MWAVES!
I have Windows 95 on /dev/hda1, swap on /dev/hda2, and Red Hat 5 on /dev/hda3.
The steps, then:
Boot to Linux.
Mount Windows 95 file system (assume mount point = /dos95).
mkdir /dos95/linux
mkdir /dos95/linux/boot
mkdir /dos95/linux/boot/parms
Copy the kernel, any initrd image, and loadlin to /dos95/linux/boot/.
Reboot to Windows 95.
Edit C:/msdos.sys and add or change the following:
Logo=0
BootGUI=0
Note that msdos.sys is a text file but it needs to be made 'unhidden',
readable and writable before it can be edited. This can be done with
DOS' "attrib" command.
Edit config.sys to have multiple config menus. I have one for windows 95 and
five for Linux, like this:
------------
[menu]
menuitem=W95, Windows 95
menuitem=LINTP, Linux - ThinkPad
menuitem=LINTP3, Linux - ThinkPad Console
menuitem=LINDOC, Linux - Docked
menuitem=LINDOC3, Linux - Docked Console
menuitem=LIN1, Linux - Single User Mode
REM menudefault=W95,10
[W95]
[LINTP]
[LINDOC]
[LINTP3]
[LINDOC3]
[LIN1]
[COMMON]
FILES=30
REM Please read README.TXT in C:\MWW subdirectory before changing the DOS= statement.
DOS=HIGH,UMB
DEVICE=C:\MWW\MANAGER\MWD50430.EXE
SHELL=c:\command.com /e:2048
-------------------
The important things are the SHELL and DEVICE statements.
Then change autoexec.bat. Basically everything in there originally should be
done ONLY when Windows 95 is booted. Then you add new things specifically
for Linux. Mine is as follows
---------------
@ECHO OFF
if "%CONFIG%" == "W95" goto W95
REM
REM Linux stuff
REM
SET MWPATH=C:\MWW\DLL;C:\MWW\MWGAMES;C:\MWW\DSP
SET BLASTER=A220 I5 D1
SET MWROOT=C:\MWW
SET LIBPATH=C:\MWW\DLL
SET PATH=C:\WINDOWS;C:\MWW\DLL;
CALL MWAVE START NOSHOW
c:\linux\boot\loadlin.exe @c:\linux\boot\parms\%CONFIG%.par
:W95
REM
REM Windows 95 stuff
REM
c:\toolkit\guard
SET MSINPUT=C:\MSINPUT
SET MWPATH=C:\MWW\DLL;C:\MWW\MWGAMES;C:\MWW\DSP
REM The following is used by DOS games to recognize Sound Blaster hardware.
REM If hardware settings are changed, please change this line as well.
REM See the Mwave README file for instructions.
SET BLASTER=A220 I5 D1
SET MWROOT=C:\MWW
SET LIBPATH=C:\MWW\DLL
SET PATH=C:\WINDOWS;C:\WINDOWS\COMMAND;E:\ORAWIN95\BIN;f:\msdev\bin;e:\v30\bin.dbg;v:\devt\v30\bin;c:\JavaSDK\Bin;C:\MWW\DLL;
SET INCLUDE=f:\MSDEV\INCLUDE;F:\MSDEV\MFC\INCLUDE
SET LIB=F:\MSDEV\LIB;F:\MSDEV\MFC\LIB
win
------------------------
Now build a file in c:\linux\boot\parms for each Linux config that you have.
For example, my LINDOC3 config is for a docked Thinkpad at runlevel 3 with no
initrd image, and has a parameter file named LINDOC3.PAR in c:\linux\boot\parms:
-----------------------
# LOADLIN @param_file image=other_image root=/dev/other
#
# Linux Console in docking station
#
c:\linux\boot\zImage.krn # First value must be filename of Linux kernel.
root=/dev/hda3 # device which gets mounted as root FS
ro # Other kernel arguments go here.
apm=off
doc=yes
3
-----------------------
The doc=yes parameter is an environment variable used by my init scripts, not
a kernel argument.
However, the apm=off parameter IS a kernel argument! APM, at least in my setup,
causes the kernel to crash when loaded via loadlin (but NOT when loaded via
LILO). The APM stuff COULD be forced out of the kernel via the kernel compile
options. Instead, I got an unofficial patch to the APM drivers that allows them
to be dynamically deactivated via kernel arguments. Whatever you chose to
document, APM, it seems, MUST be off for setups like mine.
Now make sure C:\MWW\MWCONFIG.REF looks like this:
----------------------
[NativeDOS]
Default=SB1.5
SBInputSource=CD
SYNTH=FM
QSound=OFF
Reverb=OFF
Chorus=OFF
ReverbDepth=5
ChorusDepth=5
SBInputVolume=5
SBMainVolume=10
SBWaveVolume=10
SBSynthVolume=10
WaveTableVolume=10
AudioPowerDriver=ON
[FastCFG]
Show=No
HideOption=Off
-----------------------------
OR the Default= line COULD be
Default=SBPRO
Reboot to Windows 95 and choose Linux. When booted, use sndconfig to configure
the sound modules and voilà - ThinkPad sound with Linux.
Now the gotchas - you can either have CD sound OR Mixers but not both. That's a
problem with the SB1.5 (CD sound) or SBPRO (Mixers) settings. No one knows why
this is!
For some reason MPEG3 files, when played through mpg123, sound like they
are playing at 1/8th speed - not very useful! If you have ANY insight
on why this second thing might be happening, I would be grateful.
===========================================================
_/ _/_/_/_/
_/_/ _/_/ _/
_/ _/_/ _/_/_/_/ Martin John Bartlett
_/ _/ _/ _/ (martin@nitram.demon.co.uk)
_/ _/_/_/_/
_/
_/ _/
_/_/
===========================================================

51
Documentation/sound/oss/oss-parameters.txt Normal file
View file

@ -0,0 +1,51 @@
OSS Kernel Parameters
~~~~~~~~~~~~~~~~~~~~~
See Documentation/kernel-parameters.txt for general information on
specifying module parameters.
This document may not be entirely up to date and comprehensive. The command
"modinfo -p ${modulename}" shows a current list of all parameters of a loadable
module. Loadable modules, after being loaded into the running kernel, also
reveal their parameters in /sys/module/${modulename}/parameters/. Some of these
parameters may be changed at runtime by the command
"echo -n ${value} > /sys/module/${modulename}/parameters/${parm}".
ad1848= [HW,OSS]
Format: <io>,<irq>,<dma>,<dma2>,<type>
aedsp16= [HW,OSS] Audio Excel DSP 16
Format: <io>,<irq>,<dma>,<mss_io>,<mpu_io>,<mpu_irq>
See also header of sound/oss/aedsp16.c.
dmasound= [HW,OSS] Sound subsystem buffers
mpu401= [HW,OSS]
Format: <io>,<irq>
opl3= [HW,OSS]
Format: <io>
pas2= [HW,OSS] Format:
<io>,<irq>,<dma>,<dma16>,<sb_io>,<sb_irq>,<sb_dma>,<sb_dma16>
pss= [HW,OSS] Personal Sound System (ECHO ESC614)
Format:
<io>,<mss_io>,<mss_irq>,<mss_dma>,<mpu_io>,<mpu_irq>
sscape= [HW,OSS]
Format: <io>,<irq>,<dma>,<mpu_io>,<mpu_irq>
trix= [HW,OSS] MediaTrix AudioTrix Pro
Format:
<io>,<irq>,<dma>,<dma2>,<sb_io>,<sb_irq>,<sb_dma>,<mpu_io>,<mpu_irq>
uart401= [HW,OSS]
Format: <io>,<irq>
uart6850= [HW,OSS]
Format: <io>,<irq>
waveartist= [HW,OSS]
Format: <io>,<irq>,<dma>,<dma2>

30
Documentation/sound/oss/ultrasound Normal file
View file

@ -0,0 +1,30 @@
modprobe sound
insmod ad1848
insmod gus io=* irq=* dma=* ...
This loads the driver for the Gravis Ultrasound family of sound cards.
The gus module takes the following arguments
io I/O address of the Ultrasound card (eg. io=0x220)
irq IRQ of the Sound Blaster card
dma DMA channel for the Sound Blaster
dma16 2nd DMA channel, only needed for full duplex operation
type 1 for PnP card
gus16 1 for using 16 bit sampling daughter board
no_wave_dma Set to disable DMA usage for wavetable (see note)
db16 ???
no_wave_dma option
This option defaults to a value of 0, which allows the Ultrasound wavetable
DSP to use DMA for playback and downloading samples. This is the same
as the old behaviour. If set to 1, no DMA is needed for downloading samples,
and allows owners of a GUS MAX to make use of simultaneous digital audio
(/dev/dsp), MIDI, and wavetable playback.
If you have problems in recording with GUS MAX, you could try to use
just one 8 bit DMA channel. Recording will not work with one DMA
channel if it's a 16 bit one.

293
Documentation/sound/oss/vwsnd Normal file
View file

@ -0,0 +1,293 @@
vwsnd - Sound driver for the Silicon Graphics 320 and 540 Visual
Workstations' onboard audio.
Copyright 1999 Silicon Graphics, Inc. All rights reserved.
At the time of this writing, March 1999, there are two models of
Visual Workstation, the 320 and the 540. This document only describes
those models. Future Visual Workstation models may have different
sound capabilities, and this driver will probably not work on those
boxes.
The Visual Workstation has an Analog Devices AD1843 "SoundComm" audio
codec chip. The AD1843 is accessed through the Cobalt I/O ASIC, also
known as Lithium. This driver programs both chips.
==============================================================================
QUICK CONFIGURATION
# insmod soundcore
# insmod vwsnd
==============================================================================
I/O CONNECTIONS
On the Visual Workstation, only three of the AD1843 inputs are hooked
up. The analog line in jacks are connected to the AD1843's AUX1
input. The CD audio lines are connected to the AD1843's AUX2 input.
The microphone jack is connected to the AD1843's MIC input. The mic
jack is mono, but the signal is delivered to both the left and right
MIC inputs. You can record in stereo from the mic input, but you will
get the same signal on both channels (within the limits of A/D
accuracy). Full scale on the Line input is +/- 2.0 V. Full scale on
the MIC input is 20 dB less, or +/- 0.2 V.
The AD1843's LOUT1 outputs are connected to the Line Out jacks. The
AD1843's HPOUT outputs are connected to the speaker/headphone jack.
LOUT2 is not connected. Line out's maximum level is +/- 2.0 V peak to
peak. The speaker/headphone out's maximum is +/- 4.0 V peak to peak.
The AD1843's PCM input channel and one of its output channels (DAC1)
are connected to Lithium. The other output channel (DAC2) is not
connected.
==============================================================================
CAPABILITIES
The AD1843 has PCM input and output (Pulse Code Modulation, also known
as wavetable). PCM input and output can be mono or stereo in any of
four formats. The formats are 16 bit signed and 8 bit unsigned,
u-Law, and A-Law format. Any sample rate from 4 KHz to 49 KHz is
available, in 1 Hz increments.
The AD1843 includes an analog mixer that can mix all three input
signals (line, mic and CD) into the analog outputs. The mixer has a
separate gain control and mute switch for each input.
There are two outputs, line out and speaker/headphone out. They
always produce the same signal, and the speaker always has 3 dB more
gain than the line out. The speaker/headphone output can be muted,
but this driver does not export that function.
The hardware can sync audio to the video clock, but this driver does
not have a way to specify syncing to video.
==============================================================================
PROGRAMMING
This section explains the API supported by the driver. Also see the
Open Sound Programming Guide at http://www.opensound.com/pguide/ .
This section assumes familiarity with that document.
The driver has two interfaces, an I/O interface and a mixer interface.
There is no MIDI or sequencer capability.
==============================================================================
PROGRAMMING PCM I/O
The I/O interface is usually accessed as /dev/audio or /dev/dsp.
Using the standard Open Sound System (OSS) ioctl calls, the sample
rate, number of channels, and sample format may be set within the
limitations described above. The driver supports triggering. It also
supports getting the input and output pointers with one-sample
accuracy.
The SNDCTL_DSP_GETCAP ioctl returns these capabilities.
DSP_CAP_DUPLEX - driver supports full duplex.
DSP_CAP_TRIGGER - driver supports triggering.
DSP_CAP_REALTIME - values returned by SNDCTL_DSP_GETIPTR
and SNDCTL_DSP_GETOPTR are accurate to a few samples.
Memory mapping (mmap) is not implemented.
The driver permits subdivided fragment sizes from 64 to 4096 bytes.
The number of fragments can be anything from 3 fragments to however
many fragments fit into 124 kilobytes. It is up to the user to
determine how few/small fragments can be used without introducing
glitches with a given workload. Linux is not realtime, so we can't
promise anything. (sigh...)
When this driver is switched into or out of mu-Law or A-Law mode on
output, it may produce an audible click. This is unavoidable. To
prevent clicking, use signed 16-bit mode instead, and convert from
mu-Law or A-Law format in software.
==============================================================================
PROGRAMMING THE MIXER INTERFACE
The mixer interface is usually accessed as /dev/mixer. It is accessed
through ioctls. The mixer allows the application to control gain or
mute several audio signal paths, and also allows selection of the
recording source.
Each of the constants described here can be read using the
MIXER_READ(SOUND_MIXER_xxx) ioctl. Those that are not read-only can
also be written using the MIXER_WRITE(SOUND_MIXER_xxx) ioctl. In most
cases, <sys/soundcard.h> defines constants SOUND_MIXER_READ_xxx and
SOUND_MIXER_WRITE_xxx which work just as well.
SOUND_MIXER_CAPS Read-only
This is a mask of optional driver capabilities that are implemented.
This driver's only capability is SOUND_CAP_EXCL_INPUT, which means
that only one recording source can be active at a time.
SOUND_MIXER_DEVMASK Read-only
This is a mask of the sound channels. This driver's channels are PCM,
LINE, MIC, CD, and RECLEV.
SOUND_MIXER_STEREODEVS Read-only
This is a mask of which sound channels are capable of stereo. All
channels are capable of stereo. (But see caveat on MIC input in I/O
CONNECTIONS section above).
SOUND_MIXER_OUTMASK Read-only
This is a mask of channels that route inputs through to outputs.
Those are LINE, MIC, and CD.
SOUND_MIXER_RECMASK Read-only
This is a mask of channels that can be recording sources. Those are
PCM, LINE, MIC, CD.
SOUND_MIXER_PCM Default: 0x5757 (0 dB)
This is the gain control for PCM output. The left and right channel
gain are controlled independently. This gain control has 64 levels,
which range from -82.5 dB to +12.0 dB in 1.5 dB steps. Those 64
levels are mapped onto 100 levels at the ioctl, see below.
SOUND_MIXER_LINE Default: 0x4a4a (0 dB)
This is the gain control for mixing the Line In source into the
outputs. The left and right channel gain are controlled
independently. This gain control has 32 levels, which range from
-34.5 dB to +12.0 dB in 1.5 dB steps. Those 32 levels are mapped onto
100 levels at the ioctl, see below.
SOUND_MIXER_MIC Default: 0x4a4a (0 dB)
This is the gain control for mixing the MIC source into the outputs.
The left and right channel gain are controlled independently. This
gain control has 32 levels, which range from -34.5 dB to +12.0 dB in
1.5 dB steps. Those 32 levels are mapped onto 100 levels at the
ioctl, see below.
SOUND_MIXER_CD Default: 0x4a4a (0 dB)
This is the gain control for mixing the CD audio source into the
outputs. The left and right channel gain are controlled
independently. This gain control has 32 levels, which range from
-34.5 dB to +12.0 dB in 1.5 dB steps. Those 32 levels are mapped onto
100 levels at the ioctl, see below.
SOUND_MIXER_RECLEV Default: 0 (0 dB)
This is the gain control for PCM input (RECording LEVel). The left
and right channel gain are controlled independently. This gain
control has 16 levels, which range from 0 dB to +22.5 dB in 1.5 dB
steps. Those 16 levels are mapped onto 100 levels at the ioctl, see
below.
SOUND_MIXER_RECSRC Default: SOUND_MASK_LINE
This is a mask of currently selected PCM input sources (RECording
SouRCes). Because the AD1843 can only have a single recording source
at a time, only one bit at a time can be set in this mask. The
allowable values are SOUND_MASK_PCM, SOUND_MASK_LINE, SOUND_MASK_MIC,
or SOUND_MASK_CD. Selecting SOUND_MASK_PCM sets up internal
resampling which is useful for loopback testing and for hardware
sample rate conversion. But software sample rate conversion is
probably faster, so I don't know how useful that is.
SOUND_MIXER_OUTSRC DEFAULT: SOUND_MASK_LINE|SOUND_MASK_MIC|SOUND_MASK_CD
This is a mask of sources that are currently passed through to the
outputs. Those sources whose bits are not set are muted.
==============================================================================
GAIN CONTROL
There are five gain controls listed above. Each has 16, 32, or 64
steps. Each control has 1.5 dB of gain per step. Each control is
stereo.
The OSS defines the argument to a channel gain ioctl as having two
components, left and right, each of which ranges from 0 to 100. The
two components are packed into the same word, with the left side gain
in the least significant byte, and the right side gain in the second
least significant byte. In C, we would say this.
#include <assert.h>
...
assert(leftgain >= 0 && leftgain <= 100);
assert(rightgain >= 0 && rightgain <= 100);
arg = leftgain | rightgain << 8;
So each OSS gain control has 101 steps. But the hardware has 16, 32,
or 64 steps. The hardware steps are spread across the 101 OSS steps
nearly evenly. The conversion formulas are like this, given N equals
16, 32, or 64.
int round = N/2 - 1;
OSS_gain_steps = (hw_gain_steps * 100 + round) / (N - 1);
hw_gain_steps = (OSS_gain_steps * (N - 1) + round) / 100;
Here is a snippet of C code that will return the left and right gain
of any channel in dB. Pass it one of the predefined gain_desc_t
structures to access any of the five channels' gains.
typedef struct gain_desc {
float min_gain;
float gain_step;
int nbits;
int chan;
} gain_desc_t;
const gain_desc_t gain_pcm = { -82.5, 1.5, 6, SOUND_MIXER_PCM };
const gain_desc_t gain_line = { -34.5, 1.5, 5, SOUND_MIXER_LINE };
const gain_desc_t gain_mic = { -34.5, 1.5, 5, SOUND_MIXER_MIC };
const gain_desc_t gain_cd = { -34.5, 1.5, 5, SOUND_MIXER_CD };
const gain_desc_t gain_reclev = { 0.0, 1.5, 4, SOUND_MIXER_RECLEV };
int get_gain_dB(int fd, const gain_desc_t *gp,
float *left, float *right)
{
int word;
int lg, rg;
int mask = (1 << gp->nbits) - 1;
if (ioctl(fd, MIXER_READ(gp->chan), &word) != 0)
return -1; /* fail */
lg = word & 0xFF;
rg = word >> 8 & 0xFF;
lg = (lg * mask + mask / 2) / 100;
rg = (rg * mask + mask / 2) / 100;
*left = gp->min_gain + gp->gain_step * lg;
*right = gp->min_gain + gp->gain_step * rg;
return 0;
}
And here is the corresponding routine to set a channel's gain in dB.
int set_gain_dB(int fd, const gain_desc_t *gp, float left, float right)
{
float max_gain =
gp->min_gain + (1 << gp->nbits) * gp->gain_step;
float round = gp->gain_step / 2;
int mask = (1 << gp->nbits) - 1;
int word;
int lg, rg;
if (left < gp->min_gain || right < gp->min_gain)
return EINVAL;
lg = (left - gp->min_gain + round) / gp->gain_step;
rg = (right - gp->min_gain + round) / gp->gain_step;
if (lg >= (1 << gp->nbits) || rg >= (1 << gp->nbits))
return EINVAL;
lg = (100 * lg + mask / 2) / mask;
rg = (100 * rg + mask / 2) / mask;
word = lg | rg << 8;
return ioctl(fd, MIXER_WRITE(gp->chan), &word);
}