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Merge branch 'for-next' into for-linus 

Pull 4.15 updates to take over the previous urgent fixes.

Signed-off-by: Takashi Iwai <tiwai@suse.de>
hifive-unleashed-5.1
Takashi Iwai 2017-11-13 15:43:04 +01:00
parent 75ee94b20b 9718a29d40
commit c429bda21f
171 changed files with 452 additions and 41482 deletions
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2
Documentation/sound/hd-audio/models.rst
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@ -82,6 +82,8 @@ tpt460
Lenovo Thinkpad T460/560 setup
dual-codecs
Lenovo laptops with dual codecs
alc700-ref
Intel reference board with ALC700 codec
ALC66x/67x/892
==============

66
Documentation/sound/oss/ALS
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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
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@ -1,101 +0,0 @@
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
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@ -1,152 +0,0 @@
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
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@ -1,34 +0,0 @@
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
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@ -1,55 +0,0 @@
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

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Documentation/sound/oss/Introduction
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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)

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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

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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;
}

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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.

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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/

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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 attempt 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

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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
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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
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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
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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+
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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.

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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
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(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)

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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 compliant, ...
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
View File

@ -1,185 +0,0 @@
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
View File

@ -1,51 +0,0 @@
OSS Kernel Parameters
~~~~~~~~~~~~~~~~~~~~~
See Documentation/admin-guide/kernel-parameters.rst 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
View File

@ -1,30 +0,0 @@
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.

11
MAINTAINERS
View File

@ -527,11 +527,6 @@ W: http://ez.analog.com/community/linux-device-drivers
S: Supported
F: drivers/input/misc/adxl34x.c
AEDSP16 DRIVER
M: Riccardo Facchetti <fizban@tin.it>
S: Maintained
F: sound/oss/aedsp16.c
AF9013 MEDIA DRIVER
M: Antti Palosaari <crope@iki.fi>
L: linux-media@vger.kernel.org
@ -9199,12 +9194,6 @@ F: include/linux/dt-bindings/mux/
F: include/linux/mux/
F: drivers/mux/
MULTISOUND SOUND DRIVER
M: Andrew Veliath <andrewtv@usa.net>
S: Maintained
F: Documentation/sound/oss/MultiSound
F: sound/oss/msnd*
MULTITECH MULTIPORT CARD (ISICOM)
S: Orphan
F: drivers/tty/isicom.c

30
drivers/usb/core/urb.c
View File

@ -187,6 +187,31 @@ EXPORT_SYMBOL_GPL(usb_unanchor_urb);
/*-------------------------------------------------------------------*/
static const int pipetypes[4] = {
PIPE_CONTROL, PIPE_ISOCHRONOUS, PIPE_BULK, PIPE_INTERRUPT
};
/**
* usb_urb_ep_type_check - sanity check of endpoint in the given urb
* @urb: urb to be checked
*
* This performs a light-weight sanity check for the endpoint in the
* given urb. It returns 0 if the urb contains a valid endpoint, otherwise
* a negative error code.
*/
int usb_urb_ep_type_check(const struct urb *urb)
{
const struct usb_host_endpoint *ep;
ep = usb_pipe_endpoint(urb->dev, urb->pipe);
if (!ep)
return -EINVAL;
if (usb_pipetype(urb->pipe) != pipetypes[usb_endpoint_type(&ep->desc)])
return -EINVAL;
return 0;
}
EXPORT_SYMBOL_GPL(usb_urb_ep_type_check);
/**
* usb_submit_urb - issue an asynchronous transfer request for an endpoint
* @urb: pointer to the urb describing the request
@ -326,9 +351,6 @@ EXPORT_SYMBOL_GPL(usb_unanchor_urb);
*/
int usb_submit_urb(struct urb *urb, gfp_t mem_flags)
{
static int pipetypes[4] = {
PIPE_CONTROL, PIPE_ISOCHRONOUS, PIPE_BULK, PIPE_INTERRUPT
};
int xfertype, max;
struct usb_device *dev;
struct usb_host_endpoint *ep;
@ -444,7 +466,7 @@ int usb_submit_urb(struct urb *urb, gfp_t mem_flags)
*/
/* Check that the pipe's type matches the endpoint's type */
if (usb_pipetype(urb->pipe) != pipetypes[xfertype])
if (usb_urb_ep_type_check(urb))
dev_WARN(&dev->dev, "BOGUS urb xfer, pipe %x != type %x\n",
usb_pipetype(urb->pipe), pipetypes[xfertype]);

2
include/linux/usb.h
View File

@ -1728,6 +1728,8 @@ static inline int usb_urb_dir_out(struct urb *urb)
return (urb->transfer_flags & URB_DIR_MASK) == URB_DIR_OUT;
}
int usb_urb_ep_type_check(const struct urb *urb);
void *usb_alloc_coherent(struct usb_device *dev, size_t size,
gfp_t mem_flags, dma_addr_t *dma);
void usb_free_coherent(struct usb_device *dev, size_t size,

2
include/sound/core.h
View File

@ -133,6 +133,7 @@ struct snd_card {
struct device card_dev; /* cardX object for sysfs */
const struct attribute_group *dev_groups[4]; /* assigned sysfs attr */
bool registered; /* card_dev is registered? */
wait_queue_head_t remove_sleep;
#ifdef CONFIG_PM
unsigned int power_state; /* power state */
@ -240,6 +241,7 @@ int snd_card_new(struct device *parent, int idx, const char *xid,
struct snd_card **card_ret);
int snd_card_disconnect(struct snd_card *card);
void snd_card_disconnect_sync(struct snd_card *card);
int snd_card_free(struct snd_card *card);
int snd_card_free_when_closed(struct snd_card *card);
void snd_card_set_id(struct snd_card *card, const char *id);

3
include/sound/hdaudio.h
View File

@ -111,8 +111,7 @@ void snd_hdac_device_unregister(struct hdac_device *codec);
int snd_hdac_device_set_chip_name(struct hdac_device *codec, const char *name);
int snd_hdac_codec_modalias(struct hdac_device *hdac, char *buf, size_t size);
int snd_hdac_refresh_widgets(struct hdac_device *codec);
int snd_hdac_refresh_widget_sysfs(struct hdac_device *codec);
int snd_hdac_refresh_widgets(struct hdac_device *codec, bool sysfs);
unsigned int snd_hdac_make_cmd(struct hdac_device *codec, hda_nid_t nid,
unsigned int verb, unsigned int parm);

1
include/sound/snd_wavefront.h
View File

@ -28,6 +28,7 @@ struct _snd_wavefront_midi {
struct snd_rawmidi_substream *substream_output[2];
struct snd_rawmidi_substream *substream_input[2];
struct timer_list timer;
snd_wavefront_card_t *timer_card;
spinlock_t open;
spinlock_t virtual; /* protects isvirtual */
};

33
sound/Kconfig
View File

@ -3,25 +3,7 @@ menuconfig SOUND
depends on HAS_IOMEM
help
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
<file:Documentation/sound/oss/Introduction>. The file
<file:Documentation/sound/oss/README.OSS> contains some slightly
outdated but still useful information as well. Newer sound
driver documentation is found in <file:Documentation/sound/alsa/*>.
If you have a PnP sound card and you want to configure it at boot
time using the ISA PnP tools (read
<http://www.roestock.demon.co.uk/isapnptools/>), then you need to
compile the sound card support as a module and load that module
after the PnP configuration is finished. To do this, choose M here
and read <file:Documentation/sound/oss/README.modules>; the module
will be called soundcore.
than an occasional beep, say Y.
if SOUND
@ -114,19 +96,6 @@ source "sound/synth/Kconfig"
endif # SND
menuconfig SOUND_PRIME
tristate "Open Sound System (DEPRECATED)"
select SOUND_OSS_CORE
depends on BROKEN
help
Say 'Y' or 'M' to enable Open Sound System drivers.
if SOUND_PRIME
source "sound/oss/Kconfig"
endif # SOUND_PRIME
endif # !UML
endif # SOUND

3
sound/Makefile
View File

@ -2,8 +2,7 @@
#
obj-$(CONFIG_SOUND) += soundcore.o
obj-$(CONFIG_SOUND_PRIME) += oss/
obj-$(CONFIG_DMASOUND) += oss/
obj-$(CONFIG_DMASOUND) += oss/dmasound/
obj-$(CONFIG_SND) += core/ i2c/ drivers/ isa/ pci/ ppc/ arm/ sh/ synth/ usb/ \
firewire/ sparc/ spi/ parisc/ pcmcia/ mips/ soc/ atmel/ hda/ x86/
obj-$(CONFIG_SND_AOA) += aoa/

2
sound/core/hrtimer.c
View File

@ -127,7 +127,7 @@ static int snd_hrtimer_stop(struct snd_timer *t)
return 0;
}
static struct snd_timer_hardware hrtimer_hw = {
static const struct snd_timer_hardware hrtimer_hw __initconst = {
.flags = SNDRV_TIMER_HW_AUTO | SNDRV_TIMER_HW_TASKLET,
.open = snd_hrtimer_open,
.close = snd_hrtimer_close,

2
sound/core/hwdep.c
View File

@ -228,6 +228,8 @@ static int snd_hwdep_dsp_load(struct snd_hwdep *hw,
memset(&info, 0, sizeof(info));
if (copy_from_user(&info, _info, sizeof(info)))
return -EFAULT;
if (info.index >= 32)
return -EINVAL;
/* check whether the dsp was already loaded */
if (hw->dsp_loaded & (1 << info.index))
return -EBUSY;

32
sound/core/init.c
View File

@ -255,6 +255,7 @@ int snd_card_new(struct device *parent, int idx, const char *xid,
#ifdef CONFIG_PM
init_waitqueue_head(&card->power_sleep);
#endif
init_waitqueue_head(&card->remove_sleep);
device_initialize(&card->card_dev);
card->card_dev.parent = parent;
@ -452,6 +453,35 @@ int snd_card_disconnect(struct snd_card *card)
}
EXPORT_SYMBOL(snd_card_disconnect);
/**
* snd_card_disconnect_sync - disconnect card and wait until files get closed
* @card: card object to disconnect
*
* This calls snd_card_disconnect() for disconnecting all belonging components
* and waits until all pending files get closed.
* It assures that all accesses from user-space finished so that the driver
* can release its resources gracefully.
*/
void snd_card_disconnect_sync(struct snd_card *card)
{
int err;
err = snd_card_disconnect(card);
if (err < 0) {
dev_err(card->dev,
"snd_card_disconnect error (%d), skipping sync\n",
err);
return;
}
spin_lock_irq(&card->files_lock);
wait_event_lock_irq(card->remove_sleep,
list_empty(&card->files_list),
card->files_lock);
spin_unlock_irq(&card->files_lock);
}
EXPORT_SYMBOL_GPL(snd_card_disconnect_sync);
static int snd_card_do_free(struct snd_card *card)
{
#if IS_ENABLED(CONFIG_SND_MIXER_OSS)
@ -957,6 +987,8 @@ int snd_card_file_remove(struct snd_card *card, struct file *file)
break;
}
}
if (list_empty(&card->files_list))
wake_up_all(&card->remove_sleep);
spin_unlock(&card->files_lock);
if (!found) {
dev_err(card->dev, "card file remove problem (%p)\n", file);

2
sound/core/jack.c
View File

@ -310,7 +310,7 @@ EXPORT_SYMBOL(snd_jack_set_parent);
* @type: Jack report type for this key
* @keytype: Input layer key type to be reported
*
* Map a SND_JACK_BTN_ button type to an input layer key, allowing
* Map a SND_JACK_BTN_* button type to an input layer key, allowing
* reporting of keys on accessories via the jack abstraction. If no
* mapping is provided but keys are enabled in the jack type then
* BTN_n numeric buttons will be reported.

20
sound/core/pcm.c
View File

@ -775,6 +775,9 @@ static int _snd_pcm_new(struct snd_card *card, const char *id, int device,
.dev_register = snd_pcm_dev_register,
.dev_disconnect = snd_pcm_dev_disconnect,
};
static struct snd_device_ops internal_ops = {
.dev_free = snd_pcm_dev_free,
};
if (snd_BUG_ON(!card))
return -ENXIO;
@ -801,7 +804,8 @@ static int _snd_pcm_new(struct snd_card *card, const char *id, int device,
if (err < 0)
goto free_pcm;
err = snd_device_new(card, SNDRV_DEV_PCM, pcm, &ops);
err = snd_device_new(card, SNDRV_DEV_PCM, pcm,
internal ? &internal_ops : &ops);
if (err < 0)
goto free_pcm;
@ -1099,8 +1103,6 @@ static int snd_pcm_dev_register(struct snd_device *device)
if (snd_BUG_ON(!device || !device->device_data))
return -ENXIO;
pcm = device->device_data;
if (pcm->internal)
return 0;
mutex_lock(&register_mutex);
err = snd_pcm_add(pcm);
@ -1152,6 +1154,10 @@ static int snd_pcm_dev_disconnect(struct snd_device *device)
for (substream = pcm->streams[cidx].substream; substream; substream = substream->next) {
snd_pcm_stream_lock_irq(substream);
if (substream->runtime) {
if (snd_pcm_running(substream))
snd_pcm_stop(substream,
SNDRV_PCM_STATE_DISCONNECTED);
/* to be sure, set the state unconditionally */
substream->runtime->status->state = SNDRV_PCM_STATE_DISCONNECTED;
wake_up(&substream->runtime->sleep);
wake_up(&substream->runtime->tsleep);
@ -1159,12 +1165,10 @@ static int snd_pcm_dev_disconnect(struct snd_device *device)
snd_pcm_stream_unlock_irq(substream);
}
}
if (!pcm->internal) {
pcm_call_notify(pcm, n_disconnect);
}
pcm_call_notify(pcm, n_disconnect);
for (cidx = 0; cidx < 2; cidx++) {
if (!pcm->internal)
snd_unregister_device(&pcm->streams[cidx].dev);
snd_unregister_device(&pcm->streams[cidx].dev);
free_chmap(&pcm->streams[cidx]);
}
mutex_unlock(&pcm->open_mutex);

2
sound/core/pcm_native.c
View File

@ -195,7 +195,6 @@ EXPORT_SYMBOL_GPL(snd_pcm_stream_unlock_irqrestore);
int snd_pcm_info(struct snd_pcm_substream *substream, struct snd_pcm_info *info)
{
struct snd_pcm_runtime *runtime;
struct snd_pcm *pcm = substream->pcm;
struct snd_pcm_str *pstr = substream->pstr;
@ -211,7 +210,6 @@ int snd_pcm_info(struct snd_pcm_substream *substream, struct snd_pcm_info *info)
info->subdevices_count = pstr->substream_count;
info->subdevices_avail = pstr->substream_count - pstr->substream_opened;
strlcpy(info->subname, substream->name, sizeof(info->subname));
runtime = substream->runtime;
return 0;
}

4
sound/core/seq/seq_clientmgr.c
View File

@ -802,6 +802,10 @@ static int snd_seq_deliver_event(struct snd_seq_client *client, struct snd_seq_e
return -EMLINK;
}
if (snd_seq_ev_is_variable(event) &&
snd_BUG_ON(atomic && (event->data.ext.len & SNDRV_SEQ_EXT_USRPTR)))
return -EINVAL;
if (event->queue == SNDRV_SEQ_ADDRESS_SUBSCRIBERS ||
event->dest.client == SNDRV_SEQ_ADDRESS_SUBSCRIBERS)
result = deliver_to_subscribers(client, event, atomic, hop);

11
sound/core/timer.c
View File

@ -1069,15 +1069,17 @@ EXPORT_SYMBOL(snd_timer_global_register);
struct snd_timer_system_private {
struct timer_list tlist;
struct snd_timer *snd_timer;
unsigned long last_expires;
unsigned long last_jiffies;
unsigned long correction;
};
static void snd_timer_s_function(unsigned long data)
static void snd_timer_s_function(struct timer_list *t)
{
struct snd_timer *timer = (struct snd_timer *)data;
struct snd_timer_system_private *priv = timer->private_data;
struct snd_timer_system_private *priv = from_timer(priv, t,
tlist);
struct snd_timer *timer = priv->snd_timer;
unsigned long jiff = jiffies;
if (time_after(jiff, priv->last_expires))
priv->correction += (long)jiff - (long)priv->last_expires;
@ -1159,7 +1161,8 @@ static int snd_timer_register_system(void)
snd_timer_free(timer);
return -ENOMEM;
}
setup_timer(&priv->tlist, snd_timer_s_function, (unsigned long) timer);
priv->snd_timer = timer;
timer_setup(&priv->tlist, snd_timer_s_function, 0);
timer->private_data = priv;
timer->private_free = snd_timer_free_system;
return snd_timer_global_register(timer);

7
sound/drivers/aloop.c
View File

@ -529,9 +529,9 @@ static unsigned int loopback_pos_update(struct loopback_cable *cable)
return running;
}
static void loopback_timer_function(unsigned long data)
static void loopback_timer_function(struct timer_list *t)
{
struct loopback_pcm *dpcm = (struct loopback_pcm *)data;
struct loopback_pcm *dpcm = from_timer(dpcm, t, timer);
unsigned long flags;
spin_lock_irqsave(&dpcm->cable->lock, flags);
@ -675,8 +675,7 @@ static int loopback_open(struct snd_pcm_substream *substream)
}
dpcm->loopback = loopback;
dpcm->substream = substream;
setup_timer(&dpcm->timer, loopback_timer_function,
(unsigned long)dpcm);
timer_setup(&dpcm->timer, loopback_timer_function, 0);
cable = loopback->cables[substream->number][dev];
if (!cable) {

7
sound/drivers/dummy.c
View File

@ -306,9 +306,9 @@ static int dummy_systimer_prepare(struct snd_pcm_substream *substream)
return 0;
}
static void dummy_systimer_callback(unsigned long data)
static void dummy_systimer_callback(struct timer_list *t)
{
struct dummy_systimer_pcm *dpcm = (struct dummy_systimer_pcm *)data;
struct dummy_systimer_pcm *dpcm = from_timer(dpcm, t, timer);
unsigned long flags;
int elapsed = 0;
@ -343,8 +343,7 @@ static int dummy_systimer_create(struct snd_pcm_substream *substream)
if (!dpcm)
return -ENOMEM;
substream->runtime->private_data = dpcm;
setup_timer(&dpcm->timer, dummy_systimer_callback,
(unsigned long) dpcm);
timer_setup(&dpcm->timer, dummy_systimer_callback, 0);
spin_lock_init(&dpcm->lock);
dpcm->substream = substream;
return 0;

7
sound/drivers/mpu401/mpu401_uart.c
View File

@ -169,9 +169,9 @@ EXPORT_SYMBOL(snd_mpu401_uart_interrupt_tx);
* timer callback
* reprogram the timer and call the interrupt job
*/
static void snd_mpu401_uart_timer(unsigned long data)
static void snd_mpu401_uart_timer(struct timer_list *t)
{
struct snd_mpu401 *mpu = (struct snd_mpu401 *)data;
struct snd_mpu401 *mpu = from_timer(mpu, t, timer);
unsigned long flags;
spin_lock_irqsave(&mpu->timer_lock, flags);
@ -191,8 +191,7 @@ static void snd_mpu401_uart_add_timer (struct snd_mpu401 *mpu, int input)
spin_lock_irqsave (&mpu->timer_lock, flags);
if (mpu->timer_invoked == 0) {
setup_timer(&mpu->timer, snd_mpu401_uart_timer,
(unsigned long)mpu);
timer_setup(&mpu->timer, snd_mpu401_uart_timer, 0);
mod_timer(&mpu->timer, 1 + jiffies);
}
mpu->timer_invoked |= input ? MPU401_MODE_INPUT_TIMER :

7
sound/drivers/mtpav.c
View File

@ -406,10 +406,10 @@ static void snd_mtpav_input_trigger(struct snd_rawmidi_substream *substream, int
* timer interrupt for outputs
*/
static void snd_mtpav_output_timer(unsigned long data)
static void snd_mtpav_output_timer(struct timer_list *t)
{
unsigned long flags;
struct mtpav *chip = (struct mtpav *)data;
struct mtpav *chip = from_timer(chip, t, timer);
int p;
spin_lock_irqsave(&chip->spinlock, flags);
@ -707,8 +707,7 @@ static int snd_mtpav_probe(struct platform_device *dev)
mtp_card->share_irq = 0;
mtp_card->inmidistate = 0;
mtp_card->outmidihwport = 0xffffffff;
setup_timer(&mtp_card->timer, snd_mtpav_output_timer,
(unsigned long) mtp_card);
timer_setup(&mtp_card->timer, snd_mtpav_output_timer, 0);
card->private_free = snd_mtpav_free;

4
sound/drivers/opl3/opl3_midi.c
View File

@ -238,10 +238,10 @@ static int opl3_get_voice(struct snd_opl3 *opl3, int instr_4op,
/*
* System timer interrupt function
*/
void snd_opl3_timer_func(unsigned long data)
void snd_opl3_timer_func(struct timer_list *t)
{
struct snd_opl3 *opl3 = (struct snd_opl3 *)data;
struct snd_opl3 *opl3 = from_timer(opl3, t, tlist);
unsigned long flags;
int again = 0;
int i;

2
sound/drivers/opl3/opl3_seq.c
View File

@ -248,7 +248,7 @@ static int snd_opl3_seq_probe(struct device *_dev)
}
/* setup system timer */
setup_timer(&opl3->tlist, snd_opl3_timer_func, (unsigned long) opl3);
timer_setup(&opl3->tlist, snd_opl3_timer_func, 0);
spin_lock_init(&opl3->sys_timer_lock);
opl3->sys_timer_status = 0;

2
sound/drivers/opl3/opl3_voice.h
View File

@ -37,7 +37,7 @@ void snd_opl3_nrpn(void *p, struct snd_midi_channel *chan, struct snd_midi_chann
void snd_opl3_sysex(void *p, unsigned char *buf, int len, int parsed, struct snd_midi_channel_set *chset);
void snd_opl3_calc_volume(unsigned char *reg, int vel, struct snd_midi_channel *chan);
void snd_opl3_timer_func(unsigned long data);
void snd_opl3_timer_func(struct timer_list *t);
/* Prototypes for opl3_drums.c */
void snd_opl3_load_drums(struct snd_opl3 *opl3);

7
sound/drivers/serial-u16550.c
View File

@ -309,12 +309,12 @@ static irqreturn_t snd_uart16550_interrupt(int irq, void *dev_id)
}
/* When the polling mode, this function calls snd_uart16550_io_loop. */
static void snd_uart16550_buffer_timer(unsigned long data)
static void snd_uart16550_buffer_timer(struct timer_list *t)
{
unsigned long flags;
struct snd_uart16550 *uart;
uart = (struct snd_uart16550 *)data;
uart = from_timer(uart, t, buffer_timer);
spin_lock_irqsave(&uart->open_lock, flags);
snd_uart16550_del_timer(uart);
snd_uart16550_io_loop(uart);
@ -828,8 +828,7 @@ static int snd_uart16550_create(struct snd_card *card,
uart->prev_in = 0;
uart->rstatus = 0;
memset(uart->prev_status, 0x80, sizeof(unsigned char) * SNDRV_SERIAL_MAX_OUTS);
setup_timer(&uart->buffer_timer, snd_uart16550_buffer_timer,
(unsigned long)uart);
timer_setup(&uart->buffer_timer, snd_uart16550_buffer_timer, 0);
uart->timer_running = 0;
/* Register device */

3
sound/hda/hdac_controller.c
View File

@ -319,7 +319,8 @@ int snd_hdac_bus_parse_capabilities(struct hdac_bus *bus)
break;
default:
dev_dbg(bus->dev, "Unknown capability %d\n", cur_cap);
dev_err(bus->dev, "Unknown capability %d\n", cur_cap);
cur_cap = 0;
break;
}

43
sound/hda/hdac_device.c
View File

@ -87,7 +87,7 @@ int snd_hdac_device_init(struct hdac_device *codec, struct hdac_bus *bus,
fg = codec->afg ? codec->afg : codec->mfg;
err = snd_hdac_refresh_widgets(codec);
err = snd_hdac_refresh_widgets(codec, false);
if (err < 0)
goto error;
@ -388,11 +388,12 @@ static void setup_fg_nodes(struct hdac_device *codec)
/**
* snd_hdac_refresh_widgets - Reset the widget start/end nodes
* @codec: the codec object
* @sysfs: re-initialize sysfs tree, too
*/
int snd_hdac_refresh_widgets(struct hdac_device *codec)
int snd_hdac_refresh_widgets(struct hdac_device *codec, bool sysfs)
{
hda_nid_t start_nid;
int nums;
int nums, err;
nums = snd_hdac_get_sub_nodes(codec, codec->afg, &start_nid);
if (!start_nid || nums <= 0 || nums >= 0xff) {
@ -401,6 +402,12 @@ int snd_hdac_refresh_widgets(struct hdac_device *codec)
return -EINVAL;
}
if (sysfs) {
err = hda_widget_sysfs_reinit(codec, start_nid, nums);
if (err < 0)
return err;
}
codec->num_nodes = nums;
codec->start_nid = start_nid;
codec->end_nid = start_nid + nums;
@ -408,36 +415,6 @@ int snd_hdac_refresh_widgets(struct hdac_device *codec)
}
EXPORT_SYMBOL_GPL(snd_hdac_refresh_widgets);
/**
* snd_hdac_refresh_widget_sysfs - Reset the codec widgets and reinit the
* codec sysfs
* @codec: the codec object
*
* first we need to remove sysfs, then refresh widgets and lastly
* recreate it
*/
int snd_hdac_refresh_widget_sysfs(struct hdac_device *codec)
{
int ret;
if (device_is_registered(&codec->dev))
hda_widget_sysfs_exit(codec);
ret = snd_hdac_refresh_widgets(codec);
if (ret) {
dev_err(&codec->dev, "failed to refresh widget: %d\n", ret);
return ret;
}
if (device_is_registered(&codec->dev)) {
ret = hda_widget_sysfs_init(codec);
if (ret) {
dev_err(&codec->dev, "failed to init sysfs: %d\n", ret);
return ret;
}
}
return ret;
}
EXPORT_SYMBOL_GPL(snd_hdac_refresh_widget_sysfs);
/* return CONNLIST_LEN parameter of the given widget */
static unsigned int get_num_conns(struct hdac_device *codec, hda_nid_t nid)
{

47
sound/hda/hdac_sysfs.c
View File

@ -414,3 +414,50 @@ void hda_widget_sysfs_exit(struct hdac_device *codec)
{
widget_tree_free(codec);
}
int hda_widget_sysfs_reinit(struct hdac_device *codec,
hda_nid_t start_nid, int num_nodes)
{
struct hdac_widget_tree *tree;
hda_nid_t end_nid = start_nid + num_nodes;
hda_nid_t nid;
int i;
if (!codec->widgets)
return hda_widget_sysfs_init(codec);
tree = kmemdup(codec->widgets, sizeof(*tree), GFP_KERNEL);
if (!tree)
return -ENOMEM;
tree->nodes = kcalloc(num_nodes + 1, sizeof(*tree->nodes), GFP_KERNEL);
if (!tree->nodes) {
kfree(tree);
return -ENOMEM;
}
/* prune non-existing nodes */
for (i = 0, nid = codec->start_nid; i < codec->num_nodes; i++, nid++) {
if (nid < start_nid || nid >= end_nid)
free_widget_node(codec->widgets->nodes[i],
&widget_node_group);
}
/* add new nodes */
for (i = 0, nid = start_nid; i < num_nodes; i++, nid++) {
if (nid < codec->start_nid || nid >= codec->end_nid)
add_widget_node(tree->root, nid, &widget_node_group,
&tree->nodes[i]);
else
tree->nodes[i] =
codec->widgets->nodes[nid - codec->start_nid];
}
/* replace with the new tree */
kfree(codec->widgets->nodes);
kfree(codec->widgets);
codec->widgets = tree;
kobject_uevent(tree->root, KOBJ_CHANGE);
return 0;
}

2
sound/hda/local.h
View File

@ -28,6 +28,8 @@ static inline unsigned int get_wcaps_channels(u32 wcaps)
extern const struct attribute_group *hdac_dev_attr_groups[];
int hda_widget_sysfs_init(struct hdac_device *codec);
int hda_widget_sysfs_reinit(struct hdac_device *codec, hda_nid_t start_nid,
int num_nodes);
void hda_widget_sysfs_exit(struct hdac_device *codec);
#endif /* __HDAC_LOCAL_H */

8
sound/i2c/other/ak4117.c
View File

@ -35,7 +35,7 @@ MODULE_LICENSE("GPL");
#define AK4117_ADDR 0x00 /* fixed address */
static void snd_ak4117_timer(unsigned long data);
static void snd_ak4117_timer(struct timer_list *t);
static void reg_write(struct ak4117 *ak4117, unsigned char reg, unsigned char val)
{
@ -91,7 +91,7 @@ int snd_ak4117_create(struct snd_card *card, ak4117_read_t *read, ak4117_write_t
chip->read = read;
chip->write = write;
chip->private_data = private_data;
setup_timer(&chip->timer, snd_ak4117_timer, (unsigned long)chip);
timer_setup(&chip->timer, snd_ak4117_timer, 0);
for (reg = 0; reg < 5; reg++)
chip->regmap[reg] = pgm[reg];
@ -529,9 +529,9 @@ int snd_ak4117_check_rate_and_errors(struct ak4117 *ak4117, unsigned int flags)
return res;
}
static void snd_ak4117_timer(unsigned long data)
static void snd_ak4117_timer(struct timer_list *t)
{
struct ak4117 *chip = (struct ak4117 *)data;
struct ak4117 *chip = from_timer(chip, t, timer);
if (chip->init)
return;

6
sound/isa/sb/emu8000_pcm.c
View File

@ -193,9 +193,9 @@ static inline int emu8k_get_curpos(struct snd_emu8k_pcm *rec, int ch)
* timer interrupt handler
* check the current position and update the period if necessary.
*/
static void emu8k_pcm_timer_func(unsigned long data)
static void emu8k_pcm_timer_func(struct timer_list *t)
{
struct snd_emu8k_pcm *rec = (struct snd_emu8k_pcm *)data;
struct snd_emu8k_pcm *rec = from_timer(rec, t, timer);
int ptr, delta;
spin_lock(&rec->timer_lock);
@ -241,7 +241,7 @@ static int emu8k_pcm_open(struct snd_pcm_substream *subs)
runtime->private_data = rec;
spin_lock_init(&rec->timer_lock);
setup_timer(&rec->timer, emu8k_pcm_timer_func, (unsigned long)rec);
timer_setup(&rec->timer, emu8k_pcm_timer_func, 0);
runtime->hw = emu8k_pcm_hw;
runtime->hw.buffer_bytes_max = emu->mem_size - LOOP_BLANK_SIZE * 3;

11
sound/isa/sb/sb8_midi.c
View File

@ -138,6 +138,7 @@ static int snd_sb8dsp_midi_output_close(struct snd_rawmidi_substream *substream)
struct snd_sb *chip;
chip = substream->rmidi->private_data;
del_timer_sync(&chip->midi_timer);
spin_lock_irqsave(&chip->open_lock, flags);
chip->open &= ~(SB_OPEN_MIDI_OUTPUT | SB_OPEN_MIDI_OUTPUT_TRIGGER);
chip->midi_substream_output = NULL;
@ -209,10 +210,10 @@ static void snd_sb8dsp_midi_output_write(struct snd_rawmidi_substream *substream
}
}
static void snd_sb8dsp_midi_output_timer(unsigned long data)
static void snd_sb8dsp_midi_output_timer(struct timer_list *t)
{
struct snd_rawmidi_substream *substream = (struct snd_rawmidi_substream *) data;
struct snd_sb * chip = substream->rmidi->private_data;
struct snd_sb *chip = from_timer(chip, t, midi_timer);
struct snd_rawmidi_substream *substream = chip->midi_substream_output;
unsigned long flags;
spin_lock_irqsave(&chip->open_lock, flags);
@ -230,9 +231,6 @@ static void snd_sb8dsp_midi_output_trigger(struct snd_rawmidi_substream *substre
spin_lock_irqsave(&chip->open_lock, flags);
if (up) {
if (!(chip->open & SB_OPEN_MIDI_OUTPUT_TRIGGER)) {
setup_timer(&chip->midi_timer,
snd_sb8dsp_midi_output_timer,
(unsigned long) substream);
mod_timer(&chip->midi_timer, 1 + jiffies);
chip->open |= SB_OPEN_MIDI_OUTPUT_TRIGGER;
}
@ -275,6 +273,7 @@ int snd_sb8dsp_midi(struct snd_sb *chip, int device)
if (chip->hardware >= SB_HW_20)
rmidi->info_flags |= SNDRV_RAWMIDI_INFO_DUPLEX;
rmidi->private_data = chip;
timer_setup(&chip->midi_timer, snd_sb8dsp_midi_output_timer, 0);
chip->rmidi = rmidi;
return 0;
}

10
sound/isa/wavefront/wavefront_midi.c
View File

@ -349,10 +349,10 @@ static void snd_wavefront_midi_input_trigger(struct snd_rawmidi_substream *subst
spin_unlock_irqrestore (&midi->virtual, flags);
}
static void snd_wavefront_midi_output_timer(unsigned long data)
static void snd_wavefront_midi_output_timer(struct timer_list *t)
{
snd_wavefront_card_t *card = (snd_wavefront_card_t *)data;
snd_wavefront_midi_t *midi = &card->wavefront.midi;
snd_wavefront_midi_t *midi = from_timer(midi, t, timer);
snd_wavefront_card_t *card = midi->timer_card;
unsigned long flags;
spin_lock_irqsave (&midi->virtual, flags);
@ -383,9 +383,9 @@ static void snd_wavefront_midi_output_trigger(struct snd_rawmidi_substream *subs
if (up) {
if ((midi->mode[mpu] & MPU401_MODE_OUTPUT_TRIGGER) == 0) {
if (!midi->istimer) {
setup_timer(&midi->timer,
timer_setup(&midi->timer,
snd_wavefront_midi_output_timer,
(unsigned long) substream->rmidi->card->private_data);
0);
mod_timer(&midi->timer, 1 + jiffies);
}
midi->istimer++;

369
sound/oss/CHANGELOG
View File

@ -1,369 +0,0 @@
Note these changes relate to Hannu's code and don't include the changes
made outside of this for modularising the sound
Changelog for version 3.8o
--------------------------
Since 3.8h
- Included support for OPL3-SA1 and SoftOSS
Since 3.8
- Fixed SNDCTL_DSP_GETOSPACE
- Compatibility fixes for Linux 2.1.47
Since 3.8-beta21
- Fixed all known bugs (I think).
Since 3.8-beta8
- Lot of fixes to audio playback code in dmabuf.c
Since 3.8-beta6
- Fixed the famous Quake delay bug.
Since 3.8-beta5
- Fixed many bugs in audio playback.
Since 3.8-beta4
- Just minor changes.
Since 3.8-beta1
- Major rewrite of audio playback handling.
- Added AWE32 support by Takashi Iwai (in ./lowlevel/).
Since 3.7-beta#
- Passing of ioctl() parameters between soundcard.c and other modules has been
changed so that arg always points to kernel space.
- Some bugfixes.
Since 3.7-beta5
- Disabled MIDI input with GUS PnP (Interwave). There seems to be constant
stream of received 0x00 bytes when the MIDI receiver is enabled.
Since 3.5
- Changes almost everywhere.
- Support for OPTi 82C924-based sound cards.
Since 3.5.4-beta8
- Fixed a bug in handling of non-fragment sized writes in 16 bit/stereo mode
with GUS.
- Limited minimum fragment size with some audio devices (GUS=512 and
SB=32). These devices require more time to "recover" from processing
of each fragment.
Since 3.5.4-beta6/7
- There seems to be problems in the OPTi 82C930 so cards based on this
chip don't necessarily work yet. There are problems in detecting the
MIDI interface. Also mixer volumes may be seriously wrong on some systems.
You can safely use this driver version with C930 if it looks to work.
However please don't complain if you have problems with it. C930 support
should be fixed in future releases.
- Got initialization of GUS PnP to work. With this version GUS PnP should
work in GUS compatible mode after initialization using isapnptools.
- Fixed a bug in handling of full duplex cards in write only mode. This has
been causing "audio device opening" errors with RealAudio player.
Since 3.5.4.beta5
- Changes to OPTi 82C930 driver.
- Major changes to the Soundscape driver. The driver requires now just one
DMA channel. The extra audio/dsp device (the "Not functional" one) used
for code download in the earlier versions has been eliminated. There is now
just one /dev/dsp# device which is used both for code download and audio.
Since 3.5.4.beta4
- Minor changes.
Since 3.5.4-beta2
- Fixed silent playback with ESS 688/1688.
- Got SB16 to work without the 16 bit DMA channel (only the 8 bit one
is required for 8 and 16 bit modes).
- Added the "lowlevel" subdirectory for additional low level drivers that
are not part of USS core. See lowlevel/README for more info.
- Included support for ACI mixer (by Markus Kuhn). ACI is a mixer used in
miroPCM sound cards. See lowlevel/aci.readme for more info.
- Support for Aztech Washington chipset (AZT2316 ASIC).
Since 3.5.4-beta1
- Reduced clicking with AD1848.
- Support for OPTi 82C930. Only half duplex at this time. 16 bit playback
is sometimes just white noise (occurs randomly).
Since 3.5.2
- Major changes to the SB/Jazz16/ESS driver (most parts rewritten).
The most noticeable new feature is support for multiple SB cards at the same
time.
- Renamed sb16_midi.c to uart401.c. Also modified it to work also with
other MPU401 UART compatible cards than SB16/ESS/Jazz.
- Some changes which reduce clicking in audio playback.
- Copying policy is now GPL.
Since 3.5.1
- TB Maui initialization support
Since 3.5
- Improved handling of playback underrun situations.
Since 3.5-beta10
- Bug fixing
Since 3.5-beta9
- Fixed for compatibility with Linux 1.3.70 and later.
- Changed boot time passing of 16 bit DMA channel number to SB driver.
Since 3.5-beta8
- Minor changes
Since 3.5-beta7
- enhancements to configure program (by Jeff Tranter):
- prompts are in same format as 1.3.x Linux kernel config program
- on-line help for each question
- fixed some compile warnings detected by gcc/g++ -Wall
- minor grammatical changes to prompts
Since 3.5-beta6
- Fixed bugs in mmap() support.
- Minor changes to Maui driver.
Since 3.5-beta5
- Fixed crash after recording with ESS688. It's generally a good
idea to stop inbound DMA transfers before freeing the memory
buffer.
- Fixed handling of AD1845 codec (for example Shuttle Sound System).
- Few other fixes.
Since 3.5-beta4
- Fixed bug in handling of uninitialized instruments with GUS.
Since 3.5-beta3
- Few changes which decrease popping at end/beginning of audio playback.
Since 3.5-beta2
- Removed MAD16+CS4231 hack made in previous version since it didn't
help.
- Fixed the above bug in proper way and in proper place. Many thanks
to James Hightower.
Since 3.5-beta1
- Bug fixes.
- Full duplex audio with MAD16+CS4231 may work now. The driver configures
SB DMA of MAD16 so that it doesn't conflict with codec's DMA channels.
The side effect is that all 8 bit DMA channels (0,1,3) are populated in
duplex mode.
Since 3.5-alpha9
- Bug fixes (mostly in Jazz16 and ESS1688/688 supports).
- Temporarily disabled recording with ESS1688/688 since it causes crash.
- Changed audio buffer partitioning algorithm so that it selects
smaller fragment size than earlier. This improves real time capabilities
of the driver and makes recording to disk to work better. Unfortunately
this change breaks some programs which assume that fragments cannot be
shorter than 4096 bytes.
Since 3.5-alpha8
- Bug fixes
Since 3.5-alpha7
- Linux kernel compatible configuration (_EXPERIMENTAL_). Enable
using command "cd /linux/drivers/sound;make script" and then
just run kernel's make config normally.
- Minor fixes to the SB support. Hopefully the driver works with
all SB models now.
- Added support for ESS ES1688 "AudioDrive" based cards.
Since 3.5-alpha6
- SB Pro and SB16 supports are no longer separately selectable options.
Enabling SB enables them too.
- Changed all #ifndef EXCLUDE_xx stuff to #ifdef CONFIG_xx. Modified
configure to handle this.
- Removed initialization messages from the
modularized version. They can be enabled by using init_trace=1 in
the insmod command line (insmod sound init_trace=1).
- More AIX stuff.
- Added support for synchronizing dsp/audio devices with /dev/sequencer.
- mmap() support for dsp/audio devices.
Since 3.5-alpha5
- AIX port.
- Changed some xxx_PATCH macros in soundcard.h to work with
big endian machines.
Since 3.5-alpha4
- Removed the 'setfx' stuff from the version distributed with kernel
sources. Running 'setfx' is required again.
Since 3.5-alpha3
- Moved stuff from the 'setfx' program to the AudioTrix Pro driver.
Since 3.5-alpha2
- Modifications to makefile and configure.c. Unnecessary sources
are no longer compiled. Newly created local.h is also copied to
/etc/soundconf. "make oldconfig" reads /etc/soundconf and produces
new local.h which is compatible with current version of the driver.
- Some fixes to the SB16 support.
- Fixed random protection fault in gus_wave.c
Since 3.5-alpha1
- Modified to work with Linux-1.3.33 and later
- Some minor changes
Since 3.0.2
- Support for CS4232 based PnP cards (AcerMagic S23 etc).
- Full duplex support for some CS4231, CS4232 and AD1845 based cards
(GUS MAX, AudioTrix Pro, AcerMagic S23 and many MAD16/Mozart cards
having a codec mentioned above).
- Almost fully rewritten loadable modules support.
- Fixed some bugs.
- Huge amount of testing (more testing is still required).
- mmap() support (works with some cards). Requires much more testing.
- Sample/patch/program loading for TB Maui/Tropez. No initialization
since TB doesn't allow me to release that code.
- Using CS4231 compatible codecs as timer for /dev/music.
Since 3.0.1
- Added allocation of I/O ports, DMA channels and interrupts
to the initialization code. This may break modules support since
the driver may not free some resources on unload. Should be fixed soon.
Since 3.0
- Some important bug fixes.
- select() for /dev/dsp and /dev/audio (Linux only).
(To use select() with read, you have to call read() to start
the recording. Calling write() kills recording immediately so
use select() carefully when you are writing a half duplex app.
Full duplex mode is not implemented yet.) Select works also with
/dev/sequencer and /dev/music. Maybe with /dev/midi## too.
Since 3.0-beta2
- Minor fixes.
- Added Readme.cards
Since 3.0-beta1
- Minor fixes to the modules support.
- Eliminated call to sb_free_irq() in ad1848.c
- Rewritten MAD16&Mozart support (not tested with MAD16 Pro).
- Fix to DMA initialization of PSS cards.
- Some fixes to ad1848/cs42xx mixer support (GUS MAX, MSS, etc.)
- Fixed some bugs in the PSS driver which caused I/O errors with
the MSS mode (/dev/dsp).
Since 3.0-950506
- Recording with GUS MAX fixed. It works when the driver is configured
to use two DMA channels with GUS MAX (16 bit ones recommended).
Since 3.0-94xxxx
- Too many changes
Since 3.0-940818
- Fixes for Linux 1.1.4x.
- Disables Disney Sound System with SG NX Pro 16 (less noise).
Since 2.90-2
- Fixes to soundcard.h
- Non blocking mode to /dev/sequencer
- Experimental detection code for Ensoniq Soundscape.
Since 2.90
- Minor and major bug fixes
Since pre-3.0-940712
- GUS MAX support
- Partially working MSS/WSS support (could work with some cards).
- Hardware u-Law and A-Law support with AD1848/CS4248 and CS4231 codecs
(GUS MAX, GUS16, WSS etc). Hardware ADPCM is possible with GUS16 and
GUS MAX, but it doesn't work yet.
Since pre-3.0-940426
- AD1848/CS4248/CS4231 codec support (MSS, GUS MAX, Aztec, Orchid etc).
This codec chip is used in various sound cards. This version is developed
for the 16 bit daughtercard of GUS. It should work with other cards also
if the following requirements are met:
- The I/O, IRQ and DMA settings are jumper selectable or
the card is initialized by booting DOS before booting Linux (etc.).
- You add the IO, IRQ and DMA settings manually to the local.h.
(Just define GUS16_BASE, GUS16_IRQ and GUS16_DMA). Note that
the base address bust be the base address of the codec chip not the
card itself. For the GUS16 these are the same but most MSS compatible
cards have the codec located at card_base+4.
- Some minor changes
Since 2.5 (******* MAJOR REWRITE ***********)
This version is based on v2.3. I have tried to maintain two versions
together so that this one should have the same features than v2.5.
Something may still be missing. If you notice such things, please let me
know.
The Readme.v30 contains more details.
- /dev/midi## devices.
- /dev/sequencer2
Since 2.5-beta2
- Some fine tuning to the GUS v3.7 mixer code.
- Fixed speed limits for the plain SB (1.0 to 2.0).
Since 2.5-beta
- Fixed OPL-3 detection with SB. Caused problems with PAS16.
- GUS v3.7 mixer support.
Since 2.4
- Mixer support for Sound Galaxy NX Pro (define __SGNXPRO__ on your local.h).
- Fixed truncated sound on /dev/dsp when the device is closed.
- Linear volume mode for GUS
- Pitch bends larger than +/- 2 octaves.
- MIDI recording for SB and SB Pro. (Untested).
- Some other fixes.
- SB16 MIDI and DSP drivers only initialized if SB16 actually installed.
- Implemented better detection for OPL-3. This should be useful if you
have an old SB Pro (the non-OPL-3 one) or a SB 2.0 clone which has a OPL-3.
- SVR4.2 support by Ian Hartas. Initial ALPHA TEST version (untested).
Since 2.3b
- Fixed bug which made it impossible to make long recordings to disk.
Recording was not restarted after a buffer overflow situation.
- Limited mixer support for GUS.
- Numerous improvements to the GUS driver by Andrew Robinson. Including
some click removal etc.
Since 2.3
- Fixed some minor bugs in the SB16 driver.
Since 2.2b
- Full SB16 DSP support. 8/16 bit, mono/stereo
- The SCO and FreeBSD versions should be in sync now. There are some
problems with SB16 and GUS in the FreeBSD versions.
The DMA buffer allocation of the SCO version has been polished but
there could still be some problems. At least it hogs memory.
The DMA channel
configuration method used in the SCO/System is a hack.
- Support for the MPU emulation of the SB16.
- Some big arrays are now allocated boot time. This makes the BSS segment
smaller which makes it possible to use the full driver with
NetBSD. These arrays are not allocated if no suitable sound card is available.
- Fixed a bug in the compute_and_set_volume in gus_wave.c
- Fixed the too fast mono playback problem of SB Pro and PAS16.
Since 2.2
- Stereo recording for SB Pro. Somehow it was missing and nobody
had noticed it earlier.
- Minor polishing.
- Interpreting of boot time arguments (sound=) for Linux.
- Breakup of sb_dsp.c. Parts of the code has been moved to
sb_mixer.c and sb_midi.c
Since 2.1
- Preliminary support for SB16.
- The SB16 mixer is supported in its native mode.
- Digitized voice capability up to 44.1 kHz/8 bit/mono
(16 bit and stereo support coming in the next release).
- Fixed some bugs in the digitized voice driver for PAS16.
- Proper initialization of the SB emulation of latest PAS16 models.
- Significantly improved /dev/dsp and /dev/audio support.
- Now supports half duplex mode. It's now possible to record and
playback without closing and reopening the device.
- It's possible to use smaller buffers than earlier. There is a new
ioctl(fd, SNDCTL_DSP_SUBDIVIDE, &n) where n should be 1, 2 or 4.
This call instructs the driver to use smaller buffers. The default
buffer size (0.5 to 1.0 seconds) is divided by n. Should be called
immediately after opening the device.
Since 2.0
Just cosmetic changes.

533
sound/oss/Kconfig
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@ -1,533 +0,0 @@
# 18 Apr 1998, Michael Elizabeth Chastain, <mailto:mec@shout.net>
# More hacking for modularisation.
#
# Prompt user for primary drivers.
config SOUND_BCM_CS4297A
tristate "Crystal Sound CS4297a (for Swarm)"
depends on SIBYTE_SWARM
help
The BCM91250A has a Crystal CS4297a on synchronous serial
port B (in addition to the DB-9 serial port). Say Y or M
here to enable the sound chip instead of the UART. Also
note that CONFIG_KGDB should not be enabled at the same
time, since it also attempts to use this UART port.
config SOUND_MSNDCLAS
tristate "Support for Turtle Beach MultiSound Classic, Tahiti, Monterey"
depends on (m || !STANDALONE) && ISA
help
Say M here if you have a Turtle Beach MultiSound Classic, Tahiti or
Monterey (not for the Pinnacle or Fiji).
See <file:Documentation/sound/oss/MultiSound> for important information
about this driver. Note that it has been discontinued, but the
Voyetra Turtle Beach knowledge base entry for it is still available
at <http://www.turtlebeach.com/site/kb_ftp/790.asp>.
comment "Compiled-in MSND Classic support requires firmware during compilation."
depends on SOUND_PRIME && SOUND_MSNDCLAS=y
config MSNDCLAS_HAVE_BOOT
bool
depends on SOUND_MSNDCLAS=y && !STANDALONE
default y
config MSNDCLAS_INIT_FILE
string "Full pathname of MSNDINIT.BIN firmware file"
depends on SOUND_MSNDCLAS
default "/etc/sound/msndinit.bin"
help
The MultiSound cards have two firmware files which are required for
operation, and are not currently included. These files can be
obtained from Turtle Beach. See
<file:Documentation/sound/oss/MultiSound> for information on how to
obtain this.
config MSNDCLAS_PERM_FILE
string "Full pathname of MSNDPERM.BIN firmware file"
depends on SOUND_MSNDCLAS
default "/etc/sound/msndperm.bin"
help
The MultiSound cards have two firmware files which are required for
operation, and are not currently included. These files can be
obtained from Turtle Beach. See
<file:Documentation/sound/oss/MultiSound> for information on how to
obtain this.
config MSNDCLAS_IRQ
int "MSND Classic IRQ 5, 7, 9, 10, 11, 12"
depends on SOUND_MSNDCLAS=y
default "5"
help
Interrupt Request line for the MultiSound Classic and related cards.
config MSNDCLAS_MEM
hex "MSND Classic memory B0000, C8000, D0000, D8000, E0000, E8000"
depends on SOUND_MSNDCLAS=y
default "D0000"
help
Memory-mapped I/O base address for the MultiSound Classic and
related cards.
config MSNDCLAS_IO
hex "MSND Classic I/O 210, 220, 230, 240, 250, 260, 290, 3E0"
depends on SOUND_MSNDCLAS=y
default "290"
help
I/O port address for the MultiSound Classic and related cards.
config SOUND_MSNDPIN
tristate "Support for Turtle Beach MultiSound Pinnacle, Fiji"
depends on (m || !STANDALONE) && ISA
help
Say M here if you have a Turtle Beach MultiSound Pinnacle or Fiji.
See <file:Documentation/sound/oss/MultiSound> for important information
about this driver. Note that it has been discontinued, but the
Voyetra Turtle Beach knowledge base entry for it is still available
at <http://www.turtlebeach.com/site/kb_ftp/600.asp>.
comment "Compiled-in MSND Pinnacle support requires firmware during compilation."
depends on SOUND_PRIME && SOUND_MSNDPIN=y
config MSNDPIN_HAVE_BOOT
bool
depends on SOUND_MSNDPIN=y
default y
config MSNDPIN_INIT_FILE
string "Full pathname of PNDSPINI.BIN firmware file"
depends on SOUND_MSNDPIN
default "/etc/sound/pndspini.bin"
help
The MultiSound cards have two firmware files which are required
for operation, and are not currently included. These files can be
obtained from Turtle Beach. See
<file:Documentation/sound/oss/MultiSound> for information on how to
obtain this.
config MSNDPIN_PERM_FILE
string "Full pathname of PNDSPERM.BIN firmware file"
depends on SOUND_MSNDPIN
default "/etc/sound/pndsperm.bin"
help
The MultiSound cards have two firmware files which are required for
operation, and are not currently included. These files can be
obtained from Turtle Beach. See
<file:Documentation/sound/oss/MultiSound> for information on how to
obtain this.
config MSNDPIN_IRQ
int "MSND Pinnacle IRQ 5, 7, 9, 10, 11, 12"
depends on SOUND_MSNDPIN=y
default "5"
help
Interrupt request line for the primary synthesizer on MultiSound
Pinnacle and Fiji sound cards.
config MSNDPIN_MEM
hex "MSND Pinnacle memory B0000, C8000, D0000, D8000, E0000, E8000"
depends on SOUND_MSNDPIN=y
default "D0000"
help
Memory-mapped I/O base address for the primary synthesizer on
MultiSound Pinnacle and Fiji sound cards.
config MSNDPIN_IO
hex "MSND Pinnacle I/O 210, 220, 230, 240, 250, 260, 290, 3E0"
depends on SOUND_MSNDPIN=y
default "290"
help
Memory-mapped I/O base address for the primary synthesizer on
MultiSound Pinnacle and Fiji sound cards.
config MSNDPIN_DIGITAL
bool "MSND Pinnacle has S/PDIF I/O"
depends on SOUND_MSNDPIN=y
help
If you have the S/PDIF daughter board for the Pinnacle or Fiji,
answer Y here; otherwise, say N. If you have this, you will be able
to play and record from the S/PDIF port (digital signal). See
<file:Documentation/sound/oss/MultiSound> for information on how to make
use of this capability.
config MSNDPIN_NONPNP
bool "MSND Pinnacle non-PnP Mode"
depends on SOUND_MSNDPIN=y
help
The Pinnacle and Fiji card resources can be configured either with
PnP, or through a configuration port. Say Y here if your card is NOT
in PnP mode. For the Pinnacle, configuration in non-PnP mode allows
use of the IDE and joystick peripherals on the card as well; these
do not show up when the card is in PnP mode. Specifying zero for any
resource of a device will disable the device. If you are running the
card in PnP mode, you must say N here and use isapnptools to
configure the card's resources.
comment "MSND Pinnacle DSP section will be configured to above parameters."
depends on SOUND_MSNDPIN=y && MSNDPIN_NONPNP
config MSNDPIN_CFG
hex "MSND Pinnacle config port 250,260,270"
depends on MSNDPIN_NONPNP
default "250"
help
This is the port which the Pinnacle and Fiji uses to configure the
card's resources when not in PnP mode. If your card is in PnP mode,
then be sure to say N to the previous option, "MSND Pinnacle Non-PnP
Mode".
comment "Pinnacle-specific Device Configuration (0 disables)"
depends on SOUND_MSNDPIN=y && MSNDPIN_NONPNP
config MSNDPIN_MPU_IO
hex "MSND Pinnacle MPU I/O (e.g. 330)"
depends on MSNDPIN_NONPNP
default "0"
help
Memory-mapped I/O base address for the Kurzweil daughterboard
synthesizer on MultiSound Pinnacle and Fiji sound cards.
config MSNDPIN_MPU_IRQ
int "MSND Pinnacle MPU IRQ (e.g. 9)"
depends on MSNDPIN_NONPNP
default "0"
help
Interrupt request number for the Kurzweil daughterboard
synthesizer on MultiSound Pinnacle and Fiji sound cards.
config MSNDPIN_IDE_IO0
hex "MSND Pinnacle IDE I/O 0 (e.g. 170)"
depends on MSNDPIN_NONPNP
default "0"
help
CD-ROM drive 0 memory-mapped I/O base address for the MultiSound
Pinnacle and Fiji sound cards.
config MSNDPIN_IDE_IO1
hex "MSND Pinnacle IDE I/O 1 (e.g. 376)"
depends on MSNDPIN_NONPNP
default "0"
help
CD-ROM drive 1 memory-mapped I/O base address for the MultiSound
Pinnacle and Fiji sound cards.
config MSNDPIN_IDE_IRQ
int "MSND Pinnacle IDE IRQ (e.g. 15)"
depends on MSNDPIN_NONPNP
default "0"
help
Interrupt request number for the IDE CD-ROM interface on the
MultiSound Pinnacle and Fiji sound cards.
config MSNDPIN_JOYSTICK_IO
hex "MSND Pinnacle joystick I/O (e.g. 200)"
depends on MSNDPIN_NONPNP
default "0"
help
Memory-mapped I/O base address for the joystick port on MultiSound
Pinnacle and Fiji sound cards.
config MSND_FIFOSIZE
int "MSND buffer size (kB)"
depends on SOUND_MSNDPIN=y || SOUND_MSNDCLAS=y
default "128"
help
Configures the size of each audio buffer, in kilobytes, for
recording and playing in the MultiSound drivers (both the Classic
and Pinnacle). Larger values reduce the chance of data overruns at
the expense of overall latency. If unsure, use the default.
menuconfig SOUND_OSS
tristate "OSS sound modules"
depends on ISA_DMA_API && (VIRT_TO_BUS || ARCH_RPC || ARCH_NETWINDER)
depends on !GENERIC_ISA_DMA_SUPPORT_BROKEN
help
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) if you haven't found a
driver for your sound card above, then pick your driver from the
list below.
if SOUND_OSS
config SOUND_TRACEINIT
bool "Verbose initialisation"
help
Verbose soundcard initialization -- affects the format of autoprobe
and initialization messages at boot time.
config SOUND_DMAP
bool "Persistent DMA buffers"
---help---
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 module.
Say Y unless you have 16MB or more RAM or a PCI sound card.
config SOUND_VMIDI
tristate "Loopback MIDI device support"
help
Support for MIDI loopback on port 1 or 2.
config SOUND_TRIX
tristate "MediaTrix AudioTrix Pro support"
help
Answer Y if you have the AudioTriX Pro sound card manufactured
by MediaTrix.
config TRIX_HAVE_BOOT
bool "Have TRXPRO.HEX firmware file"
depends on SOUND_TRIX=y && !STANDALONE
help
The MediaTrix AudioTrix Pro has an on-board microcontroller which
needs to be initialized by downloading the code from the file
TRXPRO.HEX in the DOS driver directory. If you don't have the
TRXPRO.HEX file handy you may skip this step. However, the SB and
MPU-401 modes of AudioTrix Pro will not work without this file!
config TRIX_BOOT_FILE
string "Full pathname of TRXPRO.HEX firmware file"
depends on TRIX_HAVE_BOOT
default "/etc/sound/trxpro.hex"
help
Enter the full pathname of your TRXPRO.HEX file, starting from /.
config SOUND_MSS
tristate "Microsoft Sound System support"
---help---
Again think carefully before answering Y to this question. It's
safe to answer Y if you have the original Windows Sound System card
made by Microsoft or Aztech SG 16 Pro (or NX16 Pro). Also you may
say Y in case your card is NOT among these:
ATI Stereo F/X, AdLib, Audio Excell DSP16, Cardinal DSP16,
Ensoniq SoundScape (and compatibles made by Reveal and Spea),
Gravis Ultrasound, Gravis Ultrasound ACE, Gravis Ultrasound Max,
Gravis Ultrasound with 16 bit option, Logitech Sound Man 16,
Logitech SoundMan Games, Logitech SoundMan Wave, MAD16 Pro (OPTi
82C929), Media Vision Jazz16, MediaTriX AudioTriX Pro, Microsoft
Windows Sound System (MSS/WSS), Mozart (OAK OTI-601), Orchid
SW32, Personal Sound System (PSS), Pro Audio Spectrum 16, Pro
Audio Studio 16, Pro Sonic 16, Roland MPU-401 MIDI interface,
Sound Blaster 1.0, Sound Blaster 16, Sound Blaster 16ASP, Sound
Blaster 2.0, Sound Blaster AWE32, Sound Blaster Pro, TI TM4000M
notebook, ThunderBoard, Turtle Beach Tropez, Yamaha FM
synthesizers (OPL2, OPL3 and OPL4), 6850 UART MIDI Interface.
For cards having native support in VoxWare, consult the card
specific instructions in <file:Documentation/sound/oss/README.OSS>.
Some drivers have their own MSS support and saying Y to this option
will cause a conflict.
If you compile the driver into the kernel, you have to add
"ad1848=<io>,<irq>,<dma>,<dma2>[,<type>]" to the kernel command
line.
config SOUND_MPU401
tristate "MPU-401 support (NOT for SB16)"
---help---
Be careful with this question. The MPU401 interface is supported by
all sound cards. However, some natively supported cards have their
own driver for MPU401. Enabling this MPU401 option with these cards
will cause a conflict. Also, enabling MPU401 on a system that
doesn't really have a MPU401 could cause some trouble. If your card
was in the list of supported cards, look at the card specific
instructions in the <file:Documentation/sound/oss/README.OSS> file. It
is safe to answer Y if you have a true MPU401 MIDI interface card.
If you compile the driver into the kernel, you have to add
"mpu401=<io>,<irq>" to the kernel command line.
config SOUND_PAS
tristate "ProAudioSpectrum 16 support"
---help---
Answer Y only if you have a Pro Audio Spectrum 16, ProAudio Studio
16 or Logitech SoundMan 16 sound card. Answer N if you have some
other card made by Media Vision or Logitech since those are not
PAS16 compatible. Please read <file:Documentation/sound/oss/PAS16>.
It is not necessary to add Sound Blaster support separately; it
is included in PAS support.
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.
config PAS_JOYSTICK
bool "Enable PAS16 joystick port"
depends on SOUND_PAS=y
help
Say Y here to enable the Pro Audio Spectrum 16's auxiliary joystick
port.
config SOUND_PSS
tristate "PSS (AD1848, ADSP-2115, ESC614) support"
help
Answer Y or M if you have an Orchid SW32, Cardinal DSP16, Beethoven
ADSP-16 or some other card based on the PSS chipset (AD1848 codec +
ADSP-2115 DSP chip + Echo ESC614 ASIC CHIP). For more information on
how to compile it into the kernel or as a module see the file
<file:Documentation/sound/oss/PSS>.
If you compile the driver into the kernel, you have to add
"pss=<io>,<mssio>,<mssirq>,<mssdma>,<mpuio>,<mpuirq>" to the kernel
command line.
config PSS_MIXER
bool "Enable PSS mixer (Beethoven ADSP-16 and other compatible)"
depends on SOUND_PSS
help
Answer Y for Beethoven ADSP-16. You may try to say Y also for other
cards if they have master volume, bass, treble, and you can't
control it under Linux. If you answer N for Beethoven ADSP-16, you
can't control master volume, bass, treble and synth volume.
If you said M to "PSS support" above, you may enable or disable this
PSS mixer with the module parameter pss_mixer. For more information
see the file <file:Documentation/sound/oss/PSS>.
config PSS_HAVE_BOOT
bool "Have DSPxxx.LD firmware file"
depends on SOUND_PSS && !STANDALONE
help
If you have the DSPxxx.LD file or SYNTH.LD file for you card, say Y
to include this file. Without this file the synth device (OPL) may
not work.
config PSS_BOOT_FILE
string "Full pathname of DSPxxx.LD firmware file"
depends on PSS_HAVE_BOOT
default "/etc/sound/dsp001.ld"
help
Enter the full pathname of your DSPxxx.LD file or SYNTH.LD file,
starting from /.
config SOUND_SB
tristate "100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support"
---help---
Answer Y if you have an original Sound Blaster card made by Creative
Labs or a 100% hardware compatible clone (like the Thunderboard or
SM Games). For an unknown card you may answer Y if the card claims
to be Sound Blaster-compatible.
Please read the file <file:Documentation/sound/oss/Soundblaster>.
You should also say Y here for cards based on the Avance Logic
ALS-007 and ALS-1X0 chips (read <file:Documentation/sound/oss/ALS>) and
for cards based on ESS chips (read
<file:Documentation/sound/oss/ESS1868> and
<file:Documentation/sound/oss/ESS>). If you have an IBM Mwave
card, say Y here and read <file:Documentation/sound/oss/mwave>.
If you compile the driver into the kernel and don't want to use
isapnp, you have to add "sb=<io>,<irq>,<dma>,<dma2>" to the kernel
command line.
You can say M here to compile this driver as a module; the module is
called sb.
config SOUND_YM3812
tristate "Yamaha FM synthesizer (YM3812/OPL-3) support"
---help---
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
<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 unsure, say Y.
config SOUND_UART6850
tristate "6850 UART support"
help
This option enables support for MIDI interfaces based on the 6850
UART chip. This interface is rarely found on sound cards. It's safe
to answer N to this question.
If you compile the driver into the kernel, you have to add
"uart6850=<io>,<irq>" to the kernel command line.
config SOUND_AEDSP16
tristate "Gallant Audio Cards (SC-6000 and SC-6600 based)"
---help---
Answer Y if you have a Gallant's Audio Excel DSP 16 card. This
driver supports Audio Excel DSP 16 but not the III nor PnP versions
of this card.
The Gallant's Audio Excel DSP 16 card can emulate either an SBPro or
a Microsoft Sound System card, so you should have said Y to either
"100% Sound Blaster compatibles (SB16/32/64, ESS, Jazz16) support"
or "Microsoft Sound System support", above, and you need to answer
the "MSS emulation" and "SBPro emulation" questions below
accordingly. You should say Y to one and only one of these two
questions.
Read the <file:Documentation/sound/oss/README.OSS> file and the head of
<file:sound/oss/aedsp16.c> as well as
<file:Documentation/sound/oss/AudioExcelDSP16> to get more information
about this driver and its configuration.
config SC6600
bool "SC-6600 based audio cards (new Audio Excel DSP 16)"
depends on SOUND_AEDSP16
help
The SC6600 is the new version of DSP mounted on the Audio Excel DSP
16 cards. Find in the manual the FCC ID of your audio card and
answer Y if you have an SC6600 DSP.
config SC6600_JOY
bool "Activate SC-6600 Joystick Interface"
depends on SC6600
help
Say Y here in order to use the joystick interface of the Audio Excel
DSP 16 card.
config SC6600_CDROM
int "SC-6600 CDROM Interface (4=None, 3=IDE, 1=Panasonic, 0=?Sony?)"
depends on SC6600
default "4"
help
This is used to activate the CD-ROM interface of the Audio Excel
DSP 16 card. Enter: 0 for Sony, 1 for Panasonic, 2 for IDE, 4 for no
CD-ROM present.
config SC6600_CDROMBASE
hex "SC-6600 CDROM Interface I/O Address"
depends on SC6600
default "0"
help
Base I/O port address for the CD-ROM interface of the Audio Excel
DSP 16 card.
config SOUND_VIDC
tristate "VIDC 16-bit sound"
depends on ARM && ARCH_ACORN
help
16-bit support for the VIDC onboard sound hardware found on Acorn
machines.
config SOUND_WAVEARTIST
tristate "Netwinder WaveArtist"
depends on ARM && ARCH_NETWINDER
help
Say Y here to include support for the Rockwell WaveArtist sound
system. This driver is mainly for the NetWinder.
config SOUND_KAHLUA
tristate "XpressAudio Sound Blaster emulation"
depends on SOUND_SB
endif # SOUND_OSS

107
sound/oss/Makefile
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@ -1,107 +0,0 @@
# Makefile for the Linux sound card driver
#
# 18 Apr 1998, Michael Elizabeth Chastain, <mailto:mec@shout.net>
# Rewritten to use lists instead of if-statements.
# Each configuration option enables a list of files.
obj-$(CONFIG_SOUND_OSS) += sound.o
# Please leave it as is, cause the link order is significant !
obj-$(CONFIG_SOUND_AEDSP16) += aedsp16.o
obj-$(CONFIG_SOUND_PSS) += pss.o ad1848.o mpu401.o
obj-$(CONFIG_SOUND_TRIX) += trix.o ad1848.o sb_lib.o uart401.o
obj-$(CONFIG_SOUND_MSS) += ad1848.o
obj-$(CONFIG_SOUND_PAS) += pas2.o sb.o sb_lib.o uart401.o
obj-$(CONFIG_SOUND_SB) += sb.o sb_lib.o uart401.o
obj-$(CONFIG_SOUND_KAHLUA) += kahlua.o
obj-$(CONFIG_SOUND_MPU401) += mpu401.o
obj-$(CONFIG_SOUND_UART6850) += uart6850.o
obj-$(CONFIG_SOUND_YM3812) += opl3.o
obj-$(CONFIG_SOUND_VMIDI) += v_midi.o
obj-$(CONFIG_SOUND_VIDC) += vidc_mod.o
obj-$(CONFIG_SOUND_WAVEARTIST) += waveartist.o
obj-$(CONFIG_SOUND_MSNDCLAS) += msnd.o msnd_classic.o
obj-$(CONFIG_SOUND_MSNDPIN) += msnd.o msnd_pinnacle.o
obj-$(CONFIG_SOUND_BCM_CS4297A) += swarm_cs4297a.o
obj-$(CONFIG_DMASOUND) += dmasound/
# Declare multi-part drivers.
sound-objs := \
dev_table.o soundcard.o \
audio.o dmabuf.o \
midi_synth.o midibuf.o \
sequencer.o sound_timer.o sys_timer.o
pas2-objs := pas2_card.o pas2_midi.o pas2_mixer.o pas2_pcm.o
sb-objs := sb_card.o
sb_lib-objs := sb_common.o sb_audio.o sb_midi.o sb_mixer.o sb_ess.o
vidc_mod-objs := vidc.o vidc_fill.o
hostprogs-y := bin2hex hex2hex
# Files generated that shall be removed upon make clean
clean-files := msndperm.c msndinit.c pndsperm.c pndspini.c \
pss_boot.h trix_boot.h
# Firmware files that need translation
#
# The translated files are protected by a file that keeps track
# of what name was used to build them. If the name changes, they
# will be forced to be remade.
#
# Turtle Beach MultiSound
ifeq ($(CONFIG_MSNDCLAS_HAVE_BOOT),y)
$(obj)/msnd_classic.o: $(obj)/msndperm.c $(obj)/msndinit.c
$(obj)/msndperm.c: $(patsubst "%", %, $(CONFIG_MSNDCLAS_PERM_FILE)) $(obj)/bin2hex
$(obj)/bin2hex msndperm < $< > $@
$(obj)/msndinit.c: $(patsubst "%", %, $(CONFIG_MSNDCLAS_INIT_FILE)) $(obj)/bin2hex
$(obj)/bin2hex msndinit < $< > $@
endif
ifeq ($(CONFIG_MSNDPIN_HAVE_BOOT),y)
$(obj)/msnd_pinnacle.o: $(obj)/pndsperm.c $(obj)/pndspini.c
$(obj)/pndsperm.c: $(patsubst "%", %, $(CONFIG_MSNDPIN_PERM_FILE)) $(obj)/bin2hex
$(obj)/bin2hex pndsperm < $< > $@
$(obj)/pndspini.c: $(patsubst "%", %, $(CONFIG_MSNDPIN_INIT_FILE)) $(obj)/bin2hex
$(obj)/bin2hex pndspini < $< > $@
endif
# PSS (ECHO-ADI2111)
$(obj)/pss.o: $(obj)/pss_boot.h
ifeq ($(CONFIG_PSS_HAVE_BOOT),y)
$(obj)/pss_boot.h: $(patsubst "%", %, $(CONFIG_PSS_BOOT_FILE)) $(obj)/bin2hex
$(obj)/bin2hex pss_synth < $< > $@
else
$(obj)/pss_boot.h:
$(Q)( \
echo 'static unsigned char * pss_synth = NULL;'; \
echo 'static int pss_synthLen = 0;'; \
) > $@
endif
# MediaTrix AudioTrix Pro
$(obj)/trix.o: $(obj)/trix_boot.h
ifeq ($(CONFIG_TRIX_HAVE_BOOT),y)
$(obj)/trix_boot.h: $(patsubst "%", %, $(CONFIG_TRIX_BOOT_FILE)) $(obj)/hex2hex
$(obj)/hex2hex -i trix_boot < $< > $@
else
$(obj)/trix_boot.h:
$(Q)( \
echo 'static unsigned char * trix_boot = NULL;'; \
echo 'static int trix_boot_len = 0;'; \
) > $@
endif

6
sound/oss/README.FIRST
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@ -1,6 +0,0 @@
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.
Alan Cox

3062
sound/oss/ad1848.c
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24
sound/oss/ad1848.h
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@ -1,24 +0,0 @@
#include <linux/interrupt.h>
#define AD_F_CS4231 0x0001 /* Returned if a CS4232 (or compatible) detected */
#define AD_F_CS4248 0x0001 /* Returned if a CS4248 (or compatible) detected */
#define AD1848_SET_XTAL 1
#define AD1848_MIXER_REROUTE 2
#define AD1848_REROUTE(oldctl, newctl) \
ad1848_control(AD1848_MIXER_REROUTE, ((oldctl)<<8)|(newctl))
int ad1848_init(char *name, struct resource *ports, int irq, int dma_playback,
int dma_capture, int share_dma, int *osp, struct module *owner);
void ad1848_unload (int io_base, int irq, int dma_playback, int dma_capture, int share_dma);
int ad1848_detect (struct resource *ports, int *flags, int *osp);
int ad1848_control(int cmd, int arg);
void attach_ms_sound(struct address_info * hw_config, struct resource *ports, struct module * owner);
int probe_ms_sound(struct address_info *hw_config, struct resource *ports);
void unload_ms_sound(struct address_info *hw_info);

253
sound/oss/ad1848_mixer.h
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@ -1,253 +0,0 @@
/*
* sound/oss/ad1848_mixer.h
*
* Definitions for the mixer of AD1848 and compatible codecs.
*/
/*
* Copyright (C) by Hannu Savolainen 1993-1997
*
* OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
* Version 2 (June 1991). See the "COPYING" file distributed with this software
* for more info.
*/
/*
* The AD1848 codec has generic input lines called Line, Aux1 and Aux2.
* Sound card manufacturers have connected actual inputs (CD, synth, line,
* etc) to these inputs in different order. Therefore it's difficult
* to assign mixer channels to these inputs correctly. The following
* contains two alternative mappings. The first one is for GUS MAX and
* the second is just a generic one (line1, line2 and line3).
* (Actually this is not a mapping but rather some kind of interleaving
* solution).
*/
#define MODE1_REC_DEVICES (SOUND_MASK_LINE3 | SOUND_MASK_MIC | \
SOUND_MASK_LINE1 | SOUND_MASK_IMIX)
#define SPRO_REC_DEVICES (SOUND_MASK_LINE | SOUND_MASK_MIC | \
SOUND_MASK_CD | SOUND_MASK_LINE1)
#define MODE1_MIXER_DEVICES (SOUND_MASK_LINE1 | SOUND_MASK_MIC | \
SOUND_MASK_LINE2 | \
SOUND_MASK_IGAIN | \
SOUND_MASK_PCM | SOUND_MASK_IMIX)
#define MODE2_MIXER_DEVICES (SOUND_MASK_LINE1 | SOUND_MASK_LINE2 | \
SOUND_MASK_MIC | \
SOUND_MASK_LINE3 | SOUND_MASK_SPEAKER | \
SOUND_MASK_IGAIN | \
SOUND_MASK_PCM | SOUND_MASK_IMIX)
#define MODE3_MIXER_DEVICES (MODE2_MIXER_DEVICES | SOUND_MASK_VOLUME)
/* OPTi 82C930 has no IMIX level control, but it can still be selected as an
* input
*/
#define C930_MIXER_DEVICES (SOUND_MASK_LINE1 | SOUND_MASK_LINE2 | \
SOUND_MASK_MIC | SOUND_MASK_VOLUME | \
SOUND_MASK_LINE3 | \
SOUND_MASK_IGAIN | SOUND_MASK_PCM)
#define SPRO_MIXER_DEVICES (SOUND_MASK_VOLUME | SOUND_MASK_PCM | \
SOUND_MASK_LINE | SOUND_MASK_SYNTH | \
SOUND_MASK_CD | SOUND_MASK_MIC | \
SOUND_MASK_SPEAKER | SOUND_MASK_LINE1 | \
SOUND_MASK_OGAIN)
struct mixer_def {
unsigned int regno:6; /* register number for volume */
unsigned int polarity:1; /* volume polarity: 0=normal, 1=reversed */
unsigned int bitpos:3; /* position of bits in register for volume */
unsigned int nbits:3; /* number of bits in register for volume */
unsigned int mutereg:6; /* register number for mute bit */
unsigned int mutepol:1; /* mute polarity: 0=normal, 1=reversed */
unsigned int mutepos:4; /* position of mute bit in register */
unsigned int recreg:6; /* register number for recording bit */
unsigned int recpol:1; /* recording polarity: 0=normal, 1=reversed */
unsigned int recpos:4; /* position of recording bit in register */
};
static char mix_cvt[101] = {
0, 0, 3, 7,10,13,16,19,21,23,26,28,30,32,34,35,37,39,40,42,
43,45,46,47,49,50,51,52,53,55,56,57,58,59,60,61,62,63,64,65,
65,66,67,68,69,70,70,71,72,73,73,74,75,75,76,77,77,78,79,79,
80,81,81,82,82,83,84,84,85,85,86,86,87,87,88,88,89,89,90,90,
91,91,92,92,93,93,94,94,95,95,96,96,96,97,97,98,98,98,99,99,
100
};
typedef struct mixer_def mixer_ent;
typedef mixer_ent mixer_ents[2];
/*
* Most of the mixer entries work in backwards. Setting the polarity field
* makes them to work correctly.
*
* The channel numbering used by individual sound cards is not fixed. Some
* cards have assigned different meanings for the AUX1, AUX2 and LINE inputs.
* The current version doesn't try to compensate this.
*/
#define MIX_ENT(name, reg_l, pola_l, pos_l, len_l, reg_r, pola_r, pos_r, len_r, mute_bit) \
[name] = {{reg_l, pola_l, pos_l, len_l, reg_l, 0, mute_bit, 0, 0, 8}, \
{reg_r, pola_r, pos_r, len_r, reg_r, 0, mute_bit, 0, 0, 8}}
#define MIX_ENT2(name, reg_l, pola_l, pos_l, len_l, mute_reg_l, mute_pola_l, mute_pos_l, \
rec_reg_l, rec_pola_l, rec_pos_l, \
reg_r, pola_r, pos_r, len_r, mute_reg_r, mute_pola_r, mute_pos_r, \
rec_reg_r, rec_pola_r, rec_pos_r) \
[name] = {{reg_l, pola_l, pos_l, len_l, mute_reg_l, mute_pola_l, mute_pos_l, \
rec_reg_l, rec_pola_l, rec_pos_l}, \
{reg_r, pola_r, pos_r, len_r, mute_reg_r, mute_pola_r, mute_pos_r, \
rec_reg_r, rec_pola_r, rec_pos_r}}
static mixer_ents ad1848_mix_devices[32] = {
MIX_ENT(SOUND_MIXER_VOLUME, 27, 1, 0, 4, 29, 1, 0, 4, 8),
MIX_ENT(SOUND_MIXER_BASS, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_TREBLE, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_SYNTH, 4, 1, 0, 5, 5, 1, 0, 5, 7),
MIX_ENT(SOUND_MIXER_PCM, 6, 1, 0, 6, 7, 1, 0, 6, 7),
MIX_ENT(SOUND_MIXER_SPEAKER, 26, 1, 0, 4, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_LINE, 18, 1, 0, 5, 19, 1, 0, 5, 7),
MIX_ENT(SOUND_MIXER_MIC, 0, 0, 5, 1, 1, 0, 5, 1, 8),
MIX_ENT(SOUND_MIXER_CD, 2, 1, 0, 5, 3, 1, 0, 5, 7),
MIX_ENT(SOUND_MIXER_IMIX, 13, 1, 2, 6, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_ALTPCM, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_RECLEV, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_IGAIN, 0, 0, 0, 4, 1, 0, 0, 4, 8),
MIX_ENT(SOUND_MIXER_OGAIN, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_LINE1, 2, 1, 0, 5, 3, 1, 0, 5, 7),
MIX_ENT(SOUND_MIXER_LINE2, 4, 1, 0, 5, 5, 1, 0, 5, 7),
MIX_ENT(SOUND_MIXER_LINE3, 18, 1, 0, 5, 19, 1, 0, 5, 7)
};
static mixer_ents iwave_mix_devices[32] = {
MIX_ENT(SOUND_MIXER_VOLUME, 25, 1, 0, 5, 27, 1, 0, 5, 8),
MIX_ENT(SOUND_MIXER_BASS, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_TREBLE, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_SYNTH, 4, 1, 0, 5, 5, 1, 0, 5, 7),
MIX_ENT(SOUND_MIXER_PCM, 6, 1, 0, 6, 7, 1, 0, 6, 7),
MIX_ENT(SOUND_MIXER_SPEAKER, 26, 1, 0, 4, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_LINE, 18, 1, 0, 5, 19, 1, 0, 5, 7),
MIX_ENT(SOUND_MIXER_MIC, 0, 0, 5, 1, 1, 0, 5, 1, 8),
MIX_ENT(SOUND_MIXER_CD, 2, 1, 0, 5, 3, 1, 0, 5, 7),
MIX_ENT(SOUND_MIXER_IMIX, 16, 1, 0, 5, 17, 1, 0, 5, 8),
MIX_ENT(SOUND_MIXER_ALTPCM, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_RECLEV, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_IGAIN, 0, 0, 0, 4, 1, 0, 0, 4, 8),
MIX_ENT(SOUND_MIXER_OGAIN, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_LINE1, 2, 1, 0, 5, 3, 1, 0, 5, 7),
MIX_ENT(SOUND_MIXER_LINE2, 4, 1, 0, 5, 5, 1, 0, 5, 7),
MIX_ENT(SOUND_MIXER_LINE3, 18, 1, 0, 5, 19, 1, 0, 5, 7)
};
static mixer_ents cs42xb_mix_devices[32] = {
/* Digital master volume actually has seven bits, but we only use
six to avoid the discontinuity when the analog gain kicks in. */
MIX_ENT(SOUND_MIXER_VOLUME, 46, 1, 0, 6, 47, 1, 0, 6, 7),
MIX_ENT(SOUND_MIXER_BASS, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_TREBLE, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_SYNTH, 4, 1, 0, 5, 5, 1, 0, 5, 7),
MIX_ENT(SOUND_MIXER_PCM, 6, 1, 0, 6, 7, 1, 0, 6, 7),
MIX_ENT(SOUND_MIXER_SPEAKER, 26, 1, 0, 4, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_LINE, 18, 1, 0, 5, 19, 1, 0, 5, 7),
MIX_ENT(SOUND_MIXER_MIC, 34, 1, 0, 5, 35, 1, 0, 5, 7),
MIX_ENT(SOUND_MIXER_CD, 2, 1, 0, 5, 3, 1, 0, 5, 7),
/* For the IMIX entry, it was not possible to use the MIX_ENT macro
because the mute bit is in different positions for the two
channels and requires reverse polarity. */
[SOUND_MIXER_IMIX] = {{13, 1, 2, 6, 13, 1, 0, 0, 0, 8},
{42, 1, 0, 6, 42, 1, 7, 0, 0, 8}},
MIX_ENT(SOUND_MIXER_ALTPCM, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_RECLEV, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_IGAIN, 0, 0, 0, 4, 1, 0, 0, 4, 8),
MIX_ENT(SOUND_MIXER_OGAIN, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_LINE1, 2, 1, 0, 5, 3, 1, 0, 5, 7),
MIX_ENT(SOUND_MIXER_LINE2, 4, 1, 0, 5, 5, 1, 0, 5, 7),
MIX_ENT(SOUND_MIXER_LINE3, 38, 1, 0, 6, 39, 1, 0, 6, 7)
};
/* OPTi 82C930 has somewhat different port addresses.
* Note: VOLUME == SPEAKER, SYNTH == LINE2, LINE == LINE3, CD == LINE1
* VOLUME, SYNTH, LINE, CD are not enabled above.
* MIC is level of mic monitoring direct to output. Same for CD, LINE, etc.
*/
static mixer_ents c930_mix_devices[32] = {
MIX_ENT(SOUND_MIXER_VOLUME, 22, 1, 1, 5, 23, 1, 1, 5, 7),
MIX_ENT(SOUND_MIXER_BASS, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_TREBLE, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_SYNTH, 4, 1, 1, 4, 5, 1, 1, 4, 7),
MIX_ENT(SOUND_MIXER_PCM, 6, 1, 0, 5, 7, 1, 0, 5, 7),
MIX_ENT(SOUND_MIXER_SPEAKER, 22, 1, 1, 5, 23, 1, 1, 5, 7),
MIX_ENT(SOUND_MIXER_LINE, 18, 1, 1, 4, 19, 1, 1, 4, 7),
MIX_ENT(SOUND_MIXER_MIC, 20, 1, 1, 4, 21, 1, 1, 4, 7),
MIX_ENT(SOUND_MIXER_CD, 2, 1, 1, 4, 3, 1, 1, 4, 7),
MIX_ENT(SOUND_MIXER_IMIX, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_ALTPCM, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_RECLEV, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_IGAIN, 0, 0, 0, 4, 1, 0, 0, 4, 8),
MIX_ENT(SOUND_MIXER_OGAIN, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT(SOUND_MIXER_LINE1, 2, 1, 1, 4, 3, 1, 1, 4, 7),
MIX_ENT(SOUND_MIXER_LINE2, 4, 1, 1, 4, 5, 1, 1, 4, 7),
MIX_ENT(SOUND_MIXER_LINE3, 18, 1, 1, 4, 19, 1, 1, 4, 7)
};
static mixer_ents spro_mix_devices[32] = {
MIX_ENT (SOUND_MIXER_VOLUME, 19, 0, 4, 4, 19, 0, 0, 4, 8),
MIX_ENT (SOUND_MIXER_BASS, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT (SOUND_MIXER_TREBLE, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT2(SOUND_MIXER_SYNTH, 4, 1, 1, 4, 23, 0, 3, 0, 0, 8,
5, 1, 1, 4, 23, 0, 3, 0, 0, 8),
MIX_ENT (SOUND_MIXER_PCM, 6, 1, 1, 4, 7, 1, 1, 4, 8),
MIX_ENT (SOUND_MIXER_SPEAKER, 18, 0, 3, 2, 0, 0, 0, 0, 8),
MIX_ENT2(SOUND_MIXER_LINE, 20, 0, 4, 4, 17, 1, 4, 16, 0, 2,
20, 0, 0, 4, 17, 1, 3, 16, 0, 1),
MIX_ENT2(SOUND_MIXER_MIC, 18, 0, 0, 3, 17, 1, 0, 16, 0, 0,
0, 0, 0, 0, 0, 0, 0, 0, 0, 0),
MIX_ENT2(SOUND_MIXER_CD, 21, 0, 4, 4, 17, 1, 2, 16, 0, 4,
21, 0, 0, 4, 17, 1, 1, 16, 0, 3),
MIX_ENT (SOUND_MIXER_IMIX, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT (SOUND_MIXER_ALTPCM, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT (SOUND_MIXER_RECLEV, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT (SOUND_MIXER_IGAIN, 0, 0, 0, 0, 0, 0, 0, 0, 8),
MIX_ENT (SOUND_MIXER_OGAIN, 17, 1, 6, 1, 0, 0, 0, 0, 8),
/* This is external wavetable */
MIX_ENT2(SOUND_MIXER_LINE1, 22, 0, 4, 4, 23, 1, 1, 23, 0, 4,
22, 0, 0, 4, 23, 1, 0, 23, 0, 5),
};
static int default_mixer_levels[32] =
{
0x3232, /* Master Volume */
0x3232, /* Bass */
0x3232, /* Treble */
0x4b4b, /* FM */
0x3232, /* PCM */
0x1515, /* PC Speaker */
0x2020, /* Ext Line */
0x1010, /* Mic */
0x4b4b, /* CD */
0x0000, /* Recording monitor */
0x4b4b, /* Second PCM */
0x4b4b, /* Recording level */
0x4b4b, /* Input gain */
0x4b4b, /* Output gain */
0x2020, /* Line1 */
0x2020, /* Line2 */
0x1515 /* Line3 (usually line in)*/
};
#define LEFT_CHN 0
#define RIGHT_CHN 1
/*
* Channel enable bits for ioctl(SOUND_MIXER_PRIVATE1)
*/
#ifndef AUDIO_SPEAKER
#define AUDIO_SPEAKER 0x01 /* Enable mono output */
#define AUDIO_HEADPHONE 0x02 /* Sparc only */
#define AUDIO_LINE_OUT 0x04 /* Sparc only */
#endif

1373
sound/oss/aedsp16.c
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985
sound/oss/audio.c
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@ -1,985 +0,0 @@
/*
* sound/oss/audio.c
*
* Device file manager for /dev/audio
*/
/*
* Copyright (C) by Hannu Savolainen 1993-1997
*
* OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
* Version 2 (June 1991). See the "COPYING" file distributed with this software
* for more info.
*/
/*
* Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
* Thomas Sailer : moved several static variables into struct audio_operations
* (which is grossly misnamed btw.) because they have the same
* lifetime as the rest in there and dynamic allocation saves
* 12k or so
* Thomas Sailer : use more logical O_NONBLOCK semantics
* Daniel Rodriksson: reworked the use of the device specific copy_user
* still generic
* Horst von Brand: Add missing #include <linux/string.h>
* Chris Rankin : Update the module-usage counter for the coprocessor,
* and decrement the counters again if we cannot open
* the audio device.
*/
#include <linux/stddef.h>
#include <linux/string.h>
#include <linux/kmod.h>
#include "sound_config.h"
#include "ulaw.h"
#include "coproc.h"
#define NEUTRAL8 0x80
#define NEUTRAL16 0x00
static int dma_ioctl(int dev, unsigned int cmd, void __user *arg);
static int set_format(int dev, int fmt)
{
if (fmt != AFMT_QUERY)
{
audio_devs[dev]->local_conversion = 0;
if (!(audio_devs[dev]->format_mask & fmt)) /* Not supported */
{
if (fmt == AFMT_MU_LAW)
{
fmt = AFMT_U8;
audio_devs[dev]->local_conversion = CNV_MU_LAW;
}
else
fmt = AFMT_U8; /* This is always supported */
}
audio_devs[dev]->audio_format = audio_devs[dev]->d->set_bits(dev, fmt);
audio_devs[dev]->local_format = fmt;
}
else
return audio_devs[dev]->local_format;
if (audio_devs[dev]->local_conversion)
return audio_devs[dev]->local_conversion;
else
return audio_devs[dev]->local_format;
}
int audio_open(int dev, struct file *file)
{
int ret;
int bits;
int dev_type = dev & 0x0f;
int mode = translate_mode(file);
const struct audio_driver *driver;
const struct coproc_operations *coprocessor;
dev = dev >> 4;
if (dev_type == SND_DEV_DSP16)
bits = 16;
else
bits = 8;
if (dev < 0 || dev >= num_audiodevs)
return -ENXIO;
driver = audio_devs[dev]->d;
if (!try_module_get(driver->owner))
return -ENODEV;
if ((ret = DMAbuf_open(dev, mode)) < 0)
goto error_1;
if ( (coprocessor = audio_devs[dev]->coproc) != NULL ) {
if (!try_module_get(coprocessor->owner))
goto error_2;
if ((ret = coprocessor->open(coprocessor->devc, COPR_PCM)) < 0) {
printk(KERN_WARNING "Sound: Can't access coprocessor device\n");
goto error_3;
}
}
audio_devs[dev]->local_conversion = 0;
if (dev_type == SND_DEV_AUDIO)
set_format(dev, AFMT_MU_LAW);
else
set_format(dev, bits);
audio_devs[dev]->audio_mode = AM_NONE;
return 0;
/*
* Clean-up stack: this is what needs (un)doing if
* we can't open the audio device ...
*/
error_3:
module_put(coprocessor->owner);
error_2:
DMAbuf_release(dev, mode);
error_1:
module_put(driver->owner);
return ret;
}
static void sync_output(int dev)
{
int p, i;
int l;
struct dma_buffparms *dmap = audio_devs[dev]->dmap_out;
if (dmap->fragment_size <= 0)
return;
dmap->flags |= DMA_POST;
/* Align the write pointer with fragment boundaries */
if ((l = dmap->user_counter % dmap->fragment_size) > 0)
{
int len;
unsigned long offs = dmap->user_counter % dmap->bytes_in_use;
len = dmap->fragment_size - l;
memset(dmap->raw_buf + offs, dmap->neutral_byte, len);
DMAbuf_move_wrpointer(dev, len);
}
/*
* Clean all unused buffer fragments.
*/
p = dmap->qtail;
dmap->flags |= DMA_POST;
for (i = dmap->qlen + 1; i < dmap->nbufs; i++)
{
p = (p + 1) % dmap->nbufs;
if (((dmap->raw_buf + p * dmap->fragment_size) + dmap->fragment_size) >
(dmap->raw_buf + dmap->buffsize))
printk(KERN_ERR "audio: Buffer error 2\n");
memset(dmap->raw_buf + p * dmap->fragment_size,
dmap->neutral_byte,
dmap->fragment_size);
}
dmap->flags |= DMA_DIRTY;
}
void audio_release(int dev, struct file *file)
{
const struct coproc_operations *coprocessor;
int mode = translate_mode(file);
dev = dev >> 4;
/*
* We do this in DMAbuf_release(). Why are we doing it
* here? Why don't we test the file mode before setting
* both flags? DMAbuf_release() does.
* ...pester...pester...pester...
*/
audio_devs[dev]->dmap_out->closing = 1;
audio_devs[dev]->dmap_in->closing = 1;
/*
* We need to make sure we allocated the dmap_out buffer
* before we go mucking around with it in sync_output().
*/
if (mode & OPEN_WRITE)
sync_output(dev);
if ( (coprocessor = audio_devs[dev]->coproc) != NULL ) {
coprocessor->close(coprocessor->devc, COPR_PCM);
module_put(coprocessor->owner);
}
DMAbuf_release(dev, mode);
module_put(audio_devs[dev]->d->owner);
}
static void translate_bytes(const unsigned char *table, unsigned char *buff, int n)
{
unsigned long i;
if (n <= 0)
return;
for (i = 0; i < n; ++i)
buff[i] = table[buff[i]];
}
int audio_write(int dev, struct file *file, const char __user *buf, int count)
{
int c, p, l, buf_size, used, returned;
int err;
char *dma_buf;
dev = dev >> 4;
p = 0;
c = count;
if(count < 0)
return -EINVAL;
if (!(audio_devs[dev]->open_mode & OPEN_WRITE))
return -EPERM;
if (audio_devs[dev]->flags & DMA_DUPLEX)
audio_devs[dev]->audio_mode |= AM_WRITE;
else
audio_devs[dev]->audio_mode = AM_WRITE;
if (!count) /* Flush output */
{
sync_output(dev);
return 0;
}
while (c)
{
if ((err = DMAbuf_getwrbuffer(dev, &dma_buf, &buf_size, !!(file->f_flags & O_NONBLOCK))) < 0)
{
/* Handle nonblocking mode */
if ((file->f_flags & O_NONBLOCK) && err == -EAGAIN)
return p? p : -EAGAIN; /* No more space. Return # of accepted bytes */
return err;
}
l = c;
if (l > buf_size)
l = buf_size;
returned = l;
used = l;
if (!audio_devs[dev]->d->copy_user)
{
if ((dma_buf + l) >
(audio_devs[dev]->dmap_out->raw_buf + audio_devs[dev]->dmap_out->buffsize))
{
printk(KERN_ERR "audio: Buffer error 3 (%lx,%d), (%lx, %d)\n", (long) dma_buf, l, (long) audio_devs[dev]->dmap_out->raw_buf, (int) audio_devs[dev]->dmap_out->buffsize);
return -EDOM;
}
if (dma_buf < audio_devs[dev]->dmap_out->raw_buf)
{
printk(KERN_ERR "audio: Buffer error 13 (%lx<%lx)\n", (long) dma_buf, (long) audio_devs[dev]->dmap_out->raw_buf);
return -EDOM;
}
if(copy_from_user(dma_buf, &(buf)[p], l))
return -EFAULT;
}
else audio_devs[dev]->d->copy_user (dev,
dma_buf, 0,
buf, p,
c, buf_size,
&used, &returned,
l);
l = returned;
if (audio_devs[dev]->local_conversion & CNV_MU_LAW)
{
translate_bytes(ulaw_dsp, (unsigned char *) dma_buf, l);
}
c -= used;
p += used;
DMAbuf_move_wrpointer(dev, l);
}
return count;
}
int audio_read(int dev, struct file *file, char __user *buf, int count)
{
int c, p, l;
char *dmabuf;
int buf_no;
dev = dev >> 4;
p = 0;
c = count;
if (!(audio_devs[dev]->open_mode & OPEN_READ))
return -EPERM;
if ((audio_devs[dev]->audio_mode & AM_WRITE) && !(audio_devs[dev]->flags & DMA_DUPLEX))
sync_output(dev);
if (audio_devs[dev]->flags & DMA_DUPLEX)
audio_devs[dev]->audio_mode |= AM_READ;
else
audio_devs[dev]->audio_mode = AM_READ;
while(c)
{
if ((buf_no = DMAbuf_getrdbuffer(dev, &dmabuf, &l, !!(file->f_flags & O_NONBLOCK))) < 0)
{
/*
* Nonblocking mode handling. Return current # of bytes
*/
if (p > 0) /* Avoid throwing away data */
return p; /* Return it instead */
if ((file->f_flags & O_NONBLOCK) && buf_no == -EAGAIN)
return -EAGAIN;
return buf_no;
}
if (l > c)
l = c;
/*
* Insert any local processing here.
*/
if (audio_devs[dev]->local_conversion & CNV_MU_LAW)
{
translate_bytes(dsp_ulaw, (unsigned char *) dmabuf, l);
}
{
char *fixit = dmabuf;
if(copy_to_user(&(buf)[p], fixit, l))
return -EFAULT;
}
DMAbuf_rmchars(dev, buf_no, l);
p += l;
c -= l;
}
return count - c;
}
int audio_ioctl(int dev, struct file *file, unsigned int cmd, void __user *arg)
{
int val, count;
unsigned long flags;
struct dma_buffparms *dmap;
int __user *p = arg;
dev = dev >> 4;
if (_IOC_TYPE(cmd) == 'C') {
if (audio_devs[dev]->coproc) /* Coprocessor ioctl */
return audio_devs[dev]->coproc->ioctl(audio_devs[dev]->coproc->devc, cmd, arg, 0);
/* else
printk(KERN_DEBUG"/dev/dsp%d: No coprocessor for this device\n", dev); */
return -ENXIO;
}
else switch (cmd)
{
case SNDCTL_DSP_SYNC:
if (!(audio_devs[dev]->open_mode & OPEN_WRITE))
return 0;
if (audio_devs[dev]->dmap_out->fragment_size == 0)
return 0;
sync_output(dev);
DMAbuf_sync(dev);
DMAbuf_reset(dev);
return 0;
case SNDCTL_DSP_POST:
if (!(audio_devs[dev]->open_mode & OPEN_WRITE))
return 0;
if (audio_devs[dev]->dmap_out->fragment_size == 0)
return 0;
audio_devs[dev]->dmap_out->flags |= DMA_POST | DMA_DIRTY;
sync_output(dev);
dma_ioctl(dev, SNDCTL_DSP_POST, NULL);
return 0;
case SNDCTL_DSP_RESET:
audio_devs[dev]->audio_mode = AM_NONE;
DMAbuf_reset(dev);
return 0;
case SNDCTL_DSP_GETFMTS:
val = audio_devs[dev]->format_mask | AFMT_MU_LAW;
break;
case SNDCTL_DSP_SETFMT:
if (get_user(val, p))
return -EFAULT;
val = set_format(dev, val);
break;
case SNDCTL_DSP_GETISPACE:
if (!(audio_devs[dev]->open_mode & OPEN_READ))
return 0;
if ((audio_devs[dev]->audio_mode & AM_WRITE) && !(audio_devs[dev]->flags & DMA_DUPLEX))
return -EBUSY;
return dma_ioctl(dev, cmd, arg);
case SNDCTL_DSP_GETOSPACE:
if (!(audio_devs[dev]->open_mode & OPEN_WRITE))
return -EPERM;
if ((audio_devs[dev]->audio_mode & AM_READ) && !(audio_devs[dev]->flags & DMA_DUPLEX))
return -EBUSY;
return dma_ioctl(dev, cmd, arg);
case SNDCTL_DSP_NONBLOCK:
spin_lock(&file->f_lock);
file->f_flags |= O_NONBLOCK;
spin_unlock(&file->f_lock);
return 0;
case SNDCTL_DSP_GETCAPS:
val = 1 | DSP_CAP_MMAP; /* Revision level of this ioctl() */
if (audio_devs[dev]->flags & DMA_DUPLEX &&
audio_devs[dev]->open_mode == OPEN_READWRITE)
val |= DSP_CAP_DUPLEX;
if (audio_devs[dev]->coproc)
val |= DSP_CAP_COPROC;
if (audio_devs[dev]->d->local_qlen) /* Device has hidden buffers */
val |= DSP_CAP_BATCH;
if (audio_devs[dev]->d->trigger) /* Supports SETTRIGGER */
val |= DSP_CAP_TRIGGER;
break;
case SOUND_PCM_WRITE_RATE:
if (get_user(val, p))
return -EFAULT;
val = audio_devs[dev]->d->set_speed(dev, val);
break;
case SOUND_PCM_READ_RATE:
val = audio_devs[dev]->d->set_speed(dev, 0);
break;
case SNDCTL_DSP_STEREO:
if (get_user(val, p))
return -EFAULT;
if (val > 1 || val < 0)
return -EINVAL;
val = audio_devs[dev]->d->set_channels(dev, val + 1) - 1;
break;
case SOUND_PCM_WRITE_CHANNELS:
if (get_user(val, p))
return -EFAULT;
val = audio_devs[dev]->d->set_channels(dev, val);
break;
case SOUND_PCM_READ_CHANNELS:
val = audio_devs[dev]->d->set_channels(dev, 0);
break;
case SOUND_PCM_READ_BITS:
val = audio_devs[dev]->d->set_bits(dev, 0);
break;
case SNDCTL_DSP_SETDUPLEX:
if (audio_devs[dev]->open_mode != OPEN_READWRITE)
return -EPERM;
return (audio_devs[dev]->flags & DMA_DUPLEX) ? 0 : -EIO;
case SNDCTL_DSP_PROFILE:
if (get_user(val, p))
return -EFAULT;
if (audio_devs[dev]->open_mode & OPEN_WRITE)
audio_devs[dev]->dmap_out->applic_profile = val;
if (audio_devs[dev]->open_mode & OPEN_READ)
audio_devs[dev]->dmap_in->applic_profile = val;
return 0;
case SNDCTL_DSP_GETODELAY:
dmap = audio_devs[dev]->dmap_out;
if (!(audio_devs[dev]->open_mode & OPEN_WRITE))
return -EINVAL;
if (!(dmap->flags & DMA_ALLOC_DONE))
{
val=0;
break;
}
spin_lock_irqsave(&dmap->lock,flags);
/* Compute number of bytes that have been played */
count = DMAbuf_get_buffer_pointer (dev, dmap, DMODE_OUTPUT);
if (count < dmap->fragment_size && dmap->qhead != 0)
count += dmap->bytes_in_use; /* Pointer wrap not handled yet */
count += dmap->byte_counter;
/* Subtract current count from the number of bytes written by app */
count = dmap->user_counter - count;
if (count < 0)
count = 0;
spin_unlock_irqrestore(&dmap->lock,flags);
val = count;
break;
default:
return dma_ioctl(dev, cmd, arg);
}
return put_user(val, p);
}
void audio_init_devices(void)
{
/*
* NOTE! This routine could be called several times during boot.
*/
}
void reorganize_buffers(int dev, struct dma_buffparms *dmap, int recording)
{
/*
* This routine breaks the physical device buffers to logical ones.
*/
struct audio_operations *dsp_dev = audio_devs[dev];
unsigned i, n;
unsigned sr, nc, sz, bsz;
sr = dsp_dev->d->set_speed(dev, 0);
nc = dsp_dev->d->set_channels(dev, 0);
sz = dsp_dev->d->set_bits(dev, 0);
if (sz == 8)
dmap->neutral_byte = NEUTRAL8;
else
dmap->neutral_byte = NEUTRAL16;
if (sr < 1 || nc < 1 || sz < 1)
{
/* printk(KERN_DEBUG "Warning: Invalid PCM parameters[%d] sr=%d, nc=%d, sz=%d\n", dev, sr, nc, sz);*/
sr = DSP_DEFAULT_SPEED;
nc = 1;
sz = 8;
}
sz = sr * nc * sz;
sz /= 8; /* #bits -> #bytes */
dmap->data_rate = sz;
if (!dmap->needs_reorg)
return;
dmap->needs_reorg = 0;
if (dmap->fragment_size == 0)
{
/* Compute the fragment size using the default algorithm */
/*
* Compute a buffer size for time not exceeding 1 second.
* Usually this algorithm gives a buffer size for 0.5 to 1.0 seconds
* of sound (using the current speed, sample size and #channels).
*/
bsz = dmap->buffsize;
while (bsz > sz)
bsz /= 2;
if (bsz == dmap->buffsize)
bsz /= 2; /* Needs at least 2 buffers */
/*
* Split the computed fragment to smaller parts. After 3.5a9
* the default subdivision is 4 which should give better
* results when recording.
*/
if (dmap->subdivision == 0) /* Not already set */
{
dmap->subdivision = 4; /* Init to the default value */
if ((bsz / dmap->subdivision) > 4096)
dmap->subdivision *= 2;
if ((bsz / dmap->subdivision) < 4096)
dmap->subdivision = 1;
}
bsz /= dmap->subdivision;
if (bsz < 16)
bsz = 16; /* Just a sanity check */
dmap->fragment_size = bsz;
}
else
{
/*
* The process has specified the buffer size with SNDCTL_DSP_SETFRAGMENT or
* the buffer size computation has already been done.
*/
if (dmap->fragment_size > (dmap->buffsize / 2))
dmap->fragment_size = (dmap->buffsize / 2);
bsz = dmap->fragment_size;
}
if (audio_devs[dev]->min_fragment)
if (bsz < (1 << audio_devs[dev]->min_fragment))
bsz = 1 << audio_devs[dev]->min_fragment;
if (audio_devs[dev]->max_fragment)
if (bsz > (1 << audio_devs[dev]->max_fragment))
bsz = 1 << audio_devs[dev]->max_fragment;
bsz &= ~0x07; /* Force size which is multiple of 8 bytes */
#ifdef OS_DMA_ALIGN_CHECK
OS_DMA_ALIGN_CHECK(bsz);
#endif
n = dmap->buffsize / bsz;
if (n > MAX_SUB_BUFFERS)
n = MAX_SUB_BUFFERS;
if (n > dmap->max_fragments)
n = dmap->max_fragments;
if (n < 2)
{
n = 2;
bsz /= 2;
}
dmap->nbufs = n;
dmap->bytes_in_use = n * bsz;
dmap->fragment_size = bsz;
dmap->max_byte_counter = (dmap->data_rate * 60 * 60) +
dmap->bytes_in_use; /* Approximately one hour */
if (dmap->raw_buf)
{
memset(dmap->raw_buf, dmap->neutral_byte, dmap->bytes_in_use);
}
for (i = 0; i < dmap->nbufs; i++)
{
dmap->counts[i] = 0;
}
dmap->flags |= DMA_ALLOC_DONE | DMA_EMPTY;
}
static int dma_subdivide(int dev, struct dma_buffparms *dmap, int fact)
{
if (fact == 0)
{
fact = dmap->subdivision;
if (fact == 0)
fact = 1;
return fact;
}
if (dmap->subdivision != 0 || dmap->fragment_size) /* Too late to change */
return -EINVAL;
if (fact > MAX_REALTIME_FACTOR)
return -EINVAL;
if (fact != 1 && fact != 2 && fact != 4 && fact != 8 && fact != 16)
return -EINVAL;
dmap->subdivision = fact;
return fact;
}
static int dma_set_fragment(int dev, struct dma_buffparms *dmap, int fact)
{
int bytes, count;
if (fact == 0)
return -EIO;
if (dmap->subdivision != 0 ||
dmap->fragment_size) /* Too late to change */
return -EINVAL;
bytes = fact & 0xffff;
count = (fact >> 16) & 0x7fff;
if (count == 0)
count = MAX_SUB_BUFFERS;
else if (count < MAX_SUB_BUFFERS)
count++;
if (bytes < 4 || bytes > 17) /* <16 || > 512k */
return -EINVAL;
if (count < 2)
return -EINVAL;
if (audio_devs[dev]->min_fragment > 0)
if (bytes < audio_devs[dev]->min_fragment)
bytes = audio_devs[dev]->min_fragment;
if (audio_devs[dev]->max_fragment > 0)
if (bytes > audio_devs[dev]->max_fragment)
bytes = audio_devs[dev]->max_fragment;
#ifdef OS_DMA_MINBITS
if (bytes < OS_DMA_MINBITS)
bytes = OS_DMA_MINBITS;
#endif
dmap->fragment_size = (1 << bytes);
dmap->max_fragments = count;
if (dmap->fragment_size > dmap->buffsize)
dmap->fragment_size = dmap->buffsize;
if (dmap->fragment_size == dmap->buffsize &&
audio_devs[dev]->flags & DMA_AUTOMODE)
dmap->fragment_size /= 2; /* Needs at least 2 buffers */
dmap->subdivision = 1; /* Disable SNDCTL_DSP_SUBDIVIDE */
return bytes | ((count - 1) << 16);
}
static int dma_ioctl(int dev, unsigned int cmd, void __user *arg)
{
struct dma_buffparms *dmap_out = audio_devs[dev]->dmap_out;
struct dma_buffparms *dmap_in = audio_devs[dev]->dmap_in;
struct dma_buffparms *dmap;
audio_buf_info info;
count_info cinfo;
int fact, ret, changed, bits, count, err;
unsigned long flags;
switch (cmd)
{
case SNDCTL_DSP_SUBDIVIDE:
ret = 0;
if (get_user(fact, (int __user *)arg))
return -EFAULT;
if (audio_devs[dev]->open_mode & OPEN_WRITE)
ret = dma_subdivide(dev, dmap_out, fact);
if (ret < 0)
return ret;
if (audio_devs[dev]->open_mode != OPEN_WRITE ||
(audio_devs[dev]->flags & DMA_DUPLEX &&
audio_devs[dev]->open_mode & OPEN_READ))
ret = dma_subdivide(dev, dmap_in, fact);
if (ret < 0)
return ret;
break;
case SNDCTL_DSP_GETISPACE:
case SNDCTL_DSP_GETOSPACE:
dmap = dmap_out;
if (cmd == SNDCTL_DSP_GETISPACE && !(audio_devs[dev]->open_mode & OPEN_READ))
return -EINVAL;
if (cmd == SNDCTL_DSP_GETOSPACE && !(audio_devs[dev]->open_mode & OPEN_WRITE))
return -EINVAL;
if (cmd == SNDCTL_DSP_GETISPACE && audio_devs[dev]->flags & DMA_DUPLEX)
dmap = dmap_in;
if (dmap->mapping_flags & DMA_MAP_MAPPED)
return -EINVAL;
if (!(dmap->flags & DMA_ALLOC_DONE))
reorganize_buffers(dev, dmap, (cmd == SNDCTL_DSP_GETISPACE));
info.fragstotal = dmap->nbufs;
if (cmd == SNDCTL_DSP_GETISPACE)
info.fragments = dmap->qlen;
else
{
if (!DMAbuf_space_in_queue(dev))
info.fragments = 0;
else
{
info.fragments = DMAbuf_space_in_queue(dev);
if (audio_devs[dev]->d->local_qlen)
{
int tmp = audio_devs[dev]->d->local_qlen(dev);
if (tmp && info.fragments)
tmp--; /*
* This buffer has been counted twice
*/
info.fragments -= tmp;
}
}
}
if (info.fragments < 0)
info.fragments = 0;
else if (info.fragments > dmap->nbufs)
info.fragments = dmap->nbufs;
info.fragsize = dmap->fragment_size;
info.bytes = info.fragments * dmap->fragment_size;
if (cmd == SNDCTL_DSP_GETISPACE && dmap->qlen)
info.bytes -= dmap->counts[dmap->qhead];
else
{
info.fragments = info.bytes / dmap->fragment_size;
info.bytes -= dmap->user_counter % dmap->fragment_size;
}
if (copy_to_user(arg, &info, sizeof(info)))
return -EFAULT;
return 0;
case SNDCTL_DSP_SETTRIGGER:
if (get_user(bits, (int __user *)arg))
return -EFAULT;
bits &= audio_devs[dev]->open_mode;
if (audio_devs[dev]->d->trigger == NULL)
return -EINVAL;
if (!(audio_devs[dev]->flags & DMA_DUPLEX) && (bits & PCM_ENABLE_INPUT) &&
(bits & PCM_ENABLE_OUTPUT))
return -EINVAL;
if (bits & PCM_ENABLE_INPUT)
{
spin_lock_irqsave(&dmap_in->lock,flags);
changed = (audio_devs[dev]->enable_bits ^ bits) & PCM_ENABLE_INPUT;
if (changed && audio_devs[dev]->go)
{
reorganize_buffers(dev, dmap_in, 1);
if ((err = audio_devs[dev]->d->prepare_for_input(dev,
dmap_in->fragment_size, dmap_in->nbufs)) < 0) {
spin_unlock_irqrestore(&dmap_in->lock,flags);
return err;
}
dmap_in->dma_mode = DMODE_INPUT;
audio_devs[dev]->enable_bits |= PCM_ENABLE_INPUT;
DMAbuf_activate_recording(dev, dmap_in);
} else
audio_devs[dev]->enable_bits &= ~PCM_ENABLE_INPUT;
spin_unlock_irqrestore(&dmap_in->lock,flags);
}
if (bits & PCM_ENABLE_OUTPUT)
{
spin_lock_irqsave(&dmap_out->lock,flags);
changed = (audio_devs[dev]->enable_bits ^ bits) & PCM_ENABLE_OUTPUT;
if (changed &&
(dmap_out->mapping_flags & DMA_MAP_MAPPED || dmap_out->qlen > 0) &&
audio_devs[dev]->go)
{
if (!(dmap_out->flags & DMA_ALLOC_DONE))
reorganize_buffers(dev, dmap_out, 0);
dmap_out->dma_mode = DMODE_OUTPUT;
audio_devs[dev]->enable_bits |= PCM_ENABLE_OUTPUT;
dmap_out->counts[dmap_out->qhead] = dmap_out->fragment_size;
DMAbuf_launch_output(dev, dmap_out);
} else
audio_devs[dev]->enable_bits &= ~PCM_ENABLE_OUTPUT;
spin_unlock_irqrestore(&dmap_out->lock,flags);
}
#if 0
if (changed && audio_devs[dev]->d->trigger)
audio_devs[dev]->d->trigger(dev, bits * audio_devs[dev]->go);
#endif
/* Falls through... */
case SNDCTL_DSP_GETTRIGGER:
ret = audio_devs[dev]->enable_bits;
break;
case SNDCTL_DSP_SETSYNCRO:
if (!audio_devs[dev]->d->trigger)
return -EINVAL;
audio_devs[dev]->d->trigger(dev, 0);
audio_devs[dev]->go = 0;
return 0;
case SNDCTL_DSP_GETIPTR:
if (!(audio_devs[dev]->open_mode & OPEN_READ))
return -EINVAL;
spin_lock_irqsave(&dmap_in->lock,flags);
cinfo.bytes = dmap_in->byte_counter;
cinfo.ptr = DMAbuf_get_buffer_pointer(dev, dmap_in, DMODE_INPUT) & ~3;
if (cinfo.ptr < dmap_in->fragment_size && dmap_in->qtail != 0)
cinfo.bytes += dmap_in->bytes_in_use; /* Pointer wrap not handled yet */
cinfo.blocks = dmap_in->qlen;
cinfo.bytes += cinfo.ptr;
if (dmap_in->mapping_flags & DMA_MAP_MAPPED)
dmap_in->qlen = 0; /* Reset interrupt counter */
spin_unlock_irqrestore(&dmap_in->lock,flags);
if (copy_to_user(arg, &cinfo, sizeof(cinfo)))
return -EFAULT;
return 0;
case SNDCTL_DSP_GETOPTR:
if (!(audio_devs[dev]->open_mode & OPEN_WRITE))
return -EINVAL;
spin_lock_irqsave(&dmap_out->lock,flags);
cinfo.bytes = dmap_out->byte_counter;
cinfo.ptr = DMAbuf_get_buffer_pointer(dev, dmap_out, DMODE_OUTPUT) & ~3;
if (cinfo.ptr < dmap_out->fragment_size && dmap_out->qhead != 0)
cinfo.bytes += dmap_out->bytes_in_use; /* Pointer wrap not handled yet */
cinfo.blocks = dmap_out->qlen;
cinfo.bytes += cinfo.ptr;
if (dmap_out->mapping_flags & DMA_MAP_MAPPED)
dmap_out->qlen = 0; /* Reset interrupt counter */
spin_unlock_irqrestore(&dmap_out->lock,flags);
if (copy_to_user(arg, &cinfo, sizeof(cinfo)))
return -EFAULT;
return 0;
case SNDCTL_DSP_GETODELAY:
if (!(audio_devs[dev]->open_mode & OPEN_WRITE))
return -EINVAL;
if (!(dmap_out->flags & DMA_ALLOC_DONE))
{
ret=0;
break;
}
spin_lock_irqsave(&dmap_out->lock,flags);
/* Compute number of bytes that have been played */
count = DMAbuf_get_buffer_pointer (dev, dmap_out, DMODE_OUTPUT);
if (count < dmap_out->fragment_size && dmap_out->qhead != 0)
count += dmap_out->bytes_in_use; /* Pointer wrap not handled yet */
count += dmap_out->byte_counter;
/* Subtract current count from the number of bytes written by app */
count = dmap_out->user_counter - count;
if (count < 0)
count = 0;
spin_unlock_irqrestore(&dmap_out->lock,flags);
ret = count;
break;
case SNDCTL_DSP_POST:
if (audio_devs[dev]->dmap_out->qlen > 0)
if (!(audio_devs[dev]->dmap_out->flags & DMA_ACTIVE))
DMAbuf_launch_output(dev, audio_devs[dev]->dmap_out);
return 0;
case SNDCTL_DSP_GETBLKSIZE:
dmap = dmap_out;
if (audio_devs[dev]->open_mode & OPEN_WRITE)
reorganize_buffers(dev, dmap_out, (audio_devs[dev]->open_mode == OPEN_READ));
if (audio_devs[dev]->open_mode == OPEN_READ ||
(audio_devs[dev]->flags & DMA_DUPLEX &&
audio_devs[dev]->open_mode & OPEN_READ))
reorganize_buffers(dev, dmap_in, (audio_devs[dev]->open_mode == OPEN_READ));
if (audio_devs[dev]->open_mode == OPEN_READ)
dmap = dmap_in;
ret = dmap->fragment_size;
break;
case SNDCTL_DSP_SETFRAGMENT:
ret = 0;
if (get_user(fact, (int __user *)arg))
return -EFAULT;
if (audio_devs[dev]->open_mode & OPEN_WRITE)
ret = dma_set_fragment(dev, dmap_out, fact);
if (ret < 0)
return ret;
if (audio_devs[dev]->open_mode == OPEN_READ ||
(audio_devs[dev]->flags & DMA_DUPLEX &&
audio_devs[dev]->open_mode & OPEN_READ))
ret = dma_set_fragment(dev, dmap_in, fact);
if (ret < 0)
return ret;
if (!arg) /* don't know what this is good for, but preserve old semantics */
return 0;
break;
default:
if (!audio_devs[dev]->d->ioctl)
return -EINVAL;
return audio_devs[dev]->d->ioctl(dev, cmd, arg);
}
return put_user(ret, (int __user *)arg);
}

39
sound/oss/bin2hex.c
View File

@ -1,39 +0,0 @@
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
int main( int argc, const char * argv [] )
{
const char * varname;
int i = 0;
int c;
int id = 0;
if(argv[1] && strcmp(argv[1],"-i")==0)
{
argv++;
argc--;
id=1;
}
if(argc==1)
{
fprintf(stderr, "bin2hex: [-i] firmware\n");
exit(1);
}
varname = argv[1];
printf( "/* automatically generated by bin2hex */\n" );
printf( "static unsigned char %s [] %s =\n{\n", varname , id?"__initdata":"");
while ( ( c = getchar( ) ) != EOF )
{
if ( i != 0 && i % 10 == 0 )
printf( "\n" );
printf( "0x%02lx,", c & 0xFFl );
i++;
}
printf( "};\nstatic int %sLen = %d;\n", varname, i );
return 0;
}

12
sound/oss/coproc.h
View File

@ -1,12 +0,0 @@
/*
* Definitions for various on board processors on the sound cards. For
* example DSP processors.
*/
/*
* Coprocessor access types
*/
#define COPR_CUSTOM 0x0001 /* Custom applications */
#define COPR_MIDI 0x0002 /* MIDI (MPU-401) emulation */
#define COPR_PCM 0x0004 /* Digitized voice applications */
#define COPR_SYNTH 0x0008 /* Music synthesis */

256
sound/oss/dev_table.c
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@ -1,256 +0,0 @@
/*
* sound/oss/dev_table.c
*
* Device call tables.
*
*
* Copyright (C) by Hannu Savolainen 1993-1997
*
* OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
* Version 2 (June 1991). See the "COPYING" file distributed with this software
* for more info.
*/
#include <linux/init.h>
#include "sound_config.h"
struct audio_operations *audio_devs[MAX_AUDIO_DEV];
EXPORT_SYMBOL(audio_devs);
int num_audiodevs;
EXPORT_SYMBOL(num_audiodevs);
struct mixer_operations *mixer_devs[MAX_MIXER_DEV];
EXPORT_SYMBOL(mixer_devs);
int num_mixers;
EXPORT_SYMBOL(num_mixers);
struct synth_operations *synth_devs[MAX_SYNTH_DEV+MAX_MIDI_DEV];
EXPORT_SYMBOL(synth_devs);
int num_synths;
struct midi_operations *midi_devs[MAX_MIDI_DEV];
EXPORT_SYMBOL(midi_devs);
int num_midis;
EXPORT_SYMBOL(num_midis);
struct sound_timer_operations *sound_timer_devs[MAX_TIMER_DEV] = {
&default_sound_timer, NULL
};
EXPORT_SYMBOL(sound_timer_devs);
int num_sound_timers = 1;
static int sound_alloc_audiodev(void);
int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver,
int driver_size, int flags, unsigned int format_mask,
void *devc, int dma1, int dma2)
{
struct audio_driver *d;
struct audio_operations *op;
int num;
if (vers != AUDIO_DRIVER_VERSION || driver_size > sizeof(struct audio_driver)) {
printk(KERN_ERR "Sound: Incompatible audio driver for %s\n", name);
return -EINVAL;
}
num = sound_alloc_audiodev();
if (num == -1) {
printk(KERN_ERR "sound: Too many audio drivers\n");
return -EBUSY;
}
d = (struct audio_driver *) (sound_mem_blocks[sound_nblocks] = vmalloc(sizeof(struct audio_driver)));
sound_nblocks++;
if (sound_nblocks >= MAX_MEM_BLOCKS)
sound_nblocks = MAX_MEM_BLOCKS - 1;
op = (struct audio_operations *) (sound_mem_blocks[sound_nblocks] = vzalloc(sizeof(struct audio_operations)));
sound_nblocks++;
if (sound_nblocks >= MAX_MEM_BLOCKS)
sound_nblocks = MAX_MEM_BLOCKS - 1;
if (d == NULL || op == NULL) {
printk(KERN_ERR "Sound: Can't allocate driver for (%s)\n", name);
sound_unload_audiodev(num);
return -ENOMEM;
}
init_waitqueue_head(&op->in_sleeper);
init_waitqueue_head(&op->out_sleeper);
init_waitqueue_head(&op->poll_sleeper);
if (driver_size < sizeof(struct audio_driver))
memset((char *) d, 0, sizeof(struct audio_driver));
memcpy((char *) d, (char *) driver, driver_size);
op->d = d;
strlcpy(op->name, name, sizeof(op->name));
op->flags = flags;
op->format_mask = format_mask;
op->devc = devc;
/*
* Hardcoded defaults
*/
audio_devs[num] = op;
DMAbuf_init(num, dma1, dma2);
audio_init_devices();
return num;
}
EXPORT_SYMBOL(sound_install_audiodrv);
int sound_install_mixer(int vers, char *name, struct mixer_operations *driver,
int driver_size, void *devc)
{
struct mixer_operations *op;
int n = sound_alloc_mixerdev();
if (n == -1) {
printk(KERN_ERR "Sound: Too many mixer drivers\n");
return -EBUSY;
}
if (vers != MIXER_DRIVER_VERSION ||
driver_size > sizeof(struct mixer_operations)) {
printk(KERN_ERR "Sound: Incompatible mixer driver for %s\n", name);
return -EINVAL;
}
/* FIXME: This leaks a mixer_operations struct every time its called
until you unload sound! */
op = (struct mixer_operations *) (sound_mem_blocks[sound_nblocks] = vzalloc(sizeof(struct mixer_operations)));
sound_nblocks++;
if (sound_nblocks >= MAX_MEM_BLOCKS)
sound_nblocks = MAX_MEM_BLOCKS - 1;
if (op == NULL) {
printk(KERN_ERR "Sound: Can't allocate mixer driver for (%s)\n", name);
return -ENOMEM;
}
memcpy((char *) op, (char *) driver, driver_size);
strlcpy(op->name, name, sizeof(op->name));
op->devc = devc;
mixer_devs[n] = op;
return n;
}
EXPORT_SYMBOL(sound_install_mixer);
void sound_unload_audiodev(int dev)
{
if (dev != -1) {
DMAbuf_deinit(dev);
audio_devs[dev] = NULL;
unregister_sound_dsp((dev<<4)+3);
}
}
EXPORT_SYMBOL(sound_unload_audiodev);
static int sound_alloc_audiodev(void)
{
int i = register_sound_dsp(&oss_sound_fops, -1);
if(i==-1)
return i;
i>>=4;
if(i>=num_audiodevs)
num_audiodevs = i + 1;
return i;
}
int sound_alloc_mididev(void)
{
int i = register_sound_midi(&oss_sound_fops, -1);
if(i==-1)
return i;
i>>=4;
if(i>=num_midis)
num_midis = i + 1;
return i;
}
EXPORT_SYMBOL(sound_alloc_mididev);
int sound_alloc_synthdev(void)
{
int i;
for (i = 0; i < MAX_SYNTH_DEV; i++) {
if (synth_devs[i] == NULL) {
if (i >= num_synths)
num_synths++;
return i;
}
}
return -1;
}
EXPORT_SYMBOL(sound_alloc_synthdev);
int sound_alloc_mixerdev(void)
{
int i = register_sound_mixer(&oss_sound_fops, -1);
if(i==-1)
return -1;
i>>=4;
if(i>=num_mixers)
num_mixers = i + 1;
return i;
}
EXPORT_SYMBOL(sound_alloc_mixerdev);
int sound_alloc_timerdev(void)
{
int i;
for (i = 0; i < MAX_TIMER_DEV; i++) {
if (sound_timer_devs[i] == NULL) {
if (i >= num_sound_timers)
num_sound_timers++;
return i;
}
}
return -1;
}
EXPORT_SYMBOL(sound_alloc_timerdev);
void sound_unload_mixerdev(int dev)
{
if (dev != -1) {
mixer_devs[dev] = NULL;
unregister_sound_mixer(dev<<4);
num_mixers--;
}
}
EXPORT_SYMBOL(sound_unload_mixerdev);
void sound_unload_mididev(int dev)
{
if (dev != -1) {
midi_devs[dev] = NULL;
unregister_sound_midi((dev<<4)+2);
}
}
EXPORT_SYMBOL(sound_unload_mididev);
void sound_unload_synthdev(int dev)
{
if (dev != -1)
synth_devs[dev] = NULL;
}
EXPORT_SYMBOL(sound_unload_synthdev);
void sound_unload_timerdev(int dev)
{
if (dev != -1)
sound_timer_devs[dev] = NULL;
}
EXPORT_SYMBOL(sound_unload_timerdev);

390
sound/oss/dev_table.h
View File

@ -1,390 +0,0 @@
/*
* dev_table.h
*
* Global definitions for device call tables
*
*
* Copyright (C) by Hannu Savolainen 1993-1997
*
* OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
* Version 2 (June 1991). See the "COPYING" file distributed with this software
* for more info.
*/
#ifndef _DEV_TABLE_H_
#define _DEV_TABLE_H_
#include <linux/spinlock.h>
/*
* Sound card numbers 27 to 999. (1 to 26 are defined in soundcard.h)
* Numbers 1000 to N are reserved for driver's internal use.
*/
#define SNDCARD_DESKPROXL 27 /* Compaq Deskpro XL */
#define SNDCARD_VIDC 28 /* ARMs VIDC */
#define SNDCARD_SBPNP 29
#define SNDCARD_SOFTOSS 36
#define SNDCARD_VMIDI 37
#define SNDCARD_OPL3SA1 38 /* Note: clash in msnd.h */
#define SNDCARD_OPL3SA1_SB 39
#define SNDCARD_OPL3SA1_MPU 40
#define SNDCARD_WAVEFRONT 41
#define SNDCARD_OPL3SA2 42
#define SNDCARD_OPL3SA2_MPU 43
#define SNDCARD_WAVEARTIST 44 /* Waveartist */
#define SNDCARD_OPL3SA2_MSS 45 /* Originally missed */
#define SNDCARD_AD1816 88
/*
* NOTE! NOTE! NOTE! NOTE!
*
* If you modify this file, please check the dev_table.c also.
*
* NOTE! NOTE! NOTE! NOTE!
*/
struct driver_info
{
char *driver_id;
int card_subtype; /* Driver specific. Usually 0 */
int card_type; /* From soundcard.h */
char *name;
void (*attach) (struct address_info *hw_config);
int (*probe) (struct address_info *hw_config);
void (*unload) (struct address_info *hw_config);
};
struct card_info
{
int card_type; /* Link (search key) to the driver list */
struct address_info config;
int enabled;
void *for_driver_use;
};
/*
* Device specific parameters (used only by dmabuf.c)
*/
#define MAX_SUB_BUFFERS (32*MAX_REALTIME_FACTOR)
#define DMODE_NONE 0
#define DMODE_OUTPUT PCM_ENABLE_OUTPUT
#define DMODE_INPUT PCM_ENABLE_INPUT
struct dma_buffparms
{
int dma_mode; /* DMODE_INPUT, DMODE_OUTPUT or DMODE_NONE */
int closing;
/*
* Pointers to raw buffers
*/
char *raw_buf;
unsigned long raw_buf_phys;
int buffsize;
/*
* Device state tables
*/
unsigned long flags;
#define DMA_BUSY 0x00000001
#define DMA_RESTART 0x00000002
#define DMA_ACTIVE 0x00000004
#define DMA_STARTED 0x00000008
#define DMA_EMPTY 0x00000010
#define DMA_ALLOC_DONE 0x00000020
#define DMA_SYNCING 0x00000040
#define DMA_DIRTY 0x00000080
#define DMA_POST 0x00000100
#define DMA_NODMA 0x00000200
#define DMA_NOTIMEOUT 0x00000400
int open_mode;
/*
* Queue parameters.
*/
int qlen;
int qhead;
int qtail;
spinlock_t lock;
int cfrag; /* Current incomplete fragment (write) */
int nbufs;
int counts[MAX_SUB_BUFFERS];
int subdivision;
int fragment_size;
int needs_reorg;
int max_fragments;
int bytes_in_use;
int underrun_count;
unsigned long byte_counter;
unsigned long user_counter;
unsigned long max_byte_counter;
int data_rate; /* Bytes/second */
int mapping_flags;
#define DMA_MAP_MAPPED 0x00000001
char neutral_byte;
int dma; /* DMA channel */
int applic_profile; /* Application profile (APF_*) */
/* Interrupt callback stuff */
void (*audio_callback) (int dev, int parm);
int callback_parm;
int buf_flags[MAX_SUB_BUFFERS];
#define BUFF_EOF 0x00000001 /* Increment eof count */
#define BUFF_DIRTY 0x00000002 /* Buffer written */
};
/*
* Structure for use with various microcontrollers and DSP processors
* in the recent sound cards.
*/
typedef struct coproc_operations
{
char name[64];
struct module *owner;
int (*open) (void *devc, int sub_device);
void (*close) (void *devc, int sub_device);
int (*ioctl) (void *devc, unsigned int cmd, void __user * arg, int local);
void (*reset) (void *devc);
void *devc; /* Driver specific info */
} coproc_operations;
struct audio_driver
{
struct module *owner;
int (*open) (int dev, int mode);
void (*close) (int dev);
void (*output_block) (int dev, unsigned long buf,
int count, int intrflag);
void (*start_input) (int dev, unsigned long buf,
int count, int intrflag);
int (*ioctl) (int dev, unsigned int cmd, void __user * arg);
int (*prepare_for_input) (int dev, int bufsize, int nbufs);
int (*prepare_for_output) (int dev, int bufsize, int nbufs);
void (*halt_io) (int dev);
int (*local_qlen)(int dev);
void (*copy_user) (int dev,
char *localbuf, int localoffs,
const char __user *userbuf, int useroffs,
int max_in, int max_out,
int *used, int *returned,
int len);
void (*halt_input) (int dev);
void (*halt_output) (int dev);
void (*trigger) (int dev, int bits);
int (*set_speed)(int dev, int speed);
unsigned int (*set_bits)(int dev, unsigned int bits);
short (*set_channels)(int dev, short channels);
void (*postprocess_write)(int dev); /* Device spesific postprocessing for written data */
void (*preprocess_read)(int dev); /* Device spesific preprocessing for read data */
void (*mmap)(int dev);
};
struct audio_operations
{
char name[128];
int flags;
#define NOTHING_SPECIAL 0x00
#define NEEDS_RESTART 0x01
#define DMA_AUTOMODE 0x02
#define DMA_DUPLEX 0x04
#define DMA_PSEUDO_AUTOMODE 0x08
#define DMA_HARDSTOP 0x10
#define DMA_EXACT 0x40
#define DMA_NORESET 0x80
int format_mask; /* Bitmask for supported audio formats */
void *devc; /* Driver specific info */
struct audio_driver *d;
void *portc; /* Driver specific info */
struct dma_buffparms *dmap_in, *dmap_out;
struct coproc_operations *coproc;
int mixer_dev;
int enable_bits;
int open_mode;
int go;
int min_fragment; /* 0 == unlimited */
int max_fragment; /* 0 == unlimited */
int parent_dev; /* 0 -> no parent, 1 to n -> parent=parent_dev+1 */
/* fields formerly in dmabuf.c */
wait_queue_head_t in_sleeper;
wait_queue_head_t out_sleeper;
wait_queue_head_t poll_sleeper;
/* fields formerly in audio.c */
int audio_mode;
#define AM_NONE 0
#define AM_WRITE OPEN_WRITE
#define AM_READ OPEN_READ
int local_format;
int audio_format;
int local_conversion;
#define CNV_MU_LAW 0x00000001
/* large structures at the end to keep offsets small */
struct dma_buffparms dmaps[2];
};
int *load_mixer_volumes(char *name, int *levels, int present);
struct mixer_operations
{
struct module *owner;
char id[16];
char name[64];
int (*ioctl) (int dev, unsigned int cmd, void __user * arg);
void *devc;
int modify_counter;
};
struct synth_operations
{
struct module *owner;
char *id; /* Unique identifier (ASCII) max 29 char */
struct synth_info *info;
int midi_dev;
int synth_type;
int synth_subtype;
int (*open) (int dev, int mode);
void (*close) (int dev);
int (*ioctl) (int dev, unsigned int cmd, void __user * arg);
int (*kill_note) (int dev, int voice, int note, int velocity);
int (*start_note) (int dev, int voice, int note, int velocity);
int (*set_instr) (int dev, int voice, int instr);
void (*reset) (int dev);
void (*hw_control) (int dev, unsigned char *event);
int (*load_patch) (int dev, int format, const char __user *addr,
int count, int pmgr_flag);
void (*aftertouch) (int dev, int voice, int pressure);
void (*controller) (int dev, int voice, int ctrl_num, int value);
void (*panning) (int dev, int voice, int value);
void (*volume_method) (int dev, int mode);
void (*bender) (int dev, int chn, int value);
int (*alloc_voice) (int dev, int chn, int note, struct voice_alloc_info *alloc);
void (*setup_voice) (int dev, int voice, int chn);
int (*send_sysex)(int dev, unsigned char *bytes, int len);
struct voice_alloc_info alloc;
struct channel_info chn_info[16];
int emulation;
#define EMU_GM 1 /* General MIDI */
#define EMU_XG 2 /* Yamaha XG */
#define MAX_SYSEX_BUF 64
unsigned char sysex_buf[MAX_SYSEX_BUF];
int sysex_ptr;
};
struct midi_input_info
{
/* MIDI input scanner variables */
#define MI_MAX 10
volatile int m_busy;
unsigned char m_buf[MI_MAX];
unsigned char m_prev_status; /* For running status */
int m_ptr;
#define MST_INIT 0
#define MST_DATA 1
#define MST_SYSEX 2
int m_state;
int m_left;
};
struct midi_operations
{
struct module *owner;
struct midi_info info;
struct synth_operations *converter;
struct midi_input_info in_info;
int (*open) (int dev, int mode,
void (*inputintr)(int dev, unsigned char data),
void (*outputintr)(int dev)
);
void (*close) (int dev);
int (*ioctl) (int dev, unsigned int cmd, void __user * arg);
int (*outputc) (int dev, unsigned char data);
int (*start_read) (int dev);
int (*end_read) (int dev);
void (*kick)(int dev);
int (*command) (int dev, unsigned char *data);
int (*buffer_status) (int dev);
int (*prefix_cmd) (int dev, unsigned char status);
struct coproc_operations *coproc;
void *devc;
};
struct sound_lowlev_timer
{
int dev;
int priority;
unsigned int (*tmr_start)(int dev, unsigned int usecs);
void (*tmr_disable)(int dev);
void (*tmr_restart)(int dev);
};
struct sound_timer_operations
{
struct module *owner;
struct sound_timer_info info;
int priority;
int devlink;
int (*open)(int dev, int mode);
void (*close)(int dev);
int (*event)(int dev, unsigned char *ev);
unsigned long (*get_time)(int dev);
int (*ioctl) (int dev, unsigned int cmd, void __user * arg);
void (*arm_timer)(int dev, long time);
};
extern struct sound_timer_operations default_sound_timer;
extern struct audio_operations *audio_devs[MAX_AUDIO_DEV];
extern int num_audiodevs;
extern struct mixer_operations *mixer_devs[MAX_MIXER_DEV];
extern int num_mixers;
extern struct synth_operations *synth_devs[MAX_SYNTH_DEV+MAX_MIDI_DEV];
extern int num_synths;
extern struct midi_operations *midi_devs[MAX_MIDI_DEV];
extern int num_midis;
extern struct sound_timer_operations * sound_timer_devs[MAX_TIMER_DEV];
extern int num_sound_timers;
extern int sound_map_buffer (int dev, struct dma_buffparms *dmap, buffmem_desc *info);
void sound_timer_init (struct sound_lowlev_timer *t, char *name);
void sound_dma_intr (int dev, struct dma_buffparms *dmap, int chan);
#define AUDIO_DRIVER_VERSION 2
#define MIXER_DRIVER_VERSION 2
int sound_install_audiodrv(int vers, char *name, struct audio_driver *driver,
int driver_size, int flags, unsigned int format_mask,
void *devc, int dma1, int dma2);
int sound_install_mixer(int vers, char *name, struct mixer_operations *driver,
int driver_size, void *devc);
void sound_unload_audiodev(int dev);
void sound_unload_mixerdev(int dev);
void sound_unload_mididev(int dev);
void sound_unload_synthdev(int dev);
void sound_unload_timerdev(int dev);
int sound_alloc_mixerdev(void);
int sound_alloc_timerdev(void);
int sound_alloc_synthdev(void);
int sound_alloc_mididev(void);
#endif /* _DEV_TABLE_H_ */

1268
sound/oss/dmabuf.c
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File diff suppressed because it is too large Load Diff

101
sound/oss/hex2hex.c
View File

@ -1,101 +0,0 @@
/*
* hex2hex reads stdin in Intel HEX format and produces an
* (unsigned char) array which contains the bytes and writes it
* to stdout using C syntax
*/
#include <stdio.h>
#include <string.h>
#include <stdlib.h>
#define ABANDON(why) { fprintf(stderr, "%s\n", why); exit(1); }
#define MAX_SIZE (256*1024)
unsigned char buf[MAX_SIZE];
static int loadhex(FILE *inf, unsigned char *buf)
{
int l=0, c, i;
while ((c=getc(inf))!=EOF)
{
if (c == ':') /* Sync with beginning of line */
{
int n, check;
unsigned char sum;
int addr;
int linetype;
if (fscanf(inf, "%02x", &n) != 1)
ABANDON("File format error");
sum = n;
if (fscanf(inf, "%04x", &addr) != 1)
ABANDON("File format error");
sum += addr/256;
sum += addr%256;
if (fscanf(inf, "%02x", &linetype) != 1)
ABANDON("File format error");
sum += linetype;
if (linetype != 0)
continue;
for (i=0;i<n;i++)
{
if (fscanf(inf, "%02x", &c) != 1)
ABANDON("File format error");
if (addr >= MAX_SIZE)
ABANDON("File too large");
buf[addr++] = c;
if (addr > l)
l = addr;
sum += c;
}
if (fscanf(inf, "%02x", &check) != 1)
ABANDON("File format error");
sum = ~sum + 1;
if (check != sum)
ABANDON("Line checksum error");
}
}
return l;
}
int main( int argc, const char * argv [] )
{
const char * varline;
int i,l;
int id=0;
if(argv[1] && strcmp(argv[1], "-i")==0)
{
argv++;
argc--;
id=1;
}
if(argv[1]==NULL)
{
fprintf(stderr,"hex2hex: [-i] filename\n");
exit(1);
}
varline = argv[1];
l = loadhex(stdin, buf);
printf("/*\n *\t Computer generated file. Do not edit.\n */\n");
printf("static int %s_len = %d;\n", varline, l);
printf("static unsigned char %s[] %s = {\n", varline, id?"__initdata":"");
for (i=0;i<l;i++)
{
if (i) printf(",");
if (i && !(i % 16)) printf("\n");
printf("0x%02x", buf[i]);
}
printf("\n};\n\n");
return 0;
}

229
sound/oss/kahlua.c
View File

@ -1,229 +0,0 @@
/*
* Initialisation code for Cyrix/NatSemi VSA1 softaudio
*
* (C) Copyright 2003 Red Hat Inc <alan@lxorguk.ukuu.org.uk>
*
* XpressAudio(tm) is used on the Cyrix MediaGX (now NatSemi Geode) systems.
* The older version (VSA1) provides fairly good soundblaster emulation
* although there are a couple of bugs: large DMA buffers break record,
* and the MPU event handling seems suspect. VSA2 allows the native driver
* to control the AC97 audio engine directly and requires a different driver.
*
* Thanks to National Semiconductor for providing the needed information
* on the XpressAudio(tm) internals.
*
* 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.
*
* TO DO:
* Investigate whether we can portably support Cognac (5520) in the
* same manner.
*/
#include <linux/delay.h>
#include <linux/init.h>
#include <linux/module.h>
#include <linux/pci.h>
#include <linux/slab.h>
#include "sound_config.h"
#include "sb.h"
/*
* Read a soundblaster compatible mixer register.
* In this case we are actually reading an SMI trap
* not real hardware.
*/
static u8 mixer_read(unsigned long io, u8 reg)
{
outb(reg, io + 4);
udelay(20);
reg = inb(io + 5);
udelay(20);
return reg;
}
static int probe_one(struct pci_dev *pdev, const struct pci_device_id *ent)
{
struct address_info *hw_config;
unsigned long base;
void __iomem *mem;
unsigned long io;
u16 map;
u8 irq, dma8, dma16;
int oldquiet;
extern int sb_be_quiet;
base = pci_resource_start(pdev, 0);
if(base == 0UL)
return 1;
mem = ioremap(base, 128);
if (!mem)
return 1;
map = readw(mem + 0x18); /* Read the SMI enables */
iounmap(mem);
/* Map bits
0:1 * 0x20 + 0x200 = sb base
2 sb enable
3 adlib enable
5 MPU enable 0x330
6 MPU enable 0x300
The other bits may be used internally so must be masked */
io = 0x220 + 0x20 * (map & 3);
if(map & (1<<2))
printk(KERN_INFO "kahlua: XpressAudio at 0x%lx\n", io);
else
return 1;
if(map & (1<<5))
printk(KERN_INFO "kahlua: MPU at 0x300\n");
else if(map & (1<<6))
printk(KERN_INFO "kahlua: MPU at 0x330\n");
irq = mixer_read(io, 0x80) & 0x0F;
dma8 = mixer_read(io, 0x81);
// printk("IRQ=%x MAP=%x DMA=%x\n", irq, map, dma8);
if(dma8 & 0x20)
dma16 = 5;
else if(dma8 & 0x40)
dma16 = 6;
else if(dma8 & 0x80)
dma16 = 7;
else
{
printk(KERN_ERR "kahlua: No 16bit DMA enabled.\n");
return 1;
}
if(dma8 & 0x01)
dma8 = 0;
else if(dma8 & 0x02)
dma8 = 1;
else if(dma8 & 0x08)
dma8 = 3;
else
{
printk(KERN_ERR "kahlua: No 8bit DMA enabled.\n");
return 1;
}
if(irq & 1)
irq = 9;
else if(irq & 2)
irq = 5;
else if(irq & 4)
irq = 7;
else if(irq & 8)
irq = 10;
else
{
printk(KERN_ERR "kahlua: SB IRQ not set.\n");
return 1;
}
printk(KERN_INFO "kahlua: XpressAudio on IRQ %d, DMA %d, %d\n",
irq, dma8, dma16);
hw_config = kzalloc(sizeof(struct address_info), GFP_KERNEL);
if(hw_config == NULL)
{
printk(KERN_ERR "kahlua: out of memory.\n");
return 1;
}
pci_set_drvdata(pdev, hw_config);
hw_config->io_base = io;
hw_config->irq = irq;
hw_config->dma = dma8;
hw_config->dma2 = dma16;
hw_config->name = "Cyrix XpressAudio";
hw_config->driver_use_1 = SB_NO_MIDI | SB_PCI_IRQ;
if (!request_region(io, 16, "soundblaster"))
goto err_out_free;
if(sb_dsp_detect(hw_config, 0, 0, NULL)==0)
{
printk(KERN_ERR "kahlua: audio not responding.\n");
release_region(io, 16);
goto err_out_free;
}
oldquiet = sb_be_quiet;
sb_be_quiet = 1;
if(sb_dsp_init(hw_config, THIS_MODULE))
{
sb_be_quiet = oldquiet;
goto err_out_free;
}
sb_be_quiet = oldquiet;
return 0;
err_out_free:
kfree(hw_config);
return 1;
}
static void remove_one(struct pci_dev *pdev)
{
struct address_info *hw_config = pci_get_drvdata(pdev);
sb_dsp_unload(hw_config, 0);
kfree(hw_config);
}
MODULE_AUTHOR("Alan Cox");
MODULE_DESCRIPTION("Kahlua VSA1 PCI Audio");
MODULE_LICENSE("GPL");
/*
* 5530 only. The 5510/5520 decode is different.
*/
static const struct pci_device_id id_tbl[] = {
{ PCI_VDEVICE(CYRIX, PCI_DEVICE_ID_CYRIX_5530_AUDIO), 0 },
{ }
};
MODULE_DEVICE_TABLE(pci, id_tbl);
static struct pci_driver kahlua_driver = {
.name = "kahlua",
.id_table = id_tbl,
.probe = probe_one,
.remove = remove_one,
};
static int __init kahlua_init_module(void)
{
printk(KERN_INFO "Cyrix Kahlua VSA1 XpressAudio support (c) Copyright 2003 Red Hat Inc\n");
return pci_register_driver(&kahlua_driver);
}
static void kahlua_cleanup_module(void)
{
pci_unregister_driver(&kahlua_driver);
}
module_init(kahlua_init_module);
module_exit(kahlua_cleanup_module);

22
sound/oss/midi_ctrl.h
View File

@ -1,22 +0,0 @@
static unsigned char ctrl_def_values[128] =
{
0x40,0x00,0x40,0x40, 0x40,0x40,0x40,0x7f, /* 0 to 7 */
0x40,0x40,0x40,0x7f, 0x40,0x40,0x40,0x40, /* 8 to 15 */
0x40,0x40,0x40,0x40, 0x40,0x40,0x40,0x40, /* 16 to 23 */
0x40,0x40,0x40,0x40, 0x40,0x40,0x40,0x40, /* 24 to 31 */
0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 32 to 39 */
0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 40 to 47 */
0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 48 to 55 */
0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 56 to 63 */
0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 64 to 71 */
0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 72 to 79 */
0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 80 to 87 */
0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 88 to 95 */
0x00,0x00,0x7f,0x7f, 0x7f,0x7f,0x00,0x00, /* 96 to 103 */
0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 104 to 111 */
0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 112 to 119 */
0x00,0x00,0x00,0x00, 0x00,0x00,0x00,0x00, /* 120 to 127 */
};

712
sound/oss/midi_synth.c
View File

@ -1,712 +0,0 @@
/*
* sound/oss/midi_synth.c
*
* High level midi sequencer manager for dumb MIDI interfaces.
*/
/*
* Copyright (C) by Hannu Savolainen 1993-1997
*
* OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
* Version 2 (June 1991). See the "COPYING" file distributed with this software
* for more info.
*/
/*
* Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
* Andrew Veliath : fixed running status in MIDI input state machine
*/
#define USE_SEQ_MACROS
#define USE_SIMPLE_MACROS
#include "sound_config.h"
#define _MIDI_SYNTH_C_
#include "midi_synth.h"
static int midi2synth[MAX_MIDI_DEV];
static int sysex_state[MAX_MIDI_DEV] =
{0};
static unsigned char prev_out_status[MAX_MIDI_DEV];
#define STORE(cmd) \
{ \
int len; \
unsigned char obuf[8]; \
cmd; \
seq_input_event(obuf, len); \
}
#define _seqbuf obuf
#define _seqbufptr 0
#define _SEQ_ADVBUF(x) len=x
void
do_midi_msg(int synthno, unsigned char *msg, int mlen)
{
switch (msg[0] & 0xf0)
{
case 0x90:
if (msg[2] != 0)
{
STORE(SEQ_START_NOTE(synthno, msg[0] & 0x0f, msg[1], msg[2]));
break;
}
msg[2] = 64;
case 0x80:
STORE(SEQ_STOP_NOTE(synthno, msg[0] & 0x0f, msg[1], msg[2]));
break;
case 0xA0:
STORE(SEQ_KEY_PRESSURE(synthno, msg[0] & 0x0f, msg[1], msg[2]));
break;
case 0xB0:
STORE(SEQ_CONTROL(synthno, msg[0] & 0x0f,
msg[1], msg[2]));
break;
case 0xC0:
STORE(SEQ_SET_PATCH(synthno, msg[0] & 0x0f, msg[1]));
break;
case 0xD0:
STORE(SEQ_CHN_PRESSURE(synthno, msg[0] & 0x0f, msg[1]));
break;
case 0xE0:
STORE(SEQ_BENDER(synthno, msg[0] & 0x0f,
(msg[1] & 0x7f) | ((msg[2] & 0x7f) << 7)));
break;
default:
/* printk( "MPU: Unknown midi channel message %02x\n", msg[0]); */
;
}
}
EXPORT_SYMBOL(do_midi_msg);
static void
midi_outc(int midi_dev, int data)
{
int timeout;
for (timeout = 0; timeout < 3200; timeout++)
if (midi_devs[midi_dev]->outputc(midi_dev, (unsigned char) (data & 0xff)))
{
if (data & 0x80) /*
* Status byte
*/
prev_out_status[midi_dev] =
(unsigned char) (data & 0xff); /*
* Store for running status
*/
return; /*
* Mission complete
*/
}
/*
* Sorry! No space on buffers.
*/
printk("Midi send timed out\n");
}
static int
prefix_cmd(int midi_dev, unsigned char status)
{
if ((char *) midi_devs[midi_dev]->prefix_cmd == NULL)
return 1;
return midi_devs[midi_dev]->prefix_cmd(midi_dev, status);
}
static void
midi_synth_input(int orig_dev, unsigned char data)
{
int dev;
struct midi_input_info *inc;
static unsigned char len_tab[] = /* # of data bytes following a status
*/
{
2, /* 8x */
2, /* 9x */
2, /* Ax */
2, /* Bx */
1, /* Cx */
1, /* Dx */
2, /* Ex */
0 /* Fx */
};
if (orig_dev < 0 || orig_dev > num_midis || midi_devs[orig_dev] == NULL)
return;
if (data == 0xfe) /* Ignore active sensing */
return;
dev = midi2synth[orig_dev];
inc = &midi_devs[orig_dev]->in_info;
switch (inc->m_state)
{
case MST_INIT:
if (data & 0x80) /* MIDI status byte */
{
if ((data & 0xf0) == 0xf0) /* Common message */
{
switch (data)
{
case 0xf0: /* Sysex */
inc->m_state = MST_SYSEX;
break; /* Sysex */
case 0xf1: /* MTC quarter frame */
case 0xf3: /* Song select */
inc->m_state = MST_DATA;
inc->m_ptr = 1;
inc->m_left = 1;
inc->m_buf[0] = data;
break;
case 0xf2: /* Song position pointer */
inc->m_state = MST_DATA;
inc->m_ptr = 1;
inc->m_left = 2;
inc->m_buf[0] = data;
break;
default:
inc->m_buf[0] = data;
inc->m_ptr = 1;
do_midi_msg(dev, inc->m_buf, inc->m_ptr);
inc->m_ptr = 0;
inc->m_left = 0;
}
} else
{
inc->m_state = MST_DATA;
inc->m_ptr = 1;
inc->m_left = len_tab[(data >> 4) - 8];
inc->m_buf[0] = inc->m_prev_status = data;
}
} else if (inc->m_prev_status & 0x80) {
/* Data byte (use running status) */
inc->m_ptr = 2;
inc->m_buf[1] = data;
inc->m_buf[0] = inc->m_prev_status;
inc->m_left = len_tab[(inc->m_buf[0] >> 4) - 8] - 1;
if (inc->m_left > 0)
inc->m_state = MST_DATA; /* Not done yet */
else {
inc->m_state = MST_INIT;
do_midi_msg(dev, inc->m_buf, inc->m_ptr);
inc->m_ptr = 0;
}
}
break; /* MST_INIT */
case MST_DATA:
inc->m_buf[inc->m_ptr++] = data;
if (--inc->m_left <= 0)
{
inc->m_state = MST_INIT;
do_midi_msg(dev, inc->m_buf, inc->m_ptr);
inc->m_ptr = 0;
}
break; /* MST_DATA */
case MST_SYSEX:
if (data == 0xf7) /* Sysex end */
{
inc->m_state = MST_INIT;
inc->m_left = 0;
inc->m_ptr = 0;
}
break; /* MST_SYSEX */
default:
printk("MIDI%d: Unexpected state %d (%02x)\n", orig_dev, inc->m_state, (int) data);
inc->m_state = MST_INIT;
}
}
static void
leave_sysex(int dev)
{
int orig_dev = synth_devs[dev]->midi_dev;
int timeout = 0;
if (!sysex_state[dev])
return;
sysex_state[dev] = 0;
while (!midi_devs[orig_dev]->outputc(orig_dev, 0xf7) &&
timeout < 1000)
timeout++;
sysex_state[dev] = 0;
}
static void
midi_synth_output(int dev)
{
/*
* Currently NOP
*/
}
int midi_synth_ioctl(int dev, unsigned int cmd, void __user *arg)
{
/*
* int orig_dev = synth_devs[dev]->midi_dev;
*/
switch (cmd) {
case SNDCTL_SYNTH_INFO:
if (__copy_to_user(arg, synth_devs[dev]->info, sizeof(struct synth_info)))
return -EFAULT;
return 0;
case SNDCTL_SYNTH_MEMAVL:
return 0x7fffffff;
default:
return -EINVAL;
}
}
EXPORT_SYMBOL(midi_synth_ioctl);
int
midi_synth_kill_note(int dev, int channel, int note, int velocity)
{
int orig_dev = synth_devs[dev]->midi_dev;
int msg, chn;
if (note < 0 || note > 127)
return 0;
if (channel < 0 || channel > 15)
return 0;
if (velocity < 0)
velocity = 0;
if (velocity > 127)
velocity = 127;
leave_sysex(dev);
msg = prev_out_status[orig_dev] & 0xf0;
chn = prev_out_status[orig_dev] & 0x0f;
if (chn == channel && ((msg == 0x90 && velocity == 64) || msg == 0x80))
{ /*
* Use running status
*/
if (!prefix_cmd(orig_dev, note))
return 0;
midi_outc(orig_dev, note);
if (msg == 0x90) /*
* Running status = Note on
*/
midi_outc(orig_dev, 0); /*
* Note on with velocity 0 == note
* off
*/
else
midi_outc(orig_dev, velocity);
} else
{
if (velocity == 64)
{
if (!prefix_cmd(orig_dev, 0x90 | (channel & 0x0f)))
return 0;
midi_outc(orig_dev, 0x90 | (channel & 0x0f)); /*
* Note on
*/
midi_outc(orig_dev, note);
midi_outc(orig_dev, 0); /*
* Zero G
*/
} else
{
if (!prefix_cmd(orig_dev, 0x80 | (channel & 0x0f)))
return 0;
midi_outc(orig_dev, 0x80 | (channel & 0x0f)); /*
* Note off
*/
midi_outc(orig_dev, note);
midi_outc(orig_dev, velocity);
}
}
return 0;
}
EXPORT_SYMBOL(midi_synth_kill_note);
int
midi_synth_set_instr(int dev, int channel, int instr_no)
{
int orig_dev = synth_devs[dev]->midi_dev;
if (instr_no < 0 || instr_no > 127)
instr_no = 0;
if (channel < 0 || channel > 15)
return 0;
leave_sysex(dev);
if (!prefix_cmd(orig_dev, 0xc0 | (channel & 0x0f)))
return 0;
midi_outc(orig_dev, 0xc0 | (channel & 0x0f)); /*
* Program change
*/
midi_outc(orig_dev, instr_no);
return 0;
}
EXPORT_SYMBOL(midi_synth_set_instr);
int
midi_synth_start_note(int dev, int channel, int note, int velocity)
{
int orig_dev = synth_devs[dev]->midi_dev;
int msg, chn;
if (note < 0 || note > 127)
return 0;
if (channel < 0 || channel > 15)
return 0;
if (velocity < 0)
velocity = 0;
if (velocity > 127)
velocity = 127;
leave_sysex(dev);
msg = prev_out_status[orig_dev] & 0xf0;
chn = prev_out_status[orig_dev] & 0x0f;
if (chn == channel && msg == 0x90)
{ /*
* Use running status
*/
if (!prefix_cmd(orig_dev, note))
return 0;
midi_outc(orig_dev, note);
midi_outc(orig_dev, velocity);
} else
{
if (!prefix_cmd(orig_dev, 0x90 | (channel & 0x0f)))
return 0;
midi_outc(orig_dev, 0x90 | (channel & 0x0f)); /*
* Note on
*/
midi_outc(orig_dev, note);
midi_outc(orig_dev, velocity);
}
return 0;
}
EXPORT_SYMBOL(midi_synth_start_note);
void
midi_synth_reset(int dev)
{
leave_sysex(dev);
}
EXPORT_SYMBOL(midi_synth_reset);
int
midi_synth_open(int dev, int mode)
{
int orig_dev = synth_devs[dev]->midi_dev;
int err;
struct midi_input_info *inc;
if (orig_dev < 0 || orig_dev >= num_midis || midi_devs[orig_dev] == NULL)
return -ENXIO;
midi2synth[orig_dev] = dev;
sysex_state[dev] = 0;
prev_out_status[orig_dev] = 0;
if ((err = midi_devs[orig_dev]->open(orig_dev, mode,
midi_synth_input, midi_synth_output)) < 0)
return err;
inc = &midi_devs[orig_dev]->in_info;
/* save_flags(flags);
cli();
don't know against what irqhandler to protect*/
inc->m_busy = 0;
inc->m_state = MST_INIT;
inc->m_ptr = 0;
inc->m_left = 0;
inc->m_prev_status = 0x00;
/* restore_flags(flags); */
return 1;
}
EXPORT_SYMBOL(midi_synth_open);
void
midi_synth_close(int dev)
{
int orig_dev = synth_devs[dev]->midi_dev;
leave_sysex(dev);
/*
* Shut up the synths by sending just single active sensing message.
*/
midi_devs[orig_dev]->outputc(orig_dev, 0xfe);
midi_devs[orig_dev]->close(orig_dev);
}
EXPORT_SYMBOL(midi_synth_close);
void
midi_synth_hw_control(int dev, unsigned char *event)
{
}
EXPORT_SYMBOL(midi_synth_hw_control);
int
midi_synth_load_patch(int dev, int format, const char __user *addr,
int count, int pmgr_flag)
{
int orig_dev = synth_devs[dev]->midi_dev;
struct sysex_info sysex;
int i;
unsigned long left, src_offs, eox_seen = 0;
int first_byte = 1;
int hdr_size = (unsigned long) &sysex.data[0] - (unsigned long) &sysex;
leave_sysex(dev);
if (!prefix_cmd(orig_dev, 0xf0))
return 0;
/* Invalid patch format */
if (format != SYSEX_PATCH)
return -EINVAL;
/* Patch header too short */
if (count < hdr_size)
return -EINVAL;
count -= hdr_size;
/*
* Copy the header from user space
*/
if (copy_from_user(&sysex, addr, hdr_size))
return -EFAULT;
/* Sysex record too short */
if ((unsigned)count < (unsigned)sysex.len)
sysex.len = count;
left = sysex.len;
src_offs = 0;
for (i = 0; i < left && !signal_pending(current); i++)
{
unsigned char data;
if (get_user(data,
(unsigned char __user *)(addr + hdr_size + i)))
return -EFAULT;
eox_seen = (i > 0 && data & 0x80); /* End of sysex */
if (eox_seen && data != 0xf7)
data = 0xf7;
if (i == 0)
{
if (data != 0xf0)
{
printk(KERN_WARNING "midi_synth: Sysex start missing\n");
return -EINVAL;
}
}
while (!midi_devs[orig_dev]->outputc(orig_dev, (unsigned char) (data & 0xff)) &&
!signal_pending(current))
schedule();
if (!first_byte && data & 0x80)
return 0;
first_byte = 0;
}
if (!eox_seen)
midi_outc(orig_dev, 0xf7);
return 0;
}
EXPORT_SYMBOL(midi_synth_load_patch);
void midi_synth_panning(int dev, int channel, int pressure)
{
}
EXPORT_SYMBOL(midi_synth_panning);
void midi_synth_aftertouch(int dev, int channel, int pressure)
{
int orig_dev = synth_devs[dev]->midi_dev;
int msg, chn;
if (pressure < 0 || pressure > 127)
return;
if (channel < 0 || channel > 15)
return;
leave_sysex(dev);
msg = prev_out_status[orig_dev] & 0xf0;
chn = prev_out_status[orig_dev] & 0x0f;
if (msg != 0xd0 || chn != channel) /*
* Test for running status
*/
{
if (!prefix_cmd(orig_dev, 0xd0 | (channel & 0x0f)))
return;
midi_outc(orig_dev, 0xd0 | (channel & 0x0f)); /*
* Channel pressure
*/
} else if (!prefix_cmd(orig_dev, pressure))
return;
midi_outc(orig_dev, pressure);
}
EXPORT_SYMBOL(midi_synth_aftertouch);
void
midi_synth_controller(int dev, int channel, int ctrl_num, int value)
{
int orig_dev = synth_devs[dev]->midi_dev;
int chn, msg;
if (ctrl_num < 0 || ctrl_num > 127)
return;
if (channel < 0 || channel > 15)
return;
leave_sysex(dev);
msg = prev_out_status[orig_dev] & 0xf0;
chn = prev_out_status[orig_dev] & 0x0f;
if (msg != 0xb0 || chn != channel)
{
if (!prefix_cmd(orig_dev, 0xb0 | (channel & 0x0f)))
return;
midi_outc(orig_dev, 0xb0 | (channel & 0x0f));
} else if (!prefix_cmd(orig_dev, ctrl_num))
return;
midi_outc(orig_dev, ctrl_num);
midi_outc(orig_dev, value & 0x7f);
}
EXPORT_SYMBOL(midi_synth_controller);
void
midi_synth_bender(int dev, int channel, int value)
{
int orig_dev = synth_devs[dev]->midi_dev;
int msg, prev_chn;
if (channel < 0 || channel > 15)
return;
if (value < 0 || value > 16383)
return;
leave_sysex(dev);
msg = prev_out_status[orig_dev] & 0xf0;
prev_chn = prev_out_status[orig_dev] & 0x0f;
if (msg != 0xd0 || prev_chn != channel) /*
* Test for running status
*/
{
if (!prefix_cmd(orig_dev, 0xe0 | (channel & 0x0f)))
return;
midi_outc(orig_dev, 0xe0 | (channel & 0x0f));
} else if (!prefix_cmd(orig_dev, value & 0x7f))
return;
midi_outc(orig_dev, value & 0x7f);
midi_outc(orig_dev, (value >> 7) & 0x7f);
}
EXPORT_SYMBOL(midi_synth_bender);
void
midi_synth_setup_voice(int dev, int voice, int channel)
{
}
EXPORT_SYMBOL(midi_synth_setup_voice);
int
midi_synth_send_sysex(int dev, unsigned char *bytes, int len)
{
int orig_dev = synth_devs[dev]->midi_dev;
int i;
for (i = 0; i < len; i++)
{
switch (bytes[i])
{
case 0xf0: /* Start sysex */
if (!prefix_cmd(orig_dev, 0xf0))
return 0;
sysex_state[dev] = 1;
break;
case 0xf7: /* End sysex */
if (!sysex_state[dev]) /* Orphan sysex end */
return 0;
sysex_state[dev] = 0;
break;
default:
if (!sysex_state[dev])
return 0;
if (bytes[i] & 0x80) /* Error. Another message before sysex end */
{
bytes[i] = 0xf7; /* Sysex end */
sysex_state[dev] = 0;
}
}
if (!midi_devs[orig_dev]->outputc(orig_dev, bytes[i]))
{
/*
* Hardware level buffer is full. Abort the sysex message.
*/
int timeout = 0;
bytes[i] = 0xf7;
sysex_state[dev] = 0;
while (!midi_devs[orig_dev]->outputc(orig_dev, bytes[i]) &&
timeout < 1000)
timeout++;
}
if (!sysex_state[dev])
return 0;
}
return 0;
}
EXPORT_SYMBOL(midi_synth_send_sysex);

47
sound/oss/midi_synth.h
View File

@ -1,47 +0,0 @@
int midi_synth_ioctl (int dev,
unsigned int cmd, void __user * arg);
int midi_synth_kill_note (int dev, int channel, int note, int velocity);
int midi_synth_set_instr (int dev, int channel, int instr_no);
int midi_synth_start_note (int dev, int channel, int note, int volume);
void midi_synth_reset (int dev);
int midi_synth_open (int dev, int mode);
void midi_synth_close (int dev);
void midi_synth_hw_control (int dev, unsigned char *event);
int midi_synth_load_patch (int dev, int format, const char __user * addr,
int count, int pmgr_flag);
void midi_synth_panning (int dev, int channel, int pressure);
void midi_synth_aftertouch (int dev, int channel, int pressure);
void midi_synth_controller (int dev, int channel, int ctrl_num, int value);
void midi_synth_bender (int dev, int chn, int value);
void midi_synth_setup_voice (int dev, int voice, int chn);
int midi_synth_send_sysex(int dev, unsigned char *bytes,int len);
#ifndef _MIDI_SYNTH_C_
static struct synth_info std_synth_info =
{MIDI_SYNTH_NAME, 0, SYNTH_TYPE_MIDI, 0, 0, 128, 0, 128, MIDI_SYNTH_CAPS};
static struct synth_operations std_midi_synth =
{
.owner = THIS_MODULE,
.id = "MIDI",
.info = &std_synth_info,
.midi_dev = 0,
.synth_type = SYNTH_TYPE_MIDI,
.synth_subtype = 0,
.open = midi_synth_open,
.close = midi_synth_close,
.ioctl = midi_synth_ioctl,
.kill_note = midi_synth_kill_note,
.start_note = midi_synth_start_note,
.set_instr = midi_synth_set_instr,
.reset = midi_synth_reset,
.hw_control = midi_synth_hw_control,
.load_patch = midi_synth_load_patch,
.aftertouch = midi_synth_aftertouch,
.controller = midi_synth_controller,
.panning = midi_synth_panning,
.bender = midi_synth_bender,
.setup_voice = midi_synth_setup_voice,
.send_sysex = midi_synth_send_sysex
};
#endif

427
sound/oss/midibuf.c
View File

@ -1,427 +0,0 @@
/*
* sound/oss/midibuf.c
*
* Device file manager for /dev/midi#
*/
/*
* Copyright (C) by Hannu Savolainen 1993-1997
*
* OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
* Version 2 (June 1991). See the "COPYING" file distributed with this software
* for more info.
*/
/*
* Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
*/
#include <linux/stddef.h>
#include <linux/kmod.h>
#include <linux/spinlock.h>
#include <linux/sched/signal.h>
#define MIDIBUF_C
#include "sound_config.h"
/*
* Don't make MAX_QUEUE_SIZE larger than 4000
*/
#define MAX_QUEUE_SIZE 4000
static wait_queue_head_t midi_sleeper[MAX_MIDI_DEV];
static wait_queue_head_t input_sleeper[MAX_MIDI_DEV];
struct midi_buf
{
int len, head, tail;
unsigned char queue[MAX_QUEUE_SIZE];
};
struct midi_parms
{
long prech_timeout; /*
* Timeout before the first ch
*/
};
static struct midi_buf *midi_out_buf[MAX_MIDI_DEV] = {NULL};
static struct midi_buf *midi_in_buf[MAX_MIDI_DEV] = {NULL};
static struct midi_parms parms[MAX_MIDI_DEV];
static void midi_poll(unsigned long dummy);
static DEFINE_TIMER(poll_timer, midi_poll, 0, 0);
static volatile int open_devs;
static DEFINE_SPINLOCK(lock);
#define DATA_AVAIL(q) (q->len)
#define SPACE_AVAIL(q) (MAX_QUEUE_SIZE - q->len)
#define QUEUE_BYTE(q, data) \
if (SPACE_AVAIL(q)) \
{ \
unsigned long flags; \
spin_lock_irqsave(&lock, flags); \
q->queue[q->tail] = (data); \
q->len++; q->tail = (q->tail+1) % MAX_QUEUE_SIZE; \
spin_unlock_irqrestore(&lock, flags); \
}
#define REMOVE_BYTE(q, data) \
if (DATA_AVAIL(q)) \
{ \
unsigned long flags; \
spin_lock_irqsave(&lock, flags); \
data = q->queue[q->head]; \
q->len--; q->head = (q->head+1) % MAX_QUEUE_SIZE; \
spin_unlock_irqrestore(&lock, flags); \
}
static void drain_midi_queue(int dev)
{
/*
* Give the Midi driver time to drain its output queues
*/
if (midi_devs[dev]->buffer_status != NULL)
wait_event_interruptible_timeout(midi_sleeper[dev],
!midi_devs[dev]->buffer_status(dev), HZ/10);
}
static void midi_input_intr(int dev, unsigned char data)
{
if (midi_in_buf[dev] == NULL)
return;
if (data == 0xfe) /*
* Active sensing
*/
return; /*
* Ignore
*/
if (SPACE_AVAIL(midi_in_buf[dev])) {
QUEUE_BYTE(midi_in_buf[dev], data);
wake_up(&input_sleeper[dev]);
}
}
static void midi_output_intr(int dev)
{
/*
* Currently NOP
*/
}
static void midi_poll(unsigned long dummy)
{
unsigned long flags;
int dev;
spin_lock_irqsave(&lock, flags);
if (open_devs)
{
for (dev = 0; dev < num_midis; dev++)
if (midi_devs[dev] != NULL && midi_out_buf[dev] != NULL)
{
while (DATA_AVAIL(midi_out_buf[dev]))
{
int ok;
int c = midi_out_buf[dev]->queue[midi_out_buf[dev]->head];
spin_unlock_irqrestore(&lock,flags);/* Give some time to others */
ok = midi_devs[dev]->outputc(dev, c);
spin_lock_irqsave(&lock, flags);
if (!ok)
break;
midi_out_buf[dev]->head = (midi_out_buf[dev]->head + 1) % MAX_QUEUE_SIZE;
midi_out_buf[dev]->len--;
}
if (DATA_AVAIL(midi_out_buf[dev]) < 100)
wake_up(&midi_sleeper[dev]);
}
poll_timer.expires = (1) + jiffies;
add_timer(&poll_timer);
/*
* Come back later
*/
}
spin_unlock_irqrestore(&lock, flags);
}
int MIDIbuf_open(int dev, struct file *file)
{
int mode, err;
dev = dev >> 4;
mode = translate_mode(file);
if (num_midis > MAX_MIDI_DEV)
{
printk(KERN_ERR "midi: Too many midi interfaces\n");
num_midis = MAX_MIDI_DEV;
}
if (dev < 0 || dev >= num_midis || midi_devs[dev] == NULL)
return -ENXIO;
/*
* Interrupts disabled. Be careful
*/
module_put(midi_devs[dev]->owner);
if ((err = midi_devs[dev]->open(dev, mode,
midi_input_intr, midi_output_intr)) < 0)
return err;
parms[dev].prech_timeout = MAX_SCHEDULE_TIMEOUT;
midi_in_buf[dev] = vmalloc(sizeof(struct midi_buf));
if (midi_in_buf[dev] == NULL)
{
printk(KERN_WARNING "midi: Can't allocate buffer\n");
midi_devs[dev]->close(dev);
return -EIO;
}
midi_in_buf[dev]->len = midi_in_buf[dev]->head = midi_in_buf[dev]->tail = 0;
midi_out_buf[dev] = vmalloc(sizeof(struct midi_buf));
if (midi_out_buf[dev] == NULL)
{
printk(KERN_WARNING "midi: Can't allocate buffer\n");
midi_devs[dev]->close(dev);
vfree(midi_in_buf[dev]);
midi_in_buf[dev] = NULL;
return -EIO;
}
midi_out_buf[dev]->len = midi_out_buf[dev]->head = midi_out_buf[dev]->tail = 0;
open_devs++;
init_waitqueue_head(&midi_sleeper[dev]);
init_waitqueue_head(&input_sleeper[dev]);
if (open_devs < 2) /* This was first open */
{
poll_timer.expires = 1 + jiffies;
add_timer(&poll_timer); /* Start polling */
}
return err;
}
void MIDIbuf_release(int dev, struct file *file)
{
int mode;
dev = dev >> 4;
mode = translate_mode(file);
if (dev < 0 || dev >= num_midis || midi_devs[dev] == NULL)
return;
/*
* Wait until the queue is empty
*/
if (mode != OPEN_READ)
{
midi_devs[dev]->outputc(dev, 0xfe); /*
* Active sensing to shut the
* devices
*/
wait_event_interruptible(midi_sleeper[dev],
!DATA_AVAIL(midi_out_buf[dev]));
/*
* Sync
*/
drain_midi_queue(dev); /*
* Ensure the output queues are empty
*/
}
midi_devs[dev]->close(dev);
open_devs--;
if (open_devs == 0)
del_timer_sync(&poll_timer);
vfree(midi_in_buf[dev]);
vfree(midi_out_buf[dev]);
midi_in_buf[dev] = NULL;
midi_out_buf[dev] = NULL;
module_put(midi_devs[dev]->owner);
}
int MIDIbuf_write(int dev, struct file *file, const char __user *buf, int count)
{
int c, n, i;
unsigned char tmp_data;
dev = dev >> 4;
if (!count)
return 0;
c = 0;
while (c < count)
{
n = SPACE_AVAIL(midi_out_buf[dev]);
if (n == 0) { /*
* No space just now.
*/
if (file->f_flags & O_NONBLOCK) {
c = -EAGAIN;
goto out;
}
if (wait_event_interruptible(midi_sleeper[dev],
SPACE_AVAIL(midi_out_buf[dev])))
{
c = -EINTR;
goto out;
}
n = SPACE_AVAIL(midi_out_buf[dev]);
}
if (n > (count - c))
n = count - c;
for (i = 0; i < n; i++)
{
/* BROKE BROKE BROKE - CAN'T DO THIS WITH CLI !! */
/* yes, think the same, so I removed the cli() brackets
QUEUE_BYTE is protected against interrupts */
if (copy_from_user((char *) &tmp_data, &(buf)[c], 1)) {
c = -EFAULT;
goto out;
}
QUEUE_BYTE(midi_out_buf[dev], tmp_data);
c++;
}
}
out:
return c;
}
int MIDIbuf_read(int dev, struct file *file, char __user *buf, int count)
{
int n, c = 0;
unsigned char tmp_data;
dev = dev >> 4;
if (!DATA_AVAIL(midi_in_buf[dev])) { /*
* No data yet, wait
*/
if (file->f_flags & O_NONBLOCK) {
c = -EAGAIN;
goto out;
}
wait_event_interruptible_timeout(input_sleeper[dev],
DATA_AVAIL(midi_in_buf[dev]),
parms[dev].prech_timeout);
if (signal_pending(current))
c = -EINTR; /* The user is getting restless */
}
if (c == 0 && DATA_AVAIL(midi_in_buf[dev])) /*
* Got some bytes
*/
{
n = DATA_AVAIL(midi_in_buf[dev]);
if (n > count)
n = count;
c = 0;
while (c < n)
{
char *fixit;
REMOVE_BYTE(midi_in_buf[dev], tmp_data);
fixit = (char *) &tmp_data;
/* BROKE BROKE BROKE */
/* yes removed the cli() brackets again
should q->len,tail&head be atomic_t? */
if (copy_to_user(&(buf)[c], fixit, 1)) {
c = -EFAULT;
goto out;
}
c++;
}
}
out:
return c;
}
int MIDIbuf_ioctl(int dev, struct file *file,
unsigned int cmd, void __user *arg)
{
int val;
dev = dev >> 4;
if (((cmd >> 8) & 0xff) == 'C')
{
if (midi_devs[dev]->coproc) /* Coprocessor ioctl */
return midi_devs[dev]->coproc->ioctl(midi_devs[dev]->coproc->devc, cmd, arg, 0);
/* printk("/dev/midi%d: No coprocessor for this device\n", dev);*/
return -ENXIO;
}
else
{
switch (cmd)
{
case SNDCTL_MIDI_PRETIME:
if (get_user(val, (int __user *)arg))
return -EFAULT;
if (val < 0)
val = 0;
val = (HZ * val) / 10;
parms[dev].prech_timeout = val;
return put_user(val, (int __user *)arg);
default:
if (!midi_devs[dev]->ioctl)
return -EINVAL;
return midi_devs[dev]->ioctl(dev, cmd, arg);
}
}
}
/* No kernel lock - fine */
unsigned int MIDIbuf_poll(int dev, struct file *file, poll_table * wait)
{
unsigned int mask = 0;
dev = dev >> 4;
/* input */
poll_wait(file, &input_sleeper[dev], wait);
if (DATA_AVAIL(midi_in_buf[dev]))
mask |= POLLIN | POLLRDNORM;
/* output */
poll_wait(file, &midi_sleeper[dev], wait);
if (!SPACE_AVAIL(midi_out_buf[dev]))
mask |= POLLOUT | POLLWRNORM;
return mask;
}
int MIDIbuf_avail(int dev)
{
if (midi_in_buf[dev])
return DATA_AVAIL (midi_in_buf[dev]);
return 0;
}
EXPORT_SYMBOL(MIDIbuf_avail);

1804
sound/oss/mpu401.c
View File

File diff suppressed because it is too large Load Diff

11
sound/oss/mpu401.h
View File

@ -1,11 +0,0 @@
/* From uart401.c */
int probe_uart401 (struct address_info *hw_config, struct module *owner);
void unload_uart401 (struct address_info *hw_config);
irqreturn_t uart401intr (int irq, void *dev_id);
/* From mpu401.c */
int probe_mpu401(struct address_info *hw_config, struct resource *ports);
int attach_mpu401(struct address_info * hw_config, struct module *owner);
void unload_mpu401(struct address_info *hw_info);

413
sound/oss/msnd.c
View File

@ -1,413 +0,0 @@
/*********************************************************************
*
* msnd.c - Driver Base
*
* Turtle Beach MultiSound Sound Card Driver for Linux
*
* Copyright (C) 1998 Andrew Veliath
*
* 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 of the License, 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.
*
********************************************************************/
#include <linux/module.h>
#include <linux/kernel.h>
#include <linux/vmalloc.h>
#include <linux/types.h>
#include <linux/delay.h>
#include <linux/mm.h>
#include <linux/init.h>
#include <linux/interrupt.h>
#include <asm/io.h>
#include <linux/uaccess.h>
#include <linux/spinlock.h>
#include <asm/irq.h>
#include "msnd.h"
#define LOGNAME "msnd"
#define MSND_MAX_DEVS 4
static multisound_dev_t *devs[MSND_MAX_DEVS];
static int num_devs;
int msnd_register(multisound_dev_t *dev)
{
int i;
for (i = 0; i < MSND_MAX_DEVS; ++i)
if (devs[i] == NULL)
break;
if (i == MSND_MAX_DEVS)
return -ENOMEM;
devs[i] = dev;
++num_devs;
return 0;
}
void msnd_unregister(multisound_dev_t *dev)
{
int i;
for (i = 0; i < MSND_MAX_DEVS; ++i)
if (devs[i] == dev)
break;
if (i == MSND_MAX_DEVS) {
printk(KERN_WARNING LOGNAME ": Unregistering unknown device\n");
return;
}
devs[i] = NULL;
--num_devs;
}
void msnd_init_queue(void __iomem *base, int start, int size)
{
writew(PCTODSP_BASED(start), base + JQS_wStart);
writew(PCTODSP_OFFSET(size) - 1, base + JQS_wSize);
writew(0, base + JQS_wHead);
writew(0, base + JQS_wTail);
}
void msnd_fifo_init(msnd_fifo *f)
{
f->data = NULL;
}
void msnd_fifo_free(msnd_fifo *f)
{
vfree(f->data);
f->data = NULL;
}
int msnd_fifo_alloc(msnd_fifo *f, size_t n)
{
msnd_fifo_free(f);
f->data = vmalloc(n);
f->n = n;
f->tail = 0;
f->head = 0;
f->len = 0;
if (!f->data)
return -ENOMEM;
return 0;
}
void msnd_fifo_make_empty(msnd_fifo *f)
{
f->len = f->tail = f->head = 0;
}
int msnd_fifo_write_io(msnd_fifo *f, char __iomem *buf, size_t len)
{
int count = 0;
while ((count < len) && (f->len != f->n)) {
int nwritten;
if (f->head <= f->tail) {
nwritten = len - count;
if (nwritten > f->n - f->tail)
nwritten = f->n - f->tail;
}
else {
nwritten = f->head - f->tail;
if (nwritten > len - count)
nwritten = len - count;
}
memcpy_fromio(f->data + f->tail, buf, nwritten);
count += nwritten;
buf += nwritten;
f->len += nwritten;
f->tail += nwritten;
f->tail %= f->n;
}
return count;
}
int msnd_fifo_write(msnd_fifo *f, const char *buf, size_t len)
{
int count = 0;
while ((count < len) && (f->len != f->n)) {
int nwritten;
if (f->head <= f->tail) {
nwritten = len - count;
if (nwritten > f->n - f->tail)
nwritten = f->n - f->tail;
}
else {
nwritten = f->head - f->tail;
if (nwritten > len - count)
nwritten = len - count;
}
memcpy(f->data + f->tail, buf, nwritten);
count += nwritten;
buf += nwritten;
f->len += nwritten;
f->tail += nwritten;
f->tail %= f->n;
}
return count;
}
int msnd_fifo_read_io(msnd_fifo *f, char __iomem *buf, size_t len)
{
int count = 0;
while ((count < len) && (f->len > 0)) {
int nread;
if (f->tail <= f->head) {
nread = len - count;
if (nread > f->n - f->head)
nread = f->n - f->head;
}
else {
nread = f->tail - f->head;
if (nread > len - count)
nread = len - count;
}
memcpy_toio(buf, f->data + f->head, nread);
count += nread;
buf += nread;
f->len -= nread;
f->head += nread;
f->head %= f->n;
}
return count;
}
int msnd_fifo_read(msnd_fifo *f, char *buf, size_t len)
{
int count = 0;
while ((count < len) && (f->len > 0)) {
int nread;
if (f->tail <= f->head) {
nread = len - count;
if (nread > f->n - f->head)
nread = f->n - f->head;
}
else {
nread = f->tail - f->head;
if (nread > len - count)
nread = len - count;
}
memcpy(buf, f->data + f->head, nread);
count += nread;
buf += nread;
f->len -= nread;
f->head += nread;
f->head %= f->n;
}
return count;
}
static int msnd_wait_TXDE(multisound_dev_t *dev)
{
register unsigned int io = dev->io;
register int timeout = 1000;
while(timeout-- > 0)
if (msnd_inb(io + HP_ISR) & HPISR_TXDE)
return 0;
return -EIO;
}
static int msnd_wait_HC0(multisound_dev_t *dev)
{
register unsigned int io = dev->io;
register int timeout = 1000;
while(timeout-- > 0)
if (!(msnd_inb(io + HP_CVR) & HPCVR_HC))
return 0;
return -EIO;
}
int msnd_send_dsp_cmd(multisound_dev_t *dev, BYTE cmd)
{
unsigned long flags;
spin_lock_irqsave(&dev->lock, flags);
if (msnd_wait_HC0(dev) == 0) {
msnd_outb(cmd, dev->io + HP_CVR);
spin_unlock_irqrestore(&dev->lock, flags);
return 0;
}
spin_unlock_irqrestore(&dev->lock, flags);
printk(KERN_DEBUG LOGNAME ": Send DSP command timeout\n");
return -EIO;
}
int msnd_send_word(multisound_dev_t *dev, unsigned char high,
unsigned char mid, unsigned char low)
{
register unsigned int io = dev->io;
if (msnd_wait_TXDE(dev) == 0) {
msnd_outb(high, io + HP_TXH);
msnd_outb(mid, io + HP_TXM);
msnd_outb(low, io + HP_TXL);
return 0;
}
printk(KERN_DEBUG LOGNAME ": Send host word timeout\n");
return -EIO;
}
int msnd_upload_host(multisound_dev_t *dev, char *bin, int len)
{
int i;
if (len % 3 != 0) {
printk(KERN_WARNING LOGNAME ": Upload host data not multiple of 3!\n");
return -EINVAL;
}
for (i = 0; i < len; i += 3)
if (msnd_send_word(dev, bin[i], bin[i + 1], bin[i + 2]) != 0)
return -EIO;
msnd_inb(dev->io + HP_RXL);
msnd_inb(dev->io + HP_CVR);
return 0;
}
int msnd_enable_irq(multisound_dev_t *dev)
{
unsigned long flags;
if (dev->irq_ref++)
return 0;
printk(KERN_DEBUG LOGNAME ": Enabling IRQ\n");
spin_lock_irqsave(&dev->lock, flags);
if (msnd_wait_TXDE(dev) == 0) {
msnd_outb(msnd_inb(dev->io + HP_ICR) | HPICR_TREQ, dev->io + HP_ICR);
if (dev->type == msndClassic)
msnd_outb(dev->irqid, dev->io + HP_IRQM);
msnd_outb(msnd_inb(dev->io + HP_ICR) & ~HPICR_TREQ, dev->io + HP_ICR);
msnd_outb(msnd_inb(dev->io + HP_ICR) | HPICR_RREQ, dev->io + HP_ICR);
enable_irq(dev->irq);
msnd_init_queue(dev->DSPQ, dev->dspq_data_buff, dev->dspq_buff_size);
spin_unlock_irqrestore(&dev->lock, flags);
return 0;
}
spin_unlock_irqrestore(&dev->lock, flags);
printk(KERN_DEBUG LOGNAME ": Enable IRQ failed\n");
return -EIO;
}
int msnd_disable_irq(multisound_dev_t *dev)
{
unsigned long flags;
if (--dev->irq_ref > 0)
return 0;
if (dev->irq_ref < 0)
printk(KERN_DEBUG LOGNAME ": IRQ ref count is %d\n", dev->irq_ref);
printk(KERN_DEBUG LOGNAME ": Disabling IRQ\n");
spin_lock_irqsave(&dev->lock, flags);
if (msnd_wait_TXDE(dev) == 0) {
msnd_outb(msnd_inb(dev->io + HP_ICR) & ~HPICR_RREQ, dev->io + HP_ICR);
if (dev->type == msndClassic)
msnd_outb(HPIRQ_NONE, dev->io + HP_IRQM);
disable_irq(dev->irq);
spin_unlock_irqrestore(&dev->lock, flags);
return 0;
}
spin_unlock_irqrestore(&dev->lock, flags);
printk(KERN_DEBUG LOGNAME ": Disable IRQ failed\n");
return -EIO;
}
#ifndef LINUX20
EXPORT_SYMBOL(msnd_register);
EXPORT_SYMBOL(msnd_unregister);
EXPORT_SYMBOL(msnd_init_queue);
EXPORT_SYMBOL(msnd_fifo_init);
EXPORT_SYMBOL(msnd_fifo_free);
EXPORT_SYMBOL(msnd_fifo_alloc);
EXPORT_SYMBOL(msnd_fifo_make_empty);
EXPORT_SYMBOL(msnd_fifo_write_io);
EXPORT_SYMBOL(msnd_fifo_read_io);
EXPORT_SYMBOL(msnd_fifo_write);
EXPORT_SYMBOL(msnd_fifo_read);
EXPORT_SYMBOL(msnd_send_dsp_cmd);
EXPORT_SYMBOL(msnd_send_word);
EXPORT_SYMBOL(msnd_upload_host);
EXPORT_SYMBOL(msnd_enable_irq);
EXPORT_SYMBOL(msnd_disable_irq);
#endif
#ifdef MODULE
MODULE_AUTHOR ("Andrew Veliath <andrewtv@usa.net>");
MODULE_DESCRIPTION ("Turtle Beach MultiSound Driver Base");
MODULE_LICENSE("GPL");
int init_module(void)
{
return 0;
}
void cleanup_module(void)
{
}
#endif

278
sound/oss/msnd.h
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@ -1,278 +0,0 @@
/*********************************************************************
*
* msnd.h
*
* Turtle Beach MultiSound Sound Card Driver for Linux
*
* Some parts of this header file were derived from the Turtle Beach
* MultiSound Driver Development Kit.
*
* Copyright (C) 1998 Andrew Veliath
* Copyright (C) 1993 Turtle Beach Systems, Inc.
*
* 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 of the License, 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.
*
********************************************************************/
#ifndef __MSND_H
#define __MSND_H
#define VERSION "0.8.3.1"
#define DEFSAMPLERATE DSP_DEFAULT_SPEED
#define DEFSAMPLESIZE AFMT_U8
#define DEFCHANNELS 1
#define DEFFIFOSIZE 128
#define SNDCARD_MSND 38
#define SRAM_BANK_SIZE 0x8000
#define SRAM_CNTL_START 0x7F00
#define DSP_BASE_ADDR 0x4000
#define DSP_BANK_BASE 0x4000
#define HP_ICR 0x00
#define HP_CVR 0x01
#define HP_ISR 0x02
#define HP_IVR 0x03
#define HP_NU 0x04
#define HP_INFO 0x04
#define HP_TXH 0x05
#define HP_RXH 0x05
#define HP_TXM 0x06
#define HP_RXM 0x06
#define HP_TXL 0x07
#define HP_RXL 0x07
#define HP_ICR_DEF 0x00
#define HP_CVR_DEF 0x12
#define HP_ISR_DEF 0x06
#define HP_IVR_DEF 0x0f
#define HP_NU_DEF 0x00
#define HP_IRQM 0x09
#define HPR_BLRC 0x08
#define HPR_SPR1 0x09
#define HPR_SPR2 0x0A
#define HPR_TCL0 0x0B
#define HPR_TCL1 0x0C
#define HPR_TCL2 0x0D
#define HPR_TCL3 0x0E
#define HPR_TCL4 0x0F
#define HPICR_INIT 0x80
#define HPICR_HM1 0x40
#define HPICR_HM0 0x20
#define HPICR_HF1 0x10
#define HPICR_HF0 0x08
#define HPICR_TREQ 0x02
#define HPICR_RREQ 0x01
#define HPCVR_HC 0x80
#define HPISR_HREQ 0x80
#define HPISR_DMA 0x40
#define HPISR_HF3 0x10
#define HPISR_HF2 0x08
#define HPISR_TRDY 0x04
#define HPISR_TXDE 0x02
#define HPISR_RXDF 0x01
#define HPIO_290 0
#define HPIO_260 1
#define HPIO_250 2
#define HPIO_240 3
#define HPIO_230 4
#define HPIO_220 5
#define HPIO_210 6
#define HPIO_3E0 7
#define HPMEM_NONE 0
#define HPMEM_B000 1
#define HPMEM_C800 2
#define HPMEM_D000 3
#define HPMEM_D400 4
#define HPMEM_D800 5
#define HPMEM_E000 6
#define HPMEM_E800 7
#define HPIRQ_NONE 0
#define HPIRQ_5 1
#define HPIRQ_7 2
#define HPIRQ_9 3
#define HPIRQ_10 4
#define HPIRQ_11 5
#define HPIRQ_12 6
#define HPIRQ_15 7
#define HIMT_PLAY_DONE 0x00
#define HIMT_RECORD_DONE 0x01
#define HIMT_MIDI_EOS 0x02
#define HIMT_MIDI_OUT 0x03
#define HIMT_MIDI_IN_UCHAR 0x0E
#define HIMT_DSP 0x0F
#define HDEX_BASE 0x92
#define HDEX_PLAY_START (0 + HDEX_BASE)
#define HDEX_PLAY_STOP (1 + HDEX_BASE)
#define HDEX_PLAY_PAUSE (2 + HDEX_BASE)
#define HDEX_PLAY_RESUME (3 + HDEX_BASE)
#define HDEX_RECORD_START (4 + HDEX_BASE)
#define HDEX_RECORD_STOP (5 + HDEX_BASE)
#define HDEX_MIDI_IN_START (6 + HDEX_BASE)
#define HDEX_MIDI_IN_STOP (7 + HDEX_BASE)
#define HDEX_MIDI_OUT_START (8 + HDEX_BASE)
#define HDEX_MIDI_OUT_STOP (9 + HDEX_BASE)
#define HDEX_AUX_REQ (10 + HDEX_BASE)
#define HIWORD(l) ((WORD)((((DWORD)(l)) >> 16) & 0xFFFF))
#define LOWORD(l) ((WORD)(DWORD)(l))
#define HIBYTE(w) ((BYTE)(((WORD)(w) >> 8) & 0xFF))
#define LOBYTE(w) ((BYTE)(w))
#define MAKELONG(low,hi) ((long)(((WORD)(low))|(((DWORD)((WORD)(hi)))<<16)))
#define MAKEWORD(low,hi) ((WORD)(((BYTE)(low))|(((WORD)((BYTE)(hi)))<<8)))
#define PCTODSP_OFFSET(w) (USHORT)((w)/2)
#define PCTODSP_BASED(w) (USHORT)(((w)/2) + DSP_BASE_ADDR)
#define DSPTOPC_BASED(w) (((w) - DSP_BASE_ADDR) * 2)
#ifdef SLOWIO
#define msnd_outb outb_p
#define msnd_inb inb_p
#else
#define msnd_outb outb
#define msnd_inb inb
#endif
/* JobQueueStruct */
#define JQS_wStart 0x00
#define JQS_wSize 0x02
#define JQS_wHead 0x04
#define JQS_wTail 0x06
#define JQS__size 0x08
/* DAQueueDataStruct */
#define DAQDS_wStart 0x00
#define DAQDS_wSize 0x02
#define DAQDS_wFormat 0x04
#define DAQDS_wSampleSize 0x06
#define DAQDS_wChannels 0x08
#define DAQDS_wSampleRate 0x0A
#define DAQDS_wIntMsg 0x0C
#define DAQDS_wFlags 0x0E
#define DAQDS__size 0x10
typedef u8 BYTE;
typedef u16 USHORT;
typedef u16 WORD;
typedef u32 DWORD;
typedef void __iomem * LPDAQD;
/* Generic FIFO */
typedef struct {
size_t n, len;
char *data;
int head, tail;
} msnd_fifo;
typedef struct multisound_dev {
/* Linux device info */
char *name;
int dsp_minor, mixer_minor;
int ext_midi_dev, hdr_midi_dev;
/* Hardware resources */
int io, numio;
int memid, irqid;
int irq, irq_ref;
unsigned char info;
void __iomem *base;
/* Motorola 56k DSP SMA */
void __iomem *SMA;
void __iomem *DAPQ, *DARQ, *MODQ, *MIDQ, *DSPQ;
void __iomem *pwDSPQData, *pwMIDQData, *pwMODQData;
int dspq_data_buff, dspq_buff_size;
/* State variables */
enum { msndClassic, msndPinnacle } type;
fmode_t mode;
unsigned long flags;
#define F_RESETTING 0
#define F_HAVEDIGITAL 1
#define F_AUDIO_WRITE_INUSE 2
#define F_WRITING 3
#define F_WRITEBLOCK 4
#define F_WRITEFLUSH 5
#define F_AUDIO_READ_INUSE 6
#define F_READING 7
#define F_READBLOCK 8
#define F_EXT_MIDI_INUSE 9
#define F_HDR_MIDI_INUSE 10
#define F_DISABLE_WRITE_NDELAY 11
wait_queue_head_t writeblock;
wait_queue_head_t readblock;
wait_queue_head_t writeflush;
spinlock_t lock;
int nresets;
unsigned long recsrc;
int left_levels[32];
int right_levels[32];
int mixer_mod_count;
int calibrate_signal;
int play_sample_size, play_sample_rate, play_channels;
int play_ndelay;
int rec_sample_size, rec_sample_rate, rec_channels;
int rec_ndelay;
BYTE bCurrentMidiPatch;
/* Digital audio FIFOs */
msnd_fifo DAPF, DARF;
int fifosize;
int last_playbank, last_recbank;
/* MIDI in callback */
void (*midi_in_interrupt)(struct multisound_dev *);
} multisound_dev_t;
#ifndef mdelay
# define mdelay(a) udelay((a) * 1000)
#endif
int msnd_register(multisound_dev_t *dev);
void msnd_unregister(multisound_dev_t *dev);
void msnd_init_queue(void __iomem *, int start, int size);
void msnd_fifo_init(msnd_fifo *f);
void msnd_fifo_free(msnd_fifo *f);
int msnd_fifo_alloc(msnd_fifo *f, size_t n);
void msnd_fifo_make_empty(msnd_fifo *f);
int msnd_fifo_write_io(msnd_fifo *f, char __iomem *buf, size_t len);
int msnd_fifo_read_io(msnd_fifo *f, char __iomem *buf, size_t len);
int msnd_fifo_write(msnd_fifo *f, const char *buf, size_t len);
int msnd_fifo_read(msnd_fifo *f, char *buf, size_t len);
int msnd_send_dsp_cmd(multisound_dev_t *dev, BYTE cmd);
int msnd_send_word(multisound_dev_t *dev, unsigned char high,
unsigned char mid, unsigned char low);
int msnd_upload_host(multisound_dev_t *dev, char *bin, int len);
int msnd_enable_irq(multisound_dev_t *dev);
int msnd_disable_irq(multisound_dev_t *dev);
#endif /* __MSND_H */

3
sound/oss/msnd_classic.c
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/* The work is in msnd_pinnacle.c, just define MSND_CLASSIC before it. */
#define MSND_CLASSIC
#include "msnd_pinnacle.c"

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sound/oss/msnd_classic.h
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/*********************************************************************
*
* msnd_classic.h
*
* Turtle Beach MultiSound Sound Card Driver for Linux
*
* Some parts of this header file were derived from the Turtle Beach
* MultiSound Driver Development Kit.
*
* Copyright (C) 1998 Andrew Veliath
* Copyright (C) 1993 Turtle Beach Systems, Inc.
*
* 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 of the License, 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.
*
********************************************************************/
#ifndef __MSND_CLASSIC_H
#define __MSND_CLASSIC_H
#define DSP_NUMIO 0x10
#define HP_MEMM 0x08
#define HP_BITM 0x0E
#define HP_WAIT 0x0D
#define HP_DSPR 0x0A
#define HP_PROR 0x0B
#define HP_BLKS 0x0C
#define HPPRORESET_OFF 0
#define HPPRORESET_ON 1
#define HPDSPRESET_OFF 0
#define HPDSPRESET_ON 1
#define HPBLKSEL_0 0
#define HPBLKSEL_1 1
#define HPWAITSTATE_0 0
#define HPWAITSTATE_1 1
#define HPBITMODE_16 0
#define HPBITMODE_8 1
#define HIDSP_INT_PLAY_UNDER 0x00
#define HIDSP_INT_RECORD_OVER 0x01
#define HIDSP_INPUT_CLIPPING 0x02
#define HIDSP_MIDI_IN_OVER 0x10
#define HIDSP_MIDI_OVERRUN_ERR 0x13
#define HDEXAR_CLEAR_PEAKS 1
#define HDEXAR_IN_SET_POTS 2
#define HDEXAR_AUX_SET_POTS 3
#define HDEXAR_CAL_A_TO_D 4
#define HDEXAR_RD_EXT_DSP_BITS 5
#define TIME_PRO_RESET_DONE 0x028A
#define TIME_PRO_SYSEX 0x0040
#define TIME_PRO_RESET 0x0032
#define AGND 0x01
#define SIGNAL 0x02
#define EXT_DSP_BIT_DCAL 0x0001
#define EXT_DSP_BIT_MIDI_CON 0x0002
#define BUFFSIZE 0x8000
#define HOSTQ_SIZE 0x40
#define SRAM_CNTL_START 0x7F00
#define SMA_STRUCT_START 0x7F40
#define DAP_BUFF_SIZE 0x2400
#define DAR_BUFF_SIZE 0x2000
#define DAPQ_STRUCT_SIZE 0x10
#define DARQ_STRUCT_SIZE 0x10
#define DAPQ_BUFF_SIZE (3 * 0x10)
#define DARQ_BUFF_SIZE (3 * 0x10)
#define MODQ_BUFF_SIZE 0x400
#define MIDQ_BUFF_SIZE 0x200
#define DSPQ_BUFF_SIZE 0x40
#define DAPQ_DATA_BUFF 0x6C00
#define DARQ_DATA_BUFF 0x6C30
#define MODQ_DATA_BUFF 0x6C60
#define MIDQ_DATA_BUFF 0x7060
#define DSPQ_DATA_BUFF 0x7260
#define DAPQ_OFFSET SRAM_CNTL_START
#define DARQ_OFFSET (SRAM_CNTL_START + 0x08)
#define MODQ_OFFSET (SRAM_CNTL_START + 0x10)
#define MIDQ_OFFSET (SRAM_CNTL_START + 0x18)
#define DSPQ_OFFSET (SRAM_CNTL_START + 0x20)
#define MOP_SYNTH 0x10
#define MOP_EXTOUT 0x32
#define MOP_EXTTHRU 0x02
#define MOP_OUTMASK 0x01
#define MIP_EXTIN 0x01
#define MIP_SYNTH 0x00
#define MIP_INMASK 0x32
/* Classic SMA Common Data */
#define SMA_wCurrPlayBytes 0x0000
#define SMA_wCurrRecordBytes 0x0002
#define SMA_wCurrPlayVolLeft 0x0004
#define SMA_wCurrPlayVolRight 0x0006
#define SMA_wCurrInVolLeft 0x0008
#define SMA_wCurrInVolRight 0x000a
#define SMA_wUser_3 0x000c
#define SMA_wUser_4 0x000e
#define SMA_dwUser_5 0x0010
#define SMA_dwUser_6 0x0014
#define SMA_wUser_7 0x0018
#define SMA_wReserved_A 0x001a
#define SMA_wReserved_B 0x001c
#define SMA_wReserved_C 0x001e
#define SMA_wReserved_D 0x0020
#define SMA_wReserved_E 0x0022
#define SMA_wReserved_F 0x0024
#define SMA_wReserved_G 0x0026
#define SMA_wReserved_H 0x0028
#define SMA_wCurrDSPStatusFlags 0x002a
#define SMA_wCurrHostStatusFlags 0x002c
#define SMA_wCurrInputTagBits 0x002e
#define SMA_wCurrLeftPeak 0x0030
#define SMA_wCurrRightPeak 0x0032
#define SMA_wExtDSPbits 0x0034
#define SMA_bExtHostbits 0x0036
#define SMA_bBoardLevel 0x0037
#define SMA_bInPotPosRight 0x0038
#define SMA_bInPotPosLeft 0x0039
#define SMA_bAuxPotPosRight 0x003a
#define SMA_bAuxPotPosLeft 0x003b
#define SMA_wCurrMastVolLeft 0x003c
#define SMA_wCurrMastVolRight 0x003e
#define SMA_bUser_12 0x0040
#define SMA_bUser_13 0x0041
#define SMA_wUser_14 0x0042
#define SMA_wUser_15 0x0044
#define SMA_wCalFreqAtoD 0x0046
#define SMA_wUser_16 0x0048
#define SMA_wUser_17 0x004a
#define SMA__size 0x004c
#ifdef HAVE_DSPCODEH
# include "msndperm.c"
# include "msndinit.c"
# define PERMCODE msndperm
# define INITCODE msndinit
# define PERMCODESIZE sizeof(msndperm)
# define INITCODESIZE sizeof(msndinit)
#else
# ifndef CONFIG_MSNDCLAS_INIT_FILE
# define CONFIG_MSNDCLAS_INIT_FILE \
"/etc/sound/msndinit.bin"
# endif
# ifndef CONFIG_MSNDCLAS_PERM_FILE
# define CONFIG_MSNDCLAS_PERM_FILE \
"/etc/sound/msndperm.bin"
# endif
# define PERMCODEFILE CONFIG_MSNDCLAS_PERM_FILE
# define INITCODEFILE CONFIG_MSNDCLAS_INIT_FILE
# define PERMCODE dspini
# define INITCODE permini
# define PERMCODESIZE sizeof_dspini
# define INITCODESIZE sizeof_permini
#endif
#define LONGNAME "MultiSound (Classic/Monterey/Tahiti)"
#endif /* __MSND_CLASSIC_H */

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sound/oss/msnd_pinnacle.c
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sound/oss/msnd_pinnacle.h
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/*********************************************************************
*
* msnd_pinnacle.h
*
* Turtle Beach MultiSound Sound Card Driver for Linux
*
* Some parts of this header file were derived from the Turtle Beach
* MultiSound Driver Development Kit.
*
* Copyright (C) 1998 Andrew Veliath
* Copyright (C) 1993 Turtle Beach Systems, Inc.
*
* 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 of the License, 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.
*
********************************************************************/
#ifndef __MSND_PINNACLE_H
#define __MSND_PINNACLE_H
#define DSP_NUMIO 0x08
#define IREG_LOGDEVICE 0x07
#define IREG_ACTIVATE 0x30
#define LD_ACTIVATE 0x01
#define LD_DISACTIVATE 0x00
#define IREG_EECONTROL 0x3F
#define IREG_MEMBASEHI 0x40
#define IREG_MEMBASELO 0x41
#define IREG_MEMCONTROL 0x42
#define IREG_MEMRANGEHI 0x43
#define IREG_MEMRANGELO 0x44
#define MEMTYPE_8BIT 0x00
#define MEMTYPE_16BIT 0x02
#define MEMTYPE_RANGE 0x00
#define MEMTYPE_HIADDR 0x01
#define IREG_IO0_BASEHI 0x60
#define IREG_IO0_BASELO 0x61
#define IREG_IO1_BASEHI 0x62
#define IREG_IO1_BASELO 0x63
#define IREG_IRQ_NUMBER 0x70
#define IREG_IRQ_TYPE 0x71
#define IRQTYPE_HIGH 0x02
#define IRQTYPE_LOW 0x00
#define IRQTYPE_LEVEL 0x01
#define IRQTYPE_EDGE 0x00
#define HP_DSPR 0x04
#define HP_BLKS 0x04
#define HPDSPRESET_OFF 2
#define HPDSPRESET_ON 0
#define HPBLKSEL_0 2
#define HPBLKSEL_1 3
#define HIMT_DAT_OFF 0x03
#define HIDSP_PLAY_UNDER 0x00
#define HIDSP_INT_PLAY_UNDER 0x01
#define HIDSP_SSI_TX_UNDER 0x02
#define HIDSP_RECQ_OVERFLOW 0x08
#define HIDSP_INT_RECORD_OVER 0x09
#define HIDSP_SSI_RX_OVERFLOW 0x0a
#define HIDSP_MIDI_IN_OVER 0x10
#define HIDSP_MIDI_FRAME_ERR 0x11
#define HIDSP_MIDI_PARITY_ERR 0x12
#define HIDSP_MIDI_OVERRUN_ERR 0x13
#define HIDSP_INPUT_CLIPPING 0x20
#define HIDSP_MIX_CLIPPING 0x30
#define HIDSP_DAT_IN_OFF 0x21
#define HDEXAR_SET_ANA_IN 0
#define HDEXAR_CLEAR_PEAKS 1
#define HDEXAR_IN_SET_POTS 2
#define HDEXAR_AUX_SET_POTS 3
#define HDEXAR_CAL_A_TO_D 4
#define HDEXAR_RD_EXT_DSP_BITS 5
#define HDEXAR_SET_SYNTH_IN 4
#define HDEXAR_READ_DAT_IN 5
#define HDEXAR_MIC_SET_POTS 6
#define HDEXAR_SET_DAT_IN 7
#define HDEXAR_SET_SYNTH_48 8
#define HDEXAR_SET_SYNTH_44 9
#define TIME_PRO_RESET_DONE 0x028A
#define TIME_PRO_SYSEX 0x001E
#define TIME_PRO_RESET 0x0032
#define AGND 0x01
#define SIGNAL 0x02
#define EXT_DSP_BIT_DCAL 0x0001
#define EXT_DSP_BIT_MIDI_CON 0x0002
#define BUFFSIZE 0x8000
#define HOSTQ_SIZE 0x40
#define SRAM_CNTL_START 0x7F00
#define SMA_STRUCT_START 0x7F40
#define DAP_BUFF_SIZE 0x2400
#define DAR_BUFF_SIZE 0x2000
#define DAPQ_STRUCT_SIZE 0x10
#define DARQ_STRUCT_SIZE 0x10
#define DAPQ_BUFF_SIZE (3 * 0x10)
#define DARQ_BUFF_SIZE (3 * 0x10)
#define MODQ_BUFF_SIZE 0x400
#define MIDQ_BUFF_SIZE 0x800
#define DSPQ_BUFF_SIZE 0x5A0
#define DAPQ_DATA_BUFF 0x6C00
#define DARQ_DATA_BUFF 0x6C30
#define MODQ_DATA_BUFF 0x6C60
#define MIDQ_DATA_BUFF 0x7060
#define DSPQ_DATA_BUFF 0x7860
#define DAPQ_OFFSET SRAM_CNTL_START
#define DARQ_OFFSET (SRAM_CNTL_START + 0x08)
#define MODQ_OFFSET (SRAM_CNTL_START + 0x10)
#define MIDQ_OFFSET (SRAM_CNTL_START + 0x18)
#define DSPQ_OFFSET (SRAM_CNTL_START + 0x20)
#define MOP_WAVEHDR 0
#define MOP_EXTOUT 1
#define MOP_HWINIT 0xfe
#define MOP_NONE 0xff
#define MOP_MAX 1
#define MIP_EXTIN 0
#define MIP_WAVEHDR 1
#define MIP_HWINIT 0xfe
#define MIP_MAX 1
/* Pinnacle/Fiji SMA Common Data */
#define SMA_wCurrPlayBytes 0x0000
#define SMA_wCurrRecordBytes 0x0002
#define SMA_wCurrPlayVolLeft 0x0004
#define SMA_wCurrPlayVolRight 0x0006
#define SMA_wCurrInVolLeft 0x0008
#define SMA_wCurrInVolRight 0x000a
#define SMA_wCurrMHdrVolLeft 0x000c
#define SMA_wCurrMHdrVolRight 0x000e
#define SMA_dwCurrPlayPitch 0x0010
#define SMA_dwCurrPlayRate 0x0014
#define SMA_wCurrMIDIIOPatch 0x0018
#define SMA_wCurrPlayFormat 0x001a
#define SMA_wCurrPlaySampleSize 0x001c
#define SMA_wCurrPlayChannels 0x001e
#define SMA_wCurrPlaySampleRate 0x0020
#define SMA_wCurrRecordFormat 0x0022
#define SMA_wCurrRecordSampleSize 0x0024
#define SMA_wCurrRecordChannels 0x0026
#define SMA_wCurrRecordSampleRate 0x0028
#define SMA_wCurrDSPStatusFlags 0x002a
#define SMA_wCurrHostStatusFlags 0x002c
#define SMA_wCurrInputTagBits 0x002e
#define SMA_wCurrLeftPeak 0x0030
#define SMA_wCurrRightPeak 0x0032
#define SMA_bMicPotPosLeft 0x0034
#define SMA_bMicPotPosRight 0x0035
#define SMA_bMicPotMaxLeft 0x0036
#define SMA_bMicPotMaxRight 0x0037
#define SMA_bInPotPosLeft 0x0038
#define SMA_bInPotPosRight 0x0039
#define SMA_bAuxPotPosLeft 0x003a
#define SMA_bAuxPotPosRight 0x003b
#define SMA_bInPotMaxLeft 0x003c
#define SMA_bInPotMaxRight 0x003d
#define SMA_bAuxPotMaxLeft 0x003e
#define SMA_bAuxPotMaxRight 0x003f
#define SMA_bInPotMaxMethod 0x0040
#define SMA_bAuxPotMaxMethod 0x0041
#define SMA_wCurrMastVolLeft 0x0042
#define SMA_wCurrMastVolRight 0x0044
#define SMA_wCalFreqAtoD 0x0046
#define SMA_wCurrAuxVolLeft 0x0048
#define SMA_wCurrAuxVolRight 0x004a
#define SMA_wCurrPlay1VolLeft 0x004c
#define SMA_wCurrPlay1VolRight 0x004e
#define SMA_wCurrPlay2VolLeft 0x0050
#define SMA_wCurrPlay2VolRight 0x0052
#define SMA_wCurrPlay3VolLeft 0x0054
#define SMA_wCurrPlay3VolRight 0x0056
#define SMA_wCurrPlay4VolLeft 0x0058
#define SMA_wCurrPlay4VolRight 0x005a
#define SMA_wCurrPlay1PeakLeft 0x005c
#define SMA_wCurrPlay1PeakRight 0x005e
#define SMA_wCurrPlay2PeakLeft 0x0060
#define SMA_wCurrPlay2PeakRight 0x0062
#define SMA_wCurrPlay3PeakLeft 0x0064
#define SMA_wCurrPlay3PeakRight 0x0066
#define SMA_wCurrPlay4PeakLeft 0x0068
#define SMA_wCurrPlay4PeakRight 0x006a
#define SMA_wCurrPlayPeakLeft 0x006c
#define SMA_wCurrPlayPeakRight 0x006e
#define SMA_wCurrDATSR 0x0070
#define SMA_wCurrDATRXCHNL 0x0072
#define SMA_wCurrDATTXCHNL 0x0074
#define SMA_wCurrDATRXRate 0x0076
#define SMA_dwDSPPlayCount 0x0078
#define SMA__size 0x007c
#ifdef HAVE_DSPCODEH
# include "pndsperm.c"
# include "pndspini.c"
# define PERMCODE pndsperm
# define INITCODE pndspini
# define PERMCODESIZE sizeof(pndsperm)
# define INITCODESIZE sizeof(pndspini)
#else
# ifndef CONFIG_MSNDPIN_INIT_FILE
# define CONFIG_MSNDPIN_INIT_FILE \
"/etc/sound/pndspini.bin"
# endif
# ifndef CONFIG_MSNDPIN_PERM_FILE
# define CONFIG_MSNDPIN_PERM_FILE \
"/etc/sound/pndsperm.bin"
# endif
# define PERMCODEFILE CONFIG_MSNDPIN_PERM_FILE
# define INITCODEFILE CONFIG_MSNDPIN_INIT_FILE
# define PERMCODE dspini
# define INITCODE permini
# define PERMCODESIZE sizeof_dspini
# define INITCODESIZE sizeof_permini
#endif
#define LONGNAME "MultiSound (Pinnacle/Fiji)"
#endif /* __MSND_PINNACLE_H */

1255
sound/oss/opl3.c
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File diff suppressed because it is too large Load Diff

246
sound/oss/opl3_hw.h
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@ -1,246 +0,0 @@
/*
* opl3_hw.h - Definitions of the OPL-3 registers
*
*
* Copyright (C) by Hannu Savolainen 1993-1997
*
* OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
* Version 2 (June 1991). See the "COPYING" file distributed with this software
* for more info.
*
*
* The OPL-3 mode is switched on by writing 0x01, to the offset 5
* of the right side.
*
* Another special register at the right side is at offset 4. It contains
* a bit mask defining which voices are used as 4 OP voices.
*
* The percussive mode is implemented in the left side only.
*
* With the above exceptions the both sides can be operated independently.
*
* A 4 OP voice can be created by setting the corresponding
* bit at offset 4 of the right side.
*
* For example setting the rightmost bit (0x01) changes the
* first voice on the right side to the 4 OP mode. The fourth
* voice is made inaccessible.
*
* If a voice is set to the 2 OP mode, it works like 2 OP modes
* of the original YM3812 (AdLib). In addition the voice can
* be connected the left, right or both stereo channels. It can
* even be left unconnected. This works with 4 OP voices also.
*
* The stereo connection bits are located in the FEEDBACK_CONNECTION
* register of the voice (0xC0-0xC8). In 4 OP voices these bits are
* in the second half of the voice.
*/
/*
* Register numbers for the global registers
*/
#define TEST_REGISTER 0x01
#define ENABLE_WAVE_SELECT 0x20
#define TIMER1_REGISTER 0x02
#define TIMER2_REGISTER 0x03
#define TIMER_CONTROL_REGISTER 0x04 /* Left side */
#define IRQ_RESET 0x80
#define TIMER1_MASK 0x40
#define TIMER2_MASK 0x20
#define TIMER1_START 0x01
#define TIMER2_START 0x02
#define CONNECTION_SELECT_REGISTER 0x04 /* Right side */
#define RIGHT_4OP_0 0x01
#define RIGHT_4OP_1 0x02
#define RIGHT_4OP_2 0x04
#define LEFT_4OP_0 0x08
#define LEFT_4OP_1 0x10
#define LEFT_4OP_2 0x20
#define OPL3_MODE_REGISTER 0x05 /* Right side */
#define OPL3_ENABLE 0x01
#define OPL4_ENABLE 0x02
#define KBD_SPLIT_REGISTER 0x08 /* Left side */
#define COMPOSITE_SINE_WAVE_MODE 0x80 /* Don't use with OPL-3? */
#define KEYBOARD_SPLIT 0x40
#define PERCOSSION_REGISTER 0xbd /* Left side only */
#define TREMOLO_DEPTH 0x80
#define VIBRATO_DEPTH 0x40
#define PERCOSSION_ENABLE 0x20
#define BASSDRUM_ON 0x10
#define SNAREDRUM_ON 0x08
#define TOMTOM_ON 0x04
#define CYMBAL_ON 0x02
#define HIHAT_ON 0x01
/*
* Offsets to the register banks for operators. To get the
* register number just add the operator offset to the bank offset
*
* AM/VIB/EG/KSR/Multiple (0x20 to 0x35)
*/
#define AM_VIB 0x20
#define TREMOLO_ON 0x80
#define VIBRATO_ON 0x40
#define SUSTAIN_ON 0x20
#define KSR 0x10 /* Key scaling rate */
#define MULTIPLE_MASK 0x0f /* Frequency multiplier */
/*
* KSL/Total level (0x40 to 0x55)
*/
#define KSL_LEVEL 0x40
#define KSL_MASK 0xc0 /* Envelope scaling bits */
#define TOTAL_LEVEL_MASK 0x3f /* Strength (volume) of OP */
/*
* Attack / Decay rate (0x60 to 0x75)
*/
#define ATTACK_DECAY 0x60
#define ATTACK_MASK 0xf0
#define DECAY_MASK 0x0f
/*
* Sustain level / Release rate (0x80 to 0x95)
*/
#define SUSTAIN_RELEASE 0x80
#define SUSTAIN_MASK 0xf0
#define RELEASE_MASK 0x0f
/*
* Wave select (0xE0 to 0xF5)
*/
#define WAVE_SELECT 0xe0
/*
* Offsets to the register banks for voices. Just add to the
* voice number to get the register number.
*
* F-Number low bits (0xA0 to 0xA8).
*/
#define FNUM_LOW 0xa0
/*
* F-number high bits / Key on / Block (octave) (0xB0 to 0xB8)
*/
#define KEYON_BLOCK 0xb0
#define KEYON_BIT 0x20
#define BLOCKNUM_MASK 0x1c
#define FNUM_HIGH_MASK 0x03
/*
* Feedback / Connection (0xc0 to 0xc8)
*
* These registers have two new bits when the OPL-3 mode
* is selected. These bits controls connecting the voice
* to the stereo channels. For 4 OP voices this bit is
* defined in the second half of the voice (add 3 to the
* register offset).
*
* For 4 OP voices the connection bit is used in the
* both halves (gives 4 ways to connect the operators).
*/
#define FEEDBACK_CONNECTION 0xc0
#define FEEDBACK_MASK 0x0e /* Valid just for 1st OP of a voice */
#define CONNECTION_BIT 0x01
/*
* In the 4 OP mode there is four possible configurations how the
* operators can be connected together (in 2 OP modes there is just
* AM or FM). The 4 OP connection mode is defined by the rightmost
* bit of the FEEDBACK_CONNECTION (0xC0-0xC8) on the both halves.
*
* First half Second half Mode
*
* +---+
* v |
* 0 0 >+-1-+--2--3--4-->
*
*
*
* +---+
* | |
* 0 1 >+-1-+--2-+
* |->
* >--3----4-+
*
* +---+
* | |
* 1 0 >+-1-+-----+
* |->
* >--2--3--4-+
*
* +---+
* | |
* 1 1 >+-1-+--+
* |
* >--2--3-+->
* |
* >--4----+
*/
#define STEREO_BITS 0x30 /* OPL-3 only */
#define VOICE_TO_LEFT 0x10
#define VOICE_TO_RIGHT 0x20
/*
* Definition table for the physical voices
*/
struct physical_voice_info {
unsigned char voice_num;
unsigned char voice_mode; /* 0=unavailable, 2=2 OP, 4=4 OP */
unsigned short ioaddr; /* I/O port (left or right side) */
unsigned char op[4]; /* Operator offsets */
};
/*
* There is 18 possible 2 OP voices
* (9 in the left and 9 in the right).
* The first OP is the modulator and 2nd is the carrier.
*
* The first three voices in the both sides may be connected
* with another voice to a 4 OP voice. For example voice 0
* can be connected with voice 3. The operators of voice 3 are
* used as operators 3 and 4 of the new 4 OP voice.
* In this case the 2 OP voice number 0 is the 'first half' and
* voice 3 is the second.
*/
#define USE_LEFT 0
#define USE_RIGHT 1
static struct physical_voice_info pv_map[18] =
{
/* No Mode Side OP1 OP2 OP3 OP4 */
/* --------------------------------------------------- */
{ 0, 2, USE_LEFT, {0x00, 0x03, 0x08, 0x0b}},
{ 1, 2, USE_LEFT, {0x01, 0x04, 0x09, 0x0c}},
{ 2, 2, USE_LEFT, {0x02, 0x05, 0x0a, 0x0d}},
{ 3, 2, USE_LEFT, {0x08, 0x0b, 0x00, 0x00}},
{ 4, 2, USE_LEFT, {0x09, 0x0c, 0x00, 0x00}},
{ 5, 2, USE_LEFT, {0x0a, 0x0d, 0x00, 0x00}},
{ 6, 2, USE_LEFT, {0x10, 0x13, 0x00, 0x00}}, /* Used by percussive voices */
{ 7, 2, USE_LEFT, {0x11, 0x14, 0x00, 0x00}}, /* if the percussive mode */
{ 8, 2, USE_LEFT, {0x12, 0x15, 0x00, 0x00}}, /* is selected */
{ 0, 2, USE_RIGHT, {0x00, 0x03, 0x08, 0x0b}},
{ 1, 2, USE_RIGHT, {0x01, 0x04, 0x09, 0x0c}},
{ 2, 2, USE_RIGHT, {0x02, 0x05, 0x0a, 0x0d}},
{ 3, 2, USE_RIGHT, {0x08, 0x0b, 0x00, 0x00}},
{ 4, 2, USE_RIGHT, {0x09, 0x0c, 0x00, 0x00}},
{ 5, 2, USE_RIGHT, {0x0a, 0x0d, 0x00, 0x00}},
{ 6, 2, USE_RIGHT, {0x10, 0x13, 0x00, 0x00}},
{ 7, 2, USE_RIGHT, {0x11, 0x14, 0x00, 0x00}},
{ 8, 2, USE_RIGHT, {0x12, 0x15, 0x00, 0x00}}
};
/*
* DMA buffer calls
*/

45
sound/oss/os.h
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@ -1,45 +0,0 @@
#define ALLOW_SELECT
#undef NO_INLINE_ASM
#define SHORT_BANNERS
#define MANUAL_PNP
#undef DO_TIMINGS
#include <linux/module.h>
#ifdef __KERNEL__
#include <linux/string.h>
#include <linux/fs.h>
#include <asm/dma.h>
#include <asm/io.h>
#include <asm/param.h>
#include <linux/sched.h>
#include <linux/slab.h>
#include <linux/ioport.h>
#include <asm/page.h>
#include <linux/vmalloc.h>
#include <linux/uaccess.h>
#include <linux/poll.h>
#include <linux/pci.h>
#endif
#include <linux/soundcard.h>
#define FALSE 0
#define TRUE 1
extern int sound_alloc_dma(int chn, char *deviceID);
extern int sound_open_dma(int chn, char *deviceID);
extern void sound_free_dma(int chn);
extern void sound_close_dma(int chn);
extern void reprogram_timer(void);
#define USE_AUTOINIT_DMA
extern void *sound_mem_blocks[1024];
extern int sound_nblocks;
#undef PSEUDO_DMA_AUTOINIT
#define ALLOW_BUFFER_MAPPING
extern const struct file_operations oss_sound_fops;

20
sound/oss/pas2.h
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@ -1,20 +0,0 @@
/* From pas_card.c */
int pas_set_intr(int mask);
int pas_remove_intr(int mask);
unsigned char pas_read(int ioaddr);
void pas_write(unsigned char data, int ioaddr);
/* From pas_audio.c */
void pas_pcm_interrupt(unsigned char status, int cause);
void pas_pcm_init(struct address_info *hw_config);
/* From pas_mixer.c */
int pas_init_mixer(void);
/* From pas_midi.c */
void pas_midi_init(void);
void pas_midi_interrupt(void);
/* From pas2_mixer.c*/
void mix_write(unsigned char data, int ioaddr);

458
sound/oss/pas2_card.c
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/*
* sound/oss/pas2_card.c
*
* Detection routine for the Pro Audio Spectrum cards.
*/
#include <linux/init.h>
#include <linux/interrupt.h>
#include <linux/module.h>
#include <linux/spinlock.h>
#include "sound_config.h"
#include "pas2.h"
#include "sb.h"
static unsigned char dma_bits[] = {
4, 1, 2, 3, 0, 5, 6, 7
};
static unsigned char irq_bits[] = {
0, 0, 1, 2, 3, 4, 5, 6, 0, 1, 7, 8, 9, 0, 10, 11
};
static unsigned char sb_irq_bits[] = {
0x00, 0x00, 0x08, 0x10, 0x00, 0x18, 0x00, 0x20,
0x00, 0x08, 0x28, 0x30, 0x38, 0, 0
};
static unsigned char sb_dma_bits[] = {
0x00, 0x40, 0x80, 0xC0, 0, 0, 0, 0
};
/*
* The Address Translation code is used to convert I/O register addresses to
* be relative to the given base -register
*/
int pas_translate_code = 0;
static int pas_intr_mask;
static int pas_irq;
static int pas_sb_base;
DEFINE_SPINLOCK(pas_lock);
#ifndef CONFIG_PAS_JOYSTICK
static bool joystick;
#else
static bool joystick = 1;
#endif
#ifdef SYMPHONY_PAS
static bool symphony = 1;
#else
static bool symphony;
#endif
#ifdef BROKEN_BUS_CLOCK
static bool broken_bus_clock = 1;
#else
static bool broken_bus_clock;
#endif
static struct address_info cfg;
static struct address_info cfg2;
char pas_model = 0;
static char *pas_model_names[] = {
"",
"Pro AudioSpectrum+",
"CDPC",
"Pro AudioSpectrum 16",
"Pro AudioSpectrum 16D"
};
/*
* pas_read() and pas_write() are equivalents of inb and outb
* These routines perform the I/O address translation required
* to support other than the default base address
*/
unsigned char pas_read(int ioaddr)
{
return inb(ioaddr + pas_translate_code);
}
void pas_write(unsigned char data, int ioaddr)
{
outb((data), ioaddr + pas_translate_code);
}
/******************* Begin of the Interrupt Handler ********************/
static irqreturn_t pasintr(int irq, void *dev_id)
{
int status;
status = pas_read(0x0B89);
pas_write(status, 0x0B89); /* Clear interrupt */
if (status & 0x08)
{
pas_pcm_interrupt(status, 1);
status &= ~0x08;
}
if (status & 0x10)
{
pas_midi_interrupt();
status &= ~0x10;
}
return IRQ_HANDLED;
}
int pas_set_intr(int mask)
{
if (!mask)
return 0;
pas_intr_mask |= mask;
pas_write(pas_intr_mask, 0x0B8B);
return 0;
}
int pas_remove_intr(int mask)
{
if (!mask)
return 0;
pas_intr_mask &= ~mask;
pas_write(pas_intr_mask, 0x0B8B);
return 0;
}
/******************* End of the Interrupt handler **********************/
/******************* Begin of the Initialization Code ******************/
static int __init config_pas_hw(struct address_info *hw_config)
{
char ok = 1;
unsigned int_ptrs; /* scsi/sound interrupt pointers */
pas_irq = hw_config->irq;
pas_write(0x00, 0x0B8B);
pas_write(0x36, 0x138B);
pas_write(0x36, 0x1388);
pas_write(0, 0x1388);
pas_write(0x74, 0x138B);
pas_write(0x74, 0x1389);
pas_write(0, 0x1389);
pas_write(0x80 | 0x40 | 0x20 | 1, 0x0B8A);
pas_write(0x80 | 0x20 | 0x10 | 0x08 | 0x01, 0xF8A);
pas_write(0x01 | 0x02 | 0x04 | 0x10 /*
* |
* 0x80
*/ , 0xB88);
pas_write(0x80 | (joystick ? 0x40 : 0), 0xF388);
if (pas_irq < 0 || pas_irq > 15)
{
printk(KERN_ERR "PAS16: Invalid IRQ %d", pas_irq);
hw_config->irq=-1;
ok = 0;
}
else
{
int_ptrs = pas_read(0xF38A);
int_ptrs = (int_ptrs & 0xf0) | irq_bits[pas_irq];
pas_write(int_ptrs, 0xF38A);
if (!irq_bits[pas_irq])
{
printk(KERN_ERR "PAS16: Invalid IRQ %d", pas_irq);
hw_config->irq=-1;
ok = 0;
}
else
{
if (request_irq(pas_irq, pasintr, 0, "PAS16",hw_config) < 0) {
printk(KERN_ERR "PAS16: Cannot allocate IRQ %d\n",pas_irq);
hw_config->irq=-1;
ok = 0;
}
}
}
if (hw_config->dma < 0 || hw_config->dma > 7)
{
printk(KERN_ERR "PAS16: Invalid DMA selection %d", hw_config->dma);
hw_config->dma=-1;
ok = 0;
}
else
{
pas_write(dma_bits[hw_config->dma], 0xF389);
if (!dma_bits[hw_config->dma])
{
printk(KERN_ERR "PAS16: Invalid DMA selection %d", hw_config->dma);
hw_config->dma=-1;
ok = 0;
}
else
{
if (sound_alloc_dma(hw_config->dma, "PAS16"))
{
printk(KERN_ERR "pas2_card.c: Can't allocate DMA channel\n");
hw_config->dma=-1;
ok = 0;
}
}
}
/*
* This fixes the timing problems of the PAS due to the Symphony chipset
* as per Media Vision. Only define this if your PAS doesn't work correctly.
*/
if(symphony)
{
outb((0x05), 0xa8);
outb((0x60), 0xa9);
}
if(broken_bus_clock)
pas_write(0x01 | 0x10 | 0x20 | 0x04, 0x8388);
else
/*
* pas_write(0x01, 0x8388);
*/
pas_write(0x01 | 0x10 | 0x20, 0x8388);
pas_write(0x18, 0x838A); /* ??? */
pas_write(0x20 | 0x01, 0x0B8A); /* Mute off, filter = 17.897 kHz */
pas_write(8, 0xBF8A);
mix_write(0x80 | 5, 0x078B);
mix_write(5, 0x078B);
{
struct address_info *sb_config;
sb_config = &cfg2;
if (sb_config->io_base)
{
unsigned char irq_dma;
/*
* Turn on Sound Blaster compatibility
* bit 1 = SB emulation
* bit 0 = MPU401 emulation (CDPC only :-( )
*/
pas_write(0x02, 0xF788);
/*
* "Emulation address"
*/
pas_write((sb_config->io_base >> 4) & 0x0f, 0xF789);
pas_sb_base = sb_config->io_base;
if (!sb_dma_bits[sb_config->dma])
printk(KERN_ERR "PAS16 Warning: Invalid SB DMA %d\n\n", sb_config->dma);
if (!sb_irq_bits[sb_config->irq])
printk(KERN_ERR "PAS16 Warning: Invalid SB IRQ %d\n\n", sb_config->irq);
irq_dma = sb_dma_bits[sb_config->dma] |
sb_irq_bits[sb_config->irq];
pas_write(irq_dma, 0xFB8A);
}
else
pas_write(0x00, 0xF788);
}
if (!ok)
printk(KERN_WARNING "PAS16: Driver not enabled\n");
return ok;
}
static int __init detect_pas_hw(struct address_info *hw_config)
{
unsigned char board_id, foo;
/*
* WARNING: Setting an option like W:1 or so that disables warm boot reset
* of the card will screw up this detect code something fierce. Adding code
* to handle this means possibly interfering with other cards on the bus if
* you have something on base port 0x388. SO be forewarned.
*/
outb((0xBC), 0x9A01); /* Activate first board */
outb((hw_config->io_base >> 2), 0x9A01); /* Set base address */
pas_translate_code = hw_config->io_base - 0x388;
pas_write(1, 0xBF88); /* Select one wait states */
board_id = pas_read(0x0B8B);
if (board_id == 0xff)
return 0;
/*
* We probably have a PAS-series board, now check for a PAS16-series board
* by trying to change the board revision bits. PAS16-series hardware won't
* let you do this - the bits are read-only.
*/
foo = board_id ^ 0xe0;
pas_write(foo, 0x0B8B);
foo = pas_read(0x0B8B);
pas_write(board_id, 0x0B8B);
if (board_id != foo)
return 0;
pas_model = pas_read(0xFF88);
return pas_model;
}
static void __init attach_pas_card(struct address_info *hw_config)
{
pas_irq = hw_config->irq;
if (detect_pas_hw(hw_config))
{
if ((pas_model = pas_read(0xFF88)))
{
char temp[100];
if (pas_model < 0 ||
pas_model >= ARRAY_SIZE(pas_model_names)) {
printk(KERN_ERR "pas2 unrecognized model.\n");
return;
}
sprintf(temp,
"%s rev %d", pas_model_names[(int) pas_model],
pas_read(0x2789));
conf_printf(temp, hw_config);
}
if (config_pas_hw(hw_config))
{
pas_pcm_init(hw_config);
pas_midi_init();
pas_init_mixer();
}
}
}
static inline int __init probe_pas(struct address_info *hw_config)
{
return detect_pas_hw(hw_config);
}
static void __exit unload_pas(struct address_info *hw_config)
{
extern int pas_audiodev;
extern int pas2_mididev;
if (hw_config->dma>0)
sound_free_dma(hw_config->dma);
if (hw_config->irq>0)
free_irq(hw_config->irq, hw_config);
if(pas_audiodev!=-1)
sound_unload_mixerdev(audio_devs[pas_audiodev]->mixer_dev);
if(pas2_mididev!=-1)
sound_unload_mididev(pas2_mididev);
if(pas_audiodev!=-1)
sound_unload_audiodev(pas_audiodev);
}
static int __initdata io = -1;
static int __initdata irq = -1;
static int __initdata dma = -1;
static int __initdata dma16 = -1; /* Set this for modules that need it */
static int __initdata sb_io = 0;
static int __initdata sb_irq = -1;
static int __initdata sb_dma = -1;
static int __initdata sb_dma16 = -1;
module_param_hw(io, int, ioport, 0);
module_param_hw(irq, int, irq, 0);
module_param_hw(dma, int, dma, 0);
module_param_hw(dma16, int, dma, 0);
module_param_hw(sb_io, int, ioport, 0);
module_param_hw(sb_irq, int, irq, 0);
module_param_hw(sb_dma, int, dma, 0);
module_param_hw(sb_dma16, int, dma, 0);
module_param(joystick, bool, 0);
module_param(symphony, bool, 0);
module_param(broken_bus_clock, bool, 0);
MODULE_LICENSE("GPL");
static int __init init_pas2(void)
{
printk(KERN_INFO "Pro Audio Spectrum driver Copyright (C) by Hannu Savolainen 1993-1996\n");
cfg.io_base = io;
cfg.irq = irq;
cfg.dma = dma;
cfg.dma2 = dma16;
cfg2.io_base = sb_io;
cfg2.irq = sb_irq;
cfg2.dma = sb_dma;
cfg2.dma2 = sb_dma16;
if (cfg.io_base == -1 || cfg.dma == -1 || cfg.irq == -1) {
printk(KERN_INFO "I/O, IRQ, DMA and type are mandatory\n");
return -EINVAL;
}
if (!probe_pas(&cfg))
return -ENODEV;
attach_pas_card(&cfg);
return 0;
}
static void __exit cleanup_pas2(void)
{
unload_pas(&cfg);
}
module_init(init_pas2);
module_exit(cleanup_pas2);
#ifndef MODULE
static int __init setup_pas2(char *str)
{
/* io, irq, dma, dma2, sb_io, sb_irq, sb_dma, sb_dma2 */
int ints[9];
str = get_options(str, ARRAY_SIZE(ints), ints);
io = ints[1];
irq = ints[2];
dma = ints[3];
dma16 = ints[4];
sb_io = ints[5];
sb_irq = ints[6];
sb_dma = ints[7];
sb_dma16 = ints[8];
return 1;
}
__setup("pas2=", setup_pas2);
#endif

262
sound/oss/pas2_midi.c
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@ -1,262 +0,0 @@
/*
* sound/oss/pas2_midi.c
*
* The low level driver for the PAS Midi Interface.
*/
/*
* Copyright (C) by Hannu Savolainen 1993-1997
*
* OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
* Version 2 (June 1991). See the "COPYING" file distributed with this software
* for more info.
*
* Bartlomiej Zolnierkiewicz : Added __init to pas_init_mixer()
*/
#include <linux/init.h>
#include <linux/spinlock.h>
#include "sound_config.h"
#include "pas2.h"
extern spinlock_t pas_lock;
static int midi_busy, input_opened;
static int my_dev;
int pas2_mididev=-1;
static unsigned char tmp_queue[256];
static volatile int qlen;
static volatile unsigned char qhead, qtail;
static void (*midi_input_intr) (int dev, unsigned char data);
static int pas_midi_open(int dev, int mode,
void (*input) (int dev, unsigned char data),
void (*output) (int dev)
)
{
int err;
unsigned long flags;
unsigned char ctrl;
if (midi_busy)
return -EBUSY;
/*
* Reset input and output FIFO pointers
*/
pas_write(0x20 | 0x40,
0x178b);
spin_lock_irqsave(&pas_lock, flags);
if ((err = pas_set_intr(0x10)) < 0)
{
spin_unlock_irqrestore(&pas_lock, flags);
return err;
}
/*
* Enable input available and output FIFO empty interrupts
*/
ctrl = 0;
input_opened = 0;
midi_input_intr = input;
if (mode == OPEN_READ || mode == OPEN_READWRITE)
{
ctrl |= 0x04; /* Enable input */
input_opened = 1;
}
if (mode == OPEN_WRITE || mode == OPEN_READWRITE)
{
ctrl |= 0x08 | 0x10; /* Enable output */
}
pas_write(ctrl, 0x178b);
/*
* Acknowledge any pending interrupts
*/
pas_write(0xff, 0x1B88);
spin_unlock_irqrestore(&pas_lock, flags);
midi_busy = 1;
qlen = qhead = qtail = 0;
return 0;
}
static void pas_midi_close(int dev)
{
/*
* Reset FIFO pointers, disable intrs
*/
pas_write(0x20 | 0x40, 0x178b);
pas_remove_intr(0x10);
midi_busy = 0;
}
static int dump_to_midi(unsigned char midi_byte)
{
int fifo_space, x;
fifo_space = ((x = pas_read(0x1B89)) >> 4) & 0x0f;
/*
* The MIDI FIFO space register and it's documentation is nonunderstandable.
* There seem to be no way to differentiate between buffer full and buffer
* empty situations. For this reason we don't never write the buffer
* completely full. In this way we can assume that 0 (or is it 15)
* means that the buffer is empty.
*/
if (fifo_space < 2 && fifo_space != 0) /* Full (almost) */
return 0; /* Ask upper layers to retry after some time */
pas_write(midi_byte, 0x178A);
return 1;
}
static int pas_midi_out(int dev, unsigned char midi_byte)
{
unsigned long flags;
/*
* Drain the local queue first
*/
spin_lock_irqsave(&pas_lock, flags);
while (qlen && dump_to_midi(tmp_queue[qhead]))
{
qlen--;
qhead++;
}
spin_unlock_irqrestore(&pas_lock, flags);
/*
* Output the byte if the local queue is empty.
*/
if (!qlen)
if (dump_to_midi(midi_byte))
return 1;
/*
* Put to the local queue
*/
if (qlen >= 256)
return 0; /* Local queue full */
spin_lock_irqsave(&pas_lock, flags);
tmp_queue[qtail] = midi_byte;
qlen++;
qtail++;
spin_unlock_irqrestore(&pas_lock, flags);
return 1;
}
static int pas_midi_start_read(int dev)
{
return 0;
}
static int pas_midi_end_read(int dev)
{
return 0;
}
static void pas_midi_kick(int dev)
{
}
static int pas_buffer_status(int dev)
{
return qlen;
}
#define MIDI_SYNTH_NAME "Pro Audio Spectrum Midi"
#define MIDI_SYNTH_CAPS SYNTH_CAP_INPUT
#include "midi_synth.h"
static struct midi_operations pas_midi_operations =
{
.owner = THIS_MODULE,
.info = {"Pro Audio Spectrum", 0, 0, SNDCARD_PAS},
.converter = &std_midi_synth,
.in_info = {0},
.open = pas_midi_open,
.close = pas_midi_close,
.outputc = pas_midi_out,
.start_read = pas_midi_start_read,
.end_read = pas_midi_end_read,
.kick = pas_midi_kick,
.buffer_status = pas_buffer_status,
};
void __init pas_midi_init(void)
{
int dev = sound_alloc_mididev();
if (dev == -1)
{
printk(KERN_WARNING "pas_midi_init: Too many midi devices detected\n");
return;
}
std_midi_synth.midi_dev = my_dev = dev;
midi_devs[dev] = &pas_midi_operations;
pas2_mididev = dev;
sequencer_init();
}
void pas_midi_interrupt(void)
{
unsigned char stat;
int i, incount;
stat = pas_read(0x1B88);
if (stat & 0x04) /* Input data available */
{
incount = pas_read(0x1B89) & 0x0f; /* Input FIFO size */
if (!incount)
incount = 16;
for (i = 0; i < incount; i++)
if (input_opened)
{
midi_input_intr(my_dev, pas_read(0x178A));
} else
pas_read(0x178A); /* Flush */
}
if (stat & (0x08 | 0x10))
{
spin_lock(&pas_lock);/* called in irq context */
while (qlen && dump_to_midi(tmp_queue[qhead]))
{
qlen--;
qhead++;
}
spin_unlock(&pas_lock);
}
if (stat & 0x40)
{
printk(KERN_WARNING "MIDI output overrun %x,%x\n", pas_read(0x1B89), stat);
}
pas_write(stat, 0x1B88); /* Acknowledge interrupts */
}

327
sound/oss/pas2_mixer.c
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@ -1,327 +0,0 @@
/*
* sound/oss/pas2_mixer.c
*
* Mixer routines for the Pro Audio Spectrum cards.
*/
/*
* Copyright (C) by Hannu Savolainen 1993-1997
*
* OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
* Version 2 (June 1991). See the "COPYING" file distributed with this software
* for more info.
*/
/*
* Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
* Bartlomiej Zolnierkiewicz : added __init to pas_init_mixer()
*/
#include <linux/init.h>
#include "sound_config.h"
#include "pas2.h"
extern int pas_translate_code;
extern char pas_model;
extern int *pas_osp;
extern int pas_audiodev;
static int rec_devices = (SOUND_MASK_MIC); /* Default recording source */
static int mode_control;
#define POSSIBLE_RECORDING_DEVICES (SOUND_MASK_SYNTH | SOUND_MASK_SPEAKER | SOUND_MASK_LINE | SOUND_MASK_MIC | \
SOUND_MASK_CD | SOUND_MASK_ALTPCM)
#define SUPPORTED_MIXER_DEVICES (SOUND_MASK_SYNTH | SOUND_MASK_PCM | SOUND_MASK_SPEAKER | SOUND_MASK_LINE | SOUND_MASK_MIC | \
SOUND_MASK_CD | SOUND_MASK_ALTPCM | SOUND_MASK_IMIX | \
SOUND_MASK_VOLUME | SOUND_MASK_BASS | SOUND_MASK_TREBLE | SOUND_MASK_RECLEV)
static int *levels;
static int default_levels[32] =
{
0x3232, /* Master Volume */
0x3232, /* Bass */
0x3232, /* Treble */
0x5050, /* FM */
0x4b4b, /* PCM */
0x3232, /* PC Speaker */
0x4b4b, /* Ext Line */
0x4b4b, /* Mic */
0x4b4b, /* CD */
0x6464, /* Recording monitor */
0x4b4b, /* SB PCM */
0x6464 /* Recording level */
};
void
mix_write(unsigned char data, int ioaddr)
{
/*
* The Revision D cards have a problem with their MVA508 interface. The
* kludge-o-rama fix is to make a 16-bit quantity with identical LSB and
* MSBs out of the output byte and to do a 16-bit out to the mixer port -
* 1. We need to do this because it isn't timing problem but chip access
* sequence problem.
*/
if (pas_model == 4)
{
outw(data | (data << 8), (ioaddr + pas_translate_code) - 1);
outb((0x80), 0);
} else
pas_write(data, ioaddr);
}
static int
mixer_output(int right_vol, int left_vol, int div, int bits,
int mixer) /* Input or output mixer */
{
int left = left_vol * div / 100;
int right = right_vol * div / 100;
if (bits & 0x10)
{
left |= mixer;
right |= mixer;
}
if (bits == 0x03 || bits == 0x04)
{
mix_write(0x80 | bits, 0x078B);
mix_write(left, 0x078B);
right_vol = left_vol;
} else
{
mix_write(0x80 | 0x20 | bits, 0x078B);
mix_write(left, 0x078B);
mix_write(0x80 | 0x40 | bits, 0x078B);
mix_write(right, 0x078B);
}
return (left_vol | (right_vol << 8));
}
static void
set_mode(int new_mode)
{
mix_write(0x80 | 0x05, 0x078B);
mix_write(new_mode, 0x078B);
mode_control = new_mode;
}
static int
pas_mixer_set(int whichDev, unsigned int level)
{
int left, right, devmask, changed, i, mixer = 0;
left = level & 0x7f;
right = (level & 0x7f00) >> 8;
if (whichDev < SOUND_MIXER_NRDEVICES) {
if ((1 << whichDev) & rec_devices)
mixer = 0x20;
else
mixer = 0x00;
}
switch (whichDev)
{
case SOUND_MIXER_VOLUME: /* Master volume (0-63) */
levels[whichDev] = mixer_output(right, left, 63, 0x01, 0);
break;
/*
* Note! Bass and Treble are mono devices. Will use just the left
* channel.
*/
case SOUND_MIXER_BASS: /* Bass (0-12) */
levels[whichDev] = mixer_output(right, left, 12, 0x03, 0);
break;
case SOUND_MIXER_TREBLE: /* Treble (0-12) */
levels[whichDev] = mixer_output(right, left, 12, 0x04, 0);
break;
case SOUND_MIXER_SYNTH: /* Internal synthesizer (0-31) */
levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x00, mixer);
break;
case SOUND_MIXER_PCM: /* PAS PCM (0-31) */
levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x05, mixer);
break;
case SOUND_MIXER_ALTPCM: /* SB PCM (0-31) */
levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x07, mixer);
break;
case SOUND_MIXER_SPEAKER: /* PC speaker (0-31) */
levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x06, mixer);
break;
case SOUND_MIXER_LINE: /* External line (0-31) */
levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x02, mixer);
break;
case SOUND_MIXER_CD: /* CD (0-31) */
levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x03, mixer);
break;
case SOUND_MIXER_MIC: /* External microphone (0-31) */
levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x04, mixer);
break;
case SOUND_MIXER_IMIX: /* Recording monitor (0-31) (Output mixer only) */
levels[whichDev] = mixer_output(right, left, 31, 0x10 | 0x01,
0x00);
break;
case SOUND_MIXER_RECLEV: /* Recording level (0-15) */
levels[whichDev] = mixer_output(right, left, 15, 0x02, 0);
break;
case SOUND_MIXER_RECSRC:
devmask = level & POSSIBLE_RECORDING_DEVICES;
changed = devmask ^ rec_devices;
rec_devices = devmask;
for (i = 0; i < SOUND_MIXER_NRDEVICES; i++)
if (changed & (1 << i))
{
pas_mixer_set(i, levels[i]);
}
return rec_devices;
break;
default:
return -EINVAL;
}
return (levels[whichDev]);
}
/*****/
static void
pas_mixer_reset(void)
{
int foo;
for (foo = 0; foo < SOUND_MIXER_NRDEVICES; foo++)
pas_mixer_set(foo, levels[foo]);
set_mode(0x04 | 0x01);
}
static int pas_mixer_ioctl(int dev, unsigned int cmd, void __user *arg)
{
int level,v ;
int __user *p = (int __user *)arg;
if (cmd == SOUND_MIXER_PRIVATE1) { /* Set loudness bit */
if (get_user(level, p))
return -EFAULT;
if (level == -1) /* Return current settings */
level = (mode_control & 0x04);
else {
mode_control &= ~0x04;
if (level)
mode_control |= 0x04;
set_mode(mode_control);
}
level = !!level;
return put_user(level, p);
}
if (cmd == SOUND_MIXER_PRIVATE2) { /* Set enhance bit */
if (get_user(level, p))
return -EFAULT;
if (level == -1) { /* Return current settings */
if (!(mode_control & 0x03))
level = 0;
else
level = ((mode_control & 0x03) + 1) * 20;
} else {
int i = 0;
level &= 0x7f;
if (level)
i = (level / 20) - 1;
mode_control &= ~0x03;
mode_control |= i & 0x03;
set_mode(mode_control);
if (i)
i = (i + 1) * 20;
level = i;
}
return put_user(level, p);
}
if (cmd == SOUND_MIXER_PRIVATE3) { /* Set mute bit */
if (get_user(level, p))
return -EFAULT;
if (level == -1) /* Return current settings */
level = !(pas_read(0x0B8A) & 0x20);
else {
if (level)
pas_write(pas_read(0x0B8A) & (~0x20), 0x0B8A);
else
pas_write(pas_read(0x0B8A) | 0x20, 0x0B8A);
level = !(pas_read(0x0B8A) & 0x20);
}
return put_user(level, p);
}
if (((cmd >> 8) & 0xff) == 'M') {
if (get_user(v, p))
return -EFAULT;
if (_SIOC_DIR(cmd) & _SIOC_WRITE) {
v = pas_mixer_set(cmd & 0xff, v);
} else {
switch (cmd & 0xff) {
case SOUND_MIXER_RECSRC:
v = rec_devices;
break;
case SOUND_MIXER_STEREODEVS:
v = SUPPORTED_MIXER_DEVICES & ~(SOUND_MASK_BASS | SOUND_MASK_TREBLE);
break;
case SOUND_MIXER_DEVMASK:
v = SUPPORTED_MIXER_DEVICES;
break;
case SOUND_MIXER_RECMASK:
v = POSSIBLE_RECORDING_DEVICES & SUPPORTED_MIXER_DEVICES;
break;
case SOUND_MIXER_CAPS:
v = 0; /* No special capabilities */
break;
default:
v = levels[cmd & 0xff];
break;
}
}
return put_user(v, p);
}
return -EINVAL;
}
static struct mixer_operations pas_mixer_operations =
{
.owner = THIS_MODULE,
.id = "PAS16",
.name = "Pro Audio Spectrum 16",
.ioctl = pas_mixer_ioctl
};
int __init
pas_init_mixer(void)
{
int d;
levels = load_mixer_volumes("PAS16_1", default_levels, 1);
pas_mixer_reset();
if ((d = sound_alloc_mixerdev()) != -1)
{
audio_devs[pas_audiodev]->mixer_dev = d;
mixer_devs[d] = &pas_mixer_operations;
}
return 1;
}

419
sound/oss/pas2_pcm.c
View File

@ -1,419 +0,0 @@
/*
* pas2_pcm.c Audio routines for PAS16
*
*
* Copyright (C) by Hannu Savolainen 1993-1997
*
* OSS/Free for Linux is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
* Version 2 (June 1991). See the "COPYING" file distributed with this software
* for more info.
*
*
* Thomas Sailer : ioctl code reworked (vmalloc/vfree removed)
* Alan Cox : Swatted a double allocation of device bug. Made a few
* more things module options.
* Bartlomiej Zolnierkiewicz : Added __init to pas_pcm_init()
*/
#include <linux/init.h>
#include <linux/spinlock.h>
#include <linux/timex.h>
#include "sound_config.h"
#include "pas2.h"
#define PAS_PCM_INTRBITS (0x08)
/*
* Sample buffer timer interrupt enable
*/
#define PCM_NON 0
#define PCM_DAC 1
#define PCM_ADC 2
static unsigned long pcm_speed; /* sampling rate */
static unsigned char pcm_channels = 1; /* channels (1 or 2) */
static unsigned char pcm_bits = 8; /* bits/sample (8 or 16) */
static unsigned char pcm_filter; /* filter FLAG */
static unsigned char pcm_mode = PCM_NON;
static unsigned long pcm_count;
static unsigned short pcm_bitsok = 8; /* mask of OK bits */
static int pcm_busy;
int pas_audiodev = -1;
static int open_mode;
extern spinlock_t pas_lock;
static int pcm_set_speed(int arg)
{
int foo, tmp;
unsigned long flags;
if (arg == 0)
return pcm_speed;
if (arg > 44100)
arg = 44100;
if (arg < 5000)
arg = 5000;
if (pcm_channels & 2)
{
foo = ((PIT_TICK_RATE / 2) + (arg / 2)) / arg;
arg = ((PIT_TICK_RATE / 2) + (foo / 2)) / foo;
}
else
{
foo = (PIT_TICK_RATE + (arg / 2)) / arg;
arg = (PIT_TICK_RATE + (foo / 2)) / foo;
}
pcm_speed = arg;
tmp = pas_read(0x0B8A);
/*
* Set anti-aliasing filters according to sample rate. You really *NEED*
* to enable this feature for all normal recording unless you want to
* experiment with aliasing effects.
* These filters apply to the selected "recording" source.
* I (pfw) don't know the encoding of these 5 bits. The values shown
* come from the SDK found on ftp.uwp.edu:/pub/msdos/proaudio/.
*
* I cleared bit 5 of these values, since that bit controls the master
* mute flag. (Olav Wölfelschneider)
*
*/
#if !defined NO_AUTO_FILTER_SET
tmp &= 0xe0;
if (pcm_speed >= 2 * 17897)
tmp |= 0x01;
else if (pcm_speed >= 2 * 15909)
tmp |= 0x02;
else if (pcm_speed >= 2 * 11931)
tmp |= 0x09;
else if (pcm_speed >= 2 * 8948)
tmp |= 0x11;
else if (pcm_speed >= 2 * 5965)
tmp |= 0x19;
else if (pcm_speed >= 2 * 2982)
tmp |= 0x04;
pcm_filter = tmp;
#endif
spin_lock_irqsave(&pas_lock, flags);
pas_write(tmp & ~(0x40 | 0x80), 0x0B8A);
pas_write(0x00 | 0x30 | 0x04, 0x138B);
pas_write(foo & 0xff, 0x1388);
pas_write((foo >> 8) & 0xff, 0x1388);
pas_write(tmp, 0x0B8A);
spin_unlock_irqrestore(&pas_lock, flags);
return pcm_speed;
}
static int pcm_set_channels(int arg)
{
if ((arg != 1) && (arg != 2))
return pcm_channels;
if (arg != pcm_channels)
{
pas_write(pas_read(0xF8A) ^ 0x20, 0xF8A);
pcm_channels = arg;
pcm_set_speed(pcm_speed); /* The speed must be reinitialized */
}
return pcm_channels;
}
static int pcm_set_bits(int arg)
{
if (arg == 0)
return pcm_bits;
if ((arg & pcm_bitsok) != arg)
return pcm_bits;
if (arg != pcm_bits)
{
pas_write(pas_read(0x8389) ^ 0x04, 0x8389);
pcm_bits = arg;
}
return pcm_bits;
}
static int pas_audio_ioctl(int dev, unsigned int cmd, void __user *arg)
{
int val, ret;
int __user *p = arg;
switch (cmd)
{
case SOUND_PCM_WRITE_RATE:
if (get_user(val, p))
return -EFAULT;
ret = pcm_set_speed(val);
break;
case SOUND_PCM_READ_RATE:
ret = pcm_speed;
break;
case SNDCTL_DSP_STEREO:
if (get_user(val, p))
return -EFAULT;
ret = pcm_set_channels(val + 1) - 1;
break;
case SOUND_PCM_WRITE_CHANNELS:
if (get_user(val, p))
return -EFAULT;
ret = pcm_set_channels(val);
break;
case SOUND_PCM_READ_CHANNELS:
ret = pcm_channels;
break;
case SNDCTL_DSP_SETFMT:
if (get_user(val, p))
return -EFAULT;
ret = pcm_set_bits(val);
break;
case SOUND_PCM_READ_BITS:
ret = pcm_bits;
break;
default:
return -EINVAL;
}
return put_user(ret, p);
}
static void pas_audio_reset(int dev)
{
pas_write(pas_read(0xF8A) & ~0x40, 0xF8A); /* Disable PCM */
}
static int pas_audio_open(int dev, int mode)
{
int err;
unsigned long flags;
spin_lock_irqsave(&pas_lock, flags);
if (pcm_busy)
{
spin_unlock_irqrestore(&pas_lock, flags);
return -EBUSY;
}
pcm_busy = 1;
spin_unlock_irqrestore(&pas_lock, flags);
if ((err = pas_set_intr(PAS_PCM_INTRBITS)) < 0)
return err;
pcm_count = 0;
open_mode = mode;
return 0;
}
static void pas_audio_close(int dev)
{
unsigned long flags;
spin_lock_irqsave(&pas_lock, flags);
pas_audio_reset(dev);
pas_remove_intr(PAS_PCM_INTRBITS);
pcm_mode = PCM_NON;
pcm_busy = 0;
spin_unlock_irqrestore(&pas_lock, flags);
}
static void pas_audio_output_block(int dev, unsigned long buf, int count,
int intrflag)
{
unsigned long flags, cnt;
cnt = count;
if (audio_devs[dev]->dmap_out->dma > 3)
cnt >>= 1;
if (audio_devs[dev]->flags & DMA_AUTOMODE &&
intrflag &&
cnt == pcm_count)
return;
spin_lock_irqsave(&pas_lock, flags);
pas_write(pas_read(0xF8A) & ~0x40,
0xF8A);
/* DMAbuf_start_dma (dev, buf, count, DMA_MODE_WRITE); */
if (audio_devs[dev]->dmap_out->dma > 3)
count >>= 1;
if (count != pcm_count)
{
pas_write(pas_read(0x0B8A) & ~0x80, 0x0B8A);
pas_write(0x40 | 0x30 | 0x04, 0x138B);
pas_write(count & 0xff, 0x1389);
pas_write((count >> 8) & 0xff, 0x1389);
pas_write(pas_read(0x0B8A) | 0x80, 0x0B8A);
pcm_count = count;
}
pas_write(pas_read(0x0B8A) | 0x80 | 0x40, 0x0B8A);
#ifdef NO_TRIGGER
pas_write(pas_read(0xF8A) | 0x40 | 0x10, 0xF8A);
#endif
pcm_mode = PCM_DAC;
spin_unlock_irqrestore(&pas_lock, flags);
}
static void pas_audio_start_input(int dev, unsigned long buf, int count,
int intrflag)
{
unsigned long flags;
int cnt;
cnt = count;
if (audio_devs[dev]->dmap_out->dma > 3)
cnt >>= 1;
if (audio_devs[pas_audiodev]->flags & DMA_AUTOMODE &&
intrflag &&
cnt == pcm_count)
return;
spin_lock_irqsave(&pas_lock, flags);
/* DMAbuf_start_dma (dev, buf, count, DMA_MODE_READ); */
if (audio_devs[dev]->dmap_out->dma > 3)
count >>= 1;
if (count != pcm_count)
{
pas_write(pas_read(0x0B8A) & ~0x80, 0x0B8A);
pas_write(0x40 | 0x30 | 0x04, 0x138B);
pas_write(count & 0xff, 0x1389);
pas_write((count >> 8) & 0xff, 0x1389);
pas_write(pas_read(0x0B8A) | 0x80, 0x0B8A);
pcm_count = count;
}
pas_write(pas_read(0x0B8A) | 0x80 | 0x40, 0x0B8A);
#ifdef NO_TRIGGER
pas_write((pas_read(0xF8A) | 0x40) & ~0x10, 0xF8A);
#endif
pcm_mode = PCM_ADC;
spin_unlock_irqrestore(&pas_lock, flags);
}
#ifndef NO_TRIGGER
static void pas_audio_trigger(int dev, int state)
{
unsigned long flags;
spin_lock_irqsave(&pas_lock, flags);
state &= open_mode;
if (state & PCM_ENABLE_OUTPUT)
pas_write(pas_read(0xF8A) | 0x40 | 0x10, 0xF8A);
else if (state & PCM_ENABLE_INPUT)
pas_write((pas_read(0xF8A) | 0x40) & ~0x10, 0xF8A);
else
pas_write(pas_read(0xF8A) & ~0x40, 0xF8A);
spin_unlock_irqrestore(&pas_lock, flags);
}
#endif
static int pas_audio_prepare_for_input(int dev, int bsize, int bcount)
{
pas_audio_reset(dev);
return 0;
}
static int pas_audio_prepare_for_output(int dev, int bsize, int bcount)
{
pas_audio_reset(dev);
return 0;
}
static struct audio_driver pas_audio_driver =
{
.owner = THIS_MODULE,
.open = pas_audio_open,
.close = pas_audio_close,
.output_block = pas_audio_output_block,
.start_input = pas_audio_start_input,
.ioctl = pas_audio_ioctl,
.prepare_for_input = pas_audio_prepare_for_input,
.prepare_for_output = pas_audio_prepare_for_output,
.halt_io = pas_audio_reset,
.trigger = pas_audio_trigger
};
void __init pas_pcm_init(struct address_info *hw_config)
{
pcm_bitsok = 8;
if (pas_read(0xEF8B) & 0x08)
pcm_bitsok |= 16;
pcm_set_speed(DSP_DEFAULT_SPEED);
if ((pas_audiodev = sound_install_audiodrv(AUDIO_DRIVER_VERSION,
"Pro Audio Spectrum",
&pas_audio_driver,
sizeof(struct audio_driver),
DMA_AUTOMODE,
AFMT_U8 | AFMT_S16_LE,
NULL,
hw_config->dma,
hw_config->dma)) < 0)
printk(KERN_WARNING "PAS16: Too many PCM devices available\n");
}
void pas_pcm_interrupt(unsigned char status, int cause)
{
if (cause == 1)
{
/*
* Halt the PCM first. Otherwise we don't have time to start a new
* block before the PCM chip proceeds to the next sample
*/
if (!(audio_devs[pas_audiodev]->flags & DMA_AUTOMODE))
pas_write(pas_read(0xF8A) & ~0x40, 0xF8A);
switch (pcm_mode)
{
case PCM_DAC:
DMAbuf_outputintr(pas_audiodev, 1);
break;
case PCM_ADC:
DMAbuf_inputintr(pas_audiodev);
break;
default:
printk(KERN_WARNING "PAS: Unexpected PCM interrupt\n");
}
}
}

1270
sound/oss/pss.c
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185
sound/oss/sb.h
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@ -1,185 +0,0 @@
#define DSP_RESET (devc->base + 0x6)
#define DSP_READ (devc->base + 0xA)
#define DSP_WRITE (devc->base + 0xC)
#define DSP_COMMAND (devc->base + 0xC)
#define DSP_STATUS (devc->base + 0xC)
#define DSP_DATA_AVAIL (devc->base + 0xE)
#define DSP_DATA_AVL16 (devc->base + 0xF)
#define MIXER_ADDR (devc->base + 0x4)
#define MIXER_DATA (devc->base + 0x5)
#define OPL3_LEFT (devc->base + 0x0)
#define OPL3_RIGHT (devc->base + 0x2)
#define OPL3_BOTH (devc->base + 0x8)
/* DSP Commands */
#define DSP_CMD_SPKON 0xD1
#define DSP_CMD_SPKOFF 0xD3
#define DSP_CMD_DMAON 0xD0
#define DSP_CMD_DMAOFF 0xD4
#define IMODE_NONE 0
#define IMODE_OUTPUT PCM_ENABLE_OUTPUT
#define IMODE_INPUT PCM_ENABLE_INPUT
#define IMODE_INIT 3
#define IMODE_MIDI 4
#define NORMAL_MIDI 0
#define UART_MIDI 1
/*
* Device models
*/
#define MDL_NONE 0
#define MDL_SB1 1 /* SB1.0 or 1.5 */
#define MDL_SB2 2 /* SB2.0 */
#define MDL_SB201 3 /* SB2.01 */
#define MDL_SBPRO 4 /* SB Pro */
#define MDL_SB16 5 /* SB16/32/AWE */
#define MDL_SBPNP 6 /* SB16/32/AWE PnP */
#define MDL_JAZZ 10 /* Media Vision Jazz16 */
#define MDL_SMW 11 /* Logitech SoundMan Wave (Jazz16) */
#define MDL_ESS 12 /* ESS ES688 and ES1688 */
#define MDL_AZTECH 13 /* Aztech Sound Galaxy family */
#define MDL_ES1868MIDI 14 /* MIDI port of ESS1868 */
#define MDL_AEDSP 15 /* Audio Excel DSP 16 */
#define MDL_ESSPCI 16 /* ESS PCI card */
#define MDL_YMPCI 17 /* Yamaha PCI sb in emulation */
#define SUBMDL_ALS007 42 /* ALS-007 differs from SB16 only in mixer */
/* register assignment */
#define SUBMDL_ALS100 43 /* ALS-100 allows sampling rates of up */
/* to 48kHz */
/*
* Config flags
*/
#define SB_NO_MIDI 0x00000001
#define SB_NO_MIXER 0x00000002
#define SB_NO_AUDIO 0x00000004
#define SB_NO_RECORDING 0x00000008 /* No audio recording */
#define SB_MIDI_ONLY (SB_NO_AUDIO|SB_NO_MIXER)
#define SB_PCI_IRQ 0x00000010 /* PCI shared IRQ */
struct mixer_def {
unsigned int regno: 8;
unsigned int bitoffs:4;
unsigned int nbits:4;
};
typedef struct mixer_def mixer_tab[32][2];
typedef struct mixer_def mixer_ent;
struct sb_module_options
{
int esstype; /* ESS chip type */
int acer; /* Do acer notebook init? */
int sm_games; /* Logitech soundman games? */
};
typedef struct sb_devc {
int dev;
/* Hardware parameters */
int *osp;
int minor, major;
int type;
int model, submodel;
int caps;
# define SBCAP_STEREO 0x00000001
# define SBCAP_16BITS 0x00000002
/* Hardware resources */
int base;
int irq;
int dma8, dma16;
int pcibase; /* For ESS Maestro etc */
/* State variables */
int opened;
/* new audio fields for full duplex support */
int fullduplex;
int duplex;
int speed, bits, channels;
volatile int irq_ok;
volatile int intr_active, irq_mode;
/* duplicate audio fields for full duplex support */
volatile int intr_active_16, irq_mode_16;
/* Mixer fields */
int *levels;
mixer_tab *iomap;
size_t iomap_sz; /* number or records in the iomap table */
int mixer_caps, recmask, outmask, supported_devices;
int supported_rec_devices, supported_out_devices;
int my_mixerdev;
int sbmixnum;
/* Audio fields */
unsigned long trg_buf;
int trigger_bits;
int trg_bytes;
int trg_intrflag;
int trg_restart;
/* duplicate audio fields for full duplex support */
unsigned long trg_buf_16;
int trigger_bits_16;
int trg_bytes_16;
int trg_intrflag_16;
int trg_restart_16;
unsigned char tconst;
/* MIDI fields */
int my_mididev;
int input_opened;
int midi_broken;
void (*midi_input_intr) (int dev, unsigned char data);
void *midi_irq_cookie; /* IRQ cookie for the midi */
spinlock_t lock;
struct sb_module_options sbmo; /* Module options */
} sb_devc;
/*
* PCI card types
*/
#define SB_PCI_ESSMAESTRO 1 /* ESS Maestro Legacy */
#define SB_PCI_YAMAHA 2 /* Yamaha Legacy */
/*
* Functions
*/
int sb_dsp_command (sb_devc *devc, unsigned char val);
int sb_dsp_get_byte(sb_devc * devc);
int sb_dsp_reset (sb_devc *devc);
void sb_setmixer (sb_devc *devc, unsigned int port, unsigned int value);
unsigned int sb_getmixer (sb_devc *devc, unsigned int port);
int sb_dsp_detect (struct address_info *hw_config, int pci, int pciio, struct sb_module_options *sbmo);
int sb_dsp_init (struct address_info *hw_config, struct module *owner);
void sb_dsp_unload(struct address_info *hw_config, int sbmpu);
int sb_mixer_init(sb_devc *devc, struct module *owner);
void sb_mixer_unload(sb_devc *devc);
void sb_mixer_set_stereo (sb_devc *devc, int mode);
void smw_mixer_init(sb_devc *devc);
void sb_dsp_midi_init (sb_devc *devc, struct module *owner);
void sb_audio_init (sb_devc *devc, char *name, struct module *owner);
void sb_midi_interrupt (sb_devc *devc);
void sb_chgmixer (sb_devc * devc, unsigned int reg, unsigned int mask, unsigned int val);
int sb_common_mixer_set(sb_devc * devc, int dev, int left, int right);
int sb_audio_open(int dev, int mode);
void sb_audio_close(int dev);
/* From sb_common.c */
void sb_dsp_disable_midi(int port);
int probe_sbmpu (struct address_info *hw_config, struct module *owner);
void unload_sbmpu (struct address_info *hw_config);
void unload_sb16(struct address_info *hw_info);
void unload_sb16midi(struct address_info *hw_info);

1097
sound/oss/sb_audio.c
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354
sound/oss/sb_card.c
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@ -1,354 +0,0 @@
/*
* sound/oss/sb_card.c
*
* Detection routine for the ISA Sound Blaster and compatible sound
* cards.
*
* This file is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
* Version 2 (June 1991). See the "COPYING" file distributed with this
* software for more info.
*
* This is a complete rewrite of the detection routines. This was
* prompted by the PnP API change during v2.5 and the ugly state the
* code was in.
*
* Copyright (C) by Paul Laufer 2002. Based on code originally by
* Hannu Savolainen which was modified by many others over the
* years. Authors specifically mentioned in the previous version were:
* Daniel Stone, Alessandro Zummo, Jeff Garzik, Arnaldo Carvalho de
* Melo, Daniel Church, and myself.
*
* 02-05-2003 Original Release, Paul Laufer <paul@laufernet.com>
* 02-07-2003 Bug made it into first release. Take two.
*/
#include <linux/module.h>
#include <linux/moduleparam.h>
#include <linux/slab.h>
#include <linux/init.h>
#include "sound_config.h"
#include "sb_mixer.h"
#include "sb.h"
#ifdef CONFIG_PNP
#include <linux/pnp.h>
#endif /* CONFIG_PNP */
#include "sb_card.h"
MODULE_DESCRIPTION("OSS Soundblaster ISA PnP and legacy sound driver");
MODULE_LICENSE("GPL");
extern void *smw_free;
static int __initdata mpu_io = 0;
static int __initdata io = -1;
static int __initdata irq = -1;
static int __initdata dma = -1;
static int __initdata dma16 = -1;
static int __initdata type = 0; /* Can set this to a specific card type */
static int __initdata esstype = 0; /* ESS chip type */
static int __initdata acer = 0; /* Do acer notebook init? */
static int __initdata sm_games = 0; /* Logitech soundman games? */
static struct sb_card_config *legacy = NULL;
#ifdef CONFIG_PNP
static int pnp_registered;
static int __initdata pnp = 1;
/*
static int __initdata uart401 = 0;
*/
#else
static int __initdata pnp = 0;
#endif
module_param_hw(io, int, ioport, 000);
MODULE_PARM_DESC(io, "Soundblaster i/o base address (0x220,0x240,0x260,0x280)");
module_param_hw(irq, int, irq, 000);
MODULE_PARM_DESC(irq, "IRQ (5,7,9,10)");
module_param_hw(dma, int, dma, 000);
MODULE_PARM_DESC(dma, "8-bit DMA channel (0,1,3)");
module_param_hw(dma16, int, dma, 000);
MODULE_PARM_DESC(dma16, "16-bit DMA channel (5,6,7)");
module_param_hw(mpu_io, int, ioport, 000);
MODULE_PARM_DESC(mpu_io, "MPU base address");
module_param(type, int, 000);
MODULE_PARM_DESC(type, "You can set this to specific card type (doesn't " \
"work with pnp)");
module_param(sm_games, int, 000);
MODULE_PARM_DESC(sm_games, "Enable support for Logitech soundman games " \
"(doesn't work with pnp)");
module_param(esstype, int, 000);
MODULE_PARM_DESC(esstype, "ESS chip type (doesn't work with pnp)");
module_param(acer, int, 000);
MODULE_PARM_DESC(acer, "Set this to detect cards in some ACER notebooks "\
"(doesn't work with pnp)");
#ifdef CONFIG_PNP
module_param(pnp, int, 000);
MODULE_PARM_DESC(pnp, "Went set to 0 will disable detection using PnP. "\
"Default is 1.\n");
/* Not done yet.... */
/*
module_param(uart401, int, 000);
MODULE_PARM_DESC(uart401, "When set to 1, will attempt to detect and enable"\
"the mpu on some clones");
*/
#endif /* CONFIG_PNP */
/* OSS subsystem card registration shared by PnP and legacy routines */
static int sb_register_oss(struct sb_card_config *scc, struct sb_module_options *sbmo)
{
if (!request_region(scc->conf.io_base, 16, "soundblaster")) {
printk(KERN_ERR "sb: ports busy.\n");
kfree(scc);
return -EBUSY;
}
if (!sb_dsp_detect(&scc->conf, 0, 0, sbmo)) {
release_region(scc->conf.io_base, 16);
printk(KERN_ERR "sb: Failed DSP Detect.\n");
kfree(scc);
return -ENODEV;
}
if(!sb_dsp_init(&scc->conf, THIS_MODULE)) {
printk(KERN_ERR "sb: Failed DSP init.\n");
kfree(scc);
return -ENODEV;
}
if(scc->mpucnf.io_base > 0) {
scc->mpu = 1;
printk(KERN_INFO "sb: Turning on MPU\n");
if(!probe_sbmpu(&scc->mpucnf, THIS_MODULE))
scc->mpu = 0;
}
return 1;
}
static void sb_unload(struct sb_card_config *scc)
{
sb_dsp_unload(&scc->conf, 0);
if(scc->mpu)
unload_sbmpu(&scc->mpucnf);
kfree(scc);
}
/* Register legacy card with OSS subsystem */
static int __init sb_init_legacy(void)
{
struct sb_module_options sbmo = {0};
if((legacy = kzalloc(sizeof(struct sb_card_config), GFP_KERNEL)) == NULL) {
printk(KERN_ERR "sb: Error: Could not allocate memory\n");
return -ENOMEM;
}
legacy->conf.io_base = io;
legacy->conf.irq = irq;
legacy->conf.dma = dma;
legacy->conf.dma2 = dma16;
legacy->conf.card_subtype = type;
legacy->mpucnf.io_base = mpu_io;
legacy->mpucnf.irq = -1;
legacy->mpucnf.dma = -1;
legacy->mpucnf.dma2 = -1;
sbmo.esstype = esstype;
sbmo.sm_games = sm_games;
sbmo.acer = acer;
return sb_register_oss(legacy, &sbmo);
}
#ifdef CONFIG_PNP
/* Populate the OSS subsystem structures with information from PnP */
static void sb_dev2cfg(struct pnp_dev *dev, struct sb_card_config *scc)
{
scc->conf.io_base = -1;
scc->conf.irq = -1;
scc->conf.dma = -1;
scc->conf.dma2 = -1;
scc->mpucnf.io_base = -1;
scc->mpucnf.irq = -1;
scc->mpucnf.dma = -1;
scc->mpucnf.dma2 = -1;
/* All clones layout their PnP tables differently and some use
different logical devices for the MPU */
if(!strncmp("CTL",scc->card_id,3)) {
scc->conf.io_base = pnp_port_start(dev,0);
scc->conf.irq = pnp_irq(dev,0);
scc->conf.dma = pnp_dma(dev,0);
scc->conf.dma2 = pnp_dma(dev,1);
scc->mpucnf.io_base = pnp_port_start(dev,1);
return;
}
if(!strncmp("tBA",scc->card_id,3)) {
scc->conf.io_base = pnp_port_start(dev,0);
scc->conf.irq = pnp_irq(dev,0);
scc->conf.dma = pnp_dma(dev,0);
scc->conf.dma2 = pnp_dma(dev,1);
return;
}
if(!strncmp("ESS",scc->card_id,3)) {
scc->conf.io_base = pnp_port_start(dev,0);
scc->conf.irq = pnp_irq(dev,0);
scc->conf.dma = pnp_dma(dev,0);
scc->conf.dma2 = pnp_dma(dev,1);
scc->mpucnf.io_base = pnp_port_start(dev,2);
return;
}
if(!strncmp("CMI",scc->card_id,3)) {
scc->conf.io_base = pnp_port_start(dev,0);
scc->conf.irq = pnp_irq(dev,0);
scc->conf.dma = pnp_dma(dev,0);
scc->conf.dma2 = pnp_dma(dev,1);
return;
}
if(!strncmp("RWB",scc->card_id,3)) {
scc->conf.io_base = pnp_port_start(dev,0);
scc->conf.irq = pnp_irq(dev,0);
scc->conf.dma = pnp_dma(dev,0);
return;
}
if(!strncmp("ALS",scc->card_id,3)) {
if(!strncmp("ALS0007",scc->card_id,7)) {
scc->conf.io_base = pnp_port_start(dev,0);
scc->conf.irq = pnp_irq(dev,0);
scc->conf.dma = pnp_dma(dev,0);
} else {
scc->conf.io_base = pnp_port_start(dev,0);
scc->conf.irq = pnp_irq(dev,0);
scc->conf.dma = pnp_dma(dev,1);
scc->conf.dma2 = pnp_dma(dev,0);
}
return;
}
if(!strncmp("RTL",scc->card_id,3)) {
scc->conf.io_base = pnp_port_start(dev,0);
scc->conf.irq = pnp_irq(dev,0);
scc->conf.dma = pnp_dma(dev,1);
scc->conf.dma2 = pnp_dma(dev,0);
}
}
static unsigned int sb_pnp_devices;
/* Probe callback function for the PnP API */
static int sb_pnp_probe(struct pnp_card_link *card, const struct pnp_card_device_id *card_id)
{
struct sb_card_config *scc;
struct sb_module_options sbmo = {0}; /* Default to 0 for PnP */
struct pnp_dev *dev = pnp_request_card_device(card, card_id->devs[0].id, NULL);
if(!dev){
return -EBUSY;
}
if((scc = kzalloc(sizeof(struct sb_card_config), GFP_KERNEL)) == NULL) {
printk(KERN_ERR "sb: Error: Could not allocate memory\n");
return -ENOMEM;
}
printk(KERN_INFO "sb: PnP: Found Card Named = \"%s\", Card PnP id = " \
"%s, Device PnP id = %s\n", card->card->name, card_id->id,
dev->id->id);
scc->card_id = card_id->id;
scc->dev_id = dev->id->id;
sb_dev2cfg(dev, scc);
printk(KERN_INFO "sb: PnP: Detected at: io=0x%x, irq=%d, " \
"dma=%d, dma16=%d\n", scc->conf.io_base, scc->conf.irq,
scc->conf.dma, scc->conf.dma2);
pnp_set_card_drvdata(card, scc);
sb_pnp_devices++;
return sb_register_oss(scc, &sbmo);
}
static void sb_pnp_remove(struct pnp_card_link *card)
{
struct sb_card_config *scc = pnp_get_card_drvdata(card);
if(!scc)
return;
printk(KERN_INFO "sb: PnP: Removing %s\n", scc->card_id);
sb_unload(scc);
}
static struct pnp_card_driver sb_pnp_driver = {
.name = "OSS SndBlstr", /* 16 character limit */
.id_table = sb_pnp_card_table,
.probe = sb_pnp_probe,
.remove = sb_pnp_remove,
};
MODULE_DEVICE_TABLE(pnp_card, sb_pnp_card_table);
#endif /* CONFIG_PNP */
static void sb_unregister_all(void)
{
#ifdef CONFIG_PNP
if (pnp_registered)
pnp_unregister_card_driver(&sb_pnp_driver);
#endif
}
static int __init sb_init(void)
{
int lres = 0;
int pres = 0;
printk(KERN_INFO "sb: Init: Starting Probe...\n");
if(io != -1 && irq != -1 && dma != -1) {
printk(KERN_INFO "sb: Probing legacy card with io=%x, "\
"irq=%d, dma=%d, dma16=%d\n",io, irq, dma, dma16);
lres = sb_init_legacy();
} else if((io != -1 || irq != -1 || dma != -1) ||
(!pnp && (io == -1 && irq == -1 && dma == -1)))
printk(KERN_ERR "sb: Error: At least io, irq, and dma "\
"must be set for legacy cards.\n");
#ifdef CONFIG_PNP
if(pnp) {
int err = pnp_register_card_driver(&sb_pnp_driver);
if (!err)
pnp_registered = 1;
pres = sb_pnp_devices;
}
#endif
printk(KERN_INFO "sb: Init: Done\n");
/* If either PnP or Legacy registered a card then return
* success */
if (pres == 0 && lres <= 0) {
sb_unregister_all();
return -ENODEV;
}
return 0;
}
static void __exit sb_exit(void)
{
printk(KERN_INFO "sb: Unloading...\n");
/* Unload legacy card */
if (legacy) {
printk (KERN_INFO "sb: Unloading legacy card\n");
sb_unload(legacy);
}
sb_unregister_all();
vfree(smw_free);
smw_free = NULL;
}
module_init(sb_init);
module_exit(sb_exit);

149
sound/oss/sb_card.h
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/*
* sound/oss/sb_card.h
*
* This file is distributed under the GNU GENERAL PUBLIC LICENSE (GPL)
* Version 2 (June 1991). See the "COPYING" file distributed with this
* software for more info.
*
* 02-05-2002 Original Release, Paul Laufer <paul@laufernet.com>
*/
struct sb_card_config {
struct address_info conf;
struct address_info mpucnf;
const char *card_id;
const char *dev_id;
int mpu;
};
#ifdef CONFIG_PNP
/*
* SoundBlaster PnP tables and structures.
*/
/* Card PnP ID Table */
static struct pnp_card_device_id sb_pnp_card_table[] = {
/* Sound Blaster 16 */
{.id = "CTL0024", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster 16 */
{.id = "CTL0025", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster 16 */
{.id = "CTL0026", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster 16 */
{.id = "CTL0027", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster 16 */
{.id = "CTL0028", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster 16 */
{.id = "CTL0029", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster 16 */
{.id = "CTL002a", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster 16 */
{.id = "CTL002b", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster 16 */
{.id = "CTL002c", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster 16 */
{.id = "CTL00ed", .driver_data = 0, .devs = { {.id="CTL0041"}, } },
/* Sound Blaster 16 */
{.id = "CTL0086", .driver_data = 0, .devs = { {.id="CTL0041"}, } },
/* Sound Blaster Vibra16S */
{.id = "CTL0051", .driver_data = 0, .devs = { {.id="CTL0001"}, } },
/* Sound Blaster Vibra16C */
{.id = "CTL0070", .driver_data = 0, .devs = { {.id="CTL0001"}, } },
/* Sound Blaster Vibra16CL */
{.id = "CTL0080", .driver_data = 0, .devs = { {.id="CTL0041"}, } },
/* Sound Blaster Vibra16CL */
{.id = "CTL00F0", .driver_data = 0, .devs = { {.id="CTL0043"}, } },
/* Sound Blaster AWE 32 */
{.id = "CTL0039", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster AWE 32 */
{.id = "CTL0042", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster AWE 32 */
{.id = "CTL0043", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster AWE 32 */
{.id = "CTL0044", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster AWE 32 */
{.id = "CTL0045", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster AWE 32 */
{.id = "CTL0046", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster AWE 32 */
{.id = "CTL0047", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster AWE 32 */
{.id = "CTL0048", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster AWE 32 */
{.id = "CTL0054", .driver_data = 0, .devs = { {.id="CTL0031"}, } },
/* Sound Blaster AWE 32 */
{.id = "CTL009C", .driver_data = 0, .devs = { {.id="CTL0041"}, } },
/* Createive SB32 PnP */
{.id = "CTL009F", .driver_data = 0, .devs = { {.id="CTL0041"}, } },
/* Sound Blaster AWE 64 */
{.id = "CTL009D", .driver_data = 0, .devs = { {.id="CTL0042"}, } },
/* Sound Blaster AWE 64 Gold */
{.id = "CTL009E", .driver_data = 0, .devs = { {.id="CTL0044"}, } },
/* Sound Blaster AWE 64 Gold */
{.id = "CTL00B2", .driver_data = 0, .devs = { {.id="CTL0044"}, } },
/* Sound Blaster AWE 64 */
{.id = "CTL00C1", .driver_data = 0, .devs = { {.id="CTL0042"}, } },
/* Sound Blaster AWE 64 */
{.id = "CTL00C3", .driver_data = 0, .devs = { {.id="CTL0045"}, } },
/* Sound Blaster AWE 64 */
{.id = "CTL00C5", .driver_data = 0, .devs = { {.id="CTL0045"}, } },
/* Sound Blaster AWE 64 */
{.id = "CTL00C7", .driver_data = 0, .devs = { {.id="CTL0045"}, } },
/* Sound Blaster AWE 64 */
{.id = "CTL00E4", .driver_data = 0, .devs = { {.id="CTL0045"}, } },
/* Sound Blaster AWE 64 */
{.id = "CTL00E9", .driver_data = 0, .devs = { {.id="CTL0045"}, } },
/* ESS 1868 */
{.id = "ESS0968", .driver_data = 0, .devs = { {.id="ESS0968"}, } },
/* ESS 1868 */
{.id = "ESS1868", .driver_data = 0, .devs = { {.id="ESS1868"}, } },
/* ESS 1868 */
{.id = "ESS1868", .driver_data = 0, .devs = { {.id="ESS8611"}, } },
/* ESS 1869 PnP AudioDrive */
{.id = "ESS0003", .driver_data = 0, .devs = { {.id="ESS1869"}, } },
/* ESS 1869 */
{.id = "ESS1869", .driver_data = 0, .devs = { {.id="ESS1869"}, } },
/* ESS 1878 */
{.id = "ESS1878", .driver_data = 0, .devs = { {.id="ESS1878"}, } },
/* ESS 1879 */
{.id = "ESS1879", .driver_data = 0, .devs = { {.id="ESS1879"}, } },
/* CMI 8330 SoundPRO */
{.id = "CMI0001", .driver_data = 0, .devs = { {.id="@X@0001"},
{.id="@H@0001"},
{.id="@@@0001"}, } },
/* Diamond DT0197H */
{.id = "RWR1688", .driver_data = 0, .devs = { {.id="@@@0001"},
{.id="@X@0001"},
{.id="@H@0001"}, } },
/* ALS007 */
{.id = "ALS0007", .driver_data = 0, .devs = { {.id="@@@0001"},
{.id="@X@0001"},
{.id="@H@0001"}, } },
/* ALS100 */
{.id = "ALS0001", .driver_data = 0, .devs = { {.id="@@@0001"},
{.id="@X@0001"},
{.id="@H@0001"}, } },
/* ALS110 */
{.id = "ALS0110", .driver_data = 0, .devs = { {.id="@@@1001"},
{.id="@X@1001"},
{.id="@H@0001"}, } },
/* ALS120 */
{.id = "ALS0120", .driver_data = 0, .devs = { {.id="@@@2001"},
{.id="@X@2001"},
{.id="@H@0001"}, } },
/* ALS200 */
{.id = "ALS0200", .driver_data = 0, .devs = { {.id="@@@0020"},
{.id="@X@0030"},
{.id="@H@0001"}, } },
/* ALS200 */
{.id = "RTL3000", .driver_data = 0, .devs = { {.id="@@@2001"},
{.id="@X@2001"},
{.id="@H@0001"}, } },
/* Sound Blaster 16 (Virtual PC 2004) */
{.id = "tBA03b0", .driver_data = 0, .devs = { {.id="PNPb003"}, } },
/* -end- */
{.id = "", }
};
#endif

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sound/oss/sb_common.c
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sound/oss/sb_ess.c
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