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support for OSS.

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Ben Gras 2009-10-01 16:36:14 +00:00
parent cb6f6a94f7
commit cb50e7e135
39 changed files with 4325 additions and 4 deletions
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5
commands/scripts/MAKEDEV.sh
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@ -240,9 +240,8 @@ do
# Audio devices.
#
$e mknod audio c 13 0
$e mknod rec c 13 1
$e mknod mixer c 13 2
$e chmod 666 audio rec mixer
$e mknod mixer c 13 1
$e chmod 666 audio mixer
;;
random|urandom)
# random data generator.

2
drivers/memory/ramdisk/proto.sh
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@ -3,7 +3,7 @@ PATH=/bin:/sbin:/usr/bin:/usr/sbin
sed -n '1,/@DEV/p' <proto | grep -v @DEV@
(
cd /dev
ls -aln | grep '^[bc]' | egrep -v ' (fd1|fd0p|tcp|eth|ip|udp|tty[pq]|pty)' | \
ls -aln | grep '^[bc]' | egrep -v ' (fd1|fd0p|tcp|eth|ip|udp|tty[pq]|pty)' | grep -v 13, | \
sed -e 's/^[bc]/& /' -e 's/rw-/6/g' -e 's/r--/4/g' \
-e 's/-w-/2/g' -e 's/---/0/g' | \
awk '{ printf "\t\t%s %s--%s %d %d %d %d \n", $11, $1, $2, $4, $5, $6, $7; }'

27
etc/drivers.conf
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@ -421,3 +421,30 @@ driver ipc
;
};
driver osscore
{
system
PRIVCTL # 4
DEVIO # 21
UMAP # 14
IRQCTL # 19
DEVIO # 21
SDEVIO # 22
SETALARM # 24
TIMES # 25
GETINFO # 26
SAFECOPYFROM # 31
SAFECOPYTO # 32
SETGRANT # 34
PROFBUF # 38
SYSCTL
;
pci class
4/1 # Multimedia / Audio device
;
ipc
SYSTEM PM RS LOG TTY DS VFS VM
pci inet amddev
;
uid 0;
};

37
include/midiparser.h Normal file
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@ -0,0 +1,37 @@
/*
* Purpose: Definitions for the MIDI message parser
*/
/*
* This file is part of Open Sound System
*
* Copyright (C) 4Front Technologies 1996-2008.
*
* This software is released under the BSD license.
* See the COPYING file included in the main directory of this source
* distribution for the license terms and conditions
*/
typedef struct midiparser_common midiparser_common_t, *midiparser_common_p;
#define CAT_VOICE 0
#define CAT_MTC 1
#define CAT_SYSEX 2
#define CAT_CHN 3
#define CAT_REALTIME 4
typedef void (*midiparser_callback_t) (void *context, int category,
unsigned char msg, unsigned char ch,
unsigned char *parms, int len);
typedef void (*midiparser_mtc_callback_t) (void *context,
oss_mtc_data_t * mtc);
extern midiparser_common_p midiparser_create (midiparser_callback_t callback,
void *comntext);
extern void midiparser_unalloc (midiparser_common_p common);
extern void midiparser_mtc_callback (midiparser_common_p common,
midiparser_mtc_callback_t callback);
extern void midiparser_input (midiparser_common_p common, unsigned char data);
extern void midiparser_input_buf (midiparser_common_p common,
unsigned char *data, int len);

2140
include/sys/soundcard.h Normal file
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File diff suppressed because it is too large Load diff

33
man/man1/ossinfo.1 Normal file
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@ -0,0 +1,33 @@
." Automatically generated text
.TH 1 "August 31, 2006" "OSS" "User Commands"
.SH NAME
ossinfo - Open Sound System information/status program
.SH SYNOPSIS
ossinfo [-Aaeghmpx] [-v #]
.SH DESCRIPTION
The ossinfo program displays OSS device information.
.SH OPTIONS
-v# Verbose output. Number indicates level of verobisity (0-9).
-p Display only physical audio/midi devices
-g Display ALL audio/midi/mixer devices (physical and virtual)
-a Display audio device files
-A Display audio device files (for applications using O_EXCL)
-e Display all audio engines
-m Display only the MIDI devices
-x Display only the mixer devices
-h Display help.
.SH FILES
/usr/bin/ossinfo
.SH SEE ALSO
ossdevlinks(1), ossmix(1), ossxmix(1)
The Getting information about devices section of the OSS Programmer's Guide
(device_discovery(2)) gives instructions for getting device information
in applications.
.SH AUTHOR
4Front Technologies

133
man/man1/ossmix.1 Normal file
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@ -0,0 +1,133 @@
." Automatically generated text
.TH 1 "August 31, 2006" "OSS" "User Commands"
.SH NAME
ossmix - Open Sound System command-line mixer program.
.SH SYNOPSIS
ossmix [-d <dev#>] [-chqD] [control name] [value]
.SH DESCRIPTION
ossmix is a simple command-line mixer utility that is used to display the mixer
settings of physical and virtual audio devices. OSS version 4 has an extended
mixer API which supports some device specific features that may not available
using other mixer applications.
.SH OPTIOMS
-D Display device information.
-c Dump mixer settings for all mixers.
-h Display usage information.
-q Quiet mode.
-v[1|2] Verbose mode. -v2 prints more detailed infoamation than -v1.
ctrl# value Change value of a mixer control.
<no arg> Display current/possible settings.
.SH USAGE
ossmix without any arguments displays the current settings of the
default mixer device (usually the motherboard sound chip). This
printout can also be used to find out the supported control names and
their possible values. Currently all controls accept an ON/OFF value, a
mono value (0 to 100) or a stereo value (left:right where both channel
volumes can be between 0 and 100). The value can also be expressed in a
relative form (e.g. +1 to add 1 to the previous volume).
The following is a sample printout produced by ossmix:
Selected mixer 0/Creative AudioPCI
Known controls are:
vol <both/leftvol>[:<rightvol>] (currently 50:50)
pcm <both/leftvol>[:<rightvol>] (currently 50:50)
speaker <monovol> (currently 21)
line <both/leftvol>[:<rightvol>] (currently 32:32)
line.rec ON|OFF (currently OFF)
mic <monovol> (currently 16)
mic.rec ON|OFF (currently ON)
cd <both/leftvol>[:<rightvol>] (currently 100:100)
cd.rec ON|OFF (currently OFF)
pcm2 <both/leftvol>[:<rightvol>] (currently 75:75)
line1 <both/leftvol>[:<rightvol>] (currently 32:32)
line1.rec ON|OFF (currently OFF)
line2 <monovol> (currently 32)
line2.rec ON|OFF (currently OFF)
line3 <monovol> (currently 0)
line3.rec ON|OFF (currently OFF)
mic.micboost ON|OFF (currently ON)
mic.micbias ON|OFF (currently ON)
mute.pcmmute ON|OFF (currently OFF)
mute.pcm2mute ON|OFF (currently OFF)
mute.micmute ON|OFF (currently OFF)
mute.cdmute ON|OFF (currently OFF)
mute.linemute ON|OFF (currently OFF)
mute.line1mute ON|OFF (currently OFF)
mute.line2mute ON|OFF (currently OFF)
mute.line3mute ON|OFF (currently OFF)
.SH SELECTING MIXER DEVICE
It's possible to select the mixer device by using the -d<mixernumber>
command line argument. This argument (when used) should be the first one
on the command line. By default the mixer number 0 will be accessed.
To find the available mixer devices, type ossinfo -x and look
under the Mixers heading for available mixer devices.
.SH CHANGING MIXER SETTINGS
Changing the values is done just like with the original "mixer" applet.
For example:
ossmix pcm 50:60
The above sets the pcm control (audio playback volume) so that the left
channel volume is 50 and the right channel volume is 60. With just
"ossmix pcm 50" the both channel volumes will be set to 50.
In addition to the old mixer there are now some (usually ON/OFF) settings.
These settings are device specific and don't work with all soundcards.
The easiest way to find them out is to start ossmix without command line
arguments (other than -d#).
Some control names contain a dot ("."). This dot is required when changing
the value. For example: "ossmix -d0 mic.micboost ON".
."USING OSSMIX WITH A MIDI CONTROLLED MIXER
."The ossmix program has capability to listen MIDI main volume controller
."messages from a MIDI port. You can assign a ossmix control to each MIDI
."channel. After receiving a channel main volume change message ossmix will then
."change the mixer level of the volume control assigned to the channel. In this
."mode ossmix will not exit (you need to kill it manually).
."
."This mode is very useful if you need to make several rapid mixer changes
."simultaneously.
."
."To use this mode you need to give the MIDI device file and a list of the
."volume sliders on command line. For example:
."
." ossmix -d1 -m/dev/midi00 vol mic pcm line gain.out1/2 gain.in3/4
."
."After that the MIDI channels will be assigned in the following way:
."
." Ch 0 = "vol"
." Ch 1 = "mic"
." Ch 2 = "pcm"
." Ch 3 = "line"
." Ch 4 = "gain.out1/2"
." Ch 5 = "gain.in3/4"
."
."Other MIDI channels (6 to 15) will be ignored.
."
."Only mono and stereo slider type controls can be assigned to MIDI channels.
."Both stereo channels will be set to the same volume (there is no balance
."support).
."
."After starting ossmix you should move the sliders on the external fader box
."so that ossmix can figure out their current settings.
."
."At this moment only MIDI fader boxes that send only main volume change messages
."are supported (any other MIDI data will make ossmix to behave incorrectly).
."For example the FM3 MIDI Mixer (AKA "FaderBaby") by JLCooper is compatible
."with ossmix.
."
.SH FILES
/usr/bin/ossmix
.SH SEE ALSO
ossdevlinks(1), ossxmix(1), savemixer(1)
.SH AUTHOR
4Front Technologies

55
man/man1/ossplay.1 Normal file
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@ -0,0 +1,55 @@
." Automatically generated text
.TH 1 "August 31, 2006" "OSS" "User Commands"
.SH NAME
ossplay - Open Sound System playback program.
.SH SYNOPSIS
ossplay [-FRhlvq] [-S secs ] [ -c channels ] [ -d devname ]
[ -f fmtname | ? ] [ -g gain ] [ -o playtarget | ? ]
[ -s rate ] filename | - ...
.SH DESCRIPTION
ossplay plays raw PCM, Microsoft RIFF (.wav), Sun ULaw (.au), Mac AIFF (.aif)
and other types of audio files. By default the application will try to
determine the audio file's format and play audio based on the stored
inforation about sample format, number of channels and sampling rate.
.SH OPTIONS
-v Verbose output. Multiple invocations increase the level
of verbosity.
-q Quiet (no information printed).
-l Loop playback indefinately.
-d<devname> Select <devname> as the device (eg -d/dev/dsp2).
-s<rate> Select the playback rate for raw PCM audio (eg -s48000).
-c<channels Select the number of channels 1=mono 2=stereo, 4, 6, 8, etc.
-f<fmtname> Select the input format (eg -fU8 or -fS16_BE).
-f? Prints the list of supported format names.
-o<playtarget> Selects the play target name if the device supports multiple
play targets (such as front, rear, side).
-o? Prints the list of available play targets.
-g<gain> Amplify all played samples by percentage given as argument.
100 (default) means normal signal level, 200 means double level.
-F Treat all input as raw PCM data.
-R Disable redirection to virtual mixer engines and sample
rate/format conversions. Should not be used unless absolutely
necessary.
-S<secs> Start playing at <secs> seconds from start of file.
The argument can contain a fractional part (e.g. -S1.2)
-h Display usage information.
.SH INTERRUPT
Sending a SIGQUIT (Ctrl-\\ in most terminals) will make ossplay stop playing
the currently played file and skip to the next file.
.SH NOTES
The ossplay executable is the same as the ossrecord executable.
Behaviour is decided by the name used to invoke the program.
.SH SEE ALSO
ossrecord(1), ossmix(1), ossxmix(1)
.SH FILES
/usr/bin/ossplay
.SH AUTHOR
4Front Technologies

103
man/man1/ossrecord.1 Normal file
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@ -0,0 +1,103 @@
." Automatically generated text
.TH 1 "August 31, 2006" "OSS" "User Commands"
.SH NAME
ossrecord - Open Sound System recording program.
.SH USAGE
ossrecord [options] filename
.SH DESCRIPTION
The ossrecord program records audio in Microsoft RIFF (wav) format. It
will record from any input that's currently set as the recording source
by the ossxmix/ossmix mixer programs. With the -l option, you also get
a level meter that will display VU levels in a character mode.
The filename parameter is name of the (.wav) file to be produced. Output can be
sent to stdout by giving - as the file name.
.SH OPTIONS
-s<rate> Select the recording rate for raw PCM audio (eg -s48000).
-c<channels> Select the number of channels 1=mono 2=stereo, 4, 6, 8, etc.
-d<devname> Select <devname> as the device (eg -d/dev/dsp2).
-f<fmt> Select the output sample format (eg -fS32_LE or -fMU_LAW)
-f? Prints the list of supported format names.
-F<cnt> Select the container format (eg WAV or AU). Default is WAV.
-F? Prints the list of supported container formats.
-R Open audio device in raw mode to disable virtual mixing and
sample rate/format conversions. Can be used when recording
from a digital source (S/PDIF input).
-v Verbose output.
-l Display level meters (character based).
-i<recsrc|?> Select the recording source or display available recording
sources if '?' is supplied.
e.g. ossrecord -i? may display:
vol
line (currently selected)
mic
cd
aux1
phone
mono
video
-m<nfiles> Repeat the recording operation <nfiles> times. The filename
argument must have %d (or %02d) somewhere in the file to
guarantee unique filenames. If no %d is given then subsequent
recordings will overwrite the previous one(s). This option is
useful only with loopback audio devices or if the -t option
is used.
-r<command> This option launches the <command> in background after
recording the file has completed. The name of the recorded file
will be given as the (only) command line argument. When the -m
option is used the script will run in parallel while recording
the next file. See the COMMAND SCRIPT section (below) for more
info.
-g<gain> Amplify recorded samples by percentage given as argument.
100 (default) means normal signal level, 200 means double level.
Only supported in 16 and 32 bit modes.
-t<maxsecs> Do not record more than <maxsecs> seconds in a single recording
operation.
-L<level> Set the recording level to <level>.
-O Allow overwriting of file when recording.
-h Display usage instructions.
.SH COMMAND SCRIPT
The -r command line argument makes it possible to execute a
script or program after recording of the wave file is finished.
Below is a simple scell script that does MP3 encoding using
lame.
#!/bin/sh
WAVENAME=$1
MP3NAME=$1.mp3
lame -m s -h --preset studio $WAVENAME $MP3NAME
exit 0
Another example script for ossrecord is a simple CGI script for live MP3
streaming (from /dev/dsp).
#!/bin/sh
echo Content-Type: audio/mp3
echo
ossrecord -S -b16 -s48 - | lame -m j - -
exit 0
.SH NOTES
The ossrecord executable is the same as the ossplay executable.
Behaviour is decided by the name used to invoke the program.
.SH SEE ALSO
ossplay(1), ossmix(1), ossxmix(1)
.SH FILES
/usr/bin/ossrecord
.SH AUTHOR
4Front Technologies

92
man/man1/osstest.1 Normal file
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@ -0,0 +1,92 @@
." Automatically generated text
.TH 1 "August 31, 2006" "OSS" "User Commands"
.SH NAME
osstest - Open Sound System audio self test applet.
.SH DESCRIPTION
The osstest applet is a simple test application that can be used to test
functionality of the sound hardware installed in the system.
osstest performs a playback test for each installed audio device. If there
are any "machine detectable" problems they will be reported. You will first
hear an audio sample played on the left speaker, then the right speaker and
finally in stereo on both speakers.
It's user's responsibility to listen if the test sound is audible. If no
sound output can be heard the possible reason is one of the following:
1. An error was reported by osstest. In this case there will usually not be
any sound output. The error needs to be fixed before running osstest
again.
2. There is no headphones or speakers connected. Or the connection is not
made correctly.
3. The mixer volume level is set to a too low value. By default it should
be OK. The mixer level can be adjusted using the mixer, ossmix and ossxmix
utilities distributed with OSS.
4. Some notebooks have nonstandard volume control and/or speaker selection
hardware that is not supported by OSS. It's very likely that OSS doesn't
support such vendor specific additions.
If no errors were reported and the test sound was audible it means that
OSS and your sound hardware is functioning correctly. If you still encounter
problems with some sound applications the reason is almost certainly in
the application. Check it's configuration or try to use another equivivalent
application.
If you are having problems with JDS, KDE and/or Gnome system sounds, you need
to make sure that OSS gets started before the GUI environment. Refer to your
operating system's startup procedures.
.SH SAMPLE RATE DRIFT
The osstest utility measures a sample rate drift value after playing
back the test sound. Ideally it should be 0% but in practice there
will be an error of few percents. 0% means that the 48000 Hz test file
was played exactly at 48000 Hz sampling rate.
The sample rate measurement is based on the system timer which has limited
precision. It's likely that less than 1% differenc between the nominal and
the measured sampling rates are actually caused by an error in the measurement.
For this reason the drift reported by osstest should not be used as any kind of
quality measurement. However if the drift is very large it means that there is
something wrong in the system. The oscillator chip used with the sound chip is
broken or the system clock is running at a wrong speed.
.SH USING OSSTEST MANUALLY
The osstest utility is located in the /usr/bin directory. It can be run
manually to test functionality of OSS and your sound hardware. When invoked
without any command line parameters osstest performs the default test on all
devices. However it will skip some of the devices base on the following rules.
.IP \(bu 3
It is possible to test just one of the available audio devices by giving
its number on command line (for example osstest 1). Use the device index
numbers reported by "ossinfo -a".
.IP \(bu 3
Use the -l command line option to loop the test infinitely.
.IP \(bu 3
Virtual mixer devices will not be tested. Use the -V command line option to
force test of virtual devices.
.IP \(bu 3
The actual (physical) audio devices will be tested directly (bypassing
virtual mixer). If you want to test playback through vmix then use the
-V option.
.IP \(bu 3
Multiple device files related with the same physical device will not
be tested. Only the first one gets tested while the remaining ones will be
skipped. At this moment there is no way to force osstest to test this kind of
devices.
.IP \(bu 3
Only stereo devices will be tested. Future versions of osstest will be
able to test mono and multi channel devices too. Also osstest requires that
the device supports the 16 bit signed format and 48kHz sampling rate.
.SH FILES
/usr/bin/osstest
.SH SEE ALSO
savemixer(1)
.SH AUTHOR
4Front Technologies

38
man/man1/soundoff.1 Normal file
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@ -0,0 +1,38 @@
." Automatically generated text
.TH 1 "August 31, 2006" "OSS" "OSS System Administration Commands"
.SH NAME
soundoff - Stop Open Sound System
.SH DESCRIPTION
The soundoff command can be used to stop Open Sound System and to unload the
kernel modules related with it.
There are no command line arguments. Only the super user (root) can use this
command.
Open Sound System can be loaded by executing the soundon command.
.SH SAVING THE MIXER AND CONTROL PANEL SETTINGS AUTOMATICALLY
By default soundoff will save the current mixer and control panel settings
automatically each time soundoff is executed. The saved settings will be
restored automatically when soundon is executed next time.
This automatic save feature can be disabled by editing /usr/lib/oss/etc/userdefs
and by changing the line containing "autosave_mixer yes" to
"autosave_mixer no". After this the mixer settings will only be saved when
the savemixer command is executed (by super user).
.SH FILES
/usr/lib/oss/etc/userdefs
/usr/sbin/soundoff
/usr/lib/oss/etc/installed_drivers.
.SH SEE ALSO
soundon(1)
ossdetect(1)
ossdevlinks(1)
.SH AUTHOR
4Front Technologies

28
man/man1/soundon.1 Normal file
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@ -0,0 +1,28 @@
." Automatically generated text
.TH 1 "August 31, 2006" "OSS" "OSS System Administration Commands"
.SH NAME
soundon - Start Open Sound System
.SH DESCRIPTION
The soundon command is used to load the OSS core module (osscore) and the
low level drivers for the sound devices detected in ths system (by ossdetect).
The list of low level sound device drivers to load is located in
/usr/lib/oss/etc/installed_drivers which is maintained by the ossdetect command.
There are no command line arguments. Only the super user (root) can use this
command.
Open Sound System can be unloaded by executing the soundoff command.
.SH FILES
/usr/sbin/soundon
/usr/lib/oss/etc/installed_drivers.
.SH SEE ALSO
soundoff(1)
ossdetect(1)
ossdevlinks(1)
.SH AUTHOR
4Front Technologies

21
man/man7/oss_atiaudio.7 Normal file
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@ -0,0 +1,21 @@
." Automatically generated text
.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_atiaudio - ATI IXP southbridge audio driver.
.SH DESCRIPTION
Open Sound System driver for ATI IXP 150/200/250 audio controller
ATI IXP device characteristics:
o 8/16 bit playback/record
o mono/stereo/4ch/5.1ch playback
o 8KHz to 48Khz sample rate.
.SH OPTIONS
None
.SH FILES
/usr/lib/oss/conf/oss_atiaudio.conf Device configuration file
.SH AUTHOR
4Front Technologies

51
man/man7/oss_audigyls.7 Normal file
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@ -0,0 +1,51 @@
." Automatically generated text
.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_audigyls - Creative Labs CA106 (AudigyLS/SBLive 24bit) driver.
.SH DESCRIPTION
Open Sound System driver for Creative Labs Audigy2-LS and SBLive 24bit 7.1
soundcards.
Audigy-LS device characteristics:
o 8/16/24 bit playback/record
o mono/stereo/4/5.1 playback
o 8KHz to 192Khz sample rate.
.SH AUDIGYLS MODELS
There are 2 models of the AudigyLS device: one with an AC97 codec called the
AudigyLS and the one without called the SBLive 7.1. Essentially they are
the same chip but behave a bit differently.
When playing AC3 on the AudigyLS (the one with the AC97 mixer) - you
need to ensure that the igain slider is set to 0.
.SH AUDIGYLS MIXER
.IP \(bu 3
The AudigyLS has 4 mixer controls for each channel.
.IP \(bu 3
The "spread" button will simply duplicate the front audio on the other 3
channels so that every speaker is playing what the front L/R is playing.
.IP \(bu 3
LoopBack recording allows you to capture any channel that's playing audio.
.IP \(bu 3
Record Volume slider just adjusts the input gain.
.IP \(bu 3
Record Source selector selects the input.
.SH OPTIONS
.IP \(bu 3
audigyls_spdif_enable=0|1
The Audigy LS has a versa-jack (orange) that can be set as SPDIF output
or the Side-Surround left/right speakers in a 7.1 setup.
When set as SPDIF, you can get play PCM/AC3 audio to a Dolby(R) capable
receiver.
.SH FILES
/usr/lib/oss/conf/oss_audigyls.conf Device configuration file
.SH AUTHOR
4Front Technologies

22
man/man7/oss_audiopci.7 Normal file
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@ -0,0 +1,22 @@
." Automatically generated text
.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_audiopci - Creative/Ensoniq Audiopci - ES1370 audio driver.
.SH DESCRIPTION
Open Sound System driver for Creative AudioPCI ES1370 (also sold as SBPCI128)
audio controllers
ES1370 device characteristics:
o 8/16 bit playback/record
o mono/stereo playback/recording
o 8KHz to 48Khz sample rate.
.SH OPTIONS
None
.SH FILES
/usr/lib/oss/conf/oss_audiopci.conf Device configuration file
.SH AUTHOR
4Front Technologies

60
man/man7/oss_cmi878x.7 Normal file
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@ -0,0 +1,60 @@
." Automatically generated text
.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_cmi878x - CMedia CMI8788 audio driver.
.SH DESCRIPTION
Open Sound System driver for CMedia Electronics CMI8788 audio
CMI87xx device characteristics:
o 8/16/24/32 bit playback/record
o mono/stereo/4ch/5.1/7.1 playback
o 8KHz to 192Khz sample rate.
The CMedia 8788 device provides 3 types of devices. The first devices is
a Multichannel full duplex device. The second device provides Front Panel
audio access and the SPDIF device provide SPDIF (Digital) audio I/O.
.SH MIXER PANEL
The CMedia chip provides some unique features that are set up
by the Mixer chip. There are 3 mixer devices presented to the user.
Main Mixer Panel (/dev/mixer0)
.IP \(bu 3
Master Mixer panel is for controlling output volumes for each of the 8
channels.
.IP \(bu 3
Monitor buttons will allow you to monitor the input from the Rear Panel
inputs, Front Panel Inputs and SPDIF IN.
.IP \(bu 3
Speaker-Spread function duplicates the front channel output on all 8
speakers.
.IP \(bu 3
SPDIF Loopback simply takes SPDIF Input and Plays it out the SPDIF Output.
panel.
AC97 Input Mixer Panel (/dev/mixer1)
This mixer panel is used to switch between the various inputs like line-in,
mic, cd. When the Rear Panel Monitor button is check marked in the Main
mixer panel, the IGAIN slider in this panel controls the level of the input
that can be hear on the speakers.
AC97 Front Panel Mixer (/dev/mixer2)
This mixer controls the front panel ac97 device. It can be used to control
all the volumes and inputs as well as SPDIF output on the front panel device.
.SH OPTIONS
None
.SH FILES
/usr/lib/oss/conf/oss_cmi878x.conf Device configuration file
.SH AUTHOR
4Front Technologies

63
man/man7/oss_cmpci.7 Normal file
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." Automatically generated text
.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_cmpci - CMedia CMI8738/8768 audio driver.
.SH DESCRIPTION
Open Sound System driver for CMedia Electronics CMI8738/8768 audio
CMI87xx device characteristics:
o 8/16 bit playback/record
o mono/stereo/4ch/5.1ch playback
o 8KHz to 48Khz sample rate.
.SH MIXER PANEL
The CMedia chip provides some unique features that are set up
by the Mixer chip. Running ossxmix will display the CMI8738 mixer
panel.
Most of the sliders and buttons are self evident. However there
are some options that need explaining:
Dual Dac: Enabling this button sets the CMPCI device as two
separate output devices with /dev/dsp1 audio going to the front and
/dev/dsp0 going to the rear outputs. Separate audio streams can
be send to the device simultaneously.
Speaker Mode: The audio can be sent just to the front speakers or
it can be sent simultaneously to all speakers in the "Spread" mode.
AC3 passthrough only works on Models 037 and higher. This is because of
a hardware bug in the earlier models so check the model number
(ossinfo -a).
SPDIF:
.IP \(bu 3
Enable will enable SPDIF output.
.IP \(bu 3
Rec will allow you to record from the SPDIF device. Note that when
you have SPDIF recording enabled, you cannot play 4/6 channel audio.
.IP \(bu 3
Polarity - certain models require you to flip the bit otherwise you
get distorted audio.
.IP \(bu 3
IMon - monitor input via SPDIF in.
.IP \(bu 3
Optical - sets the SPDIF to Optical (TOSLINK) or RCA Jacks interface.
.SH OPTIONS
None
.SH FILES
/usr/lib/oss/conf/oss_cmpci.conf Device configuration file
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_cs4281 - Cirrus Logic CS4281 driver
.SH DESCRIPTION
Open Sound System driver for Cirrus Logic (Crystal Semicoductor) CS4281 audio
controller.
CS4281 device characteristics:
o 8/16 bit playback/record
o mono/stereo playback/recording
o 8KHz to 48Khz sample rate.
.SH OPTIONS
None
.SH FILES
/usr/lib/oss/conf/oss_cs4281.conf Device configuration file
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_cs461x - Cirrus Logic CS461x/CS4280 audio driver.
.SH DESCRIPTION
Open Sound System driver for Crystal Semiconductor (Cirrus Logic) CS4280 and
461x, audio controllers.
CS4280 device characteristics:
o 8/16 bit playback/record
o mono/stereo playback/recording
o 8KHz to 48Khz sample rate.
.SH OPTIONS
.IP \(bu 3
cs461x_clk_run_fix=0|1 (feature not used anylonger)
Certain IBM Thinkpads required the CLK_RUN bit flipped in order to wake up
the audio device.
.SH FILES
/usr/lib/oss/conf/oss_cs461x.conf Device configuration file
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_digi96 - RME Digi96 professional audio driver.
.SH DESCRIPTION
Audio driver for the RME Digi96 family of profressional audio controllers.
- Only 16 and 24/32 bit audio formats are supported.
- All Digi96 family members support 32kHz, 44.1kHz, 48kHz, 64kHz, 88.2kHz and
96kHz. sampling rates.
.SH MIXER PANEL
Note! For recording you need to set digi96.sync to INTERNAL and the values
of digi96.mode and digi96.input to match your studio setup. Otherwise
recordings will fail with I/O error.
There are several settings that can be changed using the ossmix program
shipped with OSS. Note that some features don't work with all Digi96
family members. For example ADAT mode is not supported by the base
model.
.IP \(bu 3
digi96.mode <SPDIF|AESEBU|ADAT>:
This setting controls the output mode which can be S/PDIF (consumer),
AES/EBU (professional) or ADAT. The input mode is detected automatically.
If ADAT input is detected the output mode will be switched to ADAT
automatically (this doesn't work in the other direction).
.IP \(bu 3
digi96.sync <EXTERNAL|INTERNAL>:
This setting tells if the playback sampling rate is based on the internal
oscillator or the sample rate detected in the input port. See also the
definition of the digi96.worldclk setting.
.IP \(bu 3
digi96.input <OPTICAL|COAXIAL|INTERNAL|XLR>: Selects the active input.
.IP \(bu 3
digi96.sel <BYPASS|NORMAL>:
When set to BYPASS the input signal will be routed directly to the
output (also sets digi96.sync automatically to EXTERNAL). In this mode
audio data written to /dev/dsp will be muted.
.IP \(bu 3
digi96.worldclk ON|OFF:
Setting this control to ON will enable the optional worldclock input as
the sample rate source (overrides the digi96.sync setting).
.IP \(bu 3
digi96.emph ON|OFF:
Enables/disables the de-emphasis option on the analog (monitor) output
connector.
.IP \(bu 3
digi96.data <AUDIO|AC3>:
Specifies if the output signal is audio or AC3 data (sets the non-audio
bit in the channel status data).
.SH OPTIONS
None
.SH FILES
/usr/lib/oss/conf/oss_digi96.conf Device configuration file
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_emu10k1x - Creative Labs P16x (EMU10K1X) driver.
.SH DESCRIPTION
Open Sound System driver for Creative Labs SBLive 5.1 Dell OEM version
soundcards. The device has a chipset called the EMU10K1X and is not the same
as the SBLive EMU10K1/EMU10K2 audio processors found in the SBLive! and Audigy
soundcards.
EMU10K1X device characteristics:
o 8/16/24 bit playback/record
o mono/stereo/4/5.1 playback
o 8KHz to 192Khz sample rate.
.SH OPTIONS
.IP \(bu 3
emu10k1x_spdif_enable=<0|1>
The EMU10K1X has a versa-jack (orange) that can be set as SPDIF output
or the Side-Surround left/right speakers in a 5.1 setup.
When set as SPDIF, you can get play PCM/AC3 audio to a Dolby(R) capable
receiver.
.SH FILES
/usr/lib/oss/conf/oss_emu10k1x.conf Device configuration file
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_envy24 - ICE Envy24 audio device driver.
.SH DESCRIPTION
Open Sound System driver for Envy24 based audio cards such as the
M-Audio Delta Series, Terratec EWS88 Series, Hoontech DSP24.
ENVY24 device characteristics:
.IP \(bu 3
8/16 bit playback/record
.IP \(bu 3
mono/stereo/4ch/5.1ch/7.1ch playback
.IP \(bu 3
mono/sterero recording
.IP \(bu 3
8KHz to 192Khz sample rate.
ENVY24 AUDIO DEVICES
Audio devices:
0: M Audio Delta 1010 out1/2
1: M Audio Delta 1010 out3/4
2: M Audio Delta 1010 out5/6
3: M Audio Delta 1010 out7/8
4: M Audio Delta 1010 S/PDIF out
5: M Audio Delta 1010 in1/2
6: M Audio Delta 1010 in3/4
7: M Audio Delta 1010 in5/6
8: M Audio Delta 1010 in7/8
9: M Audio Delta 1010 S/PDIF in
10: M Audio Delta 1010 input from mon. mixer
11: M Audio Delta 1010 (all outputs)
12: M Audio Delta 1010 (all inputs)
Synth devices:
Midi devices:
0: M Audio Delta 1010
Timers:
0: System clock
Mixers:
0: M Audio Delta 1010
The actual /dev/dsp# numbers may be different on your system. Check the right
ones by looking at the output procuced by "ossinfo -a" command. With the
above configuration you can use /dev/dsp0 to /dev/dsp4 for playback of stereo
streams. If you play mono files the signal will be output only from the left
channel. /dev/dsp0 to /dev/dsp3 are connected to the analog outputs while
/dev/dsp4 is the S/PDIF output.
The /dev/dsp5 to /dev/dsp10 device files can be used for recording. /dev/dsp5
to /dev/dsp8 are the analog inputs. /dev/dsp11 and /dev/dsp12 are raw
input/output device files. They will be described in detail in the "Raw I/O
devices" section below.
It's also possible to make OSS to create individual device files for every
channel this creates twice as many device files than the default setting. To
do this just append envy24_skipdevs=1 to the oss_envy24.conf file. This is useful
only if you are working on mono rather than stereo signals. However please
note that setting envy24_skipdevs=1 does _NOT_ lock the device files to one
channel mode, the application can still set them to stereo or multi channel
mode if it likes.
It is possible to set all device files to mono only mode by setting
envy24_skipdevs=1 and envy24_force_mono=1. However this mode disables stereo
and multi channel usage for all devices so in general it should not be used.
.SH
By default the driver will create output devices before the input ones. By
setting envy24_swapdevs=1 in oss_envy24.conf you can ask OSS to create the device
files in opposite order i.e. input device files before the output ones. This
may be useful when using RealProducer.
As a workaround to a bug in RealProducer you also need to create some dummy
mixer devices by defining envy24_realencoder_hack=1 in oss_envy24.conf. Without
these extra mixer devices RealProducer will not be able to access other than
the first input device.
.SH DEVICE MANAGEMENT
By default OSS creates a large number of device files for each envy24 card.
This may be a problem when multiple cards need to be used in the same system.
Adding the envy24_devmask option to oss_envy24.conf should help
in most cases because it removes the devices that are actually not needed in
the system.
The envy24_devmask number is the SUM of the following values:
1: Create primary (analog/ADAT/TDIF) outputs.
2: Create primary (analog/ADAT/TDIF) inputs.
4: Create S/PDIF outputs.
8: Create S/PDIF inputs.
16: Create monitor input device.
32: Create the raw input and output devices.
For example envy24_devmask=12 (4+8) creates only the S/PDIF devices.
To enable all possible (current or future) device files set envy24_devmask
to 65535 (default).
If possible make your application to open the right device file
(/dev/dsp0 to /dev/dsp10) explicitly. It's also possible to use the
default devicefile (/dev/dsp) since OSS now supports automatic device
allocation (it opens the first available input or output devicefile
depending on the open mode).
The channel allocation mechanism between device files is very flexible.
Even there is a device file for every stereo pair (or a mono channel)
it's possible to use any of the device file to access multiple channels.
For example an application can open /dev/dsp0 and set the number of
channels to 10. In this way the application can play all 10 channels
(or any number between 1 and 10) simultaneously (the samples will be
interleaved).
There is simple automatic syncstart feature when using multiple
applications at the same time. Playback will not start before all
currently open devices files have started the playback operation.
The same mechanism works for recording (recording and playback
operations are fully independent).
The Envy24 driver supports 8, 16 and 24/32 bit sample formats.
.SH SAMPLING RATE
Envy24 based cards are multi channel devices and all the channels share the
same sampling rate. For this reason the sampling rate is normally locked to the
value selected using ossmix. However OSS supports some other methods for
changing the sampling rate. There are four ways to change the sampling rate.
BASIC METHOD:
Since all input and output channels of Envy24 work at the same sampling rate
it's not possible for the applications to select the rate themselves. Instead
the sampling rate is always locked to the currently selected rate. This rate
selection can be changed using the ossmix program shipped with OSS.
For example:
ossmix envy24.rate 48000
sets the sampling rate to 48000 Hz (default). The possible alternatives are
- 8000
- 9600
- 11025
- 12000
- 16000
- 22050
- 24000
- 32000
- 44100
- 48000
- 88200
- 96000
When using S/PDIF inputs/outputs only the sampling rates 32000, 44100, 48000, 88200 or 96000 should be used.
.SH EXTERNAL SYNC
It's possible to lock the sampling rate to the S/PDIF or world clock inputs
by setting the envy24.sync setting in ossmix to SPDIF or WCLOCK. However
the envy24.rate setting should be set manually to match the rate being used
(there is no autodetection for that).
.SH NONLOCKED METHOD
It's also possible to turn the envy24.ratelock setting to OFF using ossmix.
After that the first application that opens the device can change the sampling
rate. However great care should be taken that this application gets the
recording/playback process fully started before any of the other
applications open their devices. Otherwise all devices will be locked to 8Khz.
Also keep in mind that subsequent applications will be forced to use the
sampling rate set by the first one.
.SH SOFTWARE SRC
OSS contains a very high quality software based sample rate converter.
It can be enabled by setting envy24.src to ON using ossmix.
After that OSS can do on-fly sample rate conversions between the actual
"hardware" sampling rate and the sampling rates used by the applications. In
this way every application may use different sampling rate. However there are
some drawbacks in this method:
.IP \(bu 3
The hardware rate needs to be 44100, 48000 or 96000 Hz.
.IP \(bu 3
The software SRC algorithm consumes some CPU time (1% to 20% per audio
channel depending on the CPU speed and sampling rates). For this reason this
method may be useless in multi channel use with anything else but the fastest
high end CPUs.
.IP \(bu 3
Only mono and stereo (1 or 2 channel) streams are supported.
.IP \(bu 3
The SRC algorithm does cause minor artifacts to the sound (SNR is around 60 dB).
.SH RAW IO DEVICES
These device files provide an alternative way to access Envy24 based devices.
With these devices it's possible to bypass the dual buffering required by the
"normal" input-output device files described above. This means that also the
mmap() feature is available and that the latencies caused by dual buffering
are gone. So these device files work much like "ordinary" soundcards. However
due to multi channel professional nature of the Envy24 chip there are some very
fundamental differences. This means that these device files can only be used
with applications that are aware of them.
The differences from normal audio device files are:
1. The sample format will always be 32 bit msb aligned (AFMT_S32_LE). Trying to
use any other sample format will cause unexpected results.
2. Number of channels is fixed and cannot be changed. The output device has
always 10 channels (0 to 7 are analog outputs and 8 to 9 are the digital
outputs). This assignment will be used even with cards that don't support
digital (or analog) outputs at all. If the actual hardware being used has
less channels the unused ones will be discarded (however they will be fed to
the on board monitor mixer).
The input device is fixed to 12 channels. Channels 0 to 7 are analog inputs.
Channels 8 to 9 are digital inputs. Channels 10 and 11 are for the result
signal from the on board monitor mixer.
.SH DIGITAL MONITOR MIXER
All Envy24 based cards have a built in monitor mixer. It can be used to mix
allinput and output signals together. The result can be recorded from the
"input from mon mixer" device (device 10 in the /dev/sndstat example above).
The monitor mix signal can also be routed to any of the outputs (including
S/PDIF and the "consumer" AC97 output of Terratec EWS88MT/D and any other card
that support s it).
The settings in the gain.* group of ossmix are used to change the levels of all
inputs and outputs in the digital monitor mixer. The possible values are
between 0 (minimum) and 144 (maximum).
OSS permits using all 10 possible output channels of the monitor mixer even
with cards that have less physical outputs. These "virtual" outputs are only
sent to the monitor mixer and their signal is only present in the monitor mixer
output. To enable these "virtual" channels set the envy24_virtualout parameter
to 1 in oss_envy24.conf. This option has no effect with Delta1010, EWS88MT and
other cards that have 10 "real" outputs.
.SH SYNC SOURCE
On cards with S/PDIF and/or World Clock inputs it's possible to select the
sync source using
ossmix envy24.sync
The possible choices are:
.IP \(bu 3
INTERNAL: Use the internal sampling rate as defined by envy24.rate
.IP \(bu 3
SPDIF: Use the S/PDIF input as the clock source. The envy24.rate setting
must be set manually to match the actual input sampling rate.
.IP \(bu 3
WCLOCK: Like SPDIF but uses the world clock input signal (Delta 1010 only).
.SH OUTPUT ROUTINGS
Output routing of output ports can be changed by changing the route.* settings
using ossmix. The possible choices are:
.IP \(bu 3
DMA: Playback from the associated /dev/dsp# device.
.IP \(bu 3
MONITOR: Output of the digital mixer (only out1/2 and S/PDIF).
.IP \(bu 3
IN1/2 to IN9/10 or IN1 to IN10: Loopback from the analog inputs
.IP \(bu 3
SPDIFL or SPDIFR or SPDIF: Loopback from the S/PDIF input.
.SH PEAK METERS
Envy24 based cards have peak meters for the input and output ports of the
digital monitor mixer. ossmix can show these values under the peak.* group
(these settings are read only). The values are between 0 (minimum) and 255
(maximum). At this moment the only applications that supports these peak meters
are ossmix and ossxmix.
.SH AUDIO LATENCY
IDE disk and CD-ROM drives may cause some interrupt latency problems which
may cause dropouts in recording/playback with Envy24 based cards. For this
reason ensure that DMA is turned on for the disk drive.
Another method to solve the dropout problems is making the fragment size used
by the driver longer. This can be done by adding envy24_nfrags=N to the
oss_envy24.conf file. By default N is 16. Values 2, 4 or 8 make the fragments
longer which should cure the dropout problems. However this may cause
latency problems with some applications. Values 32 and 64 decrease the
latencies but may cause dropouts with IDE.
.SH OPTIONS
.IP \(bu 3
envy24_skipdevs: It's also possible to make OSS to create individual device
files for every channel. This creates twice as many device files than the
default setting.
Values: 1, 0 Default: 0
.IP \(bu 3
envy24_swapdevs: By default the driver will create output devices before the
input ones. You can force the input devices to be configured before output
devices.
Values: 1, 0 Default: 0
.IP \(bu 3
envy24_realencoder_hack: RealProducer wants to see a mixer device in
/dev/mixer. This option allows you to define a dummy /dev/mixer mixer device.
Envy24 Mixer device doesn't provide any consumer level soundcard compatibility
so this dummy mixer fools RealProducer into thinking it's running on a consumer
soundcard like SB Pro or SBLive.
Values: 1, 0 Default: 0
.IP \(bu 3
envy24_gain_sliders: With some devices it's possible to change the gain
controllers to be continuous sliders instead of just enumerated ones.
Values: 1, 0 Default: 0
.IP \(bu 3
envy24_nfrags: To solve the dropout problems make the fragment size used by
the driver longer. By default is 16. Values 2, 4 or 8 make the fragments longer
which should cure the dropout problems. However this may cause latency problems
with some applications. Values 32 and 64 decrease the latencies but may cause
dropouts with IDE drives.
Values: 2-64 Default: 16
.IP \(bu 3
envy24_virtualout: OSS permits using all 10 possible output channels of the
monitor mixer even with cards that have less physical outputs. These "virtual"
outputs are only sent to the monitor mixer and their signal is only present in
the monitor mixer output. This has no effect for Delta1010 or Terratec EWS88MT.
Values: 1, 0 Default: 0
.IP \(bu 3
envy24_force_mono: It is possible to set all device files to mono only mode
by setting envy24_skipdevs=1 and envy24_force_mono=1. However this mode
disables stereo and multi channel usage for all devices so in general it should
not be used.
Values: 1, 0 Default: 0
.SH FILES
/usr/lib/oss/conf/oss_envy24.conf Device configuration file
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_envy24ht - VIA Envy24HT/PT audio driver.
.SH DESCRIPTION
Open Sound System driver for Envy24HT, Envy24HT-S, Envy24PT based sound
cards.
Envy24HT device characteristics:
o 8/16 bit playback/record
o mono/stereo/4ch/5.1ch/7.1ch playback
o mono/sterero recording
o 8KHz to 192Khz sample rate.
.SH OPTIONS
o envy24ht_model = -1|0|1
Select the Model number if the card isn't autodetected
Values: 0 = Envy24ht 1=Envy24PT/HT-s compatible -1=Autodetect Default: -1
o envy24ht_fake_mixer = 0|1
Some old applications may refuse to run if they don't find some legacy mixer
controls the envy24ht chip doesn't support. A "fake" legacy mixer can be
enabled to make such applications to run. However these fake legacy controls
will be permanently bound to full level.
Values: 0 = Disabled 1 = Enabled. Default: 0.
.SH FILES
/usr/lib/oss/conf/oss_envy24ht.conf Device configuration file
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_fmedia - Forte Media FM801 driver.
.SH DESCRIPTION
Open Sound System driver for Forte Media FM801/FM801-AU audio controllers.
FM801 device characteristics:
o 8/16 bit playback/record
o mono/stereo/4ch/5.1ch playback
o mono/sterero recording
o 8KHz to 48Khz sample rate.
.SH OPTIONS
.IP \(bu 3
fmedia_mpu_irq=<xx>
Set the IRQ for the UART401 MPU. Refer to device conf file (see below) for
valid IRQs.
.SH FILES
/usr/lib/oss/conf/oss_fmedia.conf Device configuration file
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_geode - National Semiconductor Geode audio driver.
.SH DESCRIPTION
Open Sound System driver for National Semiconductor Geode/CS5530/CS5536 audio
controllers.
Geode device characteristics:
o 8/16 bit playback/record
o mono/stereo playback/recording
o 8KHz to 48Khz sample rate.
.SH NOTES
Some old Geode CPUs are not able to handle heavy computational loads.
If your audio streams are use a lot of CPU, you can start getting garbled audio
since the OSS Sample Rate Convertor is CPU intensive. Setting vmix0-src to
OFF will allow you to play audio but only at a fixed rate set via vmixctl
(Default: 48Khz).
.SH FILES
/usr/lib/oss/conf/oss_geode.conf Device configuration file
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_hdaudio - Intel High Definition Audio (AZALIA)
.SH DESCRIPTION
Open Sound System driver for Intels high definition audio known as
"Azalia". This driver supports Intel 915/925 chipsets with the
Realtek ALC880 and CMedia 9880 8 channel codecs.
The HDA driver supports:
o 8-96Khz Playback/Recording
o 8 or 16 or 32 bits
o 2, 4, 6 or 8 channel audio.
o SPDIF digital output and Input
o AC3 passthrough
.SH HDAUDIO MIXER
The Intel HDA mixer is a new type of mixer that doesn't have
the normal volume controls found on AC97 or legacy SB devices.
The HDA mixer presents a concept of Jacks and you can configure
any jack to be either an output or an input jack.
Some motherboards may not correctly initialize the jacks according
to their color and functionality but in general here's the
configuration that should generally be followed:
o Orange = Center/LFE o Blue = Line-in
o Black = Rear o Green = Front
o Grey = Side o Pink = Mic
Some Azalia codecs support front panel connectors and so if you see
fp-green and fp-pink connectors, then these are for front panel
speaker and mic/line-in Jacks.
There is a function selector for most of the analog audio jacks (for example
connector.pink.mode). This selector is used to control if the jack is used
as an input (microphone or line in) or output (front, rear, side, speaker,
etc).
.SH KNOWN PROBLEMS
In general Azalia based systems (laptops/motherboards) would require a custom
driver to work properly. Due to enormous number of different systems it is not
possible to develop such custom drivers for all systems. A generic driver is
used for systems that don't have dedicated drivers.
Unfortunately the mixer and control panel interface (see ossmix(1))
for "generic" systems is very cryptic and difficult to
understand. To solve problems with volumes or signal routing you need to
start ossxmix(1) and change the controls one at time until you get the desired
effect.
.SH OPTIONS
.IP \(bu 3
hdaudio_jacksense enables jack sensing mode when the hdaudio driver is
loaded. In this mode all I/O pin's that are not
in use will be disabled as well as the mixer controls
that are related with them. In this way the
mixer/control panel will become more intuitive.
However OSS will need to be restarted with soundoff;
soundon every time new inputs or outputs are attached
to the audio jacks. Default : 0.
NOTE! hdaudio_jacksense=1 works only in some systems.
Many laptops and motherboards don't support jack
sensing.
.IP \(bu 3
hdaudio_noskip Disable skipping unconnected jack. All mixer controls
will be shown, even for disabled I/O pins.
Can get values 0-7. 1-7 is a bitmask, where every bit
masks a different check. Bit 3 (= value 4) overrides
jacksense check too.
Default: 0 - unconnected jacks are skipped.
.SH FILES
/usr/lib/oss/conf/oss_hdaudio.conf Device configuration file
.SH AUTHOR
4Front Technologies

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." Automatically generated text
.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_ich - Intel ICH/SiS7012/Nvidia/AMD audio device driver.
.SH DESCRIPTION
Open Sound System driver for Intel ICH, nVidia Nforce, AMD and SiS 7012
devices.
.SH OPTIONS
.IP \(bu 3
intelpci_rate_tuning=<NNN> (default is 240)
Some Compaq Deskpro models (EN and EX at least) and certain Dell models
play and record audio at a higher speed than what is expected. If you have
an Intel815 motherboard with an AD1885 you can try setting the parameter
to 240, 280 or 330 and see which works for your system. The way to figure
out the the right intelpci_rate_tuning value is using the osstest application.
It reports a sample rate drift value ("Sample rate drift" or "srate drift").
Use the following formula (round the result to the nearest integer):
<intelpci_rate_tuning = (240*(drift+100))/100>
.IP \(bu 3
intelpci_force_mmio=<0|1> (default is 0=Disable)
This option can be used to force the ICH4/ICH5 and ICH6 controllers to
run in memory mapped mode to free up I/O address space.
.IP \(bu 3
ich_jacksense=<0|1> (default is 0)
Force use of jacksensing on some AD198x mixers.
.SH FILES
/usr/lib/oss/conf/oss_ich.conf Device configuration file.
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_sblive - Creative Labs Sound Blaster Live/Audigy family driver.
.SH DESCRIPTION
Open Sound System driver for Creative Labs Sound Blaster Live!, Audigy,
Audigy2, Audigy2-Value and sound cards.
The sblive driver supports:
o 8-48Khz Playback/Recording
o 8 or 16 bits
o SPDIF digital output and Input
o Multi channel 5.1 (Live!) and 7.1 (Audigy) output.
AC3 passthrough is only supported on Audigy series of the soundcards.
.SH OTHER SIMILAR CARDS
There are several Sound Blaster cards that are also called as Live or
Audigy. However these cards are based on entirely different hardware design
and they are not compatible with this driver.
.IP \(bu 3
Sound Blaster Live 5.1 card is used in some Dell machines but it's
driven by the emu10k1x driver.
.IP \(bu 3
Sound Blaster AudigyLS and Live 7.1 models are driven by the audigyls
driver of OSS.
.SH SBLIVE COMBO SPDIF AND AUDIO JACKS
Most models of Live! and Audigy cards have an orange combo jack that is
used both for the analog center/LFE output and for digital DIN (S/PDIF)
output. The output mode is selected by a driver configuration option
(seel below) which should be set to proper value depending on the actual
speaker configuration.
.IP \(bu 3
Noisy analog center/LFE output. The orange combo jack at the rear plate
of the Live/Audigy card is shared between the digital DIN and the analog
center/LFE outputs. In digital DIN mode (default) you will hear very noisy
output from the speakers connected to this output jack. If you have analog
center/LFE (subwoofer) speakers connected then you need to turn off the
sblive_digital_din (or audigy_digital_din) option.
.IP \(bu 3
There is a new configuration option to enable/disable the "digital DIN"
output. By default the digital DIN interface is enabled which disables the
center/LFE analog output (uses the same combo jack). By setting the
sblive_digital_din (or audigy_digital_din) option to 0 you can enable the
analog C/LFE output feature. When digital DIN is disabled you can still get
S/PDIF (or AC3) output from the digital (optical/coax) outputs of the
optional livedrive unit.
.SH SBLIVE MIXER
SB Live cards have actually two mixer chips. In OSS both of them are
controlled together. However only limited set of features can be controlled
using ordinary mixer programs (such as the mixer applet included in OSS).
Majority of features can only be accessed using the ossmix and ossxmix
programs included in OSS.
The AC97 mixer is used to control volumes of the back bracket inputs (mic and
line in) and the _analog_ CD input connector on the soundcard. The 'mic'
volume controls the level of the rear bracket microphone input sent directly
to the front (only front) speakers. The 'line' and 'cd' controls do the same
for the back bracket line in connector and the on board analog CD input
connector. It's usually recommended to set these volumes to 0.
Another function of the AC97 mixer is selecting the signal that is passed to
the master mixer (for example for recording). One of the 'mic', 'line' or
'cd' signals can be routed to the master mixer by selecting that device as
the recording source in the AC97 mixer. The 'rec' volume control slider can
be used to adjust the signal strength. The 'igain' control doesn't usually
have any effect but some hardware revisions may use it for controlling the
microphone recording level.
.SH SBLIVE MASTER MIXER
Other mixer functions are handled by the DSP engine of the EMU 10k1 chip.
Most input signals (including all digital signals and LiveDrive inputs).
There are only two master mixer settings that can be controlled using all
mixer programs. The 'vol' setting is the master output volume that affects
both the front and rear speakers and the headphone output (digital output
volumes are not affected). The 'pcm' setting controls volumes of all PCM
playback channels (/dev/dsp#).
In addition to volume sliders most inputs have a stereo VU meter pair
(only in ossxmix) that can be used to monitor the input and to adjust the
input levels properly.
The master mixer consists of several sections that are:
.IP \(bu 3
Primary section: This section has two settings. The "spkmode" setting
selects how front/rear speakers are used for PCM playback (outputs from
programs using /dev/dsp#). The possible settings are FRONT, REAR and
FRONT+REAR. The default is FRONT+REAR. Change this setting if you like to
get PCM playback only from front or rear speakers. The "autoreset" flag is
used to control the "/dev" section.
.IP \(bu 3
"/dev" section: This section controls the volumes of each /dev/dsp# device
file supported by the device (there are 8 of them at this moment). These
volumes will return back to maximum every time the device is opened. However
this can be disabled by setting the 'autoreset' option to OFF. The ossxmix
program has special ability to show the application using the particular
/dev/dsp device (for layout reasons only the first 4 characters of the
program name are shown).
.IP \(bu 3
The equalizer section: This section controls the graphic equalizer for
front speakers only.
.IP \(bu 3
The front rear, and record sections: These three identical sections control
the levels of external inputs and PCM playback (/dev/dsp# devices) to be
sent to the front/rear speakers and to the recording device.
The CD Analog audio will only be heard from the FRONT speakers.
.SH SBLIVE RECORDING
Before recording anything you need to set the volumes in the recording
section properly. To enable recording from the AC97 connected inputs
(mic, line in and analog CD) use the AC97 mixer to select the desired input
and then tune the input level using the rec (and igain) setting.
Finally set the 'ac97' slider in the record section of the master mixer so
that the recording level is suitable.
The OSS drivers permit recording any application that's currently playing.
To record audio that's playing on any of the SB Live channels:
.IP \(bu 3
Turn down the AC97 control in the "record" section. This prevents any audio
being fed to the soundcard from MIC/Line-in/CD-in from getting mixed with
the audio produced by the application that's currently playing.
.IP \(bu 3
Type ossrecord -s<sampling rate> -b<bits/sample> -c<channels> test.wav
.IP \(bu 3
To stop recording press <Ctrl-c> and then you can play back the test.wav
file using ossplay command.
RECORDING ISSUES:
In most cases noise is caused by the microphone input or some other
(unused) input. Use the ossxmix program to turn off all unused inputs and
finally save the current mixer settings (see below).
Hint: Look at the VU meter panels of ossxmix. It's usually very easy to
locate the noise source by looking which input has some signal coming from
it.
WARNING! If you turn off some of the signals in recording section or the
AC97 mixer section this affects all subsequent recordings. Remember
to raise the volume prior doing any recording. After that decrease
the volumes again if necessary.
.SH SBLIVE HARDWARE MIXING
You can use /dev/oss/oss_sblive0/ pcm0-pcm7 to play multiple audio programs
using the hardware mixing.
Simply specify the device name with the application. A simple test is
to do the following:
ossplay -d/dev/oss/oss_sblive0/pcm0 <file1.wav> &
ossplay -d/dev/oss/oss_sblive0/pcm1 <file2.wav> &
ossplay -d/dev/oss/oss_sblive0/pcm2 <file3.wav> &
You should hear all three wav files playing simultaneously.
NOTE: Some apps may desire the old /dev/dspN names. e.g. /dev/dsp0 - /dev/dsp7.
NOTE: You can increase the number of output devices from the standard 8 devices
to 32 device. For this, run soundconf, select Set configuration options and
look for the entry "sblive_device", now type any number between 1 and 32
for the number of channels you wish. You can also do this manually by editing
oss_sblive.conf and inserting sblive_devices=XX entry,
e.g.: sblive_devices=27
.SH CDROM CONNNECTIONS
There are two alternative ways to connect audio signal from a CD-ROM drive tor
the SB Live soundcard. You can use a (three wire) analog cable or a (2 wire)
digital cable. OSS now supports both of these choices. Note that there are
separate mixer settings for both of these connections.
The analog CD-ROM wire is connected to the AC97 code chip and this method
works in most cases. To route the analog CD -input to the (front) speakers
you need to raise the volume of the 'cd' control in mixer. However if you
like to hear the analog CD input both from the front and rears speakers you
need to do this in slightly different way (please read the description of
the mixer above).
The digital connection works only with CD-ROM drives that has support for it.
Note that some CD-ROM drives having this digital output connector use a
different signal level than the one required by SB Live. This means that the
digital connection doesn't work with all CD-ROM drives (no sound). If you
have problems with the digital connection you should use the analog one.
When using the the digital CD input you may need to adjust the 'digcd'
volumes using ossxmix (or ossmix).
It should be noted that SB Live works internally at 48 kHz. This means that
all S/PDIF input signals are automatically sample rate converted to 48 kHz.
If you record from a 44.1 kHz (CD-ROM) and save the result to a 44.1 kHz
file the signal will be sample rate converted twice. First from the 44.1 kHz
input to internal 48 kHz and then back to 44.1 kHz. While the sample rate
converter of SB Live is very precise this will cause some change. This should
not be any problem when doing audio recordings but it may cause unwanted
results when transferring digital data (such as AC3/DTS) using the S/PDIF
the interface.
.SH CONFIGURATION OPTIONS
.IP \(bu 3
sblive_digital_din=<0|1> - This option is to enable/disable the "digital DIN"
output of SB Live. By default the digital DIN interface is disabled which
enables the center/LFE analog output (uses the same combo jack). By
setting the sblive_digital_din option to 0 you can enable the analog
Center/LFE output feature. When digital DIN is disabled you can still
get S/PDIF (or AC3) output from the digital (optical/coax) outputs of the
optional livedrive unit. Default: 0=analog output.
.IP \(bu 3
audigy_digital_din=<0|1> - same as "sblive_digital_din" option except for
the Audigy soundcards. Default: 1=digital output.
.IP \(bu 3
sblive_devices=<1..32> - Number of audio devices to be configured.
.SH LIMITATION
.IP \(bu 3
SB Live! devices will not work in Sparc systems due to PCI addressing
limitations. Only Audigy/Audigy2 models work under Sparc.
.IP \(bu 3
EMU Wavetable MIDI synthesizer is not supported
.IP \(bu 3
AC3 passthrough only supported on Audigy/Audigy but not on SB Live! devices.
.SH FILES
/usr/lib/oss/conf/oss_sblive.conf Device configuration file
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_sbpci - Creative Labs ES1371 audio driver.
.SH DESCRIPTION
Open Sound System driver for Creative Labs ES1371/ES1373/5880, Ectiva 1938
audio controllers.
APCI97 device characteristics:
o 8/16 bit playback/record
o mono/stereo playback/recording
o 8KHz to 48Khz sample rate
APCI97 MIXER EXTENSIONS
Dual Dac mode: This feature turns the APCI97 into two output devices with
the output going to front and rear speakers independantly (however volume
control is global).
Speaker Mode: This feature allows you to either have the audio coming out
the front speakers or you can have audio duplicated on rear speakers. This
mode is disabled when Dual Dac mode is enabled.
SPDIF: This button enables or disables SPDIF output.
.SH OPTIONS
.IP \(bu 3
apci97_latency=<NNN>
Certain models of the ES1371 sound devices will sound distorted playing stereo
audio and setting the PCI latency fixes the problem
.IP \(bu 3
apci_spdif=0|1
Certain models like the SB 4.1D/SB PCI128D have SPDIF output jacks and
this setting enables the output device.
.SH FILES
/usr/lib/oss/conf/oss_sbpci.conf Device configuration file
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_sbxfi - SoundBlaster X-Fi audio driver
.SH DESCRIPTION
Open Sound System driver for the SoundBlaster X-Fi cards.
.SH OPTIONS
.IP \(bu 3
sbxfi_type Override X-Fi type autodetection. Values:
0 - Autodetect type
1 - Sound Blaster X-Fi (SB046x/067x/076x)
2 - Sound Blaster X-Fi (SB073x)
3 - Sound Blaster X-Fi (SB055x)
4 - Sound Blaster X-Fi (UAA)
Default : 0.
.SH FILES
/usr/lib/oss/conf/oss_sbxfi.conf Device configuration file.
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_solo - ESS Solo-1 audio driver
.SH DESCRIPTION
Open Sound System driver for ESS Solo1/1938/1968 audio controllers.
ESS Solo1 device characteristics:
o 8/16 bit playback/record
o mono/stereo playback/recording
o 8KHz to 48Khz sample rate.
.SH OPTIONS
None
.SH FILES
/usr/lib/oss/conf/oss_solo.conf Device configuration file
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_trident - SiS7018, 4Dwave, ALIM5451 audio driver.
.SH DESCRIPTION
Open Sound System driver for Trident 4DWave DX/NX, SiS7018 and ALI5451 audio
controllers.
Trident device characteristics:
o 8/16 bit playback/record
o mono/stereo playback/recording
o 8KHz to 48Khz sample rate.
o Upto 8 hardware channels to mixing audio streams
.SH OPTIONS
trident_mpu_ioaddr=<xxx>
Set the MPU I/O address. Refer to the driver.conf file for valid addresses.
.SH LIMITATIONS
.IP \(bu 3
Due to PCI addressing limitations any add-on cards based on these chips
will not work under Sparc. The only exception is the ALI5451 chip that is
used on the main boards on many Sparc based systems.
.SH FILES
/usr/lib/oss/conf/oss_trident.conf Device configuration file
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_via823x - Open Sound System driver for VIA 8233/8235/8237 audio controllers.
.SH DESCRIPTION
VIA823x device characteristics:
o 8/16 bit playback/record
o mono/stereo/4/5.1 (Dolby) playback
o mono/stereo recording
o 8KHz to 48Khz sample rate.
o SPDIF input/output capability based on AC97 codec attech to the
VIA823x controller.
.SH OPTIONS
None
.SH FILES
/usr/lib/oss/conf/oss_via823x.conf Device configuration
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_via97 - VIA 82C686 audio driver
.SH DESCRIPTION
Open Sound System driver for VIA 82C686A and 82C686B audio controllers.
VIA97 device characteristics:
o 8/16 bit playback/record
o mono/stereo playback/recording
o 8KHz to 48Khz sample rate.
.SH OPTIONS
None
.SH FILES
/usr/lib/oss/conf/oss_via97.conf Device configuration file.
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
oss_ymf7xx - Yamaha DS-XG audio driver.
.SH DESCRIPTION
Open Sound System driver for Yamaha DS-XG YMF724/740/744/754 audio controllers.
ymf7xx device characteristics:
o 8/16 bit playback/record
o mono/stereo/4 channel playback
o mono/stereo recording
o 8KHz to 48Khz sample rate.
.SH MIXER
The Yamaha DSXG models 744 and 754 supports SPDIF and AC3 multichannel output
and the Mixer extentions will allow you to enable/disable SPDIF output.
.SH CONFIG OPTIONS
.IP \(bu 3
yamaha_mpu_ioaddr=<xxx>
MPU I/O address. Refer to the device conf file (see below) for legal values.
.IP \(bu 3
yamaha_mpu_irq=<xx>
MPU IRQ. Refer to device conf file (see below) for legal values
.IP \(bu 3
yamaha_fm_ioaddr=<xxx>
Yamaha FM SYnthesizer IO address. Refer to driver conf file (see below) for
possible values.
.SH FILES
/usr/lib/oss/conf/oss_ymf7xx.conf Device configuration file
.SH AUTHOR
4Front Technologies

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.TH 7 "August 31, 2006" "OSS" "OSS Devices"
.SH NAME
osscore - Open Sound Sytem core audio framework.
.SH DESCRIPTION
Open Sound System core audio support psudo driver. This driver implements the core Open Sound System API for audio, midi, mixer and synth functions. This driver also implements the OS driver interface as well as device configuration and setup.
More information on programming on the OSS API is avaialable at:
http://manuals.opensound.com/
.SH OPTIONS
.IP \(bu 3
ac97_recselect When set to 1 this option enables independent
recording source selection for the left and the right channel
on AC97 devices. In this way it's possible to record audio
streams so that (for example) the left channel signal comes
from the microphone and
the right channel signal from line-in. However when this
option is enabled it's only possible to select the recording
source by using a fully OSS 4.0 compatible mixer program such
as ossxmix.
Default: 0 - recording source is common to both channels.
.IP \(bu 3
ac97_amplifier When set to 1 this option enables the speaker power
amplifier feature of AC97 codec (if available).
Some boards have this inverted, so this feature can be
disabled by setting this option to 0.
Affects all AC97 based audio devices in the system.
Default: -1=autodetect correct setting.
.IP \(bu 3
cooked_enable By default OSS will let applications to use sampling
rates and formats that are not supported by the hardware.
Instead OSS performs the necessary format conversions in
software. Applications that don't tolerate this kind of
conversions usually disable them by using features of the OSS
API (SNDCTL_DSP_COOKEDMODE). If this option is set to 0 then
the format conversions will be disabled for all applications
and devices unless the application explicitly enables them.
Default: 1 - conversions are enabled.
This option should not be changed without very strong reasons.
.IP \(bu 3
detect_trace Internal debugging (do not change). Default: 0
.IP \(bu 3
dma_buffsize By default OSS will allocate audio DMA buffers with some
system dependent default size (usually 64k but sometimes
smaller). It is possible to change this default allocation by
setting this option. Value of 0 means that the default size
will be used. Value between 16 and 128 (kilobytes) can be used
if the default size is not suitable for some reason.
Default: 0 - system dependent buffsize.
This option mustn't be changed unless it's absolutely necessary.
.IP \(bu 3
max_intrate Set the maximum number of interrupts per second.
Default: 100 interrupts per second which equals to about
10 msec minimum latencies.
.IP \(bu 3
vmix_disabled The virtual mixer subsystem can be disabled by setting
this configuration option to 1.
Default: 0 - virtual mixer is enabled.
.IP \(bu 3
vmix_loopdevs Optionally the virtual mixer subsystem can create
special loopback audio devices that can be used to record the
output mix sent to the device. This option tells how many
loopback devices will be created (0, 1 or 2). If there are
multiple audio devices in the system the all of them will have
the same number of loopback devices.
Default: 0 - no loopback devices are created.
This setting should be left to 0 unless there are specific
reasons to enable the loopback devices.
.IP \(bu 3
vmix_no_autoattach By default (0) the low level
drivers for most sound cards will automatically
attach virtual mixer (vmix) to the primary audio devices of the cards.
In some situations it may be necessary to attach virtual mixer using
nonstandard parameters. If vmix_no_autoattach is set to 1 then user
can use vmixctl attach command to attach virtual mixer manually to
the device(s).
Default: 0 - Automatically attach virtual mixer.
.SH FILES
/usr/lib/oss/conf/osscore.conf Device configuration file
.SH AUTHOR
4Front Technologies

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.TH 8 "August 31, 2006" "OSS" "System Administration Commands"
.SH NAME
ossinfo - Open Sound System legacy device management utility.
.SH DESCRIPTION
The ossdevlinks utility creates and manages old style (legacy) device files
for OSS audio, MIDI and mixer devices.
In previous versioms OSS used "flat" device numbering for the device files
(for example /dev/dsp0 to /dev/dspN). OSS version 4.0 and later uses different
naming scheme. The ossdevlinks utility is used to manage the legacy device
names as symbolic links to the new style devices.
.SH OPTIONS
Normally ossdevlinks is used without command line arguments. However
there are few command line options.
-v Verbose output
-r Reset the legacy device numbering (do not use).
The -r option will invalidate audio device selections in the setup files
of various applications. This is considered highly undesirable. Applications
using wrong audio devices may cause serious security and privacy problems.
For this reason the -r option should never be used unless there are no other
ways to recover from serious audio/sound related problems. After that users
should review the audio settings of all the audio applications they are using.
.SH
.SH FILES
/usr/sbin/ossdevlinks
/usr/lib/oss/etc/legacy_devices
/dev/dspN
/dev/midiNN
/dev/mixerN
.SH AUTHOR
4Front Technologies

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.TH 8 "August 31, 2006" "OSS" "System Administration Commands"
.SH NAME
savemixer - Open Sound System program for saving and restoring mixer settings.
.SH SYNOPSIS
savemixer [-LVv] [-f <fname>]
.SH DESCRIPTION
The savemixer program saves mixer settings. It can also load saved mixer
settings back into the mixer.
Running this program without any parameters will save the current mixer
settings into /etc/oss/mixer.save or $OSSLIBDIR/etc/mixer.save file.
OSSLIBDIR is decided by reading /etc/oss.conf, and defaults to /usr/lib/oss.
.SH AUTOMATIC SAVE
By default the soundoff command will automatically run savemixer to save
the active mixer settings. See the manual page for soundoff(1) if you
like to turn this feature off.
.SH OPTIONS
-f<fname> Use <fname> as setting file.
-L Loads saved mixer and device map information from mixer.save.
-V Version information.
-v Verbose output.
.SH SEE ALSO
soundoff(1), soundon(1), ossdetect(1), ossdevlinks(1), ossmix(1), ossxmix(1)
.SH FILES
/etc/oss.conf
/usr/sbin/savemixer
/usr/lib/oss/etc/mixer.save
/usr/lib/oss/etc/dspdevs.map
/usr/lib/oss/etc/applist.conf
.SH AUTHOR
4Front Technologies

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.TH 8 "August 31, 2006" "OSS" "System Administration Commands"
.SH NAME
vmixctl - Open Sound System utility to control the vmix subsystem.
.SH SYNOPSIS
o vmixctl attach [attach_options...] audiodev [inputdev]
o vmixctl detach [attach_options...] audiodev
o vmixctl rate audiodev samplerate
.SH DESCRIPTION
The vmixctl program can be used to attach or detach the virtual mixer subsystem
(vmix) to/from audio devices. In addition it can be used to control vmix
related parameters such as the sampling rate to be used with the device.
By default most OSS drivers will attach virtual mixer to the primary audio device
of the sound card (or motherboard audio chip) when the device is attached.
However possible secondary audio devices (engines) will not have vmix
attached by default. In additional professional audio devices will be
attached without vmix because mixing may cause some unwanted distortion
to the signal.
.SH ATTACHING VMIX TO AN AUDIO DEVICE
There are two forms of vmixctl attach command:
o vmixctl attach audiodev
This alternative is to be used with devices that support only output or
have a single audio device file that supports full duplex.
o vmixctl attach audiodev inputdev
The second form is to be used with devices that have separate output and
input device files. The "audiodev" parameter defines the output device and
the "inputdev" parameter is the device file to be used for input direction.
Note that both device files must belong to the same "physical" sound card.
In some cases it might be possible to use one sound card for playback and
another for recording. However this configuration is not supported and the
result may not be functional.
To find out the right device file names (audiodev and inputdev) you can use
the "ossinfo -a" command.
.SH ATTACH OPTIONS
o -r Disable recording functionality. By default vmix will suppor
recording if the master device(s) support it.
o -p Do not preallocate client engines. By default vmix will
preallocate first 4 (out of 8) client engines when attaching
to the device. The remaining engines will be allocated
on-demand if there are more concurrent applications that
use the device.
o -V Make client devices visible (have private device nodes under /dev).
o -c <n> Preallocate <n> client engines instead of 4. However -p
option makes this option ineffective.
.SH EXAMPLES
o vmixctl attach /dev/oss/oss_envy240/pcm0
o vmixctl attach /dev/oss/oss_envy240/pcm0 /dev/oss/oss_envy240/pcmin0
.SH SETTING THE SAMPLING RATE USED BY VMIX
The virtual mixer subsystem will set the physical audio devce(s) to use
fixed sampling rate that is 48000 Hz by default. It is possible to use
"vmixctl rate audiodev" to switch vmix to use some different rate with this
device (pair). You should use "ossinfo -a -v2" to verify that the sampling rate
is actually supported by the device. Otherwise the actual device may enforce
vmix to use the nearest supported rate (or some default rate).
The "audiodev" parameter is the device file name (see ossinfo -a) that is
used for playback. The input device name doesn't need to be specified.
Note that some professional audio devices may be locked to external sampling
rate or some fixed rate (defined in ossmix/ossxmis). In such case the rate is
not changeable by vmixctl.
.SH EXAMPLE
o vmixctl rate /dev/oss/oss_envy240/pcm0
.SH DETACHING VMIX FROM AN AUDIO DEVICE
It is possible to detach vmix from an audio device if it causes problems
with applications by using "vmix detach audiodev".
It is not possible to detach and (re)attach vmix to the same device more
than few times. Use the vmix-enable setting in the control panel
(ossxmix or ossmix) to disable/re-enable vmix if you need to do it
repeatedly. Use vmix detach only if you need to attach virtual mixer using
different parameters.
.SH EXAMPLE
o vmix detach /dev/oss/oss_envy240/pcm0
.SH POSSIBLE BUGS
o The control panel elements related with vmix are not removed from the
mixer API when vmix is detached. This may be somehow confusing.
.SH SEE ALSO
soundoff(1), soundon(1), ossmix(1), ossxmix(1)
.SH FILES
/usr/sbin/vmixct
.SH AUTHOR
4Front Technologies