NOTES ON KERNEL OSS-EMULATION
=============================
Jan. 22, 2004 Takashi Iwai <tiwai@suse.de>
Modules
=======
ALSA provides a powerful OSS emulation on the kernel.
The OSS emulation for PCM, mixer and sequencer devices is implemented
as add-on kernel modules, snd-pcm-oss, snd-mixer-oss and snd-seq-oss.
When you need to access the OSS PCM, mixer or sequencer devices, the
corresponding module has to be loaded.
These modules are loaded automatically when the corresponding service
is called. The alias is defined sound-service-x-y, where x and y are
the card number and the minor unit number. Usually you don't have to
define these aliases by yourself.
Only necessary step for auto-loading of OSS modules is to define the
card alias in /etc/modprobe.d/alsa.conf, such as
alias sound-slot-0 snd-emu10k1
As the second card, define sound-slot-1 as well.
Note that you can't use the aliased name as the target name (i.e.
"alias sound-slot-0 snd-card-0" doesn't work any more like the old
modutils).
The currently available OSS configuration is shown in
/proc/asound/oss/sndstat. This shows in the same syntax of
/dev/sndstat, which is available on the commercial OSS driver.
On ALSA, you can symlink /dev/sndstat to this proc file.
Please note that the devices listed in this proc file appear only
after the corresponding OSS-emulation module is loaded. Don't worry
even if "NOT ENABLED IN CONFIG" is shown in it.
Device Mapping
==============
ALSA supports the following OSS device files:
PCM:
/dev/dspX
/dev/adspX
Mixer:
/dev/mixerX
MIDI:
/dev/midi0X
/dev/amidi0X
Sequencer:
/dev/sequencer
/dev/sequencer2 (aka /dev/music)
where X is the card number from 0 to 7.
(NOTE: Some distributions have the device files like /dev/midi0 and
/dev/midi1. They are NOT for OSS but for tclmidi, which is
a totally different thing.)
Unlike the real OSS, ALSA cannot use the device files more than the
assigned ones. For example, the first card cannot use /dev/dsp1 or
/dev/dsp2, but only /dev/dsp0 and /dev/adsp0.
As seen above, PCM and MIDI may have two devices. Usually, the first
PCM device (hw:0,0 in ALSA) is mapped to /dev/dsp and the secondary
device (hw:0,1) to /dev/adsp (if available). For MIDI, /dev/midi and
/dev/amidi, respectively.
You can change this device mapping via the module options of
snd-pcm-oss and snd-rawmidi. In the case of PCM, the following
options are available for snd-pcm-oss:
dsp_map PCM device number assigned to /dev/dspX
(default = 0)
adsp_map PCM device number assigned to /dev/adspX
(default = 1)
For example, to map the third PCM device (hw:0,2) to /dev/adsp0,
define like this:
options snd-pcm-oss adsp_map=2
The options take arrays. For configuring the second card, specify
two entries separated by comma. For example, to map the third PCM
device on the second card to /dev/adsp1, define like below:
options snd-pcm-oss adsp_map=0,2
To change the mapping of MIDI devices, the following options are
available for snd-rawmidi:
midi_map MIDI device number assigned to /dev/midi0X
(default = 0)
amidi_map MIDI device number assigned to /dev/amidi0X
(default = 1)
For example, to assign the third MIDI device on the first card to
/dev/midi00, define as follows:
options snd-rawmidi midi_map=2
PCM Mode
========
As default, ALSA emulates the OSS PCM with so-called plugin layer,
i.e. tries to convert the sample format, rate or channels
automatically when the card doesn't support it natively.
This will lead to some problems for some applications like quake or
wine, especially if they use the card only in the MMAP mode.
In such a case, you can change the behavior of PCM per application by
writing a command to the proc file. There is a proc file for each PCM
stream, /proc/asound/cardX/pcmY[cp]/oss, where X is the card number
(zero-based), Y the PCM device number (zero-based), and 'p' is for
playback and 'c' for capture, respectively. Note that this proc file
exists only after snd-pcm-oss module is loaded.
The command sequence has the following syntax:
app_name fragments fragment_size [options]
app_name is the name of application with (higher priority) or without
path.
fragments specifies the number of fragments or zero if no specific
number is given.
fragment_size is the size of fragment in bytes or zero if not given.
options is the optional parameters. The following options are
available:
disable the application tries to open a pcm device for
this channel but does not want to use it.
direct don't use plugins
block force block open mode
non-block force non-block open mode
partial-frag write also partial fragments (affects playback only)
no-silence do not fill silence ahead to avoid clicks
The disable option is useful when one stream direction (playback or
capture) is not handled correctly by the application although the
hardware itself does support both directions.
The direct option is used, as mentioned above, to bypass the automatic
conversion and useful for MMAP-applications.
For example, to playback the first PCM device without plugins for
quake, send a command via echo like the following:
% echo "quake 0 0 direct" > /proc/asound/card0/pcm0p/oss
While quake wants only playback, you may append the second command
to notify driver that only this direction is about to be allocated:
% echo "quake 0 0 disable" > /proc/asound/card0/pcm0c/oss
The permission of proc files depend on the module options of snd.
As default it's set as root, so you'll likely need to be superuser for
sending the command above.
The block and non-block options are used to change the behavior of
opening the device file.
As default, ALSA behaves as original OSS drivers, i.e. does not block
the file when it's busy. The -EBUSY error is returned in this case.
This blocking behavior can be changed globally via nonblock_open
module option of snd-pcm-oss. For using the blocking mode as default
for OSS devices, define like the following:
options snd-pcm-oss nonblock_open=0
The partial-frag and no-silence commands have been added recently.
Both commands are for optimization use only. The former command
specifies to invoke the write transfer only when the whole fragment is
filled. The latter stops writing the silence data ahead
automatically. Both are disabled as default.
You can check the currently defined configuration by reading the proc
file. The read image can be sent to the proc file again, hence you
can save the current configuration
% cat /proc/asound/card0/pcm0p/oss > /somewhere/oss-cfg
and restore it like
% cat /somewhere/oss-cfg > /proc/asound/card0/pcm0p/oss
Also, for clearing all the current configuration, send "erase" command
as below:
% echo "erase" > /proc/asound/card0/pcm0p/oss
Mixer Elements
==============
Since ALSA has completely different mixer interface, the emulation of
OSS mixer is relatively complicated. ALSA builds up a mixer element
from several different ALSA (mixer) controls based on the name
string. For example, the volume element SOUND_MIXER_PCM is composed
from "PCM Playback Volume" and "PCM Playback Switch" controls for the
playback direction and from "PCM Capture Volume" and "PCM Capture
Switch" for the capture directory (if exists). When the PCM volume of
OSS is changed, all the volume and switch controls above are adjusted
automatically.
As default, ALSA uses the following control for OSS volumes:
OSS volume ALSA control Index
-----------------------------------------------------
SOUND_MIXER_VOLUME Master 0
SOUND_MIXER_BASS Tone Control - Bass 0
SOUND_MIXER_TREBLE Tone Control - Treble 0
SOUND_MIXER_SYNTH Synth 0
SOUND_MIXER_PCM PCM 0
SOUND_MIXER_SPEAKER PC Speaker 0
SOUND_MIXER_LINE Line 0
SOUND_MIXER_MIC Mic 0
SOUND_MIXER_CD CD 0
SOUND_MIXER_IMIX Monitor Mix 0
SOUND_MIXER_ALTPCM PCM 1
SOUND_MIXER_RECLEV (not assigned)
SOUND_MIXER_IGAIN Capture 0
SOUND_MIXER_OGAIN Playback 0
SOUND_MIXER_LINE1 Aux 0
SOUND_MIXER_LINE2 Aux 1
SOUND_MIXER_LINE3 Aux 2
SOUND_MIXER_DIGITAL1 Digital 0
SOUND_MIXER_DIGITAL2 Digital 1
SOUND_MIXER_DIGITAL3 Digital 2
SOUND_MIXER_PHONEIN Phone 0
SOUND_MIXER_PHONEOUT Phone 1
SOUND_MIXER_VIDEO Video 0
SOUND_MIXER_RADIO Radio 0
SOUND_MIXER_MONITOR Monitor 0
The second column is the base-string of the corresponding ALSA
control. In fact, the controls with "XXX [Playback|Capture]
[Volume|Switch]" will be checked in addition.
The current assignment of these mixer elements is listed in the proc
file, /proc/asound/cardX/oss_mixer, which will be like the following
VOLUME "Master" 0
BASS "" 0
TREBLE "" 0
SYNTH "" 0
PCM "PCM" 0
...
where the first column is the OSS volume element, the second column
the base-string of the corresponding ALSA control, and the third the
control index. When the string is empty, it means that the
corresponding OSS control is not available.
For changing the assignment, you can write the configuration to this
proc file. For example, to map "Wave Playback" to the PCM volume,
send the command like the following:
% echo 'VOLUME "Wave Playback" 0' > /proc/asound/card0/oss_mixer
The command is exactly as same as listed in the proc file. You can
change one or more elements, one volume per line. In the last
example, both "Wave Playback Volume" and "Wave Playback Switch" will
be affected when PCM volume is changed.
Like the case of PCM proc file, the permission of proc files depend on
the module options of snd. you'll likely need to be superuser for
sending the command above.
As well as in the case of PCM proc file, you can save and restore the
current mixer configuration by reading and writing the whole file
image.
Duplex Streams
==============
Note that when attempting to use a single device file for playback and
capture, the OSS API provides no way to set the format, sample rate or
number of channels different in each direction. Thus
io_handle = open("device", O_RDWR)
will only function correctly if the values are the same in each direction.
To use different values in the two directions, use both
input_handle = open("device", O_RDONLY)
output_handle = open("device", O_WRONLY)
and set the values for the corresponding handle.
Unsupported Features
====================
MMAP on ICE1712 driver
----------------------
ICE1712 supports only the unconventional format, interleaved
10-channels 24bit (packed in 32bit) format. Therefore you cannot mmap
the buffer as the conventional (mono or 2-channels, 8 or 16bit) format
on OSS.
