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