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author | Takashi Iwai <tiwai@suse.de> | 2016-11-09 16:18:14 +0100 |
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committer | Takashi Iwai <tiwai@suse.de> | 2016-11-10 18:09:26 +0100 |
commit | 0c266c4b707e263ac74e0bd4330b4193fa46c434 (patch) | |
tree | 98a0bb57b23259e45324a8c85ba92e08567d50a2 /Documentation/sound/designs | |
parent | efe541c230e41106ce912e096e19518630e810db (diff) | |
download | linux-0c266c4b707e263ac74e0bd4330b4193fa46c434.tar.gz linux-0c266c4b707e263ac74e0bd4330b4193fa46c434.tar.bz2 linux-0c266c4b707e263ac74e0bd4330b4193fa46c434.zip |
ALSA: doc: ReSTize OSS-Emulation document
A simple conversion from a text file.
Put to designs subdirectory.
Signed-off-by: Takashi Iwai <tiwai@suse.de>
Diffstat (limited to 'Documentation/sound/designs')
-rw-r--r-- | Documentation/sound/designs/index.rst | 1 | ||||
-rw-r--r-- | Documentation/sound/designs/oss-emulation.rst | 336 |
2 files changed, 337 insertions, 0 deletions
diff --git a/Documentation/sound/designs/index.rst b/Documentation/sound/designs/index.rst index dd4fcbcb2b92..5b3033c556d0 100644 --- a/Documentation/sound/designs/index.rst +++ b/Documentation/sound/designs/index.rst @@ -7,3 +7,4 @@ Designs and Implementations channel-mapping-api procfile powersave + oss-emulation diff --git a/Documentation/sound/designs/oss-emulation.rst b/Documentation/sound/designs/oss-emulation.rst new file mode 100644 index 000000000000..e8dcb9633e7b --- /dev/null +++ b/Documentation/sound/designs/oss-emulation.rst @@ -0,0 +1,336 @@ +============================= +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. |