A simple conversion from a text file. Put to designs subdirectory.
Signed-off-by: Takashi Iwai tiwai@suse.de --- Documentation/sound/designs/index.rst | 1 + .../oss-emulation.rst} | 169 ++++++++++++--------- 2 files changed, 101 insertions(+), 69 deletions(-) rename Documentation/sound/{alsa/OSS-Emulation.txt => designs/oss-emulation.rst} (70%)
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/alsa/OSS-Emulation.txt b/Documentation/sound/designs/oss-emulation.rst similarity index 70% rename from Documentation/sound/alsa/OSS-Emulation.txt rename to Documentation/sound/designs/oss-emulation.rst index 152ca2a3f1bd..e8dcb9633e7b 100644 --- a/Documentation/sound/alsa/OSS-Emulation.txt +++ b/Documentation/sound/designs/oss-emulation.rst @@ -1,7 +1,8 @@ - NOTES ON KERNEL OSS-EMULATION - ============================= +============================= +Notes on Kernel OSS-Emulation +=============================
- Jan. 22, 2004 Takashi Iwai tiwai@suse.de +Jan. 22, 2004 Takashi Iwai tiwai@suse.de
Modules @@ -14,18 +15,18 @@ 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 +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 +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. +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 +``alias sound-slot-0 snd-card-0`` doesn't work any more like the old modutils).
The currently available OSS configuration is shown in @@ -42,6 +43,7 @@ Device Mapping ==============
ALSA supports the following OSS device files: +::
PCM: /dev/dspX @@ -61,48 +63,55 @@ ALSA supports the following OSS device files: 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.) +/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 +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) +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, +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) +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
@@ -118,43 +127,52 @@ 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 +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 +``app_name`` is the name of application with (higher priority) or without path. -fragments specifies the number of fragments or zero if no specific +``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 +``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 +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 +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
@@ -171,10 +189,11 @@ 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. +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 @@ -183,15 +202,18 @@ 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 +Also, for clearing all the current configuration, send ``erase`` command as below: +::
% echo "erase" > /proc/asound/card0/pcm0p/oss
@@ -211,40 +233,43 @@ 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 +==================== ===================== ===== +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. +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 @@ -261,6 +286,7 @@ 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
@@ -284,12 +310,18 @@ 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.
@@ -302,4 +334,3 @@ 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. -