Author: pluto Date: Wed Jun 22 15:45:01 2005 GMT
Module: SOURCES Tag: LINUX_2_6
---- Log message:
- post 2.6.12 fixes.
---- Files affected:
SOURCES:
linux-2.6-alsa-1.0.9plus-20050622.patch (NONE -> 1.1.2.1) (NEW)
---- Diffs:
================================================================
Index: SOURCES/linux-2.6-alsa-1.0.9plus-20050622.patch
diff -u /dev/null SOURCES/linux-2.6-alsa-1.0.9plus-20050622.patch:1.1.2.1
--- /dev/null Wed Jun 22 17:45:01 2005
+++ SOURCES/linux-2.6-alsa-1.0.9plus-20050622.patch Wed Jun 22 17:44:56 2005
@@ -0,0 +1,22914 @@
+
+ Documentation/sound/alsa/ALSA-Configuration.txt | 127
+ Documentation/sound/alsa/CMIPCI.txt | 41
+ Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl | 6
+ Documentation/sound/alsa/emu10k1-jack.txt | 74
+ Documentation/sound/alsa/hdspm.txt | 362 +
+ include/sound/ac97_codec.h | 8
+ include/sound/asound.h | 16
+ include/sound/control.h | 2
+ include/sound/emu10k1.h | 42
+ include/sound/gus.h | 23
+ include/sound/hdspm.h | 131
+ include/sound/pcm.h | 32
+ include/sound/seq_midi_event.h | 2
+ include/sound/seq_virmidi.h | 1
+ include/sound/timer.h | 2
+ include/sound/version.h | 4
+ sound/Kconfig | 5
+ sound/arm/Kconfig | 6
+ sound/arm/Makefile | 3
+ sound/arm/aaci.c | 968 ++
+ sound/arm/aaci.h | 246
+ sound/arm/devdma.c | 81
+ sound/arm/devdma.h | 3
+ sound/core/control.c | 4
+ sound/core/memalloc.c | 201
+ sound/core/oss/pcm_oss.c | 33
+ sound/core/oss/pcm_plugin.c | 5
+ sound/core/pcm.c | 3
+ sound/core/pcm_lib.c | 52
+ sound/core/pcm_memory.c | 1
+ sound/core/pcm_misc.c | 16
+ sound/core/pcm_native.c | 74
+ sound/core/seq/oss/seq_oss_synth.c | 24
+ sound/core/seq/seq_dummy.c | 5
+ sound/core/seq/seq_midi.c | 2
+ sound/core/seq/seq_midi_event.c | 6
+ sound/core/seq/seq_queue.c | 3
+ sound/core/seq/seq_queue.h | 1
+ sound/core/seq/seq_timer.c | 3
+ sound/core/seq/seq_timer.h | 2
+ sound/core/seq/seq_virmidi.c | 8
+ sound/core/sound.c | 1
+ sound/core/timer.c | 100
+ sound/core/timer_compat.c | 5
+ sound/drivers/vx/vx_pcm.c | 12
+ sound/i2c/tea6330t.c | 3
+ sound/isa/Kconfig | 1
+ sound/isa/ad1816a/ad1816a.c | 2
+ sound/isa/cs423x/cs4236.c | 3
+ sound/isa/gus/gus_io.c | 14
+ sound/isa/gus/gus_main.c | 3
+ sound/isa/gus/gus_mem.c | 12
+ sound/isa/gus/gus_pcm.c | 3
+ sound/isa/gus/gus_reset.c | 3
+ sound/isa/gus/gus_synth.c | 3
+ sound/isa/gus/gus_tables.h | 4
+ sound/isa/gus/gus_volume.c | 8
+ sound/pci/Kconfig | 13
+ sound/pci/ac97/ac97_codec.c | 71
+ sound/pci/ac97/ac97_patch.c | 585 +
+ sound/pci/ac97/ac97_patch.h | 1
+ sound/pci/ali5451/ali5451.c | 283
+ sound/pci/als4000.c | 4
+ sound/pci/atiixp.c | 6
+ sound/pci/atiixp_modem.c | 42
+ sound/pci/au88x0/au88x0.c | 2
+ sound/pci/azt3328.c | 2
+ sound/pci/bt87x.c | 2
+ sound/pci/ca0106/ca0106.h | 72
+ sound/pci/ca0106/ca0106_main.c | 211
+ sound/pci/ca0106/ca0106_mixer.c | 76
+ sound/pci/ca0106/ca0106_proc.c | 31
+ sound/pci/cmipci.c | 159
+ sound/pci/cs4281.c | 10
+ sound/pci/cs46xx/cs46xx.c | 2
+ sound/pci/cs46xx/cs46xx_lib.c | 3
+ sound/pci/emu10k1/emu10k1.c | 2
+ sound/pci/emu10k1/emu10k1_main.c | 192
+ sound/pci/emu10k1/emu10k1x.c | 8
+ sound/pci/emu10k1/emufx.c | 56
+ sound/pci/emu10k1/emumixer.c | 14
+ sound/pci/emu10k1/emupcm.c | 6
+ sound/pci/emu10k1/emuproc.c | 89
+ sound/pci/emu10k1/irq.c | 46
+ sound/pci/emu10k1/p16v.c | 367 +
+ sound/pci/ens1370.c | 2
+ sound/pci/es1938.c | 2
+ sound/pci/es1968.c | 3
+ sound/pci/fm801.c | 3
+ sound/pci/hda/Makefile | 2
+ sound/pci/hda/hda_codec.c | 206
+ sound/pci/hda/hda_codec.h | 30
+ sound/pci/hda/hda_generic.c | 14
+ sound/pci/hda/hda_intel.c | 119
+ sound/pci/hda/hda_local.h | 37
+ sound/pci/hda/hda_patch.h | 3
+ sound/pci/hda/hda_proc.c | 56
+ sound/pci/hda/patch_analog.c | 693 +-
+ sound/pci/hda/patch_cmedia.c | 218
+ sound/pci/hda/patch_realtek.c | 2787 +++++---
+ sound/pci/hda/patch_sigmatel.c | 666 +
+ sound/pci/ice1712/amp.c | 30
+ sound/pci/ice1712/amp.h | 16
+ sound/pci/ice1712/ice1712.c | 2
+ sound/pci/ice1712/ice1712.h | 5
+ sound/pci/ice1712/ice1724.c | 2
+ sound/pci/ice1712/phase.c | 728 ++
+ sound/pci/ice1712/phase.h | 19
+ sound/pci/ice1712/vt1720_mobo.c | 9
+ sound/pci/ice1712/vt1720_mobo.h | 4
+ sound/pci/intel8x0.c | 156
+ sound/pci/intel8x0m.c | 80
+ sound/pci/korg1212/korg1212.c | 2
+ sound/pci/maestro3.c | 222
+ sound/pci/mixart/mixart.c | 2
+ sound/pci/nm256/nm256.c | 2
+ sound/pci/rme32.c | 2
+ sound/pci/rme96.c | 2
+ sound/pci/rme9652/Makefile | 2
+ sound/pci/rme9652/hdsp.c | 30
+ sound/pci/rme9652/hdspm.c | 3671
+++++++++++
+ sound/pci/rme9652/rme9652.c | 16
+ sound/pci/sonicvibes.c | 2
+ sound/pci/trident/trident.c | 5
+ sound/pci/via82xx.c | 145
+ sound/pci/via82xx_modem.c | 38
+ sound/pci/vx222/vx222.c | 2
+ sound/pci/ymfpci/ymfpci.c | 2
+ sound/pci/ymfpci/ymfpci_main.c | 35
+ sound/pcmcia/vx/vx_entry.c | 3
+ sound/synth/emux/emux_effect.c | 6
+ sound/usb/Kconfig | 1
+ sound/usb/usbaudio.c | 308
+ sound/usb/usbaudio.h | 11
+ sound/usb/usbmidi.c | 128
+ sound/usb/usbmixer.c | 588 +
+ sound/usb/usbmixer_maps.c | 126
+ sound/usb/usbquirks.h | 298
+ sound/usb/usx2y/usbusx2y.c | 2
+ sound/usb/usx2y/usbusx2yaudio.c | 6
+ 140 files changed, 13941 insertions(+), 2768 deletions(-)
+
+diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt
b/Documentation/sound/alsa/ALSA-Configuration.txt
+--- a/Documentation/sound/alsa/ALSA-Configuration.txt
++++ b/Documentation/sound/alsa/ALSA-Configuration.txt
+@@ -615,9 +615,11 @@ Prior to version 0.9.0rc4 options had a
+ Module snd-hda-intel
+ --------------------
+
+- Module for Intel HD Audio (ICH6, ICH6M, ICH7)
++ Module for Intel HD Audio (ICH6, ICH6M, ICH7), ATI SB450,
++ VIA VT8251/VT8237A
+
+ model - force the model name
++ position_fix - Fix DMA pointer (0 = FIFO size, 1 = none, 2 = POSBUF)
+
+ Module supports up to 8 cards.
+
+@@ -635,6 +637,10 @@ Prior to version 0.9.0rc4 options had a
+ 5stack 5-jack in back, 2-jack in front
+ 5stack-digout 5-jack in back, 2-jack in front, a SPDIF out
+ w810 3-jack
++ z71v 3-jack (HP shared SPDIF)
++ asus 3-jack
++ uniwill 3-jack
++ F1734 2-jack
+
+ CMI9880
+ minimal 3-jack in back
+@@ -642,6 +648,15 @@ Prior to version 0.9.0rc4 options had a
+ full 6-jack in back, 2-jack in front
+ full_dig 6-jack in back, 2-jack in front, SPDIF I/O
+ allout 5-jack in back, 2-jack in front, SPDIF out
++ auto auto-config reading BIOS (default)
++
++ Note 2: If you get click noises on output, try the module option
++ position_fix=1 or 2. position_fix=1 will use the SD_LPIB
++ register value without FIFO size correction as the current
++ DMA pointer. position_fix=2 will make the driver to use
++ the position buffer instead of reading SD_LPIB register.
++ (Usually SD_LPLIB register is more accurate than the
++ position buffer.)
+
+ Module snd-hdsp
+ ---------------
+@@ -660,7 +675,19 @@ Prior to version 0.9.0rc4 options had a
+ module did formerly. It will allocate the buffers in advance
+ when any HDSP cards are found. To make the buffer
+ allocation sure, load snd-page-alloc module in the early
+- stage of boot sequence.
++ stage of boot sequence. See "Early Buffer Allocation"
++ section.
++
++ Module snd-hdspm
++ ----------------
++
++ Module for RME HDSP MADI board.
++
++ precise_ptr - Enable precise pointer, or disable.
++ line_outs_monitor - Send playback streams to analog outs by default.
++ enable_monitor - Enable Analog Out on Channel 63/64 by default.
++
++ See hdspm.txt for details.
+
+ Module snd-ice1712
+ ------------------
+@@ -677,15 +704,19 @@ Prior to version 0.9.0rc4 options had a
+ * TerraTec EWS 88D
+ * TerraTec EWX 24/96
+ * TerraTec DMX 6Fire
++ * TerraTec Phase 88
+ * Hoontech SoundTrack DSP 24
+ * Hoontech SoundTrack DSP 24 Value
+ * Hoontech SoundTrack DSP 24 Media 7.1
++ * Event Electronics, EZ8
+ * Digigram VX442
++ * Lionstracs, Mediastaton
+
+ model - Use the given board model, one of the following:
+ delta1010, dio2496, delta66, delta44, audiophile, delta410,
+ delta1010lt, vx442, ewx2496, ews88mt, ews88mt_new, ews88d,
+- dmx6fire, dsp24, dsp24_value, dsp24_71, ez8
++ dmx6fire, dsp24, dsp24_value, dsp24_71, ez8,
++ phase88, mediastation
+ omni - Omni I/O support for MidiMan M-Audio Delta44/66
+ cs8427_timeout - reset timeout for the CS8427 chip (S/PDIF transciever)
+ in msec resolution, default value is 500 (0.5 sec)
+@@ -694,20 +725,46 @@ Prior to version 0.9.0rc4 options had a
+ is not used with all Envy24 based cards (for example in the MidiMan Delta
+ serie).
+
++ Note: The supported board is detected by reading EEPROM or PCI
++ SSID (if EEPROM isn't available). You can override the
++ model by passing "model" module option in case that the
++ driver isn't configured properly or you want to try another
++ type for testing.
++
+ Module snd-ice1724
+ ------------------
+
+- Module for Envy24HT (VT/ICE1724) based PCI sound cards.
++ Module for Envy24HT (VT/ICE1724), Envy24PT (VT1720) based PCI sound cards.
+ * MidiMan M Audio Revolution 7.1
+ * AMP Ltd AUDIO2000
+- * TerraTec Aureon Sky-5.1, Space-7.1
++ * TerraTec Aureon 5.1 Sky
++ * TerraTec Aureon 7.1 Space
++ * TerraTec Aureon 7.1 Universe
++ * TerraTec Phase 22
++ * TerraTec Phase 28
++ * AudioTrak Prodigy 7.1
++ * AudioTrak Prodigy 192
++ * Pontis MS300
++ * Albatron K8X800 Pro II
++ * Chaintech ZNF3-150
++ * Chaintech ZNF3-250
++ * Chaintech 9CJS
++ * Chaintech AV-710
++ * Shuttle SN25P
+
+ model - Use the given board model, one of the following:
+- revo71, amp2000, prodigy71, aureon51, aureon71,
+- k8x800
++ revo71, amp2000, prodigy71, prodigy192, aureon51,
++ aureon71, universe, k8x800, phase22, phase28, ms300,
++ av710
+
+ Module supports up to 8 cards and autoprobe.
+
++ Note: The supported board is detected by reading EEPROM or PCI
++ SSID (if EEPROM isn't available). You can override the
++ model by passing "model" module option in case that the
++ driver isn't configured properly or you want to try another
++ type for testing.
++
+ Module snd-intel8x0
+ -------------------
+
+@@ -1026,7 +1083,8 @@ Prior to version 0.9.0rc4 options had a
+ module did formerly. It will allocate the buffers in advance
+ when any RME9652 cards are found. To make the buffer
+ allocation sure, load snd-page-alloc module in the early
+- stage of boot sequence.
++ stage of boot sequence. See "Early Buffer Allocation"
++ section.
+
+ Module snd-sa11xx-uda1341 (on arm only)
+ ---------------------------------------
+@@ -1211,16 +1269,18 @@ Prior to version 0.9.0rc4 options had a
+ ------------------
+
+ Module for AC'97 motherboards based on VIA 82C686A/686B, 8233,
+- 8233A, 8233C, 8235 (south) bridge.
++ 8233A, 8233C, 8235, 8237 (south) bridge.
+
+ mpu_port - 0x300,0x310,0x320,0x330, otherwise obtain BIOS setup
+ [VIA686A/686B only]
+ joystick - Enable joystick (default off) [VIA686A/686B only]
+ ac97_clock - AC'97 codec clock base (default 48000Hz)
+ dxs_support - support DXS channels,
+- 0 = auto (defalut), 1 = enable, 2 = disable,
+- 3 = 48k only, 4 = no VRA
+- [VIA8233/C,8235 only]
++ 0 = auto (default), 1 = enable, 2 = disable,
++ 3 = 48k only, 4 = no VRA, 5 = enable any sample
++ rate and different sample rates on different
++ channels
++ [VIA8233/C, 8235, 8237 only]
+ ac97_quirk - AC'97 workaround for strange hardware
+ See the description of intel8x0 module for details.
+
+@@ -1232,18 +1292,23 @@ Prior to version 0.9.0rc4 options had a
+ default value 1.4. Then the interrupt number will be
+ assigned under 15. You might also upgrade your BIOS.
+
+- Note: VIA8233/5 (not VIA8233A) can support DXS (direct sound)
++ Note: VIA8233/5/7 (not VIA8233A) can support DXS (direct sound)
+ channels as the first PCM. On these channels, up to 4
+- streams can be played at the same time.
++ streams can be played at the same time, and the controller
++ can perform sample rate conversion with separate rates for
++ each channel.
+ As default (dxs_support = 0), 48k fixed rate is chosen
+ except for the known devices since the output is often
+ noisy except for 48k on some mother boards due to the
+ bug of BIOS.
+- Please try once dxs_support=1 and if it works on other
++ Please try once dxs_support=5 and if it works on other
+ sample rates (e.g. 44.1kHz of mp3 playback), please let us
+ know the PCI subsystem vendor/device id's (output of
+ "lspci -nv").
+- If it doesn't work, try dxs_support=4. If it still doesn't
++ If dxs_support=5 does not work, try dxs_support=4; if it
++ doesn't work too, try dxs_support=1. (dxs_support=1 is
++ usually for old motherboards. The correct implementated
++ board should work with 4 or 5.) If it still doesn't
+ work and the default setting is ok, dxs_support=3 is the
+ right choice. If the default setting doesn't work at all,
+ try dxs_support=2 to disable the DXS channels.
+@@ -1497,6 +1562,36 @@ Proc interfaces (/proc/asound)
+ echo "rvplayer 0 0 block" > /proc/asound/card0/pcm0p/oss
+
+
++Early Buffer Allocation
++=======================
++
++Some drivers (e.g. hdsp) require the large contiguous buffers, and
++sometimes it's too late to find such spaces when the driver module is
++actually loaded due to memory fragmentation. You can pre-allocate the
++PCM buffers by loading snd-page-alloc module and write commands to its
++proc file in prior, for example, in the early boot stage like
++/etc/init.d/*.local scripts.
++
++Reading the proc file /proc/drivers/snd-page-alloc shows the current
++usage of page allocation. In writing, you can send the following
++commands to the snd-page-alloc driver:
++
++ - add VENDOR DEVICE MASK SIZE BUFFERS
++
++ VENDOR and DEVICE are PCI vendor and device IDs. They take
++ integer numbers (0x prefix is needed for the hex).
++ MASK is the PCI DMA mask. Pass 0 if not restricted.
++ SIZE is the size of each buffer to allocate. You can pass
++ k and m suffix for KB and MB. The max number is 16MB.
++ BUFFERS is the number of buffers to allocate. It must be greater
++ than 0. The max number is 4.
++
++ - erase
++
++ This will erase the all pre-allocated buffers which are not in
++ use.
++
++
+ Links
+ =====
+
+diff --git a/Documentation/sound/alsa/CMIPCI.txt
b/Documentation/sound/alsa/CMIPCI.txt
+--- a/Documentation/sound/alsa/CMIPCI.txt
++++ b/Documentation/sound/alsa/CMIPCI.txt
+@@ -89,19 +89,22 @@ and use the interleaved 4 channel data.
+
+ There are some control switchs affecting to the speaker connections:
+
+-"Line-In As Rear" - As mentioned above, the line-in jack is used
+- for the rear (3th and 4th channels) output.
+-"Line-In As Bass" - The line-in jack is used for the bass (5th
+- and 6th channels) output.
+-"Mic As Center/LFE" - The mic jack is used for the bass output.
+- If this switch is on, you cannot use a microphone as a capture
+- source, of course.
+-
++"Line-In Mode" - an enum control to change the behavior of line-in
++ jack. Either "Line-In", "Rear Output" or "Bass Output" can
++ be selected. The last item is available only with model 039
++ or newer.
++ When "Rear Output" is chosen, the surround channels 3 and 4
++ are output to line-in jack.
++"Mic-In Mode" - an enum control to change the behavior of mic-in
++ jack. Either "Mic-In" or "Center/LFE Output" can be
++ selected.
++ When "Center/LFE Output" is chosen, the center and bass
++ channels (channels 5 and 6) are output to mic-in jack.
+
+ Digital I/O
+ -----------
+
+-The CM8x38 provides the excellent SPDIF capability with very chip
++The CM8x38 provides the excellent SPDIF capability with very cheap
+ price (yes, that's the reason I bought the card :)
+
+ The SPDIF playback and capture are done via the third PCM device
+@@ -122,8 +125,9 @@ respectively, so you cannot playback bot
+ simultaneously.
+
+ To enable SPDIF output, you need to turn on "IEC958 Output Switch"
+-control via mixer or alsactl. Then you'll see the red light on from
+-the card so you know that's working obviously :)
++control via mixer or alsactl ("IEC958" is the official name of
++so-called S/PDIF). Then you'll see the red light on from the card so
++you know that's working obviously :)
+ The SPDIF input is always enabled, so you can hear SPDIF input data
+ from line-out with "IEC958 In Monitor" switch at any time (see
+ below).
+@@ -205,9 +209,10 @@ In addition to the standard SB mixer, CM
+ MIDI CONTROLLER
+ ---------------
+
+-The MPU401-UART interface is enabled as default only for the first
+-(CMIPCI) card. You need to set module option "midi_port" properly
+-for the 2nd (CMIPCI) card.
++The MPU401-UART interface is disabled as default. You need to set
++module option "mpu_port" with a valid I/O port address to enable the
++MIDI support. The valid I/O ports are 0x300, 0x310, 0x320 and 0x330.
++Choose the value which doesn't conflict with other cards.
+
+ There is _no_ hardware wavetable function on this chip (except for
+ OPL3 synth below).
+@@ -229,9 +234,11 @@ I don't know why..
+ Joystick and Modem
+ ------------------
+
+-The joystick and modem should be available by enabling the control
+-switch "Joystick" and "Modem" respectively. But I myself have never
+-tested them yet.
++The legacy joystick is supported. To enable the joystick support, pass
++joystick_port=1 module option. The value 1 means the auto-detection.
++If the auto-detection fails, try to pass the exact I/O address.
++
++The modem is enabled dynamically via a card control switch "Modem".
+
+
+ Debugging Information
+diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
+--- a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
++++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
+@@ -371,7 +371,7 @@
+ <listitem><para>create <function>probe()</function>
callback.</para></listitem>
+ <listitem><para>create <function>remove()</function>
callback.</para></listitem>
+ <listitem><para>create pci_driver table which contains the three
pointers above.</para></listitem>
+- <listitem><para>create <function>init()</function> function just
calling <function>pci_module_init()</function> to register the pci_driver table
defined above.</para></listitem>
++ <listitem><para>create <function>init()</function> function just
calling <function>pci_register_driver()</function> to register the pci_driver
table defined above.</para></listitem>
+ <listitem><para>create <function>exit()</function> function to call
<function>pci_unregister_driver()</function> function.</para></listitem>
+ </itemizedlist>
+ </para>
+@@ -1198,7 +1198,7 @@
+ /* initialization of the module */
+ static int __init alsa_card_mychip_init(void)
+ {
+- return pci_module_init(&driver);
++ return pci_register_driver(&driver);
+ }
+
+ /* clean up the module */
+@@ -1654,7 +1654,7 @@
+ <![CDATA[
+ static int __init alsa_card_mychip_init(void)
+ {
+- return pci_module_init(&driver);
++ return pci_register_driver(&driver);
+ }
+
+ static void __exit alsa_card_mychip_exit(void)
+diff --git a/Documentation/sound/alsa/emu10k1-jack.txt
b/Documentation/sound/alsa/emu10k1-jack.txt
+new file mode 100644
+--- /dev/null
++++ b/Documentation/sound/alsa/emu10k1-jack.txt
+@@ -0,0 +1,74 @@
++This document is a guide to using the emu10k1 based devices with JACK for low
++latency, multichannel recording functionality. All of my recent work to allow
++Linux users to use the full capabilities of their hardware has been inspired
++by the kX Project. Without their work I never would have discovered the true
++power of this hardware.
++
++ http://www.kxproject.com
++ - Lee Revell, 2005.03.30
++
++Low latency, multichannel audio with JACK and the emu10k1/emu10k2
++-----------------------------------------------------------------
++
++Until recently, emu10k1 users on Linux did not have access to the same low
++latency, multichannel features offered by the "kX ASIO" feature of their
++Windows driver. As of ALSA 1.0.9 this is no more!
++
++For those unfamiliar with kX ASIO, this consists of 16 capture and 16 playback
++channels. With a post 2.6.9 Linux kernel, latencies down to 64 (1.33 ms) or
++even 32 (0.66ms) frames should work well.
++
++The configuration is slightly more involved than on Windows, as you have to
++select the correct device for JACK to use. Actually, for qjackctl users it's
++fairly self explanatory - select Duplex, then for capture and playback select
++the multichannel devices, set the in and out channels to 16, and the sample
++rate to 48000Hz. The command line looks like this:
++
++/usr/local/bin/jackd -R -dalsa -r48000 -p64 -n2 -D -Chw:0,2 -Phw:0,3 -S
++
++This will give you 16 input ports and 16 output ports.
++
++The 16 output ports map onto the 16 FX buses (or the first 16 of 64, for the
++Audigy). The mapping from FX bus to physical output is described in
++SB-Live-mixer.txt (or Audigy-mixer.txt).
++
++The 16 input ports are connected to the 16 physical inputs. Contrary to
++popular belief, all emu10k1 cards are multichannel cards. Which of these
++input channels have physical inputs connected to them depends on the card
++model. Trial and error is highly recommended; the pinout diagrams
++for the card have been reverse engineered by some enterprising kX users and
are
++available on the internet. Meterbridge is helpful here, and the kX forums are
++packed with useful information.
++
++Each input port will either correspond to a digital (SPDIF) input, an analog
++input, or nothing. The one exception is the SBLive! 5.1. On these devices,
++the second and third input ports are wired to the center/LFE output. You will
++still see 16 capture channels, but only 14 are available for recording inputs.
++
++This chart, borrowed from kxfxlib/da_asio51.cpp, describes the mapping of JACK
++ports to FXBUS2 (multitrack recording input) and EXTOUT (physical output)
++channels.
++
++/*JACK (& ASIO) mappings on 10k1 5.1 SBLive cards:
++--------------------------------------------
++JACK Epilog FXBUS2(nr)
++--------------------------------------------
++capture_1 asio14 FXBUS2(0xe)
++capture_2 asio15 FXBUS2(0xf)
++capture_3 asio0 FXBUS2(0x0)
++~capture_4 Center EXTOUT(0x11) // mapped to by Center
++~capture_5 LFE EXTOUT(0x12) // mapped to by LFE
++capture_6 asio3 FXBUS2(0x3)
++capture_7 asio4 FXBUS2(0x4)
++capture_8 asio5 FXBUS2(0x5)
++capture_9 asio6 FXBUS2(0x6)
++capture_10 asio7 FXBUS2(0x7)
++capture_11 asio8 FXBUS2(0x8)
++capture_12 asio9 FXBUS2(0x9)
++capture_13 asio10 FXBUS2(0xa)
++capture_14 asio11 FXBUS2(0xb)
++capture_15 asio12 FXBUS2(0xc)
++capture_16 asio13 FXBUS2(0xd)
++*/
++
++TODO: describe use of ld10k1/qlo10k1 in conjunction with JACK
+diff --git a/Documentation/sound/alsa/hdspm.txt
b/Documentation/sound/alsa/hdspm.txt
+new file mode 100644
+--- /dev/null
++++ b/Documentation/sound/alsa/hdspm.txt
+@@ -0,0 +1,362 @@
++Software Interface ALSA-DSP MADI Driver
++
++(translated from German, so no good English ;-),
++2004 - winfried ritsch
++
++
++
++ Full functionality has been added to the driver. Since some of
++ the Controls and startup-options are ALSA-Standard and only the
++ special Controls are described and discussed below.
++
++
++ hardware functionality:
++
++
++ Audio transmission:
++
++ number of channels -- depends on transmission mode
++
++ The number of channels chosen is from 1..Nmax. The reason to
++ use for a lower number of channels is only resource allocation,
++ since unused DMA channels are disabled and less memory is
++ allocated. So also the throughput of the PCI system can be
++ scaled. (Only important for low performance boards).
<<Diff was trimmed, longer than 597 lines>>
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