Merge upstream (approx. 2.6.12-git8) into 'janitor' branch of netdev-2.6.

5 files changed, 574 insertions, 36 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 71ef0498d5e0..104a994b8289 100644
--- 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 'snd_' prefix. This was removed.
   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 'snd_' prefix. This was removed.
 	  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 'snd_' prefix. This was removed.
 	  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 'snd_' prefix. This was removed.
           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 'snd_' prefix. This was removed.
                         * 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 'snd_' prefix. This was removed.
     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 'snd_' prefix. This was removed.
           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 'snd_' prefix. This was removed.
   ------------------
 
     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 'snd_' prefix. This was removed.
           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
index 4a7df771b806..1872e24442a4 100644
--- 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 both analog and digital streams
 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, CM8x38 provides more functions.
 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
index e789475304b6..db0b7d2dc477 100644
--- 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
index 000000000000..751d45036a05
--- /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
index 000000000000..7a67ff71a9f8
--- /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).
+
+       Single Speed -- 1..64 channels 
+
+		 (Note: Choosing the 56channel mode for transmission or as
+		 receiver, only 56 are transmitted/received over the MADI, but
+		 all 64 channels are available for the mixer, so channel count
+		 for the driver)
+
+       Double Speed -- 1..32 channels
+
+		 Note: Choosing the 56-channel mode for
+		 transmission/receive-mode , only 28 are transmitted/received
+		 over the MADI, but all 32 channels are available for the mixer,
+		 so channel count for the driver
+
+
+       Quad Speed -- 1..16 channels 
+
+		 Note: Choosing the 56-channel mode for
+		 transmission/receive-mode , only 14 are transmitted/received
+		 over the MADI, but all 16 channels are available for the mixer,
+		 so channel count for the driver
+
+     Format -- signed 32 Bit Little Endian (SNDRV_PCM_FMTBIT_S32_LE)
+
+     Sample Rates --
+
+       Single Speed -- 32000, 44100, 48000
+
+       Double Speed -- 64000, 88200, 96000 (untested)
+
+       Quad Speed -- 128000, 176400, 192000 (untested)
+
+     access-mode -- MMAP (memory mapped), Not interleaved
+     (PCM_NON-INTERLEAVED)
+
+     buffer-sizes -- 64,128,256,512,1024,2048,8192 Samples
+
+     fragments -- 2
+
+     Hardware-pointer -- 2 Modi
+
+
+		 The Card supports the readout of the actual Buffer-pointer,
+		 where DMA reads/writes. Since of the bulk mode of PCI it is only
+		 64 Byte accurate. SO it is not really usable for the
+		 ALSA-mid-level functions (here the buffer-ID gives a better
+		 result), but if MMAP is used by the application. Therefore it
+		 can be configured at load-time with the parameter
+		 precise-pointer.
+
+
+		 (Hint: Experimenting I found that the pointer is maximum 64 to
+		 large never to small. So if you subtract 64 you always have a
+		 safe pointer for writing, which is used on this mode inside
+		 ALSA. In theory now you can get now a latency as low as 16
+		 Samples, which is a quarter of the interrupt possibilities.)
+
+       Precise Pointer -- off
+					interrupt used for pointer-calculation
+
+       Precise Pointer -- on
+					hardware pointer used.
+
+   Controller:
+
+
+	  Since DSP-MADI-Mixer has 8152 Fader, it does not make sense to
+	  use the standard mixer-controls, since this would break most of
+	  (especially graphic) ALSA-Mixer GUIs. So Mixer control has be
+	  provided by a 2-dimensional controller using the
+	  hwdep-interface. 
+
+     Also all 128+256 Peak and RMS-Meter can be accessed via the
+     hwdep-interface. Since it could be a performance problem always
+     copying and converting Peak and RMS-Levels even if you just need
+     one, I decided to export the hardware structure, so that of
+     needed some driver-guru can implement a memory-mapping of mixer
+     or peak-meters over ioctl, or also to do only copying and no
+     conversion. A test-application shows the usage of the controller.
+
+    Latency Controls --- not implemented !!!
+
+
+	   Note: Within the windows-driver the latency is accessible of a
+	   control-panel, but buffer-sizes are controlled with ALSA from
+	   hwparams-calls and should not be changed in run-state, I did not
+	   implement it here.
+
+
+    System Clock -- suspended !!!!
+
+        Name -- "System Clock Mode"
+
+        Access -- Read Write
+
+        Values -- "Master" "Slave"
+
+
+		  !!!! This is a hardware-function but is in conflict with the
+		  Clock-source controller, which is a kind of ALSA-standard. I
+		  makes sense to set the card to a special mode (master at some
+		  frequency or slave), since even not using an Audio-application
+		  a studio should have working synchronisations setup. So use
+		  Clock-source-controller instead !!!!
+
+    Clock Source  
+
+       Name -- "Sample Clock Source"
+
+       Access -- Read Write
+
+       Values -- "AutoSync", "Internal 32.0 kHz", "Internal 44.1 kHz",
+       "Internal 48.0 kHz", "Internal 64.0 kHz", "Internal 88.2 kHz",
+       "Internal 96.0 kHz"
+
+		 Choose between Master at a specific Frequency and so also the
+		 Speed-mode or Slave (Autosync). Also see  "Preferred Sync Ref"
+
+
+       !!!! This is no pure hardware function but was implemented by
+       ALSA by some ALSA-drivers before, so I use it also. !!!
+
+
+    Preferred Sync Ref
+
+       Name -- "Preferred Sync Reference"
+
+       Access -- Read Write
+
+       Values -- "Word" "MADI"
+
+
+		 Within the Auto-sync-Mode the preferred Sync Source can be
+		 chosen. If it is not available another is used if possible.
+
+		 Note: Since MADI has a much higher bit-rate than word-clock, the
+		 card should synchronise better in MADI Mode. But since the
+		 RME-PLL is very good, there are almost no problems with
+		 word-clock too. I never found a difference.
+
+
+    TX 64 channel --- 
+
+       Name -- "TX 64 channels mode"
+
+       Access -- Read Write
+
+       Values -- 0 1
+
+		 Using 64-channel-modus (1) or 56-channel-modus for
+		 MADI-transmission (0).
+
+
+		 Note: This control is for output only. Input-mode is detected
+		 automatically from hardware sending MADI.
+
+
+    Clear TMS ---
+
+       Name -- "Clear Track Marker"
+
+       Access -- Read Write
+
+       Values -- 0 1
+
+
+		 Don't use to lower 5 Audio-bits on AES as additional Bits.
+        
+
+    Safe Mode oder Auto Input --- 
+
+       Name -- "Safe Mode"
+
+       Access -- Read Write
+
+       Values -- 0 1
+
+       (default on)
+
+		 If on (1), then if either the optical or coaxial connection
+		 has a failure, there is a takeover to the working one, with no
+		 sample failure. Its only useful if you use the second as a
+		 backup connection.
+
+    Input --- 
+
+       Name -- "Input Select"
+
+       Access -- Read Write
+
+       Values -- optical coaxial
+
+
+		 Choosing the Input, optical or coaxial. If Safe-mode is active,
+		 this is the preferred Input.
+
+-------------- Mixer ----------------------
+
+    Mixer
+
+       Name -- "Mixer"
+
+       Access -- Read Write
+
+       Values - <channel-number 0-127> <Value 0-65535>
+
+
+		 Here as a first value the channel-index is taken to get/set the
+		 corresponding mixer channel, where 0-63 are the input to output
+		 fader and 64-127 the playback to outputs fader. Value 0
+		 is channel muted 0 and 32768 an amplification of  1.
+
+    Chn 1-64
+
+       fast mixer for the ALSA-mixer utils. The diagonal of the
+       mixer-matrix is implemented from playback to output.
+       
+
+    Line Out
+
+       Name  -- "Line Out"
+
+       Access -- Read Write
+
+       Values -- 0 1
+
+		 Switching on and off the analog out, which has nothing to do
+		 with mixing or routing. the analog outs reflects channel 63,64.
+
+
+--- information (only read access):
+ 
+    Sample Rate
+
+       Name -- "System Sample Rate"
+
+       Access -- Read-only
+
+		 getting the sample rate.
+
+
+    External Rate measured
+
+       Name -- "External Rate"
+
+       Access -- Read only
+
+
+		 Should be "Autosync Rate", but Name used is
+		 ALSA-Scheme. External Sample frequency liked used on Autosync is
+		 reported.
+
+
+    MADI Sync Status
+
+       Name -- "MADI Sync Lock Status"
+
+       Access -- Read
+
+       Values -- 0,1,2
+
+       MADI-Input is 0=Unlocked, 1=Locked, or 2=Synced.
+
+
+    Word Clock Sync Status
+
+       Name -- "Word Clock Lock Status"
+
+       Access -- Read
+
+       Values -- 0,1,2
+
+       Word Clock Input is 0=Unlocked, 1=Locked, or 2=Synced.
+
+    AutoSync
+
+       Name -- "AutoSync Reference"
+
+       Access -- Read
+
+       Values -- "WordClock", "MADI", "None"
+
+		 Sync-Reference is either "WordClock", "MADI" or none.
+
+   RX 64ch --- noch nicht implementiert
+
+       MADI-Receiver is in 64 channel mode oder 56 channel mode.
+
+
+   AB_inp   --- not tested 
+
+		 Used input for Auto-Input.
+
+
+   actual Buffer Position --- not implemented
+
+	   !!! this is a ALSA internal function, so no control is used !!!
+
+
+
+Calling Parameter:
+
+   index int array (min = 1, max = 8), 
+     "Index value for RME HDSPM interface." card-index within ALSA
+
+     note: ALSA-standard
+
+   id string array (min = 1, max = 8), 
+     "ID string for RME HDSPM interface."
+
+     note: ALSA-standard
+
+   enable int array (min = 1, max = 8), 
+     "Enable/disable specific HDSPM sound-cards."
+
+     note: ALSA-standard
+
+   precise_ptr int array (min = 1, max = 8), 
+     "Enable precise pointer, or disable."
+
+     note: Use only when the application supports this (which is a special case).
+
+   line_outs_monitor int array (min = 1, max = 8), 
+     "Send playback streams to analog outs by default."
+
+
+	  note: each playback channel is mixed to the same numbered output
+	  channel (routed). This is against the ALSA-convention, where all
+	  channels have to be muted on after loading the driver, but was
+	  used before on other cards, so i historically use it again)
+
+
+
+   enable_monitor int array (min = 1, max = 8), 
+     "Enable Analog Out on Channel 63/64 by default."
+
+      note: here the analog output is enabled (but not routed).
\ No newline at end of file