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-rw-r--r--Documentation/sound/alsa/ALSA-Configuration.txt127
-rw-r--r--Documentation/sound/alsa/CMIPCI.txt41
-rw-r--r--Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl6
-rw-r--r--Documentation/sound/alsa/emu10k1-jack.txt74
-rw-r--r--Documentation/sound/alsa/hdspm.txt362
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
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