diff options
Diffstat (limited to 'Documentation/sound/alsa')
-rw-r--r-- | Documentation/sound/alsa/ALSA-Configuration.txt | 127 | ||||
-rw-r--r-- | Documentation/sound/alsa/CMIPCI.txt | 41 | ||||
-rw-r--r-- | Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl | 6 | ||||
-rw-r--r-- | Documentation/sound/alsa/emu10k1-jack.txt | 74 | ||||
-rw-r--r-- | Documentation/sound/alsa/hdspm.txt | 362 |
5 files changed, 574 insertions, 36 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 71ef0498d5e..104a994b828 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 4a7df771b80..1872e24442a 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 e789475304b..db0b7d2dc47 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 00000000000..751d45036a0 --- /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 00000000000..7a67ff71a9f --- /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).
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