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authorLinus Torvalds <torvalds@ppc970.osdl.org>2005-04-16 15:20:36 -0700
committerLinus Torvalds <torvalds@ppc970.osdl.org>2005-04-16 15:20:36 -0700
commit1da177e4c3f41524e886b7f1b8a0c1fc7321cac2 (patch)
tree0bba044c4ce775e45a88a51686b5d9f90697ea9d /Documentation/sound/alsa
downloadlinux-stericsson-2.6.12-rc2.tar.gz
Linux-2.6.12-rc2v2.6.12-rc2
Initial git repository build. I'm not bothering with the full history, even though we have it. We can create a separate "historical" git archive of that later if we want to, and in the meantime it's about 3.2GB when imported into git - space that would just make the early git days unnecessarily complicated, when we don't have a lot of good infrastructure for it. Let it rip!
Diffstat (limited to 'Documentation/sound/alsa')
-rw-r--r--Documentation/sound/alsa/ALSA-Configuration.txt1505
-rw-r--r--Documentation/sound/alsa/Audigy-mixer.txt345
-rw-r--r--Documentation/sound/alsa/Bt87x.txt78
-rw-r--r--Documentation/sound/alsa/CMIPCI.txt242
-rw-r--r--Documentation/sound/alsa/ControlNames.txt84
-rw-r--r--Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl100
-rw-r--r--Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl6045
-rw-r--r--Documentation/sound/alsa/Joystick.txt86
-rw-r--r--Documentation/sound/alsa/MIXART.txt100
-rw-r--r--Documentation/sound/alsa/OSS-Emulation.txt297
-rw-r--r--Documentation/sound/alsa/Procfile.txt191
-rw-r--r--Documentation/sound/alsa/SB-Live-mixer.txt356
-rw-r--r--Documentation/sound/alsa/VIA82xx-mixer.txt8
-rw-r--r--Documentation/sound/alsa/hda_codec.txt299
-rw-r--r--Documentation/sound/alsa/seq_oss.html409
-rw-r--r--Documentation/sound/alsa/serial-u16550.txt88
16 files changed, 10233 insertions, 0 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
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index 000000000000..71ef0498d5e0
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+++ b/Documentation/sound/alsa/ALSA-Configuration.txt
@@ -0,0 +1,1505 @@
+
+ Advanced Linux Sound Architecture - Driver
+ ==========================================
+ Configuration guide
+
+
+Kernel Configuration
+====================
+
+To enable ALSA support you need at least to build the kernel with
+primary sound card support (CONFIG_SOUND). Since ALSA can emulate OSS,
+you don't have to choose any of the OSS modules.
+
+Enable "OSS API emulation" (CONFIG_SND_OSSEMUL) and both OSS mixer and
+PCM supports if you want to run OSS applications with ALSA.
+
+If you want to support the WaveTable functionality on cards such as
+SB Live! then you need to enable "Sequencer support"
+(CONFIG_SND_SEQUENCER).
+
+To make ALSA debug messages more verbose, enable the "Verbose printk"
+and "Debug" options. To check for memory leaks, turn on "Debug memory"
+too. "Debug detection" will add checks for the detection of cards.
+
+Please note that all the ALSA ISA drivers support the Linux isapnp API
+(if the card supports ISA PnP). You don't need to configure the cards
+using isapnptools.
+
+
+Creating ALSA devices
+=====================
+
+This depends on your distribution, but normally you use the /dev/MAKEDEV
+script to create the necessary device nodes. On some systems you use a
+script named 'snddevices'.
+
+
+Module parameters
+=================
+
+The user can load modules with options. If the module supports more than
+one card and you have more than one card of the same type then you can
+specify multiple values for the option separated by commas.
+
+Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed.
+
+ Module snd
+ ----------
+
+ The core ALSA module. It is used by all ALSA card drivers.
+ It takes the following options which have global effects.
+
+ major - major number for sound driver
+ - Default: 116
+ cards_limit
+ - limiting card index for auto-loading (1-8)
+ - Default: 1
+ - For auto-loading more than one card, specify this
+ option together with snd-card-X aliases.
+ device_mode
+ - permission mask for dynamic sound device filesystem
+ - This is available only when DEVFS is enabled
+ - Default: 0666
+ - E.g.: device_mode=0660
+
+
+ Module snd-pcm-oss
+ ------------------
+
+ The PCM OSS emulation module.
+ This module takes options which change the mapping of devices.
+
+ dsp_map - PCM device number maps assigned to the 1st OSS device.
+ - Default: 0
+ adsp_map - PCM device number maps assigned to the 2st OSS device.
+ - Default: 1
+ nonblock_open
+ - Don't block opening busy PCM devices.
+
+ For example, when dsp_map=2, /dev/dsp will be mapped to PCM #2 of
+ the card #0. Similarly, when adsp_map=0, /dev/adsp will be mapped
+ to PCM #0 of the card #0.
+ For changing the second or later card, specify the option with
+ commas, such like "dsp_map=0,1".
+
+ nonblock_open option is used to change the behavior of the PCM
+ regarding opening the device. When this option is non-zero,
+ opening a busy OSS PCM device won't be blocked but return
+ immediately with EAGAIN (just like O_NONBLOCK flag).
+
+ Module snd-rawmidi
+ ------------------
+
+ This module takes options which change the mapping of devices.
+ similar to those of the snd-pcm-oss module.
+
+ midi_map - MIDI device number maps assigned to the 1st OSS device.
+ - Default: 0
+ amidi_map - MIDI device number maps assigned to the 2st OSS device.
+ - Default: 1
+
+ Common parameters for top sound card modules
+ --------------------------------------------
+
+ Each of top level sound card module takes the following options.
+
+ index - index (slot #) of sound card
+ - Values: 0 through 7 or negative
+ - If nonnegative, assign that index number
+ - if negative, interpret as a bitmask of permissible
+ indices; the first free permitted index is assigned
+ - Default: -1
+ id - card ID (identifier or name)
+ - Can be up to 15 characters long
+ - Default: the card type
+ - A directory by this name is created under /proc/asound/
+ containing information about the card
+ - This ID can be used instead of the index number in
+ identifying the card
+ enable - enable card
+ - Default: enabled, for PCI and ISA PnP cards
+
+ Module snd-ad1816a
+ ------------------
+
+ Module for sound cards based on Analog Devices AD1816A/AD1815 ISA chips.
+
+ port - port # for AD1816A chip (PnP setup)
+ mpu_port - port # for MPU-401 UART (PnP setup)
+ fm_port - port # for OPL3 (PnP setup)
+ irq - IRQ # for AD1816A chip (PnP setup)
+ mpu_irq - IRQ # for MPU-401 UART (PnP setup)
+ dma1 - first DMA # for AD1816A chip (PnP setup)
+ dma2 - second DMA # for AD1816A chip (PnP setup)
+
+ Module supports up to 8 cards, autoprobe and PnP.
+
+ Module snd-ad1848
+ -----------------
+
+ Module for sound cards based on AD1848/AD1847/CS4248 ISA chips.
+
+ port - port # for AD1848 chip
+ irq - IRQ # for AD1848 chip
+ dma1 - DMA # for AD1848 chip (0,1,3)
+
+ Module supports up to 8 cards. This module does not support autoprobe
+ thus main port must be specified!!! Other ports are optional.
+
+ Module snd-ali5451
+ ------------------
+
+ Module for ALi M5451 PCI chip.
+
+ pcm_channels - Number of hardware channels assigned for PCM
+ spdif - Support SPDIF I/O
+ - Default: disabled
+
+ Module supports autoprobe and multiple chips (max 8).
+
+ The power-management is supported.
+
+ Module snd-als100
+ -----------------
+
+ Module for sound cards based on Avance Logic ALS100/ALS120 ISA chips.
+
+ port - port # for ALS100 (SB16) chip (PnP setup)
+ irq - IRQ # for ALS100 (SB16) chip (PnP setup)
+ dma8 - 8-bit DMA # for ALS100 (SB16) chip (PnP setup)
+ dma16 - 16-bit DMA # for ALS100 (SB16) chip (PnP setup)
+ mpu_port - port # for MPU-401 UART (PnP setup)
+ mpu_irq - IRQ # for MPU-401 (PnP setup)
+ fm_port - port # for OPL3 FM (PnP setup)
+
+ Module supports up to 8 cards, autoprobe and PnP.
+
+ Module snd-als4000
+ ------------------
+
+ Module for sound cards based on Avance Logic ALS4000 PCI chip.
+
+ joystick_port - port # for legacy joystick support.
+ 0 = disabled (default), 1 = auto-detect
+
+ Module supports up to 8 cards, autoprobe and PnP.
+
+ Module snd-atiixp
+ -----------------
+
+ Module for ATI IXP 150/200/250 AC97 controllers.
+
+ ac97_clock - AC'97 clock (defalut = 48000)
+ ac97_quirk - AC'97 workaround for strange hardware
+ See the description of intel8x0 module for details.
+ spdif_aclink - S/PDIF transfer over AC-link (default = 1)
+
+ This module supports up to 8 cards and autoprobe.
+
+ Module snd-atiixp-modem
+ -----------------------
+
+ Module for ATI IXP 150/200/250 AC97 modem controllers.
+
+ Module supports up to 8 cards.
+
+ Note: The default index value of this module is -2, i.e. the first
+ slot is excluded.
+
+ Module snd-au8810, snd-au8820, snd-au8830
+ -----------------------------------------
+
+ Module for Aureal Vortex, Vortex2 and Advantage device.
+
+ pcifix - Control PCI workarounds
+ 0 = Disable all workarounds
+ 1 = Force the PCI latency of the Aureal card to 0xff
+ 2 = Force the Extend PCI#2 Internal Master for Efficient
+ Handling of Dummy Requests on the VIA KT133 AGP Bridge
+ 3 = Force both settings
+ 255 = Autodetect what is required (default)
+
+ This module supports all ADB PCM channels, ac97 mixer, SPDIF, hardware
+ EQ, mpu401, gameport. A3D and wavetable support are still in development.
+ Development and reverse engineering work is being coordinated at
+ http://savannah.nongnu.org/projects/openvortex/
+ SPDIF output has a copy of the AC97 codec output, unless you use the
+ "spdif" pcm device, which allows raw data passthru.
+ The hardware EQ hardware and SPDIF is only present in the Vortex2 and
+ Advantage.
+
+ Note: Some ALSA mixer applicactions don't handle the SPDIF samplerate
+ control correctly. If you have problems regarding this, try
+ another ALSA compliant mixer (alsamixer works).
+
+ Module snd-azt2320
+ ------------------
+
+ Module for sound cards based on Aztech System AZT2320 ISA chip (PnP only).
+
+ port - port # for AZT2320 chip (PnP setup)
+ wss_port - port # for WSS (PnP setup)
+ mpu_port - port # for MPU-401 UART (PnP setup)
+ fm_port - FM port # for AZT2320 chip (PnP setup)
+ irq - IRQ # for AZT2320 (WSS) chip (PnP setup)
+ mpu_irq - IRQ # for MPU-401 UART (PnP setup)
+ dma1 - 1st DMA # for AZT2320 (WSS) chip (PnP setup)
+ dma2 - 2nd DMA # for AZT2320 (WSS) chip (PnP setup)
+
+ Module supports up to 8 cards, PnP and autoprobe.
+
+ Module snd-azt3328
+ ------------------
+
+ Module for sound cards based on Aztech AZF3328 PCI chip.
+
+ joystick - Enable joystick (default off)
+
+ Module supports up to 8 cards.
+
+ Module snd-bt87x
+ ----------------
+
+ Module for video cards based on Bt87x chips.
+
+ digital_rate - Override the default digital rate (Hz)
+ load_all - Load the driver even if the card model isn't known
+
+ Module supports up to 8 cards.
+
+ Note: The default index value of this module is -2, i.e. the first
+ slot is excluded.
+
+ Module snd-ca0106
+ -----------------
+
+ Module for Creative Audigy LS and SB Live 24bit
+
+ Module supports up to 8 cards.
+
+
+ Module snd-cmi8330
+ ------------------
+
+ Module for sound cards based on C-Media CMI8330 ISA chips.
+
+ wssport - port # for CMI8330 chip (WSS)
+ wssirq - IRQ # for CMI8330 chip (WSS)
+ wssdma - first DMA # for CMI8330 chip (WSS)
+ sbport - port # for CMI8330 chip (SB16)
+ sbirq - IRQ # for CMI8330 chip (SB16)
+ sbdma8 - 8bit DMA # for CMI8330 chip (SB16)
+ sbdma16 - 16bit DMA # for CMI8330 chip (SB16)
+
+ Module supports up to 8 cards and autoprobe.
+
+ Module snd-cmipci
+ -----------------
+
+ Module for C-Media CMI8338 and 8738 PCI sound cards.
+
+ mpu_port - 0x300,0x310,0x320,0x330, 0 = disable (default)
+ fm_port - 0x388 (default), 0 = disable (default)
+ soft_ac3 - Sofware-conversion of raw SPDIF packets (model 033 only)
+ (default = 1)
+ joystick_port - Joystick port address (0 = disable, 1 = auto-detect)
+
+ Module supports autoprobe and multiple chips (max 8).
+
+ Module snd-cs4231
+ -----------------
+
+ Module for sound cards based on CS4231 ISA chips.
+
+ port - port # for CS4231 chip
+ mpu_port - port # for MPU-401 UART (optional), -1 = disable
+ irq - IRQ # for CS4231 chip
+ mpu_irq - IRQ # for MPU-401 UART
+ dma1 - first DMA # for CS4231 chip
+ dma2 - second DMA # for CS4231 chip
+
+ Module supports up to 8 cards. This module does not support autoprobe
+ thus main port must be specified!!! Other ports are optional.
+
+ The power-management is supported.
+
+ Module snd-cs4232
+ -----------------
+
+ Module for sound cards based on CS4232/CS4232A ISA chips.
+
+ port - port # for CS4232 chip (PnP setup - 0x534)
+ cport - control port # for CS4232 chip (PnP setup - 0x120,0x210,0xf00)
+ mpu_port - port # for MPU-401 UART (PnP setup - 0x300), -1 = disable
+ fm_port - FM port # for CS4232 chip (PnP setup - 0x388), -1 = disable
+ irq - IRQ # for CS4232 chip (5,7,9,11,12,15)
+ mpu_irq - IRQ # for MPU-401 UART (9,11,12,15)
+ dma1 - first DMA # for CS4232 chip (0,1,3)
+ dma2 - second DMA # for Yamaha CS4232 chip (0,1,3), -1 = disable
+ isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
+
+ Module supports up to 8 cards. This module does not support autoprobe
+ thus main port must be specified!!! Other ports are optional.
+
+ The power-management is supported.
+
+ Module snd-cs4236
+ -----------------
+
+ Module for sound cards based on CS4235/CS4236/CS4236B/CS4237B/
+ CS4238B/CS4239 ISA chips.
+
+ port - port # for CS4236 chip (PnP setup - 0x534)
+ cport - control port # for CS4236 chip (PnP setup - 0x120,0x210,0xf00)
+ mpu_port - port # for MPU-401 UART (PnP setup - 0x300), -1 = disable
+ fm_port - FM port # for CS4236 chip (PnP setup - 0x388), -1 = disable
+ irq - IRQ # for CS4236 chip (5,7,9,11,12,15)
+ mpu_irq - IRQ # for MPU-401 UART (9,11,12,15)
+ dma1 - first DMA # for CS4236 chip (0,1,3)
+ dma2 - second DMA # for CS4236 chip (0,1,3), -1 = disable
+ isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
+
+ Module supports up to 8 cards. This module does not support autoprobe
+ (if ISA PnP is not used) thus main port and control port must be
+ specified!!! Other ports are optional.
+
+ The power-management is supported.
+
+ Module snd-cs4281
+ -----------------
+
+ Module for Cirrus Logic CS4281 soundchip.
+
+ dual_codec - Secondary codec ID (0 = disable, default)
+
+ Module supports up to 8 cards.
+
+ The power-management is supported.
+
+ Module snd-cs46xx
+ -----------------
+
+ Module for PCI sound cards based on CS4610/CS4612/CS4614/CS4615/CS4622/
+ CS4624/CS4630/CS4280 PCI chips.
+
+ external_amp - Force to enable external amplifer.
+ thinkpad - Force to enable Thinkpad's CLKRUN control.
+ mmap_valid - Support OSS mmap mode (default = 0).
+
+ Module supports up to 8 cards and autoprobe.
+ Usually external amp and CLKRUN controls are detected automatically
+ from PCI sub vendor/device ids. If they don't work, give the options
+ above explicitly.
+
+ The power-management is supported.
+
+ Module snd-dt019x
+ -----------------
+
+ Module for Diamond Technologies DT-019X / Avance Logic ALS-007 (PnP
+ only)
+
+ port - Port # (PnP setup)
+ mpu_port - Port # for MPU-401 (PnP setup)
+ fm_port - Port # for FM OPL-3 (PnP setup)
+ irq - IRQ # (PnP setup)
+ mpu_irq - IRQ # for MPU-401 (PnP setup)
+ dma8 - DMA # (PnP setup)
+
+ Module supports up to 8 cards. This module is enabled only with
+ ISA PnP support.
+
+ Module snd-dummy
+ ----------------
+
+ Module for the dummy sound card. This "card" doesn't do any output
+ or input, but you may use this module for any application which
+ requires a sound card (like RealPlayer).
+
+ Module snd-emu10k1
+ ------------------
+
+ Module for EMU10K1/EMU10k2 based PCI sound cards.
+ * Sound Blaster Live!
+ * Sound Blaster PCI 512
+ * Emu APS (partially supported)
+ * Sound Blaster Audigy
+
+ extin - bitmap of available external inputs for FX8010 (see bellow)
+ extout - bitmap of available external outputs for FX8010 (see bellow)
+ seq_ports - allocated sequencer ports (4 by default)
+ max_synth_voices - limit of voices used for wavetable (64 by default)
+ max_buffer_size - specifies the maximum size of wavetable/pcm buffers
+ given in MB unit. Default value is 128.
+ enable_ir - enable IR
+
+ Module supports up to 8 cards and autoprobe.
+
+ Input & Output configurations [extin/extout]
+ * Creative Card wo/Digital out [0x0003/0x1f03]
+ * Creative Card w/Digital out [0x0003/0x1f0f]
+ * Creative Card w/Digital CD in [0x000f/0x1f0f]
+ * Creative Card wo/Digital out + LiveDrive [0x3fc3/0x1fc3]
+ * Creative Card w/Digital out + LiveDrive [0x3fc3/0x1fcf]
+ * Creative Card w/Digital CD in + LiveDrive [0x3fcf/0x1fcf]
+ * Creative Card wo/Digital out + Digital I/O 2 [0x0fc3/0x1f0f]
+ * Creative Card w/Digital out + Digital I/O 2 [0x0fc3/0x1f0f]
+ * Creative Card w/Digital CD in + Digital I/O 2 [0x0fcf/0x1f0f]
+ * Creative Card 5.1/w Digital out + LiveDrive [0x3fc3/0x1fff]
+ * Creative Card 5.1 (c) 2003 [0x3fc3/0x7cff]
+ * Creative Card all ins and outs [0x3fff/0x7fff]
+
+ Module snd-emu10k1x
+ -------------------
+
+ Module for Creative Emu10k1X (SB Live Dell OEM version)
+
+ Module supports up to 8 cards.
+
+ Module snd-ens1370
+ ------------------
+
+ Module for Ensoniq AudioPCI ES1370 PCI sound cards.
+ * SoundBlaster PCI 64
+ * SoundBlaster PCI 128
+
+ joystick - Enable joystick (default off)
+
+ Module supports up to 8 cards and autoprobe.
+
+ Module snd-ens1371
+ ------------------
+
+ Module for Ensoniq AudioPCI ES1371 PCI sound cards.
+ * SoundBlaster PCI 64
+ * SoundBlaster PCI 128
+ * SoundBlaster Vibra PCI
+
+ joystick_port - port # for joystick (0x200,0x208,0x210,0x218),
+ 0 = disable (default), 1 = auto-detect
+
+ Module supports up to 8 cards and autoprobe.
+
+ Module snd-es968
+ ----------------
+
+ Module for sound cards based on ESS ES968 chip (PnP only).
+
+ port - port # for ES968 (SB8) chip (PnP setup)
+ irq - IRQ # for ES968 (SB8) chip (PnP setup)
+ dma1 - DMA # for ES968 (SB8) chip (PnP setup)
+
+ Module supports up to 8 cards, PnP and autoprobe.
+
+ Module snd-es1688
+ -----------------
+
+ Module for ESS AudioDrive ES-1688 and ES-688 sound cards.
+
+ port - port # for ES-1688 chip (0x220,0x240,0x260)
+ mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default)
+ irq - IRQ # for ES-1688 chip (5,7,9,10)
+ mpu_irq - IRQ # for MPU-401 port (5,7,9,10)
+ dma8 - DMA # for ES-1688 chip (0,1,3)
+
+ Module supports up to 8 cards and autoprobe (without MPU-401 port).
+
+ Module snd-es18xx
+ -----------------
+
+ Module for ESS AudioDrive ES-18xx sound cards.
+
+ port - port # for ES-18xx chip (0x220,0x240,0x260)
+ mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable (default)
+ fm_port - port # for FM (optional, not used)
+ irq - IRQ # for ES-18xx chip (5,7,9,10)
+ dma1 - first DMA # for ES-18xx chip (0,1,3)
+ dma2 - first DMA # for ES-18xx chip (0,1,3)
+ isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
+
+ Module supports up to 8 cards ISA PnP and autoprobe (without MPU-401 port
+ if native ISA PnP routines are not used).
+ When dma2 is equal with dma1, the driver works as half-duplex.
+
+ The power-management is supported.
+
+ Module snd-es1938
+ -----------------
+
+ Module for sound cards based on ESS Solo-1 (ES1938,ES1946) chips.
+
+ Module supports up to 8 cards and autoprobe.
+
+ Module snd-es1968
+ -----------------
+
+ Module for sound cards based on ESS Maestro-1/2/2E (ES1968/ES1978) chips.
+
+ total_bufsize - total buffer size in kB (1-4096kB)
+ pcm_substreams_p - playback channels (1-8, default=2)
+ pcm_substreams_c - capture channels (1-8, default=0)
+ clock - clock (0 = auto-detection)
+ use_pm - support the power-management (0 = off, 1 = on,
+ 2 = auto (default))
+ enable_mpu - enable MPU401 (0 = off, 1 = on, 2 = auto (default))
+ joystick - enable joystick (default off)
+
+ Module supports up to 8 cards and autoprobe.
+
+ The power-management is supported.
+
+ Module snd-fm801
+ ----------------
+
+ Module for ForteMedia FM801 based PCI sound cards.
+
+ tea575x_tuner - Enable TEA575x tuner
+ - 1 = MediaForte 256-PCS
+ - 2 = MediaForte 256-PCPR
+ - 3 = MediaForte 64-PCR
+ - High 16-bits are video (radio) device number + 1
+ - example: 0x10002 (MediaForte 256-PCPR, device 1)
+
+ Module supports up to 8 cards and autoprobe.
+
+ Module snd-gusclassic
+ ---------------------
+
+ Module for Gravis UltraSound Classic sound card.
+
+ port - port # for GF1 chip (0x220,0x230,0x240,0x250,0x260)
+ irq - IRQ # for GF1 chip (3,5,9,11,12,15)
+ dma1 - DMA # for GF1 chip (1,3,5,6,7)
+ dma2 - DMA # for GF1 chip (1,3,5,6,7,-1=disable)
+ joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V)
+ voices - GF1 voices limit (14-32)
+ pcm_voices - reserved PCM voices
+
+ Module supports up to 8 cards and autoprobe.
+
+ Module snd-gusextreme
+ ---------------------
+
+ Module for Gravis UltraSound Extreme (Synergy ViperMax) sound card.
+
+ port - port # for ES-1688 chip (0x220,0x230,0x240,0x250,0x260)
+ gf1_port - port # for GF1 chip (0x210,0x220,0x230,0x240,0x250,0x260,0x270)
+ mpu_port - port # for MPU-401 port (0x300,0x310,0x320,0x330), -1 = disable
+ irq - IRQ # for ES-1688 chip (5,7,9,10)
+ gf1_irq - IRQ # for GF1 chip (3,5,9,11,12,15)
+ mpu_irq - IRQ # for MPU-401 port (5,7,9,10)
+ dma8 - DMA # for ES-1688 chip (0,1,3)
+ dma1 - DMA # for GF1 chip (1,3,5,6,7)
+ joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V)
+ voices - GF1 voices limit (14-32)
+ pcm_voices - reserved PCM voices
+
+ Module supports up to 8 cards and autoprobe (without MPU-401 port).
+
+ Module snd-gusmax
+ -----------------
+
+ Module for Gravis UltraSound MAX sound card.
+
+ port - port # for GF1 chip (0x220,0x230,0x240,0x250,0x260)
+ irq - IRQ # for GF1 chip (3,5,9,11,12,15)
+ dma1 - DMA # for GF1 chip (1,3,5,6,7)
+ dma2 - DMA # for GF1 chip (1,3,5,6,7,-1=disable)
+ joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V)
+ voices - GF1 voices limit (14-32)
+ pcm_voices - reserved PCM voices
+
+ Module supports up to 8 cards and autoprobe.
+
+ Module snd-hda-intel
+ --------------------
+
+ Module for Intel HD Audio (ICH6, ICH6M, ICH7)
+
+ model - force the model name
+
+ Module supports up to 8 cards.
+
+ Each codec may have a model table for different configurations.
+ If your machine isn't listed there, the default (usually minimal)
+ configuration is set up. You can pass "model=<name>" option to
+ specify a certain model in such a case. There are different
+ models depending on the codec chip.
+
+ Model name Description
+ ---------- -----------
+ ALC880
+ 3stack 3-jack in back and a headphone out
+ 3stack-digout 3-jack in back, a HP out and a SPDIF out
+ 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
+
+ CMI9880
+ minimal 3-jack in back
+ min_fp 3-jack in back, 2-jack in front
+ 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
+
+ Module snd-hdsp
+ ---------------
+
+ Module for RME Hammerfall DSP audio interface(s)
+
+ Module supports up to 8 cards.
+
+ Note: The firmware data can be automatically loaded via hotplug
+ when CONFIG_FW_LOADER is set. Otherwise, you need to load
+ the firmware via hdsploader utility included in alsa-tools
+ package.
+ The firmware data is found in alsa-firmware package.
+
+ Note: snd-page-alloc module does the job which snd-hammerfall-mem
+ 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.
+
+ Module snd-ice1712
+ ------------------
+
+ Module for Envy24 (ICE1712) based PCI sound cards.
+ * MidiMan M Audio Delta 1010
+ * MidiMan M Audio Delta 1010LT
+ * MidiMan M Audio Delta DiO 2496
+ * MidiMan M Audio Delta 66
+ * MidiMan M Audio Delta 44
+ * MidiMan M Audio Delta 410
+ * MidiMan M Audio Audiophile 2496
+ * TerraTec EWS 88MT
+ * TerraTec EWS 88D
+ * TerraTec EWX 24/96
+ * TerraTec DMX 6Fire
+ * Hoontech SoundTrack DSP 24
+ * Hoontech SoundTrack DSP 24 Value
+ * Hoontech SoundTrack DSP 24 Media 7.1
+ * Digigram VX442
+
+ 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
+ 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)
+
+ Module supports up to 8 cards and autoprobe. Note: The consumer part
+ is not used with all Envy24 based cards (for example in the MidiMan Delta
+ serie).
+
+ Module snd-ice1724
+ ------------------
+
+ Module for Envy24HT (VT/ICE1724) based PCI sound cards.
+ * MidiMan M Audio Revolution 7.1
+ * AMP Ltd AUDIO2000
+ * TerraTec Aureon Sky-5.1, Space-7.1
+
+ model - Use the given board model, one of the following:
+ revo71, amp2000, prodigy71, aureon51, aureon71,
+ k8x800
+
+ Module supports up to 8 cards and autoprobe.
+
+ Module snd-intel8x0
+ -------------------
+
+ Module for AC'97 motherboards from Intel and compatibles.
+ * Intel i810/810E, i815, i820, i830, i84x, MX440
+ * SiS 7012 (SiS 735)
+ * NVidia NForce, NForce2
+ * AMD AMD768, AMD8111
+ * ALi m5455
+
+ ac97_clock - AC'97 codec clock base (0 = auto-detect)
+ ac97_quirk - AC'97 workaround for strange hardware
+ The following strings are accepted:
+ default = don't override the default setting
+ disable = disable the quirk
+ hp_only = use headphone control as master
+ swap_hp = swap headphone and master controls
+ swap_surround = swap master and surround controls
+ ad_sharing = for AD1985, turn on OMS bit and use headphone
+ alc_jack = for ALC65x, turn on the jack sense mode
+ inv_eapd = inverted EAPD implementation
+ mute_led = bind EAPD bit for turning on/off mute LED
+ For backward compatibility, the corresponding integer
+ value -1, 0, ... are accepted, too.
+ buggy_irq - Enable workaround for buggy interrupts on some
+ motherboards (default off)
+
+ Module supports autoprobe and multiple bus-master chips (max 8).
+
+ Note: the latest driver supports auto-detection of chip clock.
+ if you still encounter too fast playback, specify the clock
+ explicitly via the module option "ac97_clock=41194".
+
+ Joystick/MIDI ports are not supported by this driver. If your
+ motherboard has these devices, use the ns558 or snd-mpu401
+ modules, respectively.
+
+ The ac97_quirk option is used to enable/override the workaround
+ for specific devices. Some hardware have swapped output pins
+ between Master and Headphone, or Surround. The driver provides
+ the auto-detection of known problematic devices, but some might
+ be unknown or wrongly detected. In such a case, pass the proper
+ value with this option.
+
+ The power-management is supported.
+
+ Module snd-intel8x0m
+ --------------------
+
+ Module for Intel ICH (i8x0) chipset MC97 modems.
+
+ ac97_clock - AC'97 codec clock base (0 = auto-detect)
+
+ This module supports up to 8 cards and autoprobe.
+
+ Note: The default index value of this module is -2, i.e. the first
+ slot is excluded.
+
+ Module snd-interwave
+ --------------------
+
+ Module for Gravis UltraSound PnP, Dynasonic 3-D/Pro, STB Sound Rage 32
+ and other sound cards based on AMD InterWave (tm) chip.
+
+ port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260)
+ irq - IRQ # for InterWave chip (3,5,9,11,12,15)
+ dma1 - DMA # for InterWave chip (0,1,3,5,6,7)
+ dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable)
+ joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V)
+ midi - 1 = MIDI UART enable, 0 = MIDI UART disable (default)
+ pcm_voices - reserved PCM voices for the synthesizer (default 2)
+ effect - 1 = InterWave effects enable (default 0);
+ requires 8 voices
+
+ Module supports up to 8 cards, autoprobe and ISA PnP.
+
+ Module snd-interwave-stb
+ ------------------------
+
+ Module for UltraSound 32-Pro (sound card from STB used by Compaq)
+ and other sound cards based on AMD InterWave (tm) chip with TEA6330T
+ circuit for extended control of bass, treble and master volume.
+
+ port - port # for InterWave chip (0x210,0x220,0x230,0x240,0x250,0x260)
+ port_tc - tone control (i2c bus) port # for TEA6330T chip (0x350,0x360,0x370,0x380)
+ irq - IRQ # for InterWave chip (3,5,9,11,12,15)
+ dma1 - DMA # for InterWave chip (0,1,3,5,6,7)
+ dma2 - DMA # for InterWave chip (0,1,3,5,6,7,-1=disable)
+ joystick_dac - 0 to 31, (0.59V-4.52V or 0.389V-2.98V)
+ midi - 1 = MIDI UART enable, 0 = MIDI UART disable (default)
+ pcm_voices - reserved PCM voices for the synthesizer (default 2)
+ effect - 1 = InterWave effects enable (default 0);
+ requires 8 voices
+
+ Module supports up to 8 cards, autoprobe and ISA PnP.
+
+ Module snd-korg1212
+ -------------------
+
+ Module for Korg 1212 IO PCI card
+
+ Module supports up to 8 cards.
+
+ Module snd-maestro3
+ -------------------
+
+ Module for Allegro/Maestro3 chips
+
+ external_amp - enable external amp (enabled by default)
+ amp_gpio - GPIO pin number for external amp (0-15) or
+ -1 for default pin (8 for allegro, 1 for
+ others)
+
+ Module supports autoprobe and multiple chips (max 8).
+
+ Note: the binding of amplifier is dependent on hardware.
+ If there is no sound even though all channels are unmuted, try to
+ specify other gpio connection via amp_gpio option.
+ For example, a Panasonic notebook might need "amp_gpio=0x0d"
+ option.
+
+ The power-management is supported.
+
+ Module snd-mixart
+ -----------------
+
+ Module for Digigram miXart8 sound cards.
+
+ Module supports multiple cards.
+ Note: One miXart8 board will be represented as 4 alsa cards.
+ See MIXART.txt for details.
+
+ When the driver is compiled as a module and the hotplug firmware
+ is supported, the firmware data is loaded via hotplug automatically.
+ Install the necessary firmware files in alsa-firmware package.
+ When no hotplug fw loader is available, you need to load the
+ firmware via mixartloader utility in alsa-tools package.
+
+ Module snd-mpu401
+ -----------------
+
+ Module for MPU-401 UART devices.
+
+ port - port number or -1 (disable)
+ irq - IRQ number or -1 (disable)
+ pnp - PnP detection - 0 = disable, 1 = enable (default)
+
+ Module supports multiple devices (max 8) and PnP.
+
+ Module snd-mtpav
+ ----------------
+
+ Module for MOTU MidiTimePiece AV multiport MIDI (on the parallel
+ port).
+
+ port - I/O port # for MTPAV (0x378,0x278, default=0x378)
+ irq - IRQ # for MTPAV (7,5, default=7)
+ hwports - number of supported hardware ports, default=8.
+
+ Module supports only 1 card. This module has no enable option.
+
+ Module snd-nm256
+ ----------------
+
+ Module for NeoMagic NM256AV/ZX chips
+
+ playback_bufsize - max playback frame size in kB (4-128kB)
+ capture_bufsize - max capture frame size in kB (4-128kB)
+ force_ac97 - 0 or 1 (disabled by default)
+ buffer_top - specify buffer top address
+ use_cache - 0 or 1 (disabled by default)
+ vaio_hack - alias buffer_top=0x25a800
+ reset_workaround - enable AC97 RESET workaround for some laptops
+
+ Module supports autoprobe and multiple chips (max 8).
+
+ The power-management is supported.
+
+ Note: on some notebooks the buffer address cannot be detected
+ automatically, or causes hang-up during initialization.
+ In such a case, specify the buffer top address explicity via
+ buffer_top option.
+ For example,
+ Sony F250: buffer_top=0x25a800
+ Sony F270: buffer_top=0x272800
+ The driver supports only ac97 codec. It's possible to force
+ to initialize/use ac97 although it's not detected. In such a
+ case, use force_ac97=1 option - but *NO* guarantee whether it
+ works!
+
+ Note: The NM256 chip can be linked internally with non-AC97
+ codecs. This driver supports only the AC97 codec, and won't work
+ with machines with other (most likely CS423x or OPL3SAx) chips,
+ even though the device is detected in lspci. In such a case, try
+ other drivers, e.g. snd-cs4232 or snd-opl3sa2. Some has ISA-PnP
+ but some doesn't have ISA PnP. You'll need to speicfy isapnp=0
+ and proper hardware parameters in the case without ISA PnP.
+
+ Note: some laptops need a workaround for AC97 RESET. For the
+ known hardware like Dell Latitude LS and Sony PCG-F305, this
+ workaround is enabled automatically. For other laptops with a
+ hard freeze, you can try reset_workaround=1 option.
+
+ Note: This driver is really crappy. It's a porting from the
+ OSS driver, which is a result of black-magic reverse engineering.
+ The detection of codec will fail if the driver is loaded *after*
+ X-server as described above. You might be able to force to load
+ the module, but it may result in hang-up. Hence, make sure that
+ you load this module *before* X if you encounter this kind of
+ problem.
+
+ Module snd-opl3sa2
+ ------------------
+
+ Module for Yamaha OPL3-SA2/SA3 sound cards.
+
+ port - control port # for OPL3-SA chip (0x370)
+ sb_port - SB port # for OPL3-SA chip (0x220,0x240)
+ wss_port - WSS port # for OPL3-SA chip (0x530,0xe80,0xf40,0x604)
+ midi_port - port # for MPU-401 UART (0x300,0x330), -1 = disable
+ fm_port - FM port # for OPL3-SA chip (0x388), -1 = disable
+ irq - IRQ # for OPL3-SA chip (5,7,9,10)
+ dma1 - first DMA # for Yamaha OPL3-SA chip (0,1,3)
+ dma2 - second DMA # for Yamaha OPL3-SA chip (0,1,3), -1 = disable
+ isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
+
+ Module supports up to 8 cards and ISA PnP. This module does not support
+ autoprobe (if ISA PnP is not used) thus all ports must be specified!!!
+
+ The power-management is supported.
+
+ Module snd-opti92x-ad1848
+ -------------------------
+
+ Module for sound cards based on OPTi 82c92x and Analog Devices AD1848 chips.
+ Module works with OAK Mozart cards as well.
+
+ port - port # for WSS chip (0x530,0xe80,0xf40,0x604)
+ mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330)
+ fm_port - port # for OPL3 device (0x388)
+ irq - IRQ # for WSS chip (5,7,9,10,11)
+ mpu_irq - IRQ # for MPU-401 UART (5,7,9,10)
+ dma1 - first DMA # for WSS chip (0,1,3)
+
+ This module supports only one card, autoprobe and PnP.
+
+ Module snd-opti92x-cs4231
+ -------------------------
+
+ Module for sound cards based on OPTi 82c92x and Crystal CS4231 chips.
+
+ port - port # for WSS chip (0x530,0xe80,0xf40,0x604)
+ mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330)
+ fm_port - port # for OPL3 device (0x388)
+ irq - IRQ # for WSS chip (5,7,9,10,11)
+ mpu_irq - IRQ # for MPU-401 UART (5,7,9,10)
+ dma1 - first DMA # for WSS chip (0,1,3)
+ dma2 - second DMA # for WSS chip (0,1,3)
+
+ This module supports only one card, autoprobe and PnP.
+
+ Module snd-opti93x
+ ------------------
+
+ Module for sound cards based on OPTi 82c93x chips.
+
+ port - port # for WSS chip (0x530,0xe80,0xf40,0x604)
+ mpu_port - port # for MPU-401 UART (0x300,0x310,0x320,0x330)
+ fm_port - port # for OPL3 device (0x388)
+ irq - IRQ # for WSS chip (5,7,9,10,11)
+ mpu_irq - IRQ # for MPU-401 UART (5,7,9,10)
+ dma1 - first DMA # for WSS chip (0,1,3)
+ dma2 - second DMA # for WSS chip (0,1,3)
+
+ This module supports only one card, autoprobe and PnP.
+
+ Module snd-powermac (on ppc only)
+ ---------------------------------
+
+ Module for PowerMac, iMac and iBook on-board soundchips
+
+ enable_beep - enable beep using PCM (enabled as default)
+
+ Module supports autoprobe a chip.
+
+ Note: the driver may have problems regarding endianess.
+
+ The power-management is supported.
+
+ Module snd-rme32
+ ----------------
+
+ Module for RME Digi32, Digi32 Pro and Digi32/8 (Sek'd Prodif32,
+ Prodif96 and Prodif Gold) sound cards.
+
+ Module supports up to 8 cards.
+
+ Module snd-rme96
+ ----------------
+
+ Module for RME Digi96, Digi96/8 and Digi96/8 PRO/PAD/PST sound cards.
+
+ Module supports up to 8 cards.
+
+ Module snd-rme9652
+ ------------------
+
+ Module for RME Digi9652 (Hammerfall, Hammerfall-Light) sound cards.
+
+ precise_ptr - Enable precise pointer (doesn't work reliably).
+ (default = 0)
+
+ Module supports up to 8 cards.
+
+ Note: snd-page-alloc module does the job which snd-hammerfall-mem
+ 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.
+
+ Module snd-sa11xx-uda1341 (on arm only)
+ ---------------------------------------
+
+ Module for Philips UDA1341TS on Compaq iPAQ H3600 sound card.
+
+ Module supports only one card.
+ Module has no enable and index options.
+
+ Module snd-sb8
+ --------------
+
+ Module for 8-bit SoundBlaster cards: SoundBlaster 1.0,
+ SoundBlaster 2.0,
+ SoundBlaster Pro
+
+ port - port # for SB DSP chip (0x220,0x240,0x260)
+ irq - IRQ # for SB DSP chip (5,7,9,10)
+ dma8 - DMA # for SB DSP chip (1,3)
+
+ Module supports up to 8 cards and autoprobe.
+
+ Module snd-sb16 and snd-sbawe
+ -----------------------------
+
+ Module for 16-bit SoundBlaster cards: SoundBlaster 16 (PnP),
+ SoundBlaster AWE 32 (PnP),
+ SoundBlaster AWE 64 PnP
+
+ port - port # for SB DSP 4.x chip (0x220,0x240,0x260)
+ mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disable
+ awe_port - base port # for EMU8000 synthesizer (0x620,0x640,0x660)
+ (snd-sbawe module only)
+ irq - IRQ # for SB DSP 4.x chip (5,7,9,10)
+ dma8 - 8-bit DMA # for SB DSP 4.x chip (0,1,3)
+ dma16 - 16-bit DMA # for SB DSP 4.x chip (5,6,7)
+ mic_agc - Mic Auto-Gain-Control - 0 = disable, 1 = enable (default)
+ csp - ASP/CSP chip support - 0 = disable (default), 1 = enable
+ isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
+
+ Module supports up to 8 cards, autoprobe and ISA PnP.
+
+ Note: To use Vibra16X cards in 16-bit half duplex mode, you must
+ disable 16bit DMA with dma16 = -1 module parameter.
+ Also, all Sound Blaster 16 type cards can operate in 16-bit
+ half duplex mode through 8-bit DMA channel by disabling their
+ 16-bit DMA channel.
+
+ Module snd-sgalaxy
+ ------------------
+
+ Module for Aztech Sound Galaxy sound card.
+
+ sbport - Port # for SB16 interface (0x220,0x240)
+ wssport - Port # for WSS interface (0x530,0xe80,0xf40,0x604)
+ irq - IRQ # (7,9,10,11)
+ dma1 - DMA #
+
+ Module supports up to 8 cards.
+
+ Module snd-sscape
+ -----------------
+
+ Module for ENSONIQ SoundScape PnP cards.
+
+ port - Port # (PnP setup)
+ irq - IRQ # (PnP setup)
+ mpu_irq - MPU-401 IRQ # (PnP setup)
+ dma - DMA # (PnP setup)
+
+ Module supports up to 8 cards. ISA PnP must be enabled.
+ You need sscape_ctl tool in alsa-tools package for loading
+ the microcode.
+
+ Module snd-sun-amd7930 (on sparc only)
+ --------------------------------------
+
+ Module for AMD7930 sound chips found on Sparcs.
+
+ Module supports up to 8 cards.
+
+ Module snd-sun-cs4231 (on sparc only)
+ -------------------------------------
+
+ Module for CS4231 sound chips found on Sparcs.
+
+ Module supports up to 8 cards.
+
+ Module snd-wavefront
+ --------------------
+
+ Module for Turtle Beach Maui, Tropez and Tropez+ sound cards.
+
+ cs4232_pcm_port - Port # for CS4232 PCM interface.
+ cs4232_pcm_irq - IRQ # for CS4232 PCM interface (5,7,9,11,12,15).
+ cs4232_mpu_port - Port # for CS4232 MPU-401 interface.
+ cs4232_mpu_irq - IRQ # for CS4232 MPU-401 interface (9,11,12,15).
+ use_cs4232_midi - Use CS4232 MPU-401 interface
+ (inaccessibly located inside your computer)
+ ics2115_port - Port # for ICS2115
+ ics2115_irq - IRQ # for ICS2115
+ fm_port - FM OPL-3 Port #
+ dma1 - DMA1 # for CS4232 PCM interface.
+ dma2 - DMA2 # for CS4232 PCM interface.
+ isapnp - ISA PnP detection - 0 = disable, 1 = enable (default)
+
+ Module supports up to 8 cards and ISA PnP.
+
+ Module snd-sonicvibes
+ ---------------------
+
+ Module for S3 SonicVibes PCI sound cards.
+ * PINE Schubert 32 PCI
+
+ reverb - Reverb Enable - 1 = enable, 0 = disable (default)
+ - SoundCard must have onboard SRAM for this.
+ mge - Mic Gain Enable - 1 = enable, 0 = disable (default)
+
+ Module supports up to 8 cards and autoprobe.
+
+ Module snd-serial-u16550
+ ------------------------
+
+ Module for UART16550A serial MIDI ports.
+
+ port - port # for UART16550A chip
+ irq - IRQ # for UART16550A chip, -1 = poll mode
+ speed - speed in bauds (9600,19200,38400,57600,115200)
+ 38400 = default
+ base - base for divisor in bauds (57600,115200,230400,460800)
+ 115200 = default
+ outs - number of MIDI ports in a serial port (1-4)
+ 1 = default
+ adaptor - Type of adaptor.
+ 0 = Soundcanvas, 1 = MS-124T, 2 = MS-124W S/A,
+ 3 = MS-124W M/B, 4 = Generic
+
+ Module supports up to 8 cards. This module does not support autoprobe
+ thus the main port must be specified!!! Other options are optional.
+
+ Module snd-trident
+ ------------------
+
+ Module for Trident 4DWave DX/NX sound cards.
+ * Best Union Miss Melody 4DWave PCI
+ * HIS 4DWave PCI
+ * Warpspeed ONSpeed 4DWave PCI
+ * AzTech PCI 64-Q3D
+ * Addonics SV 750
+ * CHIC True Sound 4Dwave
+ * Shark Predator4D-PCI
+ * Jaton SonicWave 4D
+
+ pcm_channels - max channels (voices) reserved for PCM
+ wavetable_size - max wavetable size in kB (4-?kb)
+
+ Module supports up to 8 cards and autoprobe.
+
+ The power-management is supported.
+
+ Module snd-usb-audio
+ --------------------
+
+ Module for USB audio and USB MIDI devices.
+
+ vid - Vendor ID for the device (optional)
+ pid - Product ID for the device (optional)
+
+ This module supports up to 8 cards, autoprobe and hotplugging.
+
+ Module snd-usb-usx2y
+ --------------------
+
+ Module for Tascam USB US-122, US-224 and US-428 devices.
+
+ This module supports up to 8 cards, autoprobe and hotplugging.
+
+ Note: you need to load the firmware via usx2yloader utility included
+ in alsa-tools and alsa-firmware packages.
+
+ Module snd-via82xx
+ ------------------
+
+ Module for AC'97 motherboards based on VIA 82C686A/686B, 8233,
+ 8233A, 8233C, 8235 (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]
+ ac97_quirk - AC'97 workaround for strange hardware
+ See the description of intel8x0 module for details.
+
+ Module supports autoprobe and multiple bus-master chips (max 8).
+
+ Note: on some SMP motherboards like MSI 694D the interrupts might
+ not be generated properly. In such a case, please try to
+ set the SMP (or MPS) version on BIOS to 1.1 instead of
+ 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)
+ channels as the first PCM. On these channels, up to 4
+ streams can be played at the same time.
+ 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
+ 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
+ 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.
+ In any cases, please let us know the result and the
+ subsystem vendor/device ids.
+
+ Note: for the MPU401 on VIA823x, use snd-mpu401 driver
+ additonally. The mpu_port option is for VIA686 chips only.
+
+ Module snd-via82xx-modem
+ ------------------------
+
+ Module for VIA82xx AC97 modem
+
+ ac97_clock - AC'97 codec clock base (default 48000Hz)
+
+ Module supports up to 8 cards.
+
+ Note: The default index value of this module is -2, i.e. the first
+ slot is excluded.
+
+ Module snd-virmidi
+ ------------------
+
+ Module for virtual rawmidi devices.
+ This module creates virtual rawmidi devices which communicate
+ to the corresponding ALSA sequencer ports.
+
+ midi_devs - MIDI devices # (1-8, default=4)
+
+ Module supports up to 8 cards.
+
+ Module snd-vx222
+ ----------------
+
+ Module for Digigram VX-Pocket VX222, V222 v2 and Mic cards.
+
+ mic - Enable Microphone on V222 Mic (NYI)
+ ibl - Capture IBL size. (default = 0, minimum size)
+
+ Module supports up to 8 cards.
+
+ When the driver is compiled as a module and the hotplug firmware
+ is supported, the firmware data is loaded via hotplug automatically.
+ Install the necessary firmware files in alsa-firmware package.
+ When no hotplug fw loader is available, you need to load the
+ firmware via vxloader utility in alsa-tools package. To invoke
+ vxloader automatically, add the following to /etc/modprobe.conf
+
+ install snd-vx222 /sbin/modprobe --first-time -i snd-vx222 && /usr/bin/vxloader
+
+ (for 2.2/2.4 kernels, add "post-install /usr/bin/vxloader" to
+ /etc/modules.conf, instead.)
+ IBL size defines the interrupts period for PCM. The smaller size
+ gives smaller latency but leads to more CPU consumption, too.
+ The size is usually aligned to 126. As default (=0), the smallest
+ size is chosen. The possible IBL values can be found in
+ /proc/asound/cardX/vx-status proc file.
+
+ Module snd-vxpocket
+ -------------------
+
+ Module for Digigram VX-Pocket VX2 PCMCIA card.
+
+ ibl - Capture IBL size. (default = 0, minimum size)
+
+ Module supports up to 8 cards. The module is compiled only when
+ PCMCIA is supported on kernel.
+
+ To activate the driver via the card manager, you'll need to set
+ up /etc/pcmcia/vxpocket.conf. See the sound/pcmcia/vx/vxpocket.c.
+
+ When the driver is compiled as a module and the hotplug firmware
+ is supported, the firmware data is loaded via hotplug automatically.
+ Install the necessary firmware files in alsa-firmware package.
+ When no hotplug fw loader is available, you need to load the
+ firmware via vxloader utility in alsa-tools package.
+
+ About capture IBL, see the description of snd-vx222 module.
+
+ Note: the driver is build only when CONFIG_ISA is set.
+
+ Module snd-vxp440
+ -----------------
+
+ Module for Digigram VX-Pocket 440 PCMCIA card.
+
+ ibl - Capture IBL size. (default = 0, minimum size)
+
+ Module supports up to 8 cards. The module is compiled only when
+ PCMCIA is supported on kernel.
+
+ To activate the driver via the card manager, you'll need to set
+ up /etc/pcmcia/vxp440.conf. See the sound/pcmcia/vx/vxp440.c.
+
+ When the driver is compiled as a module and the hotplug firmware
+ is supported, the firmware data is loaded via hotplug automatically.
+ Install the necessary firmware files in alsa-firmware package.
+ When no hotplug fw loader is available, you need to load the
+ firmware via vxloader utility in alsa-tools package.
+
+ About capture IBL, see the description of snd-vx222 module.
+
+ Note: the driver is build only when CONFIG_ISA is set.
+
+ Module snd-ymfpci
+ -----------------
+
+ Module for Yamaha PCI chips (YMF72x, YMF74x & YMF75x).
+
+ mpu_port - 0x300,0x330,0x332,0x334, 0 (disable) by default,
+ 1 (auto-detect for YMF744/754 only)
+ fm_port - 0x388,0x398,0x3a0,0x3a8, 0 (disable) by default
+ 1 (auto-detect for YMF744/754 only)
+ joystick_port - 0x201,0x202,0x204,0x205, 0 (disable) by default,
+ 1 (auto-detect)
+ rear_switch - enable shared rear/line-in switch (bool)
+
+ Module supports autoprobe and multiple chips (max 8).
+
+ The power-management is supported.
+
+ Module snd-pdaudiocf
+ --------------------
+
+ Module for Sound Core PDAudioCF sound card.
+
+ Note: the driver is build only when CONFIG_ISA is set.
+
+
+Configuring Non-ISAPNP Cards
+============================
+
+When the kernel is configured with ISA-PnP support, the modules
+supporting the isapnp cards will have module options "isapnp".
+If this option is set, *only* the ISA-PnP devices will be probed.
+For probing the non ISA-PnP cards, you have to pass "isapnp=0" option
+together with the proper i/o and irq configuration.
+
+When the kernel is configured without ISA-PnP support, isapnp option
+will be not built in.
+
+
+Module Autoloading Support
+==========================
+
+The ALSA drivers can be loaded automatically on demand by defining
+module aliases. The string 'snd-card-%1' is requested for ALSA native
+devices where %i is sound card number from zero to seven.
+
+To auto-load an ALSA driver for OSS services, define the string
+'sound-slot-%i' where %i means the slot number for OSS, which
+corresponds to the card index of ALSA. Usually, define this
+as the the same card module.
+
+An example configuration for a single emu10k1 card is like below:
+----- /etc/modprobe.conf
+alias snd-card-0 snd-emu10k1
+alias sound-slot-0 snd-emu10k1
+----- /etc/modprobe.conf
+
+The available number of auto-loaded sound cards depends on the module
+option "cards_limit" of snd module. As default it's set to 1.
+To enable the auto-loading of multiple cards, specify the number of
+sound cards in that option.
+
+When multiple cards are available, it'd better to specify the index
+number for each card via module option, too, so that the order of
+cards is kept consistent.
+
+An example configuration for two sound cards is like below:
+
+----- /etc/modprobe.conf
+# ALSA portion
+options snd cards_limit=2
+alias snd-card-0 snd-interwave
+alias snd-card-1 snd-ens1371
+options snd-interwave index=0
+options snd-ens1371 index=1
+# OSS/Free portion
+alias sound-slot-0 snd-interwave
+alias sound-slot-1 snd-ens1371
+----- /etc/moprobe.conf
+
+In this example, the interwave card is always loaded as the first card
+(index 0) and ens1371 as the second (index 1).
+
+
+ALSA PCM devices to OSS devices mapping
+=======================================
+
+/dev/snd/pcmC0D0[c|p] -> /dev/audio0 (/dev/audio) -> minor 4
+/dev/snd/pcmC0D0[c|p] -> /dev/dsp0 (/dev/dsp) -> minor 3
+/dev/snd/pcmC0D1[c|p] -> /dev/adsp0 (/dev/adsp) -> minor 12
+/dev/snd/pcmC1D0[c|p] -> /dev/audio1 -> minor 4+16 = 20
+/dev/snd/pcmC1D0[c|p] -> /dev/dsp1 -> minor 3+16 = 19
+/dev/snd/pcmC1D1[c|p] -> /dev/adsp1 -> minor 12+16 = 28
+/dev/snd/pcmC2D0[c|p] -> /dev/audio2 -> minor 4+32 = 36
+/dev/snd/pcmC2D0[c|p] -> /dev/dsp2 -> minor 3+32 = 39
+/dev/snd/pcmC2D1[c|p] -> /dev/adsp2 -> minor 12+32 = 44
+
+The first number from /dev/snd/pcmC{X}D{Y}[c|p] expression means
+sound card number and second means device number. The ALSA devices
+have either 'c' or 'p' suffix indicating the direction, capture and
+playback, respectively.
+
+Please note that the device mapping above may be varied via the module
+options of snd-pcm-oss module.
+
+
+DEVFS support
+=============
+
+The ALSA driver fully supports the devfs extension.
+You should add lines below to your devfsd.conf file:
+
+LOOKUP snd MODLOAD ACTION snd
+REGISTER ^sound/.* PERMISSIONS root.audio 660
+REGISTER ^snd/.* PERMISSIONS root.audio 660
+
+Warning: These lines assume that you have the audio group in your system.
+ Otherwise replace audio word with another group name (root for
+ example).
+
+
+Proc interfaces (/proc/asound)
+==============================
+
+/proc/asound/card#/pcm#[cp]/oss
+-------------------------------
+ String "erase" - erase all additional informations about OSS applications
+ String "<app_name> <fragments> <fragment_size> [<options>]"
+
+ <app_name> - name of application with (higher priority) or without path
+ <fragments> - number of fragments or zero if auto
+ <fragment_size> - size of fragment in bytes or zero if auto
+ <options> - optional parameters
+ - disable the application tries to open a pcm device for
+ this channel but does not want to use it.
+ (Cause a bug or mmap needs)
+ It's good for Quake etc...
+ - direct don't use plugins
+ - block force block mode (rvplayer)
+ - non-block force non-block mode
+ - whole-frag write only whole fragments (optimization affecting
+ playback only)
+ - no-silence do not fill silence ahead to avoid clicks
+
+ Example: echo "x11amp 128 16384" > /proc/asound/card0/pcm0p/oss
+ echo "squake 0 0 disable" > /proc/asound/card0/pcm0c/oss
+ echo "rvplayer 0 0 block" > /proc/asound/card0/pcm0p/oss
+
+
+Links
+=====
+
+ ALSA project homepage
+ http://www.alsa-project.org
+
diff --git a/Documentation/sound/alsa/Audigy-mixer.txt b/Documentation/sound/alsa/Audigy-mixer.txt
new file mode 100644
index 000000000000..5132fd95e074
--- /dev/null
+++ b/Documentation/sound/alsa/Audigy-mixer.txt
@@ -0,0 +1,345 @@
+
+ Sound Blaster Audigy mixer / default DSP code
+ ===========================================
+
+This is based on SB-Live-mixer.txt.
+
+The EMU10K2 chips have a DSP part which can be programmed to support
+various ways of sample processing, which is described here.
+(This acticle does not deal with the overall functionality of the
+EMU10K2 chips. See the manuals section for further details.)
+
+The ALSA driver programs this portion of chip by default code
+(can be altered later) which offers the following functionality:
+
+
+1) Digital mixer controls
+-------------------------
+
+These controls are built using the DSP instructions. They offer extended
+functionality. Only the default build-in code in the ALSA driver is described
+here. Note that the controls work as attenuators: the maximum value is the
+neutral position leaving the signal unchanged. Note that if the same destination
+is mentioned in multiple controls, the signal is accumulated and can be wrapped
+(set to maximal or minimal value without checking of overflow).
+
+
+Explanation of used abbreviations:
+
+DAC - digital to analog converter
+ADC - analog to digital converter
+I2S - one-way three wire serial bus for digital sound by Philips Semiconductors
+ (this standard is used for connecting standalone DAC and ADC converters)
+LFE - low frequency effects (subwoofer signal)
+AC97 - a chip containing an analog mixer, DAC and ADC converters
+IEC958 - S/PDIF
+FX-bus - the EMU10K2 chip has an effect bus containing 64 accumulators.
+ Each of the synthesizer voices can feed its output to these accumulators
+ and the DSP microcontroller can operate with the resulting sum.
+
+name='PCM Front Playback Volume',index=0
+
+This control is used to attenuate samples for left and right front PCM FX-bus
+accumulators. ALSA uses accumulators 8 and 9 for left and right front PCM
+samples for 5.1 playback. The result samples are forwarded to the front DAC PCM
+slots of the Philips DAC.
+
+name='PCM Surround Playback Volume',index=0
+
+This control is used to attenuate samples for left and right surround PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right surround PCM
+samples for 5.1 playback. The result samples are forwarded to the surround DAC PCM
+slots of the Philips DAC.
+
+name='PCM Center Playback Volume',index=0
+
+This control is used to attenuate samples for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample for 5.1 playback. The result sample
+is forwarded to the center DAC PCM slot of the Philips DAC.
+
+name='PCM LFE Playback Volume',index=0
+
+This control is used to attenuate sample for LFE PCM FX-bus accumulator.
+ALSA uses accumulator 7 for LFE PCM sample for 5.1 playback. The result sample
+is forwarded to the LFE DAC PCM slot of the Philips DAC.
+
+name='PCM Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples for
+stereo playback. The result samples are forwarded to the front DAC PCM slots
+of the Philips DAC.
+
+name='PCM Capture Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Music Playback Volume',index=0
+
+This control is used to attenuate samples for left and right MIDI FX-bus
+accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Music Capture Volume',index=0
+
+These controls are used to attenuate samples for left and right MIDI FX-bus
+accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Mic Playback Volume',index=0
+
+This control is used to attenuate samples for left and right Mic input.
+For Mic input is used AC97 codec. The result samples are forwarded to
+the front DAC PCM slots of the Philips DAC. Samples are forwarded to Mic
+capture FIFO (device 1 - 16bit/8KHz mono) too without volume control.
+
+name='Mic Capture Volume',index=0
+
+This control is used to attenuate samples for left and right Mic input.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Audigy CD Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the Philips DAC.
+
+name='Audigy CD Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+name='IEC958 Optical Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 optical
+digital input. The result samples are forwarded to the front DAC PCM slots
+of the Philips DAC.
+
+name='IEC958 Optical Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 optical
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+name='Line2 Playback Volume',index=0
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC.
+
+name='Line2 Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Analog Mix Playback Volume',index=0
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs from Philips ADC. The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC. This contains mix from analog sources
+like CD, Line In, Aux, ....
+
+name='Analog Mix Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs Philips ADC. The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Aux2 Playback Volume',index=0
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the front
+DAC PCM slots of the Philips DAC.
+
+name='Aux2 Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the AudigyDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Front Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate samples for left and right front speakers of
+this mix.
+
+name='Surround Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate samples for left and right surround speakers of
+this mix.
+
+name='Center Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate sample for center speaker of this mix.
+
+name='LFE Playback Volume',index=0
+
+All stereo signals are mixed together and mirrored to surround, center and LFE.
+This control is used to attenuate sample for LFE speaker of this mix.
+
+name='Tone Control - Switch',index=0
+
+This control turns the tone control on or off. The samples for front, rear
+and center / LFE outputs are affected.
+
+name='Tone Control - Bass',index=0
+
+This control sets the bass intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Tone Control - Treble',index=0
+
+This control sets the treble intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Master Playback Volume',index=0
+
+This control is used to attenuate samples for front, surround, center and
+LFE outputs.
+
+name='IEC958 Optical Raw Playback Switch',index=0
+
+If this switch is on, then the samples for the IEC958 (S/PDIF) digital
+output are taken only from the raw FX8010 PCM, otherwise standard front
+PCM samples are taken.
+
+
+2) PCM stream related controls
+------------------------------
+
+name='EMU10K1 PCM Volume',index 0-31
+
+Channel volume attenuation in range 0-0xffff. The maximum value (no
+attenuation) is default. The channel mapping for three values is
+as follows:
+
+ 0 - mono, default 0xffff (no attenuation)
+ 1 - left, default 0xffff (no attenuation)
+ 2 - right, default 0xffff (no attenuation)
+
+name='EMU10K1 PCM Send Routing',index 0-31
+
+This control specifies the destination - FX-bus accumulators. There 24
+values with this mapping:
+
+ 0 - mono, A destination (FX-bus 0-63), default 0
+ 1 - mono, B destination (FX-bus 0-63), default 1
+ 2 - mono, C destination (FX-bus 0-63), default 2
+ 3 - mono, D destination (FX-bus 0-63), default 3
+ 4 - mono, E destination (FX-bus 0-63), default 0
+ 5 - mono, F destination (FX-bus 0-63), default 0
+ 6 - mono, G destination (FX-bus 0-63), default 0
+ 7 - mono, H destination (FX-bus 0-63), default 0
+ 8 - left, A destination (FX-bus 0-63), default 0
+ 9 - left, B destination (FX-bus 0-63), default 1
+ 10 - left, C destination (FX-bus 0-63), default 2
+ 11 - left, D destination (FX-bus 0-63), default 3
+ 12 - left, E destination (FX-bus 0-63), default 0
+ 13 - left, F destination (FX-bus 0-63), default 0
+ 14 - left, G destination (FX-bus 0-63), default 0
+ 15 - left, H destination (FX-bus 0-63), default 0
+ 16 - right, A destination (FX-bus 0-63), default 0
+ 17 - right, B destination (FX-bus 0-63), default 1
+ 18 - right, C destination (FX-bus 0-63), default 2
+ 19 - right, D destination (FX-bus 0-63), default 3
+ 20 - right, E destination (FX-bus 0-63), default 0
+ 21 - right, F destination (FX-bus 0-63), default 0
+ 22 - right, G destination (FX-bus 0-63), default 0
+ 23 - right, H destination (FX-bus 0-63), default 0
+
+Don't forget that it's illegal to assign a channel to the same FX-bus accumulator
+more than once (it means 0=0 && 1=0 is an invalid combination).
+
+name='EMU10K1 PCM Send Volume',index 0-31
+
+It specifies the attenuation (amount) for given destination in range 0-255.
+The channel mapping is following:
+
+ 0 - mono, A destination attn, default 255 (no attenuation)
+ 1 - mono, B destination attn, default 255 (no attenuation)
+ 2 - mono, C destination attn, default 0 (mute)
+ 3 - mono, D destination attn, default 0 (mute)
+ 4 - mono, E destination attn, default 0 (mute)
+ 5 - mono, F destination attn, default 0 (mute)
+ 6 - mono, G destination attn, default 0 (mute)
+ 7 - mono, H destination attn, default 0 (mute)
+ 8 - left, A destination attn, default 255 (no attenuation)
+ 9 - left, B destination attn, default 0 (mute)
+ 10 - left, C destination attn, default 0 (mute)
+ 11 - left, D destination attn, default 0 (mute)
+ 12 - left, E destination attn, default 0 (mute)
+ 13 - left, F destination attn, default 0 (mute)
+ 14 - left, G destination attn, default 0 (mute)
+ 15 - left, H destination attn, default 0 (mute)
+ 16 - right, A destination attn, default 0 (mute)
+ 17 - right, B destination attn, default 255 (no attenuation)
+ 18 - right, C destination attn, default 0 (mute)
+ 19 - right, D destination attn, default 0 (mute)
+ 20 - right, E destination attn, default 0 (mute)
+ 21 - right, F destination attn, default 0 (mute)
+ 22 - right, G destination attn, default 0 (mute)
+ 23 - right, H destination attn, default 0 (mute)
+
+
+
+4) MANUALS/PATENTS:
+-------------------
+
+ftp://opensource.creative.com/pub/doc
+-------------------------------------
+
+ Files:
+ LM4545.pdf AC97 Codec
+
+ m2049.pdf The EMU10K1 Digital Audio Processor
+
+ hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects
+
+
+WIPO Patents
+------------
+ Patent numbers:
+ WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999)
+ streams
+
+ WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
+
+ WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction
+ Execution and Audio Data Sequencing (Jan. 14, 1999)
+
+
+US Patents (http://www.uspto.gov/)
+----------------------------------
+
+ US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
+
+ US 5928342 Audio Effects Processor integrated on a single chip (Jul. 27, 1999)
+ with a multiport memory onto which multiple asynchronous
+ digital sound samples can be concurrently loaded
+
+ US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
+
+ US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000)
+
+ US 6138207 Interpolation looping of audio samples in cache connected to (Oct. 24, 2000)
+ system bus with prioritization and modification of bus transfers
+ in accordance with loop ends and minimum block sizes
+
+ US 6151670 Method for conserving memory storage using a (Nov. 21, 2000)
+ pool of short term memory registers
+
+ US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001)
+ a common interrupt by associating programs to GP registers,
+ defining interrupt register, polling GP registers, and invoking
+ callback routine associated with defined interrupt register
diff --git a/Documentation/sound/alsa/Bt87x.txt b/Documentation/sound/alsa/Bt87x.txt
new file mode 100644
index 000000000000..11edb2fd2a5a
--- /dev/null
+++ b/Documentation/sound/alsa/Bt87x.txt
@@ -0,0 +1,78 @@
+Intro
+=====
+
+You might have noticed that the bt878 grabber cards have actually
+_two_ PCI functions:
+
+$ lspci
+[ ... ]
+00:0a.0 Multimedia video controller: Brooktree Corporation Bt878 (rev 02)
+00:0a.1 Multimedia controller: Brooktree Corporation Bt878 (rev 02)
+[ ... ]
+
+The first does video, it is backward compatible to the bt848. The second
+does audio. snd-bt87x is a driver for the second function. It's a sound
+driver which can be used for recording sound (and _only_ recording, no
+playback). As most TV cards come with a short cable which can be plugged
+into your sound card's line-in you probably don't need this driver if all
+you want to do is just watching TV...
+
+Some cards do not bother to connect anything to the audio input pins of
+the chip, and some other cards use the audio function to transport MPEG
+video data, so it's quite possible that audio recording may not work
+with your card.
+
+
+Driver Status
+=============
+
+The driver is now stable. However, it doesn't know about many TV cards,
+and it refuses to load for cards it doesn't know.
+
+If the driver complains ("Unknown TV card found, the audio driver will
+not load"), you can specify the load_all=1 option to force the driver to
+try to use the audio capture function of your card. If the frequency of
+recorded data is not right, try to specify the digital_rate option with
+other values than the default 32000 (often it's 44100 or 64000).
+
+If you have an unknown card, please mail the ID and board name to
+<alsa-devel@lists.sf.net>, regardless of whether audio capture works or
+not, so that future versions of this driver know about your card.
+
+
+Audio modes
+===========
+
+The chip knows two different modes (digital/analog). snd-bt87x
+registers two PCM devices, one for each mode. They cannot be used at
+the same time.
+
+
+Digital audio mode
+==================
+
+The first device (hw:X,0) gives you 16 bit stereo sound. The sample
+rate depends on the external source which feeds the Bt87x with digital
+sound via I2S interface.
+
+
+Analog audio mode (A/D)
+=======================
+
+The second device (hw:X,1) gives you 8 or 16 bit mono sound. Supported
+sample rates are between 119466 and 448000 Hz (yes, these numbers are
+that high). If you've set the CONFIG_SND_BT87X_OVERCLOCK option, the
+maximum sample rate is 1792000 Hz, but audio data becomes unusable
+beyond 896000 Hz on my card.
+
+The chip has three analog inputs. Consequently you'll get a mixer
+device to control these.
+
+
+Have fun,
+
+ Clemens
+
+
+Written by Clemens Ladisch <clemens@ladisch.de>
+big parts copied from btaudio.txt by Gerd Knorr <kraxel@bytesex.org>
diff --git a/Documentation/sound/alsa/CMIPCI.txt b/Documentation/sound/alsa/CMIPCI.txt
new file mode 100644
index 000000000000..4a7df771b806
--- /dev/null
+++ b/Documentation/sound/alsa/CMIPCI.txt
@@ -0,0 +1,242 @@
+ Brief Notes on C-Media 8738/8338 Driver
+ =======================================
+
+ Takashi Iwai <tiwai@suse.de>
+
+
+Front/Rear Multi-channel Playback
+---------------------------------
+
+CM8x38 chip can use ADC as the second DAC so that two different stereo
+channels can be used for front/rear playbacks. Since there are two
+DACs, both streams are handled independently unlike the 4/6ch multi-
+channel playbacks in the section below.
+
+As default, ALSA driver assigns the first PCM device (i.e. hw:0,0 for
+card#0) for front and 4/6ch playbacks, while the second PCM device
+(hw:0,1) is assigned to the second DAC for rear playback.
+
+There are slight difference between two DACs.
+
+- The first DAC supports U8 and S16LE formats, while the second DAC
+ supports only S16LE.
+- The seconde DAC supports only two channel stereo.
+
+Please note that the CM8x38 DAC doesn't support continuous playback
+rate but only fixed rates: 5512, 8000, 11025, 16000, 22050, 32000,
+44100 and 48000 Hz.
+
+The rear output can be heard only when "Four Channel Mode" switch is
+disabled. Otherwise no signal will be routed to the rear speakers.
+As default it's turned on.
+
+*** WARNING ***
+When "Four Channel Mode" switch is off, the output from rear speakers
+will be FULL VOLUME regardless of Master and PCM volumes.
+This might damage your audio equipment. Please disconnect speakers
+before your turn off this switch.
+*** WARNING ***
+
+[ Well.. I once got the output with correct volume (i.e. same with the
+ front one) and was so excited. It was even with "Four Channel" bit
+ on and "double DAC" mode. Actually I could hear separate 4 channels
+ from front and rear speakers! But.. after reboot, all was gone.
+ It's a very pity that I didn't save the register dump at that
+ time.. Maybe there is an unknown register to achieve this... ]
+
+If your card has an extra output jack for the rear output, the rear
+playback should be routed there as default. If not, there is a
+control switch in the driver "Line-In As Rear", which you can change
+via alsamixer or somewhat else. When this switch is on, line-in jack
+is used as rear output.
+
+There are two more controls regarding to the rear output.
+The "Exchange DAC" switch is used to exchange front and rear playback
+routes, i.e. the 2nd DAC is output from front output.
+
+
+4/6 Multi-Channel Playback
+--------------------------
+
+The recent CM8738 chips support for the 4/6 multi-channel playback
+function. This is useful especially for AC3 decoding.
+
+When the multi-channel is supported, the driver name has a suffix
+"-MC" such like "CMI8738-MC6". You can check this name from
+/proc/asound/cards.
+
+When the 4/6-ch output is enabled, the second DAC accepts up to 6 (or
+4) channels. While the dual DAC supports two different rates or
+formats, the 4/6-ch playback supports only the same condition for all
+channels. Since the multi-channel playback mode uses both DACs, you
+cannot operate with full-duplex.
+
+The 4.0 and 5.1 modes are defined as the pcm "surround40" and "surround51"
+in alsa-lib. For example, you can play a WAV file with 6 channels like
+
+ % aplay -Dsurround51 sixchannels.wav
+
+For programmin the 4/6 channel playback, you need to specify the PCM
+channels as you like and set the format S16LE. For example, for playback
+with 4 channels,
+
+ snd_pcm_hw_params_set_access(pcm, hw, SND_PCM_ACCESS_RW_INTERLEAVED);
+ // or mmap if you like
+ snd_pcm_hw_params_set_format(pcm, hw, SND_PCM_FORMAT_S16_LE);
+ snd_pcm_hw_params_set_channels(pcm, hw, 4);
+
+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.
+
+
+Digital I/O
+-----------
+
+The CM8x38 provides the excellent SPDIF capability with very chip
+price (yes, that's the reason I bought the card :)
+
+The SPDIF playback and capture are done via the third PCM device
+(hw:0,2). Usually this is assigned to the PCM device "spdif".
+The available rates are 44100 and 48000 Hz.
+For playback with aplay, you can run like below:
+
+ % aplay -Dhw:0,2 foo.wav
+
+or
+
+ % aplay -Dspdif foo.wav
+
+24bit format is also supported experimentally.
+
+The playback and capture over SPDIF use normal DAC and ADC,
+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 :)
+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).
+
+You can play via SPDIF even with the first device (hw:0,0),
+but SPDIF is enabled only when the proper format (S16LE), sample rate
+(441100 or 48000) and channels (2) are used. Otherwise it's turned
+off. (Also don't forget to turn on "IEC958 Output Switch", too.)
+
+
+Additionally there are relevant control switches:
+
+"IEC958 Mix Analog" - Mix analog PCM playback and FM-OPL/3 streams and
+ output through SPDIF. This switch appears only on old chip
+ models (CM8738 033 and 037).
+ Note: without this control you can output PCM to SPDIF.
+ This is "mixing" of streams, so e.g. it's not for AC3 output
+ (see the next section).
+
+"IEC958 In Select" - Select SPDIF input, the internal CD-in (false)
+ and the external input (true).
+
+"IEC958 Loop" - SPDIF input data is loop back into SPDIF
+ output (aka bypass)
+
+"IEC958 Copyright" - Set the copyright bit.
+
+"IEC958 5V" - Select 0.5V (coax) or 5V (optical) interface.
+ On some cards this doesn't work and you need to change the
+ configuration with hardware dip-switch.
+
+"IEC958 In Monitor" - SPDIF input is routed to DAC.
+
+"IEC958 In Phase Inverse" - Set SPDIF input format as inverse.
+ [FIXME: this doesn't work on all chips..]
+
+"IEC958 In Valid" - Set input validity flag detection.
+
+Note: When "PCM Playback Switch" is on, you'll hear the digital output
+stream through analog line-out.
+
+
+The AC3 (RAW DIGITAL) OUTPUT
+----------------------------
+
+The driver supports raw digital (typically AC3) i/o over SPDIF. This
+can be toggled via IEC958 playback control, but usually you need to
+access it via alsa-lib. See alsa-lib documents for more details.
+
+On the raw digital mode, the "PCM Playback Switch" is automatically
+turned off so that non-audio data is heard from the analog line-out.
+Similarly the following switches are off: "IEC958 Mix Analog" and
+"IEC958 Loop". The switches are resumed after closing the SPDIF PCM
+device automatically to the previous state.
+
+On the model 033, AC3 is implemented by the software conversion in
+the alsa-lib. If you need to bypass the software conversion of IEC958
+subframes, pass the "soft_ac3=0" module option. This doesn't matter
+on the newer models.
+
+
+ANALOG MIXER INTERFACE
+----------------------
+
+The mixer interface on CM8x38 is similar to SB16.
+There are Master, PCM, Synth, CD, Line, Mic and PC Speaker playback
+volumes. Synth, CD, Line and Mic have playback and capture switches,
+too, as well as SB16.
+
+In addition to the standard SB mixer, CM8x38 provides more functions.
+- PCM playback switch
+- PCM capture switch (to capture the data sent to DAC)
+- Mic Boost switch
+- Mic capture volume
+- Aux playback volume/switch and capture switch
+- 3D control switch
+
+
+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.
+
+There is _no_ hardware wavetable function on this chip (except for
+OPL3 synth below).
+What's said as MIDI synth on Windows is a software synthesizer
+emulation. On Linux use TiMidity or other softsynth program for
+playing MIDI music.
+
+
+FM OPL/3 Synth
+--------------
+
+The FM OPL/3 is also enabled as default only for the first card.
+Set "fm_port" module option for more cards.
+
+The output quality of FM OPL/3 is, however, very weird.
+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.
+
+
+Debugging Information
+---------------------
+
+The registers are shown in /proc/asound/cardX/cmipci. If you have any
+problem (especially unexpected behavior of mixer), please attach the
+output of this proc file together with the bug report.
diff --git a/Documentation/sound/alsa/ControlNames.txt b/Documentation/sound/alsa/ControlNames.txt
new file mode 100644
index 000000000000..5b18298e9495
--- /dev/null
+++ b/Documentation/sound/alsa/ControlNames.txt
@@ -0,0 +1,84 @@
+This document describes standard names of mixer controls.
+
+Syntax: SOURCE [DIRECTION] FUNCTION
+
+DIRECTION:
+ <nothing> (both directions)
+ Playback
+ Capture
+ Bypass Playback
+ Bypass Capture
+
+FUNCTION:
+ Switch (on/off switch)
+ Volume
+ Route (route control, hardware specific)
+
+SOURCE:
+ Master
+ Master Mono
+ Hardware Master
+ Headphone
+ PC Speaker
+ Phone
+ Phone Input
+ Phone Output
+ Synth
+ FM
+ Mic
+ Line
+ CD
+ Video
+ Zoom Video
+ Aux
+ PCM
+ PCM Front
+ PCM Rear
+ PCM Pan
+ Loopback
+ Analog Loopback (D/A -> A/D loopback)
+ Digital Loopback (playback -> capture loopback - without analog path)
+ Mono
+ Mono Output
+ Multi
+ ADC
+ Wave
+ Music
+ I2S
+ IEC958
+
+Exceptions:
+ [Digital] Capture Source
+ [Digital] Capture Switch (aka input gain switch)
+ [Digital] Capture Volume (aka input gain volume)
+ [Digital] Playback Switch (aka output gain switch)
+ [Digital] Playback Volume (aka output gain volume)
+ Tone Control - Switch
+ Tone Control - Bass
+ Tone Control - Treble
+ 3D Control - Switch
+ 3D Control - Center
+ 3D Control - Depth
+ 3D Control - Wide
+ 3D Control - Space
+ 3D Control - Level
+ Mic Boost [(?dB)]
+
+PCM interface:
+
+ Sample Clock Source { "Word", "Internal", "AutoSync" }
+ Clock Sync Status { "Lock", "Sync", "No Lock" }
+ External Rate /* external capture rate */
+ Capture Rate /* capture rate taken from external source */
+
+IEC958 (S/PDIF) interface:
+
+ IEC958 [...] [Playback|Capture] Switch /* turn on/off the IEC958 interface */
+ IEC958 [...] [Playback|Capture] Volume /* digital volume control */
+ IEC958 [...] [Playback|Capture] Default /* default or global value - read/write */
+ IEC958 [...] [Playback|Capture] Mask /* consumer and professional mask */
+ IEC958 [...] [Playback|Capture] Con Mask /* consumer mask */
+ IEC958 [...] [Playback|Capture] Pro Mask /* professional mask */
+ IEC958 [...] [Playback|Capture] PCM Stream /* the settings assigned to a PCM stream */
+ IEC958 Q-subcode [Playback|Capture] Default /* Q-subcode bits */
+ IEC958 Preamble [Playback|Capture] Default /* burst preamble words (4*16bits) */
diff --git a/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl b/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl
new file mode 100644
index 000000000000..1f3ae3e32d69
--- /dev/null
+++ b/Documentation/sound/alsa/DocBook/alsa-driver-api.tmpl
@@ -0,0 +1,100 @@
+<!DOCTYPE book PUBLIC "-//OASIS//DTD DocBook V4.1//EN">
+
+<book>
+<?dbhtml filename="index.html">
+
+<!-- ****************************************************** -->
+<!-- Header -->
+<!-- ****************************************************** -->
+ <bookinfo>
+ <title>The ALSA Driver API</title>
+
+ <legalnotice>
+ <para>
+ This document is free; you can redistribute it and/or modify it
+ under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2 of the License, or
+ (at your option) any later version.
+ </para>
+
+ <para>
+ This document is distributed in the hope that it will be useful,
+ but <emphasis>WITHOUT ANY WARRANTY</emphasis>; without even the
+ implied warranty of <emphasis>MERCHANTABILITY or FITNESS FOR A
+ PARTICULAR PURPOSE</emphasis>. See the GNU General Public License
+ for more details.
+ </para>
+
+ <para>
+ You should have received a copy of the GNU General Public
+ License along with this program; if not, write to the Free
+ Software Foundation, Inc., 59 Temple Place, Suite 330, Boston,
+ MA 02111-1307 USA
+ </para>
+ </legalnotice>
+
+ </bookinfo>
+
+ <chapter><title>Management of Cards and Devices</title>
+ <sect1><title>Card Managment</title>
+!Esound/core/init.c
+ </sect1>
+ <sect1><title>Device Components</title>
+!Esound/core/device.c
+ </sect1>
+ <sect1><title>KMOD and Device File Entries</title>
+!Esound/core/sound.c
+ </sect1>
+ <sect1><title>Memory Management Helpers</title>
+!Esound/core/memory.c
+!Esound/core/memalloc.c
+ </sect1>
+ </chapter>
+ <chapter><title>PCM API</title>
+ <sect1><title>PCM Core</title>
+!Esound/core/pcm.c
+!Esound/core/pcm_lib.c
+!Esound/core/pcm_native.c
+ </sect1>
+ <sect1><title>PCM Format Helpers</title>
+!Esound/core/pcm_misc.c
+ </sect1>
+ <sect1><title>PCM Memory Managment</title>
+!Esound/core/pcm_memory.c
+ </sect1>
+ </chapter>
+ <chapter><title>Control/Mixer API</title>
+ <sect1><title>General Control Interface</title>
+!Esound/core/control.c
+ </sect1>
+ <sect1><title>AC97 Codec API</title>
+!Esound/pci/ac97/ac97_codec.c
+!Esound/pci/ac97/ac97_pcm.c
+ </sect1>
+ </chapter>
+ <chapter><title>MIDI API</title>
+ <sect1><title>Raw MIDI API</title>
+!Esound/core/rawmidi.c
+ </sect1>
+ <sect1><title>MPU401-UART API</title>
+!Esound/drivers/mpu401/mpu401_uart.c
+ </sect1>
+ </chapter>
+ <chapter><title>Proc Info API</title>
+ <sect1><title>Proc Info Interface</title>
+!Esound/core/info.c
+ </sect1>
+ </chapter>
+ <chapter><title>Miscellaneous Functions</title>
+ <sect1><title>Hardware-Dependent Devices API</title>
+!Esound/core/hwdep.c
+ </sect1>
+ <sect1><title>ISA DMA Helpers</title>
+!Esound/core/isadma.c
+ </sect1>
+ <sect1><title>Other Helper Macros</title>
+!Iinclude/sound/core.h
+ </sect1>
+ </chapter>
+
+</book>
diff --git a/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
new file mode 100644
index 000000000000..e789475304b6
--- /dev/null
+++ b/Documentation/sound/alsa/DocBook/writing-an-alsa-driver.tmpl
@@ -0,0 +1,6045 @@
+<!DOCTYPE book PUBLIC "-//OASIS//DTD DocBook V4.1//EN">
+
+<book>
+<?dbhtml filename="index.html">
+
+<!-- ****************************************************** -->
+<!-- Header -->
+<!-- ****************************************************** -->
+ <bookinfo>
+ <title>Writing an ALSA Driver</title>
+ <author>
+ <firstname>Takashi</firstname>
+ <surname>Iwai</surname>
+ <affiliation>
+ <address>
+ <email>tiwai@suse.de</email>
+ </address>
+ </affiliation>
+ </author>
+
+ <date>March 6, 2005</date>
+ <edition>0.3.4</edition>
+
+ <abstract>
+ <para>
+ This document describes how to write an ALSA (Advanced Linux
+ Sound Architecture) driver.
+ </para>
+ </abstract>
+
+ <legalnotice>
+ <para>
+ Copyright (c) 2002-2004 Takashi Iwai <email>tiwai@suse.de</email>
+ </para>
+
+ <para>
+ This document is free; you can redistribute it and/or modify it
+ under the terms of the GNU General Public License as published by
+ the Free Software Foundation; either version 2 of the License, or
+ (at your option) any later version.
+ </para>
+
+ <para>
+ This document is distributed in the hope that it will be useful,
+ but <emphasis>WITHOUT ANY WARRANTY</emphasis>; without even the
+ implied warranty of <emphasis>MERCHANTABILITY or FITNESS FOR A
+ PARTICULAR PURPOSE</emphasis>. See the GNU General Public License
+ for more details.
+ </para>
+
+ <para>
+ You should have received a copy of the GNU General Public
+ License along with this program; if not, write to the Free
+ Software Foundation, Inc., 59 Temple Place, Suite 330, Boston,
+ MA 02111-1307 USA
+ </para>
+ </legalnotice>
+
+ </bookinfo>
+
+<!-- ****************************************************** -->
+<!-- Preface -->
+<!-- ****************************************************** -->
+ <preface id="preface">
+ <title>Preface</title>
+ <para>
+ This document describes how to write an
+ <ulink url="http://www.alsa-project.org/"><citetitle>
+ ALSA (Advanced Linux Sound Architecture)</citetitle></ulink>
+ driver. The document focuses mainly on the PCI soundcard.
+ In the case of other device types, the API might
+ be different, too. However, at least the ALSA kernel API is
+ consistent, and therefore it would be still a bit help for
+ writing them.
+ </para>
+
+ <para>
+ The target of this document is ones who already have enough
+ skill of C language and have the basic knowledge of linux
+ kernel programming. This document doesn't explain the general
+ topics of linux kernel codes and doesn't cover the detail of
+ implementation of each low-level driver. It describes only how is
+ the standard way to write a PCI sound driver on ALSA.
+ </para>
+
+ <para>
+ If you are already familiar with the older ALSA ver.0.5.x, you
+ can check the drivers such as <filename>es1938.c</filename> or
+ <filename>maestro3.c</filename> which have also almost the same
+ code-base in the ALSA 0.5.x tree, so you can compare the differences.
+ </para>
+
+ <para>
+ This document is still a draft version. Any feedbacks and
+ corrections, please!!
+ </para>
+ </preface>
+
+
+<!-- ****************************************************** -->
+<!-- File Tree Structure -->
+<!-- ****************************************************** -->
+ <chapter id="file-tree">
+ <title>File Tree Structure</title>
+
+ <section id="file-tree-general">
+ <title>General</title>
+ <para>
+ The ALSA drivers are provided in the two ways.
+ </para>
+
+ <para>
+ One is the trees provided as a tarball or via cvs from the
+ ALSA's ftp site, and another is the 2.6 (or later) Linux kernel
+ tree. To synchronize both, the ALSA driver tree is split into
+ two different trees: alsa-kernel and alsa-driver. The former
+ contains purely the source codes for the Linux 2.6 (or later)
+ tree. This tree is designed only for compilation on 2.6 or
+ later environment. The latter, alsa-driver, contains many subtle
+ files for compiling the ALSA driver on the outside of Linux
+ kernel like configure script, the wrapper functions for older,
+ 2.2 and 2.4 kernels, to adapt the latest kernel API,
+ and additional drivers which are still in development or in
+ tests. The drivers in alsa-driver tree will be moved to
+ alsa-kernel (eventually 2.6 kernel tree) once when they are
+ finished and confirmed to work fine.
+ </para>
+
+ <para>
+ The file tree structure of ALSA driver is depicted below. Both
+ alsa-kernel and alsa-driver have almost the same file
+ structure, except for <quote>core</quote> directory. It's
+ named as <quote>acore</quote> in alsa-driver tree.
+
+ <example>
+ <title>ALSA File Tree Structure</title>
+ <literallayout>
+ sound
+ /core
+ /oss
+ /seq
+ /oss
+ /instr
+ /ioctl32
+ /include
+ /drivers
+ /mpu401
+ /opl3
+ /i2c
+ /l3
+ /synth
+ /emux
+ /pci
+ /(cards)
+ /isa
+ /(cards)
+ /arm
+ /ppc
+ /sparc
+ /usb
+ /pcmcia /(cards)
+ /oss
+ </literallayout>
+ </example>
+ </para>
+ </section>
+
+ <section id="file-tree-core-directory">
+ <title>core directory</title>
+ <para>
+ This directory contains the middle layer, that is, the heart
+ of ALSA drivers. In this directory, the native ALSA modules are
+ stored. The sub-directories contain different modules and are
+ dependent upon the kernel config.
+ </para>
+
+ <section id="file-tree-core-directory-oss">
+ <title>core/oss</title>
+
+ <para>
+ The codes for PCM and mixer OSS emulation modules are stored
+ in this directory. The rawmidi OSS emulation is included in
+ the ALSA rawmidi code since it's quite small. The sequencer
+ code is stored in core/seq/oss directory (see
+ <link linkend="file-tree-core-directory-seq-oss"><citetitle>
+ below</citetitle></link>).
+ </para>
+ </section>
+
+ <section id="file-tree-core-directory-ioctl32">
+ <title>core/ioctl32</title>
+
+ <para>
+ This directory contains the 32bit-ioctl wrappers for 64bit
+ architectures such like x86-64, ppc64 and sparc64. For 32bit
+ and alpha architectures, these are not compiled.
+ </para>
+ </section>
+
+ <section id="file-tree-core-directory-seq">
+ <title>core/seq</title>
+ <para>
+ This and its sub-directories are for the ALSA
+ sequencer. This directory contains the sequencer core and
+ primary sequencer modules such like snd-seq-midi,
+ snd-seq-virmidi, etc. They are compiled only when
+ <constant>CONFIG_SND_SEQUENCER</constant> is set in the kernel
+ config.
+ </para>
+ </section>
+
+ <section id="file-tree-core-directory-seq-oss">
+ <title>core/seq/oss</title>
+ <para>
+ This contains the OSS sequencer emulation codes.
+ </para>
+ </section>
+
+ <section id="file-tree-core-directory-deq-instr">
+ <title>core/seq/instr</title>
+ <para>
+ This directory contains the modules for the sequencer
+ instrument layer.
+ </para>
+ </section>
+ </section>
+
+ <section id="file-tree-include-directory">
+ <title>include directory</title>
+ <para>
+ This is the place for the public header files of ALSA drivers,
+ which are to be exported to the user-space, or included by
+ several files at different directories. Basically, the private
+ header files should not be placed in this directory, but you may
+ still find files there, due to historical reason :)
+ </para>
+ </section>
+
+ <section id="file-tree-drivers-directory">
+ <title>drivers directory</title>
+ <para>
+ This directory contains the codes shared among different drivers
+ on the different architectures. They are hence supposed not to be
+ architecture-specific.
+ For example, the dummy pcm driver and the serial MIDI
+ driver are found in this directory. In the sub-directories,
+ there are the codes for components which are independent from
+ bus and cpu architectures.
+ </para>
+
+ <section id="file-tree-drivers-directory-mpu401">
+ <title>drivers/mpu401</title>
+ <para>
+ The MPU401 and MPU401-UART modules are stored here.
+ </para>
+ </section>
+
+ <section id="file-tree-drivers-directory-opl3">
+ <title>drivers/opl3 and opl4</title>
+ <para>
+ The OPL3 and OPL4 FM-synth stuff is found here.
+ </para>
+ </section>
+ </section>
+
+ <section id="file-tree-i2c-directory">
+ <title>i2c directory</title>
+ <para>
+ This contains the ALSA i2c components.
+ </para>
+
+ <para>
+ Although there is a standard i2c layer on Linux, ALSA has its
+ own i2c codes for some cards, because the soundcard needs only a
+ simple operation and the standard i2c API is too complicated for
+ such a purpose.
+ </para>
+
+ <section id="file-tree-i2c-directory-l3">
+ <title>i2c/l3</title>
+ <para>
+ This is a sub-directory for ARM L3 i2c.
+ </para>
+ </section>
+ </section>
+
+ <section id="file-tree-synth-directory">
+ <title>synth directory</title>
+ <para>
+ This contains the synth middle-level modules.
+ </para>
+
+ <para>
+ So far, there is only Emu8000/Emu10k1 synth driver under
+ synth/emux sub-directory.
+ </para>
+ </section>
+
+ <section id="file-tree-pci-directory">
+ <title>pci directory</title>
+ <para>
+ This and its sub-directories hold the top-level card modules
+ for PCI soundcards and the codes specific to the PCI BUS.
+ </para>
+
+ <para>
+ The drivers compiled from a single file is stored directly on
+ pci directory, while the drivers with several source files are
+ stored on its own sub-directory (e.g. emu10k1, ice1712).
+ </para>
+ </section>
+
+ <section id="file-tree-isa-directory">
+ <title>isa directory</title>
+ <para>
+ This and its sub-directories hold the top-level card modules
+ for ISA soundcards.
+ </para>
+ </section>
+
+ <section id="file-tree-arm-ppc-sparc-directories">
+ <title>arm, ppc, and sparc directories</title>
+ <para>
+ These are for the top-level card modules which are
+ specific to each given architecture.
+ </para>
+ </section>
+
+ <section id="file-tree-usb-directory">
+ <title>usb directory</title>
+ <para>
+ This contains the USB-audio driver. On the latest version, the
+ USB MIDI driver is integrated together with usb-audio driver.
+ </para>
+ </section>
+
+ <section id="file-tree-pcmcia-directory">
+ <title>pcmcia directory</title>
+ <para>
+ The PCMCIA, especially PCCard drivers will go here. CardBus
+ drivers will be on pci directory, because its API is identical
+ with the standard PCI cards.
+ </para>
+ </section>
+
+ <section id="file-tree-oss-directory">
+ <title>oss directory</title>
+ <para>
+ The OSS/Lite source files are stored here on Linux 2.6 (or
+ later) tree. (In the ALSA driver tarball, it's empty, of course :)
+ </para>
+ </section>
+ </chapter>
+
+
+<!-- ****************************************************** -->
+<!-- Basic Flow for PCI Drivers -->
+<!-- ****************************************************** -->
+ <chapter id="basic-flow">
+ <title>Basic Flow for PCI Drivers</title>
+
+ <section id="basic-flow-outline">
+ <title>Outline</title>
+ <para>
+ The minimum flow of PCI soundcard is like the following:
+
+ <itemizedlist>
+ <listitem><para>define the PCI ID table (see the section
+ <link linkend="pci-resource-entries"><citetitle>PCI Entries
+ </citetitle></link>).</para></listitem>
+ <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>exit()</function> function to call <function>pci_unregister_driver()</function> function.</para></listitem>
+ </itemizedlist>
+ </para>
+ </section>
+
+ <section id="basic-flow-example">
+ <title>Full Code Example</title>
+ <para>
+ The code example is shown below. Some parts are kept
+ unimplemented at this moment but will be filled in the
+ succeeding sections. The numbers in comment lines of
+ <function>snd_mychip_probe()</function> function are the
+ markers.
+
+ <example>
+ <title>Basic Flow for PCI Drivers Example</title>
+ <programlisting>
+<![CDATA[
+ #include <sound/driver.h>
+ #include <linux/init.h>
+ #include <linux/pci.h>
+ #include <linux/slab.h>
+ #include <sound/core.h>
+ #include <sound/initval.h>
+
+ /* module parameters (see "Module Parameters") */
+ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+
+ /* definition of the chip-specific record */
+ typedef struct snd_mychip mychip_t;
+ struct snd_mychip {
+ snd_card_t *card;
+ // rest of implementation will be in the section
+ // "PCI Resource Managements"
+ };
+
+ /* chip-specific destructor
+ * (see "PCI Resource Managements")
+ */
+ static int snd_mychip_free(mychip_t *chip)
+ {
+ .... // will be implemented later...
+ }
+
+ /* component-destructor
+ * (see "Management of Cards and Components")
+ */
+ static int snd_mychip_dev_free(snd_device_t *device)
+ {
+ mychip_t *chip = device->device_data;
+ return snd_mychip_free(chip);
+ }
+
+ /* chip-specific constructor
+ * (see "Management of Cards and Components")
+ */
+ static int __devinit snd_mychip_create(snd_card_t *card,
+ struct pci_dev *pci,
+ mychip_t **rchip)
+ {
+ mychip_t *chip;
+ int err;
+ static snd_device_ops_t ops = {
+ .dev_free = snd_mychip_dev_free,
+ };
+
+ *rchip = NULL;
+
+ // check PCI availability here
+ // (see "PCI Resource Managements")
+ ....
+
+ /* allocate a chip-specific data with zero filled */
+ chip = kcalloc(1, sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL)
+ return -ENOMEM;
+
+ chip->card = card;
+
+ // rest of initialization here; will be implemented
+ // later, see "PCI Resource Managements"
+ ....
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
+ chip, &ops)) < 0) {
+ snd_mychip_free(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *rchip = chip;
+ return 0;
+ }
+
+ /* constructor -- see "Constructor" sub-section */
+ static int __devinit snd_mychip_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+ {
+ static int dev;
+ snd_card_t *card;
+ mychip_t *chip;
+ int err;
+
+ /* (1) */
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+
+ /* (2) */
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+ if (card == NULL)
+ return -ENOMEM;
+
+ /* (3) */
+ if ((err = snd_mychip_create(card, pci, &chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ /* (4) */
+ strcpy(card->driver, "My Chip");
+ strcpy(card->shortname, "My Own Chip 123");
+ sprintf(card->longname, "%s at 0x%lx irq %i",
+ card->shortname, chip->ioport, chip->irq);
+
+ /* (5) */
+ .... // implemented later
+
+ /* (6) */
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+
+ /* (7) */
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+ }
+
+ /* destructor -- see "Destructor" sub-section */
+ static void __devexit snd_mychip_remove(struct pci_dev *pci)
+ {
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+ }
+]]>
+ </programlisting>
+ </example>
+ </para>
+ </section>
+
+ <section id="basic-flow-constructor">
+ <title>Constructor</title>
+ <para>
+ The real constructor of PCI drivers is probe callback. The
+ probe callback and other component-constructors which are called
+ from probe callback should be defined with
+ <parameter>__devinit</parameter> prefix. You
+ cannot use <parameter>__init</parameter> prefix for them,
+ because any PCI device could be a hotplug device.
+ </para>
+
+ <para>
+ In the probe callback, the following scheme is often used.
+ </para>
+
+ <section id="basic-flow-constructor-device-index">
+ <title>1) Check and increment the device index.</title>
+ <para>
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int dev;
+ ....
+ if (dev >= SNDRV_CARDS)
+ return -ENODEV;
+ if (!enable[dev]) {
+ dev++;
+ return -ENOENT;
+ }
+]]>
+ </programlisting>
+ </informalexample>
+
+ where enable[dev] is the module option.
+ </para>
+
+ <para>
+ At each time probe callback is called, check the
+ availability of the device. If not available, simply increment
+ the device index and returns. dev will be incremented also
+ later (<link
+ linkend="basic-flow-constructor-set-pci"><citetitle>step
+ 7</citetitle></link>).
+ </para>
+ </section>
+
+ <section id="basic-flow-constructor-create-card">
+ <title>2) Create a card instance</title>
+ <para>
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_card_t *card;
+ ....
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, 0);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The detail will be explained in the section
+ <link linkend="card-management-card-instance"><citetitle>
+ Management of Cards and Components</citetitle></link>.
+ </para>
+ </section>
+
+ <section id="basic-flow-constructor-create-main">
+ <title>3) Create a main component</title>
+ <para>
+ In this part, the PCI resources are allocated.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ mychip_t *chip;
+ ....
+ if ((err = snd_mychip_create(card, pci, &chip)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+]]>
+ </programlisting>
+ </informalexample>
+
+ The detail will be explained in the section <link
+ linkend="pci-resource"><citetitle>PCI Resource
+ Managements</citetitle></link>.
+ </para>
+ </section>
+
+ <section id="basic-flow-constructor-main-component">
+ <title>4) Set the driver ID and name strings.</title>
+ <para>
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ strcpy(card->driver, "My Chip");
+ strcpy(card->shortname, "My Own Chip 123");
+ sprintf(card->longname, "%s at 0x%lx irq %i",
+ card->shortname, chip->ioport, chip->irq);
+]]>
+ </programlisting>
+ </informalexample>
+
+ The driver field holds the minimal ID string of the
+ chip. This is referred by alsa-lib's configurator, so keep it
+ simple but unique.
+ Even the same driver can have different driver IDs to
+ distinguish the functionality of each chip type.
+ </para>
+
+ <para>
+ The shortname field is a string shown as more verbose
+ name. The longname field contains the information which is
+ shown in <filename>/proc/asound/cards</filename>.
+ </para>
+ </section>
+
+ <section id="basic-flow-constructor-create-other">
+ <title>5) Create other components, such as mixer, MIDI, etc.</title>
+ <para>
+ Here you define the basic components such as
+ <link linkend="pcm-interface"><citetitle>PCM</citetitle></link>,
+ mixer (e.g. <link linkend="api-ac97"><citetitle>AC97</citetitle></link>),
+ MIDI (e.g. <link linkend="midi-interface"><citetitle>MPU-401</citetitle></link>),
+ and other interfaces.
+ Also, if you want a <link linkend="proc-interface"><citetitle>proc
+ file</citetitle></link>, define it here, too.
+ </para>
+ </section>
+
+ <section id="basic-flow-constructor-register-card">
+ <title>6) Register the card instance.</title>
+ <para>
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ if ((err = snd_card_register(card)) < 0) {
+ snd_card_free(card);
+ return err;
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ Will be explained in the section <link
+ linkend="card-management-registration"><citetitle>Management
+ of Cards and Components</citetitle></link>, too.
+ </para>
+ </section>
+
+ <section id="basic-flow-constructor-set-pci">
+ <title>7) Set the PCI driver data and return zero.</title>
+ <para>
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ pci_set_drvdata(pci, card);
+ dev++;
+ return 0;
+]]>
+ </programlisting>
+ </informalexample>
+
+ In the above, the card record is stored. This pointer is
+ referred in the remove callback and power-management
+ callbacks, too.
+ </para>
+ </section>
+ </section>
+
+ <section id="basic-flow-destructor">
+ <title>Destructor</title>
+ <para>
+ The destructor, remove callback, simply releases the card
+ instance. Then the ALSA middle layer will release all the
+ attached components automatically.
+ </para>
+
+ <para>
+ It would be typically like the following:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static void __devexit snd_mychip_remove(struct pci_dev *pci)
+ {
+ snd_card_free(pci_get_drvdata(pci));
+ pci_set_drvdata(pci, NULL);
+ }
+]]>
+ </programlisting>
+ </informalexample>
+
+ The above code assumes that the card pointer is set to the PCI
+ driver data.
+ </para>
+ </section>
+
+ <section id="basic-flow-header-files">
+ <title>Header Files</title>
+ <para>
+ For the above example, at least the following include files
+ are necessary.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ #include <sound/driver.h>
+ #include <linux/init.h>
+ #include <linux/pci.h>
+ #include <linux/slab.h>
+ #include <sound/core.h>
+ #include <sound/initval.h>
+]]>
+ </programlisting>
+ </informalexample>
+
+ where the last one is necessary only when module options are
+ defined in the source file. If the codes are split to several
+ files, the file without module options don't need them.
+ </para>
+
+ <para>
+ In addition to them, you'll need
+ <filename>&lt;linux/interrupt.h&gt;</filename> for the interrupt
+ handling, and <filename>&lt;asm/io.h&gt;</filename> for the i/o
+ access. If you use <function>mdelay()</function> or
+ <function>udelay()</function> functions, you'll need to include
+ <filename>&lt;linux/delay.h&gt;</filename>, too.
+ </para>
+
+ <para>
+ The ALSA interfaces like PCM or control API are defined in other
+ header files as <filename>&lt;sound/xxx.h&gt;</filename>.
+ They have to be included after
+ <filename>&lt;sound/core.h&gt;</filename>.
+ </para>
+
+ </section>
+ </chapter>
+
+
+<!-- ****************************************************** -->
+<!-- Management of Cards and Components -->
+<!-- ****************************************************** -->
+ <chapter id="card-management">
+ <title>Management of Cards and Components</title>
+
+ <section id="card-management-card-instance">
+ <title>Card Instance</title>
+ <para>
+ For each soundcard, a <quote>card</quote> record must be allocated.
+ </para>
+
+ <para>
+ A card record is the headquarters of the soundcard. It manages
+ the list of whole devices (components) on the soundcard, such as
+ PCM, mixers, MIDI, synthesizer, and so on. Also, the card
+ record holds the ID and the name strings of the card, manages
+ the root of proc files, and controls the power-management states
+ and hotplug disconnections. The component list on the card
+ record is used to manage the proper releases of resources at
+ destruction.
+ </para>
+
+ <para>
+ As mentioned above, to create a card instance, call
+ <function>snd_card_new()</function>.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_card_t *card;
+ card = snd_card_new(index, id, module, extra_size);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The function takes four arguments, the card-index number, the
+ id string, the module pointer (usually
+ <constant>THIS_MODULE</constant>),
+ and the size of extra-data space. The last argument is used to
+ allocate card-&gt;private_data for the
+ chip-specific data. Note that this data
+ <emphasis>is</emphasis> allocated by
+ <function>snd_card_new()</function>.
+ </para>
+ </section>
+
+ <section id="card-management-component">
+ <title>Components</title>
+ <para>
+ After the card is created, you can attach the components
+ (devices) to the card instance. On ALSA driver, a component is
+ represented as a <type>snd_device_t</type> object.
+ A component can be a PCM instance, a control interface, a raw
+ MIDI interface, etc. Each of such instances has one component
+ entry.
+ </para>
+
+ <para>
+ A component can be created via
+ <function>snd_device_new()</function> function.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_device_new(card, SNDRV_DEV_XXX, chip, &ops);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ This takes the card pointer, the device-level
+ (<constant>SNDRV_DEV_XXX</constant>), the data pointer, and the
+ callback pointers (<parameter>&amp;ops</parameter>). The
+ device-level defines the type of components and the order of
+ registration and de-registration. For most of components, the
+ device-level is already defined. For a user-defined component,
+ you can use <constant>SNDRV_DEV_LOWLEVEL</constant>.
+ </para>
+
+ <para>
+ This function itself doesn't allocate the data space. The data
+ must be allocated manually beforehand, and its pointer is passed
+ as the argument. This pointer is used as the identifier
+ (<parameter>chip</parameter> in the above example) for the
+ instance.
+ </para>
+
+ <para>
+ Each ALSA pre-defined component such as ac97 or pcm calls
+ <function>snd_device_new()</function> inside its
+ constructor. The destructor for each component is defined in the
+ callback pointers. Hence, you don't need to take care of
+ calling a destructor for such a component.
+ </para>
+
+ <para>
+ If you would like to create your own component, you need to
+ set the destructor function to dev_free callback in
+ <parameter>ops</parameter>, so that it can be released
+ automatically via <function>snd_card_free()</function>. The
+ example will be shown later as an implementation of a
+ chip-specific data.
+ </para>
+ </section>
+
+ <section id="card-management-chip-specific">
+ <title>Chip-Specific Data</title>
+ <para>
+ The chip-specific information, e.g. the i/o port address, its
+ resource pointer, or the irq number, is stored in the
+ chip-specific record.
+ Usually, the chip-specific record is typedef'ed as
+ <type>xxx_t</type> like the following:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ typedef struct snd_mychip mychip_t;
+ struct snd_mychip {
+ ....
+ };
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ In general, there are two ways to allocate the chip record.
+ </para>
+
+ <section id="card-management-chip-specific-snd-card-new">
+ <title>1. Allocating via <function>snd_card_new()</function>.</title>
+ <para>
+ As mentioned above, you can pass the extra-data-length to the 4th argument of <function>snd_card_new()</function>, i.e.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, sizeof(mychip_t));
+]]>
+ </programlisting>
+ </informalexample>
+
+ whether <type>mychip_t</type> is the type of the chip record.
+ </para>
+
+ <para>
+ In return, the allocated record can be accessed as
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ mychip_t *chip = (mychip_t *)card->private_data;
+]]>
+ </programlisting>
+ </informalexample>
+
+ With this method, you don't have to allocate twice.
+ The record is released together with the card instance.
+ </para>
+ </section>
+
+ <section id="card-management-chip-specific-allocate-extra">
+ <title>2. Allocating an extra device.</title>
+
+ <para>
+ After allocating a card instance via
+ <function>snd_card_new()</function> (with
+ <constant>NULL</constant> on the 4th arg), call
+ <function>kcalloc()</function>.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_card_t *card;
+ mychip_t *chip;
+ card = snd_card_new(index[dev], id[dev], THIS_MODULE, NULL);
+ .....
+ chip = kcalloc(1, sizeof(*chip), GFP_KERNEL);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The chip record should have the field to hold the card
+ pointer at least,
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ struct snd_mychip {
+ snd_card_t *card;
+ ....
+ };
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ Then, set the card pointer in the returned chip instance.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ chip->card = card;
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ Next, initialize the fields, and register this chip
+ record as a low-level device with a specified
+ <parameter>ops</parameter>,
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static snd_device_ops_t ops = {
+ .dev_free = snd_mychip_dev_free,
+ };
+ ....
+ snd_device_new(card, SNDRV_DEV_LOWLEVEL, chip, &ops);
+]]>
+ </programlisting>
+ </informalexample>
+
+ <function>snd_mychip_dev_free()</function> is the
+ device-destructor function, which will call the real
+ destructor.
+ </para>
+
+ <para>
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int snd_mychip_dev_free(snd_device_t *device)
+ {
+ mychip_t *chip = device->device_data;
+ return snd_mychip_free(chip);
+ }
+]]>
+ </programlisting>
+ </informalexample>
+
+ where <function>snd_mychip_free()</function> is the real destructor.
+ </para>
+ </section>
+ </section>
+
+ <section id="card-management-registration">
+ <title>Registration and Release</title>
+ <para>
+ After all components are assigned, register the card instance
+ by calling <function>snd_card_register()</function>. The access
+ to the device files are enabled at this point. That is, before
+ <function>snd_card_register()</function> is called, the
+ components are safely inaccessible from external side. If this
+ call fails, exit the probe function after releasing the card via
+ <function>snd_card_free()</function>.
+ </para>
+
+ <para>
+ For releasing the card instance, you can call simply
+ <function>snd_card_free()</function>. As already mentioned, all
+ components are released automatically by this call.
+ </para>
+
+ <para>
+ As further notes, the destructors (both
+ <function>snd_mychip_dev_free</function> and
+ <function>snd_mychip_free</function>) cannot be defined with
+ <parameter>__devexit</parameter> prefix, because they may be
+ called from the constructor, too, at the false path.
+ </para>
+
+ <para>
+ For a device which allows hotplugging, you can use
+ <function>snd_card_free_in_thread</function>. This one will
+ postpone the destruction and wait in a kernel-thread until all
+ devices are closed.
+ </para>
+
+ </section>
+
+ </chapter>
+
+
+<!-- ****************************************************** -->
+<!-- PCI Resource Managements -->
+<!-- ****************************************************** -->
+ <chapter id="pci-resource">
+ <title>PCI Resource Managements</title>
+
+ <section id="pci-resource-example">
+ <title>Full Code Example</title>
+ <para>
+ In this section, we'll finish the chip-specific constructor,
+ destructor and PCI entries. The example code is shown first,
+ below.
+
+ <example>
+ <title>PCI Resource Managements Example</title>
+ <programlisting>
+<![CDATA[
+ struct snd_mychip {
+ snd_card_t *card;
+ struct pci_dev *pci;
+
+ unsigned long port;
+ int irq;
+ };
+
+ static int snd_mychip_free(mychip_t *chip)
+ {
+ /* disable hardware here if any */
+ .... // (not implemented in this document)
+
+ /* release the irq */
+ if (chip->irq >= 0)
+ free_irq(chip->irq, (void *)chip);
+ /* release the i/o ports & memory */
+ pci_release_regions(chip->pci);
+ /* disable the PCI entry */
+ pci_disable_device(chip->pci);
+ /* release the data */
+ kfree(chip);
+ return 0;
+ }
+
+ /* chip-specific constructor */
+ static int __devinit snd_mychip_create(snd_card_t *card,
+ struct pci_dev *pci,
+ mychip_t **rchip)
+ {
+ mychip_t *chip;
+ int err;
+ static snd_device_ops_t ops = {
+ .dev_free = snd_mychip_dev_free,
+ };
+
+ *rchip = NULL;
+
+ /* initialize the PCI entry */
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ /* check PCI availability (28bit DMA) */
+ if (pci_set_dma_mask(pci, 0x0fffffff) < 0 ||
+ pci_set_consistent_dma_mask(pci, 0x0fffffff) < 0) {
+ printk(KERN_ERR "error to set 28bit mask DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+ chip = kcalloc(1, sizeof(*chip), GFP_KERNEL);
+ if (chip == NULL) {
+ pci_disable_device(pci);
+ return -ENOMEM;
+ }
+
+ /* initialize the stuff */
+ chip->card = card;
+ chip->pci = pci;
+ chip->irq = -1;
+
+ /* (1) PCI resource allocation */
+ if ((err = pci_request_regions(pci, "My Chip")) < 0) {
+ kfree(chip);
+ pci_disable_device(pci);
+ return err;
+ }
+ chip->port = pci_resource_start(pci, 0);
+ if (request_irq(pci->irq, snd_mychip_interrupt,
+ SA_INTERRUPT|SA_SHIRQ, "My Chip",
+ (void *)chip)) {
+ printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
+ snd_mychip_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+
+ /* (2) initialization of the chip hardware */
+ .... // (not implemented in this document)
+
+ if ((err = snd_device_new(card, SNDRV_DEV_LOWLEVEL,
+ chip, &ops)) < 0) {
+ snd_mychip_free(chip);
+ return err;
+ }
+
+ snd_card_set_dev(card, &pci->dev);
+
+ *rchip = chip;
+ return 0;
+ }
+
+ /* PCI IDs */
+ static struct pci_device_id snd_mychip_ids[] = {
+ { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR,
+ PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, },
+ ....
+ { 0, }
+ };
+ MODULE_DEVICE_TABLE(pci, snd_mychip_ids);
+
+ /* pci_driver definition */
+ static struct pci_driver driver = {
+ .name = "My Own Chip",
+ .id_table = snd_mychip_ids,
+ .probe = snd_mychip_probe,
+ .remove = __devexit_p(snd_mychip_remove),
+ };
+
+ /* initialization of the module */
+ static int __init alsa_card_mychip_init(void)
+ {
+ return pci_module_init(&driver);
+ }
+
+ /* clean up the module */
+ static void __exit alsa_card_mychip_exit(void)
+ {
+ pci_unregister_driver(&driver);
+ }
+
+ module_init(alsa_card_mychip_init)
+ module_exit(alsa_card_mychip_exit)
+
+ EXPORT_NO_SYMBOLS; /* for old kernels only */
+]]>
+ </programlisting>
+ </example>
+ </para>
+ </section>
+
+ <section id="pci-resource-some-haftas">
+ <title>Some Hafta's</title>
+ <para>
+ The allocation of PCI resources is done in the
+ <function>probe()</function> function, and usually an extra
+ <function>xxx_create()</function> function is written for this
+ purpose.
+ </para>
+
+ <para>
+ In the case of PCI devices, you have to call at first
+ <function>pci_enable_device()</function> function before
+ allocating resources. Also, you need to set the proper PCI DMA
+ mask to limit the accessed i/o range. In some cases, you might
+ need to call <function>pci_set_master()</function> function,
+ too.
+ </para>
+
+ <para>
+ Suppose the 28bit mask, and the code to be added would be like:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ if ((err = pci_enable_device(pci)) < 0)
+ return err;
+ if (pci_set_dma_mask(pci, 0x0fffffff) < 0 ||
+ pci_set_consistent_dma_mask(pci, 0x0fffffff) < 0) {
+ printk(KERN_ERR "error to set 28bit mask DMA\n");
+ pci_disable_device(pci);
+ return -ENXIO;
+ }
+
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+ </section>
+
+ <section id="pci-resource-resource-allocation">
+ <title>Resource Allocation</title>
+ <para>
+ The allocation of I/O ports and irqs are done via standard kernel
+ functions. Unlike ALSA ver.0.5.x., there are no helpers for
+ that. And these resources must be released in the destructor
+ function (see below). Also, on ALSA 0.9.x, you don't need to
+ allocate (pseudo-)DMA for PCI like ALSA 0.5.x.
+ </para>
+
+ <para>
+ Now assume that this PCI device has an I/O port with 8 bytes
+ and an interrupt. Then <type>mychip_t</type> will have the
+ following fields:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ struct snd_mychip {
+ snd_card_t *card;
+
+ unsigned long port;
+ int irq;
+ };
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ For an i/o port (and also a memory region), you need to have
+ the resource pointer for the standard resource management. For
+ an irq, you have to keep only the irq number (integer). But you
+ need to initialize this number as -1 before actual allocation,
+ since irq 0 is valid. The port address and its resource pointer
+ can be initialized as null by
+ <function>kcalloc()</function> automatically, so you
+ don't have to take care of resetting them.
+ </para>
+
+ <para>
+ The allocation of an i/o port is done like this:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ if ((err = pci_request_regions(pci, "My Chip")) < 0) {
+ kfree(chip);
+ pci_disable_device(pci);
+ return err;
+ }
+ chip->port = pci_resource_start(pci, 0);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ <!-- obsolete -->
+ It will reserve the i/o port region of 8 bytes of the given
+ PCI device. The returned value, chip-&gt;res_port, is allocated
+ via <function>kmalloc()</function> by
+ <function>request_region()</function>. The pointer must be
+ released via <function>kfree()</function>, but there is some
+ problem regarding this. This issue will be explained more below.
+ </para>
+
+ <para>
+ The allocation of an interrupt source is done like this:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ if (request_irq(pci->irq, snd_mychip_interrupt,
+ SA_INTERRUPT|SA_SHIRQ, "My Chip",
+ (void *)chip)) {
+ printk(KERN_ERR "cannot grab irq %d\n", pci->irq);
+ snd_mychip_free(chip);
+ return -EBUSY;
+ }
+ chip->irq = pci->irq;
+]]>
+ </programlisting>
+ </informalexample>
+
+ where <function>snd_mychip_interrupt()</function> is the
+ interrupt handler defined <link
+ linkend="pcm-interface-interrupt-handler"><citetitle>later</citetitle></link>.
+ Note that chip-&gt;irq should be defined
+ only when <function>request_irq()</function> succeeded.
+ </para>
+
+ <para>
+ On the PCI bus, the interrupts can be shared. Thus,
+ <constant>SA_SHIRQ</constant> is given as the interrupt flag of
+ <function>request_irq()</function>.
+ </para>
+
+ <para>
+ The last argument of <function>request_irq()</function> is the
+ data pointer passed to the interrupt handler. Usually, the
+ chip-specific record is used for that, but you can use what you
+ like, too.
+ </para>
+
+ <para>
+ I won't define the detail of the interrupt handler at this
+ point, but at least its appearance can be explained now. The
+ interrupt handler looks usually like the following:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id,
+ struct pt_regs *regs)
+ {
+ mychip_t *chip = dev_id;
+ ....
+ return IRQ_HANDLED;
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ Now let's write the corresponding destructor for the resources
+ above. The role of destructor is simple: disable the hardware
+ (if already activated) and release the resources. So far, we
+ have no hardware part, so the disabling is not written here.
+ </para>
+
+ <para>
+ For releasing the resources, <quote>check-and-release</quote>
+ method is a safer way. For the interrupt, do like this:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ if (chip->irq >= 0)
+ free_irq(chip->irq, (void *)chip);
+]]>
+ </programlisting>
+ </informalexample>
+
+ Since the irq number can start from 0, you should initialize
+ chip-&gt;irq with a negative value (e.g. -1), so that you can
+ check the validity of the irq number as above.
+ </para>
+
+ <para>
+ When you requested I/O ports or memory regions via
+ <function>pci_request_region()</function> or
+ <function>pci_request_regions()</function> like this example,
+ release the resource(s) using the corresponding function,
+ <function>pci_release_region()</function> or
+ <function>pci_release_regions()</function>.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ pci_release_regions(chip->pci);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ When you requested manually via <function>request_region()</function>
+ or <function>request_mem_region</function>, you can release it via
+ <function>release_resource()</function>. Suppose that you keep
+ the resource pointer returned from <function>request_region()</function>
+ in chip-&gt;res_port, the release procedure looks like below:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ if (chip->res_port) {
+ release_resource(chip->res_port);
+ kfree_nocheck(chip->res_port);
+ }
+]]>
+ </programlisting>
+ </informalexample>
+
+ As you can see, the resource pointer is also to be freed
+ via <function>kfree_nocheck()</function> after
+ <function>release_resource()</function> is called. You
+ cannot use <function>kfree()</function> here, because on ALSA,
+ <function>kfree()</function> may be a wrapper to its own
+ allocator with the memory debugging. Since the resource pointer
+ is allocated externally outside the ALSA, it must be released
+ via the native
+ <function>kfree()</function>.
+ <function>kfree_nocheck()</function> is used for that; it calls
+ the native <function>kfree()</function> without wrapper.
+ </para>
+
+ <para>
+ Don't forget to call <function>pci_disable_device()</function>
+ before all finished.
+ </para>
+
+ <para>
+ And finally, release the chip-specific record.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ kfree(chip);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ Again, remember that you cannot
+ set <parameter>__devexit</parameter> prefix for this destructor.
+ </para>
+
+ <para>
+ We didn't implement the hardware-disabling part in the above.
+ If you need to do this, please note that the destructor may be
+ called even before the initialization of the chip is completed.
+ It would be better to have a flag to skip the hardware-disabling
+ if the hardware was not initialized yet.
+ </para>
+
+ <para>
+ When the chip-data is assigned to the card using
+ <function>snd_device_new()</function> with
+ <constant>SNDRV_DEV_LOWLELVEL</constant> , its destructor is
+ called at the last. That is, it is assured that all other
+ components like PCMs and controls have been already released.
+ You don't have to call stopping PCMs, etc. explicitly, but just
+ stop the hardware in the low-level.
+ </para>
+
+ <para>
+ The management of a memory-mapped region is almost as same as
+ the management of an i/o port. You'll need three fields like
+ the following:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ struct snd_mychip {
+ ....
+ unsigned long iobase_phys;
+ void __iomem *iobase_virt;
+ };
+]]>
+ </programlisting>
+ </informalexample>
+
+ and the allocation would be like below:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ if ((err = pci_request_regions(pci, "My Chip")) < 0) {
+ kfree(chip);
+ return err;
+ }
+ chip->iobase_phys = pci_resource_start(pci, 0);
+ chip->iobase_virt = ioremap_nocache(chip->iobase_phys,
+ pci_resource_len(pci, 0));
+]]>
+ </programlisting>
+ </informalexample>
+
+ and the corresponding destructor would be:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int snd_mychip_free(mychip_t *chip)
+ {
+ ....
+ if (chip->iobase_virt)
+ iounmap(chip->iobase_virt);
+ ....
+ pci_release_regions(chip->pci);
+ ....
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ </section>
+
+ <section id="pci-resource-device-struct">
+ <title>Registration of Device Struct</title>
+ <para>
+ At some point, typically after calling <function>snd_device_new()</function>,
+ you need to register the <structname>struct device</structname> of the chip
+ you're handling for udev and co. ALSA provides a macro for compatibility with
+ older kernels. Simply call like the following:
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_card_set_dev(card, &pci->dev);
+]]>
+ </programlisting>
+ </informalexample>
+ so that it stores the PCI's device pointer to the card. This will be
+ referred by ALSA core functions later when the devices are registered.
+ </para>
+ <para>
+ In the case of non-PCI, pass the proper device struct pointer of the BUS
+ instead. (In the case of legacy ISA without PnP, you don't have to do
+ anything.)
+ </para>
+ </section>
+
+ <section id="pci-resource-entries">
+ <title>PCI Entries</title>
+ <para>
+ So far, so good. Let's finish the rest of missing PCI
+ stuffs. At first, we need a
+ <structname>pci_device_id</structname> table for this
+ chipset. It's a table of PCI vendor/device ID number, and some
+ masks.
+ </para>
+
+ <para>
+ For example,
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static struct pci_device_id snd_mychip_ids[] = {
+ { PCI_VENDOR_ID_FOO, PCI_DEVICE_ID_BAR,
+ PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, },
+ ....
+ { 0, }
+ };
+ MODULE_DEVICE_TABLE(pci, snd_mychip_ids);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The first and second fields of
+ <structname>pci_device_id</structname> struct are the vendor and
+ device IDs. If you have nothing special to filter the matching
+ devices, you can use the rest of fields like above. The last
+ field of <structname>pci_device_id</structname> struct is a
+ private data for this entry. You can specify any value here, for
+ example, to tell the type of different operations per each
+ device IDs. Such an example is found in intel8x0 driver.
+ </para>
+
+ <para>
+ The last entry of this list is the terminator. You must
+ specify this all-zero entry.
+ </para>
+
+ <para>
+ Then, prepare the <structname>pci_driver</structname> record:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static struct pci_driver driver = {
+ .name = "My Own Chip",
+ .id_table = snd_mychip_ids,
+ .probe = snd_mychip_probe,
+ .remove = __devexit_p(snd_mychip_remove),
+ };
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The <structfield>probe</structfield> and
+ <structfield>remove</structfield> functions are what we already
+ defined in
+ the previous sections. The <structfield>remove</structfield> should
+ be defined with
+ <function>__devexit_p()</function> macro, so that it's not
+ defined for built-in (and non-hot-pluggable) case. The
+ <structfield>name</structfield>
+ field is the name string of this device. Note that you must not
+ use a slash <quote>/</quote> in this string.
+ </para>
+
+ <para>
+ And at last, the module entries:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int __init alsa_card_mychip_init(void)
+ {
+ return pci_module_init(&driver);
+ }
+
+ static void __exit alsa_card_mychip_exit(void)
+ {
+ pci_unregister_driver(&driver);
+ }
+
+ module_init(alsa_card_mychip_init)
+ module_exit(alsa_card_mychip_exit)
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ Note that these module entries are tagged with
+ <parameter>__init</parameter> and
+ <parameter>__exit</parameter> prefixes, not
+ <parameter>__devinit</parameter> nor
+ <parameter>__devexit</parameter>.
+ </para>
+
+ <para>
+ Oh, one thing was forgotten. If you have no exported symbols,
+ you need to declare it on 2.2 or 2.4 kernels (on 2.6 kernels
+ it's not necessary, though).
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ EXPORT_NO_SYMBOLS;
+]]>
+ </programlisting>
+ </informalexample>
+
+ That's all!
+ </para>
+ </section>
+ </chapter>
+
+
+<!-- ****************************************************** -->
+<!-- PCM Interface -->
+<!-- ****************************************************** -->
+ <chapter id="pcm-interface">
+ <title>PCM Interface</title>
+
+ <section id="pcm-interface-general">
+ <title>General</title>
+ <para>
+ The PCM middle layer of ALSA is quite powerful and it is only
+ necessary for each driver to implement the low-level functions
+ to access its hardware.
+ </para>
+
+ <para>
+ For accessing to the PCM layer, you need to include
+ <filename>&lt;sound/pcm.h&gt;</filename> above all. In addition,
+ <filename>&lt;sound/pcm_params.h&gt;</filename> might be needed
+ if you access to some functions related with hw_param.
+ </para>
+
+ <para>
+ Each card device can have up to four pcm instances. A pcm
+ instance corresponds to a pcm device file. The limitation of
+ number of instances comes only from the available bit size of
+ the linux's device number. Once when 64bit device number is
+ used, we'll have more available pcm instances.
+ </para>
+
+ <para>
+ A pcm instance consists of pcm playback and capture streams,
+ and each pcm stream consists of one or more pcm substreams. Some
+ soundcard supports the multiple-playback function. For example,
+ emu10k1 has a PCM playback of 32 stereo substreams. In this case, at
+ each open, a free substream is (usually) automatically chosen
+ and opened. Meanwhile, when only one substream exists and it was
+ already opened, the succeeding open will result in the blocking
+ or the error with <constant>EAGAIN</constant> according to the
+ file open mode. But you don't have to know the detail in your
+ driver. The PCM middle layer will take all such jobs.
+ </para>
+ </section>
+
+ <section id="pcm-interface-example">
+ <title>Full Code Example</title>
+ <para>
+ The example code below does not include any hardware access
+ routines but shows only the skeleton, how to build up the PCM
+ interfaces.
+
+ <example>
+ <title>PCM Example Code</title>
+ <programlisting>
+<![CDATA[
+ #include <sound/pcm.h>
+ ....
+
+ /* hardware definition */
+ static snd_pcm_hardware_t snd_mychip_playback_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 32768,
+ .period_bytes_min = 4096,
+ .period_bytes_max = 32768,
+ .periods_min = 1,
+ .periods_max = 1024,
+ };
+
+ /* hardware definition */
+ static snd_pcm_hardware_t snd_mychip_capture_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 32768,
+ .period_bytes_min = 4096,
+ .period_bytes_max = 32768,
+ .periods_min = 1,
+ .periods_max = 1024,
+ };
+
+ /* open callback */
+ static int snd_mychip_playback_open(snd_pcm_substream_t *substream)
+ {
+ mychip_t *chip = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+
+ runtime->hw = snd_mychip_playback_hw;
+ // more hardware-initialization will be done here
+ return 0;
+ }
+
+ /* close callback */
+ static int snd_mychip_playback_close(snd_pcm_substream_t *substream)
+ {
+ mychip_t *chip = snd_pcm_substream_chip(substream);
+ // the hardware-specific codes will be here
+ return 0;
+
+ }
+
+ /* open callback */
+ static int snd_mychip_capture_open(snd_pcm_substream_t *substream)
+ {
+ mychip_t *chip = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+
+ runtime->hw = snd_mychip_capture_hw;
+ // more hardware-initialization will be done here
+ return 0;
+ }
+
+ /* close callback */
+ static int snd_mychip_capture_close(snd_pcm_substream_t *substream)
+ {
+ mychip_t *chip = snd_pcm_substream_chip(substream);
+ // the hardware-specific codes will be here
+ return 0;
+
+ }
+
+ /* hw_params callback */
+ static int snd_mychip_pcm_hw_params(snd_pcm_substream_t *substream,
+ snd_pcm_hw_params_t * hw_params)
+ {
+ return snd_pcm_lib_malloc_pages(substream,
+ params_buffer_bytes(hw_params));
+ }
+
+ /* hw_free callback */
+ static int snd_mychip_pcm_hw_free(snd_pcm_substream_t *substream)
+ {
+ return snd_pcm_lib_free_pages(substream);
+ }
+
+ /* prepare callback */
+ static int snd_mychip_pcm_prepare(snd_pcm_substream_t *substream)
+ {
+ mychip_t *chip = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+
+ /* set up the hardware with the current configuration
+ * for example...
+ */
+ mychip_set_sample_format(chip, runtime->format);
+ mychip_set_sample_rate(chip, runtime->rate);
+ mychip_set_channels(chip, runtime->channels);
+ mychip_set_dma_setup(chip, runtime->dma_area,
+ chip->buffer_size,
+ chip->period_size);
+ return 0;
+ }
+
+ /* trigger callback */
+ static int snd_mychip_pcm_trigger(snd_pcm_substream_t *substream,
+ int cmd)
+ {
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ // do something to start the PCM engine
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ // do something to stop the PCM engine
+ break;
+ default:
+ return -EINVAL;
+ }
+ }
+
+ /* pointer callback */
+ static snd_pcm_uframes_t
+ snd_mychip_pcm_pointer(snd_pcm_substream_t *substream)
+ {
+ mychip_t *chip = snd_pcm_substream_chip(substream);
+ unsigned int current_ptr;
+
+ /* get the current hardware pointer */
+ current_ptr = mychip_get_hw_pointer(chip);
+ return current_ptr;
+ }
+
+ /* operators */
+ static snd_pcm_ops_t snd_mychip_playback_ops = {
+ .open = snd_mychip_playback_open,
+ .close = snd_mychip_playback_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_mychip_pcm_hw_params,
+ .hw_free = snd_mychip_pcm_hw_free,
+ .prepare = snd_mychip_pcm_prepare,
+ .trigger = snd_mychip_pcm_trigger,
+ .pointer = snd_mychip_pcm_pointer,
+ };
+
+ /* operators */
+ static snd_pcm_ops_t snd_mychip_capture_ops = {
+ .open = snd_mychip_capture_open,
+ .close = snd_mychip_capture_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_mychip_pcm_hw_params,
+ .hw_free = snd_mychip_pcm_hw_free,
+ .prepare = snd_mychip_pcm_prepare,
+ .trigger = snd_mychip_pcm_trigger,
+ .pointer = snd_mychip_pcm_pointer,
+ };
+
+ /*
+ * definitions of capture are omitted here...
+ */
+
+ /* create a pcm device */
+ static int __devinit snd_mychip_new_pcm(mychip_t *chip)
+ {
+ snd_pcm_t *pcm;
+ int err;
+
+ if ((err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1,
+ &pcm)) < 0)
+ return err;
+ pcm->private_data = chip;
+ strcpy(pcm->name, "My Chip");
+ chip->pcm = pcm;
+ /* set operators */
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_mychip_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_mychip_capture_ops);
+ /* pre-allocation of buffers */
+ /* NOTE: this may fail */
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ 64*1024, 64*1024);
+ return 0;
+ }
+]]>
+ </programlisting>
+ </example>
+ </para>
+ </section>
+
+ <section id="pcm-interface-constructor">
+ <title>Constructor</title>
+ <para>
+ A pcm instance is allocated by <function>snd_pcm_new()</function>
+ function. It would be better to create a constructor for pcm,
+ namely,
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int __devinit snd_mychip_new_pcm(mychip_t *chip)
+ {
+ snd_pcm_t *pcm;
+ int err;
+
+ if ((err = snd_pcm_new(chip->card, "My Chip", 0, 1, 1,
+ &pcm)) < 0)
+ return err;
+ pcm->private_data = chip;
+ strcpy(pcm->name, "My Chip");
+ chip->pcm = pcm;
+ ....
+ return 0;
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The <function>snd_pcm_new()</function> function takes the four
+ arguments. The first argument is the card pointer to which this
+ pcm is assigned, and the second is the ID string.
+ </para>
+
+ <para>
+ The third argument (<parameter>index</parameter>, 0 in the
+ above) is the index of this new pcm. It begins from zero. When
+ you will create more than one pcm instances, specify the
+ different numbers in this argument. For example,
+ <parameter>index</parameter> = 1 for the second PCM device.
+ </para>
+
+ <para>
+ The fourth and fifth arguments are the number of substreams
+ for playback and capture, respectively. Here both 1 are given in
+ the above example. When no playback or no capture is available,
+ pass 0 to the corresponding argument.
+ </para>
+
+ <para>
+ If a chip supports multiple playbacks or captures, you can
+ specify more numbers, but they must be handled properly in
+ open/close, etc. callbacks. When you need to know which
+ substream you are referring to, then it can be obtained from
+ <type>snd_pcm_substream_t</type> data passed to each callback
+ as follows:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_pcm_substream_t *substream;
+ int index = substream->number;
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ After the pcm is created, you need to set operators for each
+ pcm stream.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_PLAYBACK,
+ &snd_mychip_playback_ops);
+ snd_pcm_set_ops(pcm, SNDRV_PCM_STREAM_CAPTURE,
+ &snd_mychip_capture_ops);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The operators are defined typically like this:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static snd_pcm_ops_t snd_mychip_playback_ops = {
+ .open = snd_mychip_pcm_open,
+ .close = snd_mychip_pcm_close,
+ .ioctl = snd_pcm_lib_ioctl,
+ .hw_params = snd_mychip_pcm_hw_params,
+ .hw_free = snd_mychip_pcm_hw_free,
+ .prepare = snd_mychip_pcm_prepare,
+ .trigger = snd_mychip_pcm_trigger,
+ .pointer = snd_mychip_pcm_pointer,
+ };
+]]>
+ </programlisting>
+ </informalexample>
+
+ Each of callbacks is explained in the subsection
+ <link linkend="pcm-interface-operators"><citetitle>
+ Operators</citetitle></link>.
+ </para>
+
+ <para>
+ After setting the operators, most likely you'd like to
+ pre-allocate the buffer. For the pre-allocation, simply call
+ the following:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(chip->pci),
+ 64*1024, 64*1024);
+]]>
+ </programlisting>
+ </informalexample>
+
+ It will allocate up to 64kB buffer as default. The details of
+ buffer management will be described in the later section <link
+ linkend="buffer-and-memory"><citetitle>Buffer and Memory
+ Management</citetitle></link>.
+ </para>
+
+ <para>
+ Additionally, you can set some extra information for this pcm
+ in pcm-&gt;info_flags.
+ The available values are defined as
+ <constant>SNDRV_PCM_INFO_XXX</constant> in
+ <filename>&lt;sound/asound.h&gt;</filename>, which is used for
+ the hardware definition (described later). When your soundchip
+ supports only half-duplex, specify like this:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ pcm->info_flags = SNDRV_PCM_INFO_HALF_DUPLEX;
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+ </section>
+
+ <section id="pcm-interface-destructor">
+ <title>... And the Destructor?</title>
+ <para>
+ The destructor for a pcm instance is not always
+ necessary. Since the pcm device will be released by the middle
+ layer code automatically, you don't have to call destructor
+ explicitly.
+ </para>
+
+ <para>
+ The destructor would be necessary when you created some
+ special records internally and need to release them. In such a
+ case, set the destructor function to
+ pcm-&gt;private_free:
+
+ <example>
+ <title>PCM Instance with a Destructor</title>
+ <programlisting>
+<![CDATA[
+ static void mychip_pcm_free(snd_pcm_t *pcm)
+ {
+ mychip_t *chip = snd_pcm_chip(pcm);
+ /* free your own data */
+ kfree(chip->my_private_pcm_data);
+ // do what you like else
+ ....
+ }
+
+ static int __devinit snd_mychip_new_pcm(mychip_t *chip)
+ {
+ snd_pcm_t *pcm;
+ ....
+ /* allocate your own data */
+ chip->my_private_pcm_data = kmalloc(...);
+ /* set the destructor */
+ pcm->private_data = chip;
+ pcm->private_free = mychip_pcm_free;
+ ....
+ }
+]]>
+ </programlisting>
+ </example>
+ </para>
+ </section>
+
+ <section id="pcm-interface-runtime">
+ <title>Runtime Pointer - The Chest of PCM Information</title>
+ <para>
+ When the PCM substream is opened, a PCM runtime instance is
+ allocated and assigned to the substream. This pointer is
+ accessible via <constant>substream-&gt;runtime</constant>.
+ This runtime pointer holds the various information; it holds
+ the copy of hw_params and sw_params configurations, the buffer
+ pointers, mmap records, spinlocks, etc. Almost everyhing you
+ need for controlling the PCM can be found there.
+ </para>
+
+ <para>
+ The definition of runtime instance is found in
+ <filename>&lt;sound/pcm.h&gt;</filename>. Here is the
+ copy from the file.
+ <informalexample>
+ <programlisting>
+<![CDATA[
+struct _snd_pcm_runtime {
+ /* -- Status -- */
+ snd_pcm_substream_t *trigger_master;
+ snd_timestamp_t trigger_tstamp; /* trigger timestamp */
+ int overrange;
+ snd_pcm_uframes_t avail_max;
+ snd_pcm_uframes_t hw_ptr_base; /* Position at buffer restart */
+ snd_pcm_uframes_t hw_ptr_interrupt; /* Position at interrupt time*/
+
+ /* -- HW params -- */
+ snd_pcm_access_t access; /* access mode */
+ snd_pcm_format_t format; /* SNDRV_PCM_FORMAT_* */
+ snd_pcm_subformat_t subformat; /* subformat */
+ unsigned int rate; /* rate in Hz */
+ unsigned int channels; /* channels */
+ snd_pcm_uframes_t period_size; /* period size */
+ unsigned int periods; /* periods */
+ snd_pcm_uframes_t buffer_size; /* buffer size */
+ unsigned int tick_time; /* tick time */
+ snd_pcm_uframes_t min_align; /* Min alignment for the format */
+ size_t byte_align;
+ unsigned int frame_bits;
+ unsigned int sample_bits;
+ unsigned int info;
+ unsigned int rate_num;
+ unsigned int rate_den;
+
+ /* -- SW params -- */
+ int tstamp_timespec; /* use timeval (0) or timespec (1) */
+ snd_pcm_tstamp_t tstamp_mode; /* mmap timestamp is updated */
+ unsigned int period_step;
+ unsigned int sleep_min; /* min ticks to sleep */
+ snd_pcm_uframes_t xfer_align; /* xfer size need to be a multiple */
+ snd_pcm_uframes_t start_threshold;
+ snd_pcm_uframes_t stop_threshold;
+ snd_pcm_uframes_t silence_threshold; /* Silence filling happens when
+ noise is nearest than this */
+ snd_pcm_uframes_t silence_size; /* Silence filling size */
+ snd_pcm_uframes_t boundary; /* pointers wrap point */
+
+ snd_pcm_uframes_t silenced_start;
+ snd_pcm_uframes_t silenced_size;
+
+ snd_pcm_sync_id_t sync; /* hardware synchronization ID */
+
+ /* -- mmap -- */
+ volatile snd_pcm_mmap_status_t *status;
+ volatile snd_pcm_mmap_control_t *control;
+ atomic_t mmap_count;
+
+ /* -- locking / scheduling -- */
+ spinlock_t lock;
+ wait_queue_head_t sleep;
+ struct timer_list tick_timer;
+ struct fasync_struct *fasync;
+
+ /* -- private section -- */
+ void *private_data;
+ void (*private_free)(snd_pcm_runtime_t *runtime);
+
+ /* -- hardware description -- */
+ snd_pcm_hardware_t hw;
+ snd_pcm_hw_constraints_t hw_constraints;
+
+ /* -- interrupt callbacks -- */
+ void (*transfer_ack_begin)(snd_pcm_substream_t *substream);
+ void (*transfer_ack_end)(snd_pcm_substream_t *substream);
+
+ /* -- timer -- */
+ unsigned int timer_resolution; /* timer resolution */
+
+ /* -- DMA -- */
+ unsigned char *dma_area; /* DMA area */
+ dma_addr_t dma_addr; /* physical bus address (not accessible from main CPU) */
+ size_t dma_bytes; /* size of DMA area */
+
+ struct snd_dma_buffer *dma_buffer_p; /* allocated buffer */
+
+#if defined(CONFIG_SND_PCM_OSS) || defined(CONFIG_SND_PCM_OSS_MODULE)
+ /* -- OSS things -- */
+ snd_pcm_oss_runtime_t oss;
+#endif
+};
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ For the operators (callbacks) of each sound driver, most of
+ these records are supposed to be read-only. Only the PCM
+ middle-layer changes / updates these info. The exceptions are
+ the hardware description (hw), interrupt callbacks
+ (transfer_ack_xxx), DMA buffer information, and the private
+ data. Besides, if you use the standard buffer allocation
+ method via <function>snd_pcm_lib_malloc_pages()</function>,
+ you don't need to set the DMA buffer information by yourself.
+ </para>
+
+ <para>
+ In the sections below, important records are explained.
+ </para>
+
+ <section id="pcm-interface-runtime-hw">
+ <title>Hardware Description</title>
+ <para>
+ The hardware descriptor (<type>snd_pcm_hardware_t</type>)
+ contains the definitions of the fundamental hardware
+ configuration. Above all, you'll need to define this in
+ <link linkend="pcm-interface-operators-open-callback"><citetitle>
+ the open callback</citetitle></link>.
+ Note that the runtime instance holds the copy of the
+ descriptor, not the pointer to the existing descriptor. That
+ is, in the open callback, you can modify the copied descriptor
+ (<constant>runtime-&gt;hw</constant>) as you need. For example, if the maximum
+ number of channels is 1 only on some chip models, you can
+ still use the same hardware descriptor and change the
+ channels_max later:
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_pcm_runtime_t *runtime = substream->runtime;
+ ...
+ runtime->hw = snd_mychip_playback_hw; /* common definition */
+ if (chip->model == VERY_OLD_ONE)
+ runtime->hw.channels_max = 1;
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ Typically, you'll have a hardware descriptor like below:
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static snd_pcm_hardware_t snd_mychip_playback_hw = {
+ .info = (SNDRV_PCM_INFO_MMAP |
+ SNDRV_PCM_INFO_INTERLEAVED |
+ SNDRV_PCM_INFO_BLOCK_TRANSFER |
+ SNDRV_PCM_INFO_MMAP_VALID),
+ .formats = SNDRV_PCM_FMTBIT_S16_LE,
+ .rates = SNDRV_PCM_RATE_8000_48000,
+ .rate_min = 8000,
+ .rate_max = 48000,
+ .channels_min = 2,
+ .channels_max = 2,
+ .buffer_bytes_max = 32768,
+ .period_bytes_min = 4096,
+ .period_bytes_max = 32768,
+ .periods_min = 1,
+ .periods_max = 1024,
+ };
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ <itemizedlist>
+ <listitem><para>
+ The <structfield>info</structfield> field contains the type and
+ capabilities of this pcm. The bit flags are defined in
+ <filename>&lt;sound/asound.h&gt;</filename> as
+ <constant>SNDRV_PCM_INFO_XXX</constant>. Here, at least, you
+ have to specify whether the mmap is supported and which
+ interleaved format is supported.
+ When the mmap is supported, add
+ <constant>SNDRV_PCM_INFO_MMAP</constant> flag here. When the
+ hardware supports the interleaved or the non-interleaved
+ format, <constant>SNDRV_PCM_INFO_INTERLEAVED</constant> or
+ <constant>SNDRV_PCM_INFO_NONINTERLEAVED</constant> flag must
+ be set, respectively. If both are supported, you can set both,
+ too.
+ </para>
+
+ <para>
+ In the above example, <constant>MMAP_VALID</constant> and
+ <constant>BLOCK_TRANSFER</constant> are specified for OSS mmap
+ mode. Usually both are set. Of course,
+ <constant>MMAP_VALID</constant> is set only if the mmap is
+ really supported.
+ </para>
+
+ <para>
+ The other possible flags are
+ <constant>SNDRV_PCM_INFO_PAUSE</constant> and
+ <constant>SNDRV_PCM_INFO_RESUME</constant>. The
+ <constant>PAUSE</constant> bit means that the pcm supports the
+ <quote>pause</quote> operation, while the
+ <constant>RESUME</constant> bit means that the pcm supports
+ the <quote>suspend/resume</quote> operation. If these flags
+ are set, the <structfield>trigger</structfield> callback below
+ must handle the corresponding commands.
+ </para>
+
+ <para>
+ When the PCM substreams can be synchronized (typically,
+ synchorinized start/stop of a playback and a capture streams),
+ you can give <constant>SNDRV_PCM_INFO_SYNC_START</constant>,
+ too. In this case, you'll need to check the linked-list of
+ PCM substreams in the trigger callback. This will be
+ described in the later section.
+ </para>
+ </listitem>
+
+ <listitem>
+ <para>
+ <structfield>formats</structfield> field contains the bit-flags
+ of supported formats (<constant>SNDRV_PCM_FMTBIT_XXX</constant>).
+ If the hardware supports more than one format, give all or'ed
+ bits. In the example above, the signed 16bit little-endian
+ format is specified.
+ </para>
+ </listitem>
+
+ <listitem>
+ <para>
+ <structfield>rates</structfield> field contains the bit-flags of
+ supported rates (<constant>SNDRV_PCM_RATE_XXX</constant>).
+ When the chip supports continuous rates, pass
+ <constant>CONTINUOUS</constant> bit additionally.
+ The pre-defined rate bits are provided only for typical
+ rates. If your chip supports unconventional rates, you need to add
+ <constant>KNOT</constant> bit and set up the hardware
+ constraint manually (explained later).
+ </para>
+ </listitem>
+
+ <listitem>
+ <para>
+ <structfield>rate_min</structfield> and
+ <structfield>rate_max</structfield> define the minimal and
+ maximal sample rate. This should correspond somehow to
+ <structfield>rates</structfield> bits.
+ </para>
+ </listitem>
+
+ <listitem>
+ <para>
+ <structfield>channel_min</structfield> and
+ <structfield>channel_max</structfield>
+ define, as you might already expected, the minimal and maximal
+ number of channels.
+ </para>
+ </listitem>
+
+ <listitem>
+ <para>
+ <structfield>buffer_bytes_max</structfield> defines the
+ maximal buffer size in bytes. There is no
+ <structfield>buffer_bytes_min</structfield> field, since
+ it can be calculated from the minimal period size and the
+ minimal number of periods.
+ Meanwhile, <structfield>period_bytes_min</structfield> and
+ define the minimal and maximal size of the period in bytes.
+ <structfield>periods_max</structfield> and
+ <structfield>periods_min</structfield> define the maximal and
+ minimal number of periods in the buffer.
+ </para>
+
+ <para>
+ The <quote>period</quote> is a term, that corresponds to
+ fragment in the OSS world. The period defines the size at
+ which the PCM interrupt is generated. This size strongly
+ depends on the hardware.
+ Generally, the smaller period size will give you more
+ interrupts, that is, more controls.
+ In the case of capture, this size defines the input latency.
+ On the other hand, the whole buffer size defines the
+ output latency for the playback direction.
+ </para>
+ </listitem>
+
+ <listitem>
+ <para>
+ There is also a field <structfield>fifo_size</structfield>.
+ This specifies the size of the hardware FIFO, but it's not
+ used currently in the driver nor in the alsa-lib. So, you
+ can ignore this field.
+ </para>
+ </listitem>
+ </itemizedlist>
+ </para>
+ </section>
+
+ <section id="pcm-interface-runtime-config">
+ <title>PCM Configurations</title>
+ <para>
+ Ok, let's go back again to the PCM runtime records.
+ The most frequently referred records in the runtime instance are
+ the PCM configurations.
+ The PCM configurations are stored on runtime instance
+ after the application sends <type>hw_params</type> data via
+ alsa-lib. There are many fields copied from hw_params and
+ sw_params structs. For example,
+ <structfield>format</structfield> holds the format type
+ chosen by the application. This field contains the enum value
+ <constant>SNDRV_PCM_FORMAT_XXX</constant>.
+ </para>
+
+ <para>
+ One thing to be noted is that the configured buffer and period
+ sizes are stored in <quote>frames</quote> in the runtime
+ In the ALSA world, 1 frame = channels * samples-size.
+ For conversion between frames and bytes, you can use the
+ helper functions, <function>frames_to_bytes()</function> and
+ <function>bytes_to_frames()</function>.
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ period_bytes = frames_to_bytes(runtime, runtime->period_size);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ Also, many software parameters (sw_params) are
+ stored in frames, too. Please check the type of the field.
+ <type>snd_pcm_uframes_t</type> is for the frames as unsigned
+ integer while <type>snd_pcm_sframes_t</type> is for the frames
+ as signed integer.
+ </para>
+ </section>
+
+ <section id="pcm-interface-runtime-dma">
+ <title>DMA Buffer Information</title>
+ <para>
+ The DMA buffer is defined by the following four fields,
+ <structfield>dma_area</structfield>,
+ <structfield>dma_addr</structfield>,
+ <structfield>dma_bytes</structfield> and
+ <structfield>dma_private</structfield>.
+ The <structfield>dma_area</structfield> holds the buffer
+ pointer (the logical address). You can call
+ <function>memcpy</function> from/to
+ this pointer. Meanwhile, <structfield>dma_addr</structfield>
+ holds the physical address of the buffer. This field is
+ specified only when the buffer is a linear buffer.
+ <structfield>dma_bytes</structfield> holds the size of buffer
+ in bytes. <structfield>dma_private</structfield> is used for
+ the ALSA DMA allocator.
+ </para>
+
+ <para>
+ If you use a standard ALSA function,
+ <function>snd_pcm_lib_malloc_pages()</function>, for
+ allocating the buffer, these fields are set by the ALSA middle
+ layer, and you should <emphasis>not</emphasis> change them by
+ yourself. You can read them but not write them.
+ On the other hand, if you want to allocate the buffer by
+ yourself, you'll need to manage it in hw_params callback.
+ At least, <structfield>dma_bytes</structfield> is mandatory.
+ <structfield>dma_area</structfield> is necessary when the
+ buffer is mmapped. If your driver doesn't support mmap, this
+ field is not necessary. <structfield>dma_addr</structfield>
+ is also not mandatory. You can use
+ <structfield>dma_private</structfield> as you like, too.
+ </para>
+ </section>
+
+ <section id="pcm-interface-runtime-status">
+ <title>Running Status</title>
+ <para>
+ The running status can be referred via <constant>runtime-&gt;status</constant>.
+ This is the pointer to <type>snd_pcm_mmap_status_t</type>
+ record. For example, you can get the current DMA hardware
+ pointer via <constant>runtime-&gt;status-&gt;hw_ptr</constant>.
+ </para>
+
+ <para>
+ The DMA application pointer can be referred via
+ <constant>runtime-&gt;control</constant>, which points
+ <type>snd_pcm_mmap_control_t</type> record.
+ However, accessing directly to this value is not recommended.
+ </para>
+ </section>
+
+ <section id="pcm-interface-runtime-private">
+ <title>Private Data</title>
+ <para>
+ You can allocate a record for the substream and store it in
+ <constant>runtime-&gt;private_data</constant>. Usually, this
+ done in
+ <link linkend="pcm-interface-operators-open-callback"><citetitle>
+ the open callback</citetitle></link>.
+ Don't mix this with <constant>pcm-&gt;private_data</constant>.
+ The <constant>pcm-&gt;private_data</constant> usually points the
+ chip instance assigned statically at the creation of PCM, while the
+ <constant>runtime-&gt;private_data</constant> points a dynamic
+ data created at the PCM open callback.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int snd_xxx_open(snd_pcm_substream_t *substream)
+ {
+ my_pcm_data_t *data;
+ ....
+ data = kmalloc(sizeof(*data), GFP_KERNEL);
+ substream->runtime->private_data = data;
+ ....
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The allocated object must be released in
+ <link linkend="pcm-interface-operators-open-callback"><citetitle>
+ the close callback</citetitle></link>.
+ </para>
+ </section>
+
+ <section id="pcm-interface-runtime-intr">
+ <title>Interrupt Callbacks</title>
+ <para>
+ The field <structfield>transfer_ack_begin</structfield> and
+ <structfield>transfer_ack_end</structfield> are called at
+ the beginning and the end of
+ <function>snd_pcm_period_elapsed()</function>, respectively.
+ </para>
+ </section>
+
+ </section>
+
+ <section id="pcm-interface-operators">
+ <title>Operators</title>
+ <para>
+ OK, now let me explain the detail of each pcm callback
+ (<parameter>ops</parameter>). In general, every callback must
+ return 0 if successful, or a negative number with the error
+ number such as <constant>-EINVAL</constant> at any
+ error.
+ </para>
+
+ <para>
+ The callback function takes at least the argument with
+ <type>snd_pcm_substream_t</type> pointer. For retrieving the
+ chip record from the given substream instance, you can use the
+ following macro.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ int xxx() {
+ mychip_t *chip = snd_pcm_substream_chip(substream);
+ ....
+ }
+]]>
+ </programlisting>
+ </informalexample>
+
+ The macro reads <constant>substream-&gt;private_data</constant>,
+ which is a copy of <constant>pcm-&gt;private_data</constant>.
+ You can override the former if you need to assign different data
+ records per PCM substream. For example, cmi8330 driver assigns
+ different private_data for playback and capture directions,
+ because it uses two different codecs (SB- and AD-compatible) for
+ different directions.
+ </para>
+
+ <section id="pcm-interface-operators-open-callback">
+ <title>open callback</title>
+ <para>
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int snd_xxx_open(snd_pcm_substream_t *substream);
+]]>
+ </programlisting>
+ </informalexample>
+
+ This is called when a pcm substream is opened.
+ </para>
+
+ <para>
+ At least, here you have to initialize the runtime-&gt;hw
+ record. Typically, this is done by like this:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int snd_xxx_open(snd_pcm_substream_t *substream)
+ {
+ mychip_t *chip = snd_pcm_substream_chip(substream);
+ snd_pcm_runtime_t *runtime = substream->runtime;
+
+ runtime->hw = snd_mychip_playback_hw;
+ return 0;
+ }
+]]>
+ </programlisting>
+ </informalexample>
+
+ where <parameter>snd_mychip_playback_hw</parameter> is the
+ pre-defined hardware description.
+ </para>
+
+ <para>
+ You can allocate a private data in this callback, as described
+ in <link linkend="pcm-interface-runtime-private"><citetitle>
+ Private Data</citetitle></link> section.
+ </para>
+
+ <para>
+ If the hardware configuration needs more constraints, set the
+ hardware constraints here, too.
+ See <link linkend="pcm-interface-constraints"><citetitle>
+ Constraints</citetitle></link> for more details.
+ </para>
+ </section>
+
+ <section id="pcm-interface-operators-close-callback">
+ <title>close callback</title>
+ <para>
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int snd_xxx_close(snd_pcm_substream_t *substream);
+]]>
+ </programlisting>
+ </informalexample>
+
+ Obviously, this is called when a pcm substream is closed.
+ </para>
+
+ <para>
+ Any private instance for a pcm substream allocated in the
+ open callback will be released here.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int snd_xxx_close(snd_pcm_substream_t *substream)
+ {
+ ....
+ kfree(substream->runtime->private_data);
+ ....
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+ </section>
+
+ <section id="pcm-interface-operators-ioctl-callback">
+ <title>ioctl callback</title>
+ <para>
+ This is used for any special action to pcm ioctls. But
+ usually you can pass a generic ioctl callback,
+ <function>snd_pcm_lib_ioctl</function>.
+ </para>
+ </section>
+
+ <section id="pcm-interface-operators-hw-params-callback">
+ <title>hw_params callback</title>
+ <para>
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int snd_xxx_hw_params(snd_pcm_substream_t * substream,
+ snd_pcm_hw_params_t * hw_params);
+]]>
+ </programlisting>
+ </informalexample>
+
+ This and <structfield>hw_free</structfield> callbacks exist
+ only on ALSA 0.9.x.
+ </para>
+
+ <para>
+ This is called when the hardware parameter
+ (<structfield>hw_params</structfield>) is set
+ up by the application,
+ that is, once when the buffer size, the period size, the
+ format, etc. are defined for the pcm substream.
+ </para>
+
+ <para>
+ Many hardware set-up should be done in this callback,
+ including the allocation of buffers.
+ </para>
+
+ <para>
+ Parameters to be initialized are retrieved by
+ <function>params_xxx()</function> macros. For allocating a
+ buffer, you can call a helper function,
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_pcm_lib_malloc_pages(substream, params_buffer_bytes(hw_params));
+]]>
+ </programlisting>
+ </informalexample>
+
+ <function>snd_pcm_lib_malloc_pages()</function> is available
+ only when the DMA buffers have been pre-allocated.
+ See the section <link
+ linkend="buffer-and-memory-buffer-types"><citetitle>
+ Buffer Types</citetitle></link> for more details.
+ </para>
+
+ <para>
+ Note that this and <structfield>prepare</structfield> callbacks
+ may be called multiple times per initialization.
+ For example, the OSS emulation may
+ call these callbacks at each change via its ioctl.
+ </para>
+
+ <para>
+ Thus, you need to take care not to allocate the same buffers
+ many times, which will lead to memory leak! Calling the
+ helper function above many times is OK. It will release the
+ previous buffer automatically when it was already allocated.
+ </para>
+
+ <para>
+ Another note is that this callback is non-atomic
+ (schedulable). This is important, because the
+ <structfield>trigger</structfield> callback
+ is atomic (non-schedulable). That is, mutex or any
+ schedule-related functions are not available in
+ <structfield>trigger</structfield> callback.
+ Please see the subsection
+ <link linkend="pcm-interface-atomicity"><citetitle>
+ Atomicity</citetitle></link> for details.
+ </para>
+ </section>
+
+ <section id="pcm-interface-operators-hw-free-callback">
+ <title>hw_free callback</title>
+ <para>
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int snd_xxx_hw_free(snd_pcm_substream_t * substream);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ This is called to release the resources allocated via
+ <structfield>hw_params</structfield>. For example, releasing the
+ buffer via
+ <function>snd_pcm_lib_malloc_pages()</function> is done by
+ calling the following:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_pcm_lib_free_pages(substream);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ This function is always called before the close callback is called.
+ Also, the callback may be called multiple times, too.
+ Keep track whether the resource was already released.
+ </para>
+ </section>
+
+ <section id="pcm-interface-operators-prepare-callback">
+ <title>prepare callback</title>
+ <para>
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int snd_xxx_prepare(snd_pcm_substream_t * substream);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ This callback is called when the pcm is
+ <quote>prepared</quote>. You can set the format type, sample
+ rate, etc. here. The difference from
+ <structfield>hw_params</structfield> is that the
+ <structfield>prepare</structfield> callback will be called at each
+ time
+ <function>snd_pcm_prepare()</function> is called, i.e. when
+ recovered after underruns, etc.
+ </para>
+
+ <para>
+ Note that this callback became non-atomic since the recent version.
+ You can use schedule-related fucntions safely in this callback now.
+ </para>
+
+ <para>
+ In this and the following callbacks, you can refer to the
+ values via the runtime record,
+ substream-&gt;runtime.
+ For example, to get the current
+ rate, format or channels, access to
+ runtime-&gt;rate,
+ runtime-&gt;format or
+ runtime-&gt;channels, respectively.
+ The physical address of the allocated buffer is set to
+ runtime-&gt;dma_area. The buffer and period sizes are
+ in runtime-&gt;buffer_size and runtime-&gt;period_size,
+ respectively.
+ </para>
+
+ <para>
+ Be careful that this callback will be called many times at
+ each set up, too.
+ </para>
+ </section>
+
+ <section id="pcm-interface-operators-trigger-callback">
+ <title>trigger callback</title>
+ <para>
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int snd_xxx_trigger(snd_pcm_substream_t * substream, int cmd);
+]]>
+ </programlisting>
+ </informalexample>
+
+ This is called when the pcm is started, stopped or paused.
+ </para>
+
+ <para>
+ Which action is specified in the second argument,
+ <constant>SNDRV_PCM_TRIGGER_XXX</constant> in
+ <filename>&lt;sound/pcm.h&gt;</filename>. At least,
+ <constant>START</constant> and <constant>STOP</constant>
+ commands must be defined in this callback.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ switch (cmd) {
+ case SNDRV_PCM_TRIGGER_START:
+ // do something to start the PCM engine
+ break;
+ case SNDRV_PCM_TRIGGER_STOP:
+ // do something to stop the PCM engine
+ break;
+ default:
+ return -EINVAL;
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ When the pcm supports the pause operation (given in info
+ field of the hardware table), <constant>PAUSE_PUSE</constant>
+ and <constant>PAUSE_RELEASE</constant> commands must be
+ handled here, too. The former is the command to pause the pcm,
+ and the latter to restart the pcm again.
+ </para>
+
+ <para>
+ When the pcm supports the suspend/resume operation
+ (i.e. <constant>SNDRV_PCM_INFO_RESUME</constant> flag is set),
+ <constant>SUSPEND</constant> and <constant>RESUME</constant>
+ commands must be handled, too.
+ These commands are issued when the power-management status is
+ changed. Obviously, the <constant>SUSPEND</constant> and
+ <constant>RESUME</constant>
+ do suspend and resume of the pcm substream, and usually, they
+ are identical with <constant>STOP</constant> and
+ <constant>START</constant> commands, respectively.
+ </para>
+
+ <para>
+ As mentioned, this callback is atomic. You cannot call
+ the function going to sleep.
+ The trigger callback should be as minimal as possible,
+ just really triggering the DMA. The other stuff should be
+ initialized hw_params and prepare callbacks properly
+ beforehand.
+ </para>
+ </section>
+
+ <section id="pcm-interface-operators-pointer-callback">
+ <title>pointer callback</title>
+ <para>
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static snd_pcm_uframes_t snd_xxx_pointer(snd_pcm_substream_t * substream)
+]]>
+ </programlisting>
+ </informalexample>
+
+ This callback is called when the PCM middle layer inquires
+ the current hardware position on the buffer. The position must
+ be returned in frames (which was in bytes on ALSA 0.5.x),
+ ranged from 0 to buffer_size - 1.
+ </para>
+
+ <para>
+ This is called usually from the buffer-update routine in the
+ pcm middle layer, which is invoked when
+ <function>snd_pcm_period_elapsed()</function> is called in the
+ interrupt routine. Then the pcm middle layer updates the
+ position and calculates the available space, and wakes up the
+ sleeping poll threads, etc.
+ </para>
+
+ <para>
+ This callback is also atomic.
+ </para>
+ </section>
+
+ <section id="pcm-interface-operators-copy-silence">
+ <title>copy and silence callbacks</title>
+ <para>
+ These callbacks are not mandatory, and can be omitted in
+ most cases. These callbacks are used when the hardware buffer
+ cannot be on the normal memory space. Some chips have their
+ own buffer on the hardware which is not mappable. In such a
+ case, you have to transfer the data manually from the memory
+ buffer to the hardware buffer. Or, if the buffer is
+ non-contiguous on both physical and virtual memory spaces,
+ these callbacks must be defined, too.
+ </para>
+
+ <para>
+ If these two callbacks are defined, copy and set-silence
+ operations are done by them. The detailed will be described in
+ the later section <link
+ linkend="buffer-and-memory"><citetitle>Buffer and Memory
+ Management</citetitle></link>.
+ </para>
+ </section>
+
+ <section id="pcm-interface-operators-ack">
+ <title>ack callback</title>
+ <para>
+ This callback is also not mandatory. This callback is called
+ when the appl_ptr is updated in read or write operations.
+ Some drivers like emu10k1-fx and cs46xx need to track the
+ current appl_ptr for the internal buffer, and this callback
+ is useful only for such a purpose.
+ </para>
+ <para>
+ This callback is atomic.
+ </para>
+ </section>
+
+ <section id="pcm-interface-operators-page-callback">
+ <title>page callback</title>
+
+ <para>
+ This callback is also not mandatory. This callback is used
+ mainly for the non-contiguous buffer. The mmap calls this
+ callback to get the page address. Some examples will be
+ explained in the later section <link
+ linkend="buffer-and-memory"><citetitle>Buffer and Memory
+ Management</citetitle></link>, too.
+ </para>
+ </section>
+ </section>
+
+ <section id="pcm-interface-interrupt-handler">
+ <title>Interrupt Handler</title>
+ <para>
+ The rest of pcm stuff is the PCM interrupt handler. The
+ role of PCM interrupt handler in the sound driver is to update
+ the buffer position and to tell the PCM middle layer when the
+ buffer position goes across the prescribed period size. To
+ inform this, call <function>snd_pcm_period_elapsed()</function>
+ function.
+ </para>
+
+ <para>
+ There are several types of sound chips to generate the interrupts.
+ </para>
+
+ <section id="pcm-interface-interrupt-handler-boundary">
+ <title>Interrupts at the period (fragment) boundary</title>
+ <para>
+ This is the most frequently found type: the hardware
+ generates an interrupt at each period boundary.
+ In this case, you can call
+ <function>snd_pcm_period_elapsed()</function> at each
+ interrupt.
+ </para>
+
+ <para>
+ <function>snd_pcm_period_elapsed()</function> takes the
+ substream pointer as its argument. Thus, you need to keep the
+ substream pointer accessible from the chip instance. For
+ example, define substream field in the chip record to hold the
+ current running substream pointer, and set the pointer value
+ at open callback (and reset at close callback).
+ </para>
+
+ <para>
+ If you aquire a spinlock in the interrupt handler, and the
+ lock is used in other pcm callbacks, too, then you have to
+ release the lock before calling
+ <function>snd_pcm_period_elapsed()</function>, because
+ <function>snd_pcm_period_elapsed()</function> calls other pcm
+ callbacks inside.
+ </para>
+
+ <para>
+ A typical coding would be like:
+
+ <example>
+ <title>Interrupt Handler Case #1</title>
+ <programlisting>
+<![CDATA[
+ static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id,
+ struct pt_regs *regs)
+ {
+ mychip_t *chip = dev_id;
+ spin_lock(&chip->lock);
+ ....
+ if (pcm_irq_invoked(chip)) {
+ /* call updater, unlock before it */
+ spin_unlock(&chip->lock);
+ snd_pcm_period_elapsed(chip->substream);
+ spin_lock(&chip->lock);
+ // acknowledge the interrupt if necessary
+ }
+ ....
+ spin_unlock(&chip->lock);
+ return IRQ_HANDLED;
+ }
+]]>
+ </programlisting>
+ </example>
+ </para>
+ </section>
+
+ <section id="pcm-interface-interrupt-handler-timer">
+ <title>High-frequent timer interrupts</title>
+ <para>
+ This is the case when the hardware doesn't generate interrupts
+ at the period boundary but do timer-interrupts at the fixed
+ timer rate (e.g. es1968 or ymfpci drivers).
+ In this case, you need to check the current hardware
+ position and accumulates the processed sample length at each
+ interrupt. When the accumulated size overcomes the period
+ size, call
+ <function>snd_pcm_period_elapsed()</function> and reset the
+ accumulator.
+ </para>
+
+ <para>
+ A typical coding would be like the following.
+
+ <example>
+ <title>Interrupt Handler Case #2</title>
+ <programlisting>
+<![CDATA[
+ static irqreturn_t snd_mychip_interrupt(int irq, void *dev_id,
+ struct pt_regs *regs)
+ {
+ mychip_t *chip = dev_id;
+ spin_lock(&chip->lock);
+ ....
+ if (pcm_irq_invoked(chip)) {
+ unsigned int last_ptr, size;
+ /* get the current hardware pointer (in frames) */
+ last_ptr = get_hw_ptr(chip);
+ /* calculate the processed frames since the
+ * last update
+ */
+ if (last_ptr < chip->last_ptr)
+ size = runtime->buffer_size + last_ptr
+ - chip->last_ptr;
+ else
+ size = last_ptr - chip->last_ptr;
+ /* remember the last updated point */
+ chip->last_ptr = last_ptr;
+ /* accumulate the size */
+ chip->size += size;
+ /* over the period boundary? */
+ if (chip->size >= runtime->period_size) {
+ /* reset the accumulator */
+ chip->size %= runtime->period_size;
+ /* call updater */
+ spin_unlock(&chip->lock);
+ snd_pcm_period_elapsed(substream);
+ spin_lock(&chip->lock);
+ }
+ // acknowledge the interrupt if necessary
+ }
+ ....
+ spin_unlock(&chip->lock);
+ return IRQ_HANDLED;
+ }
+]]>
+ </programlisting>
+ </example>
+ </para>
+ </section>
+
+ <section id="pcm-interface-interrupt-handler-both">
+ <title>On calling <function>snd_pcm_period_elapsed()</function></title>
+ <para>
+ In both cases, even if more than one period are elapsed, you
+ don't have to call
+ <function>snd_pcm_period_elapsed()</function> many times. Call
+ only once. And the pcm layer will check the current hardware
+ pointer and update to the latest status.
+ </para>
+ </section>
+ </section>
+
+ <section id="pcm-interface-atomicity">
+ <title>Atomicity</title>
+ <para>
+ One of the most important (and thus difficult to debug) problem
+ on the kernel programming is the race condition.
+ On linux kernel, usually it's solved via spin-locks or
+ semaphores. In general, if the race condition may
+ happen in the interrupt handler, it's handled as atomic, and you
+ have to use spinlock for protecting the critical session. If it
+ never happens in the interrupt and it may take relatively long
+ time, you should use semaphore.
+ </para>
+
+ <para>
+ As already seen, some pcm callbacks are atomic and some are
+ not. For example, <parameter>hw_params</parameter> callback is
+ non-atomic, while <parameter>trigger</parameter> callback is
+ atomic. This means, the latter is called already in a spinlock
+ held by the PCM middle layer. Please take this atomicity into
+ account when you use a spinlock or a semaphore in the callbacks.
+ </para>
+
+ <para>
+ In the atomic callbacks, you cannot use functions which may call
+ <function>schedule</function> or go to
+ <function>sleep</function>. The semaphore and mutex do sleep,
+ and hence they cannot be used inside the atomic callbacks
+ (e.g. <parameter>trigger</parameter> callback).
+ For taking a certain delay in such a callback, please use
+ <function>udelay()</function> or <function>mdelay()</function>.
+ </para>
+
+ <para>
+ All three atomic callbacks (trigger, pointer, and ack) are
+ called with local interrupts disabled.
+ </para>
+
+ </section>
+ <section id="pcm-interface-constraints">
+ <title>Constraints</title>
+ <para>
+ If your chip supports unconventional sample rates, or only the
+ limited samples, you need to set a constraint for the
+ condition.
+ </para>
+
+ <para>
+ For example, in order to restrict the sample rates in the some
+ supported values, use
+ <function>snd_pcm_hw_constraint_list()</function>.
+ You need to call this function in the open callback.
+
+ <example>
+ <title>Example of Hardware Constraints</title>
+ <programlisting>
+<![CDATA[
+ static unsigned int rates[] =
+ {4000, 10000, 22050, 44100};
+ static snd_pcm_hw_constraint_list_t constraints_rates = {
+ .count = ARRAY_SIZE(rates),
+ .list = rates,
+ .mask = 0,
+ };
+
+ static int snd_mychip_pcm_open(snd_pcm_substream_t *substream)
+ {
+ int err;
+ ....
+ err = snd_pcm_hw_constraint_list(substream->runtime, 0,
+ SNDRV_PCM_HW_PARAM_RATE,
+ &constraints_rates);
+ if (err < 0)
+ return err;
+ ....
+ }
+]]>
+ </programlisting>
+ </example>
+ </para>
+
+ <para>
+ There are many different constraints.
+ Look in <filename>sound/pcm.h</filename> for a complete list.
+ You can even define your own constraint rules.
+ For example, let's suppose my_chip can manage a substream of 1 channel
+ if and only if the format is S16_LE, otherwise it supports any format
+ specified in the <type>snd_pcm_hardware_t</type> stucture (or in any
+ other constraint_list). You can build a rule like this:
+
+ <example>
+ <title>Example of Hardware Constraints for Channels</title>
+ <programlisting>
+<![CDATA[
+ static int hw_rule_format_by_channels(snd_pcm_hw_params_t *params,
+ snd_pcm_hw_rule_t *rule)
+ {
+ snd_interval_t *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ snd_mask_t *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ snd_mask_t fmt;
+
+ snd_mask_any(&fmt); /* Init the struct */
+ if (c->min < 2) {
+ fmt.bits[0] &= SNDRV_PCM_FMTBIT_S16_LE;
+ return snd_mask_refine(f, &fmt);
+ }
+ return 0;
+ }
+]]>
+ </programlisting>
+ </example>
+ </para>
+
+ <para>
+ Then you need to call this function to add your rule:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ hw_rule_channels_by_format, 0, SNDRV_PCM_HW_PARAM_FORMAT,
+ -1);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The rule function is called when an application sets the number of
+ channels. But an application can set the format before the number of
+ channels. Thus you also need to define the inverse rule:
+
+ <example>
+ <title>Example of Hardware Constraints for Channels</title>
+ <programlisting>
+<![CDATA[
+ static int hw_rule_channels_by_format(snd_pcm_hw_params_t *params,
+ snd_pcm_hw_rule_t *rule)
+ {
+ snd_interval_t *c = hw_param_interval(params, SNDRV_PCM_HW_PARAM_CHANNELS);
+ snd_mask_t *f = hw_param_mask(params, SNDRV_PCM_HW_PARAM_FORMAT);
+ snd_interval_t ch;
+
+ snd_interval_any(&ch);
+ if (f->bits[0] == SNDRV_PCM_FMTBIT_S16_LE) {
+ ch.min = ch.max = 1;
+ ch.integer = 1;
+ return snd_interval_refine(c, &ch);
+ }
+ return 0;
+ }
+]]>
+ </programlisting>
+ </example>
+ </para>
+
+ <para>
+ ...and in the open callback:
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_pcm_hw_rule_add(substream->runtime, 0, SNDRV_PCM_HW_PARAM_FORMAT,
+ hw_rule_format_by_channels, 0, SNDRV_PCM_HW_PARAM_CHANNELS,
+ -1);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ I won't explain more details here, rather I
+ would like to say, <quote>Luke, use the source.</quote>
+ </para>
+ </section>
+
+ </chapter>
+
+
+<!-- ****************************************************** -->
+<!-- Control Interface -->
+<!-- ****************************************************** -->
+ <chapter id="control-interface">
+ <title>Control Interface</title>
+
+ <section id="control-interface-general">
+ <title>General</title>
+ <para>
+ The control interface is used widely for many switches,
+ sliders, etc. which are accessed from the user-space. Its most
+ important use is the mixer interface. In other words, on ALSA
+ 0.9.x, all the mixer stuff is implemented on the control kernel
+ API (while there was an independent mixer kernel API on 0.5.x).
+ </para>
+
+ <para>
+ ALSA has a well-defined AC97 control module. If your chip
+ supports only the AC97 and nothing else, you can skip this
+ section.
+ </para>
+
+ <para>
+ The control API is defined in
+ <filename>&lt;sound/control.h&gt;</filename>.
+ Include this file if you add your own controls.
+ </para>
+ </section>
+
+ <section id="control-interface-definition">
+ <title>Definition of Controls</title>
+ <para>
+ For creating a new control, you need to define the three
+ callbacks: <structfield>info</structfield>,
+ <structfield>get</structfield> and
+ <structfield>put</structfield>. Then, define a
+ <type>snd_kcontrol_new_t</type> record, such as:
+
+ <example>
+ <title>Definition of a Control</title>
+ <programlisting>
+<![CDATA[
+ static snd_kcontrol_new_t my_control __devinitdata = {
+ .iface = SNDRV_CTL_ELEM_IFACE_MIXER,
+ .name = "PCM Playback Switch",
+ .index = 0,
+ .access = SNDRV_CTL_ELEM_ACCESS_READWRITE,
+ .private_values = 0xffff,
+ .info = my_control_info,
+ .get = my_control_get,
+ .put = my_control_put
+ };
+]]>
+ </programlisting>
+ </example>
+ </para>
+
+ <para>
+ Most likely the control is created via
+ <function>snd_ctl_new1()</function>, and in such a case, you can
+ add <parameter>__devinitdata</parameter> prefix to the
+ definition like above.
+ </para>
+
+ <para>
+ The <structfield>iface</structfield> field specifies the type of
+ the control,
+ <constant>SNDRV_CTL_ELEM_IFACE_XXX</constant>. There are
+ <constant>MIXER</constant>, <constant>PCM</constant>,
+ <constant>CARD</constant>, etc.
+ </para>
+
+ <para>
+ The <structfield>name</structfield> is the name identifier
+ string. On ALSA 0.9.x, the control name is very important,
+ because its role is classified from its name. There are
+ pre-defined standard control names. The details are described in
+ the subsection
+ <link linkend="control-interface-control-names"><citetitle>
+ Control Names</citetitle></link>.
+ </para>
+
+ <para>
+ The <structfield>index</structfield> field holds the index number
+ of this control. If there are several different controls with
+ the same name, they can be distinguished by the index
+ number. This is the case when
+ several codecs exist on the card. If the index is zero, you can
+ omit the definition above.
+ </para>
+
+ <para>
+ The <structfield>access</structfield> field contains the access
+ type of this control. Give the combination of bit masks,
+ <constant>SNDRV_CTL_ELEM_ACCESS_XXX</constant>, there.
+ The detailed will be explained in the subsection
+ <link linkend="control-interface-access-flags"><citetitle>
+ Access Flags</citetitle></link>.
+ </para>
+
+ <para>
+ The <structfield>private_values</structfield> field contains
+ an arbitrary long integer value for this record. When using
+ generic <structfield>info</structfield>,
+ <structfield>get</structfield> and
+ <structfield>put</structfield> callbacks, you can pass a value
+ through this field. If several small numbers are necessary, you can
+ combine them in bitwise. Or, it's possible to give a pointer
+ (casted to unsigned long) of some record to this field, too.
+ </para>
+
+ <para>
+ The other three are
+ <link linkend="control-interface-callbacks"><citetitle>
+ callback functions</citetitle></link>.
+ </para>
+ </section>
+
+ <section id="control-interface-control-names">
+ <title>Control Names</title>
+ <para>
+ There are some standards for defining the control names. A
+ control is usually defined from the three parts as
+ <quote>SOURCE DIRECTION FUNCTION</quote>.
+ </para>
+
+ <para>
+ The first, <constant>SOURCE</constant>, specifies the source
+ of the control, and is a string such as <quote>Master</quote>,
+ <quote>PCM</quote>, <quote>CD</quote> or
+ <quote>Line</quote>. There are many pre-defined sources.
+ </para>
+
+ <para>
+ The second, <constant>DIRECTION</constant>, is one of the
+ following strings according to the direction of the control:
+ <quote>Playback</quote>, <quote>Capture</quote>, <quote>Bypass
+ Playback</quote> and <quote>Bypass Capture</quote>. Or, it can
+ be omitted, meaning both playback and capture directions.
+ </para>
+
+ <para>
+ The third, <constant>FUNCTION</constant>, is one of the
+ following strings according to the function of the control:
+ <quote>Switch</quote>, <quote>Volume</quote> and
+ <quote>Route</quote>.
+ </para>
+
+ <para>
+ The example of control names are, thus, <quote>Master Capture
+ Switch</quote> or <quote>PCM Playback Volume</quote>.
+ </para>
+
+ <para>
+ There are some exceptions:
+ </para>
+
+ <section id="control-interface-control-names-global">
+ <title>Global capture and playback</title>
+ <para>
+ <quote>Capture Source</quote>, <quote>Capture Switch</quote>
+ and <quote>Capture Volume</quote> are used for the global
+ capture (input) source, switch and volume. Similarly,
+ <quote>Playback Switch</quote> and <quote>Playback
+ Volume</quote> are used for the global output gain switch and
+ volume.
+ </para>
+ </section>
+
+ <section id="control-interface-control-names-tone">
+ <title>Tone-controls</title>
+ <para>
+ tone-control switch and volumes are specified like
+ <quote>Tone Control - XXX</quote>, e.g. <quote>Tone Control -
+ Switch</quote>, <quote>Tone Control - Bass</quote>,
+ <quote>Tone Control - Center</quote>.
+ </para>
+ </section>
+
+ <section id="control-interface-control-names-3d">
+ <title>3D controls</title>
+ <para>
+ 3D-control switches and volumes are specified like <quote>3D
+ Control - XXX</quote>, e.g. <quote>3D Control -
+ Switch</quote>, <quote>3D Control - Center</quote>, <quote>3D
+ Control - Space</quote>.
+ </para>
+ </section>
+
+ <section id="control-interface-control-names-mic">
+ <title>Mic boost</title>
+ <para>
+ Mic-boost switch is set as <quote>Mic Boost</quote> or
+ <quote>Mic Boost (6dB)</quote>.
+ </para>
+
+ <para>
+ More precise information can be found in
+ <filename>Documentation/sound/alsa/ControlNames.txt</filename>.
+ </para>
+ </section>
+ </section>
+
+ <section id="control-interface-access-flags">
+ <title>Access Flags</title>
+
+ <para>
+ The access flag is the bit-flags which specifies the access type
+ of the given control. The default access type is
+ <constant>SNDRV_CTL_ELEM_ACCESS_READWRITE</constant>,
+ which means both read and write are allowed to this control.
+ When the access flag is omitted (i.e. = 0), it is
+ regarded as <constant>READWRITE</constant> access as default.
+ </para>
+
+ <para>
+ When the control is read-only, pass
+ <constant>SNDRV_CTL_ELEM_ACCESS_READ</constant> instead.
+ In this case, you don't have to define
+ <structfield>put</structfield> callback.
+ Similarly, when the control is write-only (although it's a rare
+ case), you can use <constant>WRITE</constant> flag instead, and
+ you don't need <structfield>get</structfield> callback.
+ </para>
+
+ <para>
+ If the control value changes frequently (e.g. the VU meter),
+ <constant>VOLATILE</constant> flag should be given. This means
+ that the control may be changed without
+ <link linkend="control-interface-change-notification"><citetitle>
+ notification</citetitle></link>. Applications should poll such
+ a control constantly.
+ </para>
+
+ <para>
+ When the control is inactive, set
+ <constant>INACTIVE</constant> flag, too.
+ There are <constant>LOCK</constant> and
+ <constant>OWNER</constant> flags for changing the write
+ permissions.
+ </para>
+
+ </section>
+
+ <section id="control-interface-callbacks">
+ <title>Callbacks</title>
+
+ <section id="control-interface-callbacks-info">
+ <title>info callback</title>
+ <para>
+ The <structfield>info</structfield> callback is used to get
+ the detailed information of this control. This must store the
+ values of the given <type>snd_ctl_elem_info_t</type>
+ object. For example, for a boolean control with a single
+ element will be:
+
+ <example>
+ <title>Example of info callback</title>
+ <programlisting>
+<![CDATA[
+ static int snd_myctl_info(snd_kcontrol_t *kcontrol,
+ snd_ctl_elem_info_t *uinfo)
+ {
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_BOOLEAN;
+ uinfo->count = 1;
+ uinfo->value.integer.min = 0;
+ uinfo->value.integer.max = 1;
+ return 0;
+ }
+]]>
+ </programlisting>
+ </example>
+ </para>
+
+ <para>
+ The <structfield>type</structfield> field specifies the type
+ of the control. There are <constant>BOOLEAN</constant>,
+ <constant>INTEGER</constant>, <constant>ENUMERATED</constant>,
+ <constant>BYTES</constant>, <constant>IEC958</constant> and
+ <constant>INTEGER64</constant>. The
+ <structfield>count</structfield> field specifies the
+ number of elements in this control. For example, a stereo
+ volume would have count = 2. The
+ <structfield>value</structfield> field is a union, and
+ the values stored are depending on the type. The boolean and
+ integer are identical.
+ </para>
+
+ <para>
+ The enumerated type is a bit different from others. You'll
+ need to set the string for the currently given item index.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int snd_myctl_info(snd_kcontrol_t *kcontrol,
+ snd_ctl_elem_info_t *uinfo)
+ {
+ static char *texts[4] = {
+ "First", "Second", "Third", "Fourth"
+ };
+ uinfo->type = SNDRV_CTL_ELEM_TYPE_ENUMERATED;
+ uinfo->count = 1;
+ uinfo->value.enumerated.items = 4;
+ if (uinfo->value.enumerated.item > 3)
+ uinfo->value.enumerated.item = 3;
+ strcpy(uinfo->value.enumerated.name,
+ texts[uinfo->value.enumerated.item]);
+ return 0;
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+ </section>
+
+ <section id="control-interface-callbacks-get">
+ <title>get callback</title>
+
+ <para>
+ This callback is used to read the current value of the
+ control and to return to the user-space.
+ </para>
+
+ <para>
+ For example,
+
+ <example>
+ <title>Example of get callback</title>
+ <programlisting>
+<![CDATA[
+ static int snd_myctl_get(snd_kcontrol_t *kcontrol,
+ snd_ctl_elem_value_t *ucontrol)
+ {
+ mychip_t *chip = snd_kcontrol_chip(kcontrol);
+ ucontrol->value.integer.value[0] = get_some_value(chip);
+ return 0;
+ }
+]]>
+ </programlisting>
+ </example>
+ </para>
+
+ <para>
+ Here, the chip instance is retrieved via
+ <function>snd_kcontrol_chip()</function> macro. This macro
+ converts from kcontrol-&gt;private_data to the type defined by
+ <type>chip_t</type>. The
+ kcontrol-&gt;private_data field is
+ given as the argument of <function>snd_ctl_new()</function>
+ (see the later subsection
+ <link linkend="control-interface-constructor"><citetitle>Constructor</citetitle></link>).
+ </para>
+
+ <para>
+ The <structfield>value</structfield> field is depending on
+ the type of control as well as on info callback. For example,
+ the sb driver uses this field to store the register offset,
+ the bit-shift and the bit-mask. The
+ <structfield>private_value</structfield> is set like
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ .private_value = reg | (shift << 16) | (mask << 24)
+]]>
+ </programlisting>
+ </informalexample>
+ and is retrieved in callbacks like
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int snd_sbmixer_get_single(snd_kcontrol_t *kcontrol,
+ snd_ctl_elem_value_t *ucontrol)
+ {
+ int reg = kcontrol->private_value & 0xff;
+ int shift = (kcontrol->private_value >> 16) & 0xff;
+ int mask = (kcontrol->private_value >> 24) & 0xff;
+ ....
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ In <structfield>get</structfield> callback, you have to fill all the elements if the
+ control has more than one elements,
+ i.e. <structfield>count</structfield> &gt; 1.
+ In the example above, we filled only one element
+ (<structfield>value.integer.value[0]</structfield>) since it's
+ assumed as <structfield>count</structfield> = 1.
+ </para>
+ </section>
+
+ <section id="control-interface-callbacks-put">
+ <title>put callback</title>
+
+ <para>
+ This callback is used to write a value from the user-space.
+ </para>
+
+ <para>
+ For example,
+
+ <example>
+ <title>Example of put callback</title>
+ <programlisting>
+<![CDATA[
+ static int snd_myctl_put(snd_kcontrol_t *kcontrol,
+ snd_ctl_elem_value_t *ucontrol)
+ {
+ mychip_t *chip = snd_kcontrol_chip(kcontrol);
+ int changed = 0;
+ if (chip->current_value !=
+ ucontrol->value.integer.value[0]) {
+ change_current_value(chip,
+ ucontrol->value.integer.value[0]);
+ changed = 1;
+ }
+ return changed;
+ }
+]]>
+ </programlisting>
+ </example>
+
+ As seen above, you have to return 1 if the value is
+ changed. If the value is not changed, return 0 instead.
+ If any fatal error happens, return a negative error code as
+ usual.
+ </para>
+
+ <para>
+ Like <structfield>get</structfield> callback,
+ when the control has more than one elements,
+ all elemehts must be evaluated in this callback, too.
+ </para>
+ </section>
+
+ <section id="control-interface-callbacks-all">
+ <title>Callbacks are not atomic</title>
+ <para>
+ All these three callbacks are basically not atomic.
+ </para>
+ </section>
+ </section>
+
+ <section id="control-interface-constructor">
+ <title>Constructor</title>
+ <para>
+ When everything is ready, finally we can create a new
+ control. For creating a control, there are two functions to be
+ called, <function>snd_ctl_new1()</function> and
+ <function>snd_ctl_add()</function>.
+ </para>
+
+ <para>
+ In the simplest way, you can do like this:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ if ((err = snd_ctl_add(card, snd_ctl_new1(&my_control, chip))) < 0)
+ return err;
+]]>
+ </programlisting>
+ </informalexample>
+
+ where <parameter>my_control</parameter> is the
+ <type>snd_kcontrol_new_t</type> object defined above, and chip
+ is the object pointer to be passed to
+ kcontrol-&gt;private_data
+ which can be referred in callbacks.
+ </para>
+
+ <para>
+ <function>snd_ctl_new1()</function> allocates a new
+ <type>snd_kcontrol_t</type> instance (that's why the definition
+ of <parameter>my_control</parameter> can be with
+ <parameter>__devinitdata</parameter>
+ prefix), and <function>snd_ctl_add</function> assigns the given
+ control component to the card.
+ </para>
+ </section>
+
+ <section id="control-interface-change-notification">
+ <title>Change Notification</title>
+ <para>
+ If you need to change and update a control in the interrupt
+ routine, you can call <function>snd_ctl_notify()</function>. For
+ example,
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_ctl_notify(card, SNDRV_CTL_EVENT_MASK_VALUE, id_pointer);
+]]>
+ </programlisting>
+ </informalexample>
+
+ This function takes the card pointer, the event-mask, and the
+ control id pointer for the notification. The event-mask
+ specifies the types of notification, for example, in the above
+ example, the change of control values is notified.
+ The id pointer is the pointer of <type>snd_ctl_elem_id_t</type>
+ to be notified.
+ You can find some examples in <filename>es1938.c</filename> or
+ <filename>es1968.c</filename> for hardware volume interrupts.
+ </para>
+ </section>
+
+ </chapter>
+
+
+<!-- ****************************************************** -->
+<!-- API for AC97 Codec -->
+<!-- ****************************************************** -->
+ <chapter id="api-ac97">
+ <title>API for AC97 Codec</title>
+
+ <section>
+ <title>General</title>
+ <para>
+ The ALSA AC97 codec layer is a well-defined one, and you don't
+ have to write many codes to control it. Only low-level control
+ routines are necessary. The AC97 codec API is defined in
+ <filename>&lt;sound/ac97_codec.h&gt;</filename>.
+ </para>
+ </section>
+
+ <section id="api-ac97-example">
+ <title>Full Code Example</title>
+ <para>
+ <example>
+ <title>Example of AC97 Interface</title>
+ <programlisting>
+<![CDATA[
+ struct snd_mychip {
+ ....
+ ac97_t *ac97;
+ ....
+ };
+
+ static unsigned short snd_mychip_ac97_read(ac97_t *ac97,
+ unsigned short reg)
+ {
+ mychip_t *chip = ac97->private_data;
+ ....
+ // read a register value here from the codec
+ return the_register_value;
+ }
+
+ static void snd_mychip_ac97_write(ac97_t *ac97,
+ unsigned short reg, unsigned short val)
+ {
+ mychip_t *chip = ac97->private_data;
+ ....
+ // write the given register value to the codec
+ }
+
+ static int snd_mychip_ac97(mychip_t *chip)
+ {
+ ac97_bus_t *bus;
+ ac97_template_t ac97;
+ int err;
+ static ac97_bus_ops_t ops = {
+ .write = snd_mychip_ac97_write,
+ .read = snd_mychip_ac97_read,
+ };
+
+ if ((err = snd_ac97_bus(chip->card, 0, &ops, NULL, &bus)) < 0)
+ return err;
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ return snd_ac97_mixer(bus, &ac97, &chip->ac97);
+ }
+
+]]>
+ </programlisting>
+ </example>
+ </para>
+ </section>
+
+ <section id="api-ac97-constructor">
+ <title>Constructor</title>
+ <para>
+ For creating an ac97 instance, first call <function>snd_ac97_bus</function>
+ with an <type>ac97_bus_ops_t</type> record with callback functions.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ ac97_bus_t *bus;
+ static ac97_bus_ops_t ops = {
+ .write = snd_mychip_ac97_write,
+ .read = snd_mychip_ac97_read,
+ };
+
+ snd_ac97_bus(card, 0, &ops, NULL, &pbus);
+]]>
+ </programlisting>
+ </informalexample>
+
+ The bus record is shared among all belonging ac97 instances.
+ </para>
+
+ <para>
+ And then call <function>snd_ac97_mixer()</function> with an <type>ac97_template_t</type>
+ record together with the bus pointer created above.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ ac97_template_t ac97;
+ int err;
+
+ memset(&ac97, 0, sizeof(ac97));
+ ac97.private_data = chip;
+ snd_ac97_mixer(bus, &ac97, &chip->ac97);
+]]>
+ </programlisting>
+ </informalexample>
+
+ where chip-&gt;ac97 is the pointer of a newly created
+ <type>ac97_t</type> instance.
+ In this case, the chip pointer is set as the private data, so that
+ the read/write callback functions can refer to this chip instance.
+ This instance is not necessarily stored in the chip
+ record. When you need to change the register values from the
+ driver, or need the suspend/resume of ac97 codecs, keep this
+ pointer to pass to the corresponding functions.
+ </para>
+ </section>
+
+ <section id="api-ac97-callbacks">
+ <title>Callbacks</title>
+ <para>
+ The standard callbacks are <structfield>read</structfield> and
+ <structfield>write</structfield>. Obviously they
+ correspond to the functions for read and write accesses to the
+ hardware low-level codes.
+ </para>
+
+ <para>
+ The <structfield>read</structfield> callback returns the
+ register value specified in the argument.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static unsigned short snd_mychip_ac97_read(ac97_t *ac97,
+ unsigned short reg)
+ {
+ mychip_t *chip = ac97->private_data;
+ ....
+ return the_register_value;
+ }
+]]>
+ </programlisting>
+ </informalexample>
+
+ Here, the chip can be cast from ac97-&gt;private_data.
+ </para>
+
+ <para>
+ Meanwhile, the <structfield>write</structfield> callback is
+ used to set the register value.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static void snd_mychip_ac97_write(ac97_t *ac97,
+ unsigned short reg, unsigned short val)
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ These callbacks are non-atomic like the callbacks of control API.
+ </para>
+
+ <para>
+ There are also other callbacks:
+ <structfield>reset</structfield>,
+ <structfield>wait</structfield> and
+ <structfield>init</structfield>.
+ </para>
+
+ <para>
+ The <structfield>reset</structfield> callback is used to reset
+ the codec. If the chip requires a special way of reset, you can
+ define this callback.
+ </para>
+
+ <para>
+ The <structfield>wait</structfield> callback is used for a
+ certain wait at the standard initialization of the codec. If the
+ chip requires the extra wait-time, define this callback.
+ </para>
+
+ <para>
+ The <structfield>init</structfield> callback is used for
+ additional initialization of the codec.
+ </para>
+ </section>
+
+ <section id="api-ac97-updating-registers">
+ <title>Updating Registers in The Driver</title>
+ <para>
+ If you need to access to the codec from the driver, you can
+ call the following functions:
+ <function>snd_ac97_write()</function>,
+ <function>snd_ac97_read()</function>,
+ <function>snd_ac97_update()</function> and
+ <function>snd_ac97_update_bits()</function>.
+ </para>
+
+ <para>
+ Both <function>snd_ac97_write()</function> and
+ <function>snd_ac97_update()</function> functions are used to
+ set a value to the given register
+ (<constant>AC97_XXX</constant>). The difference between them is
+ that <function>snd_ac97_update()</function> doesn't write a
+ value if the given value has been already set, while
+ <function>snd_ac97_write()</function> always rewrites the
+ value.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_ac97_write(ac97, AC97_MASTER, 0x8080);
+ snd_ac97_update(ac97, AC97_MASTER, 0x8080);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ <function>snd_ac97_read()</function> is used to read the value
+ of the given register. For example,
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ value = snd_ac97_read(ac97, AC97_MASTER);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ <function>snd_ac97_update_bits()</function> is used to update
+ some bits of the given register.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_ac97_update_bits(ac97, reg, mask, value);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ Also, there is a function to change the sample rate (of a
+ certain register such as
+ <constant>AC97_PCM_FRONT_DAC_RATE</constant>) when VRA or
+ DRA is supported by the codec:
+ <function>snd_ac97_set_rate()</function>.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_ac97_set_rate(ac97, AC97_PCM_FRONT_DAC_RATE, 44100);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The following registers are available for setting the rate:
+ <constant>AC97_PCM_MIC_ADC_RATE</constant>,
+ <constant>AC97_PCM_FRONT_DAC_RATE</constant>,
+ <constant>AC97_PCM_LR_ADC_RATE</constant>,
+ <constant>AC97_SPDIF</constant>. When the
+ <constant>AC97_SPDIF</constant> is specified, the register is
+ not really changed but the corresponding IEC958 status bits will
+ be updated.
+ </para>
+ </section>
+
+ <section id="api-ac97-clock-adjustment">
+ <title>Clock Adjustment</title>
+ <para>
+ On some chip, the clock of the codec isn't 48000 but using a
+ PCI clock (to save a quartz!). In this case, change the field
+ bus-&gt;clock to the corresponding
+ value. For example, intel8x0
+ and es1968 drivers have the auto-measurement function of the
+ clock.
+ </para>
+ </section>
+
+ <section id="api-ac97-proc-files">
+ <title>Proc Files</title>
+ <para>
+ The ALSA AC97 interface will create a proc file such as
+ <filename>/proc/asound/card0/codec97#0/ac97#0-0</filename> and
+ <filename>ac97#0-0+regs</filename>. You can refer to these files to
+ see the current status and registers of the codec.
+ </para>
+ </section>
+
+ <section id="api-ac97-multiple-codecs">
+ <title>Multiple Codecs</title>
+ <para>
+ When there are several codecs on the same card, you need to
+ call <function>snd_ac97_new()</function> multiple times with
+ ac97.num=1 or greater. The <structfield>num</structfield> field
+ specifies the codec
+ number.
+ </para>
+
+ <para>
+ If you have set up multiple codecs, you need to either write
+ different callbacks for each codec or check
+ ac97-&gt;num in the
+ callback routines.
+ </para>
+ </section>
+
+ </chapter>
+
+
+<!-- ****************************************************** -->
+<!-- MIDI (MPU401-UART) Interface -->
+<!-- ****************************************************** -->
+ <chapter id="midi-interface">
+ <title>MIDI (MPU401-UART) Interface</title>
+
+ <section id="midi-interface-general">
+ <title>General</title>
+ <para>
+ Many soundcards have built-in MIDI (MPU401-UART)
+ interfaces. When the soundcard supports the standard MPU401-UART
+ interface, most likely you can use the ALSA MPU401-UART API. The
+ MPU401-UART API is defined in
+ <filename>&lt;sound/mpu401.h&gt;</filename>.
+ </para>
+
+ <para>
+ Some soundchips have similar but a little bit different
+ implementation of mpu401 stuff. For example, emu10k1 has its own
+ mpu401 routines.
+ </para>
+ </section>
+
+ <section id="midi-interface-constructor">
+ <title>Constructor</title>
+ <para>
+ For creating a rawmidi object, call
+ <function>snd_mpu401_uart_new()</function>.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_rawmidi_t *rmidi;
+ snd_mpu401_uart_new(card, 0, MPU401_HW_MPU401, port, integrated,
+ irq, irq_flags, &rmidi);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The first argument is the card pointer, and the second is the
+ index of this component. You can create up to 8 rawmidi
+ devices.
+ </para>
+
+ <para>
+ The third argument is the type of the hardware,
+ <constant>MPU401_HW_XXX</constant>. If it's not a special one,
+ you can use <constant>MPU401_HW_MPU401</constant>.
+ </para>
+
+ <para>
+ The 4th argument is the i/o port address. Many
+ backward-compatible MPU401 has an i/o port such as 0x330. Or, it
+ might be a part of its own PCI i/o region. It depends on the
+ chip design.
+ </para>
+
+ <para>
+ When the i/o port address above is a part of the PCI i/o
+ region, the MPU401 i/o port might have been already allocated
+ (reserved) by the driver itself. In such a case, pass non-zero
+ to the 5th argument
+ (<parameter>integrated</parameter>). Otherwise, pass 0 to it,
+ and
+ the mpu401-uart layer will allocate the i/o ports by itself.
+ </para>
+
+ <para>
+ Usually, the port address corresponds to the command port and
+ port + 1 corresponds to the data port. If not, you may change
+ the <structfield>cport</structfield> field of
+ <type>mpu401_t</type> manually
+ afterward. However, <type>mpu401_t</type> pointer is not
+ returned explicitly by
+ <function>snd_mpu401_uart_new()</function>. You need to cast
+ rmidi-&gt;private_data to
+ <type>mpu401_t</type> explicitly,
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ mpu401_t *mpu;
+ mpu = rmidi->private_data;
+]]>
+ </programlisting>
+ </informalexample>
+
+ and reset the cport as you like:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ mpu->cport = my_own_control_port;
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The 6th argument specifies the irq number for UART. If the irq
+ is already allocated, pass 0 to the 7th argument
+ (<parameter>irq_flags</parameter>). Otherwise, pass the flags
+ for irq allocation
+ (<constant>SA_XXX</constant> bits) to it, and the irq will be
+ reserved by the mpu401-uart layer. If the card doesn't generates
+ UART interrupts, pass -1 as the irq number. Then a timer
+ interrupt will be invoked for polling.
+ </para>
+ </section>
+
+ <section id="midi-interface-interrupt-handler">
+ <title>Interrupt Handler</title>
+ <para>
+ When the interrupt is allocated in
+ <function>snd_mpu401_uart_new()</function>, the private
+ interrupt handler is used, hence you don't have to do nothing
+ else than creating the mpu401 stuff. Otherwise, you have to call
+ <function>snd_mpu401_uart_interrupt()</function> explicitly when
+ a UART interrupt is invoked and checked in your own interrupt
+ handler.
+ </para>
+
+ <para>
+ In this case, you need to pass the private_data of the
+ returned rawmidi object from
+ <function>snd_mpu401_uart_new()</function> as the second
+ argument of <function>snd_mpu401_uart_interrupt()</function>.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_mpu401_uart_interrupt(irq, rmidi->private_data, regs);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+ </section>
+
+ </chapter>
+
+
+<!-- ****************************************************** -->
+<!-- RawMIDI Interface -->
+<!-- ****************************************************** -->
+ <chapter id="rawmidi-interface">
+ <title>RawMIDI Interface</title>
+
+ <section id="rawmidi-interface-overview">
+ <title>Overview</title>
+
+ <para>
+ The raw MIDI interface is used for hardware MIDI ports that can
+ be accessed as a byte stream. It is not used for synthesizer
+ chips that do not directly understand MIDI.
+ </para>
+
+ <para>
+ ALSA handles file and buffer management. All you have to do is
+ to write some code to move data between the buffer and the
+ hardware.
+ </para>
+
+ <para>
+ The rawmidi API is defined in
+ <filename>&lt;sound/rawmidi.h&gt;</filename>.
+ </para>
+ </section>
+
+ <section id="rawmidi-interface-constructor">
+ <title>Constructor</title>
+
+ <para>
+ To create a rawmidi device, call the
+ <function>snd_rawmidi_new</function> function:
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_rawmidi_t *rmidi;
+ err = snd_rawmidi_new(chip->card, "MyMIDI", 0, outs, ins, &rmidi);
+ if (err < 0)
+ return err;
+ rmidi->private_data = chip;
+ strcpy(rmidi->name, "My MIDI");
+ rmidi->info_flags = SNDRV_RAWMIDI_INFO_OUTPUT |
+ SNDRV_RAWMIDI_INFO_INPUT |
+ SNDRV_RAWMIDI_INFO_DUPLEX;
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The first argument is the card pointer, the second argument is
+ the ID string.
+ </para>
+
+ <para>
+ The third argument is the index of this component. You can
+ create up to 8 rawmidi devices.
+ </para>
+
+ <para>
+ The fourth and fifth arguments are the number of output and
+ input substreams, respectively, of this device. (A substream is
+ the equivalent of a MIDI port.)
+ </para>
+
+ <para>
+ Set the <structfield>info_flags</structfield> field to specify
+ the capabilities of the device.
+ Set <constant>SNDRV_RAWMIDI_INFO_OUTPUT</constant> if there is
+ at least one output port,
+ <constant>SNDRV_RAWMIDI_INFO_INPUT</constant> if there is at
+ least one input port,
+ and <constant>SNDRV_RAWMIDI_INFO_DUPLEX</constant> if the device
+ can handle output and input at the same time.
+ </para>
+
+ <para>
+ After the rawmidi device is created, you need to set the
+ operators (callbacks) for each substream. There are helper
+ functions to set the operators for all substream of a device:
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_OUTPUT, &snd_mymidi_output_ops);
+ snd_rawmidi_set_ops(rmidi, SNDRV_RAWMIDI_STREAM_INPUT, &snd_mymidi_input_ops);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The operators are usually defined like this:
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static snd_rawmidi_ops_t snd_mymidi_output_ops = {
+ .open = snd_mymidi_output_open,
+ .close = snd_mymidi_output_close,
+ .trigger = snd_mymidi_output_trigger,
+ };
+]]>
+ </programlisting>
+ </informalexample>
+ These callbacks are explained in the <link
+ linkend="rawmidi-interface-callbacks"><citetitle>Callbacks</citetitle></link>
+ section.
+ </para>
+
+ <para>
+ If there is more than one substream, you should give each one a
+ unique name:
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ struct list_head *list;
+ snd_rawmidi_substream_t *substream;
+ list_for_each(list, &rmidi->streams[SNDRV_RAWMIDI_STREAM_OUTPUT].substreams) {
+ substream = list_entry(list, snd_rawmidi_substream_t, list);
+ sprintf(substream->name, "My MIDI Port %d", substream->number + 1);
+ }
+ /* same for SNDRV_RAWMIDI_STREAM_INPUT */
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+ </section>
+
+ <section id="rawmidi-interface-callbacks">
+ <title>Callbacks</title>
+
+ <para>
+ In all callbacks, the private data that you've set for the
+ rawmidi device can be accessed as
+ substream-&gt;rmidi-&gt;private_data.
+ <!-- <code> isn't available before DocBook 4.3 -->
+ </para>
+
+ <para>
+ If there is more than one port, your callbacks can determine the
+ port index from the snd_rawmidi_substream_t data passed to each
+ callback:
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_rawmidi_substream_t *substream;
+ int index = substream->number;
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <section id="rawmidi-interface-op-open">
+ <title><function>open</function> callback</title>
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int snd_xxx_open(snd_rawmidi_substream_t *substream);
+]]>
+ </programlisting>
+ </informalexample>
+
+ <para>
+ This is called when a substream is opened.
+ You can initialize the hardware here, but you should not yet
+ start transmitting/receiving data.
+ </para>
+ </section>
+
+ <section id="rawmidi-interface-op-close">
+ <title><function>close</function> callback</title>
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int snd_xxx_close(snd_rawmidi_substream_t *substream);
+]]>
+ </programlisting>
+ </informalexample>
+
+ <para>
+ Guess what.
+ </para>
+
+ <para>
+ The <function>open</function> and <function>close</function>
+ callbacks of a rawmidi device are serialized with a mutex,
+ and can sleep.
+ </para>
+ </section>
+
+ <section id="rawmidi-interface-op-trigger-out">
+ <title><function>trigger</function> callback for output
+ substreams</title>
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static void snd_xxx_output_trigger(snd_rawmidi_substream_t *substream, int up);
+]]>
+ </programlisting>
+ </informalexample>
+
+ <para>
+ This is called with a nonzero <parameter>up</parameter>
+ parameter when there is some data in the substream buffer that
+ must be transmitted.
+ </para>
+
+ <para>
+ To read data from the buffer, call
+ <function>snd_rawmidi_transmit_peek</function>. It will
+ return the number of bytes that have been read; this will be
+ less than the number of bytes requested when there is no more
+ data in the buffer.
+ After the data has been transmitted successfully, call
+ <function>snd_rawmidi_transmit_ack</function> to remove the
+ data from the substream buffer:
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ unsigned char data;
+ while (snd_rawmidi_transmit_peek(substream, &data, 1) == 1) {
+ if (mychip_try_to_transmit(data))
+ snd_rawmidi_transmit_ack(substream, 1);
+ else
+ break; /* hardware FIFO full */
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ If you know beforehand that the hardware will accept data, you
+ can use the <function>snd_rawmidi_transmit</function> function
+ which reads some data and removes it from the buffer at once:
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ while (mychip_transmit_possible()) {
+ unsigned char data;
+ if (snd_rawmidi_transmit(substream, &data, 1) != 1)
+ break; /* no more data */
+ mychip_transmit(data);
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ If you know beforehand how many bytes you can accept, you can
+ use a buffer size greater than one with the
+ <function>snd_rawmidi_transmit*</function> functions.
+ </para>
+
+ <para>
+ The <function>trigger</function> callback must not sleep. If
+ the hardware FIFO is full before the substream buffer has been
+ emptied, you have to continue transmitting data later, either
+ in an interrupt handler, or with a timer if the hardware
+ doesn't have a MIDI transmit interrupt.
+ </para>
+
+ <para>
+ The <function>trigger</function> callback is called with a
+ zero <parameter>up</parameter> parameter when the transmission
+ of data should be aborted.
+ </para>
+ </section>
+
+ <section id="rawmidi-interface-op-trigger-in">
+ <title><function>trigger</function> callback for input
+ substreams</title>
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static void snd_xxx_input_trigger(snd_rawmidi_substream_t *substream, int up);
+]]>
+ </programlisting>
+ </informalexample>
+
+ <para>
+ This is called with a nonzero <parameter>up</parameter>
+ parameter to enable receiving data, or with a zero
+ <parameter>up</parameter> parameter do disable receiving data.
+ </para>
+
+ <para>
+ The <function>trigger</function> callback must not sleep; the
+ actual reading of data from the device is usually done in an
+ interrupt handler.
+ </para>
+
+ <para>
+ When data reception is enabled, your interrupt handler should
+ call <function>snd_rawmidi_receive</function> for all received
+ data:
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ void snd_mychip_midi_interrupt(...)
+ {
+ while (mychip_midi_available()) {
+ unsigned char data;
+ data = mychip_midi_read();
+ snd_rawmidi_receive(substream, &data, 1);
+ }
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+ </section>
+
+ <section id="rawmidi-interface-op-drain">
+ <title><function>drain</function> callback</title>
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static void snd_xxx_drain(snd_rawmidi_substream_t *substream);
+]]>
+ </programlisting>
+ </informalexample>
+
+ <para>
+ This is only used with output substreams. This function should wait
+ until all data read from the substream buffer has been transmitted.
+ This ensures that the device can be closed and the driver unloaded
+ without losing data.
+ </para>
+
+ <para>
+ This callback is optional. If you do not set
+ <structfield>drain</structfield> in the snd_rawmidi_ops_t
+ structure, ALSA will simply wait for 50&nbsp;milliseconds
+ instead.
+ </para>
+ </section>
+ </section>
+
+ </chapter>
+
+
+<!-- ****************************************************** -->
+<!-- Miscellaneous Devices -->
+<!-- ****************************************************** -->
+ <chapter id="misc-devices">
+ <title>Miscellaneous Devices</title>
+
+ <section id="misc-devices-opl3">
+ <title>FM OPL3</title>
+ <para>
+ The FM OPL3 is still used on many chips (mainly for backward
+ compatibility). ALSA has a nice OPL3 FM control layer, too. The
+ OPL3 API is defined in
+ <filename>&lt;sound/opl3.h&gt;</filename>.
+ </para>
+
+ <para>
+ FM registers can be directly accessed through direct-FM API,
+ defined in <filename>&lt;sound/asound_fm.h&gt;</filename>. In
+ ALSA native mode, FM registers are accessed through
+ Hardware-Dependant Device direct-FM extension API, whereas in
+ OSS compatible mode, FM registers can be accessed with OSS
+ direct-FM compatible API on <filename>/dev/dmfmX</filename> device.
+ </para>
+
+ <para>
+ For creating the OPL3 component, you have two functions to
+ call. The first one is a constructor for <type>opl3_t</type>
+ instance.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ opl3_t *opl3;
+ snd_opl3_create(card, lport, rport, OPL3_HW_OPL3_XXX,
+ integrated, &opl3);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The first argument is the card pointer, the second one is the
+ left port address, and the third is the right port address. In
+ most cases, the right port is placed at the left port + 2.
+ </para>
+
+ <para>
+ The fourth argument is the hardware type.
+ </para>
+
+ <para>
+ When the left and right ports have been already allocated by
+ the card driver, pass non-zero to the fifth argument
+ (<parameter>integrated</parameter>). Otherwise, opl3 module will
+ allocate the specified ports by itself.
+ </para>
+
+ <para>
+ When the accessing to the hardware requires special method
+ instead of the standard I/O access, you can create opl3 instance
+ separately with <function>snd_opl3_new()</function>.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ opl3_t *opl3;
+ snd_opl3_new(card, OPL3_HW_OPL3_XXX, &opl3);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ Then set <structfield>command</structfield>,
+ <structfield>private_data</structfield> and
+ <structfield>private_free</structfield> for the private
+ access function, the private data and the destructor.
+ The l_port and r_port are not necessarily set. Only the
+ command must be set properly. You can retrieve the data
+ from opl3-&gt;private_data field.
+ </para>
+
+ <para>
+ After creating the opl3 instance via <function>snd_opl3_new()</function>,
+ call <function>snd_opl3_init()</function> to initialize the chip to the
+ proper state. Note that <function>snd_opl3_create()</function> always
+ calls it internally.
+ </para>
+
+ <para>
+ If the opl3 instance is created successfully, then create a
+ hwdep device for this opl3.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_hwdep_t *opl3hwdep;
+ snd_opl3_hwdep_new(opl3, 0, 1, &opl3hwdep);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The first argument is the <type>opl3_t</type> instance you
+ created, and the second is the index number, usually 0.
+ </para>
+
+ <para>
+ The third argument is the index-offset for the sequencer
+ client assigned to the OPL3 port. When there is an MPU401-UART,
+ give 1 for here (UART always takes 0).
+ </para>
+ </section>
+
+ <section id="misc-devices-hardware-dependent">
+ <title>Hardware-Dependent Devices</title>
+ <para>
+ Some chips need the access from the user-space for special
+ controls or for loading the micro code. In such a case, you can
+ create a hwdep (hardware-dependent) device. The hwdep API is
+ defined in <filename>&lt;sound/hwdep.h&gt;</filename>. You can
+ find examples in opl3 driver or
+ <filename>isa/sb/sb16_csp.c</filename>.
+ </para>
+
+ <para>
+ Creation of the <type>hwdep</type> instance is done via
+ <function>snd_hwdep_new()</function>.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_hwdep_t *hw;
+ snd_hwdep_new(card, "My HWDEP", 0, &hw);
+]]>
+ </programlisting>
+ </informalexample>
+
+ where the third argument is the index number.
+ </para>
+
+ <para>
+ You can then pass any pointer value to the
+ <parameter>private_data</parameter>.
+ If you assign a private data, you should define the
+ destructor, too. The destructor function is set to
+ <structfield>private_free</structfield> field.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ mydata_t *p = kmalloc(sizeof(*p), GFP_KERNEL);
+ hw->private_data = p;
+ hw->private_free = mydata_free;
+]]>
+ </programlisting>
+ </informalexample>
+
+ and the implementation of destructor would be:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static void mydata_free(snd_hwdep_t *hw)
+ {
+ mydata_t *p = hw->private_data;
+ kfree(p);
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The arbitrary file operations can be defined for this
+ instance. The file operators are defined in
+ <parameter>ops</parameter> table. For example, assume that
+ this chip needs an ioctl.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ hw->ops.open = mydata_open;
+ hw->ops.ioctl = mydata_ioctl;
+ hw->ops.release = mydata_release;
+]]>
+ </programlisting>
+ </informalexample>
+
+ And implement the callback functions as you like.
+ </para>
+ </section>
+
+ <section id="misc-devices-IEC958">
+ <title>IEC958 (S/PDIF)</title>
+ <para>
+ Usually the controls for IEC958 devices are implemented via
+ control interface. There is a macro to compose a name string for
+ IEC958 controls, <function>SNDRV_CTL_NAME_IEC958()</function>
+ defined in <filename>&lt;include/asound.h&gt;</filename>.
+ </para>
+
+ <para>
+ There are some standard controls for IEC958 status bits. These
+ controls use the type <type>SNDRV_CTL_ELEM_TYPE_IEC958</type>,
+ and the size of element is fixed as 4 bytes array
+ (value.iec958.status[x]). For <structfield>info</structfield>
+ callback, you don't specify
+ the value field for this type (the count field must be set,
+ though).
+ </para>
+
+ <para>
+ <quote>IEC958 Playback Con Mask</quote> is used to return the
+ bit-mask for the IEC958 status bits of consumer mode. Similarly,
+ <quote>IEC958 Playback Pro Mask</quote> returns the bitmask for
+ professional mode. They are read-only controls, and are defined
+ as MIXER controls (iface =
+ <constant>SNDRV_CTL_ELEM_IFACE_MIXER</constant>).
+ </para>
+
+ <para>
+ Meanwhile, <quote>IEC958 Playback Default</quote> control is
+ defined for getting and setting the current default IEC958
+ bits. Note that this one is usually defined as a PCM control
+ (iface = <constant>SNDRV_CTL_ELEM_IFACE_PCM</constant>),
+ although in some places it's defined as a MIXER control.
+ </para>
+
+ <para>
+ In addition, you can define the control switches to
+ enable/disable or to set the raw bit mode. The implementation
+ will depend on the chip, but the control should be named as
+ <quote>IEC958 xxx</quote>, preferably using
+ <function>SNDRV_CTL_NAME_IEC958()</function> macro.
+ </para>
+
+ <para>
+ You can find several cases, for example,
+ <filename>pci/emu10k1</filename>,
+ <filename>pci/ice1712</filename>, or
+ <filename>pci/cmipci.c</filename>.
+ </para>
+ </section>
+
+ </chapter>
+
+
+<!-- ****************************************************** -->
+<!-- Buffer and Memory Management -->
+<!-- ****************************************************** -->
+ <chapter id="buffer-and-memory">
+ <title>Buffer and Memory Management</title>
+
+ <section id="buffer-and-memory-buffer-types">
+ <title>Buffer Types</title>
+ <para>
+ ALSA provides several different buffer allocation functions
+ depending on the bus and the architecture. All these have a
+ consistent API. The allocation of physically-contiguous pages is
+ done via
+ <function>snd_malloc_xxx_pages()</function> function, where xxx
+ is the bus type.
+ </para>
+
+ <para>
+ The allocation of pages with fallback is
+ <function>snd_malloc_xxx_pages_fallback()</function>. This
+ function tries to allocate the specified pages but if the pages
+ are not available, it tries to reduce the page sizes until the
+ enough space is found.
+ </para>
+
+ <para>
+ For releasing the space, call
+ <function>snd_free_xxx_pages()</function> function.
+ </para>
+
+ <para>
+ Usually, ALSA drivers try to allocate and reserve
+ a large contiguous physical space
+ at the time the module is loaded for the later use.
+ This is called <quote>pre-allocation</quote>.
+ As already written, you can call the following function at the
+ construction of pcm instance (in the case of PCI bus).
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_pcm_lib_preallocate_pages_for_all(pcm, SNDRV_DMA_TYPE_DEV,
+ snd_dma_pci_data(pci), size, max);
+]]>
+ </programlisting>
+ </informalexample>
+
+ where <parameter>size</parameter> is the byte size to be
+ pre-allocated and the <parameter>max</parameter> is the maximal
+ size to be changed via <filename>prealloc</filename> proc file.
+ The allocator will try to get as large area as possible
+ within the given size.
+ </para>
+
+ <para>
+ The second argument (type) and the third argument (device pointer)
+ are dependent on the bus.
+ In the case of ISA bus, pass <function>snd_dma_isa_data()</function>
+ as the third argument with <constant>SNDRV_DMA_TYPE_DEV</constant> type.
+ For the continuous buffer unrelated to the bus can be pre-allocated
+ with <constant>SNDRV_DMA_TYPE_CONTINUOUS</constant> type and the
+ <function>snd_dma_continuous_data(GFP_KERNEL)</function> device pointer,
+ whereh <constant>GFP_KERNEL</constant> is the kernel allocation flag to
+ use. For the SBUS, <constant>SNDRV_DMA_TYPE_SBUS</constant> and
+ <function>snd_dma_sbus_data(sbus_dev)</function> are used instead.
+ For the PCI scatter-gather buffers, use
+ <constant>SNDRV_DMA_TYPE_DEV_SG</constant> with
+ <function>snd_dma_pci_data(pci)</function>
+ (see the section
+ <link linkend="buffer-and-memory-non-contiguous"><citetitle>Non-Contiguous Buffers
+ </citetitle></link>).
+ </para>
+
+ <para>
+ Once when the buffer is pre-allocated, you can use the
+ allocator in the <structfield>hw_params</structfield> callback
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_pcm_lib_malloc_pages(substream, size);
+]]>
+ </programlisting>
+ </informalexample>
+
+ Note that you have to pre-allocate to use this function.
+ </para>
+ </section>
+
+ <section id="buffer-and-memory-external-hardware">
+ <title>External Hardware Buffers</title>
+ <para>
+ Some chips have their own hardware buffers and the DMA
+ transfer from the host memory is not available. In such a case,
+ you need to either 1) copy/set the audio data directly to the
+ external hardware buffer, or 2) make an intermediate buffer and
+ copy/set the data from it to the external hardware buffer in
+ interrupts (or in tasklets, preferably).
+ </para>
+
+ <para>
+ The first case works fine if the external hardware buffer is enough
+ large. This method doesn't need any extra buffers and thus is
+ more effective. You need to define the
+ <structfield>copy</structfield> and
+ <structfield>silence</structfield> callbacks for
+ the data transfer. However, there is a drawback: it cannot
+ be mmapped. The examples are GUS's GF1 PCM or emu8000's
+ wavetable PCM.
+ </para>
+
+ <para>
+ The second case allows the mmap of the buffer, although you have
+ to handle an interrupt or a tasklet for transferring the data
+ from the intermediate buffer to the hardware buffer. You can find an
+ example in vxpocket driver.
+ </para>
+
+ <para>
+ Another case is that the chip uses a PCI memory-map
+ region for the buffer instead of the host memory. In this case,
+ mmap is available only on certain architectures like intel. In
+ non-mmap mode, the data cannot be transferred as the normal
+ way. Thus you need to define <structfield>copy</structfield> and
+ <structfield>silence</structfield> callbacks as well
+ as in the cases above. The examples are found in
+ <filename>rme32.c</filename> and <filename>rme96.c</filename>.
+ </para>
+
+ <para>
+ The implementation of <structfield>copy</structfield> and
+ <structfield>silence</structfield> callbacks depends upon
+ whether the hardware supports interleaved or non-interleaved
+ samples. The <structfield>copy</structfield> callback is
+ defined like below, a bit
+ differently depending whether the direction is playback or
+ capture:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int playback_copy(snd_pcm_substream_t *substream, int channel,
+ snd_pcm_uframes_t pos, void *src, snd_pcm_uframes_t count);
+ static int capture_copy(snd_pcm_substream_t *substream, int channel,
+ snd_pcm_uframes_t pos, void *dst, snd_pcm_uframes_t count);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ In the case of interleaved samples, the second argument
+ (<parameter>channel</parameter>) is not used. The third argument
+ (<parameter>pos</parameter>) points the
+ current position offset in frames.
+ </para>
+
+ <para>
+ The meaning of the fourth argument is different between
+ playback and capture. For playback, it holds the source data
+ pointer, and for capture, it's the destination data pointer.
+ </para>
+
+ <para>
+ The last argument is the number of frames to be copied.
+ </para>
+
+ <para>
+ What you have to do in this callback is again different
+ between playback and capture directions. In the case of
+ playback, you do: copy the given amount of data
+ (<parameter>count</parameter>) at the specified pointer
+ (<parameter>src</parameter>) to the specified offset
+ (<parameter>pos</parameter>) on the hardware buffer. When
+ coded like memcpy-like way, the copy would be like:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ my_memcpy(my_buffer + frames_to_bytes(runtime, pos), src,
+ frames_to_bytes(runtime, count));
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ For the capture direction, you do: copy the given amount of
+ data (<parameter>count</parameter>) at the specified offset
+ (<parameter>pos</parameter>) on the hardware buffer to the
+ specified pointer (<parameter>dst</parameter>).
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ my_memcpy(dst, my_buffer + frames_to_bytes(runtime, pos),
+ frames_to_bytes(runtime, count));
+]]>
+ </programlisting>
+ </informalexample>
+
+ Note that both of the position and the data amount are given
+ in frames.
+ </para>
+
+ <para>
+ In the case of non-interleaved samples, the implementation
+ will be a bit more complicated.
+ </para>
+
+ <para>
+ You need to check the channel argument, and if it's -1, copy
+ the whole channels. Otherwise, you have to copy only the
+ specified channel. Please check
+ <filename>isa/gus/gus_pcm.c</filename> as an example.
+ </para>
+
+ <para>
+ The <structfield>silence</structfield> callback is also
+ implemented in a similar way.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int silence(snd_pcm_substream_t *substream, int channel,
+ snd_pcm_uframes_t pos, snd_pcm_uframes_t count);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The meanings of arguments are identical with the
+ <structfield>copy</structfield>
+ callback, although there is no <parameter>src/dst</parameter>
+ argument. In the case of interleaved samples, the channel
+ argument has no meaning, as well as on
+ <structfield>copy</structfield> callback.
+ </para>
+
+ <para>
+ The role of <structfield>silence</structfield> callback is to
+ set the given amount
+ (<parameter>count</parameter>) of silence data at the
+ specified offset (<parameter>pos</parameter>) on the hardware
+ buffer. Suppose that the data format is signed (that is, the
+ silent-data is 0), and the implementation using a memset-like
+ function would be like:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ my_memcpy(my_buffer + frames_to_bytes(runtime, pos), 0,
+ frames_to_bytes(runtime, count));
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ In the case of non-interleaved samples, again, the
+ implementation becomes a bit more complicated. See, for example,
+ <filename>isa/gus/gus_pcm.c</filename>.
+ </para>
+ </section>
+
+ <section id="buffer-and-memory-non-contiguous">
+ <title>Non-Contiguous Buffers</title>
+ <para>
+ If your hardware supports the page table like emu10k1 or the
+ buffer descriptors like via82xx, you can use the scatter-gather
+ (SG) DMA. ALSA provides an interface for handling SG-buffers.
+ The API is provided in <filename>&lt;sound/pcm.h&gt;</filename>.
+ </para>
+
+ <para>
+ For creating the SG-buffer handler, call
+ <function>snd_pcm_lib_preallocate_pages()</function> or
+ <function>snd_pcm_lib_preallocate_pages_for_all()</function>
+ with <constant>SNDRV_DMA_TYPE_DEV_SG</constant>
+ in the PCM constructor like other PCI pre-allocator.
+ You need to pass the <function>snd_dma_pci_data(pci)</function>,
+ where pci is the struct <structname>pci_dev</structname> pointer
+ of the chip as well.
+ The <type>snd_sg_buf_t</type> instance is created as
+ substream-&gt;dma_private. You can cast
+ the pointer like:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_pcm_sgbuf_t *sgbuf = (snd_pcm_sgbuf_t*)substream->dma_private;
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ Then call <function>snd_pcm_lib_malloc_pages()</function>
+ in <structfield>hw_params</structfield> callback
+ as well as in the case of normal PCI buffer.
+ The SG-buffer handler will allocate the non-contiguous kernel
+ pages of the given size and map them onto the virtually contiguous
+ memory. The virtual pointer is addressed in runtime-&gt;dma_area.
+ The physical address (runtime-&gt;dma_addr) is set to zero,
+ because the buffer is physically non-contigous.
+ The physical address table is set up in sgbuf-&gt;table.
+ You can get the physical address at a certain offset via
+ <function>snd_pcm_sgbuf_get_addr()</function>.
+ </para>
+
+ <para>
+ When a SG-handler is used, you need to set
+ <function>snd_pcm_sgbuf_ops_page</function> as
+ the <structfield>page</structfield> callback.
+ (See <link linkend="pcm-interface-operators-page-callback">
+ <citetitle>page callback section</citetitle></link>.)
+ </para>
+
+ <para>
+ For releasing the data, call
+ <function>snd_pcm_lib_free_pages()</function> in the
+ <structfield>hw_free</structfield> callback as usual.
+ </para>
+ </section>
+
+ <section id="buffer-and-memory-vmalloced">
+ <title>Vmalloc'ed Buffers</title>
+ <para>
+ It's possible to use a buffer allocated via
+ <function>vmalloc</function>, for example, for an intermediate
+ buffer. Since the allocated pages are not contiguous, you need
+ to set the <structfield>page</structfield> callback to obtain
+ the physical address at every offset.
+ </para>
+
+ <para>
+ The implementation of <structfield>page</structfield> callback
+ would be like this:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ #include <linux/vmalloc.h>
+
+ /* get the physical page pointer on the given offset */
+ static struct page *mychip_page(snd_pcm_substream_t *substream,
+ unsigned long offset)
+ {
+ void *pageptr = substream->runtime->dma_area + offset;
+ return vmalloc_to_page(pageptr);
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+ </section>
+
+ </chapter>
+
+
+<!-- ****************************************************** -->
+<!-- Proc Interface -->
+<!-- ****************************************************** -->
+ <chapter id="proc-interface">
+ <title>Proc Interface</title>
+ <para>
+ ALSA provides an easy interface for procfs. The proc files are
+ very useful for debugging. I recommend you set up proc files if
+ you write a driver and want to get a running status or register
+ dumps. The API is found in
+ <filename>&lt;sound/info.h&gt;</filename>.
+ </para>
+
+ <para>
+ For creating a proc file, call
+ <function>snd_card_proc_new()</function>.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_info_entry_t *entry;
+ int err = snd_card_proc_new(card, "my-file", &entry);
+]]>
+ </programlisting>
+ </informalexample>
+
+ where the second argument specifies the proc-file name to be
+ created. The above example will create a file
+ <filename>my-file</filename> under the card directory,
+ e.g. <filename>/proc/asound/card0/my-file</filename>.
+ </para>
+
+ <para>
+ Like other components, the proc entry created via
+ <function>snd_card_proc_new()</function> will be registered and
+ released automatically in the card registration and release
+ functions.
+ </para>
+
+ <para>
+ When the creation is successful, the function stores a new
+ instance at the pointer given in the third argument.
+ It is initialized as a text proc file for read only. For using
+ this proc file as a read-only text file as it is, set the read
+ callback with a private data via
+ <function>snd_info_set_text_ops()</function>.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_info_set_text_ops(entry, chip, read_size, my_proc_read);
+]]>
+ </programlisting>
+ </informalexample>
+
+ where the second argument (<parameter>chip</parameter>) is the
+ private data to be used in the callbacks. The third parameter
+ specifies the read buffer size and the fourth
+ (<parameter>my_proc_read</parameter>) is the callback function, which
+ is defined like
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static void my_proc_read(snd_info_entry_t *entry,
+ snd_info_buffer_t *buffer);
+]]>
+ </programlisting>
+ </informalexample>
+
+ </para>
+
+ <para>
+ In the read callback, use <function>snd_iprintf()</function> for
+ output strings, which works just like normal
+ <function>printf()</function>. For example,
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static void my_proc_read(snd_info_entry_t *entry,
+ snd_info_buffer_t *buffer)
+ {
+ chip_t *chip = entry->private_data;
+
+ snd_iprintf(buffer, "This is my chip!\n");
+ snd_iprintf(buffer, "Port = %ld\n", chip->port);
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The file permission can be changed afterwards. As default, it's
+ set as read only for all users. If you want to add the write
+ permission to the user (root as default), set like below:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ entry->mode = S_IFREG | S_IRUGO | S_IWUSR;
+]]>
+ </programlisting>
+ </informalexample>
+
+ and set the write buffer size and the callback
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ entry->c.text.write_size = 256;
+ entry->c.text.write = my_proc_write;
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The buffer size for read is set to 1024 implicitly by
+ <function>snd_info_set_text_ops()</function>. It should suffice
+ in most cases (the size will be aligned to
+ <constant>PAGE_SIZE</constant> anyway), but if you need to handle
+ very large text files, you can set it explicitly, too.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ entry->c.text.read_size = 65536;
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ For the write callback, you can use
+ <function>snd_info_get_line()</function> to get a text line, and
+ <function>snd_info_get_str()</function> to retrieve a string from
+ the line. Some examples are found in
+ <filename>core/oss/mixer_oss.c</filename>, core/oss/and
+ <filename>pcm_oss.c</filename>.
+ </para>
+
+ <para>
+ For a raw-data proc-file, set the attributes like the following:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static struct snd_info_entry_ops my_file_io_ops = {
+ .read = my_file_io_read,
+ };
+
+ entry->content = SNDRV_INFO_CONTENT_DATA;
+ entry->private_data = chip;
+ entry->c.ops = &my_file_io_ops;
+ entry->size = 4096;
+ entry->mode = S_IFREG | S_IRUGO;
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The callback is much more complicated than the text-file
+ version. You need to use a low-level i/o functions such as
+ <function>copy_from/to_user()</function> to transfer the
+ data.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static long my_file_io_read(snd_info_entry_t *entry,
+ void *file_private_data,
+ struct file *file,
+ char *buf,
+ unsigned long count,
+ unsigned long pos)
+ {
+ long size = count;
+ if (pos + size > local_max_size)
+ size = local_max_size - pos;
+ if (copy_to_user(buf, local_data + pos, size))
+ return -EFAULT;
+ return size;
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ </chapter>
+
+
+<!-- ****************************************************** -->
+<!-- Power Management -->
+<!-- ****************************************************** -->
+ <chapter id="power-management">
+ <title>Power Management</title>
+ <para>
+ If the chip is supposed to work with with suspend/resume
+ functions, you need to add the power-management codes to the
+ driver. The additional codes for the power-management should be
+ <function>ifdef</function>'ed with
+ <constant>CONFIG_PM</constant>.
+ </para>
+
+ <para>
+ ALSA provides the common power-management layer. Each card driver
+ needs to have only low-level suspend and resume callbacks.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ #ifdef CONFIG_PM
+ static int snd_my_suspend(snd_card_t *card, pm_message_t state)
+ {
+ .... // do things for suspsend
+ return 0;
+ }
+ static int snd_my_resume(snd_card_t *card)
+ {
+ .... // do things for suspsend
+ return 0;
+ }
+ #endif
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The scheme of the real suspend job is as following.
+
+ <orderedlist>
+ <listitem><para>Retrieve the chip data from pm_private_data field.</para></listitem>
+ <listitem><para>Call <function>snd_pcm_suspend_all()</function> to suspend the running PCM streams.</para></listitem>
+ <listitem><para>Save the register values if necessary.</para></listitem>
+ <listitem><para>Stop the hardware if necessary.</para></listitem>
+ <listitem><para>Disable the PCI device by calling <function>pci_disable_device()</function>.</para></listitem>
+ </orderedlist>
+ </para>
+
+ <para>
+ A typical code would be like:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int mychip_suspend(snd_card_t *card, pm_message_t state)
+ {
+ /* (1) */
+ mychip_t *chip = card->pm_private_data;
+ /* (2) */
+ snd_pcm_suspend_all(chip->pcm);
+ /* (3) */
+ snd_mychip_save_registers(chip);
+ /* (4) */
+ snd_mychip_stop_hardware(chip);
+ /* (5) */
+ pci_disable_device(chip->pci);
+ return 0;
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The scheme of the real resume job is as following.
+
+ <orderedlist>
+ <listitem><para>Retrieve the chip data from pm_private_data field.</para></listitem>
+ <listitem><para>Enable the pci device again by calling
+ <function>pci_enable_device()</function>.</para></listitem>
+ <listitem><para>Re-initialize the chip.</para></listitem>
+ <listitem><para>Restore the saved registers if necessary.</para></listitem>
+ <listitem><para>Resume the mixer, e.g. calling
+ <function>snd_ac97_resume()</function>.</para></listitem>
+ <listitem><para>Restart the hardware (if any).</para></listitem>
+ </orderedlist>
+ </para>
+
+ <para>
+ A typical code would be like:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static void mychip_resume(mychip_t *chip)
+ {
+ /* (1) */
+ mychip_t *chip = card->pm_private_data;
+ /* (2) */
+ pci_enable_device(chip->pci);
+ /* (3) */
+ snd_mychip_reinit_chip(chip);
+ /* (4) */
+ snd_mychip_restore_registers(chip);
+ /* (5) */
+ snd_ac97_resume(chip->ac97);
+ /* (6) */
+ snd_mychip_restart_chip(chip);
+ return 0;
+ }
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ OK, we have all callbacks now. Let's set up them now. In the
+ initialization of the card, add the following:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int __devinit snd_mychip_probe(struct pci_dev *pci,
+ const struct pci_device_id *pci_id)
+ {
+ ....
+ snd_card_t *card;
+ mychip_t *chip;
+ ....
+ snd_card_set_pm_callback(card, snd_my_suspend, snd_my_resume, chip);
+ ....
+ }
+]]>
+ </programlisting>
+ </informalexample>
+
+ Here you don't have to put ifdef CONFIG_PM around, since it's already
+ checked in the header and expanded to empty if not needed.
+ </para>
+
+ <para>
+ If you need a space for saving the registers, you'll need to
+ allocate the buffer for it here, too, since it would be fatal
+ if you cannot allocate a memory in the suspend phase.
+ The allocated buffer should be released in the corresponding
+ destructor.
+ </para>
+
+ <para>
+ And next, set suspend/resume callbacks to the pci_driver,
+ This can be done by passing a macro SND_PCI_PM_CALLBACKS
+ in the pci_driver struct. This macro is expanded to the correct
+ (global) callbacks if CONFIG_PM is set.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static struct pci_driver driver = {
+ .name = "My Chip",
+ .id_table = snd_my_ids,
+ .probe = snd_my_probe,
+ .remove = __devexit_p(snd_my_remove),
+ SND_PCI_PM_CALLBACKS
+ };
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ </chapter>
+
+
+<!-- ****************************************************** -->
+<!-- Module Parameters -->
+<!-- ****************************************************** -->
+ <chapter id="module-parameters">
+ <title>Module Parameters</title>
+ <para>
+ There are standard module options for ALSA. At least, each
+ module should have <parameter>index</parameter>,
+ <parameter>id</parameter> and <parameter>enable</parameter>
+ options.
+ </para>
+
+ <para>
+ If the module supports multiple cards (usually up to
+ 8 = <constant>SNDRV_CARDS</constant> cards), they should be
+ arrays. The default initial values are defined already as
+ constants for ease of programming:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ static int index[SNDRV_CARDS] = SNDRV_DEFAULT_IDX;
+ static char *id[SNDRV_CARDS] = SNDRV_DEFAULT_STR;
+ static int enable[SNDRV_CARDS] = SNDRV_DEFAULT_ENABLE_PNP;
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ If the module supports only a single card, they could be single
+ variables, instead. <parameter>enable</parameter> option is not
+ always necessary in this case, but it wouldn't be so bad to have a
+ dummy option for compatibility.
+ </para>
+
+ <para>
+ The module parameters must be declared with the standard
+ <function>module_param()()</function>,
+ <function>module_param_array()()</function> and
+ <function>MODULE_PARM_DESC()</function> macros.
+ </para>
+
+ <para>
+ The typical coding would be like below:
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ #define CARD_NAME "My Chip"
+
+ module_param_array(index, int, NULL, 0444);
+ MODULE_PARM_DESC(index, "Index value for " CARD_NAME " soundcard.");
+ module_param_array(id, charp, NULL, 0444);
+ MODULE_PARM_DESC(id, "ID string for " CARD_NAME " soundcard.");
+ module_param_array(enable, bool, NULL, 0444);
+ MODULE_PARM_DESC(enable, "Enable " CARD_NAME " soundcard.");
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ Also, don't forget to define the module description, classes,
+ license and devices. Especially, the recent modprobe requires to
+ define the module license as GPL, etc., otherwise the system is
+ shown as <quote>tainted</quote>.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ MODULE_DESCRIPTION("My Chip");
+ MODULE_LICENSE("GPL");
+ MODULE_SUPPORTED_DEVICE("{{Vendor,My Chip Name}}");
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ </chapter>
+
+
+<!-- ****************************************************** -->
+<!-- How To Put Your Driver -->
+<!-- ****************************************************** -->
+ <chapter id="how-to-put-your-driver">
+ <title>How To Put Your Driver Into ALSA Tree</title>
+ <section>
+ <title>General</title>
+ <para>
+ So far, you've learned how to write the driver codes.
+ And you might have a question now: how to put my own
+ driver into the ALSA driver tree?
+ Here (finally :) the standard procedure is described briefly.
+ </para>
+
+ <para>
+ Suppose that you'll create a new PCI driver for the card
+ <quote>xyz</quote>. The card module name would be
+ snd-xyz. The new driver is usually put into alsa-driver
+ tree, <filename>alsa-driver/pci</filename> directory in
+ the case of PCI cards.
+ Then the driver is evaluated, audited and tested
+ by developers and users. After a certain time, the driver
+ will go to alsa-kernel tree (to the corresponding directory,
+ such as <filename>alsa-kernel/pci</filename>) and eventually
+ integrated into Linux 2.6 tree (the directory would be
+ <filename>linux/sound/pci</filename>).
+ </para>
+
+ <para>
+ In the following sections, the driver code is supposed
+ to be put into alsa-driver tree. The two cases are assumed:
+ a driver consisting of a single source file and one consisting
+ of several source files.
+ </para>
+ </section>
+
+ <section>
+ <title>Driver with A Single Source File</title>
+ <para>
+ <orderedlist>
+ <listitem>
+ <para>
+ Modify alsa-driver/pci/Makefile
+ </para>
+
+ <para>
+ Suppose you have a file xyz.c. Add the following
+ two lines
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd-xyz-objs := xyz.o
+ obj-$(CONFIG_SND_XYZ) += snd-xyz.o
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+ </listitem>
+
+ <listitem>
+ <para>
+ Create the Kconfig entry
+ </para>
+
+ <para>
+ Add the new entry of Kconfig for your xyz driver.
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ config SND_XYZ
+ tristate "Foobar XYZ"
+ depends on SND
+ select SND_PCM
+ help
+ Say Y here to include support for Foobar XYZ soundcard.
+
+ To compile this driver as a module, choose M here: the module
+ will be called snd-xyz.
+]]>
+ </programlisting>
+ </informalexample>
+
+ the line, select SND_PCM, specifies that the driver xyz supports
+ PCM. In addition to SND_PCM, the following components are
+ supported for select command:
+ SND_RAWMIDI, SND_TIMER, SND_HWDEP, SND_MPU401_UART,
+ SND_OPL3_LIB, SND_OPL4_LIB, SND_VX_LIB, SND_AC97_CODEC.
+ Add the select command for each supported component.
+ </para>
+
+ <para>
+ Note that some selections imply the lowlevel selections.
+ For example, PCM includes TIMER, MPU401_UART includes RAWMIDI,
+ AC97_CODEC includes PCM, and OPL3_LIB includes HWDEP.
+ You don't need to give the lowlevel selections again.
+ </para>
+
+ <para>
+ For the details of Kconfig script, refer to the kbuild
+ documentation.
+ </para>
+
+ </listitem>
+
+ <listitem>
+ <para>
+ Run cvscompile script to re-generate the configure script and
+ build the whole stuff again.
+ </para>
+ </listitem>
+ </orderedlist>
+ </para>
+ </section>
+
+ <section>
+ <title>Drivers with Several Source Files</title>
+ <para>
+ Suppose that the driver snd-xyz have several source files.
+ They are located in the new subdirectory,
+ pci/xyz.
+
+ <orderedlist>
+ <listitem>
+ <para>
+ Add a new directory (<filename>xyz</filename>) in
+ <filename>alsa-driver/pci/Makefile</filename> like below
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ obj-$(CONFIG_SND) += xyz/
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+ </listitem>
+
+ <listitem>
+ <para>
+ Under the directory <filename>xyz</filename>, create a Makefile
+
+ <example>
+ <title>Sample Makefile for a driver xyz</title>
+ <programlisting>
+<![CDATA[
+ ifndef SND_TOPDIR
+ SND_TOPDIR=../..
+ endif
+
+ include $(SND_TOPDIR)/toplevel.config
+ include $(SND_TOPDIR)/Makefile.conf
+
+ snd-xyz-objs := xyz.o abc.o def.o
+
+ obj-$(CONFIG_SND_XYZ) += snd-xyz.o
+
+ include $(SND_TOPDIR)/Rules.make
+]]>
+ </programlisting>
+ </example>
+ </para>
+ </listitem>
+
+ <listitem>
+ <para>
+ Create the Kconfig entry
+ </para>
+
+ <para>
+ This procedure is as same as in the last section.
+ </para>
+ </listitem>
+
+ <listitem>
+ <para>
+ Run cvscompile script to re-generate the configure script and
+ build the whole stuff again.
+ </para>
+ </listitem>
+ </orderedlist>
+ </para>
+ </section>
+
+ </chapter>
+
+<!-- ****************************************************** -->
+<!-- Useful Functions -->
+<!-- ****************************************************** -->
+ <chapter id="useful-functions">
+ <title>Useful Functions</title>
+
+ <section id="useful-functions-snd-printk">
+ <title><function>snd_printk()</function> and friends</title>
+ <para>
+ ALSA provides a verbose version of
+ <function>printk()</function> function. If a kernel config
+ <constant>CONFIG_SND_VERBOSE_PRINTK</constant> is set, this
+ function prints the given message together with the file name
+ and the line of the caller. The <constant>KERN_XXX</constant>
+ prefix is processed as
+ well as the original <function>printk()</function> does, so it's
+ recommended to add this prefix, e.g.
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_printk(KERN_ERR "Oh my, sorry, it's extremely bad!\n");
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ There are also <function>printk()</function>'s for
+ debugging. <function>snd_printd()</function> can be used for
+ general debugging purposes. If
+ <constant>CONFIG_SND_DEBUG</constant> is set, this function is
+ compiled, and works just like
+ <function>snd_printk()</function>. If the ALSA is compiled
+ without the debugging flag, it's ignored.
+ </para>
+
+ <para>
+ <function>snd_printdd()</function> is compiled in only when
+ <constant>CONFIG_SND_DEBUG_DETECT</constant> is set. Please note
+ that <constant>DEBUG_DETECT</constant> is not set as default
+ even if you configure the alsa-driver with
+ <option>--with-debug=full</option> option. You need to give
+ explicitly <option>--with-debug=detect</option> option instead.
+ </para>
+ </section>
+
+ <section id="useful-functions-snd-assert">
+ <title><function>snd_assert()</function></title>
+ <para>
+ <function>snd_assert()</function> macro is similar with the
+ normal <function>assert()</function> macro. For example,
+
+ <informalexample>
+ <programlisting>
+<![CDATA[
+ snd_assert(pointer != NULL, return -EINVAL);
+]]>
+ </programlisting>
+ </informalexample>
+ </para>
+
+ <para>
+ The first argument is the expression to evaluate, and the
+ second argument is the action if it fails. When
+ <constant>CONFIG_SND_DEBUG</constant>, is set, it will show an
+ error message such as <computeroutput>BUG? (xxx) (called from
+ yyy)</computeroutput>. When no debug flag is set, this is
+ ignored.
+ </para>
+ </section>
+
+ <section id="useful-functions-snd-runtime-check">
+ <title><function>snd_runtime_check()</function></title>
+ <para>
+ This macro is quite similar with
+ <function>snd_assert()</function>. Unlike
+ <function>snd_assert()</function>, the expression is always
+ evaluated regardless of
+ <constant>CONFIG_SND_DEBUG</constant>. When
+ <constant>CONFIG_SND_DEBUG</constant> is set, the macro will
+ show a message like <computeroutput>ERROR (xx) (called from
+ yyy)</computeroutput>.
+ </para>
+ </section>
+
+ <section id="useful-functions-snd-bug">
+ <title><function>snd_BUG()</function></title>
+ <para>
+ It calls <function>snd_assert(0,)</function> -- that is, just
+ prints the error message at the point. It's useful to show that
+ a fatal error happens there.
+ </para>
+ </section>
+ </chapter>
+
+
+<!-- ****************************************************** -->
+<!-- Acknowledgments -->
+<!-- ****************************************************** -->
+ <chapter id="acknowledments">
+ <title>Acknowledgments</title>
+ <para>
+ I would like to thank Phil Kerr for his help for improvement and
+ corrections of this document.
+ </para>
+ <para>
+ Kevin Conder reformatted the original plain-text to the
+ DocBook format.
+ </para>
+ <para>
+ Giuliano Pochini corrected typos and contributed the example codes
+ in the hardware constraints section.
+ </para>
+ </chapter>
+
+
+</book>
diff --git a/Documentation/sound/alsa/Joystick.txt b/Documentation/sound/alsa/Joystick.txt
new file mode 100644
index 000000000000..ccda41b10f8a
--- /dev/null
+++ b/Documentation/sound/alsa/Joystick.txt
@@ -0,0 +1,86 @@
+Analog Joystick Support on ALSA Drivers
+=======================================
+ Oct. 14, 2003
+ Takashi Iwai <tiwai@suse.de>
+
+General
+-------
+
+First of all, you need to enable GAMEPORT support on Linux kernel for
+using a joystick with the ALSA driver. For the details of gameport
+support, refer to Documentation/input/joystick.txt.
+
+The joystick support of ALSA drivers is different between ISA and PCI
+cards. In the case of ISA (PnP) cards, it's usually handled by the
+independent module (ns558). Meanwhile, the ALSA PCI drivers have the
+built-in gameport support. Hence, when the ALSA PCI driver is built
+in the kernel, CONFIG_GAMEPORT must be 'y', too. Otherwise, the
+gameport support on that card will be (silently) disabled.
+
+Some adapter modules probe the physical connection of the device at
+the load time. It'd be safer to plug in the joystick device before
+loading the module.
+
+
+PCI Cards
+---------
+
+For PCI cards, the joystick is enabled when the appropriate module
+option is specified. Some drivers don't need options, and the
+joystick support is always enabled. In the former ALSA version, there
+was a dynamic control API for the joystick activation. It was
+changed, however, to the static module options because of the system
+stability and the resource management.
+
+The following PCI drivers support the joystick natively.
+
+ Driver Module Option Available Values
+ ---------------------------------------------------------------------------
+ als4000 joystick_port 0 = disable (default), 1 = auto-detect,
+ manual: any address (e.g. 0x200)
+ au88x0 N/A N/A
+ azf3328 joystick 0 = disable, 1 = enable, -1 = auto (default)
+ ens1370 joystick 0 = disable (default), 1 = enable
+ ens1371 joystick_port 0 = disable (default), 1 = auto-detect,
+ manual: 0x200, 0x208, 0x210, 0x218
+ cmipci joystick_port 0 = disable (default), 1 = auto-detect,
+ manual: any address (e.g. 0x200)
+ cs4281 N/A N/A
+ cs46xx N/A N/A
+ es1938 N/A N/A
+ es1968 joystick 0 = disable (default), 1 = enable
+ sonicvibes N/A N/A
+ trident N/A N/A
+ via82xx(*1) joystick 0 = disable (default), 1 = enable
+ ymfpci joystick_port 0 = disable (default), 1 = auto-detect,
+ manual: 0x201, 0x202, 0x204, 0x205(*2)
+ ---------------------------------------------------------------------------
+
+ *1) VIA686A/B only
+ *2) With YMF744/754 chips, the port address can be chosen arbitrarily
+
+The following drivers don't support gameport natively, but there are
+additional modules. Load the corresponding module to add the gameport
+support.
+
+ Driver Additional Module
+ -----------------------------
+ emu10k1 emu10k1-gp
+ fm801 fm801-gp
+ -----------------------------
+
+Note: the "pcigame" and "cs461x" modules are for the OSS drivers only.
+ These ALSA drivers (cs46xx, trident and au88x0) have the
+ built-in gameport support.
+
+As mentioned above, ALSA PCI drivers have the built-in gameport
+support, so you don't have to load ns558 module. Just load "joydev"
+and the appropriate adapter module (e.g. "analog").
+
+
+ISA Cards
+---------
+
+ALSA ISA drivers don't have the built-in gameport support.
+Instead, you need to load "ns558" module in addition to "joydev" and
+the adapter module (e.g. "analog").
diff --git a/Documentation/sound/alsa/MIXART.txt b/Documentation/sound/alsa/MIXART.txt
new file mode 100644
index 000000000000..5cb970612870
--- /dev/null
+++ b/Documentation/sound/alsa/MIXART.txt
@@ -0,0 +1,100 @@
+ Alsa driver for Digigram miXart8 and miXart8AES/EBU soundcards
+ Digigram <alsa@digigram.com>
+
+
+GENERAL
+=======
+
+The miXart8 is a multichannel audio processing and mixing soundcard
+that has 4 stereo audio inputs and 4 stereo audio outputs.
+The miXart8AES/EBU is the same with a add-on card that offers further
+4 digital stereo audio inputs and outputs.
+Furthermore the add-on card offers external clock synchronisation
+(AES/EBU, Word Clock, Time Code and Video Synchro)
+
+The mainboard has a PowerPC that offers onboard mpeg encoding and
+decoding, samplerate conversions and various effects.
+
+The driver don't work properly at all until the certain firmwares
+are loaded, i.e. no PCM nor mixer devices will appear.
+Use the mixartloader that can be found in the alsa-tools package.
+
+
+VERSION 0.1.0
+=============
+
+One miXart8 board will be represented as 4 alsa cards, each with 1
+stereo analog capture 'pcm0c' and 1 stereo analog playback 'pcm0p' device.
+With a miXart8AES/EBU there is in addition 1 stereo digital input
+'pcm1c' and 1 stereo digital output 'pcm1p' per card.
+
+Formats
+-------
+U8, S16_LE, S16_BE, S24_3LE, S24_3BE, FLOAT_LE, FLOAT_BE
+Sample rates : 8000 - 48000 Hz continously
+
+Playback
+--------
+For instance the playback devices are configured to have max. 4
+substreams performing hardware mixing. This could be changed to a
+maximum of 24 substreams if wished.
+Mono files will be played on the left and right channel. Each channel
+can be muted for each stream to use 8 analog/digital outputs seperately.
+
+Capture
+-------
+There is one substream per capture device. For instance only stereo
+formats are supported.
+
+Mixer
+-----
+<Master> and <Master Capture> : analog volume control of playback and capture PCM.
+<PCM 0-3> and <PCM Capture> : digital volume control of each analog substream.
+<AES 0-3> and <AES Capture> : digital volume control of each AES/EBU substream.
+<Monitoring> : Loopback from 'pcm0c' to 'pcm0p' with digital volume
+and mute control.
+
+Rem : for best audio quality try to keep a 0 attenuation on the PCM
+and AES volume controls which is set by 219 in the range from 0 to 255
+(about 86% with alsamixer)
+
+
+NOT YET IMPLEMENTED
+===================
+
+- external clock support (AES/EBU, Word Clock, Time Code, Video Sync)
+- MPEG audio formats
+- mono record
+- on-board effects and samplerate conversions
+- linked streams
+
+
+FIRMWARE
+========
+
+[As of 2.6.11, the firmware can be loaded automatically with hotplug
+ when CONFIG_FW_LOADER is set. The mixartloader is necessary only
+ for older versions or when you build the driver into kernel.]
+
+For loading the firmware automatically after the module is loaded, use
+the post-install command. For example, add the following entry to
+/etc/modprobe.conf for miXart driver:
+
+ install snd-mixart /sbin/modprobe --first-time -i snd-mixart && \
+ /usr/bin/mixartloader
+(for 2.2/2.4 kernels, add "post-install snd-mixart /usr/bin/vxloader" to
+ /etc/modules.conf, instead.)
+
+The firmware binaries are installed on /usr/share/alsa/firmware
+(or /usr/local/share/alsa/firmware, depending to the prefix option of
+configure). There will be a miXart.conf file, which define the dsp image
+files.
+
+The firmware files are copyright by Digigram SA
+
+
+COPYRIGHT
+=========
+
+Copyright (c) 2003 Digigram SA <alsa@digigram.com>
+Distributalbe under GPL.
diff --git a/Documentation/sound/alsa/OSS-Emulation.txt b/Documentation/sound/alsa/OSS-Emulation.txt
new file mode 100644
index 000000000000..ec2a02541d5b
--- /dev/null
+++ b/Documentation/sound/alsa/OSS-Emulation.txt
@@ -0,0 +1,297 @@
+ NOTES ON KERNEL OSS-EMULATION
+ =============================
+
+ Jan. 22, 2004 Takashi Iwai <tiwai@suse.de>
+
+
+Modules
+=======
+
+ALSA provides a powerful OSS emulation on the kernel.
+The OSS emulation for PCM, mixer and sequencer devices is implemented
+as add-on kernel modules, snd-pcm-oss, snd-mixer-oss and snd-seq-oss.
+When you need to access the OSS PCM, mixer or sequencer devices, the
+corresponding module has to be loaded.
+
+These modules are loaded automatically when the corresponding service
+is called. The alias is defined sound-service-x-y, where x and y are
+the card number and the minor unit number. Usually you don't have to
+define these aliases by yourself.
+
+Only necessary step for auto-loading of OSS modules is to define the
+card alias in /etc/modprobe.conf, such as
+
+ alias sound-slot-0 snd-emu10k1
+
+As the second card, define sound-slot-1 as well.
+Note that you can't use the aliased name as the target name (i.e.
+"alias sound-slot-0 snd-card-0" doesn't work any more like the old
+modutils).
+
+The currently available OSS configuration is shown in
+/proc/asound/oss/sndstat. This shows in the same syntax of
+/dev/sndstat, which is available on the commercial OSS driver.
+On ALSA, you can symlink /dev/sndstat to this proc file.
+
+Please note that the devices listed in this proc file appear only
+after the corresponding OSS-emulation module is loaded. Don't worry
+even if "NOT ENABLED IN CONFIG" is shown in it.
+
+
+Device Mapping
+==============
+
+ALSA supports the following OSS device files:
+
+ PCM:
+ /dev/dspX
+ /dev/adspX
+
+ Mixer:
+ /dev/mixerX
+
+ MIDI:
+ /dev/midi0X
+ /dev/amidi0X
+
+ Sequencer:
+ /dev/sequencer
+ /dev/sequencer2 (aka /dev/music)
+
+where X is the card number from 0 to 7.
+
+(NOTE: Some distributions have the device files like /dev/midi0 and
+ /dev/midi1. They are NOT for OSS but for tclmidi, which is
+ a totally different thing.)
+
+Unlike the real OSS, ALSA cannot use the device files more than the
+assigned ones. For example, the first card cannot use /dev/dsp1 or
+/dev/dsp2, but only /dev/dsp0 and /dev/adsp0.
+
+As seen above, PCM and MIDI may have two devices. Usually, the first
+PCM device (hw:0,0 in ALSA) is mapped to /dev/dsp and the secondary
+device (hw:0,1) to /dev/adsp (if available). For MIDI, /dev/midi and
+/dev/amidi, respectively.
+
+You can change this device mapping via the module options of
+snd-pcm-oss and snd-rawmidi. In the case of PCM, the following
+options are available for snd-pcm-oss:
+
+ dsp_map PCM device number assigned to /dev/dspX
+ (default = 0)
+ adsp_map PCM device number assigned to /dev/adspX
+ (default = 1)
+
+For example, to map the third PCM device (hw:0,2) to /dev/adsp0,
+define like this:
+
+ options snd-pcm-oss adsp_map=2
+
+The options take arrays. For configuring the second card, specify
+two entries separated by comma. For example, to map the third PCM
+device on the second card to /dev/adsp1, define like below:
+
+ options snd-pcm-oss adsp_map=0,2
+
+To change the mapping of MIDI devices, the following options are
+available for snd-rawmidi:
+
+ midi_map MIDI device number assigned to /dev/midi0X
+ (default = 0)
+ amidi_map MIDI device number assigned to /dev/amidi0X
+ (default = 1)
+
+For example, to assign the third MIDI device on the first card to
+/dev/midi00, define as follows:
+
+ options snd-rawmidi midi_map=2
+
+
+PCM Mode
+========
+
+As default, ALSA emulates the OSS PCM with so-called plugin layer,
+i.e. tries to convert the sample format, rate or channels
+automatically when the card doesn't support it natively.
+This will lead to some problems for some applications like quake or
+wine, especially if they use the card only in the MMAP mode.
+
+In such a case, you can change the behavior of PCM per application by
+writing a command to the proc file. There is a proc file for each PCM
+stream, /proc/asound/cardX/pcmY[cp]/oss, where X is the card number
+(zero-based), Y the PCM device number (zero-based), and 'p' is for
+playback and 'c' for capture, respectively. Note that this proc file
+exists only after snd-pcm-oss module is loaded.
+
+The command sequence has the following syntax:
+
+ app_name fragments fragment_size [options]
+
+app_name is the name of application with (higher priority) or without
+path.
+fragments specifies the number of fragments or zero if no specific
+number is given.
+fragment_size is the size of fragment in bytes or zero if not given.
+options is the optional parameters. The following options are
+available:
+
+ disable the application tries to open a pcm device for
+ this channel but does not want to use it.
+ direct don't use plugins
+ block force block open mode
+ non-block force non-block open mode
+ partial-frag write also partial fragments (affects playback only)
+ no-silence do not fill silence ahead to avoid clicks
+
+The disable option is useful when one stream direction (playback or
+capture) is not handled correctly by the application although the
+hardware itself does support both directions.
+The direct option is used, as mentioned above, to bypass the automatic
+conversion and useful for MMAP-applications.
+For example, to playback the first PCM device without plugins for
+quake, send a command via echo like the following:
+
+ % echo "quake 0 0 direct" > /proc/asound/card0/pcm0p/oss
+
+While quake wants only playback, you may append the second command
+to notify driver that only this direction is about to be allocated:
+
+ % echo "quake 0 0 disable" > /proc/asound/card0/pcm0c/oss
+
+The permission of proc files depend on the module options of snd.
+As default it's set as root, so you'll likely need to be superuser for
+sending the command above.
+
+The block and non-block options are used to change the behavior of
+opening the device file.
+
+As default, ALSA behaves as original OSS drivers, i.e. does not block
+the file when it's busy. The -EBUSY error is returned in this case.
+
+This blocking behavior can be changed globally via nonblock_open
+module option of snd-pcm-oss. For using the blocking mode as default
+for OSS devices, define like the following:
+
+ options snd-pcm-oss nonblock_open=0
+
+The partial-frag and no-silence commands have been added recently.
+Both commands are for optimization use only. The former command
+specifies to invoke the write transfer only when the whole fragment is
+filled. The latter stops writing the silence data ahead
+automatically. Both are disabled as default.
+
+You can check the currently defined configuration by reading the proc
+file. The read image can be sent to the proc file again, hence you
+can save the current configuration
+
+ % cat /proc/asound/card0/pcm0p/oss > /somewhere/oss-cfg
+
+and restore it like
+
+ % cat /somewhere/oss-cfg > /proc/asound/card0/pcm0p/oss
+
+Also, for clearing all the current configuration, send "erase" command
+as below:
+
+ % echo "erase" > /proc/asound/card0/pcm0p/oss
+
+
+Mixer Elements
+==============
+
+Since ALSA has completely different mixer interface, the emulation of
+OSS mixer is relatively complicated. ALSA builds up a mixer element
+from several different ALSA (mixer) controls based on the name
+string. For example, the volume element SOUND_MIXER_PCM is composed
+from "PCM Playback Volume" and "PCM Playback Switch" controls for the
+playback direction and from "PCM Capture Volume" and "PCM Capture
+Switch" for the capture directory (if exists). When the PCM volume of
+OSS is changed, all the volume and switch controls above are adjusted
+automatically.
+
+As default, ALSA uses the following control for OSS volumes:
+
+ OSS volume ALSA control Index
+ -----------------------------------------------------
+ SOUND_MIXER_VOLUME Master 0
+ SOUND_MIXER_BASS Tone Control - Bass 0
+ SOUND_MIXER_TREBLE Tone Control - Treble 0
+ SOUND_MIXER_SYNTH Synth 0
+ SOUND_MIXER_PCM PCM 0
+ SOUND_MIXER_SPEAKER PC Speaker 0
+ SOUND_MIXER_LINE Line 0
+ SOUND_MIXER_MIC Mic 0
+ SOUND_MIXER_CD CD 0
+ SOUND_MIXER_IMIX Monitor Mix 0
+ SOUND_MIXER_ALTPCM PCM 1
+ SOUND_MIXER_RECLEV (not assigned)
+ SOUND_MIXER_IGAIN Capture 0
+ SOUND_MIXER_OGAIN Playback 0
+ SOUND_MIXER_LINE1 Aux 0
+ SOUND_MIXER_LINE2 Aux 1
+ SOUND_MIXER_LINE3 Aux 2
+ SOUND_MIXER_DIGITAL1 Digital 0
+ SOUND_MIXER_DIGITAL2 Digital 1
+ SOUND_MIXER_DIGITAL3 Digital 2
+ SOUND_MIXER_PHONEIN Phone 0
+ SOUND_MIXER_PHONEOUT Phone 1
+ SOUND_MIXER_VIDEO Video 0
+ SOUND_MIXER_RADIO Radio 0
+ SOUND_MIXER_MONITOR Monitor 0
+
+The second column is the base-string of the corresponding ALSA
+control. In fact, the controls with "XXX [Playback|Capture]
+[Volume|Switch]" will be checked in addition.
+
+The current assignment of these mixer elements is listed in the proc
+file, /proc/asound/cardX/oss_mixer, which will be like the following
+
+ VOLUME "Master" 0
+ BASS "" 0
+ TREBLE "" 0
+ SYNTH "" 0
+ PCM "PCM" 0
+ ...
+
+where the first column is the OSS volume element, the second column
+the base-string of the corresponding ALSA control, and the third the
+control index. When the string is empty, it means that the
+corresponding OSS control is not available.
+
+For changing the assignment, you can write the configuration to this
+proc file. For example, to map "Wave Playback" to the PCM volume,
+send the command like the following:
+
+ % echo 'VOLUME "Wave Playback" 0' > /proc/asound/card0/oss_mixer
+
+The command is exactly as same as listed in the proc file. You can
+change one or more elements, one volume per line. In the last
+example, both "Wave Playback Volume" and "Wave Playback Switch" will
+be affected when PCM volume is changed.
+
+Like the case of PCM proc file, the permission of proc files depend on
+the module options of snd. you'll likely need to be superuser for
+sending the command above.
+
+As well as in the case of PCM proc file, you can save and restore the
+current mixer configuration by reading and writing the whole file
+image.
+
+
+Unsupported Features
+====================
+
+MMAP on ICE1712 driver
+----------------------
+ICE1712 supports only the unconventional format, interleaved
+10-channels 24bit (packed in 32bit) format. Therefore you cannot mmap
+the buffer as the conventional (mono or 2-channels, 8 or 16bit) format
+on OSS.
+
+USB devices
+-----------
+Some USB devices support only 24bit format packed in 3bytes. This
+format is not supported by OSS and no conversion is provided by kernel
+OSS emulation. You can use the user-space OSS emulation via libaoss
+instead.
+
diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt
new file mode 100644
index 000000000000..25c5d648aef6
--- /dev/null
+++ b/Documentation/sound/alsa/Procfile.txt
@@ -0,0 +1,191 @@
+ Proc Files of ALSA Drivers
+ ==========================
+ Takashi Iwai <tiwai@suse.de>
+
+General
+-------
+
+ALSA has its own proc tree, /proc/asound. Many useful information are
+found in this tree. When you encounter a problem and need debugging,
+check the files listed in the following sections.
+
+Each card has its subtree cardX, where X is from 0 to 7. The
+card-specific files are stored in the card* subdirectories.
+
+
+Global Information
+------------------
+
+cards
+ Shows the list of currently configured ALSA drivers,
+ index, the id string, short and long descriptions.
+
+version
+ Shows the version string and compile date.
+
+modules
+ Lists the module of each card
+
+devices
+ Lists the ALSA native device mappings.
+
+meminfo
+ Shows the status of allocated pages via ALSA drivers.
+ Appears only when CONFIG_SND_DEBUG=y.
+
+hwdep
+ Lists the currently available hwdep devices in format of
+ <card>-<device>: <name>
+
+pcm
+ Lists the currently available PCM devices in format of
+ <card>-<device>: <id>: <name> : <sub-streams>
+
+timer
+ Lists the currently available timer devices
+
+
+oss/devices
+ Lists the OSS device mappings.
+
+oss/sndstat
+ Provides the output compatible with /dev/sndstat.
+ You can symlink this to /dev/sndstat.
+
+
+Card Specific Files
+-------------------
+
+The card-specific files are found in /proc/asound/card* directories.
+Some drivers (e.g. cmipci) have their own proc entries for the
+register dump, etc (e.g. /proc/asound/card*/cmipci shows the register
+dump). These files would be really helpful for debugging.
+
+When PCM devices are available on this card, you can see directories
+like pcm0p or pcm1c. They hold the PCM information for each PCM
+stream. The number after 'pcm' is the PCM device number from 0, and
+the last 'p' or 'c' means playback or capture direction. The files in
+this subtree is described later.
+
+The status of MIDI I/O is found in midi* files. It shows the device
+name and the received/transmitted bytes through the MIDI device.
+
+When the card is equipped with AC97 codecs, there are codec97#*
+subdirectories (desribed later).
+
+When the OSS mixer emulation is enabled (and the module is loaded),
+oss_mixer file appears here, too. This shows the current mapping of
+OSS mixer elements to the ALSA control elements. You can change the
+mapping by writing to this device. Read OSS-Emulation.txt for
+details.
+
+
+PCM Proc Files
+--------------
+
+card*/pcm*/info
+ The general information of this PCM device: card #, device #,
+ substreams, etc.
+
+card*/pcm*/xrun_debug
+ This file appears when CONFIG_SND_DEBUG=y.
+ This shows the status of xrun (= buffer overrun/xrun) debug of
+ ALSA PCM middle layer, as an integer from 0 to 2. The value
+ can be changed by writing to this file, such as
+
+ # cat 2 > /proc/asound/card0/pcm0p/xrun_debug
+
+ When this value is greater than 0, the driver will show the
+ messages to kernel log when an xrun is detected. The debug
+ message is shown also when the invalid H/W pointer is detected
+ at the update of periods (usually called from the interrupt
+ handler).
+
+ When this value is greater than 1, the driver will show the
+ stack trace additionally. This may help the debugging.
+
+card*/pcm*/sub*/info
+ The general information of this PCM sub-stream.
+
+card*/pcm*/sub*/status
+ The current status of this PCM sub-stream, elapsed time,
+ H/W position, etc.
+
+card*/pcm*/sub*/hw_params
+ The hardware parameters set for this sub-stream.
+
+card*/pcm*/sub*/sw_params
+ The soft parameters set for this sub-stream.
+
+card*/pcm*/sub*/prealloc
+ The buffer pre-allocation information.
+
+
+AC97 Codec Information
+----------------------
+
+card*/codec97#*/ac97#?-?
+ Shows the general information of this AC97 codec chip, such as
+ name, capabilities, set up.
+
+card*/codec97#0/ac97#?-?+regs
+ Shows the AC97 register dump. Useful for debugging.
+
+ When CONFIG_SND_DEBUG is enabled, you can write to this file for
+ changing an AC97 register directly. Pass two hex numbers.
+ For example,
+
+ # echo 02 9f1f > /proc/asound/card0/codec97#0/ac97#0-0+regs
+
+
+Sequencer Information
+---------------------
+
+seq/drivers
+ Lists the currently available ALSA sequencer drivers.
+
+seq/clients
+ Shows the list of currently available sequencer clinets and
+ ports. The connection status and the running status are shown
+ in this file, too.
+
+seq/queues
+ Lists the currently allocated/running sequener queues.
+
+seq/timer
+ Lists the currently allocated/running sequencer timers.
+
+seq/oss
+ Lists the OSS-compatible sequencer stuffs.
+
+
+Help For Debugging?
+-------------------
+
+When the problem is related with PCM, first try to turn on xrun_debug
+mode. This will give you the kernel messages when and where xrun
+happened.
+
+If it's really a bug, report it with the following information
+
+ - the name of the driver/card, show in /proc/asound/cards
+ - the reigster dump, if available (e.g. card*/cmipci)
+
+when it's a PCM problem,
+
+ - set-up of PCM, shown in hw_parms, sw_params, and status in the PCM
+ sub-stream directory
+
+when it's a mixer problem,
+
+ - AC97 proc files, codec97#*/* files
+
+for USB audio/midi,
+
+ - output of lsusb -v
+ - stream* files in card directory
+
+
+The ALSA bug-tracking system is found at:
+
+ https://bugtrack.alsa-project.org/alsa-bug/
diff --git a/Documentation/sound/alsa/SB-Live-mixer.txt b/Documentation/sound/alsa/SB-Live-mixer.txt
new file mode 100644
index 000000000000..651adaf60473
--- /dev/null
+++ b/Documentation/sound/alsa/SB-Live-mixer.txt
@@ -0,0 +1,356 @@
+
+ Sound Blaster Live mixer / default DSP code
+ ===========================================
+
+
+The EMU10K1 chips have a DSP part which can be programmed to support
+various ways of sample processing, which is described here.
+(This acticle does not deal with the overall functionality of the
+EMU10K1 chips. See the manuals section for further details.)
+
+The ALSA driver programs this portion of chip by default code
+(can be altered later) which offers the following functionality:
+
+
+1) IEC958 (S/PDIF) raw PCM
+--------------------------
+
+This PCM device (it's the 4th PCM device (index 3!) and first subdevice
+(index 0) for a given card) allows to forward 48kHz, stereo, 16-bit
+little endian streams without any modifications to the digital output
+(coaxial or optical). The universal interface allows the creation of up
+to 8 raw PCM devices operating at 48kHz, 16-bit little endian. It would
+be easy to add support for multichannel devices to the current code,
+but the conversion routines exist only for stereo (2-channel streams)
+at the time.
+
+Look to tram_poke routines in lowlevel/emu10k1/emufx.c for more details.
+
+
+2) Digital mixer controls
+-------------------------
+
+These controls are built using the DSP instructions. They offer extended
+functionality. Only the default build-in code in the ALSA driver is described
+here. Note that the controls work as attenuators: the maximum value is the
+neutral position leaving the signal unchanged. Note that if the same destination
+is mentioned in multiple controls, the signal is accumulated and can be wrapped
+(set to maximal or minimal value without checking of overflow).
+
+
+Explanation of used abbreviations:
+
+DAC - digital to analog converter
+ADC - analog to digital converter
+I2S - one-way three wire serial bus for digital sound by Philips Semiconductors
+ (this standard is used for connecting standalone DAC and ADC converters)
+LFE - low frequency effects (subwoofer signal)
+AC97 - a chip containing an analog mixer, DAC and ADC converters
+IEC958 - S/PDIF
+FX-bus - the EMU10K1 chip has an effect bus containing 16 accumulators.
+ Each of the synthesizer voices can feed its output to these accumulators
+ and the DSP microcontroller can operate with the resulting sum.
+
+
+name='Wave Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Wave Surround Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result samples are forwarded to the rear I2S DACs. These DACs operates
+separately (they are not inside the AC97 codec).
+
+name='Wave Center Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM samples.
+The result is mixed to mono signal (single channel) and forwarded to
+the ??rear?? right DAC PCM slot of the AC97 codec.
+
+name='Wave LFE Playback Volume',index=0
+
+This control is used to attenuate samples for left and right PCM FX-bus
+accumulators. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is mixed to mono signal (single channel) and forwarded to
+the ??rear?? left DAC PCM slot of the AC97 codec.
+
+name='Wave Capture Volume',index=0
+name='Wave Capture Switch',index=0
+
+These controls are used to attenuate samples for left and right PCM FX-bus
+accumulator. ALSA uses accumulators 0 and 1 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Music Playback Volume',index=0
+
+This control is used to attenuate samples for left and right MIDI FX-bus
+accumulators. ALSA uses accumulators 4 and 5 for left and right MIDI samples.
+The result samples are forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Music Capture Volume',index=0
+name='Music Capture Switch',index=0
+
+These controls are used to attenuate samples for left and right MIDI FX-bus
+accumulator. ALSA uses accumulators 4 and 5 for left and right PCM.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Surround Playback Volume',index=0
+
+This control is used to attenuate samples for left and right rear PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
+The result samples are forwarded to the rear I2S DACs. These DACs operate
+separately (they are not inside the AC97 codec).
+
+name='Surround Capture Volume',index=0
+name='Surround Capture Switch',index=0
+
+These controls are used to attenuate samples for left and right rear PCM FX-bus
+accumulators. ALSA uses accumulators 2 and 3 for left and right rear PCM samples.
+The result is forwarded to the ADC capture FIFO (thus to the standard capture
+PCM device).
+
+name='Center Playback Volume',index=0
+
+This control is used to attenuate sample for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
+to the ??rear?? right DAC PCM slot of the AC97 codec.
+
+name='LFE Playback Volume',index=0
+
+This control is used to attenuate sample for center PCM FX-bus accumulator.
+ALSA uses accumulator 6 for center PCM sample. The result sample is forwarded
+to the ??rear?? left DAC PCM slot of the AC97 codec.
+
+name='AC97 Playback Volume',index=0
+
+This control is used to attenuate samples for left and right front ADC PCM slots
+of the AC97 codec. The result samples are forwarded to the front DAC PCM
+slots of the AC97 codec.
+********************************************************************************
+*** Note: This control should be zero for the standard operations, otherwise ***
+*** a digital loopback is activated. ***
+********************************************************************************
+
+name='AC97 Capture Volume',index=0
+
+This control is used to attenuate samples for left and right front ADC PCM slots
+of the AC97 codec. The result is forwarded to the ADC capture FIFO (thus to
+the standard capture PCM device).
+********************************************************************************
+*** Note: This control should be 100 (maximal value), otherwise no analog ***
+*** inputs of the AC97 codec can be captured (recorded). ***
+********************************************************************************
+
+name='IEC958 TTL Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='IEC958 TTL Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 TTL
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+name='Zoom Video Playback Volume',index=0
+
+This control is used to attenuate samples from left and right zoom video
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the front DAC PCM slots of the AC97 codec.
+
+name='Zoom Video Capture Volume',index=0
+
+This control is used to attenuate samples from left and right zoom video
+digital inputs (usually used by a CDROM drive). The result samples are
+forwarded to the ADC capture FIFO (thus to the standard capture PCM device).
+
+name='IEC958 LiveDrive Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 optical
+digital input. The result samples are forwarded to the front DAC PCM slots
+of the AC97 codec.
+
+name='IEC958 LiveDrive Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 optical
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+name='IEC958 Coaxial Playback Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 coaxial
+digital inputs. The result samples are forwarded to the front DAC PCM slots
+of the AC97 codec.
+
+name='IEC958 Coaxial Capture Volume',index=0
+
+This control is used to attenuate samples from left and right IEC958 coaxial
+digital inputs. The result samples are forwarded to the ADC capture FIFO
+(thus to the standard capture PCM device).
+
+name='Line LiveDrive Playback Volume',index=0
+name='Line LiveDrive Playback Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the LiveDrive). The result samples are forwarded to the front
+DAC PCM slots of the AC97 codec.
+
+name='Line LiveDrive Capture Volume',index=1
+name='Line LiveDrive Capture Volume',index=1
+
+This control is used to attenuate samples from left and right I2S ADC
+inputs (on the LiveDrive). The result samples are forwarded to the ADC
+capture FIFO (thus to the standard capture PCM device).
+
+name='Tone Control - Switch',index=0
+
+This control turns the tone control on or off. The samples for front, rear
+and center / LFE outputs are affected.
+
+name='Tone Control - Bass',index=0
+
+This control sets the bass intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='Tone Control - Treble',index=0
+
+This control sets the treble intensity. There is no neutral value!!
+When the tone control code is activated, the samples are always modified.
+The closest value to pure signal is 20.
+
+name='IEC958 Optical Raw Playback Switch',index=0
+
+If this switch is on, then the samples for the IEC958 (S/PDIF) digital
+output are taken only from the raw FX8010 PCM, otherwise standard front
+PCM samples are taken.
+
+name='Headphone Playback Volume',index=1
+
+This control attenuates the samples for the headphone output.
+
+name='Headphone Center Playback Switch',index=1
+
+If this switch is on, then the sample for the center PCM is put to the
+left headphone output (useful for SB Live cards without separate center/LFE
+output).
+
+name='Headphone LFE Playback Switch',index=1
+
+If this switch is on, then the sample for the center PCM is put to the
+right headphone output (useful for SB Live cards without separate center/LFE
+output).
+
+
+3) PCM stream related controls
+------------------------------
+
+name='EMU10K1 PCM Volume',index 0-31
+
+Channel volume attenuation in range 0-0xffff. The maximum value (no
+attenuation) is default. The channel mapping for three values is
+as follows:
+
+ 0 - mono, default 0xffff (no attenuation)
+ 1 - left, default 0xffff (no attenuation)
+ 2 - right, default 0xffff (no attenuation)
+
+name='EMU10K1 PCM Send Routing',index 0-31
+
+This control specifies the destination - FX-bus accumulators. There are
+twelve values with this mapping:
+
+ 0 - mono, A destination (FX-bus 0-15), default 0
+ 1 - mono, B destination (FX-bus 0-15), default 1
+ 2 - mono, C destination (FX-bus 0-15), default 2
+ 3 - mono, D destination (FX-bus 0-15), default 3
+ 4 - left, A destination (FX-bus 0-15), default 0
+ 5 - left, B destination (FX-bus 0-15), default 1
+ 6 - left, C destination (FX-bus 0-15), default 2
+ 7 - left, D destination (FX-bus 0-15), default 3
+ 8 - right, A destination (FX-bus 0-15), default 0
+ 9 - right, B destination (FX-bus 0-15), default 1
+ 10 - right, C destination (FX-bus 0-15), default 2
+ 11 - right, D destination (FX-bus 0-15), default 3
+
+Don't forget that it's illegal to assign a channel to the same FX-bus accumulator
+more than once (it means 0=0 && 1=0 is an invalid combination).
+
+name='EMU10K1 PCM Send Volume',index 0-31
+
+It specifies the attenuation (amount) for given destination in range 0-255.
+The channel mapping is following:
+
+ 0 - mono, A destination attn, default 255 (no attenuation)
+ 1 - mono, B destination attn, default 255 (no attenuation)
+ 2 - mono, C destination attn, default 0 (mute)
+ 3 - mono, D destination attn, default 0 (mute)
+ 4 - left, A destination attn, default 255 (no attenuation)
+ 5 - left, B destination attn, default 0 (mute)
+ 6 - left, C destination attn, default 0 (mute)
+ 7 - left, D destination attn, default 0 (mute)
+ 8 - right, A destination attn, default 0 (mute)
+ 9 - right, B destination attn, default 255 (no attenuation)
+ 10 - right, C destination attn, default 0 (mute)
+ 11 - right, D destination attn, default 0 (mute)
+
+
+
+4) MANUALS/PATENTS:
+-------------------
+
+ftp://opensource.creative.com/pub/doc
+-------------------------------------
+
+ Files:
+ LM4545.pdf AC97 Codec
+
+ m2049.pdf The EMU10K1 Digital Audio Processor
+
+ hog63.ps FX8010 - A DSP Chip Architecture for Audio Effects
+
+
+WIPO Patents
+------------
+ Patent numbers:
+ WO 9901813 (A1) Audio Effects Processor with multiple asynchronous (Jan. 14, 1999)
+ streams
+
+ WO 9901814 (A1) Processor with Instruction Set for Audio Effects (Jan. 14, 1999)
+
+ WO 9901953 (A1) Audio Effects Processor having Decoupled Instruction
+ Execution and Audio Data Sequencing (Jan. 14, 1999)
+
+
+US Patents (http://www.uspto.gov/)
+----------------------------------
+
+ US 5925841 Digital Sampling Instrument employing cache memory (Jul. 20, 1999)
+
+ US 5928342 Audio Effects Processor integrated on a single chip (Jul. 27, 1999)
+ with a multiport memory onto which multiple asynchronous
+ digital sound samples can be concurrently loaded
+
+ US 5930158 Processor with Instruction Set for Audio Effects (Jul. 27, 1999)
+
+ US 6032235 Memory initialization circuit (Tram) (Feb. 29, 2000)
+
+ US 6138207 Interpolation looping of audio samples in cache connected to (Oct. 24, 2000)
+ system bus with prioritization and modification of bus transfers
+ in accordance with loop ends and minimum block sizes
+
+ US 6151670 Method for conserving memory storage using a (Nov. 21, 2000)
+ pool of short term memory registers
+
+ US 6195715 Interrupt control for multiple programs communicating with (Feb. 27, 2001)
+ a common interrupt by associating programs to GP registers,
+ defining interrupt register, polling GP registers, and invoking
+ callback routine associated with defined interrupt register
diff --git a/Documentation/sound/alsa/VIA82xx-mixer.txt b/Documentation/sound/alsa/VIA82xx-mixer.txt
new file mode 100644
index 000000000000..1b0ac06ba95d
--- /dev/null
+++ b/Documentation/sound/alsa/VIA82xx-mixer.txt
@@ -0,0 +1,8 @@
+
+ VIA82xx mixer
+ =============
+
+On many VIA82xx boards, the 'Input Source Select' mixer control does not work.
+Setting it to 'Input2' on such boards will cause recording to hang, or fail
+with EIO (input/output error) via OSS emulation. This control should be left
+at 'Input1' for such cards.
diff --git a/Documentation/sound/alsa/hda_codec.txt b/Documentation/sound/alsa/hda_codec.txt
new file mode 100644
index 000000000000..e9d07b8f1acb
--- /dev/null
+++ b/Documentation/sound/alsa/hda_codec.txt
@@ -0,0 +1,299 @@
+Notes on Universal Interface for Intel High Definition Audio Codec
+------------------------------------------------------------------
+
+Takashi Iwai <tiwai@suse.de>
+
+
+[Still a draft version]
+
+
+General
+=======
+
+The snd-hda-codec module supports the generic access function for the
+High Definition (HD) audio codecs. It's designed to be independent
+from the controller code like ac97 codec module. The real accessors
+from/to the controller must be implemented in the lowlevel driver.
+
+The structure of this module is similar with ac97_codec module.
+Each codec chip belongs to a bus class which communicates with the
+controller.
+
+
+Initialization of Bus Instance
+==============================
+
+The card driver has to create struct hda_bus at first. The template
+struct should be filled and passed to the constructor:
+
+struct hda_bus_template {
+ void *private_data;
+ struct pci_dev *pci;
+ const char *modelname;
+ struct hda_bus_ops ops;
+};
+
+The card driver can set and use the private_data field to retrieve its
+own data in callback functions. The pci field is used when the patch
+needs to check the PCI subsystem IDs, so on. For non-PCI system, it
+doesn't have to be set, of course.
+The modelname field specifies the board's specific configuration. The
+string is passed to the codec parser, and it depends on the parser how
+the string is used.
+These fields, private_data, pci and modelname are all optional.
+
+The ops field contains the callback functions as the following:
+
+struct hda_bus_ops {
+ int (*command)(struct hda_codec *codec, hda_nid_t nid, int direct,
+ unsigned int verb, unsigned int parm);
+ unsigned int (*get_response)(struct hda_codec *codec);
+ void (*private_free)(struct hda_bus *);
+};
+
+The command callback is called when the codec module needs to send a
+VERB to the controller. It's always a single command.
+The get_response callback is called when the codec requires the answer
+for the last command. These two callbacks are mandatory and have to
+be given.
+The last, private_free callback, is optional. It's called in the
+destructor to release any necessary data in the lowlevel driver.
+
+The bus instance is created via snd_hda_bus_new(). You need to pass
+the card instance, the template, and the pointer to store the
+resultant bus instance.
+
+int snd_hda_bus_new(snd_card_t *card, const struct hda_bus_template *temp,
+ struct hda_bus **busp);
+
+It returns zero if successful. A negative return value means any
+error during creation.
+
+
+Creation of Codec Instance
+==========================
+
+Each codec chip on the board is then created on the BUS instance.
+To create a codec instance, call snd_hda_codec_new().
+
+int snd_hda_codec_new(struct hda_bus *bus, unsigned int codec_addr,
+ struct hda_codec **codecp);
+
+The first argument is the BUS instance, the second argument is the
+address of the codec, and the last one is the pointer to store the
+resultant codec instance (can be NULL if not needed).
+
+The codec is stored in a linked list of bus instance. You can follow
+the codec list like:
+
+ struct list_head *p;
+ struct hda_codec *codec;
+ list_for_each(p, &bus->codec_list) {
+ codec = list_entry(p, struct hda_codec, list);
+ ...
+ }
+
+The codec isn't initialized at this stage properly. The
+initialization sequence is called when the controls are built later.
+
+
+Codec Access
+============
+
+To access codec, use snd_codec_read() and snd_codec_write().
+snd_hda_param_read() is for reading parameters.
+For writing a sequence of verbs, use snd_hda_sequence_write().
+
+To retrieve the number of sub nodes connected to the given node, use
+snd_hda_get_sub_nodes(). The connection list can be obtained via
+snd_hda_get_connections() call.
+
+When an unsolicited event happens, pass the event via
+snd_hda_queue_unsol_event() so that the codec routines will process it
+later.
+
+
+(Mixer) Controls
+================
+
+To create mixer controls of all codecs, call
+snd_hda_build_controls(). It then builds the mixers and does
+initialization stuff on each codec.
+
+
+PCM Stuff
+=========
+
+snd_hda_build_pcms() gives the necessary information to create PCM
+streams. When it's called, each codec belonging to the bus stores
+codec->num_pcms and codec->pcm_info fields. The num_pcms indicates
+the number of elements in pcm_info array. The card driver is supposed
+to traverse the codec linked list, read the pcm information in
+pcm_info array, and build pcm instances according to them.
+
+The pcm_info array contains the following record:
+
+/* PCM information for each substream */
+struct hda_pcm_stream {
+ unsigned int substreams; /* number of substreams, 0 = not exist */
+ unsigned int channels_min; /* min. number of channels */
+ unsigned int channels_max; /* max. number of channels */
+ hda_nid_t nid; /* default NID to query rates/formats/bps, or set up */
+ u32 rates; /* supported rates */
+ u64 formats; /* supported formats (SNDRV_PCM_FMTBIT_) */
+ unsigned int maxbps; /* supported max. bit per sample */
+ struct hda_pcm_ops ops;
+};
+
+/* for PCM creation */
+struct hda_pcm {
+ char *name;
+ struct hda_pcm_stream stream[2];
+};
+
+The name can be passed to snd_pcm_new(). The stream field contains
+the information for playback (SNDRV_PCM_STREAM_PLAYBACK = 0) and
+capture (SNDRV_PCM_STREAM_CAPTURE = 1) directions. The card driver
+should pass substreams to snd_pcm_new() for the number of substreams
+to create.
+
+The channels_min, channels_max, rates and formats should be copied to
+runtime->hw record. They and maxbps fields are used also to compute
+the format value for the HDA codec and controller. Call
+snd_hda_calc_stream_format() to get the format value.
+
+The ops field contains the following callback functions:
+
+struct hda_pcm_ops {
+ int (*open)(struct hda_pcm_stream *info, struct hda_codec *codec,
+ snd_pcm_substream_t *substream);
+ int (*close)(struct hda_pcm_stream *info, struct hda_codec *codec,
+ snd_pcm_substream_t *substream);
+ int (*prepare)(struct hda_pcm_stream *info, struct hda_codec *codec,
+ unsigned int stream_tag, unsigned int format,
+ snd_pcm_substream_t *substream);
+ int (*cleanup)(struct hda_pcm_stream *info, struct hda_codec *codec,
+ snd_pcm_substream_t *substream);
+};
+
+All are non-NULL, so you can call them safely without NULL check.
+
+The open callback should be called in PCM open after runtime->hw is
+set up. It may override some setting and constraints additionally.
+Similarly, the close callback should be called in the PCM close.
+
+The prepare callback should be called in PCM prepare. This will set
+up the codec chip properly for the operation. The cleanup should be
+called in hw_free to clean up the configuration.
+
+The caller should check the return value, at least for open and
+prepare callbacks. When a negative value is returned, some error
+occurred.
+
+
+Proc Files
+==========
+
+Each codec dumps the widget node information in
+/proc/asound/card*/codec#* file. This information would be really
+helpful for debugging. Please provide its contents together with the
+bug report.
+
+
+Power Management
+================
+
+It's simple:
+Call snd_hda_suspend() in the PM suspend callback.
+Call snd_hda_resume() in the PM resume callback.
+
+
+Codec Preset (Patch)
+====================
+
+To set up and handle the codec functionality fully, each codec may
+have a codec preset (patch). It's defined in struct hda_codec_preset:
+
+ struct hda_codec_preset {
+ unsigned int id;
+ unsigned int mask;
+ unsigned int subs;
+ unsigned int subs_mask;
+ unsigned int rev;
+ const char *name;
+ int (*patch)(struct hda_codec *codec);
+ };
+
+When the codec id and codec subsystem id match with the given id and
+subs fields bitwise (with bitmask mask and subs_mask), the callback
+patch is called. The patch callback should initialize the codec and
+set the codec->patch_ops field. This is defined as below:
+
+ struct hda_codec_ops {
+ int (*build_controls)(struct hda_codec *codec);
+ int (*build_pcms)(struct hda_codec *codec);
+ int (*init)(struct hda_codec *codec);
+ void (*free)(struct hda_codec *codec);
+ void (*unsol_event)(struct hda_codec *codec, unsigned int res);
+ #ifdef CONFIG_PM
+ int (*suspend)(struct hda_codec *codec, pm_message_t state);
+ int (*resume)(struct hda_codec *codec);
+ #endif
+ };
+
+The build_controls callback is called from snd_hda_build_controls().
+Similarly, the build_pcms callback is called from
+snd_hda_build_pcms(). The init callback is called after
+build_controls to initialize the hardware.
+The free callback is called as a destructor.
+
+The unsol_event callback is called when an unsolicited event is
+received.
+
+The suspend and resume callbacks are for power management.
+
+Each entry can be NULL if not necessary to be called.
+
+
+Generic Parser
+==============
+
+When the device doesn't match with any given presets, the widgets are
+parsed via th generic parser (hda_generic.c). Its support is
+limited: no multi-channel support, for example.
+
+
+Digital I/O
+===========
+
+Call snd_hda_create_spdif_out_ctls() from the patch to create controls
+related with SPDIF out. In the patch resume callback, call
+snd_hda_resume_spdif().
+
+
+Helper Functions
+================
+
+snd_hda_get_codec_name() stores the codec name on the given string.
+
+snd_hda_check_board_config() can be used to obtain the configuration
+information matching with the device. Define the table with struct
+hda_board_config entries (zero-terminated), and pass it to the
+function. The function checks the modelname given as a module
+parameter, and PCI subsystem IDs. If the matching entry is found, it
+returns the config field value.
+
+snd_hda_add_new_ctls() can be used to create and add control entries.
+Pass the zero-terminated array of snd_kcontrol_new_t. The same array
+can be passed to snd_hda_resume_ctls() for resume.
+Note that this will call control->put callback of these entries. So,
+put callback should check codec->in_resume and force to restore the
+given value if it's non-zero even if the value is identical with the
+cached value.
+
+Macros HDA_CODEC_VOLUME(), HDA_CODEC_MUTE() and their variables can be
+used for the entry of snd_kcontrol_new_t.
+
+The input MUX helper callbacks for such a control are provided, too:
+snd_hda_input_mux_info() and snd_hda_input_mux_put(). See
+patch_realtek.c for example.
diff --git a/Documentation/sound/alsa/seq_oss.html b/Documentation/sound/alsa/seq_oss.html
new file mode 100644
index 000000000000..d9776cf60c07
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+<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN">
+<HTML>
+<HEAD>
+ <TITLE>OSS Sequencer Emulation on ALSA</TITLE>
+</HEAD>
+<BODY>
+
+<CENTER>
+<H1>
+
+<HR WIDTH="100%"></H1></CENTER>
+
+<CENTER>
+<H1>
+OSS Sequencer Emulation on ALSA</H1></CENTER>
+
+<HR WIDTH="100%">
+<P>Copyright (c) 1998,1999 by Takashi Iwai
+<TT><A HREF="mailto:iwai@ww.uni-erlangen.de">&lt;iwai@ww.uni-erlangen.de></A></TT>
+<P>ver.0.1.8; Nov. 16, 1999
+<H2>
+
+<HR WIDTH="100%"></H2>
+
+<H2>
+1. Description</H2>
+This directory contains the OSS sequencer emulation driver on ALSA. Note
+that this program is still in the development state.
+<P>What this does - it provides the emulation of the OSS sequencer, access
+via
+<TT>/dev/sequencer</TT> and <TT>/dev/music</TT> devices.
+The most of applications using OSS can run if the appropriate ALSA
+sequencer is prepared.
+<P>The following features are emulated by this driver:
+<UL>
+<LI>
+Normal sequencer and MIDI events:</LI>
+
+<BR>They are converted to the ALSA sequencer events, and sent to the corresponding
+port.
+<LI>
+Timer events:</LI>
+
+<BR>The timer is not selectable by ioctl. The control rate is fixed to
+100 regardless of HZ. That is, even on Alpha system, a tick is always
+1/100 second. The base rate and tempo can be changed in <TT>/dev/music</TT>.
+
+<LI>
+Patch loading:</LI>
+
+<BR>It purely depends on the synth drivers whether it's supported since
+the patch loading is realized by callback to the synth driver.
+<LI>
+I/O controls:</LI>
+
+<BR>Most of controls are accepted. Some controls
+are dependent on the synth driver, as well as even on original OSS.</UL>
+Furthermore, you can find the following advanced features:
+<UL>
+<LI>
+Better queue mechanism:</LI>
+
+<BR>The events are queued before processing them.
+<LI>
+Multiple applications:</LI>
+
+<BR>You can run two or more applications simultaneously (even for OSS sequencer)!
+However, each MIDI device is exclusive - that is, if a MIDI device is opened
+once by some application, other applications can't use it. No such a restriction
+in synth devices.
+<LI>
+Real-time event processing:</LI>
+
+<BR>The events can be processed in real time without using out of bound
+ioctl. To switch to real-time mode, send ABSTIME 0 event. The followed
+events will be processed in real-time without queued. To switch off the
+real-time mode, send RELTIME 0 event.
+<LI>
+<TT>/proc</TT> interface:</LI>
+
+<BR>The status of applications and devices can be shown via <TT>/proc/asound/seq/oss</TT>
+at any time. In the later version, configuration will be changed via <TT>/proc</TT>
+interface, too.</UL>
+
+<H2>
+2. Installation</H2>
+Run configure script with both sequencer support (<TT>--with-sequencer=yes</TT>)
+and OSS emulation (<TT>--with-oss=yes</TT>) options. A module <TT>snd-seq-oss.o</TT>
+will be created. If the synth module of your sound card supports for OSS
+emulation (so far, only Emu8000 driver), this module will be loaded automatically.
+Otherwise, you need to load this module manually.
+<P>At beginning, this module probes all the MIDI ports which have been
+already connected to the sequencer. Once after that, the creation and deletion
+of ports are watched by announcement mechanism of ALSA sequencer.
+<P>The available synth and MIDI devices can be found in proc interface.
+Run "<TT>cat /proc/asound/seq/oss</TT>", and check the devices. For example,
+if you use an AWE64 card, you'll see like the following:
+<PRE>&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; OSS sequencer emulation version 0.1.8
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ALSA client number 63
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; ALSA receiver port 0
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Number of applications: 0
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Number of synth devices: 1
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; synth 0: [EMU8000]
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; type 0x1 : subtype 0x20 : voices 32
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capabilties : ioctl enabled / load_patch enabled
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; Number of MIDI devices: 3
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; midi 0: [Emu8000 Port-0] ALSA port 65:0
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capability write / opened none
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; midi 1: [Emu8000 Port-1] ALSA port 65:1
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capability write / opened none
+
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; midi 2: [0: MPU-401 (UART)] ALSA port 64:0
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; capability read/write / opened none</PRE>
+Note that the device number may be different from the information of
+<TT>/proc/asound/oss-devices</TT>
+or ones of the original OSS driver. Use the device number listed in <TT>/proc/asound/seq/oss</TT>
+to play via OSS sequencer emulation.
+<H2>
+3. Using Synthesizer Devices</H2>
+Run your favorite program. I've tested playmidi-2.4, awemidi-0.4.3, gmod-3.1
+and xmp-1.1.5. You can load samples via <TT>/dev/sequencer</TT> like sfxload,
+too.
+<P>If the lowlevel driver supports multiple access to synth devices (like
+Emu8000 driver), two or more applications are allowed to run at the same
+time.
+<H2>
+4. Using MIDI Devices</H2>
+So far, only MIDI output was tested. MIDI input was not checked at all,
+but hopefully it will work. Use the device number listed in <TT>/proc/asound/seq/oss</TT>.
+Be aware that these numbers are mostly different from the list in
+<TT>/proc/asound/oss-devices</TT>.
+<H2>
+5. Module Options</H2>
+The following module options are available:
+<UL>
+<LI>
+<TT>maxqlen</TT></LI>
+
+<BR>specifies the maximum read/write queue length. This queue is private
+for OSS sequencer, so that it is independent from the queue length of ALSA
+sequencer. Default value is 1024.
+<LI>
+<TT>seq_oss_debug</TT></LI>
+
+<BR>specifies the debug level and accepts zero (= no debug message) or
+positive integer. Default value is 0.</UL>
+
+<H2>
+6. Queue Mechanism</H2>
+OSS sequencer emulation uses an ALSA priority queue. The
+events from <TT>/dev/sequencer</TT> are processed and put onto the queue
+specified by module option.
+<P>All the events from <TT>/dev/sequencer</TT> are parsed at beginning.
+The timing events are also parsed at this moment, so that the events may
+be processed in real-time. Sending an event ABSTIME 0 switches the operation
+mode to real-time mode, and sending an event RELTIME 0 switches it off.
+In the real-time mode, all events are dispatched immediately.
+<P>The queued events are dispatched to the corresponding ALSA sequencer
+ports after scheduled time by ALSA sequencer dispatcher.
+<P>If the write-queue is full, the application sleeps until a certain amount
+(as default one half) becomes empty in blocking mode. The synchronization
+to write timing was implemented, too.
+<P>The input from MIDI devices or echo-back events are stored on read FIFO
+queue. If application reads <TT>/dev/sequencer</TT> in blocking mode, the
+process will be awaked.
+
+<H2>
+7. Interface to Synthesizer Device</H2>
+
+<H3>
+7.1. Registration</H3>
+To register an OSS synthesizer device, use <TT>snd_seq_oss_synth_register</TT>
+function.
+<PRE>int snd_seq_oss_synth_register(char *name, int type, int subtype, int nvoices,
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; snd_seq_oss_callback_t *oper, void *private_data)</PRE>
+The arguments <TT>name</TT>, <TT>type</TT>, <TT>subtype</TT> and
+<TT>nvoices</TT>
+are used for making the appropriate synth_info structure for ioctl. The
+return value is an index number of this device. This index must be remembered
+for unregister. If registration is failed, -errno will be returned.
+<P>To release this device, call <TT>snd_seq_oss_synth_unregister function</TT>:
+<PRE>int snd_seq_oss_synth_unregister(int index),</PRE>
+where the <TT>index</TT> is the index number returned by register function.
+<H3>
+7.2. Callbacks</H3>
+OSS synthesizer devices have capability for sample downloading and ioctls
+like sample reset. In OSS emulation, these special features are realized
+by using callbacks. The registration argument oper is used to specify these
+callbacks. The following callback functions must be defined:
+<PRE>snd_seq_oss_callback_t:
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*open)(snd_seq_oss_arg_t *p, void *closure);
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*close)(snd_seq_oss_arg_t *p);
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*ioctl)(snd_seq_oss_arg_t *p, unsigned int cmd, unsigned long arg);
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*load_patch)(snd_seq_oss_arg_t *p, int format, const char *buf, int offs, int count);
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int (*reset)(snd_seq_oss_arg_t *p);
+Except for <TT>open</TT> and <TT>close</TT> callbacks, they are allowed
+to be NULL.
+<P>Each callback function takes the argument type snd_seq_oss_arg_t as the
+first argument.
+<PRE>struct snd_seq_oss_arg_t {
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int app_index;
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int file_mode;
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int seq_mode;
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; snd_seq_addr_t addr;
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; void *private_data;
+&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; int event_passing;
+};</PRE>
+The first three fields, <TT>app_index</TT>, <TT>file_mode</TT> and
+<TT>seq_mode</TT>
+are initialized by OSS sequencer. The <TT>app_index</TT> is the application
+index which is unique to each application opening OSS sequencer. The
+<TT>file_mode</TT>
+is bit-flags indicating the file operation mode. See
+<TT>seq_oss.h</TT>
+for its meaning. The <TT>seq_mode</TT> is sequencer operation mode. In
+the current version, only <TT>SND_OSSSEQ_MODE_SYNTH</TT> is used.
+<P>The next two fields, <TT>addr</TT> and <TT>private_data</TT>, must be
+filled by the synth driver at open callback. The <TT>addr</TT> contains
+the address of ALSA sequencer port which is assigned to this device. If
+the driver allocates memory for <TT>private_data</TT>, it must be released
+in close callback by itself.
+<P>The last field, <TT>event_passing</TT>, indicates how to translate note-on
+/ off events. In <TT>PROCESS_EVENTS</TT> mode, the note 255 is regarded
+as velocity change, and key pressure event is passed to the port. In <TT>PASS_EVENTS</TT>
+mode, all note on/off events are passed to the port without modified. <TT>PROCESS_KEYPRESS</TT>
+mode checks the note above 128 and regards it as key pressure event (mainly
+for Emu8000 driver).
+<H4>
+7.2.1. Open Callback</H4>
+The <TT>open</TT> is called at each time this device is opened by an application
+using OSS sequencer. This must not be NULL. Typically, the open callback
+does the following procedure:
+<OL>
+<LI>
+Allocate private data record.</LI>
+
+<LI>
+Create an ALSA sequencer port.</LI>
+
+<LI>
+Set the new port address on arg->addr.</LI>
+
+<LI>
+Set the private data record pointer on arg->private_data.</LI>
+</OL>
+Note that the type bit-flags in port_info of this synth port must NOT contain
+<TT>TYPE_MIDI_GENERIC</TT>
+bit. Instead, <TT>TYPE_SPECIFIC</TT> should be used. Also, <TT>CAP_SUBSCRIPTION</TT>
+bit should NOT be included, too. This is necessary to tell it from other
+normal MIDI devices. If the open procedure succeeded, return zero. Otherwise,
+return -errno.
+<H4>
+7.2.2 Ioctl Callback</H4>
+The <TT>ioctl</TT> callback is called when the sequencer receives device-specific
+ioctls. The following two ioctls should be processed by this callback:
+<UL>
+<LI>
+<TT>IOCTL_SEQ_RESET_SAMPLES</TT></LI>
+
+<BR>reset all samples on memory -- return 0
+<LI>
+<TT>IOCTL_SYNTH_MEMAVL</TT></LI>
+
+<BR>return the available memory size
+<LI>
+<TT>FM_4OP_ENABLE</TT></LI>
+
+<BR>can be ignored usually</UL>
+The other ioctls are processed inside the sequencer without passing to
+the lowlevel driver.
+<H4>
+7.2.3 Load_Patch Callback</H4>
+The <TT>load_patch</TT> callback is used for sample-downloading. This callback
+must read the data on user-space and transfer to each device. Return 0
+if succeeded, and -errno if failed. The format argument is the patch key
+in patch_info record. The buf is user-space pointer where patch_info record
+is stored. The offs can be ignored. The count is total data size of this
+sample data.
+<H4>
+7.2.4 Close Callback</H4>
+The <TT>close</TT> callback is called when this device is closed by the
+applicaion. If any private data was allocated in open callback, it must
+be released in the close callback. The deletion of ALSA port should be
+done here, too. This callback must not be NULL.
+<H4>
+7.2.5 Reset Callback</H4>
+The <TT>reset</TT> callback is called when sequencer device is reset or
+closed by applications. The callback should turn off the sounds on the
+relevant port immediately, and initialize the status of the port. If this
+callback is undefined, OSS seq sends a <TT>HEARTBEAT</TT> event to the
+port.
+<H3>
+7.3 Events</H3>
+Most of the events are processed by sequencer and translated to the adequate
+ALSA sequencer events, so that each synth device can receive by input_event
+callback of ALSA sequencer port. The following ALSA events should be implemented
+by the driver:
+<BR>&nbsp;
+<TABLE BORDER WIDTH="75%" NOSAVE >
+<TR NOSAVE>
+<TD NOSAVE><B>ALSA event</B></TD>
+
+<TD><B>Original OSS events</B></TD>
+</TR>
+
+<TR>
+<TD>NOTEON</TD>
+
+<TD>SEQ_NOTEON
+<BR>MIDI_NOTEON</TD>
+</TR>
+
+<TR>
+<TD>NOTE</TD>
+
+<TD>SEQ_NOTEOFF
+<BR>MIDI_NOTEOFF</TD>
+</TR>
+
+<TR NOSAVE>
+<TD NOSAVE>KEYPRESS</TD>
+
+<TD>MIDI_KEY_PRESSURE</TD>
+</TR>
+
+<TR NOSAVE>
+<TD>CHANPRESS</TD>
+
+<TD NOSAVE>SEQ_AFTERTOUCH
+<BR>MIDI_CHN_PRESSURE</TD>
+</TR>
+
+<TR NOSAVE>
+<TD NOSAVE>PGMCHANGE</TD>
+
+<TD NOSAVE>SEQ_PGMCHANGE
+<BR>MIDI_PGM_CHANGE</TD>
+</TR>
+
+<TR>
+<TD>PITCHBEND</TD>
+
+<TD>SEQ_CONTROLLER(CTRL_PITCH_BENDER)
+<BR>MIDI_PITCH_BEND</TD>
+</TR>
+
+<TR>
+<TD>CONTROLLER</TD>
+
+<TD>MIDI_CTL_CHANGE
+<BR>SEQ_BALANCE (with CTL_PAN)</TD>
+</TR>
+
+<TR>
+<TD>CONTROL14</TD>
+
+<TD>SEQ_CONTROLLER</TD>
+</TR>
+
+<TR>
+<TD>REGPARAM</TD>
+
+<TD>SEQ_CONTROLLER(CTRL_PITCH_BENDER_RANGE)</TD>
+</TR>
+
+<TR>
+<TD>SYSEX</TD>
+
+<TD>SEQ_SYSEX</TD>
+</TR>
+</TABLE>
+
+<P>The most of these behavior can be realized by MIDI emulation driver
+included in the Emu8000 lowlevel driver. In the future release, this module
+will be independent.
+<P>Some OSS events (<TT>SEQ_PRIVATE</TT> and <TT>SEQ_VOLUME</TT> events) are passed as event
+type SND_SEQ_OSS_PRIVATE. The OSS sequencer passes these event 8 byte
+packets without any modification. The lowlevel driver should process these
+events appropriately.
+<H2>
+8. Interface to MIDI Device</H2>
+Since the OSS emulation probes the creation and deletion of ALSA MIDI sequencer
+ports automatically by receiving announcement from ALSA sequencer, the
+MIDI devices don't need to be registered explicitly like synth devices.
+However, the MIDI port_info registered to ALSA sequencer must include a group
+name <TT>SND_SEQ_GROUP_DEVICE</TT> and a capability-bit <TT>CAP_READ</TT> or
+<TT>CAP_WRITE</TT>. Also, subscription capabilities, <TT>CAP_SUBS_READ</TT> or <TT>CAP_SUBS_WRITE</TT>,
+must be defined, too. If these conditions are not satisfied, the port is not
+registered as OSS sequencer MIDI device.
+<P>The events via MIDI devices are parsed in OSS sequencer and converted
+to the corresponding ALSA sequencer events. The input from MIDI sequencer
+is also converted to MIDI byte events by OSS sequencer. This works just
+a reverse way of seq_midi module.
+<H2>
+9. Known Problems / TODO's</H2>
+
+<UL>
+<LI>
+Patch loading via ALSA instrument layer is not implemented yet.</LI>
+</UL>
+
+</BODY>
+</HTML>
diff --git a/Documentation/sound/alsa/serial-u16550.txt b/Documentation/sound/alsa/serial-u16550.txt
new file mode 100644
index 000000000000..c1919559d509
--- /dev/null
+++ b/Documentation/sound/alsa/serial-u16550.txt
@@ -0,0 +1,88 @@
+
+ Serial UART 16450/16550 MIDI driver
+ ===================================
+
+The adaptor module parameter allows you to select either:
+
+ 0 - Roland Soundcanvas support (default)
+ 1 - Midiator MS-124T support (1)
+ 2 - Midiator MS-124W S/A mode (2)
+ 3 - MS-124W M/B mode support (3)
+ 4 - Generic device with multiple input support (4)
+
+For the Midiator MS-124W, you must set the physical M-S and A-B
+switches on the Midiator to match the driver mode you select.
+
+In Roland Soundcanvas mode, multiple ALSA raw MIDI substreams are supported
+(midiCnD0-midiCnD15). Whenever you write to a different substream, the driver
+sends the nonstandard MIDI command sequence F5 NN, where NN is the substream
+number plus 1. Roland modules use this command to switch between different
+"parts", so this feature lets you treat each part as a distinct raw MIDI
+substream. The driver provides no way to send F5 00 (no selection) or to not
+send the F5 NN command sequence at all; perhaps it ought to.
+
+Usage example for simple serial converter:
+
+ /sbin/setserial /dev/ttyS0 uart none
+ /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 speed=115200
+
+Usage example for Roland SoundCanvas with 4 MIDI ports:
+
+ /sbin/setserial /dev/ttyS0 uart none
+ /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 outs=4
+
+In MS-124T mode, one raw MIDI substream is supported (midiCnD0); the outs
+module parameter is automatically set to 1. The driver sends the same data to
+all four MIDI Out connectors. Set the A-B switch and the speed module
+parameter to match (A=19200, B=9600).
+
+Usage example for MS-124T, with A-B switch in A position:
+
+ /sbin/setserial /dev/ttyS0 uart none
+ /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=1 \
+ speed=19200
+
+In MS-124W S/A mode, one raw MIDI substream is supported (midiCnD0);
+the outs module parameter is automatically set to 1. The driver sends
+the same data to all four MIDI Out connectors at full MIDI speed.
+
+Usage example for S/A mode:
+
+ /sbin/setserial /dev/ttyS0 uart none
+ /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=2
+
+In MS-124W M/B mode, the driver supports 16 ALSA raw MIDI substreams;
+the outs module parameter is automatically set to 16. The substream
+number gives a bitmask of which MIDI Out connectors the data should be
+sent to, with midiCnD1 sending to Out 1, midiCnD2 to Out 2, midiCnD4 to
+Out 3, and midiCnD8 to Out 4. Thus midiCnD15 sends the data to all 4 ports.
+As a special case, midiCnD0 also sends to all ports, since it is not useful
+to send the data to no ports. M/B mode has extra overhead to select the MIDI
+Out for each byte, so the aggregate data rate across all four MIDI Outs is
+at most one byte every 520 us, as compared with the full MIDI data rate of
+one byte every 320 us per port.
+
+Usage example for M/B mode:
+
+ /sbin/setserial /dev/ttyS0 uart none
+ /sbin/modprobe snd-serial-u16550 port=0x3f8 irq=4 adaptor=3
+
+The MS-124W hardware's M/A mode is currently not supported. This mode allows
+the MIDI Outs to act independently at double the aggregate throughput of M/B,
+but does not allow sending the same byte simultaneously to multiple MIDI Outs.
+The M/A protocol requires the driver to twiddle the modem control lines under
+timing constraints, so it would be a bit more complicated to implement than
+the other modes.
+
+Midiator models other than MS-124W and MS-124T are currently not supported.
+Note that the suffix letter is significant; the MS-124 and MS-124B are not
+compatible, nor are the other known models MS-101, MS-101B, MS-103, and MS-114.
+I do have documentation (tim.mann@compaq.com) that partially covers these models,
+but no units to experiment with. The MS-124W support is tested with a real unit.
+The MS-124T support is untested, but should work.
+
+The Generic driver supports multiple input and output substreams over a single
+serial port. Similar to Roland Soundcanvas mode, F5 NN is used to select the
+appropriate input or output stream (depending on the data direction).
+Additionally, the CTS signal is used to regulate the data flow. The number of
+inputs is specified by the ins parameter.