Merge suse.cz:/home/perex/bk/linux-sound/linux-sound

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Documentation/sound/alsa/ALSA-Configuration.txt

+
+		Advanced Linux Sound Architecture - Driver
+		==========================================
+			    Configuration guide
+
+
+Kernel Configuration
+====================
+
+To enable the ALSA support, at least you need to build the kernel with
+the primary sound card support (CONFIG_SOUND).  Since ALSA can emulate
+the OSS, you don't have to choose any of the OSS/Free modules.  Please
+enable "OSS API emulation" (CONFIG_SND_OSSEMUL) and both OSS mixer and
+PCM supports if you want to run the OSS application with the ALSA.
+
+When you want to support the WaveTable functionality on some cards
+such like SB Live!, you need to enable "Sequencer support"
+(CONFIG_SND_SEQUENCER).
+
+For getting more verbose debug messages, turn on "Verbose printk" and
+"Debug" options.  For checking the memory leaks, you can turn on
+"Debug memory" option, too.  "Debug detection" will put more
+additional checks for the detection of cards.
+
+Please note that all the ALSA ISA drivers support Linux isapnp API (if
+the card supports).  You don't need to configure the PnP via
+isapnptools.
+
+
+Module parameters
+=================
+
+  A user can modify or set parameters at the load time of the module. If
+  the module supports more cards and you have got more than one card
+  of the same type, you may simply specify more values for the parameter,
+  delimited by commas.
+  
+  Note that module option names were changed in 0.9.0rc4. The 'snd_'
+  prefix was removed.
+
+  Module snd
+  ----------
+
+    The module snd is the ALSA core module, which is used by all ALSA
+    card drivers.  This takes the global options for creating devices,
+    etc.
+
+    major	- major # for sound driver
+		- default is 116
+    cards_limit
+		- specifies card limit # (1-8)
+		- good for kmod support if you do not want to search
+		  for soundcards which are not installed in your system
+    device_mode
+		- specifies permission mask for dynamic sound device filesystem
+		- default value = 0666
+		- for example 'device_mode=0660'
+    device_gid
+		- specifies GID number for dynamic sound device filesystem
+		- default value = 0 (root)
+    device_uid
+		- specifies UID number for dynamic sound device filesystem
+		- default value = 0 (root)
+
+  
+  Module snd-pcm-oss
+  ------------------
+
+    The PCM OSS emulation module.
+    This module takes the options to 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 the options to change the mapping of OSS
+    devices like 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)
+
+  Global parameters for top soundcard modules
+  -------------------------------------------
+
+    Each of top-level soundcard module takes some general options,
+
+    index	- 0-7 - index (slot #) for soundcard
+                - if not set or -1, first free index (slot #) is assigned
+    id		- user identification for card (up to 15 chars)
+		- default expression is 'card<index>' (for example card1)
+		- value is used for /proc/asound filesystem
+		- this value can be used by applications for identification
+		  of card if user does not want identify card with index number
+    enable  	- enable card (only first card is enabled by default)
+
+  Module snd-ad1816a
+  ------------------
+
+    Module for soundcards 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 soundcards 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
+
+    Module supports autoprobe and multiple chips (max 8).
+
+    The power-management is supported.
+
+  Module snd-als100
+  -----------------
+
+    Module for soundcards 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 soundcards based on Avance Logic ALS4000 PCI chip.
+
+    joystick_port - port # for legacy joystick support.
+                        default: 0x200 for the 1st card.
+                        0 = disabled
+    
+    Module supports up to 8 cards, autoprobe and PnP.
+
+  Module snd-azt2320
+  ------------------
+
+    Module for soundcards 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-cmi8330
+  ------------------
+
+    Module for soundcards 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 soundcards.
+
+    mpu_port	- 0x300 (default),0x310,0x320,0x330, -1 (diable)
+    fm_port     - 0x388 (default), -1 (disable)
+
+    Module supports autoprobe and multiple chips (max 8).
+    
+  Module snd-cs4231
+  -----------------
+
+    Module for soundcards 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 soundcards 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 soundcards 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 soundcards 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 soundcard. This soundcard doesn't do any output
+    or input, but you may use this module for any application which
+    requires a soundcard (like RealPlayer).
+
+  Module snd-emu10k1
+  ------------------
+
+    Module for EMU10K1/EMU10k2 based PCI soundcards.
+			* 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 all ins and outs		[0x3fff/0x1fff]
+    
+  Module snd-ens1370
+  ------------------
+
+    Module for Ensoniq AudioPCI ES1370 PCI soundcards.
+			* SoundBlaster PCI 64
+			* SoundBlaster PCI 128
+
+    Module supports up to 8 cards and autoprobe.
+    
+  Module snd-ens1371
+  ------------------
+
+    Module for Ensoniq AudioPCI ES1371 PCI soundcards.
+			* SoundBlaster PCI 64
+			* SoundBlaster PCI 128
+			* SoundBlaster Vibra PCI
+
+    Module supports up to 8 cards and autoprobe.
+    
+  Module snd-es968
+  ----------------
+
+    Module for soundcards 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 soundcards.
+
+    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 soundcards.
+
+    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 soundcards based on ESS Solo-1 (ES1938,ES1946) chips.
+
+    Module supports up to 8 cards and autoprobe.
+
+  Module snd-es1968
+  -----------------
+
+    Module for soundcards 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)
+
+    Module supports up to 8 cards and autoprobe.
+
+    The power-management is supported.
+
+  Module snd-fm801
+  ----------------
+
+    Module for ForteMedia FM801 based PCI soundcards.
+
+    Module supports up to 8 cards and autoprobe.
+    
+  Module snd-gusclassic
+  ---------------------
+
+    Module for Gravis UltraSound Classic soundcard.
+
+    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) soundcard.
+
+    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 soundcard.
+
+    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-hdsp
+  ---------------
+
+    Module for RME Hammerfall DSP audio interface(s)
+
+    precise_ptr	      - Enable precise pointer (doesn't work reliably).
+			(default = 0)
+    line_outs_monitor - Send all input and playback streams to line outs
+			by default. (default = 0)
+    force_firmware    - Force a reload of the I/O box firmware
+			(default = 0)
+
+    Module supports up to 8 cards.
+
+  Module snd-ice1712
+  ------------------
+
+    Module for Envy24 (ICE1712) based PCI soundcards.
+			* MidiMan M Audio Delta 1010
+			* MidiMan M Audio Delta DiO 2496
+			* MidiMan M Audio Delta 66
+			* MidiMan M Audio Delta 44
+			* MidiMan M Audio Audiophile 2496
+                        * TerraTec EWS 88MT
+                        * TerraTec EWS 88D
+                        * TerraTec EWX 24/96
+
+    omni	- Omni I/O support for MidiMan M-Audio Delta44/66
+
+    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-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)
+    joystick_port - Joystick port # (0 = disabled, 0x200)
+    mpu_port      - MPU401 port # (0 = disabled, 0x330,0x300)
+
+    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".
+
+    The joystick and MPU-401 are supported only certain hardwares.
+    MPU401 is experimental,  It doesn't work perfectly.
+
+    The power-management is supported.
+    
+  Module snd-interwave
+  --------------------
+
+    Module for Gravis UltraSound PnP, Dynasonic 3-D/Pro, STB Sound Rage 32
+    and other soundcards 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 (soundcard from STB used by Compaq)
+    and other soundcards 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-mpu401
+  -----------------
+
+    Module for MPU-401 UART devices.
+
+    port	- port number or -1 (disable)
+    irq		- IRQ number or -1 (disable)
+
+    Module supports multiple devices (max 8).
+    
+  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
+
+    Module supports autoprobe and multiple chips (max 8).
+    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.
+
+    The power-management is supported.
+
+  Module snd-opl3sa2
+  ------------------
+
+    Module for Yamaha OPL3-SA2/SA3 soundcards.
+
+    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 soundcards 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 soundcards 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 soundcards 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, Digi32/8 and Digi32 PRO soundcards.
+
+    Module supports up to 8 cards.
+
+  Module snd-rme96
+  ----------------
+
+    Module for RME Digi96, Digi96/8 and Digi96/8 PRO/PAD/PST soundcards.
+
+    Module supports up to 8 cards.
+
+  Module snd-rme9652
+  ------------------
+
+    Module for RME Digi9652 (Hammerfall, Hammerfall-Light) soundcards.
+
+    precise_ptr	- Enable precise pointer (doesn't work reliably).
+		  (default = 0)
+
+    Module supports up to 8 cards.
+
+  Module snd-sa11xx-uda1341 (on arm only)
+  ---------------------------------------
+
+    Module for Philips UDA1341TS on Compaq iPAQ H3600 soundcard.
+
+    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 soundcard.
+
+    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-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+ soundcards.
+
+    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 soundcards.
+			* 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 soundcards.
+			* 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-via82xx
+  ------------------
+
+    Module for AC'97 motherboards based on VIA 82C686A/686B, 8233
+    (south) bridge.
+
+    mpu_port	- 0x300,0x310,0x320,0x330, otherwise obtain BIOS setup
+    ac97_clock	- AC'97 codec clock base (default 48000Hz)
+
+    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.
+    
+  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-ymfpci
+  -----------------
+
+    Module for Yamaha PCI chips (YMF72x, YMF74x & YMF75x).
+
+    mpu_port	- 0x300,0x330,0x332,0x334, -1 (disable) by default
+    fm_port	- 0x388,0x398,0x3a0,0x3a8, -1 (disable) by default
+    rear_switch - enable shared rear/line-in switch (bool)
+
+    Module supports autoprobe and multiple chips (max 8).
+    
+    The power-management is supported.
+
+
+modprobe/kmod support
+=====================
+
+The modprobe program must know which modules are used for the
+device major numbers.
+Native ALSA devices have got default number 116. Thus a line like
+'alias char-major-116 snd' must be added to /etc/modules.conf. If you have 
+compiled the ALSA driver with the OSS/Free emulation code, then you
+will need to add lines as explained below:
+
+The ALSA driver uses soundcore multiplexer for 2.2+ kernels and OSS compatible
+devices. You should add line like 'alias char-major-14 soundcore'.
+
+Example with OSS/Free emulation turned on:
+
+----- /etc/modules.conf
+
+# ALSA portion
+alias char-major-116 snd
+# OSS/Free portion
+alias char-major-14 soundcore
+
+----- /etc/modules.conf
+
+After the main multiplexer is loaded, its code requests top-level soundcard
+module. String 'snd-card-%i' is requested for native devices where %i is
+soundcard number from zero to seven. String 'sound-slot-%i' is requested
+for native devices where %i is slot number (for ALSA owner this means soundcard
+number).
+
+----- /etc/modules.conf
+
+# ALSA portion
+alias snd-card-0 snd-interwave
+alias snd-card-1 snd-ens1371
+# OSS/Free portion
+alias sound-slot-0 snd-card-0
+alias sound-slot-1 snd-card-1
+
+----- /etc/modules.conf
+
+We are finished at this point with the configuration for ALSA native devices,
+but you may also need autoloading for ALSA's add-on OSS/Free emulation
+modules. At this time only one module does not depend on any others, thus
+must be loaded separately - snd-pcm-oss. String 'sound-service-%i-%i'
+is requested for OSS/Free service where first %i means slot number
+(e.g. card number) and second %i means service number.
+
+----- /etc/modules.conf
+
+# OSS/Free portion - card #1
+alias sound-service-0-0 snd-mixer-oss
+alias sound-service-0-1 snd-seq-oss
+alias sound-service-0-3 snd-pcm-oss
+alias sound-service-0-8 snd-seq-oss
+alias sound-service-0-12 snd-pcm-oss
+# OSS/Free portion - card #2
+alias sound-service-1-0 snd-mixer-oss
+alias sound-service-1-3 snd-pcm-oss
+alias sound-service-1-12 snd-pcm-oss
+
+----- /etc/modules.conf
+
+A complete example for Gravis UltraSound PnP soundcard:
+
+----- /etc/modules.conf
+
+# ISA PnP support (don't use IRQs 9,10,11,12,13)
+options isapnp isapnp_reserve_irq=9,10,11,12,13
+
+# ALSA native device support
+alias char-major-116 snd
+options snd major=116 cards_limit=1
+alias snd-card-0 snd-interwave
+options snd-interwave index=0 id="GusPnP"
+
+# OSS/Free setup
+alias char-major-14 soundcore
+alias sound-slot-0 snd-card-0
+alias sound-service-0-0 snd-mixer-oss
+alias sound-service-0-1 snd-seq-oss
+alias sound-service-0-3 snd-pcm-oss
+alias sound-service-0-8 snd-seq-oss
+alias sound-service-0-12 snd-pcm-oss
+
+-----
+
+A complete example if you want to use more soundcards in one machine
+(the configuration below is for Sound Blaster 16 and Gravis UltraSound Classic):
+
+----- /etc/modules.conf
+
+# ISA PnP support (don't use IRQs 9,10,11,12,13)
+# it's only an example to reserve some IRQs for another hardware
+options isapnp isapnp_reserve_irq=9,10,11,12,13
+
+# ALSA native device support
+alias char-major-116 snd
+options snd major=116 cards_limit=2
+alias snd-card-0 snd-gusclassic
+alias snd-card-1 snd-sb16
+options snd-gusclassic index=0 id="Gus" \
+        port=0x220 irq=5 dma1=6 dma2=7
+options snd-sb16 index=1 id="SB16"
+
+# OSS/Free setup
+alias char-major-14 soundcore
+alias sound-slot-0 snd-card-0
+alias sound-service-0-0 snd-mixer-oss
+alias sound-service-0-1 snd-seq-oss
+alias sound-service-0-3 snd-pcm-oss
+alias sound-service-0-8 snd-seq-oss
+alias sound-service-0-12 snd-pcm-oss
+alias sound-slot-1 snd-card-1
+alias sound-service-1-0 snd-mixer-oss
+alias sound-service-1-3 snd-pcm-oss
+alias sound-service-1-12 snd-pcm-oss
+
+-----
+
+A complete example, two Gravis UltraSound Classic soundcards are installed
+in the system:
+
+----- /etc/modules.conf
+
+# ALSA native device support
+alias char-major-116 snd
+options snd major=116 cards_limit=2
+alias snd-card-0 snd-gusclassic
+alias snd-card-1 snd-gusclassic
+options snd-gusclassic index=0,1 id="Gus1","Gus2" \
+        port=0x220,0x240 irq=5,7 dma1=1,5 dma2=3,6
+
+# OSS/Free setup
+alias char-major-14 soundcore
+alias sound-slot-0 snd-card-0
+alias sound-service-0-0 snd-mixer-oss
+alias sound-service-0-1 snd-seq-oss
+alias sound-service-0-3 snd-pcm-oss
+alias sound-service-0-8 snd-seq-oss
+alias sound-service-0-12 snd-pcm-oss
+alias sound-slot-1 snd-card-1
+alias sound-service-1-0 snd-mixer-oss
+alias sound-service-1-3 snd-pcm-oss
+alias sound-service-1-12 snd-pcm-oss
+
+-----
+
+If you want to autoclean your modules, you should put below line to your
+/etc/crontab:
+
+*/10 * * * *   root  /sbin/modprobe -rs snd-card-0 snd-card-1; /sbin/rmmod -as
+
+You may also want to extend the soundcard list to follow your requirements.
+
+
+ALSA PCM devices to OSS devices mapping
+=======================================
+
+/dev/snd/pcmC0D0  -> /dev/audio0 (/dev/audio) -> minor 4
+/dev/snd/pcmC0D0  -> /dev/dsp0 (/dev/dsp)     -> minor 3
+/dev/snd/pcmC0D1  -> /dev/adsp0 (/dev/adsp)   -> minor 12
+/dev/snd/pcmC1D0  -> /dev/audio1              -> minor 4+16 = 20
+/dev/snd/pcmC1D0  -> /dev/dsp1                -> minor 3+16 = 19
+/dev/snd/pcmC1D1  -> /dev/adsp1               -> minor 12+16 = 28
+/dev/snd/pcmC2D0  -> /dev/audio2              -> minor 4+32 = 36
+/dev/snd/pcmC2D0  -> /dev/dsp2                -> minor 3+32 = 39
+/dev/snd/pcmC2D1  -> /dev/adsp2               -> minor 12+32 = 44
+
+The first number from /dev/snd/pcmC{X}D{Y} expression means soundcard number
+and second means device number.
+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
+
+  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
+

Documentation/sound/alsa/ControlNames.txt

+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 */

drivers/net/hp100.c

@@ -2091,6 +2091,9 @@ static void hp100_update_stats(struct net_device *dev)
 
 static void hp100_misc_interrupt(struct net_device *dev)
 {
+#ifdef HP100_DEBUG_B
+	int ioaddr = dev->base_addr;
+#endif
 	struct hp100_private *lp = (struct hp100_private *) dev->priv;
 
 #ifdef HP100_DEBUG_B

include/sound/version.h

 /* include/version.h.  Generated automatically by configure.  */
 #define CONFIG_SND_VERSION "0.9.0rc6"
-#define CONFIG_SND_DATE " (Wed Dec 11 21:24:39 2002 UTC)"
+#define CONFIG_SND_DATE " (Tue Dec 17 19:01:13 2002 UTC)"

sound/core/pcm_sgbuf.c

@@ -92,6 +92,56 @@ int snd_pcm_sgbuf_delete(snd_pcm_substream_t *substream)
 	return 0;
 }
 
+/*
+ * snd_pci_alloc_page - allocate a page in the valid pci dma mask
+ *
+ * returns the virtual address and stores the physical address on
+ * addrp.  this function cannot be called from interrupt handlers or
+ * within spinlocks.
+ */
+#ifdef __i386__
+/*
+ * on ix86, we allocate a page with GFP_KERNEL to assure the
+ * allocation.  the code is almost same with kernel/i386/pci-dma.c but
+ * it allocates only a single page and checkes the validity of the
+ * page address with the given pci dma mask.
+ */
+inline static void *snd_pci_alloc_page(struct pci_dev *pci, dma_addr_t *addrp)
+{
+	void *ptr;
+	dma_addr_t addr;
+	unsigned long rmask;
+
+	if (pci)
+		rmask = ~(unsigned long)pci->dma_mask;
+	else
+		rmask = 0;
+	ptr = (void *)__get_free_page(GFP_KERNEL);
+	if (ptr) {
+		addr = virt_to_phys(ptr);
+		if (((unsigned long)addr + PAGE_SIZE - 1) & rmask) {
+			/* try to reallocate with the GFP_DMA */
+			free_page((unsigned long)ptr);
+			ptr = (void *)__get_free_page(GFP_KERNEL | GFP_DMA);
+			if (ptr) /* ok, the address must be within lower 16MB... */
+				addr = virt_to_phys(ptr);
+			else
+				addr = 0;
+		}
+	} else
+		addr = 0;
+	if (ptr)
+		memset(ptr, 0, PAGE_SIZE);
+	*addrp = addr;
+	return ptr;
+}
+#else
+/* on other architectures, call snd_malloc_pci_pages() helper function
+ * which uses pci_alloc_consistent().
+ */
+#define snd_pci_alloc_page(pci, addrp) snd_malloc_pci_pages(pci, PAGE_SIZE, addrp)
+#endif
+
 /*
  * allocate sg buffer table with the given byte size.
  * if the buffer table already exists, try to resize it.
@@ -128,7 +178,7 @@ int snd_pcm_sgbuf_alloc(snd_pcm_substream_t *substream, size_t size)
 	while (sgbuf->pages < pages) {
 		void *ptr;
 		dma_addr_t addr;
-		ptr = snd_malloc_pci_pages(sgbuf->pci, PAGE_SIZE, &addr);
+		ptr = snd_pci_alloc_page(sgbuf->pci, &addr);
 		if (! ptr)
 			return -ENOMEM;
 		sgbuf->table[sgbuf->pages].buf = ptr;

sound/drivers/mtpav.c

@@ -205,19 +205,19 @@ static int translate_subdevice_to_hwport(mtpav_t *chip, int subdev)
 
 static int translate_hwport_to_subdevice(mtpav_t *chip, int hwport)
 {
-	int port;
+	int p;
 	if (hwport <= 0x00) /* all ports */
 		return chip->num_ports + MTPAV_PIDX_BROADCAST;
 	else if (hwport <= 0x08) { /* single port */
-		port = hwport - 1;
-		if (port >= chip->num_ports)
-			port = 0;
-		return port;
+		p = hwport - 1;
+		if (p >= chip->num_ports)
+			p = 0;
+		return p;
 	} else if (hwport <= 0x10) { /* remote port */
-		port = hwport - 0x09 + chip->num_ports;
-		if (port >= chip->num_ports * 2)
-			port = chip->num_ports;
-		return port;
+		p = hwport - 0x09 + chip->num_ports;
+		if (p >= chip->num_ports * 2)
+			p = chip->num_ports;
+		return p;
 	} else if (hwport == 0x11)  /* computer port */
 		return chip->num_ports + MTPAV_PIDX_COMPUTER;
 	else  /* ADAT */
@@ -335,11 +335,11 @@ static void snd_mtpav_output_write(snd_rawmidi_substream_t * substream)
 
 static void snd_mtpav_portscan(mtpav_t *chip)	// put mtp into smart routing mode
 {
-	u8 port;
+	u8 p;
 
-	for (port = 0; port < 8; port++) {
+	for (p = 0; p < 8; p++) {
 		snd_mtpav_send_byte(chip, 0xf5);
-		snd_mtpav_send_byte(chip, port);
+		snd_mtpav_send_byte(chip, p);
 		snd_mtpav_send_byte(chip, 0xfe);
 	}
 }
@@ -350,12 +350,12 @@ static void snd_mtpav_portscan(mtpav_t *chip)	// put mtp into smart routing mode
 static int snd_mtpav_input_open(snd_rawmidi_substream_t * substream)
 {
 	unsigned long flags;
-	mtpav_port_t *port = &mtp_card->ports[substream->number];
+	mtpav_port_t *portp = &mtp_card->ports[substream->number];
 
 	//printk("mtpav port: %d opened\n", (int) substream->number);
 	spin_lock_irqsave(&mtp_card->spinlock, flags);
-	port->mode |= MTPAV_MODE_INPUT_OPENED;
-	port->input = substream;
+	portp->mode |= MTPAV_MODE_INPUT_OPENED;
+	portp->input = substream;
 	if (mtp_card->share_irq++ == 0)
 		snd_mtpav_mputreg(mtp_card, CREG, (SIGC_INTEN | SIGC_WRITE));	// enable pport interrupts
 	spin_unlock_irqrestore(&mtp_card->spinlock, flags);
@@ -368,14 +368,14 @@ static int snd_mtpav_input_open(snd_rawmidi_substream_t * substream)
 static int snd_mtpav_input_close(snd_rawmidi_substream_t *substream)
 {
 	unsigned long flags;
-	mtpav_port_t *port = &mtp_card->ports[substream->number];
+	mtpav_port_t *portp = &mtp_card->ports[substream->number];
 
-	//printk("mtpav port: %d closed\n", (int) port);
+	//printk("mtpav port: %d closed\n", (int) portp);
 
 	spin_lock_irqsave(&mtp_card->spinlock, flags);
 
-	port->mode &= (~MTPAV_MODE_INPUT_OPENED);
-	port->input = NULL;
+	portp->mode &= (~MTPAV_MODE_INPUT_OPENED);
+	portp->input = NULL;
 	if (--mtp_card->share_irq == 0)
 		snd_mtpav_mputreg(mtp_card, CREG, 0);	// disable pport interrupts
 
@@ -389,13 +389,13 @@ static int snd_mtpav_input_close(snd_rawmidi_substream_t *substream)
 static void snd_mtpav_input_trigger(snd_rawmidi_substream_t * substream, int up)
 {
 	unsigned long flags;
-	mtpav_port_t *port = &mtp_card->ports[substream->number];
+	mtpav_port_t *portp = &mtp_card->ports[substream->number];
 
 	spin_lock_irqsave(&mtp_card->spinlock, flags);
 	if (up)
-		port->mode |= MTPAV_MODE_INPUT_TRIGGERED;
+		portp->mode |= MTPAV_MODE_INPUT_TRIGGERED;
 	else
-		port->mode &= ~MTPAV_MODE_INPUT_TRIGGERED;
+		portp->mode &= ~MTPAV_MODE_INPUT_TRIGGERED;
 	spin_unlock_irqrestore(&mtp_card->spinlock, flags);
 
 }
@@ -416,9 +416,9 @@ static void snd_mtpav_output_timer(unsigned long data)
 	add_timer(&chip->timer);
 	/* process each port */
 	for (p = 0; p <= chip->num_ports * 2 + MTPAV_PIDX_BROADCAST; p++) {
-		mtpav_port_t *port = &mtp_card->ports[p];
-		if ((port->mode & MTPAV_MODE_OUTPUT_TRIGGERED) && port->output)
-			snd_mtpav_output_port_write(port, port->output);
+		mtpav_port_t *portp = &mtp_card->ports[p];
+		if ((portp->mode & MTPAV_MODE_OUTPUT_TRIGGERED) && portp->output)
+			snd_mtpav_output_port_write(portp, portp->output);
 	}
 	spin_unlock(&chip->spinlock);
 }
@@ -445,11 +445,11 @@ static void snd_mtpav_remove_output_timer(mtpav_t *chip)
 static int snd_mtpav_output_open(snd_rawmidi_substream_t * substream)
 {
 	unsigned long flags;
-	mtpav_port_t *port = &mtp_card->ports[substream->number];
+	mtpav_port_t *portp = &mtp_card->ports[substream->number];
 
 	spin_lock_irqsave(&mtp_card->spinlock, flags);
-	port->mode |= MTPAV_MODE_OUTPUT_OPENED;
-	port->output = substream;
+	portp->mode |= MTPAV_MODE_OUTPUT_OPENED;
+	portp->output = substream;
 	spin_unlock_irqrestore(&mtp_card->spinlock, flags);
 	return 0;
 };
@@ -460,11 +460,11 @@ static int snd_mtpav_output_open(snd_rawmidi_substream_t * substream)
 static int snd_mtpav_output_close(snd_rawmidi_substream_t * substream)
 {
 	unsigned long flags;
-	mtpav_port_t *port = &mtp_card->ports[substream->number];
+	mtpav_port_t *portp = &mtp_card->ports[substream->number];
 
 	spin_lock_irqsave(&mtp_card->spinlock, flags);
-	port->mode &= (~MTPAV_MODE_OUTPUT_OPENED);
-	port->output = NULL;
+	portp->mode &= (~MTPAV_MODE_OUTPUT_OPENED);
+	portp->output = NULL;
 	spin_unlock_irqrestore(&mtp_card->spinlock, flags);
 	return 0;
 };
@@ -475,17 +475,17 @@ static int snd_mtpav_output_close(snd_rawmidi_substream_t * substream)
 static void snd_mtpav_output_trigger(snd_rawmidi_substream_t * substream, int up)
 {
 	unsigned long flags;
-	mtpav_port_t *port = &mtp_card->ports[substream->number];
+	mtpav_port_t *portp = &mtp_card->ports[substream->number];
 
 	spin_lock_irqsave(&mtp_card->spinlock, flags);
 	if (up) {
-		if (! (port->mode  & MTPAV_MODE_OUTPUT_TRIGGERED)) {
+		if (! (portp->mode  & MTPAV_MODE_OUTPUT_TRIGGERED)) {
 			if (mtp_card->istimer++ == 0)
 				snd_mtpav_add_output_timer(mtp_card);
-			port->mode |= MTPAV_MODE_OUTPUT_TRIGGERED;
+			portp->mode |= MTPAV_MODE_OUTPUT_TRIGGERED;
 		}
 	} else {
-		port->mode &= ~MTPAV_MODE_OUTPUT_TRIGGERED;
+		portp->mode &= ~MTPAV_MODE_OUTPUT_TRIGGERED;
 		if (--mtp_card->istimer == 0)
 			snd_mtpav_remove_output_timer(mtp_card);
 	}
@@ -501,15 +501,15 @@ static void snd_mtpav_output_trigger(snd_rawmidi_substream_t * substream, int up
 
 static void snd_mtpav_inmidi_process(mtpav_t *mcrd, u8 inbyte)
 {
-	mtpav_port_t *port;
+	mtpav_port_t *portp;
 
 	if (mcrd->inmidiport > mcrd->num_ports * 2 + MTPAV_PIDX_BROADCAST)
 		return;
 
-	port = &mcrd->ports[mcrd->inmidiport];
-	if (port->mode & MTPAV_MODE_INPUT_TRIGGERED) {
+	portp = &mcrd->ports[mcrd->inmidiport];
+	if (portp->mode & MTPAV_MODE_INPUT_TRIGGERED) {
 		spin_unlock(&mcrd->spinlock);
-		snd_rawmidi_receive(port->input, &inbyte, 1);
+		snd_rawmidi_receive(portp->input, &inbyte, 1);
 		spin_lock(&mcrd->spinlock);
 	}
 }

sound/pci/ali5451/ali5451.c

@@ -1831,9 +1831,10 @@ static int snd_ali5451_spdif_put(snd_kcontrol_t * kcontrol, snd_ctl_elem_value_t
 	return change;
 }
 
-static snd_kcontrol_new_t snd_ali5451_mixer_spdif[] __devinit = {
+static snd_kcontrol_new_t snd_ali5451_mixer_spdif[] __devinitdata = {
 	/* spdif aplayback switch */
-	ALI5451_SPDIF(SNDRV_CTL_NAME_IEC958("",PLAYBACK,SWITCH), 0, 0),
+	/* FIXME: "IEC958 Playback Switch" may conflict with one on ac97_codec */
+	ALI5451_SPDIF("IEC958 Output switch", 0, 0),
 	/* spdif out to spdif channel */
 	ALI5451_SPDIF("IEC958 Channel Output Switch", 0, 1),
 	/* spdif in from spdif channel */
@@ -1861,7 +1862,7 @@ static int __devinit snd_ali_mixer(ali_t * codec)
 		return err;
 	}
 	if (codec->revision == ALI_5451_V02) {
-		for(idx = 0; idx < 3; idx++) {
+		for(idx = 0; idx < ARRAY_SIZE(snd_ali5451_mixer_spdif); idx++) {
 			err=snd_ctl_add(codec->card, snd_ctl_new1(&snd_ali5451_mixer_spdif[idx], codec));
 			if (err < 0) return err;
 		}
@@ -2010,10 +2011,12 @@ static int snd_ali_chip_init(ali_t *codec)
 	if (codec->revision == ALI_5451_V02) {
         	pci_dev = codec->pci_m1533;
 		pci_read_config_byte(pci_dev, 0x59, &temp);
+		temp |= 0x80;
+		pci_write_config_byte(pci_dev, 0x59, temp);
 	
 		pci_dev = codec->pci_m7101;
 		pci_read_config_byte(pci_dev, 0xb8, &temp);
-		temp |= 1 << 6;
+		temp |= 0x20;
 		pci_write_config_byte(pci_dev, 0xB8, temp);
 	}
 

sound/pci/fm801.c

@@ -164,7 +164,7 @@ struct _snd_fm801 {
 };
 
 static struct pci_device_id snd_fm801_ids[] __devinitdata = {
-	{ 0x1319, 0x0801, PCI_ANY_ID, PCI_ANY_ID, 0, 0, 0, },   /* FM801 */
+	{ 0x1319, 0x0801, PCI_ANY_ID, PCI_ANY_ID, PCI_CLASS_MULTIMEDIA_AUDIO << 8, 0xffff00, 0, },   /* FM801 */
 	{ 0, }
 };
 

sound/pci/ice1712/hoontech.h

@@ -24,7 +24,10 @@
  *
  */      
 
-#define  HOONTECH_DEVICE_DESC 	       "{Hoontech SoundTrack DSP 24},"
+#define  HOONTECH_DEVICE_DESC \
+	"{Hoontech SoundTrack DSP 24}," \
+	"{Hoontech SoundTrack DSP 24 Value}," \
+	"{Hoontech SoundTrack DSP 24 Media 7.1}," \
 
 #define ICE1712_SUBDEVICE_STDSP24		0x12141217	/* Hoontech SoundTrack Audio DSP 24 */
 #define ICE1712_SUBDEVICE_STDSP24_MEDIA7_1	0x16141217	/* Hoontech ST Audio DSP24 Media 7.1 */

sound/pci/intel8x0.c

@@ -53,6 +53,7 @@ MODULE_DEVICES("{{Intel,82801AA-ICH},"
 		"{Intel,MX440},"
 		"{SiS,SI7012},"
 		"{NVidia,NForce Audio},"
+		"{NVidia,NForce2 Audio},"
 		"{AMD,AMD768},"
 		"{AMD,AMD8111},"
 	        "{ALI,M5455}}");
@@ -127,6 +128,9 @@ MODULE_PARM_SYNTAX(mpu_port, SNDRV_ENABLED ",allows:{{0},{0x330},{0x300}},dialog
 #ifndef PCI_DEVICE_ID_NVIDIA_MCP_AUDIO
 #define PCI_DEVICE_ID_NVIDIA_MCP_AUDIO	0x01b1
 #endif
+#ifndef PCI_DEVICE_ID_NVIDIA_MCP2_AUDIO
+#define PCI_DEVICE_ID_NVIDIA_MCP2_AUDIO	0x006a
+#endif
 
 enum { DEVICE_INTEL, DEVICE_INTEL_ICH4, DEVICE_SIS, DEVICE_ALI };
 
@@ -383,6 +387,7 @@ static struct pci_device_id snd_intel8x0_ids[] __devinitdata = {
 	{ 0x8086, 0x7195, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL },	/* 440MX */
 	{ 0x1039, 0x7012, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_SIS },	/* SI7012 */
 	{ 0x10de, 0x01b1, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL },	/* NFORCE */
+	{ 0x10de, 0x006a, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL },	/* NFORCE2 */
 	{ 0x1022, 0x746d, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL },	/* AMD8111 */
 	{ 0x1022, 0x7445, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_INTEL },	/* AMD768 */
 	{ 0x10b9, 0x5455, PCI_ANY_ID, PCI_ANY_ID, 0, 0, DEVICE_ALI },   /* Ali5455 */
@@ -2298,6 +2303,7 @@ static struct shortname_table {
 	{ PCI_DEVICE_ID_INTEL_ICH4, "Intel 82801DB-ICH4" },
 	{ PCI_DEVICE_ID_SI_7012, "SiS SI7012" },
 	{ PCI_DEVICE_ID_NVIDIA_MCP_AUDIO, "NVidia NForce" },
+	{ PCI_DEVICE_ID_NVIDIA_MCP2_AUDIO, "NVidia NForce2" },
 	{ 0x746d, "AMD AMD8111" },
 	{ 0x7445, "AMD AMD768" },
 	{ 0x5455, "ALi M5455" },

sound/usb/usbaudio.c

@@ -526,7 +526,11 @@ static struct snd_urb_ops audio_urb_ops[2] = {
 /*
  * complete callback from data urb
  */
+#ifndef OLD_USB
 static void snd_complete_urb(struct urb *urb, struct pt_regs *regs)
+#else
+static void snd_complete_urb(struct urb *urb)
+#endif
 {
 	snd_urb_ctx_t *ctx = (snd_urb_ctx_t *)urb->context;
 	snd_usb_substream_t *subs = ctx->subs;
@@ -551,7 +555,11 @@ static void snd_complete_urb(struct urb *urb, struct pt_regs *regs)
 /*
  * complete callback from sync urb
  */
+#ifndef OLD_USB
 static void snd_complete_sync_urb(struct urb *urb, struct pt_regs *regs)
+#else
+static void snd_complete_sync_urb(struct urb *urb)
+#endif
 {
 	snd_urb_ctx_t *ctx = (snd_urb_ctx_t *)urb->context;
 	snd_usb_substream_t *subs = ctx->subs;
@@ -583,6 +591,9 @@ static int deactivate_urbs(snd_usb_substream_t *subs)
 
 	subs->running = 0;
 
+	if (subs->stream->chip->shutdown) /* to be sure... */
+		return 0;
+
 #ifndef SND_USB_ASYNC_UNLINK
 	if (in_interrupt())
 		return 0;
@@ -870,7 +881,7 @@ static int init_substream_urbs(snd_usb_substream_t *subs, snd_pcm_runtime_t *run
 		u->urb->transfer_flags = URB_ISO_ASAP | UNLINK_FLAGS;
 		u->urb->number_of_packets = u->packets;
 		u->urb->context = u;
-		u->urb->complete = snd_complete_urb;
+		u->urb->complete = (usb_complete_t)snd_complete_urb;
 	}
 
 	if (subs->syncpipe) {
@@ -892,7 +903,7 @@ static int init_substream_urbs(snd_usb_substream_t *subs, snd_pcm_runtime_t *run
 			u->urb->transfer_flags = URB_ISO_ASAP | UNLINK_FLAGS;
 			u->urb->number_of_packets = u->packets;
 			u->urb->context = u;
-			u->urb->complete = snd_complete_sync_urb;
+			u->urb->complete = (usb_complete_t)snd_complete_sync_urb;
 		}
 	}
 	return 0;
@@ -2253,9 +2264,6 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr)
 	snd_usb_audio_t *chip;
 	snd_card_t *card;
 	struct list_head *p;
-	snd_usb_stream_t *as;
-	snd_usb_substream_t *subs;
-	int idx;
 
 	if (ptr == (void *)-1)
 		return;
@@ -2265,22 +2273,21 @@ static void snd_usb_audio_disconnect(struct usb_device *dev, void *ptr)
 	down(&register_mutex);
 	chip->shutdown = 1;
 	chip->num_interfaces--;
-	if (chip->num_interfaces <= 0)
+	if (chip->num_interfaces <= 0) {
 		snd_card_disconnect(card);
+		/* release the pcm resources */
 		list_for_each(p, &chip->pcm_list) {
+			snd_usb_stream_t *as;
+			int idx;
 			as = list_entry(p, snd_usb_stream_t, list);
 			for (idx = 0; idx < 2; idx++) {
+				snd_usb_substream_t *subs;
 				subs = &as->substream[idx];
 				if (!subs->num_formats)
 					continue;
 				release_substream_urbs(subs);
-			if (subs->interface >= 0) {
-				usb_set_interface(subs->dev, subs->interface, 0);
-				subs->interface = -1;
 			}
 		}
-	}
-	if (chip->num_interfaces <= 0) {
 		up(&register_mutex);
 		snd_card_free_in_thread(card);
 	} else {

sound/usb/usbmidi.c

@@ -172,7 +172,11 @@ static void snd_usbmidi_input_packet(snd_usb_midi_in_endpoint_t* ep,
 /*
  * Processes the data read from the device.
  */
+#ifndef OLD_USB
 static void snd_usbmidi_in_urb_complete(struct urb* urb, struct pt_regs *regs)
+#else
+static void snd_usbmidi_in_urb_complete(struct urb* urb)
+#endif
 {
 	snd_usb_midi_in_endpoint_t* ep = snd_magic_cast(snd_usb_midi_in_endpoint_t, urb->context, return);
 
@@ -197,7 +201,11 @@ static void snd_usbmidi_in_urb_complete(struct urb* urb, struct pt_regs *regs)
 /*
  * Converts the data read from a Midiman device to standard USB MIDI packets.
  */
+#ifndef OLD_USB
 static void snd_usbmidi_in_midiman_complete(struct urb* urb, struct pt_regs *regs)
+#else
+static void snd_usbmidi_in_midiman_complete(struct urb* urb)
+#endif
 {
 	if (urb->status == 0) {
 		uint8_t* buffer = (uint8_t*)urb->transfer_buffer;
@@ -223,10 +231,18 @@ static void snd_usbmidi_in_midiman_complete(struct urb* urb, struct pt_regs *reg
 			}
 		}
 	}
+#ifndef OLD_USB
 	snd_usbmidi_in_urb_complete(urb, regs);
+#else
+	snd_usbmidi_in_urb_complete(urb);
+#endif
 }
 
+#ifndef OLD_USB
 static void snd_usbmidi_out_urb_complete(struct urb* urb, struct pt_regs *regs)
+#else
+static void snd_usbmidi_out_urb_complete(struct urb* urb)
+#endif
 {
 	snd_usb_midi_out_endpoint_t* ep = snd_magic_cast(snd_usb_midi_out_endpoint_t, urb->context, return);
 
@@ -451,8 +467,10 @@ static void snd_usbmidi_output_trigger(snd_rawmidi_substream_t* substream, int u
 	usbmidi_out_port_t* port = (usbmidi_out_port_t*)substream->runtime->private_data;
 
 	port->active = up;
-	if (up)
+	if (up) {
+		if (! port->ep->umidi->chip->shutdown) /* to be sure... */
 			tasklet_hi_schedule(&port->ep->tasklet);
+	}
 }
 
 static int snd_usbmidi_input_open(snd_rawmidi_substream_t* substream)
@@ -489,6 +507,7 @@ static void snd_usbmidi_in_endpoint_delete(snd_usb_midi_in_endpoint_t* ep)
 {
 	if (ep->urb) {
 		if (ep->urb->transfer_buffer) {
+			if (! ep->umidi->chip->shutdown) /* to be sure */
 				usb_unlink_urb(ep->urb);
 			kfree(ep->urb->transfer_buffer);
 		}
@@ -587,10 +606,12 @@ static int snd_usbmidi_in_endpoint_create(snd_usb_midi_t* umidi,
 	}
 	if (int_epd)
 		usb_fill_int_urb(ep->urb, umidi->chip->dev, pipe, buffer, length,
-			     snd_usbmidi_in_urb_complete, ep, int_epd->bInterval);
+				 (usb_complete_t)snd_usbmidi_in_urb_complete,
+				 ep, int_epd->bInterval);
 	else
 		usb_fill_bulk_urb(ep->urb, umidi->chip->dev, pipe, buffer, length,
-			      snd_usbmidi_in_urb_complete, ep);
+				  (usb_complete_t)snd_usbmidi_in_urb_complete,
+				  ep);
 
 	rep->in = ep;
 	return 0;
@@ -615,6 +636,7 @@ static void snd_usbmidi_out_endpoint_delete(snd_usb_midi_out_endpoint_t* ep)
 		tasklet_kill(&ep->tasklet);
 	if (ep->urb) {
 		if (ep->urb->transfer_buffer) {
+			if (! ep->umidi->chip->shutdown) /* to be sure */
 				usb_unlink_urb(ep->urb);
 			kfree(ep->urb->transfer_buffer);
 		}
@@ -654,7 +676,8 @@ static int snd_usbmidi_out_endpoint_create(snd_usb_midi_t* umidi,
 		return -ENOMEM;
 	}
 	usb_fill_bulk_urb(ep->urb, umidi->chip->dev, pipe, buffer,
-		      ep->max_transfer, snd_usbmidi_out_urb_complete, ep);
+			  ep->max_transfer,
+			  (usb_complete_t)snd_usbmidi_out_urb_complete, ep);
 
 	spin_lock_init(&ep->buffer_lock);
 	tasklet_init(&ep->tasklet, snd_usbmidi_out_tasklet, (unsigned long)ep);
@@ -944,7 +967,7 @@ static int snd_usbmidi_create_endpoints_midiman(snd_usb_midi_t* umidi,
 	err = snd_usbmidi_in_endpoint_create(umidi, &ep_info, &umidi->endpoints[0]);
 	if (err < 0)
 		return err;
-	umidi->endpoints[0].in->urb->complete = snd_usbmidi_in_midiman_complete;
+	umidi->endpoints[0].in->urb->complete = (usb_complete_t)snd_usbmidi_in_midiman_complete;
 
 	if (endpoint->out_cables > 0x0001) {
 		ep_info.epnum = get_endpoint(hostif, 4)->bEndpointAddress & USB_ENDPOINT_NUMBER_MASK;

sound/usb/usbmixer.c

@@ -296,7 +296,7 @@ static int get_ctl_value(usb_mixer_elem_info_t *cval, int request, int validx, i
 			return 0;
 		}
 	}
-	snd_printdd(KERN_ERR "cannot get ctl value: req = 0x%x, idx = 0x%x, val = 0x%x, type = %d\n", request, validx, cval->ctrlif | (cval->id << 8), cval->val_type);
+	snd_printdd(KERN_ERR "cannot get ctl value: req = 0x%x, wValue = 0x%x, wIndex = 0x%x, type = %d\n", request, validx, cval->ctrlif | (cval->id << 8), cval->val_type);
 	return -EINVAL;
 }
 
@@ -331,6 +331,7 @@ static int set_ctl_value(usb_mixer_elem_info_t *cval, int request, int validx, i
 				    validx, cval->ctrlif | (cval->id << 8),
 				    buf, val_len, HZ / 10) >= 0)
 			return 0;
+	snd_printdd(KERN_ERR "cannot set ctl value: req = 0x%x, wValue = 0x%x, wIndex = 0x%x, type = %d, data = 0x%x/0x%x\n", request, validx, cval->ctrlif | (cval->id << 8), cval->val_type, buf[0], buf[1]);
 	return -EINVAL;
 }
 
@@ -571,11 +572,11 @@ static void usb_mixer_elem_free(snd_kcontrol_t *kctl)
 /*
  * retrieve the minimum and maximum values for the specified control
  */
-static int get_min_max(usb_mixer_elem_info_t *cval)
+static int get_min_max(usb_mixer_elem_info_t *cval, int default_min)
 {
 	/* for failsafe */
-	cval->min = 0;
-	cval->max = 1;
+	cval->min = default_min;
+	cval->max = cval->min + 1;
 	cval->res = 1;
 
 	if (cval->val_type == USB_MIXER_BOOLEAN ||
@@ -634,7 +635,7 @@ static int mixer_ctl_feature_info(snd_kcontrol_t *kcontrol, snd_ctl_elem_info_t
 		uinfo->value.integer.max = 1;
 	} else {
 		if (! cval->initialized)
-			get_min_max(cval);
+			get_min_max(cval,  0);
 		uinfo->value.integer.min = 0;
 		uinfo->value.integer.max = (cval->max - cval->min) / cval->res;
 	}
@@ -784,7 +785,7 @@ static void build_feature_ctl(mixer_build_t *state, unsigned char *desc,
 	}
 
 	/* get min/max values */
-	get_min_max(cval);
+	get_min_max(cval, 0);
 
 	kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval);
 	if (! kctl) {
@@ -952,7 +953,7 @@ static void build_mixer_unit_ctl(mixer_build_t *state, unsigned char *desc,
 	}
 
 	/* get min/max values */
-	get_min_max(cval);
+	get_min_max(cval, 0);
 
 	kctl = snd_ctl_new1(&usb_feature_unit_ctl, cval);
 	if (! kctl) {
@@ -1060,6 +1061,7 @@ struct procunit_value_info {
 	int control;
 	char *suffix;
 	int val_type;
+	int min_value;
 };
 
 struct procunit_info {
@@ -1070,12 +1072,12 @@ struct procunit_info {
 
 static struct procunit_value_info updown_proc_info[] = {
 	{ USB_PROC_UPDOWN_SWITCH, "Switch", USB_MIXER_BOOLEAN },
-	{ USB_PROC_UPDOWN_MODE_SEL, "Mode Select", USB_MIXER_U8 },
+	{ USB_PROC_UPDOWN_MODE_SEL, "Mode Select", USB_MIXER_U8, 1 },
 	{ 0 }
 };
 static struct procunit_value_info prologic_proc_info[] = {
 	{ USB_PROC_PROLOGIC_SWITCH, "Switch", USB_MIXER_BOOLEAN },
-	{ USB_PROC_PROLOGIC_MODE_SEL, "Mode Select", USB_MIXER_U8 },
+	{ USB_PROC_PROLOGIC_MODE_SEL, "Mode Select", USB_MIXER_U8, 1 },
 	{ 0 }
 };
 static struct procunit_value_info threed_enh_proc_info[] = {
@@ -1173,7 +1175,7 @@ static int build_audio_procunit(mixer_build_t *state, int unitid, unsigned char
 		cval->channels = 1;
 
 		/* get min/max values */
-		get_min_max(cval);
+		get_min_max(cval, valinfo->min_value);
 
 		kctl = snd_ctl_new1(&mixer_procunit_ctl, cval);
 		if (! kctl) {