diff options
Diffstat (limited to 'Documentation/sound/alsa')
-rw-r--r-- | Documentation/sound/alsa/ALSA-Configuration.txt | 84 | ||||
-rw-r--r-- | Documentation/sound/alsa/HD-Audio-Models.txt | 20 | ||||
-rw-r--r-- | Documentation/sound/alsa/HD-Audio.txt | 10 | ||||
-rw-r--r-- | Documentation/sound/alsa/Procfile.txt | 8 | ||||
-rw-r--r-- | Documentation/sound/alsa/alsa-parameters.txt | 135 | ||||
-rw-r--r-- | Documentation/sound/alsa/soc/DAI.txt | 2 | ||||
-rw-r--r-- | Documentation/sound/alsa/soc/codec.txt | 2 | ||||
-rw-r--r-- | Documentation/sound/alsa/soc/platform.txt | 2 |
8 files changed, 239 insertions, 24 deletions
diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt index 2075bbb8b3e..d0eb696d32e 100644 --- a/Documentation/sound/alsa/ALSA-Configuration.txt +++ b/Documentation/sound/alsa/ALSA-Configuration.txt @@ -300,6 +300,74 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. control correctly. If you have problems regarding this, try another ALSA compliant mixer (alsamixer works). + Module snd-azt1605 + ------------------ + + Module for Aztech Sound Galaxy soundcards based on the Aztech AZT1605 + chipset. + + port - port # for BASE (0x220,0x240,0x260,0x280) + wss_port - port # for WSS (0x530,0x604,0xe80,0xf40) + irq - IRQ # for WSS (7,9,10,11) + dma1 - DMA # for WSS playback (0,1,3) + dma2 - DMA # for WSS capture (0,1), -1 = disabled (default) + mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) + mpu_irq - IRQ # for MPU-401 UART (3,5,7,9), -1 = disabled (default) + fm_port - port # for OPL3 (0x388), -1 = disabled (default) + + This module supports multiple cards. It does not support autoprobe: port, + wss_port, irq and dma1 have to be specified. The other values are + optional. + + "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) + or the value stored in the card's EEPROM for cards that have an EEPROM and + their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can + be choosen freely from the options enumerated above. + + If dma2 is specified and different from dma1, the card will operate in + full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to + enable capture since only channels 0 and 1 are available for capture. + + Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 + mpu_port=0x330 mpu_irq=9 fm_port=0x388". + + Whatever IRQ and DMA channels you pick, be sure to reserve them for + legacy ISA in your BIOS. + + Module snd-azt2316 + ------------------ + + Module for Aztech Sound Galaxy soundcards based on the Aztech AZT2316 + chipset. + + port - port # for BASE (0x220,0x240,0x260,0x280) + wss_port - port # for WSS (0x530,0x604,0xe80,0xf40) + irq - IRQ # for WSS (7,9,10,11) + dma1 - DMA # for WSS playback (0,1,3) + dma2 - DMA # for WSS capture (0,1), -1 = disabled (default) + mpu_port - port # for MPU-401 UART (0x300,0x330), -1 = disabled (default) + mpu_irq - IRQ # for MPU-401 UART (5,7,9,10), -1 = disabled (default) + fm_port - port # for OPL3 (0x388), -1 = disabled (default) + + This module supports multiple cards. It does not support autoprobe: port, + wss_port, irq and dma1 have to be specified. The other values are + optional. + + "port" needs to match the BASE ADDRESS jumper on the card (0x220 or 0x240) + or the value stored in the card's EEPROM for cards that have an EEPROM and + their "CONFIG MODE" jumper set to "EEPROM SETTING". The other values can + be choosen freely from the options enumerated above. + + If dma2 is specified and different from dma1, the card will operate in + full-duplex mode. When dma1=3, only dma2=0 is valid and the only way to + enable capture since only channels 0 and 1 are available for capture. + + Generic settings are "port=0x220 wss_port=0x530 irq=10 dma1=1 dma2=0 + mpu_port=0x330 mpu_irq=9 fm_port=0x388". + + Whatever IRQ and DMA channels you pick, be sure to reserve them for + legacy ISA in your BIOS. + Module snd-aw2 -------------- @@ -1285,7 +1353,7 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. about this driver. Note that it has been discontinued, but the Voyetra Turtle Beach knowledge base entry for it is still available at - http://www.turtlebeach.com/site/kb_ftp/790.asp + http://www.turtlebeach.com Module snd-msnd-pinnacle ------------------------ @@ -1641,20 +1709,6 @@ Prior to version 0.9.0rc4 options had a 'snd_' prefix. This was removed. This card is also known as Audio Excel DSP 16 or Zoltrix AV302. - 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 # - - This module supports multiple cards. - - The power-management is supported. - Module snd-sscape ----------------- diff --git a/Documentation/sound/alsa/HD-Audio-Models.txt b/Documentation/sound/alsa/HD-Audio-Models.txt index 1d38b0dfba9..37c6aad5e59 100644 --- a/Documentation/sound/alsa/HD-Audio-Models.txt +++ b/Documentation/sound/alsa/HD-Audio-Models.txt @@ -83,8 +83,8 @@ ALC269 ====== basic Basic preset quanta Quanta FL1 - eeepc-p703 ASUS Eeepc P703 P900A - eeepc-p901 ASUS Eeepc P901 S101 + laptop-amic Laptops with analog-mic input + laptop-dmic Laptops with digital-mic input fujitsu FSC Amilo lifebook Fujitsu Lifebook S6420 auto auto-config reading BIOS (default) @@ -109,11 +109,18 @@ ALC662/663/272 asus-mode4 ASUS asus-mode5 ASUS asus-mode6 ASUS + asus-mode7 ASUS + asus-mode8 ASUS dell Dell with ALC272 dell-zm1 Dell ZM1 with ALC272 samsung-nc10 Samsung NC10 mini notebook auto auto-config reading BIOS (default) +ALC680 +====== + base Base model (ASUS NX90) + auto auto-config reading BIOS (default) + ALC882/883/885/888/889 ====================== 3stack-dig 3-jack with SPDIF I/O @@ -282,15 +289,19 @@ Conexant 5051 hp HP Spartan laptop hp-dv6736 HP dv6736 hp-f700 HP Compaq Presario F700 + ideapad Lenovo IdeaPad laptop lenovo-x200 Lenovo X200 laptop toshiba Toshiba Satellite M300 Conexant 5066 ============= laptop Basic Laptop config (default) + hp-laptop HP laptops, e g G60 dell-laptop Dell laptops + dell-vostro Dell Vostro olpc-xo-1_5 OLPC XO 1.5 ideapad Lenovo IdeaPad U150 + thinkpad Lenovo Thinkpad STAC9200 ======== @@ -398,6 +409,7 @@ STAC92HD83* mic-ref Reference board with power management for ports dell-s14 Dell laptop hp HP laptops with (inverted) mute-LED + hp-dv7-4000 HP dv-7 4000 auto BIOS setup (default) STAC9872 @@ -410,3 +422,7 @@ Cirrus Logic CS4206/4207 mbp55 MacBook Pro 5,5 imac27 IMac 27 Inch auto BIOS setup (default) + +VIA VT17xx/VT18xx/VT20xx +======================== + auto BIOS setup (default) diff --git a/Documentation/sound/alsa/HD-Audio.txt b/Documentation/sound/alsa/HD-Audio.txt index bdafdbd3256..c82beb00763 100644 --- a/Documentation/sound/alsa/HD-Audio.txt +++ b/Documentation/sound/alsa/HD-Audio.txt @@ -57,9 +57,11 @@ dead. However, this detection isn't perfect on some devices. In such a case, you can change the default method via `position_fix` option. `position_fix=1` means to use LPIB method explicitly. -`position_fix=2` means to use the position-buffer. 0 is the default -value, the automatic check and fallback to LPIB as described in the -above. If you get a problem of repeated sounds, this option might +`position_fix=2` means to use the position-buffer. +`position_fix=3` means to use a combination of both methods, needed +for some VIA and ATI controllers. 0 is the default value for all other +controllers, the automatic check and fallback to LPIB as described in +the above. If you get a problem of repeated sounds, this option might help. In addition to that, every controller is known to be broken regarding @@ -562,7 +564,7 @@ compare the codec registers directly. Send a bug report either the followings: kernel-bugzilla:: - http://bugme.linux-foundation.org/ + https://bugzilla.kernel.org/ alsa-devel ML:: alsa-devel@alsa-project.org diff --git a/Documentation/sound/alsa/Procfile.txt b/Documentation/sound/alsa/Procfile.txt index 07301de12cc..7fcd1ad96fc 100644 --- a/Documentation/sound/alsa/Procfile.txt +++ b/Documentation/sound/alsa/Procfile.txt @@ -103,6 +103,8 @@ card*/pcm*/xrun_debug bit 2 = Enable additional jiffies check bit 3 = Log hwptr update at each period interrupt bit 4 = Log hwptr update at each snd_pcm_update_hw_ptr() + bit 5 = Show last 10 positions on error + bit 6 = Do above only once When the bit 0 is set, the driver will show the messages to kernel log when an xrun is detected. The debug message is @@ -122,6 +124,12 @@ card*/pcm*/xrun_debug Bits 3 and 4 are for logging the hwptr records. Note that these will give flood of kernel messages. + When bit 5 is set, the driver logs the last 10 xrun errors and + the proc file shows each jiffies, position, period_size, + buffer_size, old_hw_ptr, and hw_ptr_base values. + + When bit 6 is set, the full xrun log is shown only once. + card*/pcm*/sub*/info The general information of this PCM sub-stream. diff --git a/Documentation/sound/alsa/alsa-parameters.txt b/Documentation/sound/alsa/alsa-parameters.txt new file mode 100644 index 00000000000..0fa40679b08 --- /dev/null +++ b/Documentation/sound/alsa/alsa-parameters.txt @@ -0,0 +1,135 @@ + ALSA Kernel Parameters + ~~~~~~~~~~~~~~~~~~~~~~ + +See Documentation/kernel-parameters.txt for general information on +specifying module parameters. + +This document may not be entirely up to date and comprehensive. The command +"modinfo -p ${modulename}" shows a current list of all parameters of a loadable +module. Loadable modules, after being loaded into the running kernel, also +reveal their parameters in /sys/module/${modulename}/parameters/. Some of these +parameters may be changed at runtime by the command +"echo -n ${value} > /sys/module/${modulename}/parameters/${parm}". + + + snd-ad1816a= [HW,ALSA] + + snd-ad1848= [HW,ALSA] + + snd-ali5451= [HW,ALSA] + + snd-als100= [HW,ALSA] + + snd-als4000= [HW,ALSA] + + snd-azt2320= [HW,ALSA] + + snd-cmi8330= [HW,ALSA] + + snd-cmipci= [HW,ALSA] + + snd-cs4231= [HW,ALSA] + + snd-cs4232= [HW,ALSA] + + snd-cs4236= [HW,ALSA] + + snd-cs4281= [HW,ALSA] + + snd-cs46xx= [HW,ALSA] + + snd-dt019x= [HW,ALSA] + + snd-dummy= [HW,ALSA] + + snd-emu10k1= [HW,ALSA] + + snd-ens1370= [HW,ALSA] + + snd-ens1371= [HW,ALSA] + + snd-es968= [HW,ALSA] + + snd-es1688= [HW,ALSA] + + snd-es18xx= [HW,ALSA] + + snd-es1938= [HW,ALSA] + + snd-es1968= [HW,ALSA] + + snd-fm801= [HW,ALSA] + + snd-gusclassic= [HW,ALSA] + + snd-gusextreme= [HW,ALSA] + + snd-gusmax= [HW,ALSA] + + snd-hdsp= [HW,ALSA] + + snd-ice1712= [HW,ALSA] + + snd-intel8x0= [HW,ALSA] + + snd-interwave= [HW,ALSA] + + snd-interwave-stb= + [HW,ALSA] + + snd-korg1212= [HW,ALSA] + + snd-maestro3= [HW,ALSA] + + snd-mpu401= [HW,ALSA] + + snd-mtpav= [HW,ALSA] + + snd-nm256= [HW,ALSA] + + snd-opl3sa2= [HW,ALSA] + + snd-opti92x-ad1848= + [HW,ALSA] + + snd-opti92x-cs4231= + [HW,ALSA] + + snd-opti93x= [HW,ALSA] + + snd-pmac= [HW,ALSA] + + snd-rme32= [HW,ALSA] + + snd-rme96= [HW,ALSA] + + snd-rme9652= [HW,ALSA] + + snd-sb8= [HW,ALSA] + + snd-sb16= [HW,ALSA] + + snd-sbawe= [HW,ALSA] + + snd-serial= [HW,ALSA] + + snd-sgalaxy= [HW,ALSA] + + snd-sonicvibes= [HW,ALSA] + + snd-sun-amd7930= + [HW,ALSA] + + snd-sun-cs4231= [HW,ALSA] + + snd-trident= [HW,ALSA] + + snd-usb-audio= [HW,ALSA,USB] + + snd-via82xx= [HW,ALSA] + + snd-virmidi= [HW,ALSA] + + snd-wavefront= [HW,ALSA] + + snd-ymfpci= [HW,ALSA] diff --git a/Documentation/sound/alsa/soc/DAI.txt b/Documentation/sound/alsa/soc/DAI.txt index 0ebd7ea9706..c9679264c55 100644 --- a/Documentation/sound/alsa/soc/DAI.txt +++ b/Documentation/sound/alsa/soc/DAI.txt @@ -13,7 +13,7 @@ frame (FRAME) (usually 48kHz) is always driven by the controller. Each AC97 frame is 21uS long and is divided into 13 time slots. The AC97 specification can be found at :- -http://www.intel.com/design/chipsets/audio/ac97_r23.pdf +http://www.intel.com/p/en_US/business/design I2S diff --git a/Documentation/sound/alsa/soc/codec.txt b/Documentation/sound/alsa/soc/codec.txt index 1e95342ed72..37ba3a72cb7 100644 --- a/Documentation/sound/alsa/soc/codec.txt +++ b/Documentation/sound/alsa/soc/codec.txt @@ -143,7 +143,7 @@ struct snd_soc_ops { }; Please refer to the ALSA driver PCM documentation for details. -http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm +http://www.alsa-project.org/~iwai/writing-an-alsa-driver/ 5 - DAPM description. diff --git a/Documentation/sound/alsa/soc/platform.txt b/Documentation/sound/alsa/soc/platform.txt index b681d17fc38..06d835987c6 100644 --- a/Documentation/sound/alsa/soc/platform.txt +++ b/Documentation/sound/alsa/soc/platform.txt @@ -39,7 +39,7 @@ struct snd_soc_platform { }; Please refer to the ALSA driver documentation for details of audio DMA. -http://www.alsa-project.org/~iwai/writing-an-alsa-driver/c436.htm +http://www.alsa-project.org/~iwai/writing-an-alsa-driver/ An example DMA driver is soc/pxa/pxa2xx-pcm.c |