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-rw-r--r--Documentation/sound/alsa/ALSA-Configuration.txt84
-rw-r--r--Documentation/sound/alsa/HD-Audio-Models.txt20
-rw-r--r--Documentation/sound/alsa/HD-Audio.txt10
-rw-r--r--Documentation/sound/alsa/Procfile.txt8
-rw-r--r--Documentation/sound/alsa/alsa-parameters.txt135
-rw-r--r--Documentation/sound/alsa/soc/DAI.txt2
-rw-r--r--Documentation/sound/alsa/soc/codec.txt2
-rw-r--r--Documentation/sound/alsa/soc/platform.txt2
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