Merge branch 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6

* 'for-linus' of git://git.kernel.org/pub/scm/linux/kernel/git/tiwai/sound-2.6: (365 commits)
  ALSA: hda - Disable sticky PCM stream assignment for AD codecs
  ALSA: usb - Creative USB X-Fi volume knob support
  ALSA: ca0106: Use card specific dac id for mute controls.
  ALSA: ca0106: Allow different sound cards to use different SPI channel mappings.
  ALSA: ca0106: Create a nice spot for mapping channels to dacs.
  ALSA: ca0106: Move enabling of front dac out of hardcoded setup sequence.
  ALSA: ca0106: Pull out dac powering routine into separate function.
  ALSA: ca0106 - add Sound Blaster 5.1vx info.
  ASoC: tlv320dac33: Use usleep_range for delays
  ALSA: usb-audio: add Novation Launchpad support
  ALSA: hda - Add workarounds for CT-IBG controllers
  ALSA: hda - Fix wrong TLV mute bit for STAC/IDT codecs
  ASoC: tpa6130a2: Error handling for broken chip
  ASoC: max98088: Staticise m98088_eq_band
  ASoC: soc-core: Fix codec->name memory leak
  ALSA: hda - Apply ideapad quirk to Acer laptops with Cxt5066
  ALSA: hda - Add some workarounds for Creative IBG
  ALSA: hda - Fix wrong SPDIF NID assignment for CA0110
  ALSA: hda - Fix codec rename rules for ALC662-compatible codecs
  ALSA: hda - Add alc_init_jacks() call to other codecs
  ...

2 files changed, 73 insertions, 17 deletions

diff --git a/Documentation/sound/alsa/ALSA-Configuration.txt b/Documentation/sound/alsa/ALSA-Configuration.txt
index 7f4dcebda9c6..d0eb696d32e8 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
   --------------
 
@@ -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.txt b/Documentation/sound/alsa/HD-Audio.txt
index 278cc2122ea0..c82beb007634 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