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authorLinus Torvalds <torvalds@linux-foundation.org>2010-10-25 08:32:05 -0700
committerLinus Torvalds <torvalds@linux-foundation.org>2010-10-25 08:32:05 -0700
commit33081adf8b89d5a716d7e1c60171768d39795b39 (patch)
tree275de58bbbb5f7ddffcdc087844cfc7fbe4315be /Documentation
parentc55960499f810357a29659b32d6ea594abee9237 (diff)
parent506ecbca71d07fa327dd986be1682e90885678ee (diff)
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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 ...
Diffstat (limited to 'Documentation')
-rw-r--r--Documentation/sound/alsa/ALSA-Configuration.txt82
-rw-r--r--Documentation/sound/alsa/HD-Audio.txt8
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