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author | Linus Torvalds <torvalds@linux-foundation.org> | 2010-10-25 08:32:05 -0700 |
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committer | Linus Torvalds <torvalds@linux-foundation.org> | 2010-10-25 08:32:05 -0700 |
commit | 33081adf8b89d5a716d7e1c60171768d39795b39 (patch) | |
tree | 275de58bbbb5f7ddffcdc087844cfc7fbe4315be /Documentation | |
parent | c55960499f810357a29659b32d6ea594abee9237 (diff) | |
parent | 506ecbca71d07fa327dd986be1682e90885678ee (diff) | |
download | linux-33081adf8b89d5a716d7e1c60171768d39795b39.tar.gz linux-33081adf8b89d5a716d7e1c60171768d39795b39.tar.bz2 linux-33081adf8b89d5a716d7e1c60171768d39795b39.zip |
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.txt | 82 | ||||
-rw-r--r-- | Documentation/sound/alsa/HD-Audio.txt | 8 |
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 |